From gcd at i.ph Sun Feb 1 00:30:39 2009 From: gcd at i.ph (Nandy Dagondon) Date: Sun, 1 Feb 2009 16:30:39 +0800 Subject: [Freeswitch-users] TDM400 FXO can dialout only once In-Reply-To: <7d0bfd8c0901261639x7c53f256yabf7347b5d22e7e4@mail.gmail.com> References: <7d0bfd8c0901250021q2969fe26u5ea79c3957989ffd@mail.gmail.com> <191c3a030901251818g284054ben60c0bfa61157824@mail.gmail.com> <7d0bfd8c0901252305v5a33fa02w4bc02deaa0c1b14b@mail.gmail.com> <87f2f3b90901260201r2e4d42a3x30367eee64b0e13b@mail.gmail.com> <7d0bfd8c0901260532g50deb359v3c92f5bb372f11a4@mail.gmail.com> <191c3a030901260833l2e07bfffkca9400d33a5483fe@mail.gmail.com> <7d0bfd8c0901261639x7c53f256yabf7347b5d22e7e4@mail.gmail.com> Message-ID: <7d0bfd8c0902010030n6b5d3c35k7539fac38e07300d@mail.gmail.com> hi everybody, i created [ph] tone definition per raul's suggestion and changed /etc/zaptel.conf entries to: tonezone=ph defaultzone=ph but it didn't solve the problem. i captured the console log during start-up and shutdown. i noticed openzap related errors during shutdown. here's the snippet of the log: STARTUP --------------- 2009-02-01 15:58:10 [NOTICE] zap_io.c:2517 zap_global_init() Modules configured: 1 2009-02-01 15:58:10 [INFO] zap_io.c:2341 zap_load_module() Loading IO from /opt/freeswitch/mod/ozmod_zt.so 2009-02-01 15:58:10 [INFO] zap_io.c:2127 load_config() auto-loaded 'zt' 2009-02-01 15:58:10 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 1 as OpenZAP device 1:1 fd:39 2009-02-01 15:58:10 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 2 as OpenZAP device 2:1 fd:40 2009-02-01 15:58:10 [INFO] zap_io.c:2265 load_config() Configured 2 channel(s) 2009-02-01 15:58:10 [INFO] zap_io.c:2358 zap_load_module() Loading SIG from /opt/freeswitch/mod/ozmod_analog.so 2009-02-01 15:58:10 [INFO] zap_io.c:2474 zap_configure_span() auto-loaded 'analog' 2009-02-01 15:58:10 [CONSOLE] switch_loadable_module.c:857 switch_loadable_module_load_file() Successfully Loaded [mod_openzap] --- DIDN'T MAKE ANY CALL --- SHUTDOWN ------------------ 2009-02-01 15:59:07 [NOTICE] switch_loadable_module.c:536 switch_loadable_module_unprocess() Deleting API Function 'oz' 2009-02-01 15:59:07 [CONSOLE] switch_loadable_module.c:1231 do_shutdown() Stopping: mod_openzap 2009-02-01 15:59:07 [INFO] zap_io.c:256 zap_channel_destroy() Closing channel zt:1:1 fd:39 2009-02-01 15:59:07 [INFO] zap_io.c:256 zap_channel_destroy() Closing channel zt:2:1 fd:40 2009-02-01 15:59:08 [ERR] ozmod_analog.c:899 zap_analog_run() Failure Polling event! [no matching descriptor] 2009-02-01 15:59:08 [ERR] ozmod_analog.c:899 zap_analog_run() Failure Polling event! [no matching descriptor] 2009-02-01 15:59:08 [INFO] zap_io.c:2456 zap_unload_modules() Unloading /opt/freeswitch/mod/ozmod_analog.so 2009-02-01 15:59:08 [INFO] zap_io.c:2441 zap_unload_modules() Unloading IO zt 2009-02-01 15:59:08 [INFO] zap_io.c:2456 zap_unload_modules() Unloading /opt/freeswitch/mod/ozmod_zt.so i also notice the same ERR flag during shutdown after making test calls. any suggestion what to do next? tks for your assistance. rgds, -nandy On Tue, Jan 27, 2009 at 8:39 AM, Nandy Dagondon wrote: > i tested the SVN trunk version. still the same behaviour. > -nandy > > > On Tue, Jan 27, 2009 at 12:33 AM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> The unhanded type message just means that mod_openzap does not do anything >> with the TONE_DETECTED event that was passed >> up from the ozmod_analog. >> >> On Mon, Jan 26, 2009 at 7:32 AM, Nandy Dagondon wrote: >> >>> that's great. yes, i'm in the philippines. there's a difference in >>> dialtone - it's 425 Hz. >>> -nandy >>> >>> >>> >>> On Mon, Jan 26, 2009 at 6:01 PM, Michael Collins wrote: >>> >>>> I have a TDM400 clone and I will see if I can reproduce these >>>> symptoms. BTW, are you in the Philippines? Is there any difference in >>>> the dial tone there than in the US? >>>> -MC >>>> >>>> On Sun, Jan 25, 2009 at 11:05 PM, Nandy Dagondon wrote: >>>> > i monitored the line using another phone. there's indeed dialtone in >>>> all >>>> > attempts. >>>> > i see TONE_DETECTED in the first call but i wonder there's a WARNING >>>> message >>>> > immediately following [WARNING] mod_openzap.c:1196 on_fxo_signal() >>>> Unhandled >>>> > type for channel 2:1. >>>> > the dialtone freq should be okay since it's detected in the first >>>> call.could >>>> > the WARNING message gives us a hint of a possible problem other than >>>> the >>>> > dialtone freq? >>>> > >>>> > okay, i'll try the SVN version next. >>>> > >>>> > >>>> > On Mon, Jan 26, 2009 at 10:18 AM, Anthony Minessale >>>> > wrote: >>>> >> >>>> >> Its not detecting a dial tone on the failure case. >>>> >> Before dialing it waits until it picks up dialtone. >>>> >> Try the svn trunk version to see if it works any better or verify >>>> there is >>>> >> a dialtone on the line. >>>> >> >>>> >> On Jan 25, 2009 6:19 PM, "Nandy Dagondon" wrote: >>>> >> >>>> >> hi everybody, >>>> >> >>>> >> i installed FS 1.0.2 to an Atom mobo (Intel D945GCLF2). it's working >>>> using >>>> >> IP phones, softphones and digium FXS port. but there's a problem in >>>> dialing >>>> >> out to PSTN using digium tdm400 fxo - it works fine on the first >>>> attempt >>>> >> (after starting FS) but it fails on the subsequent attempts. i tested >>>> to >>>> >> call using the FXS port and IP phone. same problem. >>>> >> >>>> >> before i place any call, i checked >oz dump 2 1 (show current state >>>> = >>>> >> DOWN, last state = DOWN) >>>> >> >>>> >> in the first call, there's this message: >>>> >> [WARNING] mod_openzap.c:1196 on_fxo_signal() Unhandled type for >>>> channel >>>> >> 2:1 >>>> >> but >>>> >> >>>> >> then i hangup. checked >oz dump 21 (show current state=DOWN, last >>>> >> state=HANGUP) >>>> >> >>>> >> in the 2nd (and subsequent) attempts, the fxo just goes off-hook but >>>> >> doesn't send the dtmf tones. >>>> >> >oz dump 2 1 (shows current state = DIALING, last state = DOWN) >>>> >> >>>> >> has anyone encountered this problem before? i appreciate for any help >>>> to >>>> >> correct this problem. >>>> >> >>>> >> tks, >>>> >> nandy >>>> >> >>>> >> >>>> >> Environment: >>>> >> ================== >>>> >> kernel 2.6.18-92.1.22.el5 >>>> >> FS 1.0.2 >>>> >> zaptel 1.4.11 >>>> >> oslec >>>> >> digium TDM400P Rev. I (1-FXS, 2-FXO,3-4 vacant) >>>> >> >>>> >> zaptel.conf >>>> >> ======== >>>> >> loadzone = us >>>> >> defaultzone=us >>>> >> channels=1-2 >>>> >> alaw=1-4 >>>> >> fxsks=2 >>>> >> fxoks=1 >>>> >> >>>> >> >>>> >> openzap.conf.xml: >>>> >> =============== >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> openzap.conf >>>> >> ========== >>>> >> [span zt] >>>> >> name => OpenZAP FXS >>>> >> number => 1 >>>> >> fxs-channel => 1 >>>> >> >>>> >> [span zt] >>>> >> name => OpenZAP FXO >>>> >> number => 2 >>>> >> fxo-channel => 2 >>>> >> >>>> >> tones.conf (the dialtone and ring tone is set to Philipping tones) >>>> >> ======== >>>> >> [us] >>>> >> generate-dial => v=-7;%(1000,0,425) >>>> >> detect-dial => 425 >>>> >> >>>> >> generate-ring => v=-7;%(1000,4000,425,480) >>>> >> detect-ring => 425,480 >>>> >> >>>> >> generate-busy => v=-7;%(500,500,480,620) >>>> >> detect-busy => 480,620 >>>> >> >>>> >> generate-attn => v=0;%(200,300,1400,1800) >>>> >> detect-attn => 1400,1800 >>>> >> >>>> >> generate-callwaiting-sas => v=0;%(300,10000,440) >>>> >> detect-callwaiting-sas => 440 >>>> >> >>>> >> generate-callwaiting-cas => v=0;%(80,0,2750,2130) >>>> >> detect-callwaiting-cas => 2750,2130 >>>> >> >>>> >> detect-fail1 => 913.8 >>>> >> detect-fail2 => 1370.6 >>>> >> detect-fail3 => 776.7 >>>> >> >>>> >> LOG OF FIRST CALL (OK) >>>> >> ==================== >>>> >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:152 >>>> >> switch_core_standard_on_execute() OpenZAP/1:1/93400534 Execute >>>> >> bridge(openzap/2/1/3400534) >>>> >> 2009-01-25 10:35:58 [DEBUG] mod_openzap.c:340 tech_init() Set codec >>>> PCMU >>>> >> 20ms >>>> >> 2009-01-25 10:35:58 [DEBUG] mod_openzap.c:1017 >>>> channel_outgoing_channel() >>>> >> Connect outbound channel OpenZAP/2:1/3400534 >>>> >> 2009-01-25 10:35:58 [NOTICE] switch_channel.c:565 >>>> >> switch_channel_set_name() New Channel OpenZAP/2:1/3400534 >>>> >> [e5f12114-ea88-11dd-9f5c-290fb4a527a4] >>>> >> 2009-01-25 10:35:58 [DEBUG] mod_openzap.c:1026 >>>> channel_outgoing_channel() >>>> >> (OpenZAP/2:1/3400534) State Change CS_NEW -> CS_INIT >>>> >> 2009-01-25 10:35:58 [DEBUG] switch_core_session.c:807 >>>> >> switch_core_session_signal_state_change() Send signal >>>> OpenZAP/2:1/3400534 >>>> >> [BREAK] >>>> >> 2009-01-25 10:35:58 [DEBUG] ozmod_analog.c:52 >>>> analog_fxo_outgoing_call() >>>> >> Changing state on 2:1 from DOWN to DIALING >>>> >> 2009-01-25 10:35:58 [DEBUG] ozmod_analog.c:239 >>>> zap_analog_channel_run() >>>> >> ANALOG CHANNEL thread starting. >>>> >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:379 >>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change >>>> CS_INIT >>>> >> 2009-01-25 10:35:58 [DEBUG] ozmod_analog.c:410 >>>> zap_analog_channel_run() >>>> >> Executing state handler on 2:1 for DIALING >>>> >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:444 >>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State INIT >>>> >> 2009-01-25 10:35:58 [DEBUG] mod_openzap.c:364 channel_on_init() >>>> >> (OpenZAP/2:1/3400534) State Change CS_INIT -> CS_ROUTING >>>> >> 2009-01-25 10:35:58 [DEBUG] switch_core_session.c:807 >>>> >> switch_core_session_signal_state_change() Send signal >>>> OpenZAP/2:1/3400534 >>>> >> [BREAK] >>>> >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:444 >>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State INIT going to >>>> sleep >>>> >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:379 >>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change >>>> >> CS_ROUTING >>>> >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:447 >>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State ROUTING >>>> >> 2009-01-25 10:35:58 [DEBUG] mod_openzap.c:387 channel_on_routing() >>>> >> OpenZAP/2:1/3400534 CHANNEL ROUTING >>>> >> 2009-01-25 10:35:58 [DEBUG] switch_ivr_originate.c:58 >>>> >> originate_on_routing() (OpenZAP/2:1/3400534) State Change CS_ROUTING >>>> -> >>>> >> CS_CONSUME_MEDIA >>>> >> 2009-01-25 10:35:58 [DEBUG] switch_core_session.c:807 >>>> >> switch_core_session_signal_state_change() Send signal >>>> OpenZAP/2:1/3400534 >>>> >> [BREAK] >>>> >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:447 >>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State ROUTING going >>>> to sleep >>>> >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:379 >>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change >>>> >> CS_CONSUME_MEDIA >>>> >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:466 >>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State CONSUME_MEDIA >>>> >> 2009-01-25 10:35:59 [DEBUG] ozmod_analog.c:615 >>>> zap_analog_channel_run() >>>> >> Detected tone DIAL on 2:1 >>>> >> 2009-01-25 10:35:59 [DEBUG] mod_openzap.c:1153 on_fxo_signal() got >>>> FXO sig >>>> >> 2:1 [TONE_DETECTED] >>>> >> 2009-01-25 10:35:59 [WARNING] mod_openzap.c:1196 on_fxo_signal() >>>> Unhandled >>>> >> type for channel 2:1 >>>> >> 2009-01-25 10:35:59 [DEBUG] zap_io.c:1179 >>>> zchan_activate_dtmf_buffer() >>>> >> Created DTMF Buffer! >>>> >> 2009-01-25 10:35:59 [DEBUG] zap_io.c:1715 handle_dtmf() 2:1 GENERATE >>>> DTMF >>>> >> [3400534] >>>> >> 2009-01-25 10:36:03 [DEBUG] ozmod_analog.c:302 >>>> zap_analog_channel_run() >>>> >> Changing state on 2:1 from DIALING to UP >>>> >> 2009-01-25 10:36:03 [DEBUG] ozmod_analog.c:410 >>>> zap_analog_channel_run() >>>> >> Executing state handler on 2:1 for UP >>>> >> 2009-01-25 10:36:03 [DEBUG] mod_openzap.c:1153 on_fxo_signal() got >>>> FXO sig >>>> >> 2:1 [UP] >>>> >> 2009-01-25 10:36:03 [DEBUG] switch_channel.c:1710 >>>> >> switch_channel_perform_mark_answered() Send signal >>>> OpenZAP/1:1/93400534 >>>> >> [BREAK] >>>> >> 2009-01-25 10:36:03 [NOTICE] mod_openzap.c:1180 on_fxo_signal() >>>> Channel >>>> >> [OpenZAP/2:1/3400534] has been answered >>>> >> 2009-01-25 10:36:03 [DEBUG] switch_channel.c:177 >>>> >> switch_channel_audio_sync() OpenZAP/2:1/3400534 receive message >>>> [AUDIO_SYNC] >>>> >> 2009-01-25 10:36:03 [DEBUG] switch_channel.c:1768 >>>> >> switch_channel_perform_answer() OpenZAP/1:1/93400534 receive message >>>> >> [ANSWER] >>>> >> 2009-01-25 10:36:03 [DEBUG] mod_openzap.c:813 >>>> >> channel_receive_message_fxs() Changing state on 1:1 from IDLE to UP >>>> >> 2009-01-25 10:36:03 [NOTICE] mod_openzap.c:814 >>>> >> channel_receive_message_fxs() Channel [OpenZAP/1:1/93400534] has been >>>> >> answered >>>> >> 2009-01-25 10:36:03 [DEBUG] switch_channel.c:177 >>>> >> switch_channel_audio_sync() OpenZAP/1:1/93400534 receive message >>>> >> [AUDIO_SYNC] >>>> >> 2009-01-25 10:36:03 [DEBUG] switch_core_session.c:511 >>>> >> switch_core_session_perform_receive_message() Send signal >>>> >> OpenZAP/1:1/93400534 [BREAK] >>>> >> 2009-01-25 10:36:03 [DEBUG] switch_ivr_originate.c:1627 >>>> >> switch_ivr_originate() Originate Resulted in Success: >>>> [OpenZAP/2:1/3400534] >>>> >> 2009-01-25 10:36:03 [DEBUG] switch_channel.c:177 >>>> >> switch_channel_audio_sync() OpenZAP/2:1/3400534 receive message >>>> [AUDIO_SYNC] >>>> >> 2009-01-25 10:36:03 [DEBUG] switch_channel.c:177 >>>> >> switch_channel_audio_sync() OpenZAP/1:1/93400534 receive message >>>> >> [AUDIO_SYNC] >>>> >> 2009-01-25 10:36:03 [DEBUG] switch_ivr_bridge.c:862 >>>> >> switch_ivr_multi_threaded_bridge() OpenZAP/2:1/3400534 receive >>>> message >>>> >> [BRIDGE] >>>> >> 2009-01-25 10:36:03 [DEBUG] switch_core_session.c:511 >>>> >> switch_core_session_perform_receive_message() Send signal >>>> >> OpenZAP/2:1/3400534 [BREAK] >>>> >> 2009-01-25 10:36:03 [DEBUG] switch_ivr_bridge.c:869 >>>> >> switch_ivr_multi_threaded_bridge() OpenZAP/1:1/93400534 receive >>>> message >>>> >> [BRIDGE] >>>> >> 2009-01-25 10:36:03 [DEBUG] switch_core_session.c:511 >>>> >> switch_core_session_perform_receive_message() Send signal >>>> >> OpenZAP/1:1/93400534 [BREAK] >>>> >> 2009-01-25 10:36:03 [DEBUG] switch_ivr_bridge.c:913 >>>> >> switch_ivr_multi_threaded_bridge() (OpenZAP/2:1/3400534) State Change >>>> >> CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA >>>> >> 2009-01-25 10:36:03 [DEBUG] switch_core_session.c:807 >>>> >> switch_core_session_signal_state_change() Send signal >>>> OpenZAP/2:1/3400534 >>>> >> [BREAK] >>>> >> 2009-01-25 10:36:03 [DEBUG] ozmod_analog.c:410 >>>> zap_analog_channel_run() >>>> >> Executing state handler on 1:1 for UP >>>> >> 2009-01-25 10:36:03 [DEBUG] mod_openzap.c:1212 on_fxs_signal() got >>>> FXS sig >>>> >> [UP] >>>> >> 2009-01-25 10:36:03 [DEBUG] switch_core_state_machine.c:466 >>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State CONSUME_MEDIA >>>> going to >>>> >> sleep >>>> >> 2009-01-25 10:36:03 [DEBUG] switch_core_state_machine.c:379 >>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change >>>> >> CS_EXCHANGE_MEDIA >>>> >> 2009-01-25 10:36:03 [DEBUG] switch_core_state_machine.c:457 >>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State EXCHANGE_MEDIA >>>> >> 2009-01-25 10:36:03 [DEBUG] mod_openzap.c:511 >>>> channel_on_exchange_media() >>>> >> CHANNEL EXCHANGE_MEDIA >>>> >> 2009-01-25 10:36:16 [DEBUG] ozmod_analog.c:744 process_event() EVENT >>>> >> [ONHOOK][1:1] STATE [UP] >>>> >> 2009-01-25 10:36:16 [DEBUG] ozmod_analog.c:780 process_event() >>>> Changing >>>> >> state on 1:1 from UP to DOWN >>>> >> 2009-01-25 10:36:16 [DEBUG] ozmod_analog.c:410 >>>> zap_analog_channel_run() >>>> >> Executing state handler on 1:1 for DOWN >>>> >> 2009-01-25 10:36:16 [DEBUG] mod_openzap.c:1212 on_fxs_signal() got >>>> FXS sig >>>> >> [STOP] >>>> >> 2009-01-25 10:36:16 [NOTICE] mod_openzap.c:1300 on_fxs_signal() >>>> Hangup >>>> >> OpenZAP/1:1/93400534 [CS_EXECUTE] [NORMAL_CLEARING] >>>> >> 2009-01-25 10:36:16 [DEBUG] switch_channel.c:1494 >>>> >> switch_channel_perform_hangup() Send signal OpenZAP/1:1/93400534 >>>> [KILL] >>>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_session.c:807 >>>> >> switch_core_session_signal_state_change() Send signal >>>> OpenZAP/1:1/93400534 >>>> >> [BREAK] >>>> >> 2009-01-25 10:36:16 [DEBUG] zap_io.c:1125 zap_channel_done() channel >>>> done >>>> >> 1:1 >>>> >> 2009-01-25 10:36:16 [DEBUG] ozmod_analog.c:726 >>>> zap_analog_channel_run() >>>> >> ANALOG CHANNEL 1:1 thread ended. >>>> >> 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:360 >>>> audio_bridge_thread() >>>> >> OpenZAP/1:1/93400534 ending bridge by request from read function >>>> >> 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:435 >>>> audio_bridge_thread() >>>> >> BRIDGE THREAD DONE [OpenZAP/1:1/93400534] >>>> >> 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:439 >>>> audio_bridge_thread() >>>> >> Send signal OpenZAP/2:1/3400534 [BREAK] >>>> >> 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:354 >>>> audio_bridge_thread() >>>> >> OpenZAP/1:1/93400534 ending bridge by request from write function >>>> >> 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:409 >>>> audio_bridge_thread() >>>> >> OpenZAP/2:1/3400534 receive message [UNBRIDGE] >>>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_session.c:511 >>>> >> switch_core_session_perform_receive_message() Send signal >>>> >> OpenZAP/2:1/3400534 [BREAK] >>>> >> 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:435 >>>> audio_bridge_thread() >>>> >> BRIDGE THREAD DONE [OpenZAP/2:1/3400534] >>>> >> 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:439 >>>> audio_bridge_thread() >>>> >> Send signal OpenZAP/1:1/93400534 [BREAK] >>>> >> 2009-01-25 10:36:16 [NOTICE] switch_ivr_bridge.c:470 >>>> >> audio_bridge_on_exchange_media() Hangup OpenZAP/2:1/3400534 >>>> >> [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] >>>> >> 2009-01-25 10:36:16 [DEBUG] switch_channel.c:1494 >>>> >> switch_channel_perform_hangup() Send signal OpenZAP/2:1/3400534 >>>> [KILL] >>>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_session.c:807 >>>> >> switch_core_session_signal_state_change() Send signal >>>> OpenZAP/2:1/3400534 >>>> >> [BREAK] >>>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:457 >>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State EXCHANGE_MEDIA >>>> going >>>> >> to sleep >>>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:379 >>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change >>>> >> CS_HANGUP >>>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:410 >>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State HANGUP >>>> >> 2009-01-25 10:36:16 [DEBUG] mod_openzap.c:429 channel_on_hangup() >>>> Changing >>>> >> state on 2:1 from UP to HANGUP >>>> >> 2009-01-25 10:36:16 [DEBUG] mod_openzap.c:472 channel_on_hangup() >>>> >> OpenZAP/2:1/3400534 CHANNEL HANGUP >>>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:46 >>>> >> switch_core_standard_on_hangup() OpenZAP/2:1/3400534 Standard HANGUP, >>>> cause: >>>> >> NORMAL_CLEARING >>>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:410 >>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State HANGUP going to >>>> sleep >>>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_session.c:939 >>>> >> switch_core_session_thread() Session 2 (OpenZAP/2:1/3400534) Locked, >>>> Waiting >>>> >> on external entities >>>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:454 >>>> >> switch_core_session_run() (OpenZAP/1:1/93400534) State EXECUTE going >>>> to >>>> >> sleep >>>> >> 2009-01-25 10:36:16 [NOTICE] switch_core_session.c:957 >>>> >> switch_core_session_thread() Session 2 (OpenZAP/2:1/3400534) Ended >>>> >> 2009-01-25 10:36:16 [NOTICE] switch_core_session.c:959 >>>> >> switch_core_session_thread() Close Channel OpenZAP/2:1/3400534 >>>> [CS_HANGUP] >>>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:379 >>>> >> switch_core_session_run() (OpenZAP/1:1/93400534) Running State Change >>>> >> CS_HANGUP >>>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:410 >>>> >> switch_core_session_run() (OpenZAP/1:1/93400534) State HANGUP >>>> >> 2009-01-25 10:36:16 [DEBUG] mod_openzap.c:472 channel_on_hangup() >>>> >> OpenZAP/1:1/93400534 CHANNEL HANGUP >>>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:46 >>>> >> switch_core_standard_on_hangup() OpenZAP/1:1/93400534 Standard >>>> HANGUP, >>>> >> cause: NORMAL_CLEARING >>>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:410 >>>> >> switch_core_session_run() (OpenZAP/1:1/93400534) State HANGUP going >>>> to sleep >>>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_session.c:939 >>>> >> switch_core_session_thread() Session 1 (OpenZAP/1:1/93400534) Locked, >>>> >> Waiting on external entities >>>> >> 2009-01-25 10:36:16 [NOTICE] switch_core_session.c:957 >>>> >> switch_core_session_thread() Session 1 (OpenZAP/1:1/93400534) Ended >>>> >> 2009-01-25 10:36:16 [NOTICE] switch_core_session.c:959 >>>> >> switch_core_session_thread() Close Channel OpenZAP/1:1/93400534 >>>> [CS_HANGUP] >>>> >> 2009-01-25 10:36:16 [DEBUG] ozmod_analog.c:410 >>>> zap_analog_channel_run() >>>> >> Executing state handler on 2:1 for HANGUP >>>> >> 2009-01-25 10:36:17 [DEBUG] ozmod_analog.c:351 >>>> zap_analog_channel_run() >>>> >> Changing state on 2:1 from HANGUP to DOWN >>>> >> 2009-01-25 10:36:17 [DEBUG] ozmod_analog.c:410 >>>> zap_analog_channel_run() >>>> >> Executing state handler on 2:1 for DOWN >>>> >> 2009-01-25 10:36:17 [DEBUG] mod_openzap.c:1153 on_fxo_signal() got >>>> FXO sig >>>> >> 2:1 [STOP] >>>> >> 2009-01-25 10:36:17 [DEBUG] zap_io.c:1125 zap_channel_done() channel >>>> done >>>> >> 2:1 >>>> >> 2009-01-25 10:36:17 [DEBUG] ozmod_analog.c:726 >>>> zap_analog_channel_run() >>>> >> ANALOG CHANNEL 2:1 thread ended. >>>> >> >>>> >> LOG OF FAILED CALLS >>>> >> ================== >>>> >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:152 >>>> >> switch_core_standard_on_execute() OpenZAP/1:1/93400534 Execute >>>> >> bridge(openzap/2/1/3400534) >>>> >> 2009-01-25 10:36:55 [DEBUG] mod_openzap.c:340 tech_init() Set codec >>>> PCMU >>>> >> 20ms >>>> >> 2009-01-25 10:36:55 [DEBUG] mod_openzap.c:1017 >>>> channel_outgoing_channel() >>>> >> Connect outbound channel OpenZAP/2:1/3400534 >>>> >> 2009-01-25 10:36:55 [NOTICE] switch_channel.c:565 >>>> >> switch_channel_set_name() New Channel OpenZAP/2:1/3400534 >>>> >> [079f5420-ea89-11dd-9f5c-290fb4a527a4] >>>> >> 2009-01-25 10:36:55 [DEBUG] mod_openzap.c:1026 >>>> channel_outgoing_channel() >>>> >> (OpenZAP/2:1/3400534) State Change CS_NEW -> CS_INIT >>>> >> 2009-01-25 10:36:55 [DEBUG] switch_core_session.c:807 >>>> >> switch_core_session_signal_state_change() Send signal >>>> OpenZAP/2:1/3400534 >>>> >> [BREAK] >>>> >> 2009-01-25 10:36:55 [DEBUG] ozmod_analog.c:52 >>>> analog_fxo_outgoing_call() >>>> >> Changing state on 2:1 from DOWN to DIALING >>>> >> 2009-01-25 10:36:55 [DEBUG] ozmod_analog.c:239 >>>> zap_analog_channel_run() >>>> >> ANALOG CHANNEL thread starting. >>>> >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:379 >>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change >>>> CS_INIT >>>> >> 2009-01-25 10:36:55 [DEBUG] ozmod_analog.c:410 >>>> zap_analog_channel_run() >>>> >> Executing state handler on 2:1 for DIALING >>>> >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:444 >>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State INIT >>>> >> 2009-01-25 10:36:55 [DEBUG] mod_openzap.c:364 channel_on_init() >>>> >> (OpenZAP/2:1/3400534) State Change CS_INIT -> CS_ROUTING >>>> >> 2009-01-25 10:36:55 [DEBUG] switch_core_session.c:807 >>>> >> switch_core_session_signal_state_change() Send signal >>>> OpenZAP/2:1/3400534 >>>> >> [BREAK] >>>> >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:444 >>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State INIT going to >>>> sleep >>>> >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:379 >>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change >>>> >> CS_ROUTING >>>> >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:447 >>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State ROUTING >>>> >> 2009-01-25 10:36:55 [DEBUG] mod_openzap.c:387 channel_on_routing() >>>> >> OpenZAP/2:1/3400534 CHANNEL ROUTING >>>> >> 2009-01-25 10:36:55 [DEBUG] switch_ivr_originate.c:58 >>>> >> originate_on_routing() (OpenZAP/2:1/3400534) State Change CS_ROUTING >>>> -> >>>> >> CS_CONSUME_MEDIA >>>> >> 2009-01-25 10:36:55 [DEBUG] switch_core_session.c:807 >>>> >> switch_core_session_signal_state_change() Send signal >>>> OpenZAP/2:1/3400534 >>>> >> [BREAK] >>>> >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:447 >>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State ROUTING going >>>> to sleep >>>> >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:379 >>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change >>>> >> CS_CONSUME_MEDIA >>>> >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:466 >>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State CONSUME_MEDIA >>>> >> 2009-01-25 10:37:08 [DEBUG] ozmod_analog.c:744 process_event() EVENT >>>> >> [ONHOOK][1:1] STATE [IDLE] >>>> >> 2009-01-25 10:37:08 [DEBUG] ozmod_analog.c:780 process_event() >>>> Changing >>>> >> state on 1:1 from IDLE to DOWN >>>> >> 2009-01-25 10:37:08 [DEBUG] ozmod_analog.c:410 >>>> zap_analog_channel_run() >>>> >> Executing state handler on 1:1 for DOWN >>>> >> 2009-01-25 10:37:08 [DEBUG] mod_openzap.c:1212 on_fxs_signal() got >>>> FXS sig >>>> >> [STOP] >>>> >> 2009-01-25 10:37:08 [NOTICE] mod_openzap.c:1300 on_fxs_signal() >>>> Hangup >>>> >> OpenZAP/1:1/93400534 [CS_EXECUTE] [NORMAL_CLEARING] >>>> >> 2009-01-25 10:37:08 [DEBUG] switch_channel.c:1494 >>>> >> switch_channel_perform_hangup() Send signal OpenZAP/1:1/93400534 >>>> [KILL] >>>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_session.c:807 >>>> >> switch_core_session_signal_state_change() Send signal >>>> OpenZAP/1:1/93400534 >>>> >> [BREAK] >>>> >> 2009-01-25 10:37:08 [DEBUG] zap_io.c:1125 zap_channel_done() channel >>>> done >>>> >> 1:1 >>>> >> 2009-01-25 10:37:08 [DEBUG] ozmod_analog.c:726 >>>> zap_analog_channel_run() >>>> >> ANALOG CHANNEL 1:1 thread ended. >>>> >> 2009-01-25 10:37:08 [NOTICE] switch_ivr_originate.c:1566 >>>> >> switch_ivr_originate() Hangup OpenZAP/2:1/3400534 [CS_CONSUME_MEDIA] >>>> >> [ORIGINATOR_CANCEL] >>>> >> 2009-01-25 10:37:08 [DEBUG] switch_channel.c:1494 >>>> >> switch_channel_perform_hangup() Send signal OpenZAP/2:1/3400534 >>>> [KILL] >>>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_session.c:807 >>>> >> switch_core_session_signal_state_change() Send signal >>>> OpenZAP/2:1/3400534 >>>> >> [BREAK] >>>> >> 2009-01-25 10:37:08 [DEBUG] switch_ivr_originate.c:1691 >>>> >> switch_ivr_originate() Originate Cancelled by originator termination >>>> Cause: >>>> >> 487 [ORIGINATOR_CANCEL] >>>> >> 2009-01-25 10:37:08 [INFO] mod_dptools.c:1909 audio_bridge_function() >>>> >> Originate Failed. Cause: ORIGINATOR_CANCEL >>>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:454 >>>> >> switch_core_session_run() (OpenZAP/1:1/93400534) State EXECUTE going >>>> to >>>> >> sleep >>>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:379 >>>> >> switch_core_session_run() (OpenZAP/1:1/93400534) Running State Change >>>> >> CS_HANGUP >>>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:410 >>>> >> switch_core_session_run() (OpenZAP/1:1/93400534) State HANGUP >>>> >> 2009-01-25 10:37:08 [DEBUG] mod_openzap.c:472 channel_on_hangup() >>>> >> OpenZAP/1:1/93400534 CHANNEL HANGUP >>>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:46 >>>> >> switch_core_standard_on_hangup() OpenZAP/1:1/93400534 Standard >>>> HANGUP, >>>> >> cause: NORMAL_CLEARING >>>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:410 >>>> >> switch_core_session_run() (OpenZAP/1:1/93400534) State HANGUP going >>>> to sleep >>>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_session.c:939 >>>> >> switch_core_session_thread() Session 3 (OpenZAP/1:1/93400534) Locked, >>>> >> Waiting on external entities >>>> >> 2009-01-25 10:37:08 [NOTICE] switch_core_session.c:957 >>>> >> switch_core_session_thread() Session 3 (OpenZAP/1:1/93400534) Ended >>>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:466 >>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State CONSUME_MEDIA >>>> going to >>>> >> sleep >>>> >> 2009-01-25 10:37:08 [NOTICE] switch_core_session.c:959 >>>> >> switch_core_session_thread() Close Channel OpenZAP/1:1/93400534 >>>> [CS_HANGUP] >>>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:379 >>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change >>>> >> CS_HANGUP >>>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:410 >>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State HANGUP >>>> >> 2009-01-25 10:37:08 [DEBUG] mod_openzap.c:429 channel_on_hangup() >>>> Changing >>>> >> state on 2:1 from DIALING to HANGUP >>>> >> 2009-01-25 10:37:08 [DEBUG] mod_openzap.c:472 channel_on_hangup() >>>> >> OpenZAP/2:1/3400534 CHANNEL HANGUP >>>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:46 >>>> >> switch_core_standard_on_hangup() OpenZAP/2:1/3400534 Standard HANGUP, >>>> cause: >>>> >> ORIGINATOR_CANCEL >>>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:410 >>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State HANGUP going to >>>> sleep >>>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_session.c:939 >>>> >> switch_core_session_thread() Session 4 (OpenZAP/2:1/3400534) Locked, >>>> Waiting >>>> >> on external entities >>>> >> 2009-01-25 10:37:08 [NOTICE] switch_core_session.c:957 >>>> >> switch_core_session_thread() Session 4 (OpenZAP/2:1/3400534) Ended >>>> >> 2009-01-25 10:37:08 [NOTICE] switch_core_session.c:959 >>>> >> switch_core_session_thread() Close Channel OpenZAP/2:1/3400534 >>>> [CS_HANGUP] >>>> >> 2009-01-25 10:37:08 [DEBUG] ozmod_analog.c:410 >>>> zap_analog_channel_run() >>>> >> Executing state handler on 2:1 for HANGUP >>>> >> 2009-01-25 10:37:09 [DEBUG] ozmod_analog.c:351 >>>> zap_analog_channel_run() >>>> >> Changing state on 2:1 from HANGUP to DOWN >>>> >> 2009-01-25 10:37:09 [DEBUG] ozmod_analog.c:410 >>>> zap_analog_channel_run() >>>> >> Executing state handler on 2:1 for DOWN >>>> >> 2009-01-25 10:37:09 [DEBUG] mod_openzap.c:1153 on_fxo_signal() got >>>> FXO sig >>>> >> 2:1 [STOP] >>>> >> 2009-01-25 10:37:09 [DEBUG] zap_io.c:1125 zap_channel_done() channel >>>> done >>>> >> 2:1 >>>> >> 2009-01-25 10:37:09 [DEBUG] ozmod_analog.c:726 >>>> zap_analog_channel_run() >>>> >> ANALOG CHANNEL 2:1 thread ended. >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> _______________________________________________ >>>> >> Freeswitch-users mailing list >>>> >> Freeswitch-users at lists.freeswitch.org >>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> >> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> >> http://www.freeswitch.org >>>> >> >>>> >> >>>> >> _______________________________________________ >>>> >> Freeswitch-users mailing list >>>> >> Freeswitch-users at lists.freeswitch.org >>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> >> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> >> http://www.freeswitch.org >>>> >> >>>> > >>>> > >>>> > _______________________________________________ >>>> > Freeswitch-users mailing list >>>> > Freeswitch-users at lists.freeswitch.org >>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> > UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> > http://www.freeswitch.org >>>> > >>>> > >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:213-799-1400 >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090201/c925d46d/attachment-0001.html From gcd at i.ph Sun Feb 1 01:17:39 2009 From: gcd at i.ph (Nandy Dagondon) Date: Sun, 1 Feb 2009 17:17:39 +0800 Subject: [Freeswitch-users] TDM400 FXO can dialout only once In-Reply-To: <7d0bfd8c0902010030n6b5d3c35k7539fac38e07300d@mail.gmail.com> References: <7d0bfd8c0901250021q2969fe26u5ea79c3957989ffd@mail.gmail.com> <191c3a030901251818g284054ben60c0bfa61157824@mail.gmail.com> <7d0bfd8c0901252305v5a33fa02w4bc02deaa0c1b14b@mail.gmail.com> <87f2f3b90901260201r2e4d42a3x30367eee64b0e13b@mail.gmail.com> <7d0bfd8c0901260532g50deb359v3c92f5bb372f11a4@mail.gmail.com> <191c3a030901260833l2e07bfffkca9400d33a5483fe@mail.gmail.com> <7d0bfd8c0901261639x7c53f256yabf7347b5d22e7e4@mail.gmail.com> <7d0bfd8c0902010030n6b5d3c35k7539fac38e07300d@mail.gmail.com> Message-ID: <7d0bfd8c0902010117h7e26a3a6x2112225cb5951cd8@mail.gmail.com> hi, i found a major one. this time i deliberately set the dialtone freq to US std on my PH definition. i expect FS wont dial at all. but to my surprise, the problem is gone!! i checked the log. it indicates successful detection of DIALTONE. going on further. i noticed FXO wont hangup on busy tone. one possibility is the volume settings. the default is -7. how many dBm is this? and what is the dB equivalent per increment? tks n rgds, nandy On Sun, Feb 1, 2009 at 4:30 PM, Nandy Dagondon wrote: > hi everybody, > > i created [ph] tone definition per raul's suggestion and changed > /etc/zaptel.conf entries to: > tonezone=ph > defaultzone=ph > > but it didn't solve the problem. > i captured the console log during start-up and shutdown. i noticed openzap > related errors during shutdown. here's the snippet of the log: > > STARTUP > --------------- > 2009-02-01 15:58:10 [NOTICE] zap_io.c:2517 zap_global_init() Modules > configured: 1 > 2009-02-01 15:58:10 [INFO] zap_io.c:2341 zap_load_module() Loading IO from > /opt/freeswitch/mod/ozmod_zt.so > 2009-02-01 15:58:10 [INFO] zap_io.c:2127 load_config() auto-loaded 'zt' > 2009-02-01 15:58:10 [INFO] ozmod_zt.c:186 zt_open_range() configuring > device /dev/zap/channel channel 1 as OpenZAP device 1:1 fd:39 > 2009-02-01 15:58:10 [INFO] ozmod_zt.c:186 zt_open_range() configuring > device /dev/zap/channel channel 2 as OpenZAP device 2:1 fd:40 > 2009-02-01 15:58:10 [INFO] zap_io.c:2265 load_config() Configured 2 > channel(s) > 2009-02-01 15:58:10 [INFO] zap_io.c:2358 zap_load_module() Loading SIG from > /opt/freeswitch/mod/ozmod_analog.so > 2009-02-01 15:58:10 [INFO] zap_io.c:2474 zap_configure_span() auto-loaded > 'analog' > 2009-02-01 15:58:10 [CONSOLE] switch_loadable_module.c:857 > switch_loadable_module_load_file() Successfully Loaded [mod_openzap] > > --- DIDN'T MAKE ANY CALL --- > > SHUTDOWN > ------------------ > 2009-02-01 15:59:07 [NOTICE] switch_loadable_module.c:536 > switch_loadable_module_unprocess() Deleting API Function 'oz' > 2009-02-01 15:59:07 [CONSOLE] switch_loadable_module.c:1231 do_shutdown() > Stopping: mod_openzap > 2009-02-01 15:59:07 [INFO] zap_io.c:256 zap_channel_destroy() Closing > channel zt:1:1 fd:39 > 2009-02-01 15:59:07 [INFO] zap_io.c:256 zap_channel_destroy() Closing > channel zt:2:1 fd:40 > 2009-02-01 15:59:08 [ERR] ozmod_analog.c:899 zap_analog_run() Failure > Polling event! [no matching descriptor] > 2009-02-01 15:59:08 [ERR] ozmod_analog.c:899 zap_analog_run() Failure > Polling event! [no matching descriptor] > 2009-02-01 15:59:08 [INFO] zap_io.c:2456 zap_unload_modules() Unloading > /opt/freeswitch/mod/ozmod_analog.so > 2009-02-01 15:59:08 [INFO] zap_io.c:2441 zap_unload_modules() Unloading IO > zt > 2009-02-01 15:59:08 [INFO] zap_io.c:2456 zap_unload_modules() Unloading > /opt/freeswitch/mod/ozmod_zt.so > > i also notice the same ERR flag during shutdown after making test calls. > > any suggestion what to do next? > > tks for your assistance. > > rgds, > -nandy > > > On Tue, Jan 27, 2009 at 8:39 AM, Nandy Dagondon wrote: > >> i tested the SVN trunk version. still the same behaviour. >> -nandy >> >> >> On Tue, Jan 27, 2009 at 12:33 AM, Anthony Minessale < >> anthony.minessale at gmail.com> wrote: >> >>> The unhanded type message just means that mod_openzap does not do >>> anything with the TONE_DETECTED event that was passed >>> up from the ozmod_analog. >>> >>> On Mon, Jan 26, 2009 at 7:32 AM, Nandy Dagondon wrote: >>> >>>> that's great. yes, i'm in the philippines. there's a difference in >>>> dialtone - it's 425 Hz. >>>> -nandy >>>> >>>> >>>> >>>> On Mon, Jan 26, 2009 at 6:01 PM, Michael Collins wrote: >>>> >>>>> I have a TDM400 clone and I will see if I can reproduce these >>>>> symptoms. BTW, are you in the Philippines? Is there any difference in >>>>> the dial tone there than in the US? >>>>> -MC >>>>> >>>>> On Sun, Jan 25, 2009 at 11:05 PM, Nandy Dagondon wrote: >>>>> > i monitored the line using another phone. there's indeed dialtone in >>>>> all >>>>> > attempts. >>>>> > i see TONE_DETECTED in the first call but i wonder there's a WARNING >>>>> message >>>>> > immediately following [WARNING] mod_openzap.c:1196 on_fxo_signal() >>>>> Unhandled >>>>> > type for channel 2:1. >>>>> > the dialtone freq should be okay since it's detected in the first >>>>> call.could >>>>> > the WARNING message gives us a hint of a possible problem other than >>>>> the >>>>> > dialtone freq? >>>>> > >>>>> > okay, i'll try the SVN version next. >>>>> > >>>>> > >>>>> > On Mon, Jan 26, 2009 at 10:18 AM, Anthony Minessale >>>>> > wrote: >>>>> >> >>>>> >> Its not detecting a dial tone on the failure case. >>>>> >> Before dialing it waits until it picks up dialtone. >>>>> >> Try the svn trunk version to see if it works any better or verify >>>>> there is >>>>> >> a dialtone on the line. >>>>> >> >>>>> >> On Jan 25, 2009 6:19 PM, "Nandy Dagondon" wrote: >>>>> >> >>>>> >> hi everybody, >>>>> >> >>>>> >> i installed FS 1.0.2 to an Atom mobo (Intel D945GCLF2). it's working >>>>> using >>>>> >> IP phones, softphones and digium FXS port. but there's a problem in >>>>> dialing >>>>> >> out to PSTN using digium tdm400 fxo - it works fine on the first >>>>> attempt >>>>> >> (after starting FS) but it fails on the subsequent attempts. i >>>>> tested to >>>>> >> call using the FXS port and IP phone. same problem. >>>>> >> >>>>> >> before i place any call, i checked >oz dump 2 1 (show current state >>>>> = >>>>> >> DOWN, last state = DOWN) >>>>> >> >>>>> >> in the first call, there's this message: >>>>> >> [WARNING] mod_openzap.c:1196 on_fxo_signal() Unhandled type for >>>>> channel >>>>> >> 2:1 >>>>> >> but >>>>> >> >>>>> >> then i hangup. checked >oz dump 21 (show current state=DOWN, last >>>>> >> state=HANGUP) >>>>> >> >>>>> >> in the 2nd (and subsequent) attempts, the fxo just goes off-hook but >>>>> >> doesn't send the dtmf tones. >>>>> >> >oz dump 2 1 (shows current state = DIALING, last state = DOWN) >>>>> >> >>>>> >> has anyone encountered this problem before? i appreciate for any >>>>> help to >>>>> >> correct this problem. >>>>> >> >>>>> >> tks, >>>>> >> nandy >>>>> >> >>>>> >> >>>>> >> Environment: >>>>> >> ================== >>>>> >> kernel 2.6.18-92.1.22.el5 >>>>> >> FS 1.0.2 >>>>> >> zaptel 1.4.11 >>>>> >> oslec >>>>> >> digium TDM400P Rev. I (1-FXS, 2-FXO,3-4 vacant) >>>>> >> >>>>> >> zaptel.conf >>>>> >> ======== >>>>> >> loadzone = us >>>>> >> defaultzone=us >>>>> >> channels=1-2 >>>>> >> alaw=1-4 >>>>> >> fxsks=2 >>>>> >> fxoks=1 >>>>> >> >>>>> >> >>>>> >> openzap.conf.xml: >>>>> >> =============== >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> openzap.conf >>>>> >> ========== >>>>> >> [span zt] >>>>> >> name => OpenZAP FXS >>>>> >> number => 1 >>>>> >> fxs-channel => 1 >>>>> >> >>>>> >> [span zt] >>>>> >> name => OpenZAP FXO >>>>> >> number => 2 >>>>> >> fxo-channel => 2 >>>>> >> >>>>> >> tones.conf (the dialtone and ring tone is set to Philipping tones) >>>>> >> ======== >>>>> >> [us] >>>>> >> generate-dial => v=-7;%(1000,0,425) >>>>> >> detect-dial => 425 >>>>> >> >>>>> >> generate-ring => v=-7;%(1000,4000,425,480) >>>>> >> detect-ring => 425,480 >>>>> >> >>>>> >> generate-busy => v=-7;%(500,500,480,620) >>>>> >> detect-busy => 480,620 >>>>> >> >>>>> >> generate-attn => v=0;%(200,300,1400,1800) >>>>> >> detect-attn => 1400,1800 >>>>> >> >>>>> >> generate-callwaiting-sas => v=0;%(300,10000,440) >>>>> >> detect-callwaiting-sas => 440 >>>>> >> >>>>> >> generate-callwaiting-cas => v=0;%(80,0,2750,2130) >>>>> >> detect-callwaiting-cas => 2750,2130 >>>>> >> >>>>> >> detect-fail1 => 913.8 >>>>> >> detect-fail2 => 1370.6 >>>>> >> detect-fail3 => 776.7 >>>>> >> >>>>> >> LOG OF FIRST CALL (OK) >>>>> >> ==================== >>>>> >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:152 >>>>> >> switch_core_standard_on_execute() OpenZAP/1:1/93400534 Execute >>>>> >> bridge(openzap/2/1/3400534) >>>>> >> 2009-01-25 10:35:58 [DEBUG] mod_openzap.c:340 tech_init() Set codec >>>>> PCMU >>>>> >> 20ms >>>>> >> 2009-01-25 10:35:58 [DEBUG] mod_openzap.c:1017 >>>>> channel_outgoing_channel() >>>>> >> Connect outbound channel OpenZAP/2:1/3400534 >>>>> >> 2009-01-25 10:35:58 [NOTICE] switch_channel.c:565 >>>>> >> switch_channel_set_name() New Channel OpenZAP/2:1/3400534 >>>>> >> [e5f12114-ea88-11dd-9f5c-290fb4a527a4] >>>>> >> 2009-01-25 10:35:58 [DEBUG] mod_openzap.c:1026 >>>>> channel_outgoing_channel() >>>>> >> (OpenZAP/2:1/3400534) State Change CS_NEW -> CS_INIT >>>>> >> 2009-01-25 10:35:58 [DEBUG] switch_core_session.c:807 >>>>> >> switch_core_session_signal_state_change() Send signal >>>>> OpenZAP/2:1/3400534 >>>>> >> [BREAK] >>>>> >> 2009-01-25 10:35:58 [DEBUG] ozmod_analog.c:52 >>>>> analog_fxo_outgoing_call() >>>>> >> Changing state on 2:1 from DOWN to DIALING >>>>> >> 2009-01-25 10:35:58 [DEBUG] ozmod_analog.c:239 >>>>> zap_analog_channel_run() >>>>> >> ANALOG CHANNEL thread starting. >>>>> >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:379 >>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change >>>>> CS_INIT >>>>> >> 2009-01-25 10:35:58 [DEBUG] ozmod_analog.c:410 >>>>> zap_analog_channel_run() >>>>> >> Executing state handler on 2:1 for DIALING >>>>> >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:444 >>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State INIT >>>>> >> 2009-01-25 10:35:58 [DEBUG] mod_openzap.c:364 channel_on_init() >>>>> >> (OpenZAP/2:1/3400534) State Change CS_INIT -> CS_ROUTING >>>>> >> 2009-01-25 10:35:58 [DEBUG] switch_core_session.c:807 >>>>> >> switch_core_session_signal_state_change() Send signal >>>>> OpenZAP/2:1/3400534 >>>>> >> [BREAK] >>>>> >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:444 >>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State INIT going to >>>>> sleep >>>>> >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:379 >>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change >>>>> >> CS_ROUTING >>>>> >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:447 >>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State ROUTING >>>>> >> 2009-01-25 10:35:58 [DEBUG] mod_openzap.c:387 channel_on_routing() >>>>> >> OpenZAP/2:1/3400534 CHANNEL ROUTING >>>>> >> 2009-01-25 10:35:58 [DEBUG] switch_ivr_originate.c:58 >>>>> >> originate_on_routing() (OpenZAP/2:1/3400534) State Change CS_ROUTING >>>>> -> >>>>> >> CS_CONSUME_MEDIA >>>>> >> 2009-01-25 10:35:58 [DEBUG] switch_core_session.c:807 >>>>> >> switch_core_session_signal_state_change() Send signal >>>>> OpenZAP/2:1/3400534 >>>>> >> [BREAK] >>>>> >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:447 >>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State ROUTING going >>>>> to sleep >>>>> >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:379 >>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change >>>>> >> CS_CONSUME_MEDIA >>>>> >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:466 >>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State CONSUME_MEDIA >>>>> >> 2009-01-25 10:35:59 [DEBUG] ozmod_analog.c:615 >>>>> zap_analog_channel_run() >>>>> >> Detected tone DIAL on 2:1 >>>>> >> 2009-01-25 10:35:59 [DEBUG] mod_openzap.c:1153 on_fxo_signal() got >>>>> FXO sig >>>>> >> 2:1 [TONE_DETECTED] >>>>> >> 2009-01-25 10:35:59 [WARNING] mod_openzap.c:1196 on_fxo_signal() >>>>> Unhandled >>>>> >> type for channel 2:1 >>>>> >> 2009-01-25 10:35:59 [DEBUG] zap_io.c:1179 >>>>> zchan_activate_dtmf_buffer() >>>>> >> Created DTMF Buffer! >>>>> >> 2009-01-25 10:35:59 [DEBUG] zap_io.c:1715 handle_dtmf() 2:1 GENERATE >>>>> DTMF >>>>> >> [3400534] >>>>> >> 2009-01-25 10:36:03 [DEBUG] ozmod_analog.c:302 >>>>> zap_analog_channel_run() >>>>> >> Changing state on 2:1 from DIALING to UP >>>>> >> 2009-01-25 10:36:03 [DEBUG] ozmod_analog.c:410 >>>>> zap_analog_channel_run() >>>>> >> Executing state handler on 2:1 for UP >>>>> >> 2009-01-25 10:36:03 [DEBUG] mod_openzap.c:1153 on_fxo_signal() got >>>>> FXO sig >>>>> >> 2:1 [UP] >>>>> >> 2009-01-25 10:36:03 [DEBUG] switch_channel.c:1710 >>>>> >> switch_channel_perform_mark_answered() Send signal >>>>> OpenZAP/1:1/93400534 >>>>> >> [BREAK] >>>>> >> 2009-01-25 10:36:03 [NOTICE] mod_openzap.c:1180 on_fxo_signal() >>>>> Channel >>>>> >> [OpenZAP/2:1/3400534] has been answered >>>>> >> 2009-01-25 10:36:03 [DEBUG] switch_channel.c:177 >>>>> >> switch_channel_audio_sync() OpenZAP/2:1/3400534 receive message >>>>> [AUDIO_SYNC] >>>>> >> 2009-01-25 10:36:03 [DEBUG] switch_channel.c:1768 >>>>> >> switch_channel_perform_answer() OpenZAP/1:1/93400534 receive message >>>>> >> [ANSWER] >>>>> >> 2009-01-25 10:36:03 [DEBUG] mod_openzap.c:813 >>>>> >> channel_receive_message_fxs() Changing state on 1:1 from IDLE to UP >>>>> >> 2009-01-25 10:36:03 [NOTICE] mod_openzap.c:814 >>>>> >> channel_receive_message_fxs() Channel [OpenZAP/1:1/93400534] has >>>>> been >>>>> >> answered >>>>> >> 2009-01-25 10:36:03 [DEBUG] switch_channel.c:177 >>>>> >> switch_channel_audio_sync() OpenZAP/1:1/93400534 receive message >>>>> >> [AUDIO_SYNC] >>>>> >> 2009-01-25 10:36:03 [DEBUG] switch_core_session.c:511 >>>>> >> switch_core_session_perform_receive_message() Send signal >>>>> >> OpenZAP/1:1/93400534 [BREAK] >>>>> >> 2009-01-25 10:36:03 [DEBUG] switch_ivr_originate.c:1627 >>>>> >> switch_ivr_originate() Originate Resulted in Success: >>>>> [OpenZAP/2:1/3400534] >>>>> >> 2009-01-25 10:36:03 [DEBUG] switch_channel.c:177 >>>>> >> switch_channel_audio_sync() OpenZAP/2:1/3400534 receive message >>>>> [AUDIO_SYNC] >>>>> >> 2009-01-25 10:36:03 [DEBUG] switch_channel.c:177 >>>>> >> switch_channel_audio_sync() OpenZAP/1:1/93400534 receive message >>>>> >> [AUDIO_SYNC] >>>>> >> 2009-01-25 10:36:03 [DEBUG] switch_ivr_bridge.c:862 >>>>> >> switch_ivr_multi_threaded_bridge() OpenZAP/2:1/3400534 receive >>>>> message >>>>> >> [BRIDGE] >>>>> >> 2009-01-25 10:36:03 [DEBUG] switch_core_session.c:511 >>>>> >> switch_core_session_perform_receive_message() Send signal >>>>> >> OpenZAP/2:1/3400534 [BREAK] >>>>> >> 2009-01-25 10:36:03 [DEBUG] switch_ivr_bridge.c:869 >>>>> >> switch_ivr_multi_threaded_bridge() OpenZAP/1:1/93400534 receive >>>>> message >>>>> >> [BRIDGE] >>>>> >> 2009-01-25 10:36:03 [DEBUG] switch_core_session.c:511 >>>>> >> switch_core_session_perform_receive_message() Send signal >>>>> >> OpenZAP/1:1/93400534 [BREAK] >>>>> >> 2009-01-25 10:36:03 [DEBUG] switch_ivr_bridge.c:913 >>>>> >> switch_ivr_multi_threaded_bridge() (OpenZAP/2:1/3400534) State >>>>> Change >>>>> >> CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA >>>>> >> 2009-01-25 10:36:03 [DEBUG] switch_core_session.c:807 >>>>> >> switch_core_session_signal_state_change() Send signal >>>>> OpenZAP/2:1/3400534 >>>>> >> [BREAK] >>>>> >> 2009-01-25 10:36:03 [DEBUG] ozmod_analog.c:410 >>>>> zap_analog_channel_run() >>>>> >> Executing state handler on 1:1 for UP >>>>> >> 2009-01-25 10:36:03 [DEBUG] mod_openzap.c:1212 on_fxs_signal() got >>>>> FXS sig >>>>> >> [UP] >>>>> >> 2009-01-25 10:36:03 [DEBUG] switch_core_state_machine.c:466 >>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State CONSUME_MEDIA >>>>> going to >>>>> >> sleep >>>>> >> 2009-01-25 10:36:03 [DEBUG] switch_core_state_machine.c:379 >>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change >>>>> >> CS_EXCHANGE_MEDIA >>>>> >> 2009-01-25 10:36:03 [DEBUG] switch_core_state_machine.c:457 >>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State EXCHANGE_MEDIA >>>>> >> 2009-01-25 10:36:03 [DEBUG] mod_openzap.c:511 >>>>> channel_on_exchange_media() >>>>> >> CHANNEL EXCHANGE_MEDIA >>>>> >> 2009-01-25 10:36:16 [DEBUG] ozmod_analog.c:744 process_event() EVENT >>>>> >> [ONHOOK][1:1] STATE [UP] >>>>> >> 2009-01-25 10:36:16 [DEBUG] ozmod_analog.c:780 process_event() >>>>> Changing >>>>> >> state on 1:1 from UP to DOWN >>>>> >> 2009-01-25 10:36:16 [DEBUG] ozmod_analog.c:410 >>>>> zap_analog_channel_run() >>>>> >> Executing state handler on 1:1 for DOWN >>>>> >> 2009-01-25 10:36:16 [DEBUG] mod_openzap.c:1212 on_fxs_signal() got >>>>> FXS sig >>>>> >> [STOP] >>>>> >> 2009-01-25 10:36:16 [NOTICE] mod_openzap.c:1300 on_fxs_signal() >>>>> Hangup >>>>> >> OpenZAP/1:1/93400534 [CS_EXECUTE] [NORMAL_CLEARING] >>>>> >> 2009-01-25 10:36:16 [DEBUG] switch_channel.c:1494 >>>>> >> switch_channel_perform_hangup() Send signal OpenZAP/1:1/93400534 >>>>> [KILL] >>>>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_session.c:807 >>>>> >> switch_core_session_signal_state_change() Send signal >>>>> OpenZAP/1:1/93400534 >>>>> >> [BREAK] >>>>> >> 2009-01-25 10:36:16 [DEBUG] zap_io.c:1125 zap_channel_done() channel >>>>> done >>>>> >> 1:1 >>>>> >> 2009-01-25 10:36:16 [DEBUG] ozmod_analog.c:726 >>>>> zap_analog_channel_run() >>>>> >> ANALOG CHANNEL 1:1 thread ended. >>>>> >> 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:360 >>>>> audio_bridge_thread() >>>>> >> OpenZAP/1:1/93400534 ending bridge by request from read function >>>>> >> 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:435 >>>>> audio_bridge_thread() >>>>> >> BRIDGE THREAD DONE [OpenZAP/1:1/93400534] >>>>> >> 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:439 >>>>> audio_bridge_thread() >>>>> >> Send signal OpenZAP/2:1/3400534 [BREAK] >>>>> >> 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:354 >>>>> audio_bridge_thread() >>>>> >> OpenZAP/1:1/93400534 ending bridge by request from write function >>>>> >> 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:409 >>>>> audio_bridge_thread() >>>>> >> OpenZAP/2:1/3400534 receive message [UNBRIDGE] >>>>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_session.c:511 >>>>> >> switch_core_session_perform_receive_message() Send signal >>>>> >> OpenZAP/2:1/3400534 [BREAK] >>>>> >> 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:435 >>>>> audio_bridge_thread() >>>>> >> BRIDGE THREAD DONE [OpenZAP/2:1/3400534] >>>>> >> 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:439 >>>>> audio_bridge_thread() >>>>> >> Send signal OpenZAP/1:1/93400534 [BREAK] >>>>> >> 2009-01-25 10:36:16 [NOTICE] switch_ivr_bridge.c:470 >>>>> >> audio_bridge_on_exchange_media() Hangup OpenZAP/2:1/3400534 >>>>> >> [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] >>>>> >> 2009-01-25 10:36:16 [DEBUG] switch_channel.c:1494 >>>>> >> switch_channel_perform_hangup() Send signal OpenZAP/2:1/3400534 >>>>> [KILL] >>>>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_session.c:807 >>>>> >> switch_core_session_signal_state_change() Send signal >>>>> OpenZAP/2:1/3400534 >>>>> >> [BREAK] >>>>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:457 >>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State EXCHANGE_MEDIA >>>>> going >>>>> >> to sleep >>>>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:379 >>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change >>>>> >> CS_HANGUP >>>>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:410 >>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State HANGUP >>>>> >> 2009-01-25 10:36:16 [DEBUG] mod_openzap.c:429 channel_on_hangup() >>>>> Changing >>>>> >> state on 2:1 from UP to HANGUP >>>>> >> 2009-01-25 10:36:16 [DEBUG] mod_openzap.c:472 channel_on_hangup() >>>>> >> OpenZAP/2:1/3400534 CHANNEL HANGUP >>>>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:46 >>>>> >> switch_core_standard_on_hangup() OpenZAP/2:1/3400534 Standard >>>>> HANGUP, cause: >>>>> >> NORMAL_CLEARING >>>>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:410 >>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State HANGUP going >>>>> to sleep >>>>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_session.c:939 >>>>> >> switch_core_session_thread() Session 2 (OpenZAP/2:1/3400534) Locked, >>>>> Waiting >>>>> >> on external entities >>>>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:454 >>>>> >> switch_core_session_run() (OpenZAP/1:1/93400534) State EXECUTE going >>>>> to >>>>> >> sleep >>>>> >> 2009-01-25 10:36:16 [NOTICE] switch_core_session.c:957 >>>>> >> switch_core_session_thread() Session 2 (OpenZAP/2:1/3400534) Ended >>>>> >> 2009-01-25 10:36:16 [NOTICE] switch_core_session.c:959 >>>>> >> switch_core_session_thread() Close Channel OpenZAP/2:1/3400534 >>>>> [CS_HANGUP] >>>>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:379 >>>>> >> switch_core_session_run() (OpenZAP/1:1/93400534) Running State >>>>> Change >>>>> >> CS_HANGUP >>>>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:410 >>>>> >> switch_core_session_run() (OpenZAP/1:1/93400534) State HANGUP >>>>> >> 2009-01-25 10:36:16 [DEBUG] mod_openzap.c:472 channel_on_hangup() >>>>> >> OpenZAP/1:1/93400534 CHANNEL HANGUP >>>>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:46 >>>>> >> switch_core_standard_on_hangup() OpenZAP/1:1/93400534 Standard >>>>> HANGUP, >>>>> >> cause: NORMAL_CLEARING >>>>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:410 >>>>> >> switch_core_session_run() (OpenZAP/1:1/93400534) State HANGUP going >>>>> to sleep >>>>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_session.c:939 >>>>> >> switch_core_session_thread() Session 1 (OpenZAP/1:1/93400534) >>>>> Locked, >>>>> >> Waiting on external entities >>>>> >> 2009-01-25 10:36:16 [NOTICE] switch_core_session.c:957 >>>>> >> switch_core_session_thread() Session 1 (OpenZAP/1:1/93400534) Ended >>>>> >> 2009-01-25 10:36:16 [NOTICE] switch_core_session.c:959 >>>>> >> switch_core_session_thread() Close Channel OpenZAP/1:1/93400534 >>>>> [CS_HANGUP] >>>>> >> 2009-01-25 10:36:16 [DEBUG] ozmod_analog.c:410 >>>>> zap_analog_channel_run() >>>>> >> Executing state handler on 2:1 for HANGUP >>>>> >> 2009-01-25 10:36:17 [DEBUG] ozmod_analog.c:351 >>>>> zap_analog_channel_run() >>>>> >> Changing state on 2:1 from HANGUP to DOWN >>>>> >> 2009-01-25 10:36:17 [DEBUG] ozmod_analog.c:410 >>>>> zap_analog_channel_run() >>>>> >> Executing state handler on 2:1 for DOWN >>>>> >> 2009-01-25 10:36:17 [DEBUG] mod_openzap.c:1153 on_fxo_signal() got >>>>> FXO sig >>>>> >> 2:1 [STOP] >>>>> >> 2009-01-25 10:36:17 [DEBUG] zap_io.c:1125 zap_channel_done() channel >>>>> done >>>>> >> 2:1 >>>>> >> 2009-01-25 10:36:17 [DEBUG] ozmod_analog.c:726 >>>>> zap_analog_channel_run() >>>>> >> ANALOG CHANNEL 2:1 thread ended. >>>>> >> >>>>> >> LOG OF FAILED CALLS >>>>> >> ================== >>>>> >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:152 >>>>> >> switch_core_standard_on_execute() OpenZAP/1:1/93400534 Execute >>>>> >> bridge(openzap/2/1/3400534) >>>>> >> 2009-01-25 10:36:55 [DEBUG] mod_openzap.c:340 tech_init() Set codec >>>>> PCMU >>>>> >> 20ms >>>>> >> 2009-01-25 10:36:55 [DEBUG] mod_openzap.c:1017 >>>>> channel_outgoing_channel() >>>>> >> Connect outbound channel OpenZAP/2:1/3400534 >>>>> >> 2009-01-25 10:36:55 [NOTICE] switch_channel.c:565 >>>>> >> switch_channel_set_name() New Channel OpenZAP/2:1/3400534 >>>>> >> [079f5420-ea89-11dd-9f5c-290fb4a527a4] >>>>> >> 2009-01-25 10:36:55 [DEBUG] mod_openzap.c:1026 >>>>> channel_outgoing_channel() >>>>> >> (OpenZAP/2:1/3400534) State Change CS_NEW -> CS_INIT >>>>> >> 2009-01-25 10:36:55 [DEBUG] switch_core_session.c:807 >>>>> >> switch_core_session_signal_state_change() Send signal >>>>> OpenZAP/2:1/3400534 >>>>> >> [BREAK] >>>>> >> 2009-01-25 10:36:55 [DEBUG] ozmod_analog.c:52 >>>>> analog_fxo_outgoing_call() >>>>> >> Changing state on 2:1 from DOWN to DIALING >>>>> >> 2009-01-25 10:36:55 [DEBUG] ozmod_analog.c:239 >>>>> zap_analog_channel_run() >>>>> >> ANALOG CHANNEL thread starting. >>>>> >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:379 >>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change >>>>> CS_INIT >>>>> >> 2009-01-25 10:36:55 [DEBUG] ozmod_analog.c:410 >>>>> zap_analog_channel_run() >>>>> >> Executing state handler on 2:1 for DIALING >>>>> >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:444 >>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State INIT >>>>> >> 2009-01-25 10:36:55 [DEBUG] mod_openzap.c:364 channel_on_init() >>>>> >> (OpenZAP/2:1/3400534) State Change CS_INIT -> CS_ROUTING >>>>> >> 2009-01-25 10:36:55 [DEBUG] switch_core_session.c:807 >>>>> >> switch_core_session_signal_state_change() Send signal >>>>> OpenZAP/2:1/3400534 >>>>> >> [BREAK] >>>>> >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:444 >>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State INIT going to >>>>> sleep >>>>> >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:379 >>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change >>>>> >> CS_ROUTING >>>>> >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:447 >>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State ROUTING >>>>> >> 2009-01-25 10:36:55 [DEBUG] mod_openzap.c:387 channel_on_routing() >>>>> >> OpenZAP/2:1/3400534 CHANNEL ROUTING >>>>> >> 2009-01-25 10:36:55 [DEBUG] switch_ivr_originate.c:58 >>>>> >> originate_on_routing() (OpenZAP/2:1/3400534) State Change CS_ROUTING >>>>> -> >>>>> >> CS_CONSUME_MEDIA >>>>> >> 2009-01-25 10:36:55 [DEBUG] switch_core_session.c:807 >>>>> >> switch_core_session_signal_state_change() Send signal >>>>> OpenZAP/2:1/3400534 >>>>> >> [BREAK] >>>>> >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:447 >>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State ROUTING going >>>>> to sleep >>>>> >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:379 >>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change >>>>> >> CS_CONSUME_MEDIA >>>>> >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:466 >>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State CONSUME_MEDIA >>>>> >> 2009-01-25 10:37:08 [DEBUG] ozmod_analog.c:744 process_event() EVENT >>>>> >> [ONHOOK][1:1] STATE [IDLE] >>>>> >> 2009-01-25 10:37:08 [DEBUG] ozmod_analog.c:780 process_event() >>>>> Changing >>>>> >> state on 1:1 from IDLE to DOWN >>>>> >> 2009-01-25 10:37:08 [DEBUG] ozmod_analog.c:410 >>>>> zap_analog_channel_run() >>>>> >> Executing state handler on 1:1 for DOWN >>>>> >> 2009-01-25 10:37:08 [DEBUG] mod_openzap.c:1212 on_fxs_signal() got >>>>> FXS sig >>>>> >> [STOP] >>>>> >> 2009-01-25 10:37:08 [NOTICE] mod_openzap.c:1300 on_fxs_signal() >>>>> Hangup >>>>> >> OpenZAP/1:1/93400534 [CS_EXECUTE] [NORMAL_CLEARING] >>>>> >> 2009-01-25 10:37:08 [DEBUG] switch_channel.c:1494 >>>>> >> switch_channel_perform_hangup() Send signal OpenZAP/1:1/93400534 >>>>> [KILL] >>>>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_session.c:807 >>>>> >> switch_core_session_signal_state_change() Send signal >>>>> OpenZAP/1:1/93400534 >>>>> >> [BREAK] >>>>> >> 2009-01-25 10:37:08 [DEBUG] zap_io.c:1125 zap_channel_done() channel >>>>> done >>>>> >> 1:1 >>>>> >> 2009-01-25 10:37:08 [DEBUG] ozmod_analog.c:726 >>>>> zap_analog_channel_run() >>>>> >> ANALOG CHANNEL 1:1 thread ended. >>>>> >> 2009-01-25 10:37:08 [NOTICE] switch_ivr_originate.c:1566 >>>>> >> switch_ivr_originate() Hangup OpenZAP/2:1/3400534 [CS_CONSUME_MEDIA] >>>>> >> [ORIGINATOR_CANCEL] >>>>> >> 2009-01-25 10:37:08 [DEBUG] switch_channel.c:1494 >>>>> >> switch_channel_perform_hangup() Send signal OpenZAP/2:1/3400534 >>>>> [KILL] >>>>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_session.c:807 >>>>> >> switch_core_session_signal_state_change() Send signal >>>>> OpenZAP/2:1/3400534 >>>>> >> [BREAK] >>>>> >> 2009-01-25 10:37:08 [DEBUG] switch_ivr_originate.c:1691 >>>>> >> switch_ivr_originate() Originate Cancelled by originator termination >>>>> Cause: >>>>> >> 487 [ORIGINATOR_CANCEL] >>>>> >> 2009-01-25 10:37:08 [INFO] mod_dptools.c:1909 >>>>> audio_bridge_function() >>>>> >> Originate Failed. Cause: ORIGINATOR_CANCEL >>>>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:454 >>>>> >> switch_core_session_run() (OpenZAP/1:1/93400534) State EXECUTE going >>>>> to >>>>> >> sleep >>>>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:379 >>>>> >> switch_core_session_run() (OpenZAP/1:1/93400534) Running State >>>>> Change >>>>> >> CS_HANGUP >>>>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:410 >>>>> >> switch_core_session_run() (OpenZAP/1:1/93400534) State HANGUP >>>>> >> 2009-01-25 10:37:08 [DEBUG] mod_openzap.c:472 channel_on_hangup() >>>>> >> OpenZAP/1:1/93400534 CHANNEL HANGUP >>>>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:46 >>>>> >> switch_core_standard_on_hangup() OpenZAP/1:1/93400534 Standard >>>>> HANGUP, >>>>> >> cause: NORMAL_CLEARING >>>>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:410 >>>>> >> switch_core_session_run() (OpenZAP/1:1/93400534) State HANGUP going >>>>> to sleep >>>>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_session.c:939 >>>>> >> switch_core_session_thread() Session 3 (OpenZAP/1:1/93400534) >>>>> Locked, >>>>> >> Waiting on external entities >>>>> >> 2009-01-25 10:37:08 [NOTICE] switch_core_session.c:957 >>>>> >> switch_core_session_thread() Session 3 (OpenZAP/1:1/93400534) Ended >>>>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:466 >>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State CONSUME_MEDIA >>>>> going to >>>>> >> sleep >>>>> >> 2009-01-25 10:37:08 [NOTICE] switch_core_session.c:959 >>>>> >> switch_core_session_thread() Close Channel OpenZAP/1:1/93400534 >>>>> [CS_HANGUP] >>>>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:379 >>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change >>>>> >> CS_HANGUP >>>>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:410 >>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State HANGUP >>>>> >> 2009-01-25 10:37:08 [DEBUG] mod_openzap.c:429 channel_on_hangup() >>>>> Changing >>>>> >> state on 2:1 from DIALING to HANGUP >>>>> >> 2009-01-25 10:37:08 [DEBUG] mod_openzap.c:472 channel_on_hangup() >>>>> >> OpenZAP/2:1/3400534 CHANNEL HANGUP >>>>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:46 >>>>> >> switch_core_standard_on_hangup() OpenZAP/2:1/3400534 Standard >>>>> HANGUP, cause: >>>>> >> ORIGINATOR_CANCEL >>>>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:410 >>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State HANGUP going >>>>> to sleep >>>>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_session.c:939 >>>>> >> switch_core_session_thread() Session 4 (OpenZAP/2:1/3400534) Locked, >>>>> Waiting >>>>> >> on external entities >>>>> >> 2009-01-25 10:37:08 [NOTICE] switch_core_session.c:957 >>>>> >> switch_core_session_thread() Session 4 (OpenZAP/2:1/3400534) Ended >>>>> >> 2009-01-25 10:37:08 [NOTICE] switch_core_session.c:959 >>>>> >> switch_core_session_thread() Close Channel OpenZAP/2:1/3400534 >>>>> [CS_HANGUP] >>>>> >> 2009-01-25 10:37:08 [DEBUG] ozmod_analog.c:410 >>>>> zap_analog_channel_run() >>>>> >> Executing state handler on 2:1 for HANGUP >>>>> >> 2009-01-25 10:37:09 [DEBUG] ozmod_analog.c:351 >>>>> zap_analog_channel_run() >>>>> >> Changing state on 2:1 from HANGUP to DOWN >>>>> >> 2009-01-25 10:37:09 [DEBUG] ozmod_analog.c:410 >>>>> zap_analog_channel_run() >>>>> >> Executing state handler on 2:1 for DOWN >>>>> >> 2009-01-25 10:37:09 [DEBUG] mod_openzap.c:1153 on_fxo_signal() got >>>>> FXO sig >>>>> >> 2:1 [STOP] >>>>> >> 2009-01-25 10:37:09 [DEBUG] zap_io.c:1125 zap_channel_done() channel >>>>> done >>>>> >> 2:1 >>>>> >> 2009-01-25 10:37:09 [DEBUG] ozmod_analog.c:726 >>>>> zap_analog_channel_run() >>>>> >> ANALOG CHANNEL 2:1 thread ended. >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> _______________________________________________ >>>>> >> Freeswitch-users mailing list >>>>> >> Freeswitch-users at lists.freeswitch.org >>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> >> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> >> http://www.freeswitch.org >>>>> >> >>>>> >> >>>>> >> _______________________________________________ >>>>> >> Freeswitch-users mailing list >>>>> >> Freeswitch-users at lists.freeswitch.org >>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> >> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> >> http://www.freeswitch.org >>>>> >> >>>>> > >>>>> > >>>>> > _______________________________________________ >>>>> > Freeswitch-users mailing list >>>>> > Freeswitch-users at lists.freeswitch.org >>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> > UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> > http://www.freeswitch.org >>>>> > >>>>> > >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> Anthony Minessale II >>> >>> FreeSWITCH http://www.freeswitch.org/ >>> ClueCon http://www.cluecon.com/ >>> >>> AIM: anthm >>> MSN:anthony_minessale at hotmail.com >>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>> IRC: irc.freenode.net #freeswitch >>> >>> FreeSWITCH Developer Conference >>> sip:888 at conference.freeswitch.org >>> iax:guest at conference.freeswitch.org/888 >>> googletalk:conf+888 at conference.freeswitch.org >>> pstn:213-799-1400 >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090201/88e279e2/attachment-0001.html From pmhshz at gmail.com Sun Feb 1 02:02:04 2009 From: pmhshz at gmail.com (shehzad p) Date: Sun, 1 Feb 2009 02:02:04 -0800 (PST) Subject: [Freeswitch-users] Freeswitch freezes on increasing call traffic In-Reply-To: <191c3a030901310850s553f8c11oc4e559a710def6d2@mail.gmail.com> References: <21701744.post@talk.nabble.com> <191c3a030901280601t3e185ec2m76295110e4888837@mail.gmail.com> <21729863.post@talk.nabble.com> <191c3a030901290831y2ff329b3maee6c1aafa3c68e2@mail.gmail.com> <21745312.post@talk.nabble.com> <21749375.post@talk.nabble.com> <191c3a030901300818i5d85a7aawa0c53b4e54892f40@mail.gmail.com> <21761523.post@talk.nabble.com> <191c3a030901310850s553f8c11oc4e559a710def6d2@mail.gmail.com> Message-ID: <21773300.post@talk.nabble.com> Hi Anthony, How can I handle Garbage collector related things. Is there any way to look out where js variables are causing some kind of fault related to Garbage collector? There are some loops but they finish in hardly two or three iterations. I didn't check session.ready() before/after fetching values from database (odbc), which might block until response from server comes back. might this cause fault as you said? Thanks for help... Anthony Minessale-2 wrote: > > Clearly you have an issue with your javascript code. > > You have the Garbage collector blocking in every thread. > > Are you doing any endless loops in your code where you do not check > session.ready() as a condition for > continuing the script? > > any time session.ready() fails you must immediately exit. > > Are you using session.execute to execute long blocking operations like > bridging many calls or entering a conference? > You should avoid doing this as all the collective scripts on the system > share a common Garbage Collector provided by the > JS engine and it can lead to the exact issues you describe if the code is > not properly designed. > > What else does you script do that are things provided by FS such as > playing > files and executing applications. > > > > On Sat, Jan 31, 2009 at 3:44 AM, shehzad p wrote: > >> >> Hi Anthony, >> >> Freeswitch 1.0.2 was crashed on last test again... >> BT is on http://pastebin.freeswitch.org/6979. >> >> I tried to use scripts/freeswitch-gcore, to capture resident memory, but >> before the command complete the process system hanged up so only half >> output >> was captured.< http://www.nabble.com/file/p21761523/gcore-fs.txt >> gcore-fs.txt > >> >> >> Now I have checkout from trunk and will post back if any new thing >> found... >> >> Thanks, >> msp >> >> Anthony Minessale-2 wrote: >> > >> > if you are using unix you can use the supplied script >> > >> > scripts/freeswitch-gcore >> > >> > to capture a copy of the resident memory and I can have a look perhaps. >> > >> > Trunk is safe for production as we are in beta stage for a release of >> > 1.0.3 >> > at this time. >> > >> > >> > >> > On Fri, Jan 30, 2009 at 9:29 AM, shehzad p wrote: >> > >> >> >> >> When freeswitch freezes, we can't connect to it to check sps status, >> >> but once we were able to connect and at that time it was showing 0/0 >> sps. >> >> >> >> thanks... >> >> >> >> shehzad p wrote: >> >> > >> >> > Thanks, Anthony >> >> > >> >> > In my previous test sps did not changed, >> >> > but in recent test sps was dropped to 0 itself (as below). >> >> > =============================================================== >> >> > UP 0 years, 0 days, 5 hours, 1 minute, 53 seconds, 878 milliseconds, >> >> 190 >> >> > microseconds >> >> > 5474 session(s) since startup >> >> > 75 session(s) 0/0 >> >> > ============================================================= >> >> > >> >> > My system is 32 bit. >> >> > CPU is Intel(R) Xeon(R) CPU X3220 @ 2.40GHz >> >> > And RAM is 4GB >> >> > >> >> > Output of ulimit -a is: >> >> > ulimit -a: (set after first test) >> >> > core file size (blocks, -c) unlimited >> >> > data seg size (kbytes, -d) unlimited >> >> > max nice (-e) 20 >> >> > file size (blocks, -f) unlimited >> >> > pending signals (-i) unlimited >> >> > max locked memory (kbytes, -l) unlimited >> >> > max memory size (kbytes, -m) unlimited >> >> > open files (-n) 999999 >> >> > pipe size (512 bytes, -p) 8 >> >> > POSIX message queues (bytes, -q) unlimited >> >> > max rt priority (-r) unlimited >> >> > stack size (kbytes, -s) 244 >> >> > cpu time (seconds, -t) unlimited >> >> > max user processes (-u) unlimited >> >> > virtual memory (kbytes, -v) unlimited >> >> > file locks (-x) unlimited >> >> > =================================================== >> >> > >> >> > >> >> > BTW using trunk on production system is safe? >> >> > >> >> > Warm thanks for kind responses... >> >> > >> >> > >> >> > >> >> > Anthony Minessale-2 wrote: >> >> >> >> >> >> When you get it in that state what do you see when you execute >> >> >> >> >> >> fsctl sps >> >> >> >> >> >> is the sps a very low number? >> >> >> >> >> >> Did the sps drop by itself from the value you originally set it to? >> >> >> >> >> >> Are you using 32 bit? >> >> >> >> >> >> if so try all of these commands in your shell before starting FS >> >> >> >> >> >> ulimit -c unlimited >> >> >> ulimit -d unlimited >> >> >> ulimit -f unlimited >> >> >> ulimit -i unlimited >> >> >> ulimit -n 999999 >> >> >> ulimit -q unlimited >> >> >> ulimit -u unlimited >> >> >> ulimit -v unlimited >> >> >> ulimit -x unlimited >> >> >> ulimit -s 244 >> >> >> ulimit -l unlimited >> >> >> >> >> >> >> >> >> DO NOT put them in a script unless you source the script with . >> >> >> . myscript or they will be undone instantly when the script exits >> >> >> >> >> >> BTW, I said to try latest trunk not 1.0.2 We can only debug the >> >> >> development >> >> >> code at this point. >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> On Thu, Jan 29, 2009 at 10:06 AM, shehzad p >> wrote: >> >> >> >> >> >>> >> >> >>> Hi Anthony, >> >> >>> >> >> >>> I found interesting result while testing Freeswitch, and it might >> be >> >> >>> cause >> >> >>> of freezing out of freeswitch, >> >> >>> >> >> >>> I updated my system (as you told) with latest stable version >> >> Freeswitch >> >> >>> 1.0.2 >> >> >>> First of all I set sps to 100, >> >> >>> Then I sends call approximately 100 per seconds, Freeswitch works >> >> fine >> >> >>> and >> >> >>> handles all the calls very well. >> >> >>> >> >> >>> After that I send 130 calls per seconds, and magic happen now, >> >> >>> Freeswitch >> >> >>> handles first 100 calls only. >> >> >>> all the preceding calls were failed (even not appeared in >> freeswitch >> >> >>> console >> >> >>> why?) >> >> >>> >> >> >>> When I put ngrep trace, System responds with 503 Maximum Calls In >> >> >>> Progress. >> >> >>> (as below) >> >> >>> ########################################################### >> >> >>> # >> >> >>> U FSFSFSFSFS -> GWGWGWGWGW >> >> >>> SIP/2.0 503 Maximum Calls In Progress. >> >> >>> Via: SIP/2.0/UDP GWGWGWGWGW;branch=z9hG4bK53eafabb;rport=5060. >> >> >>> From: "99999" ;tag=as2e10c170. >> >> >>> To: ;tag=K3jSUFrDHpmmB. >> >> >>> Call-ID: 0feb229e58afe7be17110deb361dc234 at GWGWGWGWGW. >> >> >>> CSeq: 102 INVITE. >> >> >>> Retry-After: 300. >> >> >>> User-Agent: FreeSWITCH-mod_sofia/1.0.2-exported. >> >> >>> Accept: application/sdp. >> >> >>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, >> SUBSCRIBE, >> >> >>> NOTIFY, >> >> >>> REFER, UPDATE, REGISTER, INFO, PUBLISH. >> >> >>> Supported: timer, precondition, path, replaces. >> >> >>> Allow-Events: talk, presence, dialog, call-info, sla, >> >> >>> include-session-description, presence.winfo, message-summary, >> refer. >> >> >>> Content-Length: 0. >> >> >>> . >> >> >>> >> ##################################################################### >> >> >>> >> >> >>> >> >> >>> Now another issue to note down is that, >> >> >>> After all above happened and active calls comes to zero, >> >> >>> I just make a single call which also fails with response 503 - >> >> Maximum >> >> >>> Calls >> >> >>> In Progress. >> >> >>> >> >> >>> >> >> >>> Is this intended behaviour, should I increase SPS to overcome >> this. >> >> or >> >> >>> something like bug. >> >> >>> >> >> >>> Please let me know what should be the resolution for this. >> >> >>> >> >> >>> Thanks, >> >> >>> msp >> >> >>> >> >> >>> >> >> >>> >> >> >>> Anthony Minessale-2 wrote: >> >> >>> > >> >> >>> > Also remember, >> >> >>> > Actually completely uninstall and erase /usr/local/freeswitch >> and >> >> the >> >> >>> > 1.0.1 >> >> >>> > source tree and freshly install the new one. >> >> >>> > If you try to upgrade on top of a release with trunk it will >> cause >> >> >>> more >> >> >>> > problems for you. >> >> >>> > >> >> >>> > >> >> >>> > On Wed, Jan 28, 2009 at 3:11 AM, Ken Rice >> >> >>> wrote: >> >> >>> > >> >> >>> >> Upgrade to trunk... Many many issues have been resolved since >> >> 1.0.1 >> >> >>> was >> >> >>> >> the >> >> >>> >> current release >> >> >>> >> >> >> >>> >> >> >> >>> >> > From: shehzad p >> >> >>> >> > Reply-To: >> >> >>> >> > Date: Wed, 28 Jan 2009 00:54:13 -0800 (PST) >> >> >>> >> > To: >> >> >>> >> > Subject: [Freeswitch-users] Freeswitch freezes on increasing >> >> call >> >> >>> >> traffic >> >> >>> >> > >> >> >>> >> > >> >> >>> >> > Hi all, >> >> >>> >> > >> >> >>> >> > Yesterday my Freeswitch server faced a problem when call >> traffic >> >> >>> >> increased >> >> >>> >> > to more than 100. >> >> >>> >> > >> >> >>> >> > When I start Freeswitch, it works fine and then after some >> time >> >> >>> >> > (approximately 15 to 20 minutes) it stops functioning (means >> no >> >> >>> call >> >> >>> >> is >> >> >>> >> > being processed, no CLI command is working and it just >> freezes) >> >> >>> until >> >> >>> I >> >> >>> >> > restart the freeswitch. >> >> >>> >> > >> >> >>> >> > I am using Freeswitch 1.0.1. >> >> >>> >> > Debug (gdb) trace as on wiki page >> >> >>> >> > >> >> http://wiki.freeswitch.org/wiki/Debugging_Freeswitch#fs_debug.shis >> >> >>> >> attached >> >> >>> >> > http://www.nabble.com/file/p21701744/fs_debgu.txt >> fs_debgu.txt >> >> >>> >> > -- >> >> >>> >> > View this message in context: >> >> >>> >> > >> >> >>> >> >> >> >>> >> >> >> http://www.nabble.com/Freeswitch-freezes-on-increasing-call-traffic-tp21701744 >> >> >>> >> > p21701744.html >> >> >>> >> > Sent from the Freeswitch-users mailing list archive at >> >> Nabble.com. >> >> >>> >> > >> >> >>> >> > >> >> >>> >> > _______________________________________________ >> >> >>> >> > Freeswitch-users mailing list >> >> >>> >> > Freeswitch-users at lists.freeswitch.org >> >> >>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >>> >> > >> >> >>> >> UNSUBSCRIBE: >> >> >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> >>> >> > http://www.freeswitch.org >> >> >>> >> >> >> >>> >> >> >> >>> >> >> >> >>> >> _______________________________________________ >> >> >>> >> Freeswitch-users mailing list >> >> >>> >> Freeswitch-users at lists.freeswitch.org >> >> >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >>> >> UNSUBSCRIBE: >> >> >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> >>> >> http://www.freeswitch.org >> >> >>> >> >> >> >>> > >> >> >>> > >> >> >>> > >> >> >>> > -- >> >> >>> > Anthony Minessale II >> >> >>> > >> >> >>> > FreeSWITCH http://www.freeswitch.org/ >> >> >>> > ClueCon http://www.cluecon.com/ >> >> >>> > >> >> >>> > AIM: anthm >> >> >>> > >> >> MSN:anthony_minessale at hotmail.com >> >> >> > >> >> >>> >> >> >> >> >> > >> >> >< >> >> >>> >> >> >> MSN%3Aanthony_minessale at hotmail.com >> >> > >> >> >> >> >> > >> >> > >> >> >>> > >> >> >>> > >> >> >>> >> >> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> >> > >> >> >> >> >> > >> >> > >> >> >>> >> >> >> >> >> > >> >> >> >> >> > >> >> > >> >> >>> > >> >> >>> > IRC: irc.freenode.net #freeswitch >> >> >>> > >> >> >>> > FreeSWITCH Developer Conference >> >> >>> > >> >> sip:888 at conference.freeswitch.org >> >> >> > >> >> >>> >> >> >> >> >> > >> >> >< >> >> >>> >> >> >> sip%3A888 at conference.freeswitch.org >> >> > >> >> >> >> >> > >> >> > >> >> >>> > >> >> >>> > iax:guest at conference.freeswitch.org/888 >> >> >>> > >> >> >>> >> >> >> googletalk:conf+888 at conference.freeswitch.org >> >> > >> >> >> >> >> > >> >> > >> >> >>> >> >> >> >> >> > >> >> >> >> >> > >> >> > >> >> >>> > >> >> >>> > pstn:213-799-1400 >> >> >>> > >> >> >>> > _______________________________________________ >> >> >>> > Freeswitch-users mailing list >> >> >>> > Freeswitch-users at lists.freeswitch.org >> >> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >>> > >> >> >>> UNSUBSCRIBE: >> >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> >>> > http://www.freeswitch.org >> >> >>> > >> >> >>> > >> >> >>> >> >> >>> -- >> >> >>> View this message in context: >> >> >>> >> >> >> http://www.nabble.com/Freeswitch-freezes-on-increasing-call-traffic-tp21701744p21729863.html >> >> >>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >>> >> >> >>> >> >> >>> _______________________________________________ >> >> >>> Freeswitch-users mailing list >> >> >>> Freeswitch-users at lists.freeswitch.org >> >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >>> UNSUBSCRIBE: >> >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> >>> http://www.freeswitch.org >> >> >>> >> >> >> >> >> >> >> >> >> >> >> >> -- >> >> >> Anthony Minessale II >> >> >> >> >> >> FreeSWITCH http://www.freeswitch.org/ >> >> >> ClueCon http://www.cluecon.com/ >> >> >> >> >> >> AIM: anthm >> >> >> >> MSN:anthony_minessale at hotmail.com >> >> >> >> >< >> >> >> MSN%3Aanthony_minessale at hotmail.com >> >> > >> >> > >> >> >> >> >> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> >> > >> >> >> >> >> > >> >> > >> >> >> IRC: irc.freenode.net #freeswitch >> >> >> >> >> >> FreeSWITCH Developer Conference >> >> >> >> sip:888 at conference.freeswitch.org >> >> >> >> >< >> >> >> sip%3A888 at conference.freeswitch.org >> >> > >> >> > >> >> >> iax:guest at conference.freeswitch.org/888 >> >> >> >> >> >> googletalk:conf+888 at conference.freeswitch.org >> >> > >> >> >> >> >> > >> >> > >> >> >> pstn:213-799-1400 >> >> >> >> >> >> _______________________________________________ >> >> >> Freeswitch-users mailing list >> >> >> Freeswitch-users at lists.freeswitch.org >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> UNSUBSCRIBE: >> >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> >> http://www.freeswitch.org >> >> >> >> >> >> >> >> > >> >> > >> >> >> >> -- >> >> View this message in context: >> >> >> http://www.nabble.com/Freeswitch-freezes-on-increasing-call-traffic-tp21701744p21749375.html >> >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> >> >> >> _______________________________________________ >> >> Freeswitch-users mailing list >> >> Freeswitch-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> >> >> > >> > >> > >> > -- >> > Anthony Minessale II >> > >> > FreeSWITCH http://www.freeswitch.org/ >> > ClueCon http://www.cluecon.com/ >> > >> > AIM: anthm >> > MSN:anthony_minessale at hotmail.com >> < >> MSN%3Aanthony_minessale at hotmail.com >> > >> > >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> >> > >> > IRC: irc.freenode.net #freeswitch >> > >> > FreeSWITCH Developer Conference >> > sip:888 at conference.freeswitch.org >> < >> sip%3A888 at conference.freeswitch.org >> > >> > iax:guest at conference.freeswitch.org/888 >> > >> googletalk:conf+888 at conference.freeswitch.org >> >> > >> > pstn:213-799-1400 >> > >> > _______________________________________________ >> > Freeswitch-users mailing list >> > Freeswitch-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> >> -- >> View this message in context: >> http://www.nabble.com/Freeswitch-freezes-on-increasing-call-traffic-tp21701744p21761523.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/Freeswitch-freezes-on-increasing-call-traffic-tp21701744p21773300.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From blansky at interwise.com Sun Feb 1 02:37:26 2009 From: blansky at interwise.com (Boris Lansky) Date: Sun, 1 Feb 2009 12:37:26 +0200 Subject: [Freeswitch-users] Outbound Proxy configuration In-Reply-To: <4981D3F8.2030207@freeswitch.org> References: <7160CD9F-F06E-4964-943A-F4B0C12150CC@jerris.com> <4981D3F8.2030207@freeswitch.org> Message-ID: I have checked the "fs_path" usage ... again (I have done it before I have issued my question as well). And once again I can't understand why this thing is useful for me. I have found only two small examples refer the issue that use "fs_path" for a an API call. What I need is a real configuration example that shows configuration of a Proxy Server for all outbound calls going out from a Free Switch. I will real appreciate if I will get an exact answer and not just a general link to a FS doc. Regards, Boris Lansky Unified Communications Telephony Team AT&T Unified Communications Phone: +972.3.976.7604 Fax: +972.3.976.7712 blansky at interwise.com From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Raymond Chandler Sent: Thursday, January 29, 2009 6:06 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Outbound Proxy configuration Boris Lansky wrote: Sorry for the stupid question but in what configuration file should I add such line "sofia/foo/user at that.domain ;fs_path=sip:proxy.this.domain" ? appology accepted... look in the dialplan, there should be plenty of documentations on using the dialplan on our wiki... wiki.freeswitch.org... look for sofia syntax too on the mod_sofia page -Ray -------------- next part -------------- An HTML attachment was scrubbed... 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Name: not available Type: image/png Size: 16211 bytes Desc: image001.png Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090201/3f1f8beb/attachment-0001.png From pmhshz at gmail.com Sun Feb 1 05:35:07 2009 From: pmhshz at gmail.com (shehzad p) Date: Sun, 1 Feb 2009 05:35:07 -0800 (PST) Subject: [Freeswitch-users] Freeswitch freezes on increasing call traffic In-Reply-To: <21773300.post@talk.nabble.com> References: <21701744.post@talk.nabble.com> <191c3a030901280601t3e185ec2m76295110e4888837@mail.gmail.com> <21729863.post@talk.nabble.com> <191c3a030901290831y2ff329b3maee6c1aafa3c68e2@mail.gmail.com> <21745312.post@talk.nabble.com> <21749375.post@talk.nabble.com> <191c3a030901300818i5d85a7aawa0c53b4e54892f40@mail.gmail.com> <21761523.post@talk.nabble.com> <191c3a030901310850s553f8c11oc4e559a710def6d2@mail.gmail.com> <21773300.post@talk.nabble.com> Message-ID: <21775256.post@talk.nabble.com> Hi anthony, There are so many crash occured, on FS 1.0.1 Server, all of them were same (means BT was same) but this last one was looking totally different so I feel that it should be posted... http://www.nabble.com/file/p21775256/bt_full_new.txt bt_full_new.txt shehzad p wrote: > > Hi Anthony, > > How can I handle Garbage collector related things. Is there any way to > look out where js variables are causing some kind of fault related to > Garbage collector? > > There are some loops but they finish in hardly two or three iterations. > > I didn't check session.ready() before/after fetching values from database > (odbc), which might block until response from server comes back. might > this cause fault as you said? > > Thanks for help... > > > > > Anthony Minessale-2 wrote: >> >> Clearly you have an issue with your javascript code. >> >> You have the Garbage collector blocking in every thread. >> >> Are you doing any endless loops in your code where you do not check >> session.ready() as a condition for >> continuing the script? >> >> any time session.ready() fails you must immediately exit. >> >> Are you using session.execute to execute long blocking operations like >> bridging many calls or entering a conference? >> You should avoid doing this as all the collective scripts on the system >> share a common Garbage Collector provided by the >> JS engine and it can lead to the exact issues you describe if the code is >> not properly designed. >> >> What else does you script do that are things provided by FS such as >> playing >> files and executing applications. >> >> >> >> On Sat, Jan 31, 2009 at 3:44 AM, shehzad p wrote: >> >>> >>> Hi Anthony, >>> >>> Freeswitch 1.0.2 was crashed on last test again... >>> BT is on http://pastebin.freeswitch.org/6979. >>> >>> I tried to use scripts/freeswitch-gcore, to capture resident memory, but >>> before the command complete the process system hanged up so only half >>> output >>> was captured.< http://www.nabble.com/file/p21761523/gcore-fs.txt >>> gcore-fs.txt > >>> >>> >>> Now I have checkout from trunk and will post back if any new thing >>> found... >>> >>> Thanks, >>> msp >>> >>> Anthony Minessale-2 wrote: >>> > >>> > if you are using unix you can use the supplied script >>> > >>> > scripts/freeswitch-gcore >>> > >>> > to capture a copy of the resident memory and I can have a look >>> perhaps. >>> > >>> > Trunk is safe for production as we are in beta stage for a release of >>> > 1.0.3 >>> > at this time. >>> > >>> > >>> > >>> > On Fri, Jan 30, 2009 at 9:29 AM, shehzad p wrote: >>> > >>> >> >>> >> When freeswitch freezes, we can't connect to it to check sps status, >>> >> but once we were able to connect and at that time it was showing 0/0 >>> sps. >>> >> >>> >> thanks... >>> >> >>> >> shehzad p wrote: >>> >> > >>> >> > Thanks, Anthony >>> >> > >>> >> > In my previous test sps did not changed, >>> >> > but in recent test sps was dropped to 0 itself (as below). >>> >> > =============================================================== >>> >> > UP 0 years, 0 days, 5 hours, 1 minute, 53 seconds, 878 >>> milliseconds, >>> >> 190 >>> >> > microseconds >>> >> > 5474 session(s) since startup >>> >> > 75 session(s) 0/0 >>> >> > ============================================================= >>> >> > >>> >> > My system is 32 bit. >>> >> > CPU is Intel(R) Xeon(R) CPU X3220 @ 2.40GHz >>> >> > And RAM is 4GB >>> >> > >>> >> > Output of ulimit -a is: >>> >> > ulimit -a: (set after first test) >>> >> > core file size (blocks, -c) unlimited >>> >> > data seg size (kbytes, -d) unlimited >>> >> > max nice (-e) 20 >>> >> > file size (blocks, -f) unlimited >>> >> > pending signals (-i) unlimited >>> >> > max locked memory (kbytes, -l) unlimited >>> >> > max memory size (kbytes, -m) unlimited >>> >> > open files (-n) 999999 >>> >> > pipe size (512 bytes, -p) 8 >>> >> > POSIX message queues (bytes, -q) unlimited >>> >> > max rt priority (-r) unlimited >>> >> > stack size (kbytes, -s) 244 >>> >> > cpu time (seconds, -t) unlimited >>> >> > max user processes (-u) unlimited >>> >> > virtual memory (kbytes, -v) unlimited >>> >> > file locks (-x) unlimited >>> >> > =================================================== >>> >> > >>> >> > >>> >> > BTW using trunk on production system is safe? >>> >> > >>> >> > Warm thanks for kind responses... >>> >> > >>> >> > >>> >> > >>> >> > Anthony Minessale-2 wrote: >>> >> >> >>> >> >> When you get it in that state what do you see when you execute >>> >> >> >>> >> >> fsctl sps >>> >> >> >>> >> >> is the sps a very low number? >>> >> >> >>> >> >> Did the sps drop by itself from the value you originally set it >>> to? >>> >> >> >>> >> >> Are you using 32 bit? >>> >> >> >>> >> >> if so try all of these commands in your shell before starting FS >>> >> >> >>> >> >> ulimit -c unlimited >>> >> >> ulimit -d unlimited >>> >> >> ulimit -f unlimited >>> >> >> ulimit -i unlimited >>> >> >> ulimit -n 999999 >>> >> >> ulimit -q unlimited >>> >> >> ulimit -u unlimited >>> >> >> ulimit -v unlimited >>> >> >> ulimit -x unlimited >>> >> >> ulimit -s 244 >>> >> >> ulimit -l unlimited >>> >> >> >>> >> >> >>> >> >> DO NOT put them in a script unless you source the script with . >>> >> >> . myscript or they will be undone instantly when the script exits >>> >> >> >>> >> >> BTW, I said to try latest trunk not 1.0.2 We can only debug the >>> >> >> development >>> >> >> code at this point. >>> >> >> >>> >> >> >>> >> >> >>> >> >> >>> >> >> >>> >> >> On Thu, Jan 29, 2009 at 10:06 AM, shehzad p >>> wrote: >>> >> >> >>> >> >>> >>> >> >>> Hi Anthony, >>> >> >>> >>> >> >>> I found interesting result while testing Freeswitch, and it might >>> be >>> >> >>> cause >>> >> >>> of freezing out of freeswitch, >>> >> >>> >>> >> >>> I updated my system (as you told) with latest stable version >>> >> Freeswitch >>> >> >>> 1.0.2 >>> >> >>> First of all I set sps to 100, >>> >> >>> Then I sends call approximately 100 per seconds, Freeswitch works >>> >> fine >>> >> >>> and >>> >> >>> handles all the calls very well. >>> >> >>> >>> >> >>> After that I send 130 calls per seconds, and magic happen now, >>> >> >>> Freeswitch >>> >> >>> handles first 100 calls only. >>> >> >>> all the preceding calls were failed (even not appeared in >>> freeswitch >>> >> >>> console >>> >> >>> why?) >>> >> >>> >>> >> >>> When I put ngrep trace, System responds with 503 Maximum Calls In >>> >> >>> Progress. >>> >> >>> (as below) >>> >> >>> ########################################################### >>> >> >>> # >>> >> >>> U FSFSFSFSFS -> GWGWGWGWGW >>> >> >>> SIP/2.0 503 Maximum Calls In Progress. >>> >> >>> Via: SIP/2.0/UDP GWGWGWGWGW;branch=z9hG4bK53eafabb;rport=5060. >>> >> >>> From: "99999" ;tag=as2e10c170. >>> >> >>> To: ;tag=K3jSUFrDHpmmB. >>> >> >>> Call-ID: 0feb229e58afe7be17110deb361dc234 at GWGWGWGWGW. >>> >> >>> CSeq: 102 INVITE. >>> >> >>> Retry-After: 300. >>> >> >>> User-Agent: FreeSWITCH-mod_sofia/1.0.2-exported. >>> >> >>> Accept: application/sdp. >>> >> >>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, >>> SUBSCRIBE, >>> >> >>> NOTIFY, >>> >> >>> REFER, UPDATE, REGISTER, INFO, PUBLISH. >>> >> >>> Supported: timer, precondition, path, replaces. >>> >> >>> Allow-Events: talk, presence, dialog, call-info, sla, >>> >> >>> include-session-description, presence.winfo, message-summary, >>> refer. >>> >> >>> Content-Length: 0. >>> >> >>> . >>> >> >>> >>> ##################################################################### >>> >> >>> >>> >> >>> >>> >> >>> Now another issue to note down is that, >>> >> >>> After all above happened and active calls comes to zero, >>> >> >>> I just make a single call which also fails with response 503 - >>> >> Maximum >>> >> >>> Calls >>> >> >>> In Progress. >>> >> >>> >>> >> >>> >>> >> >>> Is this intended behaviour, should I increase SPS to overcome >>> this. >>> >> or >>> >> >>> something like bug. >>> >> >>> >>> >> >>> Please let me know what should be the resolution for this. >>> >> >>> >>> >> >>> Thanks, >>> >> >>> msp >>> >> >>> >>> >> >>> >>> >> >>> >>> >> >>> Anthony Minessale-2 wrote: >>> >> >>> > >>> >> >>> > Also remember, >>> >> >>> > Actually completely uninstall and erase /usr/local/freeswitch >>> and >>> >> the >>> >> >>> > 1.0.1 >>> >> >>> > source tree and freshly install the new one. >>> >> >>> > If you try to upgrade on top of a release with trunk it will >>> cause >>> >> >>> more >>> >> >>> > problems for you. >>> >> >>> > >>> >> >>> > >>> >> >>> > On Wed, Jan 28, 2009 at 3:11 AM, Ken Rice >>> >>> >> >>> wrote: >>> >> >>> > >>> >> >>> >> Upgrade to trunk... Many many issues have been resolved since >>> >> 1.0.1 >>> >> >>> was >>> >> >>> >> the >>> >> >>> >> current release >>> >> >>> >> >>> >> >>> >> >>> >> >>> >> > From: shehzad p >>> >> >>> >> > Reply-To: >>> >> >>> >> > Date: Wed, 28 Jan 2009 00:54:13 -0800 (PST) >>> >> >>> >> > To: >>> >> >>> >> > Subject: [Freeswitch-users] Freeswitch freezes on >>> increasing >>> >> call >>> >> >>> >> traffic >>> >> >>> >> > >>> >> >>> >> > >>> >> >>> >> > Hi all, >>> >> >>> >> > >>> >> >>> >> > Yesterday my Freeswitch server faced a problem when call >>> traffic >>> >> >>> >> increased >>> >> >>> >> > to more than 100. >>> >> >>> >> > >>> >> >>> >> > When I start Freeswitch, it works fine and then after some >>> time >>> >> >>> >> > (approximately 15 to 20 minutes) it stops functioning >>> (means >>> no >>> >> >>> call >>> >> >>> >> is >>> >> >>> >> > being processed, no CLI command is working and it just >>> freezes) >>> >> >>> until >>> >> >>> I >>> >> >>> >> > restart the freeswitch. >>> >> >>> >> > >>> >> >>> >> > I am using Freeswitch 1.0.1. >>> >> >>> >> > Debug (gdb) trace as on wiki page >>> >> >>> >> > >>> >> http://wiki.freeswitch.org/wiki/Debugging_Freeswitch#fs_debug.shis >>> >> >>> >> attached >>> >> >>> >> > http://www.nabble.com/file/p21701744/fs_debgu.txt >>> fs_debgu.txt >>> >> >>> >> > -- >>> >> >>> >> > View this message in context: >>> >> >>> >> > >>> >> >>> >> >>> >> >>> >>> >> >>> http://www.nabble.com/Freeswitch-freezes-on-increasing-call-traffic-tp21701744 >>> >> >>> >> > p21701744.html >>> >> >>> >> > Sent from the Freeswitch-users mailing list archive at >>> >> Nabble.com. >>> >> >>> >> > >>> >> >>> >> > >>> >> >>> >> > _______________________________________________ >>> >> >>> >> > Freeswitch-users mailing list >>> >> >>> >> > Freeswitch-users at lists.freeswitch.org >>> >> >>> >> > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >> >>> >> > >>> >> >>> >> UNSUBSCRIBE: >>> >> >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >> >>> >> > http://www.freeswitch.org >>> >> >>> >> >>> >> >>> >> >>> >> >>> >> >>> >> >>> >> _______________________________________________ >>> >> >>> >> Freeswitch-users mailing list >>> >> >>> >> Freeswitch-users at lists.freeswitch.org >>> >> >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >> >>> >> UNSUBSCRIBE: >>> >> >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >> >>> >> http://www.freeswitch.org >>> >> >>> >> >>> >> >>> > >>> >> >>> > >>> >> >>> > >>> >> >>> > -- >>> >> >>> > Anthony Minessale II >>> >> >>> > >>> >> >>> > FreeSWITCH http://www.freeswitch.org/ >>> >> >>> > ClueCon http://www.cluecon.com/ >>> >> >>> > >>> >> >>> > AIM: anthm >>> >> >>> > >>> >> MSN:anthony_minessale at hotmail.com >>> >>> >>> > >>> >> >>> >>> >> >>> >>> >>> > >>> >> >< >>> >> >>> >>> >> >>> MSN%3Aanthony_minessale at hotmail.com >>> >>> > >>> >> >>> >>> >>> > >>> >> > >>> >> >>> > >>> >> >>> > >>> >> >>> >>> >> >>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>> >>> > >>> >> >>> >>> >>> > >>> >> > >>> >> >>> >>> >> >>> >>> >>> > >>> >> >>> >>> >>> > >>> >> > >>> >> >>> > >>> >> >>> > IRC: irc.freenode.net #freeswitch >>> >> >>> > >>> >> >>> > FreeSWITCH Developer Conference >>> >> >>> > >>> >> sip:888 at conference.freeswitch.org >>> >>> >>> > >>> >> >>> >>> >> >>> >>> >>> > >>> >> >< >>> >> >>> >>> >> >>> sip%3A888 at conference.freeswitch.org >>> >>> > >>> >> >>> >>> >>> > >>> >> > >>> >> >>> > >>> >> >>> > iax:guest at conference.freeswitch.org/888 >>> >> >>> > >>> >> >>> >>> >> >>> googletalk:conf+888 at conference.freeswitch.org >>> >>> > >>> >> >>> >>> >>> > >>> >> > >>> >> >>> >>> >> >>> >>> >>> > >>> >> >>> >>> >>> > >>> >> > >>> >> >>> > >>> >> >>> > pstn:213-799-1400 >>> >> >>> > >>> >> >>> > _______________________________________________ >>> >> >>> > Freeswitch-users mailing list >>> >> >>> > Freeswitch-users at lists.freeswitch.org >>> >> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >> >>> > >>> >> >>> UNSUBSCRIBE: >>> >> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >> >>> > http://www.freeswitch.org >>> >> >>> > >>> >> >>> > >>> >> >>> >>> >> >>> -- >>> >> >>> View this message in context: >>> >> >>> >>> >> >>> http://www.nabble.com/Freeswitch-freezes-on-increasing-call-traffic-tp21701744p21729863.html >>> >> >>> Sent from the Freeswitch-users mailing list archive at >>> Nabble.com. >>> >> >>> >>> >> >>> >>> >> >>> _______________________________________________ >>> >> >>> Freeswitch-users mailing list >>> >> >>> Freeswitch-users at lists.freeswitch.org >>> >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >> >>> UNSUBSCRIBE: >>> >> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >> >>> http://www.freeswitch.org >>> >> >>> >>> >> >> >>> >> >> >>> >> >> >>> >> >> -- >>> >> >> Anthony Minessale II >>> >> >> >>> >> >> FreeSWITCH http://www.freeswitch.org/ >>> >> >> ClueCon http://www.cluecon.com/ >>> >> >> >>> >> >> AIM: anthm >>> >> >> >>> MSN:anthony_minessale at hotmail.com >>> >> >>> >>> >< >>> >> >>> MSN%3Aanthony_minessale at hotmail.com >>> >>> > >>> >> > >>> >> >> >>> >> >>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>> >>> > >>> >> >>> >>> >>> > >>> >> > >>> >> >> IRC: irc.freenode.net #freeswitch >>> >> >> >>> >> >> FreeSWITCH Developer Conference >>> >> >> >>> sip:888 at conference.freeswitch.org >>> >> >>> >>> >< >>> >> >>> sip%3A888 at conference.freeswitch.org >>> >>> > >>> >> > >>> >> >> iax:guest at conference.freeswitch.org/888 >>> >> >> >>> >> >>> googletalk:conf+888 at conference.freeswitch.org >>> >>> > >>> >> >>> >>> >>> > >>> >> > >>> >> >> pstn:213-799-1400 >>> >> >> >>> >> >> _______________________________________________ >>> >> >> Freeswitch-users mailing list >>> >> >> Freeswitch-users at lists.freeswitch.org >>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >> >> UNSUBSCRIBE: >>> >> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >> >> http://www.freeswitch.org >>> >> >> >>> >> >> >>> >> > >>> >> > >>> >> >>> >> -- >>> >> View this message in context: >>> >> >>> http://www.nabble.com/Freeswitch-freezes-on-increasing-call-traffic-tp21701744p21749375.html >>> >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>> >> >>> >> >>> >> _______________________________________________ >>> >> Freeswitch-users mailing list >>> >> Freeswitch-users at lists.freeswitch.org >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >> UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >> http://www.freeswitch.org >>> >> >>> > >>> > >>> > >>> > -- >>> > Anthony Minessale II >>> > >>> > FreeSWITCH http://www.freeswitch.org/ >>> > ClueCon http://www.cluecon.com/ >>> > >>> > AIM: anthm >>> > MSN:anthony_minessale at hotmail.com >>> < >>> MSN%3Aanthony_minessale at hotmail.com >>> > >>> > >>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>> >>> > >>> > IRC: irc.freenode.net #freeswitch >>> > >>> > FreeSWITCH Developer Conference >>> > sip:888 at conference.freeswitch.org >>> < >>> sip%3A888 at conference.freeswitch.org >>> > >>> > iax:guest at conference.freeswitch.org/888 >>> > >>> googletalk:conf+888 at conference.freeswitch.org >>> >>> > >>> > pstn:213-799-1400 >>> > >>> > _______________________________________________ >>> > Freeswitch-users mailing list >>> > Freeswitch-users at lists.freeswitch.org >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> > >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> > http://www.freeswitch.org >>> > >>> > >>> >>> -- >>> View this message in context: >>> http://www.nabble.com/Freeswitch-freezes-on-increasing-call-traffic-tp21701744p21761523.html >>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:213-799-1400 >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> > > -- View this message in context: http://www.nabble.com/Freeswitch-freezes-on-increasing-call-traffic-tp21701744p21775256.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From brian at freeswitch.org Sun Feb 1 08:43:50 2009 From: brian at freeswitch.org (Brian West) Date: Sun, 1 Feb 2009 10:43:50 -0600 Subject: [Freeswitch-users] Freeswitch freezes on increasing call traffic In-Reply-To: <21775256.post@talk.nabble.com> References: <21701744.post@talk.nabble.com> <191c3a030901280601t3e185ec2m76295110e4888837@mail.gmail.com> <21729863.post@talk.nabble.com> <191c3a030901290831y2ff329b3maee6c1aafa3c68e2@mail.gmail.com> <21745312.post@talk.nabble.com> <21749375.post@talk.nabble.com> <191c3a030901300818i5d85a7aawa0c53b4e54892f40@mail.gmail.com> <21761523.post@talk.nabble.com> <191c3a030901310850s553f8c11oc4e559a710def6d2@mail.gmail.com> <21773300.post@talk.nabble.com> <21775256.post@talk.nabble.com> Message-ID: <30BF0913-A90C-4472-9A70-6CD6AEE2AD08@freeswitch.org> I highly recommend you upgrade to 1.03RC1 or SVN Trunk. Chances are we have already fixed these issues. /b On Feb 1, 2009, at 7:35 AM, shehzad p wrote: > > Hi anthony, > > There are so many crash occured, on FS 1.0.1 Server, all of them > were same > (means BT was same) > but this last one was looking totally different so I feel that it > should be > posted... > http://www.nabble.com/file/p21775256/bt_full_new.txt bt_full_new.txt -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090201/3297f1f6/attachment.html From peder at networkoblivion.com Sun Feb 1 10:03:11 2009 From: peder at networkoblivion.com (peder at networkoblivion.com) Date: Sun, 01 Feb 2009 12:03:11 -0600 Subject: [Freeswitch-users] Cisco 7975G and XML options for G722 In-Reply-To: <897947BF-9886-47BB-B8BB-8DE8849437F0@freeswitch.org> References: <0C8E2714-6D96-49DB-93CD-0D377E4DAE96@freeswitch.org> <3885f4fe0901312115k1309573lb160ecbc83f8d57@mail.gmail.com> <897947BF-9886-47BB-B8BB-8DE8849437F0@freeswitch.org> Message-ID: <4985E3DF.1000805@networkoblivion.com> I have an xml config file for the 79x1, but there is no mention of wideband and/or g722. I found a doc on CCO that says that the 7941 and 7961 support g722 for sccp and sip, so if you find some xml that mentions g722, I would appreciate seeing it as I have a couple of those phones. I know there is a "" entry, so maybe you just add it there like on the 79x0 versions 'preferred_codec: "g729"'. Peder Brian West wrote: > I found it finally. Now if I had the full XML I could pick apart for > all the options that would help too. > > /b > > On Jan 31, 2009, at 11:15 PM, Ron McCarthy wrote: > >> Wow, can't even find the tech doc via my CCO login even. Gotta love >> Cisco and the support for SIP... >> >> Probably have to find someone with CCM and look at a config that CCM >> made up, as that should have it if they enabled for the phone. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > ' From anthony.minessale at gmail.com Sun Feb 1 10:19:00 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sun, 1 Feb 2009 12:19:00 -0600 Subject: [Freeswitch-users] Outbound Proxy configuration In-Reply-To: References: <7160CD9F-F06E-4964-943A-F4B0C12150CC@jerris.com> <4981D3F8.2030207@freeswitch.org> Message-ID: <191c3a030902011019k94e5fa3i1a37c2c0b2e66e52@mail.gmail.com> They probably saw your signature that says you work for AT&T and assumed you could follow the documentation since AT&T is a telephone company. What they were trying to explain was if you want to send a call out of a sofia profile you can set the proxy on the fly with that extra parameter added to the dial string anywhere it's used. The explanation requires you understand the basic operation of freeswitch. the uri contained in fs_path indicates the sip address of a proxy server. On Sun, Feb 1, 2009 at 4:37 AM, Boris Lansky wrote: > I have checked the "fs_path" usage ? again (I have done it before I have > issued my question as well). And once again I can't understand why this > thing is useful for me. I have found only two small examples refer the issue > that use "fs_path" for a an API call. What I need is a real configuration > example that shows configuration of a Proxy Server for all outbound calls > going out from a Free Switch. I will real appreciate if I will get an exact > answer and not just a general link to a FS doc. > > > > Regards, > > > > *Boris Lansky* > *Unified Communications Telephony Team* > > *AT&T Unified Communications > *Phone: +972.3.976.7604 > > Fax: +972.3.976.7712 > > blansky at interwise.com > > [image: cid:image001.png at 01C8E00D.2EC2BC90] > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Raymond > Chandler > *Sent:* Thursday, January 29, 2009 6:06 PM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Outbound Proxy configuration > > > > Boris Lansky wrote: > > Sorry for the stupid question but in what configuration file should I add such line "sofia/foo/user at that.domain ;fs_path=sip:proxy.this.domain" ? > > > > appology accepted... look in the dialplan, there should be plenty of > documentations on using the dialplan on our wiki... wiki.freeswitch.org... > look for sofia syntax too on the mod_sofia page > > -Ray > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... 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Name: not available Type: image/png Size: 16211 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090201/0fdbe3e8/attachment-0001.png From brian at freeswitch.org Sun Feb 1 10:33:29 2009 From: brian at freeswitch.org (Brian West) Date: Sun, 1 Feb 2009 12:33:29 -0600 Subject: [Freeswitch-users] Cisco 7975G and XML options for G722 In-Reply-To: <4985E3DF.1000805@networkoblivion.com> References: <0C8E2714-6D96-49DB-93CD-0D377E4DAE96@freeswitch.org> <3885f4fe0901312115k1309573lb160ecbc83f8d57@mail.gmail.com> <897947BF-9886-47BB-B8BB-8DE8849437F0@freeswitch.org> <4985E3DF.1000805@networkoblivion.com> Message-ID: Cisco 7970 and the G.722 codecThe G.722 codec is disabled by default on the 7970G phone. - To enable the G722 codec, add the following line inside of the context: 2 - Add the following line within the context: 1 On a config generated by CUCM 6, the 'advertiseG722Codec' context usually appears on a new line following the context. - It is also useful to add the following lines inside of the context, but these are purely optional: 0 0 0 1 From voip-info. /b On Feb 1, 2009, at 12:03 PM, peder at networkoblivion.com wrote: > I have an xml config file for the 79x1, but there is no mention of > wideband and/or g722. I found a doc on CCO that says that the 7941 > and > 7961 support g722 for sccp and sip, so if you find some xml that > mentions g722, I would appreciate seeing it as I have a couple of > those > phones. I know there is a "" > entry, so > maybe you just add it there like on the 79x0 versions > 'preferred_codec: > "g729"'. > > Peder > > > Brian West wrote: >> I found it finally. Now if I had the full XML I could pick apart >> for >> all the options that would help too. >> >> /b >> >> On Jan 31, 2009, at 11:15 PM, Ron McCarthy wrote: >> >>> Wow, can't even find the tech doc via my CCO login even. Gotta love >>> Cisco and the support for SIP... >>> >>> Probably have to find someone with CCM and look at a config that CCM >>> made up, as that should have it if they enabled for the phone. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > ' > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From gkuri at ieee.org Sun Feb 1 12:55:07 2009 From: gkuri at ieee.org (Gabriel Kuri) Date: Sun, 01 Feb 2009 12:55:07 -0800 Subject: [Freeswitch-users] Cisco 7975G and XML options for G722 In-Reply-To: <897947BF-9886-47BB-B8BB-8DE8849437F0@freeswitch.org> References: <0C8E2714-6D96-49DB-93CD-0D377E4DAE96@freeswitch.org> <3885f4fe0901312115k1309573lb160ecbc83f8d57@mail.gmail.com> <897947BF-9886-47BB-B8BB-8DE8849437F0@freeswitch.org> Message-ID: <49860C2B.7020801@ieee.org> I don't know if this helps, but I attached a config file generated by CUCM for a 7975. I don't believe CUCM writes out all the possible config options into the XML file, although it does write out quite a bit. If you're looking for another option, let me know and I can see if I can enable it and write out the config again. Gabe Brian West wrote: > I found it finally. Now if I had the full XML I could pick apart for > all the options that would help too. > > /b > > On Jan 31, 2009, at 11:15 PM, Ron McCarthy wrote: > >> Wow, can't even find the tech doc via my CCO login even. Gotta love >> Cisco and the support for SIP... >> >> Probably have to find someone with CCM and look at a config that CCM >> made up, as that should have it if they enabled for the phone. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- A non-text attachment was scrubbed... Name: SEP000E03123456.cnf.xml Type: text/xml Size: 7980 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090201/bce27527/attachment.xml From anthony.minessale at gmail.com Sun Feb 1 10:54:59 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sun, 1 Feb 2009 12:54:59 -0600 Subject: [Freeswitch-users] TDM400 FXO can dialout only once In-Reply-To: <7d0bfd8c0902010117h7e26a3a6x2112225cb5951cd8@mail.gmail.com> References: <7d0bfd8c0901250021q2969fe26u5ea79c3957989ffd@mail.gmail.com> <191c3a030901251818g284054ben60c0bfa61157824@mail.gmail.com> <7d0bfd8c0901252305v5a33fa02w4bc02deaa0c1b14b@mail.gmail.com> <87f2f3b90901260201r2e4d42a3x30367eee64b0e13b@mail.gmail.com> <7d0bfd8c0901260532g50deb359v3c92f5bb372f11a4@mail.gmail.com> <191c3a030901260833l2e07bfffkca9400d33a5483fe@mail.gmail.com> <7d0bfd8c0901261639x7c53f256yabf7347b5d22e7e4@mail.gmail.com> <7d0bfd8c0902010030n6b5d3c35k7539fac38e07300d@mail.gmail.com> <7d0bfd8c0902010117h7e26a3a6x2112225cb5951cd8@mail.gmail.com> Message-ID: <191c3a030902011054k1d3df502ydca8ae1b645abb10@mail.gmail.com> the value is dB already -7dB but for the detection section you do not specify anything but the list of frequencies needed to detect. detect-ring => 440,480 this means it needs to detect a 440+480 to know there is a dialtone, the generate value has nothing to do with it. openzap does not currently do busy detection to detect a hangup, but up in freeswitch you can use the tone_detect app to do this. On Sun, Feb 1, 2009 at 3:17 AM, Nandy Dagondon wrote: > hi, > > i found a major one. this time i deliberately set the dialtone freq to US > std on my PH definition. i expect FS wont dial at all. but to my surprise, > the problem is gone!! i checked the log. it indicates successful detection > of DIALTONE. > > going on further. i noticed FXO wont hangup on busy tone. > > one possibility is the volume settings. the default is -7. how many dBm is > this? and what is the dB equivalent per increment? > > tks n rgds, > nandy > > > On Sun, Feb 1, 2009 at 4:30 PM, Nandy Dagondon wrote: > >> hi everybody, >> >> i created [ph] tone definition per raul's suggestion and changed >> /etc/zaptel.conf entries to: >> tonezone=ph >> defaultzone=ph >> >> but it didn't solve the problem. >> i captured the console log during start-up and shutdown. i noticed openzap >> related errors during shutdown. here's the snippet of the log: >> >> STARTUP >> --------------- >> 2009-02-01 15:58:10 [NOTICE] zap_io.c:2517 zap_global_init() Modules >> configured: 1 >> 2009-02-01 15:58:10 [INFO] zap_io.c:2341 zap_load_module() Loading IO from >> /opt/freeswitch/mod/ozmod_zt.so >> 2009-02-01 15:58:10 [INFO] zap_io.c:2127 load_config() auto-loaded 'zt' >> 2009-02-01 15:58:10 [INFO] ozmod_zt.c:186 zt_open_range() configuring >> device /dev/zap/channel channel 1 as OpenZAP device 1:1 fd:39 >> 2009-02-01 15:58:10 [INFO] ozmod_zt.c:186 zt_open_range() configuring >> device /dev/zap/channel channel 2 as OpenZAP device 2:1 fd:40 >> 2009-02-01 15:58:10 [INFO] zap_io.c:2265 load_config() Configured 2 >> channel(s) >> 2009-02-01 15:58:10 [INFO] zap_io.c:2358 zap_load_module() Loading SIG >> from /opt/freeswitch/mod/ozmod_analog.so >> 2009-02-01 15:58:10 [INFO] zap_io.c:2474 zap_configure_span() auto-loaded >> 'analog' >> 2009-02-01 15:58:10 [CONSOLE] switch_loadable_module.c:857 >> switch_loadable_module_load_file() Successfully Loaded [mod_openzap] >> >> --- DIDN'T MAKE ANY CALL --- >> >> SHUTDOWN >> ------------------ >> 2009-02-01 15:59:07 [NOTICE] switch_loadable_module.c:536 >> switch_loadable_module_unprocess() Deleting API Function 'oz' >> 2009-02-01 15:59:07 [CONSOLE] switch_loadable_module.c:1231 do_shutdown() >> Stopping: mod_openzap >> 2009-02-01 15:59:07 [INFO] zap_io.c:256 zap_channel_destroy() Closing >> channel zt:1:1 fd:39 >> 2009-02-01 15:59:07 [INFO] zap_io.c:256 zap_channel_destroy() Closing >> channel zt:2:1 fd:40 >> 2009-02-01 15:59:08 [ERR] ozmod_analog.c:899 zap_analog_run() Failure >> Polling event! [no matching descriptor] >> 2009-02-01 15:59:08 [ERR] ozmod_analog.c:899 zap_analog_run() Failure >> Polling event! [no matching descriptor] >> 2009-02-01 15:59:08 [INFO] zap_io.c:2456 zap_unload_modules() Unloading >> /opt/freeswitch/mod/ozmod_analog.so >> 2009-02-01 15:59:08 [INFO] zap_io.c:2441 zap_unload_modules() Unloading IO >> zt >> 2009-02-01 15:59:08 [INFO] zap_io.c:2456 zap_unload_modules() Unloading >> /opt/freeswitch/mod/ozmod_zt.so >> >> i also notice the same ERR flag during shutdown after making test calls. >> >> any suggestion what to do next? >> >> tks for your assistance. >> >> rgds, >> -nandy >> >> >> On Tue, Jan 27, 2009 at 8:39 AM, Nandy Dagondon wrote: >> >>> i tested the SVN trunk version. still the same behaviour. >>> -nandy >>> >>> >>> On Tue, Jan 27, 2009 at 12:33 AM, Anthony Minessale < >>> anthony.minessale at gmail.com> wrote: >>> >>>> The unhanded type message just means that mod_openzap does not do >>>> anything with the TONE_DETECTED event that was passed >>>> up from the ozmod_analog. >>>> >>>> On Mon, Jan 26, 2009 at 7:32 AM, Nandy Dagondon wrote: >>>> >>>>> that's great. yes, i'm in the philippines. there's a difference in >>>>> dialtone - it's 425 Hz. >>>>> -nandy >>>>> >>>>> >>>>> >>>>> On Mon, Jan 26, 2009 at 6:01 PM, Michael Collins wrote: >>>>> >>>>>> I have a TDM400 clone and I will see if I can reproduce these >>>>>> symptoms. BTW, are you in the Philippines? Is there any difference in >>>>>> the dial tone there than in the US? >>>>>> -MC >>>>>> >>>>>> On Sun, Jan 25, 2009 at 11:05 PM, Nandy Dagondon wrote: >>>>>> > i monitored the line using another phone. there's indeed dialtone in >>>>>> all >>>>>> > attempts. >>>>>> > i see TONE_DETECTED in the first call but i wonder there's a WARNING >>>>>> message >>>>>> > immediately following [WARNING] mod_openzap.c:1196 on_fxo_signal() >>>>>> Unhandled >>>>>> > type for channel 2:1. >>>>>> > the dialtone freq should be okay since it's detected in the first >>>>>> call.could >>>>>> > the WARNING message gives us a hint of a possible problem other than >>>>>> the >>>>>> > dialtone freq? >>>>>> > >>>>>> > okay, i'll try the SVN version next. >>>>>> > >>>>>> > >>>>>> > On Mon, Jan 26, 2009 at 10:18 AM, Anthony Minessale >>>>>> > wrote: >>>>>> >> >>>>>> >> Its not detecting a dial tone on the failure case. >>>>>> >> Before dialing it waits until it picks up dialtone. >>>>>> >> Try the svn trunk version to see if it works any better or verify >>>>>> there is >>>>>> >> a dialtone on the line. >>>>>> >> >>>>>> >> On Jan 25, 2009 6:19 PM, "Nandy Dagondon" wrote: >>>>>> >> >>>>>> >> hi everybody, >>>>>> >> >>>>>> >> i installed FS 1.0.2 to an Atom mobo (Intel D945GCLF2). it's >>>>>> working using >>>>>> >> IP phones, softphones and digium FXS port. but there's a problem in >>>>>> dialing >>>>>> >> out to PSTN using digium tdm400 fxo - it works fine on the first >>>>>> attempt >>>>>> >> (after starting FS) but it fails on the subsequent attempts. i >>>>>> tested to >>>>>> >> call using the FXS port and IP phone. same problem. >>>>>> >> >>>>>> >> before i place any call, i checked >oz dump 2 1 (show current >>>>>> state = >>>>>> >> DOWN, last state = DOWN) >>>>>> >> >>>>>> >> in the first call, there's this message: >>>>>> >> [WARNING] mod_openzap.c:1196 on_fxo_signal() Unhandled type for >>>>>> channel >>>>>> >> 2:1 >>>>>> >> but >>>>>> >> >>>>>> >> then i hangup. checked >oz dump 21 (show current state=DOWN, last >>>>>> >> state=HANGUP) >>>>>> >> >>>>>> >> in the 2nd (and subsequent) attempts, the fxo just goes off-hook >>>>>> but >>>>>> >> doesn't send the dtmf tones. >>>>>> >> >oz dump 2 1 (shows current state = DIALING, last state = DOWN) >>>>>> >> >>>>>> >> has anyone encountered this problem before? i appreciate for any >>>>>> help to >>>>>> >> correct this problem. >>>>>> >> >>>>>> >> tks, >>>>>> >> nandy >>>>>> >> >>>>>> >> >>>>>> >> Environment: >>>>>> >> ================== >>>>>> >> kernel 2.6.18-92.1.22.el5 >>>>>> >> FS 1.0.2 >>>>>> >> zaptel 1.4.11 >>>>>> >> oslec >>>>>> >> digium TDM400P Rev. I (1-FXS, 2-FXO,3-4 vacant) >>>>>> >> >>>>>> >> zaptel.conf >>>>>> >> ======== >>>>>> >> loadzone = us >>>>>> >> defaultzone=us >>>>>> >> channels=1-2 >>>>>> >> alaw=1-4 >>>>>> >> fxsks=2 >>>>>> >> fxoks=1 >>>>>> >> >>>>>> >> >>>>>> >> openzap.conf.xml: >>>>>> >> =============== >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> >> openzap.conf >>>>>> >> ========== >>>>>> >> [span zt] >>>>>> >> name => OpenZAP FXS >>>>>> >> number => 1 >>>>>> >> fxs-channel => 1 >>>>>> >> >>>>>> >> [span zt] >>>>>> >> name => OpenZAP FXO >>>>>> >> number => 2 >>>>>> >> fxo-channel => 2 >>>>>> >> >>>>>> >> tones.conf (the dialtone and ring tone is set to Philipping tones) >>>>>> >> ======== >>>>>> >> [us] >>>>>> >> generate-dial => v=-7;%(1000,0,425) >>>>>> >> detect-dial => 425 >>>>>> >> >>>>>> >> generate-ring => v=-7;%(1000,4000,425,480) >>>>>> >> detect-ring => 425,480 >>>>>> >> >>>>>> >> generate-busy => v=-7;%(500,500,480,620) >>>>>> >> detect-busy => 480,620 >>>>>> >> >>>>>> >> generate-attn => v=0;%(200,300,1400,1800) >>>>>> >> detect-attn => 1400,1800 >>>>>> >> >>>>>> >> generate-callwaiting-sas => v=0;%(300,10000,440) >>>>>> >> detect-callwaiting-sas => 440 >>>>>> >> >>>>>> >> generate-callwaiting-cas => v=0;%(80,0,2750,2130) >>>>>> >> detect-callwaiting-cas => 2750,2130 >>>>>> >> >>>>>> >> detect-fail1 => 913.8 >>>>>> >> detect-fail2 => 1370.6 >>>>>> >> detect-fail3 => 776.7 >>>>>> >> >>>>>> >> LOG OF FIRST CALL (OK) >>>>>> >> ==================== >>>>>> >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:152 >>>>>> >> switch_core_standard_on_execute() OpenZAP/1:1/93400534 Execute >>>>>> >> bridge(openzap/2/1/3400534) >>>>>> >> 2009-01-25 10:35:58 [DEBUG] mod_openzap.c:340 tech_init() Set codec >>>>>> PCMU >>>>>> >> 20ms >>>>>> >> 2009-01-25 10:35:58 [DEBUG] mod_openzap.c:1017 >>>>>> channel_outgoing_channel() >>>>>> >> Connect outbound channel OpenZAP/2:1/3400534 >>>>>> >> 2009-01-25 10:35:58 [NOTICE] switch_channel.c:565 >>>>>> >> switch_channel_set_name() New Channel OpenZAP/2:1/3400534 >>>>>> >> [e5f12114-ea88-11dd-9f5c-290fb4a527a4] >>>>>> >> 2009-01-25 10:35:58 [DEBUG] mod_openzap.c:1026 >>>>>> channel_outgoing_channel() >>>>>> >> (OpenZAP/2:1/3400534) State Change CS_NEW -> CS_INIT >>>>>> >> 2009-01-25 10:35:58 [DEBUG] switch_core_session.c:807 >>>>>> >> switch_core_session_signal_state_change() Send signal >>>>>> OpenZAP/2:1/3400534 >>>>>> >> [BREAK] >>>>>> >> 2009-01-25 10:35:58 [DEBUG] ozmod_analog.c:52 >>>>>> analog_fxo_outgoing_call() >>>>>> >> Changing state on 2:1 from DOWN to DIALING >>>>>> >> 2009-01-25 10:35:58 [DEBUG] ozmod_analog.c:239 >>>>>> zap_analog_channel_run() >>>>>> >> ANALOG CHANNEL thread starting. >>>>>> >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:379 >>>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) Running State >>>>>> Change CS_INIT >>>>>> >> 2009-01-25 10:35:58 [DEBUG] ozmod_analog.c:410 >>>>>> zap_analog_channel_run() >>>>>> >> Executing state handler on 2:1 for DIALING >>>>>> >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:444 >>>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State INIT >>>>>> >> 2009-01-25 10:35:58 [DEBUG] mod_openzap.c:364 channel_on_init() >>>>>> >> (OpenZAP/2:1/3400534) State Change CS_INIT -> CS_ROUTING >>>>>> >> 2009-01-25 10:35:58 [DEBUG] switch_core_session.c:807 >>>>>> >> switch_core_session_signal_state_change() Send signal >>>>>> OpenZAP/2:1/3400534 >>>>>> >> [BREAK] >>>>>> >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:444 >>>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State INIT going to >>>>>> sleep >>>>>> >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:379 >>>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) Running State >>>>>> Change >>>>>> >> CS_ROUTING >>>>>> >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:447 >>>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State ROUTING >>>>>> >> 2009-01-25 10:35:58 [DEBUG] mod_openzap.c:387 channel_on_routing() >>>>>> >> OpenZAP/2:1/3400534 CHANNEL ROUTING >>>>>> >> 2009-01-25 10:35:58 [DEBUG] switch_ivr_originate.c:58 >>>>>> >> originate_on_routing() (OpenZAP/2:1/3400534) State Change >>>>>> CS_ROUTING -> >>>>>> >> CS_CONSUME_MEDIA >>>>>> >> 2009-01-25 10:35:58 [DEBUG] switch_core_session.c:807 >>>>>> >> switch_core_session_signal_state_change() Send signal >>>>>> OpenZAP/2:1/3400534 >>>>>> >> [BREAK] >>>>>> >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:447 >>>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State ROUTING going >>>>>> to sleep >>>>>> >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:379 >>>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) Running State >>>>>> Change >>>>>> >> CS_CONSUME_MEDIA >>>>>> >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:466 >>>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State CONSUME_MEDIA >>>>>> >> 2009-01-25 10:35:59 [DEBUG] ozmod_analog.c:615 >>>>>> zap_analog_channel_run() >>>>>> >> Detected tone DIAL on 2:1 >>>>>> >> 2009-01-25 10:35:59 [DEBUG] mod_openzap.c:1153 on_fxo_signal() got >>>>>> FXO sig >>>>>> >> 2:1 [TONE_DETECTED] >>>>>> >> 2009-01-25 10:35:59 [WARNING] mod_openzap.c:1196 on_fxo_signal() >>>>>> Unhandled >>>>>> >> type for channel 2:1 >>>>>> >> 2009-01-25 10:35:59 [DEBUG] zap_io.c:1179 >>>>>> zchan_activate_dtmf_buffer() >>>>>> >> Created DTMF Buffer! >>>>>> >> 2009-01-25 10:35:59 [DEBUG] zap_io.c:1715 handle_dtmf() 2:1 >>>>>> GENERATE DTMF >>>>>> >> [3400534] >>>>>> >> 2009-01-25 10:36:03 [DEBUG] ozmod_analog.c:302 >>>>>> zap_analog_channel_run() >>>>>> >> Changing state on 2:1 from DIALING to UP >>>>>> >> 2009-01-25 10:36:03 [DEBUG] ozmod_analog.c:410 >>>>>> zap_analog_channel_run() >>>>>> >> Executing state handler on 2:1 for UP >>>>>> >> 2009-01-25 10:36:03 [DEBUG] mod_openzap.c:1153 on_fxo_signal() got >>>>>> FXO sig >>>>>> >> 2:1 [UP] >>>>>> >> 2009-01-25 10:36:03 [DEBUG] switch_channel.c:1710 >>>>>> >> switch_channel_perform_mark_answered() Send signal >>>>>> OpenZAP/1:1/93400534 >>>>>> >> [BREAK] >>>>>> >> 2009-01-25 10:36:03 [NOTICE] mod_openzap.c:1180 on_fxo_signal() >>>>>> Channel >>>>>> >> [OpenZAP/2:1/3400534] has been answered >>>>>> >> 2009-01-25 10:36:03 [DEBUG] switch_channel.c:177 >>>>>> >> switch_channel_audio_sync() OpenZAP/2:1/3400534 receive message >>>>>> [AUDIO_SYNC] >>>>>> >> 2009-01-25 10:36:03 [DEBUG] switch_channel.c:1768 >>>>>> >> switch_channel_perform_answer() OpenZAP/1:1/93400534 receive >>>>>> message >>>>>> >> [ANSWER] >>>>>> >> 2009-01-25 10:36:03 [DEBUG] mod_openzap.c:813 >>>>>> >> channel_receive_message_fxs() Changing state on 1:1 from IDLE to UP >>>>>> >> 2009-01-25 10:36:03 [NOTICE] mod_openzap.c:814 >>>>>> >> channel_receive_message_fxs() Channel [OpenZAP/1:1/93400534] has >>>>>> been >>>>>> >> answered >>>>>> >> 2009-01-25 10:36:03 [DEBUG] switch_channel.c:177 >>>>>> >> switch_channel_audio_sync() OpenZAP/1:1/93400534 receive message >>>>>> >> [AUDIO_SYNC] >>>>>> >> 2009-01-25 10:36:03 [DEBUG] switch_core_session.c:511 >>>>>> >> switch_core_session_perform_receive_message() Send signal >>>>>> >> OpenZAP/1:1/93400534 [BREAK] >>>>>> >> 2009-01-25 10:36:03 [DEBUG] switch_ivr_originate.c:1627 >>>>>> >> switch_ivr_originate() Originate Resulted in Success: >>>>>> [OpenZAP/2:1/3400534] >>>>>> >> 2009-01-25 10:36:03 [DEBUG] switch_channel.c:177 >>>>>> >> switch_channel_audio_sync() OpenZAP/2:1/3400534 receive message >>>>>> [AUDIO_SYNC] >>>>>> >> 2009-01-25 10:36:03 [DEBUG] switch_channel.c:177 >>>>>> >> switch_channel_audio_sync() OpenZAP/1:1/93400534 receive message >>>>>> >> [AUDIO_SYNC] >>>>>> >> 2009-01-25 10:36:03 [DEBUG] switch_ivr_bridge.c:862 >>>>>> >> switch_ivr_multi_threaded_bridge() OpenZAP/2:1/3400534 receive >>>>>> message >>>>>> >> [BRIDGE] >>>>>> >> 2009-01-25 10:36:03 [DEBUG] switch_core_session.c:511 >>>>>> >> switch_core_session_perform_receive_message() Send signal >>>>>> >> OpenZAP/2:1/3400534 [BREAK] >>>>>> >> 2009-01-25 10:36:03 [DEBUG] switch_ivr_bridge.c:869 >>>>>> >> switch_ivr_multi_threaded_bridge() OpenZAP/1:1/93400534 receive >>>>>> message >>>>>> >> [BRIDGE] >>>>>> >> 2009-01-25 10:36:03 [DEBUG] switch_core_session.c:511 >>>>>> >> switch_core_session_perform_receive_message() Send signal >>>>>> >> OpenZAP/1:1/93400534 [BREAK] >>>>>> >> 2009-01-25 10:36:03 [DEBUG] switch_ivr_bridge.c:913 >>>>>> >> switch_ivr_multi_threaded_bridge() (OpenZAP/2:1/3400534) State >>>>>> Change >>>>>> >> CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA >>>>>> >> 2009-01-25 10:36:03 [DEBUG] switch_core_session.c:807 >>>>>> >> switch_core_session_signal_state_change() Send signal >>>>>> OpenZAP/2:1/3400534 >>>>>> >> [BREAK] >>>>>> >> 2009-01-25 10:36:03 [DEBUG] ozmod_analog.c:410 >>>>>> zap_analog_channel_run() >>>>>> >> Executing state handler on 1:1 for UP >>>>>> >> 2009-01-25 10:36:03 [DEBUG] mod_openzap.c:1212 on_fxs_signal() got >>>>>> FXS sig >>>>>> >> [UP] >>>>>> >> 2009-01-25 10:36:03 [DEBUG] switch_core_state_machine.c:466 >>>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State CONSUME_MEDIA >>>>>> going to >>>>>> >> sleep >>>>>> >> 2009-01-25 10:36:03 [DEBUG] switch_core_state_machine.c:379 >>>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) Running State >>>>>> Change >>>>>> >> CS_EXCHANGE_MEDIA >>>>>> >> 2009-01-25 10:36:03 [DEBUG] switch_core_state_machine.c:457 >>>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State >>>>>> EXCHANGE_MEDIA >>>>>> >> 2009-01-25 10:36:03 [DEBUG] mod_openzap.c:511 >>>>>> channel_on_exchange_media() >>>>>> >> CHANNEL EXCHANGE_MEDIA >>>>>> >> 2009-01-25 10:36:16 [DEBUG] ozmod_analog.c:744 process_event() >>>>>> EVENT >>>>>> >> [ONHOOK][1:1] STATE [UP] >>>>>> >> 2009-01-25 10:36:16 [DEBUG] ozmod_analog.c:780 process_event() >>>>>> Changing >>>>>> >> state on 1:1 from UP to DOWN >>>>>> >> 2009-01-25 10:36:16 [DEBUG] ozmod_analog.c:410 >>>>>> zap_analog_channel_run() >>>>>> >> Executing state handler on 1:1 for DOWN >>>>>> >> 2009-01-25 10:36:16 [DEBUG] mod_openzap.c:1212 on_fxs_signal() got >>>>>> FXS sig >>>>>> >> [STOP] >>>>>> >> 2009-01-25 10:36:16 [NOTICE] mod_openzap.c:1300 on_fxs_signal() >>>>>> Hangup >>>>>> >> OpenZAP/1:1/93400534 [CS_EXECUTE] [NORMAL_CLEARING] >>>>>> >> 2009-01-25 10:36:16 [DEBUG] switch_channel.c:1494 >>>>>> >> switch_channel_perform_hangup() Send signal OpenZAP/1:1/93400534 >>>>>> [KILL] >>>>>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_session.c:807 >>>>>> >> switch_core_session_signal_state_change() Send signal >>>>>> OpenZAP/1:1/93400534 >>>>>> >> [BREAK] >>>>>> >> 2009-01-25 10:36:16 [DEBUG] zap_io.c:1125 zap_channel_done() >>>>>> channel done >>>>>> >> 1:1 >>>>>> >> 2009-01-25 10:36:16 [DEBUG] ozmod_analog.c:726 >>>>>> zap_analog_channel_run() >>>>>> >> ANALOG CHANNEL 1:1 thread ended. >>>>>> >> 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:360 >>>>>> audio_bridge_thread() >>>>>> >> OpenZAP/1:1/93400534 ending bridge by request from read function >>>>>> >> 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:435 >>>>>> audio_bridge_thread() >>>>>> >> BRIDGE THREAD DONE [OpenZAP/1:1/93400534] >>>>>> >> 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:439 >>>>>> audio_bridge_thread() >>>>>> >> Send signal OpenZAP/2:1/3400534 [BREAK] >>>>>> >> 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:354 >>>>>> audio_bridge_thread() >>>>>> >> OpenZAP/1:1/93400534 ending bridge by request from write function >>>>>> >> 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:409 >>>>>> audio_bridge_thread() >>>>>> >> OpenZAP/2:1/3400534 receive message [UNBRIDGE] >>>>>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_session.c:511 >>>>>> >> switch_core_session_perform_receive_message() Send signal >>>>>> >> OpenZAP/2:1/3400534 [BREAK] >>>>>> >> 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:435 >>>>>> audio_bridge_thread() >>>>>> >> BRIDGE THREAD DONE [OpenZAP/2:1/3400534] >>>>>> >> 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:439 >>>>>> audio_bridge_thread() >>>>>> >> Send signal OpenZAP/1:1/93400534 [BREAK] >>>>>> >> 2009-01-25 10:36:16 [NOTICE] switch_ivr_bridge.c:470 >>>>>> >> audio_bridge_on_exchange_media() Hangup OpenZAP/2:1/3400534 >>>>>> >> [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] >>>>>> >> 2009-01-25 10:36:16 [DEBUG] switch_channel.c:1494 >>>>>> >> switch_channel_perform_hangup() Send signal OpenZAP/2:1/3400534 >>>>>> [KILL] >>>>>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_session.c:807 >>>>>> >> switch_core_session_signal_state_change() Send signal >>>>>> OpenZAP/2:1/3400534 >>>>>> >> [BREAK] >>>>>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:457 >>>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State >>>>>> EXCHANGE_MEDIA going >>>>>> >> to sleep >>>>>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:379 >>>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) Running State >>>>>> Change >>>>>> >> CS_HANGUP >>>>>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:410 >>>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State HANGUP >>>>>> >> 2009-01-25 10:36:16 [DEBUG] mod_openzap.c:429 channel_on_hangup() >>>>>> Changing >>>>>> >> state on 2:1 from UP to HANGUP >>>>>> >> 2009-01-25 10:36:16 [DEBUG] mod_openzap.c:472 channel_on_hangup() >>>>>> >> OpenZAP/2:1/3400534 CHANNEL HANGUP >>>>>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:46 >>>>>> >> switch_core_standard_on_hangup() OpenZAP/2:1/3400534 Standard >>>>>> HANGUP, cause: >>>>>> >> NORMAL_CLEARING >>>>>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:410 >>>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State HANGUP going >>>>>> to sleep >>>>>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_session.c:939 >>>>>> >> switch_core_session_thread() Session 2 (OpenZAP/2:1/3400534) >>>>>> Locked, Waiting >>>>>> >> on external entities >>>>>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:454 >>>>>> >> switch_core_session_run() (OpenZAP/1:1/93400534) State EXECUTE >>>>>> going to >>>>>> >> sleep >>>>>> >> 2009-01-25 10:36:16 [NOTICE] switch_core_session.c:957 >>>>>> >> switch_core_session_thread() Session 2 (OpenZAP/2:1/3400534) Ended >>>>>> >> 2009-01-25 10:36:16 [NOTICE] switch_core_session.c:959 >>>>>> >> switch_core_session_thread() Close Channel OpenZAP/2:1/3400534 >>>>>> [CS_HANGUP] >>>>>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:379 >>>>>> >> switch_core_session_run() (OpenZAP/1:1/93400534) Running State >>>>>> Change >>>>>> >> CS_HANGUP >>>>>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:410 >>>>>> >> switch_core_session_run() (OpenZAP/1:1/93400534) State HANGUP >>>>>> >> 2009-01-25 10:36:16 [DEBUG] mod_openzap.c:472 channel_on_hangup() >>>>>> >> OpenZAP/1:1/93400534 CHANNEL HANGUP >>>>>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:46 >>>>>> >> switch_core_standard_on_hangup() OpenZAP/1:1/93400534 Standard >>>>>> HANGUP, >>>>>> >> cause: NORMAL_CLEARING >>>>>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:410 >>>>>> >> switch_core_session_run() (OpenZAP/1:1/93400534) State HANGUP going >>>>>> to sleep >>>>>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_session.c:939 >>>>>> >> switch_core_session_thread() Session 1 (OpenZAP/1:1/93400534) >>>>>> Locked, >>>>>> >> Waiting on external entities >>>>>> >> 2009-01-25 10:36:16 [NOTICE] switch_core_session.c:957 >>>>>> >> switch_core_session_thread() Session 1 (OpenZAP/1:1/93400534) Ended >>>>>> >> 2009-01-25 10:36:16 [NOTICE] switch_core_session.c:959 >>>>>> >> switch_core_session_thread() Close Channel OpenZAP/1:1/93400534 >>>>>> [CS_HANGUP] >>>>>> >> 2009-01-25 10:36:16 [DEBUG] ozmod_analog.c:410 >>>>>> zap_analog_channel_run() >>>>>> >> Executing state handler on 2:1 for HANGUP >>>>>> >> 2009-01-25 10:36:17 [DEBUG] ozmod_analog.c:351 >>>>>> zap_analog_channel_run() >>>>>> >> Changing state on 2:1 from HANGUP to DOWN >>>>>> >> 2009-01-25 10:36:17 [DEBUG] ozmod_analog.c:410 >>>>>> zap_analog_channel_run() >>>>>> >> Executing state handler on 2:1 for DOWN >>>>>> >> 2009-01-25 10:36:17 [DEBUG] mod_openzap.c:1153 on_fxo_signal() got >>>>>> FXO sig >>>>>> >> 2:1 [STOP] >>>>>> >> 2009-01-25 10:36:17 [DEBUG] zap_io.c:1125 zap_channel_done() >>>>>> channel done >>>>>> >> 2:1 >>>>>> >> 2009-01-25 10:36:17 [DEBUG] ozmod_analog.c:726 >>>>>> zap_analog_channel_run() >>>>>> >> ANALOG CHANNEL 2:1 thread ended. >>>>>> >> >>>>>> >> LOG OF FAILED CALLS >>>>>> >> ================== >>>>>> >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:152 >>>>>> >> switch_core_standard_on_execute() OpenZAP/1:1/93400534 Execute >>>>>> >> bridge(openzap/2/1/3400534) >>>>>> >> 2009-01-25 10:36:55 [DEBUG] mod_openzap.c:340 tech_init() Set codec >>>>>> PCMU >>>>>> >> 20ms >>>>>> >> 2009-01-25 10:36:55 [DEBUG] mod_openzap.c:1017 >>>>>> channel_outgoing_channel() >>>>>> >> Connect outbound channel OpenZAP/2:1/3400534 >>>>>> >> 2009-01-25 10:36:55 [NOTICE] switch_channel.c:565 >>>>>> >> switch_channel_set_name() New Channel OpenZAP/2:1/3400534 >>>>>> >> [079f5420-ea89-11dd-9f5c-290fb4a527a4] >>>>>> >> 2009-01-25 10:36:55 [DEBUG] mod_openzap.c:1026 >>>>>> channel_outgoing_channel() >>>>>> >> (OpenZAP/2:1/3400534) State Change CS_NEW -> CS_INIT >>>>>> >> 2009-01-25 10:36:55 [DEBUG] switch_core_session.c:807 >>>>>> >> switch_core_session_signal_state_change() Send signal >>>>>> OpenZAP/2:1/3400534 >>>>>> >> [BREAK] >>>>>> >> 2009-01-25 10:36:55 [DEBUG] ozmod_analog.c:52 >>>>>> analog_fxo_outgoing_call() >>>>>> >> Changing state on 2:1 from DOWN to DIALING >>>>>> >> 2009-01-25 10:36:55 [DEBUG] ozmod_analog.c:239 >>>>>> zap_analog_channel_run() >>>>>> >> ANALOG CHANNEL thread starting. >>>>>> >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:379 >>>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) Running State >>>>>> Change CS_INIT >>>>>> >> 2009-01-25 10:36:55 [DEBUG] ozmod_analog.c:410 >>>>>> zap_analog_channel_run() >>>>>> >> Executing state handler on 2:1 for DIALING >>>>>> >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:444 >>>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State INIT >>>>>> >> 2009-01-25 10:36:55 [DEBUG] mod_openzap.c:364 channel_on_init() >>>>>> >> (OpenZAP/2:1/3400534) State Change CS_INIT -> CS_ROUTING >>>>>> >> 2009-01-25 10:36:55 [DEBUG] switch_core_session.c:807 >>>>>> >> switch_core_session_signal_state_change() Send signal >>>>>> OpenZAP/2:1/3400534 >>>>>> >> [BREAK] >>>>>> >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:444 >>>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State INIT going to >>>>>> sleep >>>>>> >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:379 >>>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) Running State >>>>>> Change >>>>>> >> CS_ROUTING >>>>>> >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:447 >>>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State ROUTING >>>>>> >> 2009-01-25 10:36:55 [DEBUG] mod_openzap.c:387 channel_on_routing() >>>>>> >> OpenZAP/2:1/3400534 CHANNEL ROUTING >>>>>> >> 2009-01-25 10:36:55 [DEBUG] switch_ivr_originate.c:58 >>>>>> >> originate_on_routing() (OpenZAP/2:1/3400534) State Change >>>>>> CS_ROUTING -> >>>>>> >> CS_CONSUME_MEDIA >>>>>> >> 2009-01-25 10:36:55 [DEBUG] switch_core_session.c:807 >>>>>> >> switch_core_session_signal_state_change() Send signal >>>>>> OpenZAP/2:1/3400534 >>>>>> >> [BREAK] >>>>>> >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:447 >>>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State ROUTING going >>>>>> to sleep >>>>>> >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:379 >>>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) Running State >>>>>> Change >>>>>> >> CS_CONSUME_MEDIA >>>>>> >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:466 >>>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State CONSUME_MEDIA >>>>>> >> 2009-01-25 10:37:08 [DEBUG] ozmod_analog.c:744 process_event() >>>>>> EVENT >>>>>> >> [ONHOOK][1:1] STATE [IDLE] >>>>>> >> 2009-01-25 10:37:08 [DEBUG] ozmod_analog.c:780 process_event() >>>>>> Changing >>>>>> >> state on 1:1 from IDLE to DOWN >>>>>> >> 2009-01-25 10:37:08 [DEBUG] ozmod_analog.c:410 >>>>>> zap_analog_channel_run() >>>>>> >> Executing state handler on 1:1 for DOWN >>>>>> >> 2009-01-25 10:37:08 [DEBUG] mod_openzap.c:1212 on_fxs_signal() got >>>>>> FXS sig >>>>>> >> [STOP] >>>>>> >> 2009-01-25 10:37:08 [NOTICE] mod_openzap.c:1300 on_fxs_signal() >>>>>> Hangup >>>>>> >> OpenZAP/1:1/93400534 [CS_EXECUTE] [NORMAL_CLEARING] >>>>>> >> 2009-01-25 10:37:08 [DEBUG] switch_channel.c:1494 >>>>>> >> switch_channel_perform_hangup() Send signal OpenZAP/1:1/93400534 >>>>>> [KILL] >>>>>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_session.c:807 >>>>>> >> switch_core_session_signal_state_change() Send signal >>>>>> OpenZAP/1:1/93400534 >>>>>> >> [BREAK] >>>>>> >> 2009-01-25 10:37:08 [DEBUG] zap_io.c:1125 zap_channel_done() >>>>>> channel done >>>>>> >> 1:1 >>>>>> >> 2009-01-25 10:37:08 [DEBUG] ozmod_analog.c:726 >>>>>> zap_analog_channel_run() >>>>>> >> ANALOG CHANNEL 1:1 thread ended. >>>>>> >> 2009-01-25 10:37:08 [NOTICE] switch_ivr_originate.c:1566 >>>>>> >> switch_ivr_originate() Hangup OpenZAP/2:1/3400534 >>>>>> [CS_CONSUME_MEDIA] >>>>>> >> [ORIGINATOR_CANCEL] >>>>>> >> 2009-01-25 10:37:08 [DEBUG] switch_channel.c:1494 >>>>>> >> switch_channel_perform_hangup() Send signal OpenZAP/2:1/3400534 >>>>>> [KILL] >>>>>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_session.c:807 >>>>>> >> switch_core_session_signal_state_change() Send signal >>>>>> OpenZAP/2:1/3400534 >>>>>> >> [BREAK] >>>>>> >> 2009-01-25 10:37:08 [DEBUG] switch_ivr_originate.c:1691 >>>>>> >> switch_ivr_originate() Originate Cancelled by originator >>>>>> termination Cause: >>>>>> >> 487 [ORIGINATOR_CANCEL] >>>>>> >> 2009-01-25 10:37:08 [INFO] mod_dptools.c:1909 >>>>>> audio_bridge_function() >>>>>> >> Originate Failed. Cause: ORIGINATOR_CANCEL >>>>>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:454 >>>>>> >> switch_core_session_run() (OpenZAP/1:1/93400534) State EXECUTE >>>>>> going to >>>>>> >> sleep >>>>>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:379 >>>>>> >> switch_core_session_run() (OpenZAP/1:1/93400534) Running State >>>>>> Change >>>>>> >> CS_HANGUP >>>>>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:410 >>>>>> >> switch_core_session_run() (OpenZAP/1:1/93400534) State HANGUP >>>>>> >> 2009-01-25 10:37:08 [DEBUG] mod_openzap.c:472 channel_on_hangup() >>>>>> >> OpenZAP/1:1/93400534 CHANNEL HANGUP >>>>>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:46 >>>>>> >> switch_core_standard_on_hangup() OpenZAP/1:1/93400534 Standard >>>>>> HANGUP, >>>>>> >> cause: NORMAL_CLEARING >>>>>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:410 >>>>>> >> switch_core_session_run() (OpenZAP/1:1/93400534) State HANGUP going >>>>>> to sleep >>>>>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_session.c:939 >>>>>> >> switch_core_session_thread() Session 3 (OpenZAP/1:1/93400534) >>>>>> Locked, >>>>>> >> Waiting on external entities >>>>>> >> 2009-01-25 10:37:08 [NOTICE] switch_core_session.c:957 >>>>>> >> switch_core_session_thread() Session 3 (OpenZAP/1:1/93400534) Ended >>>>>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:466 >>>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State CONSUME_MEDIA >>>>>> going to >>>>>> >> sleep >>>>>> >> 2009-01-25 10:37:08 [NOTICE] switch_core_session.c:959 >>>>>> >> switch_core_session_thread() Close Channel OpenZAP/1:1/93400534 >>>>>> [CS_HANGUP] >>>>>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:379 >>>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) Running State >>>>>> Change >>>>>> >> CS_HANGUP >>>>>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:410 >>>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State HANGUP >>>>>> >> 2009-01-25 10:37:08 [DEBUG] mod_openzap.c:429 channel_on_hangup() >>>>>> Changing >>>>>> >> state on 2:1 from DIALING to HANGUP >>>>>> >> 2009-01-25 10:37:08 [DEBUG] mod_openzap.c:472 channel_on_hangup() >>>>>> >> OpenZAP/2:1/3400534 CHANNEL HANGUP >>>>>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:46 >>>>>> >> switch_core_standard_on_hangup() OpenZAP/2:1/3400534 Standard >>>>>> HANGUP, cause: >>>>>> >> ORIGINATOR_CANCEL >>>>>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:410 >>>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State HANGUP going >>>>>> to sleep >>>>>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_session.c:939 >>>>>> >> switch_core_session_thread() Session 4 (OpenZAP/2:1/3400534) >>>>>> Locked, Waiting >>>>>> >> on external entities >>>>>> >> 2009-01-25 10:37:08 [NOTICE] switch_core_session.c:957 >>>>>> >> switch_core_session_thread() Session 4 (OpenZAP/2:1/3400534) Ended >>>>>> >> 2009-01-25 10:37:08 [NOTICE] switch_core_session.c:959 >>>>>> >> switch_core_session_thread() Close Channel OpenZAP/2:1/3400534 >>>>>> [CS_HANGUP] >>>>>> >> 2009-01-25 10:37:08 [DEBUG] ozmod_analog.c:410 >>>>>> zap_analog_channel_run() >>>>>> >> Executing state handler on 2:1 for HANGUP >>>>>> >> 2009-01-25 10:37:09 [DEBUG] ozmod_analog.c:351 >>>>>> zap_analog_channel_run() >>>>>> >> Changing state on 2:1 from HANGUP to DOWN >>>>>> >> 2009-01-25 10:37:09 [DEBUG] ozmod_analog.c:410 >>>>>> zap_analog_channel_run() >>>>>> >> Executing state handler on 2:1 for DOWN >>>>>> >> 2009-01-25 10:37:09 [DEBUG] mod_openzap.c:1153 on_fxo_signal() got >>>>>> FXO sig >>>>>> >> 2:1 [STOP] >>>>>> >> 2009-01-25 10:37:09 [DEBUG] zap_io.c:1125 zap_channel_done() >>>>>> channel done >>>>>> >> 2:1 >>>>>> >> 2009-01-25 10:37:09 [DEBUG] ozmod_analog.c:726 >>>>>> zap_analog_channel_run() >>>>>> >> ANALOG CHANNEL 2:1 thread ended. >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> >> _______________________________________________ >>>>>> >> Freeswitch-users mailing list >>>>>> >> Freeswitch-users at lists.freeswitch.org >>>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> >> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> >> http://www.freeswitch.org >>>>>> >> >>>>>> >> >>>>>> >> _______________________________________________ >>>>>> >> Freeswitch-users mailing list >>>>>> >> Freeswitch-users at lists.freeswitch.org >>>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> >> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> >> http://www.freeswitch.org >>>>>> >> >>>>>> > >>>>>> > >>>>>> > _______________________________________________ >>>>>> > Freeswitch-users mailing list >>>>>> > Freeswitch-users at lists.freeswitch.org >>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> > UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> > http://www.freeswitch.org >>>>>> > >>>>>> > >>>>>> >>>>>> _______________________________________________ >>>>>> Freeswitch-users mailing list >>>>>> Freeswitch-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>> >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> -- >>>> Anthony Minessale II >>>> >>>> FreeSWITCH http://www.freeswitch.org/ >>>> ClueCon http://www.cluecon.com/ >>>> >>>> AIM: anthm >>>> MSN:anthony_minessale at hotmail.com >>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>>> IRC: irc.freenode.net #freeswitch >>>> >>>> FreeSWITCH Developer Conference >>>> sip:888 at conference.freeswitch.org >>>> iax:guest at conference.freeswitch.org/888 >>>> googletalk:conf+888 at conference.freeswitch.org >>>> pstn:213-799-1400 >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090201/e94b380d/attachment-0001.html From brian at freeswitch.org Sun Feb 1 10:59:04 2009 From: brian at freeswitch.org (Brian West) Date: Sun, 1 Feb 2009 12:59:04 -0600 Subject: [Freeswitch-users] Cisco 7975G and XML options for G722 In-Reply-To: <49860C2B.7020801@ieee.org> References: <0C8E2714-6D96-49DB-93CD-0D377E4DAE96@freeswitch.org> <3885f4fe0901312115k1309573lb160ecbc83f8d57@mail.gmail.com> <897947BF-9886-47BB-B8BB-8DE8849437F0@freeswitch.org> <49860C2B.7020801@ieee.org> Message-ID: <1D9807B8-61ED-41B8-9AC3-C6C6BD306BE5@freeswitch.org> Why yes it does ;) /b On Feb 1, 2009, at 2:55 PM, Gabriel Kuri wrote: > 722 From pmhshz at gmail.com Sun Feb 1 21:26:56 2009 From: pmhshz at gmail.com (shehzad p) Date: Sun, 1 Feb 2009 21:26:56 -0800 (PST) Subject: [Freeswitch-users] Freeswitch freezes on increasing call traffic In-Reply-To: <30BF0913-A90C-4472-9A70-6CD6AEE2AD08@freeswitch.org> References: <21701744.post@talk.nabble.com> <191c3a030901280601t3e185ec2m76295110e4888837@mail.gmail.com> <21729863.post@talk.nabble.com> <191c3a030901290831y2ff329b3maee6c1aafa3c68e2@mail.gmail.com> <21745312.post@talk.nabble.com> <21749375.post@talk.nabble.com> <191c3a030901300818i5d85a7aawa0c53b4e54892f40@mail.gmail.com> <21761523.post@talk.nabble.com> <191c3a030901310850s553f8c11oc4e559a710def6d2@mail.gmail.com> <21773300.post@talk.nabble.com> <21775256.post@talk.nabble.com> <30BF0913-A90C-4472-9A70-6CD6AEE2AD08@freeswitch.org> Message-ID: <21784344.post@talk.nabble.com> Hi Brian, I am first looking to modify the script, as there are several things to modify (thanks to Anthony) then paralally I will test 1.0.3 RC1 also and any new thing come out then I will post update here... Thanks, msp Brian West-3 wrote: > > I highly recommend you upgrade to 1.03RC1 or SVN Trunk. Chances are > we have already fixed these issues. > > /b > > On Feb 1, 2009, at 7:35 AM, shehzad p wrote: > >> >> Hi anthony, >> >> There are so many crash occured, on FS 1.0.1 Server, all of them >> were same >> (means BT was same) >> but this last one was looking totally different so I feel that it >> should be >> posted... >> http://www.nabble.com/file/p21775256/bt_full_new.txt bt_full_new.txt > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/Freeswitch-freezes-on-increasing-call-traffic-tp21701744p21784344.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From brian at freeswitch.org Sun Feb 1 21:37:59 2009 From: brian at freeswitch.org (Brian West) Date: Sun, 1 Feb 2009 23:37:59 -0600 Subject: [Freeswitch-users] Freeswitch freezes on increasing call traffic In-Reply-To: <21784344.post@talk.nabble.com> References: <21701744.post@talk.nabble.com> <191c3a030901280601t3e185ec2m76295110e4888837@mail.gmail.com> <21729863.post@talk.nabble.com> <191c3a030901290831y2ff329b3maee6c1aafa3c68e2@mail.gmail.com> <21745312.post@talk.nabble.com> <21749375.post@talk.nabble.com> <191c3a030901300818i5d85a7aawa0c53b4e54892f40@mail.gmail.com> <21761523.post@talk.nabble.com> <191c3a030901310850s553f8c11oc4e559a710def6d2@mail.gmail.com> <21773300.post@talk.nabble.com> <21775256.post@talk.nabble.com> <30BF0913-A90C-4472-9A70-6CD6AEE2AD08@freeswitch.org> <21784344.post@talk.nabble.com> Message-ID: <54089955-A273-48CD-8657-D2A1CD4E0B89@freeswitch.org> The RC1 tarball has all the SVN dirs so you can "make current" on it. /b On Feb 1, 2009, at 11:26 PM, shehzad p wrote: > > Hi Brian, > > I am first looking to modify the script, as there are several things > to > modify (thanks to Anthony) then paralally I will test 1.0.3 RC1 also > and any > new thing come out then I will post update here... > > Thanks, > msp From kawarod at laposte.net Sun Feb 1 23:04:17 2009 From: kawarod at laposte.net (rod) Date: Mon, 02 Feb 2009 11:04:17 +0400 Subject: [Freeswitch-users] Strange Performance when using as SBC In-Reply-To: <191c3a030901300556s534823a1if4b79b4279962ed1@mail.gmail.com> References: <4982EAD5.1010200@laposte.net> <191c3a030901300556s534823a1if4b79b4279962ed1@mail.gmail.com> Message-ID: <49869AF1.1070903@laposte.net> Thanks Anthony, the setup is like this: sipp server ---- FS 1 ---- FS2 FS1 is the AMD CPU that has only one extension in dialplan that bridges 9999 to FS2. 9999 is the first extension in FS2 dialplan that plays moh, FS2 has no CPU pbm. FS1 is maxing out at 60 bridged calls without your option -hp. Using -hp, I'm now able to bridge 200 concurrent calls (a great improvement) and the system is still reactive. CPU load is high but not 100% and as the system responds well, I think that doesn't matter. The 2GB of memory are completely consumed (top command shows 700MB for FS process). I understand that FS1 server is not the best hardware platform, and I'm waiting for new 4 cores server for testing. I will update those numbers when testing with the new hardware. regards, rod. Anthony Minessale wrote: > Which of the 2 machines has the load issue? You said it was one box > calling the other. > > You have 2 major things against you, single CPU and AMD, but you > should at least be able to get in the vicinity of 800-1000 calls on a > box like that. > > Are you calling the default 9999? It's not really an appropriate > extension for load testing. > On the terminating box you should set up a manual extension that is > the first one in the dial plan > to play a wav file from preferably a ram disk or /tmp > > If you do plan on using this in production accept nothing less than a > multi-core intel machine with at least 4 cores, the more cores the > better because that parallel processing is where FS gets it's atvantage. > > > > On Fri, Jan 30, 2009 at 5:56 AM, rod > wrote: > > Dear list, > > I've been playing with freeswitch for some time (2 months) and the > fact > is that I'm very pleased with the functionnalities of this software. > > I'd like to use FS as a SBC handling media and I'm doing some > tests with > sipp to load the machine but I'm unable to bridge more than 60 calls > without seeing the CPU being loaded at 100%. I'm sure something is > going > wrong with my setup but I'm unable to see what. > > The test machine has the following specs: > Athlon XP 3500+ with 2GB of memory (I know this is not a high end > machine :p) > > Freeswitch:/opt/freeswitch/log# cat /proc/cpuinfo > processor : 0 > vendor_id : AuthenticAMD > cpu family : 15 > model : 95 > model name : AMD Athlon(tm) 64 Processor 3500+ > stepping : 2 > cpu MHz : 2199.973 > cache size : 512 KB > fpu : yes > fpu_exception : yes > cpuid level : 1 > wp : yes > flags : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge > mca cmov pat pse36 clflush mmx fxsr sse sse2 syscall nx mmxext > fxsr_opt > rdtscp lm 3dnowext 3dnow up rep_good pni cx16 lahf_lm svm extapic > cr8_legacy > bogomips : 4402.97 > TLB size : 1024 4K pages > clflush size : 64 > cache_alignment : 64 > address sizes : 40 bits physical, 48 bits virtual > power management: ts fid vid ttp tm stc > > I installed FS on a fresh debian 64: > Linux Freeswitch 2.6.26-1-amd64 #1 SMP Sat Jan 10 17:57:00 UTC 2009 > x86_64 GNU/Linux > > I set the ulimit parameters like those on the website: > freeswitch at internal> ... > Freeswitch:/opt/free-svn/bin# ulimit -a > core file size (blocks, -c) unlimited > data seg size (kbytes, -d) unlimited > scheduling priority (-e) 0 > file size (blocks, -f) unlimited > pending signals (-i) unlimited > max locked memory (kbytes, -l) unlimited > max memory size (kbytes, -m) unlimited > open files (-n) 999999 > pipe size (512 bytes, -p) 8 > POSIX message queues (bytes, -q) unlimited > real-time priority (-r) 0 > stack size (kbytes, -s) 244 > cpu time (seconds, -t) unlimited > max user processes (-u) unlimited > virtual memory (kbytes, -v) unlimited > file locks (-x) unlimited > > > My network setup is the following: > > SIPP machine (10.10.10.1/24)----------------vlan > 55 > ----------(10.10.10.254/24 ) FS > (10.10.20.254/24)-------------- > vlan56 > -------------------(10.10.20.100/24 ) > OTHER STOCK FS > > > I launched sipp with: > sipp -sn uac_pcap -s 9999 -r 10 -l 80 -d 60000 -mi 10.10.10.1 -i > 10.10.10.1 -mp 25000 10.10.10.254:5060 > > The dialplan on FS is very simple: > > > > > > > > > data="sofia/external/9999 at 10.10.20.100 "/> > > > > > > > FreeSWITCH Version 1.0.trunk (11560M) Started. > Crash Protection [Disabled] > Max Sessions[1000] > Session Rate[100] > SQL [Enabled] > > > The test is very simple: sipp dial 9999 that matches in my FS dialplan > and this is bridged to an other FS machine playing music on hold. > When I launch "top" I see after 30 to 40 s that FS consumes all > the CPU > ressources (with a mean of 50-60 % before), with 80 calls. > When I set 70 calls, I have to wait 70-80 s before seeing the same > issue. > > Presence is set to false on the 2 profile. > > I have the same issue with FS 1.0.2 that' s why I tried FS 11560. > > When I use the FS machine as a router to test the packet per second > performance, I'm reaching 100Mbps with 8000pps in each direction (from > vlan 55 to vlan56) with less than 12% CPU. So that I don't think > there's > an issue with the network. > > Here is an "mpstat -P ALL 1" to show you what's happening suddenly > with > 70 bridge calls: > 12:31:26 CPU %user %nice %sys %iowait %irq %soft > %steal %idle intr/s > 12:31:27 all 3,00 0,00 3,00 0,00 1,00 4,00 > 0,00 89,00 6241,00 > 12:31:27 0 3,00 0,00 3,00 0,00 1,00 4,00 > 0,00 89,00 6241,00 > > 12:31:27 CPU %user %nice %sys %iowait %irq %soft > %steal %idle intr/s > 12:31:28 all 14,14 0,00 56,57 0,00 2,02 5,05 > 0,00 22,22 6035,35 > 12:31:28 0 14,14 0,00 56,57 0,00 2,02 5,05 > 0,00 22,22 6035,35 > > 12:31:28 CPU %user %nice %sys %iowait %irq %soft > %steal %idle intr/s > 12:31:29 all 24,75 0,00 67,33 0,00 0,99 6,93 > 0,00 0,00 5483,17 > 12:31:29 0 24,75 0,00 67,33 0,00 0,99 6,93 > 0,00 0,00 5483,17 > > > The CPU is going from 89% idle to 0% in less than 2 seconds. > > I know that I don't have to expect too much from this kind of > hardware, > but it seems strange that the CPU power vanished so suddenly. > > Thanks a lot for the guys that have read this long mail :p > > kind regards, > rod > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From krice at freeswitch.org Sun Feb 1 23:17:50 2009 From: krice at freeswitch.org (Ken Rice) Date: Mon, 2 Feb 2009 01:17:50 -0600 Subject: [Freeswitch-users] Strange Performance when using as SBC In-Reply-To: <49869AF1.1070903@laposte.net> References: <4982EAD5.1010200@laposte.net> <191c3a030901300556s534823a1if4b79b4279962ed1@mail.gmail.com> <49869AF1.1070903@laposte.net> Message-ID: <94b568920902012317h8411fccg216435380f62d649@mail.gmail.com> Dont forget there are several things you can do to increase performance... 1) where possible use bypass media or media proxy modes 2) mount freeswitch/db as a ram drive (if you are using voicemail with the internal FS DBs you'll need a way to make this persistant across reboots) 3) see the wiki for setting reasonable ulimits 4) (this is my oppinion others may vary) dont use mod_cdr_csv 5) turn off (or reduce logging) in switch.conf.xml all of these thing can greatly improve performance. On Mon, Feb 2, 2009 at 1:04 AM, rod wrote: > Thanks Anthony, > > the setup is like this: > > sipp server ---- FS 1 ---- FS2 > > FS1 is the AMD CPU that has only one extension in dialplan that bridges > 9999 to FS2. 9999 is the first extension in FS2 dialplan that plays moh, > FS2 has no CPU pbm. > > FS1 is maxing out at 60 bridged calls without your option -hp. > > Using -hp, I'm now able to bridge 200 concurrent calls (a great > improvement) and the system is still reactive. CPU load is high but not > 100% and as the system responds well, I think that doesn't matter. The > 2GB of memory are completely consumed (top command shows 700MB for FS > process). > > I understand that FS1 server is not the best hardware platform, and I'm > waiting for new 4 cores server for testing. > I will update those numbers when testing with the new hardware. > > regards, > rod. > > Anthony Minessale wrote: > > Which of the 2 machines has the load issue? You said it was one box > > calling the other. > > > > You have 2 major things against you, single CPU and AMD, but you > > should at least be able to get in the vicinity of 800-1000 calls on a > > box like that. > > > > Are you calling the default 9999? It's not really an appropriate > > extension for load testing. > > On the terminating box you should set up a manual extension that is > > the first one in the dial plan > > to play a wav file from preferably a ram disk or /tmp > > > > If you do plan on using this in production accept nothing less than a > > multi-core intel machine with at least 4 cores, the more cores the > > better because that parallel processing is where FS gets it's atvantage. > > > > > > > > On Fri, Jan 30, 2009 at 5:56 AM, rod > > wrote: > > > > Dear list, > > > > I've been playing with freeswitch for some time (2 months) and the > > fact > > is that I'm very pleased with the functionnalities of this software. > > > > I'd like to use FS as a SBC handling media and I'm doing some > > tests with > > sipp to load the machine but I'm unable to bridge more than 60 calls > > without seeing the CPU being loaded at 100%. I'm sure something is > > going > > wrong with my setup but I'm unable to see what. > > > > The test machine has the following specs: > > Athlon XP 3500+ with 2GB of memory (I know this is not a high end > > machine :p) > > > > Freeswitch:/opt/freeswitch/log# cat /proc/cpuinfo > > processor : 0 > > vendor_id : AuthenticAMD > > cpu family : 15 > > model : 95 > > model name : AMD Athlon(tm) 64 Processor 3500+ > > stepping : 2 > > cpu MHz : 2199.973 > > cache size : 512 KB > > fpu : yes > > fpu_exception : yes > > cpuid level : 1 > > wp : yes > > flags : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr > pge > > mca cmov pat pse36 clflush mmx fxsr sse sse2 syscall nx mmxext > > fxsr_opt > > rdtscp lm 3dnowext 3dnow up rep_good pni cx16 lahf_lm svm extapic > > cr8_legacy > > bogomips : 4402.97 > > TLB size : 1024 4K pages > > clflush size : 64 > > cache_alignment : 64 > > address sizes : 40 bits physical, 48 bits virtual > > power management: ts fid vid ttp tm stc > > > > I installed FS on a fresh debian 64: > > Linux Freeswitch 2.6.26-1-amd64 #1 SMP Sat Jan 10 17:57:00 UTC 2009 > > x86_64 GNU/Linux > > > > I set the ulimit parameters like those on the website: > > freeswitch at internal> ... > > Freeswitch:/opt/free-svn/bin# ulimit -a > > core file size (blocks, -c) unlimited > > data seg size (kbytes, -d) unlimited > > scheduling priority (-e) 0 > > file size (blocks, -f) unlimited > > pending signals (-i) unlimited > > max locked memory (kbytes, -l) unlimited > > max memory size (kbytes, -m) unlimited > > open files (-n) 999999 > > pipe size (512 bytes, -p) 8 > > POSIX message queues (bytes, -q) unlimited > > real-time priority (-r) 0 > > stack size (kbytes, -s) 244 > > cpu time (seconds, -t) unlimited > > max user processes (-u) unlimited > > virtual memory (kbytes, -v) unlimited > > file locks (-x) unlimited > > > > > > My network setup is the following: > > > > SIPP machine (10.10.10.1/24)----------------vlan > > 55 > > ----------(10.10.10.254/24 ) FS > > (10.10.20.254/24)-------------- > > vlan56 > > -------------------(10.10.20.100/24 ) > > OTHER STOCK FS > > > > > > I launched sipp with: > > sipp -sn uac_pcap -s 9999 -r 10 -l 80 -d 60000 -mi 10.10.10.1 -i > > 10.10.10.1 -mp 25000 10.10.10.254:5060 > > > > The dialplan on FS is very simple: > > > > > > > > > > > > > > > > > > > data="sofia/external/9999 at 10.10.20.100 "/> > > > > > > > > > > > > > > FreeSWITCH Version 1.0.trunk (11560M) Started. > > Crash Protection [Disabled] > > Max Sessions[1000] > > Session Rate[100] > > SQL [Enabled] > > > > > > The test is very simple: sipp dial 9999 that matches in my FS > dialplan > > and this is bridged to an other FS machine playing music on hold. > > When I launch "top" I see after 30 to 40 s that FS consumes all > > the CPU > > ressources (with a mean of 50-60 % before), with 80 calls. > > When I set 70 calls, I have to wait 70-80 s before seeing the same > > issue. > > > > Presence is set to false on the 2 profile. > > > > I have the same issue with FS 1.0.2 that' s why I tried FS 11560. > > > > When I use the FS machine as a router to test the packet per second > > performance, I'm reaching 100Mbps with 8000pps in each direction > (from > > vlan 55 to vlan56) with less than 12% CPU. So that I don't think > > there's > > an issue with the network. > > > > Here is an "mpstat -P ALL 1" to show you what's happening suddenly > > with > > 70 bridge calls: > > 12:31:26 CPU %user %nice %sys %iowait %irq %soft > > %steal %idle intr/s > > 12:31:27 all 3,00 0,00 3,00 0,00 1,00 4,00 > > 0,00 89,00 6241,00 > > 12:31:27 0 3,00 0,00 3,00 0,00 1,00 4,00 > > 0,00 89,00 6241,00 > > > > 12:31:27 CPU %user %nice %sys %iowait %irq %soft > > %steal %idle intr/s > > 12:31:28 all 14,14 0,00 56,57 0,00 2,02 5,05 > > 0,00 22,22 6035,35 > > 12:31:28 0 14,14 0,00 56,57 0,00 2,02 5,05 > > 0,00 22,22 6035,35 > > > > 12:31:28 CPU %user %nice %sys %iowait %irq %soft > > %steal %idle intr/s > > 12:31:29 all 24,75 0,00 67,33 0,00 0,99 6,93 > > 0,00 0,00 5483,17 > > 12:31:29 0 24,75 0,00 67,33 0,00 0,99 6,93 > > 0,00 0,00 5483,17 > > > > > > The CPU is going from 89% idle to 0% in less than 2 seconds. > > > > I know that I don't have to expect too much from this kind of > > hardware, > > but it seems strange that the CPU power vanished so suddenly. > > > > Thanks a lot for the guys that have read this long mail :p > > > > kind regards, > > rod > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com > > > > > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > > > > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > > > > > iax:guest at conference.freeswitch.org/888 > > > > googletalk:conf+888 at conference.freeswitch.org > > > > > > pstn:213-799-1400 > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090202/e5cef9ec/attachment-0001.html From kawarod at laposte.net Sun Feb 1 23:36:35 2009 From: kawarod at laposte.net (rod) Date: Mon, 02 Feb 2009 11:36:35 +0400 Subject: [Freeswitch-users] Strange Performance when using as SBC In-Reply-To: <94b568920902012317h8411fccg216435380f62d649@mail.gmail.com> References: <4982EAD5.1010200@laposte.net> <191c3a030901300556s534823a1if4b79b4279962ed1@mail.gmail.com> <49869AF1.1070903@laposte.net> <94b568920902012317h8411fccg216435380f62d649@mail.gmail.com> Message-ID: <4986A283.3090707@laposte.net> Hi Ken, 1) I'd like to use FS to hide topology, so bypass media is not possible 2) done 3) done 4) not used 5) i'm using this ins switch.xml -> , if you think an other log level is more suitable. Regarding logging, I can see in console and in the freeswitch.log that there is still a lot of NOTICE logging, see below: 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:960 switch_core_session_thread() Session 8721 (sofia/internal/sipp at 10.10.10.1:5060) Ended 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:962 switch_core_session_thread() Close Channel sofia/internal/sipp at 10.10.10.1:5060 [CS_HANGUP] 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:960 switch_core_session_thread() Session 8722 (sofia/external/9998 at 10.10.20.100) Ended 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:962 switch_core_session_thread() Close Channel sofia/external/9998 at 10.10.20.100 [CS_HANGUP] 2009-02-02 08:33:56 [NOTICE] sofia.c:3164 sofia_handle_sip_i_state() Channel [sofia/external/9998 at 10.10.20.100] has been answered 2009-02-02 08:33:56 [WARNING] mod_sofia.c:740 sofia_read_frame() Changing codec ptime to 30. I bet you have a linksys/sipura =D Do you have any idea where I can switch off this kind of logging. I thought it should be in /dialplan/internal.xml, but I see that in internal.xml -> thanks a lot for your suggestion. regards, rod Ken Rice wrote: > Dont forget there are several things you can do to increase performance... > > 1) where possible use bypass media or media proxy modes > 2) mount freeswitch/db as a ram drive (if you are using voicemail with > the internal FS DBs you'll need a way to make this persistant across > reboots) > 3) see the wiki for setting reasonable ulimits > 4) (this is my oppinion others may vary) dont use mod_cdr_csv > 5) turn off (or reduce logging) in switch.conf.xml > > all of these thing can greatly improve performance. > > On Mon, Feb 2, 2009 at 1:04 AM, rod > wrote: > > Thanks Anthony, > > the setup is like this: > > sipp server ---- FS 1 ---- FS2 > > FS1 is the AMD CPU that has only one extension in dialplan that > bridges > 9999 to FS2. 9999 is the first extension in FS2 dialplan that > plays moh, > FS2 has no CPU pbm. > > FS1 is maxing out at 60 bridged calls without your option -hp. > > Using -hp, I'm now able to bridge 200 concurrent calls (a great > improvement) and the system is still reactive. CPU load is high > but not > 100% and as the system responds well, I think that doesn't matter. The > 2GB of memory are completely consumed (top command shows 700MB for FS > process). > > I understand that FS1 server is not the best hardware platform, > and I'm > waiting for new 4 cores server for testing. > I will update those numbers when testing with the new hardware. > > regards, > rod. > > Anthony Minessale wrote: > > Which of the 2 machines has the load issue? You said it was one box > > calling the other. > > > > You have 2 major things against you, single CPU and AMD, but you > > should at least be able to get in the vicinity of 800-1000 calls > on a > > box like that. > > > > Are you calling the default 9999? It's not really an appropriate > > extension for load testing. > > On the terminating box you should set up a manual extension that is > > the first one in the dial plan > > to play a wav file from preferably a ram disk or /tmp > > > > If you do plan on using this in production accept nothing less > than a > > multi-core intel machine with at least 4 cores, the more cores the > > better because that parallel processing is where FS gets it's > atvantage. > > > > > > > > On Fri, Jan 30, 2009 at 5:56 AM, rod > > >> wrote: > > > > Dear list, > > > > I've been playing with freeswitch for some time (2 months) > and the > > fact > > is that I'm very pleased with the functionnalities of this > software. > > > > I'd like to use FS as a SBC handling media and I'm doing some > > tests with > > sipp to load the machine but I'm unable to bridge more than > 60 calls > > without seeing the CPU being loaded at 100%. I'm sure > something is > > going > > wrong with my setup but I'm unable to see what. > > > > The test machine has the following specs: > > Athlon XP 3500+ with 2GB of memory (I know this is not a > high end > > machine :p) > > > > Freeswitch:/opt/freeswitch/log# cat /proc/cpuinfo > > processor : 0 > > vendor_id : AuthenticAMD > > cpu family : 15 > > model : 95 > > model name : AMD Athlon(tm) 64 Processor 3500+ > > stepping : 2 > > cpu MHz : 2199.973 > > cache size : 512 KB > > fpu : yes > > fpu_exception : yes > > cpuid level : 1 > > wp : yes > > flags : fpu vme de pse tsc msr pae mce cx8 apic > sep mtrr pge > > mca cmov pat pse36 clflush mmx fxsr sse sse2 syscall nx mmxext > > fxsr_opt > > rdtscp lm 3dnowext 3dnow up rep_good pni cx16 lahf_lm svm > extapic > > cr8_legacy > > bogomips : 4402.97 > > TLB size : 1024 4K pages > > clflush size : 64 > > cache_alignment : 64 > > address sizes : 40 bits physical, 48 bits virtual > > power management: ts fid vid ttp tm stc > > > > I installed FS on a fresh debian 64: > > Linux Freeswitch 2.6.26-1-amd64 #1 SMP Sat Jan 10 17:57:00 > UTC 2009 > > x86_64 GNU/Linux > > > > I set the ulimit parameters like those on the website: > > freeswitch at internal> ... > > Freeswitch:/opt/free-svn/bin# ulimit -a > > core file size (blocks, -c) unlimited > > data seg size (kbytes, -d) unlimited > > scheduling priority (-e) 0 > > file size (blocks, -f) unlimited > > pending signals (-i) unlimited > > max locked memory (kbytes, -l) unlimited > > max memory size (kbytes, -m) unlimited > > open files (-n) 999999 > > pipe size (512 bytes, -p) 8 > > POSIX message queues (bytes, -q) unlimited > > real-time priority (-r) 0 > > stack size (kbytes, -s) 244 > > cpu time (seconds, -t) unlimited > > max user processes (-u) unlimited > > virtual memory (kbytes, -v) unlimited > > file locks (-x) unlimited > > > > > > My network setup is the following: > > > > SIPP machine (10.10.10.1/24)----------------vlan > > > 55 > > ----------(10.10.10.254/24 > ) FS > > (10.10.20.254/24)-------------- > > > vlan56 > > -------------------(10.10.20.100/24 > ) > > OTHER STOCK FS > > > > > > I launched sipp with: > > sipp -sn uac_pcap -s 9999 -r 10 -l 80 -d 60000 -mi > 10.10.10.1 -i > > 10.10.10.1 -mp 25000 10.10.10.254:5060 > > > > > The dialplan on FS is very simple: > > > > > > > > > > > > > > > > > > > data="sofia/external/9999 at 10.10.20.100 > >"/> > > > > > > > > > > > > > > FreeSWITCH Version 1.0.trunk (11560M) Started. > > Crash Protection [Disabled] > > Max Sessions[1000] > > Session Rate[100] > > SQL [Enabled] > > > > > > The test is very simple: sipp dial 9999 that matches in my > FS dialplan > > and this is bridged to an other FS machine playing music on > hold. > > When I launch "top" I see after 30 to 40 s that FS consumes all > > the CPU > > ressources (with a mean of 50-60 % before), with 80 calls. > > When I set 70 calls, I have to wait 70-80 s before seeing > the same > > issue. > > > > Presence is set to false on the 2 profile. > > > > I have the same issue with FS 1.0.2 that' s why I tried FS > 11560. > > > > When I use the FS machine as a router to test the packet per > second > > performance, I'm reaching 100Mbps with 8000pps in each > direction (from > > vlan 55 to vlan56) with less than 12% CPU. So that I don't think > > there's > > an issue with the network. > > > > Here is an "mpstat -P ALL 1" to show you what's happening > suddenly > > with > > 70 bridge calls: > > 12:31:26 CPU %user %nice %sys %iowait %irq %soft > > %steal %idle intr/s > > 12:31:27 all 3,00 0,00 3,00 0,00 1,00 4,00 > > 0,00 89,00 6241,00 > > 12:31:27 0 3,00 0,00 3,00 0,00 1,00 4,00 > > 0,00 89,00 6241,00 > > > > 12:31:27 CPU %user %nice %sys %iowait %irq %soft > > %steal %idle intr/s > > 12:31:28 all 14,14 0,00 56,57 0,00 2,02 5,05 > > 0,00 22,22 6035,35 > > 12:31:28 0 14,14 0,00 56,57 0,00 2,02 5,05 > > 0,00 22,22 6035,35 > > > > 12:31:28 CPU %user %nice %sys %iowait %irq %soft > > %steal %idle intr/s > > 12:31:29 all 24,75 0,00 67,33 0,00 0,99 6,93 > > 0,00 0,00 5483,17 > > 12:31:29 0 24,75 0,00 67,33 0,00 0,99 6,93 > > 0,00 0,00 5483,17 > > > > > > The CPU is going from 89% idle to 0% in less than 2 seconds. > > > > I know that I don't have to expect too much from this kind of > > hardware, > > but it seems strange that the CPU power vanished so suddenly. > > > > Thanks a lot for the guys that have read this long mail :p > > > > kind regards, > > rod > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com > > > > > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > > > > > IRC: irc.freenode.net > #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > > > > > iax:guest at conference.freeswitch.org/888 > > > > > googletalk:conf+888 at conference.freeswitch.org > > > > > > pstn:213-799-1400 > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From blansky at interwise.com Mon Feb 2 00:33:19 2009 From: blansky at interwise.com (Boris Lansky) Date: Mon, 2 Feb 2009 10:33:19 +0200 Subject: [Freeswitch-users] Outbound Proxy configuration In-Reply-To: <191c3a030902011019k94e5fa3i1a37c2c0b2e66e52@mail.gmail.com> References: <7160CD9F-F06E-4964-943A-F4B0C12150CC@jerris.com><4981D3F8.2030207@freeswitch.org> <191c3a030902011019k94e5fa3i1a37c2c0b2e66e52@mail.gmail.com> Message-ID: Thanks Anthony, A little explanation of yours was very helpful. The directive works for me. Regards, Boris Lansky Unified Communications Telephony Team AT&T Unified Communications Phone: +972.3.976.7604 Fax: +972.3.976.7712 blansky at interwise.com From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Sunday, February 01, 2009 8:19 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Outbound Proxy configuration They probably saw your signature that says you work for AT&T and assumed you could follow the documentation since AT&T is a telephone company. What they were trying to explain was if you want to send a call out of a sofia profile you can set the proxy on the fly with that extra parameter added to the dial string anywhere it's used. The explanation requires you understand the basic operation of freeswitch. the uri contained in fs_path indicates the sip address of a proxy server. On Sun, Feb 1, 2009 at 4:37 AM, Boris Lansky wrote: I have checked the "fs_path" usage ... again (I have done it before I have issued my question as well). And once again I can't understand why this thing is useful for me. I have found only two small examples refer the issue that use "fs_path" for a an API call. What I need is a real configuration example that shows configuration of a Proxy Server for all outbound calls going out from a Free Switch. I will real appreciate if I will get an exact answer and not just a general link to a FS doc. Regards, Boris Lansky Unified Communications Telephony Team AT&T Unified Communications Phone: +972.3.976.7604 Fax: +972.3.976.7712 blansky at interwise.com From: freeswitch-users-bounces at lists.freeswitch.org [mailto: freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Raymond Chandler Sent: Thursday, January 29, 2009 6:06 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Outbound Proxy configuration Boris Lansky wrote: Sorry for the stupid question but in what configuration file should I add such line "sofia/foo/user at that.domain ;fs_path=sip:proxy.this.domain" ? appology accepted... look in the dialplan, there should be plenty of documentations on using the dialplan on our wiki... wiki.freeswitch.org... look for sofia syntax too on the mod_sofia page -Ray _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090202/3cea3f07/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/png Size: 16211 bytes Desc: image001.png Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090202/3cea3f07/attachment-0001.png From sias at cpdata.co.za Mon Feb 2 00:51:57 2009 From: sias at cpdata.co.za (Sias Mey) Date: Mon, 2 Feb 2009 10:51:57 +0200 Subject: [Freeswitch-users] Conference dialing and uuid In-Reply-To: <191c3a030901300605r78dd8943w7fb1075ef851e1d@mail.gmail.com> References: <20090130103944.GA23536@cpdata.co.za> <20090130133315.GB23536@cpdata.co.za> <191c3a030901300605r78dd8943w7fb1075ef851e1d@mail.gmail.com> Message-ID: <20090202085157.GA3555@cpdata.co.za> Yes ... yes indeed I can. That works quite a bit better than generating 4 channels and getting massively confused with what uuid does what... but now im stuck without ringback again :-(. In my conference dial string I send: {ringback=\'%(400,200,400,450)\',transfer_ringback=\'%(400,200,400,450)\', .... }sofia/internal/1001 at xxx.xxx.xxx.xxx A dump of all the channel variables shows ringback is set to %25(400,200,400,450)%3B%25(400,2200,400,450) %25(400,200,400,450)%3B%25(400,2200,400,450) This seems ok to me but I still dont get any ringback. Thanks again for answering all the anoying questions from the same guy :-P, Sias On Fri, Jan 30, 2009 at 08:05:07AM -0600, Anthony Minessale wrote: > you should be able to use {} in the dial command > you also should be able to do > originate {...}sofia/profile/[1]user at domain.com > conference:@ inline > to the api interface > > On Fri, Jan 30, 2009 at 7:33 AM, Sias Mey <[2]sias at cpdata.co.za> wrote: > > Hi Brian, > Hmmm Ill do some more testing on it later. But I got a destination > out > of order when I tried. Right now Im busy implementing the string > checking. Which seems like it will work out ok, but is clearly not > ideal. > Thanks for the replay > > On Fri, Jan 30, 2009 at 04:45:54AM -0600, Brian West wrote: > > What wasn't working about this? The {} can be used everywhere > without > > a problem... Maybe you can provide more details on this. > > > > /b > > > > > > > > > > On Jan 30, 2009, at 4:39 AM, Sias Mey wrote: > > > > > > > > I couldent find a way of setting channel variables or executing > > > javascript directly on the conference dial since it expects and > > > endpoint > > > and the {} syntax produced an error. So now I am using the Loopback > > > inteface to register some values. > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > [3]Freeswitch-users at lists.freeswitch.org > > [4]http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:[5]http://lists.freeswitch.org/mailman/options/freeswitch-u > sers > > [6]http://www.freeswitch.org > _______________________________________________ > Freeswitch-users mailing list > [7]Freeswitch-users at lists.freeswitch.org > [8]http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:[9]http://lists.freeswitch.org/mailman/options/freeswitch-u > sers > [10]http://www.freeswitch.org > > -- > Anthony Minessale II > FreeSWITCH [11]http://www.freeswitch.org/ > ClueCon [12]http://www.cluecon.com/ > AIM: anthm > [13]MSN:anthony_minessale at hotmail.com > GTALK/JABBER/[14]PAYPAL:anthony.minessale at gmail.com > IRC: [15]irc.freenode.net #freeswitch > FreeSWITCH Developer Conference > [16]sip:888 at conference.freeswitch.org > [17]iax:guest at conference.freeswitch.org/888 > [18]googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > References > > 1. mailto:user at domain.com > 2. mailto:sias at cpdata.co.za > 3. mailto:Freeswitch-users at lists.freeswitch.org > 4. http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > 5. http://lists.freeswitch.org/mailman/options/freeswitch-users > 6. http://www.freeswitch.org/ > 7. mailto:Freeswitch-users at lists.freeswitch.org > 8. http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > 9. http://lists.freeswitch.org/mailman/options/freeswitch-users > 10. http://www.freeswitch.org/ > 11. http://www.freeswitch.org/ > 12. http://www.cluecon.com/ > 13. mailto:MSN%3Aanthony_minessale at hotmail.com > 14. mailto:PAYPAL%3Aanthony.minessale at gmail.com > 15. http://irc.freenode.net/ > 16. mailto:sip%3A888 at conference.freeswitch.org > 17. http://iax:guest at conference.freeswitch.org/888 > 18. mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Mon Feb 2 00:55:42 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 2 Feb 2009 02:55:42 -0600 Subject: [Freeswitch-users] Conference dialing and uuid In-Reply-To: <20090202085157.GA3555@cpdata.co.za> References: <20090130103944.GA23536@cpdata.co.za> <20090130133315.GB23536@cpdata.co.za> <191c3a030901300605r78dd8943w7fb1075ef851e1d@mail.gmail.com> <20090202085157.GA3555@cpdata.co.za> Message-ID: <0F863750-5C77-4CD9-ADC6-CFDDF2B149B4@freeswitch.org> You can't get ringback dialing out from a conference its not possible as it is now. /b On Feb 2, 2009, at 2:51 AM, Sias Mey wrote: > Yes ... yes indeed I can. > > That works quite a bit better than generating 4 channels and getting > massively confused with what uuid does what... but now im stuck > without > ringback again :-(. > > In my conference dial string I send: > {ringback=\'%(400,200,400,450)\',transfer_ringback= > \'%(400,200,400,450)\', > .... }sofia/internal/1001 at xxx.xxx.xxx.xxx > > A dump of all the channel variables shows ringback is set to > > %25(400,200,400,450)%3B%25(400,2200,400,450) > %25(400,200,400,450)%3B%25(400,2200,400,450) transfer_ringback> > > This seems ok to me but I still dont get any ringback. > > Thanks again for answering all the anoying questions from the same guy > :-P, > Sias -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090202/ede8482f/attachment.html From jaybinks at gmail.com Mon Feb 2 01:09:10 2009 From: jaybinks at gmail.com (jay binks) Date: Mon, 2 Feb 2009 19:09:10 +1000 Subject: [Freeswitch-users] Strange Performance when using as SBC In-Reply-To: <4986A283.3090707@laposte.net> References: <4982EAD5.1010200@laposte.net> <191c3a030901300556s534823a1if4b79b4279962ed1@mail.gmail.com> <49869AF1.1070903@laposte.net> <94b568920902012317h8411fccg216435380f62d649@mail.gmail.com> <4986A283.3090707@laposte.net> Message-ID: for topology hiding, use proxy media. it means FS ignores the RTP stream totally, and just passes it through. On Mon, Feb 2, 2009 at 5:36 PM, rod wrote: > Hi Ken, > > 1) I'd like to use FS to hide topology, so bypass media is not possible > 2) done > 3) done > 4) not used > 5) i'm using this ins switch.xml -> value="info"/>, if you think an other log level is more suitable. > > Regarding logging, I can see in console and in the freeswitch.log that > there is still a lot of NOTICE logging, see below: > 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:960 > switch_core_session_thread() Session 8721 > (sofia/internal/sipp at 10.10.10.1:5060) Ended > 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:962 > switch_core_session_thread() Close Channel > sofia/internal/sipp at 10.10.10.1:5060 [CS_HANGUP] > 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:960 > switch_core_session_thread() Session 8722 > (sofia/external/9998 at 10.10.20.100) Ended > 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:962 > switch_core_session_thread() Close Channel > sofia/external/9998 at 10.10.20.100 [CS_HANGUP] > 2009-02-02 08:33:56 [NOTICE] sofia.c:3164 sofia_handle_sip_i_state() > Channel [sofia/external/9998 at 10.10.20.100] has been answered > 2009-02-02 08:33:56 [WARNING] mod_sofia.c:740 sofia_read_frame() > Changing codec ptime to 30. I bet you have a linksys/sipura =D > > Do you have any idea where I can switch off this kind of logging. I > thought it should be in /dialplan/internal.xml, but I see that in > internal.xml -> > > thanks a lot for your suggestion. > > regards, > rod > > Ken Rice wrote: > > Dont forget there are several things you can do to increase > performance... > > > > 1) where possible use bypass media or media proxy modes > > 2) mount freeswitch/db as a ram drive (if you are using voicemail with > > the internal FS DBs you'll need a way to make this persistant across > > reboots) > > 3) see the wiki for setting reasonable ulimits > > 4) (this is my oppinion others may vary) dont use mod_cdr_csv > > 5) turn off (or reduce logging) in switch.conf.xml > > > > all of these thing can greatly improve performance. > > > > On Mon, Feb 2, 2009 at 1:04 AM, rod > > wrote: > > > > Thanks Anthony, > > > > the setup is like this: > > > > sipp server ---- FS 1 ---- FS2 > > > > FS1 is the AMD CPU that has only one extension in dialplan that > > bridges > > 9999 to FS2. 9999 is the first extension in FS2 dialplan that > > plays moh, > > FS2 has no CPU pbm. > > > > FS1 is maxing out at 60 bridged calls without your option -hp. > > > > Using -hp, I'm now able to bridge 200 concurrent calls (a great > > improvement) and the system is still reactive. CPU load is high > > but not > > 100% and as the system responds well, I think that doesn't matter. > The > > 2GB of memory are completely consumed (top command shows 700MB for FS > > process). > > > > I understand that FS1 server is not the best hardware platform, > > and I'm > > waiting for new 4 cores server for testing. > > I will update those numbers when testing with the new hardware. > > > > regards, > > rod. > > > > Anthony Minessale wrote: > > > Which of the 2 machines has the load issue? You said it was one box > > > calling the other. > > > > > > You have 2 major things against you, single CPU and AMD, but you > > > should at least be able to get in the vicinity of 800-1000 calls > > on a > > > box like that. > > > > > > Are you calling the default 9999? It's not really an appropriate > > > extension for load testing. > > > On the terminating box you should set up a manual extension that is > > > the first one in the dial plan > > > to play a wav file from preferably a ram disk or /tmp > > > > > > If you do plan on using this in production accept nothing less > > than a > > > multi-core intel machine with at least 4 cores, the more cores the > > > better because that parallel processing is where FS gets it's > > atvantage. > > > > > > > > > > > > On Fri, Jan 30, 2009 at 5:56 AM, rod > > > > >> wrote: > > > > > > Dear list, > > > > > > I've been playing with freeswitch for some time (2 months) > > and the > > > fact > > > is that I'm very pleased with the functionnalities of this > > software. > > > > > > I'd like to use FS as a SBC handling media and I'm doing some > > > tests with > > > sipp to load the machine but I'm unable to bridge more than > > 60 calls > > > without seeing the CPU being loaded at 100%. I'm sure > > something is > > > going > > > wrong with my setup but I'm unable to see what. > > > > > > The test machine has the following specs: > > > Athlon XP 3500+ with 2GB of memory (I know this is not a > > high end > > > machine :p) > > > > > > Freeswitch:/opt/freeswitch/log# cat /proc/cpuinfo > > > processor : 0 > > > vendor_id : AuthenticAMD > > > cpu family : 15 > > > model : 95 > > > model name : AMD Athlon(tm) 64 Processor 3500+ > > > stepping : 2 > > > cpu MHz : 2199.973 > > > cache size : 512 KB > > > fpu : yes > > > fpu_exception : yes > > > cpuid level : 1 > > > wp : yes > > > flags : fpu vme de pse tsc msr pae mce cx8 apic > > sep mtrr pge > > > mca cmov pat pse36 clflush mmx fxsr sse sse2 syscall nx mmxext > > > fxsr_opt > > > rdtscp lm 3dnowext 3dnow up rep_good pni cx16 lahf_lm svm > > extapic > > > cr8_legacy > > > bogomips : 4402.97 > > > TLB size : 1024 4K pages > > > clflush size : 64 > > > cache_alignment : 64 > > > address sizes : 40 bits physical, 48 bits virtual > > > power management: ts fid vid ttp tm stc > > > > > > I installed FS on a fresh debian 64: > > > Linux Freeswitch 2.6.26-1-amd64 #1 SMP Sat Jan 10 17:57:00 > > UTC 2009 > > > x86_64 GNU/Linux > > > > > > I set the ulimit parameters like those on the website: > > > freeswitch at internal> ... > > > Freeswitch:/opt/free-svn/bin# ulimit -a > > > core file size (blocks, -c) unlimited > > > data seg size (kbytes, -d) unlimited > > > scheduling priority (-e) 0 > > > file size (blocks, -f) unlimited > > > pending signals (-i) unlimited > > > max locked memory (kbytes, -l) unlimited > > > max memory size (kbytes, -m) unlimited > > > open files (-n) 999999 > > > pipe size (512 bytes, -p) 8 > > > POSIX message queues (bytes, -q) unlimited > > > real-time priority (-r) 0 > > > stack size (kbytes, -s) 244 > > > cpu time (seconds, -t) unlimited > > > max user processes (-u) unlimited > > > virtual memory (kbytes, -v) unlimited > > > file locks (-x) unlimited > > > > > > > > > My network setup is the following: > > > > > > SIPP machine (10.10.10.1/24)----------------vlan > > > > > 55 > > > ----------(10.10.10.254/24 > > ) FS > > > (10.10.20.254/24)-------------- > > > > > vlan56 > > > -------------------(10.10.20.100/24 > > ) > > > OTHER STOCK FS > > > > > > > > > I launched sipp with: > > > sipp -sn uac_pcap -s 9999 -r 10 -l 80 -d 60000 -mi > > 10.10.10.1 -i > > > 10.10.10.1 -mp 25000 10.10.10.254:5060 > > > > > > > > The dialplan on FS is very simple: > > > > > > > > > > > > > > > > > > > > > > > > > > > > > data="sofia/external/9999 at 10.10.20.100 > > > >"/> > > > > > > > > > > > > > > > > > > > > > FreeSWITCH Version 1.0.trunk (11560M) Started. > > > Crash Protection [Disabled] > > > Max Sessions[1000] > > > Session Rate[100] > > > SQL [Enabled] > > > > > > > > > The test is very simple: sipp dial 9999 that matches in my > > FS dialplan > > > and this is bridged to an other FS machine playing music on > > hold. > > > When I launch "top" I see after 30 to 40 s that FS consumes > all > > > the CPU > > > ressources (with a mean of 50-60 % before), with 80 calls. > > > When I set 70 calls, I have to wait 70-80 s before seeing > > the same > > > issue. > > > > > > Presence is set to false on the 2 profile. > > > > > > I have the same issue with FS 1.0.2 that' s why I tried FS > > 11560. > > > > > > When I use the FS machine as a router to test the packet per > > second > > > performance, I'm reaching 100Mbps with 8000pps in each > > direction (from > > > vlan 55 to vlan56) with less than 12% CPU. So that I don't > think > > > there's > > > an issue with the network. > > > > > > Here is an "mpstat -P ALL 1" to show you what's happening > > suddenly > > > with > > > 70 bridge calls: > > > 12:31:26 CPU %user %nice %sys %iowait %irq > %soft > > > %steal %idle intr/s > > > 12:31:27 all 3,00 0,00 3,00 0,00 1,00 > 4,00 > > > 0,00 89,00 6241,00 > > > 12:31:27 0 3,00 0,00 3,00 0,00 1,00 > 4,00 > > > 0,00 89,00 6241,00 > > > > > > 12:31:27 CPU %user %nice %sys %iowait %irq > %soft > > > %steal %idle intr/s > > > 12:31:28 all 14,14 0,00 56,57 0,00 2,02 > 5,05 > > > 0,00 22,22 6035,35 > > > 12:31:28 0 14,14 0,00 56,57 0,00 2,02 > 5,05 > > > 0,00 22,22 6035,35 > > > > > > 12:31:28 CPU %user %nice %sys %iowait %irq > %soft > > > %steal %idle intr/s > > > 12:31:29 all 24,75 0,00 67,33 0,00 0,99 > 6,93 > > > 0,00 0,00 5483,17 > > > 12:31:29 0 24,75 0,00 67,33 0,00 0,99 > 6,93 > > > 0,00 0,00 5483,17 > > > > > > > > > The CPU is going from 89% idle to 0% in less than 2 seconds. > > > > > > I know that I don't have to expect too much from this kind of > > > hardware, > > > but it seems strange that the CPU power vanished so suddenly. > > > > > > Thanks a lot for the guys that have read this long mail :p > > > > > > kind regards, > > > rod > > > > > > > > > _______________________________________________ > > > Freeswitch-users mailing list > > > Freeswitch-users at lists.freeswitch.org > > > > > > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > > > > > > > > > > -- > > > Anthony Minessale II > > > > > > FreeSWITCH http://www.freeswitch.org/ > > > ClueCon http://www.cluecon.com/ > > > > > > AIM: anthm > > > MSN:anthony_minessale at hotmail.com > > > > > > > > > > >> > > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > > > > > > > > > >> > > > IRC: irc.freenode.net > > #freeswitch > > > > > > FreeSWITCH Developer Conference > > > sip:888 at conference.freeswitch.org > > > > > > > > > > >> > > > iax:guest at conference.freeswitch.org/888 > > > > > > > > googletalk:conf+888 at conference.freeswitch.org > > > > > > > > > > >> > > > pstn:213-799-1400 > > > > > > ------------------------------------------------------------------------ > > > > > > _______________________________________________ > > > Freeswitch-users mailing list > > > Freeswitch-users at lists.freeswitch.org > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Sincerely Jay -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090202/c0181163/attachment-0001.html From krice at freeswitch.org Mon Feb 2 01:09:46 2009 From: krice at freeswitch.org (Ken Rice) Date: Mon, 02 Feb 2009 03:09:46 -0600 Subject: [Freeswitch-users] Strange Performance when using as SBC In-Reply-To: <4986A283.3090707@laposte.net> Message-ID: If you don't have to transcode, using proxy media mode will still save you some CPU time. This is 1/2 way between bypass media and the default media interactive mode. The other draw back to this mode is if you are using FS to clean up RTP and DTMF you loose those functions but they are not needed in most use cases. As far as the log level goes, I found that once I had things stable setting the loglevel to helped a good deal... Info is probably a bit too high of a loglevel I would probably go for CRIT or ERR (2 or 1 respectively) if you insist on leaving logging turned on... On a busy system these can and will generate a good deal of activity (and disk IO if using mod_logfile) Ken > From: rod > Reply-To: > Date: Mon, 02 Feb 2009 11:36:35 +0400 > To: > Subject: Re: [Freeswitch-users] Strange Performance when using as SBC > > Hi Ken, > > 1) I'd like to use FS to hide topology, so bypass media is not possible > 2) done > 3) done > 4) not used > 5) i'm using this ins switch.xml -> value="info"/>, if you think an other log level is more suitable. > > Regarding logging, I can see in console and in the freeswitch.log that > there is still a lot of NOTICE logging, see below: > 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:960 > switch_core_session_thread() Session 8721 > (sofia/internal/sipp at 10.10.10.1:5060) Ended > 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:962 > switch_core_session_thread() Close Channel > sofia/internal/sipp at 10.10.10.1:5060 [CS_HANGUP] > 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:960 > switch_core_session_thread() Session 8722 > (sofia/external/9998 at 10.10.20.100) Ended > 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:962 > switch_core_session_thread() Close Channel > sofia/external/9998 at 10.10.20.100 [CS_HANGUP] > 2009-02-02 08:33:56 [NOTICE] sofia.c:3164 sofia_handle_sip_i_state() > Channel [sofia/external/9998 at 10.10.20.100] has been answered > 2009-02-02 08:33:56 [WARNING] mod_sofia.c:740 sofia_read_frame() > Changing codec ptime to 30. I bet you have a linksys/sipura =D > > Do you have any idea where I can switch off this kind of logging. I > thought it should be in /dialplan/internal.xml, but I see that in > internal.xml -> > > thanks a lot for your suggestion. > > regards, > rod > > Ken Rice wrote: >> Dont forget there are several things you can do to increase performance... >> >> 1) where possible use bypass media or media proxy modes >> 2) mount freeswitch/db as a ram drive (if you are using voicemail with >> the internal FS DBs you'll need a way to make this persistant across >> reboots) >> 3) see the wiki for setting reasonable ulimits >> 4) (this is my oppinion others may vary) dont use mod_cdr_csv >> 5) turn off (or reduce logging) in switch.conf.xml >> >> all of these thing can greatly improve performance. >> >> On Mon, Feb 2, 2009 at 1:04 AM, rod > > wrote: >> >> Thanks Anthony, >> >> the setup is like this: >> >> sipp server ---- FS 1 ---- FS2 >> >> FS1 is the AMD CPU that has only one extension in dialplan that >> bridges >> 9999 to FS2. 9999 is the first extension in FS2 dialplan that >> plays moh, >> FS2 has no CPU pbm. >> >> FS1 is maxing out at 60 bridged calls without your option -hp. >> >> Using -hp, I'm now able to bridge 200 concurrent calls (a great >> improvement) and the system is still reactive. CPU load is high >> but not >> 100% and as the system responds well, I think that doesn't matter. The >> 2GB of memory are completely consumed (top command shows 700MB for FS >> process). >> >> I understand that FS1 server is not the best hardware platform, >> and I'm >> waiting for new 4 cores server for testing. >> I will update those numbers when testing with the new hardware. >> >> regards, >> rod. >> >> Anthony Minessale wrote: >>> Which of the 2 machines has the load issue? You said it was one box >>> calling the other. >>> >>> You have 2 major things against you, single CPU and AMD, but you >>> should at least be able to get in the vicinity of 800-1000 calls >> on a >>> box like that. >>> >>> Are you calling the default 9999? It's not really an appropriate >>> extension for load testing. >>> On the terminating box you should set up a manual extension that is >>> the first one in the dial plan >>> to play a wav file from preferably a ram disk or /tmp >>> >>> If you do plan on using this in production accept nothing less >> than a >>> multi-core intel machine with at least 4 cores, the more cores the >>> better because that parallel processing is where FS gets it's >> atvantage. >>> >>> >>> >>> On Fri, Jan 30, 2009 at 5:56 AM, rod > >>> >> wrote: >>> >>> Dear list, >>> >>> I've been playing with freeswitch for some time (2 months) >> and the >>> fact >>> is that I'm very pleased with the functionnalities of this >> software. >>> >>> I'd like to use FS as a SBC handling media and I'm doing some >>> tests with >>> sipp to load the machine but I'm unable to bridge more than >> 60 calls >>> without seeing the CPU being loaded at 100%. I'm sure >> something is >>> going >>> wrong with my setup but I'm unable to see what. >>> >>> The test machine has the following specs: >>> Athlon XP 3500+ with 2GB of memory (I know this is not a >> high end >>> machine :p) >>> >>> Freeswitch:/opt/freeswitch/log# cat /proc/cpuinfo >>> processor : 0 >>> vendor_id : AuthenticAMD >>> cpu family : 15 >>> model : 95 >>> model name : AMD Athlon(tm) 64 Processor 3500+ >>> stepping : 2 >>> cpu MHz : 2199.973 >>> cache size : 512 KB >>> fpu : yes >>> fpu_exception : yes >>> cpuid level : 1 >>> wp : yes >>> flags : fpu vme de pse tsc msr pae mce cx8 apic >> sep mtrr pge >>> mca cmov pat pse36 clflush mmx fxsr sse sse2 syscall nx mmxext >>> fxsr_opt >>> rdtscp lm 3dnowext 3dnow up rep_good pni cx16 lahf_lm svm >> extapic >>> cr8_legacy >>> bogomips : 4402.97 >>> TLB size : 1024 4K pages >>> clflush size : 64 >>> cache_alignment : 64 >>> address sizes : 40 bits physical, 48 bits virtual >>> power management: ts fid vid ttp tm stc >>> >>> I installed FS on a fresh debian 64: >>> Linux Freeswitch 2.6.26-1-amd64 #1 SMP Sat Jan 10 17:57:00 >> UTC 2009 >>> x86_64 GNU/Linux >>> >>> I set the ulimit parameters like those on the website: >>> freeswitch at internal> ... >>> Freeswitch:/opt/free-svn/bin# ulimit -a >>> core file size (blocks, -c) unlimited >>> data seg size (kbytes, -d) unlimited >>> scheduling priority (-e) 0 >>> file size (blocks, -f) unlimited >>> pending signals (-i) unlimited >>> max locked memory (kbytes, -l) unlimited >>> max memory size (kbytes, -m) unlimited >>> open files (-n) 999999 >>> pipe size (512 bytes, -p) 8 >>> POSIX message queues (bytes, -q) unlimited >>> real-time priority (-r) 0 >>> stack size (kbytes, -s) 244 >>> cpu time (seconds, -t) unlimited >>> max user processes (-u) unlimited >>> virtual memory (kbytes, -v) unlimited >>> file locks (-x) unlimited >>> >>> >>> My network setup is the following: >>> >>> SIPP machine (10.10.10.1/24)----------------vlan >> >>> 55 >>> ----------(10.10.10.254/24 >> ) FS >>> (10.10.20.254/24)-------------- >> >>> vlan56 >>> -------------------(10.10.20.100/24 >> ) >>> OTHER STOCK FS >>> >>> >>> I launched sipp with: >>> sipp -sn uac_pcap -s 9999 -r 10 -l 80 -d 60000 -mi >> 10.10.10.1 -i >>> 10.10.10.1 -mp 25000 10.10.10.254:5060 >> >>> >>> The dialplan on FS is very simple: >>> >>> >>> >>> >>> >>> >>> >>> >>> >> data="sofia/external/9999 at 10.10.20.100 >> > >"/> >>> >>> >>> >>> >>> >>> >>> FreeSWITCH Version 1.0.trunk (11560M) Started. >>> Crash Protection [Disabled] >>> Max Sessions[1000] >>> Session Rate[100] >>> SQL [Enabled] >>> >>> >>> The test is very simple: sipp dial 9999 that matches in my >> FS dialplan >>> and this is bridged to an other FS machine playing music on >> hold. >>> When I launch "top" I see after 30 to 40 s that FS consumes all >>> the CPU >>> ressources (with a mean of 50-60 % before), with 80 calls. >>> When I set 70 calls, I have to wait 70-80 s before seeing >> the same >>> issue. >>> >>> Presence is set to false on the 2 profile. >>> >>> I have the same issue with FS 1.0.2 that' s why I tried FS >> 11560. >>> >>> When I use the FS machine as a router to test the packet per >> second >>> performance, I'm reaching 100Mbps with 8000pps in each >> direction (from >>> vlan 55 to vlan56) with less than 12% CPU. So that I don't think >>> there's >>> an issue with the network. >>> >>> Here is an "mpstat -P ALL 1" to show you what's happening >> suddenly >>> with >>> 70 bridge calls: >>> 12:31:26 CPU %user %nice %sys %iowait %irq %soft >>> %steal %idle intr/s >>> 12:31:27 all 3,00 0,00 3,00 0,00 1,00 4,00 >>> 0,00 89,00 6241,00 >>> 12:31:27 0 3,00 0,00 3,00 0,00 1,00 4,00 >>> 0,00 89,00 6241,00 >>> >>> 12:31:27 CPU %user %nice %sys %iowait %irq %soft >>> %steal %idle intr/s >>> 12:31:28 all 14,14 0,00 56,57 0,00 2,02 5,05 >>> 0,00 22,22 6035,35 >>> 12:31:28 0 14,14 0,00 56,57 0,00 2,02 5,05 >>> 0,00 22,22 6035,35 >>> >>> 12:31:28 CPU %user %nice %sys %iowait %irq %soft >>> %steal %idle intr/s >>> 12:31:29 all 24,75 0,00 67,33 0,00 0,99 6,93 >>> 0,00 0,00 5483,17 >>> 12:31:29 0 24,75 0,00 67,33 0,00 0,99 6,93 >>> 0,00 0,00 5483,17 >>> >>> >>> The CPU is going from 89% idle to 0% in less than 2 seconds. >>> >>> I know that I don't have to expect too much from this kind of >>> hardware, >>> but it seems strange that the CPU power vanished so suddenly. >>> >>> Thanks a lot for the guys that have read this long mail :p >>> >>> kind regards, >>> rod >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >> >>> > > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >>> >>> -- >>> Anthony Minessale II >>> >>> FreeSWITCH http://www.freeswitch.org/ >>> ClueCon http://www.cluecon.com/ >>> >>> AIM: anthm >>> MSN:anthony_minessale at hotmail.com >> >>> > > >>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> >>> > > >>> IRC: irc.freenode.net >> #freeswitch >>> >>> FreeSWITCH Developer Conference >>> sip:888 at conference.freeswitch.org >> >>> > > >>> iax:guest at conference.freeswitch.org/888 >> >>> >>> googletalk:conf+888 at conference.freeswitch.org >> >>> > > >>> pstn:213-799-1400 >>> >> ------------------------------------------------------------------------ >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From kawarod at laposte.net Mon Feb 2 02:00:12 2009 From: kawarod at laposte.net (rod) Date: Mon, 02 Feb 2009 14:00:12 +0400 Subject: [Freeswitch-users] Strange Performance when using as SBC In-Reply-To: References: Message-ID: <4986C42C.7030700@laposte.net> Hi Ken, Jay, thanks for pointing to proxy media, I will test. Ken, you are right, I was brain damaged (a stupid mistake) when setting INFO cause this kind of level could be very verbose. I'm switching to CRIT or ERR. Thanks guys, rod. thanks for Ken Rice wrote: > If you don't have to transcode, using proxy media mode will still save you > some CPU time. This is 1/2 way between bypass media and the default media > interactive mode. The other draw back to this mode is if you are using FS to > clean up RTP and DTMF you loose those functions but they are not needed in > most use cases. > > As far as the log level goes, I found that once I had things stable setting > the loglevel to helped a good deal... Info is probably a bit too high of a > loglevel I would probably go for CRIT or ERR (2 or 1 respectively) if you > insist on leaving logging turned on... On a busy system these can and will > generate a good deal of activity (and disk IO if using mod_logfile) > > Ken > > > >> From: rod >> Reply-To: >> Date: Mon, 02 Feb 2009 11:36:35 +0400 >> To: >> Subject: Re: [Freeswitch-users] Strange Performance when using as SBC >> >> Hi Ken, >> >> 1) I'd like to use FS to hide topology, so bypass media is not possible >> 2) done >> 3) done >> 4) not used >> 5) i'm using this ins switch.xml -> > value="info"/>, if you think an other log level is more suitable. >> >> Regarding logging, I can see in console and in the freeswitch.log that >> there is still a lot of NOTICE logging, see below: >> 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:960 >> switch_core_session_thread() Session 8721 >> (sofia/internal/sipp at 10.10.10.1:5060) Ended >> 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:962 >> switch_core_session_thread() Close Channel >> sofia/internal/sipp at 10.10.10.1:5060 [CS_HANGUP] >> 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:960 >> switch_core_session_thread() Session 8722 >> (sofia/external/9998 at 10.10.20.100) Ended >> 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:962 >> switch_core_session_thread() Close Channel >> sofia/external/9998 at 10.10.20.100 [CS_HANGUP] >> 2009-02-02 08:33:56 [NOTICE] sofia.c:3164 sofia_handle_sip_i_state() >> Channel [sofia/external/9998 at 10.10.20.100] has been answered >> 2009-02-02 08:33:56 [WARNING] mod_sofia.c:740 sofia_read_frame() >> Changing codec ptime to 30. I bet you have a linksys/sipura =D >> >> Do you have any idea where I can switch off this kind of logging. I >> thought it should be in /dialplan/internal.xml, but I see that in >> internal.xml -> >> >> thanks a lot for your suggestion. >> >> regards, >> rod >> >> Ken Rice wrote: >> >>> Dont forget there are several things you can do to increase performance... >>> >>> 1) where possible use bypass media or media proxy modes >>> 2) mount freeswitch/db as a ram drive (if you are using voicemail with >>> the internal FS DBs you'll need a way to make this persistant across >>> reboots) >>> 3) see the wiki for setting reasonable ulimits >>> 4) (this is my oppinion others may vary) dont use mod_cdr_csv >>> 5) turn off (or reduce logging) in switch.conf.xml >>> >>> all of these thing can greatly improve performance. >>> >>> On Mon, Feb 2, 2009 at 1:04 AM, rod >> > wrote: >>> >>> Thanks Anthony, >>> >>> the setup is like this: >>> >>> sipp server ---- FS 1 ---- FS2 >>> >>> FS1 is the AMD CPU that has only one extension in dialplan that >>> bridges >>> 9999 to FS2. 9999 is the first extension in FS2 dialplan that >>> plays moh, >>> FS2 has no CPU pbm. >>> >>> FS1 is maxing out at 60 bridged calls without your option -hp. >>> >>> Using -hp, I'm now able to bridge 200 concurrent calls (a great >>> improvement) and the system is still reactive. CPU load is high >>> but not >>> 100% and as the system responds well, I think that doesn't matter. The >>> 2GB of memory are completely consumed (top command shows 700MB for FS >>> process). >>> >>> I understand that FS1 server is not the best hardware platform, >>> and I'm >>> waiting for new 4 cores server for testing. >>> I will update those numbers when testing with the new hardware. >>> >>> regards, >>> rod. >>> >>> Anthony Minessale wrote: >>> >>>> Which of the 2 machines has the load issue? You said it was one box >>>> calling the other. >>>> >>>> You have 2 major things against you, single CPU and AMD, but you >>>> should at least be able to get in the vicinity of 800-1000 calls >>>> >>> on a >>> >>>> box like that. >>>> >>>> Are you calling the default 9999? It's not really an appropriate >>>> extension for load testing. >>>> On the terminating box you should set up a manual extension that is >>>> the first one in the dial plan >>>> to play a wav file from preferably a ram disk or /tmp >>>> >>>> If you do plan on using this in production accept nothing less >>>> >>> than a >>> >>>> multi-core intel machine with at least 4 cores, the more cores the >>>> better because that parallel processing is where FS gets it's >>>> >>> atvantage. >>> >>>> >>>> On Fri, Jan 30, 2009 at 5:56 AM, rod >>> >>> >>> >>>> >> wrote: >>>> >>>> Dear list, >>>> >>>> I've been playing with freeswitch for some time (2 months) >>>> >>> and the >>> >>>> fact >>>> is that I'm very pleased with the functionnalities of this >>>> >>> software. >>> >>>> I'd like to use FS as a SBC handling media and I'm doing some >>>> tests with >>>> sipp to load the machine but I'm unable to bridge more than >>>> >>> 60 calls >>> >>>> without seeing the CPU being loaded at 100%. I'm sure >>>> >>> something is >>> >>>> going >>>> wrong with my setup but I'm unable to see what. >>>> >>>> The test machine has the following specs: >>>> Athlon XP 3500+ with 2GB of memory (I know this is not a >>>> >>> high end >>> >>>> machine :p) >>>> >>>> Freeswitch:/opt/freeswitch/log# cat /proc/cpuinfo >>>> processor : 0 >>>> vendor_id : AuthenticAMD >>>> cpu family : 15 >>>> model : 95 >>>> model name : AMD Athlon(tm) 64 Processor 3500+ >>>> stepping : 2 >>>> cpu MHz : 2199.973 >>>> cache size : 512 KB >>>> fpu : yes >>>> fpu_exception : yes >>>> cpuid level : 1 >>>> wp : yes >>>> flags : fpu vme de pse tsc msr pae mce cx8 apic >>>> >>> sep mtrr pge >>> >>>> mca cmov pat pse36 clflush mmx fxsr sse sse2 syscall nx mmxext >>>> fxsr_opt >>>> rdtscp lm 3dnowext 3dnow up rep_good pni cx16 lahf_lm svm >>>> >>> extapic >>> >>>> cr8_legacy >>>> bogomips : 4402.97 >>>> TLB size : 1024 4K pages >>>> clflush size : 64 >>>> cache_alignment : 64 >>>> address sizes : 40 bits physical, 48 bits virtual >>>> power management: ts fid vid ttp tm stc >>>> >>>> I installed FS on a fresh debian 64: >>>> Linux Freeswitch 2.6.26-1-amd64 #1 SMP Sat Jan 10 17:57:00 >>>> >>> UTC 2009 >>> >>>> x86_64 GNU/Linux >>>> >>>> I set the ulimit parameters like those on the website: >>>> freeswitch at internal> ... >>>> Freeswitch:/opt/free-svn/bin# ulimit -a >>>> core file size (blocks, -c) unlimited >>>> data seg size (kbytes, -d) unlimited >>>> scheduling priority (-e) 0 >>>> file size (blocks, -f) unlimited >>>> pending signals (-i) unlimited >>>> max locked memory (kbytes, -l) unlimited >>>> max memory size (kbytes, -m) unlimited >>>> open files (-n) 999999 >>>> pipe size (512 bytes, -p) 8 >>>> POSIX message queues (bytes, -q) unlimited >>>> real-time priority (-r) 0 >>>> stack size (kbytes, -s) 244 >>>> cpu time (seconds, -t) unlimited >>>> max user processes (-u) unlimited >>>> virtual memory (kbytes, -v) unlimited >>>> file locks (-x) unlimited >>>> >>>> >>>> My network setup is the following: >>>> >>>> SIPP machine (10.10.10.1/24)----------------vlan >>>> >>> >>> >>>> 55 >>>> ----------(10.10.10.254/24 >>>> >>> ) FS >>> >>>> (10.10.20.254/24)-------------- >>>> >>> >>> >>>> vlan56 >>>> -------------------(10.10.20.100/24 >>>> >>> ) >>> >>>> OTHER STOCK FS >>>> >>>> >>>> I launched sipp with: >>>> sipp -sn uac_pcap -s 9999 -r 10 -l 80 -d 60000 -mi >>>> >>> 10.10.10.1 -i >>> >>>> 10.10.10.1 -mp 25000 10.10.10.254:5060 >>>> >>> >>> >>>> The dialplan on FS is very simple: >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>> data="sofia/external/9999 at 10.10.20.100 >>>> >>> >> >"/> >>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> FreeSWITCH Version 1.0.trunk (11560M) Started. >>>> Crash Protection [Disabled] >>>> Max Sessions[1000] >>>> Session Rate[100] >>>> SQL [Enabled] >>>> >>>> >>>> The test is very simple: sipp dial 9999 that matches in my >>>> >>> FS dialplan >>> >>>> and this is bridged to an other FS machine playing music on >>>> >>> hold. >>> >>>> When I launch "top" I see after 30 to 40 s that FS consumes all >>>> the CPU >>>> ressources (with a mean of 50-60 % before), with 80 calls. >>>> When I set 70 calls, I have to wait 70-80 s before seeing >>>> >>> the same >>> >>>> issue. >>>> >>>> Presence is set to false on the 2 profile. >>>> >>>> I have the same issue with FS 1.0.2 that' s why I tried FS >>>> >>> 11560. >>> >>>> When I use the FS machine as a router to test the packet per >>>> >>> second >>> >>>> performance, I'm reaching 100Mbps with 8000pps in each >>>> >>> direction (from >>> >>>> vlan 55 to vlan56) with less than 12% CPU. So that I don't think >>>> there's >>>> an issue with the network. >>>> >>>> Here is an "mpstat -P ALL 1" to show you what's happening >>>> >>> suddenly >>> >>>> with >>>> 70 bridge calls: >>>> 12:31:26 CPU %user %nice %sys %iowait %irq %soft >>>> %steal %idle intr/s >>>> 12:31:27 all 3,00 0,00 3,00 0,00 1,00 4,00 >>>> 0,00 89,00 6241,00 >>>> 12:31:27 0 3,00 0,00 3,00 0,00 1,00 4,00 >>>> 0,00 89,00 6241,00 >>>> >>>> 12:31:27 CPU %user %nice %sys %iowait %irq %soft >>>> %steal %idle intr/s >>>> 12:31:28 all 14,14 0,00 56,57 0,00 2,02 5,05 >>>> 0,00 22,22 6035,35 >>>> 12:31:28 0 14,14 0,00 56,57 0,00 2,02 5,05 >>>> 0,00 22,22 6035,35 >>>> >>>> 12:31:28 CPU %user %nice %sys %iowait %irq %soft >>>> %steal %idle intr/s >>>> 12:31:29 all 24,75 0,00 67,33 0,00 0,99 6,93 >>>> 0,00 0,00 5483,17 >>>> 12:31:29 0 24,75 0,00 67,33 0,00 0,99 6,93 >>>> 0,00 0,00 5483,17 >>>> >>>> >>>> The CPU is going from 89% idle to 0% in less than 2 seconds. >>>> >>>> I know that I don't have to expect too much from this kind of >>>> hardware, >>>> but it seems strange that the CPU power vanished so suddenly. >>>> >>>> Thanks a lot for the guys that have read this long mail :p >>>> >>>> kind regards, >>>> rod >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> >>> >>> >>>> >>> >>> > >>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> >>>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >>>> http://www.freeswitch.org >>>> >>>> >>>> >>>> >>>> -- >>>> Anthony Minessale II >>>> >>>> FreeSWITCH http://www.freeswitch.org/ >>>> ClueCon http://www.cluecon.com/ >>>> >>>> AIM: anthm >>>> MSN:anthony_minessale at hotmail.com >>>> >>> >>> >>>> >>> >>> > >>> >>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>>> >>> >>> >>>> >>> >>> > >>> >>>> IRC: irc.freenode.net >>>> >>> #freeswitch >>> >>>> FreeSWITCH Developer Conference >>>> sip:888 at conference.freeswitch.org >>>> >>> >>> >>>> >>> >>> > >>> >>>> iax:guest at conference.freeswitch.org/888 >>>> >>> >>> >>>> >>>> googletalk:conf+888 at conference.freeswitch.org >>>> >>> >>> >>>> >>> >>> > >>> >>>> pstn:213-799-1400 >>>> >>>> >>> ------------------------------------------------------------------------ >>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> >>> >>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> >>>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >>>> http://www.freeswitch.org >>>> >>>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> ------------------------------------------------------------------------ >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > From saeedahmad1981 at gmail.com Mon Feb 2 02:21:06 2009 From: saeedahmad1981 at gmail.com (Saeed Ahmed) Date: Mon, 2 Feb 2009 11:21:06 +0100 Subject: [Freeswitch-users] Strange Performance when using as SBC In-Reply-To: <4986C42C.7030700@laposte.net> References: <4986C42C.7030700@laposte.net> Message-ID: <1C31435E19AB4E57981CF2AAC2CA74AA@SaeedLaptop> Hi Rod, Could you please share how you configured Sipp & FS to create a test environment? Especially the dial plan, sofia settings etc..., actually I am a newbie. I want to test it on a single FS machine. Kind Regards Saeed -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of rod Sent: Monday, February 02, 2009 11:00 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Strange Performance when using as SBC Hi Ken, Jay, thanks for pointing to proxy media, I will test. Ken, you are right, I was brain damaged (a stupid mistake) when setting INFO cause this kind of level could be very verbose. I'm switching to CRIT or ERR. Thanks guys, rod. thanks for Ken Rice wrote: > If you don't have to transcode, using proxy media mode will still save you > some CPU time. This is 1/2 way between bypass media and the default media > interactive mode. The other draw back to this mode is if you are using FS to > clean up RTP and DTMF you loose those functions but they are not needed in > most use cases. > > As far as the log level goes, I found that once I had things stable setting > the loglevel to helped a good deal... Info is probably a bit too high of a > loglevel I would probably go for CRIT or ERR (2 or 1 respectively) if you > insist on leaving logging turned on... On a busy system these can and will > generate a good deal of activity (and disk IO if using mod_logfile) > > Ken > > > >> From: rod >> Reply-To: >> Date: Mon, 02 Feb 2009 11:36:35 +0400 >> To: >> Subject: Re: [Freeswitch-users] Strange Performance when using as SBC >> >> Hi Ken, >> >> 1) I'd like to use FS to hide topology, so bypass media is not possible >> 2) done >> 3) done >> 4) not used >> 5) i'm using this ins switch.xml -> > value="info"/>, if you think an other log level is more suitable. >> >> Regarding logging, I can see in console and in the freeswitch.log that >> there is still a lot of NOTICE logging, see below: >> 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:960 >> switch_core_session_thread() Session 8721 >> (sofia/internal/sipp at 10.10.10.1:5060) Ended >> 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:962 >> switch_core_session_thread() Close Channel >> sofia/internal/sipp at 10.10.10.1:5060 [CS_HANGUP] >> 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:960 >> switch_core_session_thread() Session 8722 >> (sofia/external/9998 at 10.10.20.100) Ended >> 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:962 >> switch_core_session_thread() Close Channel >> sofia/external/9998 at 10.10.20.100 [CS_HANGUP] >> 2009-02-02 08:33:56 [NOTICE] sofia.c:3164 sofia_handle_sip_i_state() >> Channel [sofia/external/9998 at 10.10.20.100] has been answered >> 2009-02-02 08:33:56 [WARNING] mod_sofia.c:740 sofia_read_frame() >> Changing codec ptime to 30. I bet you have a linksys/sipura =D >> >> Do you have any idea where I can switch off this kind of logging. I >> thought it should be in /dialplan/internal.xml, but I see that in >> internal.xml -> >> >> thanks a lot for your suggestion. >> >> regards, >> rod >> >> Ken Rice wrote: >> >>> Dont forget there are several things you can do to increase performance... >>> >>> 1) where possible use bypass media or media proxy modes >>> 2) mount freeswitch/db as a ram drive (if you are using voicemail with >>> the internal FS DBs you'll need a way to make this persistant across >>> reboots) >>> 3) see the wiki for setting reasonable ulimits >>> 4) (this is my oppinion others may vary) dont use mod_cdr_csv >>> 5) turn off (or reduce logging) in switch.conf.xml >>> >>> all of these thing can greatly improve performance. >>> >>> On Mon, Feb 2, 2009 at 1:04 AM, rod >> > wrote: >>> >>> Thanks Anthony, >>> >>> the setup is like this: >>> >>> sipp server ---- FS 1 ---- FS2 >>> >>> FS1 is the AMD CPU that has only one extension in dialplan that >>> bridges >>> 9999 to FS2. 9999 is the first extension in FS2 dialplan that >>> plays moh, >>> FS2 has no CPU pbm. >>> >>> FS1 is maxing out at 60 bridged calls without your option -hp. >>> >>> Using -hp, I'm now able to bridge 200 concurrent calls (a great >>> improvement) and the system is still reactive. CPU load is high >>> but not >>> 100% and as the system responds well, I think that doesn't matter. The >>> 2GB of memory are completely consumed (top command shows 700MB for FS >>> process). >>> >>> I understand that FS1 server is not the best hardware platform, >>> and I'm >>> waiting for new 4 cores server for testing. >>> I will update those numbers when testing with the new hardware. >>> >>> regards, >>> rod. >>> >>> Anthony Minessale wrote: >>> >>>> Which of the 2 machines has the load issue? You said it was one box >>>> calling the other. >>>> >>>> You have 2 major things against you, single CPU and AMD, but you >>>> should at least be able to get in the vicinity of 800-1000 calls >>>> >>> on a >>> >>>> box like that. >>>> >>>> Are you calling the default 9999? It's not really an appropriate >>>> extension for load testing. >>>> On the terminating box you should set up a manual extension that is >>>> the first one in the dial plan >>>> to play a wav file from preferably a ram disk or /tmp >>>> >>>> If you do plan on using this in production accept nothing less >>>> >>> than a >>> >>>> multi-core intel machine with at least 4 cores, the more cores the >>>> better because that parallel processing is where FS gets it's >>>> >>> atvantage. >>> >>>> >>>> On Fri, Jan 30, 2009 at 5:56 AM, rod >>> >>> >>> >>>> >> wrote: >>>> >>>> Dear list, >>>> >>>> I've been playing with freeswitch for some time (2 months) >>>> >>> and the >>> >>>> fact >>>> is that I'm very pleased with the functionnalities of this >>>> >>> software. >>> >>>> I'd like to use FS as a SBC handling media and I'm doing some >>>> tests with >>>> sipp to load the machine but I'm unable to bridge more than >>>> >>> 60 calls >>> >>>> without seeing the CPU being loaded at 100%. I'm sure >>>> >>> something is >>> >>>> going >>>> wrong with my setup but I'm unable to see what. >>>> >>>> The test machine has the following specs: >>>> Athlon XP 3500+ with 2GB of memory (I know this is not a >>>> >>> high end >>> >>>> machine :p) >>>> >>>> Freeswitch:/opt/freeswitch/log# cat /proc/cpuinfo >>>> processor : 0 >>>> vendor_id : AuthenticAMD >>>> cpu family : 15 >>>> model : 95 >>>> model name : AMD Athlon(tm) 64 Processor 3500+ >>>> stepping : 2 >>>> cpu MHz : 2199.973 >>>> cache size : 512 KB >>>> fpu : yes >>>> fpu_exception : yes >>>> cpuid level : 1 >>>> wp : yes >>>> flags : fpu vme de pse tsc msr pae mce cx8 apic >>>> >>> sep mtrr pge >>> >>>> mca cmov pat pse36 clflush mmx fxsr sse sse2 syscall nx mmxext >>>> fxsr_opt >>>> rdtscp lm 3dnowext 3dnow up rep_good pni cx16 lahf_lm svm >>>> >>> extapic >>> >>>> cr8_legacy >>>> bogomips : 4402.97 >>>> TLB size : 1024 4K pages >>>> clflush size : 64 >>>> cache_alignment : 64 >>>> address sizes : 40 bits physical, 48 bits virtual >>>> power management: ts fid vid ttp tm stc >>>> >>>> I installed FS on a fresh debian 64: >>>> Linux Freeswitch 2.6.26-1-amd64 #1 SMP Sat Jan 10 17:57:00 >>>> >>> UTC 2009 >>> >>>> x86_64 GNU/Linux >>>> >>>> I set the ulimit parameters like those on the website: >>>> freeswitch at internal> ... >>>> Freeswitch:/opt/free-svn/bin# ulimit -a >>>> core file size (blocks, -c) unlimited >>>> data seg size (kbytes, -d) unlimited >>>> scheduling priority (-e) 0 >>>> file size (blocks, -f) unlimited >>>> pending signals (-i) unlimited >>>> max locked memory (kbytes, -l) unlimited >>>> max memory size (kbytes, -m) unlimited >>>> open files (-n) 999999 >>>> pipe size (512 bytes, -p) 8 >>>> POSIX message queues (bytes, -q) unlimited >>>> real-time priority (-r) 0 >>>> stack size (kbytes, -s) 244 >>>> cpu time (seconds, -t) unlimited >>>> max user processes (-u) unlimited >>>> virtual memory (kbytes, -v) unlimited >>>> file locks (-x) unlimited >>>> >>>> >>>> My network setup is the following: >>>> >>>> SIPP machine (10.10.10.1/24)----------------vlan >>>> >>> >>> >>>> 55 >>>> ----------(10.10.10.254/24 >>>> >>> ) FS >>> >>>> (10.10.20.254/24)-------------- >>>> >>> >>> >>>> vlan56 >>>> -------------------(10.10.20.100/24 >>>> >>> ) >>> >>>> OTHER STOCK FS >>>> >>>> >>>> I launched sipp with: >>>> sipp -sn uac_pcap -s 9999 -r 10 -l 80 -d 60000 -mi >>>> >>> 10.10.10.1 -i >>> >>>> 10.10.10.1 -mp 25000 10.10.10.254:5060 >>>> >>> >>> >>>> The dialplan on FS is very simple: >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>> data="sofia/external/9999 at 10.10.20.100 >>>> >>> >> >"/> >>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> FreeSWITCH Version 1.0.trunk (11560M) Started. >>>> Crash Protection [Disabled] >>>> Max Sessions[1000] >>>> Session Rate[100] >>>> SQL [Enabled] >>>> >>>> >>>> The test is very simple: sipp dial 9999 that matches in my >>>> >>> FS dialplan >>> >>>> and this is bridged to an other FS machine playing music on >>>> >>> hold. >>> >>>> When I launch "top" I see after 30 to 40 s that FS consumes all >>>> the CPU >>>> ressources (with a mean of 50-60 % before), with 80 calls. >>>> When I set 70 calls, I have to wait 70-80 s before seeing >>>> >>> the same >>> >>>> issue. >>>> >>>> Presence is set to false on the 2 profile. >>>> >>>> I have the same issue with FS 1.0.2 that' s why I tried FS >>>> >>> 11560. >>> >>>> When I use the FS machine as a router to test the packet per >>>> >>> second >>> >>>> performance, I'm reaching 100Mbps with 8000pps in each >>>> >>> direction (from >>> >>>> vlan 55 to vlan56) with less than 12% CPU. So that I don't think >>>> there's >>>> an issue with the network. >>>> >>>> Here is an "mpstat -P ALL 1" to show you what's happening >>>> >>> suddenly >>> >>>> with >>>> 70 bridge calls: >>>> 12:31:26 CPU %user %nice %sys %iowait %irq %soft >>>> %steal %idle intr/s >>>> 12:31:27 all 3,00 0,00 3,00 0,00 1,00 4,00 >>>> 0,00 89,00 6241,00 >>>> 12:31:27 0 3,00 0,00 3,00 0,00 1,00 4,00 >>>> 0,00 89,00 6241,00 >>>> >>>> 12:31:27 CPU %user %nice %sys %iowait %irq %soft >>>> %steal %idle intr/s >>>> 12:31:28 all 14,14 0,00 56,57 0,00 2,02 5,05 >>>> 0,00 22,22 6035,35 >>>> 12:31:28 0 14,14 0,00 56,57 0,00 2,02 5,05 >>>> 0,00 22,22 6035,35 >>>> >>>> 12:31:28 CPU %user %nice %sys %iowait %irq %soft >>>> %steal %idle intr/s >>>> 12:31:29 all 24,75 0,00 67,33 0,00 0,99 6,93 >>>> 0,00 0,00 5483,17 >>>> 12:31:29 0 24,75 0,00 67,33 0,00 0,99 6,93 >>>> 0,00 0,00 5483,17 >>>> >>>> >>>> The CPU is going from 89% idle to 0% in less than 2 seconds. >>>> >>>> I know that I don't have to expect too much from this kind of >>>> hardware, >>>> but it seems strange that the CPU power vanished so suddenly. >>>> >>>> Thanks a lot for the guys that have read this long mail :p >>>> >>>> kind regards, >>>> rod >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> >>> >>> >>>> >>> >>> > >>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> >>>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >>>> http://www.freeswitch.org >>>> >>>> >>>> >>>> >>>> -- >>>> Anthony Minessale II >>>> >>>> FreeSWITCH http://www.freeswitch.org/ >>>> ClueCon http://www.cluecon.com/ >>>> >>>> AIM: anthm >>>> MSN:anthony_minessale at hotmail.com >>>> >>> >>> >>>> >>> >>> > >>> >>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>>> >>> >>> >>>> >>> >>> > >>> >>>> IRC: irc.freenode.net >>>> >>> #freeswitch >>> >>>> FreeSWITCH Developer Conference >>>> sip:888 at conference.freeswitch.org >>>> >>> >>> >>>> >>> >>> > >>> >>>> iax:guest at conference.freeswitch.org/888 >>>> >>> >>> >>>> >>>> googletalk:conf+888 at conference.freeswitch.org >>>> >>> >>> >>>> >>> >>> > >>> >>>> pstn:213-799-1400 >>>> >>>> >>> ------------------------------------------------------------------------ >>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> >>> >>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> >>>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >>>> http://www.freeswitch.org >>>> >>>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> ------------------------------------------------------------------------ >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From sias at cpdata.co.za Mon Feb 2 02:24:59 2009 From: sias at cpdata.co.za (Sias Mey) Date: Mon, 2 Feb 2009 12:24:59 +0200 Subject: [Freeswitch-users] Conference dialing and uuid In-Reply-To: <0F863750-5C77-4CD9-ADC6-CFDDF2B149B4@freeswitch.org> References: <20090130103944.GA23536@cpdata.co.za> <20090130133315.GB23536@cpdata.co.za> <191c3a030901300605r78dd8943w7fb1075ef851e1d@mail.gmail.com> <20090202085157.GA3555@cpdata.co.za> <0F863750-5C77-4CD9-ADC6-CFDDF2B149B4@freeswitch.org> Message-ID: <20090202102459.GA4179@cpdata.co.za> Aaah ok. Thanks for clearing that up. So using loopback is still the only real workable sollution for me, since that generates ringback from and alternative endpoint and plays it into the conference. I might play with some javascript that streams ring into the channel eventually but for now the string comparisons at least get me the right uuid. Thank you again, Sias On Mon, Feb 02, 2009 at 02:55:42AM -0600, Brian West wrote: > You can't get ringback dialing out from a conference its not possible > as it is now. > > /b > > On Feb 2, 2009, at 2:51 AM, Sias Mey wrote: > > Yes ... yes indeed I can. > That works quite a bit better than generating 4 channels and getting > massively confused with what uuid does what... but now im stuck > without > ringback again :-(. > In my conference dial string I send: > {ringback=\'%(400,200,400,450)\',transfer_ringback=\'%(400,200,400,4 > 50)\', > .... [1]}sofia/internal/1001 at xxx.xxx.xxx.xxx > A dump of all the channel variables shows ringback is set to > %25(400,200,400,450)%3B%25(400,2200,400,450) > %25(400,200,400,450)%3B%25(400,2200,400,450) transfer_ringback> > This seems ok to me but I still dont get any ringback. > Thanks again for answering all the anoying questions from the same > guy > :-P, > Sias > > References > > 1. mailto:}sofia/internal/1001 at xxx.xxx.xxx.xxx > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Mon Feb 2 02:29:25 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 2 Feb 2009 04:29:25 -0600 Subject: [Freeswitch-users] Conference dialing and uuid In-Reply-To: <20090202102459.GA4179@cpdata.co.za> References: <20090130103944.GA23536@cpdata.co.za> <20090130133315.GB23536@cpdata.co.za> <191c3a030901300605r78dd8943w7fb1075ef851e1d@mail.gmail.com> <20090202085157.GA3555@cpdata.co.za> <0F863750-5C77-4CD9-ADC6-CFDDF2B149B4@freeswitch.org> <20090202102459.GA4179@cpdata.co.za> Message-ID: <9FE1733C-A46A-4A17-8AEE-077DE8010235@freeswitch.org> Loopback will not work in that case either. If the far end plays ringback inband you should hear that if you use the conference dial api call. /b On Feb 2, 2009, at 4:24 AM, Sias Mey wrote: > Aaah ok. > > Thanks for clearing that up. > > So using loopback is still the only real workable sollution for me, > since that generates ringback from and alternative endpoint and > plays it > into the conference. > > I might play with some javascript that streams ring into the channel > eventually but for now the string comparisons at least get me the > right > uuid. > > Thank you again, > Sias From leon at scarlet-internet.nl Mon Feb 2 03:42:05 2009 From: leon at scarlet-internet.nl (Leon de Rooij) Date: Mon, 2 Feb 2009 12:42:05 +0100 Subject: [Freeswitch-users] LDAP Integration In-Reply-To: <4983C258.6080705@skopis.com> References: <49417123.10709@ydeasolutions.com.br> <49417538.9040203@ydeasolutions.com.br> <200812120842.00808.hads@nice.net.nz> <49418790.60001@ydeasolutions.com.br> <87f2f3b90812111241q3b16b307lbf4d1251c7d8aad7@mail.gmail.com> <494198F3.10806@ydeasolutions.com.br> <4947A61F.6060806@ydeasolutions.com.br> <49485AD1.5070708@skopis.com> <4949025F.9040008@ydeasolutions.com.br> <49621532.5080003@ydeasolutions.com.br> <4962D64D.3080809@skopis.com> <2DCF79D8-E2B5-43EB-93D1-EED92506E8DF@scarlet-internet.nl> <4983C258.6080705@skopis.com> Message-ID: <87556643-1B94-40AE-B84C-871AEA593593@scarlet-internet.nl> On Jan 31, 2009, at 4:15 AM, John Skopis (Lists) wrote: > Leon de Rooij wrote: >> Hi John, >> >> I've been trying to get your mod_xml_ldap module running, but didn't >> get very far yet.. >> >> What is the official way to get the module built ? >> > > The official way to build all fs modules is to uncomment the entry in > modules.conf. > > If you want to build a specific module there are targets > > make mod_name-clean > make mod_name-install Thanks, I'll try that. > as for mod_xml_ldap, I really do not feel that it is as quality as I > would expect a production quality module to be. I understand, it's just that I'm very interested in it as we're using ldap everywhere over here. >> I tried modifying trunk/freeswitch.spec so that >> >> XML_INT_MODULES contains xml_int/mod_xml_ldap >> >> There's also a directories/mod_ldap in DISABLED_MODULES in the same >> file, but I don't suppose it's necessary to enable it, or is it ? >> > > mod_ldap is a separate module, implementing the directory interface, > not > to be confused with the "directory", which is queried for user + > domain > configuration (e.g., conf/directory/default.xml). > > perhaps it should be renamed to mod_dbi? > >> The mod_xml_ldap doesn't get built by running make make or dpkg- >> buildpackage from trunk/ >> >> Also I tried building it from the module directory itself, but then I >> get the following error: >> >> fsbuilder at sv:~/trunk/src/mod/xml_int/mod_xml_ldap$ make >> Compiling mod_xml_ldap.c... >> cc1: warnings being treated as errors >> mod_xml_ldap.c: In function 'xml_ldap_search': >> mod_xml_ldap.c:356: warning: cast from pointer to integer of >> different >> size >> make[1]: *** [mod_xml_ldap.o] Error 1 >> make: *** [all] Error 1 >> > > > > I have been working on a new module called mod_entity that works off a > simple description of an xml entitiy (domain, user, extension, > condition, action, anti-action currently) querying a db backend via > the > directory interface for fields used to build the entity. It still > needs > a bit of work but I am hoping to get a patch together this weekend. I > will post it to the freeswitch-dev list asking for comments. > > Off the top of my head at least the wishlist TODO is: > > implement connection pooling for mod_directory > > implement a cache either as a module used by an xml_int mod or in > switch_xml to cache a switch_xml_t > > >> (Also I had to apt-get install libsasl2 libsasl2-dev, otherwise make >> from this dir errored with missing sasl/sasl.h) >> >> Can you see what I'm doing wrong ? >> >> (I'm using svn rev 11560) >> >> thanks & regards, >> >> Leon >> >> On Jan 6, 2009, at 4:55 AM, John Skopis (Lists) wrote: >> >>> Vinicius Kobashi wrote: >>>> hi ppl. >>>> >>>> i tried hard to make it work, but still i couldnt find a complete >>>> openldap scheme that provides these information, and i still >>>> could't >>>> find out where to put these configuration... >>>> >>>> can anyone help me? >>>> >>>> thankz! >>>> >>>> vinicius escreveu: >>>>> thankz! >>>>> >>>>> ill set my openldap to provide these information.. >>>>> >>>>> but these about these binding settings... where should i set them? >>>>> >>>>> best regards >>>>> >>>>> John Skopis (Lists) wrote: >>>>>> vinicius wrote: >>>>>> >>>>>>> hi ppl.. i tried to find something at google, but i couldnt >>>>>>> manage to find >>>>>>> anything. >>>>>>> i still dont know what to do to make the mod_xml_ldap work. >>>>>>> i couldnt find information about how to build a config file for >>>>>>> the >>>>>>> module, and where to store it... >>>>>>> >>>>>>> can anyone give me a help? >>>>>>> >>>>>>> >>>>>> Be advised mod_xml_ldap is probably not production quality and >>>>>> will >>>>>> undoubtedly change, eventually at least. >>>>>> >>>>>> Here is what I used once: >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>> bindings="configuration"/> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> which should/probably/might work with ldap objects like these: >>>>>> >>>>>> dn: cn=John Skopis,ou=people,dc=example >>>>>> objectClass: person >>>>>> objectClass: inetOrgPerson >>>>>> objectClass: organizationalPerson >>>>>> objectClass: FreeSWITCH-Exten-Object >>>>>> objectClass: top >>>>>> cn: John Skopis >>>>>> sn: Skopis >>>>>> givenName: John >>>>>> FSid: 1001 >>>>>> FSmailbox: 1001 >>>>>> FSpassword: 1234 >>>>>> FSvm-password: 1001 >>>>>> FSemail-addr: john+fs at skopis.com >>>>>> FSvm-email-all-messages: TRUE >>>>>> FSvm-delete-file: TRUE >>>>>> FSvm-attach-file: TRUE >>>>>> >>>>>> dn: SIPIdentityUserName=1001,ou=h350,dc=example >>>>>> objectClass: person >>>>>> objectClass: SIPIdentity >>>>>> objectClass: top >>>>>> cn: 1001 >>>>>> sn: 1001 >>>>>> SIPIdentitySIPURI: sip:1001 at 172.16.75.129 >>>>>> SIPIdentityRegistrarAddress: 172.16.75.128 >>>>>> SIPIdentityProxyAddress: 172.16.75.128 >>>>>> SIPIdentityPassword: 1234 >>>>>> SIPIdentityUserName: 1001 >>>>>> SIPIdentityServiceLevel: premium >>>>>> >>>>>> >>> Again, the module is not production quality. Hopefully I will >>> conjurer >>> the time and know-how to put something decent together eventually. >>> >>> To load configuration for any fs module you need to define the XML >>> configuration element under the section "configuration". >>> >>> A good starting point is the file >>> $PREFIX/conf/freeswitch.xml >>> >>> http://wiki.freeswitch.org/wiki/Freeswitch.xml >>> >>> Also take a look at $PREFIX/logs/freeswitch.xml.fsxml >>> >>> to load mod_xml_ldap you would need to add something like this to >>> modules.conf.xml >>> >>> >>> >>> and create an xml_ldap.conf.xml in >>> $PREFIX/autoload_configs/xml_ldap.conf.xml >>> >>> >>> ... >>> >>> >>> The ITU is doing some work called h.350: >>> http://www.itu.int/ITU-T/studygroups/com16/h350/index.html >>> >>> Here is what I was working with: >>> attributetype ( 1.3.6.1.4.1.65535.2.1.1 NAME 'FSid' >>> DESC 'FreeSWITCH Extension ID' >>> EQUALITY caseIgnoreIA5Match >>> SYNTAX 1.3.6.1.4.1.1466.115.121.1.26 ) >>> >>> attributetype ( 1.3.6.1.4.1.65535.2.1.2 NAME 'FSmailbox' >>> DESC 'FreeSWITCH Extension Mailbox' >>> EQUALITY caseIgnoreIA5Match >>> SYNTAX 1.3.6.1.4.1.1466.115.121.1.26 ) >>> >>> attributetype ( 1.3.6.1.4.1.65535.2.1.3 NAME 'FSpassword' >>> DESC 'FreeSWITCH Password' >>> EQUALITY caseExactIA5Match >>> SYNTAX 1.3.6.1.4.1.1466.115.121.1.26 >>> SINGLE-VALUE ) >>> >>> attributetype ( 1.3.6.1.4.1.65535.2.1.4 NAME 'FSa1hash' >>> DESC 'FreeSWITCH Crypted Password' >>> EQUALITY caseExactIA5Match >>> SYNTAX 1.3.6.1.4.1.1466.115.121.1.26 >>> SINGLE-VALUE ) >>> >>> attributetype ( 1.3.6.1.4.1.65535.2.1.5 NAME 'FSvm-password' >>> DESC 'FreeSWITCH VoiceMail Password' >>> EQUALITY integerMatch >>> SYNTAX 1.3.6.1.4.1.1466.115.121.1.27 >>> SINGLE-VALUE ) >>> >>> attributetype ( 1.3.6.1.4.1.65535.2.1.6 NAME 'FSemail-addr' >>> DESC 'E-mail address to send voicemail' >>> EQUALITY caseIgnoreIA5Match >>> SYNTAX 1.3.6.1.4.1.1466.115.121.1.26 ) >>> >>> attributetype ( 1.3.6.1.4.1.65535.2.1.7 NAME 'FSvm-email-all- >>> messages' >>> DESC 'FreeSWITCH Email All Mesages' >>> EQUALITY booleanMatch >>> SYNTAX 1.3.6.1.4.1.1466.115.121.1.7 >>> SINGLE-VALUE ) >>> >>> attributetype ( 1.3.6.1.4.1.65535.2.1.8 NAME 'FSvm-delete-file' >>> DESC 'FreeSWITCH VoiceMail Delete File' >>> EQUALITY booleanMatch >>> SYNTAX 1.3.6.1.4.1.1466.115.121.1.7 >>> SINGLE-VALUE ) >>> >>> attributetype ( 1.3.6.1.4.1.65535.2.1.9 NAME 'FSvm-attach-file' >>> DESC 'FreeSWITCH VoiceMail Attach file' >>> EQUALITY booleanMatch >>> SYNTAX 1.3.6.1.4.1.1466.115.121.1.7 >>> SINGLE-VALUE ) >>> >>> >>> >>> >>> >>> objectclass ( 1.3.6.1.4.1.65535.2.2.1 NAME 'FreeSWITCH-Exten-Object' >>> SUP top AUXILIARY >>> DESC '%obj_desc%' >>> MUST ( FSid $ FSpassword ) >>> MAY ( FSmailbox $ FSa1hash $ FSvm-password $ FSemail-addr $ >>> FSvm-email-all-messages $ FSvm-delete-file $ FSvm-attach-file ) ) >>> >>> hth >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From pmhshz at gmail.com Mon Feb 2 04:07:57 2009 From: pmhshz at gmail.com (shehzad p) Date: Mon, 2 Feb 2009 04:07:57 -0800 (PST) Subject: [Freeswitch-users] Call Variable not available when call hangup Message-ID: <21788550.post@talk.nabble.com> Hi all, I need to process some CDR variables in Dialplan, like call duration, Answered time etc. but when I place info application after bridge, it is not listing them properly as below: =========================================== Caller-Channel-Created-Time: [1233573341672157] Caller-Channel-Answered-Time: [1233573342712939] Caller-Channel-Hangup-Time: [0] ========================================== Here Hangup time is 0, So how can I find actual values? --I know that we can use xml_cdr or cdr_csv, but my current need is to get those values from dialplan itself so that can be passed to some script... thanks, msp -- View this message in context: http://www.nabble.com/Call-Variable-not-available-when-call-hangup-tp21788550p21788550.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From krice at freeswitch.org Mon Feb 2 04:11:28 2009 From: krice at freeswitch.org (Ken Rice) Date: Mon, 02 Feb 2009 06:11:28 -0600 Subject: [Freeswitch-users] Call Variable not available when call hangup In-Reply-To: <21788550.post@talk.nabble.com> Message-ID: As I told you on IRC, the call is not completed at that stage... So there is no hangup time... You must post process the call or figure out your own start answer and stop times.... > From: shehzad p > Reply-To: > Date: Mon, 2 Feb 2009 04:07:57 -0800 (PST) > To: > Subject: [Freeswitch-users] Call Variable not available when call hangup > > > Hi all, > > I need to process some CDR variables in Dialplan, like call duration, > Answered time etc. > but when I place info application after bridge, it is not listing them > properly as below: > =========================================== > Caller-Channel-Created-Time: [1233573341672157] > Caller-Channel-Answered-Time: [1233573342712939] > Caller-Channel-Hangup-Time: [0] > ========================================== > Here Hangup time is 0, So how can I find actual values? > > --I know that we can use xml_cdr or cdr_csv, but my current need is to get > those values from dialplan itself so that can be passed to some script... > > > thanks, > msp > -- > View this message in context: > http://www.nabble.com/Call-Variable-not-available-when-call-hangup-tp21788550p > 21788550.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From kawarod at laposte.net Mon Feb 2 04:36:44 2009 From: kawarod at laposte.net (rod) Date: Mon, 02 Feb 2009 16:36:44 +0400 Subject: [Freeswitch-users] Strange Performance when using as SBC In-Reply-To: <4986C42C.7030700@laposte.net> References: <4986C42C.7030700@laposte.net> Message-ID: <4986E8DC.20707@laposte.net> Some update: - I switched log level to "err" - I'm now using proxy-media - and I erased the directive answer in the dialplan (useless and seems that it consumes lots of CPU, don't know why) the dialplan now looks like this: instead of The box is now able to bridge 300 calls with 20-30% of free CPU. I will run a long term test. I see this error in the log and don't understand exactly if somebody could help (I'm running latest trunk 11592M): 2009-02-02 13:29:54 [ERR] switch_core_io.c:117 switch_core_session_read_frame() sofia/external/9998 at 10.10.20.100 has no read codec. regards, rodrigue rod wrote: > Hi Ken, Jay, > > thanks for pointing to proxy media, I will test. > > Ken, you are right, I was brain damaged (a stupid mistake) when setting > INFO cause this kind of level could be very verbose. I'm switching to > CRIT or ERR. > > Thanks guys, > rod. > > thanks for > > Ken Rice wrote: > >> If you don't have to transcode, using proxy media mode will still save you >> some CPU time. This is 1/2 way between bypass media and the default media >> interactive mode. The other draw back to this mode is if you are using FS to >> clean up RTP and DTMF you loose those functions but they are not needed in >> most use cases. >> >> As far as the log level goes, I found that once I had things stable setting >> the loglevel to helped a good deal... Info is probably a bit too high of a >> loglevel I would probably go for CRIT or ERR (2 or 1 respectively) if you >> insist on leaving logging turned on... On a busy system these can and will >> generate a good deal of activity (and disk IO if using mod_logfile) >> >> Ken >> >> >> >> >>> From: rod >>> Reply-To: >>> Date: Mon, 02 Feb 2009 11:36:35 +0400 >>> To: >>> Subject: Re: [Freeswitch-users] Strange Performance when using as SBC >>> >>> Hi Ken, >>> >>> 1) I'd like to use FS to hide topology, so bypass media is not possible >>> 2) done >>> 3) done >>> 4) not used >>> 5) i'm using this ins switch.xml -> >> value="info"/>, if you think an other log level is more suitable. >>> >>> Regarding logging, I can see in console and in the freeswitch.log that >>> there is still a lot of NOTICE logging, see below: >>> 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:960 >>> switch_core_session_thread() Session 8721 >>> (sofia/internal/sipp at 10.10.10.1:5060) Ended >>> 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:962 >>> switch_core_session_thread() Close Channel >>> sofia/internal/sipp at 10.10.10.1:5060 [CS_HANGUP] >>> 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:960 >>> switch_core_session_thread() Session 8722 >>> (sofia/external/9998 at 10.10.20.100) Ended >>> 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:962 >>> switch_core_session_thread() Close Channel >>> sofia/external/9998 at 10.10.20.100 [CS_HANGUP] >>> 2009-02-02 08:33:56 [NOTICE] sofia.c:3164 sofia_handle_sip_i_state() >>> Channel [sofia/external/9998 at 10.10.20.100] has been answered >>> 2009-02-02 08:33:56 [WARNING] mod_sofia.c:740 sofia_read_frame() >>> Changing codec ptime to 30. I bet you have a linksys/sipura =D >>> >>> Do you have any idea where I can switch off this kind of logging. I >>> thought it should be in /dialplan/internal.xml, but I see that in >>> internal.xml -> >>> >>> thanks a lot for your suggestion. >>> >>> regards, >>> rod >>> >>> Ken Rice wrote: >>> >>> >>>> Dont forget there are several things you can do to increase performance... >>>> >>>> 1) where possible use bypass media or media proxy modes >>>> 2) mount freeswitch/db as a ram drive (if you are using voicemail with >>>> the internal FS DBs you'll need a way to make this persistant across >>>> reboots) >>>> 3) see the wiki for setting reasonable ulimits >>>> 4) (this is my oppinion others may vary) dont use mod_cdr_csv >>>> 5) turn off (or reduce logging) in switch.conf.xml >>>> >>>> all of these thing can greatly improve performance. >>>> >>>> On Mon, Feb 2, 2009 at 1:04 AM, rod >>> > wrote: >>>> >>>> Thanks Anthony, >>>> >>>> the setup is like this: >>>> >>>> sipp server ---- FS 1 ---- FS2 >>>> >>>> FS1 is the AMD CPU that has only one extension in dialplan that >>>> bridges >>>> 9999 to FS2. 9999 is the first extension in FS2 dialplan that >>>> plays moh, >>>> FS2 has no CPU pbm. >>>> >>>> FS1 is maxing out at 60 bridged calls without your option -hp. >>>> >>>> Using -hp, I'm now able to bridge 200 concurrent calls (a great >>>> improvement) and the system is still reactive. CPU load is high >>>> but not >>>> 100% and as the system responds well, I think that doesn't matter. The >>>> 2GB of memory are completely consumed (top command shows 700MB for FS >>>> process). >>>> >>>> I understand that FS1 server is not the best hardware platform, >>>> and I'm >>>> waiting for new 4 cores server for testing. >>>> I will update those numbers when testing with the new hardware. >>>> >>>> regards, >>>> rod. >>>> >>>> Anthony Minessale wrote: >>>> >>>> >>>>> Which of the 2 machines has the load issue? You said it was one box >>>>> calling the other. >>>>> >>>>> You have 2 major things against you, single CPU and AMD, but you >>>>> should at least be able to get in the vicinity of 800-1000 calls >>>>> >>>>> >>>> on a >>>> >>>> >>>>> box like that. >>>>> >>>>> Are you calling the default 9999? It's not really an appropriate >>>>> extension for load testing. >>>>> On the terminating box you should set up a manual extension that is >>>>> the first one in the dial plan >>>>> to play a wav file from preferably a ram disk or /tmp >>>>> >>>>> If you do plan on using this in production accept nothing less >>>>> >>>>> >>>> than a >>>> >>>> >>>>> multi-core intel machine with at least 4 cores, the more cores the >>>>> better because that parallel processing is where FS gets it's >>>>> >>>>> >>>> atvantage. >>>> >>>> >>>>> On Fri, Jan 30, 2009 at 5:56 AM, rod >>>> >>>>> >>>> >>>> >>>> >>>>> >> wrote: >>>>> >>>>> Dear list, >>>>> >>>>> I've been playing with freeswitch for some time (2 months) >>>>> >>>>> >>>> and the >>>> >>>> >>>>> fact >>>>> is that I'm very pleased with the functionnalities of this >>>>> >>>>> >>>> software. >>>> >>>> >>>>> I'd like to use FS as a SBC handling media and I'm doing some >>>>> tests with >>>>> sipp to load the machine but I'm unable to bridge more than >>>>> >>>>> >>>> 60 calls >>>> >>>> >>>>> without seeing the CPU being loaded at 100%. I'm sure >>>>> >>>>> >>>> something is >>>> >>>> >>>>> going >>>>> wrong with my setup but I'm unable to see what. >>>>> >>>>> The test machine has the following specs: >>>>> Athlon XP 3500+ with 2GB of memory (I know this is not a >>>>> >>>>> >>>> high end >>>> >>>> >>>>> machine :p) >>>>> >>>>> Freeswitch:/opt/freeswitch/log# cat /proc/cpuinfo >>>>> processor : 0 >>>>> vendor_id : AuthenticAMD >>>>> cpu family : 15 >>>>> model : 95 >>>>> model name : AMD Athlon(tm) 64 Processor 3500+ >>>>> stepping : 2 >>>>> cpu MHz : 2199.973 >>>>> cache size : 512 KB >>>>> fpu : yes >>>>> fpu_exception : yes >>>>> cpuid level : 1 >>>>> wp : yes >>>>> flags : fpu vme de pse tsc msr pae mce cx8 apic >>>>> >>>>> >>>> sep mtrr pge >>>> >>>> >>>>> mca cmov pat pse36 clflush mmx fxsr sse sse2 syscall nx mmxext >>>>> fxsr_opt >>>>> rdtscp lm 3dnowext 3dnow up rep_good pni cx16 lahf_lm svm >>>>> >>>>> >>>> extapic >>>> >>>> >>>>> cr8_legacy >>>>> bogomips : 4402.97 >>>>> TLB size : 1024 4K pages >>>>> clflush size : 64 >>>>> cache_alignment : 64 >>>>> address sizes : 40 bits physical, 48 bits virtual >>>>> power management: ts fid vid ttp tm stc >>>>> >>>>> I installed FS on a fresh debian 64: >>>>> Linux Freeswitch 2.6.26-1-amd64 #1 SMP Sat Jan 10 17:57:00 >>>>> >>>>> >>>> UTC 2009 >>>> >>>> >>>>> x86_64 GNU/Linux >>>>> >>>>> I set the ulimit parameters like those on the website: >>>>> freeswitch at internal> ... >>>>> Freeswitch:/opt/free-svn/bin# ulimit -a >>>>> core file size (blocks, -c) unlimited >>>>> data seg size (kbytes, -d) unlimited >>>>> scheduling priority (-e) 0 >>>>> file size (blocks, -f) unlimited >>>>> pending signals (-i) unlimited >>>>> max locked memory (kbytes, -l) unlimited >>>>> max memory size (kbytes, -m) unlimited >>>>> open files (-n) 999999 >>>>> pipe size (512 bytes, -p) 8 >>>>> POSIX message queues (bytes, -q) unlimited >>>>> real-time priority (-r) 0 >>>>> stack size (kbytes, -s) 244 >>>>> cpu time (seconds, -t) unlimited >>>>> max user processes (-u) unlimited >>>>> virtual memory (kbytes, -v) unlimited >>>>> file locks (-x) unlimited >>>>> >>>>> >>>>> My network setup is the following: >>>>> >>>>> SIPP machine (10.10.10.1/24)----------------vlan >>>>> >>>>> >>>> >>>> >>>> >>>>> 55 >>>>> ----------(10.10.10.254/24 >>>>> >>>>> >>>> ) FS >>>> >>>> >>>>> (10.10.20.254/24)-------------- >>>>> >>>>> >>>> >>>> >>>> >>>>> vlan56 >>>>> -------------------(10.10.20.100/24 >>>>> >>>>> >>>> ) >>>> >>>> >>>>> OTHER STOCK FS >>>>> >>>>> >>>>> I launched sipp with: >>>>> sipp -sn uac_pcap -s 9999 -r 10 -l 80 -d 60000 -mi >>>>> >>>>> >>>> 10.10.10.1 -i >>>> >>>> >>>>> 10.10.10.1 -mp 25000 10.10.10.254:5060 >>>>> >>>>> >>>> >>>> >>>> >>>>> The dialplan on FS is very simple: >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>> data="sofia/external/9999 at 10.10.20.100 >>>>> >>>>> >>>> >>> >"/> >>>> >>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> FreeSWITCH Version 1.0.trunk (11560M) Started. >>>>> Crash Protection [Disabled] >>>>> Max Sessions[1000] >>>>> Session Rate[100] >>>>> SQL [Enabled] >>>>> >>>>> >>>>> The test is very simple: sipp dial 9999 that matches in my >>>>> >>>>> >>>> FS dialplan >>>> >>>> >>>>> and this is bridged to an other FS machine playing music on >>>>> >>>>> >>>> hold. >>>> >>>> >>>>> When I launch "top" I see after 30 to 40 s that FS consumes all >>>>> the CPU >>>>> ressources (with a mean of 50-60 % before), with 80 calls. >>>>> When I set 70 calls, I have to wait 70-80 s before seeing >>>>> >>>>> >>>> the same >>>> >>>> >>>>> issue. >>>>> >>>>> Presence is set to false on the 2 profile. >>>>> >>>>> I have the same issue with FS 1.0.2 that' s why I tried FS >>>>> >>>>> >>>> 11560. >>>> >>>> >>>>> When I use the FS machine as a router to test the packet per >>>>> >>>>> >>>> second >>>> >>>> >>>>> performance, I'm reaching 100Mbps with 8000pps in each >>>>> >>>>> >>>> direction (from >>>> >>>> >>>>> vlan 55 to vlan56) with less than 12% CPU. So that I don't think >>>>> there's >>>>> an issue with the network. >>>>> >>>>> Here is an "mpstat -P ALL 1" to show you what's happening >>>>> >>>>> >>>> suddenly >>>> >>>> >>>>> with >>>>> 70 bridge calls: >>>>> 12:31:26 CPU %user %nice %sys %iowait %irq %soft >>>>> %steal %idle intr/s >>>>> 12:31:27 all 3,00 0,00 3,00 0,00 1,00 4,00 >>>>> 0,00 89,00 6241,00 >>>>> 12:31:27 0 3,00 0,00 3,00 0,00 1,00 4,00 >>>>> 0,00 89,00 6241,00 >>>>> >>>>> 12:31:27 CPU %user %nice %sys %iowait %irq %soft >>>>> %steal %idle intr/s >>>>> 12:31:28 all 14,14 0,00 56,57 0,00 2,02 5,05 >>>>> 0,00 22,22 6035,35 >>>>> 12:31:28 0 14,14 0,00 56,57 0,00 2,02 5,05 >>>>> 0,00 22,22 6035,35 >>>>> >>>>> 12:31:28 CPU %user %nice %sys %iowait %irq %soft >>>>> %steal %idle intr/s >>>>> 12:31:29 all 24,75 0,00 67,33 0,00 0,99 6,93 >>>>> 0,00 0,00 5483,17 >>>>> 12:31:29 0 24,75 0,00 67,33 0,00 0,99 6,93 >>>>> 0,00 0,00 5483,17 >>>>> >>>>> >>>>> The CPU is going from 89% idle to 0% in less than 2 seconds. >>>>> >>>>> I know that I don't have to expect too much from this kind of >>>>> hardware, >>>>> but it seems strange that the CPU power vanished so suddenly. >>>>> >>>>> Thanks a lot for the guys that have read this long mail :p >>>>> >>>>> kind regards, >>>>> rod >>>>> >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> >>>>> >>>> >>>>> >>>> > >>>> >>>> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> >>>>> >>>>> >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> >>>> >>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> >>>>> >>>>> -- >>>>> Anthony Minessale II >>>>> >>>>> FreeSWITCH http://www.freeswitch.org/ >>>>> ClueCon http://www.cluecon.com/ >>>>> >>>>> AIM: anthm >>>>> MSN:anthony_minessale at hotmail.com >>>>> >>>>> >>>> >>>> >>>> >>>>> >>>> >>>>> >>>> > >>>> >>>> >>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>>>> >>>>> >>>> >>>> >>>> >>>>> >>>> >>>>> >>>> > >>>> >>>> >>>>> IRC: irc.freenode.net >>>>> >>>>> >>>> #freeswitch >>>> >>>> >>>>> FreeSWITCH Developer Conference >>>>> sip:888 at conference.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> >>>>> >>>> >>>>> >>>> > >>>> >>>> >>>>> iax:guest at conference.freeswitch.org/888 >>>>> >>>>> >>>> >>>> >>>> >>>>> >>>>> googletalk:conf+888 at conference.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> >>>>> >>>> >>>>> >>>> > >>>> >>>> >>>>> pstn:213-799-1400 >>>>> >>>>> >>>>> >>>> ------------------------------------------------------------------------ >>>> >>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> >>>>> >>>>> >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> >>>> >>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> ------------------------------------------------------------------------ >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > From pmhshz at gmail.com Mon Feb 2 04:53:16 2009 From: pmhshz at gmail.com (shehzad p) Date: Mon, 2 Feb 2009 04:53:16 -0800 (PST) Subject: [Freeswitch-users] Call Variable not available when call hangup In-Reply-To: <21788550.post@talk.nabble.com> References: <21788550.post@talk.nabble.com> Message-ID: <21789152.post@talk.nabble.com> Is there any settings that when call hangup control can be transferred to another context and these CDR values can be accessible there? (just like in Asterisk, h extension) shehzad p wrote: > > Hi all, > > I need to process some CDR variables in Dialplan, like call duration, > Answered time etc. > but when I place info application after bridge, it is not listing them > properly as below: > =========================================== > Caller-Channel-Created-Time: [1233573341672157] > Caller-Channel-Answered-Time: [1233573342712939] > Caller-Channel-Hangup-Time: [0] > ========================================== > Here Hangup time is 0, So how can I find actual values? > > --I know that we can use xml_cdr or cdr_csv, but my current need is to get > those values from dialplan itself so that can be passed to some script... > > > thanks, > msp > -- View this message in context: http://www.nabble.com/Call-Variable-not-available-when-call-hangup-tp21788550p21789152.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From kawarod at laposte.net Mon Feb 2 04:52:57 2009 From: kawarod at laposte.net (rod) Date: Mon, 02 Feb 2009 16:52:57 +0400 Subject: [Freeswitch-users] Strange Performance when using as SBC In-Reply-To: <1C31435E19AB4E57981CF2AAC2CA74AA@SaeedLaptop> References: <4986C42C.7030700@laposte.net> <1C31435E19AB4E57981CF2AAC2CA74AA@SaeedLaptop> Message-ID: <4986ECA9.3040707@laposte.net> Hi Saeed, I just created an account to share my setup on the wiki. I will detail all the steps for a clean install of a debian64 lenny with FS used as a SBC (next step is to try the new LCR module :) )and what I'm doing do stress the server. I wrote nothing at this time so please be patient, I'm waiting for my new hardware so that I will detail as much as possible what I'll do. For beginning I suggest you reading the start page on the wiki, especially these pages: -http://wiki.freeswitch.org/wiki/Getting_Started_Guide -http://wiki.freeswitch.org/wiki/Dialplan_XML maybe you could tell more about the linux distribution you're using so that I can give you some pointers for sipp... regards. rod. Saeed Ahmed wrote: > Hi Rod, > > Could you please share how you configured Sipp & FS to create a test > environment? Especially the dial plan, sofia settings etc..., actually I am > a newbie. I want to test it on a single FS machine. > > Kind Regards > Saeed > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of rod > Sent: Monday, February 02, 2009 11:00 AM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Strange Performance when using as SBC > > Hi Ken, Jay, > > thanks for pointing to proxy media, I will test. > > Ken, you are right, I was brain damaged (a stupid mistake) when setting > INFO cause this kind of level could be very verbose. I'm switching to > CRIT or ERR. > > Thanks guys, > rod. > > thanks for > > Ken Rice wrote: > >> If you don't have to transcode, using proxy media mode will still save you >> some CPU time. This is 1/2 way between bypass media and the default media >> interactive mode. The other draw back to this mode is if you are using FS >> > to > >> clean up RTP and DTMF you loose those functions but they are not needed in >> most use cases. >> >> As far as the log level goes, I found that once I had things stable >> > setting > >> the loglevel to helped a good deal... Info is probably a bit too high of a >> loglevel I would probably go for CRIT or ERR (2 or 1 respectively) if you >> insist on leaving logging turned on... On a busy system these can and will >> generate a good deal of activity (and disk IO if using mod_logfile) >> >> Ken >> >> >> >> >>> From: rod >>> Reply-To: >>> Date: Mon, 02 Feb 2009 11:36:35 +0400 >>> To: >>> Subject: Re: [Freeswitch-users] Strange Performance when using as SBC >>> >>> Hi Ken, >>> >>> 1) I'd like to use FS to hide topology, so bypass media is not possible >>> 2) done >>> 3) done >>> 4) not used >>> 5) i'm using this ins switch.xml -> >> value="info"/>, if you think an other log level is more suitable. >>> >>> Regarding logging, I can see in console and in the freeswitch.log that >>> there is still a lot of NOTICE logging, see below: >>> 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:960 >>> switch_core_session_thread() Session 8721 >>> (sofia/internal/sipp at 10.10.10.1:5060) Ended >>> 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:962 >>> switch_core_session_thread() Close Channel >>> sofia/internal/sipp at 10.10.10.1:5060 [CS_HANGUP] >>> 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:960 >>> switch_core_session_thread() Session 8722 >>> (sofia/external/9998 at 10.10.20.100) Ended >>> 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:962 >>> switch_core_session_thread() Close Channel >>> sofia/external/9998 at 10.10.20.100 [CS_HANGUP] >>> 2009-02-02 08:33:56 [NOTICE] sofia.c:3164 sofia_handle_sip_i_state() >>> Channel [sofia/external/9998 at 10.10.20.100] has been answered >>> 2009-02-02 08:33:56 [WARNING] mod_sofia.c:740 sofia_read_frame() >>> Changing codec ptime to 30. I bet you have a linksys/sipura =D >>> >>> Do you have any idea where I can switch off this kind of logging. I >>> thought it should be in /dialplan/internal.xml, but I see that in >>> internal.xml -> >>> >>> thanks a lot for your suggestion. >>> >>> regards, >>> rod >>> >>> Ken Rice wrote: >>> >>> >>>> Dont forget there are several things you can do to increase >>>> > performance... > >>>> 1) where possible use bypass media or media proxy modes >>>> 2) mount freeswitch/db as a ram drive (if you are using voicemail with >>>> the internal FS DBs you'll need a way to make this persistant across >>>> reboots) >>>> 3) see the wiki for setting reasonable ulimits >>>> 4) (this is my oppinion others may vary) dont use mod_cdr_csv >>>> 5) turn off (or reduce logging) in switch.conf.xml >>>> >>>> all of these thing can greatly improve performance. >>>> >>>> On Mon, Feb 2, 2009 at 1:04 AM, rod >>> > wrote: >>>> >>>> Thanks Anthony, >>>> >>>> the setup is like this: >>>> >>>> sipp server ---- FS 1 ---- FS2 >>>> >>>> FS1 is the AMD CPU that has only one extension in dialplan that >>>> bridges >>>> 9999 to FS2. 9999 is the first extension in FS2 dialplan that >>>> plays moh, >>>> FS2 has no CPU pbm. >>>> >>>> FS1 is maxing out at 60 bridged calls without your option -hp. >>>> >>>> Using -hp, I'm now able to bridge 200 concurrent calls (a great >>>> improvement) and the system is still reactive. CPU load is high >>>> but not >>>> 100% and as the system responds well, I think that doesn't matter. >>>> > The > >>>> 2GB of memory are completely consumed (top command shows 700MB for >>>> > FS > >>>> process). >>>> >>>> I understand that FS1 server is not the best hardware platform, >>>> and I'm >>>> waiting for new 4 cores server for testing. >>>> I will update those numbers when testing with the new hardware. >>>> >>>> regards, >>>> rod. >>>> >>>> Anthony Minessale wrote: >>>> >>>> >>>>> Which of the 2 machines has the load issue? You said it was one box >>>>> calling the other. >>>>> >>>>> You have 2 major things against you, single CPU and AMD, but you >>>>> should at least be able to get in the vicinity of 800-1000 calls >>>>> >>>>> >>>> on a >>>> >>>> >>>>> box like that. >>>>> >>>>> Are you calling the default 9999? It's not really an appropriate >>>>> extension for load testing. >>>>> On the terminating box you should set up a manual extension that is >>>>> the first one in the dial plan >>>>> to play a wav file from preferably a ram disk or /tmp >>>>> >>>>> If you do plan on using this in production accept nothing less >>>>> >>>>> >>>> than a >>>> >>>> >>>>> multi-core intel machine with at least 4 cores, the more cores the >>>>> better because that parallel processing is where FS gets it's >>>>> >>>>> >>>> atvantage. >>>> >>>> >>>>> On Fri, Jan 30, 2009 at 5:56 AM, rod >>>> >>>>> >>>> >>>> >>>> >>>>> >> wrote: >>>>> >>>>> Dear list, >>>>> >>>>> I've been playing with freeswitch for some time (2 months) >>>>> >>>>> >>>> and the >>>> >>>> >>>>> fact >>>>> is that I'm very pleased with the functionnalities of this >>>>> >>>>> >>>> software. >>>> >>>> >>>>> I'd like to use FS as a SBC handling media and I'm doing some >>>>> tests with >>>>> sipp to load the machine but I'm unable to bridge more than >>>>> >>>>> >>>> 60 calls >>>> >>>> >>>>> without seeing the CPU being loaded at 100%. I'm sure >>>>> >>>>> >>>> something is >>>> >>>> >>>>> going >>>>> wrong with my setup but I'm unable to see what. >>>>> >>>>> The test machine has the following specs: >>>>> Athlon XP 3500+ with 2GB of memory (I know this is not a >>>>> >>>>> >>>> high end >>>> >>>> >>>>> machine :p) >>>>> >>>>> Freeswitch:/opt/freeswitch/log# cat /proc/cpuinfo >>>>> processor : 0 >>>>> vendor_id : AuthenticAMD >>>>> cpu family : 15 >>>>> model : 95 >>>>> model name : AMD Athlon(tm) 64 Processor 3500+ >>>>> stepping : 2 >>>>> cpu MHz : 2199.973 >>>>> cache size : 512 KB >>>>> fpu : yes >>>>> fpu_exception : yes >>>>> cpuid level : 1 >>>>> wp : yes >>>>> flags : fpu vme de pse tsc msr pae mce cx8 apic >>>>> >>>>> >>>> sep mtrr pge >>>> >>>> >>>>> mca cmov pat pse36 clflush mmx fxsr sse sse2 syscall nx mmxext >>>>> fxsr_opt >>>>> rdtscp lm 3dnowext 3dnow up rep_good pni cx16 lahf_lm svm >>>>> >>>>> >>>> extapic >>>> >>>> >>>>> cr8_legacy >>>>> bogomips : 4402.97 >>>>> TLB size : 1024 4K pages >>>>> clflush size : 64 >>>>> cache_alignment : 64 >>>>> address sizes : 40 bits physical, 48 bits virtual >>>>> power management: ts fid vid ttp tm stc >>>>> >>>>> I installed FS on a fresh debian 64: >>>>> Linux Freeswitch 2.6.26-1-amd64 #1 SMP Sat Jan 10 17:57:00 >>>>> >>>>> >>>> UTC 2009 >>>> >>>> >>>>> x86_64 GNU/Linux >>>>> >>>>> I set the ulimit parameters like those on the website: >>>>> freeswitch at internal> ... >>>>> Freeswitch:/opt/free-svn/bin# ulimit -a >>>>> core file size (blocks, -c) unlimited >>>>> data seg size (kbytes, -d) unlimited >>>>> scheduling priority (-e) 0 >>>>> file size (blocks, -f) unlimited >>>>> pending signals (-i) unlimited >>>>> max locked memory (kbytes, -l) unlimited >>>>> max memory size (kbytes, -m) unlimited >>>>> open files (-n) 999999 >>>>> pipe size (512 bytes, -p) 8 >>>>> POSIX message queues (bytes, -q) unlimited >>>>> real-time priority (-r) 0 >>>>> stack size (kbytes, -s) 244 >>>>> cpu time (seconds, -t) unlimited >>>>> max user processes (-u) unlimited >>>>> virtual memory (kbytes, -v) unlimited >>>>> file locks (-x) unlimited >>>>> >>>>> >>>>> My network setup is the following: >>>>> >>>>> SIPP machine (10.10.10.1/24)----------------vlan >>>>> >>>>> >>>> >>>> >>>> >>>>> 55 >>>>> ----------(10.10.10.254/24 >>>>> >>>>> >>>> ) FS >>>> >>>> >>>>> (10.10.20.254/24)-------------- >>>>> >>>>> >>>> >>>> >>>> >>>>> vlan56 >>>>> -------------------(10.10.20.100/24 >>>>> >>>>> >>>> ) >>>> >>>> >>>>> OTHER STOCK FS >>>>> >>>>> >>>>> I launched sipp with: >>>>> sipp -sn uac_pcap -s 9999 -r 10 -l 80 -d 60000 -mi >>>>> >>>>> >>>> 10.10.10.1 -i >>>> >>>> >>>>> 10.10.10.1 -mp 25000 10.10.10.254:5060 >>>>> >>>>> >>>> >>>> >>>> >>>>> The dialplan on FS is very simple: >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>> data="sofia/external/9999 at 10.10.20.100 >>>>> >>>>> >>>> >>> >"/> >>>> >>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> FreeSWITCH Version 1.0.trunk (11560M) Started. >>>>> Crash Protection [Disabled] >>>>> Max Sessions[1000] >>>>> Session Rate[100] >>>>> SQL [Enabled] >>>>> >>>>> >>>>> The test is very simple: sipp dial 9999 that matches in my >>>>> >>>>> >>>> FS dialplan >>>> >>>> >>>>> and this is bridged to an other FS machine playing music on >>>>> >>>>> >>>> hold. >>>> >>>> >>>>> When I launch "top" I see after 30 to 40 s that FS consumes all >>>>> the CPU >>>>> ressources (with a mean of 50-60 % before), with 80 calls. >>>>> When I set 70 calls, I have to wait 70-80 s before seeing >>>>> >>>>> >>>> the same >>>> >>>> >>>>> issue. >>>>> >>>>> Presence is set to false on the 2 profile. >>>>> >>>>> I have the same issue with FS 1.0.2 that' s why I tried FS >>>>> >>>>> >>>> 11560. >>>> >>>> >>>>> When I use the FS machine as a router to test the packet per >>>>> >>>>> >>>> second >>>> >>>> >>>>> performance, I'm reaching 100Mbps with 8000pps in each >>>>> >>>>> >>>> direction (from >>>> >>>> >>>>> vlan 55 to vlan56) with less than 12% CPU. So that I don't think >>>>> there's >>>>> an issue with the network. >>>>> >>>>> Here is an "mpstat -P ALL 1" to show you what's happening >>>>> >>>>> >>>> suddenly >>>> >>>> >>>>> with >>>>> 70 bridge calls: >>>>> 12:31:26 CPU %user %nice %sys %iowait %irq %soft >>>>> %steal %idle intr/s >>>>> 12:31:27 all 3,00 0,00 3,00 0,00 1,00 4,00 >>>>> 0,00 89,00 6241,00 >>>>> 12:31:27 0 3,00 0,00 3,00 0,00 1,00 4,00 >>>>> 0,00 89,00 6241,00 >>>>> >>>>> 12:31:27 CPU %user %nice %sys %iowait %irq %soft >>>>> %steal %idle intr/s >>>>> 12:31:28 all 14,14 0,00 56,57 0,00 2,02 5,05 >>>>> 0,00 22,22 6035,35 >>>>> 12:31:28 0 14,14 0,00 56,57 0,00 2,02 5,05 >>>>> 0,00 22,22 6035,35 >>>>> >>>>> 12:31:28 CPU %user %nice %sys %iowait %irq %soft >>>>> %steal %idle intr/s >>>>> 12:31:29 all 24,75 0,00 67,33 0,00 0,99 6,93 >>>>> 0,00 0,00 5483,17 >>>>> 12:31:29 0 24,75 0,00 67,33 0,00 0,99 6,93 >>>>> 0,00 0,00 5483,17 >>>>> >>>>> >>>>> The CPU is going from 89% idle to 0% in less than 2 seconds. >>>>> >>>>> I know that I don't have to expect too much from this kind of >>>>> hardware, >>>>> but it seems strange that the CPU power vanished so suddenly. >>>>> >>>>> Thanks a lot for the guys that have read this long mail :p >>>>> >>>>> kind regards, >>>>> rod >>>>> >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> >>>>> >>>> >>>>> >>>> > >>>> >>>> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> >>>>> >>>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >>>> >>>> >>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> >>>>> >>>>> -- >>>>> Anthony Minessale II >>>>> >>>>> FreeSWITCH http://www.freeswitch.org/ >>>>> ClueCon http://www.cluecon.com/ >>>>> >>>>> AIM: anthm >>>>> MSN:anthony_minessale at hotmail.com >>>>> >>>>> >>>> >>>> >>>> >>>>> >>>> >>>>> >>>> > >>>> >>>> >>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>>>> >>>>> >>>> >>>> >>>> >>>>> >>>> >>>>> >>>> > >>>> >>>> >>>>> IRC: irc.freenode.net >>>>> >>>>> >>>> #freeswitch >>>> >>>> >>>>> FreeSWITCH Developer Conference >>>>> sip:888 at conference.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> >>>>> >>>> >>>>> >>>> > >>>> >>>> >>>>> iax:guest at conference.freeswitch.org/888 >>>>> >>>>> >>>> >>>> >>>> >>>>> >>>>> googletalk:conf+888 at conference.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> >>>>> >>>> >>>>> >>>> > >>>> >>>> >>>>> pstn:213-799-1400 >>>>> >>>>> >>>>> > ------------------------------------------------------------------------ > >>>> >>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> >>>>> >>>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >>>> >>>> >>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> >>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >>>> http://www.freeswitch.org >>>> >>>> >>>> ------------------------------------------------------------------------ >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > From palletboy at gmail.com Mon Feb 2 05:02:58 2009 From: palletboy at gmail.com (J. G.) Date: Mon, 2 Feb 2009 08:02:58 -0500 Subject: [Freeswitch-users] Phonebooth? Message-ID: <3093591d0902020502s6f726ba8h819402da0705a76a@mail.gmail.com> I got a group email from Anders Brownworth this weekend regarding him donating Phonebooth to the FreePBX project? Wonder what the impact will be.. -- ----- Jason Gehman General Manager North Voice Communications www.NorthVC.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090202/7cc6cf1f/attachment.html From saeedahmad1981 at gmail.com Mon Feb 2 05:03:55 2009 From: saeedahmad1981 at gmail.com (Saeed Ahmed) Date: Mon, 2 Feb 2009 14:03:55 +0100 Subject: [Freeswitch-users] Strange Performance when using as SBC In-Reply-To: <4986ECA9.3040707@laposte.net> References: <4986C42C.7030700@laposte.net><1C31435E19AB4E57981CF2AAC2CA74AA@SaeedLaptop> <4986ECA9.3040707@laposte.net> Message-ID: <9232C06D5362494791AF713E1DF61343@SaeedLaptop> Thanks rod for a quick answer, FS is installed on Ubuntu Server. I am planning to replace Nextone SBC with FS, Later I'll also use openZAP to communicate with TDM but this all depends how much calls it can take, or maybe we can also do something in clustering environment ( I am not sure about it). But thanks again and any further help will be highly appreciated. Kind Regards Saeed Ahmed Tariq -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of rod Sent: Monday, February 02, 2009 1:53 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Strange Performance when using as SBC Hi Saeed, I just created an account to share my setup on the wiki. I will detail all the steps for a clean install of a debian64 lenny with FS used as a SBC (next step is to try the new LCR module :) )and what I'm doing do stress the server. I wrote nothing at this time so please be patient, I'm waiting for my new hardware so that I will detail as much as possible what I'll do. For beginning I suggest you reading the start page on the wiki, especially these pages: -http://wiki.freeswitch.org/wiki/Getting_Started_Guide -http://wiki.freeswitch.org/wiki/Dialplan_XML maybe you could tell more about the linux distribution you're using so that I can give you some pointers for sipp... regards. rod. Saeed Ahmed wrote: > Hi Rod, > > Could you please share how you configured Sipp & FS to create a test > environment? Especially the dial plan, sofia settings etc..., actually I am > a newbie. I want to test it on a single FS machine. > > Kind Regards > Saeed > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of rod > Sent: Monday, February 02, 2009 11:00 AM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Strange Performance when using as SBC > > Hi Ken, Jay, > > thanks for pointing to proxy media, I will test. > > Ken, you are right, I was brain damaged (a stupid mistake) when setting > INFO cause this kind of level could be very verbose. I'm switching to > CRIT or ERR. > > Thanks guys, > rod. > > thanks for > > Ken Rice wrote: > >> If you don't have to transcode, using proxy media mode will still save you >> some CPU time. This is 1/2 way between bypass media and the default media >> interactive mode. The other draw back to this mode is if you are using FS >> > to > >> clean up RTP and DTMF you loose those functions but they are not needed in >> most use cases. >> >> As far as the log level goes, I found that once I had things stable >> > setting > >> the loglevel to helped a good deal... Info is probably a bit too high of a >> loglevel I would probably go for CRIT or ERR (2 or 1 respectively) if you >> insist on leaving logging turned on... On a busy system these can and will >> generate a good deal of activity (and disk IO if using mod_logfile) >> >> Ken >> >> >> >> >>> From: rod >>> Reply-To: >>> Date: Mon, 02 Feb 2009 11:36:35 +0400 >>> To: >>> Subject: Re: [Freeswitch-users] Strange Performance when using as SBC >>> >>> Hi Ken, >>> >>> 1) I'd like to use FS to hide topology, so bypass media is not possible >>> 2) done >>> 3) done >>> 4) not used >>> 5) i'm using this ins switch.xml -> >> value="info"/>, if you think an other log level is more suitable. >>> >>> Regarding logging, I can see in console and in the freeswitch.log that >>> there is still a lot of NOTICE logging, see below: >>> 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:960 >>> switch_core_session_thread() Session 8721 >>> (sofia/internal/sipp at 10.10.10.1:5060) Ended >>> 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:962 >>> switch_core_session_thread() Close Channel >>> sofia/internal/sipp at 10.10.10.1:5060 [CS_HANGUP] >>> 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:960 >>> switch_core_session_thread() Session 8722 >>> (sofia/external/9998 at 10.10.20.100) Ended >>> 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:962 >>> switch_core_session_thread() Close Channel >>> sofia/external/9998 at 10.10.20.100 [CS_HANGUP] >>> 2009-02-02 08:33:56 [NOTICE] sofia.c:3164 sofia_handle_sip_i_state() >>> Channel [sofia/external/9998 at 10.10.20.100] has been answered >>> 2009-02-02 08:33:56 [WARNING] mod_sofia.c:740 sofia_read_frame() >>> Changing codec ptime to 30. I bet you have a linksys/sipura =D >>> >>> Do you have any idea where I can switch off this kind of logging. I >>> thought it should be in /dialplan/internal.xml, but I see that in >>> internal.xml -> >>> >>> thanks a lot for your suggestion. >>> >>> regards, >>> rod >>> >>> Ken Rice wrote: >>> >>> >>>> Dont forget there are several things you can do to increase >>>> > performance... > >>>> 1) where possible use bypass media or media proxy modes >>>> 2) mount freeswitch/db as a ram drive (if you are using voicemail with >>>> the internal FS DBs you'll need a way to make this persistant across >>>> reboots) >>>> 3) see the wiki for setting reasonable ulimits >>>> 4) (this is my oppinion others may vary) dont use mod_cdr_csv >>>> 5) turn off (or reduce logging) in switch.conf.xml >>>> >>>> all of these thing can greatly improve performance. >>>> >>>> On Mon, Feb 2, 2009 at 1:04 AM, rod >>> > wrote: >>>> >>>> Thanks Anthony, >>>> >>>> the setup is like this: >>>> >>>> sipp server ---- FS 1 ---- FS2 >>>> >>>> FS1 is the AMD CPU that has only one extension in dialplan that >>>> bridges >>>> 9999 to FS2. 9999 is the first extension in FS2 dialplan that >>>> plays moh, >>>> FS2 has no CPU pbm. >>>> >>>> FS1 is maxing out at 60 bridged calls without your option -hp. >>>> >>>> Using -hp, I'm now able to bridge 200 concurrent calls (a great >>>> improvement) and the system is still reactive. CPU load is high >>>> but not >>>> 100% and as the system responds well, I think that doesn't matter. >>>> > The > >>>> 2GB of memory are completely consumed (top command shows 700MB for >>>> > FS > >>>> process). >>>> >>>> I understand that FS1 server is not the best hardware platform, >>>> and I'm >>>> waiting for new 4 cores server for testing. >>>> I will update those numbers when testing with the new hardware. >>>> >>>> regards, >>>> rod. >>>> >>>> Anthony Minessale wrote: >>>> >>>> >>>>> Which of the 2 machines has the load issue? You said it was one box >>>>> calling the other. >>>>> >>>>> You have 2 major things against you, single CPU and AMD, but you >>>>> should at least be able to get in the vicinity of 800-1000 calls >>>>> >>>>> >>>> on a >>>> >>>> >>>>> box like that. >>>>> >>>>> Are you calling the default 9999? It's not really an appropriate >>>>> extension for load testing. >>>>> On the terminating box you should set up a manual extension that is >>>>> the first one in the dial plan >>>>> to play a wav file from preferably a ram disk or /tmp >>>>> >>>>> If you do plan on using this in production accept nothing less >>>>> >>>>> >>>> than a >>>> >>>> >>>>> multi-core intel machine with at least 4 cores, the more cores the >>>>> better because that parallel processing is where FS gets it's >>>>> >>>>> >>>> atvantage. >>>> >>>> >>>>> On Fri, Jan 30, 2009 at 5:56 AM, rod >>>> >>>>> >>>> >>>> >>>> >>>>> >> wrote: >>>>> >>>>> Dear list, >>>>> >>>>> I've been playing with freeswitch for some time (2 months) >>>>> >>>>> >>>> and the >>>> >>>> >>>>> fact >>>>> is that I'm very pleased with the functionnalities of this >>>>> >>>>> >>>> software. >>>> >>>> >>>>> I'd like to use FS as a SBC handling media and I'm doing some >>>>> tests with >>>>> sipp to load the machine but I'm unable to bridge more than >>>>> >>>>> >>>> 60 calls >>>> >>>> >>>>> without seeing the CPU being loaded at 100%. I'm sure >>>>> >>>>> >>>> something is >>>> >>>> >>>>> going >>>>> wrong with my setup but I'm unable to see what. >>>>> >>>>> The test machine has the following specs: >>>>> Athlon XP 3500+ with 2GB of memory (I know this is not a >>>>> >>>>> >>>> high end >>>> >>>> >>>>> machine :p) >>>>> >>>>> Freeswitch:/opt/freeswitch/log# cat /proc/cpuinfo >>>>> processor : 0 >>>>> vendor_id : AuthenticAMD >>>>> cpu family : 15 >>>>> model : 95 >>>>> model name : AMD Athlon(tm) 64 Processor 3500+ >>>>> stepping : 2 >>>>> cpu MHz : 2199.973 >>>>> cache size : 512 KB >>>>> fpu : yes >>>>> fpu_exception : yes >>>>> cpuid level : 1 >>>>> wp : yes >>>>> flags : fpu vme de pse tsc msr pae mce cx8 apic >>>>> >>>>> >>>> sep mtrr pge >>>> >>>> >>>>> mca cmov pat pse36 clflush mmx fxsr sse sse2 syscall nx mmxext >>>>> fxsr_opt >>>>> rdtscp lm 3dnowext 3dnow up rep_good pni cx16 lahf_lm svm >>>>> >>>>> >>>> extapic >>>> >>>> >>>>> cr8_legacy >>>>> bogomips : 4402.97 >>>>> TLB size : 1024 4K pages >>>>> clflush size : 64 >>>>> cache_alignment : 64 >>>>> address sizes : 40 bits physical, 48 bits virtual >>>>> power management: ts fid vid ttp tm stc >>>>> >>>>> I installed FS on a fresh debian 64: >>>>> Linux Freeswitch 2.6.26-1-amd64 #1 SMP Sat Jan 10 17:57:00 >>>>> >>>>> >>>> UTC 2009 >>>> >>>> >>>>> x86_64 GNU/Linux >>>>> >>>>> I set the ulimit parameters like those on the website: >>>>> freeswitch at internal> ... >>>>> Freeswitch:/opt/free-svn/bin# ulimit -a >>>>> core file size (blocks, -c) unlimited >>>>> data seg size (kbytes, -d) unlimited >>>>> scheduling priority (-e) 0 >>>>> file size (blocks, -f) unlimited >>>>> pending signals (-i) unlimited >>>>> max locked memory (kbytes, -l) unlimited >>>>> max memory size (kbytes, -m) unlimited >>>>> open files (-n) 999999 >>>>> pipe size (512 bytes, -p) 8 >>>>> POSIX message queues (bytes, -q) unlimited >>>>> real-time priority (-r) 0 >>>>> stack size (kbytes, -s) 244 >>>>> cpu time (seconds, -t) unlimited >>>>> max user processes (-u) unlimited >>>>> virtual memory (kbytes, -v) unlimited >>>>> file locks (-x) unlimited >>>>> >>>>> >>>>> My network setup is the following: >>>>> >>>>> SIPP machine (10.10.10.1/24)----------------vlan >>>>> >>>>> >>>> >>>> >>>> >>>>> 55 >>>>> ----------(10.10.10.254/24 >>>>> >>>>> >>>> ) FS >>>> >>>> >>>>> (10.10.20.254/24)-------------- >>>>> >>>>> >>>> >>>> >>>> >>>>> vlan56 >>>>> -------------------(10.10.20.100/24 >>>>> >>>>> >>>> ) >>>> >>>> >>>>> OTHER STOCK FS >>>>> >>>>> >>>>> I launched sipp with: >>>>> sipp -sn uac_pcap -s 9999 -r 10 -l 80 -d 60000 -mi >>>>> >>>>> >>>> 10.10.10.1 -i >>>> >>>> >>>>> 10.10.10.1 -mp 25000 10.10.10.254:5060 >>>>> >>>>> >>>> >>>> >>>> >>>>> The dialplan on FS is very simple: >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>> data="sofia/external/9999 at 10.10.20.100 >>>>> >>>>> >>>> >>> >"/> >>>> >>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> FreeSWITCH Version 1.0.trunk (11560M) Started. >>>>> Crash Protection [Disabled] >>>>> Max Sessions[1000] >>>>> Session Rate[100] >>>>> SQL [Enabled] >>>>> >>>>> >>>>> The test is very simple: sipp dial 9999 that matches in my >>>>> >>>>> >>>> FS dialplan >>>> >>>> >>>>> and this is bridged to an other FS machine playing music on >>>>> >>>>> >>>> hold. >>>> >>>> >>>>> When I launch "top" I see after 30 to 40 s that FS consumes all >>>>> the CPU >>>>> ressources (with a mean of 50-60 % before), with 80 calls. >>>>> When I set 70 calls, I have to wait 70-80 s before seeing >>>>> >>>>> >>>> the same >>>> >>>> >>>>> issue. >>>>> >>>>> Presence is set to false on the 2 profile. >>>>> >>>>> I have the same issue with FS 1.0.2 that' s why I tried FS >>>>> >>>>> >>>> 11560. >>>> >>>> >>>>> When I use the FS machine as a router to test the packet per >>>>> >>>>> >>>> second >>>> >>>> >>>>> performance, I'm reaching 100Mbps with 8000pps in each >>>>> >>>>> >>>> direction (from >>>> >>>> >>>>> vlan 55 to vlan56) with less than 12% CPU. So that I don't think >>>>> there's >>>>> an issue with the network. >>>>> >>>>> Here is an "mpstat -P ALL 1" to show you what's happening >>>>> >>>>> >>>> suddenly >>>> >>>> >>>>> with >>>>> 70 bridge calls: >>>>> 12:31:26 CPU %user %nice %sys %iowait %irq %soft >>>>> %steal %idle intr/s >>>>> 12:31:27 all 3,00 0,00 3,00 0,00 1,00 4,00 >>>>> 0,00 89,00 6241,00 >>>>> 12:31:27 0 3,00 0,00 3,00 0,00 1,00 4,00 >>>>> 0,00 89,00 6241,00 >>>>> >>>>> 12:31:27 CPU %user %nice %sys %iowait %irq %soft >>>>> %steal %idle intr/s >>>>> 12:31:28 all 14,14 0,00 56,57 0,00 2,02 5,05 >>>>> 0,00 22,22 6035,35 >>>>> 12:31:28 0 14,14 0,00 56,57 0,00 2,02 5,05 >>>>> 0,00 22,22 6035,35 >>>>> >>>>> 12:31:28 CPU %user %nice %sys %iowait %irq %soft >>>>> %steal %idle intr/s >>>>> 12:31:29 all 24,75 0,00 67,33 0,00 0,99 6,93 >>>>> 0,00 0,00 5483,17 >>>>> 12:31:29 0 24,75 0,00 67,33 0,00 0,99 6,93 >>>>> 0,00 0,00 5483,17 >>>>> >>>>> >>>>> The CPU is going from 89% idle to 0% in less than 2 seconds. >>>>> >>>>> I know that I don't have to expect too much from this kind of >>>>> hardware, >>>>> but it seems strange that the CPU power vanished so suddenly. >>>>> >>>>> Thanks a lot for the guys that have read this long mail :p >>>>> >>>>> kind regards, >>>>> rod >>>>> >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> >>>>> >>>> >>>>> >>>> > >>>> >>>> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> >>>>> >>>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >>>> >>>> >>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> >>>>> >>>>> -- >>>>> Anthony Minessale II >>>>> >>>>> FreeSWITCH http://www.freeswitch.org/ >>>>> ClueCon http://www.cluecon.com/ >>>>> >>>>> AIM: anthm >>>>> MSN:anthony_minessale at hotmail.com >>>>> >>>>> >>>> >>>> >>>> >>>>> >>>> >>>>> >>>> > >>>> >>>> >>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>>>> >>>>> >>>> >>>> >>>> >>>>> >>>> >>>>> >>>> > >>>> >>>> >>>>> IRC: irc.freenode.net >>>>> >>>>> >>>> #freeswitch >>>> >>>> >>>>> FreeSWITCH Developer Conference >>>>> sip:888 at conference.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> >>>>> >>>> >>>>> >>>> > >>>> >>>> >>>>> iax:guest at conference.freeswitch.org/888 >>>>> >>>>> >>>> >>>> >>>> >>>>> >>>>> googletalk:conf+888 at conference.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> >>>>> >>>> >>>>> >>>> > >>>> >>>> >>>>> pstn:213-799-1400 >>>>> >>>>> >>>>> > ------------------------------------------------------------------------ > >>>> >>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> >>>>> >>>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >>>> >>>> >>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> >>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >>>> http://www.freeswitch.org >>>> >>>> >>>> ------------------------------------------------------------------------ >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From anthony.minessale at gmail.com Mon Feb 2 05:57:43 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 2 Feb 2009 07:57:43 -0600 Subject: [Freeswitch-users] Strange Performance when using as SBC In-Reply-To: <9232C06D5362494791AF713E1DF61343@SaeedLaptop> References: <4986C42C.7030700@laposte.net> <1C31435E19AB4E57981CF2AAC2CA74AA@SaeedLaptop> <4986ECA9.3040707@laposte.net> <9232C06D5362494791AF713E1DF61343@SaeedLaptop> Message-ID: <191c3a030902020557v70d3777fqb83e591385a64a59@mail.gmail.com> if you want to use ubuntu, be sure to use hardy and not intrepid. On Mon, Feb 2, 2009 at 7:03 AM, Saeed Ahmed wrote: > Thanks rod for a quick answer, > > FS is installed on Ubuntu Server. > > I am planning to replace Nextone SBC with FS, Later I'll also use openZAP > to > communicate with TDM but this all depends how much calls it can take, or > maybe we can also do something in clustering environment ( I am not sure > about it). But thanks again and any further help will be highly > appreciated. > > > Kind Regards > Saeed Ahmed Tariq > > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of rod > Sent: Monday, February 02, 2009 1:53 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Strange Performance when using as SBC > > Hi Saeed, > > I just created an account to share my setup on the wiki. I will detail > all the steps for a clean install of a debian64 lenny with FS used as a > SBC (next step is to try the new LCR module :) )and what I'm doing do > stress the server. > > I wrote nothing at this time so please be patient, I'm waiting for my > new hardware so that I will detail as much as possible what I'll do. > > For beginning I suggest you reading the start page on the wiki, > especially these pages: > -http://wiki.freeswitch.org/wiki/Getting_Started_Guide > -http://wiki.freeswitch.org/wiki/Dialplan_XML > > maybe you could tell more about the linux distribution you're using so > that I can give you some pointers for sipp... > > regards. > rod. > > > Saeed Ahmed wrote: > > Hi Rod, > > > > Could you please share how you configured Sipp & FS to create a test > > environment? Especially the dial plan, sofia settings etc..., actually I > am > > a newbie. I want to test it on a single FS machine. > > > > Kind Regards > > Saeed > > -----Original Message----- > > From: freeswitch-users-bounces at lists.freeswitch.org > > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of rod > > Sent: Monday, February 02, 2009 11:00 AM > > To: freeswitch-users at lists.freeswitch.org > > Subject: Re: [Freeswitch-users] Strange Performance when using as SBC > > > > Hi Ken, Jay, > > > > thanks for pointing to proxy media, I will test. > > > > Ken, you are right, I was brain damaged (a stupid mistake) when setting > > INFO cause this kind of level could be very verbose. I'm switching to > > CRIT or ERR. > > > > Thanks guys, > > rod. > > > > thanks for > > > > Ken Rice wrote: > > > >> If you don't have to transcode, using proxy media mode will still save > you > >> some CPU time. This is 1/2 way between bypass media and the default > media > >> interactive mode. The other draw back to this mode is if you are using > FS > >> > > to > > > >> clean up RTP and DTMF you loose those functions but they are not needed > in > >> most use cases. > >> > >> As far as the log level goes, I found that once I had things stable > >> > > setting > > > >> the loglevel to helped a good deal... Info is probably a bit too high of > a > >> loglevel I would probably go for CRIT or ERR (2 or 1 respectively) if > you > >> insist on leaving logging turned on... On a busy system these can and > will > >> generate a good deal of activity (and disk IO if using mod_logfile) > >> > >> Ken > >> > >> > >> > >> > >>> From: rod > >>> Reply-To: > >>> Date: Mon, 02 Feb 2009 11:36:35 +0400 > >>> To: > >>> Subject: Re: [Freeswitch-users] Strange Performance when using as SBC > >>> > >>> Hi Ken, > >>> > >>> 1) I'd like to use FS to hide topology, so bypass media is not possible > >>> 2) done > >>> 3) done > >>> 4) not used > >>> 5) i'm using this ins switch.xml -> >>> value="info"/>, if you think an other log level is more suitable. > >>> > >>> Regarding logging, I can see in console and in the freeswitch.log that > >>> there is still a lot of NOTICE logging, see below: > >>> 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:960 > >>> switch_core_session_thread() Session 8721 > >>> (sofia/internal/sipp at 10.10.10.1:5060) Ended > >>> 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:962 > >>> switch_core_session_thread() Close Channel > >>> sofia/internal/sipp at 10.10.10.1:5060 [CS_HANGUP] > >>> 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:960 > >>> switch_core_session_thread() Session 8722 > >>> (sofia/external/9998 at 10.10.20.100) Ended > >>> 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:962 > >>> switch_core_session_thread() Close Channel > >>> sofia/external/9998 at 10.10.20.100 [CS_HANGUP] > >>> 2009-02-02 08:33:56 [NOTICE] sofia.c:3164 sofia_handle_sip_i_state() > >>> Channel [sofia/external/9998 at 10.10.20.100] has been answered > >>> 2009-02-02 08:33:56 [WARNING] mod_sofia.c:740 sofia_read_frame() > >>> Changing codec ptime to 30. I bet you have a linksys/sipura =D > >>> > >>> Do you have any idea where I can switch off this kind of logging. I > >>> thought it should be in /dialplan/internal.xml, but I see that in > >>> internal.xml -> > >>> > >>> thanks a lot for your suggestion. > >>> > >>> regards, > >>> rod > >>> > >>> Ken Rice wrote: > >>> > >>> > >>>> Dont forget there are several things you can do to increase > >>>> > > performance... > > > >>>> 1) where possible use bypass media or media proxy modes > >>>> 2) mount freeswitch/db as a ram drive (if you are using voicemail with > >>>> the internal FS DBs you'll need a way to make this persistant across > >>>> reboots) > >>>> 3) see the wiki for setting reasonable ulimits > >>>> 4) (this is my oppinion others may vary) dont use mod_cdr_csv > >>>> 5) turn off (or reduce logging) in switch.conf.xml > >>>> > >>>> all of these thing can greatly improve performance. > >>>> > >>>> On Mon, Feb 2, 2009 at 1:04 AM, rod >>>> > wrote: > >>>> > >>>> Thanks Anthony, > >>>> > >>>> the setup is like this: > >>>> > >>>> sipp server ---- FS 1 ---- FS2 > >>>> > >>>> FS1 is the AMD CPU that has only one extension in dialplan that > >>>> bridges > >>>> 9999 to FS2. 9999 is the first extension in FS2 dialplan that > >>>> plays moh, > >>>> FS2 has no CPU pbm. > >>>> > >>>> FS1 is maxing out at 60 bridged calls without your option -hp. > >>>> > >>>> Using -hp, I'm now able to bridge 200 concurrent calls (a great > >>>> improvement) and the system is still reactive. CPU load is high > >>>> but not > >>>> 100% and as the system responds well, I think that doesn't matter. > >>>> > > The > > > >>>> 2GB of memory are completely consumed (top command shows 700MB for > >>>> > > FS > > > >>>> process). > >>>> > >>>> I understand that FS1 server is not the best hardware platform, > >>>> and I'm > >>>> waiting for new 4 cores server for testing. > >>>> I will update those numbers when testing with the new hardware. > >>>> > >>>> regards, > >>>> rod. > >>>> > >>>> Anthony Minessale wrote: > >>>> > >>>> > >>>>> Which of the 2 machines has the load issue? You said it was one box > >>>>> calling the other. > >>>>> > >>>>> You have 2 major things against you, single CPU and AMD, but you > >>>>> should at least be able to get in the vicinity of 800-1000 calls > >>>>> > >>>>> > >>>> on a > >>>> > >>>> > >>>>> box like that. > >>>>> > >>>>> Are you calling the default 9999? It's not really an appropriate > >>>>> extension for load testing. > >>>>> On the terminating box you should set up a manual extension that is > >>>>> the first one in the dial plan > >>>>> to play a wav file from preferably a ram disk or /tmp > >>>>> > >>>>> If you do plan on using this in production accept nothing less > >>>>> > >>>>> > >>>> than a > >>>> > >>>> > >>>>> multi-core intel machine with at least 4 cores, the more cores the > >>>>> better because that parallel processing is where FS gets it's > >>>>> > >>>>> > >>>> atvantage. > >>>> > >>>> > >>>>> On Fri, Jan 30, 2009 at 5:56 AM, rod >>>>> > >>>>> > >>>> > >>>> > >>>> > >>>>> >> wrote: > >>>>> > >>>>> Dear list, > >>>>> > >>>>> I've been playing with freeswitch for some time (2 months) > >>>>> > >>>>> > >>>> and the > >>>> > >>>> > >>>>> fact > >>>>> is that I'm very pleased with the functionnalities of this > >>>>> > >>>>> > >>>> software. > >>>> > >>>> > >>>>> I'd like to use FS as a SBC handling media and I'm doing some > >>>>> tests with > >>>>> sipp to load the machine but I'm unable to bridge more than > >>>>> > >>>>> > >>>> 60 calls > >>>> > >>>> > >>>>> without seeing the CPU being loaded at 100%. I'm sure > >>>>> > >>>>> > >>>> something is > >>>> > >>>> > >>>>> going > >>>>> wrong with my setup but I'm unable to see what. > >>>>> > >>>>> The test machine has the following specs: > >>>>> Athlon XP 3500+ with 2GB of memory (I know this is not a > >>>>> > >>>>> > >>>> high end > >>>> > >>>> > >>>>> machine :p) > >>>>> > >>>>> Freeswitch:/opt/freeswitch/log# cat /proc/cpuinfo > >>>>> processor : 0 > >>>>> vendor_id : AuthenticAMD > >>>>> cpu family : 15 > >>>>> model : 95 > >>>>> model name : AMD Athlon(tm) 64 Processor 3500+ > >>>>> stepping : 2 > >>>>> cpu MHz : 2199.973 > >>>>> cache size : 512 KB > >>>>> fpu : yes > >>>>> fpu_exception : yes > >>>>> cpuid level : 1 > >>>>> wp : yes > >>>>> flags : fpu vme de pse tsc msr pae mce cx8 apic > >>>>> > >>>>> > >>>> sep mtrr pge > >>>> > >>>> > >>>>> mca cmov pat pse36 clflush mmx fxsr sse sse2 syscall nx mmxext > >>>>> fxsr_opt > >>>>> rdtscp lm 3dnowext 3dnow up rep_good pni cx16 lahf_lm svm > >>>>> > >>>>> > >>>> extapic > >>>> > >>>> > >>>>> cr8_legacy > >>>>> bogomips : 4402.97 > >>>>> TLB size : 1024 4K pages > >>>>> clflush size : 64 > >>>>> cache_alignment : 64 > >>>>> address sizes : 40 bits physical, 48 bits virtual > >>>>> power management: ts fid vid ttp tm stc > >>>>> > >>>>> I installed FS on a fresh debian 64: > >>>>> Linux Freeswitch 2.6.26-1-amd64 #1 SMP Sat Jan 10 17:57:00 > >>>>> > >>>>> > >>>> UTC 2009 > >>>> > >>>> > >>>>> x86_64 GNU/Linux > >>>>> > >>>>> I set the ulimit parameters like those on the website: > >>>>> freeswitch at internal> ... > >>>>> Freeswitch:/opt/free-svn/bin# ulimit -a > >>>>> core file size (blocks, -c) unlimited > >>>>> data seg size (kbytes, -d) unlimited > >>>>> scheduling priority (-e) 0 > >>>>> file size (blocks, -f) unlimited > >>>>> pending signals (-i) unlimited > >>>>> max locked memory (kbytes, -l) unlimited > >>>>> max memory size (kbytes, -m) unlimited > >>>>> open files (-n) 999999 > >>>>> pipe size (512 bytes, -p) 8 > >>>>> POSIX message queues (bytes, -q) unlimited > >>>>> real-time priority (-r) 0 > >>>>> stack size (kbytes, -s) 244 > >>>>> cpu time (seconds, -t) unlimited > >>>>> max user processes (-u) unlimited > >>>>> virtual memory (kbytes, -v) unlimited > >>>>> file locks (-x) unlimited > >>>>> > >>>>> > >>>>> My network setup is the following: > >>>>> > >>>>> SIPP machine (10.10.10.1/24)----------------vlan > >>>>> > >>>>> > >>>> > >>>> > >>>> > >>>>> 55 > >>>>> ----------(10.10.10.254/24 > >>>>> > >>>>> > >>>> ) FS > >>>> > >>>> > >>>>> (10.10.20.254/24)-------------- > >>>>> > >>>>> > >>>> > >>>> > >>>> > >>>>> vlan56 > >>>>> -------------------(10.10.20.100/24 > >>>>> > >>>>> > >>>> ) > >>>> > >>>> > >>>>> OTHER STOCK FS > >>>>> > >>>>> > >>>>> I launched sipp with: > >>>>> sipp -sn uac_pcap -s 9999 -r 10 -l 80 -d 60000 -mi > >>>>> > >>>>> > >>>> 10.10.10.1 -i > >>>> > >>>> > >>>>> 10.10.10.1 -mp 25000 10.10.10.254:5060 > >>>>> > >>>>> > >>>> > >>>> > >>>> > >>>>> The dialplan on FS is very simple: > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> >>>>> data="sofia/external/9999 at 10.10.20.100 > >>>>> > >>>>> > >>>> >>>> >"/> > >>>> > >>>> > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> FreeSWITCH Version 1.0.trunk (11560M) Started. > >>>>> Crash Protection [Disabled] > >>>>> Max Sessions[1000] > >>>>> Session Rate[100] > >>>>> SQL [Enabled] > >>>>> > >>>>> > >>>>> The test is very simple: sipp dial 9999 that matches in my > >>>>> > >>>>> > >>>> FS dialplan > >>>> > >>>> > >>>>> and this is bridged to an other FS machine playing music on > >>>>> > >>>>> > >>>> hold. > >>>> > >>>> > >>>>> When I launch "top" I see after 30 to 40 s that FS consumes all > >>>>> the CPU > >>>>> ressources (with a mean of 50-60 % before), with 80 calls. > >>>>> When I set 70 calls, I have to wait 70-80 s before seeing > >>>>> > >>>>> > >>>> the same > >>>> > >>>> > >>>>> issue. > >>>>> > >>>>> Presence is set to false on the 2 profile. > >>>>> > >>>>> I have the same issue with FS 1.0.2 that' s why I tried FS > >>>>> > >>>>> > >>>> 11560. > >>>> > >>>> > >>>>> When I use the FS machine as a router to test the packet per > >>>>> > >>>>> > >>>> second > >>>> > >>>> > >>>>> performance, I'm reaching 100Mbps with 8000pps in each > >>>>> > >>>>> > >>>> direction (from > >>>> > >>>> > >>>>> vlan 55 to vlan56) with less than 12% CPU. So that I don't think > >>>>> there's > >>>>> an issue with the network. > >>>>> > >>>>> Here is an "mpstat -P ALL 1" to show you what's happening > >>>>> > >>>>> > >>>> suddenly > >>>> > >>>> > >>>>> with > >>>>> 70 bridge calls: > >>>>> 12:31:26 CPU %user %nice %sys %iowait %irq %soft > >>>>> %steal %idle intr/s > >>>>> 12:31:27 all 3,00 0,00 3,00 0,00 1,00 4,00 > >>>>> 0,00 89,00 6241,00 > >>>>> 12:31:27 0 3,00 0,00 3,00 0,00 1,00 4,00 > >>>>> 0,00 89,00 6241,00 > >>>>> > >>>>> 12:31:27 CPU %user %nice %sys %iowait %irq %soft > >>>>> %steal %idle intr/s > >>>>> 12:31:28 all 14,14 0,00 56,57 0,00 2,02 5,05 > >>>>> 0,00 22,22 6035,35 > >>>>> 12:31:28 0 14,14 0,00 56,57 0,00 2,02 5,05 > >>>>> 0,00 22,22 6035,35 > >>>>> > >>>>> 12:31:28 CPU %user %nice %sys %iowait %irq %soft > >>>>> %steal %idle intr/s > >>>>> 12:31:29 all 24,75 0,00 67,33 0,00 0,99 6,93 > >>>>> 0,00 0,00 5483,17 > >>>>> 12:31:29 0 24,75 0,00 67,33 0,00 0,99 6,93 > >>>>> 0,00 0,00 5483,17 > >>>>> > >>>>> > >>>>> The CPU is going from 89% idle to 0% in less than 2 seconds. > >>>>> > >>>>> I know that I don't have to expect too much from this kind of > >>>>> hardware, > >>>>> but it seems strange that the CPU power vanished so suddenly. > >>>>> > >>>>> Thanks a lot for the guys that have read this long mail :p > >>>>> > >>>>> kind regards, > >>>>> rod > >>>>> > >>>>> > >>>>> _______________________________________________ > >>>>> Freeswitch-users mailing list > >>>>> Freeswitch-users at lists.freeswitch.org > >>>>> > >>>>> > >>>> > >>>> > >>>> > >>>>> >>>>> > >>>>> > >>>> > > >>>> > >>>> > >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>>>> > >>>>> > >>>>> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > >>>> > >>>> > >>>>> http://www.freeswitch.org > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> -- > >>>>> Anthony Minessale II > >>>>> > >>>>> FreeSWITCH http://www.freeswitch.org/ > >>>>> ClueCon http://www.cluecon.com/ > >>>>> > >>>>> AIM: anthm > >>>>> MSN:anthony_minessale at hotmail.com > >>>>> > >>>>> > >>>> > > > >>>> > >>>> > >>>>> > >>>>> > >>>>> > >>>> > >> > >>>> > >>>> > >>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > >>>>> > >>>>> > >>>> > > > >>>> > >>>> > >>>>> > >>>>> > >>>>> > >>>> > >> > >>>> > >>>> > >>>>> IRC: irc.freenode.net > >>>>> > >>>>> > >>>> #freeswitch > >>>> > >>>> > >>>>> FreeSWITCH Developer Conference > >>>>> sip:888 at conference.freeswitch.org > >>>>> > >>>>> > >>>> > > > >>>> > >>>> > >>>>> > >>>>> > >>>>> > >>>> > >> > >>>> > >>>> > >>>>> iax:guest at conference.freeswitch.org/888 > >>>>> > >>>>> > >>>> > >>>> > >>>> > >>>>> > >>>>> googletalk:conf+888 at conference.freeswitch.org > >>>>> > >>>>> > >>>> > > > >>>> > >>>> > >>>>> > >>>>> > >>>>> > >>>> > >> > >>>> > >>>> > >>>>> pstn:213-799-1400 > >>>>> > >>>>> > >>>>> > > ------------------------------------------------------------------------ > > > >>>> > >>>> > >>>>> _______________________________________________ > >>>>> Freeswitch-users mailing list > >>>>> Freeswitch-users at lists.freeswitch.org > >>>>> > >>>>> > >>>> > >>>> > >>>> > >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>>>> > >>>>> > >>>>> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > >>>> > >>>> > >>>>> http://www.freeswitch.org > >>>>> > >>>>> > >>>>> > >>>> _______________________________________________ > >>>> Freeswitch-users mailing list > >>>> Freeswitch-users at lists.freeswitch.org > >>>> > >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>>> > >>>> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > >>>> http://www.freeswitch.org > >>>> > >>>> > >>>> > ------------------------------------------------------------------------ > >>>> > >>>> _______________________________________________ > >>>> Freeswitch-users mailing list > >>>> Freeswitch-users at lists.freeswitch.org > >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >>>> http://www.freeswitch.org > >>>> > >>>> > >>>> > >>> _______________________________________________ > >>> Freeswitch-users mailing list > >>> Freeswitch-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> http://www.freeswitch.org > >>> > >>> > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > >> > >> > >> > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090202/e58251ab/attachment-0001.html From anthony.minessale at gmail.com Mon Feb 2 06:01:25 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 2 Feb 2009 08:01:25 -0600 Subject: [Freeswitch-users] Conference dialing and uuid In-Reply-To: <9FE1733C-A46A-4A17-8AEE-077DE8010235@freeswitch.org> References: <20090130103944.GA23536@cpdata.co.za> <20090130133315.GB23536@cpdata.co.za> <191c3a030901300605r78dd8943w7fb1075ef851e1d@mail.gmail.com> <20090202085157.GA3555@cpdata.co.za> <0F863750-5C77-4CD9-ADC6-CFDDF2B149B4@freeswitch.org> <20090202102459.GA4179@cpdata.co.za> <9FE1733C-A46A-4A17-8AEE-077DE8010235@freeswitch.org> Message-ID: <191c3a030902020601v1109b1e9he867e9f1e1d4a401@mail.gmail.com> you could set the conference moh sound to be tone_stream::// with the teletone spec for ring sound and it use ignore_early_media=true in your originates so the first caller would hear ringback until the 2nd one arrived. On Mon, Feb 2, 2009 at 4:29 AM, Brian West wrote: > Loopback will not work in that case either. If the far end plays > ringback inband you should hear that if you use the conference dial > api call. > > /b > > On Feb 2, 2009, at 4:24 AM, Sias Mey wrote: > > > Aaah ok. > > > > Thanks for clearing that up. > > > > So using loopback is still the only real workable sollution for me, > > since that generates ringback from and alternative endpoint and > > plays it > > into the conference. > > > > I might play with some javascript that streams ring into the channel > > eventually but for now the string comparisons at least get me the > > right > > uuid. > > > > Thank you again, > > Sias > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090202/1754895a/attachment.html From anthony.minessale at gmail.com Mon Feb 2 06:06:14 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 2 Feb 2009 08:06:14 -0600 Subject: [Freeswitch-users] Call Variable not available when call hangup In-Reply-To: <21789152.post@talk.nabble.com> References: <21788550.post@talk.nabble.com> <21789152.post@talk.nabble.com> Message-ID: <191c3a030902020606r1a42ef44n7a73bd1e5157392e@mail.gmail.com> the leg you are running the script on is not hungup, the other leg of the call is. If it was hungup you would not be executing the script. Asterisk and the h ext and the whole dead-agi thing are all poor design showing it's teeth. We do not support anything like it. You can however try this: (see the link below) http://www.nabble.com/Re:-Conference-javascript-and-hanuphooks-giving-me-headaches-p21614840.html On Mon, Feb 2, 2009 at 6:53 AM, shehzad p wrote: > > Is there any settings that when call hangup control can be transferred to > another context and these CDR values can be accessible there? (just like in > Asterisk, h extension) > > shehzad p wrote: > > > > Hi all, > > > > I need to process some CDR variables in Dialplan, like call duration, > > Answered time etc. > > but when I place info application after bridge, it is not listing them > > properly as below: > > =========================================== > > Caller-Channel-Created-Time: [1233573341672157] > > Caller-Channel-Answered-Time: [1233573342712939] > > Caller-Channel-Hangup-Time: [0] > > ========================================== > > Here Hangup time is 0, So how can I find actual values? > > > > --I know that we can use xml_cdr or cdr_csv, but my current need is to > get > > those values from dialplan itself so that can be passed to some script... > > > > > > thanks, > > msp > > > > -- > View this message in context: > http://www.nabble.com/Call-Variable-not-available-when-call-hangup-tp21788550p21789152.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090202/6d471696/attachment.html From saeedahmad1981 at gmail.com Mon Feb 2 06:29:49 2009 From: saeedahmad1981 at gmail.com (Saeed Ahmed) Date: Mon, 2 Feb 2009 15:29:49 +0100 Subject: [Freeswitch-users] Strange Performance when using as SBC In-Reply-To: <191c3a030902020557v70d3777fqb83e591385a64a59@mail.gmail.com> References: <4986C42C.7030700@laposte.net><1C31435E19AB4E57981CF2AAC2CA74AA@SaeedLaptop><4986ECA9.3040707@laposte.net><9232C06D5362494791AF713E1DF61343@SaeedLaptop> <191c3a030902020557v70d3777fqb83e591385a64a59@mail.gmail.com> Message-ID: <0AF29744A97A482CB167B5C1F185F8B6@SaeedLaptop> Its Ubuntu 8.04 Hardy, 2.6.24-16 kernel. I hope it will be OK Kind Regards Saeed Ahmed Tariq _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Monday, February 02, 2009 2:58 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Strange Performance when using as SBC if you want to use ubuntu, be sure to use hardy and not intrepid. On Mon, Feb 2, 2009 at 7:03 AM, Saeed Ahmed wrote: Thanks rod for a quick answer, FS is installed on Ubuntu Server. I am planning to replace Nextone SBC with FS, Later I'll also use openZAP to communicate with TDM but this all depends how much calls it can take, or maybe we can also do something in clustering environment ( I am not sure about it). But thanks again and any further help will be highly appreciated. Kind Regards Saeed Ahmed Tariq -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of rod Sent: Monday, February 02, 2009 1:53 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Strange Performance when using as SBC Hi Saeed, I just created an account to share my setup on the wiki. I will detail all the steps for a clean install of a debian64 lenny with FS used as a SBC (next step is to try the new LCR module :) )and what I'm doing do stress the server. I wrote nothing at this time so please be patient, I'm waiting for my new hardware so that I will detail as much as possible what I'll do. For beginning I suggest you reading the start page on the wiki, especially these pages: -http://wiki.freeswitch.org/wiki/Getting_Started_Guide -http://wiki.freeswitch.org/wiki/Dialplan_XML maybe you could tell more about the linux distribution you're using so that I can give you some pointers for sipp... regards. rod. Saeed Ahmed wrote: > Hi Rod, > > Could you please share how you configured Sipp & FS to create a test > environment? Especially the dial plan, sofia settings etc..., actually I am > a newbie. I want to test it on a single FS machine. > > Kind Regards > Saeed > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of rod > Sent: Monday, February 02, 2009 11:00 AM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Strange Performance when using as SBC > > Hi Ken, Jay, > > thanks for pointing to proxy media, I will test. > > Ken, you are right, I was brain damaged (a stupid mistake) when setting > INFO cause this kind of level could be very verbose. I'm switching to > CRIT or ERR. > > Thanks guys, > rod. > > thanks for > > Ken Rice wrote: > >> If you don't have to transcode, using proxy media mode will still save you >> some CPU time. This is 1/2 way between bypass media and the default media >> interactive mode. The other draw back to this mode is if you are using FS >> > to > >> clean up RTP and DTMF you loose those functions but they are not needed in >> most use cases. >> >> As far as the log level goes, I found that once I had things stable >> > setting > >> the loglevel to helped a good deal... Info is probably a bit too high of a >> loglevel I would probably go for CRIT or ERR (2 or 1 respectively) if you >> insist on leaving logging turned on... On a busy system these can and will >> generate a good deal of activity (and disk IO if using mod_logfile) >> >> Ken >> >> >> >> >>> From: rod >>> Reply-To: >>> Date: Mon, 02 Feb 2009 11:36:35 +0400 >>> To: >>> Subject: Re: [Freeswitch-users] Strange Performance when using as SBC >>> >>> Hi Ken, >>> >>> 1) I'd like to use FS to hide topology, so bypass media is not possible >>> 2) done >>> 3) done >>> 4) not used >>> 5) i'm using this ins switch.xml -> >> value="info"/>, if you think an other log level is more suitable. >>> >>> Regarding logging, I can see in console and in the freeswitch.log that >>> there is still a lot of NOTICE logging, see below: >>> 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:960 >>> switch_core_session_thread() Session 8721 >>> (sofia/internal/sipp at 10.10.10.1:5060) Ended >>> 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:962 >>> switch_core_session_thread() Close Channel >>> sofia/internal/sipp at 10.10.10.1:5060 [CS_HANGUP] >>> 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:960 >>> switch_core_session_thread() Session 8722 >>> (sofia/external/9998 at 10.10.20.100) Ended >>> 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:962 >>> switch_core_session_thread() Close Channel >>> sofia/external/9998 at 10.10.20.100 [CS_HANGUP] >>> 2009-02-02 08:33:56 [NOTICE] sofia.c:3164 sofia_handle_sip_i_state() >>> Channel [sofia/external/9998 at 10.10.20.100] has been answered >>> 2009-02-02 08:33:56 [WARNING] mod_sofia.c:740 sofia_read_frame() >>> Changing codec ptime to 30. I bet you have a linksys/sipura =D >>> >>> Do you have any idea where I can switch off this kind of logging. I >>> thought it should be in /dialplan/internal.xml, but I see that in >>> internal.xml -> >>> >>> thanks a lot for your suggestion. >>> >>> regards, >>> rod >>> >>> Ken Rice wrote: >>> >>> >>>> Dont forget there are several things you can do to increase >>>> > performance... > >>>> 1) where possible use bypass media or media proxy modes >>>> 2) mount freeswitch/db as a ram drive (if you are using voicemail with >>>> the internal FS DBs you'll need a way to make this persistant across >>>> reboots) >>>> 3) see the wiki for setting reasonable ulimits >>>> 4) (this is my oppinion others may vary) dont use mod_cdr_csv >>>> 5) turn off (or reduce logging) in switch.conf.xml >>>> >>>> all of these thing can greatly improve performance. >>>> >>>> On Mon, Feb 2, 2009 at 1:04 AM, rod >>> > wrote: >>>> >>>> Thanks Anthony, >>>> >>>> the setup is like this: >>>> >>>> sipp server ---- FS 1 ---- FS2 >>>> >>>> FS1 is the AMD CPU that has only one extension in dialplan that >>>> bridges >>>> 9999 to FS2. 9999 is the first extension in FS2 dialplan that >>>> plays moh, >>>> FS2 has no CPU pbm. >>>> >>>> FS1 is maxing out at 60 bridged calls without your option -hp. >>>> >>>> Using -hp, I'm now able to bridge 200 concurrent calls (a great >>>> improvement) and the system is still reactive. CPU load is high >>>> but not >>>> 100% and as the system responds well, I think that doesn't matter. >>>> > The > >>>> 2GB of memory are completely consumed (top command shows 700MB for >>>> > FS > >>>> process). >>>> >>>> I understand that FS1 server is not the best hardware platform, >>>> and I'm >>>> waiting for new 4 cores server for testing. >>>> I will update those numbers when testing with the new hardware. >>>> >>>> regards, >>>> rod. >>>> >>>> Anthony Minessale wrote: >>>> >>>> >>>>> Which of the 2 machines has the load issue? You said it was one box >>>>> calling the other. >>>>> >>>>> You have 2 major things against you, single CPU and AMD, but you >>>>> should at least be able to get in the vicinity of 800-1000 calls >>>>> >>>>> >>>> on a >>>> >>>> >>>>> box like that. >>>>> >>>>> Are you calling the default 9999? It's not really an appropriate >>>>> extension for load testing. >>>>> On the terminating box you should set up a manual extension that is >>>>> the first one in the dial plan >>>>> to play a wav file from preferably a ram disk or /tmp >>>>> >>>>> If you do plan on using this in production accept nothing less >>>>> >>>>> >>>> than a >>>> >>>> >>>>> multi-core intel machine with at least 4 cores, the more cores the >>>>> better because that parallel processing is where FS gets it's >>>>> >>>>> >>>> atvantage. >>>> >>>> >>>>> On Fri, Jan 30, 2009 at 5:56 AM, rod >>>> >>>>> >>>> >>>> >>>> >>>>> >> wrote: >>>>> >>>>> Dear list, >>>>> >>>>> I've been playing with freeswitch for some time (2 months) >>>>> >>>>> >>>> and the >>>> >>>> >>>>> fact >>>>> is that I'm very pleased with the functionnalities of this >>>>> >>>>> >>>> software. >>>> >>>> >>>>> I'd like to use FS as a SBC handling media and I'm doing some >>>>> tests with >>>>> sipp to load the machine but I'm unable to bridge more than >>>>> >>>>> >>>> 60 calls >>>> >>>> >>>>> without seeing the CPU being loaded at 100%. I'm sure >>>>> >>>>> >>>> something is >>>> >>>> >>>>> going >>>>> wrong with my setup but I'm unable to see what. >>>>> >>>>> The test machine has the following specs: >>>>> Athlon XP 3500+ with 2GB of memory (I know this is not a >>>>> >>>>> >>>> high end >>>> >>>> >>>>> machine :p) >>>>> >>>>> Freeswitch:/opt/freeswitch/log# cat /proc/cpuinfo >>>>> processor : 0 >>>>> vendor_id : AuthenticAMD >>>>> cpu family : 15 >>>>> model : 95 >>>>> model name : AMD Athlon(tm) 64 Processor 3500+ >>>>> stepping : 2 >>>>> cpu MHz : 2199.973 >>>>> cache size : 512 KB >>>>> fpu : yes >>>>> fpu_exception : yes >>>>> cpuid level : 1 >>>>> wp : yes >>>>> flags : fpu vme de pse tsc msr pae mce cx8 apic >>>>> >>>>> >>>> sep mtrr pge >>>> >>>> >>>>> mca cmov pat pse36 clflush mmx fxsr sse sse2 syscall nx mmxext >>>>> fxsr_opt >>>>> rdtscp lm 3dnowext 3dnow up rep_good pni cx16 lahf_lm svm >>>>> >>>>> >>>> extapic >>>> >>>> >>>>> cr8_legacy >>>>> bogomips : 4402.97 >>>>> TLB size : 1024 4K pages >>>>> clflush size : 64 >>>>> cache_alignment : 64 >>>>> address sizes : 40 bits physical, 48 bits virtual >>>>> power management: ts fid vid ttp tm stc >>>>> >>>>> I installed FS on a fresh debian 64: >>>>> Linux Freeswitch 2.6.26-1-amd64 #1 SMP Sat Jan 10 17:57:00 >>>>> >>>>> >>>> UTC 2009 >>>> >>>> >>>>> x86_64 GNU/Linux >>>>> >>>>> I set the ulimit parameters like those on the website: >>>>> freeswitch at internal> ... >>>>> Freeswitch:/opt/free-svn/bin# ulimit -a >>>>> core file size (blocks, -c) unlimited >>>>> data seg size (kbytes, -d) unlimited >>>>> scheduling priority (-e) 0 >>>>> file size (blocks, -f) unlimited >>>>> pending signals (-i) unlimited >>>>> max locked memory (kbytes, -l) unlimited >>>>> max memory size (kbytes, -m) unlimited >>>>> open files (-n) 999999 >>>>> pipe size (512 bytes, -p) 8 >>>>> POSIX message queues (bytes, -q) unlimited >>>>> real-time priority (-r) 0 >>>>> stack size (kbytes, -s) 244 >>>>> cpu time (seconds, -t) unlimited >>>>> max user processes (-u) unlimited >>>>> virtual memory (kbytes, -v) unlimited >>>>> file locks (-x) unlimited >>>>> >>>>> >>>>> My network setup is the following: >>>>> >>>>> SIPP machine (10.10.10.1/24)----------------vlan >>>>> >>>>> >>>> >>>> >>>> >>>>> 55 >>>>> ----------(10.10.10.254/24 >>>>> >>>>> >>>> ) FS >>>> >>>> >>>>> (10.10.20.254/24)-------------- >>>>> >>>>> >>>> >>>> >>>> >>>>> vlan56 >>>>> -------------------(10.10.20.100/24 >>>>> >>>>> >>>> ) >>>> >>>> >>>>> OTHER STOCK FS >>>>> >>>>> >>>>> I launched sipp with: >>>>> sipp -sn uac_pcap -s 9999 -r 10 -l 80 -d 60000 -mi >>>>> >>>>> >>>> 10.10.10.1 -i >>>> >>>> >>>>> 10.10.10.1 -mp 25000 10.10.10.254:5060 >>>>> >>>>> >>>> >>>> >>>> >>>>> The dialplan on FS is very simple: >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>> data="sofia/external/9999 at 10.10.20.100 >>>>> >>>>> >>>> >>> >"/> >>>> >>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> FreeSWITCH Version 1.0.trunk (11560M) Started. >>>>> Crash Protection [Disabled] >>>>> Max Sessions[1000] >>>>> Session Rate[100] >>>>> SQL [Enabled] >>>>> >>>>> >>>>> The test is very simple: sipp dial 9999 that matches in my >>>>> >>>>> >>>> FS dialplan >>>> >>>> >>>>> and this is bridged to an other FS machine playing music on >>>>> >>>>> >>>> hold. >>>> >>>> >>>>> When I launch "top" I see after 30 to 40 s that FS consumes all >>>>> the CPU >>>>> ressources (with a mean of 50-60 % before), with 80 calls. >>>>> When I set 70 calls, I have to wait 70-80 s before seeing >>>>> >>>>> >>>> the same >>>> >>>> >>>>> issue. >>>>> >>>>> Presence is set to false on the 2 profile. >>>>> >>>>> I have the same issue with FS 1.0.2 that' s why I tried FS >>>>> >>>>> >>>> 11560. >>>> >>>> >>>>> When I use the FS machine as a router to test the packet per >>>>> >>>>> >>>> second >>>> >>>> >>>>> performance, I'm reaching 100Mbps with 8000pps in each >>>>> >>>>> >>>> direction (from >>>> >>>> >>>>> vlan 55 to vlan56) with less than 12% CPU. So that I don't think >>>>> there's >>>>> an issue with the network. >>>>> >>>>> Here is an "mpstat -P ALL 1" to show you what's happening >>>>> >>>>> >>>> suddenly >>>> >>>> >>>>> with >>>>> 70 bridge calls: >>>>> 12:31:26 CPU %user %nice %sys %iowait %irq %soft >>>>> %steal %idle intr/s >>>>> 12:31:27 all 3,00 0,00 3,00 0,00 1,00 4,00 >>>>> 0,00 89,00 6241,00 >>>>> 12:31:27 0 3,00 0,00 3,00 0,00 1,00 4,00 >>>>> 0,00 89,00 6241,00 >>>>> >>>>> 12:31:27 CPU %user %nice %sys %iowait %irq %soft >>>>> %steal %idle intr/s >>>>> 12:31:28 all 14,14 0,00 56,57 0,00 2,02 5,05 >>>>> 0,00 22,22 6035,35 >>>>> 12:31:28 0 14,14 0,00 56,57 0,00 2,02 5,05 >>>>> 0,00 22,22 6035,35 >>>>> >>>>> 12:31:28 CPU %user %nice %sys %iowait %irq %soft >>>>> %steal %idle intr/s >>>>> 12:31:29 all 24,75 0,00 67,33 0,00 0,99 6,93 >>>>> 0,00 0,00 5483,17 >>>>> 12:31:29 0 24,75 0,00 67,33 0,00 0,99 6,93 >>>>> 0,00 0,00 5483,17 >>>>> >>>>> >>>>> The CPU is going from 89% idle to 0% in less than 2 seconds. >>>>> >>>>> I know that I don't have to expect too much from this kind of >>>>> hardware, >>>>> but it seems strange that the CPU power vanished so suddenly. >>>>> >>>>> Thanks a lot for the guys that have read this long mail :p >>>>> >>>>> kind regards, >>>>> rod >>>>> >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> >>>>> >>>> >>>>> >>>> > >>>> >>>> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> >>>>> >>>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >>>> >>>> >>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> >>>>> >>>>> -- >>>>> Anthony Minessale II >>>>> >>>>> FreeSWITCH http://www.freeswitch.org/ >>>>> ClueCon http://www.cluecon.com/ >>>>> >>>>> AIM: anthm >>>>> MSN:anthony_minessale at hotmail.com >>>>> >>>>> >>>> > >>>> >>>> >>>>> >>>>> >>>>> >>>> >> >>>> >>>> >>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>>>> >>>>> >>>> > >>>> >>>> >>>>> >>>>> >>>>> >>>> >> >>>> >>>> >>>>> IRC: irc.freenode.net >>>>> >>>>> >>>> #freeswitch >>>> >>>> >>>>> FreeSWITCH Developer Conference >>>>> sip:888 at conference.freeswitch.org >>>>> >>>>> >>>> > >>>> >>>> >>>>> >>>>> >>>>> >>>> >> >>>> >>>> >>>>> iax:guest at conference.freeswitch.org/888 >>>>> >>>>> >>>> >>>> >>>> >>>>> >>>>> googletalk:conf+888 at conference.freeswitch.org >>>>> >>>>> >>>> > >>>> >>>> >>>>> >>>>> >>>>> >>>> >> >>>> >>>> >>>>> pstn:213-799-1400 >>>>> >>>>> >>>>> > ------------------------------------------------------------------------ > >>>> >>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> >>>>> >>>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >>>> >>>> >>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> >>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >>>> http://www.freeswitch.org >>>> >>>> >>>> ------------------------------------------------------------------------ >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 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URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090202/72e54e09/attachment-0001.html From kawarod at laposte.net Mon Feb 2 06:33:22 2009 From: kawarod at laposte.net (rod) Date: Mon, 02 Feb 2009 18:33:22 +0400 Subject: [Freeswitch-users] Strange Performance when using as SBC In-Reply-To: <9232C06D5362494791AF713E1DF61343@SaeedLaptop> References: <4986C42C.7030700@laposte.net><1C31435E19AB4E57981CF2AAC2CA74AA@SaeedLaptop> <4986ECA9.3040707@laposte.net> <9232C06D5362494791AF713E1DF61343@SaeedLaptop> Message-ID: <49870432.5050301@laposte.net> Hi Saeed, Here is a first draft of what I did to install FS on my server. Configuration are not present, they'll be in a next release :p http://wiki.freeswitch.org/wiki/SBC_Setup My aim is to setup FS as a SBC, I hope this page could be a great startup point for others. I will update regularly based on what I did. Saeed, why are you replacing your Nextone, it's said to be one of the best commercial SBC on the market. regards. Saeed Ahmed wrote: > Thanks rod for a quick answer, > > FS is installed on Ubuntu Server. > > I am planning to replace Nextone SBC with FS, Later I'll also use openZAP to > communicate with TDM but this all depends how much calls it can take, or > maybe we can also do something in clustering environment ( I am not sure > about it). But thanks again and any further help will be highly appreciated. > > > Kind Regards > Saeed Ahmed Tariq > > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of rod > Sent: Monday, February 02, 2009 1:53 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Strange Performance when using as SBC > > Hi Saeed, > > I just created an account to share my setup on the wiki. I will detail > all the steps for a clean install of a debian64 lenny with FS used as a > SBC (next step is to try the new LCR module :) )and what I'm doing do > stress the server. > > I wrote nothing at this time so please be patient, I'm waiting for my > new hardware so that I will detail as much as possible what I'll do. > > For beginning I suggest you reading the start page on the wiki, > especially these pages: > -http://wiki.freeswitch.org/wiki/Getting_Started_Guide > -http://wiki.freeswitch.org/wiki/Dialplan_XML > > maybe you could tell more about the linux distribution you're using so > that I can give you some pointers for sipp... > > regards. > rod. > > > Saeed Ahmed wrote: > >> Hi Rod, >> >> Could you please share how you configured Sipp & FS to create a test >> environment? Especially the dial plan, sofia settings etc..., actually I >> > am > >> a newbie. I want to test it on a single FS machine. >> >> Kind Regards >> Saeed >> -----Original Message----- >> From: freeswitch-users-bounces at lists.freeswitch.org >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of rod >> Sent: Monday, February 02, 2009 11:00 AM >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] Strange Performance when using as SBC >> >> Hi Ken, Jay, >> >> thanks for pointing to proxy media, I will test. >> >> Ken, you are right, I was brain damaged (a stupid mistake) when setting >> INFO cause this kind of level could be very verbose. I'm switching to >> CRIT or ERR. >> >> Thanks guys, >> rod. >> >> thanks for >> >> Ken Rice wrote: >> >> >>> If you don't have to transcode, using proxy media mode will still save >>> > you > >>> some CPU time. This is 1/2 way between bypass media and the default media >>> interactive mode. The other draw back to this mode is if you are using FS >>> >>> >> to >> >> >>> clean up RTP and DTMF you loose those functions but they are not needed >>> > in > >>> most use cases. >>> >>> As far as the log level goes, I found that once I had things stable >>> >>> >> setting >> >> >>> the loglevel to helped a good deal... Info is probably a bit too high of >>> > a > >>> loglevel I would probably go for CRIT or ERR (2 or 1 respectively) if you >>> insist on leaving logging turned on... On a busy system these can and >>> > will > >>> generate a good deal of activity (and disk IO if using mod_logfile) >>> >>> Ken >>> >>> >>> >>> >>> >>>> From: rod >>>> Reply-To: >>>> Date: Mon, 02 Feb 2009 11:36:35 +0400 >>>> To: >>>> Subject: Re: [Freeswitch-users] Strange Performance when using as SBC >>>> >>>> Hi Ken, >>>> >>>> 1) I'd like to use FS to hide topology, so bypass media is not possible >>>> 2) done >>>> 3) done >>>> 4) not used >>>> 5) i'm using this ins switch.xml -> >>> value="info"/>, if you think an other log level is more suitable. >>>> >>>> Regarding logging, I can see in console and in the freeswitch.log that >>>> there is still a lot of NOTICE logging, see below: >>>> 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:960 >>>> switch_core_session_thread() Session 8721 >>>> (sofia/internal/sipp at 10.10.10.1:5060) Ended >>>> 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:962 >>>> switch_core_session_thread() Close Channel >>>> sofia/internal/sipp at 10.10.10.1:5060 [CS_HANGUP] >>>> 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:960 >>>> switch_core_session_thread() Session 8722 >>>> (sofia/external/9998 at 10.10.20.100) Ended >>>> 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:962 >>>> switch_core_session_thread() Close Channel >>>> sofia/external/9998 at 10.10.20.100 [CS_HANGUP] >>>> 2009-02-02 08:33:56 [NOTICE] sofia.c:3164 sofia_handle_sip_i_state() >>>> Channel [sofia/external/9998 at 10.10.20.100] has been answered >>>> 2009-02-02 08:33:56 [WARNING] mod_sofia.c:740 sofia_read_frame() >>>> Changing codec ptime to 30. I bet you have a linksys/sipura =D >>>> >>>> Do you have any idea where I can switch off this kind of logging. I >>>> thought it should be in /dialplan/internal.xml, but I see that in >>>> internal.xml -> >>>> >>>> thanks a lot for your suggestion. >>>> >>>> regards, >>>> rod >>>> >>>> Ken Rice wrote: >>>> >>>> >>>> >>>>> Dont forget there are several things you can do to increase >>>>> >>>>> >> performance... >> >> >>>>> 1) where possible use bypass media or media proxy modes >>>>> 2) mount freeswitch/db as a ram drive (if you are using voicemail with >>>>> the internal FS DBs you'll need a way to make this persistant across >>>>> reboots) >>>>> 3) see the wiki for setting reasonable ulimits >>>>> 4) (this is my oppinion others may vary) dont use mod_cdr_csv >>>>> 5) turn off (or reduce logging) in switch.conf.xml >>>>> >>>>> all of these thing can greatly improve performance. >>>>> >>>>> On Mon, Feb 2, 2009 at 1:04 AM, rod >>>> > wrote: >>>>> >>>>> Thanks Anthony, >>>>> >>>>> the setup is like this: >>>>> >>>>> sipp server ---- FS 1 ---- FS2 >>>>> >>>>> FS1 is the AMD CPU that has only one extension in dialplan that >>>>> bridges >>>>> 9999 to FS2. 9999 is the first extension in FS2 dialplan that >>>>> plays moh, >>>>> FS2 has no CPU pbm. >>>>> >>>>> FS1 is maxing out at 60 bridged calls without your option -hp. >>>>> >>>>> Using -hp, I'm now able to bridge 200 concurrent calls (a great >>>>> improvement) and the system is still reactive. CPU load is high >>>>> but not >>>>> 100% and as the system responds well, I think that doesn't matter. >>>>> >>>>> >> The >> >> >>>>> 2GB of memory are completely consumed (top command shows 700MB for >>>>> >>>>> >> FS >> >> >>>>> process). >>>>> >>>>> I understand that FS1 server is not the best hardware platform, >>>>> and I'm >>>>> waiting for new 4 cores server for testing. >>>>> I will update those numbers when testing with the new hardware. >>>>> >>>>> regards, >>>>> rod. >>>>> >>>>> Anthony Minessale wrote: >>>>> >>>>> >>>>> >>>>>> Which of the 2 machines has the load issue? You said it was one box >>>>>> calling the other. >>>>>> >>>>>> You have 2 major things against you, single CPU and AMD, but you >>>>>> should at least be able to get in the vicinity of 800-1000 calls >>>>>> >>>>>> >>>>>> >>>>> on a >>>>> >>>>> >>>>> >>>>>> box like that. >>>>>> >>>>>> Are you calling the default 9999? It's not really an appropriate >>>>>> extension for load testing. >>>>>> On the terminating box you should set up a manual extension that is >>>>>> the first one in the dial plan >>>>>> to play a wav file from preferably a ram disk or /tmp >>>>>> >>>>>> If you do plan on using this in production accept nothing less >>>>>> >>>>>> >>>>>> >>>>> than a >>>>> >>>>> >>>>> >>>>>> multi-core intel machine with at least 4 cores, the more cores the >>>>>> better because that parallel processing is where FS gets it's >>>>>> >>>>>> >>>>>> >>>>> atvantage. >>>>> >>>>> >>>>> >>>>>> On Fri, Jan 30, 2009 at 5:56 AM, rod >>>>> >>>>>> >>>>>> >>>>> >>>>> >>>>> >>>>> >>>>>> >> wrote: >>>>>> >>>>>> Dear list, >>>>>> >>>>>> I've been playing with freeswitch for some time (2 months) >>>>>> >>>>>> >>>>>> >>>>> and the >>>>> >>>>> >>>>> >>>>>> fact >>>>>> is that I'm very pleased with the functionnalities of this >>>>>> >>>>>> >>>>>> >>>>> software. >>>>> >>>>> >>>>> >>>>>> I'd like to use FS as a SBC handling media and I'm doing some >>>>>> tests with >>>>>> sipp to load the machine but I'm unable to bridge more than >>>>>> >>>>>> >>>>>> >>>>> 60 calls >>>>> >>>>> >>>>> >>>>>> without seeing the CPU being loaded at 100%. I'm sure >>>>>> >>>>>> >>>>>> >>>>> something is >>>>> >>>>> >>>>> >>>>>> going >>>>>> wrong with my setup but I'm unable to see what. >>>>>> >>>>>> The test machine has the following specs: >>>>>> Athlon XP 3500+ with 2GB of memory (I know this is not a >>>>>> >>>>>> >>>>>> >>>>> high end >>>>> >>>>> >>>>> >>>>>> machine :p) >>>>>> >>>>>> Freeswitch:/opt/freeswitch/log# cat /proc/cpuinfo >>>>>> processor : 0 >>>>>> vendor_id : AuthenticAMD >>>>>> cpu family : 15 >>>>>> model : 95 >>>>>> model name : AMD Athlon(tm) 64 Processor 3500+ >>>>>> stepping : 2 >>>>>> cpu MHz : 2199.973 >>>>>> cache size : 512 KB >>>>>> fpu : yes >>>>>> fpu_exception : yes >>>>>> cpuid level : 1 >>>>>> wp : yes >>>>>> flags : fpu vme de pse tsc msr pae mce cx8 apic >>>>>> >>>>>> >>>>>> >>>>> sep mtrr pge >>>>> >>>>> >>>>> >>>>>> mca cmov pat pse36 clflush mmx fxsr sse sse2 syscall nx mmxext >>>>>> fxsr_opt >>>>>> rdtscp lm 3dnowext 3dnow up rep_good pni cx16 lahf_lm svm >>>>>> >>>>>> >>>>>> >>>>> extapic >>>>> >>>>> >>>>> >>>>>> cr8_legacy >>>>>> bogomips : 4402.97 >>>>>> TLB size : 1024 4K pages >>>>>> clflush size : 64 >>>>>> cache_alignment : 64 >>>>>> address sizes : 40 bits physical, 48 bits virtual >>>>>> power management: ts fid vid ttp tm stc >>>>>> >>>>>> I installed FS on a fresh debian 64: >>>>>> Linux Freeswitch 2.6.26-1-amd64 #1 SMP Sat Jan 10 17:57:00 >>>>>> >>>>>> >>>>>> >>>>> UTC 2009 >>>>> >>>>> >>>>> >>>>>> x86_64 GNU/Linux >>>>>> >>>>>> I set the ulimit parameters like those on the website: >>>>>> freeswitch at internal> ... >>>>>> Freeswitch:/opt/free-svn/bin# ulimit -a >>>>>> core file size (blocks, -c) unlimited >>>>>> data seg size (kbytes, -d) unlimited >>>>>> scheduling priority (-e) 0 >>>>>> file size (blocks, -f) unlimited >>>>>> pending signals (-i) unlimited >>>>>> max locked memory (kbytes, -l) unlimited >>>>>> max memory size (kbytes, -m) unlimited >>>>>> open files (-n) 999999 >>>>>> pipe size (512 bytes, -p) 8 >>>>>> POSIX message queues (bytes, -q) unlimited >>>>>> real-time priority (-r) 0 >>>>>> stack size (kbytes, -s) 244 >>>>>> cpu time (seconds, -t) unlimited >>>>>> max user processes (-u) unlimited >>>>>> virtual memory (kbytes, -v) unlimited >>>>>> file locks (-x) unlimited >>>>>> >>>>>> >>>>>> My network setup is the following: >>>>>> >>>>>> SIPP machine (10.10.10.1/24)----------------vlan >>>>>> >>>>>> >>>>>> >>>>> >>>>> >>>>> >>>>> >>>>>> 55 >>>>>> ----------(10.10.10.254/24 >>>>>> >>>>>> >>>>>> >>>>> ) FS >>>>> >>>>> >>>>> >>>>>> (10.10.20.254/24)-------------- >>>>>> >>>>>> >>>>>> >>>>> >>>>> >>>>> >>>>> >>>>>> vlan56 >>>>>> -------------------(10.10.20.100/24 >>>>>> >>>>>> >>>>>> >>>>> ) >>>>> >>>>> >>>>> >>>>>> OTHER STOCK FS >>>>>> >>>>>> >>>>>> I launched sipp with: >>>>>> sipp -sn uac_pcap -s 9999 -r 10 -l 80 -d 60000 -mi >>>>>> >>>>>> >>>>>> >>>>> 10.10.10.1 -i >>>>> >>>>> >>>>> >>>>>> 10.10.10.1 -mp 25000 10.10.10.254:5060 >>>>>> >>>>>> >>>>>> >>>>> >>>>> >>>>> >>>>> >>>>>> The dialplan on FS is very simple: >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>> data="sofia/external/9999 at 10.10.20.100 >>>>>> >>>>>> >>>>>> >>>>> >>>> >"/> >>>>> >>>>> >>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> FreeSWITCH Version 1.0.trunk (11560M) Started. >>>>>> Crash Protection [Disabled] >>>>>> Max Sessions[1000] >>>>>> Session Rate[100] >>>>>> SQL [Enabled] >>>>>> >>>>>> >>>>>> The test is very simple: sipp dial 9999 that matches in my >>>>>> >>>>>> >>>>>> >>>>> FS dialplan >>>>> >>>>> >>>>> >>>>>> and this is bridged to an other FS machine playing music on >>>>>> >>>>>> >>>>>> >>>>> hold. >>>>> >>>>> >>>>> >>>>>> When I launch "top" I see after 30 to 40 s that FS consumes all >>>>>> the CPU >>>>>> ressources (with a mean of 50-60 % before), with 80 calls. >>>>>> When I set 70 calls, I have to wait 70-80 s before seeing >>>>>> >>>>>> >>>>>> >>>>> the same >>>>> >>>>> >>>>> >>>>>> issue. >>>>>> >>>>>> Presence is set to false on the 2 profile. >>>>>> >>>>>> I have the same issue with FS 1.0.2 that' s why I tried FS >>>>>> >>>>>> >>>>>> >>>>> 11560. >>>>> >>>>> >>>>> >>>>>> When I use the FS machine as a router to test the packet per >>>>>> >>>>>> >>>>>> >>>>> second >>>>> >>>>> >>>>> >>>>>> performance, I'm reaching 100Mbps with 8000pps in each >>>>>> >>>>>> >>>>>> >>>>> direction (from >>>>> >>>>> >>>>> >>>>>> vlan 55 to vlan56) with less than 12% CPU. So that I don't think >>>>>> there's >>>>>> an issue with the network. >>>>>> >>>>>> Here is an "mpstat -P ALL 1" to show you what's happening >>>>>> >>>>>> >>>>>> >>>>> suddenly >>>>> >>>>> >>>>> >>>>>> with >>>>>> 70 bridge calls: >>>>>> 12:31:26 CPU %user %nice %sys %iowait %irq %soft >>>>>> %steal %idle intr/s >>>>>> 12:31:27 all 3,00 0,00 3,00 0,00 1,00 4,00 >>>>>> 0,00 89,00 6241,00 >>>>>> 12:31:27 0 3,00 0,00 3,00 0,00 1,00 4,00 >>>>>> 0,00 89,00 6241,00 >>>>>> >>>>>> 12:31:27 CPU %user %nice %sys %iowait %irq %soft >>>>>> %steal %idle intr/s >>>>>> 12:31:28 all 14,14 0,00 56,57 0,00 2,02 5,05 >>>>>> 0,00 22,22 6035,35 >>>>>> 12:31:28 0 14,14 0,00 56,57 0,00 2,02 5,05 >>>>>> 0,00 22,22 6035,35 >>>>>> >>>>>> 12:31:28 CPU %user %nice %sys %iowait %irq %soft >>>>>> %steal %idle intr/s >>>>>> 12:31:29 all 24,75 0,00 67,33 0,00 0,99 6,93 >>>>>> 0,00 0,00 5483,17 >>>>>> 12:31:29 0 24,75 0,00 67,33 0,00 0,99 6,93 >>>>>> 0,00 0,00 5483,17 >>>>>> >>>>>> >>>>>> The CPU is going from 89% idle to 0% in less than 2 seconds. >>>>>> >>>>>> I know that I don't have to expect too much from this kind of >>>>>> hardware, >>>>>> but it seems strange that the CPU power vanished so suddenly. >>>>>> >>>>>> Thanks a lot for the guys that have read this long mail :p >>>>>> >>>>>> kind regards, >>>>>> rod >>>>>> >>>>>> >>>>>> _______________________________________________ >>>>>> Freeswitch-users mailing list >>>>>> Freeswitch-users at lists.freeswitch.org >>>>>> >>>>>> >>>>>> >>>>> >>>>> >>>>> >>>>> >>>>>> >>>>> >>>>>> >>>>>> >>>>> > >>>>> >>>>> >>>>> >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> >>>>>> >>>>>> >>>>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> >>>>> >>>>> >>>>> >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> -- >>>>>> Anthony Minessale II >>>>>> >>>>>> FreeSWITCH http://www.freeswitch.org/ >>>>>> ClueCon http://www.cluecon.com/ >>>>>> >>>>>> AIM: anthm >>>>>> MSN:anthony_minessale at hotmail.com >>>>>> >>>>>> >>>>>> >>>>> >>>>> >>>>> >>>>> >>>>>> >>>>> >>>>>> >>>>>> >>>>> > >>>>> >>>>> >>>>> >>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>>>>> >>>>>> >>>>>> >>>>> >>>>> >>>>> >>>>> >>>>>> >>>>> >>>>>> >>>>>> >>>>> > >>>>> >>>>> >>>>> >>>>>> IRC: irc.freenode.net >>>>>> >>>>>> >>>>>> >>>>> #freeswitch >>>>> >>>>> >>>>> >>>>>> FreeSWITCH Developer Conference >>>>>> sip:888 at conference.freeswitch.org >>>>>> >>>>>> >>>>>> >>>>> >>>>> >>>>> >>>>> >>>>>> >>>>> >>>>>> >>>>>> >>>>> > >>>>> >>>>> >>>>> >>>>>> iax:guest at conference.freeswitch.org/888 >>>>>> >>>>>> >>>>>> >>>>> >>>>> >>>>> >>>>> >>>>>> >>>>>> googletalk:conf+888 at conference.freeswitch.org >>>>>> >>>>>> >>>>>> >>>>> >>>>> >>>>> >>>>> >>>>>> >>>>> >>>>>> >>>>>> >>>>> > >>>>> >>>>> >>>>> >>>>>> pstn:213-799-1400 >>>>>> >>>>>> >>>>>> >>>>>> >> ------------------------------------------------------------------------ >> >> >>>>> >>>>> >>>>> >>>>>> _______________________________________________ >>>>>> Freeswitch-users mailing list >>>>>> Freeswitch-users at lists.freeswitch.org >>>>>> >>>>>> >>>>>> >>>>> >>>>> >>>>> >>>>> >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> >>>>>> >>>>>> >>>>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> >>>>> >>>>> >>>>> >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>>> >>>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> >>>>> >>>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> >>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> >>>>> > ------------------------------------------------------------------------ > >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> >>>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> >>>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >>> >>> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > From saeedahmad1981 at gmail.com Mon Feb 2 07:15:01 2009 From: saeedahmad1981 at gmail.com (Saeed Ahmed) Date: Mon, 2 Feb 2009 16:15:01 +0100 Subject: [Freeswitch-users] Strange Performance when using as SBC In-Reply-To: <49870432.5050301@laposte.net> References: <4986C42C.7030700@laposte.net><1C31435E19AB4E57981CF2AAC2CA74AA@SaeedLaptop> <4986ECA9.3040707@laposte.net><9232C06D5362494791AF713E1DF61343@SaeedLaptop> <49870432.5050301@laposte.net> Message-ID: <47EAE16A9F2547488578BCF6C0884BBD@SaeedLaptop> Thanks Rod, Its really helpful contribution. @Nextone: I don't want to say much about it, but simply I am not happy with it, have you heard someone satisfied with NX who also owns it? Kind Regards Saeed Ahmed Tariq -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of rod Sent: Monday, February 02, 2009 3:33 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Strange Performance when using as SBC Hi Saeed, Here is a first draft of what I did to install FS on my server. Configuration are not present, they'll be in a next release :p http://wiki.freeswitch.org/wiki/SBC_Setup My aim is to setup FS as a SBC, I hope this page could be a great startup point for others. I will update regularly based on what I did. Saeed, why are you replacing your Nextone, it's said to be one of the best commercial SBC on the market. regards. Saeed Ahmed wrote: > Thanks rod for a quick answer, > > FS is installed on Ubuntu Server. > > I am planning to replace Nextone SBC with FS, Later I'll also use openZAP to > communicate with TDM but this all depends how much calls it can take, or > maybe we can also do something in clustering environment ( I am not sure > about it). But thanks again and any further help will be highly appreciated. > > > Kind Regards > Saeed Ahmed Tariq > > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of rod > Sent: Monday, February 02, 2009 1:53 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Strange Performance when using as SBC > > Hi Saeed, > > I just created an account to share my setup on the wiki. I will detail > all the steps for a clean install of a debian64 lenny with FS used as a > SBC (next step is to try the new LCR module :) )and what I'm doing do > stress the server. > > I wrote nothing at this time so please be patient, I'm waiting for my > new hardware so that I will detail as much as possible what I'll do. > > For beginning I suggest you reading the start page on the wiki, > especially these pages: > -http://wiki.freeswitch.org/wiki/Getting_Started_Guide > -http://wiki.freeswitch.org/wiki/Dialplan_XML > > maybe you could tell more about the linux distribution you're using so > that I can give you some pointers for sipp... > > regards. > rod. > > > Saeed Ahmed wrote: > >> Hi Rod, >> >> Could you please share how you configured Sipp & FS to create a test >> environment? Especially the dial plan, sofia settings etc..., actually I >> > am > >> a newbie. I want to test it on a single FS machine. >> >> Kind Regards >> Saeed >> -----Original Message----- >> From: freeswitch-users-bounces at lists.freeswitch.org >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of rod >> Sent: Monday, February 02, 2009 11:00 AM >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] Strange Performance when using as SBC >> >> Hi Ken, Jay, >> >> thanks for pointing to proxy media, I will test. >> >> Ken, you are right, I was brain damaged (a stupid mistake) when setting >> INFO cause this kind of level could be very verbose. I'm switching to >> CRIT or ERR. >> >> Thanks guys, >> rod. >> >> thanks for >> >> Ken Rice wrote: >> >> >>> If you don't have to transcode, using proxy media mode will still save >>> > you > >>> some CPU time. This is 1/2 way between bypass media and the default media >>> interactive mode. The other draw back to this mode is if you are using FS >>> >>> >> to >> >> >>> clean up RTP and DTMF you loose those functions but they are not needed >>> > in > >>> most use cases. >>> >>> As far as the log level goes, I found that once I had things stable >>> >>> >> setting >> >> >>> the loglevel to helped a good deal... Info is probably a bit too high of >>> > a > >>> loglevel I would probably go for CRIT or ERR (2 or 1 respectively) if you >>> insist on leaving logging turned on... On a busy system these can and >>> > will > >>> generate a good deal of activity (and disk IO if using mod_logfile) >>> >>> Ken >>> >>> >>> >>> >>> >>>> From: rod >>>> Reply-To: >>>> Date: Mon, 02 Feb 2009 11:36:35 +0400 >>>> To: >>>> Subject: Re: [Freeswitch-users] Strange Performance when using as SBC >>>> >>>> Hi Ken, >>>> >>>> 1) I'd like to use FS to hide topology, so bypass media is not possible >>>> 2) done >>>> 3) done >>>> 4) not used >>>> 5) i'm using this ins switch.xml -> >>> value="info"/>, if you think an other log level is more suitable. >>>> >>>> Regarding logging, I can see in console and in the freeswitch.log that >>>> there is still a lot of NOTICE logging, see below: >>>> 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:960 >>>> switch_core_session_thread() Session 8721 >>>> (sofia/internal/sipp at 10.10.10.1:5060) Ended >>>> 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:962 >>>> switch_core_session_thread() Close Channel >>>> sofia/internal/sipp at 10.10.10.1:5060 [CS_HANGUP] >>>> 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:960 >>>> switch_core_session_thread() Session 8722 >>>> (sofia/external/9998 at 10.10.20.100) Ended >>>> 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:962 >>>> switch_core_session_thread() Close Channel >>>> sofia/external/9998 at 10.10.20.100 [CS_HANGUP] >>>> 2009-02-02 08:33:56 [NOTICE] sofia.c:3164 sofia_handle_sip_i_state() >>>> Channel [sofia/external/9998 at 10.10.20.100] has been answered >>>> 2009-02-02 08:33:56 [WARNING] mod_sofia.c:740 sofia_read_frame() >>>> Changing codec ptime to 30. I bet you have a linksys/sipura =D >>>> >>>> Do you have any idea where I can switch off this kind of logging. I >>>> thought it should be in /dialplan/internal.xml, but I see that in >>>> internal.xml -> >>>> >>>> thanks a lot for your suggestion. >>>> >>>> regards, >>>> rod >>>> >>>> Ken Rice wrote: >>>> >>>> >>>> >>>>> Dont forget there are several things you can do to increase >>>>> >>>>> >> performance... >> >> >>>>> 1) where possible use bypass media or media proxy modes >>>>> 2) mount freeswitch/db as a ram drive (if you are using voicemail with >>>>> the internal FS DBs you'll need a way to make this persistant across >>>>> reboots) >>>>> 3) see the wiki for setting reasonable ulimits >>>>> 4) (this is my oppinion others may vary) dont use mod_cdr_csv >>>>> 5) turn off (or reduce logging) in switch.conf.xml >>>>> >>>>> all of these thing can greatly improve performance. >>>>> >>>>> On Mon, Feb 2, 2009 at 1:04 AM, rod >>>> > wrote: >>>>> >>>>> Thanks Anthony, >>>>> >>>>> the setup is like this: >>>>> >>>>> sipp server ---- FS 1 ---- FS2 >>>>> >>>>> FS1 is the AMD CPU that has only one extension in dialplan that >>>>> bridges >>>>> 9999 to FS2. 9999 is the first extension in FS2 dialplan that >>>>> plays moh, >>>>> FS2 has no CPU pbm. >>>>> >>>>> FS1 is maxing out at 60 bridged calls without your option -hp. >>>>> >>>>> Using -hp, I'm now able to bridge 200 concurrent calls (a great >>>>> improvement) and the system is still reactive. CPU load is high >>>>> but not >>>>> 100% and as the system responds well, I think that doesn't matter. >>>>> >>>>> >> The >> >> >>>>> 2GB of memory are completely consumed (top command shows 700MB for >>>>> >>>>> >> FS >> >> >>>>> process). >>>>> >>>>> I understand that FS1 server is not the best hardware platform, >>>>> and I'm >>>>> waiting for new 4 cores server for testing. >>>>> I will update those numbers when testing with the new hardware. >>>>> >>>>> regards, >>>>> rod. >>>>> >>>>> Anthony Minessale wrote: >>>>> >>>>> >>>>> >>>>>> Which of the 2 machines has the load issue? You said it was one box >>>>>> calling the other. >>>>>> >>>>>> You have 2 major things against you, single CPU and AMD, but you >>>>>> should at least be able to get in the vicinity of 800-1000 calls >>>>>> >>>>>> >>>>>> >>>>> on a >>>>> >>>>> >>>>> >>>>>> box like that. >>>>>> >>>>>> Are you calling the default 9999? It's not really an appropriate >>>>>> extension for load testing. >>>>>> On the terminating box you should set up a manual extension that is >>>>>> the first one in the dial plan >>>>>> to play a wav file from preferably a ram disk or /tmp >>>>>> >>>>>> If you do plan on using this in production accept nothing less >>>>>> >>>>>> >>>>>> >>>>> than a >>>>> >>>>> >>>>> >>>>>> multi-core intel machine with at least 4 cores, the more cores the >>>>>> better because that parallel processing is where FS gets it's >>>>>> >>>>>> >>>>>> >>>>> atvantage. >>>>> >>>>> >>>>> >>>>>> On Fri, Jan 30, 2009 at 5:56 AM, rod >>>>> >>>>>> >>>>>> >>>>> >>>>> >>>>> >>>>> >>>>>> >> wrote: >>>>>> >>>>>> Dear list, >>>>>> >>>>>> I've been playing with freeswitch for some time (2 months) >>>>>> >>>>>> >>>>>> >>>>> and the >>>>> >>>>> >>>>> >>>>>> fact >>>>>> is that I'm very pleased with the functionnalities of this >>>>>> >>>>>> >>>>>> >>>>> software. >>>>> >>>>> >>>>> >>>>>> I'd like to use FS as a SBC handling media and I'm doing some >>>>>> tests with >>>>>> sipp to load the machine but I'm unable to bridge more than >>>>>> >>>>>> >>>>>> >>>>> 60 calls >>>>> >>>>> >>>>> >>>>>> without seeing the CPU being loaded at 100%. I'm sure >>>>>> >>>>>> >>>>>> >>>>> something is >>>>> >>>>> >>>>> >>>>>> going >>>>>> wrong with my setup but I'm unable to see what. >>>>>> >>>>>> The test machine has the following specs: >>>>>> Athlon XP 3500+ with 2GB of memory (I know this is not a >>>>>> >>>>>> >>>>>> >>>>> high end >>>>> >>>>> >>>>> >>>>>> machine :p) >>>>>> >>>>>> Freeswitch:/opt/freeswitch/log# cat /proc/cpuinfo >>>>>> processor : 0 >>>>>> vendor_id : AuthenticAMD >>>>>> cpu family : 15 >>>>>> model : 95 >>>>>> model name : AMD Athlon(tm) 64 Processor 3500+ >>>>>> stepping : 2 >>>>>> cpu MHz : 2199.973 >>>>>> cache size : 512 KB >>>>>> fpu : yes >>>>>> fpu_exception : yes >>>>>> cpuid level : 1 >>>>>> wp : yes >>>>>> flags : fpu vme de pse tsc msr pae mce cx8 apic >>>>>> >>>>>> >>>>>> >>>>> sep mtrr pge >>>>> >>>>> >>>>> >>>>>> mca cmov pat pse36 clflush mmx fxsr sse sse2 syscall nx mmxext >>>>>> fxsr_opt >>>>>> rdtscp lm 3dnowext 3dnow up rep_good pni cx16 lahf_lm svm >>>>>> >>>>>> >>>>>> >>>>> extapic >>>>> >>>>> >>>>> >>>>>> cr8_legacy >>>>>> bogomips : 4402.97 >>>>>> TLB size : 1024 4K pages >>>>>> clflush size : 64 >>>>>> cache_alignment : 64 >>>>>> address sizes : 40 bits physical, 48 bits virtual >>>>>> power management: ts fid vid ttp tm stc >>>>>> >>>>>> I installed FS on a fresh debian 64: >>>>>> Linux Freeswitch 2.6.26-1-amd64 #1 SMP Sat Jan 10 17:57:00 >>>>>> >>>>>> >>>>>> >>>>> UTC 2009 >>>>> >>>>> >>>>> >>>>>> x86_64 GNU/Linux >>>>>> >>>>>> I set the ulimit parameters like those on the website: >>>>>> freeswitch at internal> ... >>>>>> Freeswitch:/opt/free-svn/bin# ulimit -a >>>>>> core file size (blocks, -c) unlimited >>>>>> data seg size (kbytes, -d) unlimited >>>>>> scheduling priority (-e) 0 >>>>>> file size (blocks, -f) unlimited >>>>>> pending signals (-i) unlimited >>>>>> max locked memory (kbytes, -l) unlimited >>>>>> max memory size (kbytes, -m) unlimited >>>>>> open files (-n) 999999 >>>>>> pipe size (512 bytes, -p) 8 >>>>>> POSIX message queues (bytes, -q) unlimited >>>>>> real-time priority (-r) 0 >>>>>> stack size (kbytes, -s) 244 >>>>>> cpu time (seconds, -t) unlimited >>>>>> max user processes (-u) unlimited >>>>>> virtual memory (kbytes, -v) unlimited >>>>>> file locks (-x) unlimited >>>>>> >>>>>> >>>>>> My network setup is the following: >>>>>> >>>>>> SIPP machine (10.10.10.1/24)----------------vlan >>>>>> >>>>>> >>>>>> >>>>> >>>>> >>>>> >>>>> >>>>>> 55 >>>>>> ----------(10.10.10.254/24 >>>>>> >>>>>> >>>>>> >>>>> ) FS >>>>> >>>>> >>>>> >>>>>> (10.10.20.254/24)-------------- >>>>>> >>>>>> >>>>>> >>>>> >>>>> >>>>> >>>>> >>>>>> vlan56 >>>>>> -------------------(10.10.20.100/24 >>>>>> >>>>>> >>>>>> >>>>> ) >>>>> >>>>> >>>>> >>>>>> OTHER STOCK FS >>>>>> >>>>>> >>>>>> I launched sipp with: >>>>>> sipp -sn uac_pcap -s 9999 -r 10 -l 80 -d 60000 -mi >>>>>> >>>>>> >>>>>> >>>>> 10.10.10.1 -i >>>>> >>>>> >>>>> >>>>>> 10.10.10.1 -mp 25000 10.10.10.254:5060 >>>>>> >>>>>> >>>>>> >>>>> >>>>> >>>>> >>>>> >>>>>> The dialplan on FS is very simple: >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>> data="sofia/external/9999 at 10.10.20.100 >>>>>> >>>>>> >>>>>> >>>>> >>>> >"/> >>>>> >>>>> >>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> FreeSWITCH Version 1.0.trunk (11560M) Started. >>>>>> Crash Protection [Disabled] >>>>>> Max Sessions[1000] >>>>>> Session Rate[100] >>>>>> SQL [Enabled] >>>>>> >>>>>> >>>>>> The test is very simple: sipp dial 9999 that matches in my >>>>>> >>>>>> >>>>>> >>>>> FS dialplan >>>>> >>>>> >>>>> >>>>>> and this is bridged to an other FS machine playing music on >>>>>> >>>>>> >>>>>> >>>>> hold. >>>>> >>>>> >>>>> >>>>>> When I launch "top" I see after 30 to 40 s that FS consumes all >>>>>> the CPU >>>>>> ressources (with a mean of 50-60 % before), with 80 calls. >>>>>> When I set 70 calls, I have to wait 70-80 s before seeing >>>>>> >>>>>> >>>>>> >>>>> the same >>>>> >>>>> >>>>> >>>>>> issue. >>>>>> >>>>>> Presence is set to false on the 2 profile. >>>>>> >>>>>> I have the same issue with FS 1.0.2 that' s why I tried FS >>>>>> >>>>>> >>>>>> >>>>> 11560. >>>>> >>>>> >>>>> >>>>>> When I use the FS machine as a router to test the packet per >>>>>> >>>>>> >>>>>> >>>>> second >>>>> >>>>> >>>>> >>>>>> performance, I'm reaching 100Mbps with 8000pps in each >>>>>> >>>>>> >>>>>> >>>>> direction (from >>>>> >>>>> >>>>> >>>>>> vlan 55 to vlan56) with less than 12% CPU. So that I don't think >>>>>> there's >>>>>> an issue with the network. >>>>>> >>>>>> Here is an "mpstat -P ALL 1" to show you what's happening >>>>>> >>>>>> >>>>>> >>>>> suddenly >>>>> >>>>> >>>>> >>>>>> with >>>>>> 70 bridge calls: >>>>>> 12:31:26 CPU %user %nice %sys %iowait %irq %soft >>>>>> %steal %idle intr/s >>>>>> 12:31:27 all 3,00 0,00 3,00 0,00 1,00 4,00 >>>>>> 0,00 89,00 6241,00 >>>>>> 12:31:27 0 3,00 0,00 3,00 0,00 1,00 4,00 >>>>>> 0,00 89,00 6241,00 >>>>>> >>>>>> 12:31:27 CPU %user %nice %sys %iowait %irq %soft >>>>>> %steal %idle intr/s >>>>>> 12:31:28 all 14,14 0,00 56,57 0,00 2,02 5,05 >>>>>> 0,00 22,22 6035,35 >>>>>> 12:31:28 0 14,14 0,00 56,57 0,00 2,02 5,05 >>>>>> 0,00 22,22 6035,35 >>>>>> >>>>>> 12:31:28 CPU %user %nice %sys %iowait %irq %soft >>>>>> %steal %idle intr/s >>>>>> 12:31:29 all 24,75 0,00 67,33 0,00 0,99 6,93 >>>>>> 0,00 0,00 5483,17 >>>>>> 12:31:29 0 24,75 0,00 67,33 0,00 0,99 6,93 >>>>>> 0,00 0,00 5483,17 >>>>>> >>>>>> >>>>>> The CPU is going from 89% idle to 0% in less than 2 seconds. >>>>>> >>>>>> I know that I don't have to expect too much from this kind of >>>>>> hardware, >>>>>> but it seems strange that the CPU power vanished so suddenly. >>>>>> >>>>>> Thanks a lot for the guys that have read this long mail :p >>>>>> >>>>>> kind regards, >>>>>> rod >>>>>> >>>>>> >>>>>> _______________________________________________ >>>>>> Freeswitch-users mailing list >>>>>> Freeswitch-users at lists.freeswitch.org >>>>>> >>>>>> >>>>>> >>>>> >>>>> >>>>> >>>>> >>>>>> >>>>> >>>>>> >>>>>> >>>>> > >>>>> >>>>> >>>>> >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> >>>>>> >>>>>> >>>>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> >>>>> >>>>> >>>>> >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> -- >>>>>> Anthony Minessale II >>>>>> >>>>>> FreeSWITCH http://www.freeswitch.org/ >>>>>> ClueCon http://www.cluecon.com/ >>>>>> >>>>>> AIM: anthm >>>>>> MSN:anthony_minessale at hotmail.com >>>>>> >>>>>> >>>>>> >>>>> >>>>> >>>>> >>>>> >>>>>> >>>>> >>>>>> >>>>>> >>>>> > >>>>> >>>>> >>>>> >>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>>>>> >>>>>> >>>>>> >>>>> >>>>> >>>>> >>>>> >>>>>> >>>>> >>>>>> >>>>>> >>>>> > >>>>> >>>>> >>>>> >>>>>> IRC: irc.freenode.net >>>>>> >>>>>> >>>>>> >>>>> #freeswitch >>>>> >>>>> >>>>> >>>>>> FreeSWITCH Developer Conference >>>>>> sip:888 at conference.freeswitch.org >>>>>> >>>>>> >>>>>> >>>>> >>>>> >>>>> >>>>> >>>>>> >>>>> >>>>>> >>>>>> >>>>> > >>>>> >>>>> >>>>> >>>>>> iax:guest at conference.freeswitch.org/888 >>>>>> >>>>>> >>>>>> >>>>> >>>>> >>>>> >>>>> >>>>>> >>>>>> googletalk:conf+888 at conference.freeswitch.org >>>>>> >>>>>> >>>>>> >>>>> >>>>> >>>>> >>>>> >>>>>> >>>>> >>>>>> >>>>>> >>>>> > >>>>> >>>>> >>>>> >>>>>> pstn:213-799-1400 >>>>>> >>>>>> >>>>>> >>>>>> >> ------------------------------------------------------------------------ >> >> >>>>> >>>>> >>>>> >>>>>> _______________________________________________ >>>>>> Freeswitch-users mailing list >>>>>> Freeswitch-users at lists.freeswitch.org >>>>>> >>>>>> >>>>>> >>>>> >>>>> >>>>> >>>>> >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> >>>>>> >>>>>> >>>>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> >>>>> >>>>> >>>>> >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>>> >>>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> >>>>> >>>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> >>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> >>>>> > ------------------------------------------------------------------------ > >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> >>>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> >>>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >>> >>> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From pmhshz at gmail.com Mon Feb 2 07:21:32 2009 From: pmhshz at gmail.com (shehzad p) Date: Mon, 2 Feb 2009 07:21:32 -0800 (PST) Subject: [Freeswitch-users] Call Variable not available when call hangup In-Reply-To: <191c3a030902020606r1a42ef44n7a73bd1e5157392e@mail.gmail.com> References: <21788550.post@talk.nabble.com> <21789152.post@talk.nabble.com> <191c3a030902020606r1a42ef44n7a73bd1e5157392e@mail.gmail.com> Message-ID: <21791503.post@talk.nabble.com> one question is that when javascript is being called from dial plan, I get the session object already available, It is for A leg of channel, So when javascript is called after Bridge how can I get the session object for B leg also? Anthony Minessale-2 wrote: > > the leg you are running the script on is not hungup, the other leg of the > call is. > > If it was hungup you would not be executing the script. > > Asterisk and the h ext and the whole dead-agi thing are all poor design > showing it's teeth. > We do not support anything like it. > > > You can however try this: (see the link below) > > http://www.nabble.com/Re:-Conference-javascript-and-hanuphooks-giving-me-headaches-p21614840.html > > > > On Mon, Feb 2, 2009 at 6:53 AM, shehzad p wrote: > >> >> Is there any settings that when call hangup control can be transferred to >> another context and these CDR values can be accessible there? (just like >> in >> Asterisk, h extension) >> >> shehzad p wrote: >> > >> > Hi all, >> > >> > I need to process some CDR variables in Dialplan, like call duration, >> > Answered time etc. >> > but when I place info application after bridge, it is not listing them >> > properly as below: >> > =========================================== >> > Caller-Channel-Created-Time: [1233573341672157] >> > Caller-Channel-Answered-Time: [1233573342712939] >> > Caller-Channel-Hangup-Time: [0] >> > ========================================== >> > Here Hangup time is 0, So how can I find actual values? >> > >> > --I know that we can use xml_cdr or cdr_csv, but my current need is to >> get >> > those values from dialplan itself so that can be passed to some >> script... >> > >> > >> > thanks, >> > msp >> > >> >> -- >> View this message in context: >> http://www.nabble.com/Call-Variable-not-available-when-call-hangup-tp21788550p21789152.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/Call-Variable-not-available-when-call-hangup-tp21788550p21791503.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From clif at eugeneweb.com Sun Feb 1 13:24:20 2009 From: clif at eugeneweb.com (clif at eugeneweb.com) Date: Sun, 1 Feb 2009 13:24:20 -0800 (PST) Subject: [Freeswitch-users] How do I set my FS internal ip address to a "static" value. Message-ID: Hi Gang, I've been struggleing with this also. Actually I can get it to bind to my address, the problem is it randomly drops my calls. :-( I have a FS running on a box with a static IP and I can start a call between two extensions and it will go for hours. Then I add anther interface say eth0:0 with a new static IP and reconfigure my phones and FS to use that, and the calls drop after about 15-20 mins. Though it's pretty random. Here is my setup. I have Debian Linux 2.6.23.1 kernel, and freeswitch-1.0.1. Here is my /etc/network/interfaces: # /etc/network/interfaces -- configuration file for ifup(8), ifdown(8) # The loopback interface auto lo iface lo inet loopback # The first network card - this entry was created during the Debian installation auto eth0 eth0:0 iface eth0 inet dhcp iface eth0:0 inet static address 192.168.0.249 netmask 255.255.255.0 gateway 192.168.0.254 The only change I made to the FS config is in Vars.xml. I added this line close to the top: Here is the console log of the call being dropped: freeswitch at archive> sofia status API CALL [sofia(status)] output: Name Type Data State ================================================================================================= external profile sip:mod_sofia at 67.171.158.226:5080 RUNNING (0) internal profile sip:mod_sofia at 192.168.0.249:5060 RUNNING (2) nat profile sip:mod_sofia at 67.171.158.226:5070 RUNNING (0) default alias internal ALIASED outbound alias external ALIASED 192.168.0.249 alias internal ALIASED ================================================================================================= 3 profiles 3 aliases freeswitch at archive> 2009-02-01 13:23:19 [NOTICE] sofia_glue.c:2634 sofia_glue_restart_all_profiles() Reload XML [Success] 2009-02-01 13:23:19 [INFO] mod_enum.c:817 event_handler() ENUM Reloaded 2009-02-01 13:23:19 [NOTICE] mod_sofia.c:568 sofia_read_frame() Hangup sofia/internal/1003 at 192.168.0.53:5060;user=phone;transport=udp;fs_nat=yes [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 2009-02-01 13:23:19 [NOTICE] switch_ivr_bridge.c:820 switch_ivr_multi_threaded_bridge() Hangup sofia/internal/1001 at 192.168.0.249 [CS_EXECUTE] [NORMAL_CLEARING] 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:807 switch_core_session_thread() Session 6 (sofia/internal/1003 at 192.168.0.53:5060;user=phone;transport=udp;fs_nat=yes) Ended 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:809 switch_core_session_thread() Close Channel sofia/internal/1003 at 192.168.0.53:5060;user=phone;transport=udp;fs_nat=yes [CS_HANGUP] 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:807 switch_core_session_thread() Session 5 (sofia/internal/1001 at 192.168.0.249) Ended 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:809 switch_core_session_thread() Close Channel sofia/internal/1001 at 192.168.0.249 [CS_HANGUP] 2009-02-01 13:23:19 [NOTICE] sofia.c:645 sofia_profile_thread_run() waiting for worker thread 2009-02-01 13:23:19 [NOTICE] sofia.c:645 sofia_profile_thread_run() waiting for worker thread 2009-02-01 13:23:19 [NOTICE] sofia.c:1865 config_sofia() Adding Alias [192.168.0.249] for profile [internal] 2009-02-01 13:23:19 [NOTICE] sofia.c:1865 config_sofia() Adding Alias [default] for profile [internal] 2009-02-01 13:23:19 [NOTICE] sofia.c:1875 config_sofia() Started Profile internal [sofia_reg_internal] 2009-02-01 13:23:20 [NOTICE] sofia.c:1865 config_sofia() Adding Alias [outbound] for profile [external] 2009-02-01 13:23:20 [NOTICE] sofia.c:1875 config_sofia() Started Profile external [sofia_reg_external] 2009-02-01 13:23:20 [NOTICE] sofia.c:645 sofia_profile_thread_run() waiting for worker thread 2009-02-01 13:23:20 [NOTICE] sofia.c:1875 config_sofia() Started Profile nat [sofia_reg_nat] sofia status API CALL [sofia(status)] output: Name Type Data State ================================================================================================= external profile sip:mod_sofia at 67.171.158.226:5080 RUNNING (0) internal profile sip:mod_sofia at 192.168.0.249:5060 RUNNING (0) outbound alias external ALIASED 192.168.0.249 alias internal ALIASED nat profile sip:mod_sofia at 67.171.158.226:5070 RUNNING (0) default alias internal ALIASED ================================================================================================= 3 profiles 3 aliases There is an older thread that says one should set but in this (later) thread is says only Jingleling usese that variable. ie. see: http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg00695.html http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg07345.html So what do you think causes this? What is the correct way? ;-) Thanks, Clif From anthony.minessale at gmail.com Mon Feb 2 07:41:05 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 2 Feb 2009 09:41:05 -0600 Subject: [Freeswitch-users] Call Variable not available when call hangup In-Reply-To: <21791503.post@talk.nabble.com> References: <21788550.post@talk.nabble.com> <21789152.post@talk.nabble.com> <191c3a030902020606r1a42ef44n7a73bd1e5157392e@mail.gmail.com> <21791503.post@talk.nabble.com> Message-ID: <191c3a030902020741k779e2488o38ca578a3b40e9ad@mail.gmail.com> you can't that's why i said it was a horrible approach. That's also why i posted you the instructions on the only elegant solution to your problem. On Mon, Feb 2, 2009 at 9:21 AM, shehzad p wrote: > > > one question is that when javascript is being called from dial plan, I get > the session object already available, It is for A leg of channel, > So when javascript is called after Bridge how can I get the session object > for B leg also? > > > Anthony Minessale-2 wrote: > > > > the leg you are running the script on is not hungup, the other leg of the > > call is. > > > > If it was hungup you would not be executing the script. > > > > Asterisk and the h ext and the whole dead-agi thing are all poor design > > showing it's teeth. > > We do not support anything like it. > > > > > > You can however try this: (see the link below) > > > > > http://www.nabble.com/Re:-Conference-javascript-and-hanuphooks-giving-me-headaches-p21614840.html > > > > > > > > On Mon, Feb 2, 2009 at 6:53 AM, shehzad p wrote: > > > >> > >> Is there any settings that when call hangup control can be transferred > to > >> another context and these CDR values can be accessible there? (just like > >> in > >> Asterisk, h extension) > >> > >> shehzad p wrote: > >> > > >> > Hi all, > >> > > >> > I need to process some CDR variables in Dialplan, like call duration, > >> > Answered time etc. > >> > but when I place info application after bridge, it is not listing them > >> > properly as below: > >> > =========================================== > >> > Caller-Channel-Created-Time: [1233573341672157] > >> > Caller-Channel-Answered-Time: [1233573342712939] > >> > Caller-Channel-Hangup-Time: [0] > >> > ========================================== > >> > Here Hangup time is 0, So how can I find actual values? > >> > > >> > --I know that we can use xml_cdr or cdr_csv, but my current need is to > >> get > >> > those values from dialplan itself so that can be passed to some > >> script... > >> > > >> > > >> > thanks, > >> > msp > >> > > >> > >> -- > >> View this message in context: > >> > http://www.nabble.com/Call-Variable-not-available-when-call-hangup-tp21788550p21789152.html > >> Sent from the Freeswitch-users mailing list archive at Nabble.com. > >> > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com < > MSN%3Aanthony_minessale at hotmail.com > > > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > > > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org < > sip%3A888 at conference.freeswitch.org > > > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > > > > pstn:213-799-1400 > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > View this message in context: > http://www.nabble.com/Call-Variable-not-available-when-call-hangup-tp21788550p21791503.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090202/2d430e44/attachment.html From brian at freeswitch.org Mon Feb 2 07:41:39 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 2 Feb 2009 09:41:39 -0600 Subject: [Freeswitch-users] How do I set my FS internal ip address to a "static" value. In-Reply-To: References: Message-ID: <05AC6B23-7F34-4C89-8BD0-1744BEF20B4C@freeswitch.org> you need to add this setting to sofia.conf.xml You'll also need to edit the sofia profiles and input the exact IP you wish it to bind to. The params are sip-ip and rtp-ip. /b On Feb 1, 2009, at 3:24 PM, clif at eugeneweb.com wrote: > Hi Gang, > > I've been struggleing with this also. Actually I can get it to bind > to my > address, the problem is it randomly drops my calls. :-( > > I have a FS running on a box with a static IP and I can start a call > between > two extensions and it will go for hours. Then I add anther interface > say eth0:0 > with a new static IP and reconfigure my phones and FS to use that, > and the > calls drop after about 15-20 mins. Though it's pretty random. > > Here is my setup. I have Debian Linux 2.6.23.1 kernel, and > freeswitch-1.0.1. > Here is my /etc/network/interfaces: > > # /etc/network/interfaces -- configuration file for ifup(8), ifdown(8) > > # The loopback interface > auto lo > iface lo inet loopback > > # The first network card - this entry was created during the Debian > installation > auto eth0 eth0:0 > iface eth0 inet dhcp > iface eth0:0 inet static > address 192.168.0.249 > netmask 255.255.255.0 > gateway 192.168.0.254 > > The only change I made to the FS config is in Vars.xml. I added this > line close > to the top: > > > > Here is the console log of the call being dropped: > > freeswitch at archive> sofia status > API CALL [sofia(status)] output: > Name Type Data > State > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > ====================================================================== > external profile sip:mod_sofia at 67.171.158.226:5080 > RUNNING (0) > internal profile sip:mod_sofia at 192.168.0.249:5060 > RUNNING (2) > nat profile sip:mod_sofia at 67.171.158.226:5070 > RUNNING (0) > default alias internal > ALIASED > outbound alias external > ALIASED > 192.168.0.249 alias internal > ALIASED > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > ====================================================================== > 3 profiles 3 aliases > > freeswitch at archive> 2009-02-01 13:23:19 [NOTICE] sofia_glue.c:2634 > sofia_glue_restart_all_profiles() Reload XML [Success] > 2009-02-01 13:23:19 [INFO] mod_enum.c:817 event_handler() ENUM > Reloaded > 2009-02-01 13:23:19 [NOTICE] mod_sofia.c:568 sofia_read_frame() Hangup > sofia/internal/ > 1003 at 192.168.0.53:5060;user=phone;transport=udp;fs_nat=yes > [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] > 2009-02-01 13:23:19 [NOTICE] switch_ivr_bridge.c:820 > switch_ivr_multi_threaded_bridge() Hangup sofia/internal/1001 at 192.168.0.249 > [CS_EXECUTE] [NORMAL_CLEARING] > 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:807 > switch_core_session_thread() Session 6 > (sofia/internal/ > 1003 at 192.168.0.53:5060;user=phone;transport=udp;fs_nat=yes) > Ended > 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:809 > switch_core_session_thread() Close Channel > sofia/internal/ > 1003 at 192.168.0.53:5060;user=phone;transport=udp;fs_nat=yes > [CS_HANGUP] > 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:807 > switch_core_session_thread() Session 5 (sofia/internal/1001 at 192.168.0.249 > ) > Ended > 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:809 > switch_core_session_thread() Close Channel sofia/internal/1001 at 192.168.0.249 > [CS_HANGUP] > 2009-02-01 13:23:19 [NOTICE] sofia.c:645 sofia_profile_thread_run() > waiting for > worker thread > 2009-02-01 13:23:19 [NOTICE] sofia.c:645 sofia_profile_thread_run() > waiting for > worker thread > 2009-02-01 13:23:19 [NOTICE] sofia.c:1865 config_sofia() Adding Alias > [192.168.0.249] for profile [internal] > 2009-02-01 13:23:19 [NOTICE] sofia.c:1865 config_sofia() Adding > Alias [default] > for profile [internal] > 2009-02-01 13:23:19 [NOTICE] sofia.c:1875 config_sofia() Started > Profile > internal [sofia_reg_internal] > 2009-02-01 13:23:20 [NOTICE] sofia.c:1865 config_sofia() Adding Alias > [outbound] for profile [external] > 2009-02-01 13:23:20 [NOTICE] sofia.c:1875 config_sofia() Started > Profile > external [sofia_reg_external] > 2009-02-01 13:23:20 [NOTICE] sofia.c:645 sofia_profile_thread_run() > waiting for > worker thread > 2009-02-01 13:23:20 [NOTICE] sofia.c:1875 config_sofia() Started > Profile nat > [sofia_reg_nat] > sofia status > API CALL [sofia(status)] output: > Name Type Data > State > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > ====================================================================== > external profile sip:mod_sofia at 67.171.158.226:5080 > RUNNING (0) > internal profile sip:mod_sofia at 192.168.0.249:5060 > RUNNING (0) > outbound alias external > ALIASED > 192.168.0.249 alias internal > ALIASED > nat profile sip:mod_sofia at 67.171.158.226:5070 > RUNNING (0) > default alias internal > ALIASED > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > ====================================================================== > 3 profiles 3 aliases > > There is an older thread that says one should set > > but in this (later) thread is says only Jingleling usese that > variable. > ie. see: > http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg00695.html > http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg07345.html > > So what do you think causes this? What is the correct way? ;-) > > > Thanks, > Clif > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From helmut.kuper at ewetel.de Mon Feb 2 10:00:48 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Mon, 02 Feb 2009 19:00:48 +0100 Subject: [Freeswitch-users] mod_voicemail: Limiting the number how often a menu is repeated Message-ID: <498734D0.5060004@ewetel.de> Hello, today I searched for a way to limit the number of menu repeatings in mod_voicemail to let's say 3 times and when it reached the limit voicemail should abort. But I couldn't find a hint. Any ideas? regards helmut From helmut.kuper at ewetel.de Mon Feb 2 10:04:11 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Mon, 02 Feb 2009 19:04:11 +0100 Subject: [Freeswitch-users] Q931 decoding Update In-Reply-To: <49830576.6080907@ewetel.de> References: <4972046E.8020102@ewetel.de> <49802B7F.70100@ewetel.de> <49818AE9.7030605@ewetel.de> <200901291202.41144.stkn@freeswitch.org> <191c3a030901290549x29c87976k68b3f9b455f00d0f@mail.gmail.com> <4981B8E6.9070708@ewetel.de> <5A2ED995-CF05-4AEF-9103-43BC84465514@freeswitch.org> <4981D6F8.4060409@ewetel.de> <191c3a030901290832k27a18b30u7bc531157fd89b19@mail.gmail.com> <49830576.6080907@ewetel.de> Message-ID: <4987359B.2020402@ewetel.de> Hello, today I uploaded a little patch for openzap concerning missed linking of the pcap library. So loading ozmod_isdn failed with some kind of "unknown symbol pcap_flush_dump" error message. This keeps mod_openzap from loading at FS startup. regards helmut From peder at networkoblivion.com Mon Feb 2 12:01:36 2009 From: peder at networkoblivion.com (peder at networkoblivion.com) Date: Mon, 02 Feb 2009 14:01:36 -0600 Subject: [Freeswitch-users] Strange Performance when using as SBC In-Reply-To: <191c3a030902020557v70d3777fqb83e591385a64a59@mail.gmail.com> References: <4986C42C.7030700@laposte.net> <1C31435E19AB4E57981CF2AAC2CA74AA@SaeedLaptop> <4986ECA9.3040707@laposte.net> <9232C06D5362494791AF713E1DF61343@SaeedLaptop> <191c3a030902020557v70d3777fqb83e591385a64a59@mail.gmail.com> Message-ID: <49875120.1040502@networkoblivion.com> What is wrong with Intrepid? Anthony Minessale wrote: > if you want to use ubuntu, be sure to use hardy and not intrepid. > > On Mon, Feb 2, 2009 at 7:03 AM, Saeed Ahmed > wrote: > > Thanks rod for a quick answer, > > FS is installed on Ubuntu Server. > > I am planning to replace Nextone SBC with FS, Later I'll also use > openZAP to > communicate with TDM but this all depends how much calls it can take, or > maybe we can also do something in clustering environment ( I am not sure > about it). But thanks again and any further help will be highly > appreciated. > > > Kind Regards > Saeed Ahmed Tariq > > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > > [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of rod > Sent: Monday, February 02, 2009 1:53 PM > To: freeswitch-users at lists.freeswitch.org > > Subject: Re: [Freeswitch-users] Strange Performance when using as SBC > > Hi Saeed, > > I just created an account to share my setup on the wiki. I will detail > all the steps for a clean install of a debian64 lenny with FS used as a > SBC (next step is to try the new LCR module :) )and what I'm doing do > stress the server. > > I wrote nothing at this time so please be patient, I'm waiting for my > new hardware so that I will detail as much as possible what I'll do. > > For beginning I suggest you reading the start page on the wiki, > especially these pages: > -http://wiki.freeswitch.org/wiki/Getting_Started_Guide > -http://wiki.freeswitch.org/wiki/Dialplan_XML > > maybe you could tell more about the linux distribution you're using so > that I can give you some pointers for sipp... > > regards. > rod. > > > Saeed Ahmed wrote: > > Hi Rod, > > > > Could you please share how you configured Sipp & FS to create a test > > environment? Especially the dial plan, sofia settings etc..., > actually I > am > > a newbie. I want to test it on a single FS machine. > > > > Kind Regards > > Saeed > > -----Original Message----- > > From: freeswitch-users-bounces at lists.freeswitch.org > > > [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of rod > > Sent: Monday, February 02, 2009 11:00 AM > > To: freeswitch-users at lists.freeswitch.org > > > Subject: Re: [Freeswitch-users] Strange Performance when using as SBC > > > > Hi Ken, Jay, > > > > thanks for pointing to proxy media, I will test. > > > > Ken, you are right, I was brain damaged (a stupid mistake) when > setting > > INFO cause this kind of level could be very verbose. I'm switching to > > CRIT or ERR. > > > > Thanks guys, > > rod. > > > > thanks for > > > > Ken Rice wrote: > > > >> If you don't have to transcode, using proxy media mode will > still save > you > >> some CPU time. This is 1/2 way between bypass media and the > default media > >> interactive mode. The other draw back to this mode is if you are > using FS > >> > > to > > > >> clean up RTP and DTMF you loose those functions but they are not > needed > in > >> most use cases. > >> > >> As far as the log level goes, I found that once I had things stable > >> > > setting > > > >> the loglevel to helped a good deal... Info is probably a bit too > high of > a > >> loglevel I would probably go for CRIT or ERR (2 or 1 > respectively) if you > >> insist on leaving logging turned on... On a busy system these > can and > will > >> generate a good deal of activity (and disk IO if using mod_logfile) > >> > >> Ken > >> > >> > >> > >> > >>> From: rod > > >>> Reply-To: > > >>> Date: Mon, 02 Feb 2009 11:36:35 +0400 > >>> To: > > >>> Subject: Re: [Freeswitch-users] Strange Performance when using > as SBC > >>> > >>> Hi Ken, > >>> > >>> 1) I'd like to use FS to hide topology, so bypass media is not > possible > >>> 2) done > >>> 3) done > >>> 4) not used > >>> 5) i'm using this ins switch.xml -> >>> value="info"/>, if you think an other log level is more suitable. > >>> > >>> Regarding logging, I can see in console and in the > freeswitch.log that > >>> there is still a lot of NOTICE logging, see below: > >>> 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:960 > >>> switch_core_session_thread() Session 8721 > >>> (sofia/internal/sipp at 10.10.10.1:5060 > ) Ended > >>> 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:962 > >>> switch_core_session_thread() Close Channel > >>> sofia/internal/sipp at 10.10.10.1:5060 > [CS_HANGUP] > >>> 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:960 > >>> switch_core_session_thread() Session 8722 > >>> (sofia/external/9998 at 10.10.20.100 ) Ended > >>> 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:962 > >>> switch_core_session_thread() Close Channel > >>> sofia/external/9998 at 10.10.20.100 > [CS_HANGUP] > >>> 2009-02-02 08:33:56 [NOTICE] sofia.c:3164 > sofia_handle_sip_i_state() > >>> Channel [sofia/external/9998 at 10.10.20.100 > ] has been answered > >>> 2009-02-02 08:33:56 [WARNING] mod_sofia.c:740 sofia_read_frame() > >>> Changing codec ptime to 30. I bet you have a linksys/sipura =D > >>> > >>> Do you have any idea where I can switch off this kind of logging. I > >>> thought it should be in /dialplan/internal.xml, but I see that in > >>> internal.xml -> > >>> > >>> thanks a lot for your suggestion. > >>> > >>> regards, > >>> rod > >>> > >>> Ken Rice wrote: > >>> > >>> > >>>> Dont forget there are several things you can do to increase > >>>> > > performance... > > > >>>> 1) where possible use bypass media or media proxy modes > >>>> 2) mount freeswitch/db as a ram drive (if you are using > voicemail with > >>>> the internal FS DBs you'll need a way to make this persistant > across > >>>> reboots) > >>>> 3) see the wiki for setting reasonable ulimits > >>>> 4) (this is my oppinion others may vary) dont use mod_cdr_csv > >>>> 5) turn off (or reduce logging) in switch.conf.xml > >>>> > >>>> all of these thing can greatly improve performance. > >>>> > >>>> On Mon, Feb 2, 2009 at 1:04 AM, rod > >>>> >> wrote: > >>>> > >>>> Thanks Anthony, > >>>> > >>>> the setup is like this: > >>>> > >>>> sipp server ---- FS 1 ---- FS2 > >>>> > >>>> FS1 is the AMD CPU that has only one extension in dialplan > that > >>>> bridges > >>>> 9999 to FS2. 9999 is the first extension in FS2 dialplan that > >>>> plays moh, > >>>> FS2 has no CPU pbm. > >>>> > >>>> FS1 is maxing out at 60 bridged calls without your option -hp. > >>>> > >>>> Using -hp, I'm now able to bridge 200 concurrent calls (a > great > >>>> improvement) and the system is still reactive. CPU load is > high > >>>> but not > >>>> 100% and as the system responds well, I think that doesn't > matter. > >>>> > > The > > > >>>> 2GB of memory are completely consumed (top command shows > 700MB for > >>>> > > FS > > > >>>> process). > >>>> > >>>> I understand that FS1 server is not the best hardware > platform, > >>>> and I'm > >>>> waiting for new 4 cores server for testing. > >>>> I will update those numbers when testing with the new > hardware. > >>>> > >>>> regards, > >>>> rod. > >>>> > >>>> Anthony Minessale wrote: > >>>> > >>>> > >>>>> Which of the 2 machines has the load issue? You said it was > one box > >>>>> calling the other. > >>>>> > >>>>> You have 2 major things against you, single CPU and AMD, but you > >>>>> should at least be able to get in the vicinity of 800-1000 calls > >>>>> > >>>>> > >>>> on a > >>>> > >>>> > >>>>> box like that. > >>>>> > >>>>> Are you calling the default 9999? It's not really an appropriate > >>>>> extension for load testing. > >>>>> On the terminating box you should set up a manual extension > that is > >>>>> the first one in the dial plan > >>>>> to play a wav file from preferably a ram disk or /tmp > >>>>> > >>>>> If you do plan on using this in production accept nothing less > >>>>> > >>>>> > >>>> than a > >>>> > >>>> > >>>>> multi-core intel machine with at least 4 cores, the more > cores the > >>>>> better because that parallel processing is where FS gets it's > >>>>> > >>>>> > >>>> atvantage. > >>>> > >>>> > >>>>> On Fri, Jan 30, 2009 at 5:56 AM, rod > >>>>> > >>>>> > >>>> > > >>>> > >>>> > >>>>> > >>> wrote: > >>>>> > >>>>> Dear list, > >>>>> > >>>>> I've been playing with freeswitch for some time (2 months) > >>>>> > >>>>> > >>>> and the > >>>> > >>>> > >>>>> fact > >>>>> is that I'm very pleased with the functionnalities of this > >>>>> > >>>>> > >>>> software. > >>>> > >>>> > >>>>> I'd like to use FS as a SBC handling media and I'm doing some > >>>>> tests with > >>>>> sipp to load the machine but I'm unable to bridge more than > >>>>> > >>>>> > >>>> 60 calls > >>>> > >>>> > >>>>> without seeing the CPU being loaded at 100%. I'm sure > >>>>> > >>>>> > >>>> something is > >>>> > >>>> > >>>>> going > >>>>> wrong with my setup but I'm unable to see what. > >>>>> > >>>>> The test machine has the following specs: > >>>>> Athlon XP 3500+ with 2GB of memory (I know this is not a > >>>>> > >>>>> > >>>> high end > >>>> > >>>> > >>>>> machine :p) > >>>>> > >>>>> Freeswitch:/opt/freeswitch/log# cat /proc/cpuinfo > >>>>> processor : 0 > >>>>> vendor_id : AuthenticAMD > >>>>> cpu family : 15 > >>>>> model : 95 > >>>>> model name : AMD Athlon(tm) 64 Processor 3500+ > >>>>> stepping : 2 > >>>>> cpu MHz : 2199.973 > >>>>> cache size : 512 KB > >>>>> fpu : yes > >>>>> fpu_exception : yes > >>>>> cpuid level : 1 > >>>>> wp : yes > >>>>> flags : fpu vme de pse tsc msr pae mce cx8 apic > >>>>> > >>>>> > >>>> sep mtrr pge > >>>> > >>>> > >>>>> mca cmov pat pse36 clflush mmx fxsr sse sse2 syscall nx > mmxext > >>>>> fxsr_opt > >>>>> rdtscp lm 3dnowext 3dnow up rep_good pni cx16 lahf_lm svm > >>>>> > >>>>> > >>>> extapic > >>>> > >>>> > >>>>> cr8_legacy > >>>>> bogomips : 4402.97 > >>>>> TLB size : 1024 4K pages > >>>>> clflush size : 64 > >>>>> cache_alignment : 64 > >>>>> address sizes : 40 bits physical, 48 bits virtual > >>>>> power management: ts fid vid ttp tm stc > >>>>> > >>>>> I installed FS on a fresh debian 64: > >>>>> Linux Freeswitch 2.6.26-1-amd64 #1 SMP Sat Jan 10 17:57:00 > >>>>> > >>>>> > >>>> UTC 2009 > >>>> > >>>> > >>>>> x86_64 GNU/Linux > >>>>> > >>>>> I set the ulimit parameters like those on the website: > >>>>> freeswitch at internal> ... > >>>>> Freeswitch:/opt/free-svn/bin# ulimit -a > >>>>> core file size (blocks, -c) unlimited > >>>>> data seg size (kbytes, -d) unlimited > >>>>> scheduling priority (-e) 0 > >>>>> file size (blocks, -f) unlimited > >>>>> pending signals (-i) unlimited > >>>>> max locked memory (kbytes, -l) unlimited > >>>>> max memory size (kbytes, -m) unlimited > >>>>> open files (-n) 999999 > >>>>> pipe size (512 bytes, -p) 8 > >>>>> POSIX message queues (bytes, -q) unlimited > >>>>> real-time priority (-r) 0 > >>>>> stack size (kbytes, -s) 244 > >>>>> cpu time (seconds, -t) unlimited > >>>>> max user processes (-u) unlimited > >>>>> virtual memory (kbytes, -v) unlimited > >>>>> file locks (-x) unlimited > >>>>> > >>>>> > >>>>> My network setup is the following: > >>>>> > >>>>> SIPP machine (10.10.10.1/24)----------------vlan > > >>>>> > >>>>> > >>>> > >>>> > >>>> > >>>>> 55 > >>>>> ----------(10.10.10.254/24 > > >>>>> > >>>>> > >>>> ) FS > >>>> > >>>> > >>>>> (10.10.20.254/24)-------------- > > >>>>> > >>>>> > >>>> > >>>> > >>>> > >>>>> vlan56 > >>>>> -------------------(10.10.20.100/24 > > >>>>> > >>>>> > >>>> ) > >>>> > >>>> > >>>>> OTHER STOCK FS > >>>>> > >>>>> > >>>>> I launched sipp with: > >>>>> sipp -sn uac_pcap -s 9999 -r 10 -l 80 -d 60000 -mi > >>>>> > >>>>> > >>>> 10.10.10.1 -i > >>>> > >>>> > >>>>> 10.10.10.1 -mp 25000 10.10.10.254:5060 > > >>>>> > >>>>> > >>>> > >>>> > >>>> > >>>>> The dialplan on FS is very simple: > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> expression="^9999$"> > >>>>> > >>>>> >>>>> data="sofia/external/9999 at 10.10.20.100 > > >>>>> > >>>>> > >>>> > > > >>>> >>"/> > >>>> > >>>> > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> FreeSWITCH Version 1.0.trunk (11560M) Started. > >>>>> Crash Protection [Disabled] > >>>>> Max Sessions[1000] > >>>>> Session Rate[100] > >>>>> SQL [Enabled] > >>>>> > >>>>> > >>>>> The test is very simple: sipp dial 9999 that matches in my > >>>>> > >>>>> > >>>> FS dialplan > >>>> > >>>> > >>>>> and this is bridged to an other FS machine playing music on > >>>>> > >>>>> > >>>> hold. > >>>> > >>>> > >>>>> When I launch "top" I see after 30 to 40 s that FS > consumes all > >>>>> the CPU > >>>>> ressources (with a mean of 50-60 % before), with 80 calls. > >>>>> When I set 70 calls, I have to wait 70-80 s before seeing > >>>>> > >>>>> > >>>> the same > >>>> > >>>> > >>>>> issue. > >>>>> > >>>>> Presence is set to false on the 2 profile. > >>>>> > >>>>> I have the same issue with FS 1.0.2 that' s why I tried FS > >>>>> > >>>>> > >>>> 11560. > >>>> > >>>> > >>>>> When I use the FS machine as a router to test the packet per > >>>>> > >>>>> > >>>> second > >>>> > >>>> > >>>>> performance, I'm reaching 100Mbps with 8000pps in each > >>>>> > >>>>> > >>>> direction (from > >>>> > >>>> > >>>>> vlan 55 to vlan56) with less than 12% CPU. So that I > don't think > >>>>> there's > >>>>> an issue with the network. > >>>>> > >>>>> Here is an "mpstat -P ALL 1" to show you what's happening > >>>>> > >>>>> > >>>> suddenly > >>>> > >>>> > >>>>> with > >>>>> 70 bridge calls: > >>>>> 12:31:26 CPU %user %nice %sys %iowait %irq > %soft > >>>>> %steal %idle intr/s > >>>>> 12:31:27 all 3,00 0,00 3,00 0,00 1,00 > 4,00 > >>>>> 0,00 89,00 6241,00 > >>>>> 12:31:27 0 3,00 0,00 3,00 0,00 1,00 > 4,00 > >>>>> 0,00 89,00 6241,00 > >>>>> > >>>>> 12:31:27 CPU %user %nice %sys %iowait %irq > %soft > >>>>> %steal %idle intr/s > >>>>> 12:31:28 all 14,14 0,00 56,57 0,00 2,02 > 5,05 > >>>>> 0,00 22,22 6035,35 > >>>>> 12:31:28 0 14,14 0,00 56,57 0,00 2,02 > 5,05 > >>>>> 0,00 22,22 6035,35 > >>>>> > >>>>> 12:31:28 CPU %user %nice %sys %iowait %irq > %soft > >>>>> %steal %idle intr/s > >>>>> 12:31:29 all 24,75 0,00 67,33 0,00 0,99 > 6,93 > >>>>> 0,00 0,00 5483,17 > >>>>> 12:31:29 0 24,75 0,00 67,33 0,00 0,99 > 6,93 > >>>>> 0,00 0,00 5483,17 > >>>>> > >>>>> > >>>>> The CPU is going from 89% idle to 0% in less than 2 seconds. > >>>>> > >>>>> I know that I don't have to expect too much from this kind of > >>>>> hardware, > >>>>> but it seems strange that the CPU power vanished so suddenly. > >>>>> > >>>>> Thanks a lot for the guys that have read this long mail :p > >>>>> > >>>>> kind regards, > >>>>> rod > >>>>> > >>>>> > >>>>> _______________________________________________ > >>>>> Freeswitch-users mailing list > >>>>> Freeswitch-users at lists.freeswitch.org > > >>>>> > >>>>> > >>>> > > >>>> > >>>> > >>>>> > >>>>> > >>>>> > >>>> >> > >>>> > >>>> > >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>>>> > >>>>> > >>>>> > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > >>>> > >>>> > >>>>> http://www.freeswitch.org > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> -- > >>>>> Anthony Minessale II > >>>>> > >>>>> FreeSWITCH http://www.freeswitch.org/ > >>>>> ClueCon http://www.cluecon.com/ > >>>>> > >>>>> AIM: anthm > >>>>> MSN:anthony_minessale at hotmail.com > > >>>>> > >>>>> > >>>> > > >>>> > >>>> > >>>>> > >>>>> > >>>>> > >>>> >> > >>>> > >>>> > >>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > >>>>> > >>>>> > >>>> > > >>>> > >>>> > >>>>> > >>>>> > >>>>> > >>>> >> > >>>> > >>>> > >>>>> IRC: irc.freenode.net > > >>>>> > >>>>> > >>>> #freeswitch > >>>> > >>>> > >>>>> FreeSWITCH Developer Conference > >>>>> sip:888 at conference.freeswitch.org > > >>>>> > >>>>> > >>>> > > >>>> > >>>> > >>>>> > >>>>> > >>>>> > >>>> >> > >>>> > >>>> > >>>>> iax:guest at conference.freeswitch.org/888 > > >>>>> > >>>>> > >>>> > >>>> > >>>> > >>>>> > >>>>> googletalk:conf+888 at conference.freeswitch.org > > >>>>> > >>>>> > >>>> > > >>>> > >>>> > >>>>> > >>>>> > >>>>> > >>>> > >> > >>>> > >>>> > >>>>> pstn:213-799-1400 > >>>>> > >>>>> > >>>>> > > > ------------------------------------------------------------------------ > > > >>>> > >>>> > >>>>> _______________________________________________ > >>>>> Freeswitch-users mailing list > >>>>> Freeswitch-users at lists.freeswitch.org > > >>>>> > >>>>> > >>>> > > >>>> > >>>> > >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>>>> > >>>>> > >>>>> > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > >>>> > >>>> > >>>>> http://www.freeswitch.org > >>>>> > >>>>> > >>>>> > >>>> _______________________________________________ > >>>> Freeswitch-users mailing list > >>>> Freeswitch-users at lists.freeswitch.org > > >>>> > > >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>>> > >>>> > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > >>>> http://www.freeswitch.org > >>>> > >>>> > >>>> > ------------------------------------------------------------------------ > >>>> > >>>> _______________________________________________ > >>>> Freeswitch-users mailing list > >>>> Freeswitch-users at lists.freeswitch.org > > >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >>>> http://www.freeswitch.org > >>>> > >>>> > >>>> > >>> _______________________________________________ > >>> Freeswitch-users mailing list > >>> Freeswitch-users at lists.freeswitch.org > > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> http://www.freeswitch.org > >>> > >>> > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > >> > >> > >> > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Mon Feb 2 12:10:22 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 2 Feb 2009 14:10:22 -0600 Subject: [Freeswitch-users] Strange Performance when using as SBC In-Reply-To: <49875120.1040502@networkoblivion.com> References: <4986C42C.7030700@laposte.net> <1C31435E19AB4E57981CF2AAC2CA74AA@SaeedLaptop> <4986ECA9.3040707@laposte.net> <9232C06D5362494791AF713E1DF61343@SaeedLaptop> <191c3a030902020557v70d3777fqb83e591385a64a59@mail.gmail.com> <49875120.1040502@networkoblivion.com> Message-ID: Its too bleeding edge and you had better know what you're doing if you use it. It comes with libtool 2.2 which you can't use to build FreeSWITCH. /b On Feb 2, 2009, at 2:01 PM, peder at networkoblivion.com wrote: > What is wrong with Intrepid? From raul at etellicom.com Mon Feb 2 12:34:45 2009 From: raul at etellicom.com (Raul Fragoso) Date: Mon, 02 Feb 2009 18:34:45 -0200 Subject: [Freeswitch-users] Strange Performance when using as SBC In-Reply-To: References: <4986C42C.7030700@laposte.net> <1C31435E19AB4E57981CF2AAC2CA74AA@SaeedLaptop> <4986ECA9.3040707@laposte.net> <9232C06D5362494791AF713E1DF61343@SaeedLaptop> <191c3a030902020557v70d3777fqb83e591385a64a59@mail.gmail.com> <49875120.1040502@networkoblivion.com> Message-ID: <1233606885.28870.5.camel@stargate> Yes, that's exactly the issue: libtool. Provided that you use libtool 1.5.22-4 or some other 1.5.x version, FS seems to work fine with Intrepid. -- Raul On Mon, 2009-02-02 at 14:10 -0600, Brian West wrote: > Its too bleeding edge and you had better know what you're doing if you > use it. It comes with libtool 2.2 which you can't use to build > FreeSWITCH. > > /b > > On Feb 2, 2009, at 2:01 PM, peder at networkoblivion.com wrote: > > > What is wrong with Intrepid? > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From raul at etellicom.com Mon Feb 2 12:48:14 2009 From: raul at etellicom.com (Raul Fragoso) Date: Mon, 02 Feb 2009 18:48:14 -0200 Subject: [Freeswitch-users] Strange Performance when using as SBC In-Reply-To: References: <4986C42C.7030700@laposte.net> <1C31435E19AB4E57981CF2AAC2CA74AA@SaeedLaptop> <4986ECA9.3040707@laposte.net> <9232C06D5362494791AF713E1DF61343@SaeedLaptop> <191c3a030902020557v70d3777fqb83e591385a64a59@mail.gmail.com> <49875120.1040502@networkoblivion.com> Message-ID: <1233607694.28870.9.camel@stargate> In addition to libtool, you may have issues with the latest packages of gcc and some other tools that FS will need. In any case, it's better to not use Intrepid at all ;-) Use Hardy as suggested and you will be happy. On Mon, 2009-02-02 at 14:10 -0600, Brian West wrote: > Its too bleeding edge and you had better know what you're doing if you > use it. It comes with libtool 2.2 which you can't use to build > FreeSWITCH. > > /b > > On Feb 2, 2009, at 2:01 PM, peder at networkoblivion.com wrote: > > > What is wrong with Intrepid? > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From hads at nice.net.nz Mon Feb 2 13:02:34 2009 From: hads at nice.net.nz (Hadley Rich) Date: Tue, 3 Feb 2009 10:02:34 +1300 Subject: [Freeswitch-users] Strange Performance when using as SBC In-Reply-To: <1233607694.28870.9.camel@stargate> References: <1233607694.28870.9.camel@stargate> Message-ID: <200902031002.34785.hads@nice.net.nz> On Tue, 03 Feb 2009 09:48:14 Raul Fragoso wrote: > In addition to libtool, you may have issues with the latest packages of > gcc and some other tools that FS will need. In any case, it's better to > not use Intrepid at all ;-) Use Hardy as suggested and you will be > happy. You shouldn't have any issues. I've used Intrepid on a VM to compile and test FreeSWITCH quite a bit and haven't run across any issues at all after downgrading libtool. That said I would also recommend Hardy LTS for production servers. hads -- http://nicegear.co.nz VoIP, DVB and other Linux compatible hardware. From brian at freeswitch.org Mon Feb 2 13:05:40 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 2 Feb 2009 15:05:40 -0600 Subject: [Freeswitch-users] Strange Performance when using as SBC In-Reply-To: <200902031002.34785.hads@nice.net.nz> References: <1233607694.28870.9.camel@stargate> <200902031002.34785.hads@nice.net.nz> Message-ID: <46AB18C7-C28B-4DA7-BBAE-1BBAF8ECB430@freeswitch.org> gcc 4.3.2 caused a segfault to appear in openzap due to over optimization... so yes it can bite you. :) /b On Feb 2, 2009, at 3:02 PM, Hadley Rich wrote: > You shouldn't have any issues. I've used Intrepid on a VM to compile > and test > FreeSWITCH quite a bit and haven't run across any issues at all after > downgrading libtool. > > That said I would also recommend Hardy LTS for production servers. > > hads From mike at jerris.com Mon Feb 2 13:48:55 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 2 Feb 2009 16:48:55 -0500 Subject: [Freeswitch-users] Q931 decoding Update In-Reply-To: <4987359B.2020402@ewetel.de> References: <4972046E.8020102@ewetel.de> <49802B7F.70100@ewetel.de> <49818AE9.7030605@ewetel.de> <200901291202.41144.stkn@freeswitch.org> <191c3a030901290549x29c87976k68b3f9b455f00d0f@mail.gmail.com> <4981B8E6.9070708@ewetel.de> <5A2ED995-CF05-4AEF-9103-43BC84465514@freeswitch.org> <4981D6F8.4060409@ewetel.de> <191c3a030901290832k27a18b30u7bc531157fd89b19@mail.gmail.com> <49830576.6080907@ewetel.de> <4987359B.2020402@ewetel.de> Message-ID: <3D009334-DCFD-4AE5-A446-6C8165F2D77D@jerris.com> We need to add more than this including detection in openzap configure.in if libpcap is available (headers and lib) and if not, disabling the functionality. MIke On Feb 2, 2009, at 1:04 PM, Helmut Kuper wrote: > Hello, > > today I uploaded a little patch for openzap concerning missed > linking of > the pcap library. So loading ozmod_isdn failed with some kind of > "unknown symbol pcap_flush_dump" error message. This keeps mod_openzap > from loading at FS startup. > > regards > helmut > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From jaybinks at gmail.com Mon Feb 2 14:20:20 2009 From: jaybinks at gmail.com (jay binks) Date: Tue, 3 Feb 2009 08:20:20 +1000 Subject: [Freeswitch-users] Strange Performance when using as SBC In-Reply-To: <49870432.5050301@laposte.net> References: <4986C42C.7030700@laposte.net> <1C31435E19AB4E57981CF2AAC2CA74AA@SaeedLaptop> <4986ECA9.3040707@laposte.net> <9232C06D5362494791AF713E1DF61343@SaeedLaptop> <49870432.5050301@laposte.net> Message-ID: Rod, that wiki article is Awesome ! real good to see guides with start to finish steps. cant wait to see the next installment of your guide :) Jay On Tue, Feb 3, 2009 at 12:33 AM, rod wrote: > Hi Saeed, > > Here is a first draft of what I did to install FS on my server. > Configuration are not present, they'll be in a next release :p > http://wiki.freeswitch.org/wiki/SBC_Setup > > My aim is to setup FS as a SBC, I hope this page could be a great > startup point for others. I will update regularly based on what I did. > > Saeed, why are you replacing your Nextone, it's said to be one of the > best commercial SBC on the market. > > regards. > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090203/b84c521a/attachment.html From nik.middleton at noblesolutions.co.uk Mon Feb 2 15:35:51 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Mon, 2 Feb 2009 23:35:51 -0000 Subject: [Freeswitch-users] Generating calls from external source Message-ID: Hi Guys, As a long time Asterisk user, I'm looking into freeswitch as an alternative mainly due to (list multiple reasons here) Can anyone give me a pointer as to how I would achieve the following? I need to replicate an emergency broadcast system currently running under Asterisk. At the moment, I run through a Mysql database and using the manager API, issues an Originate command to dial a number. When the call is answered, a message is played, and the recipient has the option of hitting a digit to confirm receipt. I then call an AGI script to update the database. Is this fairly easy to do in Freeswitch? Not looking for code, just some pointers as to what's available to do the above / Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090202/82eebf5b/attachment.html From msc at freeswitch.org Mon Feb 2 17:16:43 2009 From: msc at freeswitch.org (Michael S Collins) Date: Mon, 2 Feb 2009 17:16:43 -0800 Subject: [Freeswitch-users] Generating calls from external source In-Reply-To: References: Message-ID: Nik, Welcome to FreeSWITCH! The short answer is "yes, FS can do that." The first thing that you should do is unlearn "the Asterisk way" of thinking. Usually there is an elegant way of doing things in FS that wasn't possible in Ast. I would recommend that you start by looking at the event socket, which is somewhat analogous to the AMI only cooler. :) I have personally done something similar to this using the event socket and a Perl script. The key is to learn the syntax of the originate command. (definitely hit the wiki and IRC channel) Are you using TDM cards for this? Just curious. -MC (IRC nick: mercutioviz) Sent from my iPhone On Feb 2, 2009, at 3:35 PM, "Nik Middleton" wrote: > Hi Guys, > > > > As a long time Asterisk user, I?m looking into freeswitch as an alte > rnative mainly due to (list multiple reasons here) > > > > Can anyone give me a pointer as to how I would achieve the following? > > > > I need to replicate an emergency broadcast system currently running > under Asterisk. > > > > At the moment, I run through a Mysql database and using the manager > API, issues an Originate command to dial a number. > > > > When the call is answered, a message is played, and the recipient > has the option of hitting a digit to confirm receipt. I then call > an AGI script to update the database. > > > > Is this fairly easy to do in Freeswitch? > > > > Not looking for code, just some pointers as to what?s available to d > o the above / > > > > Regards, > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090202/14b335d2/attachment-0001.html From ajlong at worldlink.net Mon Feb 2 19:05:17 2009 From: ajlong at worldlink.net (Adam Long) Date: Mon, 2 Feb 2009 22:05:17 -0500 Subject: [Freeswitch-users] FreeSwitch setup as a "Dumb" SBC Message-ID: <019501c985ac$4f00ee60$ed02cb20$@net> Hi Guys, I've been working at setting up a couple of FreeSwitch nodes as a topology hiding SBCs that handles both ingress traffic from my providers/peers and pass traffic up to an openser router that then routes call across the cluster of SBCs through which they reach the destination. I have OpenSIPS/SER setup doing DB route lookups and ENUM with LCR/Serial forking etc. My question is what would be the best way to send a call out to a destination choosen by the OpenSER router? For example: SIP Provider -- > SBC --- > OpenSER ---- ( route lookup returns 123.123.123.4 as dest ) -- > SBC --- > 123.123.123.4 I was thinking something along the lines of adding a "X-Route-To: +1NXXNXXXXXX at 123.123.123.4" with openser and then something like this in the SBC. Is this a wise approach, is there anything I could do to do this better? I'd like to keep the logic in the SBCs as simple as possible. I am pretty familiar with SIP but my knowledge fades when it gets into the nitty gritty of routing. ie the Contact: and Via: headers and all that good stuff. I should also state I have two profiles defined one for the internal/private "core" network and one for the outside "external" network. Any thoughts on this at all would be greatly appreciated. Am I missing something in the SIP spec that would allow for this is a standardized way? Regards, -Adam -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090202/1c9fdb9e/attachment.html From jason at voicesession.com Mon Feb 2 18:21:25 2009 From: jason at voicesession.com (lee jason) Date: Tue, 3 Feb 2009 10:21:25 +0800 Subject: [Freeswitch-users] Application language to support C or C++? Message-ID: <2cbf225c0902021821u2bc8b622ka5d9f94f05a968ae@mail.gmail.com> Hi All, I saw the applications using FreeSwitch library can be written in JavaScript, Perl, Python and Lua but I need to use Linux C or C++ for applications, Is FreeSwitch can supported it? Where can I get the sample codes? My Linux platform is base on Fedora. Thanks a lot. Jason -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090203/cd4c9ffb/attachment.html From brian at freeswitch.org Mon Feb 2 19:16:47 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 2 Feb 2009 21:16:47 -0600 Subject: [Freeswitch-users] Application language to support C or C++? In-Reply-To: <2cbf225c0902021821u2bc8b622ka5d9f94f05a968ae@mail.gmail.com> References: <2cbf225c0902021821u2bc8b622ka5d9f94f05a968ae@mail.gmail.com> Message-ID: <87A290D3-137D-4E54-A308-4A3D568104E2@freeswitch.org> Do you want to write modules in c++ for FreeSWITCH? If so then yes you can write modules in c++... If thats not what you mean please clarify. /b On Feb 2, 2009, at 8:21 PM, lee jason wrote: > Hi All, > > I saw the applications using FreeSwitch library can be written > in JavaScript, Perl, Python and Lua but I need to use Linux C or C+ > + for applications, Is FreeSwitch can supported it? Where can I get > the sample codes? My Linux platform is base on Fedora. > > > Thanks a lot. > > Jason From krice at freeswitch.org Mon Feb 2 20:24:44 2009 From: krice at freeswitch.org (Ken Rice) Date: Mon, 02 Feb 2009 22:24:44 -0600 Subject: [Freeswitch-users] Application language to support C or C++? In-Reply-To: <2cbf225c0902021821u2bc8b622ka5d9f94f05a968ae@mail.gmail.com> Message-ID: FreeSwitch is written in C mainly and some things like mod_opal are written in C++, you can create your own modules in C... Grab the source and look around its pretty straight forward From: lee jason Reply-To: Date: Tue, 3 Feb 2009 10:21:25 +0800 To: Subject: [Freeswitch-users] Application language to support C or C++? Hi All, I saw the applications using FreeSwitch library can be written in JavaScript, Perl, Python and Lua but I need to use Linux C or C++ for applications, Is FreeSwitch can supported it? Where can I get the sample codes? My Linux platform is base on Fedora. Thanks a lot. Jason _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090202/af71a030/attachment.html From krice at freeswitch.org Mon Feb 2 20:28:17 2009 From: krice at freeswitch.org (Ken Rice) Date: Mon, 02 Feb 2009 22:28:17 -0600 Subject: [Freeswitch-users] FreeSwitch setup as a "Dumb" SBC In-Reply-To: <019501c985ac$4f00ee60$ed02cb20$@net> Message-ID: Yes you can do that, but there is nothing that says you cant have FreeSWITCH just do those lookups and ENUM (FS Supports ENUM out of the box) and do the exact same thing so it would work like Provider -> ingress SBC -> egress SBC/Registration Server -> customer Loosing a whole hop in the process From: Adam Long Reply-To: Date: Mon, 2 Feb 2009 22:05:17 -0500 To: Subject: [Freeswitch-users] FreeSwitch setup as a "Dumb" SBC Hi Guys, I?ve been working at setting up a couple of FreeSwitch nodes as a topology hiding SBCs that handles both ingress traffic from my providers/peers and pass traffic up to an openser router that then routes call across the cluster of SBCs through which they reach the destination. I have OpenSIPS/SER setup doing DB route lookups and ENUM with LCR/Serial forking etc. My question is what would be the best way to send a call out to a destination choosen by the OpenSER router? For example: SIP Provider -- > SBC --- > OpenSER ---- ( route lookup returns 123.123.123.4 as dest ) -- > SBC --- > 123.123.123.4 I was thinking something along the lines of adding a ?X-Route-To: +1NXXNXXXXXX at 123.123.123.4? with openser and then something like this in the SBC? Is this a wise approach, is there anything I could do to do this better? I?d like to keep the logic in the SBCs as simple as possible. I am pretty familiar with SIP but my knowledge fades when it gets into the nitty gritty of routing? ie the Contact: and Via: headers and all that good stuff. I should also state I have two profiles defined one for the internal/private ?core? network and one for the outside ?external? network. Any thoughts on this at all would be greatly appreciated. Am I missing something in the SIP spec that would allow for this is a standardized way? Regards, -Adam _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090202/be0fb682/attachment.html From kawarod at laposte.net Mon Feb 2 22:33:11 2009 From: kawarod at laposte.net (rod) Date: Tue, 03 Feb 2009 10:33:11 +0400 Subject: [Freeswitch-users] FreeSwitch setup as a "Dumb" SBC In-Reply-To: <019501c985ac$4f00ee60$ed02cb20$@net> References: <019501c985ac$4f00ee60$ed02cb20$@net> Message-ID: <4987E527.1040909@laposte.net> Hi Adam, I'm in the process of using FS as a SBC. For the route lookup, I do it using OpenSER carrierroute, without having to flow through SBC---Openser---SBC. I'm using carrierroute at this time cause I need more than 200 000 routing entries and carrierroute has been tested with twice this number. Here is the setup: - install openser and carrierroute and make openser listening on 127.0.0.1:5062 (for example) on your SBC - populate carrierroute table What I do to use carrierroute module from FS is to use a specific X-header (X-LOOKUP). In the dialplan, in the default context, I have something like this: The process is simple: the export "sip_h_X-ROUTE=LOOKUP" had a sip header X-ROUTE=LOOKUP then I bridge the call to 127.0.0.1:5062 (openser process) In openser I have a route block that checks the presence of header LOOKUP and openser sends a "604: unable to route call" if the prefix is not found, or a "302: with the IP of the gateway found" In FS, you can get the IP using the variable "${sip_redirect_contact_host_0}". Then I transfer this to the context ROUTING, where the check condition is based on the LOOKUP header that has been rewritten with this variable. I will document all this setup (installation of openser/carrierroute and config file of FS and openser) on a wiki page I start writing yesterday, so please be indulgent and patient. The next step is to test the scalability of this. I'm a very bad programmer, so that's the only way for me to contribute to FS, and as I see many people interested for an SBC setup, I think it could be great if we share our work/knowlegde. The wiki page is there: http://wiki.freeswitch.org/wiki/SBC_Setup regards, rod. Adam Long wrote: > > Hi Guys, > > I?ve been working at setting up a couple of FreeSwitch nodes as a > topology hiding SBCs that handles both ingress traffic from my > > providers/peers and pass traffic up to an openser router that then > routes call across the cluster of SBCs through which they reach the > destination. > > I have OpenSIPS/SER setup doing DB route lookups and ENUM with > LCR/Serial forking etc. > > My question is what would be the best way to send a call out to a > destination choosen by the OpenSER router? > > For example: > > SIP Provider -- > SBC --- > OpenSER ---- ( route lookup returns > 123.123.123.4 as dest ) -- > SBC --- > 123.123.123.4 > > I was thinking something along the lines of adding a ?X-Route-To: > +1NXXNXXXXXX@ 123.123.123.4? with openser > > and then something like this in the SBC? > > > > > > > > > > > > Is this a wise approach, is there anything I could do to do this better? > > I?d like to keep the logic in the SBCs as simple as possible. > > I am pretty familiar with SIP but my knowledge fades when it gets into > the nitty gritty of routing? ie the Contact: and Via: headers > > and all that good stuff. > > I should also state I have two profiles defined one for the > internal/private ?core? network and one for the outside ?external? > network. > > Any thoughts on this at all would be greatly appreciated. > > Am I missing something in the SIP spec that would allow for this is a > standardized way? > > Regards, > > -Adam > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From sias at cpdata.co.za Mon Feb 2 23:22:21 2009 From: sias at cpdata.co.za (Sias Mey) Date: Tue, 3 Feb 2009 09:22:21 +0200 Subject: [Freeswitch-users] Conference dialing and uuid In-Reply-To: <191c3a030902020601v1109b1e9he867e9f1e1d4a401@mail.gmail.com> References: <20090130103944.GA23536@cpdata.co.za> <20090130133315.GB23536@cpdata.co.za> <191c3a030901300605r78dd8943w7fb1075ef851e1d@mail.gmail.com> <20090202085157.GA3555@cpdata.co.za> <0F863750-5C77-4CD9-ADC6-CFDDF2B149B4@freeswitch.org> <20090202102459.GA4179@cpdata.co.za> <9FE1733C-A46A-4A17-8AEE-077DE8010235@freeswitch.org> <191c3a030902020601v1109b1e9he867e9f1e1d4a401@mail.gmail.com> Message-ID: <20090203072221.GD16105@cpdata.co.za> Actually loopback does work. however as I said it generates a pair of extra channels. Hmmm I was trying to generate and extra call to a JS script that generated a teletone ring in an on_ring_execute for the second call however it seems to stop execution of the call itself. Event though I use api commands to originate and then transfer it into the conference so that I have direct access to its uuid. I think changeing the moh might be a bit simpler however and elimite some CoreDB stuff I was doing to keep track of the calls ring generating call (what a sentance). On Mon, Feb 02, 2009 at 08:01:25AM -0600, Anthony Minessale wrote: > you could set the conference moh sound to be tone_stream::// with the > teletone spec for ring sound and it use ignore_early_media=true in your > originates so the first caller would hear ringback until the 2nd one > arrived. > > On Mon, Feb 2, 2009 at 4:29 AM, Brian West <[1]brian at freeswitch.org> > wrote: > > Loopback will not work in that case either. If the far end plays > ringback inband you should hear that if you use the conference dial > api call. > /b > > On Feb 2, 2009, at 4:24 AM, Sias Mey wrote: > > Aaah ok. > > > > Thanks for clearing that up. > > > > So using loopback is still the only real workable sollution for me, > > since that generates ringback from and alternative endpoint and > > plays it > > into the conference. > > > > I might play with some javascript that streams ring into the channel > > eventually but for now the string comparisons at least get me the > > right > > uuid. > > > > Thank you again, > > Sias > > _______________________________________________ > Freeswitch-users mailing list > [2]Freeswitch-users at lists.freeswitch.org > [3]http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:[4]http://lists.freeswitch.org/mailman/options/freeswitch-u > sers > [5]http://www.freeswitch.org > > -- > Anthony Minessale II > FreeSWITCH [6]http://www.freeswitch.org/ > ClueCon [7]http://www.cluecon.com/ > AIM: anthm > [8]MSN:anthony_minessale at hotmail.com > GTALK/JABBER/[9]PAYPAL:anthony.minessale at gmail.com > IRC: [10]irc.freenode.net #freeswitch > FreeSWITCH Developer Conference > [11]sip:888 at conference.freeswitch.org > [12]iax:guest at conference.freeswitch.org/888 > [13]googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > References > > 1. mailto:brian at freeswitch.org > 2. mailto:Freeswitch-users at lists.freeswitch.org > 3. http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > 4. http://lists.freeswitch.org/mailman/options/freeswitch-users > 5. http://www.freeswitch.org/ > 6. http://www.freeswitch.org/ > 7. http://www.cluecon.com/ > 8. mailto:MSN%3Aanthony_minessale at hotmail.com > 9. mailto:PAYPAL%3Aanthony.minessale at gmail.com > 10. http://irc.freenode.net/ > 11. mailto:sip%3A888 at conference.freeswitch.org > 12. http://iax:guest at conference.freeswitch.org/888 > 13. mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From sias at cpdata.co.za Tue Feb 3 00:25:30 2009 From: sias at cpdata.co.za (Sias Mey) Date: Tue, 3 Feb 2009 10:25:30 +0200 Subject: [Freeswitch-users] Conference dialing and uuid In-Reply-To: <20090203072221.GD16105@cpdata.co.za> References: <20090130103944.GA23536@cpdata.co.za> <20090130133315.GB23536@cpdata.co.za> <191c3a030901300605r78dd8943w7fb1075ef851e1d@mail.gmail.com> <20090202085157.GA3555@cpdata.co.za> <0F863750-5C77-4CD9-ADC6-CFDDF2B149B4@freeswitch.org> <20090202102459.GA4179@cpdata.co.za> <9FE1733C-A46A-4A17-8AEE-077DE8010235@freeswitch.org> <191c3a030902020601v1109b1e9he867e9f1e1d4a401@mail.gmail.com> <20090203072221.GD16105@cpdata.co.za> Message-ID: <20090203082530.GA17166@cpdata.co.za> Hmmm no MOH wont work... since I am planning on pulling more than just 2 members into the conference and I still need ringback for the later members as well. Is there a direct way for me to use conference play to play teletone directly? or should I just records some ringing if I want to use that? And lastly for my own sanity ;-) why would the following in a on_ring_execute stop execution of the call at that point? call = argv[1]; conf = argv[2]; consoleLog("info","Making ringback channel for uuid : "+ session.uuid +"\n"); var ringuuid = apiExecute("originate","loopback/ringback-conf="+ conf +"-conf &park()") //I tried with and without a exit() at the end It seems to stop media detection??(not really sure about the term) for the call that executes this script. Freeswitch doesent recognize the pickup of that call and thus it doesent get bridged into the conference. when I uuid_kill the call that gets originated everything else starts happening again. Oh Im running this in FS ver. 1.0.trunk (11226:11561M) and that loopback points to and ringback.js is use("TeleTone"); session.answer(); var tts = new TeleTone(session); tts.addTone("u", 400.0, 450.0, 0.0); tts.addTone("r", 440.0, 480.0, 0.0); var RESET = "v=2000;>=0;+=0;"; var UK_RING = RESET + "L=2;u(400,200);u(400,2200)"; var US_RING = RESET + "r(2000,4000)"; while(session.ready()) { console_log("making UK ring\n"); for (x = 0 ; x < 2 ; x++) { tts.generate(UK_RING); } } A slight bastardisation of the teletone JS example. I would expected the new channel that is created via a api originate to be completely seperate from the JS I create it in. (thats why I use api instead of creating a new session, although I should probably try that as well). I use some CoreDB stuff to keep tabs on the uuid for the originated call so that I can uuid_kill it in the on_answer_script but as mentioned... the on_answer only executes after I uuid_kill the originated channel in the cli... Thanks again guys, Specially since it seems you two are always the ones that get back to me. On Tue, Feb 03, 2009 at 09:22:21AM +0200, Sias Mey wrote: > Actually loopback does work. > however as I said it generates a pair of extra channels. > > Hmmm I was trying to generate and extra call to a JS script that > generated a teletone ring in an on_ring_execute for the second call > however it seems to stop execution of the call itself. Event though I > use api commands to originate and then transfer it into the conference > so that I have direct access to its uuid. > > I think changeing the moh might be a bit simpler however and elimite > some CoreDB stuff I was doing to keep track of the calls ring generating > call (what a sentance). > > On Mon, Feb 02, 2009 at 08:01:25AM -0600, Anthony Minessale wrote: > > you could set the conference moh sound to be tone_stream::// with the > > teletone spec for ring sound and it use ignore_early_media=true in your > > originates so the first caller would hear ringback until the 2nd one > > arrived. > > > > On Mon, Feb 2, 2009 at 4:29 AM, Brian West <[1]brian at freeswitch.org> > > wrote: > > > > Loopback will not work in that case either. If the far end plays > > ringback inband you should hear that if you use the conference dial > > api call. > > /b > > > > On Feb 2, 2009, at 4:24 AM, Sias Mey wrote: > > > Aaah ok. > > > > > > Thanks for clearing that up. > > > > > > So using loopback is still the only real workable sollution for me, > > > since that generates ringback from and alternative endpoint and > > > plays it > > > into the conference. > > > > > > I might play with some javascript that streams ring into the channel > > > eventually but for now the string comparisons at least get me the > > > right > > > uuid. > > > > > > Thank you again, > > > Sias > > > > _______________________________________________ > > Freeswitch-users mailing list > > [2]Freeswitch-users at lists.freeswitch.org > > [3]http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:[4]http://lists.freeswitch.org/mailman/options/freeswitch-u > > sers > > [5]http://www.freeswitch.org > > > > -- > > Anthony Minessale II > > FreeSWITCH [6]http://www.freeswitch.org/ > > ClueCon [7]http://www.cluecon.com/ > > AIM: anthm > > [8]MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/[9]PAYPAL:anthony.minessale at gmail.com > > IRC: [10]irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > > [11]sip:888 at conference.freeswitch.org > > [12]iax:guest at conference.freeswitch.org/888 > > [13]googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > > > > References > > > > 1. mailto:brian at freeswitch.org > > 2. mailto:Freeswitch-users at lists.freeswitch.org > > 3. http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > 4. http://lists.freeswitch.org/mailman/options/freeswitch-users > > 5. http://www.freeswitch.org/ > > 6. http://www.freeswitch.org/ > > 7. http://www.cluecon.com/ > > 8. mailto:MSN%3Aanthony_minessale at hotmail.com > > 9. mailto:PAYPAL%3Aanthony.minessale at gmail.com > > 10. http://irc.freenode.net/ > > 11. mailto:sip%3A888 at conference.freeswitch.org > > 12. http://iax:guest at conference.freeswitch.org/888 > > 13. mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From kawarod at laposte.net Tue Feb 3 01:11:24 2009 From: kawarod at laposte.net (rod) Date: Tue, 03 Feb 2009 13:11:24 +0400 Subject: [Freeswitch-users] Strange Performance when using as SBC In-Reply-To: References: <4986C42C.7030700@laposte.net> <1C31435E19AB4E57981CF2AAC2CA74AA@SaeedLaptop> <4986ECA9.3040707@laposte.net> <9232C06D5362494791AF713E1DF61343@SaeedLaptop> <49870432.5050301@laposte.net> Message-ID: <49880A3C.3010506@laposte.net> Hi all, I completed the wiki page with the comments I made in the posts: Re: [Freeswitch-users] FreeSwitch setup as a "Dumb" SBC I detailed how to setup mysql/kamailio/carrierroute to use the carrierroute module of kamailio for LCR. I wrote this page using my memory and history of the linux commands. Maybe some things are missing and I will update as soon as I get my new servers for reinstallation. I have to cleanup the way it is displayed, cause it lacks some wiki rules. If some would like to contribute, they are welcome. http://wiki.freeswitch.org/wiki/SBC_Setup regards, rod jay binks wrote: > Rod, > that wiki article is Awesome ! > > real good to see guides with start to finish steps. > cant wait to see the next installment of your guide :) > > Jay > > On Tue, Feb 3, 2009 at 12:33 AM, rod > wrote: > > Hi Saeed, > > Here is a first draft of what I did to install FS on my server. > Configuration are not present, they'll be in a next release :p > http://wiki.freeswitch.org/wiki/SBC_Setup > > My aim is to setup FS as a SBC, I hope this page could be a great > startup point for others. I will update regularly based on what I did. > > Saeed, why are you replacing your Nextone, it's said to be one of the > best commercial SBC on the market. > > regards. > > From nik.middleton at noblesolutions.co.uk Tue Feb 3 01:28:34 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Tue, 3 Feb 2009 09:28:34 -0000 Subject: [Freeswitch-users] Generating calls from external source In-Reply-To: References: Message-ID: Thanks for that, coming from a C++ background it's a refreshing change to be looking at something that seems logical and efficient. I'd briefly looked at the event socket and wondered if that was the way to go. I presume that there's some sort of event generation that can trigger and external process as well somewhere, though all I need to do is update mysql (hopefully using some sort of pooled connection) I'm not using a TDM card, I have a direct interconnect with the PSTN breakout provider with 1,500 channels available to me. I'm finding Asterisk proving to be less than stable at high call volumes and load values spike at more than 100 calls with billing/accounting in place, hence my interest in FS. The only thing that's concerning me is XML at the moment. Lots of code and very wordy. I'm sure I'll appreciate why XML given time Regards, ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael S Collins Sent: 03 February 2009 01:17 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Generating calls from external source Nik, Welcome to FreeSWITCH! The short answer is "yes, FS can do that." The first thing that you should do is unlearn "the Asterisk way" of thinking. Usually there is an elegant way of doing things in FS that wasn't possible in Ast. I would recommend that you start by looking at the event socket, which is somewhat analogous to the AMI only cooler. :) I have personally done something similar to this using the event socket and a Perl script. The key is to learn the syntax of the originate command. (definitely hit the wiki and IRC channel) Are you using TDM cards for this? Just curious. -MC (IRC nick: mercutioviz) Sent from my iPhone On Feb 2, 2009, at 3:35 PM, "Nik Middleton" wrote: Hi Guys, As a long time Asterisk user, I'm looking into freeswitch as an alternative mainly due to (list multiple reasons here) Can anyone give me a pointer as to how I would achieve the following? I need to replicate an emergency broadcast system currently running under Asterisk. At the moment, I run through a Mysql database and using the manager API, issues an Originate command to dial a number. When the call is answered, a message is played, and the recipient has the option of hitting a digit to confirm receipt. I then call an AGI script to update the database. Is this fairly easy to do in Freeswitch? Not looking for code, just some pointers as to what's available to do the above / Regards, _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090203/fe01adb3/attachment-0001.html From dave at 3c.co.uk Tue Feb 3 01:46:00 2009 From: dave at 3c.co.uk (David Knell) Date: Tue, 3 Feb 2009 09:46:00 +0000 Subject: [Freeswitch-users] Generating calls from external source In-Reply-To: References: Message-ID: <40B58CBF-9D0F-43A3-B36F-BD05916756E4@3c.co.uk> Hi Nik, Here's a snipped in Perl that launches an outbound call: if (my $sock = IO::Socket::INET->new(Proto =>'tcp', PeerAddr => '127.0.0.1', PeerPort => 8021)) { print $sock "auth XXX\n\n"; print $sock "api originate {softivr_id=$siid,src_softivr_id=$siid,softivr_outdial=true}sofia/frombt/$ntd at 1.2.3.4 $service\n\n"; $sock->close(); } - it does no error checking or anything, but (line by line) it: - opens a socket to the event socket interface - authenticates - issues an originate which dials out to the number in $ntd. The bits in {} set a bunch of variables on the channel, which are used by the software which processes the call later on. The call is linked to the extension in $service - FS looks this up in the dialplan - which handles our end. - closes the socket Cheers -- Dave > Thanks for that, coming from a C++ background it?s a refreshing > change to be looking at something that seems logical and efficient. > > I?d briefly looked at the event socket and wondered if that was the > way to go. I presume that there?s some sort of event generation > that can trigger and external process as well somewhere, though all > I need to do is update mysql (hopefully using some sort of pooled > connection) > > I?m not using a TDM card, I have a direct interconnect with the PSTN > breakout provider with 1,500 channels available to me. I?m finding > Asterisk proving to be less than stable at high call volumes and > load values spike at more than 100 calls with billing/accounting in > place, hence my interest in FS. The only thing that?s concerning me > is XML at the moment. Lots of code and very wordy. I?m sure I?ll > appreciate why XML given time > > Regards, > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Michael S Collins > Sent: 03 February 2009 01:17 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Generating calls from external source > > Nik, > > Welcome to FreeSWITCH! The short answer is "yes, FS can do that." > The first thing that you should do is unlearn "the Asterisk way" of > thinking. Usually there is an elegant way of doing things in FS that > wasn't possible in Ast. > > I would recommend that you start by looking at the event socket, > which is somewhat analogous to the AMI only cooler. :) I have > personally done something similar to this using the event socket and > a Perl script. The key is to learn the syntax of the originate > command. (definitely hit the wiki and IRC channel) > Are you using TDM cards for this? Just curious. > > -MC (IRC nick: mercutioviz) > > Sent from my iPhone > > On Feb 2, 2009, at 3:35 PM, "Nik Middleton" > wrote: >> Hi Guys, >> >> As a long time Asterisk user, I?m looking into freeswitch as an >> alternative mainly due to (list multiple reasons here) >> >> Can anyone give me a pointer as to how I would achieve the following? >> >> I need to replicate an emergency broadcast system currently running >> under Asterisk. >> >> At the moment, I run through a Mysql database and using the manager >> API, issues an Originate command to dial a number. >> >> When the call is answered, a message is played, and the recipient >> has the option of hitting a digit to confirm receipt. I then call >> an AGI script to update the database. >> >> Is this fairly easy to do in Freeswitch? >> >> Not looking for code, just some pointers as to what?s available to >> do the above / >> >> Regards, >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090203/b539deb4/attachment.html From leon at scarlet-internet.nl Tue Feb 3 03:55:41 2009 From: leon at scarlet-internet.nl (Leon de Rooij) Date: Tue, 3 Feb 2009 12:55:41 +0100 Subject: [Freeswitch-users] debuild breaks since the last few days Message-ID: <0B3BFADA-4589-48AD-9F18-EEBDC6BAB740@scarlet-internet.nl> Hi all, I've been trying to build new debs, but debuild seems to break.. I tried trunk rev 11608 and 1.0.3RC-1 and tried building the packages with: debuild -i -us -uc -b (which worked before) And now it breaks at openzap with: cc1: warnings being treated as errors src/ozmod/ozmod_isdn/ozmod_isdn.c: In function 'writeQ931PacketToPcap': src/ozmod/ozmod_isdn/ozmod_isdn.c:220: warning: implicit declaration of function 'pcap_dump_flush' make[7]: *** [src/ozmod/ozmod_isdn/ozmod_isdn.o] Error 1 make[7]: Leaving directory `/home/fsbuilder/freeswitch-1.0.3RC1/libs/ openzap' make[6]: *** [../libopenzap.so] Error 2 make[6]: Leaving directory `/home/fsbuilder/freeswitch-1.0.3RC1/libs/ openzap/mod_openzap' make[5]: *** [all] Error 1 make[5]: Leaving directory `/home/fsbuilder/freeswitch-1.0.3RC1/libs/ openzap/mod_openzap' make[4]: *** [../../libs/openzap/mod_openzap-all] Error 1 make[4]: Leaving directory `/home/fsbuilder/freeswitch-1.0.3RC1/src/mod' make[3]: *** [all-recursive] Error 1 make[3]: Leaving directory `/home/fsbuilder/freeswitch-1.0.3RC1/src' Making all in build make[3]: Entering directory `/home/fsbuilder/freeswitch-1.0.3RC1/build' +-------- FreeSWITCH Build Complete -----------+ + FreeSWITCH has been successfully built. + + Install by running: + + + + /usr/bin/make install + +----------------------------------------------+ make[3]: Leaving directory `/home/fsbuilder/freeswitch-1.0.3RC1/build' make[2]: *** [all-recursive] Error 1 make[2]: Leaving directory `/home/fsbuilder/freeswitch-1.0.3RC1' make[1]: *** [all] Error 2 make[1]: Leaving directory `/home/fsbuilder/freeswitch-1.0.3RC1' make: *** [build-stamp] Error 2 debuild: fatal error at line 1247: debian/rules build failed Does anyone know how to fix this ? thanks, Leon From saeedahmad1981 at gmail.com Tue Feb 3 04:41:03 2009 From: saeedahmad1981 at gmail.com (Saeed Ahmed) Date: Tue, 3 Feb 2009 13:41:03 +0100 Subject: [Freeswitch-users] Strange Performance when using as SBC In-Reply-To: <49870432.5050301@laposte.net> References: <4986C42C.7030700@laposte.net><1C31435E19AB4E57981CF2AAC2CA74AA@SaeedLaptop> <4986ECA9.3040707@laposte.net><9232C06D5362494791AF713E1DF61343@SaeedLaptop> <49870432.5050301@laposte.net> Message-ID: Hi rod, It's really amazing! Well described! Could you please explain a bit why we used Kamailio? Kind Regards Saeed Ahmed Tariq -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of rod Sent: Monday, February 02, 2009 3:33 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Strange Performance when using as SBC Hi Saeed, Here is a first draft of what I did to install FS on my server. Configuration are not present, they'll be in a next release :p http://wiki.freeswitch.org/wiki/SBC_Setup My aim is to setup FS as a SBC, I hope this page could be a great startup point for others. I will update regularly based on what I did. Saeed, why are you replacing your Nextone, it's said to be one of the best commercial SBC on the market. regards. Saeed Ahmed wrote: > Thanks rod for a quick answer, > > FS is installed on Ubuntu Server. > > I am planning to replace Nextone SBC with FS, Later I'll also use openZAP to > communicate with TDM but this all depends how much calls it can take, or > maybe we can also do something in clustering environment ( I am not sure > about it). But thanks again and any further help will be highly appreciated. > > > Kind Regards > Saeed Ahmed Tariq > > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of rod > Sent: Monday, February 02, 2009 1:53 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Strange Performance when using as SBC > > Hi Saeed, > > I just created an account to share my setup on the wiki. I will detail > all the steps for a clean install of a debian64 lenny with FS used as a > SBC (next step is to try the new LCR module :) )and what I'm doing do > stress the server. > > I wrote nothing at this time so please be patient, I'm waiting for my > new hardware so that I will detail as much as possible what I'll do. > > For beginning I suggest you reading the start page on the wiki, > especially these pages: > -http://wiki.freeswitch.org/wiki/Getting_Started_Guide > -http://wiki.freeswitch.org/wiki/Dialplan_XML > > maybe you could tell more about the linux distribution you're using so > that I can give you some pointers for sipp... > > regards. > rod. > > > Saeed Ahmed wrote: > >> Hi Rod, >> >> Could you please share how you configured Sipp & FS to create a test >> environment? Especially the dial plan, sofia settings etc..., actually I >> > am > >> a newbie. I want to test it on a single FS machine. >> >> Kind Regards >> Saeed >> -----Original Message----- >> From: freeswitch-users-bounces at lists.freeswitch.org >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of rod >> Sent: Monday, February 02, 2009 11:00 AM >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] Strange Performance when using as SBC >> >> Hi Ken, Jay, >> >> thanks for pointing to proxy media, I will test. >> >> Ken, you are right, I was brain damaged (a stupid mistake) when setting >> INFO cause this kind of level could be very verbose. I'm switching to >> CRIT or ERR. >> >> Thanks guys, >> rod. >> >> thanks for >> >> Ken Rice wrote: >> >> >>> If you don't have to transcode, using proxy media mode will still save >>> > you > >>> some CPU time. This is 1/2 way between bypass media and the default media >>> interactive mode. The other draw back to this mode is if you are using FS >>> >>> >> to >> >> >>> clean up RTP and DTMF you loose those functions but they are not needed >>> > in > >>> most use cases. >>> >>> As far as the log level goes, I found that once I had things stable >>> >>> >> setting >> >> >>> the loglevel to helped a good deal... Info is probably a bit too high of >>> > a > >>> loglevel I would probably go for CRIT or ERR (2 or 1 respectively) if you >>> insist on leaving logging turned on... On a busy system these can and >>> > will > >>> generate a good deal of activity (and disk IO if using mod_logfile) >>> >>> Ken >>> >>> >>> >>> >>> >>>> From: rod >>>> Reply-To: >>>> Date: Mon, 02 Feb 2009 11:36:35 +0400 >>>> To: >>>> Subject: Re: [Freeswitch-users] Strange Performance when using as SBC >>>> >>>> Hi Ken, >>>> >>>> 1) I'd like to use FS to hide topology, so bypass media is not possible >>>> 2) done >>>> 3) done >>>> 4) not used >>>> 5) i'm using this ins switch.xml -> >>> value="info"/>, if you think an other log level is more suitable. >>>> >>>> Regarding logging, I can see in console and in the freeswitch.log that >>>> there is still a lot of NOTICE logging, see below: >>>> 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:960 >>>> switch_core_session_thread() Session 8721 >>>> (sofia/internal/sipp at 10.10.10.1:5060) Ended >>>> 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:962 >>>> switch_core_session_thread() Close Channel >>>> sofia/internal/sipp at 10.10.10.1:5060 [CS_HANGUP] >>>> 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:960 >>>> switch_core_session_thread() Session 8722 >>>> (sofia/external/9998 at 10.10.20.100) Ended >>>> 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:962 >>>> switch_core_session_thread() Close Channel >>>> sofia/external/9998 at 10.10.20.100 [CS_HANGUP] >>>> 2009-02-02 08:33:56 [NOTICE] sofia.c:3164 sofia_handle_sip_i_state() >>>> Channel [sofia/external/9998 at 10.10.20.100] has been answered >>>> 2009-02-02 08:33:56 [WARNING] mod_sofia.c:740 sofia_read_frame() >>>> Changing codec ptime to 30. I bet you have a linksys/sipura =D >>>> >>>> Do you have any idea where I can switch off this kind of logging. I >>>> thought it should be in /dialplan/internal.xml, but I see that in >>>> internal.xml -> >>>> >>>> thanks a lot for your suggestion. >>>> >>>> regards, >>>> rod >>>> >>>> Ken Rice wrote: >>>> >>>> >>>> >>>>> Dont forget there are several things you can do to increase >>>>> >>>>> >> performance... >> >> >>>>> 1) where possible use bypass media or media proxy modes >>>>> 2) mount freeswitch/db as a ram drive (if you are using voicemail with >>>>> the internal FS DBs you'll need a way to make this persistant across >>>>> reboots) >>>>> 3) see the wiki for setting reasonable ulimits >>>>> 4) (this is my oppinion others may vary) dont use mod_cdr_csv >>>>> 5) turn off (or reduce logging) in switch.conf.xml >>>>> >>>>> all of these thing can greatly improve performance. >>>>> >>>>> On Mon, Feb 2, 2009 at 1:04 AM, rod >>>> > wrote: >>>>> >>>>> Thanks Anthony, >>>>> >>>>> the setup is like this: >>>>> >>>>> sipp server ---- FS 1 ---- FS2 >>>>> >>>>> FS1 is the AMD CPU that has only one extension in dialplan that >>>>> bridges >>>>> 9999 to FS2. 9999 is the first extension in FS2 dialplan that >>>>> plays moh, >>>>> FS2 has no CPU pbm. >>>>> >>>>> FS1 is maxing out at 60 bridged calls without your option -hp. >>>>> >>>>> Using -hp, I'm now able to bridge 200 concurrent calls (a great >>>>> improvement) and the system is still reactive. CPU load is high >>>>> but not >>>>> 100% and as the system responds well, I think that doesn't matter. >>>>> >>>>> >> The >> >> >>>>> 2GB of memory are completely consumed (top command shows 700MB for >>>>> >>>>> >> FS >> >> >>>>> process). >>>>> >>>>> I understand that FS1 server is not the best hardware platform, >>>>> and I'm >>>>> waiting for new 4 cores server for testing. >>>>> I will update those numbers when testing with the new hardware. >>>>> >>>>> regards, >>>>> rod. >>>>> >>>>> Anthony Minessale wrote: >>>>> >>>>> >>>>> >>>>>> Which of the 2 machines has the load issue? You said it was one box >>>>>> calling the other. >>>>>> >>>>>> You have 2 major things against you, single CPU and AMD, but you >>>>>> should at least be able to get in the vicinity of 800-1000 calls >>>>>> >>>>>> >>>>>> >>>>> on a >>>>> >>>>> >>>>> >>>>>> box like that. >>>>>> >>>>>> Are you calling the default 9999? It's not really an appropriate >>>>>> extension for load testing. >>>>>> On the terminating box you should set up a manual extension that is >>>>>> the first one in the dial plan >>>>>> to play a wav file from preferably a ram disk or /tmp >>>>>> >>>>>> If you do plan on using this in production accept nothing less >>>>>> >>>>>> >>>>>> >>>>> than a >>>>> >>>>> >>>>> >>>>>> multi-core intel machine with at least 4 cores, the more cores the >>>>>> better because that parallel processing is where FS gets it's >>>>>> >>>>>> >>>>>> >>>>> atvantage. >>>>> >>>>> >>>>> >>>>>> On Fri, Jan 30, 2009 at 5:56 AM, rod >>>>> >>>>>> >>>>>> >>>>> >>>>> >>>>> >>>>> >>>>>> >> wrote: >>>>>> >>>>>> Dear list, >>>>>> >>>>>> I've been playing with freeswitch for some time (2 months) >>>>>> >>>>>> >>>>>> >>>>> and the >>>>> >>>>> >>>>> >>>>>> fact >>>>>> is that I'm very pleased with the functionnalities of this >>>>>> >>>>>> >>>>>> >>>>> software. >>>>> >>>>> >>>>> >>>>>> I'd like to use FS as a SBC handling media and I'm doing some >>>>>> tests with >>>>>> sipp to load the machine but I'm unable to bridge more than >>>>>> >>>>>> >>>>>> >>>>> 60 calls >>>>> >>>>> >>>>> >>>>>> without seeing the CPU being loaded at 100%. I'm sure >>>>>> >>>>>> >>>>>> >>>>> something is >>>>> >>>>> >>>>> >>>>>> going >>>>>> wrong with my setup but I'm unable to see what. >>>>>> >>>>>> The test machine has the following specs: >>>>>> Athlon XP 3500+ with 2GB of memory (I know this is not a >>>>>> >>>>>> >>>>>> >>>>> high end >>>>> >>>>> >>>>> >>>>>> machine :p) >>>>>> >>>>>> Freeswitch:/opt/freeswitch/log# cat /proc/cpuinfo >>>>>> processor : 0 >>>>>> vendor_id : AuthenticAMD >>>>>> cpu family : 15 >>>>>> model : 95 >>>>>> model name : AMD Athlon(tm) 64 Processor 3500+ >>>>>> stepping : 2 >>>>>> cpu MHz : 2199.973 >>>>>> cache size : 512 KB >>>>>> fpu : yes >>>>>> fpu_exception : yes >>>>>> cpuid level : 1 >>>>>> wp : yes >>>>>> flags : fpu vme de pse tsc msr pae mce cx8 apic >>>>>> >>>>>> >>>>>> >>>>> sep mtrr pge >>>>> >>>>> >>>>> >>>>>> mca cmov pat pse36 clflush mmx fxsr sse sse2 syscall nx mmxext >>>>>> fxsr_opt >>>>>> rdtscp lm 3dnowext 3dnow up rep_good pni cx16 lahf_lm svm >>>>>> >>>>>> >>>>>> >>>>> extapic >>>>> >>>>> >>>>> >>>>>> cr8_legacy >>>>>> bogomips : 4402.97 >>>>>> TLB size : 1024 4K pages >>>>>> clflush size : 64 >>>>>> cache_alignment : 64 >>>>>> address sizes : 40 bits physical, 48 bits virtual >>>>>> power management: ts fid vid ttp tm stc >>>>>> >>>>>> I installed FS on a fresh debian 64: >>>>>> Linux Freeswitch 2.6.26-1-amd64 #1 SMP Sat Jan 10 17:57:00 >>>>>> >>>>>> >>>>>> >>>>> UTC 2009 >>>>> >>>>> >>>>> >>>>>> x86_64 GNU/Linux >>>>>> >>>>>> I set the ulimit parameters like those on the website: >>>>>> freeswitch at internal> ... >>>>>> Freeswitch:/opt/free-svn/bin# ulimit -a >>>>>> core file size (blocks, -c) unlimited >>>>>> data seg size (kbytes, -d) unlimited >>>>>> scheduling priority (-e) 0 >>>>>> file size (blocks, -f) unlimited >>>>>> pending signals (-i) unlimited >>>>>> max locked memory (kbytes, -l) unlimited >>>>>> max memory size (kbytes, -m) unlimited >>>>>> open files (-n) 999999 >>>>>> pipe size (512 bytes, -p) 8 >>>>>> POSIX message queues (bytes, -q) unlimited >>>>>> real-time priority (-r) 0 >>>>>> stack size (kbytes, -s) 244 >>>>>> cpu time (seconds, -t) unlimited >>>>>> max user processes (-u) unlimited >>>>>> virtual memory (kbytes, -v) unlimited >>>>>> file locks (-x) unlimited >>>>>> >>>>>> >>>>>> My network setup is the following: >>>>>> >>>>>> SIPP machine (10.10.10.1/24)----------------vlan >>>>>> >>>>>> >>>>>> >>>>> >>>>> >>>>> >>>>> >>>>>> 55 >>>>>> ----------(10.10.10.254/24 >>>>>> >>>>>> >>>>>> >>>>> ) FS >>>>> >>>>> >>>>> >>>>>> (10.10.20.254/24)-------------- >>>>>> >>>>>> >>>>>> >>>>> >>>>> >>>>> >>>>> >>>>>> vlan56 >>>>>> -------------------(10.10.20.100/24 >>>>>> >>>>>> >>>>>> >>>>> ) >>>>> >>>>> >>>>> >>>>>> OTHER STOCK FS >>>>>> >>>>>> >>>>>> I launched sipp with: >>>>>> sipp -sn uac_pcap -s 9999 -r 10 -l 80 -d 60000 -mi >>>>>> >>>>>> >>>>>> >>>>> 10.10.10.1 -i >>>>> >>>>> >>>>> >>>>>> 10.10.10.1 -mp 25000 10.10.10.254:5060 >>>>>> >>>>>> >>>>>> >>>>> >>>>> >>>>> >>>>> >>>>>> The dialplan on FS is very simple: >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>> data="sofia/external/9999 at 10.10.20.100 >>>>>> >>>>>> >>>>>> >>>>> >>>> >"/> >>>>> >>>>> >>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> FreeSWITCH Version 1.0.trunk (11560M) Started. >>>>>> Crash Protection [Disabled] >>>>>> Max Sessions[1000] >>>>>> Session Rate[100] >>>>>> SQL [Enabled] >>>>>> >>>>>> >>>>>> The test is very simple: sipp dial 9999 that matches in my >>>>>> >>>>>> >>>>>> >>>>> FS dialplan >>>>> >>>>> >>>>> >>>>>> and this is bridged to an other FS machine playing music on >>>>>> >>>>>> >>>>>> >>>>> hold. >>>>> >>>>> >>>>> >>>>>> When I launch "top" I see after 30 to 40 s that FS consumes all >>>>>> the CPU >>>>>> ressources (with a mean of 50-60 % before), with 80 calls. >>>>>> When I set 70 calls, I have to wait 70-80 s before seeing >>>>>> >>>>>> >>>>>> >>>>> the same >>>>> >>>>> >>>>> >>>>>> issue. >>>>>> >>>>>> Presence is set to false on the 2 profile. >>>>>> >>>>>> I have the same issue with FS 1.0.2 that' s why I tried FS >>>>>> >>>>>> >>>>>> >>>>> 11560. >>>>> >>>>> >>>>> >>>>>> When I use the FS machine as a router to test the packet per >>>>>> >>>>>> >>>>>> >>>>> second >>>>> >>>>> >>>>> >>>>>> performance, I'm reaching 100Mbps with 8000pps in each >>>>>> >>>>>> >>>>>> >>>>> direction (from >>>>> >>>>> >>>>> >>>>>> vlan 55 to vlan56) with less than 12% CPU. So that I don't think >>>>>> there's >>>>>> an issue with the network. >>>>>> >>>>>> Here is an "mpstat -P ALL 1" to show you what's happening >>>>>> >>>>>> >>>>>> >>>>> suddenly >>>>> >>>>> >>>>> >>>>>> with >>>>>> 70 bridge calls: >>>>>> 12:31:26 CPU %user %nice %sys %iowait %irq %soft >>>>>> %steal %idle intr/s >>>>>> 12:31:27 all 3,00 0,00 3,00 0,00 1,00 4,00 >>>>>> 0,00 89,00 6241,00 >>>>>> 12:31:27 0 3,00 0,00 3,00 0,00 1,00 4,00 >>>>>> 0,00 89,00 6241,00 >>>>>> >>>>>> 12:31:27 CPU %user %nice %sys %iowait %irq %soft >>>>>> %steal %idle intr/s >>>>>> 12:31:28 all 14,14 0,00 56,57 0,00 2,02 5,05 >>>>>> 0,00 22,22 6035,35 >>>>>> 12:31:28 0 14,14 0,00 56,57 0,00 2,02 5,05 >>>>>> 0,00 22,22 6035,35 >>>>>> >>>>>> 12:31:28 CPU %user %nice %sys %iowait %irq %soft >>>>>> %steal %idle intr/s >>>>>> 12:31:29 all 24,75 0,00 67,33 0,00 0,99 6,93 >>>>>> 0,00 0,00 5483,17 >>>>>> 12:31:29 0 24,75 0,00 67,33 0,00 0,99 6,93 >>>>>> 0,00 0,00 5483,17 >>>>>> >>>>>> >>>>>> The CPU is going from 89% idle to 0% in less than 2 seconds. >>>>>> >>>>>> I know that I don't have to expect too much from this kind of >>>>>> hardware, >>>>>> but it seems strange that the CPU power vanished so suddenly. >>>>>> >>>>>> Thanks a lot for the guys that have read this long mail :p >>>>>> >>>>>> kind regards, >>>>>> rod >>>>>> >>>>>> >>>>>> _______________________________________________ >>>>>> Freeswitch-users mailing list >>>>>> Freeswitch-users at lists.freeswitch.org >>>>>> >>>>>> >>>>>> >>>>> >>>>> >>>>> >>>>> >>>>>> >>>>> >>>>>> >>>>>> >>>>> > >>>>> >>>>> >>>>> >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> >>>>>> >>>>>> >>>>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> >>>>> >>>>> >>>>> >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> -- >>>>>> Anthony Minessale II >>>>>> >>>>>> FreeSWITCH http://www.freeswitch.org/ >>>>>> ClueCon http://www.cluecon.com/ >>>>>> >>>>>> AIM: anthm >>>>>> MSN:anthony_minessale at hotmail.com >>>>>> >>>>>> >>>>>> >>>>> >>>>> >>>>> >>>>> >>>>>> >>>>> >>>>>> >>>>>> >>>>> > >>>>> >>>>> >>>>> >>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>>>>> >>>>>> >>>>>> >>>>> >>>>> >>>>> >>>>> >>>>>> >>>>> >>>>>> >>>>>> >>>>> > >>>>> >>>>> >>>>> >>>>>> IRC: irc.freenode.net >>>>>> >>>>>> >>>>>> >>>>> #freeswitch >>>>> >>>>> >>>>> >>>>>> FreeSWITCH Developer Conference >>>>>> sip:888 at conference.freeswitch.org >>>>>> >>>>>> >>>>>> >>>>> >>>>> >>>>> >>>>> >>>>>> >>>>> >>>>>> >>>>>> >>>>> > >>>>> >>>>> >>>>> >>>>>> iax:guest at conference.freeswitch.org/888 >>>>>> >>>>>> >>>>>> >>>>> >>>>> >>>>> >>>>> >>>>>> >>>>>> googletalk:conf+888 at conference.freeswitch.org >>>>>> >>>>>> >>>>>> >>>>> >>>>> >>>>> >>>>> >>>>>> >>>>> >>>>>> >>>>>> >>>>> > >>>>> >>>>> >>>>> >>>>>> pstn:213-799-1400 >>>>>> >>>>>> >>>>>> >>>>>> >> ------------------------------------------------------------------------ >> >> >>>>> >>>>> >>>>> >>>>>> _______________________________________________ >>>>>> Freeswitch-users mailing list >>>>>> Freeswitch-users at lists.freeswitch.org >>>>>> >>>>>> >>>>>> >>>>> >>>>> >>>>> >>>>> >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> >>>>>> >>>>>> >>>>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> >>>>> >>>>> >>>>> >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>>> >>>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> >>>>> >>>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> >>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> >>>>> > ------------------------------------------------------------------------ > >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> >>>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> >>>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >>> >>> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From raul at etellicom.com Tue Feb 3 05:00:28 2009 From: raul at etellicom.com (Raul Fragoso) Date: Tue, 03 Feb 2009 11:00:28 -0200 Subject: [Freeswitch-users] debuild breaks since the last few days In-Reply-To: <0B3BFADA-4589-48AD-9F18-EEBDC6BAB740@scarlet-internet.nl> References: <0B3BFADA-4589-48AD-9F18-EEBDC6BAB740@scarlet-internet.nl> Message-ID: <1233666028.24619.0.camel@stargate> I believe that installing the libpcap and libpcap-dev packages may fix your problem. -- Raul On Tue, 2009-02-03 at 12:55 +0100, Leon de Rooij wrote: > Hi all, > > I've been trying to build new debs, but debuild seems to break.. > > I tried trunk rev 11608 and 1.0.3RC-1 and tried building the packages > with: > > debuild -i -us -uc -b > > (which worked before) > > And now it breaks at openzap with: > > cc1: warnings being treated as errors > src/ozmod/ozmod_isdn/ozmod_isdn.c: In function 'writeQ931PacketToPcap': > src/ozmod/ozmod_isdn/ozmod_isdn.c:220: warning: implicit declaration > of function 'pcap_dump_flush' > make[7]: *** [src/ozmod/ozmod_isdn/ozmod_isdn.o] Error 1 > make[7]: Leaving directory `/home/fsbuilder/freeswitch-1.0.3RC1/libs/ > openzap' > make[6]: *** [../libopenzap.so] Error 2 > make[6]: Leaving directory `/home/fsbuilder/freeswitch-1.0.3RC1/libs/ > openzap/mod_openzap' > make[5]: *** [all] Error 1 > make[5]: Leaving directory `/home/fsbuilder/freeswitch-1.0.3RC1/libs/ > openzap/mod_openzap' > make[4]: *** [../../libs/openzap/mod_openzap-all] Error 1 > make[4]: Leaving directory `/home/fsbuilder/freeswitch-1.0.3RC1/src/mod' > make[3]: *** [all-recursive] Error 1 > make[3]: Leaving directory `/home/fsbuilder/freeswitch-1.0.3RC1/src' > Making all in build > make[3]: Entering directory `/home/fsbuilder/freeswitch-1.0.3RC1/build' > +-------- FreeSWITCH Build Complete -----------+ > + FreeSWITCH has been successfully built. + > + Install by running: + > + + > + /usr/bin/make install + > +----------------------------------------------+ > make[3]: Leaving directory `/home/fsbuilder/freeswitch-1.0.3RC1/build' > make[2]: *** [all-recursive] Error 1 > make[2]: Leaving directory `/home/fsbuilder/freeswitch-1.0.3RC1' > make[1]: *** [all] Error 2 > make[1]: Leaving directory `/home/fsbuilder/freeswitch-1.0.3RC1' > make: *** [build-stamp] Error 2 > debuild: fatal error at line 1247: > debian/rules build failed > > Does anyone know how to fix this ? > > thanks, > > Leon > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From helmut.kuper at ewetel.de Tue Feb 3 05:03:31 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Tue, 03 Feb 2009 14:03:31 +0100 Subject: [Freeswitch-users] debuild breaks since the last few days In-Reply-To: <0B3BFADA-4589-48AD-9F18-EEBDC6BAB740@scarlet-internet.nl> References: <0B3BFADA-4589-48AD-9F18-EEBDC6BAB740@scarlet-internet.nl> Message-ID: <498840A3.30509@ewetel.de> Hello, yes, you have openzap upgraded to r632. Then recompile it. Make sure you have libpcap installed and pcap devel files regards helmut Am 03.02.2009 12:55, schrieb Leon de Rooij: > Hi all, > > I've been trying to build new debs, but debuild seems to break.. > > I tried trunk rev 11608 and 1.0.3RC-1 and tried building the packages > with: > > debuild -i -us -uc -b > > (which worked before) > > And now it breaks at openzap with: > > cc1: warnings being treated as errors > src/ozmod/ozmod_isdn/ozmod_isdn.c: In function 'writeQ931PacketToPcap': > src/ozmod/ozmod_isdn/ozmod_isdn.c:220: warning: implicit declaration > of function 'pcap_dump_flush' > make[7]: *** [src/ozmod/ozmod_isdn/ozmod_isdn.o] Error 1 > make[7]: Leaving directory `/home/fsbuilder/freeswitch-1.0.3RC1/libs/ > openzap' > make[6]: *** [../libopenzap.so] Error 2 > make[6]: Leaving directory `/home/fsbuilder/freeswitch-1.0.3RC1/libs/ > openzap/mod_openzap' > make[5]: *** [all] Error 1 > make[5]: Leaving directory `/home/fsbuilder/freeswitch-1.0.3RC1/libs/ > openzap/mod_openzap' > make[4]: *** [../../libs/openzap/mod_openzap-all] Error 1 > make[4]: Leaving directory `/home/fsbuilder/freeswitch-1.0.3RC1/src/mod' > make[3]: *** [all-recursive] Error 1 > make[3]: Leaving directory `/home/fsbuilder/freeswitch-1.0.3RC1/src' > Making all in build > make[3]: Entering directory `/home/fsbuilder/freeswitch-1.0.3RC1/build' > +-------- FreeSWITCH Build Complete -----------+ > + FreeSWITCH has been successfully built. + > + Install by running: + > + + > + /usr/bin/make install + > +----------------------------------------------+ > make[3]: Leaving directory `/home/fsbuilder/freeswitch-1.0.3RC1/build' > make[2]: *** [all-recursive] Error 1 > make[2]: Leaving directory `/home/fsbuilder/freeswitch-1.0.3RC1' > make[1]: *** [all] Error 2 > make[1]: Leaving directory `/home/fsbuilder/freeswitch-1.0.3RC1' > make: *** [build-stamp] Error 2 > debuild: fatal error at line 1247: > debian/rules build failed > > Does anyone know how to fix this ? > > thanks, > > Leon > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > -- Mit freundlichen Gr??en Helmut Kuper Finanzdienstleistungen und Entwicklung Telefax: (0441) 8000-2799 mailto:helmut.kuper at ewetel.de ___________________________________ EWE TEL GmbH Cloppenburger Stra?e 310 26133 Oldenburg EWE TEL GmbH Handelsregister Amtsgericht Oldenburg HRB 3723 Vorsitzender des Aufsichtsrates: Heiko Harms Gesch?ftsf?hrung: Hans-Joachim Iken (Vorsitzender), Dr. Norbert Schulz, Dirk Thole Homepage: http://www.ewetel.de ___________________________________ From raul at etellicom.com Tue Feb 3 05:12:28 2009 From: raul at etellicom.com (Raul Fragoso) Date: Tue, 03 Feb 2009 11:12:28 -0200 Subject: [Freeswitch-users] Generating calls from external source In-Reply-To: <40B58CBF-9D0F-43A3-B36F-BD05916756E4@3c.co.uk> References: <40B58CBF-9D0F-43A3-B36F-BD05916756E4@3c.co.uk> Message-ID: <1233666748.24619.8.camel@stargate> In addition do David's suggestion, you probably want to have your application to watch for some specific events after the call is originated and take action based on them. For example, you could watch for the CHANNEL_ANSWER event and play some audio file waiting for some digit, which is generated by the DTMF event. To watch only for those specific events, you should do the following just after authentication (still using Perl as an example, but the mod_event_socket is language agnostic), then you will receive those events from FreeSWITCH through the socket stream: ... print $sock "auth XXX\n\n"; print $sock "event plain CHANNEL_ANSWER DTMF\n\n"; ... To see a list of available events, please look at the following wiki pages: http://wiki.freeswitch.org/wiki/Mod_event_socket#event http://wiki.freeswitch.org/wiki/Event_list Regards, Raul On Tue, 2009-02-03 at 09:46 +0000, David Knell wrote: > Hi Nik, > > > Here's a snipped in Perl that launches an outbound call: > > > if (my $sock = IO::Socket::INET->new(Proto =>'tcp', PeerAddr => > '127.0.0.1', PeerPort => 8021)) { > print $sock "auth XXX\n\n"; > print $sock "api originate {softivr_id=$siid,src_softivr_id= > $siid,softivr_outdial=true}sofia/frombt/$ntd at 1.2.3.4 $service\n\n"; > $sock->close(); > } > > > - it does no error checking or anything, but (line by line) it: > - opens a socket to the event socket interface > - authenticates > - issues an originate which dials out to the number in $ntd. The > bits in {} set a bunch of variables on the channel, which are used by > the software which processes the call later on. The call is linked to > the extension in $service - FS looks this up in the dialplan - which > handles our end. > - closes the socket > > > Cheers -- > > > Dave > > > > > Thanks for that, coming from a C++ background it?s a refreshing > > change to be looking at something that seems logical and efficient. > > > > I?d briefly looked at the event socket and wondered if that was the > > way to go. I presume that there?s some sort of event generation > > that can trigger and external process as well somewhere, though all > > I need to do is update mysql (hopefully using some sort of pooled > > connection) > > > > I?m not using a TDM card, I have a direct interconnect with the PSTN > > breakout provider with 1,500 channels available to me. I?m finding > > Asterisk proving to be less than stable at high call volumes and > > load values spike at more than 100 calls with billing/accounting in > > place, hence my interest in FS. The only thing that?s concerning me > > is XML at the moment. Lots of code and very wordy. I?m sure I?ll > > appreciate why XML given time > > > > Regards, > > > > > > ____________________________________________________________________ > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael S Collins > > Sent: 03 February 2009 01:17 > > To: freeswitch-users at lists.freeswitch.org > > Subject: Re: [Freeswitch-users] Generating calls from external > > source > > > > Nik, > > > > Welcome to FreeSWITCH! The short answer is "yes, FS can do that." > > The first thing that you should do is unlearn "the Asterisk way" of > > thinking. Usually there is an elegant way of doing things in FS that > > wasn't possible in Ast. > > > > I would recommend that you start by looking at the event socket, > > which is somewhat analogous to the AMI only cooler. :) I have > > personally done something similar to this using the event socket and > > a Perl script. The key is to learn the syntax of the originate > > command. (definitely hit the wiki and IRC channel) > > Are you using TDM cards for this? Just curious. > > > > -MC (IRC nick: mercutioviz) > > > > Sent from my iPhone > > > > On Feb 2, 2009, at 3:35 PM, "Nik Middleton" > > wrote: > > > Hi Guys, > > > > > > As a long time Asterisk user, I?m looking into freeswitch as an > > > alternative mainly due to (list multiple reasons here) > > > > > > Can anyone give me a pointer as to how I would achieve the > > > following? > > > > > > I need to replicate an emergency broadcast system currently > > > running under Asterisk. > > > > > > At the moment, I run through a Mysql database and using the > > > manager API, issues an Originate command to dial a number. > > > > > > When the call is answered, a message is played, and the recipient > > > has the option of hitting a digit to confirm receipt. I then call > > > an AGI script to update the database. > > > > > > Is this fairly easy to do in Freeswitch? > > > > > > Not looking for code, just some pointers as to what?s available to > > > do the above / > > > > > > Regards, > > > _______________________________________________ > > > Freeswitch-users mailing list > > > Freeswitch-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From kawarod at laposte.net Tue Feb 3 05:27:35 2009 From: kawarod at laposte.net (rod) Date: Tue, 03 Feb 2009 17:27:35 +0400 Subject: [Freeswitch-users] Strange Performance when using as SBC In-Reply-To: References: <4986C42C.7030700@laposte.net><1C31435E19AB4E57981CF2AAC2CA74AA@SaeedLaptop> <4986ECA9.3040707@laposte.net><9232C06D5362494791AF713E1DF61343@SaeedLaptop> <49870432.5050301@laposte.net> Message-ID: <49884647.2080306@laposte.net> Hi Saaed, thanks for encouraging. I'm using Kamailio to get access to the carrierroute module. Carrierroute is a module that is able to handle very large routing table (excerpt from carrierroute page: "This modules scales up to more than a few million users, and is able to handle more than several hundred thousand routing table entries", Greatings to Henning Westerholt). When I did my first test with FS, LCR module was not available and as I'm not a programmer I had to deal with existing tools and being able to handle a route table with approx 160 000 entries. I'm not a programmer so I relies on SIP (which I understand better than C or C++ :p) and the possibility to define specific header to exchange message between FS and Kamailio at the cost of just an extra SIP invite parsing (maybe a bad thing for very very high call per second rate) So if you follow the setup on the wiki, FS will pass the number to examine, and Kamailio will send the best route to use depending on probability (for load sharing, eg: 10% on a gateway, 20% on an other and 70% on the last one) and matching longest prefix. Then FS uses those route. You could also update the kamailio database and then issue a "kamctl cr reload" to load the new routing table. Maybe this is not the best setup, but my aim is to share what I did so that we could converge to the best solution to use FS as a SBC, that's why I provided also some indications to optimize FS based on what I read on the list and the wiki. The next steps are scalability testing, maybe a php (or whatever else) frontend to populate carrierroute table depending on the cost of many carriers (any people willing to contribute, don't rely on me for this :o), FS redundancy (I'd like to use LVS and some tools like sipsack to check the SIP process, but I'm far from having done any interesting things on that) that is lacking against commercial SBC, some scripts to graph the number of calls... (please an SNMP module :p) An other way to achieve LCR could be to use the new LCR module, and I think that Ken Rice on this list can provide advices for a high performance LCR setup. I subscribed to this list a long time ago, and my feeling is that FS is a great piece of software with a great community, so that I decided that it could be great to contribute. regards, rod Saeed Ahmed wrote: > Hi rod, > > It's really amazing! Well described! > > Could you please explain a bit why we used Kamailio? > > Kind Regards > Saeed Ahmed Tariq > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of rod > Sent: Monday, February 02, 2009 3:33 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Strange Performance when using as SBC > > Hi Saeed, > > Here is a first draft of what I did to install FS on my server. > Configuration are not present, they'll be in a next release :p > http://wiki.freeswitch.org/wiki/SBC_Setup > > My aim is to setup FS as a SBC, I hope this page could be a great > startup point for others. I will update regularly based on what I did. > > Saeed, why are you replacing your Nextone, it's said to be one of the > best commercial SBC on the market. > > regards. > > Saeed Ahmed wrote: > >> Thanks rod for a quick answer, >> >> FS is installed on Ubuntu Server. >> >> I am planning to replace Nextone SBC with FS, Later I'll also use openZAP >> > to > >> communicate with TDM but this all depends how much calls it can take, or >> maybe we can also do something in clustering environment ( I am not sure >> about it). But thanks again and any further help will be highly >> > appreciated. > >> Kind Regards >> Saeed Ahmed Tariq >> >> >> >> -----Original Message----- >> From: freeswitch-users-bounces at lists.freeswitch.org >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of rod >> Sent: Monday, February 02, 2009 1:53 PM >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] Strange Performance when using as SBC >> >> Hi Saeed, >> >> I just created an account to share my setup on the wiki. I will detail >> all the steps for a clean install of a debian64 lenny with FS used as a >> SBC (next step is to try the new LCR module :) )and what I'm doing do >> stress the server. >> >> I wrote nothing at this time so please be patient, I'm waiting for my >> new hardware so that I will detail as much as possible what I'll do. >> >> For beginning I suggest you reading the start page on the wiki, >> especially these pages: >> -http://wiki.freeswitch.org/wiki/Getting_Started_Guide >> -http://wiki.freeswitch.org/wiki/Dialplan_XML >> >> maybe you could tell more about the linux distribution you're using so >> that I can give you some pointers for sipp... >> >> regards. >> rod. >> >> >> Saeed Ahmed wrote: >> >> >>> Hi Rod, >>> >>> Could you please share how you configured Sipp & FS to create a test >>> environment? Especially the dial plan, sofia settings etc..., actually I >>> >>> >> am >> >> >>> a newbie. I want to test it on a single FS machine. >>> >>> Kind Regards >>> Saeed >>> -----Original Message----- >>> From: freeswitch-users-bounces at lists.freeswitch.org >>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of rod >>> Sent: Monday, February 02, 2009 11:00 AM >>> To: freeswitch-users at lists.freeswitch.org >>> Subject: Re: [Freeswitch-users] Strange Performance when using as SBC >>> >>> Hi Ken, Jay, >>> >>> thanks for pointing to proxy media, I will test. >>> >>> Ken, you are right, I was brain damaged (a stupid mistake) when setting >>> INFO cause this kind of level could be very verbose. I'm switching to >>> CRIT or ERR. >>> >>> Thanks guys, >>> rod. >>> >>> thanks for >>> >>> Ken Rice wrote: >>> >>> >>> >>>> If you don't have to transcode, using proxy media mode will still save >>>> >>>> >> you >> >> >>>> some CPU time. This is 1/2 way between bypass media and the default >>>> > media > >>>> interactive mode. The other draw back to this mode is if you are using >>>> > FS > >>>> >>>> >>>> >>> to >>> >>> >>> >>>> clean up RTP and DTMF you loose those functions but they are not needed >>>> >>>> >> in >> >> >>>> most use cases. >>>> >>>> As far as the log level goes, I found that once I had things stable >>>> >>>> >>>> >>> setting >>> >>> >>> >>>> the loglevel to helped a good deal... Info is probably a bit too high of >>>> >>>> >> a >> >> >>>> loglevel I would probably go for CRIT or ERR (2 or 1 respectively) if >>>> > you > >>>> insist on leaving logging turned on... On a busy system these can and >>>> >>>> >> will >> >> >>>> generate a good deal of activity (and disk IO if using mod_logfile) >>>> >>>> Ken >>>> >>>> >>>> >>>> >>>> >>>> >>>>> From: rod >>>>> Reply-To: >>>>> Date: Mon, 02 Feb 2009 11:36:35 +0400 >>>>> To: >>>>> Subject: Re: [Freeswitch-users] Strange Performance when using as SBC >>>>> >>>>> Hi Ken, >>>>> >>>>> 1) I'd like to use FS to hide topology, so bypass media is not possible >>>>> 2) done >>>>> 3) done >>>>> 4) not used >>>>> 5) i'm using this ins switch.xml -> >>>> value="info"/>, if you think an other log level is more suitable. >>>>> >>>>> Regarding logging, I can see in console and in the freeswitch.log that >>>>> there is still a lot of NOTICE logging, see below: >>>>> 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:960 >>>>> switch_core_session_thread() Session 8721 >>>>> (sofia/internal/sipp at 10.10.10.1:5060) Ended >>>>> 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:962 >>>>> switch_core_session_thread() Close Channel >>>>> sofia/internal/sipp at 10.10.10.1:5060 [CS_HANGUP] >>>>> 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:960 >>>>> switch_core_session_thread() Session 8722 >>>>> (sofia/external/9998 at 10.10.20.100) Ended >>>>> 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:962 >>>>> switch_core_session_thread() Close Channel >>>>> sofia/external/9998 at 10.10.20.100 [CS_HANGUP] >>>>> 2009-02-02 08:33:56 [NOTICE] sofia.c:3164 sofia_handle_sip_i_state() >>>>> Channel [sofia/external/9998 at 10.10.20.100] has been answered >>>>> 2009-02-02 08:33:56 [WARNING] mod_sofia.c:740 sofia_read_frame() >>>>> Changing codec ptime to 30. I bet you have a linksys/sipura =D >>>>> >>>>> Do you have any idea where I can switch off this kind of logging. I >>>>> thought it should be in /dialplan/internal.xml, but I see that in >>>>> internal.xml -> >>>>> >>>>> thanks a lot for your suggestion. >>>>> >>>>> regards, >>>>> rod >>>>> >>>>> Ken Rice wrote: >>>>> >>>>> >>>>> >>>>> >>>>>> Dont forget there are several things you can do to increase >>>>>> >>>>>> >>>>>> >>> performance... >>> >>> >>> >>>>>> 1) where possible use bypass media or media proxy modes >>>>>> 2) mount freeswitch/db as a ram drive (if you are using voicemail with >>>>>> the internal FS DBs you'll need a way to make this persistant across >>>>>> reboots) >>>>>> 3) see the wiki for setting reasonable ulimits >>>>>> 4) (this is my oppinion others may vary) dont use mod_cdr_csv >>>>>> 5) turn off (or reduce logging) in switch.conf.xml >>>>>> >>>>>> all of these thing can greatly improve performance. >>>>>> >>>>>> On Mon, Feb 2, 2009 at 1:04 AM, rod >>>>> > wrote: >>>>>> >>>>>> Thanks Anthony, >>>>>> >>>>>> the setup is like this: >>>>>> >>>>>> sipp server ---- FS 1 ---- FS2 >>>>>> >>>>>> FS1 is the AMD CPU that has only one extension in dialplan that >>>>>> bridges >>>>>> 9999 to FS2. 9999 is the first extension in FS2 dialplan that >>>>>> plays moh, >>>>>> FS2 has no CPU pbm. >>>>>> >>>>>> FS1 is maxing out at 60 bridged calls without your option -hp. >>>>>> >>>>>> Using -hp, I'm now able to bridge 200 concurrent calls (a great >>>>>> improvement) and the system is still reactive. CPU load is high >>>>>> but not >>>>>> 100% and as the system responds well, I think that doesn't matter. >>>>>> >>>>>> >>>>>> >>> The >>> >>> >>> >>>>>> 2GB of memory are completely consumed (top command shows 700MB for >>>>>> >>>>>> >>>>>> >>> FS >>> >>> >>> >>>>>> process). >>>>>> >>>>>> I understand that FS1 server is not the best hardware platform, >>>>>> and I'm >>>>>> waiting for new 4 cores server for testing. >>>>>> I will update those numbers when testing with the new hardware. >>>>>> >>>>>> regards, >>>>>> rod. >>>>>> >>>>>> Anthony Minessale wrote: >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> Which of the 2 machines has the load issue? You said it was one box >>>>>>> calling the other. >>>>>>> >>>>>>> You have 2 major things against you, single CPU and AMD, but you >>>>>>> should at least be able to get in the vicinity of 800-1000 calls >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> on a >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> box like that. >>>>>>> >>>>>>> Are you calling the default 9999? It's not really an appropriate >>>>>>> extension for load testing. >>>>>>> On the terminating box you should set up a manual extension that is >>>>>>> the first one in the dial plan >>>>>>> to play a wav file from preferably a ram disk or /tmp >>>>>>> >>>>>>> If you do plan on using this in production accept nothing less >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> than a >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> multi-core intel machine with at least 4 cores, the more cores the >>>>>>> better because that parallel processing is where FS gets it's >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> atvantage. >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> On Fri, Jan 30, 2009 at 5:56 AM, rod >>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> >> wrote: >>>>>>> >>>>>>> Dear list, >>>>>>> >>>>>>> I've been playing with freeswitch for some time (2 months) >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> and the >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> fact >>>>>>> is that I'm very pleased with the functionnalities of this >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> software. >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> I'd like to use FS as a SBC handling media and I'm doing some >>>>>>> tests with >>>>>>> sipp to load the machine but I'm unable to bridge more than >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> 60 calls >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> without seeing the CPU being loaded at 100%. I'm sure >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> something is >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> going >>>>>>> wrong with my setup but I'm unable to see what. >>>>>>> >>>>>>> The test machine has the following specs: >>>>>>> Athlon XP 3500+ with 2GB of memory (I know this is not a >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> high end >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> machine :p) >>>>>>> >>>>>>> Freeswitch:/opt/freeswitch/log# cat /proc/cpuinfo >>>>>>> processor : 0 >>>>>>> vendor_id : AuthenticAMD >>>>>>> cpu family : 15 >>>>>>> model : 95 >>>>>>> model name : AMD Athlon(tm) 64 Processor 3500+ >>>>>>> stepping : 2 >>>>>>> cpu MHz : 2199.973 >>>>>>> cache size : 512 KB >>>>>>> fpu : yes >>>>>>> fpu_exception : yes >>>>>>> cpuid level : 1 >>>>>>> wp : yes >>>>>>> flags : fpu vme de pse tsc msr pae mce cx8 apic >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> sep mtrr pge >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> mca cmov pat pse36 clflush mmx fxsr sse sse2 syscall nx mmxext >>>>>>> fxsr_opt >>>>>>> rdtscp lm 3dnowext 3dnow up rep_good pni cx16 lahf_lm svm >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> extapic >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> cr8_legacy >>>>>>> bogomips : 4402.97 >>>>>>> TLB size : 1024 4K pages >>>>>>> clflush size : 64 >>>>>>> cache_alignment : 64 >>>>>>> address sizes : 40 bits physical, 48 bits virtual >>>>>>> power management: ts fid vid ttp tm stc >>>>>>> >>>>>>> I installed FS on a fresh debian 64: >>>>>>> Linux Freeswitch 2.6.26-1-amd64 #1 SMP Sat Jan 10 17:57:00 >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> UTC 2009 >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> x86_64 GNU/Linux >>>>>>> >>>>>>> I set the ulimit parameters like those on the website: >>>>>>> freeswitch at internal> ... >>>>>>> Freeswitch:/opt/free-svn/bin# ulimit -a >>>>>>> core file size (blocks, -c) unlimited >>>>>>> data seg size (kbytes, -d) unlimited >>>>>>> scheduling priority (-e) 0 >>>>>>> file size (blocks, -f) unlimited >>>>>>> pending signals (-i) unlimited >>>>>>> max locked memory (kbytes, -l) unlimited >>>>>>> max memory size (kbytes, -m) unlimited >>>>>>> open files (-n) 999999 >>>>>>> pipe size (512 bytes, -p) 8 >>>>>>> POSIX message queues (bytes, -q) unlimited >>>>>>> real-time priority (-r) 0 >>>>>>> stack size (kbytes, -s) 244 >>>>>>> cpu time (seconds, -t) unlimited >>>>>>> max user processes (-u) unlimited >>>>>>> virtual memory (kbytes, -v) unlimited >>>>>>> file locks (-x) unlimited >>>>>>> >>>>>>> >>>>>>> My network setup is the following: >>>>>>> >>>>>>> SIPP machine (10.10.10.1/24)----------------vlan >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> 55 >>>>>>> ----------(10.10.10.254/24 >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> ) FS >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> (10.10.20.254/24)-------------- >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> vlan56 >>>>>>> -------------------(10.10.20.100/24 >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> ) >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> OTHER STOCK FS >>>>>>> >>>>>>> >>>>>>> I launched sipp with: >>>>>>> sipp -sn uac_pcap -s 9999 -r 10 -l 80 -d 60000 -mi >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> 10.10.10.1 -i >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> 10.10.10.1 -mp 25000 10.10.10.254:5060 >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> The dialplan on FS is very simple: >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> data="sofia/external/9999 at 10.10.20.100 >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> >>>>> >"/> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> FreeSWITCH Version 1.0.trunk (11560M) Started. >>>>>>> Crash Protection [Disabled] >>>>>>> Max Sessions[1000] >>>>>>> Session Rate[100] >>>>>>> SQL [Enabled] >>>>>>> >>>>>>> >>>>>>> The test is very simple: sipp dial 9999 that matches in my >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> FS dialplan >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> and this is bridged to an other FS machine playing music on >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> hold. >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> When I launch "top" I see after 30 to 40 s that FS consumes all >>>>>>> the CPU >>>>>>> ressources (with a mean of 50-60 % before), with 80 calls. >>>>>>> When I set 70 calls, I have to wait 70-80 s before seeing >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> the same >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> issue. >>>>>>> >>>>>>> Presence is set to false on the 2 profile. >>>>>>> >>>>>>> I have the same issue with FS 1.0.2 that' s why I tried FS >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> 11560. >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> When I use the FS machine as a router to test the packet per >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> second >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> performance, I'm reaching 100Mbps with 8000pps in each >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> direction (from >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> vlan 55 to vlan56) with less than 12% CPU. So that I don't think >>>>>>> there's >>>>>>> an issue with the network. >>>>>>> >>>>>>> Here is an "mpstat -P ALL 1" to show you what's happening >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> suddenly >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> with >>>>>>> 70 bridge calls: >>>>>>> 12:31:26 CPU %user %nice %sys %iowait %irq %soft >>>>>>> %steal %idle intr/s >>>>>>> 12:31:27 all 3,00 0,00 3,00 0,00 1,00 4,00 >>>>>>> 0,00 89,00 6241,00 >>>>>>> 12:31:27 0 3,00 0,00 3,00 0,00 1,00 4,00 >>>>>>> 0,00 89,00 6241,00 >>>>>>> >>>>>>> 12:31:27 CPU %user %nice %sys %iowait %irq %soft >>>>>>> %steal %idle intr/s >>>>>>> 12:31:28 all 14,14 0,00 56,57 0,00 2,02 5,05 >>>>>>> 0,00 22,22 6035,35 >>>>>>> 12:31:28 0 14,14 0,00 56,57 0,00 2,02 5,05 >>>>>>> 0,00 22,22 6035,35 >>>>>>> >>>>>>> 12:31:28 CPU %user %nice %sys %iowait %irq %soft >>>>>>> %steal %idle intr/s >>>>>>> 12:31:29 all 24,75 0,00 67,33 0,00 0,99 6,93 >>>>>>> 0,00 0,00 5483,17 >>>>>>> 12:31:29 0 24,75 0,00 67,33 0,00 0,99 6,93 >>>>>>> 0,00 0,00 5483,17 >>>>>>> >>>>>>> >>>>>>> The CPU is going from 89% idle to 0% in less than 2 seconds. >>>>>>> >>>>>>> I know that I don't have to expect too much from this kind of >>>>>>> hardware, >>>>>>> but it seems strange that the CPU power vanished so suddenly. >>>>>>> >>>>>>> Thanks a lot for the guys that have read this long mail :p >>>>>>> >>>>>>> kind regards, >>>>>>> rod >>>>>>> >>>>>>> >>>>>>> _______________________________________________ >>>>>>> Freeswitch-users mailing list >>>>>>> Freeswitch-users at lists.freeswitch.org >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> >>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> > >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >>> >>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> -- >>>>>>> Anthony Minessale II >>>>>>> >>>>>>> FreeSWITCH http://www.freeswitch.org/ >>>>>>> ClueCon http://www.cluecon.com/ >>>>>>> >>>>>>> AIM: anthm >>>>>>> MSN:anthony_minessale at hotmail.com >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> >>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> > >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> >>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> > >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> IRC: irc.freenode.net >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> #freeswitch >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> FreeSWITCH Developer Conference >>>>>>> sip:888 at conference.freeswitch.org >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> >>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> > >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> iax:guest at conference.freeswitch.org/888 >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> >>>>>>> googletalk:conf+888 at conference.freeswitch.org >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> >>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> > >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> pstn:213-799-1400 >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>> ------------------------------------------------------------------------ >>> >>> >>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> _______________________________________________ >>>>>>> Freeswitch-users mailing list >>>>>>> Freeswitch-users at lists.freeswitch.org >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >>> >>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> _______________________________________________ >>>>>> Freeswitch-users mailing list >>>>>> Freeswitch-users at lists.freeswitch.org >>>>>> >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> >>>>>> >>>>>> >>>>>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >>> >>> >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >> ------------------------------------------------------------------------ >> >> >>>>>> _______________________________________________ >>>>>> Freeswitch-users mailing list >>>>>> Freeswitch-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> >>>>>> >>>>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> >>>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> >>>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> >>>> >>>> >>>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >>> >>> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > From raul at etellicom.com Tue Feb 3 05:33:49 2009 From: raul at etellicom.com (Raul Fragoso) Date: Tue, 03 Feb 2009 11:33:49 -0200 Subject: [Freeswitch-users] Application language to support C or C++? In-Reply-To: <2cbf225c0902021821u2bc8b622ka5d9f94f05a968ae@mail.gmail.com> References: <2cbf225c0902021821u2bc8b622ka5d9f94f05a968ae@mail.gmail.com> Message-ID: <1233668029.24619.29.camel@stargate> Depending on what you want to do, I suggest having a look at mod_event_socket: http://wiki.freeswitch.org/wiki/Mod_event_socket That module is a socket based interface that provides a vast range of options to control FreeSWITCH and its applications. Just for the record, my application is entirely written in C++ and uses FreeSWITCH as a back-end for providing PBX functionality through a combination of mod_event_socket and mod_xml_curl. Regards, Raul On Tue, 2009-02-03 at 10:21 +0800, lee jason wrote: > Hi All, > > I saw the applications using FreeSwitch library can be written > in JavaScript, Perl, Python and Lua but I need to use Linux C or C++ > for applications, Is FreeSwitch can supported it? Where can I get the > sample codes? My Linux platform is base on Fedora. > > > > > Thanks a lot. > > > Jason > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From cstomi.levlist at gmail.com Tue Feb 3 05:51:29 2009 From: cstomi.levlist at gmail.com (Tamas Cseke) Date: Tue, 03 Feb 2009 14:51:29 +0100 Subject: [Freeswitch-users] fifo problem Message-ID: <49884BE1.2000603@gmail.com> Hello, We have a problem with mod_fifo. we monitor fifo push event on event socket, call consumer with originate & fifo out nowait Similar like fifo_outbound works, but we have an external strategy for consumer selection (eg.: skill-based routing) The problem is when a caller stops waiting in fifo, the originated calls kepp ringing the consumer, and when the consumer answer the call, he or she may grab somebody else from the fifo, which is a problem because the callers are identified and some data (eg name, phonenumber is shown for the consumer). so it can happen these data will be wrong. We tried to resolve this issue by a call tracking in the external script using event socket. we pushes a variable into the CHANNEL_ORIGINATE event calling the consumer containing the caller uuid. and if the caller aborts the fifo, we hangup the consumer call with (uuid_kill) But it's not prefect becasue it can happen that the consumer pop another caller from the fifo. and we hangup this call, so as a side-effect we loosing another caller. Could anybody advise a solution for this please? we thinking about to have a fifo_caller_uuid variable, that we set before calling fifo with the out method. and if this uuid is in the top of the fifo then pop it else don't pop anybody. it seems to be a hack anyway.... Thanks in advance, Tamas From anthony.minessale at gmail.com Tue Feb 3 05:54:29 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 3 Feb 2009 07:54:29 -0600 Subject: [Freeswitch-users] Conference dialing and uuid In-Reply-To: <20090203082530.GA17166@cpdata.co.za> References: <20090130103944.GA23536@cpdata.co.za> <20090130133315.GB23536@cpdata.co.za> <191c3a030901300605r78dd8943w7fb1075ef851e1d@mail.gmail.com> <20090202085157.GA3555@cpdata.co.za> <0F863750-5C77-4CD9-ADC6-CFDDF2B149B4@freeswitch.org> <20090202102459.GA4179@cpdata.co.za> <9FE1733C-A46A-4A17-8AEE-077DE8010235@freeswitch.org> <191c3a030902020601v1109b1e9he867e9f1e1d4a401@mail.gmail.com> <20090203072221.GD16105@cpdata.co.za> <20090203082530.GA17166@cpdata.co.za> Message-ID: <191c3a030902030554y13e3dd7aj2eeaa43b86b1ced4@mail.gmail.com> There is a file format called tone_stream that I was trying to explain yesterday. tone_stream:// or tone_stream://path=/path/to/text_file.ttml you can use this to play tones anywhere a filename is supposed to go. I guess loopback really is your only option if you must generate ringback. Typically, whatever gateway you are calling out over will go into early media and start playing the real ringback. You should not execute any apps during the on_ring_execute that block, (playing audio etc) Media has not even been established at that point and you have nobody to play the audio to anyway, But you will block from that point until the application you chose has ended so you should only execute small apps that return immediately such as setting a variable etc. As for ringback I think you have the whole thing reversed in your head. the ringback vars etc only apply to the origination (a) leg of a call. If you make an inbound call set the ringback variable and then call bridge, the ringback var is parsed on that inbound leg and the dialout process of the bridge app involves 2 channels the A leg and the B leg. When the B leg gets a ring indication and the A leg detects it, it will begin to play the ringback sound you chose back to the originator of that inbound leg. In the conference or using originate situation, you are doing an outbound call with no relevant inbound call, so there is nothing to generate ringback to. That's why loopback works because it cross connects an outbound call back to an inbound call which gives the bridge app everything it needs to be able to generate artificial ringback. On Tue, Feb 3, 2009 at 2:25 AM, Sias Mey wrote: > Hmmm no MOH wont work... since I am planning on pulling more than just 2 > members into the conference and I still need ringback for the later > members as well. > > Is there a direct way for me to use conference play > to play teletone directly? or should I just records some ringing if I > want to use that? > > And lastly for my own sanity ;-) why would the following in a > on_ring_execute stop execution of the call at that point? > > call = argv[1]; > conf = argv[2]; > > consoleLog("info","Making ringback channel for uuid : "+ session.uuid > +"\n"); > var ringuuid = apiExecute("originate","loopback/ringback-conf="+ conf > +"-conf &park()") > > //I tried with and without a exit() at the end > > It seems to stop media detection??(not really sure about the term) for the > call that executes this > script. > > Freeswitch doesent recognize the pickup of that call and thus it doesent > get bridged into the conference. when I uuid_kill the call that gets > originated everything else starts happening again. > > Oh Im running this in FS ver. 1.0.trunk (11226:11561M) > > and that loopback points to > > > > > > > and ringback.js is > > use("TeleTone"); > session.answer(); > var tts = new TeleTone(session); > > tts.addTone("u", 400.0, 450.0, 0.0); > tts.addTone("r", 440.0, 480.0, 0.0); > > var RESET = "v=2000;>=0;+=0;"; > var UK_RING = RESET + "L=2;u(400,200);u(400,2200)"; > var US_RING = RESET + "r(2000,4000)"; > > while(session.ready()) { > console_log("making UK ring\n"); > for (x = 0 ; x < 2 ; x++) { > tts.generate(UK_RING); > } > } > > A slight bastardisation of the teletone JS example. > > I would expected the new channel that is created via a api originate to > be completely seperate from the JS I create it in. (thats why I use api > instead of creating a new session, although I should probably try that > as well). > > I use some CoreDB stuff to keep tabs on the uuid for the originated call > so that I can uuid_kill it in the on_answer_script but as mentioned... > the on_answer only executes after I uuid_kill the originated channel in > the cli... > > Thanks again guys, > Specially since it seems you two are always the ones that get back to > me. > > On Tue, Feb 03, 2009 at 09:22:21AM +0200, Sias Mey wrote: > > Actually loopback does work. > > however as I said it generates a pair of extra channels. > > > > Hmmm I was trying to generate and extra call to a JS script that > > generated a teletone ring in an on_ring_execute for the second call > > however it seems to stop execution of the call itself. Event though I > > use api commands to originate and then transfer it into the conference > > so that I have direct access to its uuid. > > > > I think changeing the moh might be a bit simpler however and elimite > > some CoreDB stuff I was doing to keep track of the calls ring generating > > call (what a sentance). > > > > On Mon, Feb 02, 2009 at 08:01:25AM -0600, Anthony Minessale wrote: > > > you could set the conference moh sound to be tone_stream::// with > the > > > teletone spec for ring sound and it use ignore_early_media=true in > your > > > originates so the first caller would hear ringback until the 2nd one > > > arrived. > > > > > > On Mon, Feb 2, 2009 at 4:29 AM, Brian West <[1]brian at freeswitch.org > > > > > wrote: > > > > > > Loopback will not work in that case either. If the far end plays > > > ringback inband you should hear that if you use the conference > dial > > > api call. > > > /b > > > > > > On Feb 2, 2009, at 4:24 AM, Sias Mey wrote: > > > > Aaah ok. > > > > > > > > Thanks for clearing that up. > > > > > > > > So using loopback is still the only real workable sollution for > me, > > > > since that generates ringback from and alternative endpoint and > > > > plays it > > > > into the conference. > > > > > > > > I might play with some javascript that streams ring into the > channel > > > > eventually but for now the string comparisons at least get me the > > > > right > > > > uuid. > > > > > > > > Thank you again, > > > > Sias > > > > > > _______________________________________________ > > > Freeswitch-users mailing list > > > [2]Freeswitch-users at lists.freeswitch.org > > > [3]http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:[4] > http://lists.freeswitch.org/mailman/options/freeswitch-u > > > sers > > > [5]http://www.freeswitch.org > > > > > > -- > > > Anthony Minessale II > > > FreeSWITCH [6]http://www.freeswitch.org/ > > > ClueCon [7]http://www.cluecon.com/ > > > AIM: anthm > > > [8]MSN:anthony_minessale at hotmail.com > > > GTALK/JABBER/[9]PAYPAL:anthony.minessale at gmail.com > > > IRC: [10]irc.freenode.net #freeswitch > > > FreeSWITCH Developer Conference > > > [11]sip:888 at conference.freeswitch.org > > > [12]iax:guest at conference.freeswitch.org/888 > > > [13]googletalk:conf+888 at conference.freeswitch.org > > > pstn:213-799-1400 > > > > > > References > > > > > > 1. mailto:brian at freeswitch.org > > > 2. mailto:Freeswitch-users at lists.freeswitch.org > > > 3. http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > 4. http://lists.freeswitch.org/mailman/options/freeswitch-users > > > 5. http://www.freeswitch.org/ > > > 6. http://www.freeswitch.org/ > > > 7. http://www.cluecon.com/ > > > 8. mailto:MSN%3Aanthony_minessale at hotmail.com > > > 9. mailto:PAYPAL%3Aanthony.minessale at gmail.com > > > 10. http://irc.freenode.net/ > > > 11. mailto:sip%3A888 at conference.freeswitch.org > > > 12. http://iax:guest at conference.freeswitch.org/888 > > > 13. mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org > > > > > _______________________________________________ > > > Freeswitch-users mailing list > > > Freeswitch-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090203/0e6d61e3/attachment-0001.html From anthony.minessale at gmail.com Tue Feb 3 06:09:32 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 3 Feb 2009 08:09:32 -0600 Subject: [Freeswitch-users] Generating calls from external source In-Reply-To: <1233666748.24619.8.camel@stargate> References: <40B58CBF-9D0F-43A3-B36F-BD05916756E4@3c.co.uk> <1233666748.24619.8.camel@stargate> Message-ID: <191c3a030902030609i5c3f415fhed501aa492d29cd3@mail.gmail.com> There is also an event socket library written in C called esl that is in the fs tree in the libs directory. This has the ability to establish connections both inbound and outbound from FS. There is also a perl module FreeSWITCH::Client that mr collins may be interested in in the tree as well. On Tue, Feb 3, 2009 at 7:12 AM, Raul Fragoso wrote: > In addition do David's suggestion, you probably want to have your > application to watch for some specific events after the call is > originated and take action based on them. For example, you could watch > for the CHANNEL_ANSWER event and play some audio file waiting for some > digit, which is generated by the DTMF event. > To watch only for those specific events, you should do the following > just after authentication (still using Perl as an example, but the > mod_event_socket is language agnostic), then you will receive those > events from FreeSWITCH through the socket stream: > > ... > print $sock "auth XXX\n\n"; > print $sock "event plain CHANNEL_ANSWER DTMF\n\n"; > ... > > To see a list of available events, please look at the following wiki > pages: > http://wiki.freeswitch.org/wiki/Mod_event_socket#event > http://wiki.freeswitch.org/wiki/Event_list > > Regards, > > Raul > > On Tue, 2009-02-03 at 09:46 +0000, David Knell wrote: > > Hi Nik, > > > > > > Here's a snipped in Perl that launches an outbound call: > > > > > > if (my $sock = IO::Socket::INET->new(Proto =>'tcp', PeerAddr => > > '127.0.0.1', PeerPort => 8021)) { > > print $sock "auth XXX\n\n"; > > print $sock "api originate {softivr_id=$siid,src_softivr_id= > > $siid,softivr_outdial=true}sofia/frombt/$ntd at 1.2.3.4 $service\n\n"; > > $sock->close(); > > } > > > > > > - it does no error checking or anything, but (line by line) it: > > - opens a socket to the event socket interface > > - authenticates > > - issues an originate which dials out to the number in $ntd. The > > bits in {} set a bunch of variables on the channel, which are used by > > the software which processes the call later on. The call is linked to > > the extension in $service - FS looks this up in the dialplan - which > > handles our end. > > - closes the socket > > > > > > Cheers -- > > > > > > Dave > > > > > > > > > Thanks for that, coming from a C++ background it's a refreshing > > > change to be looking at something that seems logical and efficient. > > > > > > I'd briefly looked at the event socket and wondered if that was the > > > way to go. I presume that there's some sort of event generation > > > that can trigger and external process as well somewhere, though all > > > I need to do is update mysql (hopefully using some sort of pooled > > > connection) > > > > > > I'm not using a TDM card, I have a direct interconnect with the PSTN > > > breakout provider with 1,500 channels available to me. I'm finding > > > Asterisk proving to be less than stable at high call volumes and > > > load values spike at more than 100 calls with billing/accounting in > > > place, hence my interest in FS. The only thing that's concerning me > > > is XML at the moment. Lots of code and very wordy. I'm sure I'll > > > appreciate why XML given time > > > > > > Regards, > > > > > > > > > ____________________________________________________________________ > > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael S > Collins > > > Sent: 03 February 2009 01:17 > > > To: freeswitch-users at lists.freeswitch.org > > > Subject: Re: [Freeswitch-users] Generating calls from external > > > source > > > > > > Nik, > > > > > > Welcome to FreeSWITCH! The short answer is "yes, FS can do that." > > > The first thing that you should do is unlearn "the Asterisk way" of > > > thinking. Usually there is an elegant way of doing things in FS that > > > wasn't possible in Ast. > > > > > > I would recommend that you start by looking at the event socket, > > > which is somewhat analogous to the AMI only cooler. :) I have > > > personally done something similar to this using the event socket and > > > a Perl script. The key is to learn the syntax of the originate > > > command. (definitely hit the wiki and IRC channel) > > > Are you using TDM cards for this? Just curious. > > > > > > -MC (IRC nick: mercutioviz) > > > > > > Sent from my iPhone > > > > > > On Feb 2, 2009, at 3:35 PM, "Nik Middleton" > > > wrote: > > > > Hi Guys, > > > > > > > > As a long time Asterisk user, I'm looking into freeswitch as an > > > > alternative mainly due to (list multiple reasons here) > > > > > > > > Can anyone give me a pointer as to how I would achieve the > > > > following? > > > > > > > > I need to replicate an emergency broadcast system currently > > > > running under Asterisk. > > > > > > > > At the moment, I run through a Mysql database and using the > > > > manager API, issues an Originate command to dial a number. > > > > > > > > When the call is answered, a message is played, and the recipient > > > > has the option of hitting a digit to confirm receipt. I then call > > > > an AGI script to update the database. > > > > > > > > Is this fairly easy to do in Freeswitch? > > > > > > > > Not looking for code, just some pointers as to what's available to > > > > do the above / > > > > > > > > Regards, > > > > _______________________________________________ > > > > Freeswitch-users mailing list > > > > Freeswitch-users at lists.freeswitch.org > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > http://www.freeswitch.org > > > _______________________________________________ > > > Freeswitch-users mailing list > > > Freeswitch-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090203/586ea54a/attachment.html From sias at cpdata.co.za Tue Feb 3 06:16:21 2009 From: sias at cpdata.co.za (Sias Mey) Date: Tue, 3 Feb 2009 16:16:21 +0200 Subject: [Freeswitch-users] Conference dialing and uuid In-Reply-To: <191c3a030902030554y13e3dd7aj2eeaa43b86b1ced4@mail.gmail.com> References: <20090130133315.GB23536@cpdata.co.za> <191c3a030901300605r78dd8943w7fb1075ef851e1d@mail.gmail.com> <20090202085157.GA3555@cpdata.co.za> <0F863750-5C77-4CD9-ADC6-CFDDF2B149B4@freeswitch.org> <20090202102459.GA4179@cpdata.co.za> <9FE1733C-A46A-4A17-8AEE-077DE8010235@freeswitch.org> <191c3a030902020601v1109b1e9he867e9f1e1d4a401@mail.gmail.com> <20090203072221.GD16105@cpdata.co.za> <20090203082530.GA17166@cpdata.co.za> <191c3a030902030554y13e3dd7aj2eeaa43b86b1ced4@mail.gmail.com> Message-ID: <20090203141621.GA8916@cpdata.co.za> Hmm ok ... Ill try that In my head though the api call to originate shouldent block? but I assume since it does my head is wrong. Thanks you for the explanation. I think you can put this one to bed now :-P On Tue, Feb 03, 2009 at 07:54:29AM -0600, Anthony Minessale wrote: > There is a file format called tone_stream that I was trying to explain > yesterday. > tone_stream:// > or > tone_stream://path=/path/to/text_file.ttml > you can use this to play tones anywhere a filename is supposed to go. > I guess loopback really is your only option if you must generate > ringback. > Typically, whatever gateway you are calling out over will go into early > media and start playing the real ringback. > You should not execute any apps during the on_ring_execute that block, > (playing audio etc) > Media has not even been established at that point and you have nobody > to play the audio to anyway, > But you will block from that point until the application you chose has > ended so you should only execute small apps that > return immediately such as setting a variable etc. > As for ringback I think you have the whole thing reversed in your > head. > the ringback vars etc only apply to the origination (a) leg of a call. > If you make an inbound call set the ringback variable and then call > bridge, the ringback var is parsed on that inbound leg > and the dialout process of the bridge app involves 2 channels the A leg > and the B leg. When the B leg gets a ring indication and the A leg > detects it, it will begin to play the ringback sound you chose back to > the originator of that inbound leg. > In the conference or using originate situation, you are doing an > outbound call with no relevant inbound call, so there is nothing > to generate ringback to. That's why loopback works because it cross > connects an outbound call back to an inbound call which gives the > bridge app everything it needs to be able to generate artificial > ringback. > > On Tue, Feb 3, 2009 at 2:25 AM, Sias Mey <[1]sias at cpdata.co.za> wrote: > > Hmmm no MOH wont work... since I am planning on pulling more than > just 2 > members into the conference and I still need ringback for the later > members as well. > Is there a direct way for me to use conference play > > to play teletone directly? or should I just records some ringing if > I > want to use that? > And lastly for my own sanity ;-) why would the following in a > on_ring_execute stop execution of the call at that point? > call = argv[1]; > conf = argv[2]; > consoleLog("info","Making ringback channel for uuid : "+ > session.uuid > +"\n"); > var ringuuid = apiExecute("originate","loopback/ringback-conf="+ > conf +"-conf &park()") > //I tried with and without a exit() at the end > It seems to stop media detection??(not really sure about the term) > for the call that executes this > script. > Freeswitch doesent recognize the pickup of that call and thus it > doesent > get bridged into the conference. when I uuid_kill the call that gets > originated everything else starts happening again. > Oh Im running this in FS ver. 1.0.trunk (11226:11561M) > and that loopback points to > > expression="^ringback-conf=(.*)$"> > > > > and ringback.js is > use("TeleTone"); > session.answer(); > var tts = new TeleTone(session); > tts.addTone("u", 400.0, 450.0, 0.0); > tts.addTone("r", 440.0, 480.0, 0.0); > var RESET = "v=2000;>=0;+=0;"; > var UK_RING = RESET + "L=2;u(400,200);u(400,2200)"; > var US_RING = RESET + "r(2000,4000)"; > while(session.ready()) { > console_log("making UK ring\n"); > for (x = 0 ; x < 2 ; x++) { > tts.generate(UK_RING); > } > } > A slight bastardisation of the teletone JS example. > I would expected the new channel that is created via a api originate > to > be completely seperate from the JS I create it in. (thats why I use > api > instead of creating a new session, although I should probably try > that > as well). > I use some CoreDB stuff to keep tabs on the uuid for the originated > call > so that I can uuid_kill it in the on_answer_script but as > mentioned... > the on_answer only executes after I uuid_kill the originated channel > in > the cli... > Thanks again guys, > Specially since it seems you two are always the ones that get back > to > me. > > On Tue, Feb 03, 2009 at 09:22:21AM +0200, Sias Mey wrote: > > Actually loopback does work. > > however as I said it generates a pair of extra channels. > > > > Hmmm I was trying to generate and extra call to a JS script that > > generated a teletone ring in an on_ring_execute for the second call > > however it seems to stop execution of the call itself. Event though I > > use api commands to originate and then transfer it into the > conference > > so that I have direct access to its uuid. > > > > I think changeing the moh might be a bit simpler however and elimite > > some CoreDB stuff I was doing to keep track of the calls ring > generating > > call (what a sentance). > > > > On Mon, Feb 02, 2009 at 08:01:25AM -0600, Anthony Minessale wrote: > > > you could set the conference moh sound to be tone_stream::// > with the > > > teletone spec for ring sound and it use ignore_early_media=true > in your > > > originates so the first caller would hear ringback until the 2nd > one > > > arrived. > > > > > > On Mon, Feb 2, 2009 at 4:29 AM, Brian West > <[1][2]brian at freeswitch.org> > > > wrote: > > > > > > Loopback will not work in that case either. If the far end > plays > > > ringback inband you should hear that if you use the conference > dial > > > api call. > > > /b > > > > > > On Feb 2, 2009, at 4:24 AM, Sias Mey wrote: > > > > Aaah ok. > > > > > > > > Thanks for clearing that up. > > > > > > > > So using loopback is still the only real workable sollution > for me, > > > > since that generates ringback from and alternative endpoint > and > > > > plays it > > > > into the conference. > > > > > > > > I might play with some javascript that streams ring into the > channel > > > > eventually but for now the string comparisons at least get me > the > > > > right > > > > uuid. > > > > > > > > Thank you again, > > > > Sias > > > > > > _______________________________________________ > > > Freeswitch-users mailing list > > > [2][3]Freeswitch-users at lists.freeswitch.org > > > > [3][4]http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > UNSUBSCRIBE:[4][5]http://lists.freeswitch.org/mailman/options/freeswitc > h-u > > > sers > > > [5][6]http://www.freeswitch.org > > > > > > -- > > > Anthony Minessale II > > > FreeSWITCH [6][7]http://www.freeswitch.org/ > > > ClueCon [7][8]http://www.cluecon.com/ > > > AIM: anthm > > > [8][9]MSN:anthony_minessale at hotmail.com > > > GTALK/JABBER/[9][10]PAYPAL:anthony.minessale at gmail.com > > > IRC: [10][11]irc.freenode.net #freeswitch > > > FreeSWITCH Developer Conference > > > [11][12]sip:888 at conference.freeswitch.org > > > [12][13]iax:guest at conference.freeswitch.org/888 > > > [13][14]googletalk:conf+888 at conference.freeswitch.org > > > pstn:213-799-1400 > > > > > > References > > > > > > 1. mailto:[15]brian at freeswitch.org > > > 2. mailto:[16]Freeswitch-users at lists.freeswitch.org > > > 3. > [17]http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > 4. > [18]http://lists.freeswitch.org/mailman/options/freeswitch-users > > > 5. [19]http://www.freeswitch.org/ > > > 6. [20]http://www.freeswitch.org/ > > > 7. [21]http://www.cluecon.com/ > > > 8. mailto:[22]MSN%3Aanthony_minessale at hotmail.com > > > 9. mailto:[23]PAYPAL%3Aanthony.minessale at gmail.com > > > 10. [24]http://irc.freenode.net/ > > > 11. mailto:[25]sip%3A888 at conference.freeswitch.org > > > 12. [26]http://iax:guest at conference.freeswitch.org/888 > > > 13. mailto:[27]googletalk%3Aconf%2B888 at conference.freeswitch.org > > > > > _______________________________________________ > > > Freeswitch-users mailing list > > > [28]Freeswitch-users at lists.freeswitch.org > > > [29]http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > UNSUBSCRIBE:[30]http://lists.freeswitch.org/mailman/options/freeswitch- > users > > > [31]http://www.freeswitch.org > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > [32]Freeswitch-users at lists.freeswitch.org > > [33]http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:[34]http://lists.freeswitch.org/mailman/options/freeswitch- > users > > [35]http://www.freeswitch.org > _______________________________________________ > Freeswitch-users mailing list > [36]Freeswitch-users at lists.freeswitch.org > [37]http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:[38]http://lists.freeswitch.org/mailman/options/freeswitch- > users > [39]http://www.freeswitch.org > > -- > Anthony Minessale II > FreeSWITCH [40]http://www.freeswitch.org/ > ClueCon [41]http://www.cluecon.com/ > AIM: anthm > [42]MSN:anthony_minessale at hotmail.com > GTALK/JABBER/[43]PAYPAL:anthony.minessale at gmail.com > IRC: [44]irc.freenode.net #freeswitch > FreeSWITCH Developer Conference > [45]sip:888 at conference.freeswitch.org > [46]iax:guest at conference.freeswitch.org/888 > [47]googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > References > > 1. mailto:sias at cpdata.co.za > 2. mailto:brian at freeswitch.org > 3. mailto:Freeswitch-users at lists.freeswitch.org > 4. http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > 5. http://lists.freeswitch.org/mailman/options/freeswitch-u > 6. http://www.freeswitch.org/ > 7. http://www.freeswitch.org/ > 8. http://www.cluecon.com/ > 9. mailto:MSN%3Aanthony_minessale at hotmail.com > 10. mailto:PAYPAL%3Aanthony.minessale at gmail.com > 11. http://irc.freenode.net/ > 12. mailto:sip%3A888 at conference.freeswitch.org > 13. http://iax:guest at conference.freeswitch.org/888 > 14. mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org > 15. mailto:brian at freeswitch.org > 16. mailto:Freeswitch-users at lists.freeswitch.org > 17. http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > 18. http://lists.freeswitch.org/mailman/options/freeswitch-users > 19. http://www.freeswitch.org/ > 20. http://www.freeswitch.org/ > 21. http://www.cluecon.com/ > 22. mailto:MSN%253Aanthony_minessale at hotmail.com > 23. mailto:PAYPAL%253Aanthony.minessale at gmail.com > 24. http://irc.freenode.net/ > 25. mailto:sip%253A888 at conference.freeswitch.org > 26. http://iax:guest at conference.freeswitch.org/888 > 27. mailto:googletalk%253Aconf%252B888 at conference.freeswitch.org > 28. mailto:Freeswitch-users at lists.freeswitch.org > 29. http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > 30. http://lists.freeswitch.org/mailman/options/freeswitch-users > 31. http://www.freeswitch.org/ > 32. mailto:Freeswitch-users at lists.freeswitch.org > 33. http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > 34. http://lists.freeswitch.org/mailman/options/freeswitch-users > 35. http://www.freeswitch.org/ > 36. mailto:Freeswitch-users at lists.freeswitch.org > 37. http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > 38. http://lists.freeswitch.org/mailman/options/freeswitch-users > 39. http://www.freeswitch.org/ > 40. http://www.freeswitch.org/ > 41. http://www.cluecon.com/ > 42. mailto:MSN%3Aanthony_minessale at hotmail.com > 43. mailto:PAYPAL%3Aanthony.minessale at gmail.com > 44. http://irc.freenode.net/ > 45. mailto:sip%3A888 at conference.freeswitch.org > 46. http://iax:guest at conference.freeswitch.org/888 > 47. mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From sias at cpdata.co.za Tue Feb 3 07:11:22 2009 From: sias at cpdata.co.za (Sias Mey) Date: Tue, 3 Feb 2009 17:11:22 +0200 Subject: [Freeswitch-users] Conference dialing and uuid In-Reply-To: <20090203141621.GA8916@cpdata.co.za> References: <191c3a030901300605r78dd8943w7fb1075ef851e1d@mail.gmail.com> <20090202085157.GA3555@cpdata.co.za> <0F863750-5C77-4CD9-ADC6-CFDDF2B149B4@freeswitch.org> <20090202102459.GA4179@cpdata.co.za> <9FE1733C-A46A-4A17-8AEE-077DE8010235@freeswitch.org> <191c3a030902020601v1109b1e9he867e9f1e1d4a401@mail.gmail.com> <20090203072221.GD16105@cpdata.co.za> <20090203082530.GA17166@cpdata.co.za> <191c3a030902030554y13e3dd7aj2eeaa43b86b1ced4@mail.gmail.com> <20090203141621.GA8916@cpdata.co.za> Message-ID: <20090203151122.GB8916@cpdata.co.za> hmmm ok indeed. small mods to js files to just play a loooong tone_stream full of ringy noises and then stop them in the on answer and I have what I wanted. Thank you very very much for all your help. On Tue, Feb 03, 2009 at 04:16:21PM +0200, Sias Mey wrote: > Hmm ok ... Ill try that In my head though the api call to originate > shouldent block? but I assume since it does my head is wrong. > > Thanks you for the explanation. I think you can put this one to bed now > :-P > > On Tue, Feb 03, 2009 at 07:54:29AM -0600, Anthony Minessale wrote: > > There is a file format called tone_stream that I was trying to explain > > yesterday. > > tone_stream:// > > or > > tone_stream://path=/path/to/text_file.ttml > > you can use this to play tones anywhere a filename is supposed to go. > > I guess loopback really is your only option if you must generate > > ringback. > > Typically, whatever gateway you are calling out over will go into early > > media and start playing the real ringback. > > You should not execute any apps during the on_ring_execute that block, > > (playing audio etc) > > Media has not even been established at that point and you have nobody > > to play the audio to anyway, > > But you will block from that point until the application you chose has > > ended so you should only execute small apps that > > return immediately such as setting a variable etc. > > As for ringback I think you have the whole thing reversed in your > > head. > > the ringback vars etc only apply to the origination (a) leg of a call. > > If you make an inbound call set the ringback variable and then call > > bridge, the ringback var is parsed on that inbound leg > > and the dialout process of the bridge app involves 2 channels the A leg > > and the B leg. When the B leg gets a ring indication and the A leg > > detects it, it will begin to play the ringback sound you chose back to > > the originator of that inbound leg. > > In the conference or using originate situation, you are doing an > > outbound call with no relevant inbound call, so there is nothing > > to generate ringback to. That's why loopback works because it cross > > connects an outbound call back to an inbound call which gives the > > bridge app everything it needs to be able to generate artificial > > ringback. > > > > On Tue, Feb 3, 2009 at 2:25 AM, Sias Mey <[1]sias at cpdata.co.za> wrote: > > > > Hmmm no MOH wont work... since I am planning on pulling more than > > just 2 > > members into the conference and I still need ringback for the later > > members as well. > > Is there a direct way for me to use conference play > > > > to play teletone directly? or should I just records some ringing if > > I > > want to use that? > > And lastly for my own sanity ;-) why would the following in a > > on_ring_execute stop execution of the call at that point? > > call = argv[1]; > > conf = argv[2]; > > consoleLog("info","Making ringback channel for uuid : "+ > > session.uuid > > +"\n"); > > var ringuuid = apiExecute("originate","loopback/ringback-conf="+ > > conf +"-conf &park()") > > //I tried with and without a exit() at the end > > It seems to stop media detection??(not really sure about the term) > > for the call that executes this > > script. > > Freeswitch doesent recognize the pickup of that call and thus it > > doesent > > get bridged into the conference. when I uuid_kill the call that gets > > originated everything else starts happening again. > > Oh Im running this in FS ver. 1.0.trunk (11226:11561M) > > and that loopback points to > > > > > expression="^ringback-conf=(.*)$"> > > > > > > > > and ringback.js is > > use("TeleTone"); > > session.answer(); > > var tts = new TeleTone(session); > > tts.addTone("u", 400.0, 450.0, 0.0); > > tts.addTone("r", 440.0, 480.0, 0.0); > > var RESET = "v=2000;>=0;+=0;"; > > var UK_RING = RESET + "L=2;u(400,200);u(400,2200)"; > > var US_RING = RESET + "r(2000,4000)"; > > while(session.ready()) { > > console_log("making UK ring\n"); > > for (x = 0 ; x < 2 ; x++) { > > tts.generate(UK_RING); > > } > > } > > A slight bastardisation of the teletone JS example. > > I would expected the new channel that is created via a api originate > > to > > be completely seperate from the JS I create it in. (thats why I use > > api > > instead of creating a new session, although I should probably try > > that > > as well). > > I use some CoreDB stuff to keep tabs on the uuid for the originated > > call > > so that I can uuid_kill it in the on_answer_script but as > > mentioned... > > the on_answer only executes after I uuid_kill the originated channel > > in > > the cli... > > Thanks again guys, > > Specially since it seems you two are always the ones that get back > > to > > me. > > > > On Tue, Feb 03, 2009 at 09:22:21AM +0200, Sias Mey wrote: > > > Actually loopback does work. > > > however as I said it generates a pair of extra channels. > > > > > > Hmmm I was trying to generate and extra call to a JS script that > > > generated a teletone ring in an on_ring_execute for the second call > > > however it seems to stop execution of the call itself. Event though I > > > use api commands to originate and then transfer it into the > > conference > > > so that I have direct access to its uuid. > > > > > > I think changeing the moh might be a bit simpler however and elimite > > > some CoreDB stuff I was doing to keep track of the calls ring > > generating > > > call (what a sentance). > > > > > > On Mon, Feb 02, 2009 at 08:01:25AM -0600, Anthony Minessale wrote: > > > > you could set the conference moh sound to be tone_stream::// > > with the > > > > teletone spec for ring sound and it use ignore_early_media=true > > in your > > > > originates so the first caller would hear ringback until the 2nd > > one > > > > arrived. > > > > > > > > On Mon, Feb 2, 2009 at 4:29 AM, Brian West > > <[1][2]brian at freeswitch.org> > > > > wrote: > > > > > > > > Loopback will not work in that case either. If the far end > > plays > > > > ringback inband you should hear that if you use the conference > > dial > > > > api call. > > > > /b > > > > > > > > On Feb 2, 2009, at 4:24 AM, Sias Mey wrote: > > > > > Aaah ok. > > > > > > > > > > Thanks for clearing that up. > > > > > > > > > > So using loopback is still the only real workable sollution > > for me, > > > > > since that generates ringback from and alternative endpoint > > and > > > > > plays it > > > > > into the conference. > > > > > > > > > > I might play with some javascript that streams ring into the > > channel > > > > > eventually but for now the string comparisons at least get me > > the > > > > > right > > > > > uuid. > > > > > > > > > > Thank you again, > > > > > Sias > > > > > > > > _______________________________________________ > > > > Freeswitch-users mailing list > > > > [2][3]Freeswitch-users at lists.freeswitch.org > > > > > > [3][4]http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > > UNSUBSCRIBE:[4][5]http://lists.freeswitch.org/mailman/options/freeswitc > > h-u > > > > sers > > > > [5][6]http://www.freeswitch.org > > > > > > > > -- > > > > Anthony Minessale II > > > > FreeSWITCH [6][7]http://www.freeswitch.org/ > > > > ClueCon [7][8]http://www.cluecon.com/ > > > > AIM: anthm > > > > [8][9]MSN:anthony_minessale at hotmail.com > > > > GTALK/JABBER/[9][10]PAYPAL:anthony.minessale at gmail.com > > > > IRC: [10][11]irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > > > [11][12]sip:888 at conference.freeswitch.org > > > > [12][13]iax:guest at conference.freeswitch.org/888 > > > > [13][14]googletalk:conf+888 at conference.freeswitch.org > > > > pstn:213-799-1400 > > > > > > > > References > > > > > > > > 1. mailto:[15]brian at freeswitch.org > > > > 2. mailto:[16]Freeswitch-users at lists.freeswitch.org > > > > 3. > > [17]http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > 4. > > [18]http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > 5. [19]http://www.freeswitch.org/ > > > > 6. [20]http://www.freeswitch.org/ > > > > 7. [21]http://www.cluecon.com/ > > > > 8. mailto:[22]MSN%3Aanthony_minessale at hotmail.com > > > > 9. mailto:[23]PAYPAL%3Aanthony.minessale at gmail.com > > > > 10. [24]http://irc.freenode.net/ > > > > 11. mailto:[25]sip%3A888 at conference.freeswitch.org > > > > 12. [26]http://iax:guest at conference.freeswitch.org/888 > > > > 13. mailto:[27]googletalk%3Aconf%2B888 at conference.freeswitch.org > > > > > > > _______________________________________________ > > > > Freeswitch-users mailing list > > > > [28]Freeswitch-users at lists.freeswitch.org > > > > [29]http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > > UNSUBSCRIBE:[30]http://lists.freeswitch.org/mailman/options/freeswitch- > > users > > > > [31]http://www.freeswitch.org > > > > > > > > > _______________________________________________ > > > Freeswitch-users mailing list > > > [32]Freeswitch-users at lists.freeswitch.org > > > [33]http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > UNSUBSCRIBE:[34]http://lists.freeswitch.org/mailman/options/freeswitch- > > users > > > [35]http://www.freeswitch.org > > _______________________________________________ > > Freeswitch-users mailing list > > [36]Freeswitch-users at lists.freeswitch.org > > [37]http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:[38]http://lists.freeswitch.org/mailman/options/freeswitch- > > users > > [39]http://www.freeswitch.org > > > > -- > > Anthony Minessale II > > FreeSWITCH [40]http://www.freeswitch.org/ > > ClueCon [41]http://www.cluecon.com/ > > AIM: anthm > > [42]MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/[43]PAYPAL:anthony.minessale at gmail.com > > IRC: [44]irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > > [45]sip:888 at conference.freeswitch.org > > [46]iax:guest at conference.freeswitch.org/888 > > [47]googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > > > > References > > > > 1. mailto:sias at cpdata.co.za > > 2. mailto:brian at freeswitch.org > > 3. mailto:Freeswitch-users at lists.freeswitch.org > > 4. http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > 5. http://lists.freeswitch.org/mailman/options/freeswitch-u > > 6. http://www.freeswitch.org/ > > 7. http://www.freeswitch.org/ > > 8. http://www.cluecon.com/ > > 9. mailto:MSN%3Aanthony_minessale at hotmail.com > > 10. mailto:PAYPAL%3Aanthony.minessale at gmail.com > > 11. http://irc.freenode.net/ > > 12. mailto:sip%3A888 at conference.freeswitch.org > > 13. http://iax:guest at conference.freeswitch.org/888 > > 14. mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org > > 15. mailto:brian at freeswitch.org > > 16. mailto:Freeswitch-users at lists.freeswitch.org > > 17. http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > 18. http://lists.freeswitch.org/mailman/options/freeswitch-users > > 19. http://www.freeswitch.org/ > > 20. http://www.freeswitch.org/ > > 21. http://www.cluecon.com/ > > 22. mailto:MSN%253Aanthony_minessale at hotmail.com > > 23. mailto:PAYPAL%253Aanthony.minessale at gmail.com > > 24. http://irc.freenode.net/ > > 25. mailto:sip%253A888 at conference.freeswitch.org > > 26. http://iax:guest at conference.freeswitch.org/888 > > 27. mailto:googletalk%253Aconf%252B888 at conference.freeswitch.org > > 28. mailto:Freeswitch-users at lists.freeswitch.org > > 29. http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > 30. http://lists.freeswitch.org/mailman/options/freeswitch-users > > 31. http://www.freeswitch.org/ > > 32. mailto:Freeswitch-users at lists.freeswitch.org > > 33. http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > 34. http://lists.freeswitch.org/mailman/options/freeswitch-users > > 35. http://www.freeswitch.org/ > > 36. mailto:Freeswitch-users at lists.freeswitch.org > > 37. http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > 38. http://lists.freeswitch.org/mailman/options/freeswitch-users > > 39. http://www.freeswitch.org/ > > 40. http://www.freeswitch.org/ > > 41. http://www.cluecon.com/ > > 42. mailto:MSN%3Aanthony_minessale at hotmail.com > > 43. mailto:PAYPAL%3Aanthony.minessale at gmail.com > > 44. http://irc.freenode.net/ > > 45. mailto:sip%3A888 at conference.freeswitch.org > > 46. http://iax:guest at conference.freeswitch.org/888 > > 47. mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From anthony.minessale at gmail.com Tue Feb 3 07:19:37 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 3 Feb 2009 09:19:37 -0600 Subject: [Freeswitch-users] fifo problem In-Reply-To: <49884BE1.2000603@gmail.com> References: <49884BE1.2000603@gmail.com> Message-ID: <191c3a030902030719l2d8eb824rc1472df6d0f8b779@mail.gmail.com> you could use the intercept app to unpark the caller without using fifo out, then it would only work if the caller existed. On Tue, Feb 3, 2009 at 7:51 AM, Tamas Cseke wrote: > Hello, > > We have a problem with mod_fifo. > > we monitor fifo push event on event socket, > call consumer with originate & fifo out nowait > Similar like fifo_outbound works, but we have an external strategy for > consumer selection (eg.: skill-based routing) > > The problem is when a caller stops waiting in fifo, the originated calls > kepp ringing the consumer, and when the consumer answer the call, > he or she may grab somebody else from the fifo, which is a problem > because the callers are identified and some data (eg name, phonenumber > is shown for the consumer). > so it can happen these data will be wrong. > > We tried to resolve this issue by a call tracking in the external script > using event socket. > we pushes a variable into the CHANNEL_ORIGINATE event calling the > consumer containing the caller uuid. > and if the caller aborts the fifo, we hangup the consumer call with > (uuid_kill) > But it's not prefect becasue it can happen that the consumer pop another > caller from the fifo. > and we hangup this call, so as a side-effect we loosing another caller. > > Could anybody advise a solution for this please? > we thinking about to have a fifo_caller_uuid variable, that we set > before calling fifo with the out method. > and if this uuid is in the top of the fifo then pop it else don't pop > anybody. > it seems to be a hack anyway.... > > Thanks in advance, > Tamas > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090203/a6993e45/attachment.html From jsokulski at dotsystems.pl Tue Feb 3 05:36:19 2009 From: jsokulski at dotsystems.pl (Jacek Sokulski) Date: Tue, 03 Feb 2009 14:36:19 +0100 Subject: [Freeswitch-users] origainate through sofia gateway Message-ID: <1233668179.5354.15.camel@dotw1126.dotsystems.pl> Hello I am trying to initiate a call from javascript, it works fine for local numbers: > session1.originate(session1, "{ignore_early_media=true}user/1008 at 192.168.1.122"); but when I am trying to connect through sofia gateway, the connection is not being established: > session2.originate(session2, "sofia/gateway/halonet/0225490317"); although I can call to this number from softphone. I have also tried setting effective_caller_id_number: > session1.originate(session1, "{effective_caller_id_number=fixed0248b}sofia/gateway/halonet/0225490317"); with the same result. A configuration in the dialplan that works is: > > > > > > > > Would appreciate any help. Jacek From nik.middleton at noblesolutions.co.uk Tue Feb 3 08:20:30 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Tue, 3 Feb 2009 16:20:30 -0000 Subject: [Freeswitch-users] OPenser <-> FS Do I need this? In-Reply-To: <1233668029.24619.29.camel@stargate> References: <2cbf225c0902021821u2bc8b622ka5d9f94f05a968ae@mail.gmail.com> <1233668029.24619.29.camel@stargate> Message-ID: Newbie with FS, currently have Asterisk servers front ended by Openser Question: I have around 400 sip remote clients, if I were to deploy FS, do I need Openser? Is there any advantage in retaining Openser? Regards From msc at freeswitch.org Tue Feb 3 08:41:59 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 3 Feb 2009 08:41:59 -0800 Subject: [Freeswitch-users] Application language to support C or C++? In-Reply-To: <1233668029.24619.29.camel@stargate> References: <2cbf225c0902021821u2bc8b622ka5d9f94f05a968ae@mail.gmail.com> <1233668029.24619.29.camel@stargate> Message-ID: <87f2f3b90902030841i54f2064djd13c0fc82fc71ca1@mail.gmail.com> Lee, You also might want to take a look at some of the examples in the contrib folder in the source tree. There are several items there that use the event socket. The event socket is extremely powerful and is suitable for a wide range of applications. However, it isn't the only way to do things. You could also build an actual FreeSWITCH application like the ones found in the "mod" directory. That's a bit more involved and I don't recommend starting there unless you're C/C++ skills are well established. :) What is your application? Most likely others here have done something similar and can share with you their experiences, including what worked and what didn't work. -MC On Tue, Feb 3, 2009 at 5:33 AM, Raul Fragoso wrote: > Depending on what you want to do, I suggest having a look at > mod_event_socket: http://wiki.freeswitch.org/wiki/Mod_event_socket > That module is a socket based interface that provides a vast range of > options to control FreeSWITCH and its applications. > Just for the record, my application is entirely written in C++ and uses > FreeSWITCH as a back-end for providing PBX functionality through a > combination of mod_event_socket and mod_xml_curl. > > Regards, > > Raul > > On Tue, 2009-02-03 at 10:21 +0800, lee jason wrote: >> Hi All, >> >> I saw the applications using FreeSwitch library can be written >> in JavaScript, Perl, Python and Lua but I need to use Linux C or C++ >> for applications, Is FreeSwitch can supported it? Where can I get the >> sample codes? My Linux platform is base on Fedora. >> >> >> >> >> Thanks a lot. >> >> >> Jason >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From msc at freeswitch.org Tue Feb 3 08:52:05 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 3 Feb 2009 08:52:05 -0800 Subject: [Freeswitch-users] Generating calls from external source In-Reply-To: <191c3a030902030609i5c3f415fhed501aa492d29cd3@mail.gmail.com> References: <40B58CBF-9D0F-43A3-B36F-BD05916756E4@3c.co.uk> <1233666748.24619.8.camel@stargate> <191c3a030902030609i5c3f415fhed501aa492d29cd3@mail.gmail.com> Message-ID: <87f2f3b90902030852h6b6477e9u9025ca278ab9122f@mail.gmail.com> On Tue, Feb 3, 2009 at 6:09 AM, Anthony Minessale wrote: > There is also an event socket library written in C called esl that is in the > fs tree in the libs directory. > This has the ability to establish connections both inbound and outbound from > FS. > > There is also a perl module FreeSWITCH::Client that mr collins may be > interested in in the tree as well. As a matter of fact that is the module I used for my outbound IVR application. It simply handled the communications between my perl script and my FS instance. The script would read in pre-formatted originate strings from a text file that had been previously generated by another application. Then all I had to do was specify how many concurrent channels that I wanted - kind of like a throttle - and then I let the script go. I used the "bgapi originate" syntax so that I wouldn't have to wait to see what happened with each origination attempt. Then about every second or so I would issue an "oz dump 1" and parse the results to count how many b channels were in use. If the number of b channels in use was >= my throttle limit then I'd pause the script for 1000ms and then issue the oz dump again until the number of b channels in use had dropped down below my limit. Nothing too fancy. You're welcome to review my script, originate syntax, and dialplan entries if you are interested. -MC > > > On Tue, Feb 3, 2009 at 7:12 AM, Raul Fragoso wrote: >> >> In addition do David's suggestion, you probably want to have your >> application to watch for some specific events after the call is >> originated and take action based on them. For example, you could watch >> for the CHANNEL_ANSWER event and play some audio file waiting for some >> digit, which is generated by the DTMF event. >> To watch only for those specific events, you should do the following >> just after authentication (still using Perl as an example, but the >> mod_event_socket is language agnostic), then you will receive those >> events from FreeSWITCH through the socket stream: >> >> ... >> print $sock "auth XXX\n\n"; >> print $sock "event plain CHANNEL_ANSWER DTMF\n\n"; >> ... >> >> To see a list of available events, please look at the following wiki >> pages: >> http://wiki.freeswitch.org/wiki/Mod_event_socket#event >> http://wiki.freeswitch.org/wiki/Event_list >> >> Regards, >> >> Raul >> >> On Tue, 2009-02-03 at 09:46 +0000, David Knell wrote: >> > Hi Nik, >> > >> > >> > Here's a snipped in Perl that launches an outbound call: >> > >> > >> > if (my $sock = IO::Socket::INET->new(Proto =>'tcp', PeerAddr => >> > '127.0.0.1', PeerPort => 8021)) { >> > print $sock "auth XXX\n\n"; >> > print $sock "api originate {softivr_id=$siid,src_softivr_id= >> > $siid,softivr_outdial=true}sofia/frombt/$ntd at 1.2.3.4 $service\n\n"; >> > $sock->close(); >> > } >> > >> > >> > - it does no error checking or anything, but (line by line) it: >> > - opens a socket to the event socket interface >> > - authenticates >> > - issues an originate which dials out to the number in $ntd. The >> > bits in {} set a bunch of variables on the channel, which are used by >> > the software which processes the call later on. The call is linked to >> > the extension in $service - FS looks this up in the dialplan - which >> > handles our end. >> > - closes the socket >> > >> > >> > Cheers -- >> > >> > >> > Dave >> > >> > >> > >> > > Thanks for that, coming from a C++ background it's a refreshing >> > > change to be looking at something that seems logical and efficient. >> > > >> > > I'd briefly looked at the event socket and wondered if that was the >> > > way to go. I presume that there's some sort of event generation >> > > that can trigger and external process as well somewhere, though all >> > > I need to do is update mysql (hopefully using some sort of pooled >> > > connection) >> > > >> > > I'm not using a TDM card, I have a direct interconnect with the PSTN >> > > breakout provider with 1,500 channels available to me. I'm finding >> > > Asterisk proving to be less than stable at high call volumes and >> > > load values spike at more than 100 calls with billing/accounting in >> > > place, hence my interest in FS. The only thing that's concerning me >> > > is XML at the moment. Lots of code and very wordy. I'm sure I'll >> > > appreciate why XML given time >> > > >> > > Regards, >> > > >> > > >> > > ____________________________________________________________________ >> > > From: freeswitch-users-bounces at lists.freeswitch.org >> > > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael >> > > S Collins >> > > Sent: 03 February 2009 01:17 >> > > To: freeswitch-users at lists.freeswitch.org >> > > Subject: Re: [Freeswitch-users] Generating calls from external >> > > source >> > > >> > > Nik, >> > > >> > > Welcome to FreeSWITCH! The short answer is "yes, FS can do that." >> > > The first thing that you should do is unlearn "the Asterisk way" of >> > > thinking. Usually there is an elegant way of doing things in FS that >> > > wasn't possible in Ast. >> > > >> > > I would recommend that you start by looking at the event socket, >> > > which is somewhat analogous to the AMI only cooler. :) I have >> > > personally done something similar to this using the event socket and >> > > a Perl script. The key is to learn the syntax of the originate >> > > command. (definitely hit the wiki and IRC channel) >> > > Are you using TDM cards for this? Just curious. >> > > >> > > -MC (IRC nick: mercutioviz) >> > > >> > > Sent from my iPhone >> > > >> > > On Feb 2, 2009, at 3:35 PM, "Nik Middleton" >> > > wrote: >> > > > Hi Guys, >> > > > >> > > > As a long time Asterisk user, I'm looking into freeswitch as an >> > > > alternative mainly due to (list multiple reasons here) >> > > > >> > > > Can anyone give me a pointer as to how I would achieve the >> > > > following? >> > > > >> > > > I need to replicate an emergency broadcast system currently >> > > > running under Asterisk. >> > > > >> > > > At the moment, I run through a Mysql database and using the >> > > > manager API, issues an Originate command to dial a number. >> > > > >> > > > When the call is answered, a message is played, and the recipient >> > > > has the option of hitting a digit to confirm receipt. I then call >> > > > an AGI script to update the database. >> > > > >> > > > Is this fairly easy to do in Freeswitch? >> > > > >> > > > Not looking for code, just some pointers as to what's available to >> > > > do the above / >> > > > >> > > > Regards, >> > > > _______________________________________________ >> > > > Freeswitch-users mailing list >> > > > Freeswitch-users at lists.freeswitch.org >> > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > > > >> > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > > > http://www.freeswitch.org >> > > _______________________________________________ >> > > Freeswitch-users mailing list >> > > Freeswitch-users at lists.freeswitch.org >> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > > >> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > > http://www.freeswitch.org >> > > >> > >> > >> > _______________________________________________ >> > Freeswitch-users mailing list >> > Freeswitch-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From nik.middleton at noblesolutions.co.uk Tue Feb 3 08:53:20 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Tue, 3 Feb 2009 16:53:20 -0000 Subject: [Freeswitch-users] Generating calls from external source In-Reply-To: <1233666748.24619.8.camel@stargate> References: <40B58CBF-9D0F-43A3-B36F-BD05916756E4@3c.co.uk> <1233666748.24619.8.camel@stargate> Message-ID: Are you suggesting that I should process the call externally instead of using the dialplan? That would be neat as the audio file select could be driven from the db select for the number. I presume that I could also bridge the call to another number as well dependant on DTMF selection? Regards -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Raul Fragoso Sent: 03 February 2009 13:12 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Generating calls from external source In addition do David's suggestion, you probably want to have your application to watch for some specific events after the call is originated and take action based on them. For example, you could watch for the CHANNEL_ANSWER event and play some audio file waiting for some digit, which is generated by the DTMF event. To watch only for those specific events, you should do the following just after authentication (still using Perl as an example, but the mod_event_socket is language agnostic), then you will receive those events from FreeSWITCH through the socket stream: ... print $sock "auth XXX\n\n"; print $sock "event plain CHANNEL_ANSWER DTMF\n\n"; ... To see a list of available events, please look at the following wiki pages: http://wiki.freeswitch.org/wiki/Mod_event_socket#event http://wiki.freeswitch.org/wiki/Event_list Regards, Raul On Tue, 2009-02-03 at 09:46 +0000, David Knell wrote: > Hi Nik, > > > Here's a snipped in Perl that launches an outbound call: > > > if (my $sock = IO::Socket::INET->new(Proto =>'tcp', PeerAddr => > '127.0.0.1', PeerPort => 8021)) { > print $sock "auth XXX\n\n"; > print $sock "api originate {softivr_id=$siid,src_softivr_id= > $siid,softivr_outdial=true}sofia/frombt/$ntd at 1.2.3.4 $service\n\n"; > $sock->close(); > } > > > - it does no error checking or anything, but (line by line) it: > - opens a socket to the event socket interface > - authenticates > - issues an originate which dials out to the number in $ntd. The > bits in {} set a bunch of variables on the channel, which are used by > the software which processes the call later on. The call is linked to > the extension in $service - FS looks this up in the dialplan - which > handles our end. > - closes the socket > > > Cheers -- > > > Dave > > > > > Thanks for that, coming from a C++ background it's a refreshing > > change to be looking at something that seems logical and efficient. > > > > I'd briefly looked at the event socket and wondered if that was the > > way to go. I presume that there's some sort of event generation > > that can trigger and external process as well somewhere, though all > > I need to do is update mysql (hopefully using some sort of pooled > > connection) > > > > I'm not using a TDM card, I have a direct interconnect with the PSTN > > breakout provider with 1,500 channels available to me. I'm finding > > Asterisk proving to be less than stable at high call volumes and > > load values spike at more than 100 calls with billing/accounting in > > place, hence my interest in FS. The only thing that's concerning me > > is XML at the moment. Lots of code and very wordy. I'm sure I'll > > appreciate why XML given time > > > > Regards, > > > > > > ____________________________________________________________________ > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael S Collins > > Sent: 03 February 2009 01:17 > > To: freeswitch-users at lists.freeswitch.org > > Subject: Re: [Freeswitch-users] Generating calls from external > > source > > > > Nik, > > > > Welcome to FreeSWITCH! The short answer is "yes, FS can do that." > > The first thing that you should do is unlearn "the Asterisk way" of > > thinking. Usually there is an elegant way of doing things in FS that > > wasn't possible in Ast. > > > > I would recommend that you start by looking at the event socket, > > which is somewhat analogous to the AMI only cooler. :) I have > > personally done something similar to this using the event socket and > > a Perl script. The key is to learn the syntax of the originate > > command. (definitely hit the wiki and IRC channel) > > Are you using TDM cards for this? Just curious. > > > > -MC (IRC nick: mercutioviz) > > > > Sent from my iPhone > > > > On Feb 2, 2009, at 3:35 PM, "Nik Middleton" > > wrote: > > > Hi Guys, > > > > > > As a long time Asterisk user, I'm looking into freeswitch as an > > > alternative mainly due to (list multiple reasons here) > > > > > > Can anyone give me a pointer as to how I would achieve the > > > following? > > > > > > I need to replicate an emergency broadcast system currently > > > running under Asterisk. > > > > > > At the moment, I run through a Mysql database and using the > > > manager API, issues an Originate command to dial a number. > > > > > > When the call is answered, a message is played, and the recipient > > > has the option of hitting a digit to confirm receipt. I then call > > > an AGI script to update the database. > > > > > > Is this fairly easy to do in Freeswitch? > > > > > > Not looking for code, just some pointers as to what's available to > > > do the above / > > > > > > Regards, > > > _______________________________________________ > > > Freeswitch-users mailing list > > > Freeswitch-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From mike at jerris.com Tue Feb 3 09:02:38 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 3 Feb 2009 12:02:38 -0500 Subject: [Freeswitch-users] debuild breaks since the last few days In-Reply-To: <498840A3.30509@ewetel.de> References: <0B3BFADA-4589-48AD-9F18-EEBDC6BAB740@scarlet-internet.nl> <498840A3.30509@ewetel.de> Message-ID: <707A7340-2C7A-47E9-9F69-8CC514785D8D@jerris.com> I just finished adding full libpcap detection. Openzap will now build again with or without libpcap, of course the pcap features will not work without. Mike On Feb 3, 2009, at 8:03 AM, Helmut Kuper wrote: > Hello, > > yes, you have openzap upgraded to r632. Then recompile it. Make sure > you > have libpcap installed and pcap devel files > > regards > helmut > > > Am 03.02.2009 12:55, schrieb Leon de Rooij: >> Hi all, >> >> I've been trying to build new debs, but debuild seems to break.. >> From msc at freeswitch.org Tue Feb 3 09:05:57 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 3 Feb 2009 09:05:57 -0800 Subject: [Freeswitch-users] Conference dialing and uuid In-Reply-To: <20090203141621.GA8916@cpdata.co.za> References: <20090130133315.GB23536@cpdata.co.za> <20090202085157.GA3555@cpdata.co.za> <0F863750-5C77-4CD9-ADC6-CFDDF2B149B4@freeswitch.org> <20090202102459.GA4179@cpdata.co.za> <9FE1733C-A46A-4A17-8AEE-077DE8010235@freeswitch.org> <191c3a030902020601v1109b1e9he867e9f1e1d4a401@mail.gmail.com> <20090203072221.GD16105@cpdata.co.za> <20090203082530.GA17166@cpdata.co.za> <191c3a030902030554y13e3dd7aj2eeaa43b86b1ced4@mail.gmail.com> <20090203141621.GA8916@cpdata.co.za> Message-ID: <87f2f3b90902030905y348efa0dq90c0080e5792e63f@mail.gmail.com> On Tue, Feb 3, 2009 at 6:16 AM, Sias Mey wrote: > Hmm ok ... Ill try that In my head though the api call to originate > shouldent block? but I assume since it does my head is wrong. You can use "bgapi originate" to do it in a non-blocking way. See the very last example for the originate command: http://wiki.freeswitch.org/wiki/Mod_commands#originate -MC > > Thanks you for the explanation. I think you can put this one to bed now > :-P > > On Tue, Feb 03, 2009 at 07:54:29AM -0600, Anthony Minessale wrote: >> There is a file format called tone_stream that I was trying to explain >> yesterday. >> tone_stream:// >> or >> tone_stream://path=/path/to/text_file.ttml >> you can use this to play tones anywhere a filename is supposed to go. >> I guess loopback really is your only option if you must generate >> ringback. >> Typically, whatever gateway you are calling out over will go into early >> media and start playing the real ringback. >> You should not execute any apps during the on_ring_execute that block, >> (playing audio etc) >> Media has not even been established at that point and you have nobody >> to play the audio to anyway, >> But you will block from that point until the application you chose has >> ended so you should only execute small apps that >> return immediately such as setting a variable etc. >> As for ringback I think you have the whole thing reversed in your >> head. >> the ringback vars etc only apply to the origination (a) leg of a call. >> If you make an inbound call set the ringback variable and then call >> bridge, the ringback var is parsed on that inbound leg >> and the dialout process of the bridge app involves 2 channels the A leg >> and the B leg. When the B leg gets a ring indication and the A leg >> detects it, it will begin to play the ringback sound you chose back to >> the originator of that inbound leg. >> In the conference or using originate situation, you are doing an >> outbound call with no relevant inbound call, so there is nothing >> to generate ringback to. That's why loopback works because it cross >> connects an outbound call back to an inbound call which gives the >> bridge app everything it needs to be able to generate artificial >> ringback. >> >> On Tue, Feb 3, 2009 at 2:25 AM, Sias Mey <[1]sias at cpdata.co.za> wrote: >> >> Hmmm no MOH wont work... since I am planning on pulling more than >> just 2 >> members into the conference and I still need ringback for the later >> members as well. >> Is there a direct way for me to use conference play >> >> to play teletone directly? or should I just records some ringing if >> I >> want to use that? >> And lastly for my own sanity ;-) why would the following in a >> on_ring_execute stop execution of the call at that point? >> call = argv[1]; >> conf = argv[2]; >> consoleLog("info","Making ringback channel for uuid : "+ >> session.uuid >> +"\n"); >> var ringuuid = apiExecute("originate","loopback/ringback-conf="+ >> conf +"-conf &park()") >> //I tried with and without a exit() at the end >> It seems to stop media detection??(not really sure about the term) >> for the call that executes this >> script. >> Freeswitch doesent recognize the pickup of that call and thus it >> doesent >> get bridged into the conference. when I uuid_kill the call that gets >> originated everything else starts happening again. >> Oh Im running this in FS ver. 1.0.trunk (11226:11561M) >> and that loopback points to >> >> > expression="^ringback-conf=(.*)$"> >> >> >> >> and ringback.js is >> use("TeleTone"); >> session.answer(); >> var tts = new TeleTone(session); >> tts.addTone("u", 400.0, 450.0, 0.0); >> tts.addTone("r", 440.0, 480.0, 0.0); >> var RESET = "v=2000;>=0;+=0;"; >> var UK_RING = RESET + "L=2;u(400,200);u(400,2200)"; >> var US_RING = RESET + "r(2000,4000)"; >> while(session.ready()) { >> console_log("making UK ring\n"); >> for (x = 0 ; x < 2 ; x++) { >> tts.generate(UK_RING); >> } >> } >> A slight bastardisation of the teletone JS example. >> I would expected the new channel that is created via a api originate >> to >> be completely seperate from the JS I create it in. (thats why I use >> api >> instead of creating a new session, although I should probably try >> that >> as well). >> I use some CoreDB stuff to keep tabs on the uuid for the originated >> call >> so that I can uuid_kill it in the on_answer_script but as >> mentioned... >> the on_answer only executes after I uuid_kill the originated channel >> in >> the cli... >> Thanks again guys, >> Specially since it seems you two are always the ones that get back >> to >> me. >> >> On Tue, Feb 03, 2009 at 09:22:21AM +0200, Sias Mey wrote: >> > Actually loopback does work. >> > however as I said it generates a pair of extra channels. >> > >> > Hmmm I was trying to generate and extra call to a JS script that >> > generated a teletone ring in an on_ring_execute for the second call >> > however it seems to stop execution of the call itself. Event though I >> > use api commands to originate and then transfer it into the >> conference >> > so that I have direct access to its uuid. >> > >> > I think changeing the moh might be a bit simpler however and elimite >> > some CoreDB stuff I was doing to keep track of the calls ring >> generating >> > call (what a sentance). >> > >> > On Mon, Feb 02, 2009 at 08:01:25AM -0600, Anthony Minessale wrote: >> > > you could set the conference moh sound to be tone_stream::// >> with the >> > > teletone spec for ring sound and it use ignore_early_media=true >> in your >> > > originates so the first caller would hear ringback until the 2nd >> one >> > > arrived. >> > > >> > > On Mon, Feb 2, 2009 at 4:29 AM, Brian West >> <[1][2]brian at freeswitch.org> >> > > wrote: >> > > >> > > Loopback will not work in that case either. If the far end >> plays >> > > ringback inband you should hear that if you use the conference >> dial >> > > api call. >> > > /b >> > > >> > > On Feb 2, 2009, at 4:24 AM, Sias Mey wrote: >> > > > Aaah ok. >> > > > >> > > > Thanks for clearing that up. >> > > > >> > > > So using loopback is still the only real workable sollution >> for me, >> > > > since that generates ringback from and alternative endpoint >> and >> > > > plays it >> > > > into the conference. >> > > > >> > > > I might play with some javascript that streams ring into the >> channel >> > > > eventually but for now the string comparisons at least get me >> the >> > > > right >> > > > uuid. >> > > > >> > > > Thank you again, >> > > > Sias >> > > >> > > _______________________________________________ >> > > Freeswitch-users mailing list >> > > [2][3]Freeswitch-users at lists.freeswitch.org >> > > >> [3][4]http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > > >> UNSUBSCRIBE:[4][5]http://lists.freeswitch.org/mailman/options/freeswitc >> h-u >> > > sers >> > > [5][6]http://www.freeswitch.org >> > > >> > > -- >> > > Anthony Minessale II >> > > FreeSWITCH [6][7]http://www.freeswitch.org/ >> > > ClueCon [7][8]http://www.cluecon.com/ >> > > AIM: anthm >> > > [8][9]MSN:anthony_minessale at hotmail.com >> > > GTALK/JABBER/[9][10]PAYPAL:anthony.minessale at gmail.com >> > > IRC: [10][11]irc.freenode.net #freeswitch >> > > FreeSWITCH Developer Conference >> > > [11][12]sip:888 at conference.freeswitch.org >> > > [12][13]iax:guest at conference.freeswitch.org/888 >> > > [13][14]googletalk:conf+888 at conference.freeswitch.org >> > > pstn:213-799-1400 >> > > >> > > References >> > > >> > > 1. mailto:[15]brian at freeswitch.org >> > > 2. mailto:[16]Freeswitch-users at lists.freeswitch.org >> > > 3. >> [17]http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > > 4. >> [18]http://lists.freeswitch.org/mailman/options/freeswitch-users >> > > 5. [19]http://www.freeswitch.org/ >> > > 6. [20]http://www.freeswitch.org/ >> > > 7. [21]http://www.cluecon.com/ >> > > 8. mailto:[22]MSN%3Aanthony_minessale at hotmail.com >> > > 9. mailto:[23]PAYPAL%3Aanthony.minessale at gmail.com >> > > 10. [24]http://irc.freenode.net/ >> > > 11. mailto:[25]sip%3A888 at conference.freeswitch.org >> > > 12. [26]http://iax:guest at conference.freeswitch.org/888 >> > > 13. mailto:[27]googletalk%3Aconf%2B888 at conference.freeswitch.org >> > >> > > _______________________________________________ >> > > Freeswitch-users mailing list >> > > [28]Freeswitch-users at lists.freeswitch.org >> > > [29]http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > > >> UNSUBSCRIBE:[30]http://lists.freeswitch.org/mailman/options/freeswitch- >> users >> > > [31]http://www.freeswitch.org >> > >> > >> > _______________________________________________ >> > Freeswitch-users mailing list >> > [32]Freeswitch-users at lists.freeswitch.org >> > [33]http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > >> UNSUBSCRIBE:[34]http://lists.freeswitch.org/mailman/options/freeswitch- >> users >> > [35]http://www.freeswitch.org >> _______________________________________________ >> Freeswitch-users mailing list >> [36]Freeswitch-users at lists.freeswitch.org >> [37]http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:[38]http://lists.freeswitch.org/mailman/options/freeswitch- >> users >> [39]http://www.freeswitch.org >> >> -- >> Anthony Minessale II >> FreeSWITCH [40]http://www.freeswitch.org/ >> ClueCon [41]http://www.cluecon.com/ >> AIM: anthm >> [42]MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/[43]PAYPAL:anthony.minessale at gmail.com >> IRC: [44]irc.freenode.net #freeswitch >> FreeSWITCH Developer Conference >> [45]sip:888 at conference.freeswitch.org >> [46]iax:guest at conference.freeswitch.org/888 >> [47]googletalk:conf+888 at conference.freeswitch.org >> pstn:213-799-1400 >> >> References >> >> 1. mailto:sias at cpdata.co.za >> 2. mailto:brian at freeswitch.org >> 3. mailto:Freeswitch-users at lists.freeswitch.org >> 4. http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> 5. http://lists.freeswitch.org/mailman/options/freeswitch-u >> 6. http://www.freeswitch.org/ >> 7. http://www.freeswitch.org/ >> 8. http://www.cluecon.com/ >> 9. mailto:MSN%3Aanthony_minessale at hotmail.com >> 10. mailto:PAYPAL%3Aanthony.minessale at gmail.com >> 11. http://irc.freenode.net/ >> 12. mailto:sip%3A888 at conference.freeswitch.org >> 13. http://iax:guest at conference.freeswitch.org/888 >> 14. mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org >> 15. mailto:brian at freeswitch.org >> 16. mailto:Freeswitch-users at lists.freeswitch.org >> 17. http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> 18. http://lists.freeswitch.org/mailman/options/freeswitch-users >> 19. http://www.freeswitch.org/ >> 20. http://www.freeswitch.org/ >> 21. http://www.cluecon.com/ >> 22. mailto:MSN%253Aanthony_minessale at hotmail.com >> 23. mailto:PAYPAL%253Aanthony.minessale at gmail.com >> 24. http://irc.freenode.net/ >> 25. mailto:sip%253A888 at conference.freeswitch.org >> 26. http://iax:guest at conference.freeswitch.org/888 >> 27. mailto:googletalk%253Aconf%252B888 at conference.freeswitch.org >> 28. mailto:Freeswitch-users at lists.freeswitch.org >> 29. http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> 30. http://lists.freeswitch.org/mailman/options/freeswitch-users >> 31. http://www.freeswitch.org/ >> 32. mailto:Freeswitch-users at lists.freeswitch.org >> 33. http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> 34. http://lists.freeswitch.org/mailman/options/freeswitch-users >> 35. http://www.freeswitch.org/ >> 36. mailto:Freeswitch-users at lists.freeswitch.org >> 37. http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> 38. http://lists.freeswitch.org/mailman/options/freeswitch-users >> 39. http://www.freeswitch.org/ >> 40. http://www.freeswitch.org/ >> 41. http://www.cluecon.com/ >> 42. mailto:MSN%3Aanthony_minessale at hotmail.com >> 43. mailto:PAYPAL%3Aanthony.minessale at gmail.com >> 44. http://irc.freenode.net/ >> 45. mailto:sip%3A888 at conference.freeswitch.org >> 46. http://iax:guest at conference.freeswitch.org/888 >> 47. mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org > >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From msc at freeswitch.org Tue Feb 3 09:08:19 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 3 Feb 2009 09:08:19 -0800 Subject: [Freeswitch-users] OPenser <-> FS Do I need this? In-Reply-To: References: <2cbf225c0902021821u2bc8b622ka5d9f94f05a968ae@mail.gmail.com> <1233668029.24619.29.camel@stargate> Message-ID: <87f2f3b90902030908n629399bdmdd46f633b803b5f6@mail.gmail.com> On Tue, Feb 3, 2009 at 8:20 AM, Nik Middleton wrote: > Newbie with FS, currently have Asterisk servers front ended by Openser > > Question: I have around 400 sip remote clients, if I were to deploy FS, > do I need Openser? Is there any advantage in retaining Openser? If I may ask... why did you have OpenSER with your Asterisk deployment? Reason I ask is because some people do that "because Asterisk sucks" but others have a specific application or reason. What does OpenSER do for your Asterisk install? -MC > > Regards > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From helmut.kuper at ewetel.de Tue Feb 3 09:09:06 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Tue, 03 Feb 2009 18:09:06 +0100 Subject: [Freeswitch-users] mod_voicemail: Limiting the number how often a menu is repeated In-Reply-To: <498734D0.5060004@ewetel.de> References: <498734D0.5060004@ewetel.de> Message-ID: <49887A32.7060804@ewetel.de> Hi, has anybody an idea? regards Helmut Am 02.02.2009 19:00, schrieb Helmut Kuper: > Hello, > > today I searched for a way to limit the number of menu repeatings in > mod_voicemail to let's say 3 times and when it reached the limit > voicemail should abort. But I couldn't find a hint. Any ideas? > > > regards > helmut > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > -- Mit freundlichen Gr??en Helmut Kuper Finanzdienstleistungen und Entwicklung Telefax: (0441) 8000-2799 mailto:helmut.kuper at ewetel.de ___________________________________ EWE TEL GmbH Cloppenburger Stra?e 310 26133 Oldenburg EWE TEL GmbH Handelsregister Amtsgericht Oldenburg HRB 3723 Vorsitzender des Aufsichtsrates: Heiko Harms Gesch?ftsf?hrung: Hans-Joachim Iken (Vorsitzender), Dr. Norbert Schulz, Dirk Thole Homepage: http://www.ewetel.de ___________________________________ From sicfslist at gmail.com Tue Feb 3 09:13:19 2009 From: sicfslist at gmail.com (Shelby Ramsey) Date: Tue, 3 Feb 2009 11:13:19 -0600 Subject: [Freeswitch-users] Generating calls from external source In-Reply-To: References: <40B58CBF-9D0F-43A3-B36F-BD05916756E4@3c.co.uk> <1233666748.24619.8.camel@stargate> Message-ID: <35b355e90902030913s1cec764j46280ac2d9a162a7@mail.gmail.com> Nik, There are a lot of ways to make FS dial out and deliver messaging etc. We are going through the process of replacing * for this purpose. For us (getting started with the help of our friends here on the list) it has been pretty easy. With * we were using AMI to originate calls ... to migrate to FS we just changed that to use event_socket with bgapi to originate the call and connect the call to a context and extension. There are several ways to get the dialplan to FS after that ... a script, xml_curl, or statically configured in the conf directory. So as an example the application we have just logs into the FS socket (similar to * but much better) and then rips off calls like this: bgapi originate{$set_some_vars}sofia/external/$ANI@$IP:$PORT $EXTENSION xml $CONTEXT The beauty of it all is that: -- a lot of flexibility in what you can do (like drive the call through events) -- the CDR reporting is about 3 million times better than * -- obviously higher capacity I'd start playing with event_socket and some static dialplans to get the feel for it ... but if you have an application written already to work with * (i.e. the logic and backend) it will be very easy to migrate and you'll be glad you did it! Shelby On Tue, Feb 3, 2009 at 10:53 AM, Nik Middleton < nik.middleton at noblesolutions.co.uk> wrote: > Are you suggesting that I should process the call externally instead of > using the dialplan? That would be neat as the audio file select could > be driven from the db select for the number. I presume that I could > also bridge the call to another number as well dependant on DTMF > selection? > > Regards > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Raul > Fragoso > Sent: 03 February 2009 13:12 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Generating calls from external source > > In addition do David's suggestion, you probably want to have your > application to watch for some specific events after the call is > originated and take action based on them. For example, you could watch > for the CHANNEL_ANSWER event and play some audio file waiting for some > digit, which is generated by the DTMF event. > To watch only for those specific events, you should do the following > just after authentication (still using Perl as an example, but the > mod_event_socket is language agnostic), then you will receive those > events from FreeSWITCH through the socket stream: > > ... > print $sock "auth XXX\n\n"; > print $sock "event plain CHANNEL_ANSWER DTMF\n\n"; > ... > > To see a list of available events, please look at the following wiki > pages: > http://wiki.freeswitch.org/wiki/Mod_event_socket#event > http://wiki.freeswitch.org/wiki/Event_list > > Regards, > > Raul > > On Tue, 2009-02-03 at 09:46 +0000, David Knell wrote: > > Hi Nik, > > > > > > Here's a snipped in Perl that launches an outbound call: > > > > > > if (my $sock = IO::Socket::INET->new(Proto =>'tcp', PeerAddr => > > '127.0.0.1', PeerPort => 8021)) { > > print $sock "auth XXX\n\n"; > > print $sock "api originate {softivr_id=$siid,src_softivr_id= > > $siid,softivr_outdial=true}sofia/frombt/$ntd at 1.2.3.4 $service\n\n"; > > $sock->close(); > > } > > > > > > - it does no error checking or anything, but (line by line) it: > > - opens a socket to the event socket interface > > - authenticates > > - issues an originate which dials out to the number in $ntd. The > > bits in {} set a bunch of variables on the channel, which are used by > > the software which processes the call later on. The call is linked to > > the extension in $service - FS looks this up in the dialplan - which > > handles our end. > > - closes the socket > > > > > > Cheers -- > > > > > > Dave > > > > > > > > > Thanks for that, coming from a C++ background it's a refreshing > > > change to be looking at something that seems logical and efficient. > > > > > > I'd briefly looked at the event socket and wondered if that was the > > > way to go. I presume that there's some sort of event generation > > > that can trigger and external process as well somewhere, though all > > > I need to do is update mysql (hopefully using some sort of pooled > > > connection) > > > > > > I'm not using a TDM card, I have a direct interconnect with the PSTN > > > breakout provider with 1,500 channels available to me. I'm finding > > > Asterisk proving to be less than stable at high call volumes and > > > load values spike at more than 100 calls with billing/accounting in > > > place, hence my interest in FS. The only thing that's concerning me > > > is XML at the moment. Lots of code and very wordy. I'm sure I'll > > > appreciate why XML given time > > > > > > Regards, > > > > > > > > > ____________________________________________________________________ > > > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Michael S Collins > > > Sent: 03 February 2009 01:17 > > > To: freeswitch-users at lists.freeswitch.org > > > Subject: Re: [Freeswitch-users] Generating calls from external > > > source > > > > > > Nik, > > > > > > Welcome to FreeSWITCH! The short answer is "yes, FS can do that." > > > The first thing that you should do is unlearn "the Asterisk way" of > > > thinking. Usually there is an elegant way of doing things in FS that > > > wasn't possible in Ast. > > > > > > I would recommend that you start by looking at the event socket, > > > which is somewhat analogous to the AMI only cooler. :) I have > > > personally done something similar to this using the event socket and > > > a Perl script. The key is to learn the syntax of the originate > > > command. (definitely hit the wiki and IRC channel) > > > Are you using TDM cards for this? Just curious. > > > > > > -MC (IRC nick: mercutioviz) > > > > > > Sent from my iPhone > > > > > > On Feb 2, 2009, at 3:35 PM, "Nik Middleton" > > > wrote: > > > > Hi Guys, > > > > > > > > As a long time Asterisk user, I'm looking into freeswitch as an > > > > alternative mainly due to (list multiple reasons here) > > > > > > > > Can anyone give me a pointer as to how I would achieve the > > > > following? > > > > > > > > I need to replicate an emergency broadcast system currently > > > > running under Asterisk. > > > > > > > > At the moment, I run through a Mysql database and using the > > > > manager API, issues an Originate command to dial a number. > > > > > > > > When the call is answered, a message is played, and the recipient > > > > has the option of hitting a digit to confirm receipt. I then call > > > > an AGI script to update the database. > > > > > > > > Is this fairly easy to do in Freeswitch? > > > > > > > > Not looking for code, just some pointers as to what's available to > > > > do the above / > > > > > > > > Regards, > > > > _______________________________________________ > > > > Freeswitch-users mailing list > > > > Freeswitch-users at lists.freeswitch.org > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > http://www.freeswitch.org > > > _______________________________________________ > > > Freeswitch-users mailing list > > > Freeswitch-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090203/2dc96b6e/attachment.html From e.schmidbauer at gmail.com Tue Feb 3 09:16:59 2009 From: e.schmidbauer at gmail.com (e schmidbauer) Date: Tue, 3 Feb 2009 12:16:59 -0500 Subject: [Freeswitch-users] shoutcast skips Message-ID: <2cef777b0902030916s77339375pabc3d3c6fd9aec1a@mail.gmail.com> hey everyone. just wondering if anyone has tested recording conferences at 48000h celt to a shoutcast stream or wav file. we are able to have cd quality conferences with 3 members each using the celt codec with little or no noise disturbances or skipping. but when we try to record the conference either to a wav file or to a shoutcast stream, the quality significantly decreases due to skipping or popping noises. im not sure but maybe we are having this problem because our server doesnt have the CPU power to handle reencoding on the fly like that. we are using a 2.8ghz amd64 dual core, 4gig ddr 800 as our freeswitch server. im thinking if there is a way to record the conference as a celt audio file (instead of reencoding to mp3) that may reduce the CPU power needed and therefore solve the problem or we just need a more powerful server. could anyone recommended what kind of server we would need to handle such instances as i described above? thank you. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090203/d9a2483d/attachment-0001.html From nicolas at medularis.com Tue Feb 3 09:20:56 2009 From: nicolas at medularis.com (Nicolas Brenner) Date: Tue, 3 Feb 2009 14:20:56 -0300 Subject: [Freeswitch-users] origainate through sofia gateway In-Reply-To: <1233668179.5354.15.camel@dotw1126.dotsystems.pl> References: <1233668179.5354.15.camel@dotw1126.dotsystems.pl> Message-ID: <1b46b4e80902030920p521d6079s4c9b78d8d39a924a@mail.gmail.com> Jacek, I had a similar problem once. It actually depends on your sip gateway, but I was able to solve the problem by setting the caller id, ie: session1 = new Session(); session1.setCallerData("caller_id_name", "8280052500"); session1.setCallerData("caller_id_number", "8280052500"); session1.originate(session1, "{ignore_early_media=true}sofia/gateway/sip.ipcorp.cl/0225490317", 60); In this case, the caller_id was the number assigned to me by the external gateway. Hope it helps. Nicolas On Tue, Feb 3, 2009 at 10:36 AM, Jacek Sokulski wrote: > Hello > I am trying to initiate a call from javascript, it works fine for local numbers: > >> session1.originate(session1, "{ignore_early_media=true}user/1008 at 192.168.1.122"); > > but when I am trying to connect through sofia gateway, the connection is not being established: > >> session2.originate(session2, "sofia/gateway/halonet/0225490317"); > > although I can call to this number from softphone. > I have also tried setting effective_caller_id_number: > >> session1.originate(session1, "{effective_caller_id_number=fixed0248b}sofia/gateway/halonet/0225490317"); > > with the same result. > > A configuration in the dialplan that works is: > >> >> >> >> >> >> >> >> > > > Would appreciate any help. > Jacek > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From brian at freeswitch.org Tue Feb 3 09:23:02 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 3 Feb 2009 11:23:02 -0600 Subject: [Freeswitch-users] shoutcast skips In-Reply-To: <2cef777b0902030916s77339375pabc3d3c6fd9aec1a@mail.gmail.com> References: <2cef777b0902030916s77339375pabc3d3c6fd9aec1a@mail.gmail.com> Message-ID: <898222CF-3335-4E2D-AE79-E57CFEE7EFB8@freeswitch.org> You forgot to tell us what revision of the code you're on? /b On Feb 3, 2009, at 11:16 AM, e schmidbauer wrote: > hey everyone. just wondering if anyone has tested recording > conferences at 48000h celt to a shoutcast stream or wav file. > we are able to have cd quality conferences with 3 members each using > the celt codec with little or no noise disturbances or skipping. > but when we try to record the conference either to a wav file or to > a shoutcast stream, the quality significantly decreases due to > skipping or popping noises. > im not sure but maybe we are having this problem because our server > doesnt have the CPU power to handle reencoding on the fly like that. > we are using a 2.8ghz amd64 dual core, 4gig ddr 800 as our > freeswitch server. > im thinking if there is a way to record the conference as a celt > audio file (instead of reencoding to mp3) that may reduce the CPU > power needed and therefore solve the problem or we just need a more > powerful server. > could anyone recommended what kind of server we would need to handle > such instances as i described above? thank you. > _______________________________________________ > Freeswitch-users mailing list From brian at freeswitch.org Tue Feb 3 09:25:33 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 3 Feb 2009 11:25:33 -0600 Subject: [Freeswitch-users] origainate through sofia gateway In-Reply-To: <1b46b4e80902030920p521d6079s4c9b78d8d39a924a@mail.gmail.com> References: <1233668179.5354.15.camel@dotw1126.dotsystems.pl> <1b46b4e80902030920p521d6079s4c9b78d8d39a924a@mail.gmail.com> Message-ID: <5E2B217D-8C85-48DB-800A-535A7FAAE24B@freeswitch.org> YOU should NEVER use this method or call setCallerData at all you should use the correct methods to override the callerid. If its a B-Leg born from an A-Leg you use these on the on the A-Leg: http://wiki.freeswitch.org/wiki/Channel_Variables#effective_caller_id_name http://wiki.freeswitch.org/wiki/Channel_Variables#effective_caller_id_number If you're originating you use this: http://wiki.freeswitch.org/wiki/Channel_Variables#origination_caller_id_name http://wiki.freeswitch.org/wiki/Channel_Variables#origination_caller_id_number /b On Feb 3, 2009, at 11:20 AM, Nicolas Brenner wrote: > Jacek, > > I had a similar problem once. It actually depends on your sip gateway, > but I was able to solve the problem by setting the caller id, ie: > > session1 = new Session(); > session1.setCallerData("caller_id_name", "8280052500"); > session1.setCallerData("caller_id_number", "8280052500"); > session1.originate(session1, > "{ignore_early_media=true}sofia/gateway/sip.ipcorp.cl/0225490317", > 60); > > In this case, the caller_id was the number assigned to me by the > external gateway. > > Hope it helps. > > Nicolas From msc at freeswitch.org Tue Feb 3 09:27:57 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 3 Feb 2009 09:27:57 -0800 Subject: [Freeswitch-users] Generating calls from external source In-Reply-To: References: <40B58CBF-9D0F-43A3-B36F-BD05916756E4@3c.co.uk> <1233666748.24619.8.camel@stargate> Message-ID: <87f2f3b90902030927t2566577cm68248a16eff0246b@mail.gmail.com> On Tue, Feb 3, 2009 at 8:53 AM, Nik Middleton wrote: > Are you suggesting that I should process the call externally instead of > using the dialplan? That would be neat as the audio file select could I'm not saying you should, merely that you could. What I did was create a bunch of extensions in my dialplan that handled various steps of the IVR outbound call: start, answered, busy, not answered, SIT tones, etc. So my originate command would originate the call (A leg) and drop the B leg into the dialplan at the "start" extension and then it goes from there. It listens for early media busy or SIT tones and also does an "execute_on_answer" to the extension that does the actual IVR. (Only need the IVR on an answered call.) If the call is not answered after 25 seconds then I run a Lua script that checks for the presence of certain channel variables that I set with the "tone_detect" application (busy and SIT). If none of those are present then I assume the call went unanswered and do the post-processing. > be driven from the db select for the number. I presume that I could > also bridge the call to another number as well dependant on DTMF > selection? Yes, you can do this as well. You can build an IVR in XML or you can build in a scripting language like Lua: demo IVR: http://svn.freeswitch.org/svn/freeswitch/trunk/conf/autoload_configs/ivr.conf.xml Lua IVR info: http://wiki.freeswitch.org/wiki/IVR#Lua_IVRs Sorry if this is all a bit overwhelming, but you'll be glad that you dove in to FS because it does soooo much and does it so well. Enjoy! -MC > > Regards > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Raul > Fragoso > Sent: 03 February 2009 13:12 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Generating calls from external source > > In addition do David's suggestion, you probably want to have your > application to watch for some specific events after the call is > originated and take action based on them. For example, you could watch > for the CHANNEL_ANSWER event and play some audio file waiting for some > digit, which is generated by the DTMF event. > To watch only for those specific events, you should do the following > just after authentication (still using Perl as an example, but the > mod_event_socket is language agnostic), then you will receive those > events from FreeSWITCH through the socket stream: > > ... > print $sock "auth XXX\n\n"; > print $sock "event plain CHANNEL_ANSWER DTMF\n\n"; > ... > > To see a list of available events, please look at the following wiki > pages: > http://wiki.freeswitch.org/wiki/Mod_event_socket#event > http://wiki.freeswitch.org/wiki/Event_list > > Regards, > > Raul > > On Tue, 2009-02-03 at 09:46 +0000, David Knell wrote: >> Hi Nik, >> >> >> Here's a snipped in Perl that launches an outbound call: >> >> >> if (my $sock = IO::Socket::INET->new(Proto =>'tcp', PeerAddr => >> '127.0.0.1', PeerPort => 8021)) { >> print $sock "auth XXX\n\n"; >> print $sock "api originate {softivr_id=$siid,src_softivr_id= >> $siid,softivr_outdial=true}sofia/frombt/$ntd at 1.2.3.4 $service\n\n"; >> $sock->close(); >> } >> >> >> - it does no error checking or anything, but (line by line) it: >> - opens a socket to the event socket interface >> - authenticates >> - issues an originate which dials out to the number in $ntd. The >> bits in {} set a bunch of variables on the channel, which are used by >> the software which processes the call later on. The call is linked to >> the extension in $service - FS looks this up in the dialplan - which >> handles our end. >> - closes the socket >> >> >> Cheers -- >> >> >> Dave >> >> >> >> > Thanks for that, coming from a C++ background it's a refreshing >> > change to be looking at something that seems logical and efficient. >> > >> > I'd briefly looked at the event socket and wondered if that was the >> > way to go. I presume that there's some sort of event generation >> > that can trigger and external process as well somewhere, though all >> > I need to do is update mysql (hopefully using some sort of pooled >> > connection) >> > >> > I'm not using a TDM card, I have a direct interconnect with the PSTN >> > breakout provider with 1,500 channels available to me. I'm finding >> > Asterisk proving to be less than stable at high call volumes and >> > load values spike at more than 100 calls with billing/accounting in >> > place, hence my interest in FS. The only thing that's concerning me >> > is XML at the moment. Lots of code and very wordy. I'm sure I'll >> > appreciate why XML given time >> > >> > Regards, >> > >> > >> > ____________________________________________________________________ >> > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Michael S Collins >> > Sent: 03 February 2009 01:17 >> > To: freeswitch-users at lists.freeswitch.org >> > Subject: Re: [Freeswitch-users] Generating calls from external >> > source >> > >> > Nik, >> > >> > Welcome to FreeSWITCH! The short answer is "yes, FS can do that." >> > The first thing that you should do is unlearn "the Asterisk way" of >> > thinking. Usually there is an elegant way of doing things in FS that >> > wasn't possible in Ast. >> > >> > I would recommend that you start by looking at the event socket, >> > which is somewhat analogous to the AMI only cooler. :) I have >> > personally done something similar to this using the event socket and >> > a Perl script. The key is to learn the syntax of the originate >> > command. (definitely hit the wiki and IRC channel) >> > Are you using TDM cards for this? Just curious. >> > >> > -MC (IRC nick: mercutioviz) >> > >> > Sent from my iPhone >> > >> > On Feb 2, 2009, at 3:35 PM, "Nik Middleton" >> > wrote: >> > > Hi Guys, >> > > >> > > As a long time Asterisk user, I'm looking into freeswitch as an >> > > alternative mainly due to (list multiple reasons here) >> > > >> > > Can anyone give me a pointer as to how I would achieve the >> > > following? >> > > >> > > I need to replicate an emergency broadcast system currently >> > > running under Asterisk. >> > > >> > > At the moment, I run through a Mysql database and using the >> > > manager API, issues an Originate command to dial a number. >> > > >> > > When the call is answered, a message is played, and the recipient >> > > has the option of hitting a digit to confirm receipt. I then call >> > > an AGI script to update the database. >> > > >> > > Is this fairly easy to do in Freeswitch? >> > > >> > > Not looking for code, just some pointers as to what's available to >> > > do the above / >> > > >> > > Regards, >> > > _______________________________________________ >> > > Freeswitch-users mailing list >> > > Freeswitch-users at lists.freeswitch.org >> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > > http://www.freeswitch.org >> > _______________________________________________ >> > Freeswitch-users mailing list >> > Freeswitch-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From nicolas at medularis.com Tue Feb 3 09:29:26 2009 From: nicolas at medularis.com (Nicolas Brenner) Date: Tue, 3 Feb 2009 14:29:26 -0300 Subject: [Freeswitch-users] Generating calls from external source In-Reply-To: <35b355e90902030913s1cec764j46280ac2d9a162a7@mail.gmail.com> References: <40B58CBF-9D0F-43A3-B36F-BD05916756E4@3c.co.uk> <1233666748.24619.8.camel@stargate> <35b355e90902030913s1cec764j46280ac2d9a162a7@mail.gmail.com> Message-ID: <1b46b4e80902030929h4d4b4b39x9884f856b6028ad@mail.gmail.com> Nik, There's also a PHP library fs_sock.php under contrib in the source code. I used it to create a simple app that originates calls and then run some other sutff when it detects the call has ended. The actual call originate command is executed inside a javascript file which is run using bgapi jsrun. The js script also makes a POST request to an external URL using CURL. There's plenty to play around with, Freeswitch is really great, and mostly easy, a world of difference with *. Good luck! Nicolas On Tue, Feb 3, 2009 at 2:13 PM, Shelby Ramsey wrote: > Nik, > There are a lot of ways to make FS dial out and deliver messaging etc. We > are going through the process of replacing * for this purpose. For us > (getting started with the help of our friends here on the list) it has been > pretty easy. > With * we were using AMI to originate calls ... to migrate to FS we just > changed that to use event_socket with bgapi to originate the call and > connect the call to a context and extension. There are several ways to get > the dialplan to FS after that ... a script, xml_curl, or statically > configured in the conf directory. > So as an example the application we have just logs into the FS socket > (similar to * but much better) and then rips off calls like this: > bgapi originate{$set_some_vars}sofia/external/$ANI@$IP:$PORT $EXTENSION xml > $CONTEXT > The beauty of it all is that: > -- a lot of flexibility in what you can do (like drive the call through > events) > -- the CDR reporting is about 3 million times better than * > -- obviously higher capacity > I'd start playing with event_socket and some static dialplans to get the > feel for it ... but if you have an application written already to work with > * (i.e. the logic and backend) it will be very easy to migrate and you'll be > glad you did it! > Shelby > > > On Tue, Feb 3, 2009 at 10:53 AM, Nik Middleton > wrote: >> >> Are you suggesting that I should process the call externally instead of >> using the dialplan? That would be neat as the audio file select could >> be driven from the db select for the number. I presume that I could >> also bridge the call to another number as well dependant on DTMF >> selection? >> >> Regards >> >> >> -----Original Message----- >> From: freeswitch-users-bounces at lists.freeswitch.org >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Raul >> Fragoso >> Sent: 03 February 2009 13:12 >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] Generating calls from external source >> >> In addition do David's suggestion, you probably want to have your >> application to watch for some specific events after the call is >> originated and take action based on them. For example, you could watch >> for the CHANNEL_ANSWER event and play some audio file waiting for some >> digit, which is generated by the DTMF event. >> To watch only for those specific events, you should do the following >> just after authentication (still using Perl as an example, but the >> mod_event_socket is language agnostic), then you will receive those >> events from FreeSWITCH through the socket stream: >> >> ... >> print $sock "auth XXX\n\n"; >> print $sock "event plain CHANNEL_ANSWER DTMF\n\n"; >> ... >> >> To see a list of available events, please look at the following wiki >> pages: >> http://wiki.freeswitch.org/wiki/Mod_event_socket#event >> http://wiki.freeswitch.org/wiki/Event_list >> >> Regards, >> >> Raul >> >> On Tue, 2009-02-03 at 09:46 +0000, David Knell wrote: >> > Hi Nik, >> > >> > >> > Here's a snipped in Perl that launches an outbound call: >> > >> > >> > if (my $sock = IO::Socket::INET->new(Proto =>'tcp', PeerAddr => >> > '127.0.0.1', PeerPort => 8021)) { >> > print $sock "auth XXX\n\n"; >> > print $sock "api originate {softivr_id=$siid,src_softivr_id= >> > $siid,softivr_outdial=true}sofia/frombt/$ntd at 1.2.3.4 $service\n\n"; >> > $sock->close(); >> > } >> > >> > >> > - it does no error checking or anything, but (line by line) it: >> > - opens a socket to the event socket interface >> > - authenticates >> > - issues an originate which dials out to the number in $ntd. The >> > bits in {} set a bunch of variables on the channel, which are used by >> > the software which processes the call later on. The call is linked to >> > the extension in $service - FS looks this up in the dialplan - which >> > handles our end. >> > - closes the socket >> > >> > >> > Cheers -- >> > >> > >> > Dave >> > >> > >> > >> > > Thanks for that, coming from a C++ background it's a refreshing >> > > change to be looking at something that seems logical and efficient. >> > > >> > > I'd briefly looked at the event socket and wondered if that was the >> > > way to go. I presume that there's some sort of event generation >> > > that can trigger and external process as well somewhere, though all >> > > I need to do is update mysql (hopefully using some sort of pooled >> > > connection) >> > > >> > > I'm not using a TDM card, I have a direct interconnect with the PSTN >> > > breakout provider with 1,500 channels available to me. I'm finding >> > > Asterisk proving to be less than stable at high call volumes and >> > > load values spike at more than 100 calls with billing/accounting in >> > > place, hence my interest in FS. The only thing that's concerning me >> > > is XML at the moment. Lots of code and very wordy. I'm sure I'll >> > > appreciate why XML given time >> > > >> > > Regards, >> > > >> > > >> > > ____________________________________________________________________ >> > > From: freeswitch-users-bounces at lists.freeswitch.org >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of >> Michael S Collins >> > > Sent: 03 February 2009 01:17 >> > > To: freeswitch-users at lists.freeswitch.org >> > > Subject: Re: [Freeswitch-users] Generating calls from external >> > > source >> > > >> > > Nik, >> > > >> > > Welcome to FreeSWITCH! The short answer is "yes, FS can do that." >> > > The first thing that you should do is unlearn "the Asterisk way" of >> > > thinking. Usually there is an elegant way of doing things in FS that >> > > wasn't possible in Ast. >> > > >> > > I would recommend that you start by looking at the event socket, >> > > which is somewhat analogous to the AMI only cooler. :) I have >> > > personally done something similar to this using the event socket and >> > > a Perl script. The key is to learn the syntax of the originate >> > > command. (definitely hit the wiki and IRC channel) >> > > Are you using TDM cards for this? Just curious. >> > > >> > > -MC (IRC nick: mercutioviz) >> > > >> > > Sent from my iPhone >> > > >> > > On Feb 2, 2009, at 3:35 PM, "Nik Middleton" >> > > wrote: >> > > > Hi Guys, >> > > > >> > > > As a long time Asterisk user, I'm looking into freeswitch as an >> > > > alternative mainly due to (list multiple reasons here) >> > > > >> > > > Can anyone give me a pointer as to how I would achieve the >> > > > following? >> > > > >> > > > I need to replicate an emergency broadcast system currently >> > > > running under Asterisk. >> > > > >> > > > At the moment, I run through a Mysql database and using the >> > > > manager API, issues an Originate command to dial a number. >> > > > >> > > > When the call is answered, a message is played, and the recipient >> > > > has the option of hitting a digit to confirm receipt. I then call >> > > > an AGI script to update the database. >> > > > >> > > > Is this fairly easy to do in Freeswitch? >> > > > >> > > > Not looking for code, just some pointers as to what's available to >> > > > do the above / >> > > > >> > > > Regards, >> > > > _______________________________________________ >> > > > Freeswitch-users mailing list >> > > > Freeswitch-users at lists.freeswitch.org >> > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > > > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > > > http://www.freeswitch.org >> > > _______________________________________________ >> > > Freeswitch-users mailing list >> > > Freeswitch-users at lists.freeswitch.org >> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > > http://www.freeswitch.org >> > > >> > >> > >> > _______________________________________________ >> > Freeswitch-users mailing list >> > Freeswitch-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From nicolas at medularis.com Tue Feb 3 09:31:32 2009 From: nicolas at medularis.com (Nicolas Brenner) Date: Tue, 3 Feb 2009 14:31:32 -0300 Subject: [Freeswitch-users] origainate through sofia gateway In-Reply-To: <5E2B217D-8C85-48DB-800A-535A7FAAE24B@freeswitch.org> References: <1233668179.5354.15.camel@dotw1126.dotsystems.pl> <1b46b4e80902030920p521d6079s4c9b78d8d39a924a@mail.gmail.com> <5E2B217D-8C85-48DB-800A-535A7FAAE24B@freeswitch.org> Message-ID: <1b46b4e80902030931y6a1679afw12ec37596ba9d77d@mail.gmail.com> Oops! Well, fortunately I don't use that voip provider anymore (nor the script). Thanks Brian. Nicolas On Tue, Feb 3, 2009 at 2:25 PM, Brian West wrote: > YOU should NEVER use this method or call setCallerData at all you > should use the correct methods to override the callerid. > > If its a B-Leg born from an A-Leg you use these on the on the A-Leg: > > http://wiki.freeswitch.org/wiki/Channel_Variables#effective_caller_id_name > http://wiki.freeswitch.org/wiki/Channel_Variables#effective_caller_id_number > > If you're originating you use this: > > http://wiki.freeswitch.org/wiki/Channel_Variables#origination_caller_id_name > http://wiki.freeswitch.org/wiki/Channel_Variables#origination_caller_id_number > > /b From raul at etellicom.com Tue Feb 3 09:36:06 2009 From: raul at etellicom.com (Raul Fragoso) Date: Tue, 03 Feb 2009 15:36:06 -0200 Subject: [Freeswitch-users] Generating calls from external source In-Reply-To: References: <40B58CBF-9D0F-43A3-B36F-BD05916756E4@3c.co.uk> <1233666748.24619.8.camel@stargate> Message-ID: <1233682566.24619.34.camel@stargate> Hi Nik, That's one possibility, yes. You could use mod_xml_curl to provide the dial-plan on the fly and then use mod_event_socket to send commands to FS and process events. That's exactly what I do actually, we have an IVR engine that is driven by mod_event_socket and another module that provides the XML dial-plan through mod_xml_curl. The beauty of FS is that you have many options to tack a problem, and all of those options are very elegant. I suggest looking at mod_event_socket first and then decide if you can live with the static dial-plan or go to a more dynamic dial-plan via mod_xml_curl. Regards, Raul On Tue, 2009-02-03 at 16:53 +0000, Nik Middleton wrote: > Are you suggesting that I should process the call externally instead of > using the dialplan? That would be neat as the audio file select could > be driven from the db select for the number. I presume that I could > also bridge the call to another number as well dependant on DTMF > selection? > > Regards > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Raul > Fragoso > Sent: 03 February 2009 13:12 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Generating calls from external source > > In addition do David's suggestion, you probably want to have your > application to watch for some specific events after the call is > originated and take action based on them. For example, you could watch > for the CHANNEL_ANSWER event and play some audio file waiting for some > digit, which is generated by the DTMF event. > To watch only for those specific events, you should do the following > just after authentication (still using Perl as an example, but the > mod_event_socket is language agnostic), then you will receive those > events from FreeSWITCH through the socket stream: > > ... > print $sock "auth XXX\n\n"; > print $sock "event plain CHANNEL_ANSWER DTMF\n\n"; > ... > > To see a list of available events, please look at the following wiki > pages: > http://wiki.freeswitch.org/wiki/Mod_event_socket#event > http://wiki.freeswitch.org/wiki/Event_list > > Regards, > > Raul > > On Tue, 2009-02-03 at 09:46 +0000, David Knell wrote: > > Hi Nik, > > > > > > Here's a snipped in Perl that launches an outbound call: > > > > > > if (my $sock = IO::Socket::INET->new(Proto =>'tcp', PeerAddr => > > '127.0.0.1', PeerPort => 8021)) { > > print $sock "auth XXX\n\n"; > > print $sock "api originate {softivr_id=$siid,src_softivr_id= > > $siid,softivr_outdial=true}sofia/frombt/$ntd at 1.2.3.4 $service\n\n"; > > $sock->close(); > > } > > > > > > - it does no error checking or anything, but (line by line) it: > > - opens a socket to the event socket interface > > - authenticates > > - issues an originate which dials out to the number in $ntd. The > > bits in {} set a bunch of variables on the channel, which are used by > > the software which processes the call later on. The call is linked to > > the extension in $service - FS looks this up in the dialplan - which > > handles our end. > > - closes the socket > > > > > > Cheers -- > > > > > > Dave > > > > > > > > > Thanks for that, coming from a C++ background it's a refreshing > > > change to be looking at something that seems logical and efficient. > > > > > > I'd briefly looked at the event socket and wondered if that was the > > > way to go. I presume that there's some sort of event generation > > > that can trigger and external process as well somewhere, though all > > > I need to do is update mysql (hopefully using some sort of pooled > > > connection) > > > > > > I'm not using a TDM card, I have a direct interconnect with the PSTN > > > breakout provider with 1,500 channels available to me. I'm finding > > > Asterisk proving to be less than stable at high call volumes and > > > load values spike at more than 100 calls with billing/accounting in > > > place, hence my interest in FS. The only thing that's concerning me > > > is XML at the moment. Lots of code and very wordy. I'm sure I'll > > > appreciate why XML given time > > > > > > Regards, > > > > > > > > > ____________________________________________________________________ > > > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Michael S Collins > > > Sent: 03 February 2009 01:17 > > > To: freeswitch-users at lists.freeswitch.org > > > Subject: Re: [Freeswitch-users] Generating calls from external > > > source > > > > > > Nik, > > > > > > Welcome to FreeSWITCH! The short answer is "yes, FS can do that." > > > The first thing that you should do is unlearn "the Asterisk way" of > > > thinking. Usually there is an elegant way of doing things in FS that > > > wasn't possible in Ast. > > > > > > I would recommend that you start by looking at the event socket, > > > which is somewhat analogous to the AMI only cooler. :) I have > > > personally done something similar to this using the event socket and > > > a Perl script. The key is to learn the syntax of the originate > > > command. (definitely hit the wiki and IRC channel) > > > Are you using TDM cards for this? Just curious. > > > > > > -MC (IRC nick: mercutioviz) > > > > > > Sent from my iPhone > > > > > > On Feb 2, 2009, at 3:35 PM, "Nik Middleton" > > > wrote: > > > > Hi Guys, > > > > > > > > As a long time Asterisk user, I'm looking into freeswitch as an > > > > alternative mainly due to (list multiple reasons here) > > > > > > > > Can anyone give me a pointer as to how I would achieve the > > > > following? > > > > > > > > I need to replicate an emergency broadcast system currently > > > > running under Asterisk. > > > > > > > > At the moment, I run through a Mysql database and using the > > > > manager API, issues an Originate command to dial a number. > > > > > > > > When the call is answered, a message is played, and the recipient > > > > has the option of hitting a digit to confirm receipt. I then call > > > > an AGI script to update the database. > > > > > > > > Is this fairly easy to do in Freeswitch? > > > > > > > > Not looking for code, just some pointers as to what's available to > > > > do the above / > > > > > > > > Regards, > > > > _______________________________________________ > > > > Freeswitch-users mailing list > > > > Freeswitch-users at lists.freeswitch.org > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > http://www.freeswitch.org > > > _______________________________________________ > > > Freeswitch-users mailing list > > > Freeswitch-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From nik.middleton at noblesolutions.co.uk Tue Feb 3 10:07:32 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Tue, 3 Feb 2009 18:07:32 -0000 Subject: [Freeswitch-users] OPenser <-> FS Do I need this? In-Reply-To: <87f2f3b90902030908n629399bdmdd46f633b803b5f6@mail.gmail.com> References: <2cbf225c0902021821u2bc8b622ka5d9f94f05a968ae@mail.gmail.com><1233668029.24619.29.camel@stargate> <87f2f3b90902030908n629399bdmdd46f633b803b5f6@mail.gmail.com> Message-ID: Well Openser has better NAT handling than Asterisk for a start. In addition it takes the load off of Asterisk with regards to registrations. Further, I'm able to have multiple asterisk servers fronted by Openser Finally, I've numerous posts that * chokes with sip clients > 200. I couldn't afford to take the risk. But the biggest issue is with load spikes and asterisk. I've never gotten to the bottom of it, and believe me a lot of people far smarter then me have tried to figure it out. So... The more I can keep asterisk out of the mundane stuff the better. It's been said to me many times, that the way Asterisk is put together is fundamentally flawed and this really shows it's self under load. Not knocking Asterisk, it's served me well for the last 4 years. Heck I've got a book being published on it in a couple of months, but for me, I need a scalable solution, hence my interest in FS. I also don't see * going beyond 1.4. 1.6 as far as I can tell has a very low take-up rate, why ? well because they've changed how everything works to the extent that hardly anything written for 1.4 can port to 1.6. The syntax changes don't appear to serve any real purpose. So to get back to my original question, if FS can handle a significantly higher number of call setups, then perhaps I don't need OpenSer, that was the thrust of my post. Regards -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 03 February 2009 17:08 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] OPenser <-> FS Do I need this? On Tue, Feb 3, 2009 at 8:20 AM, Nik Middleton wrote: > Newbie with FS, currently have Asterisk servers front ended by Openser > > Question: I have around 400 sip remote clients, if I were to deploy FS, > do I need Openser? Is there any advantage in retaining Openser? If I may ask... why did you have OpenSER with your Asterisk deployment? Reason I ask is because some people do that "because Asterisk sucks" but others have a specific application or reason. What does OpenSER do for your Asterisk install? -MC > > Regards > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From krice at freeswitch.org Tue Feb 3 10:16:08 2009 From: krice at freeswitch.org (Ken Rice) Date: Tue, 03 Feb 2009 12:16:08 -0600 Subject: [Freeswitch-users] OPenser <-> FS Do I need this? In-Reply-To: Message-ID: FreeSwitch is very capable of handling high call setup loads... The question is what do you consider high setup loads? Where it is true, OpenSER/SIP/whatever its called this week can handle a much higher packet per second load then freeswitch, freeswitch on the other hand is capable of handling much more call volume then asterisk... Certain people hate when I quote numbers but I have personally deployed FreeSwitch on projects that handle (per FS Box) > 500 calls/sec (that's 2 leg calls) and in excess of concurrent calls... The real question is not can FS hang, but what at what level do you call 'high volume'... What I call high volume is a telemarketer running at 2500 calls/sec and peak concurrent channel usage in the 10,000 to 15,000 channel range K > From: Nik Middleton > Subject: Re: [Freeswitch-users] OPenser <-> FS Do I need this? > > ----SNIP > So to get back to my original question, if FS can handle a significantly > higher number of call setups, then perhaps I don't need OpenSer, that > was the thrust of my post. > ----SNIP From nik.middleton at noblesolutions.co.uk Tue Feb 3 10:31:37 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Tue, 3 Feb 2009 18:31:37 -0000 Subject: [Freeswitch-users] OPenser <-> FS Do I need this? In-Reply-To: References: Message-ID: If you're telling me that FS can handle the figures quoted, that's plenty enough for me. I have 5,000 lines PSTN /channels, possibly double that shortly. I need to fill all of them as quickly as possible and maintain that level for a given period of time. So I guess I'm in the upper medium end of the scale. Regards, -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice Sent: 03 February 2009 18:16 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] OPenser <-> FS Do I need this? FreeSwitch is very capable of handling high call setup loads... The question is what do you consider high setup loads? Where it is true, OpenSER/SIP/whatever its called this week can handle a much higher packet per second load then freeswitch, freeswitch on the other hand is capable of handling much more call volume then asterisk... Certain people hate when I quote numbers but I have personally deployed FreeSwitch on projects that handle (per FS Box) > 500 calls/sec (that's 2 leg calls) and in excess of concurrent calls... The real question is not can FS hang, but what at what level do you call 'high volume'... What I call high volume is a telemarketer running at 2500 calls/sec and peak concurrent channel usage in the 10,000 to 15,000 channel range K > From: Nik Middleton > Subject: Re: [Freeswitch-users] OPenser <-> FS Do I need this? > > ----SNIP > So to get back to my original question, if FS can handle a significantly > higher number of call setups, then perhaps I don't need OpenSer, that > was the thrust of my post. > ----SNIP _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From ajlong at worldlink.net Tue Feb 3 10:47:59 2009 From: ajlong at worldlink.net (Adam Long) Date: Tue, 3 Feb 2009 13:47:59 -0500 Subject: [Freeswitch-users] FreeSwitch setup as a "Dumb" SBC In-Reply-To: <4987E527.1040909@laposte.net> References: <019501c985ac$4f00ee60$ed02cb20$@net> <4987E527.1040909@laposte.net> Message-ID: <022001c9862f$efd4b7d0$cf7e2770$@net> Hi Rod, Great info, Thanks! Glad to see others are interested in the same concept. My reasons for SER as routing core and implementation is slightly different yet similar. I like your Redirect model, with that you are truly using your Kamailio as route server only. I would imagine very scalable. - Are you able to do any round robin, serial or parallel forking with this? - I wonder if multiple Contacts in the 302 response maybe with some logic in FreeSwitch dialplan? If so I think your design is a bit more efficient than mine as it keeps SER out of the call path. My design is little different.. it is more of a "Stateful" setup. With SER staying in call path and FreeSwitch at Edge. I do this to enable Serial Forking to a series of SBCs (FreeSwitch) geo distributed, when one of the branches is congested it forks to the next SBC (route). The FreeSwitch guys are probably right tho... with mod_easyroute and mod_lcr we could probably implement all of this in FreeSwitch without SER. I would be curious to know if anyone is doing something similar at high volumes and what sort of concurrency and cps they are able to achieve. I am a Perl and C# guy, I thought about implementing a mod_manged_lcr with memcached support. Memcache support would prob boost the scalability by a factor of 10 at least. I will let you know if I end up developing a high performance FreeSwitch route module. Right now I use memcache in a OpenSIPS perl script for my route caching and its incredibly fast and clusters well. It actually might be easier to add memcached support to mod_lcr and mod_easyroute but im not real strong in C/C++ I'll jump on IRC later and chat with some of the experts on this as I know memcache has been discussed before. I'd be curious to know if any progress has been made there already. Regards, -Adam -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of rod Sent: Tuesday, February 03, 2009 1:33 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] FreeSwitch setup as a "Dumb" SBC Hi Adam, I'm in the process of using FS as a SBC. For the route lookup, I do it using OpenSER carrierroute, without having to flow through SBC---Openser---SBC. I'm using carrierroute at this time cause I need more than 200 000 routing entries and carrierroute has been tested with twice this number. Here is the setup: - install openser and carrierroute and make openser listening on 127.0.0.1:5062 (for example) on your SBC - populate carrierroute table What I do to use carrierroute module from FS is to use a specific X-header (X-LOOKUP). In the dialplan, in the default context, I have something like this: The process is simple: the export "sip_h_X-ROUTE=LOOKUP" had a sip header X-ROUTE=LOOKUP then I bridge the call to 127.0.0.1:5062 (openser process) In openser I have a route block that checks the presence of header LOOKUP and openser sends a "604: unable to route call" if the prefix is not found, or a "302: with the IP of the gateway found" In FS, you can get the IP using the variable "${sip_redirect_contact_host_0}". Then I transfer this to the context ROUTING, where the check condition is based on the LOOKUP header that has been rewritten with this variable. I will document all this setup (installation of openser/carrierroute and config file of FS and openser) on a wiki page I start writing yesterday, so please be indulgent and patient. The next step is to test the scalability of this. I'm a very bad programmer, so that's the only way for me to contribute to FS, and as I see many people interested for an SBC setup, I think it could be great if we share our work/knowlegde. The wiki page is there: http://wiki.freeswitch.org/wiki/SBC_Setup regards, rod. Adam Long wrote: > > Hi Guys, > > I've been working at setting up a couple of FreeSwitch nodes as a > topology hiding SBCs that handles both ingress traffic from my > > providers/peers and pass traffic up to an openser router that then > routes call across the cluster of SBCs through which they reach the > destination. > > I have OpenSIPS/SER setup doing DB route lookups and ENUM with > LCR/Serial forking etc. > > My question is what would be the best way to send a call out to a > destination choosen by the OpenSER router? > > For example: > > SIP Provider -- > SBC --- > OpenSER ---- ( route lookup returns > 123.123.123.4 as dest ) -- > SBC --- > 123.123.123.4 > > I was thinking something along the lines of adding a "X-Route-To: > +1NXXNXXXXXX@ 123.123.123.4" with openser > > and then something like this in the SBC. > > > > > > > > > > > > Is this a wise approach, is there anything I could do to do this better? > > I'd like to keep the logic in the SBCs as simple as possible. > > I am pretty familiar with SIP but my knowledge fades when it gets into > the nitty gritty of routing. ie the Contact: and Via: headers > > and all that good stuff. > > I should also state I have two profiles defined one for the > internal/private "core" network and one for the outside "external" > network. > > Any thoughts on this at all would be greatly appreciated. > > Am I missing something in the SIP spec that would allow for this is a > standardized way? > > Regards, > > -Adam > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From ajlong at worldlink.net Tue Feb 3 12:24:58 2009 From: ajlong at worldlink.net (Adam Long) Date: Tue, 3 Feb 2009 15:24:58 -0500 Subject: [Freeswitch-users] mod_sofia "ReINVITE" Message-ID: <002b01c9863d$7c107470$74315d50$@net> In every one of my SIP sessions FreeSwitch appears to be inserting .. Contact: sip:mod_sofia at XXX.XXX.XXX.XXX:5060 Is this normal? I only ask as it is causing some of my end points to RE-INVITE back to this after the initial ( INVITE <---- > 100 Trying < --- > 200 OK ) call setup. If this is not normal or by design I can provide more details on configuration and dialplan. Thanks! Regards, -Adam -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090203/dd3018e1/attachment.html From e.schmidbauer at gmail.com Tue Feb 3 12:26:39 2009 From: e.schmidbauer at gmail.com (e schmidbauer) Date: Tue, 3 Feb 2009 15:26:39 -0500 Subject: [Freeswitch-users] shoutcast skips In-Reply-To: <898222CF-3335-4E2D-AE79-E57CFEE7EFB8@freeswitch.org> References: <2cef777b0902030916s77339375pabc3d3c6fd9aec1a@mail.gmail.com> <898222CF-3335-4E2D-AE79-E57CFEE7EFB8@freeswitch.org> Message-ID: <2cef777b0902031226l72f4a4ceia80ec00721456602@mail.gmail.com> im using the latest svn of freeswitch On Tue, Feb 3, 2009 at 12:23 PM, Brian West wrote: > You forgot to tell us what revision of the code you're on? > /b > > On Feb 3, 2009, at 11:16 AM, e schmidbauer wrote: > > > hey everyone. just wondering if anyone has tested recording > > conferences at 48000h celt to a shoutcast stream or wav file. > > we are able to have cd quality conferences with 3 members each using > > the celt codec with little or no noise disturbances or skipping. > > but when we try to record the conference either to a wav file or to > > a shoutcast stream, the quality significantly decreases due to > > skipping or popping noises. > > im not sure but maybe we are having this problem because our server > > doesnt have the CPU power to handle reencoding on the fly like that. > > we are using a 2.8ghz amd64 dual core, 4gig ddr 800 as our > > freeswitch server. > > im thinking if there is a way to record the conference as a celt > > audio file (instead of reencoding to mp3) that may reduce the CPU > > power needed and therefore solve the problem or we just need a more > > powerful server. > > could anyone recommended what kind of server we would need to handle > > such instances as i described above? thank you. > > _______________________________________________ > > Freeswitch-users mailing list > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090203/c90d68c5/attachment.html From krice at freeswitch.org Tue Feb 3 11:26:37 2009 From: krice at freeswitch.org (Ken Rice) Date: Tue, 03 Feb 2009 13:26:37 -0600 Subject: [Freeswitch-users] FreeSwitch setup as a "Dumb" SBC In-Reply-To: <022001c9862f$efd4b7d0$cf7e2770$@net> Message-ID: Actually I currently deploy FreeSWITCH for high volume usage using FreeSWITCH + mod_easyroute (I'm the author) and an advanced LCR module that does things like load balancing across multiple media gateways, auto route advance, and a few other nifty things... (this LCR module uses a proprietary algorithm so its not open source but it is licensable) With these things we do run OpenSER but only as a proxy to aggregate traffic heading upstream toward certain carriers (like L3 who make any IP changes a royal pain) Now to get down to some hard numbers that we have experience Equipment: DB Servers: Dell 2650 RAID 3+1 or 0+1 depending on number of Spindles, Dual 3GHz XEON (single code old slow FSB ones), 4G RAM, running Centos 5.2 and PostgreSQL 8.3 SIP Servers: Dell 1950 Dual Quad Core 2Ghz (E5335 part), 4 to 8G of RAM, GIG-E ethernet, whatever hard drive was cheap at time of order. Nothing really lives on these boxes but FreeSWITCH with mod_easyroute, mod_lcr_adv, and some CDR processing stuff DB servers feed all the route information... (yes we do the route lookups from the DB in real-time, the problem with most LCRs in doing this is an algorithm Call Rates Sustained, 500 avg cps, > 2000 calls (that's 2 legs not 1), avg invite delay 115ms (INVITE in to INVITE out measured with 'ngrep -q -t INVITE' - Note this is not a true picture of PDD as a number of other factors affect that, this is a picture of how much time we are adding on box in delaying an INVITE message) On Registrations we have experienced Registration/second rates exceeding 150 registrations per second using mod_xml_curl to feed the users directory. I suspect, this number can be greatly increased if we were to feed directory with something that cut out the apache and php over head K > From: Adam Long > Reply-To: > Date: Tue, 3 Feb 2009 13:47:59 -0500 > To: > Subject: Re: [Freeswitch-users] FreeSwitch setup as a "Dumb" SBC > > The FreeSwitch guys are probably right tho... with mod_easyroute and mod_lcr > we could probably implement all of this in FreeSwitch without SER. > I would be curious to know if anyone is doing something similar at high > volumes and what sort of concurrency and cps they > are able to achieve. From brian at freeswitch.org Tue Feb 3 12:29:19 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 3 Feb 2009 14:29:19 -0600 Subject: [Freeswitch-users] mod_sofia "ReINVITE" In-Reply-To: <002b01c9863d$7c107470$74315d50$@net> References: <002b01c9863d$7c107470$74315d50$@net> Message-ID: Yes this is normal. Your contact is mod_sofia ... why would it change? Remember its a B2Bua. Now you can put in your sofia profile but be warned it will break some devices. /b On Feb 3, 2009, at 2:24 PM, Adam Long wrote: > In every one of my SIP sessions FreeSwitch appears to be inserting ?. > > Contact: sip:mod_sofia at XXX.XXX.XXX.XXX:5060 > > Is this normal? > > I only ask as it is causing some of my end points to RE-INVITE back > to this after the initial ( INVITE <---- > 100 Trying < --- > > 200 OK ) call setup. > > If this is not normal or by design I can provide more details on > configuration and dialplan. > Thanks! > > Regards, > -Adam -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090203/a393d45f/attachment.html From brian at freeswitch.org Tue Feb 3 12:29:36 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 3 Feb 2009 14:29:36 -0600 Subject: [Freeswitch-users] shoutcast skips In-Reply-To: <2cef777b0902031226l72f4a4ceia80ec00721456602@mail.gmail.com> References: <2cef777b0902030916s77339375pabc3d3c6fd9aec1a@mail.gmail.com> <898222CF-3335-4E2D-AE79-E57CFEE7EFB8@freeswitch.org> <2cef777b0902031226l72f4a4ceia80ec00721456602@mail.gmail.com> Message-ID: "latest" isn't a number... Can you provide the exact SVN rev you're on? /b On Feb 3, 2009, at 2:26 PM, e schmidbauer wrote: > im using the latest svn of freeswitch From anthony.minessale at gmail.com Tue Feb 3 12:45:04 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 3 Feb 2009 14:45:04 -0600 Subject: [Freeswitch-users] mod_voicemail: Limiting the number how often a menu is repeated In-Reply-To: <49887A32.7060804@ewetel.de> References: <498734D0.5060004@ewetel.de> <49887A32.7060804@ewetel.de> Message-ID: <191c3a030902031245m76cca75dk48501976526423a3@mail.gmail.com> no, there is no way to do that. On Tue, Feb 3, 2009 at 11:09 AM, Helmut Kuper wrote: > Hi, > > has anybody an idea? > > regards > Helmut > > Am 02.02.2009 19:00, schrieb Helmut Kuper: > > Hello, > > > > today I searched for a way to limit the number of menu repeatings in > > mod_voicemail to let's say 3 times and when it reached the limit > > voicemail should abort. But I couldn't find a hint. Any ideas? > > > > > > regards > > helmut > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > -- > > Mit freundlichen Gr??en > Helmut Kuper > Finanzdienstleistungen und Entwicklung > Telefax: (0441) 8000-2799 > mailto:helmut.kuper at ewetel.de > ___________________________________ > EWE TEL GmbH > Cloppenburger Stra?e 310 > 26133 Oldenburg > EWE TEL GmbH > > Handelsregister Amtsgericht Oldenburg HRB 3723 > Vorsitzender des Aufsichtsrates: Heiko Harms > Gesch?ftsf?hrung: Hans-Joachim Iken (Vorsitzender), Dr. Norbert Schulz, > Dirk Thole > Homepage: http://www.ewetel.de > ___________________________________ > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090203/a228b229/attachment-0001.html From brian at freeswitch.org Tue Feb 3 12:51:49 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 3 Feb 2009 14:51:49 -0600 Subject: [Freeswitch-users] shoutcast skips In-Reply-To: <2cef777b0902031226l72f4a4ceia80ec00721456602@mail.gmail.com> References: <2cef777b0902030916s77339375pabc3d3c6fd9aec1a@mail.gmail.com> <898222CF-3335-4E2D-AE79-E57CFEE7EFB8@freeswitch.org> <2cef777b0902031226l72f4a4ceia80ec00721456602@mail.gmail.com> Message-ID: <574748E6-1EF0-4F5A-A1D6-338BAA711040@freeswitch.org> Can you get me a sample of the recording to listen to? /b On Feb 3, 2009, at 2:26 PM, e schmidbauer wrote: > im using the latest svn of freeswitch > > On Tue, Feb 3, 2009 at 12:23 PM, Brian West > wrote: > You forgot to tell us what revision of the code you're on? > /b > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090203/c47c0ff0/attachment.html From e.schmidbauer at gmail.com Tue Feb 3 12:59:51 2009 From: e.schmidbauer at gmail.com (e schmidbauer) Date: Tue, 3 Feb 2009 15:59:51 -0500 Subject: [Freeswitch-users] shoutcast skips In-Reply-To: <574748E6-1EF0-4F5A-A1D6-338BAA711040@freeswitch.org> References: <2cef777b0902030916s77339375pabc3d3c6fd9aec1a@mail.gmail.com> <898222CF-3335-4E2D-AE79-E57CFEE7EFB8@freeswitch.org> <2cef777b0902031226l72f4a4ceia80ec00721456602@mail.gmail.com> <574748E6-1EF0-4F5A-A1D6-338BAA711040@freeswitch.org> Message-ID: <2cef777b0902031259t6210b7b2w9873ced0806b98b3@mail.gmail.com> FreeSWITCH Version 1.0.trunk (11567) check out these sample recordings http://bwrl.org/recordings/2009-01-31-12-07-49.mp3 http://bwrl.org/recordings/2009-01-31-12-07-49.wav http://bwrl.org/recordings/test2.mp3 http://bwrl.org/recordings/test2.wav the conferences were recorded as wav files, i then converted them to mp3, both sound the same to me On Tue, Feb 3, 2009 at 3:51 PM, Brian West wrote: > Can you get me a sample of the recording to listen to? > /b > > On Feb 3, 2009, at 2:26 PM, e schmidbauer wrote: > > im using the latest svn of freeswitch > > On Tue, Feb 3, 2009 at 12:23 PM, Brian West wrote: > >> You forgot to tell us what revision of the code you're on? >> /b >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090203/06621b25/attachment.html From brian at freeswitch.org Tue Feb 3 13:21:05 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 3 Feb 2009 15:21:05 -0600 Subject: [Freeswitch-users] shoutcast skips In-Reply-To: <2cef777b0902031259t6210b7b2w9873ced0806b98b3@mail.gmail.com> References: <2cef777b0902030916s77339375pabc3d3c6fd9aec1a@mail.gmail.com> <898222CF-3335-4E2D-AE79-E57CFEE7EFB8@freeswitch.org> <2cef777b0902031226l72f4a4ceia80ec00721456602@mail.gmail.com> <574748E6-1EF0-4F5A-A1D6-338BAA711040@freeswitch.org> <2cef777b0902031259t6210b7b2w9873ced0806b98b3@mail.gmail.com> Message-ID: <81F41092-76A7-4DCD-8274-9584177F8752@freeswitch.org> You're doing distributed radio right? So callers are calling in with CELT from all over the place? Can you contact us on IRC because we are very interested in debugging this issue. You can get us on IRC #freeswitch on irc.freenode.net Thanks, /b On Feb 3, 2009, at 2:59 PM, e schmidbauer wrote: > FreeSWITCH Version 1.0.trunk (11567) > check out these sample recordings > http://bwrl.org/recordings/2009-01-31-12-07-49.mp3 > http://bwrl.org/recordings/2009-01-31-12-07-49.wav > http://bwrl.org/recordings/test2.mp3 > http://bwrl.org/recordings/test2.wav > > the conferences were recorded as wav files, i then converted them to > mp3, both sound the same to me -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090203/35c1f885/attachment.html From e.schmidbauer at gmail.com Tue Feb 3 13:27:42 2009 From: e.schmidbauer at gmail.com (e schmidbauer) Date: Tue, 3 Feb 2009 16:27:42 -0500 Subject: [Freeswitch-users] shoutcast skips In-Reply-To: <81F41092-76A7-4DCD-8274-9584177F8752@freeswitch.org> References: <2cef777b0902030916s77339375pabc3d3c6fd9aec1a@mail.gmail.com> <898222CF-3335-4E2D-AE79-E57CFEE7EFB8@freeswitch.org> <2cef777b0902031226l72f4a4ceia80ec00721456602@mail.gmail.com> <574748E6-1EF0-4F5A-A1D6-338BAA711040@freeswitch.org> <2cef777b0902031259t6210b7b2w9873ced0806b98b3@mail.gmail.com> <81F41092-76A7-4DCD-8274-9584177F8752@freeswitch.org> Message-ID: <2cef777b0902031327v4725f174kcaf976c04926fb21@mail.gmail.com> We are attempting distributed radio. We plan on having the hosts of the shows join the conference using CELT. But callers to the show would be joining using regular phones therefore using lower end codecs. I will be in the IRC shortly. On Tue, Feb 3, 2009 at 4:21 PM, Brian West wrote: > You're doing distributed radio right? So callers are calling in with CELT > from all over the place? Can you contact us on IRC because we are very > interested in debugging this issue. > You can get us on IRC #freeswitch on irc.freenode.net > > Thanks, > > /b > > On Feb 3, 2009, at 2:59 PM, e schmidbauer wrote: > > FreeSWITCH Version 1.0.trunk (11567) > check out these sample recordings > http://bwrl.org/recordings/2009-01-31-12-07-49.mp3 > http://bwrl.org/recordings/2009-01-31-12-07-49.wav > http://bwrl.org/recordings/test2.mp3 > http://bwrl.org/recordings/test2.wav > > the conferences were recorded as wav files, i then converted them to mp3, > both sound the same to me > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090203/b0bea706/attachment.html From kokoska.rokoska at post.cz Tue Feb 3 14:11:58 2009 From: kokoska.rokoska at post.cz (kokoska.rokoska) Date: Tue, 03 Feb 2009 23:11:58 +0100 Subject: [Freeswitch-users] FreeSwitch setup as a "Dumb" SBC In-Reply-To: References: Message-ID: <4988C12E.1090109@post.cz> Ken Rice napsal(a): ... > On Registrations we have experienced Registration/second rates exceeding 150 > registrations per second using mod_xml_curl to feed the users directory. I > suspect, this number can be greatly increased if we were to feed directory > with something that cut out the apache and php over head > If someone interested I have few numbers on Registrar performance: DB server: 2x Quad core E5345 @ 2.33GHz, 16 GiB RAM Centos 5 x86_64, MySQL 5.0 Registrar server: 2x Quad core E5345 @ 2.33GHz, 16 GiB RAM Centos 5 x86_64 Tested using sipp with 10.000 and 30.000 "users". FreeSWITCH as registrar - current trunk: 1. FreeSwitch si simply modified (code doing NAT-ping is commented out :-) 2. Directory is served through lighttpd and simple "C" binary doing one trivial select. Lighttpd runs on the same machine as FS. When I move lighhtpd to another machine, I cannot see any significat performance boost. Result: I can go up to the 470-500 reg/s. and FS is heavy overloaded and retransmissions occurs. Kamailio as registrar - 1.4.3. no TLS: 1. Kamailio runs with usrloc db_mode 3 (no caching) Result: I can go up to the 3500-3700 reg/s. and Kamailio server is at 0.3 load and all 8 cores are bellow 15 %. Without retransmissions. The limit is DB throughput. Just for "curiosity" I switched userloc to db_mode 2 (write back) and at 5000 regs/s I stopped the sipp test, because I saw the bottle neck becomes the server runnig sipp (very old P4 box). Conclusion: While I see amazing FreeSWITCH performance on INVITEs per seconds and concurrent calls (another galaxy from * point of view :-), if you have to handle lots of registrations per second, it is IMO better to use Kamailio/OpenSIPS/SER as separate registrar and "propagate" users to FS through SQL view. Hope this helps someone... Best regards, kokoska.rokoska From anthony.minessale at gmail.com Tue Feb 3 14:34:38 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 3 Feb 2009 16:34:38 -0600 Subject: [Freeswitch-users] FreeSwitch setup as a "Dumb" SBC In-Reply-To: <4988C12E.1090109@post.cz> References: <4988C12E.1090109@post.cz> Message-ID: <191c3a030902031434v1129d684j18b09a954d871f7b@mail.gmail.com> What does it look like if you serve the directory from the static xml file out of curiosity. On Tue, Feb 3, 2009 at 4:11 PM, kokoska.rokoska wrote: > Ken Rice napsal(a): > ... > > > On Registrations we have experienced Registration/second rates exceeding > 150 > > registrations per second using mod_xml_curl to feed the users directory. > I > > suspect, this number can be greatly increased if we were to feed > directory > > with something that cut out the apache and php over head > > > > If someone interested I have few numbers on Registrar performance: > > DB server: > 2x Quad core E5345 @ 2.33GHz, 16 GiB RAM > Centos 5 x86_64, MySQL 5.0 > > Registrar server: > 2x Quad core E5345 @ 2.33GHz, 16 GiB RAM > Centos 5 x86_64 > > Tested using sipp with 10.000 and 30.000 "users". > > > FreeSWITCH as registrar - current trunk: > 1. FreeSwitch si simply modified (code doing NAT-ping is commented out :-) > 2. Directory is served through lighttpd and simple "C" binary doing one > trivial select. Lighttpd runs on the same machine as FS. When I move > lighhtpd to another machine, I cannot see any significat performance boost. > > Result: I can go up to the 470-500 reg/s. and FS is heavy overloaded and > retransmissions occurs. > > > Kamailio as registrar - 1.4.3. no TLS: > 1. Kamailio runs with usrloc db_mode 3 (no caching) > > Result: I can go up to the 3500-3700 reg/s. and Kamailio server is at > 0.3 load and all 8 cores are bellow 15 %. Without retransmissions. The > limit is DB throughput. > Just for "curiosity" I switched userloc to db_mode 2 (write back) and at > 5000 regs/s I stopped the sipp test, because I saw the bottle neck > becomes the server runnig sipp (very old P4 box). > > > Conclusion: > While I see amazing FreeSWITCH performance on INVITEs per seconds and > concurrent calls (another galaxy from * point of view :-), if you have > to handle lots of registrations per second, it is IMO better to use > Kamailio/OpenSIPS/SER as separate registrar and "propagate" users to FS > through SQL view. > > Hope this helps someone... > > Best regards, > > kokoska.rokoska > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090203/3676ff23/attachment-0001.html From kokoska.rokoska at post.cz Tue Feb 3 14:51:00 2009 From: kokoska.rokoska at post.cz (kokoska.rokoska) Date: Tue, 03 Feb 2009 23:51:00 +0100 Subject: [Freeswitch-users] FreeSwitch setup as a "Dumb" SBC In-Reply-To: <191c3a030902031434v1129d684j18b09a954d871f7b@mail.gmail.com> References: <4988C12E.1090109@post.cz> <191c3a030902031434v1129d684j18b09a954d871f7b@mail.gmail.com> Message-ID: <4988CA54.4080003@post.cz> Anthony Minessale napsal(a): > What does it look like if you serve the directory from the static xml > file out of curiosity. > Good question :-) I have never thing about it, becasue I need "dynamic" users. But it should show up very impressive number :-) I'll try it tommorow (here is midnight) and let you know. BTW: I try to find some another server in colocation with higher performace. With mentioned P4 I'm affraid have no chance to stress FS with static xml directory... Thank you for your interest, Anthony! Best regards, kokoska.rokoska > On Tue, Feb 3, 2009 at 4:11 PM, kokoska.rokoska > wrote: > > Ken Rice napsal(a): > ... > > > On Registrations we have experienced Registration/second rates > exceeding 150 > > registrations per second using mod_xml_curl to feed the users > directory. I > > suspect, this number can be greatly increased if we were to feed > directory > > with something that cut out the apache and php over head > > > > If someone interested I have few numbers on Registrar performance: > > DB server: > 2x Quad core E5345 @ 2.33GHz, 16 GiB RAM > Centos 5 x86_64, MySQL 5.0 > > Registrar server: > 2x Quad core E5345 @ 2.33GHz, 16 GiB RAM > Centos 5 x86_64 > > Tested using sipp with 10.000 and 30.000 "users". > > > FreeSWITCH as registrar - current trunk: > 1. FreeSwitch si simply modified (code doing NAT-ping is commented > out :-) > 2. Directory is served through lighttpd and simple "C" binary doing one > trivial select. Lighttpd runs on the same machine as FS. When I move > lighhtpd to another machine, I cannot see any significat performance > boost. > > Result: I can go up to the 470-500 reg/s. and FS is heavy overloaded and > retransmissions occurs. > > > Kamailio as registrar - 1.4.3. no TLS: > 1. Kamailio runs with usrloc db_mode 3 (no caching) > > Result: I can go up to the 3500-3700 reg/s. and Kamailio server is at > 0.3 load and all 8 cores are bellow 15 %. Without retransmissions. The > limit is DB throughput. > Just for "curiosity" I switched userloc to db_mode 2 (write back) and at > 5000 regs/s I stopped the sipp test, because I saw the bottle neck > becomes the server runnig sipp (very old P4 box). > > > Conclusion: > While I see amazing FreeSWITCH performance on INVITEs per seconds and > concurrent calls (another galaxy from * point of view :-), if you have > to handle lots of registrations per second, it is IMO better to use > Kamailio/OpenSIPS/SER as separate registrar and "propagate" users to FS > through SQL view. > > Hope this helps someone... > > Best regards, > > kokoska.rokoska > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From krice at freeswitch.org Tue Feb 3 14:56:17 2009 From: krice at freeswitch.org (Ken Rice) Date: Tue, 03 Feb 2009 16:56:17 -0600 Subject: [Freeswitch-users] FreeSwitch setup as a "Dumb" SBC In-Reply-To: <191c3a030902031434v1129d684j18b09a954d871f7b@mail.gmail.com> Message-ID: Never tried hah... From: Anthony Minessale Reply-To: Date: Tue, 3 Feb 2009 16:34:38 -0600 To: Subject: Re: [Freeswitch-users] FreeSwitch setup as a "Dumb" SBC What does it look like if you serve the directory from the static xml file out of curiosity. On Tue, Feb 3, 2009 at 4:11 PM, kokoska.rokoska wrote: > Ken Rice napsal(a): > ... > >> > On Registrations we have experienced Registration/second rates exceeding >> 150 >> > registrations per second using mod_xml_curl to feed the users directory. I >> > suspect, this number can be greatly increased if we were to feed directory >> > with something that cut out the apache and php over head >> > > > If someone interested I have few numbers on Registrar performance: > > DB server: > 2x Quad core E5345 @ 2.33GHz, 16 GiB RAM > Centos 5 x86_64, MySQL 5.0 > > Registrar server: > 2x Quad core E5345 @ 2.33GHz, 16 GiB RAM > Centos 5 x86_64 > > Tested using sipp with 10.000 and 30.000 "users". > > > FreeSWITCH as registrar - current trunk: > 1. FreeSwitch si simply modified (code doing NAT-ping is commented out :-) > 2. Directory is served through lighttpd and simple "C" binary doing one > trivial select. Lighttpd runs on the same machine as FS. When I move > lighhtpd to another machine, I cannot see any significat performance boost. > > Result: I can go up to the 470-500 reg/s. and FS is heavy overloaded and > retransmissions occurs. > > > Kamailio as registrar - 1.4.3. no TLS: > 1. Kamailio runs with usrloc db_mode 3 (no caching) > > Result: I can go up to the 3500-3700 reg/s. and Kamailio server is at > 0.3 load and all 8 cores are bellow 15 %. Without retransmissions. The > limit is DB throughput. > Just for "curiosity" I switched userloc to db_mode 2 (write back) and at > 5000 regs/s I stopped the sipp test, because I saw the bottle neck > becomes the server runnig sipp (very old P4 box). > > > Conclusion: > While I see amazing FreeSWITCH performance on INVITEs per seconds and > concurrent calls (another galaxy from * point of view :-), if you have > to handle lots of registrations per second, it is IMO better to use > Kamailio/OpenSIPS/SER as separate registrar and "propagate" users to FS > through SQL view. > > Hope this helps someone... > > Best regards, > > kokoska.rokoska > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090203/79c8824b/attachment.html From Daniell at airg.com Tue Feb 3 18:21:07 2009 From: Daniell at airg.com (Daniel Liang) Date: Tue, 3 Feb 2009 18:21:07 -0800 Subject: [Freeswitch-users] Recording background music and voice is out of sync In-Reply-To: <022001c9862f$efd4b7d0$cf7e2770$@net> References: <019501c985ac$4f00ee60$ed02cb20$@net><4987E527.1040909@laposte.net> <022001c9862f$efd4b7d0$cf7e2770$@net> Message-ID: <0B02E756F603CC409EB553879B090CC80A23EB2F@HPEXCHVS01.exchange.airg> Hi, I was trying to record a background music with a user's voice at the same time. I did a playback and started recording. But the recorded user's voice and the background music is about 0.5 second out of sync. I also tried to use uuid_displace instead of playback, but I got the same result. I guess it was the transfer delay between freeswitch and the end user. Is there a way to avoid that? One of the solution that I can think of is to route the background music to the end user and then route it back to freeswitch and let freeswitch recorded user's voice and the routed music together. But I don't know how I can do that in freeswitch. Any idea? Thanks. Daniel From brian at freeswitch.org Tue Feb 3 18:35:42 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 3 Feb 2009 20:35:42 -0600 Subject: [Freeswitch-users] Recording background music and voice is out of sync In-Reply-To: <0B02E756F603CC409EB553879B090CC80A23EB2F@HPEXCHVS01.exchange.airg> References: <019501c985ac$4f00ee60$ed02cb20$@net><4987E527.1040909@laposte.net> <022001c9862f$efd4b7d0$cf7e2770$@net> <0B02E756F603CC409EB553879B090CC80A23EB2F@HPEXCHVS01.exchange.airg> Message-ID: <341AE5F8-20B2-4CF3-92EE-7311B3E71C7E@freeswitch.org> Can you show us an example of how you're doing this? Playback and Record aren't async so you'll need to show us how you're doing this. Also don't hijack threads you hit replay on the one "Re: [Freeswitch- users] FreeSwitch setup as a "Dumb" SBC" as the subject, deleted the subject and started a new body. That hijacks the thread and that can cause your problem to go ignored in some cases if people aren't interested in the thread topic depending on how their reader threads the emails. Please click new message and type freeswitch- users at lists.freeswitch.org in and then input your subject and body to start a new thread. Thanks, Brian West FreeSWITCH.org On Feb 3, 2009, at 8:21 PM, Daniel Liang wrote: > Hi, > > I was trying to record a background music with a user's voice at the > same time. I did a playback and started recording. But the recorded > user's voice and the background music is about 0.5 second out of > sync. I > also tried to use uuid_displace instead of playback, but I got the > same > result. From kawarod at laposte.net Tue Feb 3 22:35:37 2009 From: kawarod at laposte.net (rod) Date: Wed, 04 Feb 2009 10:35:37 +0400 Subject: [Freeswitch-users] FreeSwitch setup as a "Dumb" SBC In-Reply-To: <191c3a030902031434v1129d684j18b09a954d871f7b@mail.gmail.com> References: <4988C12E.1090109@post.cz> <191c3a030902031434v1129d684j18b09a954d871f7b@mail.gmail.com> Message-ID: <49893739.1090400@laposte.net> I did the test when I start looking at FS. With 10 000 files in conf/directory/default mounted as a ramdisk (if not in Ramdisk, the IO are too high) and an intel quad core q9550 (2.83Ghz) with 4GB RAM and the db also in Ramdisk, I was stuck at approx 150cps with a very high CPU usage. The version I used was 1.0.1, but not sure. Anthony Minessale wrote: > What does it look like if you serve the directory from the static xml > file out of curiosity. > > > On Tue, Feb 3, 2009 at 4:11 PM, kokoska.rokoska > > wrote: > > Ken Rice napsal(a): > ... > > > On Registrations we have experienced Registration/second rates > exceeding 150 > > registrations per second using mod_xml_curl to feed the users > directory. I > > suspect, this number can be greatly increased if we were to feed > directory > > with something that cut out the apache and php over head > > > > If someone interested I have few numbers on Registrar performance: > > DB server: > 2x Quad core E5345 @ 2.33GHz, 16 GiB RAM > Centos 5 x86_64, MySQL 5.0 > > Registrar server: > 2x Quad core E5345 @ 2.33GHz, 16 GiB RAM > Centos 5 x86_64 > > Tested using sipp with 10.000 and 30.000 "users". > > > FreeSWITCH as registrar - current trunk: > 1. FreeSwitch si simply modified (code doing NAT-ping is commented > out :-) > 2. Directory is served through lighttpd and simple "C" binary > doing one > trivial select. Lighttpd runs on the same machine as FS. When I move > lighhtpd to another machine, I cannot see any significat > performance boost. > > Result: I can go up to the 470-500 reg/s. and FS is heavy > overloaded and > retransmissions occurs. > > > Kamailio as registrar - 1.4.3. no TLS: > 1. Kamailio runs with usrloc db_mode 3 (no caching) > > Result: I can go up to the 3500-3700 reg/s. and Kamailio server is at > 0.3 load and all 8 cores are bellow 15 %. Without retransmissions. The > limit is DB throughput. > Just for "curiosity" I switched userloc to db_mode 2 (write back) > and at > 5000 regs/s I stopped the sipp test, because I saw the bottle neck > becomes the server runnig sipp (very old P4 box). > > > Conclusion: > While I see amazing FreeSWITCH performance on INVITEs per seconds and > concurrent calls (another galaxy from * point of view :-), if you have > to handle lots of registrations per second, it is IMO better to use > Kamailio/OpenSIPS/SER as separate registrar and "propagate" users > to FS > through SQL view. > > Hope this helps someone... > > Best regards, > > kokoska.rokoska > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From pmhshz at gmail.com Tue Feb 3 22:38:24 2009 From: pmhshz at gmail.com (shehzad p) Date: Tue, 3 Feb 2009 22:38:24 -0800 (PST) Subject: [Freeswitch-users] Freeswitch freezes on increasing call traffic In-Reply-To: <191c3a030901310850s553f8c11oc4e559a710def6d2@mail.gmail.com> References: <21701744.post@talk.nabble.com> <191c3a030901280601t3e185ec2m76295110e4888837@mail.gmail.com> <21729863.post@talk.nabble.com> <191c3a030901290831y2ff329b3maee6c1aafa3c68e2@mail.gmail.com> <21745312.post@talk.nabble.com> <21749375.post@talk.nabble.com> <191c3a030901300818i5d85a7aawa0c53b4e54892f40@mail.gmail.com> <21761523.post@talk.nabble.com> <191c3a030901310850s553f8c11oc4e559a710def6d2@mail.gmail.com> Message-ID: <21825226.post@talk.nabble.com> Hi anthony, I Modified the whole architecture of call routing system, Now after getting required routes, script exit and, control comes back to Dialplan, and call is bridged there, And call hangup, CDR is posted to cdr.php file (using xml_cdr). So now there is no blocking statement (bridge or anything like that) in current javascript, It return back control instantly. So, setting up all above architecture... First I tested FS 1.0.1 , It get crashed two times, in interval of 3 to 5 hours and simultaneous call of about 100 to 150. BT is the same as before... http://www.nabble.com/file/p21825226/bt_new_arch.txt bt_new_arch.txt Now I am also testing 1.0.3RC1, and post it back if any found. Thanks msp Clearly you have an issue with your javascript code. You have the Garbage collector blocking in every thread. Are you doing any endless loops in your code where you do not check session.ready() as a condition for continuing the script? any time session.ready() fails you must immediately exit. Are you using session.execute to execute long blocking operations like bridging many calls or entering a conference? You should avoid doing this as all the collective scripts on the system share a common Garbage Collector provided by the JS engine and it can lead to the exact issues you describe if the code is not properly designed. What else does you script do that are things provided by FS such as playing files and executing applications. -- View this message in context: http://www.nabble.com/Freeswitch-freezes-on-increasing-call-traffic-tp21701744p21825226.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From kawarod at laposte.net Tue Feb 3 22:59:48 2009 From: kawarod at laposte.net (rod) Date: Wed, 04 Feb 2009 10:59:48 +0400 Subject: [Freeswitch-users] FreeSwitch setup as a "Dumb" SBC In-Reply-To: <022001c9862f$efd4b7d0$cf7e2770$@net> References: <019501c985ac$4f00ee60$ed02cb20$@net> <4987E527.1040909@laposte.net> <022001c9862f$efd4b7d0$cf7e2770$@net> Message-ID: <49893CE4.7080300@laposte.net> Hi Adam, I detailed a bit more my previous mail on this page: http://wiki.freeswitch.org/wiki/SBC_Setup Round robin is managed by the carrierroute module. Carrierroute will reply based on the probability you defined for a route, so if you define 0.3 and 0.7 for the same prefix, your traffic will point to 2 different gateways with a probability of 30% for one and 70% for the others (If I understood well the behaviour of carrierroute). For forking, what I do is that carrierroute replies with a code and not an IP address. This code, is then used as a condition in FS and the dialplan matched could then propose serial or parallel forking (in the wiki, I detailed serial forking). The idea is that you could define many combination of GWs, eg: - code01: try IP_A then IP_B (serial) - code02: try IP_B then IP_A (serial) - code03: try IP_A and IP_C (parallel) this setup is working for me as I do not have 1000 of GWs but I need a big routing table (approx 160000). I'm sure it could be possible to use the failure route functionnality of carrierroute to define a new route when the first one failed without having to define code. The drawback of this method is that you can't define metrics/properties for a route (quality, cost, fax compliance...) in realtime, and this is where using/enhancing the native FS module mod_lcr could be better (I have no idea on how mod_lcr performs, I will give it a try). rod Adam Long wrote: > Hi Rod, > > Great info, Thanks! > Glad to see others are interested in the same concept. > My reasons for SER as routing core and implementation is slightly different > yet similar. > > I like your Redirect model, with that you are truly using your Kamailio as > route server only. I would imagine very scalable. > - Are you able to do any round robin, serial or parallel forking > with this? > - I wonder if multiple Contacts in the 302 response maybe with some > logic in FreeSwitch dialplan? > If so I think your design is a bit more efficient than mine as it keeps SER > out of the call path. > > My design is little different.. it is more of a "Stateful" setup. With SER > staying in call path and FreeSwitch at Edge. > I do this to enable Serial Forking to a series of SBCs (FreeSwitch) geo > distributed, when one of the branches is congested it > forks to the next SBC (route). > > The FreeSwitch guys are probably right tho... with mod_easyroute and mod_lcr > we could probably implement all of this in FreeSwitch without SER. > I would be curious to know if anyone is doing something similar at high > volumes and what sort of concurrency and cps they > are able to achieve. > > I am a Perl and C# guy, I thought about implementing a mod_manged_lcr with > memcached support. > Memcache support would prob boost the scalability by a factor of 10 at > least. > > I will let you know if I end up developing a high performance FreeSwitch > route module. > Right now I use memcache in a OpenSIPS perl script for my route caching and > its incredibly fast > and clusters well. > > It actually might be easier to add memcached support to mod_lcr and > mod_easyroute but im not real strong in C/C++ > > I'll jump on IRC later and chat with some of the experts on this as I know > memcache has been discussed before. > I'd be curious to know if any progress has been made there already. > > > Regards, > -Adam > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of rod > Sent: Tuesday, February 03, 2009 1:33 AM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] FreeSwitch setup as a "Dumb" SBC > > Hi Adam, > > I'm in the process of using FS as a SBC. For the route lookup, I do it > using OpenSER carrierroute, without having to flow through > SBC---Openser---SBC. I'm using carrierroute at this time cause I need > more than 200 000 routing entries and carrierroute has been tested with > twice this number. > > Here is the setup: > > - install openser and carrierroute and make openser listening on > 127.0.0.1:5062 (for example) on your SBC > - populate carrierroute table > > What I do to use carrierroute module from FS is to use a specific > X-header (X-LOOKUP). > > In the dialplan, in the default context, I have something like this: > > > > > > data="sofia/internal/${sip_req_user}@127.0.0.1:5062"/> > data="sip_h_X-ROUTE=${sip_redirect_contact_host_0}"/> > > > > > The process is simple: > the export "sip_h_X-ROUTE=LOOKUP" had a sip header X-ROUTE=LOOKUP > then I bridge the call to 127.0.0.1:5062 (openser process) > > In openser I have a route block that checks the presence of header > LOOKUP and openser sends a "604: unable to route call" if the prefix is > not found, or a "302: with the IP of the gateway found" > > In FS, you can get the IP using the variable > "${sip_redirect_contact_host_0}". Then I transfer this to the context > ROUTING, where the check condition is based on the LOOKUP header that > has been rewritten with this variable. > > I will document all this setup (installation of openser/carrierroute and > config file of FS and openser) on a wiki page I start writing yesterday, > so please be indulgent and patient. > The next step is to test the scalability of this. > > I'm a very bad programmer, so that's the only way for me to contribute > to FS, and as I see many people interested for an SBC setup, I think it > could be great if we share our work/knowlegde. > > The wiki page is there: > http://wiki.freeswitch.org/wiki/SBC_Setup > > regards, > rod. > > > > > > Adam Long wrote: > >> Hi Guys, >> >> I've been working at setting up a couple of FreeSwitch nodes as a >> topology hiding SBCs that handles both ingress traffic from my >> >> providers/peers and pass traffic up to an openser router that then >> routes call across the cluster of SBCs through which they reach the >> destination. >> >> I have OpenSIPS/SER setup doing DB route lookups and ENUM with >> LCR/Serial forking etc. >> >> My question is what would be the best way to send a call out to a >> destination choosen by the OpenSER router? >> >> For example: >> >> SIP Provider -- > SBC --- > OpenSER ---- ( route lookup returns >> 123.123.123.4 as dest ) -- > SBC --- > 123.123.123.4 >> >> I was thinking something along the lines of adding a "X-Route-To: >> +1NXXNXXXXXX@ 123.123.123.4" with openser >> >> and then something like this in the SBC. >> >> >> >> >> >> >> >> >> >> >> >> Is this a wise approach, is there anything I could do to do this better? >> >> I'd like to keep the logic in the SBCs as simple as possible. >> >> I am pretty familiar with SIP but my knowledge fades when it gets into >> the nitty gritty of routing. ie the Contact: and Via: headers >> >> and all that good stuff. >> >> I should also state I have two profiles defined one for the >> internal/private "core" network and one for the outside "external" >> network. >> >> Any thoughts on this at all would be greatly appreciated. >> >> Am I missing something in the SIP spec that would allow for this is a >> standardized way? >> >> Regards, >> >> -Adam >> >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > From kawarod at laposte.net Tue Feb 3 23:09:52 2009 From: kawarod at laposte.net (rod) Date: Wed, 04 Feb 2009 11:09:52 +0400 Subject: [Freeswitch-users] FreeSwitch setup as a "Dumb" SBC In-Reply-To: <022001c9862f$efd4b7d0$cf7e2770$@net> References: <019501c985ac$4f00ee60$ed02cb20$@net> <4987E527.1040909@laposte.net> <022001c9862f$efd4b7d0$cf7e2770$@net> Message-ID: <49893F40.9040904@laposte.net> One more thing, I worked on a setup like yours: - Kamailio as a registrar that do the routing decision - FS as a SBC What you have to do is just append an header with Kamailio and send the invite to your FS server using something like that (use of pseudo variables in Kamailio): #------------------------------------------- # PREPARE ROUTING USING REWRITING OF DOMAIN #------------------------------------------- if (is_method("INVITE") && from_uri==myself && src_ip!=10.10.10.254){ if(!cr_route("default", "0", "$rU", "$rU", "call_id")){ xlog("$ci CALLEE ROUTING FAILED: no route found"); sl_send_reply("604", "Unable to route this call"); exit; } else { xlog("$ci Route found for $rU via $rd"); } } # ----------------------------------------------------------------- # Route to FREESWITCH using domain rewriting applied above for LCR # ----------------------------------------------------------------- xlog("$ci ROUTE: $rd"); append_hf("X-ROUTE: $rd\r\n"); rewritehostport("10.10.10.254:5062"); # there you have to distribute the invite to your FS servers, take a look at the dispatcher module Using that, the FS server receiving the Invite, just need to parse the X-ROUTE header and route the call, without having to resend the call to a Kamailio server. I think you can adapt this scenario to your perl script using variable exportation and append_hf function. rod. Adam Long wrote: > Hi Rod, > > Great info, Thanks! > Glad to see others are interested in the same concept. > My reasons for SER as routing core and implementation is slightly different > yet similar. > > I like your Redirect model, with that you are truly using your Kamailio as > route server only. I would imagine very scalable. > - Are you able to do any round robin, serial or parallel forking > with this? > - I wonder if multiple Contacts in the 302 response maybe with some > logic in FreeSwitch dialplan? > If so I think your design is a bit more efficient than mine as it keeps SER > out of the call path. > > My design is little different.. it is more of a "Stateful" setup. With SER > staying in call path and FreeSwitch at Edge. > I do this to enable Serial Forking to a series of SBCs (FreeSwitch) geo > distributed, when one of the branches is congested it > forks to the next SBC (route). > > The FreeSwitch guys are probably right tho... with mod_easyroute and mod_lcr > we could probably implement all of this in FreeSwitch without SER. > I would be curious to know if anyone is doing something similar at high > volumes and what sort of concurrency and cps they > are able to achieve. > > I am a Perl and C# guy, I thought about implementing a mod_manged_lcr with > memcached support. > Memcache support would prob boost the scalability by a factor of 10 at > least. > > I will let you know if I end up developing a high performance FreeSwitch > route module. > Right now I use memcache in a OpenSIPS perl script for my route caching and > its incredibly fast > and clusters well. > > It actually might be easier to add memcached support to mod_lcr and > mod_easyroute but im not real strong in C/C++ > > I'll jump on IRC later and chat with some of the experts on this as I know > memcache has been discussed before. > I'd be curious to know if any progress has been made there already. > > > Regards, > -Adam > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of rod > Sent: Tuesday, February 03, 2009 1:33 AM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] FreeSwitch setup as a "Dumb" SBC > > Hi Adam, > > I'm in the process of using FS as a SBC. For the route lookup, I do it > using OpenSER carrierroute, without having to flow through > SBC---Openser---SBC. I'm using carrierroute at this time cause I need > more than 200 000 routing entries and carrierroute has been tested with > twice this number. > > Here is the setup: > > - install openser and carrierroute and make openser listening on > 127.0.0.1:5062 (for example) on your SBC > - populate carrierroute table > > What I do to use carrierroute module from FS is to use a specific > X-header (X-LOOKUP). > > In the dialplan, in the default context, I have something like this: > > > > > > data="sofia/internal/${sip_req_user}@127.0.0.1:5062"/> > data="sip_h_X-ROUTE=${sip_redirect_contact_host_0}"/> > > > > > The process is simple: > the export "sip_h_X-ROUTE=LOOKUP" had a sip header X-ROUTE=LOOKUP > then I bridge the call to 127.0.0.1:5062 (openser process) > > In openser I have a route block that checks the presence of header > LOOKUP and openser sends a "604: unable to route call" if the prefix is > not found, or a "302: with the IP of the gateway found" > > In FS, you can get the IP using the variable > "${sip_redirect_contact_host_0}". Then I transfer this to the context > ROUTING, where the check condition is based on the LOOKUP header that > has been rewritten with this variable. > > I will document all this setup (installation of openser/carrierroute and > config file of FS and openser) on a wiki page I start writing yesterday, > so please be indulgent and patient. > The next step is to test the scalability of this. > > I'm a very bad programmer, so that's the only way for me to contribute > to FS, and as I see many people interested for an SBC setup, I think it > could be great if we share our work/knowlegde. > > The wiki page is there: > http://wiki.freeswitch.org/wiki/SBC_Setup > > regards, > rod. > > > > > > Adam Long wrote: > >> Hi Guys, >> >> I've been working at setting up a couple of FreeSwitch nodes as a >> topology hiding SBCs that handles both ingress traffic from my >> >> providers/peers and pass traffic up to an openser router that then >> routes call across the cluster of SBCs through which they reach the >> destination. >> >> I have OpenSIPS/SER setup doing DB route lookups and ENUM with >> LCR/Serial forking etc. >> >> My question is what would be the best way to send a call out to a >> destination choosen by the OpenSER router? >> >> For example: >> >> SIP Provider -- > SBC --- > OpenSER ---- ( route lookup returns >> 123.123.123.4 as dest ) -- > SBC --- > 123.123.123.4 >> >> I was thinking something along the lines of adding a "X-Route-To: >> +1NXXNXXXXXX@ 123.123.123.4" with openser >> >> and then something like this in the SBC. >> >> >> >> >> >> >> >> >> >> >> >> Is this a wise approach, is there anything I could do to do this better? >> >> I'd like to keep the logic in the SBCs as simple as possible. >> >> I am pretty familiar with SIP but my knowledge fades when it gets into >> the nitty gritty of routing. ie the Contact: and Via: headers >> >> and all that good stuff. >> >> I should also state I have two profiles defined one for the >> internal/private "core" network and one for the outside "external" >> network. >> >> Any thoughts on this at all would be greatly appreciated. >> >> Am I missing something in the SIP spec that would allow for this is a >> standardized way? >> >> Regards, >> >> -Adam >> >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > From jsokulski at dotsystems.pl Wed Feb 4 00:51:34 2009 From: jsokulski at dotsystems.pl (Jacek Sokulski) Date: Wed, 04 Feb 2009 09:51:34 +0100 Subject: [Freeswitch-users] origainate through sofia gateway In-Reply-To: <1b46b4e80902030931y6a1679afw12ec37596ba9d77d@mail.gmail.com> References: <1233668179.5354.15.camel@dotw1126.dotsystems.pl> <1b46b4e80902030920p521d6079s4c9b78d8d39a924a@mail.gmail.com> <5E2B217D-8C85-48DB-800A-535A7FAAE24B@freeswitch.org> <1b46b4e80902030931y6a1679afw12ec37596ba9d77d@mail.gmail.com> Message-ID: <1233737494.5405.12.camel@dotw1126.dotsystems.pl> We have tried setting both effective_caller_id_number and origination_caller_id_number: session1.originate(session1,"{origination_caller_id_number=fixed0248b}sofia/gateway/halonet/0225490317",15); but the problem still exists. The solution we have found for the case when we originate two calls, local and external, is as follow: session1 = new Session(); session1.originate(session1,"user/1003 at 192.168.1.122",15);//local if(session1.ready()) { session1.execute("execute_extension","00930691688627 XML default");//external } so the external call goes through the dialplan. It does not work if both calls are external. One possible solution could be to pass the originating call through dialplan (loopback?) but we have not managed to figure out how to do it. Thanks Jacek Dnia 03-02-2009, wto o godzinie 14:31 -0300, Nicolas Brenner pisze: > Oops! Well, fortunately I don't use that voip provider anymore (nor the script). > > Thanks Brian. > > Nicolas > > On Tue, Feb 3, 2009 at 2:25 PM, Brian West wrote: > > YOU should NEVER use this method or call setCallerData at all you > > should use the correct methods to override the callerid. > > > > If its a B-Leg born from an A-Leg you use these on the on the A-Leg: > > > > http://wiki.freeswitch.org/wiki/Channel_Variables#effective_caller_id_name > > http://wiki.freeswitch.org/wiki/Channel_Variables#effective_caller_id_number > > > > If you're originating you use this: > > > > http://wiki.freeswitch.org/wiki/Channel_Variables#origination_caller_id_name > > http://wiki.freeswitch.org/wiki/Channel_Variables#origination_caller_id_number > > > > /b > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From kokoska.rokoska at post.cz Wed Feb 4 02:31:41 2009 From: kokoska.rokoska at post.cz (kokoska rokoska) Date: Wed, 04 Feb 2009 11:31:41 +0100 Subject: [Freeswitch-users] FreeSwitch setup as a "Dumb" SBC In-Reply-To: <191c3a030902031434v1129d684j18b09a954d871f7b@mail.gmail.com> References: <4988C12E.1090109@post.cz> <191c3a030902031434v1129d684j18b09a954d871f7b@mail.gmail.com> Message-ID: <49896E8D.3010609@post.cz> Anthony Minessale napsal(a): > What does it look like if you serve the directory from the static xml > file out of curiosity. > Well, I write all user infos into static xml files loaded at startup :-) For the first try (without "tuning", see below) I can't go beyond 220 reg/s - it is just about 10-12 % higher rate then with "lighty" etc. BTW: Lighty alone can serve about 2600-2700 "dynamic directory xml files from DB" per second - tested with "ab". The only difference (but big one :-) is that CPU utilization is below 15% on all cores and load is about 0.2, so machine is idle :-) It tells me, that somthing is wron with my setup :-) The only optimizations done are: 1. No logging 2. FS in "high priority" mode 3. "ulimits" applied Next (today evenings - tommorow mornings, don't know) I try to "tune" FS for better preformance - as I did with previous test last month: 1. Move FS internal SQL light to ramdisk - I'm not sure if it helps, because OS caches all HDD reads/writes, but if SQL light forces sync after every DB update/insert it can make sense - I try it. May be I move all FS dir to ramdisk. 2. Slightly change mod_sofia to disable NAT ping loop (unnecessary DB operations) and, mainly, disable retrieving and sending of NOTIFY messages containing VM info. I'll look into my notes to see what I have done before and do the same... Be patient please, as soon as I have the results, I post them here :-) Best regards, kokoska.rokoska > On Tue, Feb 3, 2009 at 4:11 PM, kokoska.rokoska > wrote: > > Ken Rice napsal(a): > ... > > > On Registrations we have experienced Registration/second rates > exceeding 150 > > registrations per second using mod_xml_curl to feed the users > directory. I > > suspect, this number can be greatly increased if we were to feed > directory > > with something that cut out the apache and php over head > > > > If someone interested I have few numbers on Registrar performance: > > DB server: > 2x Quad core E5345 @ 2.33GHz, 16 GiB RAM > Centos 5 x86_64, MySQL 5.0 > > Registrar server: > 2x Quad core E5345 @ 2.33GHz, 16 GiB RAM > Centos 5 x86_64 > > Tested using sipp with 10.000 and 30.000 "users". > > > FreeSWITCH as registrar - current trunk: > 1. FreeSwitch si simply modified (code doing NAT-ping is commented > out :-) > 2. Directory is served through lighttpd and simple "C" binary doing one > trivial select. Lighttpd runs on the same machine as FS. When I move > lighhtpd to another machine, I cannot see any significat performance > boost. > > Result: I can go up to the 470-500 reg/s. and FS is heavy overloaded and > retransmissions occurs. > > > Kamailio as registrar - 1.4.3. no TLS: > 1. Kamailio runs with usrloc db_mode 3 (no caching) > > Result: I can go up to the 3500-3700 reg/s. and Kamailio server is at > 0.3 load and all 8 cores are bellow 15 %. Without retransmissions. The > limit is DB throughput. > Just for "curiosity" I switched userloc to db_mode 2 (write back) and at > 5000 regs/s I stopped the sipp test, because I saw the bottle neck > becomes the server runnig sipp (very old P4 box). > > > Conclusion: > While I see amazing FreeSWITCH performance on INVITEs per seconds and > concurrent calls (another galaxy from * point of view :-), if you have > to handle lots of registrations per second, it is IMO better to use > Kamailio/OpenSIPS/SER as separate registrar and "propagate" users to FS > through SQL view. > > Hope this helps someone... > > Best regards, > > kokoska.rokoska > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From Claudio.Cavalera at italtel.it Wed Feb 4 03:03:12 2009 From: Claudio.Cavalera at italtel.it (Cavalera Claudio Luigi) Date: Wed, 4 Feb 2009 12:03:12 +0100 Subject: [Freeswitch-users] 16 threads didn't exit Message-ID: Hello list, I'm trying to track down a seg fault issue with a fs Revision: 11489 Here is the backtrace pastebin: http://pastebin.freeswitch.org/7009 but before digging the dump I would like to understand: am I the only one having error like this in fs console: "Error in my_thread_global_end(): 16 threads didn't exit" I'm asking this because googling around did not take me to much relation between this error and fs. In fact as you can see the error does not have the usual fs logging format with date time and logging level, it's just a yellow line printed out in console. It seems to me related on php and mysql from what I've read here http://forums.mysql.com/read.php?10,153077,206930#msg-206930 It's possible that my fs segfaults not because of this error at all, but I just wanted to inform you asap in case someone could face similar issues. I've enabled crash-protection in switch.conf.xml although I'm not sure what it does. Best Regards, Claudio Internet Email Confidentiality Footer ----------------------------------------------------------------------------------------------------- La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. ----------------------------------------------------------------------------------------------------- From sias at cpdata.co.za Wed Feb 4 03:49:55 2009 From: sias at cpdata.co.za (Sias Mey) Date: Wed, 4 Feb 2009 13:49:55 +0200 Subject: [Freeswitch-users] 16 threads didn't exit In-Reply-To: References: Message-ID: <20090204114955.GD6752@cpdata.co.za> I have seen that error myself, however I assumed it was due te me hanging up other cals from the api_hangup_hook of a related call. I use this to set a "master" call in a conference so that if it hangs up all calls in the conference hangup. On Wed, Feb 04, 2009 at 12:03:12PM +0100, Cavalera Claudio Luigi wrote: > Hello list, > I'm trying to track down a seg fault issue with a fs Revision: 11489 > Here is the backtrace pastebin: > http://pastebin.freeswitch.org/7009 > > but before digging the dump I would like to understand: am I the only > one having error like this in fs console: > "Error in my_thread_global_end(): 16 threads didn't exit" > > I'm asking this because googling around did not take me to much relation > between this error and fs. > In fact as you can see the error does not have the usual fs logging > format with date time and logging level, it's just a yellow line printed > out in console. > It seems to me related on php and mysql from what I've read here > http://forums.mysql.com/read.php?10,153077,206930#msg-206930 > > It's possible that my fs segfaults not because of this error at all, but > I just wanted to inform you asap in case someone could face similar > issues. > > I've enabled crash-protection in switch.conf.xml although I'm not sure > what it does. > > Best Regards, > Claudio > > > Internet Email Confidentiality Footer > ----------------------------------------------------------------------------------------------------- > La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. > > This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. > ----------------------------------------------------------------------------------------------------- > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From jonas.gauffin at gmail.com Wed Feb 4 04:26:04 2009 From: jonas.gauffin at gmail.com (Jonas Gauffin) Date: Wed, 4 Feb 2009 13:26:04 +0100 Subject: [Freeswitch-users] gateway Message-ID: Hello I'm trying to make outbound calls through my gateway provider. My calls got rejected and I asked them why. Apparently I need to use 5060 as source port, since they validate both my IP and the port that the messages come from. Is this possible with freeswitch? If so, what config settings should I set? Regards, Jonas -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090204/68095d5b/attachment.html From Claudio.Cavalera at italtel.it Wed Feb 4 05:28:48 2009 From: Claudio.Cavalera at italtel.it (Cavalera Claudio Luigi) Date: Wed, 4 Feb 2009 14:28:48 +0100 Subject: [Freeswitch-users] 16 threads didn't exit In-Reply-To: <20090204114955.GD6752@cpdata.co.za> Message-ID: freeswitch-users-bounces at lists.freeswitch.org wrote: > I have seen that error myself, however I assumed it was due te me > hanging up other cals from the api_hangup_hook of a related call. > > I use this to set a "master" call in a conference so that if > it hangs up > all calls in the conference hangup. > Thanks Sias for this info, if you are able to reproduce this message in a systematic way please let me know so that I can reproduce it myself. I've searched for the "threads didn't exit" message in fs code and I did not found it. I think it comes from mysql and I'm not sure it's even related to my segfault. Best regards, Claudio Internet Email Confidentiality Footer ----------------------------------------------------------------------------------------------------- La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. ----------------------------------------------------------------------------------------------------- From anthony.minessale at gmail.com Wed Feb 4 05:51:06 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 4 Feb 2009 07:51:06 -0600 Subject: [Freeswitch-users] Freeswitch freezes on increasing call traffic In-Reply-To: <21825226.post@talk.nabble.com> References: <21701744.post@talk.nabble.com> <191c3a030901280601t3e185ec2m76295110e4888837@mail.gmail.com> <21729863.post@talk.nabble.com> <191c3a030901290831y2ff329b3maee6c1aafa3c68e2@mail.gmail.com> <21745312.post@talk.nabble.com> <21749375.post@talk.nabble.com> <191c3a030901300818i5d85a7aawa0c53b4e54892f40@mail.gmail.com> <21761523.post@talk.nabble.com> <191c3a030901310850s553f8c11oc4e559a710def6d2@mail.gmail.com> <21825226.post@talk.nabble.com> Message-ID: <191c3a030902040551w5d867deta55c98bd3a3c03f8@mail.gmail.com> If you still get a crash on SVN trunk please post the bt even if you think it's the same, since it won't be exactly the same, the line numbers etc will be accurate with our development code making it easier to debug. On Wed, Feb 4, 2009 at 12:38 AM, shehzad p wrote: > > Hi anthony, > > I Modified the whole architecture of call routing system, > Now after getting required routes, script exit and, > control comes back to Dialplan, and call is bridged there, > And call hangup, CDR is posted to cdr.php file (using xml_cdr). > > So now there is no blocking statement (bridge or anything like that) in > current javascript, It return back control instantly. > > So, setting up all above architecture... > First I tested FS 1.0.1 , It get crashed two times, in interval of 3 to 5 > hours and simultaneous call of about 100 to 150. > BT is the same as before... > http://www.nabble.com/file/p21825226/bt_new_arch.txt bt_new_arch.txt > > Now I am also testing 1.0.3RC1, and post it back if any found. > > Thanks > msp > > > Clearly you have an issue with your javascript code. > > You have the Garbage collector blocking in every thread. > > Are you doing any endless loops in your code where you do not check > session.ready() as a condition for > continuing the script? > > any time session.ready() fails you must immediately exit. > > Are you using session.execute to execute long blocking operations like > bridging many calls or entering a conference? > You should avoid doing this as all the collective scripts on the system > share a common Garbage Collector provided by the > JS engine and it can lead to the exact issues you describe if the code is > not properly designed. > > What else does you script do that are things provided by FS such as playing > files and executing applications. > > > > > -- > View this message in context: > http://www.nabble.com/Freeswitch-freezes-on-increasing-call-traffic-tp21701744p21825226.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090204/deb1ce9a/attachment.html From anthony.minessale at gmail.com Wed Feb 4 06:09:07 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 4 Feb 2009 08:09:07 -0600 Subject: [Freeswitch-users] origainate through sofia gateway In-Reply-To: <1233737494.5405.12.camel@dotw1126.dotsystems.pl> References: <1233668179.5354.15.camel@dotw1126.dotsystems.pl> <1b46b4e80902030920p521d6079s4c9b78d8d39a924a@mail.gmail.com> <5E2B217D-8C85-48DB-800A-535A7FAAE24B@freeswitch.org> <1b46b4e80902030931y6a1679afw12ec37596ba9d77d@mail.gmail.com> <1233737494.5405.12.camel@dotw1126.dotsystems.pl> Message-ID: <191c3a030902040609s600798c8t720950a2e13ee05a@mail.gmail.com> Where did you learn how to use js this way? session.originate is being misused here and is depricated and may be removed. the first arg to session.originate is either undefined or a *different* session (the a leg) session1 = new Session(); session1.originate(undefined, "{ignore_early_media=true}user/ 1008 at 192.168.1.122"); session1.setVariable("effective_caller_id_number","fixed0248b"); //once you have session1 when you originate session2 you pass session1 as the arg // the effective_caller_id is taken from session1 session2 = new Session(); session2.originate(session1, "sofia/gateway/halonet/0225490317"); Anyway this whole code is depricated in favor of this: session1 = new Session("{ignore_early_media=true}user/1008 at 192.168.1.122"); if (session1.ready()) { session1.setVariable("effective_caller_id_number","fixed0248b"); session2 = new Session("sofia/gateway/halonet/0225490317", session1); } and could be further refactored down to this: session1 = new Session("{ignore_early_media=true}user/1008 at 192.168.1.122"); if (session1.ready()) { session1.setVariable("effective_caller_id_number","fixed0248b"); session1.execute("bridge", "sofia/gateway/halonet/0225490317"); } or down to this one line of code that will setup the call detached from the script and exit. var result = apiExecute("originate", "{effective_caller_id_number=fixed0248b,origination_caller_id_number=1000,ignore_early_media=true}user/ 1008 at 192.168.1.122 bridge:sofia/gateway/halonet/0225490317 inline"); if you dont care about the result and want to exit even before the call is completed. var result = apiExecute("bgapi", "originate {effective_caller_id_number=fixed0248b,origination_caller_id_number=1000,ignore_early_media=true}user/ 1008 at 192.168.1.122 bridge:sofia/gateway/halonet/0225490317 inline"); On Wed, Feb 4, 2009 at 2:51 AM, Jacek Sokulski wrote: > > We have tried setting both effective_caller_id_number and > origination_caller_id_number: > > > session1.originate(session1,"{origination_caller_id_number=fixed0248b}sofia/gateway/halonet/0225490317",15); > but the problem still exists. The solution we have found for the case > when we originate two calls, local and external, is as follow: > > session1 = new Session(); > session1.originate(session1,"user/1003 at 192.168.1.122",15);//local > if(session1.ready()) { > session1.execute("execute_extension","00930691688627 XML > default");//external > } > > so the external call goes through the dialplan. > It does not work if both calls are external. One possible solution could be > to pass the originating call through dialplan (loopback?) but we have not > managed > to figure out how to do it. > > Thanks > Jacek > > Dnia 03-02-2009, wto o godzinie 14:31 -0300, Nicolas Brenner pisze: > > Oops! Well, fortunately I don't use that voip provider anymore (nor the > script). > > > > Thanks Brian. > > > > Nicolas > > > > On Tue, Feb 3, 2009 at 2:25 PM, Brian West wrote: > > > YOU should NEVER use this method or call setCallerData at all you > > > should use the correct methods to override the callerid. > > > > > > If its a B-Leg born from an A-Leg you use these on the on the A-Leg: > > > > > > > http://wiki.freeswitch.org/wiki/Channel_Variables#effective_caller_id_name > > > > http://wiki.freeswitch.org/wiki/Channel_Variables#effective_caller_id_number > > > > > > If you're originating you use this: > > > > > > > http://wiki.freeswitch.org/wiki/Channel_Variables#origination_caller_id_name > > > > http://wiki.freeswitch.org/wiki/Channel_Variables#origination_caller_id_number > > > > > > /b > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090204/d81ea72e/attachment-0001.html From anthony.minessale at gmail.com Wed Feb 4 06:14:52 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 4 Feb 2009 08:14:52 -0600 Subject: [Freeswitch-users] gateway In-Reply-To: References: Message-ID: <191c3a030902040614q32c05f02le7e8f41dd70d1fc3@mail.gmail.com> if you are not behind any nat then as long as you run your profile on 5060, the source port on every packet will be 5060. If you *are* behind nat the nat mapping will pick a random port unless you have a firewall that allows you to set specific mappings. On Wed, Feb 4, 2009 at 6:26 AM, Jonas Gauffin wrote: > Hello > I'm trying to make outbound calls through my gateway provider. > My calls got rejected and I asked them why. > > Apparently I need to use 5060 as source port, since they validate both my > IP and the port that the messages come from. > Is this possible with freeswitch? If so, what config settings should I set? > > Regards, > Jonas > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090204/84ddc63c/attachment.html From Claudio.Cavalera at italtel.it Wed Feb 4 06:27:31 2009 From: Claudio.Cavalera at italtel.it (Cavalera Claudio Luigi) Date: Wed, 4 Feb 2009 15:27:31 +0100 Subject: [Freeswitch-users] Errors compiling trunk on a fresh system Message-ID: Hello, I'm trying to compile a brand new fs on a clean system. Revision: 11630 After the usual ./bootstrap.sh ./configure --enable-core-odbc-support I was getting this at make http://pastebin.freeswitch.org/7011 so i cd into utils under libs/sofia-sip/utils issued a doxygen -u but still getting this http://pastebin.freeswitch.org/7012 any hint on what could be? Best Regards, Claudio Internet Email Confidentiality Footer ----------------------------------------------------------------------------------------------------- La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. ----------------------------------------------------------------------------------------------------- From jsokulski at dotsystems.pl Wed Feb 4 06:46:07 2009 From: jsokulski at dotsystems.pl (Jacek Sokulski) Date: Wed, 04 Feb 2009 15:46:07 +0100 Subject: [Freeswitch-users] origainate through sofia gateway In-Reply-To: <191c3a030902040609s600798c8t720950a2e13ee05a@mail.gmail.com> References: <1233668179.5354.15.camel@dotw1126.dotsystems.pl> <1b46b4e80902030920p521d6079s4c9b78d8d39a924a@mail.gmail.com> <5E2B217D-8C85-48DB-800A-535A7FAAE24B@freeswitch.org> <1b46b4e80902030931y6a1679afw12ec37596ba9d77d@mail.gmail.com> <1233737494.5405.12.camel@dotw1126.dotsystems.pl> <191c3a030902040609s600798c8t720950a2e13ee05a@mail.gmail.com> Message-ID: <1233758767.5405.22.camel@dotw1126.dotsystems.pl> Thanks Anthony, the js snippets are very instructive. A couple of points: 1. The code with apiExecute does not work (local phone is connected, but after picking up it hungs up immediately), other examples are working fine. 2. It does not show how initiate external call without existing session. 3. How can one pass the call through dialplan? Jacek PS. we got the code probable from wiki or from this mialing list. Dnia 04-02-2009, ?ro o godzinie 08:09 -0600, Anthony Minessale pisze: > Where did you learn how to use js this way? > session.originate is being misused here and is depricated and may be > removed. > > the first arg to session.originate is either undefined or a > *different* session (the a leg) > > session1 = new Session(); > session1.originate(undefined, > "{ignore_early_media=true}user/1008 at 192.168.1.122"); > > session1.setVariable("effective_caller_id_number","fixed0248b"); > > //once you have session1 when you originate session2 you pass session1 > as the arg > // the effective_caller_id is taken from session1 > > session2 = new Session(); > session2.originate(session1, "sofia/gateway/halonet/0225490317"); > > Anyway this whole code is depricated in favor of this: > > session1 = new > Session("{ignore_early_media=true}user/1008 at 192.168.1.122"); > > if (session1.ready()) { > session1.setVariable("effective_caller_id_number","fixed0248b"); > session2 = new Session("sofia/gateway/halonet/0225490317", > session1); > } > > and could be further refactored down to this: > > session1 = new > Session("{ignore_early_media=true}user/1008 at 192.168.1.122"); > if (session1.ready()) { > session1.setVariable("effective_caller_id_number","fixed0248b"); > session1.execute("bridge", "sofia/gateway/halonet/0225490317"); > } > > or down to this one line of code that will setup the call detached > from the script and exit. > > var result = apiExecute("originate", > "{effective_caller_id_number=fixed0248b,origination_caller_id_number=1000,ignore_early_media=true}user/1008 at 192.168.1.122 bridge:sofia/gateway/halonet/0225490317 inline"); > > if you dont care about the result and want to exit even before the > call is completed. > > var result = apiExecute("bgapi", "originate > {effective_caller_id_number=fixed0248b,origination_caller_id_number=1000,ignore_early_media=true}user/1008 at 192.168.1.122 bridge:sofia/gateway/halonet/0225490317 inline"); > > > > On Wed, Feb 4, 2009 at 2:51 AM, Jacek Sokulski > wrote: > > We have tried setting both effective_caller_id_number and > origination_caller_id_number: > > session1.originate(session1,"{origination_caller_id_number=fixed0248b}sofia/gateway/halonet/0225490317",15); > but the problem still exists. The solution we have found for > the case > when we originate two calls, local and external, is as follow: > > session1 = new Session(); > session1.originate(session1,"user/1003 at 192.168.1.122",15);//local > if(session1.ready()) { > session1.execute("execute_extension","00930691688627 XML > default");//external > } > > so the external call goes through the dialplan. > It does not work if both calls are external. One possible > solution could be > to pass the originating call through dialplan (loopback?) but > we have not managed > to figure out how to do it. > > Thanks > Jacek > > Dnia 03-02-2009, wto o godzinie 14:31 -0300, Nicolas Brenner > pisze: > > > Oops! Well, fortunately I don't use that voip provider > anymore (nor the script). > > > > Thanks Brian. > > > > Nicolas > > > > On Tue, Feb 3, 2009 at 2:25 PM, Brian West > wrote: > > > YOU should NEVER use this method or call setCallerData at > all you > > > should use the correct methods to override the callerid. > > > > > > If its a B-Leg born from an A-Leg you use these on the on > the A-Leg: > > > > > > > http://wiki.freeswitch.org/wiki/Channel_Variables#effective_caller_id_name > > > > http://wiki.freeswitch.org/wiki/Channel_Variables#effective_caller_id_number > > > > > > If you're originating you use this: > > > > > > > http://wiki.freeswitch.org/wiki/Channel_Variables#origination_caller_id_name > > > > http://wiki.freeswitch.org/wiki/Channel_Variables#origination_caller_id_number > > > > > > /b > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From jonas.gauffin at gmail.com Wed Feb 4 07:20:42 2009 From: jonas.gauffin at gmail.com (Jonas Gauffin) Date: Wed, 4 Feb 2009 16:20:42 +0100 Subject: [Freeswitch-users] gateway In-Reply-To: <191c3a030902040614q32c05f02le7e8f41dd70d1fc3@mail.gmail.com> References: <191c3a030902040614q32c05f02le7e8f41dd70d1fc3@mail.gmail.com> Message-ID: I'm behind NAT. Is it FS that picks the random port, or the FW? I've mapped port 5060 to the freeswitch ip in my FW. On Wed, Feb 4, 2009 at 3:14 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > if you are not behind any nat then as long as you run your profile on 5060, > the source port on every packet will be 5060. > If you *are* behind nat the nat mapping will pick a random port unless you > have a firewall that allows you to set specific mappings. > > > On Wed, Feb 4, 2009 at 6:26 AM, Jonas Gauffin wrote: > >> Hello >> I'm trying to make outbound calls through my gateway provider. >> My calls got rejected and I asked them why. >> >> Apparently I need to use 5060 as source port, since they validate both my >> IP and the port that the messages come from. >> Is this possible with freeswitch? If so, what config settings should I >> set? >> >> Regards, >> Jonas >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090204/393aa237/attachment.html From wasim at convergence.pk Wed Feb 4 07:25:40 2009 From: wasim at convergence.pk (Wasim Baig) Date: Wed, 4 Feb 2009 20:25:40 +0500 Subject: [Freeswitch-users] gateway In-Reply-To: References: <191c3a030902040614q32c05f02le7e8f41dd70d1fc3@mail.gmail.com> Message-ID: On Wed, Feb 4, 2009 at 8:20 PM, Jonas Gauffin wrote: > I'm behind NAT. Is it FS that picks the random port, or the FW? > the FW > I've mapped port 5060 to the freeswitch ip in my FW. > thats inbound, now you need to tell your firewall to nail port 5060 on the outbound side when it comes from port 5060 of your freeswitch IP -- wasim h. baig | principal consultant | convergence pk | +92 300 8508070 | peace be upon you ... -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090204/4cde34eb/attachment-0001.html From intralanman at freeswitch.org Wed Feb 4 07:26:40 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Wed, 04 Feb 2009 10:26:40 -0500 Subject: [Freeswitch-users] gateway In-Reply-To: References: <191c3a030902040614q32c05f02le7e8f41dd70d1fc3@mail.gmail.com> Message-ID: <4989B3B0.3060407@freeswitch.org> Its the firewall... some fw's will give you port 5060 if there's nothing else in the network using 5060, but others will randomize it always. -Ray Jonas Gauffin wrote: > I'm behind NAT. > Is it FS that picks the random port, or the FW? > > I've mapped port 5060 to the freeswitch ip in my FW. > > On Wed, Feb 4, 2009 at 3:14 PM, Anthony Minessale > > wrote: > > if you are not behind any nat then as long as you run your profile > on 5060, the source port on every packet will be 5060. > If you *are* behind nat the nat mapping will pick a random port > unless you have a firewall that allows you to set specific mappings. > > > On Wed, Feb 4, 2009 at 6:26 AM, Jonas Gauffin > > wrote: > > Hello > > I'm trying to make outbound calls through my gateway provider. > My calls got rejected and I asked them why. > > Apparently I need to use 5060 as source port, since they > validate both my IP and the port that the messages come from. > Is this possible with freeswitch? If so, what config settings > should I set? > > Regards, > Jonas > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090204/eaf37c7a/attachment.html From freeswitch-users at lists.rupa.com Wed Feb 4 07:27:52 2009 From: freeswitch-users at lists.rupa.com (Rupa Schomaker (lists)) Date: Wed, 04 Feb 2009 09:27:52 -0600 Subject: [Freeswitch-users] gateway In-Reply-To: References: <191c3a030902040614q32c05f02le7e8f41dd70d1fc3@mail.gmail.com> Message-ID: <4989B3F8.6000604@lists.rupa.com> It is the firewall. Most consumer firewalls allow mapping inbound ports (probably what you describe). I don't know of any that do outbound mapping. Linux or *bsd firewalls should be able to do what you want. I'm sure a cisco with IOS could but it has been ages since I've played with that. On 2/4/2009 9:20 AM, Jonas Gauffin wrote: > I'm behind NAT. > Is it FS that picks the random port, or the FW? > > I've mapped port 5060 to the freeswitch ip in my FW. > > On Wed, Feb 4, 2009 at 3:14 PM, Anthony Minessale > > wrote: > > if you are not behind any nat then as long as you run your profile > on 5060, the source port on every packet will be 5060. > If you *are* behind nat the nat mapping will pick a random port > unless you have a firewall that allows you to set specific mappings. > > > On Wed, Feb 4, 2009 at 6:26 AM, Jonas Gauffin > > wrote: > > Hello > > I'm trying to make outbound calls through my gateway provider. > My calls got rejected and I asked them why. > > Apparently I need to use 5060 as source port, since they > validate both my IP and the port that the messages come from. > Is this possible with freeswitch? If so, what config settings > should I set? > > Regards, > Jonas > > From anthony.minessale at gmail.com Wed Feb 4 08:18:15 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 4 Feb 2009 10:18:15 -0600 Subject: [Freeswitch-users] 16 threads didn't exit In-Reply-To: References: <20090204114955.GD6752@cpdata.co.za> Message-ID: <191c3a030902040818r102dce59qb021b93857307353@mail.gmail.com> can you "make current" to rule out any issues with outdated code and maybe describe what you are using in FS such as scripting langs or anything else that was not enabled by default. 2009/2/4 Cavalera Claudio Luigi > freeswitch-users-bounces at lists.freeswitch.org wrote: > > I have seen that error myself, however I assumed it was due te me > > hanging up other cals from the api_hangup_hook of a related call. > > > > I use this to set a "master" call in a conference so that if > > it hangs up > > all calls in the conference hangup. > > > > Thanks Sias for this info, > if you are able to reproduce this message in a systematic way please let > me know so that I can reproduce it myself. > > I've searched for the "threads didn't exit" message in fs code and I did > not found it. > I think it comes from mysql and I'm not sure it's even related to my > segfault. > Best regards, > Claudio > > > Internet Email Confidentiality Footer > > ----------------------------------------------------------------------------------------------------- > La presente comunicazione, con le informazioni in essa contenute e ogni > documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' > indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete > i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, > comunicazione, divulgazione o simili basate sul contenuto di tali > informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., > D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se > avete ricevuto questa comunicazione per errore, vi preghiamo di darne > immediata notizia al mittente e di distruggere il messaggio originale e ogni > file allegato senza farne copia alcuna o riprodurne in alcun modo il > contenuto. > > This e-mail and its attachments are intended for the addressee(s) only and > are confidential and/or may contain legally privileged information. If you > have received this message by mistake or are not one of the addressees > above, you may take no action based on it, and you may not copy or show it > to anyone; please reply to this e-mail and point out the error which has > occurred. > > ----------------------------------------------------------------------------------------------------- > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090204/5699d6b4/attachment.html From anthony.minessale at gmail.com Wed Feb 4 08:22:05 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 4 Feb 2009 10:22:05 -0600 Subject: [Freeswitch-users] origainate through sofia gateway In-Reply-To: <1233758767.5405.22.camel@dotw1126.dotsystems.pl> References: <1233668179.5354.15.camel@dotw1126.dotsystems.pl> <1b46b4e80902030920p521d6079s4c9b78d8d39a924a@mail.gmail.com> <5E2B217D-8C85-48DB-800A-535A7FAAE24B@freeswitch.org> <1b46b4e80902030931y6a1679afw12ec37596ba9d77d@mail.gmail.com> <1233737494.5405.12.camel@dotw1126.dotsystems.pl> <191c3a030902040609s600798c8t720950a2e13ee05a@mail.gmail.com> <1233758767.5405.22.camel@dotw1126.dotsystems.pl> Message-ID: <191c3a030902040822o1ffededbwd592361aa09f6b46@mail.gmail.com> can you press f8 for debug and try that apiExecute and post the results? On Wed, Feb 4, 2009 at 8:46 AM, Jacek Sokulski wrote: > Thanks Anthony, > the js snippets are very instructive. > A couple of points: > 1. The code with apiExecute does not work (local phone is connected, but > after picking up it hungs up immediately), other examples are working > fine. > > 2. It does not show how initiate external call without existing session. > > 3. How can one pass the call through dialplan? > > Jacek > > PS. > we got the code probable from wiki or from this mialing list. > > Dnia 04-02-2009, ?ro o godzinie 08:09 -0600, Anthony Minessale pisze: > > Where did you learn how to use js this way? > > session.originate is being misused here and is depricated and may be > > removed. > > > > the first arg to session.originate is either undefined or a > > *different* session (the a leg) > > > > session1 = new Session(); > > session1.originate(undefined, > > "{ignore_early_media=true}user/1008 at 192.168.1.122"); > > > > session1.setVariable("effective_caller_id_number","fixed0248b"); > > > > //once you have session1 when you originate session2 you pass session1 > > as the arg > > // the effective_caller_id is taken from session1 > > > > session2 = new Session(); > > session2.originate(session1, "sofia/gateway/halonet/0225490317"); > > > > Anyway this whole code is depricated in favor of this: > > > > session1 = new > > Session("{ignore_early_media=true}user/1008 at 192.168.1.122"); > > > > if (session1.ready()) { > > session1.setVariable("effective_caller_id_number","fixed0248b"); > > session2 = new Session("sofia/gateway/halonet/0225490317", > > session1); > > } > > > > and could be further refactored down to this: > > > > session1 = new > > Session("{ignore_early_media=true}user/1008 at 192.168.1.122"); > > if (session1.ready()) { > > session1.setVariable("effective_caller_id_number","fixed0248b"); > > session1.execute("bridge", "sofia/gateway/halonet/0225490317"); > > } > > > > or down to this one line of code that will setup the call detached > > from the script and exit. > > > > var result = apiExecute("originate", > > > "{effective_caller_id_number=fixed0248b,origination_caller_id_number=1000,ignore_early_media=true}user/ > 1008 at 192.168.1.122 bridge:sofia/gateway/halonet/0225490317 inline"); > > > > if you dont care about the result and want to exit even before the > > call is completed. > > > > var result = apiExecute("bgapi", "originate > > > {effective_caller_id_number=fixed0248b,origination_caller_id_number=1000,ignore_early_media=true}user/ > 1008 at 192.168.1.122 bridge:sofia/gateway/halonet/0225490317 inline"); > > > > > > > > On Wed, Feb 4, 2009 at 2:51 AM, Jacek Sokulski > > wrote: > > > > We have tried setting both effective_caller_id_number and > > origination_caller_id_number: > > > > > session1.originate(session1,"{origination_caller_id_number=fixed0248b}sofia/gateway/halonet/0225490317",15); > > but the problem still exists. The solution we have found for > > the case > > when we originate two calls, local and external, is as follow: > > > > session1 = new Session(); > > session1.originate(session1,"user/1003 at 192.168.1.122 > ",15);//local > > if(session1.ready()) { > > session1.execute("execute_extension","00930691688627 XML > > default");//external > > } > > > > so the external call goes through the dialplan. > > It does not work if both calls are external. One possible > > solution could be > > to pass the originating call through dialplan (loopback?) but > > we have not managed > > to figure out how to do it. > > > > Thanks > > Jacek > > > > Dnia 03-02-2009, wto o godzinie 14:31 -0300, Nicolas Brenner > > pisze: > > > > > Oops! Well, fortunately I don't use that voip provider > > anymore (nor the script). > > > > > > Thanks Brian. > > > > > > Nicolas > > > > > > On Tue, Feb 3, 2009 at 2:25 PM, Brian West > > wrote: > > > > YOU should NEVER use this method or call setCallerData at > > all you > > > > should use the correct methods to override the callerid. > > > > > > > > If its a B-Leg born from an A-Leg you use these on the on > > the A-Leg: > > > > > > > > > > > http://wiki.freeswitch.org/wiki/Channel_Variables#effective_caller_id_name > > > > > > > http://wiki.freeswitch.org/wiki/Channel_Variables#effective_caller_id_number > > > > > > > > If you're originating you use this: > > > > > > > > > > > http://wiki.freeswitch.org/wiki/Channel_Variables#origination_caller_id_name > > > > > > > http://wiki.freeswitch.org/wiki/Channel_Variables#origination_caller_id_number > > > > > > > > /b > > > > > > _______________________________________________ > > > Freeswitch-users mailing list > > > Freeswitch-users at lists.freeswitch.org > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090204/6ede27db/attachment-0001.html From mike at jerris.com Wed Feb 4 08:30:38 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 04 Feb 2009 11:30:38 -0500 Subject: [Freeswitch-users] Errors compiling trunk on a fresh system In-Reply-To: Message-ID: This should now be fixed in trunk in revision 11632. Can you please test and confirm. Mike On 2/4/09 9:27 AM, "Cavalera Claudio Luigi" wrote: > Hello, > I'm trying to compile a brand new fs on a clean system. > Revision: 11630 > > After the usual ./bootstrap.sh > ./configure --enable-core-odbc-support > I was getting this at make > http://pastebin.freeswitch.org/7011 > > so > i cd into utils under libs/sofia-sip/utils > issued a doxygen -u > but still getting this > http://pastebin.freeswitch.org/7012 > > any hint on what could be? > > Best Regards, > Claudio > > From kerrada2003 at yahoo.com Wed Feb 4 08:17:12 2009 From: kerrada2003 at yahoo.com (Ali Al-Rubaie) Date: Wed, 4 Feb 2009 08:17:12 -0800 (PST) Subject: [Freeswitch-users] SIP Authentication Message-ID: <423759.89818.qm@web33708.mail.mud.yahoo.com> Hi, I have a problem in SIP registration (authentication) with FreeSWITCH server. The SIP messages are: recv 292 bytes from udp/[209.82.10.250]:3458 at 16:35:24.758862: ?? ------------------------------------------------------------------------ ?? REGISTER sip:209.82.10.235 SIP/2.0 ?? Via: SIP/2.0/UDP 209.82.10.250:1059 ?? From: sip:1001 at 209.82.10.235 ?? To: sip:1001 at 209.82.10.235 ?? Contact: sip:1001 at 209.82.10.250:1059 ?? Call-ID: fc383368-e3e0-4c32-b87b-c3127d0cc2d8 at 192.168.10.23 ?? CSeq: 597775306 REGISTER ?? Content-Length: 0 ?? Expires: 3600 ? ?? ------------------------------------------------------------------------ send 582 bytes to udp/[209.82.10.250]:1059 at 16:35:24.763948: ?? ------------------------------------------------------------------------ ?? SIP/2.0 401 Unauthorized ?? Via: SIP/2.0/UDP 209.82.10.250:1059 ?? From: sip:1001 at 209.82.10.235 ?? To: ;tag=7yam2F01ZH3vH ?? Call-ID: fc383368-e3e0-4c32-b87b-c3127d0cc2d8 at 192.168.10.23 ?? CSeq: 597775306 REGISTER ?? User-Agent: FreeSWITCH-mod_sofia/1.0.2-hacked ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH ?? Supported: timer, precondition, path, replaces ?? WWW-Authenticate: Digest realm="209.82.10.235", nonce="40b63193-85c2-4ed9-874e-c03f81be313d", algorithm=MD5, qop="auth" ?? Content-Length: 0 ? ?? ------------------------------------------------------------------------ recv 466 bytes from udp/[209.82.10.250]:3458 at 16:35:24.772834: ?? ------------------------------------------------------------------------ ?? REGISTER sip:209.82.10.235 SIP/2.0 ?? Via: SIP/2.0/UDP 209.82.10.250:1059 ?? From: sip:1001 at 209.82.10.235 ?? To: sip:1001 at 209.82.10.235 ?? Contact: sip:1001 at 209.82.10.250:1059 ?? Call-ID: fc383368-e3e0-4c32-b87b-c3127d0cc2d8 at 192.168.10.23 ?? CSeq: 597775307 REGISTER ?? Content-Length: 0 ?? Expires: 3600 ?? Authorization: Digest username="1001",realm="209.82.10.235",nonce="40b63193-85c2-4ed9-874e-c03f81be313d",response="eebe0ea43319e82cc5f6dba5877de706",uri="sip:209.82.10.235" ? ?? ------------------------------------------------------------------------ send 458 bytes to udp/[209.82.10.250]:1059 at 16:35:24.774354: ?? ------------------------------------------------------------------------ ?? SIP/2.0 403 Forbidden ?? Via: SIP/2.0/UDP 209.82.10.250:1059 ?? From: sip:1001 at 209.82.10.235 ?? To: ;tag=873c4aH5vtSFD ?? Call-ID: fc383368-e3e0-4c32-b87b-c3127d0cc2d8 at 192.168.10.23 ?? CSeq: 597775307 REGISTER ?? User-Agent: FreeSWITCH-mod_sofia/1.0.2-hacked ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH ?? Supported: timer, precondition, path, replaces ?? Content-Length: 0 ? ?? ------------------------------------------------------------------------ What I have noted is that the client does not send the values for "cnonce" and "nc" in the response. I'm not sure if this is the reason, however how this problem can be solved? Thanks, Ali -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090204/1b23e643/attachment.html From anthony.minessale at gmail.com Wed Feb 4 08:52:57 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 4 Feb 2009 10:52:57 -0600 Subject: [Freeswitch-users] SIP Authentication In-Reply-To: <423759.89818.qm@web33708.mail.mud.yahoo.com> References: <423759.89818.qm@web33708.mail.mud.yahoo.com> Message-ID: <191c3a030902040852k2c54fceekd73bbd128eb36765@mail.gmail.com> That's the reason, the missing params. The client has a bug in it. On Wed, Feb 4, 2009 at 10:17 AM, Ali Al-Rubaie wrote: > Hi, > > I have a problem in SIP registration (authentication) with FreeSWITCH > server. The SIP messages are: > > recv 292 bytes from udp/[209.82.10.250]:3458 at 16:35:24.758862: > > > ------------------------------------------------------------------------ > > REGISTER sip:209.82.10.235 SIP/2.0 > > Via: SIP/2.0/UDP 209.82.10.250:1059 > > From: sip:1001 at 209.82.10.235 > > To: sip:1001 at 209.82.10.235 > > Contact: sip:1001 at 209.82.10.250:1059 > > Call-ID: fc383368-e3e0-4c32-b87b-c3127d0cc2d8 at 192.168.10.23 > > CSeq: 597775306 REGISTER > > Content-Length: 0 > > Expires: 3600 > > > > > ------------------------------------------------------------------------ > > send 582 bytes to udp/[209.82.10.250]:1059 at 16:35:24.763948: > > > ------------------------------------------------------------------------ > > SIP/2.0 401 Unauthorized > > Via: SIP/2.0/UDP 209.82.10.250:1059 > > From: sip:1001 at 209.82.10.235 > > To: > >;tag=7yam2F01ZH3vH > > Call-ID: fc383368-e3e0-4c32-b87b-c3127d0cc2d8 at 192.168.10.23 > > CSeq: 597775306 REGISTER > > User-Agent: FreeSWITCH-mod_sofia/1.0.2-hacked > > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH > > Supported: timer, precondition, path, replaces > > WWW-Authenticate: Digest realm="209.82.10.235", > nonce="40b63193-85c2-4ed9-874e-c03f81be313d", algorithm=MD5, qop="auth" > > Content-Length: 0 > > > > > ------------------------------------------------------------------------ > > recv 466 bytes from udp/[209.82.10.250]:3458 at 16:35:24.772834: > > > ------------------------------------------------------------------------ > > REGISTER sip:209.82.10.235 SIP/2.0 > > Via: SIP/2.0/UDP 209.82.10.250:1059 > > From: sip:1001 at 209.82.10.235 > > To: sip:1001 at 209.82.10.235 > > Contact: sip:1001 at 209.82.10.250:1059 > > Call-ID: fc383368-e3e0-4c32-b87b-c3127d0cc2d8 at 192.168.10.23 > > CSeq: 597775307 REGISTER > > Content-Length: 0 > > Expires: 3600 > > Authorization: Digest > username="1001",realm="209.82.10.235",nonce="40b63193-85c2-4ed9-874e-c03f81be313d",response="eebe0ea43319e82cc5f6dba5877de706",uri="sip:209.82.10.235" > > > > > ------------------------------------------------------------------------ > > send 458 bytes to udp/[209.82.10.250]:1059 at 16:35:24.774354: > > > ------------------------------------------------------------------------ > > SIP/2.0 403 Forbidden > > Via: SIP/2.0/UDP 209.82.10.250:1059 > > From: sip:1001 at 209.82.10.235 > > To: > >;tag=873c4aH5vtSFD > > Call-ID: fc383368-e3e0-4c32-b87b-c3127d0cc2d8 at 192.168.10.23 > > CSeq: 597775307 REGISTER > > User-Agent: FreeSWITCH-mod_sofia/1.0.2-hacked > > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH > > Supported: timer, precondition, path, replaces > > Content-Length: 0 > > > > > ------------------------------------------------------------------------ > What I have noted is that the client does not send the values for "cnonce" > and "nc" in the response. I'm not sure if this is the reason, however how > this problem can be solved? > > Thanks, > > Ali > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090204/8a749fcb/attachment-0001.html From brian at freeswitch.org Wed Feb 4 08:52:45 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 4 Feb 2009 10:52:45 -0600 Subject: [Freeswitch-users] SIP Authentication In-Reply-To: <423759.89818.qm@web33708.mail.mud.yahoo.com> References: <423759.89818.qm@web33708.mail.mud.yahoo.com> Message-ID: <7DAC91F6-2FD3-464D-AA81-321EBCADC8C0@freeswitch.org> What client is this? I also notice we receive port 3458 and reply to port 1059... /b On Feb 4, 2009, at 10:17 AM, Ali Al-Rubaie wrote: > What I have noted is that the client does not send the values for > "cnonce" and "nc" in the response. I'm not sure if this is the > reason, however how this problem can be solved? > > Thanks, > > Ali From msc at freeswitch.org Wed Feb 4 09:41:07 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 4 Feb 2009 09:41:07 -0800 Subject: [Freeswitch-users] origainate through sofia gateway In-Reply-To: <191c3a030902040609s600798c8t720950a2e13ee05a@mail.gmail.com> References: <1233668179.5354.15.camel@dotw1126.dotsystems.pl> <1b46b4e80902030920p521d6079s4c9b78d8d39a924a@mail.gmail.com> <5E2B217D-8C85-48DB-800A-535A7FAAE24B@freeswitch.org> <1b46b4e80902030931y6a1679afw12ec37596ba9d77d@mail.gmail.com> <1233737494.5405.12.camel@dotw1126.dotsystems.pl> <191c3a030902040609s600798c8t720950a2e13ee05a@mail.gmail.com> Message-ID: <87f2f3b90902040941r61d669aaie949aa7cc8578a9a@mail.gmail.com> I'll make sure the substance of this is in the wiki and I'll look for references to the deprecated way and remove those. -MC On Wed, Feb 4, 2009 at 6:09 AM, Anthony Minessale wrote: > Where did you learn how to use js this way? > session.originate is being misused here and is depricated and may be > removed. > > the first arg to session.originate is either undefined or a *different* > session (the a leg) > > session1 = new Session(); > session1.originate(undefined, > "{ignore_early_media=true}user/1008 at 192.168.1.122"); > > session1.setVariable("effective_caller_id_number","fixed0248b"); > > //once you have session1 when you originate session2 you pass session1 as > the arg > // the effective_caller_id is taken from session1 > > session2 = new Session(); > session2.originate(session1, "sofia/gateway/halonet/0225490317"); > > Anyway this whole code is depricated in favor of this: > > session1 = new Session("{ignore_early_media=true}user/1008 at 192.168.1.122"); > if (session1.ready()) { > session1.setVariable("effective_caller_id_number","fixed0248b"); > session2 = new Session("sofia/gateway/halonet/0225490317", session1); > } > > and could be further refactored down to this: > > session1 = new Session("{ignore_early_media=true}user/1008 at 192.168.1.122"); > if (session1.ready()) { > session1.setVariable("effective_caller_id_number","fixed0248b"); > session1.execute("bridge", "sofia/gateway/halonet/0225490317"); > } > > or down to this one line of code that will setup the call detached from the > script and exit. > > var result = apiExecute("originate", > "{effective_caller_id_number=fixed0248b,origination_caller_id_number=1000,ignore_early_media=true}user/1008 at 192.168.1.122 > bridge:sofia/gateway/halonet/0225490317 inline"); > > if you dont care about the result and want to exit even before the call is > completed. > > var result = apiExecute("bgapi", "originate > {effective_caller_id_number=fixed0248b,origination_caller_id_number=1000,ignore_early_media=true}user/1008 at 192.168.1.122 > bridge:sofia/gateway/halonet/0225490317 inline"); > > > > On Wed, Feb 4, 2009 at 2:51 AM, Jacek Sokulski > wrote: >> >> We have tried setting both effective_caller_id_number and >> origination_caller_id_number: >> >> >> session1.originate(session1,"{origination_caller_id_number=fixed0248b}sofia/gateway/halonet/0225490317",15); >> but the problem still exists. The solution we have found for the case >> when we originate two calls, local and external, is as follow: >> >> session1 = new Session(); >> session1.originate(session1,"user/1003 at 192.168.1.122",15);//local >> if(session1.ready()) { >> session1.execute("execute_extension","00930691688627 XML >> default");//external >> } >> >> so the external call goes through the dialplan. >> It does not work if both calls are external. One possible solution could >> be >> to pass the originating call through dialplan (loopback?) but we have not >> managed >> to figure out how to do it. >> >> Thanks >> Jacek >> >> Dnia 03-02-2009, wto o godzinie 14:31 -0300, Nicolas Brenner pisze: >> > Oops! Well, fortunately I don't use that voip provider anymore (nor the >> > script). >> > >> > Thanks Brian. >> > >> > Nicolas >> > >> > On Tue, Feb 3, 2009 at 2:25 PM, Brian West wrote: >> > > YOU should NEVER use this method or call setCallerData at all you >> > > should use the correct methods to override the callerid. >> > > >> > > If its a B-Leg born from an A-Leg you use these on the on the A-Leg: >> > > >> > > >> > > http://wiki.freeswitch.org/wiki/Channel_Variables#effective_caller_id_name >> > > >> > > http://wiki.freeswitch.org/wiki/Channel_Variables#effective_caller_id_number >> > > >> > > If you're originating you use this: >> > > >> > > >> > > http://wiki.freeswitch.org/wiki/Channel_Variables#origination_caller_id_name >> > > >> > > http://wiki.freeswitch.org/wiki/Channel_Variables#origination_caller_id_number >> > > >> > > /b >> > >> > _______________________________________________ >> > Freeswitch-users mailing list >> > Freeswitch-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From Daniell at airg.com Wed Feb 4 09:43:10 2009 From: Daniell at airg.com (Daniel Liang) Date: Wed, 4 Feb 2009 09:43:10 -0800 Subject: [Freeswitch-users] Recording background music and voice is out of sync Message-ID: <0B02E756F603CC409EB553879B090CC80A23EBB5@HPEXCHVS01.exchange.airg> What I did was the following: First, I sent the playback command: call-command: execute execute-app-name: playback execute-app-arg: Then I send uuid_record (Sorry, it was not Record command): api uuid_record start 120 I also tried replacing the playback command with: api uuid_displace start 0 mux But the end results are the same. The recorded user's voice is about 0.5 second behind the expected result. Thanks, Daniel -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: February 3, 2009 6:36 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Recording background music and voice is outof sync Can you show us an example of how you're doing this? Playback and Record aren't async so you'll need to show us how you're doing this. Also don't hijack threads you hit replay on the one "Re: [Freeswitch- users] FreeSwitch setup as a "Dumb" SBC" as the subject, deleted the subject and started a new body. That hijacks the thread and that can cause your problem to go ignored in some cases if people aren't interested in the thread topic depending on how their reader threads the emails. Please click new message and type freeswitch- users at lists.freeswitch.org in and then input your subject and body to start a new thread. Thanks, Brian West FreeSWITCH.org On Feb 3, 2009, at 8:21 PM, Daniel Liang wrote: > Hi, > > I was trying to record a background music with a user's voice at the > same time. I did a playback and started recording. But the recorded > user's voice and the background music is about 0.5 second out of sync. > I also tried to use uuid_displace instead of playback, but I got the > same result. _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090204/2d1124e2/attachment.html From saigop at gmail.com Wed Feb 4 09:56:14 2009 From: saigop at gmail.com (Gopalakrishnan A.N) Date: Wed, 4 Feb 2009 23:26:14 +0530 Subject: [Freeswitch-users] Q931 decoding Update In-Reply-To: <3D009334-DCFD-4AE5-A446-6C8165F2D77D@jerris.com> References: <4972046E.8020102@ewetel.de> <200901291202.41144.stkn@freeswitch.org> <191c3a030901290549x29c87976k68b3f9b455f00d0f@mail.gmail.com> <4981B8E6.9070708@ewetel.de> <5A2ED995-CF05-4AEF-9103-43BC84465514@freeswitch.org> <4981D6F8.4060409@ewetel.de> <191c3a030901290832k27a18b30u7bc531157fd89b19@mail.gmail.com> <49830576.6080907@ewetel.de> <4987359B.2020402@ewetel.de> <3D009334-DCFD-4AE5-A446-6C8165F2D77D@jerris.com> Message-ID: <2ea4d47e0902040956v75c5472foa4649c50b7340484@mail.gmail.com> Hi, Its a awesome. Can the packet capturing be done with event socket? -- Thank you with regards, Gopal, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090204/74f6434a/attachment.html From chavpaskov at shaw.ca Wed Feb 4 09:59:48 2009 From: chavpaskov at shaw.ca (Chav Paskov) Date: Wed, 04 Feb 2009 09:59:48 -0800 Subject: [Freeswitch-users] mod_limit Message-ID: <4989D794.1010805@shaw.ca> Hi , is it possible to use mod_limit in case if the end point is not registered / gateway for example/. Regards Chav From msc at freeswitch.org Wed Feb 4 10:06:52 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 4 Feb 2009 10:06:52 -0800 Subject: [Freeswitch-users] mod_limit In-Reply-To: <4989D794.1010805@shaw.ca> References: <4989D794.1010805@shaw.ca> Message-ID: <87f2f3b90902041006v7d7d7ff2t9961274905d9d5b0@mail.gmail.com> On Wed, Feb 4, 2009 at 9:59 AM, Chav Paskov wrote: > Hi , > is it possible to use mod_limit in case if the end point is not > registered / gateway for example/. Could you add some detail to this question? What are you trying to do? (mod_limit may or may not work, but there might be another solution which is why I am asking.) -MC > Regards > Chav > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From chavpaskov at shaw.ca Wed Feb 4 10:54:56 2009 From: chavpaskov at shaw.ca (Chav Paskov) Date: Wed, 04 Feb 2009 10:54:56 -0800 Subject: [Freeswitch-users] mod_limit In-Reply-To: <87f2f3b90902041006v7d7d7ff2t9961274905d9d5b0@mail.gmail.com> References: <4989D794.1010805@shaw.ca> <87f2f3b90902041006v7d7d7ff2t9961274905d9d5b0@mail.gmail.com> Message-ID: <4989E480.1080105@shaw.ca> Michael Collins wrote: > On Wed, Feb 4, 2009 at 9:59 AM, Chav Paskov wrote: > >> Hi , >> is it possible to use mod_limit in case if the end point is not >> registered / gateway for example/. >> > > Could you add some detail to this question? What are you trying to do? > (mod_limit may or may not work, but there might be another solution > which is why I am asking.) > > -MC > > >> Regards >> Chav >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > i have few gateways under my ACL that are allowed to send calls to FS, but i want to be able to enforce "capacity" policy on the traffic coming from any one of them depending on total termination capacity on my termination end. Let say GW 1 has to be limited to make 10 simultaneous calls while GW2 could make up to 30 and so on. Regards Chav From msc at freeswitch.org Wed Feb 4 11:05:09 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 4 Feb 2009 11:05:09 -0800 Subject: [Freeswitch-users] mod_limit In-Reply-To: <4989E480.1080105@shaw.ca> References: <4989D794.1010805@shaw.ca> <87f2f3b90902041006v7d7d7ff2t9961274905d9d5b0@mail.gmail.com> <4989E480.1080105@shaw.ca> Message-ID: <87f2f3b90902041105l50f51f08t230bab8d69eefb4e@mail.gmail.com> On Wed, Feb 4, 2009 at 10:54 AM, Chav Paskov wrote: > Michael Collins wrote: >> On Wed, Feb 4, 2009 at 9:59 AM, Chav Paskov wrote: >> >>> Hi , >>> is it possible to use mod_limit in case if the end point is not >>> registered / gateway for example/. >>> >> >> Could you add some detail to this question? What are you trying to do? >> (mod_limit may or may not work, but there might be another solution >> which is why I am asking.) >> >> -MC >> >> >>> Regards >>> Chav >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > i have few gateways under my ACL that are allowed to send calls to FS, > but i want to be able to enforce "capacity" policy on the traffic > coming from any one of them depending on total termination capacity on > my termination end. > Let say GW 1 has to be limited to make 10 simultaneous calls while GW2 > could make up to 30 and so on. I'm sure that this is possible. I don't personally have a way to test all of this but I know that a number of our users are doing things like this currently. Can you hop on to the IRC channel? #freeswitch on irc.freenode.net. A lot of people there can help with this one. -MC (IRC: mercutioviz) > Regards > Chav > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From brian at freeswitch.org Wed Feb 4 11:13:54 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 4 Feb 2009 13:13:54 -0600 Subject: [Freeswitch-users] mod_limit In-Reply-To: <87f2f3b90902041105l50f51f08t230bab8d69eefb4e@mail.gmail.com> References: <4989D794.1010805@shaw.ca> <87f2f3b90902041006v7d7d7ff2t9961274905d9d5b0@mail.gmail.com> <4989E480.1080105@shaw.ca> <87f2f3b90902041105l50f51f08t230bab8d69eefb4e@mail.gmail.com> Message-ID: <49E707E6-2126-45E5-9D5D-D44D477245C5@freeswitch.org> mod_limit is just another lego brick in the set... You can put it in the dialplan before you go in our out a gateway and limit based on the realm... its no different for registered users either... I get the feeling you're trying to make this more complicated then it really is. /b PS when you reply please try to shorten the resulting text of the reply no need to have unsubscribe info in the email 10 times. On Feb 4, 2009, at 1:05 PM, Michael Collins wrote: >>> >>> >> i have few gateways under my ACL that are allowed to send calls to >> FS, >> but i want to be able to enforce "capacity" policy on the traffic >> coming from any one of them depending on total termination capacity >> on >> my termination end. >> Let say GW 1 has to be limited to make 10 simultaneous calls >> while GW2 >> could make up to 30 and so on. > > I'm sure that this is possible. I don't personally have a way to test > all of this but I know that a number of our users are doing things > like this currently. Can you hop on to the IRC channel? #freeswitch on > irc.freenode.net. A lot of people there can help with this one. > > -MC (IRC: mercutioviz) > >> Regards >> Chav From msc at freeswitch.org Wed Feb 4 12:52:29 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 4 Feb 2009 12:52:29 -0800 Subject: [Freeswitch-users] Q931 decoding Update In-Reply-To: <2ea4d47e0902040956v75c5472foa4649c50b7340484@mail.gmail.com> References: <4972046E.8020102@ewetel.de> <191c3a030901290549x29c87976k68b3f9b455f00d0f@mail.gmail.com> <4981B8E6.9070708@ewetel.de> <5A2ED995-CF05-4AEF-9103-43BC84465514@freeswitch.org> <4981D6F8.4060409@ewetel.de> <191c3a030901290832k27a18b30u7bc531157fd89b19@mail.gmail.com> <49830576.6080907@ewetel.de> <4987359B.2020402@ewetel.de> <3D009334-DCFD-4AE5-A446-6C8165F2D77D@jerris.com> <2ea4d47e0902040956v75c5472foa4649c50b7340484@mail.gmail.com> Message-ID: <87f2f3b90902041252h5ede448bq720c15ea23dd517a@mail.gmail.com> On Wed, Feb 4, 2009 at 9:56 AM, Gopalakrishnan A.N wrote: > Hi, > Its a awesome. Can the packet capturing be done with event socket? Not at this time. Would require some additional programming. Are you up for the task? ;) -MC > > -- > Thank you with regards, > Gopal, > > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From sias at cpdata.co.za Wed Feb 4 13:14:46 2009 From: sias at cpdata.co.za (Sias Mey) Date: Wed, 4 Feb 2009 23:14:46 +0200 Subject: [Freeswitch-users] 16 threads didn't exit In-Reply-To: <191c3a030902040818r102dce59qb021b93857307353@mail.gmail.com> References: <20090204114955.GD6752@cpdata.co.za> <191c3a030902040818r102dce59qb021b93857307353@mail.gmail.com> Message-ID: <20090204211445.GA15501@cpdata.co.za> Hi Anthony, I have been seeing this message for the for a couple of weeks now. And as you know(you asked me to) I have been keeping up to date. So it is definately not only a recent change. I use JS with odbc and core support. And also xml_rpc. I will do some more testing tommorow to see if I can narrow down what script and what bit of code seems to cause this. On Wed, Feb 04, 2009 at 10:18:15AM -0600, Anthony Minessale wrote: > can you "make current" to rule out any issues with outdated code and > maybe describe what you are using in FS such as scripting langs or > anything else that was not enabled by default. > > 2009/2/4 Cavalera Claudio Luigi <[1]Claudio.Cavalera at italtel.it> > > [2]freeswitch-users-bounces at lists.freeswitch.org wrote: > > I have seen that error myself, however I assumed it was due te me > > hanging up other cals from the api_hangup_hook of a related call. > > > > I use this to set a "master" call in a conference so that if > > it hangs up > > all calls in the conference hangup. > > > > Thanks Sias for this info, > if you are able to reproduce this message in a systematic way please > let > me know so that I can reproduce it myself. > I've searched for the "threads didn't exit" message in fs code and I > did > not found it. > I think it comes from mysql and I'm not sure it's even related to my > segfault. > > Best regards, > Claudio > Internet Email Confidentiality Footer > ----------------------------------------------------------------------- > ------------------------------ > La presente comunicazione, con le informazioni in essa contenute e ogni > documento o file allegato, e' rivolta unicamente alla/e persona/e cui > e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se > non siete i destinatari/autorizzati siete avvisati che qualsiasi > azione, copia, comunicazione, divulgazione o simili basate sul > contenuto di tali informazioni e' vietata e potrebbe essere contro la > legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione > dei dati personali). Se avete ricevuto questa comunicazione per errore, > vi preghiamo di darne immediata notizia al mittente e di distruggere il > messaggio originale e ogni file allegato senza farne copia alcuna o > riprodurne in alcun modo il contenuto. > This e-mail and its attachments are intended for the addressee(s) only > and are confidential and/or may contain legally privileged information. > If you have received this message by mistake or are not one of the > addressees above, you may take no action based on it, and you may not > copy or show it to anyone; please reply to this e-mail and point out > the error which has occurred. > ----------------------------------------------------------------------- > ------------------------------ > _______________________________________________ > Freeswitch-users mailing list > [3]Freeswitch-users at lists.freeswitch.org > [4]http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:[5]http://lists.freeswitch.org/mailman/options/freeswitch-u > sers > [6]http://www.freeswitch.org > > -- > Anthony Minessale II > FreeSWITCH [7]http://www.freeswitch.org/ > ClueCon [8]http://www.cluecon.com/ > AIM: anthm > [9]MSN:anthony_minessale at hotmail.com > GTALK/JABBER/[10]PAYPAL:anthony.minessale at gmail.com > IRC: [11]irc.freenode.net #freeswitch > FreeSWITCH Developer Conference > [12]sip:888 at conference.freeswitch.org > [13]iax:guest at conference.freeswitch.org/888 > [14]googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > References > > 1. mailto:Claudio.Cavalera at italtel.it > 2. mailto:freeswitch-users-bounces at lists.freeswitch.org > 3. mailto:Freeswitch-users at lists.freeswitch.org > 4. http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > 5. http://lists.freeswitch.org/mailman/options/freeswitch-users > 6. http://www.freeswitch.org/ > 7. http://www.freeswitch.org/ > 8. http://www.cluecon.com/ > 9. mailto:MSN%3Aanthony_minessale at hotmail.com > 10. mailto:PAYPAL%3Aanthony.minessale at gmail.com > 11. http://irc.freenode.net/ > 12. mailto:sip%3A888 at conference.freeswitch.org > 13. http://iax:guest at conference.freeswitch.org/888 > 14. mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From nik.middleton at noblesolutions.co.uk Wed Feb 4 14:30:09 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Wed, 4 Feb 2009 22:30:09 -0000 Subject: [Freeswitch-users] FS in ISP Mode In-Reply-To: <2ea4d47e0902040956v75c5472foa4649c50b7340484@mail.gmail.com> References: <4972046E.8020102@ewetel.de><200901291202.41144.stkn@freeswitch.org><191c3a030901290549x29c87976k68b3f9b455f00d0f@mail.gmail.com><4981B8E6.9070708@ewetel.de><5A2ED995-CF05-4AEF-9103-43BC84465514@freeswitch.org><4981D6F8.4060409@ewetel.de><191c3a030901290832k27a18b30u7bc531157fd89b19@mail.gmail.com><49830576.6080907@ewetel.de> <4987359B.2020402@ewetel.de><3D009334-DCFD-4AE5-A446-6C8165F2D77D@jerris.com> <2ea4d47e0902040956v75c5472foa4649c50b7340484@mail.gmail.com> Message-ID: Hi Guys, Excuse my ignorance, but I'm just starting with FS. I've loaded FS onto one of our servers in a datacenter. I'm registering with our PSTN breakout provider just fine, but I'm a little confused about internal/external. Given that we have no internal clients, as they're all external, should I switch the ports over so that 5060 is the external port? I think I know the answer, but would like confirmation Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090204/2b5c0a83/attachment.html From nik.middleton at noblesolutions.co.uk Wed Feb 4 14:40:11 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Wed, 4 Feb 2009 22:40:11 -0000 Subject: [Freeswitch-users] Gateway setting In-Reply-To: <2ea4d47e0902040956v75c5472foa4649c50b7340484@mail.gmail.com> References: <4972046E.8020102@ewetel.de><200901291202.41144.stkn@freeswitch.org><191c3a030901290549x29c87976k68b3f9b455f00d0f@mail.gmail.com><4981B8E6.9070708@ewetel.de><5A2ED995-CF05-4AEF-9103-43BC84465514@freeswitch.org><4981D6F8.4060409@ewetel.de><191c3a030901290832k27a18b30u7bc531157fd89b19@mail.gmail.com><49830576.6080907@ewetel.de> <4987359B.2020402@ewetel.de><3D009334-DCFD-4AE5-A446-6C8165F2D77D@jerris.com> <2ea4d47e0902040956v75c5472foa4649c50b7340484@mail.gmail.com> Message-ID: Hi Guys, Need a little help here; I connect to my PSTN provider via the LAN, Question: As the provider authenticates on IP, how do I not send a password? In the .xml file if I remove the password entry it complains Secondly, the contact should be my local address, not the public one. What do I need to do here? Finally what does FS do to determine if the status is up, is there an asterisk equivalent of qualify going on here? Regards Name My Provider Scheme Digest Realm 172.16.1.2 Username myname Password yes >From Contact To sip: myname @172.168.1.2 Proxy myname:172.168.1.2 Context default Expires 600 Freq 600 Ping 0 PingFreq 0 State NOREG Status UP -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090204/3705c4b8/attachment-0001.html From brian at freeswitch.org Wed Feb 4 14:42:05 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 4 Feb 2009 16:42:05 -0600 Subject: [Freeswitch-users] FS in ISP Mode In-Reply-To: References: <4972046E.8020102@ewetel.de><200901291202.41144.stkn@freeswitch.org><191c3a030901290549x29c87976k68b3f9b455f00d0f@mail.gmail.com><4981B8E6.9070708@ewetel.de><5A2ED995-CF05-4AEF-9103-43BC84465514@freeswitch.org><4981D6F8.4060409@ewetel.de><191c3a030901290832k27a18b30u7bc531157fd89b19@mail.gmail.com><49830576.6080907@ewetel.de> <4987359B.2020402@ewetel.de><3D009334-DCFD-4AE5-A446-6C8165F2D77D@jerris.com> <2ea4d47e0902040956v75c5472foa4649c50b7340484@mail.gmail.com> Message-ID: <513C18B7-631A-4250-8484-81DD7F75D3A1@freeswitch.org> Don't let the names of the profiles confuse you... they are just names. internal is on port 5060 has auth on... external is on 5080 and doesn't' have auth on and lets all calls into the public context without auth. Also when you post to the mailing list do not hijack a thread. Hijacking happens when you take an existing message, click reply, change the subject and start a new body. That will hijack the thread. In your case the thread was "Re: [Freeswitch-users] Q931 decoding Update". So next time please click new message and input the address freeswitch-users at lists.freeswitch.org . Thanks, Brian West FreeSWITCH.org On Feb 4, 2009, at 4:30 PM, Nik Middleton wrote: > Hi Guys, > > Excuse my ignorance, but I?m just starting with FS. > > I?ve loaded FS onto one of our servers in a datacenter. I?m > registering with our PSTN breakout provider just fine, but I?m a > little confused about internal/external. > > Given that we have no internal clients, as they?re all external, > should I switch the ports over so that 5060 is the external port? > > I think I know the answer, but would like confirmation > > Regards > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Wed Feb 4 14:48:49 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 4 Feb 2009 16:48:49 -0600 Subject: [Freeswitch-users] Gateway setting In-Reply-To: References: <4972046E.8020102@ewetel.de><200901291202.41144.stkn@freeswitch.org><191c3a030901290549x29c87976k68b3f9b455f00d0f@mail.gmail.com><4981B8E6.9070708@ewetel.de><5A2ED995-CF05-4AEF-9103-43BC84465514@freeswitch.org><4981D6F8.4060409@ewetel.de><191c3a030901290832k27a18b30u7bc531157fd89b19@mail.gmail.com><49830576.6080907@ewetel.de> <4987359B.2020402@ewetel.de><3D009334-DCFD-4AE5-A446-6C8165F2D77D@jerris.com> <2ea4d47e0902040956v75c5472foa4649c50b7340484@mail.gmail.com> Message-ID: <4C4AD6CE-35C7-4F13-80C6-6F86DA71A51E@freeswitch.org> You don't use gateways if they auth by IP... just dial sofia/profile/ number at remoteip Also please stop hijacking threads. /b On Feb 4, 2009, at 4:40 PM, Nik Middleton wrote: > Hi Guys, > > Need a little help here; I connect to my PSTN provider via the LAN, > > Question: As the provider authenticates on IP, how do I not send a > password? In the .xml file if I remove the password entry it > complains > > Secondly, the contact should be my local address, not the public > one. What do I need to do here? > > Finally what does FS do to determine if the status is up, is there > an asterisk equivalent of qualify going on here? > > Regards > > > Name My Provider > Scheme Digest > Realm 172.16.1.2 > Username myname > Password yes > From > Contact > To sip: myname @172.168.1.2 > Proxy myname:172.168.1.2 > Context default > Expires 600 > Freq 600 > Ping 0 > PingFreq 0 > State NOREG > Status UP > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090204/def37ca8/attachment.html From nik.middleton at noblesolutions.co.uk Wed Feb 4 14:50:35 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Wed, 4 Feb 2009 22:50:35 -0000 Subject: [Freeswitch-users] FS in ISP Mode Message-ID: Do apologise about the hijacking, Question: My ISP sends inbound calls via 5060, so it seems I need to renumber the ports, but that leaves my SIP end points who authenticate also needing 5060, can they be combined? Regards, -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: 04 February 2009 22:42 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] FS in ISP Mode Don't let the names of the profiles confuse you... they are just names. internal is on port 5060 has auth on... external is on 5080 and doesn't' have auth on and lets all calls into the public context without auth. Also when you post to the mailing list do not hijack a thread. Hijacking happens when you take an existing message, click reply, change the subject and start a new body. That will hijack the thread. In your case the thread was "Re: [Freeswitch-users] Q931 decoding Update". So next time please click new message and input the address freeswitch-users at lists.freeswitch.org . Thanks, Brian West FreeSWITCH.org On Feb 4, 2009, at 4:30 PM, Nik Middleton wrote: > Hi Guys, > > Excuse my ignorance, but I'm just starting with FS. > > I've loaded FS onto one of our servers in a datacenter. I'm > registering with our PSTN breakout provider just fine, but I'm a > little confused about internal/external. > > Given that we have no internal clients, as they're all external, > should I switch the ports over so that 5060 is the external port? > > I think I know the answer, but would like confirmation > > Regards > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090204/f98b8f0d/attachment-0001.html From nik.middleton at noblesolutions.co.uk Wed Feb 4 14:52:49 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Wed, 4 Feb 2009 22:52:49 -0000 Subject: [Freeswitch-users] Gateway settings Message-ID: Hi Guys, Need a little help here; I connect to my PSTN provider via the LAN, Question: As the provider authenticates on IP, how do I not send a password? In the .xml file if I remove the password entry it complains Secondly, the contact should be my local address, not the public one. What do I need to do here? Finally what does FS do to determine if the status is up, is there an asterisk equivalent of qualify going on here? Regards Name My Provider Scheme Digest Realm 172.16.1.2 Username myname Password yes >From Contact To sip: myname @172.168.1.2 Proxy myname:172.168.1.2 Context default Expires 600 Freq 600 Ping 0 PingFreq 0 State NOREG Status UP -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090204/15a3fcd1/attachment.html From nik.middleton at noblesolutions.co.uk Wed Feb 4 14:54:41 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Wed, 4 Feb 2009 22:54:41 -0000 Subject: [Freeswitch-users] Gateway setting In-Reply-To: <4C4AD6CE-35C7-4F13-80C6-6F86DA71A51E@freeswitch.org> References: <4972046E.8020102@ewetel.de><200901291202.41144.stkn@freeswitch.org><191c3a030901290549x29c87976k68b3f9b455f00d0f@mail.gmail.com><4981B8E6.9070708@ewetel.de><5A2ED995-CF05-4AEF-9103-43BC84465514@freeswitch.org><4981D6F8.4060409@ewetel.de><191c3a030901290832k27a18b30u7bc531157fd89b19@mail.gmail.com><49830576.6080907@ewetel.de><4987359B.2020402@ewetel.de><3D009334-DCFD-4AE5-A446-6C8165F2D77D@jerris.com><2ea4d47e0902040956v75c5472foa4649c50b7340484@mail.gmail.com> <4C4AD6CE-35C7-4F13-80C6-6F86DA71A51E@freeswitch.org> Message-ID: Sorry, 2 messages sent before I understood how this forum processes posts ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: 04 February 2009 22:49 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Gateway setting You don't use gateways if they auth by IP... just dial sofia/profile/number at remoteip Also please stop hijacking threads. /b On Feb 4, 2009, at 4:40 PM, Nik Middleton wrote: Hi Guys, Need a little help here; I connect to my PSTN provider via the LAN, Question: As the provider authenticates on IP, how do I not send a password? In the .xml file if I remove the password entry it complains Secondly, the contact should be my local address, not the public one. What do I need to do here? Finally what does FS do to determine if the status is up, is there an asterisk equivalent of qualify going on here? Regards Name My Provider Scheme Digest Realm 172.16.1.2 Username myname Password yes >From Contact To sip: myname @172.168.1.2 Proxy myname:172.168.1.2 Context default Expires 600 Freq 600 Ping 0 PingFreq 0 State NOREG Status UP _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090204/440f0b90/attachment-0001.html From brian at freeswitch.org Wed Feb 4 14:55:49 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 4 Feb 2009 16:55:49 -0600 Subject: [Freeswitch-users] FS in ISP Mode In-Reply-To: References: Message-ID: <05317B37-EDAF-4286-BE27-5DAB8C0A0289@freeswitch.org> If your ITSP requires you to come from 5060 on your request then they are seriously broken. But yes you can move the ports around on the profiles or turn auth to false on the internal profile if you don't require any digest auth or phones registering. /b On Feb 4, 2009, at 4:50 PM, Nik Middleton wrote: > Do apologise about the hijacking, > > Question: My ISP sends inbound calls via 5060, so it seems I need > to renumber the ports, but that leaves my SIP end points who > authenticate also needing 5060, can they be combined? > > Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090204/9f91c9aa/attachment.html From brian at freeswitch.org Wed Feb 4 14:59:57 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 4 Feb 2009 16:59:57 -0600 Subject: [Freeswitch-users] Gateway setting In-Reply-To: References: <4972046E.8020102@ewetel.de><200901291202.41144.stkn@freeswitch.org><191c3a030901290549x29c87976k68b3f9b455f00d0f@mail.gmail.com><4981B8E6.9070708@ewetel.de><5A2ED995-CF05-4AEF-9103-43BC84465514@freeswitch.org><4981D6F8.4060409@ewetel.de><191c3a030901290832k27a18b30u7bc531157fd89b19@mail.gmail.com><49830576.6080907@ewetel.de><4987359B.2020402@ewetel.de><3D009334-DCFD-4AE5-A446-6C8165F2D77D@jerris.com><2ea4d47e0902040956v75c5472foa4649c50b7340484@mail.gmail.com> <4C4AD6CE-35C7-4F13-80C6-6F86DA71A51E@freeswitch.org> Message-ID: <43782D4C-3B15-427D-98A7-F72E3CE0F922@freeswitch.org> Its ok ;) We'll get you taken care of.. you should join us on IRC... #freenode its a faster way to get help. irc.freenode.net /b On Feb 4, 2009, at 4:54 PM, Nik Middleton wrote: > Sorry, 2 messages sent before I understood how this forum processes > posts -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090204/65288a5d/attachment.html From nik.middleton at noblesolutions.co.uk Wed Feb 4 15:05:52 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Wed, 4 Feb 2009 23:05:52 -0000 Subject: [Freeswitch-users] Gateway setting In-Reply-To: <43782D4C-3B15-427D-98A7-F72E3CE0F922@freeswitch.org> References: <4972046E.8020102@ewetel.de><200901291202.41144.stkn@freeswitch.org><191c3a030901290549x29c87976k68b3f9b455f00d0f@mail.gmail.com><4981B8E6.9070708@ewetel.de><5A2ED995-CF05-4AEF-9103-43BC84465514@freeswitch.org><4981D6F8.4060409@ewetel.de><191c3a030901290832k27a18b30u7bc531157fd89b19@mail.gmail.com><49830576.6080907@ewetel.de><4987359B.2020402@ewetel.de><3D009334-DCFD-4AE5-A446-6C8165F2D77D@jerris.com><2ea4d47e0902040956v75c5472foa4649c50b7340484@mail.gmail.com><4C4AD6CE-35C7-4F13-80C6-6F86DA71A51E@freeswitch.org> <43782D4C-3B15-427D-98A7-F72E3CE0F922@freeswitch.org> Message-ID: Well try as I might, I can't connect to that server, others are fine, but I get DNS pool errors ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: 04 February 2009 23:00 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Gateway setting Its ok ;) We'll get you taken care of.. you should join us on IRC... #freenode its a faster way to get help. irc.freenode.net /b On Feb 4, 2009, at 4:54 PM, Nik Middleton wrote: Sorry, 2 messages sent before I understood how this forum processes posts -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090204/ec162a50/attachment.html From nik.middleton at noblesolutions.co.uk Wed Feb 4 15:07:58 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Wed, 4 Feb 2009 23:07:58 -0000 Subject: [Freeswitch-users] FS in ISP Mode In-Reply-To: <05317B37-EDAF-4286-BE27-5DAB8C0A0289@freeswitch.org> References: <05317B37-EDAF-4286-BE27-5DAB8C0A0289@freeswitch.org> Message-ID: Sorry, not being clear. If external user dials a geo number, my pstn provider forwards call to 5060 at my server address. They expect me to be listening on 5060 Regards ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: 04 February 2009 22:56 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] FS in ISP Mode If your ITSP requires you to come from 5060 on your request then they are seriously broken. But yes you can move the ports around on the profiles or turn auth to false on the internal profile if you don't require any digest auth or phones registering. /b On Feb 4, 2009, at 4:50 PM, Nik Middleton wrote: Do apologise about the hijacking, Question: My ISP sends inbound calls via 5060, so it seems I need to renumber the ports, but that leaves my SIP end points who authenticate also needing 5060, can they be combined? Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090204/0eafa860/attachment.html From krice at freeswitch.org Wed Feb 4 15:11:07 2009 From: krice at freeswitch.org (Ken Rice) Date: Wed, 04 Feb 2009 17:11:07 -0600 Subject: [Freeswitch-users] FS in ISP Mode In-Reply-To: Message-ID: You can reverse the ports or try to get your provider to send to 5060 or you can bind an additional IP and have the external profile listen there K From: Nik Middleton Reply-To: Date: Wed, 4 Feb 2009 23:07:58 -0000 To: Subject: Re: [Freeswitch-users] FS in ISP Mode Sorry, not being clear. If external user dials a geo number, my pstn provider forwards call to 5060 at my server address. They expect me to be listening on 5060 Regards From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: 04 February 2009 22:56 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] FS in ISP Mode If your ITSP requires you to come from 5060 on your request then they are seriously broken. But yes you can move the ports around on the profiles or turn auth to false on the internal profile if you don't require any digest auth or phones registering. /b On Feb 4, 2009, at 4:50 PM, Nik Middleton wrote: Do apologise about the hijacking, Question: My ISP sends inbound calls via 5060, so it seems I need to renumber the ports, but that leaves my SIP end points who authenticate also needing 5060, can they be combined? Regards, _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090204/8d522b32/attachment-0001.html From brian at freeswitch.org Wed Feb 4 15:11:21 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 4 Feb 2009 17:11:21 -0600 Subject: [Freeswitch-users] Gateway setting In-Reply-To: References: <4972046E.8020102@ewetel.de><200901291202.41144.stkn@freeswitch.org><191c3a030901290549x29c87976k68b3f9b455f00d0f@mail.gmail.com><4981B8E6.9070708@ewetel.de><5A2ED995-CF05-4AEF-9103-43BC84465514@freeswitch.org><4981D6F8.4060409@ewetel.de><191c3a030901290832k27a18b30u7bc531157fd89b19@mail.gmail.com><49830576.6080907@ewetel.de><4987359B.2020402@ewetel.de><3D009334-DCFD-4AE5-A446-6C8165F2D77D@jerris.com><2ea4d47e0902040956v75c5472foa4649c50b7340484@mail.gmail.com><4C4AD6CE-35C7-4F13-80C6-6F86DA71A51E@freeswitch.org> <43782D4C-3B15-427D-98A7-F72E3CE0F922@freeswitch.org> Message-ID: <217639A0-29CD-410E-B571-FAD71636D083@freeswitch.org> You'll need to get an IRC client or do it from the www.freeswitch.org site on the right. /b On Feb 4, 2009, at 5:05 PM, Nik Middleton wrote: > Well try as I might, I can?t connect to that server, others are > fine, but I get DNS pool errors > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090204/c3f69e1b/attachment.html From nik.middleton at noblesolutions.co.uk Wed Feb 4 15:15:45 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Wed, 4 Feb 2009 23:15:45 -0000 Subject: [Freeswitch-users] FS in ISP Mode In-Reply-To: References: Message-ID: Ah Ha, that would work (second IP) Regards ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice Sent: 04 February 2009 23:11 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] FS in ISP Mode You can reverse the ports or try to get your provider to send to 5060 or you can bind an additional IP and have the external profile listen there K ________________________________ From: Nik Middleton Reply-To: Date: Wed, 4 Feb 2009 23:07:58 -0000 To: Subject: Re: [Freeswitch-users] FS in ISP Mode Sorry, not being clear. If external user dials a geo number, my pstn provider forwards call to 5060 at my server address. They expect me to be listening on 5060 Regards ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: 04 February 2009 22:56 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] FS in ISP Mode If your ITSP requires you to come from 5060 on your request then they are seriously broken. But yes you can move the ports around on the profiles or turn auth to false on the internal profile if you don't require any digest auth or phones registering. /b On Feb 4, 2009, at 4:50 PM, Nik Middleton wrote: Do apologise about the hijacking, Question: My ISP sends inbound calls via 5060, so it seems I need to renumber the ports, but that leaves my SIP end points who authenticate also needing 5060, can they be combined? Regards, ________________________________ _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090204/4d319639/attachment.html From saigop at gmail.com Wed Feb 4 21:11:26 2009 From: saigop at gmail.com (Gopalakrishnan A.N) Date: Thu, 5 Feb 2009 10:41:26 +0530 Subject: [Freeswitch-users] Q931 decoding Update In-Reply-To: <87f2f3b90902041252h5ede448bq720c15ea23dd517a@mail.gmail.com> References: <4972046E.8020102@ewetel.de> <4981B8E6.9070708@ewetel.de> <5A2ED995-CF05-4AEF-9103-43BC84465514@freeswitch.org> <4981D6F8.4060409@ewetel.de> <191c3a030901290832k27a18b30u7bc531157fd89b19@mail.gmail.com> <49830576.6080907@ewetel.de> <4987359B.2020402@ewetel.de> <3D009334-DCFD-4AE5-A446-6C8165F2D77D@jerris.com> <2ea4d47e0902040956v75c5472foa4649c50b7340484@mail.gmail.com> <87f2f3b90902041252h5ede448bq720c15ea23dd517a@mail.gmail.com> Message-ID: <2ea4d47e0902042111x5aa08410oe99c6ae02df3d7de@mail.gmail.com> Yes I can do that with any integration On Thu, Feb 5, 2009 at 2:22 AM, Michael Collins wrote: > On Wed, Feb 4, 2009 at 9:56 AM, Gopalakrishnan A.N > wrote: > > Hi, > > Its a awesome. Can the packet capturing be done with event socket? > > Not at this time. Would require some additional programming. Are you > up for the task? ;) > -MC > > > > > -- > > Thank you with regards, > > Gopal, > > > > > > > > > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Thank you with regards, Gopal, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090205/8e3617a4/attachment.html From sicfslist at gmail.com Wed Feb 4 23:33:07 2009 From: sicfslist at gmail.com (Shelby Ramsey) Date: Thu, 5 Feb 2009 01:33:07 -0600 Subject: [Freeswitch-users] XML CDR ERROR ... Message-ID: <35b355e90902042333p3009d6c1md2a76355a90509bd@mail.gmail.com> Hello, I'm having it on both Fedora and Ubuntu boxes: 2009-02-05 01:26:27 [ERR] mod_xml_cdr.c:290 mod_xml_cdr_load() Open of xml_cdr.conf failed 2009-02-05 01:26:27 [CRIT] switch_loadable_module.c:839 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_xml_cdr.so **Module load routine returned an error** Details: -- Ubuntu 6.04 LTS -- Fedora 8 Tried a couple of things: -- messing with libcurl -- ./configure --without-libcurl Thanks! SDR -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090205/1ced279a/attachment.html From brian at freeswitch.org Wed Feb 4 23:42:59 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 5 Feb 2009 01:42:59 -0600 Subject: [Freeswitch-users] XML CDR ERROR ... In-Reply-To: <35b355e90902042333p3009d6c1md2a76355a90509bd@mail.gmail.com> References: <35b355e90902042333p3009d6c1md2a76355a90509bd@mail.gmail.com> Message-ID: <640834B6-0A23-4855-BB0F-F3622AA334FF@freeswitch.org> Make sure your config file is installed and issue a reloadxml then load mod_xml_cdr /b On Feb 5, 2009, at 1:33 AM, Shelby Ramsey wrote: > Hello, > > I'm having it on both Fedora and Ubuntu boxes: > > 2009-02-05 01:26:27 [ERR] mod_xml_cdr.c:290 mod_xml_cdr_load() Open > of xml_cdr.conf failed > 2009-02-05 01:26:27 [CRIT] switch_loadable_module.c:839 > switch_loadable_module_load_file() Error Loading module /usr/local/ > freeswitch/mod/mod_xml_cdr.so > **Module load routine returned an error** > > Details: > -- Ubuntu 6.04 LTS > -- Fedora 8 > > Tried a couple of things: > -- messing with libcurl > -- ./configure --without-libcurl > > Thanks! > > SDR From kawarod at laposte.net Thu Feb 5 00:25:09 2009 From: kawarod at laposte.net (rod) Date: Thu, 05 Feb 2009 12:25:09 +0400 Subject: [Freeswitch-users] FreeSwitch setup as a "Dumb" SBC In-Reply-To: <49896E8D.3010609@post.cz> References: <4988C12E.1090109@post.cz> <191c3a030902031434v1129d684j18b09a954d871f7b@mail.gmail.com> <49896E8D.3010609@post.cz> Message-ID: <498AA265.4060307@laposte.net> Hi, how many static xml files did you create for your test ? rod. kokoska rokoska wrote: > Anthony Minessale napsal(a): > >> What does it look like if you serve the directory from the static xml >> file out of curiosity. >> >> > > Well, I write all user infos into static xml files loaded at startup :-) > > For the first try (without "tuning", see below) I can't go beyond 220 > reg/s - it is just about 10-12 % higher rate then with "lighty" etc. > BTW: Lighty alone can serve about 2600-2700 "dynamic directory xml files > from DB" per second - tested with "ab". > > The only difference (but big one :-) is that CPU utilization is below > 15% on all cores and load is about 0.2, so machine is idle :-) > > It tells me, that somthing is wron with my setup :-) > > > The only optimizations done are: > > 1. No logging > 2. FS in "high priority" mode > 3. "ulimits" applied > > Next (today evenings - tommorow mornings, don't know) I try to "tune" FS > for better preformance - as I did with previous test last month: > > 1. Move FS internal SQL light to ramdisk - I'm not sure if it helps, > because OS caches all HDD reads/writes, but if SQL light forces sync > after every DB update/insert it can make sense - I try it. > May be I move all FS dir to ramdisk. > > 2. Slightly change mod_sofia to disable NAT ping loop (unnecessary DB > operations) and, mainly, disable retrieving and sending of NOTIFY > messages containing VM info. > I'll look into my notes to see what I have done before and do the same... > > > Be patient please, as soon as I have the results, I post them here :-) > > > Best regards, > > kokoska.rokoska > > > >> On Tue, Feb 3, 2009 at 4:11 PM, kokoska.rokoska > > wrote: >> >> Ken Rice napsal(a): >> ... >> >> > On Registrations we have experienced Registration/second rates >> exceeding 150 >> > registrations per second using mod_xml_curl to feed the users >> directory. I >> > suspect, this number can be greatly increased if we were to feed >> directory >> > with something that cut out the apache and php over head >> > >> >> If someone interested I have few numbers on Registrar performance: >> >> DB server: >> 2x Quad core E5345 @ 2.33GHz, 16 GiB RAM >> Centos 5 x86_64, MySQL 5.0 >> >> Registrar server: >> 2x Quad core E5345 @ 2.33GHz, 16 GiB RAM >> Centos 5 x86_64 >> >> Tested using sipp with 10.000 and 30.000 "users". >> >> >> FreeSWITCH as registrar - current trunk: >> 1. FreeSwitch si simply modified (code doing NAT-ping is commented >> out :-) >> 2. Directory is served through lighttpd and simple "C" binary doing one >> trivial select. Lighttpd runs on the same machine as FS. When I move >> lighhtpd to another machine, I cannot see any significat performance >> boost. >> >> Result: I can go up to the 470-500 reg/s. and FS is heavy overloaded and >> retransmissions occurs. >> >> >> Kamailio as registrar - 1.4.3. no TLS: >> 1. Kamailio runs with usrloc db_mode 3 (no caching) >> >> Result: I can go up to the 3500-3700 reg/s. and Kamailio server is at >> 0.3 load and all 8 cores are bellow 15 %. Without retransmissions. The >> limit is DB throughput. >> Just for "curiosity" I switched userloc to db_mode 2 (write back) and at >> 5000 regs/s I stopped the sipp test, because I saw the bottle neck >> becomes the server runnig sipp (very old P4 box). >> >> >> Conclusion: >> While I see amazing FreeSWITCH performance on INVITEs per seconds and >> concurrent calls (another galaxy from * point of view :-), if you have >> to handle lots of registrations per second, it is IMO better to use >> Kamailio/OpenSIPS/SER as separate registrar and "propagate" users to FS >> through SQL view. >> >> Hope this helps someone... >> >> Best regards, >> >> kokoska.rokoska >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> >> iax:guest at conference.freeswitch.org/888 >> >> googletalk:conf+888 at conference.freeswitch.org >> >> pstn:213-799-1400 >> >> >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > From pmhshz at gmail.com Thu Feb 5 00:42:15 2009 From: pmhshz at gmail.com (shehzad p) Date: Thu, 5 Feb 2009 00:42:15 -0800 (PST) Subject: [Freeswitch-users] Freeswitch freezes on increasing call traffic In-Reply-To: <191c3a030902040551w5d867deta55c98bd3a3c03f8@mail.gmail.com> References: <21701744.post@talk.nabble.com> <191c3a030901280601t3e185ec2m76295110e4888837@mail.gmail.com> <21729863.post@talk.nabble.com> <191c3a030901290831y2ff329b3maee6c1aafa3c68e2@mail.gmail.com> <21745312.post@talk.nabble.com> <21749375.post@talk.nabble.com> <191c3a030901300818i5d85a7aawa0c53b4e54892f40@mail.gmail.com> <21761523.post@talk.nabble.com> <191c3a030901310850s553f8c11oc4e559a710def6d2@mail.gmail.com> <21825226.post@talk.nabble.com> <191c3a030902040551w5d867deta55c98bd3a3c03f8@mail.gmail.com> Message-ID: <21847332.post@talk.nabble.com> Hi anthony, In my previous post I already attached the BT for the testing of FS 1.0.1 Posting it again, please find it on link==>http://www.nabble.com/file/p21825226/bt_new_arch.txt Now I got the same result while testing FS 1.0.3RC1, And its BT is also same... BT Link: http://www.nabble.com/file/p21847332/fs_1_0_3_bt_new_arch.log fs_1_0_3_bt_new_arch.log (Note: Same in the sens the functions listed in the sequence are almost same as before...) Anthony Minessale-2 wrote: > > If you still get a crash on SVN trunk please post the bt even if you think > it's the same, since it won't be > exactly the same, the line numbers etc will be accurate with our > development > code making it easier to debug. > > > On Wed, Feb 4, 2009 at 12:38 AM, shehzad p wrote: > >> >> Hi anthony, >> >> I Modified the whole architecture of call routing system, >> Now after getting required routes, script exit and, >> control comes back to Dialplan, and call is bridged there, >> And call hangup, CDR is posted to cdr.php file (using xml_cdr). >> >> So now there is no blocking statement (bridge or anything like that) in >> current javascript, It return back control instantly. >> >> So, setting up all above architecture... >> First I tested FS 1.0.1 , It get crashed two times, in interval of 3 to 5 >> hours and simultaneous call of about 100 to 150. >> BT is the same as before... >> http://www.nabble.com/file/p21825226/bt_new_arch.txt bt_new_arch.txt >> >> Now I am also testing 1.0.3RC1, and post it back if any found. >> >> Thanks >> msp >> >> >> Clearly you have an issue with your javascript code. >> >> You have the Garbage collector blocking in every thread. >> >> Are you doing any endless loops in your code where you do not check >> session.ready() as a condition for >> continuing the script? >> >> any time session.ready() fails you must immediately exit. >> >> Are you using session.execute to execute long blocking operations like >> bridging many calls or entering a conference? >> You should avoid doing this as all the collective scripts on the system >> share a common Garbage Collector provided by the >> JS engine and it can lead to the exact issues you describe if the code is >> not properly designed. >> >> What else does you script do that are things provided by FS such as >> playing >> files and executing applications. >> >> >> >> >> -- >> View this message in context: >> http://www.nabble.com/Freeswitch-freezes-on-increasing-call-traffic-tp21701744p21825226.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/Freeswitch-freezes-on-increasing-call-traffic-tp21701744p21847332.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From brian at freeswitch.org Thu Feb 5 00:46:55 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 5 Feb 2009 02:46:55 -0600 Subject: [Freeswitch-users] Freeswitch freezes on increasing call traffic In-Reply-To: <21847332.post@talk.nabble.com> References: <21701744.post@talk.nabble.com> <191c3a030901280601t3e185ec2m76295110e4888837@mail.gmail.com> <21729863.post@talk.nabble.com> <191c3a030901290831y2ff329b3maee6c1aafa3c68e2@mail.gmail.com> <21745312.post@talk.nabble.com> <21749375.post@talk.nabble.com> <191c3a030901300818i5d85a7aawa0c53b4e54892f40@mail.gmail.com> <21761523.post@talk.nabble.com> <191c3a030901310850s553f8c11oc4e559a710def6d2@mail.gmail.com> <21825226.post@talk.nabble.com> <191c3a030902040551w5d867deta55c98bd3a3c03f8@mail.gmail.com> <21847332.post@talk.nabble.com> Message-ID: <4C2A4389-BEF6-454B-8B5F-FDCCC76E174A@freeswitch.org> Can you give me the output of uname -a and the contents of /proc/ cpuinfo? Not sure I asked for this info already or not. Thanks, Brian On Feb 5, 2009, at 2:42 AM, shehzad p wrote: > > Hi anthony, > In my previous post I already attached the BT for the testing of FS > 1.0.1 > Posting it again, please find it on > link==>http://www.nabble.com/file/p21825226/bt_new_arch.txt > > Now I got the same result while testing FS 1.0.3RC1, And its BT is > also > same... BT Link: > http://www.nabble.com/file/p21847332/fs_1_0_3_bt_new_arch.log > fs_1_0_3_bt_new_arch.log > > (Note: Same in the sens the functions listed in the sequence are > almost same > as before...) > > > Anthony Minessale-2 wrote: >> >> If you still get a crash on SVN trunk please post the bt even if >> you think >> it's the same, since it won't be >> exactly the same, the line numbers etc will be accurate with our >> development >> code making it easier to debug. >> >> >> On Wed, Feb 4, 2009 at 12:38 AM, shehzad p wrote: >> >>> >>> Hi anthony, >>> >>> I Modified the whole architecture of call routing system, >>> Now after getting required routes, script exit and, >>> control comes back to Dialplan, and call is bridged there, >>> And call hangup, CDR is posted to cdr.php file (using xml_cdr). >>> >>> So now there is no blocking statement (bridge or anything like >>> that) in >>> current javascript, It return back control instantly. >>> >>> So, setting up all above architecture... >>> First I tested FS 1.0.1 , It get crashed two times, in interval of >>> 3 to 5 >>> hours and simultaneous call of about 100 to 150. >>> BT is the same as before... >>> http://www.nabble.com/file/p21825226/bt_new_arch.txt bt_new_arch.txt >>> >>> Now I am also testing 1.0.3RC1, and post it back if any found. >>> >>> Thanks >>> msp >>> >>> >>> Clearly you have an issue with your javascript code. >>> >>> You have the Garbage collector blocking in every thread. >>> >>> Are you doing any endless loops in your code where you do not check >>> session.ready() as a condition for >>> continuing the script? >>> >>> any time session.ready() fails you must immediately exit. >>> >>> Are you using session.execute to execute long blocking operations >>> like >>> bridging many calls or entering a conference? >>> You should avoid doing this as all the collective scripts on the >>> system >>> share a common Garbage Collector provided by the >>> JS engine and it can lead to the exact issues you describe if the >>> code is >>> not properly designed. >>> >>> What else does you script do that are things provided by FS such as >>> playing >>> files and executing applications. >>> >>> >>> >>> >>> -- >>> View this message in context: >>> http://www.nabble.com/Freeswitch-freezes-on-increasing-call-traffic-tp21701744p21825226.html >>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com > > >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com> > >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org > > >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org> > >> pstn:213-799-1400 >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > View this message in context: http://www.nabble.com/Freeswitch-freezes-on-increasing-call-traffic-tp21701744p21847332.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From kokoska.rokoska at post.cz Thu Feb 5 00:49:48 2009 From: kokoska.rokoska at post.cz (kokoska rokoska) Date: Thu, 05 Feb 2009 09:49:48 +0100 Subject: [Freeswitch-users] FreeSwitch setup as a "Dumb" SBC In-Reply-To: <498AA265.4060307@laposte.net> References: <4988C12E.1090109@post.cz> <191c3a030902031434v1129d684j18b09a954d871f7b@mail.gmail.com> <49896E8D.3010609@post.cz> <498AA265.4060307@laposte.net> Message-ID: <498AA82C.4050302@post.cz> rod napsal(a): > Hi, > > how many static xml files did you create for your test ? > > rod. > Hi rod, I created 10.000 files xml directory files, but all of them were "included" into main FreeSWITCH xml config (using preprocessor). So I hope => no disk reads, all in memory (freeswitch.xml.fsxml is memory mapped)... Best regards, kokoska.rokoska From krice at suspicious.org Thu Feb 5 00:55:08 2009 From: krice at suspicious.org (Ken Rice) Date: Thu, 05 Feb 2009 02:55:08 -0600 Subject: [Freeswitch-users] FreeSwitch setup as a "Dumb" SBC In-Reply-To: <498AA82C.4050302@post.cz> Message-ID: That file is memory mapped but it can still take an IO hit... Mounting fs/db in a ramdrive still helps out and it doesn't have to be that big of a ram drive for the testing you are doing > From: kokoska rokoska > Reply-To: > Date: Thu, 05 Feb 2009 09:49:48 +0100 > To: > Subject: Re: [Freeswitch-users] FreeSwitch setup as a "Dumb" SBC > > > > > rod napsal(a): >> Hi, >> >> how many static xml files did you create for your test ? >> >> rod. >> > > Hi rod, > > I created 10.000 files xml directory files, but all of them were > "included" into main FreeSWITCH xml config (using preprocessor). So I > hope => no disk reads, all in memory (freeswitch.xml.fsxml is memory > mapped)... > > Best regards, > > kokoska.rokoska > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From kokoska.rokoska at post.cz Thu Feb 5 01:07:37 2009 From: kokoska.rokoska at post.cz (kokoska rokoska) Date: Thu, 05 Feb 2009 10:07:37 +0100 Subject: [Freeswitch-users] FreeSwitch setup as a "Dumb" SBC In-Reply-To: References: Message-ID: <498AAC59.60407@post.cz> Ken Rice napsal(a): > That file is memory mapped but it can still take an IO hit... Mounting fs/db > in a ramdrive still helps out and it doesn't have to be that big of a ram > drive for the testing you are doing > Thank you very much, Ken, for that info! I do it like I wrote before, but I have to wait till the "blades" aren't busy. They are not only for my testing and the other users have "tasks" with higher priority than my sipp testing :-) As soon as I have the result, I'll post them. I'm pretty sure, that changes in FS code and "all in RAM" helps much :-) Best regards, kokoska.rokoska From Claudio.Cavalera at italtel.it Thu Feb 5 01:11:37 2009 From: Claudio.Cavalera at italtel.it (Cavalera Claudio Luigi) Date: Thu, 5 Feb 2009 10:11:37 +0100 Subject: [Freeswitch-users] Errors compiling trunk on a fresh system In-Reply-To: Message-ID: freeswitch-users-bounces at lists.freeswitch.org wrote: > This should now be fixed in trunk in revision 11632. Can you please > test and confirm. > > Mike I confirm it's fixed (revision 11649), thanks as always. 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If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. ----------------------------------------------------------------------------------------------------- From pmhshz at gmail.com Thu Feb 5 01:51:37 2009 From: pmhshz at gmail.com (shehzad p) Date: Thu, 5 Feb 2009 01:51:37 -0800 (PST) Subject: [Freeswitch-users] Freeswitch freezes on increasing call traffic In-Reply-To: <4C2A4389-BEF6-454B-8B5F-FDCCC76E174A@freeswitch.org> References: <21701744.post@talk.nabble.com> <191c3a030901280601t3e185ec2m76295110e4888837@mail.gmail.com> <21729863.post@talk.nabble.com> <191c3a030901290831y2ff329b3maee6c1aafa3c68e2@mail.gmail.com> <21745312.post@talk.nabble.com> <21749375.post@talk.nabble.com> <191c3a030901300818i5d85a7aawa0c53b4e54892f40@mail.gmail.com> <21761523.post@talk.nabble.com> <191c3a030901310850s553f8c11oc4e559a710def6d2@mail.gmail.com> <21825226.post@talk.nabble.com> <191c3a030902040551w5d867deta55c98bd3a3c03f8@mail.gmail.com> <21847332.post@talk.nabble.com> <4C2A4389-BEF6-454B-8B5F-FDCCC76E174A@freeswitch.org> Message-ID: <21848148.post@talk.nabble.com> HI Brian, Output of ulimit -a and /proc/cpuinfo is attached... http://www.nabble.com/file/p21848148/12_ulimit_and_cpuinfo.log 12_ulimit_and_cpuinfo.log BUT...................... I am running the freeswitch using below command (So ulimit set according to Anthony's previous post): =================================================================================== ulimit -d unlimited; ulimit -f unlimited; ulimit -i unlimited; ulimit -n 999999; ulimit -q unlimited ; ulimit -u unlimited; ulimit -v unlimited ; ulimit -x unlimited; ulimit -s 244; ulimit -l unlimited; freeswitch =================================================================================== Thanks msp Brian West-3 wrote: > > Can you give me the output of uname -a and the contents of /proc/ > cpuinfo? Not sure I asked for this info already or not. > > Thanks, > Brian > > On Feb 5, 2009, at 2:42 AM, shehzad p wrote: > >> >> Hi anthony, >> In my previous post I already attached the BT for the testing of FS >> 1.0.1 >> Posting it again, please find it on >> link==>http://www.nabble.com/file/p21825226/bt_new_arch.txt >> >> Now I got the same result while testing FS 1.0.3RC1, And its BT is >> also >> same... BT Link: >> http://www.nabble.com/file/p21847332/fs_1_0_3_bt_new_arch.log >> fs_1_0_3_bt_new_arch.log >> >> (Note: Same in the sens the functions listed in the sequence are >> almost same >> as before...) >> >> >> Anthony Minessale-2 wrote: >>> >>> If you still get a crash on SVN trunk please post the bt even if >>> you think >>> it's the same, since it won't be >>> exactly the same, the line numbers etc will be accurate with our >>> development >>> code making it easier to debug. >>> >>> >>> On Wed, Feb 4, 2009 at 12:38 AM, shehzad p wrote: >>> >>>> >>>> Hi anthony, >>>> >>>> I Modified the whole architecture of call routing system, >>>> Now after getting required routes, script exit and, >>>> control comes back to Dialplan, and call is bridged there, >>>> And call hangup, CDR is posted to cdr.php file (using xml_cdr). >>>> >>>> So now there is no blocking statement (bridge or anything like >>>> that) in >>>> current javascript, It return back control instantly. >>>> >>>> So, setting up all above architecture... >>>> First I tested FS 1.0.1 , It get crashed two times, in interval of >>>> 3 to 5 >>>> hours and simultaneous call of about 100 to 150. >>>> BT is the same as before... >>>> http://www.nabble.com/file/p21825226/bt_new_arch.txt bt_new_arch.txt >>>> >>>> Now I am also testing 1.0.3RC1, and post it back if any found. >>>> >>>> Thanks >>>> msp >>>> >>>> >>>> Clearly you have an issue with your javascript code. >>>> >>>> You have the Garbage collector blocking in every thread. >>>> >>>> Are you doing any endless loops in your code where you do not check >>>> session.ready() as a condition for >>>> continuing the script? >>>> >>>> any time session.ready() fails you must immediately exit. >>>> >>>> Are you using session.execute to execute long blocking operations >>>> like >>>> bridging many calls or entering a conference? >>>> You should avoid doing this as all the collective scripts on the >>>> system >>>> share a common Garbage Collector provided by the >>>> JS engine and it can lead to the exact issues you describe if the >>>> code is >>>> not properly designed. >>>> >>>> What else does you script do that are things provided by FS such as >>>> playing >>>> files and executing applications. >>>> >>>> >>>> >>>> >>>> -- >>>> View this message in context: >>>> http://www.nabble.com/Freeswitch-freezes-on-increasing-call-traffic-tp21701744p21825226.html >>>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> >>> -- >>> Anthony Minessale II >>> >>> FreeSWITCH http://www.freeswitch.org/ >>> ClueCon http://www.cluecon.com/ >>> >>> AIM: anthm >>> MSN:anthony_minessale at hotmail.com >> > >>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com>> > >>> IRC: irc.freenode.net #freeswitch >>> >>> FreeSWITCH Developer Conference >>> sip:888 at conference.freeswitch.org >> > >>> iax:guest at conference.freeswitch.org/888 >>> googletalk:conf+888 at conference.freeswitch.org>> > >>> pstn:213-799-1400 >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> -- >> View this message in context: >> http://www.nabble.com/Freeswitch-freezes-on-increasing-call-traffic-tp21701744p21847332.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/Freeswitch-freezes-on-increasing-call-traffic-tp21701744p21848148.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From helmut.kuper at ewetel.de Thu Feb 5 01:57:21 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Thu, 05 Feb 2009 10:57:21 +0100 Subject: [Freeswitch-users] Q931 decoding Update In-Reply-To: <2ea4d47e0902042111x5aa08410oe99c6ae02df3d7de@mail.gmail.com> References: <4972046E.8020102@ewetel.de> <4981B8E6.9070708@ewetel.de> <5A2ED995-CF05-4AEF-9103-43BC84465514@freeswitch.org> <4981D6F8.4060409@ewetel.de> <191c3a030901290832k27a18b30u7bc531157fd89b19@mail.gmail.com> <49830576.6080907@ewetel.de> <4987359B.2020402@ewetel.de> <3D009334-DCFD-4AE5-A446-6C8165F2D77D@jerris.com> <2ea4d47e0902040956v75c5472foa4649c50b7340484@mail.gmail.com> <87f2f3b90902041252h5ede448bq720c15ea23dd517a@mail.gmail.com> <2ea4d47e0902042111x5aa08410oe99c6ae02df3d7de@mail.gmail.com> Message-ID: <498AB801.7030007@ewetel.de> Hello, if you have more than one call in your q931.pcap file captured, you may like to seperate the call flows in wireshark's packet list. wireshark allows you to sort the packets by e.g. q931 call reference first. All you have to do is this: Open q931 pcap file in wireshark, goto edit->preferences...->Columns Enter a title of your new column e.g "Q931 Call Ref" in title-field. Select "Custom" from Format-field. Enter exactly "q931.call_ref" without quotes into the field next to Format-field. Then apply it and close the window. Now you have a "Q931 Call Ref" column in the packet list. Click on it and the Flows a sorted first by "q931 call reference" and second by time. regards Helmut From Laurent.Fabre at kirranet.com Thu Feb 5 03:29:10 2009 From: Laurent.Fabre at kirranet.com (Laurent Fabre) Date: Thu, 5 Feb 2009 12:29:10 +0100 Subject: [Freeswitch-users] Directory User Password Message-ID: Hi everyone, Any chance I could populate user accounts with hashed passwords instead of cleartext ? If not, which block of code should I look for to propose a patch ? Thanks in advance, Laurent -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090205/616db3b6/attachment.html From hads at nice.net.nz Thu Feb 5 03:39:06 2009 From: hads at nice.net.nz (Hadley Rich) Date: Fri, 6 Feb 2009 00:39:06 +1300 Subject: [Freeswitch-users] Directory User Password In-Reply-To: References: Message-ID: <200902060039.07078.hads@nice.net.nz> On Friday 06 February 2009 00:29:10 Laurent Fabre wrote: > Hi everyone, > > Any chance I could populate user accounts with hashed passwords instead of > cleartext ? Sure can; http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide#Basic_User hads -- http://nicegear.co.nz New Zealands Open Source Hardware Supplier From jacredit at gmail.com Wed Feb 4 22:06:17 2009 From: jacredit at gmail.com (John Hyde) Date: Wed, 4 Feb 2009 22:06:17 -0800 Subject: [Freeswitch-users] does anyone have a working FS / aastra config Message-ID: <777d76f40902042206u7c448985i8690c6df4472f3b7@mail.gmail.com> I am having problems getting an Aastra 57i to make calls through FS. the phone registers fine, but all calls fail. If i use xlite or a nokia sip phone, i have no problems. Here is a packet capture of an attempted call: http://pastebin.freeswitch.org/7039 notice packet 9, it should have been a SIP INVITE, but it turned out to be a Fragmented IP protocol The phone and FS are both on the same lan subnet, and the phone connects fine with an asterisk server on the same subnet. Is there a known config for aastra phones that I can reference, or does anyone know why I am having this issue? -- john -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090204/39a5e4c6/attachment.html From jsokulski at dotsystems.pl Thu Feb 5 05:09:15 2009 From: jsokulski at dotsystems.pl (Jacek Sokulski) Date: Thu, 05 Feb 2009 14:09:15 +0100 Subject: [Freeswitch-users] origainate through sofia gateway In-Reply-To: <191c3a030902040822o1ffededbwd592361aa09f6b46@mail.gmail.com> References: <1233668179.5354.15.camel@dotw1126.dotsystems.pl> <1b46b4e80902030920p521d6079s4c9b78d8d39a924a@mail.gmail.com> <5E2B217D-8C85-48DB-800A-535A7FAAE24B@freeswitch.org> <1b46b4e80902030931y6a1679afw12ec37596ba9d77d@mail.gmail.com> <1233737494.5405.12.camel@dotw1126.dotsystems.pl> <191c3a030902040609s600798c8t720950a2e13ee05a@mail.gmail.com> <1233758767.5405.22.camel@dotw1126.dotsystems.pl> <191c3a030902040822o1ffededbwd592361aa09f6b46@mail.gmail.com> Message-ID: <1233839355.5346.31.camel@dotw1126.dotsystems.pl> the result of apiExecute("bgapi", "originate {effective_caller_id_number=fixed0248b,origination_caller_id_number=1000,ignore_early_media=true}user/1008 at 192.168.1.122 bridge:sofia/gateway/halonet/0225490317 inline"); > /: > 2009-02-05 14:04:48 [DEBUG] switch_ivr_originate.c:777 switch_ivr_originate() variable string 0 = [effective_caller_id_number=fixed0248b] > 2009-02-05 14:04:48 [DEBUG] switch_ivr_originate.c:777 switch_ivr_originate() variable string 1 = [origination_caller_id_number=1000] > 2009-02-05 14:04:48 [DEBUG] switch_ivr_originate.c:777 switch_ivr_originate() variable string 2 = [ignore_early_media=true] > 2009-02-05 14:04:48 [DEBUG] switch_ivr_originate.c:777 switch_ivr_originate() variable string 0 = [presence_id=1008 at 192.168.1.122] > 2009-02-05 14:04:48 [NOTICE] switch_channel.c:565 switch_channel_set_name() New Channel sofia/internal/sip:1008 at 192.168.1.126:5070 [f9b18b2e-b0a9-4e24-8452-40e1cce047bd] > 2009-02-05 14:04:48 [DEBUG] mod_sofia.c:2518 sofia_outgoing_channel() (sofia/internal/sip:1008 at 192.168.1.126:5070) State Change CS_NEW -> CS_INIT > 2009-02-05 14:04:48 [DEBUG] switch_core_session.c:807 switch_core_session_signal_state_change() Send signal sofia/internal/sip:1008 at 192.168.1.126:5070 [BREAK] > 2009-02-05 14:04:48 [DEBUG] switch_core_state_machine.c:379 switch_core_session_run() (sofia/internal/sip:1008 at 192.168.1.126:5070) Running State Change CS_INIT > 2009-02-05 14:04:48 [DEBUG] switch_core_state_machine.c:444 switch_core_session_run() (sofia/internal/sip:1008 at 192.168.1.126:5070) State INIT > 2009-02-05 14:04:48 [DEBUG] mod_sofia.c:83 sofia_on_init() sofia/internal/sip:1008 at 192.168.1.126:5070 SOFIA INIT > send 1196 bytes to udp/[192.168.1.126]:5070 at 13:04:48.127004: > ------------------------------------------------------------------------ > INVITE sip:1008 at 192.168.1.126:5070 SIP/2.0 > Via: SIP/2.0/UDP 192.168.1.122;rport;branch=z9hG4bKFpQjUmp1vNHZB > Max-Forwards: 70 > From: "FreeSWITCH" ;tag=9vpFm3tBc8gSe > To: > Call-ID: 68736fc7-6e28-122c-8396-000c7680c408 > CSeq: 110801208 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.2-wyeksportowane > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH > Supported: timer, precondition, path, replaces > Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer > Min-SE: 120 > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 335 > Remote-Party-ID: "FreeSWITCH" ;screen=yes;privacy=off > > v=0 > o=FreeSWITCH 1927164999227225404 3276738624485570081 IN IP4 192.168.1.122 > s=FreeSWITCH > c=IN IP4 192.168.1.122 > t=0 0 > m=audio 27044 RTP/AVP 9 0 8 3 101 13 > a=rtpmap:9 G722/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=rtpmap:13 CN/8000 > a=ptime:20 > ------------------------------------------------------------------------ > 2009-02-05 14:04:48 [DEBUG] mod_sofia.c:111 sofia_on_init() (sofia/internal/sip:1008 at 192.168.1.126:5070) State Change CS_INIT -> CS_ROUTING > 2009-02-05 14:04:48 [DEBUG] switch_core_session.c:807 switch_core_session_signal_state_change() Send signal sofia/internal/sip:1008 at 192.168.1.126:5070 [BREAK] > 2009-02-05 14:04:48 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state() Channel sofia/internal/sip:1008 at 192.168.1.126:5070 entering state [calling] > 2009-02-05 14:04:48 [DEBUG] switch_core_state_machine.c:444 switch_core_session_run() (sofia/internal/sip:1008 at 192.168.1.126:5070) State INIT going to sleep > 2009-02-05 14:04:48 [DEBUG] switch_core_state_machine.c:379 switch_core_session_run() (sofia/internal/sip:1008 at 192.168.1.126:5070) Running State Change CS_ROUTING > 2009-02-05 14:04:48 [DEBUG] switch_core_state_machine.c:447 switch_core_session_run() (sofia/internal/sip:1008 at 192.168.1.126:5070) State ROUTING > 2009-02-05 14:04:48 [DEBUG] mod_sofia.c:130 sofia_on_routing() sofia/internal/sip:1008 at 192.168.1.126:5070 SOFIA ROUTING > 2009-02-05 14:04:48 [DEBUG] switch_ivr_originate.c:58 originate_on_routing() (sofia/internal/sip:1008 at 192.168.1.126:5070) State Change CS_ROUTING -> CS_CONSUME_MEDIA > 2009-02-05 14:04:48 [DEBUG] switch_core_session.c:807 switch_core_session_signal_state_change() Send signal sofia/internal/sip:1008 at 192.168.1.126:5070 [BREAK] > 2009-02-05 14:04:48 [DEBUG] switch_core_state_machine.c:447 switch_core_session_run() (sofia/internal/sip:1008 at 192.168.1.126:5070) State ROUTING going to sleep > 2009-02-05 14:04:48 [DEBUG] switch_core_state_machine.c:379 switch_core_session_run() (sofia/internal/sip:1008 at 192.168.1.126:5070) Running State Change CS_CONSUME_MEDIA > 2009-02-05 14:04:48 [DEBUG] switch_core_state_machine.c:466 switch_core_session_run() (sofia/internal/sip:1008 at 192.168.1.126:5070) State CONSUME_MEDIA > recv 361 bytes from udp/[192.168.1.126]:5070 at 13:04:48.130745: > ------------------------------------------------------------------------ > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 192.168.1.122;rport;branch=z9hG4bKFpQjUmp1vNHZB > From: "FreeSWITCH" ;tag=9vpFm3tBc8gSe > To: ;tag=2111843199 > Contact: > Call-ID: 68736fc7-6e28-122c-8396-000c7680c408 > CSeq: 110801208 INVITE > Server: X-Lite release 1105d > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 362 bytes from udp/[192.168.1.126]:5070 at 13:04:48.181571: > ------------------------------------------------------------------------ > SIP/2.0 180 Ringing > Via: SIP/2.0/UDP 192.168.1.122;rport;branch=z9hG4bKFpQjUmp1vNHZB > From: "FreeSWITCH" ;tag=9vpFm3tBc8gSe > To: ;tag=2111843199 > Contact: > Call-ID: 68736fc7-6e28-122c-8396-000c7680c408 > CSeq: 110801208 INVITE > Server: X-Lite release 1105d > Content-Length: 0 > > ------------------------------------------------------------------------ > 2009-02-05 14:04:48 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state() Channel sofia/internal/sip:1008 at 192.168.1.126:5070 entering state [proceeding] > 2009-02-05 14:04:48 [NOTICE] sofia.c:2596 sofia_handle_sip_i_state() Ring-Ready sofia/internal/sip:1008 at 192.168.1.126:5070! > recv 678 bytes from udp/[192.168.1.126]:5070 at 13:04:51.914714: > ------------------------------------------------------------------------ > SIP/2.0 200 Ok > Via: SIP/2.0/UDP 192.168.1.122;rport;branch=z9hG4bKFpQjUmp1vNHZB > From: "FreeSWITCH" ;tag=9vpFm3tBc8gSe > To: ;tag=2111843199 > Contact: > Call-ID: 68736fc7-6e28-122c-8396-000c7680c408 > CSeq: 110801208 INVITE > Content-Type: application/sdp > Server: X-Lite release 1105d > Content-Length: 288 > > v=0 > o=1008 1183460117 1183463903 IN IP4 192.168.1.126 > s=X-Lite > c=IN IP4 192.168.1.126 > t=0 0 > m=audio 9000 RTP/AVP 0 8 98 97 101 > a=rtpmap:0 pcmu/8000 > a=rtpmap:8 pcma/8000 > a=rtpmap:98 iLBC/8000 > a=rtpmap:97 speex/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=sendrecv > ------------------------------------------------------------------------ > send 372 bytes to udp/[192.168.1.126]:5070 at 13:04:51.915303: > ------------------------------------------------------------------------ > ACK sip:1008 at 192.168.1.126:5070 SIP/2.0 > Via: SIP/2.0/UDP 192.168.1.122;rport;branch=z9hG4bKgZgBXF74Sy7HQ > Max-Forwards: 70 > From: "FreeSWITCH" ;tag=9vpFm3tBc8gSe > To: ;tag=2111843199 > Call-ID: 68736fc7-6e28-122c-8396-000c7680c408 > CSeq: 110801208 ACK > Contact: > Content-Length: 0 > > ------------------------------------------------------------------------ > 2009-02-05 14:04:51 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state() Channel sofia/internal/sip:1008 at 192.168.1.126:5070 entering state [ready] > 2009-02-05 14:04:51 [DEBUG] sofia.c:2546 sofia_handle_sip_i_state() Remote SDP: > v=0 > o=1008 1183460117 1183463903 IN IP4 192.168.1.126 > s=X-Lite > c=IN IP4 192.168.1.126 > t=0 0 > m=audio 9000 RTP/AVP 0 8 98 97 101 > a=rtpmap:0 pcmu/8000 > a=rtpmap:8 pcma/8000 > a=rtpmap:98 iLBC/8000 > a=rtpmap:97 speex/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > > 2009-02-05 14:04:51 [DEBUG] sofia_glue.c:2509 sofia_glue_negotiate_sdp() Audio Codec Compare [pcmu:0:8000]/[G722:9:8000] > 2009-02-05 14:04:51 [DEBUG] sofia_glue.c:2509 sofia_glue_negotiate_sdp() Audio Codec Compare [pcmu:0:8000]/[PCMU:0:8000] > 2009-02-05 14:04:51 [DEBUG] sofia_glue.c:1670 sofia_glue_tech_set_codec() Set Codec sofia/internal/sip:1008 at 192.168.1.126:5070 PCMU/8000 20 ms 160 samples > 2009-02-05 14:04:51 [DEBUG] sofia_glue.c:2473 sofia_glue_negotiate_sdp() Set 2833 dtmf payload to 101 > 2009-02-05 14:04:51 [DEBUG] sofia_glue.c:1890 sofia_glue_activate_rtp() AUDIO RTP [sofia/internal/sip:1008 at 192.168.1.126:5070] 192.168.1.122 port 27044 -> 192.168.1.126 port 9000 codec: 0 ms: 20 > 2009-02-05 14:04:51 [DEBUG] switch_rtp.c:865 switch_rtp_create() Starting timer [soft] 160 bytes per 20000ms > 2009-02-05 14:04:51 [NOTICE] sofia.c:3031 sofia_handle_sip_i_state() Channel [sofia/internal/sip:1008 at 192.168.1.126:5070] has been answered > 2009-02-05 14:04:51 [DEBUG] switch_channel.c:177 switch_channel_audio_sync() sofia/internal/sip:1008 at 192.168.1.126:5070 receive message [AUDIO_SYNC] > 2009-02-05 14:04:51 [DEBUG] switch_ivr_originate.c:1627 switch_ivr_originate() Originate Resulted in Success: [sofia/internal/sip:1008 at 192.168.1.126:5070] > 2009-02-05 14:04:51 [DEBUG] switch_channel.c:177 switch_channel_audio_sync() sofia/internal/sip:1008 at 192.168.1.126:5070 receive message [AUDIO_SYNC] > 2009-02-05 14:04:51 [DEBUG] switch_ivr_originate.c:1627 switch_ivr_originate() Originate Resulted in Success: [sofia/internal/sip:1008 at 192.168.1.126:5070] > 2009-02-05 14:04:51 [DEBUG] switch_channel.c:177 switch_channel_audio_sync() sofia/internal/sip:1008 at 192.168.1.126:5070 receive message [AUDIO_SYNC] > 2009-02-05 14:04:51 [DEBUG] switch_ivr.c:1245 switch_ivr_session_transfer() (sofia/internal/sip:1008 at 192.168.1.126:5070) State Change CS_CONSUME_MEDIA -> CS_ROUTING > 2009-02-05 14:04:51 [DEBUG] switch_core_session.c:807 switch_core_session_signal_state_change() Send signal sofia/internal/sip:1008 at 192.168.1.126:5070 [BREAK] > 2009-02-05 14:04:51 [DEBUG] switch_ivr.c:1249 switch_ivr_session_transfer() sofia/internal/sip:1008 at 192.168.1.126:5070 receive message [TRANSFER] > 2009-02-05 14:04:51 [DEBUG] switch_core_session.c:511 switch_core_session_perform_receive_message() Send signal sofia/internal/sip:1008 at 192.168.1.126:5070 [BREAK] > 2009-02-05 14:04:51 [NOTICE] switch_ivr.c:1251 switch_ivr_session_transfer() Transfer sofia/internal/sip:1008 at 192.168.1.126:5070 to inline[bridge:sofia/gateway/halonet/0225490317 at default] > 2009-02-05 14:04:51 [DEBUG] switch_core_state_machine.c:466 switch_core_session_run() (sofia/internal/sip:1008 at 192.168.1.126:5070) State CONSUME_MEDIA going to sleep > 2009-02-05 14:04:51 [DEBUG] switch_core_state_machine.c:379 switch_core_session_run() (sofia/internal/sip:1008 at 192.168.1.126:5070) Running State Change CS_ROUTING > 2009-02-05 14:04:51 [DEBUG] switch_core_state_machine.c:447 switch_core_session_run() (sofia/internal/sip:1008 at 192.168.1.126:5070) State ROUTING > 2009-02-05 14:04:51 [DEBUG] mod_sofia.c:130 sofia_on_routing() sofia/internal/sip:1008 at 192.168.1.126:5070 SOFIA ROUTING > 2009-02-05 14:04:51 [DEBUG] switch_core_state_machine.c:64 switch_core_standard_on_routing() sofia/internal/sip:1008 at 192.168.1.126:5070 Standard ROUTING > 2009-02-05 14:04:51 [DEBUG] switch_core_state_machine.c:100 switch_core_standard_on_routing() (sofia/internal/sip:1008 at 192.168.1.126:5070) State Change CS_ROUTING -> CS_EXECUTE > 2009-02-05 14:04:51 [DEBUG] switch_core_session.c:807 switch_core_session_signal_state_change() Send signal sofia/internal/sip:1008 at 192.168.1.126:5070 [BREAK] > 2009-02-05 14:04:51 [DEBUG] switch_core_state_machine.c:447 switch_core_session_run() (sofia/internal/sip:1008 at 192.168.1.126:5070) State ROUTING going to sleep > 2009-02-05 14:04:51 [DEBUG] switch_core_state_machine.c:379 switch_core_session_run() (sofia/internal/sip:1008 at 192.168.1.126:5070) Running State Change CS_EXECUTE > 2009-02-05 14:04:51 [DEBUG] switch_core_state_machine.c:454 switch_core_session_run() (sofia/internal/sip:1008 at 192.168.1.126:5070) State EXECUTE > 2009-02-05 14:04:51 [DEBUG] mod_sofia.c:173 sofia_on_execute() sofia/internal/sip:1008 at 192.168.1.126:5070 SOFIA EXECUTE > 2009-02-05 14:04:51 [DEBUG] switch_core_state_machine.c:137 switch_core_standard_on_execute() sofia/internal/sip:1008 at 192.168.1.126:5070 Standard EXECUTE > 2009-02-05 14:04:51 [DEBUG] switch_core_state_machine.c:152 switch_core_standard_on_execute() sofia/internal/sip:1008 at 192.168.1.126:5070 Execute bridge(sofia/gateway/halonet/0225490317) > 2009-02-05 14:04:51 [NOTICE] switch_channel.c:565 switch_channel_set_name() New Channel sofia/external/0225490317 [bf4fece9-38fc-40fa-8e9e-91f836f05e55] > 2009-02-05 14:04:51 [DEBUG] mod_sofia.c:2518 sofia_outgoing_channel() (sofia/external/0225490317) State Change CS_NEW -> CS_INIT > 2009-02-05 14:04:51 [DEBUG] switch_core_session.c:807 switch_core_session_signal_state_change() Send signal sofia/external/0225490317 [BREAK] > 2009-02-05 14:04:51 [DEBUG] switch_core_state_machine.c:379 switch_core_session_run() (sofia/external/0225490317) Running State Change CS_INIT > 2009-02-05 14:04:51 [DEBUG] switch_core_state_machine.c:444 switch_core_session_run() (sofia/external/0225490317) State INIT > 2009-02-05 14:04:51 [DEBUG] mod_sofia.c:83 sofia_on_init() sofia/external/0225490317 SOFIA INIT > 2009-02-05 14:04:52 [DEBUG] sofia_glue.c:566 sofia_glue_ext_address_lookup() STUN Success [89.77.89.244]:[50863] > 2009-02-05 14:04:52 [DEBUG] mod_sofia.c:111 sofia_on_init() (sofia/external/0225490317) State Change CS_INIT -> CS_ROUTING > 2009-02-05 14:04:52 [DEBUG] switch_core_session.c:807 switch_core_session_signal_state_change() Send signal sofia/external/0225490317 [BREAK] > 2009-02-05 14:04:52 [DEBUG] switch_core_state_machine.c:444 switch_core_session_run() (sofia/external/0225490317) State INIT going to sleep > 2009-02-05 14:04:52 [DEBUG] switch_core_state_machine.c:379 switch_core_session_run() (sofia/external/0225490317) Running State Change CS_ROUTING > 2009-02-05 14:04:52 [DEBUG] switch_core_state_machine.c:447 switch_core_session_run() (sofia/external/0225490317) State ROUTING > 2009-02-05 14:04:52 [DEBUG] mod_sofia.c:130 sofia_on_routing() sofia/external/0225490317 SOFIA ROUTING > 2009-02-05 14:04:52 [DEBUG] switch_ivr_originate.c:58 originate_on_routing() (sofia/external/0225490317) State Change CS_ROUTING -> CS_CONSUME_MEDIA > 2009-02-05 14:04:52 [DEBUG] switch_core_session.c:807 switch_core_session_signal_state_change() Send signal sofia/external/0225490317 [BREAK] > 2009-02-05 14:04:52 [DEBUG] switch_core_state_machine.c:447 switch_core_session_run() (sofia/external/0225490317) State ROUTING going to sleep > 2009-02-05 14:04:52 [DEBUG] switch_core_state_machine.c:379 switch_core_session_run() (sofia/external/0225490317) Running State Change CS_CONSUME_MEDIA > 2009-02-05 14:04:52 [DEBUG] switch_core_state_machine.c:466 switch_core_session_run() (sofia/external/0225490317) State CONSUME_MEDIA > 2009-02-05 14:04:52 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state() Channel sofia/external/0225490317 entering state [calling] > send 1117 bytes to udp/[194.9.25.21]:5060 at 13:04:52.572008: > ------------------------------------------------------------------------ > INVITE sip:0225490317 at sip.halonet.pl SIP/2.0 > Via: SIP/2.0/UDP 83.13.98.165:5080;rport;branch=z9hG4bKyvKBDBpvBQe2p > Max-Forwards: 69 > From: "Extension 1008" ;tag=FZp353yettFNN > To: > Call-ID: 6b05cee4-6e28-122c-8396-000c7680c408 > CSeq: 110801210 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.2-wyeksportowane > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO > Supported: timer, precondition, path, replaces > Allow-Events: talk, refer > Min-SE: 120 > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 309 > Remote-Party-ID: "Extension 1008" ;screen=yes;privacy=off > > v=0 > o=FreeSWITCH 3996368035436745512 7588776177330183331 IN IP4 89.77.89.244 > s=FreeSWITCH > c=IN IP4 89.77.89.244 > t=0 0 > m=audio 50863 RTP/AVP 0 8 3 101 13 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=rtpmap:13 CN/8000 > a=ptime:20 > ------------------------------------------------------------------------ > recv 760 bytes from udp/[194.9.25.21]:5060 at 13:04:52.993246: > ------------------------------------------------------------------------ > SIP/2.0 407 Proxy Authentication Required > Via: SIP/2.0/UDP 83.13.98.165:5080;rport=5080;branch=z9hG4bKyvKBDBpvBQe2p;received=89.77.89.244 > From: "Extension 1008" ;tag=FZp353yettFNN > To: ;tag=03bd8e75ec97c9ee65c772e401792a5a.5fa3 > Call-ID: 6b05cee4-6e28-122c-8396-000c7680c408 > CSeq: 110801210 INVITE > Proxy-Authenticate: Digest realm="sip.halonet.pl", nonce="498ae51439454939d240f51ab9455e2c8505c7aa", qop="auth" > Server: Sip EXpress router (2.0.0-rc1 (i386/linux)) > Content-Length: 0 > Warning: 392 194.9.25.21:5060 "Noisy feedback tells: pid=22400 req_src_ip=89.77.89.244 req_src_port=50866 in_uri=sip:0225490317 at sip.halonet.pl out_uri=sip:0225490317 at sip.halonet.pl via_cnt==1" > > ------------------------------------------------------------------------ > send 387 bytes to udp/[194.9.25.21]:5060 at 13:04:52.993616: > ------------------------------------------------------------------------ > ACK sip:0225490317 at sip.halonet.pl SIP/2.0 > Via: SIP/2.0/UDP 83.13.98.165:5080;rport;branch=z9hG4bKyvKBDBpvBQe2p > Max-Forwards: 69 > From: "Extension 1008" ;tag=FZp353yettFNN > To: ;tag=03bd8e75ec97c9ee65c772e401792a5a.5fa3 > Call-ID: 6b05cee4-6e28-122c-8396-000c7680c408 > CSeq: 110801210 ACK > Content-Length: 0 > > ------------------------------------------------------------------------ > send 1409 bytes to udp/[194.9.25.21]:5060 at 13:04:52.994530: > ------------------------------------------------------------------------ > INVITE sip:0225490317 at sip.halonet.pl SIP/2.0 > Via: SIP/2.0/UDP 83.13.98.165:5080;rport;branch=z9hG4bKZ5c4e66Z8Z4mj > Max-Forwards: 69 > From: "Extension 1008" ;tag=FZp353yettFNN > To: > Call-ID: 6b05cee4-6e28-122c-8396-000c7680c408 > CSeq: 110801211 INVITE > Contact: > Expires: 600 > User-Agent: FreeSWITCH-mod_sofia/1.0.2-wyeksportowane > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO > Supported: timer, precondition, path, replaces > Allow-Events: talk, refer > Proxy-Authorization: Digest username="fixed0248b", realm="sip.halonet.pl", nonce="498ae51439454939d240f51ab9455e2c8505c7aa", cnonce="a1oY524oEiyWgwAMdoDECA", algorithm=MD5, uri="sip:0225490317 at sip.halonet.pl", response="f05fad749d4ff6d5a75d61e3283c3afa", qop=auth, nc=00000001 > Min-SE: 120 > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 309 > Remote-Party-ID: "Extension 1008" ;screen=yes;privacy=off > > v=0 > o=FreeSWITCH 3996368035436745512 7588776177330183331 IN IP4 89.77.89.244 > s=FreeSWITCH > c=IN IP4 89.77.89.244 > t=0 0 > m=audio 50863 RTP/AVP 0 8 3 101 13 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=rtpmap:13 CN/8000 > a=ptime:20 > ------------------------------------------------------------------------ > 2009-02-05 14:04:52 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state() Channel sofia/external/0225490317 entering state [calling] > recv 618 bytes from udp/[194.9.25.21]:5060 at 13:04:53.349700: > ------------------------------------------------------------------------ > SIP/2.0 100 trying -- your call is important to us > Via: SIP/2.0/UDP 83.13.98.165:5080;rport=5080;branch=z9hG4bKZ5c4e66Z8Z4mj;received=89.77.89.244 > From: "Extension 1008" ;tag=FZp353yettFNN > To: > Call-ID: 6b05cee4-6e28-122c-8396-000c7680c408 > CSeq: 110801211 INVITE > Server: Sip EXpress router (2.0.0-rc1 (i386/linux)) > Content-Length: 0 > Warning: 392 194.9.25.21:5060 "Noisy feedback tells: pid=22332 req_src_ip=89.77.89.244 req_src_port=50866 in_uri=sip:0225490317 at sip.halonet.pl out_uri=sip:0225490317 at 217.11.128.50:5060 via_cnt==1" > > ------------------------------------------------------------------------ > recv 469 bytes from udp/[194.9.25.21]:5060 at 13:04:53.861812: > ------------------------------------------------------------------------ > SIP/2.0 500 Internal Server Error > Via: SIP/2.0/UDP 83.13.98.165:5080;received=89.77.89.244;rport=5080;branch=z9hG4bKZ5c4e66Z8Z4mj > From: "Extension 1008" ;tag=FZp353yettFNN > To: ;tag=4961D2C4-DA7 > Date: Thu, 05 Feb 2009 13:04:39 GMT > Call-ID: 6b05cee4-6e28-122c-8396-000c7680c408 > Server: Cisco-SIPGateway/IOS-12.x > CSeq: 110801211 INVITE > Allow-Events: telephone-event > Content-Length: 0 > > ------------------------------------------------------------------------ > send 362 bytes to udp/[194.9.25.21]:5060 at 13:04:53.862199: > ------------------------------------------------------------------------ > ACK sip:0225490317 at sip.halonet.pl SIP/2.0 > Via: SIP/2.0/UDP 83.13.98.165:5080;rport;branch=z9hG4bKZ5c4e66Z8Z4mj > Max-Forwards: 69 > From: "Extension 1008" ;tag=FZp353yettFNN > To: ;tag=4961D2C4-DA7 > Call-ID: 6b05cee4-6e28-122c-8396-000c7680c408 > CSeq: 110801211 ACK > Content-Length: 0 > > ------------------------------------------------------------------------ > 2009-02-05 14:04:53 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state() Channel sofia/external/0225490317 entering state [terminated] > 2009-02-05 14:04:53 [NOTICE] sofia.c:3090 sofia_handle_sip_i_state() Hangup sofia/external/0225490317 [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE] > 2009-02-05 14:04:53 [DEBUG] switch_channel.c:1494 switch_channel_perform_hangup() Send signal sofia/external/0225490317 [KILL] > 2009-02-05 14:04:53 [DEBUG] switch_core_session.c:807 switch_core_session_signal_state_change() Send signal sofia/external/0225490317 [BREAK] > 2009-02-05 14:04:53 [DEBUG] switch_core_state_machine.c:466 switch_core_session_run() (sofia/external/0225490317) State CONSUME_MEDIA going to sleep > 2009-02-05 14:04:53 [DEBUG] switch_core_state_machine.c:379 switch_core_session_run() (sofia/external/0225490317) Running State Change CS_HANGUP > 2009-02-05 14:04:53 [DEBUG] switch_core_state_machine.c:410 switch_core_session_run() (sofia/external/0225490317) State HANGUP > 2009-02-05 14:04:53 [DEBUG] mod_sofia.c:253 sofia_on_hangup() sofia/external/0225490317 Overriding SIP cause 503 with 500 from the other leg > 2009-02-05 14:04:53 [DEBUG] mod_sofia.c:287 sofia_on_hangup() Channel sofia/external/0225490317 hanging up, cause: NORMAL_TEMPORARY_FAILURE > 2009-02-05 14:04:53 [DEBUG] switch_core_state_machine.c:46 switch_core_standard_on_hangup() sofia/external/0225490317 Standard HANGUP, cause: NORMAL_TEMPORARY_FAILURE > 2009-02-05 14:04:53 [DEBUG] switch_core_state_machine.c:410 switch_core_session_run() (sofia/external/0225490317) State HANGUP going to sleep > 2009-02-05 14:04:53 [DEBUG] switch_core_session.c:939 switch_core_session_thread() Session 16 (sofia/external/0225490317) Locked, Waiting on external entities > 2009-02-05 14:04:53 [DEBUG] switch_ivr_originate.c:1695 switch_ivr_originate() Originate Resulted in Error Cause: 41 [NORMAL_TEMPORARY_FAILURE] > 2009-02-05 14:04:53 [DEBUG] switch_channel.c:177 switch_channel_audio_sync() sofia/internal/sip:1008 at 192.168.1.126:5070 receive message [AUDIO_SYNC] > 2009-02-05 14:04:53 [NOTICE] switch_core_session.c:957 switch_core_session_thread() Session 16 (sofia/external/0225490317) Ended > 2009-02-05 14:04:53 [INFO] mod_dptools.c:1909 audio_bridge_function() Originate Failed. Cause: NORMAL_TEMPORARY_FAILURE > 2009-02-05 14:04:53 [NOTICE] mod_dptools.c:1936 audio_bridge_function() Hangup sofia/internal/sip:1008 at 192.168.1.126:5070 [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE] > 2009-02-05 14:04:53 [NOTICE] switch_core_session.c:959 switch_core_session_thread() Close Channel sofia/external/0225490317 [CS_HANGUP] > 2009-02-05 14:04:53 [DEBUG] switch_channel.c:1494 switch_channel_perform_hangup() Send signal sofia/internal/sip:1008 at 192.168.1.126:5070 [KILL] > 2009-02-05 14:04:53 [DEBUG] switch_core_session.c:807 switch_core_session_signal_state_change() Send signal sofia/internal/sip:1008 at 192.168.1.126:5070 [BREAK] > 2009-02-05 14:04:53 [DEBUG] switch_core_state_machine.c:454 switch_core_session_run() (sofia/internal/sip:1008 at 192.168.1.126:5070) State EXECUTE going to sleep > 2009-02-05 14:04:53 [DEBUG] switch_core_state_machine.c:379 switch_core_session_run() (sofia/internal/sip:1008 at 192.168.1.126:5070) Running State Change CS_HANGUP > 2009-02-05 14:04:53 [DEBUG] switch_core_state_machine.c:410 switch_core_session_run() (sofia/internal/sip:1008 at 192.168.1.126:5070) State HANGUP > 2009-02-05 14:04:53 [DEBUG] mod_sofia.c:253 sofia_on_hangup() sofia/internal/sip:1008 at 192.168.1.126:5070 Overriding SIP cause 503 with 500 from the other leg > 2009-02-05 14:04:53 [DEBUG] mod_sofia.c:287 sofia_on_hangup() Channel sofia/internal/sip:1008 at 192.168.1.126:5070 hanging up, cause: NORMAL_TEMPORARY_FAILURE > 2009-02-05 14:04:53 [DEBUG] mod_sofia.c:344 sofia_on_hangup() Sending BYE to sofia/internal/sip:1008 at 192.168.1.126:5070 > send 648 bytes to udp/[192.168.1.126]:5070 at 13:04:53.868506: > ------------------------------------------------------------------------ > BYE sip:1008 at 192.168.1.126:5070 SIP/2.0 > Via: SIP/2.0/UDP 192.168.1.122;rport;branch=z9hG4bKH893yar8p7X4j > Max-Forwards: 70 > From: "FreeSWITCH" ;tag=9vpFm3tBc8gSe > To: ;tag=2111843199 > Call-ID: 68736fc7-6e28-122c-8396-000c7680c408 > CSeq: 110801209 BYE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.2-wyeksportowane > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH > Supported: timer, precondition, path, replaces > Reason: Q.850;cause=41;text="NORMAL_TEMPORARY_FAILURE" > Content-Length: 0 > > ------------------------------------------------------------------------ > 2009-02-05 14:04:53 [DEBUG] switch_core_state_machine.c:46 switch_core_standard_on_hangup() sofia/internal/sip:1008 at 192.168.1.126:5070 Standard HANGUP, cause: NORMAL_TEMPORARY_FAILURE > 2009-02-05 14:04:53 [DEBUG] switch_core_state_machine.c:410 switch_core_session_run() (sofia/internal/sip:1008 at 192.168.1.126:5070) State HANGUP going to sleep > 2009-02-05 14:04:53 [DEBUG] switch_core_session.c:939 switch_core_session_thread() Session 15 (sofia/internal/sip:1008 at 192.168.1.126:5070) Locked, Waiting on external entities > 2009-02-05 14:04:53 [NOTICE] switch_core_session.c:957 switch_core_session_thread() Session 15 (sofia/internal/sip:1008 at 192.168.1.126:5070) Ended > 2009-02-05 14:04:53 [NOTICE] switch_core_session.c:959 switch_core_session_thread() Close Channel sofia/internal/sip:1008 at 192.168.1.126:5070 [CS_HANGUP] > recv 354 bytes from udp/[192.168.1.126]:5070 at 13:04:53.882344: > ------------------------------------------------------------------------ > SIP/2.0 200 Ok > Via: SIP/2.0/UDP 192.168.1.122;rport;branch=z9hG4bKH893yar8p7X4j > From: "FreeSWITCH" ;tag=9vpFm3tBc8gSe > To: ;tag=2111843199 > Contact: > Call-ID: 68736fc7-6e28-122c-8396-000c7680c408 > CSeq: 110801209 BYE > Server: X-Lite release 1105d > Content-Length: 0 > > ------------------------------------------------------------------------ Dnia 04-02-2009, ?ro o godzinie 10:22 -0600, Anthony Minessale pisze: > can you press f8 for debug and try that apiExecute and post the > results? > > On Wed, Feb 4, 2009 at 8:46 AM, Jacek Sokulski > wrote: > Thanks Anthony, > the js snippets are very instructive. > A couple of points: > 1. The code with apiExecute does not work (local phone is > connected, but > after picking up it hungs up immediately), other examples are > working > fine. > > 2. It does not show how initiate external call without > existing session. > > 3. How can one pass the call through dialplan? > > Jacek > > PS. > we got the code probable from wiki or from this mialing list. > > Dnia 04-02-2009, ?ro o godzinie 08:09 -0600, Anthony Minessale > pisze: > > > Where did you learn how to use js this way? > > session.originate is being misused here and is depricated > and may be > > removed. > > > > the first arg to session.originate is either undefined or a > > *different* session (the a leg) > > > > session1 = new Session(); > > session1.originate(undefined, > > "{ignore_early_media=true}user/1008 at 192.168.1.122"); > > > > > session1.setVariable("effective_caller_id_number","fixed0248b"); > > > > //once you have session1 when you originate session2 you > pass session1 > > as the arg > > // the effective_caller_id is taken from session1 > > > > session2 = new Session(); > > session2.originate(session1, > "sofia/gateway/halonet/0225490317"); > > > > Anyway this whole code is depricated in favor of this: > > > > session1 = new > > Session("{ignore_early_media=true}user/1008 at 192.168.1.122"); > > > > if (session1.ready()) { > > > session1.setVariable("effective_caller_id_number","fixed0248b"); > > session2 = new Session("sofia/gateway/halonet/0225490317", > > session1); > > } > > > > and could be further refactored down to this: > > > > session1 = new > > Session("{ignore_early_media=true}user/1008 at 192.168.1.122"); > > if (session1.ready()) { > > > session1.setVariable("effective_caller_id_number","fixed0248b"); > > session1.execute("bridge", > "sofia/gateway/halonet/0225490317"); > > } > > > > or down to this one line of code that will setup the call > detached > > from the script and exit. > > > > var result = apiExecute("originate", > > > "{effective_caller_id_number=fixed0248b,origination_caller_id_number=1000,ignore_early_media=true}user/1008 at 192.168.1.122 bridge:sofia/gateway/halonet/0225490317 inline"); > > > > if you dont care about the result and want to exit even > before the > > call is completed. > > > > var result = apiExecute("bgapi", "originate > > > {effective_caller_id_number=fixed0248b,origination_caller_id_number=1000,ignore_early_media=true}user/1008 at 192.168.1.122 bridge:sofia/gateway/halonet/0225490317 inline"); > > > > > > > > On Wed, Feb 4, 2009 at 2:51 AM, Jacek Sokulski > > wrote: > > > > We have tried setting both > effective_caller_id_number and > > origination_caller_id_number: > > > > > session1.originate(session1,"{origination_caller_id_number=fixed0248b}sofia/gateway/halonet/0225490317",15); > > but the problem still exists. The solution we have > found for > > the case > > when we originate two calls, local and external, is > as follow: > > > > session1 = new Session(); > > > session1.originate(session1,"user/1003 at 192.168.1.122",15);//local > > if(session1.ready()) { > > > session1.execute("execute_extension","00930691688627 XML > > default");//external > > } > > > > so the external call goes through the dialplan. > > It does not work if both calls are external. One > possible > > solution could be > > to pass the originating call through dialplan > (loopback?) but > > we have not managed > > to figure out how to do it. > > > > Thanks > > Jacek > > > > Dnia 03-02-2009, wto o godzinie 14:31 -0300, Nicolas > Brenner > > pisze: > > > > > Oops! Well, fortunately I don't use that voip > provider > > anymore (nor the script). > > > > > > Thanks Brian. > > > > > > Nicolas > > > > > > On Tue, Feb 3, 2009 at 2:25 PM, Brian West > > wrote: > > > > YOU should NEVER use this method or call > setCallerData at > > all you > > > > should use the correct methods to override the > callerid. > > > > > > > > If its a B-Leg born from an A-Leg you use these > on the on > > the A-Leg: > > > > > > > > > > > http://wiki.freeswitch.org/wiki/Channel_Variables#effective_caller_id_name > > > > > > > http://wiki.freeswitch.org/wiki/Channel_Variables#effective_caller_id_number > > > > > > > > If you're originating you use this: > > > > > > > > > > > http://wiki.freeswitch.org/wiki/Channel_Variables#origination_caller_id_name > > > > > > > http://wiki.freeswitch.org/wiki/Channel_Variables#origination_caller_id_number > > > > > > > > /b > > > > > > _______________________________________________ > > > Freeswitch-users mailing list > > > Freeswitch-users at lists.freeswitch.org > > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From anthony.minessale at gmail.com Thu Feb 5 06:21:32 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 5 Feb 2009 08:21:32 -0600 Subject: [Freeswitch-users] Freeswitch freezes on increasing call traffic In-Reply-To: <21848148.post@talk.nabble.com> References: <21701744.post@talk.nabble.com> <21749375.post@talk.nabble.com> <191c3a030901300818i5d85a7aawa0c53b4e54892f40@mail.gmail.com> <21761523.post@talk.nabble.com> <191c3a030901310850s553f8c11oc4e559a710def6d2@mail.gmail.com> <21825226.post@talk.nabble.com> <191c3a030902040551w5d867deta55c98bd3a3c03f8@mail.gmail.com> <21847332.post@talk.nabble.com> <4C2A4389-BEF6-454B-8B5F-FDCCC76E174A@freeswitch.org> <21848148.post@talk.nabble.com> Message-ID: <191c3a030902050621k7534e2d4n35eec6dc36fd439a@mail.gmail.com> First of all please stop using the mailing list as a bug tracker. All issues should be put into jira and managed with that. Secondly, Didn't I ask you multiple times to stop using release snapshots and please use the SVN trunk? I don't understand why you keep ignoring me and using everything but what I asked. I am not telling you to use SVN because I think it will be fixed it's so we are on the development copy of the code to get the proper line numbers etc. If you look at your 2 bt you posted, the line numbers are different on each one. What are you using on the other side of ODBC? as you can see in your bt, the call goes into ODBC then into several libs with no symbols and crashes on free. This can be a sign of corrupt memory, running out of memory or an issue in either ODBC or the database specific lib. What distro is it? What ODBC version? unixODBC? version xxx? What database driver version xxx? Is it mysl not using the proper reentrant version of the plugin? Sometimes packaged libs have bugs in them which fall out of our control. Can you build unixODBC and the plugins yourself with debug symbols so we can see if that is the cause or at the very least then we can see the debug info in the bt. please make sure you address *all* my questions in your jira report. Starting with using svn trunk, *hint* type "make current" from your rc1 distro. On Thu, Feb 5, 2009 at 3:51 AM, shehzad p wrote: > > HI Brian, > > Output of ulimit -a and /proc/cpuinfo is attached... > http://www.nabble.com/file/p21848148/12_ulimit_and_cpuinfo.log > 12_ulimit_and_cpuinfo.log > > BUT...................... > I am running the freeswitch using below command (So ulimit set according to > Anthony's previous post): > > =================================================================================== > ulimit -d unlimited; ulimit -f unlimited; ulimit -i unlimited; ulimit -n > 999999; ulimit -q unlimited ; ulimit -u unlimited; ulimit -v unlimited ; > ulimit -x unlimited; ulimit -s 244; ulimit -l unlimited; freeswitch > > =================================================================================== > > Thanks > msp > > > > Brian West-3 wrote: > > > > Can you give me the output of uname -a and the contents of /proc/ > > cpuinfo? Not sure I asked for this info already or not. > > > > Thanks, > > Brian > > > > On Feb 5, 2009, at 2:42 AM, shehzad p wrote: > > > >> > >> Hi anthony, > >> In my previous post I already attached the BT for the testing of FS > >> 1.0.1 > >> Posting it again, please find it on > >> link==>http://www.nabble.com/file/p21825226/bt_new_arch.txt > >> > >> Now I got the same result while testing FS 1.0.3RC1, And its BT is > >> also > >> same... BT Link: > >> http://www.nabble.com/file/p21847332/fs_1_0_3_bt_new_arch.log > >> fs_1_0_3_bt_new_arch.log > >> > >> (Note: Same in the sens the functions listed in the sequence are > >> almost same > >> as before...) > >> > >> > >> Anthony Minessale-2 wrote: > >>> > >>> If you still get a crash on SVN trunk please post the bt even if > >>> you think > >>> it's the same, since it won't be > >>> exactly the same, the line numbers etc will be accurate with our > >>> development > >>> code making it easier to debug. > >>> > >>> > >>> On Wed, Feb 4, 2009 at 12:38 AM, shehzad p wrote: > >>> > >>>> > >>>> Hi anthony, > >>>> > >>>> I Modified the whole architecture of call routing system, > >>>> Now after getting required routes, script exit and, > >>>> control comes back to Dialplan, and call is bridged there, > >>>> And call hangup, CDR is posted to cdr.php file (using xml_cdr). > >>>> > >>>> So now there is no blocking statement (bridge or anything like > >>>> that) in > >>>> current javascript, It return back control instantly. > >>>> > >>>> So, setting up all above architecture... > >>>> First I tested FS 1.0.1 , It get crashed two times, in interval of > >>>> 3 to 5 > >>>> hours and simultaneous call of about 100 to 150. > >>>> BT is the same as before... > >>>> http://www.nabble.com/file/p21825226/bt_new_arch.txt bt_new_arch.txt > >>>> > >>>> Now I am also testing 1.0.3RC1, and post it back if any found. > >>>> > >>>> Thanks > >>>> msp > >>>> > >>>> > >>>> Clearly you have an issue with your javascript code. > >>>> > >>>> You have the Garbage collector blocking in every thread. > >>>> > >>>> Are you doing any endless loops in your code where you do not check > >>>> session.ready() as a condition for > >>>> continuing the script? > >>>> > >>>> any time session.ready() fails you must immediately exit. > >>>> > >>>> Are you using session.execute to execute long blocking operations > >>>> like > >>>> bridging many calls or entering a conference? > >>>> You should avoid doing this as all the collective scripts on the > >>>> system > >>>> share a common Garbage Collector provided by the > >>>> JS engine and it can lead to the exact issues you describe if the > >>>> code is > >>>> not properly designed. > >>>> > >>>> What else does you script do that are things provided by FS such as > >>>> playing > >>>> files and executing applications. > >>>> > >>>> > >>>> > >>>> > >>>> -- > >>>> View this message in context: > >>>> > http://www.nabble.com/Freeswitch-freezes-on-increasing-call-traffic-tp21701744p21825226.html > >>>> Sent from the Freeswitch-users mailing list archive at Nabble.com. > >>>> > >>>> > >>>> _______________________________________________ > >>>> Freeswitch-users mailing list > >>>> Freeswitch-users at lists.freeswitch.org > >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>>> http://www.freeswitch.org > >>>> > >>> > >>> > >>> > >>> -- > >>> Anthony Minessale II > >>> > >>> FreeSWITCH http://www.freeswitch.org/ > >>> ClueCon http://www.cluecon.com/ > >>> > >>> AIM: anthm > >>> MSN:anthony_minessale at hotmail.com< > MSN%3Aanthony_minessale at hotmail.com > >>> > > >>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > >>> > > >>> IRC: irc.freenode.net #freeswitch > >>> > >>> FreeSWITCH Developer Conference > >>> sip:888 at conference.freeswitch.org< > sip%3A888 at conference.freeswitch.org > >>> > > >>> iax:guest at conference.freeswitch.org/888 > >>> googletalk:conf+888 at conference.freeswitch.org > > >>> > > >>> pstn:213-799-1400 > >>> > >>> _______________________________________________ > >>> Freeswitch-users mailing list > >>> Freeswitch-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> http://www.freeswitch.org > >>> > >>> > >> > >> -- > >> View this message in context: > >> > http://www.nabble.com/Freeswitch-freezes-on-increasing-call-traffic-tp21701744p21847332.html > >> Sent from the Freeswitch-users mailing list archive at Nabble.com. > >> > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > View this message in context: > http://www.nabble.com/Freeswitch-freezes-on-increasing-call-traffic-tp21701744p21848148.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090205/a427c469/attachment.html From pmhshz at gmail.com Thu Feb 5 06:22:49 2009 From: pmhshz at gmail.com (shehzad p) Date: Thu, 5 Feb 2009 06:22:49 -0800 (PST) Subject: [Freeswitch-users] Freeswitch freezes on increasing call traffic In-Reply-To: <21848148.post@talk.nabble.com> References: <21701744.post@talk.nabble.com> <191c3a030901280601t3e185ec2m76295110e4888837@mail.gmail.com> <21729863.post@talk.nabble.com> <191c3a030901290831y2ff329b3maee6c1aafa3c68e2@mail.gmail.com> <21745312.post@talk.nabble.com> <21749375.post@talk.nabble.com> <191c3a030901300818i5d85a7aawa0c53b4e54892f40@mail.gmail.com> <21761523.post@talk.nabble.com> <191c3a030901310850s553f8c11oc4e559a710def6d2@mail.gmail.com> <21825226.post@talk.nabble.com> <191c3a030902040551w5d867deta55c98bd3a3c03f8@mail.gmail.com> <21847332.post@talk.nabble.com> <4C2A4389-BEF6-454B-8B5F-FDCCC76E174A@freeswitch.org> <21848148.post@talk.nabble.com> Message-ID: <21852304.post@talk.nabble.com> Hi Brian, As it can be seen from the system information, there require any change in system or any suggestion... out put of uname -a is : Linux localhost.localdomain 2.6.23.1-42.fc8 #1 SMP Tue Oct 30 13:55:12 EDT 2007 i686 i686 i386 GNU/Linux Thanks, msp shehzad p wrote: > > HI Brian, > > Output of ulimit -a and /proc/cpuinfo is attached... > http://www.nabble.com/file/p21848148/12_ulimit_and_cpuinfo.log > 12_ulimit_and_cpuinfo.log > > BUT...................... > I am running the freeswitch using below command (So ulimit set according > to Anthony's previous post): > =================================================================================== > ulimit -d unlimited; ulimit -f unlimited; ulimit -i unlimited; ulimit -n > 999999; ulimit -q unlimited ; ulimit -u unlimited; ulimit -v unlimited ; > ulimit -x unlimited; ulimit -s 244; ulimit -l unlimited; freeswitch > =================================================================================== > > Thanks > msp > > > > Brian West-3 wrote: >> >> Can you give me the output of uname -a and the contents of /proc/ >> cpuinfo? Not sure I asked for this info already or not. >> >> Thanks, >> Brian >> >> On Feb 5, 2009, at 2:42 AM, shehzad p wrote: >> >>> >>> Hi anthony, >>> In my previous post I already attached the BT for the testing of FS >>> 1.0.1 >>> Posting it again, please find it on >>> link==>http://www.nabble.com/file/p21825226/bt_new_arch.txt >>> >>> Now I got the same result while testing FS 1.0.3RC1, And its BT is >>> also >>> same... BT Link: >>> http://www.nabble.com/file/p21847332/fs_1_0_3_bt_new_arch.log >>> fs_1_0_3_bt_new_arch.log >>> >>> (Note: Same in the sens the functions listed in the sequence are >>> almost same >>> as before...) >>> >>> >>> Anthony Minessale-2 wrote: >>>> >>>> If you still get a crash on SVN trunk please post the bt even if >>>> you think >>>> it's the same, since it won't be >>>> exactly the same, the line numbers etc will be accurate with our >>>> development >>>> code making it easier to debug. >>>> >>>> >>>> On Wed, Feb 4, 2009 at 12:38 AM, shehzad p wrote: >>>> >>>>> >>>>> Hi anthony, >>>>> >>>>> I Modified the whole architecture of call routing system, >>>>> Now after getting required routes, script exit and, >>>>> control comes back to Dialplan, and call is bridged there, >>>>> And call hangup, CDR is posted to cdr.php file (using xml_cdr). >>>>> >>>>> So now there is no blocking statement (bridge or anything like >>>>> that) in >>>>> current javascript, It return back control instantly. >>>>> >>>>> So, setting up all above architecture... >>>>> First I tested FS 1.0.1 , It get crashed two times, in interval of >>>>> 3 to 5 >>>>> hours and simultaneous call of about 100 to 150. >>>>> BT is the same as before... >>>>> http://www.nabble.com/file/p21825226/bt_new_arch.txt bt_new_arch.txt >>>>> >>>>> Now I am also testing 1.0.3RC1, and post it back if any found. >>>>> >>>>> Thanks >>>>> msp >>>>> >>>>> >>>>> Clearly you have an issue with your javascript code. >>>>> >>>>> You have the Garbage collector blocking in every thread. >>>>> >>>>> Are you doing any endless loops in your code where you do not check >>>>> session.ready() as a condition for >>>>> continuing the script? >>>>> >>>>> any time session.ready() fails you must immediately exit. >>>>> >>>>> Are you using session.execute to execute long blocking operations >>>>> like >>>>> bridging many calls or entering a conference? >>>>> You should avoid doing this as all the collective scripts on the >>>>> system >>>>> share a common Garbage Collector provided by the >>>>> JS engine and it can lead to the exact issues you describe if the >>>>> code is >>>>> not properly designed. >>>>> >>>>> What else does you script do that are things provided by FS such as >>>>> playing >>>>> files and executing applications. >>>>> >>>>> >>>>> >>>>> >>>>> -- >>>>> View this message in context: >>>>> http://www.nabble.com/Freeswitch-freezes-on-increasing-call-traffic-tp21701744p21825226.html >>>>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>>>> >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> >>>> -- >>>> Anthony Minessale II >>>> >>>> FreeSWITCH http://www.freeswitch.org/ >>>> ClueCon http://www.cluecon.com/ >>>> >>>> AIM: anthm >>>> MSN:anthony_minessale at hotmail.com >>> > >>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com>>> > >>>> IRC: irc.freenode.net #freeswitch >>>> >>>> FreeSWITCH Developer Conference >>>> sip:888 at conference.freeswitch.org >>> > >>>> iax:guest at conference.freeswitch.org/888 >>>> googletalk:conf+888 at conference.freeswitch.org>>> > >>>> pstn:213-799-1400 >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> -- >>> View this message in context: >>> http://www.nabble.com/Freeswitch-freezes-on-increasing-call-traffic-tp21701744p21847332.html >>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- View this message in context: http://www.nabble.com/Freeswitch-freezes-on-increasing-call-traffic-tp21701744p21852304.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From paul.degt at gmail.com Thu Feb 5 07:12:15 2009 From: paul.degt at gmail.com (paul.degt) Date: Thu, 05 Feb 2009 10:12:15 -0500 Subject: [Freeswitch-users] Freeswitch freezes on increasing call traffic In-Reply-To: <21852304.post@talk.nabble.com> References: <21701744.post@talk.nabble.com> <191c3a030901280601t3e185ec2m76295110e4888837@mail.gmail.com> <21729863.post@talk.nabble.com> <191c3a030901290831y2ff329b3maee6c1aafa3c68e2@mail.gmail.com> <21745312.post@talk.nabble.com> <21749375.post@talk.nabble.com> <191c3a030901300818i5d85a7aawa0c53b4e54892f40@mail.gmail.com> <21761523.post@talk.nabble.com> <191c3a030901310850s553f8c11oc4e559a710def6d2@mail.gmail.com> <21825226.post@talk.nabble.com> <191c3a030902040551w5d867deta55c98bd3a3c03f8@mail.gmail.com> <21847332.post@talk.nabble.com> <4C2A4389-BEF6-454B-8B5F-FDCCC76E174A@freeswitch.org> <21848148.post@talk.nabble.com> <21852304.post@talk.nabble.com> Message-ID: <498B01CF.6080902@gmail.com> Look like you use Fedora. I had a lot of issues with using Fedora as production or load test system, in my opinion it's more like work in progress than a production ready stable linux. If you cannot buy RHEL or SLES use Centos. shehzad p wrote: > Hi Brian, > > As it can be seen from the system information, there require any change in > system or any suggestion... > > out put of uname -a is : > Linux localhost.localdomain 2.6.23.1-42.fc8 #1 SMP Tue Oct 30 13:55:12 EDT > 2007 i686 i686 i386 GNU/Linux > > > Thanks, > msp > > > shehzad p wrote: > >> HI Brian, >> >> Output of ulimit -a and /proc/cpuinfo is attached... >> http://www.nabble.com/file/p21848148/12_ulimit_and_cpuinfo.log >> 12_ulimit_and_cpuinfo.log >> >> BUT...................... >> I am running the freeswitch using below command (So ulimit set according >> to Anthony's previous post): >> =================================================================================== >> ulimit -d unlimited; ulimit -f unlimited; ulimit -i unlimited; ulimit -n >> 999999; ulimit -q unlimited ; ulimit -u unlimited; ulimit -v unlimited ; >> ulimit -x unlimited; ulimit -s 244; ulimit -l unlimited; freeswitch >> =================================================================================== >> >> Thanks >> msp >> >> >> >> Brian West-3 wrote: >> >>> Can you give me the output of uname -a and the contents of /proc/ >>> cpuinfo? Not sure I asked for this info already or not. >>> >>> Thanks, >>> Brian >>> >>> On Feb 5, 2009, at 2:42 AM, shehzad p wrote: >>> >>> > From sicfslist at gmail.com Thu Feb 5 07:28:37 2009 From: sicfslist at gmail.com (Shelby Ramsey) Date: Thu, 5 Feb 2009 09:28:37 -0600 Subject: [Freeswitch-users] XML CDR ERROR ... In-Reply-To: <640834B6-0A23-4855-BB0F-F3622AA334FF@freeswitch.org> References: <35b355e90902042333p3009d6c1md2a76355a90509bd@mail.gmail.com> <640834B6-0A23-4855-BB0F-F3622AA334FF@freeswitch.org> Message-ID: <35b355e90902050728u4f0e1dbk2386518ed3985b74@mail.gmail.com> Just to make sure I'm clear: -- start FS (without in modules.conf.xml) -- but have xml_cdr.conf.xml in autoload configs -- then reload xml -- then execute load mod_xml_cdr via the api That seems like a challenging way to start FS ... SDR On Thu, Feb 5, 2009 at 1:42 AM, Brian West wrote: > Make sure your config file is installed and issue a reloadxml then > load mod_xml_cdr > > /b > > On Feb 5, 2009, at 1:33 AM, Shelby Ramsey wrote: > > > Hello, > > > > I'm having it on both Fedora and Ubuntu boxes: > > > > 2009-02-05 01:26:27 [ERR] mod_xml_cdr.c:290 mod_xml_cdr_load() Open > > of xml_cdr.conf failed > > 2009-02-05 01:26:27 [CRIT] switch_loadable_module.c:839 > > switch_loadable_module_load_file() Error Loading module /usr/local/ > > freeswitch/mod/mod_xml_cdr.so > > **Module load routine returned an error** > > > > Details: > > -- Ubuntu 6.04 LTS > > -- Fedora 8 > > > > Tried a couple of things: > > -- messing with libcurl > > -- ./configure --without-libcurl > > > > Thanks! > > > > SDR > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090205/4e4915dd/attachment.html From freeswitch-users at lists.rupa.com Thu Feb 5 07:38:54 2009 From: freeswitch-users at lists.rupa.com (Rupa Schomaker (lists)) Date: Thu, 05 Feb 2009 09:38:54 -0600 Subject: [Freeswitch-users] XML CDR ERROR ... In-Reply-To: <35b355e90902050728u4f0e1dbk2386518ed3985b74@mail.gmail.com> References: <35b355e90902042333p3009d6c1md2a76355a90509bd@mail.gmail.com> <640834B6-0A23-4855-BB0F-F3622AA334FF@freeswitch.org> <35b355e90902050728u4f0e1dbk2386518ed3985b74@mail.gmail.com> Message-ID: <498B080E.6080202@lists.rupa.com> The error is that the conf file couldn't be loaded. If you've verified teh file is actually in autoload configs dir, also ensure the file is readable by whatever user/group freeswitch is running as. Once you have your config squared away, just having hte in modules.conf.xml is sufficient. Brian was trying to assist you in debugging.... On 2/5/2009 9:28 AM, Shelby Ramsey wrote: > Just to make sure I'm clear: > -- start FS (without in modules.conf.xml) > -- but have xml_cdr.conf.xml in autoload configs > -- then reload xml > -- then execute load mod_xml_cdr via the api > > That seems like a challenging way to start FS ... > > SDR > > On Thu, Feb 5, 2009 at 1:42 AM, Brian West > wrote: > > Make sure your config file is installed and issue a reloadxml then > load mod_xml_cdr > > /b > > On Feb 5, 2009, at 1:33 AM, Shelby Ramsey wrote: > > > Hello, > > > > I'm having it on both Fedora and Ubuntu boxes: > > > > 2009-02-05 01:26:27 [ERR] mod_xml_cdr.c:290 mod_xml_cdr_load() Open > > of xml_cdr.conf failed > > 2009-02-05 01:26:27 [CRIT] switch_loadable_module.c:839 > > switch_loadable_module_load_file() Error Loading module /usr/local/ > > freeswitch/mod/mod_xml_cdr.so > > **Module load routine returned an error** > > > > Details: > > -- Ubuntu 6.04 LTS > > -- Fedora 8 > > > > Tried a couple of things: > > -- messing with libcurl > > -- ./configure --without-libcurl > > > > Thanks! > > > > SDR From anthony.minessale at gmail.com Thu Feb 5 07:40:20 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 5 Feb 2009 09:40:20 -0600 Subject: [Freeswitch-users] origainate through sofia gateway In-Reply-To: <1233839355.5346.31.camel@dotw1126.dotsystems.pl> References: <1233668179.5354.15.camel@dotw1126.dotsystems.pl> <1b46b4e80902030920p521d6079s4c9b78d8d39a924a@mail.gmail.com> <5E2B217D-8C85-48DB-800A-535A7FAAE24B@freeswitch.org> <1b46b4e80902030931y6a1679afw12ec37596ba9d77d@mail.gmail.com> <1233737494.5405.12.camel@dotw1126.dotsystems.pl> <191c3a030902040609s600798c8t720950a2e13ee05a@mail.gmail.com> <1233758767.5405.22.camel@dotw1126.dotsystems.pl> <191c3a030902040822o1ffededbwd592361aa09f6b46@mail.gmail.com> <1233839355.5346.31.camel@dotw1126.dotsystems.pl> Message-ID: <191c3a030902050740w7f7c4ef2hdb7a6a0e355c46f5@mail.gmail.com> Oddly the provider returns SIP/2.0 500 Internal Server Error to your invite. Maybe you can ask them why they do that in this particular case. On Thu, Feb 5, 2009 at 7:09 AM, Jacek Sokulski wrote: > the result of > apiExecute("bgapi", "originate > > {effective_caller_id_number=fixed0248b,origination_caller_id_number=1000,ignore_early_media=true}user/ > 1008 at 192.168.1.122 bridge:sofia/gateway/halonet/0225490317 inline"); > > /: > > 2009-02-05 14:04:48 [DEBUG] switch_ivr_originate.c:777 > switch_ivr_originate() variable string 0 = > [effective_caller_id_number=fixed0248b] > > 2009-02-05 14:04:48 [DEBUG] switch_ivr_originate.c:777 > switch_ivr_originate() variable string 1 = > [origination_caller_id_number=1000] > > 2009-02-05 14:04:48 [DEBUG] switch_ivr_originate.c:777 > switch_ivr_originate() variable string 2 = [ignore_early_media=true] > > 2009-02-05 14:04:48 [DEBUG] switch_ivr_originate.c:777 > switch_ivr_originate() variable string 0 = [presence_id=1008 at 192.168.1.122 > ] > > 2009-02-05 14:04:48 [NOTICE] switch_channel.c:565 > switch_channel_set_name() New Channel sofia/internal/ > sip:1008 at 192.168.1.126:5070 [f9b18b2e-b0a9-4e24-8452-40e1cce047bd] > > 2009-02-05 14:04:48 [DEBUG] mod_sofia.c:2518 sofia_outgoing_channel() > (sofia/internal/sip:1008 at 192.168.1.126:5070) State Change CS_NEW -> > CS_INIT > > 2009-02-05 14:04:48 [DEBUG] switch_core_session.c:807 > switch_core_session_signal_state_change() Send signal sofia/internal/ > sip:1008 at 192.168.1.126:5070 [BREAK] > > 2009-02-05 14:04:48 [DEBUG] switch_core_state_machine.c:379 > switch_core_session_run() (sofia/internal/sip:1008 at 192.168.1.126:5070) > Running State Change CS_INIT > > 2009-02-05 14:04:48 [DEBUG] switch_core_state_machine.c:444 > switch_core_session_run() (sofia/internal/sip:1008 at 192.168.1.126:5070) > State INIT > > 2009-02-05 14:04:48 [DEBUG] mod_sofia.c:83 sofia_on_init() > sofia/internal/sip:1008 at 192.168.1.126:5070 SOFIA INIT > > send 1196 bytes to udp/[192.168.1.126]:5070 at 13:04:48.127004: > > > ------------------------------------------------------------------------ > > INVITE sip:1008 at 192.168.1.126:5070 SIP/2.0 > > Via: SIP/2.0/UDP 192.168.1.122;rport;branch=z9hG4bKFpQjUmp1vNHZB > > Max-Forwards: 70 > > From: "FreeSWITCH" > >;tag=9vpFm3tBc8gSe > > To: > > Call-ID: 68736fc7-6e28-122c-8396-000c7680c408 > > CSeq: 110801208 INVITE > > Contact: > > User-Agent: FreeSWITCH-mod_sofia/1.0.2-wyeksportowane > > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH > > Supported: timer, precondition, path, replaces > > Allow-Events: talk, presence, dialog, call-info, sla, > include-session-description, presence.winfo, message-summary, refer > > Min-SE: 120 > > Content-Type: application/sdp > > Content-Disposition: session > > Content-Length: 335 > > Remote-Party-ID: "FreeSWITCH" > >;screen=yes;privacy=off > > > > v=0 > > o=FreeSWITCH 1927164999227225404 3276738624485570081 IN IP4 > 192.168.1.122 > > s=FreeSWITCH > > c=IN IP4 192.168.1.122 > > t=0 0 > > m=audio 27044 RTP/AVP 9 0 8 3 101 13 > > a=rtpmap:9 G722/8000 > > a=rtpmap:0 PCMU/8000 > > a=rtpmap:8 PCMA/8000 > > a=rtpmap:3 GSM/8000 > > a=rtpmap:101 telephone-event/8000 > > a=fmtp:101 0-16 > > a=rtpmap:13 CN/8000 > > a=ptime:20 > > > ------------------------------------------------------------------------ > > 2009-02-05 14:04:48 [DEBUG] mod_sofia.c:111 sofia_on_init() > (sofia/internal/sip:1008 at 192.168.1.126:5070) State Change CS_INIT -> > CS_ROUTING > > 2009-02-05 14:04:48 [DEBUG] switch_core_session.c:807 > switch_core_session_signal_state_change() Send signal sofia/internal/ > sip:1008 at 192.168.1.126:5070 [BREAK] > > 2009-02-05 14:04:48 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state() > Channel sofia/internal/sip:1008 at 192.168.1.126:5070 entering state > [calling] > > 2009-02-05 14:04:48 [DEBUG] switch_core_state_machine.c:444 > switch_core_session_run() (sofia/internal/sip:1008 at 192.168.1.126:5070) > State INIT going to sleep > > 2009-02-05 14:04:48 [DEBUG] switch_core_state_machine.c:379 > switch_core_session_run() (sofia/internal/sip:1008 at 192.168.1.126:5070) > Running State Change CS_ROUTING > > 2009-02-05 14:04:48 [DEBUG] switch_core_state_machine.c:447 > switch_core_session_run() (sofia/internal/sip:1008 at 192.168.1.126:5070) > State ROUTING > > 2009-02-05 14:04:48 [DEBUG] mod_sofia.c:130 sofia_on_routing() > sofia/internal/sip:1008 at 192.168.1.126:5070 SOFIA ROUTING > > 2009-02-05 14:04:48 [DEBUG] switch_ivr_originate.c:58 > originate_on_routing() (sofia/internal/sip:1008 at 192.168.1.126:5070) State > Change CS_ROUTING -> CS_CONSUME_MEDIA > > 2009-02-05 14:04:48 [DEBUG] switch_core_session.c:807 > switch_core_session_signal_state_change() Send signal sofia/internal/ > sip:1008 at 192.168.1.126:5070 [BREAK] > > 2009-02-05 14:04:48 [DEBUG] switch_core_state_machine.c:447 > switch_core_session_run() (sofia/internal/sip:1008 at 192.168.1.126:5070) > State ROUTING going to sleep > > 2009-02-05 14:04:48 [DEBUG] switch_core_state_machine.c:379 > switch_core_session_run() (sofia/internal/sip:1008 at 192.168.1.126:5070) > Running State Change CS_CONSUME_MEDIA > > 2009-02-05 14:04:48 [DEBUG] switch_core_state_machine.c:466 > switch_core_session_run() (sofia/internal/sip:1008 at 192.168.1.126:5070) > State CONSUME_MEDIA > > recv 361 bytes from udp/[192.168.1.126]:5070 at 13:04:48.130745: > > > ------------------------------------------------------------------------ > > SIP/2.0 100 Trying > > Via: SIP/2.0/UDP 192.168.1.122;rport;branch=z9hG4bKFpQjUmp1vNHZB > > From: "FreeSWITCH" > >;tag=9vpFm3tBc8gSe > > To: ;tag=2111843199 > > Contact: > > Call-ID: 68736fc7-6e28-122c-8396-000c7680c408 > > CSeq: 110801208 INVITE > > Server: X-Lite release 1105d > > Content-Length: 0 > > > > > ------------------------------------------------------------------------ > > recv 362 bytes from udp/[192.168.1.126]:5070 at 13:04:48.181571: > > > ------------------------------------------------------------------------ > > SIP/2.0 180 Ringing > > Via: SIP/2.0/UDP 192.168.1.122;rport;branch=z9hG4bKFpQjUmp1vNHZB > > From: "FreeSWITCH" > >;tag=9vpFm3tBc8gSe > > To: ;tag=2111843199 > > Contact: > > Call-ID: 68736fc7-6e28-122c-8396-000c7680c408 > > CSeq: 110801208 INVITE > > Server: X-Lite release 1105d > > Content-Length: 0 > > > > > ------------------------------------------------------------------------ > > 2009-02-05 14:04:48 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state() > Channel sofia/internal/sip:1008 at 192.168.1.126:5070 entering state > [proceeding] > > 2009-02-05 14:04:48 [NOTICE] sofia.c:2596 sofia_handle_sip_i_state() > Ring-Ready sofia/internal/sip:1008 at 192.168.1.126:5070! > > recv 678 bytes from udp/[192.168.1.126]:5070 at 13:04:51.914714: > > > ------------------------------------------------------------------------ > > SIP/2.0 200 Ok > > Via: SIP/2.0/UDP 192.168.1.122;rport;branch=z9hG4bKFpQjUmp1vNHZB > > From: "FreeSWITCH" > >;tag=9vpFm3tBc8gSe > > To: ;tag=2111843199 > > Contact: > > Call-ID: 68736fc7-6e28-122c-8396-000c7680c408 > > CSeq: 110801208 INVITE > > Content-Type: application/sdp > > Server: X-Lite release 1105d > > Content-Length: 288 > > > > v=0 > > o=1008 1183460117 1183463903 IN IP4 192.168.1.126 > > s=X-Lite > > c=IN IP4 192.168.1.126 > > t=0 0 > > m=audio 9000 RTP/AVP 0 8 98 97 101 > > a=rtpmap:0 pcmu/8000 > > a=rtpmap:8 pcma/8000 > > a=rtpmap:98 iLBC/8000 > > a=rtpmap:97 speex/8000 > > a=rtpmap:101 telephone-event/8000 > > a=fmtp:101 0-15 > > a=sendrecv > > > ------------------------------------------------------------------------ > > send 372 bytes to udp/[192.168.1.126]:5070 at 13:04:51.915303: > > > ------------------------------------------------------------------------ > > ACK sip:1008 at 192.168.1.126:5070 SIP/2.0 > > Via: SIP/2.0/UDP 192.168.1.122;rport;branch=z9hG4bKgZgBXF74Sy7HQ > > Max-Forwards: 70 > > From: "FreeSWITCH" > >;tag=9vpFm3tBc8gSe > > To: ;tag=2111843199 > > Call-ID: 68736fc7-6e28-122c-8396-000c7680c408 > > CSeq: 110801208 ACK > > Contact: > > Content-Length: 0 > > > > > ------------------------------------------------------------------------ > > 2009-02-05 14:04:51 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state() > Channel sofia/internal/sip:1008 at 192.168.1.126:5070 entering state [ready] > > 2009-02-05 14:04:51 [DEBUG] sofia.c:2546 sofia_handle_sip_i_state() > Remote SDP: > > v=0 > > o=1008 1183460117 1183463903 IN IP4 192.168.1.126 > > s=X-Lite > > c=IN IP4 192.168.1.126 > > t=0 0 > > m=audio 9000 RTP/AVP 0 8 98 97 101 > > a=rtpmap:0 pcmu/8000 > > a=rtpmap:8 pcma/8000 > > a=rtpmap:98 iLBC/8000 > > a=rtpmap:97 speex/8000 > > a=rtpmap:101 telephone-event/8000 > > a=fmtp:101 0-15 > > > > 2009-02-05 14:04:51 [DEBUG] sofia_glue.c:2509 sofia_glue_negotiate_sdp() > Audio Codec Compare [pcmu:0:8000]/[G722:9:8000] > > 2009-02-05 14:04:51 [DEBUG] sofia_glue.c:2509 sofia_glue_negotiate_sdp() > Audio Codec Compare [pcmu:0:8000]/[PCMU:0:8000] > > 2009-02-05 14:04:51 [DEBUG] sofia_glue.c:1670 sofia_glue_tech_set_codec() > Set Codec sofia/internal/sip:1008 at 192.168.1.126:5070 PCMU/8000 20 ms 160 > samples > > 2009-02-05 14:04:51 [DEBUG] sofia_glue.c:2473 sofia_glue_negotiate_sdp() > Set 2833 dtmf payload to 101 > > 2009-02-05 14:04:51 [DEBUG] sofia_glue.c:1890 sofia_glue_activate_rtp() > AUDIO RTP [sofia/internal/sip:1008 at 192.168.1.126:5070] 192.168.1.122 port > 27044 -> 192.168.1.126 port 9000 codec: 0 ms: 20 > > 2009-02-05 14:04:51 [DEBUG] switch_rtp.c:865 switch_rtp_create() Starting > timer [soft] 160 bytes per 20000ms > > 2009-02-05 14:04:51 [NOTICE] sofia.c:3031 sofia_handle_sip_i_state() > Channel [sofia/internal/sip:1008 at 192.168.1.126:5070] has been answered > > 2009-02-05 14:04:51 [DEBUG] switch_channel.c:177 > switch_channel_audio_sync() sofia/internal/sip:1008 at 192.168.1.126:5070receive message [AUDIO_SYNC] > > 2009-02-05 14:04:51 [DEBUG] switch_ivr_originate.c:1627 > switch_ivr_originate() Originate Resulted in Success: [sofia/internal/ > sip:1008 at 192.168.1.126:5070] > > 2009-02-05 14:04:51 [DEBUG] switch_channel.c:177 > switch_channel_audio_sync() sofia/internal/sip:1008 at 192.168.1.126:5070receive message [AUDIO_SYNC] > > 2009-02-05 14:04:51 [DEBUG] switch_ivr_originate.c:1627 > switch_ivr_originate() Originate Resulted in Success: [sofia/internal/ > sip:1008 at 192.168.1.126:5070] > > 2009-02-05 14:04:51 [DEBUG] switch_channel.c:177 > switch_channel_audio_sync() sofia/internal/sip:1008 at 192.168.1.126:5070receive message [AUDIO_SYNC] > > 2009-02-05 14:04:51 [DEBUG] switch_ivr.c:1245 > switch_ivr_session_transfer() (sofia/internal/sip:1008 at 192.168.1.126:5070) > State Change CS_CONSUME_MEDIA -> CS_ROUTING > > 2009-02-05 14:04:51 [DEBUG] switch_core_session.c:807 > switch_core_session_signal_state_change() Send signal sofia/internal/ > sip:1008 at 192.168.1.126:5070 [BREAK] > > 2009-02-05 14:04:51 [DEBUG] switch_ivr.c:1249 > switch_ivr_session_transfer() sofia/internal/sip:1008 at 192.168.1.126:5070receive message [TRANSFER] > > 2009-02-05 14:04:51 [DEBUG] switch_core_session.c:511 > switch_core_session_perform_receive_message() Send signal sofia/internal/ > sip:1008 at 192.168.1.126:5070 [BREAK] > > 2009-02-05 14:04:51 [NOTICE] switch_ivr.c:1251 > switch_ivr_session_transfer() Transfer sofia/internal/ > sip:1008 at 192.168.1.126:5070 to > inline[bridge:sofia/gateway/halonet/0225490317 at default] > > 2009-02-05 14:04:51 [DEBUG] switch_core_state_machine.c:466 > switch_core_session_run() (sofia/internal/sip:1008 at 192.168.1.126:5070) > State CONSUME_MEDIA going to sleep > > 2009-02-05 14:04:51 [DEBUG] switch_core_state_machine.c:379 > switch_core_session_run() (sofia/internal/sip:1008 at 192.168.1.126:5070) > Running State Change CS_ROUTING > > 2009-02-05 14:04:51 [DEBUG] switch_core_state_machine.c:447 > switch_core_session_run() (sofia/internal/sip:1008 at 192.168.1.126:5070) > State ROUTING > > 2009-02-05 14:04:51 [DEBUG] mod_sofia.c:130 sofia_on_routing() > sofia/internal/sip:1008 at 192.168.1.126:5070 SOFIA ROUTING > > 2009-02-05 14:04:51 [DEBUG] switch_core_state_machine.c:64 > switch_core_standard_on_routing() sofia/internal/ > sip:1008 at 192.168.1.126:5070 Standard ROUTING > > 2009-02-05 14:04:51 [DEBUG] switch_core_state_machine.c:100 > switch_core_standard_on_routing() (sofia/internal/ > sip:1008 at 192.168.1.126:5070) State Change CS_ROUTING -> CS_EXECUTE > > 2009-02-05 14:04:51 [DEBUG] switch_core_session.c:807 > switch_core_session_signal_state_change() Send signal sofia/internal/ > sip:1008 at 192.168.1.126:5070 [BREAK] > > 2009-02-05 14:04:51 [DEBUG] switch_core_state_machine.c:447 > switch_core_session_run() (sofia/internal/sip:1008 at 192.168.1.126:5070) > State ROUTING going to sleep > > 2009-02-05 14:04:51 [DEBUG] switch_core_state_machine.c:379 > switch_core_session_run() (sofia/internal/sip:1008 at 192.168.1.126:5070) > Running State Change CS_EXECUTE > > 2009-02-05 14:04:51 [DEBUG] switch_core_state_machine.c:454 > switch_core_session_run() (sofia/internal/sip:1008 at 192.168.1.126:5070) > State EXECUTE > > 2009-02-05 14:04:51 [DEBUG] mod_sofia.c:173 sofia_on_execute() > sofia/internal/sip:1008 at 192.168.1.126:5070 SOFIA EXECUTE > > 2009-02-05 14:04:51 [DEBUG] switch_core_state_machine.c:137 > switch_core_standard_on_execute() sofia/internal/ > sip:1008 at 192.168.1.126:5070 Standard EXECUTE > > 2009-02-05 14:04:51 [DEBUG] switch_core_state_machine.c:152 > switch_core_standard_on_execute() sofia/internal/ > sip:1008 at 192.168.1.126:5070 Execute > bridge(sofia/gateway/halonet/0225490317) > > 2009-02-05 14:04:51 [NOTICE] switch_channel.c:565 > switch_channel_set_name() New Channel sofia/external/0225490317 > [bf4fece9-38fc-40fa-8e9e-91f836f05e55] > > 2009-02-05 14:04:51 [DEBUG] mod_sofia.c:2518 sofia_outgoing_channel() > (sofia/external/0225490317) State Change CS_NEW -> CS_INIT > > 2009-02-05 14:04:51 [DEBUG] switch_core_session.c:807 > switch_core_session_signal_state_change() Send signal > sofia/external/0225490317 [BREAK] > > 2009-02-05 14:04:51 [DEBUG] switch_core_state_machine.c:379 > switch_core_session_run() (sofia/external/0225490317) Running State Change > CS_INIT > > 2009-02-05 14:04:51 [DEBUG] switch_core_state_machine.c:444 > switch_core_session_run() (sofia/external/0225490317) State INIT > > 2009-02-05 14:04:51 [DEBUG] mod_sofia.c:83 sofia_on_init() > sofia/external/0225490317 SOFIA INIT > > 2009-02-05 14:04:52 [DEBUG] sofia_glue.c:566 > sofia_glue_ext_address_lookup() STUN Success [89.77.89.244]:[50863] > > 2009-02-05 14:04:52 [DEBUG] mod_sofia.c:111 sofia_on_init() > (sofia/external/0225490317) State Change CS_INIT -> CS_ROUTING > > 2009-02-05 14:04:52 [DEBUG] switch_core_session.c:807 > switch_core_session_signal_state_change() Send signal > sofia/external/0225490317 [BREAK] > > 2009-02-05 14:04:52 [DEBUG] switch_core_state_machine.c:444 > switch_core_session_run() (sofia/external/0225490317) State INIT going to > sleep > > 2009-02-05 14:04:52 [DEBUG] switch_core_state_machine.c:379 > switch_core_session_run() (sofia/external/0225490317) Running State Change > CS_ROUTING > > 2009-02-05 14:04:52 [DEBUG] switch_core_state_machine.c:447 > switch_core_session_run() (sofia/external/0225490317) State ROUTING > > 2009-02-05 14:04:52 [DEBUG] mod_sofia.c:130 sofia_on_routing() > sofia/external/0225490317 SOFIA ROUTING > > 2009-02-05 14:04:52 [DEBUG] switch_ivr_originate.c:58 > originate_on_routing() (sofia/external/0225490317) State Change CS_ROUTING > -> CS_CONSUME_MEDIA > > 2009-02-05 14:04:52 [DEBUG] switch_core_session.c:807 > switch_core_session_signal_state_change() Send signal > sofia/external/0225490317 [BREAK] > > 2009-02-05 14:04:52 [DEBUG] switch_core_state_machine.c:447 > switch_core_session_run() (sofia/external/0225490317) State ROUTING going to > sleep > > 2009-02-05 14:04:52 [DEBUG] switch_core_state_machine.c:379 > switch_core_session_run() (sofia/external/0225490317) Running State Change > CS_CONSUME_MEDIA > > 2009-02-05 14:04:52 [DEBUG] switch_core_state_machine.c:466 > switch_core_session_run() (sofia/external/0225490317) State CONSUME_MEDIA > > 2009-02-05 14:04:52 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state() > Channel sofia/external/0225490317 entering state [calling] > > send 1117 bytes to udp/[194.9.25.21]:5060 at 13:04:52.572008: > > > ------------------------------------------------------------------------ > > INVITE sip:0225490317 at sip.halonet.plSIP/2.0 > > Via: SIP/2.0/UDP 83.13.98.165:5080;rport;branch=z9hG4bKyvKBDBpvBQe2p > > Max-Forwards: 69 > > From: "Extension 1008" > ;transport=udp>;tag=FZp353yettFNN > > To: > > > Call-ID: 6b05cee4-6e28-122c-8396-000c7680c408 > > CSeq: 110801210 INVITE > > Contact: > > User-Agent: FreeSWITCH-mod_sofia/1.0.2-wyeksportowane > > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO > > Supported: timer, precondition, path, replaces > > Allow-Events: talk, refer > > Min-SE: 120 > > Content-Type: application/sdp > > Content-Disposition: session > > Content-Length: 309 > > Remote-Party-ID: "Extension 1008" > >;screen=yes;privacy=off > > > > v=0 > > o=FreeSWITCH 3996368035436745512 7588776177330183331 IN IP4 > 89.77.89.244 > > s=FreeSWITCH > > c=IN IP4 89.77.89.244 > > t=0 0 > > m=audio 50863 RTP/AVP 0 8 3 101 13 > > a=rtpmap:0 PCMU/8000 > > a=rtpmap:8 PCMA/8000 > > a=rtpmap:3 GSM/8000 > > a=rtpmap:101 telephone-event/8000 > > a=fmtp:101 0-16 > > a=rtpmap:13 CN/8000 > > a=ptime:20 > > > ------------------------------------------------------------------------ > > recv 760 bytes from udp/[194.9.25.21]:5060 at 13:04:52.993246: > > > ------------------------------------------------------------------------ > > SIP/2.0 407 Proxy Authentication Required > > Via: SIP/2.0/UDP 83.13.98.165:5080 > ;rport=5080;branch=z9hG4bKyvKBDBpvBQe2p;received=89.77.89.244 > > From: "Extension 1008" > ;transport=udp>;tag=FZp353yettFNN > > To: > >;tag=03bd8e75ec97c9ee65c772e401792a5a.5fa3 > > Call-ID: 6b05cee4-6e28-122c-8396-000c7680c408 > > CSeq: 110801210 INVITE > > Proxy-Authenticate: Digest realm="sip.halonet.pl", > nonce="498ae51439454939d240f51ab9455e2c8505c7aa", qop="auth" > > Server: Sip EXpress router (2.0.0-rc1 (i386/linux)) > > Content-Length: 0 > > Warning: 392 194.9.25.21:5060 "Noisy feedback tells: pid=22400 > req_src_ip=89.77.89.244 req_src_port=50866 in_uri= > sip:0225490317 at sip.halonet.pl out_uri= > sip:0225490317 at sip.halonet.pl via_cnt==1" > > > > > ------------------------------------------------------------------------ > > send 387 bytes to udp/[194.9.25.21]:5060 at 13:04:52.993616: > > > ------------------------------------------------------------------------ > > ACK sip:0225490317 at sip.halonet.pl SIP/2.0 > > Via: SIP/2.0/UDP 83.13.98.165:5080;rport;branch=z9hG4bKyvKBDBpvBQe2p > > Max-Forwards: 69 > > From: "Extension 1008" > ;transport=udp>;tag=FZp353yettFNN > > To: > >;tag=03bd8e75ec97c9ee65c772e401792a5a.5fa3 > > Call-ID: 6b05cee4-6e28-122c-8396-000c7680c408 > > CSeq: 110801210 ACK > > Content-Length: 0 > > > > > ------------------------------------------------------------------------ > > send 1409 bytes to udp/[194.9.25.21]:5060 at 13:04:52.994530: > > > ------------------------------------------------------------------------ > > INVITE sip:0225490317 at sip.halonet.plSIP/2.0 > > Via: SIP/2.0/UDP 83.13.98.165:5080;rport;branch=z9hG4bKZ5c4e66Z8Z4mj > > Max-Forwards: 69 > > From: "Extension 1008" > ;transport=udp>;tag=FZp353yettFNN > > To: > > > Call-ID: 6b05cee4-6e28-122c-8396-000c7680c408 > > CSeq: 110801211 INVITE > > Contact: > > Expires: 600 > > User-Agent: FreeSWITCH-mod_sofia/1.0.2-wyeksportowane > > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO > > Supported: timer, precondition, path, replaces > > Allow-Events: talk, refer > > Proxy-Authorization: Digest username="fixed0248b", realm=" > sip.halonet.pl", nonce="498ae51439454939d240f51ab9455e2c8505c7aa", > cnonce="a1oY524oEiyWgwAMdoDECA", algorithm=MD5, uri=" > sip:0225490317 at sip.halonet.pl ", > response="f05fad749d4ff6d5a75d61e3283c3afa", qop=auth, nc=00000001 > > Min-SE: 120 > > Content-Type: application/sdp > > Content-Disposition: session > > Content-Length: 309 > > Remote-Party-ID: "Extension 1008" > >;screen=yes;privacy=off > > > > v=0 > > o=FreeSWITCH 3996368035436745512 7588776177330183331 IN IP4 > 89.77.89.244 > > s=FreeSWITCH > > c=IN IP4 89.77.89.244 > > t=0 0 > > m=audio 50863 RTP/AVP 0 8 3 101 13 > > a=rtpmap:0 PCMU/8000 > > a=rtpmap:8 PCMA/8000 > > a=rtpmap:3 GSM/8000 > > a=rtpmap:101 telephone-event/8000 > > a=fmtp:101 0-16 > > a=rtpmap:13 CN/8000 > > a=ptime:20 > > > ------------------------------------------------------------------------ > > 2009-02-05 14:04:52 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state() > Channel sofia/external/0225490317 entering state [calling] > > recv 618 bytes from udp/[194.9.25.21]:5060 at 13:04:53.349700: > > > ------------------------------------------------------------------------ > > SIP/2.0 100 trying -- your call is important to us > > Via: SIP/2.0/UDP 83.13.98.165:5080 > ;rport=5080;branch=z9hG4bKZ5c4e66Z8Z4mj;received=89.77.89.244 > > From: "Extension 1008" > ;transport=udp>;tag=FZp353yettFNN > > To: > > > Call-ID: 6b05cee4-6e28-122c-8396-000c7680c408 > > CSeq: 110801211 INVITE > > Server: Sip EXpress router (2.0.0-rc1 (i386/linux)) > > Content-Length: 0 > > Warning: 392 194.9.25.21:5060 "Noisy feedback tells: pid=22332 > req_src_ip=89.77.89.244 req_src_port=50866 in_uri= > sip:0225490317 at sip.halonet.pl out_uri= > sip:0225490317 at 217.11.128.50:5060 via_cnt==1" > > > > > ------------------------------------------------------------------------ > > recv 469 bytes from udp/[194.9.25.21]:5060 at 13:04:53.861812: > > > ------------------------------------------------------------------------ > > SIP/2.0 500 Internal Server Error > > Via: SIP/2.0/UDP 83.13.98.165:5080 > ;received=89.77.89.244;rport=5080;branch=z9hG4bKZ5c4e66Z8Z4mj > > From: "Extension 1008" > ;transport=udp>;tag=FZp353yettFNN > > To: > >;tag=4961D2C4-DA7 > > Date: Thu, 05 Feb 2009 13:04:39 GMT > > Call-ID: 6b05cee4-6e28-122c-8396-000c7680c408 > > Server: Cisco-SIPGateway/IOS-12.x > > CSeq: 110801211 INVITE > > Allow-Events: telephone-event > > Content-Length: 0 > > > > > ------------------------------------------------------------------------ > > send 362 bytes to udp/[194.9.25.21]:5060 at 13:04:53.862199: > > > ------------------------------------------------------------------------ > > ACK sip:0225490317 at sip.halonet.pl SIP/2.0 > > Via: SIP/2.0/UDP 83.13.98.165:5080;rport;branch=z9hG4bKZ5c4e66Z8Z4mj > > Max-Forwards: 69 > > From: "Extension 1008" > ;transport=udp>;tag=FZp353yettFNN > > To: > >;tag=4961D2C4-DA7 > > Call-ID: 6b05cee4-6e28-122c-8396-000c7680c408 > > CSeq: 110801211 ACK > > Content-Length: 0 > > > > > ------------------------------------------------------------------------ > > 2009-02-05 14:04:53 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state() > Channel sofia/external/0225490317 entering state [terminated] > > 2009-02-05 14:04:53 [NOTICE] sofia.c:3090 sofia_handle_sip_i_state() > Hangup sofia/external/0225490317 [CS_CONSUME_MEDIA] > [NORMAL_TEMPORARY_FAILURE] > > 2009-02-05 14:04:53 [DEBUG] switch_channel.c:1494 > switch_channel_perform_hangup() Send signal sofia/external/0225490317 [KILL] > > 2009-02-05 14:04:53 [DEBUG] switch_core_session.c:807 > switch_core_session_signal_state_change() Send signal > sofia/external/0225490317 [BREAK] > > 2009-02-05 14:04:53 [DEBUG] switch_core_state_machine.c:466 > switch_core_session_run() (sofia/external/0225490317) State CONSUME_MEDIA > going to sleep > > 2009-02-05 14:04:53 [DEBUG] switch_core_state_machine.c:379 > switch_core_session_run() (sofia/external/0225490317) Running State Change > CS_HANGUP > > 2009-02-05 14:04:53 [DEBUG] switch_core_state_machine.c:410 > switch_core_session_run() (sofia/external/0225490317) State HANGUP > > 2009-02-05 14:04:53 [DEBUG] mod_sofia.c:253 sofia_on_hangup() > sofia/external/0225490317 Overriding SIP cause 503 with 500 from the other > leg > > 2009-02-05 14:04:53 [DEBUG] mod_sofia.c:287 sofia_on_hangup() Channel > sofia/external/0225490317 hanging up, cause: NORMAL_TEMPORARY_FAILURE > > 2009-02-05 14:04:53 [DEBUG] switch_core_state_machine.c:46 > switch_core_standard_on_hangup() sofia/external/0225490317 Standard HANGUP, > cause: NORMAL_TEMPORARY_FAILURE > > 2009-02-05 14:04:53 [DEBUG] switch_core_state_machine.c:410 > switch_core_session_run() (sofia/external/0225490317) State HANGUP going to > sleep > > 2009-02-05 14:04:53 [DEBUG] switch_core_session.c:939 > switch_core_session_thread() Session 16 (sofia/external/0225490317) Locked, > Waiting on external entities > > 2009-02-05 14:04:53 [DEBUG] switch_ivr_originate.c:1695 > switch_ivr_originate() Originate Resulted in Error Cause: 41 > [NORMAL_TEMPORARY_FAILURE] > > 2009-02-05 14:04:53 [DEBUG] switch_channel.c:177 > switch_channel_audio_sync() sofia/internal/sip:1008 at 192.168.1.126:5070receive message [AUDIO_SYNC] > > 2009-02-05 14:04:53 [NOTICE] switch_core_session.c:957 > switch_core_session_thread() Session 16 (sofia/external/0225490317) Ended > > 2009-02-05 14:04:53 [INFO] mod_dptools.c:1909 audio_bridge_function() > Originate Failed. Cause: NORMAL_TEMPORARY_FAILURE > > 2009-02-05 14:04:53 [NOTICE] mod_dptools.c:1936 audio_bridge_function() > Hangup sofia/internal/sip:1008 at 192.168.1.126:5070 [CS_EXECUTE] > [NORMAL_TEMPORARY_FAILURE] > > 2009-02-05 14:04:53 [NOTICE] switch_core_session.c:959 > switch_core_session_thread() Close Channel sofia/external/0225490317 > [CS_HANGUP] > > 2009-02-05 14:04:53 [DEBUG] switch_channel.c:1494 > switch_channel_perform_hangup() Send signal sofia/internal/ > sip:1008 at 192.168.1.126:5070 [KILL] > > 2009-02-05 14:04:53 [DEBUG] switch_core_session.c:807 > switch_core_session_signal_state_change() Send signal sofia/internal/ > sip:1008 at 192.168.1.126:5070 [BREAK] > > 2009-02-05 14:04:53 [DEBUG] switch_core_state_machine.c:454 > switch_core_session_run() (sofia/internal/sip:1008 at 192.168.1.126:5070) > State EXECUTE going to sleep > > 2009-02-05 14:04:53 [DEBUG] switch_core_state_machine.c:379 > switch_core_session_run() (sofia/internal/sip:1008 at 192.168.1.126:5070) > Running State Change CS_HANGUP > > 2009-02-05 14:04:53 [DEBUG] switch_core_state_machine.c:410 > switch_core_session_run() (sofia/internal/sip:1008 at 192.168.1.126:5070) > State HANGUP > > 2009-02-05 14:04:53 [DEBUG] mod_sofia.c:253 sofia_on_hangup() > sofia/internal/sip:1008 at 192.168.1.126:5070 Overriding SIP cause 503 with > 500 from the other leg > > 2009-02-05 14:04:53 [DEBUG] mod_sofia.c:287 sofia_on_hangup() Channel > sofia/internal/sip:1008 at 192.168.1.126:5070 hanging up, cause: > NORMAL_TEMPORARY_FAILURE > > 2009-02-05 14:04:53 [DEBUG] mod_sofia.c:344 sofia_on_hangup() Sending BYE > to sofia/internal/sip:1008 at 192.168.1.126:5070 > > send 648 bytes to udp/[192.168.1.126]:5070 at 13:04:53.868506: > > > ------------------------------------------------------------------------ > > BYE sip:1008 at 192.168.1.126:5070 SIP/2.0 > > Via: SIP/2.0/UDP 192.168.1.122;rport;branch=z9hG4bKH893yar8p7X4j > > Max-Forwards: 70 > > From: "FreeSWITCH" > >;tag=9vpFm3tBc8gSe > > To: ;tag=2111843199 > > Call-ID: 68736fc7-6e28-122c-8396-000c7680c408 > > CSeq: 110801209 BYE > > Contact: > > User-Agent: FreeSWITCH-mod_sofia/1.0.2-wyeksportowane > > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH > > Supported: timer, precondition, path, replaces > > Reason: Q.850;cause=41;text="NORMAL_TEMPORARY_FAILURE" > > Content-Length: 0 > > > > > ------------------------------------------------------------------------ > > 2009-02-05 14:04:53 [DEBUG] switch_core_state_machine.c:46 > switch_core_standard_on_hangup() sofia/internal/ > sip:1008 at 192.168.1.126:5070 Standard HANGUP, cause: > NORMAL_TEMPORARY_FAILURE > > 2009-02-05 14:04:53 [DEBUG] switch_core_state_machine.c:410 > switch_core_session_run() (sofia/internal/sip:1008 at 192.168.1.126:5070) > State HANGUP going to sleep > > 2009-02-05 14:04:53 [DEBUG] switch_core_session.c:939 > switch_core_session_thread() Session 15 (sofia/internal/ > sip:1008 at 192.168.1.126:5070) Locked, Waiting on external entities > > 2009-02-05 14:04:53 [NOTICE] switch_core_session.c:957 > switch_core_session_thread() Session 15 (sofia/internal/ > sip:1008 at 192.168.1.126:5070) Ended > > 2009-02-05 14:04:53 [NOTICE] switch_core_session.c:959 > switch_core_session_thread() Close Channel sofia/internal/ > sip:1008 at 192.168.1.126:5070 [CS_HANGUP] > > recv 354 bytes from udp/[192.168.1.126]:5070 at 13:04:53.882344: > > > ------------------------------------------------------------------------ > > SIP/2.0 200 Ok > > Via: SIP/2.0/UDP 192.168.1.122;rport;branch=z9hG4bKH893yar8p7X4j > > From: "FreeSWITCH" > >;tag=9vpFm3tBc8gSe > > To: ;tag=2111843199 > > Contact: > > Call-ID: 68736fc7-6e28-122c-8396-000c7680c408 > > CSeq: 110801209 BYE > > Server: X-Lite release 1105d > > Content-Length: 0 > > > > > ------------------------------------------------------------------------ > Dnia 04-02-2009, ?ro o godzinie 10:22 -0600, Anthony Minessale pisze: > > can you press f8 for debug and try that apiExecute and post the > > results? > > > > On Wed, Feb 4, 2009 at 8:46 AM, Jacek Sokulski > > wrote: > > Thanks Anthony, > > the js snippets are very instructive. > > A couple of points: > > 1. The code with apiExecute does not work (local phone is > > connected, but > > after picking up it hungs up immediately), other examples are > > working > > fine. > > > > 2. It does not show how initiate external call without > > existing session. > > > > 3. How can one pass the call through dialplan? > > > > Jacek > > > > PS. > > we got the code probable from wiki or from this mialing list. > > > > Dnia 04-02-2009, ?ro o godzinie 08:09 -0600, Anthony Minessale > > pisze: > > > > > Where did you learn how to use js this way? > > > session.originate is being misused here and is depricated > > and may be > > > removed. > > > > > > the first arg to session.originate is either undefined or a > > > *different* session (the a leg) > > > > > > session1 = new Session(); > > > session1.originate(undefined, > > > "{ignore_early_media=true}user/1008 at 192.168.1.122"); > > > > > > > > session1.setVariable("effective_caller_id_number","fixed0248b"); > > > > > > //once you have session1 when you originate session2 you > > pass session1 > > > as the arg > > > // the effective_caller_id is taken from session1 > > > > > > session2 = new Session(); > > > session2.originate(session1, > > "sofia/gateway/halonet/0225490317"); > > > > > > Anyway this whole code is depricated in favor of this: > > > > > > session1 = new > > > Session("{ignore_early_media=true}user/1008 at 192.168.1.122"); > > > > > > if (session1.ready()) { > > > > > session1.setVariable("effective_caller_id_number","fixed0248b"); > > > session2 = new Session("sofia/gateway/halonet/0225490317", > > > session1); > > > } > > > > > > and could be further refactored down to this: > > > > > > session1 = new > > > Session("{ignore_early_media=true}user/1008 at 192.168.1.122"); > > > if (session1.ready()) { > > > > > session1.setVariable("effective_caller_id_number","fixed0248b"); > > > session1.execute("bridge", > > "sofia/gateway/halonet/0225490317"); > > > } > > > > > > or down to this one line of code that will setup the call > > detached > > > from the script and exit. > > > > > > var result = apiExecute("originate", > > > > > > "{effective_caller_id_number=fixed0248b,origination_caller_id_number=1000,ignore_early_media=true}user/ > 1008 at 192.168.1.122 bridge:sofia/gateway/halonet/0225490317 inline"); > > > > > > if you dont care about the result and want to exit even > > before the > > > call is completed. > > > > > > var result = apiExecute("bgapi", "originate > > > > > > {effective_caller_id_number=fixed0248b,origination_caller_id_number=1000,ignore_early_media=true}user/ > 1008 at 192.168.1.122 bridge:sofia/gateway/halonet/0225490317 inline"); > > > > > > > > > > > > On Wed, Feb 4, 2009 at 2:51 AM, Jacek Sokulski > > > wrote: > > > > > > We have tried setting both > > effective_caller_id_number and > > > origination_caller_id_number: > > > > > > > > > session1.originate(session1,"{origination_caller_id_number=fixed0248b}sofia/gateway/halonet/0225490317",15); > > > but the problem still exists. The solution we have > > found for > > > the case > > > when we originate two calls, local and external, is > > as follow: > > > > > > session1 = new Session(); > > > > > session1.originate(session1,"user/1003 at 192.168.1.122 > ",15);//local > > > if(session1.ready()) { > > > > > session1.execute("execute_extension","00930691688627 XML > > > default");//external > > > } > > > > > > so the external call goes through the dialplan. > > > It does not work if both calls are external. One > > possible > > > solution could be > > > to pass the originating call through dialplan > > (loopback?) but > > > we have not managed > > > to figure out how to do it. > > > > > > Thanks > > > Jacek > > > > > > Dnia 03-02-2009, wto o godzinie 14:31 -0300, Nicolas > > Brenner > > > pisze: > > > > > > > Oops! Well, fortunately I don't use that voip > > provider > > > anymore (nor the script). > > > > > > > > Thanks Brian. > > > > > > > > Nicolas > > > > > > > > On Tue, Feb 3, 2009 at 2:25 PM, Brian West > > > wrote: > > > > > YOU should NEVER use this method or call > > setCallerData at > > > all you > > > > > should use the correct methods to override the > > callerid. > > > > > > > > > > If its a B-Leg born from an A-Leg you use these > > on the on > > > the A-Leg: > > > > > > > > > > > > > > > > http://wiki.freeswitch.org/wiki/Channel_Variables#effective_caller_id_name > > > > > > > > > > > http://wiki.freeswitch.org/wiki/Channel_Variables#effective_caller_id_number > > > > > > > > > > If you're originating you use this: > > > > > > > > > > > > > > > > http://wiki.freeswitch.org/wiki/Channel_Variables#origination_caller_id_name > > > > > > > > > > > http://wiki.freeswitch.org/wiki/Channel_Variables#origination_caller_id_number > > > > > > > > > > /b > > > > > > > > _______________________________________________ > > > > Freeswitch-users mailing list > > > > Freeswitch-users at lists.freeswitch.org > > > > > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > > > > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > http://www.freeswitch.org > > > > > > > > > _______________________________________________ > > > Freeswitch-users mailing list > > > Freeswitch-users at lists.freeswitch.org > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > > > > > > > > > > > > > -- > > > Anthony Minessale II > > > > > > FreeSWITCH http://www.freeswitch.org/ > > > ClueCon http://www.cluecon.com/ > > > > > > AIM: anthm > > > MSN:anthony_minessale at hotmail.com > > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > > IRC: irc.freenode.net #freeswitch > > > > > > FreeSWITCH Developer Conference > > > sip:888 at conference.freeswitch.org > > > iax:guest at conference.freeswitch.org/888 > > > googletalk:conf+888 at conference.freeswitch.org > > > pstn:213-799-1400 > > > _______________________________________________ > > > Freeswitch-users mailing list > > > Freeswitch-users at lists.freeswitch.org > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090205/0f69fe0e/attachment-0001.html From anthony.minessale at gmail.com Thu Feb 5 07:48:12 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 5 Feb 2009 09:48:12 -0600 Subject: [Freeswitch-users] FreeSwitch setup as a "Dumb" SBC In-Reply-To: <498AAC59.60407@post.cz> References: <498AAC59.60407@post.cz> Message-ID: <191c3a030902050748p44a86740hd823aedee5d81f56@mail.gmail.com> fyi, I am pretty sure i disabled the mmap calls so it's not memory mapped it's all in ram the whole time. The fsxml file is just left behind to show you the last successful load On Thu, Feb 5, 2009 at 3:07 AM, kokoska rokoska wrote: > > > > Ken Rice napsal(a): > > That file is memory mapped but it can still take an IO hit... Mounting > fs/db > > in a ramdrive still helps out and it doesn't have to be that big of a ram > > drive for the testing you are doing > > > > Thank you very much, Ken, for that info! > > I do it like I wrote before, but I have to wait till the "blades" aren't > busy. They are not only for my testing and the other users have "tasks" > with higher priority than my sipp testing :-) > > As soon as I have the result, I'll post them. I'm pretty sure, that > changes in FS code and "all in RAM" helps much :-) > > Best regards, > > kokoska.rokoska > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090205/7563273b/attachment.html From kerrada2003 at yahoo.com Thu Feb 5 08:34:08 2009 From: kerrada2003 at yahoo.com (Ali Al-Rubaie) Date: Thu, 5 Feb 2009 08:34:08 -0800 (PST) Subject: [Freeswitch-users] SIP Authentication In-Reply-To: Message-ID: <939444.77774.qm@web33707.mail.mud.yahoo.com> We're using HelpCaster softphone. The issue here is that in Digest Authentication, if the server sends the parameter "qop" in the challenge then the client should respond with the "cnonce" parameter. The parameter "qop" is optional in Digest Auth. So the question here is that, can we configure FreeSWITCH so that it will not send "qop" in the challenge? Thanks! --- On Wed, 2/4/09, freeswitch-users-request at lists.freeswitch.org wrote: From: freeswitch-users-request at lists.freeswitch.org Subject: Freeswitch-users Digest, Vol 32, Issue 39 To: freeswitch-users at lists.freeswitch.org Date: Wednesday, February 4, 2009, 2:05 PM Send Freeswitch-users mailing list submissions to freeswitch-users at lists.freeswitch.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.freeswitch.org/mailman/listinfo/freeswitch-users or, via email, send a message with subject or body 'help' to freeswitch-users-request at lists.freeswitch.org You can reach the person managing the list at freeswitch-users-owner at lists.freeswitch.org When replying, please edit your Subject line so it is more specific than "Re: Contents of Freeswitch-users digest..." Today's Topics: 1. Re: SIP Authentication (Brian West) 2. Re: origainate through sofia gateway (Michael Collins) 3. Recording background music and voice is out of sync (Daniel Liang) 4. Re: Q931 decoding Update (Gopalakrishnan A.N) 5. mod_limit (Chav Paskov) 6. Re: mod_limit (Michael Collins) 7. Re: mod_limit (Chav Paskov) 8. Re: mod_limit (Michael Collins) ---------------------------------------------------------------------- Message: 1 Date: Wed, 4 Feb 2009 10:52:45 -0600 From: Brian West Subject: Re: [Freeswitch-users] SIP Authentication To: freeswitch-users at lists.freeswitch.org Message-ID: <7DAC91F6-2FD3-464D-AA81-321EBCADC8C0 at freeswitch.org> Content-Type: text/plain; charset=US-ASCII; format=flowed; delsp=yes What client is this? I also notice we receive port 3458 and reply to port 1059... /b On Feb 4, 2009, at 10:17 AM, Ali Al-Rubaie wrote: > What I have noted is that the client does not send the values for > "cnonce" and "nc" in the response. I'm not sure if this is the > reason, however how this problem can be solved? > > Thanks, > > Ali ------------------------------ Message: 2 Date: Wed, 4 Feb 2009 09:41:07 -0800 From: Michael Collins Subject: Re: [Freeswitch-users] origainate through sofia gateway To: freeswitch-users at lists.freeswitch.org Message-ID: <87f2f3b90902040941r61d669aaie949aa7cc8578a9a at mail.gmail.com> Content-Type: text/plain; charset=ISO-8859-1 I'll make sure the substance of this is in the wiki and I'll look for references to the deprecated way and remove those. -MC On Wed, Feb 4, 2009 at 6:09 AM, Anthony Minessale wrote: > Where did you learn how to use js this way? > session.originate is being misused here and is depricated and may be > removed. > > the first arg to session.originate is either undefined or a *different* > session (the a leg) > > session1 = new Session(); > session1.originate(undefined, > "{ignore_early_media=true}user/1008 at 192.168.1.122"); > > session1.setVariable("effective_caller_id_number","fixed0248b"); > > //once you have session1 when you originate session2 you pass session1 as > the arg > // the effective_caller_id is taken from session1 > > session2 = new Session(); > session2.originate(session1, "sofia/gateway/halonet/0225490317"); > > Anyway this whole code is depricated in favor of this: > > session1 = new Session("{ignore_early_media=true}user/1008 at 192.168.1.122"); > if (session1.ready()) { > session1.setVariable("effective_caller_id_number","fixed0248b"); > session2 = new Session("sofia/gateway/halonet/0225490317", session1); > } > > and could be further refactored down to this: > > session1 = new Session("{ignore_early_media=true}user/1008 at 192.168.1.122"); > if (session1.ready()) { > session1.setVariable("effective_caller_id_number","fixed0248b"); > session1.execute("bridge", "sofia/gateway/halonet/0225490317"); > } > > or down to this one line of code that will setup the call detached from the > script and exit. > > var result = apiExecute("originate", > "{effective_caller_id_number=fixed0248b,origination_caller_id_number=1000,ignore_early_media=true}user/1008 at 192.168.1.122 > bridge:sofia/gateway/halonet/0225490317 inline"); > > if you dont care about the result and want to exit even before the call is > completed. > > var result = apiExecute("bgapi", "originate > {effective_caller_id_number=fixed0248b,origination_caller_id_number=1000,ignore_early_media=true}user/1008 at 192.168.1.122 > bridge:sofia/gateway/halonet/0225490317 inline"); > > > > On Wed, Feb 4, 2009 at 2:51 AM, Jacek Sokulski > wrote: >> >> We have tried setting both effective_caller_id_number and >> origination_caller_id_number: >> >> >> session1.originate(session1,"{origination_caller_id_number=fixed0248b}sofia/gateway/halonet/0225490317",15); >> but the problem still exists. The solution we have found for the case >> when we originate two calls, local and external, is as follow: >> >> session1 = new Session(); >> session1.originate(session1,"user/1003 at 192.168.1.122",15);//local >> if(session1.ready()) { >> session1.execute("execute_extension","00930691688627 XML >> default");//external >> } >> >> so the external call goes through the dialplan. >> It does not work if both calls are external. One possible solution could >> be >> to pass the originating call through dialplan (loopback?) but we have not >> managed >> to figure out how to do it. >> >> Thanks >> Jacek >> >> Dnia 03-02-2009, wto o godzinie 14:31 -0300, Nicolas Brenner pisze: >> > Oops! Well, fortunately I don't use that voip provider anymore (nor the >> > script). >> > >> > Thanks Brian. >> > >> > Nicolas >> > >> > On Tue, Feb 3, 2009 at 2:25 PM, Brian West wrote: >> > > YOU should NEVER use this method or call setCallerData at all you >> > > should use the correct methods to override the callerid. >> > > >> > > If its a B-Leg born from an A-Leg you use these on the on the A-Leg: >> > > >> > > >> > > http://wiki.freeswitch.org/wiki/Channel_Variables#effective_caller_id_name >> > > >> > > http://wiki.freeswitch.org/wiki/Channel_Variables#effective_caller_id_number >> > > >> > > If you're originating you use this: >> > > >> > > >> > > http://wiki.freeswitch.org/wiki/Channel_Variables#origination_caller_id_name >> > > >> > > http://wiki.freeswitch.org/wiki/Channel_Variables#origination_caller_id_number >> > > >> > > /b >> > >> > _______________________________________________ >> > Freeswitch-users mailing list >> > Freeswitch-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > ------------------------------ Message: 3 Date: Wed, 4 Feb 2009 09:43:10 -0800 From: "Daniel Liang" Subject: [Freeswitch-users] Recording background music and voice is out of sync To: Message-ID: <0B02E756F603CC409EB553879B090CC80A23EBB5 at HPEXCHVS01.exchange.airg> Content-Type: text/plain; charset="us-ascii" What I did was the following: First, I sent the playback command: call-command: execute execute-app-name: playback execute-app-arg: Then I send uuid_record (Sorry, it was not Record command): api uuid_record start 120 I also tried replacing the playback command with: api uuid_displace start 0 mux But the end results are the same. The recorded user's voice is about 0.5 second behind the expected result. Thanks, Daniel -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: February 3, 2009 6:36 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Recording background music and voice is outof sync Can you show us an example of how you're doing this? Playback and Record aren't async so you'll need to show us how you're doing this. Also don't hijack threads you hit replay on the one "Re: [Freeswitch- users] FreeSwitch setup as a "Dumb" SBC" as the subject, deleted the subject and started a new body. That hijacks the thread and that can cause your problem to go ignored in some cases if people aren't interested in the thread topic depending on how their reader threads the emails. Please click new message and type freeswitch- users at lists.freeswitch.org in and then input your subject and body to start a new thread. Thanks, Brian West FreeSWITCH.org On Feb 3, 2009, at 8:21 PM, Daniel Liang wrote: > Hi, > > I was trying to record a background music with a user's voice at the > same time. I did a playback and started recording. But the recorded > user's voice and the background music is about 0.5 second out of sync. > I also tried to use uuid_displace instead of playback, but I got the > same result. _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090204/2d1124e2/attachment-0001.html ------------------------------ Message: 4 Date: Wed, 4 Feb 2009 23:26:14 +0530 From: "Gopalakrishnan A.N" Subject: Re: [Freeswitch-users] Q931 decoding Update To: freeswitch-users at lists.freeswitch.org Message-ID: <2ea4d47e0902040956v75c5472foa4649c50b7340484 at mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" Hi, Its a awesome. Can the packet capturing be done with event socket? -- Thank you with regards, Gopal, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090204/74f6434a/attachment-0001.html ------------------------------ Message: 5 Date: Wed, 04 Feb 2009 09:59:48 -0800 From: Chav Paskov Subject: [Freeswitch-users] mod_limit To: freeswitch-users at lists.freeswitch.org Message-ID: <4989D794.1010805 at shaw.ca> Content-Type: text/plain; charset=ISO-8859-1; format=flowed Hi , is it possible to use mod_limit in case if the end point is not registered / gateway for example/. Regards Chav ------------------------------ Message: 6 Date: Wed, 4 Feb 2009 10:06:52 -0800 From: Michael Collins Subject: Re: [Freeswitch-users] mod_limit To: freeswitch-users at lists.freeswitch.org Message-ID: <87f2f3b90902041006v7d7d7ff2t9961274905d9d5b0 at mail.gmail.com> Content-Type: text/plain; charset=ISO-8859-1 On Wed, Feb 4, 2009 at 9:59 AM, Chav Paskov wrote: > Hi , > is it possible to use mod_limit in case if the end point is not > registered / gateway for example/. Could you add some detail to this question? What are you trying to do? (mod_limit may or may not work, but there might be another solution which is why I am asking.) -MC > Regards > Chav > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > ------------------------------ Message: 7 Date: Wed, 04 Feb 2009 10:54:56 -0800 From: Chav Paskov Subject: Re: [Freeswitch-users] mod_limit To: freeswitch-users at lists.freeswitch.org Message-ID: <4989E480.1080105 at shaw.ca> Content-Type: text/plain; charset=ISO-8859-1; format=flowed Michael Collins wrote: > On Wed, Feb 4, 2009 at 9:59 AM, Chav Paskov wrote: > >> Hi , >> is it possible to use mod_limit in case if the end point is not >> registered / gateway for example/. >> > > Could you add some detail to this question? What are you trying to do? > (mod_limit may or may not work, but there might be another solution > which is why I am asking.) > > -MC > > >> Regards >> Chav >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > i have few gateways under my ACL that are allowed to send calls to FS, but i want to be able to enforce "capacity" policy on the traffic coming from any one of them depending on total termination capacity on my termination end. Let say GW 1 has to be limited to make 10 simultaneous calls while GW2 could make up to 30 and so on. Regards Chav ------------------------------ Message: 8 Date: Wed, 4 Feb 2009 11:05:09 -0800 From: Michael Collins Subject: Re: [Freeswitch-users] mod_limit To: freeswitch-users at lists.freeswitch.org Message-ID: <87f2f3b90902041105l50f51f08t230bab8d69eefb4e at mail.gmail.com> Content-Type: text/plain; charset=ISO-8859-1 On Wed, Feb 4, 2009 at 10:54 AM, Chav Paskov wrote: > Michael Collins wrote: >> On Wed, Feb 4, 2009 at 9:59 AM, Chav Paskov wrote: >> >>> Hi , >>> is it possible to use mod_limit in case if the end point is not >>> registered / gateway for example/. >>> >> >> Could you add some detail to this question? What are you trying to do? >> (mod_limit may or may not work, but there might be another solution >> which is why I am asking.) >> >> -MC >> >> >>> Regards >>> Chav >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > i have few gateways under my ACL that are allowed to send calls to FS, > but i want to be able to enforce "capacity" policy on the traffic > coming from any one of them depending on total termination capacity on > my termination end. > Let say GW 1 has to be limited to make 10 simultaneous calls while GW2 > could make up to 30 and so on. I'm sure that this is possible. I don't personally have a way to test all of this but I know that a number of our users are doing things like this currently. Can you hop on to the IRC channel? #freeswitch on irc.freenode.net. A lot of people there can help with this one. -MC (IRC: mercutioviz) > Regards > Chav > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > ------------------------------ _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org End of Freeswitch-users Digest, Vol 32, Issue 39 ************************************************ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090205/15fa27e7/attachment-0001.html From kokoska.rokoska at post.cz Thu Feb 5 08:41:01 2009 From: kokoska.rokoska at post.cz (kokoska rokoska) Date: Thu, 05 Feb 2009 17:41:01 +0100 Subject: [Freeswitch-users] FreeSwitch setup as a "Dumb" SBC In-Reply-To: <191c3a030902050748p44a86740hd823aedee5d81f56@mail.gmail.com> References: <498AAC59.60407@post.cz> <191c3a030902050748p44a86740hd823aedee5d81f56@mail.gmail.com> Message-ID: <498B169D.8080105@post.cz> Anthony Minessale napsal(a): > fyi, > > I am pretty sure i disabled the mmap calls so it's not memory mapped > it's all in ram the whole time. > The fsxml file is just left behind to show you the last successful load > Thank you very much, Anthony, for very useful info! Best regards, kokoska.rokoska From anthony.minessale at gmail.com Thu Feb 5 08:46:54 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 5 Feb 2009 10:46:54 -0600 Subject: [Freeswitch-users] SIP Authentication In-Reply-To: <939444.77774.qm@web33707.mail.mud.yahoo.com> References: <939444.77774.qm@web33707.mail.mud.yahoo.com> Message-ID: <191c3a030902050846o60047c30pa2890707eae386d6@mail.gmail.com> It's optional for us but it's mandatory for the client if we exercise the option which we have opted to always do =D There is no way in the code to disable sending it because we prefer the more secure version of SIP auth. So again it's a bug in the client for not following the protocol. It would be considered a feature in FreeSWITCH to support limping for the sake of this broken client and we currently do not have any plans for implementing this feature. On Thu, Feb 5, 2009 at 10:34 AM, Ali Al-Rubaie wrote: > > We're using HelpCaster softphone. > > The issue here is that in Digest Authentication, if the server sends the > parameter "qop" in the challenge then the client should respond with the > "cnonce" parameter. The parameter "qop" is optional in Digest Auth. So the > question here is that, can we configure FreeSWITCH so that it will not send > "qop" in the challenge? > > Thanks! > > --- On *Wed, 2/4/09, freeswitch-users-request at lists.freeswitch.org < > freeswitch-users-request at lists.freeswitch.org>* wrote: > > From: freeswitch-users-request at lists.freeswitch.org < > freeswitch-users-request at lists.freeswitch.org> > Subject: Freeswitch-users Digest, Vol 32, Issue 39 > To: freeswitch-users at lists.freeswitch.org > Date: Wednesday, February 4, 2009, 2:05 PM > > Send Freeswitch-users mailing list submissions to > freeswitch-users at lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request at lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner at lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Freeswitch-users digest..." > > > Today's Topics: > > 1. Re: SIP Authentication (Brian West) > 2. Re: origainate through sofia gateway (Michael Collins) > 3. Recording background music and voice is out of sync (Daniel Liang) > 4. Re: Q931 decoding Update (Gopalakrishnan A.N) > 5. mod_limit (Chav Paskov) > 6. Re: mod_limit (Michael Collins) > 7. Re: mod_limit (Chav > Paskov) > 8. Re: mod_limit (Michael Collins) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Wed, 4 Feb 2009 10:52:45 -0600 > From: Brian West > Subject: Re: [Freeswitch-users] SIP Authentication > To: freeswitch-users at lists.freeswitch.org > Message-ID: <7DAC91F6-2FD3-464D-AA81-321EBCADC8C0 at freeswitch.org> > Content-Type: text/plain; charset=US-ASCII; format=flowed; delsp=yes > > What client is this? I also notice we receive port 3458 and reply to > port 1059... > > /b > > On Feb 4, 2009, at 10:17 AM, Ali Al-Rubaie wrote: > > > What I have noted is that the client does not send the values for > > "cnonce" and "nc" in the response. I'm not sure if > this is the > > reason, however how this problem can be solved? > > > > Thanks, > > > > Ali > > > > > ------------------------------ > > Message: > 2 > Date: Wed, 4 Feb 2009 09:41:07 -0800 > From: Michael Collins > Subject: Re: [Freeswitch-users] origainate through sofia gateway > To: freeswitch-users at lists.freeswitch.org > Message-ID: > <87f2f3b90902040941r61d669aaie949aa7cc8578a9a at mail.gmail.com> > Content-Type: text/plain; charset=ISO-8859-1 > > I'll make sure the substance of this is in the wiki and I'll look for > references to the deprecated way and remove those. > -MC > > On Wed, Feb 4, 2009 at 6:09 AM, Anthony Minessale > wrote: > > Where did you learn how to use js this way? > > session.originate is being misused here and is depricated and may be > > removed. > > > > the first arg to session.originate is either undefined or a *different* > > session (the a leg) > > > > session1 = new Session(); > > session1.originate(undefined, > > > "{ignore_early_media=true}user/1008 at 192.168.1.122"); > > > > > session1.setVariable("effective_caller_id_number","fixed0248b"); > > > > //once you have session1 when you originate session2 you pass session1 as > > the arg > > // the effective_caller_id is taken from session1 > > > > session2 = new Session(); > > session2.originate(session1, > "sofia/gateway/halonet/0225490317"); > > > > Anyway this whole code is depricated in favor of this: > > > > session1 = new > Session("{ignore_early_media=true}user/1008 at 192.168.1.122"); > > if (session1.ready()) { > > > session1.setVariable("effective_caller_id_number","fixed0248b"); > > session2 = new Session("sofia/gateway/halonet/0225490317", > session1); > > } > > > > and could be further refactored down to this: > > > > session1 = new > Session("{ignore_early_media=true}user/1008 at 192.168.1.122"); > > if > (session1.ready()) { > > > session1.setVariable("effective_caller_id_number","fixed0248b"); > > session1.execute("bridge", > "sofia/gateway/halonet/0225490317"); > > } > > > > or down to this one line of code that will setup the call detached from > the > > script and exit. > > > > var result = apiExecute("originate", > > > "{effective_caller_id_number=fixed0248b,origination_caller_id_number=1000,ignore_early_media=true}user/1008 at 192.168.1.122 > > bridge:sofia/gateway/halonet/0225490317 inline"); > > > > if you dont care about the result and want to exit even before the call is > > completed. > > > > var result = apiExecute("bgapi", "originate > > > {effective_caller_id_number=fixed0248b,origination_caller_id_number=1000,ignore_early_media=true}user/1008 at 192.168.1.122 > > bridge:sofia/gateway/halonet/0225490317 inline"); > > > > > > > > On Wed, Feb 4, 2009 at > 2:51 AM, Jacek Sokulski > > > wrote: > >> > >> We have tried setting both effective_caller_id_number and > >> origination_caller_id_number: > >> > >> > >> > session1.originate(session1,"{origination_caller_id_number=fixed0248b}sofia/gateway/halonet/0225490317",15); > >> but the problem still exists. The solution we have found for the case > >> when we originate two calls, local and external, is as follow: > >> > >> session1 = new Session(); > >> > session1.originate(session1,"user/1003 at 192.168.1.122",15);//local > >> if(session1.ready()) { > >> session1.execute("execute_extension","00930691688627 > XML > >> default");//external > >> } > >> > >> so the external call goes through the dialplan. > >> It does not work if both calls are external. One possible solution > could > >> > be > >> to pass the originating call through dialplan (loopback?) but we have > not > >> managed > >> to figure out how to do it. > >> > >> Thanks > >> Jacek > >> > >> Dnia 03-02-2009, wto o godzinie 14:31 -0300, Nicolas Brenner pisze: > >> > Oops! Well, fortunately I don't use that voip provider > anymore (nor the > >> > script). > >> > > >> > Thanks Brian. > >> > > >> > Nicolas > >> > > >> > On Tue, Feb 3, 2009 at 2:25 PM, Brian West > wrote: > >> > > YOU should NEVER use this method or call setCallerData at > all you > >> > > should use the correct methods to override the callerid. > >> > > > >> > > If its a B-Leg born from an A-Leg you use these on the on > the A-Leg: > >> > > > >> > > > >> > > > > http://wiki.freeswitch.org/wiki/Channel_Variables#effective_caller_id_name > >> > > > >> > > > http://wiki.freeswitch.org/wiki/Channel_Variables#effective_caller_id_number > >> > > > >> > > If you're originating you use this: > >> > > > >> > > > >> > > > http://wiki.freeswitch.org/wiki/Channel_Variables#origination_caller_id_name > >> > > > >> > > > http://wiki.freeswitch.org/wiki/Channel_Variables#origination_caller_id_number > >> > > > >> > > /b > >> > > >> > _______________________________________________ > >> > Freeswitch-users mailing list > >> > Freeswitch-users at lists.freeswitch.org > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> > http://www.freeswitch.org > >> > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > ------------------------------ > > Message: 3 > Date: Wed, 4 Feb 2009 09:43:10 -0800 > From: "Daniel Liang" > Subject: [Freeswitch-users] Recording background music and voice is > out of sync > To: > Message-ID: > <0B02E756F603CC409EB553879B090CC80A23EBB5 at HPEXCHVS01.exchange.airg> > Content-Type: text/plain; charset="us-ascii" > > What I did was the following: > > First, I sent the > playback command: > > call-command: execute > execute-app-name: playback > execute-app-arg: > > Then I send uuid_record (Sorry, it was not Record command): > > api uuid_record start 120 > > I also tried replacing the playback command with: > api uuid_displace start 0 mux > > But the end results are the same. The recorded user's voice is about 0.5 > second behind the expected result. > > Thanks, > Daniel > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Brian West > Sent: February 3, 2009 6:36 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Recording background music and voice is > outof sync > > Can you show us an example of how you're doing this? Playback and > Record aren't async so > you'll need to show us how you're doing > this. > > Also don't hijack threads you hit replay on the one "Re: [Freeswitch- > users] FreeSwitch setup as a "Dumb" SBC" as the subject, deleted > the > subject and started a new body. That hijacks the thread and that can > cause your problem to go ignored in some cases if people aren't > interested in the thread topic depending on how their reader threads the > emails. > > Please click new message and type freeswitch- users at lists.freeswitch.org > in and then input your subject and body to start a new thread. > > Thanks, > Brian West > FreeSWITCH.org > > > On Feb 3, 2009, at 8:21 PM, Daniel Liang wrote: > > > Hi, > > > > I was trying to record a background music with a user's voice at the > > same time. I did a playback and started recording. But the recorded > > user's voice and the background music is about 0.5 second out of sync. > > > I also tried > to use uuid_displace instead of playback, but I got the > > same result. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090204/2d1124e2/attachment-0001.html > > > ------------------------------ > > Message: 4 > Date: Wed, 4 Feb 2009 23:26:14 +0530 > From: "Gopalakrishnan A.N" > Subject: Re: [Freeswitch-users] Q931 decoding Update > To: freeswitch-users at lists.freeswitch.org > Message-ID: > > <2ea4d47e0902040956v75c5472foa4649c50b7340484 at mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > Hi, > Its a awesome. Can the packet capturing be done with event socket? > > -- > Thank you with regards, > Gopal, > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090204/74f6434a/attachment-0001.html > > > ------------------------------ > > Message: 5 > Date: Wed, 04 Feb 2009 09:59:48 -0800 > From: Chav Paskov > Subject: [Freeswitch-users] mod_limit > To: freeswitch-users at lists.freeswitch.org > Message-ID: <4989D794.1010805 at shaw.ca> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Hi , > is it possible to use mod_limit in case if the end point is not > registered / gateway for > example/. > Regards > Chav > > > > ------------------------------ > > Message: 6 > Date: Wed, 4 Feb 2009 10:06:52 -0800 > From: Michael Collins > Subject: Re: [Freeswitch-users] mod_limit > To: freeswitch-users at lists.freeswitch.org > Message-ID: > <87f2f3b90902041006v7d7d7ff2t9961274905d9d5b0 at mail.gmail.com> > Content-Type: text/plain; charset=ISO-8859-1 > > On Wed, Feb 4, 2009 at 9:59 AM, Chav Paskov wrote: > > Hi , > > is it possible to use mod_limit in case if the end point is not > > registered / gateway for example/. > > Could you add some detail to this question? What are you trying to do? > (mod_limit may or may not work, but there might be another solution > which is why I am asking.) > > -MC > > > Regards > > Chav > > > > _______________________________________________ > > Freeswitch-users mailing list > > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > ------------------------------ > > Message: 7 > Date: Wed, 04 Feb 2009 10:54:56 -0800 > From: Chav Paskov > Subject: Re: [Freeswitch-users] mod_limit > To: freeswitch-users at lists.freeswitch.org > Message-ID: <4989E480.1080105 at shaw.ca> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Michael Collins wrote: > > On Wed, Feb 4, 2009 at 9:59 AM, Chav Paskov > wrote: > > > >> Hi , > >> is it possible to use mod_limit in case if the end point is not > >> registered / gateway for example/. > >> > > > > Could you add some detail to this question? What are you trying to do? > > > (mod_limit may or may not work, but there might be another solution > > which is why I am asking.) > > > > -MC > > > > > >> Regards > >> Chav > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > >> > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > i have few gateways under my ACL that > are allowed to send calls to FS, > but i want to be able to enforce "capacity" policy on the traffic > coming from any one of them depending on total termination capacity on > my termination end. > Let say GW 1 has to be limited to make 10 simultaneous calls while GW2 > could make up to 30 and so on. > Regards > Chav > > > > ------------------------------ > > Message: 8 > Date: Wed, 4 Feb 2009 11:05:09 -0800 > From: Michael Collins > Subject: Re: [Freeswitch-users] mod_limit > To: freeswitch-users at lists.freeswitch.org > Message-ID: > <87f2f3b90902041105l50f51f08t230bab8d69eefb4e at mail.gmail.com> > Content-Type: text/plain; charset=ISO-8859-1 > > On Wed, Feb 4, 2009 at 10:54 AM, Chav Paskov wrote: > > Michael Collins wrote: > >> On Wed, Feb 4, 2009 at 9:59 AM, Chav Paskov > wrote: > >> > >>> Hi > , > >>> is it possible to use mod_limit in case if the end point is not > >>> registered / gateway for example/. > >>> > >> > >> Could you add some detail to this question? What are you trying to do? > >> (mod_limit may or may not work, but there might be another solution > >> which is why I am asking.) > >> > >> -MC > >> > >> > >>> Regards > >>> Chav > >>> > >>> _______________________________________________ > >>> Freeswitch-users mailing list > >>> Freeswitch-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> http://www.freeswitch.org > >>> > >>> > >> > >> _______________________________________________ > >> > Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > >> > > i have few gateways under my ACL that are allowed to send calls to FS, > > but i want to be able to enforce "capacity" policy on the > traffic > > coming from any one of them depending on total termination capacity on > > my termination end. > > Let say GW 1 has to be limited to make 10 simultaneous calls while GW2 > > could make up to 30 and so on. > > I'm sure that this is possible. I don't personally have a way to test > all of this but I know that a number of our users are doing things > like this currently. Can you hop on to the IRC channel? #freeswitch on > irc.freenode.net. A lot of people there can help with > this one. > > -MC (IRC: mercutioviz) > > > Regards > > Chav > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > ------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > End of Freeswitch-users Digest, Vol 32, Issue 39 > ************************************************ > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090205/61200d9e/attachment-0001.html From kerrada2003 at yahoo.com Thu Feb 5 11:50:30 2009 From: kerrada2003 at yahoo.com (Ali Al-Rubaie) Date: Thu, 5 Feb 2009 11:50:30 -0800 (PST) Subject: [Freeswitch-users] Transcoding G723 Message-ID: <592485.77688.qm@web33706.mail.mud.yahoo.com> Hi, I need FreeSWITCH to transcode from G711 to G723 but I couldn't do so because it supports G723 only in passthru mode. So, what's the solution? Thanks, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090205/0768e009/attachment.html From kerrada2003 at yahoo.com Thu Feb 5 11:54:07 2009 From: kerrada2003 at yahoo.com (Ali Al-Rubaie) Date: Thu, 5 Feb 2009 11:54:07 -0800 (PST) Subject: [Freeswitch-users] FreeSWITCH calls database Message-ID: <254392.33195.qm@web33701.mail.mud.yahoo.com> Hi, Is there a way to find some calls' statistics in FreeSWITCH, like no. of calls, durations, calls' records..etc. Thanks, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090205/ab07641a/attachment.html From intralanman at freeswitch.org Thu Feb 5 11:58:15 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Thu, 05 Feb 2009 14:58:15 -0500 Subject: [Freeswitch-users] Transcoding G723 In-Reply-To: <592485.77688.qm@web33706.mail.mud.yahoo.com> References: <592485.77688.qm@web33706.mail.mud.yahoo.com> Message-ID: <498B44D7.4050108@freeswitch.org> Ali Al-Rubaie wrote: > So, what's the solution? > don't try to transcode to patent encumbered codecs ;-) -Ray -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090205/ceac53ab/attachment.html From krice at freeswitch.org Thu Feb 5 11:59:41 2009 From: krice at freeswitch.org (Ken Rice) Date: Thu, 05 Feb 2009 13:59:41 -0600 Subject: [Freeswitch-users] Transcoding G723 In-Reply-To: <592485.77688.qm@web33706.mail.mud.yahoo.com> Message-ID: G723 is a patent encumbered codec as is G729. A codec module for trans-coding either of these is not available at this time. G729 may become available at some point in the future, however I don?t see G723 coming available anytime soon do to facts that there is not much of a demand for it coupled with the high costs of licensing the required patents. From: Ali Al-Rubaie Reply-To: Date: Thu, 5 Feb 2009 11:50:30 -0800 (PST) To: Subject: [Freeswitch-users] Transcoding G723 Hi, I need FreeSWITCH to transcode from G711 to G723 but I couldn't do so because it supports G723 only in passthru mode. So, what's the solution? Thanks, _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090205/071b337b/attachment.html From krice at freeswitch.org Thu Feb 5 12:00:37 2009 From: krice at freeswitch.org (Ken Rice) Date: Thu, 05 Feb 2009 14:00:37 -0600 Subject: [Freeswitch-users] FreeSWITCH calls database In-Reply-To: <254392.33195.qm@web33701.mail.mud.yahoo.com> Message-ID: There are several ways to do this... See mod_xml_cdr and mod_cdr_csv K From: Ali Al-Rubaie Reply-To: Date: Thu, 5 Feb 2009 11:54:07 -0800 (PST) To: Subject: [Freeswitch-users] FreeSWITCH calls database Hi, Is there a way to find some calls' statistics in FreeSWITCH, like no. of calls, durations, calls' records..etc. Thanks, _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090205/5827cdb6/attachment.html From brian at freeswitch.org Thu Feb 5 12:03:28 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 5 Feb 2009 14:03:28 -0600 Subject: [Freeswitch-users] Transcoding G723 In-Reply-To: References: Message-ID: <2FEEE7F7-DCE7-4919-A0FD-A0042BBB7EEE@freeswitch.org> This codec patent expires in a few years anyway. /b On Feb 5, 2009, at 1:59 PM, Ken Rice wrote: > I don?t see G723 coming available anytime soon do to facts that > there is not much of a demand for it coupled with the high costs of > licensing the required patents. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090205/302077ad/attachment.html From sicfslist at gmail.com Thu Feb 5 12:26:32 2009 From: sicfslist at gmail.com (Shelby Ramsey) Date: Thu, 5 Feb 2009 14:26:32 -0600 Subject: [Freeswitch-users] Transcoding G723 In-Reply-To: <2FEEE7F7-DCE7-4919-A0FD-A0042BBB7EEE@freeswitch.org> References: <2FEEE7F7-DCE7-4919-A0FD-A0042BBB7EEE@freeswitch.org> Message-ID: <35b355e90902051226g12756d2fsf85fb996ec1ed188@mail.gmail.com> This is a tough deal ... and something that I do think keeps FS out of the same boat as proprietary solutions like Nextone, etc. In the real world (from a service provider view) you have customers who want to send one thing (i.e. g729 and g723 a lot from international carriers) and your vendors who will only accept a limited set (specifically g729 (maybe) and certainly g711 ulaw). So you have to really restrict what people can send you and in some cases it can be a deal killer. I'm seeing more and more wholesale vendors (especially smaller niche guys) getting away from accepting anything other than g711. I would be interested in seeing if there would be a way to have the RTP transverse a media processing blade like the ones offered from Audiocodes etc. Most have some method to tell the device to set up ports and bridge without being involved in the signaling itself. There are a couple of major advantages: -- removing the transcoding from the host to risc based processors -- not worrying about the licensing because it comes with the card and would support all codecs SDR -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090205/da5661de/attachment.html From nik.middleton at noblesolutions.co.uk Thu Feb 5 12:36:33 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Thu, 5 Feb 2009 20:36:33 -0000 Subject: [Freeswitch-users] Dialplan variables Message-ID: Hi Guys, Simple question, tried asking on IRC but no joy, they're too busy slating other systems. I'm trying to dial out via a remote sip gateway via the dial plan This works fine, but I'd like to wild card the extension so it matches on anything starting with a 0 the a number > 0 How do I pass the number dialed using a variable? In asterisk I would put ${EXTEN} Finally I also have the sip gateway registered Mag gateway sip:xxx at hostname.net REGED Is it possible to use the name of this gateway instead of the IP address as in 21X.XXX.XXX/XXX ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090205/997125ce/attachment-0001.html From brian at freeswitch.org Thu Feb 5 12:45:52 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 5 Feb 2009 14:45:52 -0600 Subject: [Freeswitch-users] Transcoding G723 In-Reply-To: <35b355e90902051226g12756d2fsf85fb996ec1ed188@mail.gmail.com> References: <2FEEE7F7-DCE7-4919-A0FD-A0042BBB7EEE@freeswitch.org> <35b355e90902051226g12756d2fsf85fb996ec1ed188@mail.gmail.com> Message-ID: <11C3C72A-247D-4F8C-B106-52D49949D026@freeswitch.org> If its that critical you can do it: http://jira.freeswitch.org/browse/MODCODEC-7 /b On Feb 5, 2009, at 2:26 PM, Shelby Ramsey wrote: > This is a tough deal ... and something that I do think keeps FS out > of the same boat as proprietary solutions like Nextone, etc. In the > real world (from a service provider view) you have customers who > want to send one thing (i.e. g729 and g723 a lot from international > carriers) and your vendors who will only accept a limited set > (specifically g729 (maybe) and certainly g711 ulaw). So you have to > really restrict what people can send you and in some cases it can be > a deal killer. I'm seeing more and more wholesale vendors > (especially smaller niche guys) getting away from accepting anything > other than g711. > > I would be interested in seeing if there would be a way to have the > RTP transverse a media processing blade like the ones offered from > Audiocodes etc. > Most have some method to tell the device to set up ports and bridge > without being involved in the signaling itself. > > There are a couple of major advantages: > -- removing the transcoding from the host to risc based processors > -- not worrying about the licensing because it comes with the card > and would support all codecs > > SDR > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Thu Feb 5 12:45:22 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 5 Feb 2009 14:45:22 -0600 Subject: [Freeswitch-users] Dialplan variables In-Reply-To: References: Message-ID: <4A7EAF44-59DE-4D1F-A52E-A5B37C488BBB@freeswitch.org> Nik, We did answer you twice. ${destination_number}, But you need to not approach this with an Asterisk mindset. Example: In this example the regular expression would match everything start with a 0 and capture the zero plus all digits and put it into $1, Then in the next line you use $1 to pass what the regular expression matched. This concept is a bit different vs Asterisk. /b On Feb 5, 2009, at 2:36 PM, Nik Middleton wrote: > Hi Guys, > > Simple question, tried asking on IRC but no joy, they?re too busy > slating other systems. > > I?m trying to dial out via a remote sip gateway via the dial plan > > > > > > > > This works fine, but I?d like to wild card the extension so it > matches on anything starting with a 0 the a number > 0 > > How do I pass the number dialed using a variable? In asterisk I > would put ${EXTEN} > > Finally I also have the sip gateway registered > > Mag gateway sip:xxx at hostname.net REGED > > Is it possible to use the name of this gateway instead of the IP > address as in 21X.XXX.XXX/XXX ? > > Regards > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090205/d4836e77/attachment.html From brian at freeswitch.org Thu Feb 5 12:48:07 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 5 Feb 2009 14:48:07 -0600 Subject: [Freeswitch-users] Dialplan variables In-Reply-To: References: Message-ID: <98F86FCA-5D19-4F57-A77D-9A4F57632A64@freeswitch.org> Btw I just noticed you're using a gateway... in that case you use sofia/gateway/$gatewayname_here/$1 /b On Feb 5, 2009, at 2:36 PM, Nik Middleton wrote: > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090205/63fa0b0e/attachment.html From nicolas at medularis.com Thu Feb 5 12:49:19 2009 From: nicolas at medularis.com (Nicolas Brenner) Date: Thu, 5 Feb 2009 17:49:19 -0300 Subject: [Freeswitch-users] Transcoding G723 In-Reply-To: <35b355e90902051226g12756d2fsf85fb996ec1ed188@mail.gmail.com> References: <2FEEE7F7-DCE7-4919-A0FD-A0042BBB7EEE@freeswitch.org> <35b355e90902051226g12756d2fsf85fb996ec1ed188@mail.gmail.com> Message-ID: <1b46b4e80902051249x35b6b398k2dccf78a8f3d24e1@mail.gmail.com> I had to go with Asterisk because my VoIP providers only accept G729. I love FS and I have it on standby until either my providers accept other codecs (I'm trying to convince them of using Speex), or FS can transcode G729. Anyway, congratulations to the whole development team, and everybody on this list who help other people get started with FS, this is a really great project/software/platform! On Thu, Feb 5, 2009 at 5:26 PM, Shelby Ramsey wrote: > This is a tough deal ... and something that I do think keeps FS out of the > same boat as proprietary solutions like Nextone, etc. In the real world > (from a service provider view) you have customers who want to send one thing > (i.e. g729 and g723 a lot from international carriers) and your vendors who > will only accept a limited set (specifically g729 (maybe) and certainly g711 > ulaw). So you have to really restrict what people can send you and in some > cases it can be a deal killer. I'm seeing more and more wholesale vendors > (especially smaller niche guys) getting away from accepting anything other > than g711. > I would be interested in seeing if there would be a way to have the RTP > transverse a media processing blade like the ones offered from Audiocodes > etc. > Most have some method to tell the device to set up ports and bridge without > being involved in the signaling itself. > > There are a couple of major advantages: > -- removing the transcoding from the host to risc based processors > -- not worrying about the licensing because it comes with the card and > would support all codecs > SDR > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Nicol?s Brenner From nicolas at medularis.com Thu Feb 5 12:52:03 2009 From: nicolas at medularis.com (Nicolas Brenner) Date: Thu, 5 Feb 2009 17:52:03 -0300 Subject: [Freeswitch-users] Transcoding G723 In-Reply-To: <11C3C72A-247D-4F8C-B106-52D49949D026@freeswitch.org> References: <2FEEE7F7-DCE7-4919-A0FD-A0042BBB7EEE@freeswitch.org> <35b355e90902051226g12756d2fsf85fb996ec1ed188@mail.gmail.com> <11C3C72A-247D-4F8C-B106-52D49949D026@freeswitch.org> Message-ID: <1b46b4e80902051252o1b5a5c75l9f8d242cbaeec552@mail.gmail.com> In my case, I can't use hardware transcoding since I don't have physical access to the servers, I rent them. Hence I need a pure software/IP solution. On Thu, Feb 5, 2009 at 5:45 PM, Brian West wrote: > If its that critical you can do it: http://jira.freeswitch.org/browse/MODCODEC-7 > > /b > > On Feb 5, 2009, at 2:26 PM, Shelby Ramsey wrote: > >> This is a tough deal ... and something that I do think keeps FS out >> of the same boat as proprietary solutions like Nextone, etc. In the >> real world (from a service provider view) you have customers who >> want to send one thing (i.e. g729 and g723 a lot from international >> carriers) and your vendors who will only accept a limited set >> (specifically g729 (maybe) and certainly g711 ulaw). So you have to >> really restrict what people can send you and in some cases it can be >> a deal killer. I'm seeing more and more wholesale vendors >> (especially smaller niche guys) getting away from accepting anything >> other than g711. >> >> I would be interested in seeing if there would be a way to have the >> RTP transverse a media processing blade like the ones offered from >> Audiocodes etc. >> Most have some method to tell the device to set up ports and bridge >> without being involved in the signaling itself. >> >> There are a couple of major advantages: >> -- removing the transcoding from the host to risc based processors >> -- not worrying about the licensing because it comes with the card >> and would support all codecs >> >> SDR >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Nicol?s Brenner From brian at freeswitch.org Thu Feb 5 12:54:24 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 5 Feb 2009 14:54:24 -0600 Subject: [Freeswitch-users] Transcoding G723 In-Reply-To: <1b46b4e80902051252o1b5a5c75l9f8d242cbaeec552@mail.gmail.com> References: <2FEEE7F7-DCE7-4919-A0FD-A0042BBB7EEE@freeswitch.org> <35b355e90902051226g12756d2fsf85fb996ec1ed188@mail.gmail.com> <11C3C72A-247D-4F8C-B106-52D49949D026@freeswitch.org> <1b46b4e80902051252o1b5a5c75l9f8d242cbaeec552@mail.gmail.com> Message-ID: well that'll not scale far :P /b On Feb 5, 2009, at 2:52 PM, Nicolas Brenner wrote: > In my case, I can't use hardware transcoding since I don't have > physical access to the servers, I rent them. Hence I need a pure > software/IP solution. From mike at jerris.com Thu Feb 5 12:54:33 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 5 Feb 2009 15:54:33 -0500 Subject: [Freeswitch-users] spidermonkey problems In-Reply-To: <07F43D7C-3CE2-4A55-819B-E8F1095C5C8B@jerris.com> References: <191c3a030901151548w7504e2a0j5650449e20eff557@mail.gmail.com> <07F43D7C-3CE2-4A55-819B-E8F1095C5C8B@jerris.com> Message-ID: <722A8256-ECD2-4BBA-AEA2-E0B832AB60EA@jerris.com> This should now be fixed in svn trunk. Please re-test this with trunk and confirm that all is working correctly now. Mike On Jan 16, 2009, at 12:03 PM, Michael Jerris wrote: > All long running non js code should be wrapped in the suspend/resume > gc stuff. For example: > > cb_state.ret = BOOLEAN_TO_JSVAL(JS_FALSE); > cb_state.saveDepth = JS_SuspendRequest(cx); > args.input_callback = dtmf_func; > args.buf = bp; > args.buflen = len; > switch_ivr_sleep(jss->session, ms, sync, &args); > JS_ResumeRequest(cx, cb_state.saveDepth); > > I think this is your issue. Can you please file a bug on jira for > this issue (even better with a patch) > > Mike > > > > On Jan 16, 2009, at 5:54 AM, Jonas Gauffin wrote: > >> I've found the problem. one js thread wait in socket.read >> (mod_spidermonkey_socket) on data. >> That caller have hangup, which means that the garbage collector >> waits on it to close. >> >> All new javascript sessions waits in JS_AWAIT_GC_DONE for the >> garbage collector to be done before proceeding (which means that >> all new javascript calls don't do anything after being launched). >> >> My server will not send anything until an agent gets free or the >> session hangs up (detects it through the event socket). And the >> event socket will not send that the session has been hangup until >> the socket have received anything (and the script can exit). So >> it's kind of deadlock between my server and the spidermonkey_socket. >> >> Is it possible to add an option to socket.read to make it abort if >> the session have been closed? I know that I wrote >> mod_spidermonkey_socket from the start, but I can't figure out how >> to do it. >> >> Will new sessions always wait on old ones to be garbage collected >> properly? For instance, what happens if a script have a lenghty >> post process after caller have hang up? >> >> On Fri, Jan 16, 2009 at 9:38 AM, Jonas Gauffin > > wrote: >> I've got a loop, but the first thing checked in each iteration is >> if session.ready() returns false (and in that case exit the loop). >> >> I do create sessions in the script: create, try to originate to a >> destination and then finally bridge together the caller and the new >> session. >> >> I'll try to give you more details during the day. >> >> On Fri, Jan 16, 2009 at 12:48 AM, Anthony Minessale > > wrote: >> do you have any loops in your code that might not check for >> session.ready() in a exit when its not true. >> >> The symptoms you posted would be consistent with held readlocks so >> if you got a gcore (or windows equiv) of the process you might be >> able to see what threads where doing what to hang on to the read >> lock. >> >> also are you creating sessions in the script then executing app >> with them, beware of this because the thread of the script is used >> to execute apps on a session created that way and not the session >> thread. >> >> >> >> >> On Thu, Jan 15, 2009 at 5:20 AM, Jonas Gauffin > > wrote: >> Hello >> >> I got problems with hanging spidermonkey sessions and need some >> advice on how to debug them. >> >> I've made a javascript queue application that uses >> mod_spidermonkey_socket. It works fine for a while, >> but after some calls I noticed that calls didnt get transferred to >> agents. The reason was that earlier >> calls had not been terminated properly. >> >> freeswitch at test1> hupall >> 2009-01-15 12:15:04 [CRIT] switch_core_session.c:147 >> switch_core_session_hupall() Giving up with 8 sessions remaining >> API CALL [hupall()] output: >> +OK hangup all channels with cause MANAGER_REQUEST >> >> >> freeswitch at test1> show calls >> API CALL [show(calls)] output: >> >> 0 total. >> >> >> As you can see, 8 sessions are alive, but none of them are listed >> as calls. What kind of logs should I turn on to see what is >> happening with those sessions? >> >> Thanks, >> Jonas >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:213-799-1400 >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090205/92b66141/attachment.html From sicfslist at gmail.com Thu Feb 5 13:01:17 2009 From: sicfslist at gmail.com (Shelby Ramsey) Date: Thu, 5 Feb 2009 15:01:17 -0600 Subject: [Freeswitch-users] Transcoding G723 In-Reply-To: References: <2FEEE7F7-DCE7-4919-A0FD-A0042BBB7EEE@freeswitch.org> <35b355e90902051226g12756d2fsf85fb996ec1ed188@mail.gmail.com> <11C3C72A-247D-4F8C-B106-52D49949D026@freeswitch.org> <1b46b4e80902051252o1b5a5c75l9f8d242cbaeec552@mail.gmail.com> Message-ID: <35b355e90902051301r4c2c53b6tb958774f4754e60f@mail.gmail.com> > > Brian, > Thanks for the link. Is anyone using this in the real world? I did think it was interesting that the author was from Sangoma ... SDR -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090205/8b71cfee/attachment.html From brian at freeswitch.org Thu Feb 5 13:09:00 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 5 Feb 2009 15:09:00 -0600 Subject: [Freeswitch-users] Transcoding G723 In-Reply-To: <35b355e90902051301r4c2c53b6tb958774f4754e60f@mail.gmail.com> References: <2FEEE7F7-DCE7-4919-A0FD-A0042BBB7EEE@freeswitch.org> <35b355e90902051226g12756d2fsf85fb996ec1ed188@mail.gmail.com> <11C3C72A-247D-4F8C-B106-52D49949D026@freeswitch.org> <1b46b4e80902051252o1b5a5c75l9f8d242cbaeec552@mail.gmail.com> <35b355e90902051301r4c2c53b6tb958774f4754e60f@mail.gmail.com> Message-ID: No clue.. it was just put on jira this past weekend ;) /b On Feb 5, 2009, at 3:01 PM, Shelby Ramsey wrote: > Brian, > > Thanks for the link. Is anyone using this in the real world? I did > think it was interesting that the author was from Sangoma ... > > SDR From e.schmidbauer at gmail.com Thu Feb 5 13:11:50 2009 From: e.schmidbauer at gmail.com (e schmidbauer) Date: Thu, 5 Feb 2009 16:11:50 -0500 Subject: [Freeswitch-users] shoutcast skips In-Reply-To: <2cef777b0902031327v4725f174kcaf976c04926fb21@mail.gmail.com> References: <2cef777b0902030916s77339375pabc3d3c6fd9aec1a@mail.gmail.com> <898222CF-3335-4E2D-AE79-E57CFEE7EFB8@freeswitch.org> <2cef777b0902031226l72f4a4ceia80ec00721456602@mail.gmail.com> <574748E6-1EF0-4F5A-A1D6-338BAA711040@freeswitch.org> <2cef777b0902031259t6210b7b2w9873ced0806b98b3@mail.gmail.com> <81F41092-76A7-4DCD-8274-9584177F8752@freeswitch.org> <2cef777b0902031327v4725f174kcaf976c04926fb21@mail.gmail.com> Message-ID: <2cef777b0902051311q3e404a43q2c59fe971169ac8c@mail.gmail.com> Hey just wanted to report back on this issue. I switched OS's from Ubuntu 8.10 to Centos5.2 x64.and issue seems to be resolved. Thanks for the help on this issue. Regards. Emmanuel Schmidbauer On Tue, Feb 3, 2009 at 4:27 PM, e schmidbauer wrote: > We are attempting distributed radio. We plan on having the hosts of the > shows join the conference using CELT. But callers to the show would be > joining using regular phones therefore using lower end codecs. I will be in > the IRC shortly. > > On Tue, Feb 3, 2009 at 4:21 PM, Brian West wrote: > >> You're doing distributed radio right? So callers are calling in with CELT >> from all over the place? Can you contact us on IRC because we are very >> interested in debugging this issue. >> You can get us on IRC #freeswitch on irc.freenode.net >> >> Thanks, >> >> /b >> >> On Feb 3, 2009, at 2:59 PM, e schmidbauer wrote: >> >> FreeSWITCH Version 1.0.trunk (11567) >> check out these sample recordings >> http://bwrl.org/recordings/2009-01-31-12-07-49.mp3 >> http://bwrl.org/recordings/2009-01-31-12-07-49.wav >> http://bwrl.org/recordings/test2.mp3 >> http://bwrl.org/recordings/test2.wav >> >> the conferences were recorded as wav files, i then converted them to mp3, >> both sound the same to me >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090205/55872948/attachment-0001.html From nicolas at medularis.com Thu Feb 5 13:14:40 2009 From: nicolas at medularis.com (Nicolas Brenner) Date: Thu, 5 Feb 2009 18:14:40 -0300 Subject: [Freeswitch-users] Transcoding G723 In-Reply-To: References: <2FEEE7F7-DCE7-4919-A0FD-A0042BBB7EEE@freeswitch.org> <35b355e90902051226g12756d2fsf85fb996ec1ed188@mail.gmail.com> <11C3C72A-247D-4F8C-B106-52D49949D026@freeswitch.org> <1b46b4e80902051252o1b5a5c75l9f8d242cbaeec552@mail.gmail.com> Message-ID: <1b46b4e80902051314k238d0fb3x6ee41fa8daab1d48@mail.gmail.com> Tell that to the Amazon S3 and Amazon EC2 people ;) On Thu, Feb 5, 2009 at 5:54 PM, Brian West wrote: > well that'll not scale far :P > > /b > > On Feb 5, 2009, at 2:52 PM, Nicolas Brenner wrote: > >> In my case, I can't use hardware transcoding since I don't have >> physical access to the servers, I rent them. Hence I need a pure >> software/IP solution. > > -- Nicol?s Brenner From nik.middleton at noblesolutions.co.uk Thu Feb 5 13:14:39 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Thu, 5 Feb 2009 21:14:39 -0000 Subject: [Freeswitch-users] Dialplan variables In-Reply-To: <98F86FCA-5D19-4F57-A77D-9A4F57632A64@freeswitch.org> References: <98F86FCA-5D19-4F57-A77D-9A4F57632A64@freeswitch.org> Message-ID: That didn't work, until I removed the $ in front of the gateway name as in sofia/gateway/gatewayname_here/$1 Why is that, surely it's a variable? Regards, ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: 05 February 2009 20:48 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Dialplan variables Btw I just noticed you're using a gateway... in that case you use sofia/gateway/$gatewayname_here/$1 /b On Feb 5, 2009, at 2:36 PM, Nik Middleton wrote: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090205/870bf357/attachment.html From brian at freeswitch.org Thu Feb 5 13:20:27 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 5 Feb 2009 15:20:27 -0600 Subject: [Freeswitch-users] Dialplan variables In-Reply-To: References: <98F86FCA-5D19-4F57-A77D-9A4F57632A64@freeswitch.org> Message-ID: <081F27CA-12D7-4047-8011-AED40397500A@freeswitch.org> I mean for you to replace it with your gateway name. In your case its Mag I think? /b On Feb 5, 2009, at 3:14 PM, Nik Middleton wrote: > That didn?t work, until I removed the $ in front of the gateway name > as in > > sofia/gateway/gatewayname_here/$1 > > Why is that, surely it?s a variable? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090205/a81465d7/attachment.html From brian at freeswitch.org Thu Feb 5 13:20:47 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 5 Feb 2009 15:20:47 -0600 Subject: [Freeswitch-users] shoutcast skips In-Reply-To: <2cef777b0902051311q3e404a43q2c59fe971169ac8c@mail.gmail.com> References: <2cef777b0902030916s77339375pabc3d3c6fd9aec1a@mail.gmail.com> <898222CF-3335-4E2D-AE79-E57CFEE7EFB8@freeswitch.org> <2cef777b0902031226l72f4a4ceia80ec00721456602@mail.gmail.com> <574748E6-1EF0-4F5A-A1D6-338BAA711040@freeswitch.org> <2cef777b0902031259t6210b7b2w9873ced0806b98b3@mail.gmail.com> <81F41092-76A7-4DCD-8274-9584177F8752@freeswitch.org> <2cef777b0902031327v4725f174kcaf976c04926fb21@mail.gmail.com> <2cef777b0902051311q3e404a43q2c59fe971169ac8c@mail.gmail.com> Message-ID: <06712653-71EF-4C87-AA2C-CE3825F00FA3@freeswitch.org> Anyway we can look closer at the ubuntu issue also? /b On Feb 5, 2009, at 3:11 PM, e schmidbauer wrote: > Hey just wanted to report back on this issue. I switched OS's from > Ubuntu 8.10 to Centos5.2 x64.and issue seems to be resolved. Thanks > for the help on this issue. Regards. > Emmanuel Schmidbauer From nik.middleton at noblesolutions.co.uk Thu Feb 5 14:19:19 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Thu, 5 Feb 2009 22:19:19 -0000 Subject: [Freeswitch-users] Caller ID not being passed Message-ID: Try as I might, I cannot seem to get caller ID passed to the external sip gateway This GW happily processes caller id from Asterisk If tried adding param name="caller-id-in-from" value="true" in gw definition, and even in the dial plan to no avail Can anyone shed some light on this? Regards, This is a sip debug sent 1382 bytes to udp/[XXX.XXX.XXX.XXX]:5060 at 21:45:17.404609: INVITE sip:07539000000 at mygw.net SIP/2.0 Via: SIP/2.0/UDP 87.238.75.206:5080;rport;branch=z9hG4bKm2UveeUvtNc5D Max-Forwards: 69 From: "Extension 1000" ;tag=j6m5U7g4F3XNa To: Call-ID: 1e8c5d1d-6e71-122c-fdba-001a4b0a67ca CSeq: 110816823 INVITE Contact: Expires: 600 User-Agent: FreeSWITCH-mod_sofia/1.0.2-exported Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: timer, precondition, path, replaces Allow-Events: talk, refer Proxy-Authorization: Digest username="me", realm="mygw.net", nonce="498b5f19b089db1bf9d13b2c83f45407048ede9b", algorithm=MD5, uri="sip:07539600000 at mygw.net", response="03dbbeef1387a11c7100965dcfd01052" Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 311 P-Key-Flags: keys="3" -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090205/0ff99ee7/attachment.html From anthony.minessale at gmail.com Thu Feb 5 14:20:18 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 5 Feb 2009 16:20:18 -0600 Subject: [Freeswitch-users] shoutcast skips In-Reply-To: <06712653-71EF-4C87-AA2C-CE3825F00FA3@freeswitch.org> References: <2cef777b0902030916s77339375pabc3d3c6fd9aec1a@mail.gmail.com> <898222CF-3335-4E2D-AE79-E57CFEE7EFB8@freeswitch.org> <2cef777b0902031226l72f4a4ceia80ec00721456602@mail.gmail.com> <574748E6-1EF0-4F5A-A1D6-338BAA711040@freeswitch.org> <2cef777b0902031259t6210b7b2w9873ced0806b98b3@mail.gmail.com> <81F41092-76A7-4DCD-8274-9584177F8752@freeswitch.org> <2cef777b0902031327v4725f174kcaf976c04926fb21@mail.gmail.com> <2cef777b0902051311q3e404a43q2c59fe971169ac8c@mail.gmail.com> <06712653-71EF-4C87-AA2C-CE3825F00FA3@freeswitch.org> Message-ID: <191c3a030902051420k5606076fub1a2c313dcb84693@mail.gmail.com> I changed the code in tree that probably fixed it in both cases. It should be good now. On Thu, Feb 5, 2009 at 3:20 PM, Brian West wrote: > Anyway we can look closer at the ubuntu issue also? > > /b > > On Feb 5, 2009, at 3:11 PM, e schmidbauer wrote: > > > Hey just wanted to report back on this issue. I switched OS's from > > Ubuntu 8.10 to Centos5.2 x64.and issue seems to be resolved. Thanks > > for the help on this issue. Regards. > > Emmanuel Schmidbauer > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090205/673ad786/attachment-0001.html From brian at freeswitch.org Thu Feb 5 14:29:24 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 5 Feb 2009 16:29:24 -0600 Subject: [Freeswitch-users] Caller ID not being passed In-Reply-To: References: Message-ID: Try application export /b On Feb 5, 2009, at 4:19 PM, Nik Middleton wrote: > Try as I might, I cannot seem to get caller ID passed to the > external sip gateway > > This GW happily processes caller id from Asterisk > > If tried adding param name="caller-id-in-from" value="true" in gw > definition, and even > > data="effective_caller_id_number=07539600000"/> in the dial plan to > no avail > > Can anyone shed some light on this? > > Regards, > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090205/65778ccf/attachment.html From nik.middleton at noblesolutions.co.uk Thu Feb 5 14:43:53 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Thu, 5 Feb 2009 22:43:53 -0000 Subject: [Freeswitch-users] Caller ID not being passed In-Reply-To: References: Message-ID: No good, I tried But surely, If I have the proper values in the sip phones xml files, these should be passed to the GW should they not? Regards ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: 05 February 2009 22:29 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Caller ID not being passed Try application export /b On Feb 5, 2009, at 4:19 PM, Nik Middleton wrote: Try as I might, I cannot seem to get caller ID passed to the external sip gateway This GW happily processes caller id from Asterisk If tried adding param name="caller-id-in-from" value="true" in gw definition, and even in the dial plan to no avail Can anyone shed some light on this? Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090205/076d9ec5/attachment.html From brian at freeswitch.org Thu Feb 5 14:50:18 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 5 Feb 2009 16:50:18 -0600 Subject: [Freeswitch-users] Caller ID not being passed In-Reply-To: References: Message-ID: Nope still doing it wrong. Try this: use export instead of set. > data="effective_caller_id_number=07539600000"/> -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090205/8e373dd3/attachment.html From brian at freeswitch.org Thu Feb 5 14:52:24 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 5 Feb 2009 16:52:24 -0600 Subject: [Freeswitch-users] Caller ID not being passed In-Reply-To: References: Message-ID: <356C95D2-B637-47F7-91AA-BD22C83BEC2F@freeswitch.org> Nik, Ignore me... set should have worked... You're using caller-id-in-from let me look closer at this. /b On Feb 5, 2009, at 4:43 PM, Nik Middleton wrote: > No good, I tried > > data="effective_caller_id_number=07539600000"/> > > > > But surely, If I have the proper values in the sip phones xml files, > these should be passed to the GW should they not? > > Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090205/d1ac086c/attachment-0001.html From brian at freeswitch.org Thu Feb 5 14:53:36 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 5 Feb 2009 16:53:36 -0600 Subject: [Freeswitch-users] Caller ID not being passed In-Reply-To: References: Message-ID: <2294453B-8059-4AB9-8566-44E22B859C9A@freeswitch.org> Nik, While I'm looking at this can you post your full gateway and dialplan for us to see? /b On Feb 5, 2009, at 4:43 PM, Nik Middleton wrote: > No good, I tried > > data="effective_caller_id_number=07539600000"/> > > > > But surely, If I have the proper values in the sip phones xml files, > these should be passed to the GW should they not? > > Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090205/f8822068/attachment.html From nik.middleton at noblesolutions.co.uk Thu Feb 5 15:12:18 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Thu, 5 Feb 2009 23:12:18 -0000 Subject: [Freeswitch-users] Caller ID not being passed In-Reply-To: <2294453B-8059-4AB9-8566-44E22B859C9A@freeswitch.org> References: <2294453B-8059-4AB9-8566-44E22B859C9A@freeswitch.org> Message-ID: Dial plan is as per default setup with the addition of the following. To be honest, and I'm no SIP guru, I can't see the caller-id being set in the sip headers Mag.xml ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: 05 February 2009 22:54 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Caller ID not being passed Nik, While I'm looking at this can you post your full gateway and dialplan for us to see? /b On Feb 5, 2009, at 4:43 PM, Nik Middleton wrote: No good, I tried But surely, If I have the proper values in the sip phones xml files, these should be passed to the GW should they not? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090205/cc69fe92/attachment.html From brian at freeswitch.org Thu Feb 5 15:20:35 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 5 Feb 2009 17:20:35 -0600 Subject: [Freeswitch-users] Caller ID not being passed In-Reply-To: References: <2294453B-8059-4AB9-8566-44E22B859C9A@freeswitch.org> Message-ID: <4726C6D0-897E-46D9-91F6-6D2EF646344F@freeswitch.org> I notice you're using 1.0.2 any way you can test this with 1.0.3 RC1 tarball? /b On Feb 5, 2009, at 5:12 PM, Nik Middleton wrote: > Dial plan is as per default setup with the addition of the > following. To be honest, and I?m no SIP guru, I can?t see the caller- > id being set in the sip headers > > > > > data="effective_caller_id_number=0------00000000006"/> > data="effective_caller_id_number"/> > > > > > > Mag.xml > > > > > > > > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090205/e1250d97/attachment-0001.html From nik.middleton at noblesolutions.co.uk Thu Feb 5 15:26:27 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Thu, 5 Feb 2009 23:26:27 -0000 Subject: [Freeswitch-users] Caller ID not being passed In-Reply-To: <4726C6D0-897E-46D9-91F6-6D2EF646344F@freeswitch.org> References: <2294453B-8059-4AB9-8566-44E22B859C9A@freeswitch.org> <4726C6D0-897E-46D9-91F6-6D2EF646344F@freeswitch.org> Message-ID: Yes, I'll report back tomorrow, Regards ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: 05 February 2009 23:21 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Caller ID not being passed I notice you're using 1.0.2 any way you can test this with 1.0.3 RC1 tarball? /b On Feb 5, 2009, at 5:12 PM, Nik Middleton wrote: Dial plan is as per default setup with the addition of the following. To be honest, and I'm no SIP guru, I can't see the caller-id being set in the sip headers Mag.xml -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090205/2b86ef00/attachment.html From steveu at coppice.org Thu Feb 5 15:32:01 2009 From: steveu at coppice.org (Steve Underwood) Date: Fri, 06 Feb 2009 07:32:01 +0800 Subject: [Freeswitch-users] Transcoding G723 In-Reply-To: References: <2FEEE7F7-DCE7-4919-A0FD-A0042BBB7EEE@freeswitch.org> <35b355e90902051226g12756d2fsf85fb996ec1ed188@mail.gmail.com> <11C3C72A-247D-4F8C-B106-52D49949D026@freeswitch.org> <1b46b4e80902051252o1b5a5c75l9f8d242cbaeec552@mail.gmail.com> Message-ID: <498B76F1.5070406@coppice.org> Brian West wrote: > well that'll not scale far :P > That transcoding card does 120 channels. A modern quad core CPU with a well implemented codec can do several hundred. A dual quad core chassis can do twice as much. Which one has a scaling problem? Steve > /b > > On Feb 5, 2009, at 2:52 PM, Nicolas Brenner wrote: > > >> In my case, I can't use hardware transcoding since I don't have >> physical access to the servers, I rent them. Hence I need a pure >> software/IP solution. >> From brian at freeswitch.org Thu Feb 5 15:37:24 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 5 Feb 2009 17:37:24 -0600 Subject: [Freeswitch-users] Transcoding G723 In-Reply-To: <498B76F1.5070406@coppice.org> References: <2FEEE7F7-DCE7-4919-A0FD-A0042BBB7EEE@freeswitch.org> <35b355e90902051226g12756d2fsf85fb996ec1ed188@mail.gmail.com> <11C3C72A-247D-4F8C-B106-52D49949D026@freeswitch.org> <1b46b4e80902051252o1b5a5c75l9f8d242cbaeec552@mail.gmail.com> <498B76F1.5070406@coppice.org> Message-ID: <7A658893-9EA4-4CD9-9AB6-18067FF0E445@freeswitch.org> The hardware in this case... which is why I said it wouldn't scale far :P /b On Feb 5, 2009, at 5:32 PM, Steve Underwood wrote: > That transcoding card does 120 channels. A modern quad core CPU with a > well implemented codec can do several hundred. A dual quad core > chassis > can do twice as much. Which one has a scaling problem? > > Steve From moises.silva at gmail.com Thu Feb 5 21:17:33 2009 From: moises.silva at gmail.com (Moises Silva) Date: Thu, 5 Feb 2009 23:17:33 -0600 Subject: [Freeswitch-users] Transcoding G723 In-Reply-To: <35b355e90902051301r4c2c53b6tb958774f4754e60f@mail.gmail.com> References: <2FEEE7F7-DCE7-4919-A0FD-A0042BBB7EEE@freeswitch.org> <35b355e90902051226g12756d2fsf85fb996ec1ed188@mail.gmail.com> <11C3C72A-247D-4F8C-B106-52D49949D026@freeswitch.org> <1b46b4e80902051252o1b5a5c75l9f8d242cbaeec552@mail.gmail.com> <35b355e90902051301r4c2c53b6tb958774f4754e60f@mail.gmail.com> Message-ID: At least 1 company is using it in their FS gateway for a call center of around 125 positions with their this scenario: Asterisk servers <---- IAX G711 ----> FS Gateway <--- SIP G729 ---> SIP Provider The G723 has only been tested in my laptop with an IAX connection to the FS server though. Any testing is certainly appreciated to squeeze bugs out. Moy On Thu, Feb 5, 2009 at 3:01 PM, Shelby Ramsey wrote: >> Brian, > > Thanks for the link. Is anyone using this in the real world? I did think > it was interesting that the author was from Sangoma ... > SDR > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- "I do not agree with what you have to say, but I'll defend to the death your right to say it." Voltaire From pmhshz at gmail.com Fri Feb 6 00:30:56 2009 From: pmhshz at gmail.com (shehzad p) Date: Fri, 6 Feb 2009 00:30:56 -0800 (PST) Subject: [Freeswitch-users] Freeswitch freezes on increasing call traffic In-Reply-To: <498B01CF.6080902@gmail.com> References: <21701744.post@talk.nabble.com> <191c3a030901280601t3e185ec2m76295110e4888837@mail.gmail.com> <21729863.post@talk.nabble.com> <191c3a030901290831y2ff329b3maee6c1aafa3c68e2@mail.gmail.com> <21745312.post@talk.nabble.com> <21749375.post@talk.nabble.com> <191c3a030901300818i5d85a7aawa0c53b4e54892f40@mail.gmail.com> <21761523.post@talk.nabble.com> <191c3a030901310850s553f8c11oc4e559a710def6d2@mail.gmail.com> <21825226.post@talk.nabble.com> <191c3a030902040551w5d867deta55c98bd3a3c03f8@mail.gmail.com> <21847332.post@talk.nabble.com> <4C2A4389-BEF6-454B-8B5F-FDCCC76E174A@freeswitch.org> <21848148.post@talk.nabble.com> <21852304.post@talk.nabble.com> <498B01CF.6080902@gmail.com> Message-ID: <21868458.post@talk.nabble.com> Hi all, I have opened JIRA for the same. http://jira.freeswitch.org/browse/FSCORE-285 One system is Fedora, and another one is Ubuntu. Although fs 1.0.1 was also get crashed in Ubuntu many times. Now 1.0.3.RC1 is loaded on Ubuntu, so this happen again with my Ubuntu I will surely post it. And I am now going to test latest trunk version and post back if any thing found... thanks, msp paul.degt wrote: > > Look like you use Fedora. I had a lot of issues with using Fedora as > production or load test system, in my opinion it's more like work in > progress than a production ready stable linux. If you cannot buy RHEL or > SLES use Centos. > > shehzad p wrote: >> Hi Brian, >> >> As it can be seen from the system information, there require any change >> in >> system or any suggestion... >> >> out put of uname -a is : >> Linux localhost.localdomain 2.6.23.1-42.fc8 #1 SMP Tue Oct 30 13:55:12 >> EDT >> 2007 i686 i686 i386 GNU/Linux >> >> >> Thanks, >> msp >> >> >> shehzad p wrote: >> >>> HI Brian, >>> >>> Output of ulimit -a and /proc/cpuinfo is attached... >>> http://www.nabble.com/file/p21848148/12_ulimit_and_cpuinfo.log >>> 12_ulimit_and_cpuinfo.log >>> >>> BUT...................... >>> I am running the freeswitch using below command (So ulimit set according >>> to Anthony's previous post): >>> =================================================================================== >>> ulimit -d unlimited; ulimit -f unlimited; ulimit -i unlimited; ulimit -n >>> 999999; ulimit -q unlimited ; ulimit -u unlimited; ulimit -v unlimited ; >>> ulimit -x unlimited; ulimit -s 244; ulimit -l unlimited; freeswitch >>> =================================================================================== >>> >>> Thanks >>> msp >>> >>> >>> >>> Brian West-3 wrote: >>> >>>> Can you give me the output of uname -a and the contents of /proc/ >>>> cpuinfo? Not sure I asked for this info already or not. >>>> >>>> Thanks, >>>> Brian >>>> >>>> On Feb 5, 2009, at 2:42 AM, shehzad p wrote: >>>> >>>> >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/Freeswitch-freezes-on-increasing-call-traffic-tp21701744p21868458.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From ahgindia308 at gmail.com Fri Feb 6 00:57:07 2009 From: ahgindia308 at gmail.com (Ankit Gandhi) Date: Fri, 6 Feb 2009 00:57:07 -0800 (PST) Subject: [Freeswitch-users] Freeswitch hangup cause code issue Message-ID: <21868781.post@talk.nabble.com> Hello, I want my caller to detect hangup cause as 34 so that he can try next provider according to lcr. Here is my setup. Caller -> switch (fs) -> Terminator. Now when terminator sends "503 Service Unavailable", I want to override this cause, so that the caller gets hangup case 34. (According to the terminator he is sending hangup cause 34 from his side, but in freeswitch we are getting hangup cause 41 for that call and the same hangup cause on caller side). When I tried asterisk as caller, I get hangup cause 34 in that case. But when I tried freeswitch as caller, then we are getting hangup cause 41, the same as we are getting in switch (fs). >From the switch, I tried one of this condition through javascript to override the hangup cause before sending to caller: -> session.execute("respond","503"); -> session.execute("hangup","NORMAL_CIRCUIT_CONGESTION"); -> session.execute("hangup","34"); -> session.hangup(34); In all the above cases, asterisk properly detects the hangup cause 34, but freeswitch does not detect that. It detects the same hangup cause 41 for the call. Other callers also get the same hangup cause 41 for such calls. How can I override this cause, so that the caller gets hangup cause 34 in such cases? Here is the sip trace, on the caller side returned through switch (fs). ss = switch cc = caller -------------------------------------------------------------------- U ss.ss.ss.ss:5060 -> cc.cc.cc.cc:5080 SIP/2.0 503 Service Unavailable. Via: SIP/2.0/UDP cc.cc.cc.cc:5080;rport=5080;branch=z9hG4bK8HvDU4Z88KU5m;received=122.169.29.122. From: "654321" ;tag=8p7aF18aN0mSj. To: ;tag=Nj8ypjDyvKUXe. Call-ID: fd1ab774-6ec5-122c-9fac-001cc086141d. CSeq: 110835048 INVITE. User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-hacked. Accept: application/sdp. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. Supported: timer, precondition, path, replaces. Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Reason: Q.850;cause=34;text="NORMAL_CIRCUIT_CONGESTION". Content-Length: 0. ----------------------------------------------------------------------- -- View this message in context: http://www.nabble.com/Freeswitch-hangup-cause-code-issue-tp21868781p21868781.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From helmut.kuper at ewetel.de Fri Feb 6 04:32:03 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Fri, 06 Feb 2009 13:32:03 +0100 Subject: [Freeswitch-users] mod_openzap stops working after some calls Message-ID: <498C2DC3.40701@ewetel.de> Hello, since yesterday I do a real life test with FS in a 40 sip extension environment with TDM connection via wanpipe/sangoma A104d. I detected two times the problem, that openzap stops working, while SIP calls worked. Only restarting helped. Maybe reloading of mod_openzap helps as well, but I didn't tested that, yet. I filtered FS logfile for hints of that problem and I found a growing number of this line: 2009-02-06 11:09:51 [INFO] ozmod_isdn.c:706 zap_isdn_931_34() Duplicate SETUP message(?) for Channel 1:21 ~ 1:21 in state DOWN [ignoring] Befor restart I saw such a line for *each* TDM channel and no one was able to dial out. mod_openzap handled around 40 outgoing calls until last restart . Since last restart FS runs good for 2 hours now without any duplicate SETUP in log ... Has anybody similar problems? regards Helmut From pmhshz at gmail.com Fri Feb 6 06:51:31 2009 From: pmhshz at gmail.com (shehzad p) Date: Fri, 6 Feb 2009 06:51:31 -0800 (PST) Subject: [Freeswitch-users] streamFile and read doesn't get DTMF while used with originate Message-ID: <21873924.post@talk.nabble.com> Hi all My set up is : When i originate the call from CLI, using originate command, It comes in a dialplan and from there it goes to javascript for handling simple IVR. In IVR I tested both streamFile and dialplan read application to get the DTMF from user, but it was not working, it just play the file and does nothing (streamFile doesn't call the onInput function at all).. This setup was working before, when I call a javascript from CLI, and in JS I create a session to originate the call. Is there any settings missing for my first setup, so that I can get DTMF properly... Thank.. msp -- View this message in context: http://www.nabble.com/streamFile-and-read-doesn%27t-get-DTMF-while-used-with-originate-tp21873924p21873924.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From anthony.minessale at gmail.com Fri Feb 6 07:09:08 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 6 Feb 2009 09:09:08 -0600 Subject: [Freeswitch-users] Freeswitch hangup cause code issue In-Reply-To: <21868781.post@talk.nabble.com> References: <21868781.post@talk.nabble.com> Message-ID: <191c3a030902060709m786873f9i7c31f5c0d57dce03@mail.gmail.com> you would need to provide the console output of FreeSWITCH of the entire call. with TPORT_LOG=1 env var set and console loglevel debug (press f8) On Fri, Feb 6, 2009 at 2:57 AM, Ankit Gandhi wrote: > > Hello, > I want my caller to detect hangup cause as 34 so that he can try next > provider according to lcr. > Here is my setup. > Caller -> switch (fs) -> Terminator. > Now when terminator sends "503 Service Unavailable", I want to override > this > cause, so that the caller gets hangup case 34. (According to the terminator > he is sending hangup cause 34 from his side, but in freeswitch we are > getting hangup cause 41 for that call and the same hangup cause on caller > side). > When I tried asterisk as caller, I get hangup cause 34 in that case. But > when I tried freeswitch as caller, then we are getting hangup cause 41, the > same as we are getting in switch (fs). > >From the switch, I tried one of this condition through javascript to > override the hangup cause before sending to caller: > -> session.execute("respond","503"); > -> session.execute("hangup","NORMAL_CIRCUIT_CONGESTION"); > -> session.execute("hangup","34"); > -> session.hangup(34); > In all the above cases, asterisk properly detects the hangup cause 34, but > freeswitch does not detect that. It detects the same hangup cause 41 for > the > call. Other callers also get the same hangup cause 41 for such calls. > How can I override this cause, so that the caller gets hangup cause 34 in > such cases? > > Here is the sip trace, on the caller side returned through switch (fs). > ss = switch > cc = caller > -------------------------------------------------------------------- > U ss.ss.ss.ss:5060 -> cc.cc.cc.cc:5080 > SIP/2.0 503 Service Unavailable. > Via: SIP/2.0/UDP > cc.cc.cc.cc:5080 > ;rport=5080;branch=z9hG4bK8HvDU4Z88KU5m;received=122.169.29.122. > From: "654321" > >;tag=8p7aF18aN0mSj. > To: ;tag=Nj8ypjDyvKUXe. > Call-ID: fd1ab774-6ec5-122c-9fac-001cc086141d. > CSeq: 110835048 INVITE. > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-hacked. > Accept: application/sdp. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, > REFER, UPDATE, REGISTER, INFO, PUBLISH. > Supported: timer, precondition, path, replaces. > Allow-Events: talk, presence, dialog, call-info, sla, > include-session-description, presence.winfo, message-summary, refer. > Reason: Q.850;cause=34;text="NORMAL_CIRCUIT_CONGESTION". > Content-Length: 0. > ----------------------------------------------------------------------- > > > -- > View this message in context: > http://www.nabble.com/Freeswitch-hangup-cause-code-issue-tp21868781p21868781.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090206/7cf508d5/attachment.html From sicfslist at gmail.com Fri Feb 6 07:16:11 2009 From: sicfslist at gmail.com (Shelby Ramsey) Date: Fri, 6 Feb 2009 09:16:11 -0600 Subject: [Freeswitch-users] Transcoding G723 In-Reply-To: References: <2FEEE7F7-DCE7-4919-A0FD-A0042BBB7EEE@freeswitch.org> <35b355e90902051226g12756d2fsf85fb996ec1ed188@mail.gmail.com> <11C3C72A-247D-4F8C-B106-52D49949D026@freeswitch.org> <1b46b4e80902051252o1b5a5c75l9f8d242cbaeec552@mail.gmail.com> <35b355e90902051301r4c2c53b6tb958774f4754e60f@mail.gmail.com> Message-ID: <35b355e90902060716v2dff2586t253a69c97ab1adac@mail.gmail.com> Thanks Moises. It looks like good work. When is Sangoma coming out with a similar product ... Doug told me it was in the works, then not in the works, then back in the works ... The problem is this particular card is PCI only and it will only do 120 channels .... Thanks! SDR -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090206/bd1c27fc/attachment.html From anthony.minessale at gmail.com Fri Feb 6 07:18:18 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 6 Feb 2009 09:18:18 -0600 Subject: [Freeswitch-users] Freeswitch freezes on increasing call traffic In-Reply-To: <21868458.post@talk.nabble.com> References: <21701744.post@talk.nabble.com> <191c3a030901310850s553f8c11oc4e559a710def6d2@mail.gmail.com> <21825226.post@talk.nabble.com> <191c3a030902040551w5d867deta55c98bd3a3c03f8@mail.gmail.com> <21847332.post@talk.nabble.com> <4C2A4389-BEF6-454B-8B5F-FDCCC76E174A@freeswitch.org> <21848148.post@talk.nabble.com> <21852304.post@talk.nabble.com> <498B01CF.6080902@gmail.com> <21868458.post@talk.nabble.com> Message-ID: <191c3a030902060718m2d58cab9p757c0e95d5f8c559@mail.gmail.com> I am providing you with free software and free support. I am willing to investigate you problem for you but you must be more cooperative. I took the time to ask you several questions yesterday and you have sent 4 emails since and not addressed it. Please move any further correspondence on this issue to your jira ticket. If you do not have the necessary skill to build your own ODBC and driver as the email requests, will you please try CentOS 64 bit linux with only SVN Trunk which is a platform we trust the unixODBC and its depends. Please E-mail me privately your latest JS code. Also please try to produce a minimal script that will reproduce your issue that we can reproduce. If you continue to ignore my requests I will have no choice but to close your issue. Transcript of ignored e-mail ---------------------------------- First of all please stop using the mailing list as a bug tracker. All issues should be put into jira and managed with that. Secondly, Didn't I ask you multiple times to stop using release snapshots and please use the SVN trunk? I don't understand why you keep ignoring me and using everything but what I asked. I am not telling you to use SVN because I think it will be fixed it's so we are on the development copy of the code to get the proper line numbers etc. If you look at your 2 bt you posted, the line numbers are different on each one. What are you using on the other side of ODBC? as you can see in your bt, the call goes into ODBC then into several libs with no symbols and crashes on free. This can be a sign of corrupt memory, running out of memory or an issue in either ODBC or the database specific lib. What distro is it? What ODBC version? unixODBC? version xxx? What database driver version xxx? Is it mysl not using the proper reentrant version of the plugin? Sometimes packaged libs have bugs in them which fall out of our control. Can you build unixODBC and the plugins yourself with debug symbols so we can see if that is the cause or at the very least then we can see the debug info in the bt. please make sure you address *all* my questions in your jira report. Starting with using svn trunk, *hint* type "make current" from your rc1 distro. On Fri, Feb 6, 2009 at 2:30 AM, shehzad p wrote: > > Hi all, > > I have opened JIRA for the same. > http://jira.freeswitch.org/browse/FSCORE-285 > > One system is Fedora, and another one is Ubuntu. Although fs 1.0.1 was also > get crashed in Ubuntu many times. > Now 1.0.3.RC1 is loaded on Ubuntu, so this happen again with my Ubuntu I > will surely post it. > > And I am now going to test latest trunk version and post back if any thing > found... > > thanks, > msp > > paul.degt wrote: > > > > Look like you use Fedora. I had a lot of issues with using Fedora as > > production or load test system, in my opinion it's more like work in > > progress than a production ready stable linux. If you cannot buy RHEL or > > SLES use Centos. > > > > shehzad p wrote: > >> Hi Brian, > >> > >> As it can be seen from the system information, there require any change > >> in > >> system or any suggestion... > >> > >> out put of uname -a is : > >> Linux localhost.localdomain 2.6.23.1-42.fc8 #1 SMP Tue Oct 30 13:55:12 > >> EDT > >> 2007 i686 i686 i386 GNU/Linux > >> > >> > >> Thanks, > >> msp > >> > >> > >> shehzad p wrote: > >> > >>> HI Brian, > >>> > >>> Output of ulimit -a and /proc/cpuinfo is attached... > >>> http://www.nabble.com/file/p21848148/12_ulimit_and_cpuinfo.log > >>> 12_ulimit_and_cpuinfo.log > >>> > >>> BUT...................... > >>> I am running the freeswitch using below command (So ulimit set > according > >>> to Anthony's previous post): > >>> > =================================================================================== > >>> ulimit -d unlimited; ulimit -f unlimited; ulimit -i unlimited; ulimit > -n > >>> 999999; ulimit -q unlimited ; ulimit -u unlimited; ulimit -v unlimited > ; > >>> ulimit -x unlimited; ulimit -s 244; ulimit -l unlimited; freeswitch > >>> > =================================================================================== > >>> > >>> Thanks > >>> msp > >>> > >>> > >>> > >>> Brian West-3 wrote: > >>> > >>>> Can you give me the output of uname -a and the contents of /proc/ > >>>> cpuinfo? Not sure I asked for this info already or not. > >>>> > >>>> Thanks, > >>>> Brian > >>>> > >>>> On Feb 5, 2009, at 2:42 AM, shehzad p wrote: > >>>> > >>>> > >> > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > View this message in context: > http://www.nabble.com/Freeswitch-freezes-on-increasing-call-traffic-tp21701744p21868458.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090206/75346abc/attachment-0001.html From nik.middleton at noblesolutions.co.uk Fri Feb 6 07:51:37 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Fri, 6 Feb 2009 15:51:37 -0000 Subject: [Freeswitch-users] Caller ID not being passed In-Reply-To: References: <2294453B-8059-4AB9-8566-44E22B859C9A@freeswitch.org><4726C6D0-897E-46D9-91F6-6D2EF646344F@freeswitch.org> Message-ID: Ok, It's now working as expected, looks like I had something odd set in the phone's peer definition. I'll try and back track to see what I was doing wrong, but what ever it was, was preventing the caller id from being sent in the INVITE. Regards ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Nik Middleton Sent: 05 February 2009 23:26 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Caller ID not being passed Yes, I'll report back tomorrow, Regards ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: 05 February 2009 23:21 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Caller ID not being passed I notice you're using 1.0.2 any way you can test this with 1.0.3 RC1 tarball? /b On Feb 5, 2009, at 5:12 PM, Nik Middleton wrote: Dial plan is as per default setup with the addition of the following. To be honest, and I'm no SIP guru, I can't see the caller-id being set in the sip headers Mag.xml -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090206/3eabbdc6/attachment.html From steveu at coppice.org Fri Feb 6 08:02:21 2009 From: steveu at coppice.org (Steve Underwood) Date: Sat, 07 Feb 2009 00:02:21 +0800 Subject: [Freeswitch-users] Transcoding G723 In-Reply-To: <35b355e90902060716v2dff2586t253a69c97ab1adac@mail.gmail.com> References: <2FEEE7F7-DCE7-4919-A0FD-A0042BBB7EEE@freeswitch.org> <35b355e90902051226g12756d2fsf85fb996ec1ed188@mail.gmail.com> <11C3C72A-247D-4F8C-B106-52D49949D026@freeswitch.org> <1b46b4e80902051252o1b5a5c75l9f8d242cbaeec552@mail.gmail.com> <35b355e90902051301r4c2c53b6tb958774f4754e60f@mail.gmail.com> <35b355e90902060716v2dff2586t253a69c97ab1adac@mail.gmail.com> Message-ID: <498C5F0D.9040409@coppice.org> Shelby Ramsey wrote: > Thanks Moises. It looks like good work. When is Sangoma coming out > with a similar product ... Doug told me it was in the works, then not > in the works, then back in the works ... > > The problem is this particular card is PCI only and it will only do > 120 channels .... If I were them, I'd think long and hard about such a product, and in the end probably not do it. Its the only realistic way to do G.723.1, but the market for that is not so big. For the much larger number wanting G.729, the main CPUs keep getting faster. There are 6 core Xeons now, and there are supposed to be 8 cores by the end of 2009. You can have 4 of those in a 1U chassis. Their speed is just going to keep running ahead of a card. The only real DSP that needs to be offloaded is EC, because a large number of channels of that is probably going to challenge the main CPUs for some time to come. Steve From anthony.minessale at gmail.com Fri Feb 6 08:02:41 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 6 Feb 2009 10:02:41 -0600 Subject: [Freeswitch-users] mod_openzap stops working after some calls In-Reply-To: <498C2DC3.40701@ewetel.de> References: <498C2DC3.40701@ewetel.de> Message-ID: <191c3a030902060802r424f8a22tb4887dff7c6aead5@mail.gmail.com> I think we have some trouble surviving issues. So when everything is ok we do fine but if something goes wrong we don't recover. We are still missing state timers in the q931. maybe you can use your new pcap thing to see what goes wrong =D On Fri, Feb 6, 2009 at 6:32 AM, Helmut Kuper wrote: > Hello, > > since yesterday I do a real life test with FS in a 40 sip extension > environment with TDM connection via wanpipe/sangoma A104d. > > I detected two times the problem, that openzap stops working, while SIP > calls worked. Only restarting helped. Maybe reloading of mod_openzap > helps as well, but I didn't tested that, yet. I filtered FS logfile for > hints of that problem and I found a growing number of this line: > > 2009-02-06 11:09:51 [INFO] ozmod_isdn.c:706 zap_isdn_931_34() Duplicate > SETUP message(?) for Channel 1:21 ~ 1:21 in state DOWN [ignoring] > > Befor restart I saw such a line for *each* TDM channel and no one was > able to dial out. mod_openzap handled around 40 outgoing calls until > last restart . > > Since last restart FS runs good for 2 hours now without any duplicate > SETUP in log ... > > Has anybody similar problems? > > > regards > Helmut > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090206/e4d49369/attachment.html From nik.middleton at noblesolutions.co.uk Fri Feb 6 08:09:45 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Fri, 6 Feb 2009 16:09:45 -0000 Subject: [Freeswitch-users] Call accounting - CDR's Message-ID: Hi Guys I'm looking for some pointers on how to collect CDR's and store in mysql. Is there anything built in yet? I can rate the calls as a batch process, I simply need the call data. Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090206/fd9f1ada/attachment-0001.html From freeswitch-users at lists.rupa.com Fri Feb 6 08:27:17 2009 From: freeswitch-users at lists.rupa.com (Rupa Schomaker (lists)) Date: Fri, 06 Feb 2009 10:27:17 -0600 Subject: [Freeswitch-users] Call accounting - CDR's In-Reply-To: References: Message-ID: <498C64E5.3040103@lists.rupa.com> Use the script in scripts/contrib/wasim/ as a starting point. Basically, you log to csv files, and then the script periodically picks them up and loads to your DB. This is using mysql as an example, but you can do the same with postgres as well. If you need realtime inserts, then use mod_cdr_xml and have those post to a script on a webserver that parses the xml and inserts into appropriate tables. This is what I use along with a rails app. Remember that if you do real time, you also need to periodically scrape the error directory and load those (mod_cdr_xml will save to error if it can't successfully post to your script). On 2/6/2009 10:09 AM, Nik Middleton wrote: > Hi Guys > > > > I?m looking for some pointers on how to collect CDR?s and store in > mysql. Is there anything built in yet? > > > > I can rate the calls as a batch process, I simply need the call data. > > > > Regards > From sicfslist at gmail.com Fri Feb 6 08:28:54 2009 From: sicfslist at gmail.com (Shelby Ramsey) Date: Fri, 6 Feb 2009 10:28:54 -0600 Subject: [Freeswitch-users] Transcoding G723 In-Reply-To: <498C5F0D.9040409@coppice.org> References: <2FEEE7F7-DCE7-4919-A0FD-A0042BBB7EEE@freeswitch.org> <35b355e90902051226g12756d2fsf85fb996ec1ed188@mail.gmail.com> <11C3C72A-247D-4F8C-B106-52D49949D026@freeswitch.org> <1b46b4e80902051252o1b5a5c75l9f8d242cbaeec552@mail.gmail.com> <35b355e90902051301r4c2c53b6tb958774f4754e60f@mail.gmail.com> <35b355e90902060716v2dff2586t253a69c97ab1adac@mail.gmail.com> <498C5F0D.9040409@coppice.org> Message-ID: <35b355e90902060828g2552e05i882f8a8f1a05d37c@mail.gmail.com> Steve, You definitely have a better grasp on this topic than me. But I think it's a tough sell on the host based processing ... when you look at products like what audio codes can do on a card (3 DS-3's worth of transcoding) ... but I have had a couple of soft switch vendors claim though that they could do 2,000 calls per host (but I seriously doubt it). I agree that producing a card that does 120 channels is pretty worthless ... but having something that could do say a 1000 would be very helpful. G723 is a pretty big deal internationally ... I'm even seeing crazy requests like AMR from folks trying to originate VoIP off of mobile devices in Europe. But I agree just being able to g729 --> ulaw or ulaw --> g729 would be a great first step (host based or otherwise). SDR -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090206/2ced76a5/attachment.html From sicfslist at gmail.com Fri Feb 6 08:32:36 2009 From: sicfslist at gmail.com (Shelby Ramsey) Date: Fri, 6 Feb 2009 10:32:36 -0600 Subject: [Freeswitch-users] Call accounting - CDR's In-Reply-To: References: Message-ID: <35b355e90902060832r4bb1b70y9ca850726ae1510e@mail.gmail.com> Nik, There are a bunch of ways to do this ... mod_xml_cdr posts to a url then you can parse and dump ... or you can use mod_cdr_csv which allows you to dictate exactly what you want to collect and then parse the file and dump into mysql. There are also a couple of examples here --> http://wiki.freeswitch.org/wiki/Mod_cdr for hacking up the cdr_csv.conf and making it do what you want. SDR -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090206/85911ac0/attachment.html From nik.middleton at noblesolutions.co.uk Fri Feb 6 08:59:09 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Fri, 6 Feb 2009 16:59:09 -0000 Subject: [Freeswitch-users] Call accounting - CDR's In-Reply-To: <498C64E5.3040103@lists.rupa.com> References: <498C64E5.3040103@lists.rupa.com> Message-ID: Thanks, I was confused because I saw that mod cdr had been dropped. Due anticipated call volumes, batch processing is ideal, it keeps any MySql load issues away from FS Regards, -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rupa Schomaker (lists) Sent: 06 February 2009 16:27 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Call accounting - CDR's Use the script in scripts/contrib/wasim/ as a starting point. Basically, you log to csv files, and then the script periodically picks them up and loads to your DB. This is using mysql as an example, but you can do the same with postgres as well. If you need realtime inserts, then use mod_cdr_xml and have those post to a script on a webserver that parses the xml and inserts into appropriate tables. This is what I use along with a rails app. Remember that if you do real time, you also need to periodically scrape the error directory and load those (mod_cdr_xml will save to error if it can't successfully post to your script). On 2/6/2009 10:09 AM, Nik Middleton wrote: > Hi Guys > > > > I'm looking for some pointers on how to collect CDR's and store in > mysql. Is there anything built in yet? > > > > I can rate the calls as a batch process, I simply need the call data. > > > > Regards > _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From ajlong at worldlink.net Fri Feb 6 09:30:34 2009 From: ajlong at worldlink.net (Adam Long) Date: Fri, 6 Feb 2009 12:30:34 -0500 Subject: [Freeswitch-users] Call accounting - CDR's In-Reply-To: <498C64E5.3040103@lists.rupa.com> References: <498C64E5.3040103@lists.rupa.com> Message-ID: <030001c98880$9e660f90$db322eb0$@net> Is mod_cdr_xml asynchronous ... by that I mean .. if there is a communication failure how does this effect load on the system or the call for that matter... it wouldn?t hold up the progress or delay the turn up of the call or anything would it? I understand it would write to the error directory (but my thoughts are what is the impact of this beyond the obvious IO hit) I guess a good question is what is more load intensive??? 1.) mod_cdr_csv (with batch script that loads into DB somewhere) 2.) mod_cdr_xml (posting to lighttpd on remote host inserting into DB) I'm thinking about this for a system that would be handling in excess of 200-300 call setups per second. -Adam -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rupa Schomaker (lists) Sent: Friday, February 06, 2009 11:27 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Call accounting - CDR's Use the script in scripts/contrib/wasim/ as a starting point. Basically, you log to csv files, and then the script periodically picks them up and loads to your DB. This is using mysql as an example, but you can do the same with postgres as well. If you need realtime inserts, then use mod_cdr_xml and have those post to a script on a webserver that parses the xml and inserts into appropriate tables. This is what I use along with a rails app. Remember that if you do real time, you also need to periodically scrape the error directory and load those (mod_cdr_xml will save to error if it can't successfully post to your script). On 2/6/2009 10:09 AM, Nik Middleton wrote: > Hi Guys > > > > I?m looking for some pointers on how to collect CDR?s and store in > mysql. Is there anything built in yet? > > > > I can rate the calls as a batch process, I simply need the call data. > > > > Regards > _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From krice at freeswitch.org Fri Feb 6 09:40:16 2009 From: krice at freeswitch.org (Ken Rice) Date: Fri, 06 Feb 2009 11:40:16 -0600 Subject: [Freeswitch-users] Call accounting - CDR's In-Reply-To: <030001c98880$9e660f90$db322eb0$@net> Message-ID: Mod_xml_cdr will drop a file to the file system on failure to post. You can also leverage this drop a file to the file system and run a CDR processor locally. We handle call rates in the 500+ range using the local file system as a caching mechanism and a simple PHP script to rate the CDRs and load them into a pgsql db. > From: Adam Long > Reply-To: > Date: Fri, 6 Feb 2009 12:30:34 -0500 > To: > Subject: Re: [Freeswitch-users] Call accounting - CDR's > > Is mod_cdr_xml asynchronous ... by that I mean .. if there is a communication > failure how does this effect load on the system or the call for that > matter... it wouldn?t hold up the progress or delay the turn up of the call or > anything would it? I understand it would write to the error directory (but my > thoughts are what is the impact of this beyond the obvious IO hit) I guess a > good question is what is more load intensive??? 1.) mod_cdr_csv (with batch > script that loads into DB somewhere) 2.) mod_cdr_xml (posting to lighttpd on > remote host inserting into DB) I'm thinking about this for a system that > would be handling in excess of 200-300 call setups per > second. -Adam -----Original Message----- From: > freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rupa > Schomaker (lists) Sent: Friday, February 06, 2009 11:27 AM To: > freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Call > accounting - CDR's Use the script in scripts/contrib/wasim/ as a starting > point. Basically, you log to csv files, and then the script periodically > picks them up and loads to your DB. This is using mysql as an example, > but you can do the same with postgres as well. If you need realtime inserts, > then use mod_cdr_xml and have those post to a script on a webserver that > parses the xml and inserts into appropriate tables. This is what I use along > with a rails app. Remember that if you do real time, you also need to > periodically scrape the error directory and load those (mod_cdr_xml will save > to error if it can't successfully post to your script). On 2/6/2009 10:09 AM, > Nik Middleton wrote: > Hi Guys > > > > I?m looking for some pointers on > how to collect CDR?s and store in > mysql. Is there anything built in yet? > > > > > I can rate the calls as a batch process, I simply need the call > data. > > > > Regards > > _______________________________________________ Freeswitch-users mailing > list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman > /listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/opt > ions/freeswitch-users http://www.freeswitch.org ____________________________ > ___________________ Freeswitch-users mailing > list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman > /listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/opt > ions/freeswitch-users http://www.freeswitch.org From freeswitch-users at digitaldan.com Fri Feb 6 10:59:12 2009 From: freeswitch-users at digitaldan.com (Dan) Date: Fri, 6 Feb 2009 11:59:12 -0700 (MST) Subject: [Freeswitch-users] mod_shout delay in trunk Message-ID: <11000168.01233946752072.JavaMail.root@zimbra> Hi, With the 1.0.2 release i was able to to stream a call using mod_shout to an icecast server with only a 1 or two second delay to clients. With the current trunk that delay is now 8 to 10 seconds. I thought it might have been a change to mod_shout.c. I tried tweaking a few outbound buffer sizes with no luck so I just copied the 1.0.2 version of mod_shout.c over, compiled and reinstalled the module, restarted fs and still the delay is 8 to 10 seconds. I'm a little stumped. I currently have both versions installed (trunk and 1.0.2) for testing. Both are streaming to the same icecast server. My current svn revision is 11669, the calls are coming in via sip using g.711 ulaw and it looks like lame/mod_shout is streaming it as a 16kbs, 8khz mono mp3 stream. I'm using a flash/flex applet I wrote to consume the icecast stream, although I have used totem to listen to the stream as well. Any thoughts -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090206/40be49c6/attachment.html From brian at freeswitch.org Fri Feb 6 11:11:54 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 6 Feb 2009 13:11:54 -0600 Subject: [Freeswitch-users] mod_shout delay in trunk In-Reply-To: <11000168.01233946752072.JavaMail.root@zimbra> References: <11000168.01233946752072.JavaMail.root@zimbra> Message-ID: Yes this will be normal due to buffering. Have you tested svn trunk? /b On Feb 6, 2009, at 12:59 PM, Dan wrote: > Hi, > > With the 1.0.2 release i was able to to stream a call using > mod_shout to an icecast server with only a 1 or two second delay to > clients. With the current trunk that delay is now 8 to 10 seconds. > I thought it might have been a change to mod_shout.c. I tried > tweaking a few outbound buffer sizes with no luck so I just copied > the 1.0.2 version of mod_shout.c over, compiled and reinstalled the > module, restarted fs and still the delay is 8 to 10 seconds. I'm a > little stumped. I currently have both versions installed (trunk > and 1.0.2) for testing. Both are streaming to the same icecast > server. > > My current svn revision is 11669, the calls are coming in via sip > using g.711 ulaw and it looks like lame/mod_shout is streaming it as > a 16kbs, 8khz mono mp3 stream. I'm using a flash/flex applet I > wrote to consume the icecast stream, although I have used totem to > listen to the stream as well. > > Any thoughts -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090206/56093979/attachment-0001.html From mitul at enterux.com Fri Feb 6 11:27:39 2009 From: mitul at enterux.com (Mitul Limbani) Date: Sat, 7 Feb 2009 00:57:39 +0530 Subject: [Freeswitch-users] Transcoding G723 In-Reply-To: <35b355e90902060828g2552e05i882f8a8f1a05d37c@mail.gmail.com> References: <2FEEE7F7-DCE7-4919-A0FD-A0042BBB7EEE@freeswitch.org> <35b355e90902051226g12756d2fsf85fb996ec1ed188@mail.gmail.com> <11C3C72A-247D-4F8C-B106-52D49949D026@freeswitch.org> <1b46b4e80902051252o1b5a5c75l9f8d242cbaeec552@mail.gmail.com> <35b355e90902051301r4c2c53b6tb958774f4754e60f@mail.gmail.com> <35b355e90902060716v2dff2586t253a69c97ab1adac@mail.gmail.com> <498C5F0D.9040409@coppice.org> <35b355e90902060828g2552e05i882f8a8f1a05d37c@mail.gmail.com> Message-ID: <26923699-F29A-42AB-BACE-11921955F0C2@enterux.com> Hi, I came across a company who was selling hardware which could do like 5000 G729 conversion simultanoeusly, I was like this sounds cool, they have support for asterisk, I haven't enquirer yet how they do this, but anyone wishes to buy it cost ?15000/year for the hrdware + support Any one on the list requires such high throughput calls can connect with me so that we can really test if it can work with freeswitch. Regards, Mitul Limbani, Founder & CEO, Enterux Solutions Pvt Ltd, The Enterprise Linux Company(r), http://www.enterux.com/ On 06-Feb-09, at 21:58, Shelby Ramsey wrote: > Steve, > > You definitely have a better grasp on this topic than me. But I > think it's a tough sell on the host based processing ... when you > look at products like what audio codes can do on a card (3 DS-3's > worth of transcoding) ... but I have had a couple of soft switch > vendors claim though that they could do 2,000 calls per host (but I > seriously doubt it). > > I agree that producing a card that does 120 channels is pretty > worthless ... but having something that could do say a 1000 would be > very helpful. > > G723 is a pretty big deal internationally ... I'm even seeing crazy > requests like AMR from folks trying to originate VoIP off of mobile > devices in Europe. > > But I agree just being able to g729 --> ulaw or ulaw --> g729 would > be a great first step (host based or otherwise). > > SDR > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Fri Feb 6 11:40:53 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 6 Feb 2009 13:40:53 -0600 Subject: [Freeswitch-users] Transcoding G723 In-Reply-To: <26923699-F29A-42AB-BACE-11921955F0C2@enterux.com> References: <2FEEE7F7-DCE7-4919-A0FD-A0042BBB7EEE@freeswitch.org> <35b355e90902051226g12756d2fsf85fb996ec1ed188@mail.gmail.com> <11C3C72A-247D-4F8C-B106-52D49949D026@freeswitch.org> <1b46b4e80902051252o1b5a5c75l9f8d242cbaeec552@mail.gmail.com> <35b355e90902051301r4c2c53b6tb958774f4754e60f@mail.gmail.com> <35b355e90902060716v2dff2586t253a69c97ab1adac@mail.gmail.com> <498C5F0D.9040409@coppice.org> <35b355e90902060828g2552e05i882f8a8f1a05d37c@mail.gmail.com> <26923699-F29A-42AB-BACE-11921955F0C2@enterux.com> Message-ID: PER YEAR? ARE THEY DAFT? /b On Feb 6, 2009, at 1:27 PM, Mitul Limbani wrote: > it cost ?15000/year for the hrdware + support From freeswitch-users at digitaldan.com Fri Feb 6 11:43:17 2009 From: freeswitch-users at digitaldan.com (freeswitch-users at digitaldan.com) Date: Fri, 6 Feb 2009 12:43:17 -0700 (MST) Subject: [Freeswitch-users] mod_shout delay in trunk In-Reply-To: <1047580.31233949312392.JavaMail.root@zimbra> Message-ID: <17591403.51233949397584.JavaMail.root@zimbra> I have, do you know what would have changed between 1.0.2 and trunk that would cause the buffer to change? Also if its not in mod_shout.c (which I copied from 1.0.2 to trunk for testing with no luck), where else would fs be buffering? One thing I have noticed is that in 1.0.2 as soon as the dial plan hits my record statement I see mod_shout logging that it has connected to the icecast server, in trunk it takes about 5 seconds to see the same log mesage. Below is my current svn info Path: . URL: http://svn.freeswitch.org/svn/freeswitch/trunk Repository Root: http://svn.freeswitch.org/svn Repository UUID: d0543943-73ff-0310-b7d9-9358b9ac24b2 Revision: 11669 Node Kind: directory Schedule: normal Last Changed Author: brian Last Changed Rev: 11669 Last Changed Date: 2009-02-06 11:29:51 -0700 (Fri, 06 Feb 2009) ----- Original Message ----- From: "Brian West" To: freeswitch-users at lists.freeswitch.org Sent: Friday, February 6, 2009 12:11:54 PM GMT -07:00 US/Canada Mountain Subject: Re: [Freeswitch-users] mod_shout delay in trunk Yes this will be normal due to buffering. Have you tested svn trunk? /b On Feb 6, 2009, at 12:59 PM, Dan wrote: Hi, With the 1.0.2 release i was able to to stream a call using mod_shout to an icecast server with only a 1 or two second delay to clients. With the current trunk that delay is now 8 to 10 seconds. I thought it might have been a change to mod_shout.c. I tried tweaking a few outbound buffer sizes with no luck so I just copied the 1.0.2 version of mod_shout.c over, compiled and reinstalled the module, restarted fs and still the delay is 8 to 10 seconds. I'm a little stumped. I currently have both versions installed (trunk and 1.0.2) for testing. Both are streaming to the same icecast server. My current svn revision is 11669, the calls are coming in via sip using g.711 ulaw and it looks like lame/mod_shout is streaming it as a 16kbs, 8khz mono mp3 stream. I'm using a flash/flex applet I wrote to consume the icecast stream, although I have used totem to listen to the stream as well. Any thoughts _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090206/daf345e3/attachment.html From brian at freeswitch.org Fri Feb 6 11:47:53 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 6 Feb 2009 13:47:53 -0600 Subject: [Freeswitch-users] mod_shout delay in trunk In-Reply-To: <17591403.51233949397584.JavaMail.root@zimbra> References: <17591403.51233949397584.JavaMail.root@zimbra> Message-ID: <824A1B7F-A0CE-4ED9-AEF2-97037E8F1E59@freeswitch.org> Let me clarify.. yes this is normal file buffering was added so we wouldn't thrash your hard drive with tiny bits of data when recording calls so now it buffers and writes larger chunks to disk. This is why you have this delay which is 100% normal.... is realtime a critical thing? It is shout cast so you know it doesn't have to be realtime.. in fact some clients will buffer a little bit anyway and add to it. /b On Feb 6, 2009, at 1:43 PM, freeswitch-users at digitaldan.com wrote: > I have, do you know what would have changed between 1.0.2 and trunk > that would cause the buffer to change? Also if its not in > mod_shout.c (which I copied from 1.0.2 to trunk for testing with no > luck), where else would fs be buffering? One thing I have noticed > is that in 1.0.2 as soon as the dial plan hits my record statement I > see mod_shout logging that it has connected to the icecast server, > in trunk it takes about 5 seconds to see the same log mesage. Below > is my current svn info > Path: . -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090206/eabd461d/attachment.html From freeswitch-users at digitaldan.com Fri Feb 6 12:01:23 2009 From: freeswitch-users at digitaldan.com (freeswitch-users at digitaldan.com) Date: Fri, 6 Feb 2009 13:01:23 -0700 (MST) Subject: [Freeswitch-users] mod_shout delay in trunk In-Reply-To: <16242086.81233950401907.JavaMail.root@zimbra> Message-ID: <8732241.101233950483385.JavaMail.root@zimbra> For me it is. For what I'm using it for I can tolerate around a second or two delay. I have the icecast server setup to only buffer 1K for their on-connect burst as well as my flash/flex player to only buffer 1k (yes I might as well not buffer at all, which I may end up doing). In 1.0.2 this worked very well. Is this buffer configurable? If not, where is it being set? Thanks Dan- ----- Original Message ----- From: "Brian West" To: freeswitch-users at lists.freeswitch.org Sent: Friday, February 6, 2009 12:47:53 PM GMT -07:00 US/Canada Mountain Subject: Re: [Freeswitch-users] mod_shout delay in trunk Let me clarify.. yes this is normal file buffering was added so we wouldn't thrash your hard drive with tiny bits of data when recording calls so now it buffers and writes larger chunks to disk. This is why you have this delay which is 100% normal.... is realtime a critical thing? It is shout cast so you know it doesn't have to be realtime.. in fact some clients will buffer a little bit anyway and add to it. /b On Feb 6, 2009, at 1:43 PM, freeswitch-users at digitaldan.com wrote: I have, do you know what would have changed between 1.0.2 and trunk that would cause the buffer to change? Also if its not in mod_shout.c (which I copied from 1.0.2 to trunk for testing with no luck), where else would fs be buffering? One thing I have noticed is that in 1.0.2 as soon as the dial plan hits my record statement I see mod_shout logging that it has connected to the icecast server, in trunk it takes about 5 seconds to see the same log mesage. Below is my current svn info Path: . _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090206/217ca9cf/attachment.html From anthony.minessale at gmail.com Fri Feb 6 12:07:44 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 6 Feb 2009 14:07:44 -0600 Subject: [Freeswitch-users] mod_shout delay in trunk In-Reply-To: <8732241.101233950483385.JavaMail.root@zimbra> References: <16242086.81233950401907.JavaMail.root@zimbra> <8732241.101233950483385.JavaMail.root@zimbra> Message-ID: <191c3a030902061207x24bb8f05h901f1f7ad208926c@mail.gmail.com> edit switch_ivr_play_say.c line 423 comment the line out and recompile. Tell me if it helps you and i will consider making it configurable. On Fri, Feb 6, 2009 at 2:01 PM, wrote: > For me it is. For what I'm using it for I can tolerate around a second or > two delay. I have the icecast server setup to only buffer 1K for their > on-connect burst as well as my flash/flex player to only buffer 1k (yes I > might as well not buffer at all, which I may end up doing). In 1.0.2 this > worked very well. Is this buffer configurable? If not, where is it being > set? > > Thanks > Dan- > ----- Original Message ----- > From: "Brian West" > To: freeswitch-users at lists.freeswitch.org > Sent: Friday, February 6, 2009 12:47:53 PM GMT -07:00 US/Canada Mountain > Subject: Re: [Freeswitch-users] mod_shout delay in trunk > > Let me clarify.. yes this is normal file buffering was added so we wouldn't > thrash your hard drive with tiny bits of data when recording calls so now it > buffers and writes larger chunks to disk. This is why you have this delay > which is 100% normal.... is realtime a critical thing? It is shout cast so > you know it doesn't have to be realtime.. in fact some clients will buffer a > little bit anyway and add to it. > /b > > On Feb 6, 2009, at 1:43 PM, freeswitch-users at digitaldan.com wrote: > > I have, do you know what would have changed between 1.0.2 and trunk that > would cause the buffer to change? Also if its not in mod_shout.c (which I > copied from 1.0.2 to trunk for testing with no luck), where else would fs be > buffering? One thing I have noticed is that in 1.0.2 as soon as the dial > plan hits my record statement I see mod_shout logging that it has connected > to the icecast server, in trunk it takes about 5 seconds to see the same log > mesage. Below is my current svn info > Path: . > > > > _______________________________________________ Freeswitch-users mailing > list Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090206/562d328c/attachment-0001.html From kristian.kielhofner at gmail.com Fri Feb 6 12:20:25 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Fri, 6 Feb 2009 15:20:25 -0500 Subject: [Freeswitch-users] Compiling FreeSWITCH for AstLinux Message-ID: <2d9149cd0902061220i11b87fd9se253109d7a39249a@mail.gmail.com> Hey guys, I've finally gotten around to trying to compile FreeSWITCH for AstLinux. Here is the branch I've created for it: http://astlinux.svn.sourceforge.net/viewvc/astlinux/branches/astlinux-freeswitch/package/freeswitch/ Right now it is bombing trying to run configure for libs/sqlite, as shown in the build log here: http://astbuild.star2star.com/astlinux-freeswitch-build.log Here is the config.log for sqlite: http://astbuild.star2star.com/sqlite-config.log I'll continue to dig into this but in the meantime I thought I'd get some extra eyeballs on it... Thanks! P.S. - Yes, yes I know "AstLinux" isn't the best name for a distro with FreeSWITCH. Depending on my success here I have some other ideas... -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From darren at aleph-com.net Fri Feb 6 12:29:59 2009 From: darren at aleph-com.net (Darren Wiebe) Date: Fri, 06 Feb 2009 13:29:59 -0700 Subject: [Freeswitch-users] Call accounting - CDR's In-Reply-To: References: Message-ID: <498C9DC7.6030104@aleph-com.net> Nik Middleton wrote: > > Hi Guys > > I?m looking for some pointers on how to collect CDR?s and store in > mysql. Is there anything built in yet? > > I can rate the calls as a batch process, I simply need the call data. > > Regards > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > I use mod_cdr_csv and then run a process every minute to import them into the mysql database. It seems to work well. -- Darren Wiebe darren at aleph-com.net Aleph Communications www.aleph-com.net From mgg at giagnocavo.net Fri Feb 6 12:30:38 2009 From: mgg at giagnocavo.net (Michael Giagnocavo) Date: Fri, 6 Feb 2009 15:30:38 -0500 Subject: [Freeswitch-users] Transcoding G723 In-Reply-To: <26923699-F29A-42AB-BACE-11921955F0C2@enterux.com> References: <2FEEE7F7-DCE7-4919-A0FD-A0042BBB7EEE@freeswitch.org> <35b355e90902051226g12756d2fsf85fb996ec1ed188@mail.gmail.com> <11C3C72A-247D-4F8C-B106-52D49949D026@freeswitch.org> <1b46b4e80902051252o1b5a5c75l9f8d242cbaeec552@mail.gmail.com> <35b355e90902051301r4c2c53b6tb958774f4754e60f@mail.gmail.com> <35b355e90902060716v2dff2586t253a69c97ab1adac@mail.gmail.com> <498C5F0D.9040409@coppice.org> <35b355e90902060828g2552e05i882f8a8f1a05d37c@mail.gmail.com> <26923699-F29A-42AB-BACE-11921955F0C2@enterux.com> Message-ID: <6E8D2069C08AA84A83D336E996AE4C6702362F6BCE@mse17be1.mse17.exchange.ms> $22K would buy quite a few machines with many core Xeons. I just don't see how it'd be effective at that price. Not to mention a yearly figure. The only thing to take into consideration would be the G729 licenses. But in bulk, the price should be pretty effective, even figuring in hardware. (Not to mention what things like Larrabee will mean for encoding -- 32 1.5GHz "Pentium 4 x64" cores, each with 4 threads?) And... if you're running Asterisk, um, isn't 5000 channels a wee bit over what you can handle anyways? ;) -Michael -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mitul Limbani Sent: Friday, February 06, 2009 12:28 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Transcoding G723 Hi, I came across a company who was selling hardware which could do like 5000 G729 conversion simultanoeusly, I was like this sounds cool, they have support for asterisk, I haven't enquirer yet how they do this, but anyone wishes to buy it cost ?15000/year for the hrdware + support Any one on the list requires such high throughput calls can connect with me so that we can really test if it can work with freeswitch. Regards, Mitul Limbani, Founder & CEO, Enterux Solutions Pvt Ltd, The Enterprise Linux Company(r), http://www.enterux.com/ On 06-Feb-09, at 21:58, Shelby Ramsey wrote: > Steve, > > You definitely have a better grasp on this topic than me. But I > think it's a tough sell on the host based processing ... when you > look at products like what audio codes can do on a card (3 DS-3's > worth of transcoding) ... but I have had a couple of soft switch > vendors claim though that they could do 2,000 calls per host (but I > seriously doubt it). > > I agree that producing a card that does 120 channels is pretty > worthless ... but having something that could do say a 1000 would be > very helpful. > > G723 is a pretty big deal internationally ... I'm even seeing crazy > requests like AMR from folks trying to originate VoIP off of mobile > devices in Europe. > > But I agree just being able to g729 --> ulaw or ulaw --> g729 would > be a great first step (host based or otherwise). > > SDR > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From freeswitch-users at digitaldan.com Fri Feb 6 12:36:14 2009 From: freeswitch-users at digitaldan.com (freeswitch-users at digitaldan.com) Date: Fri, 6 Feb 2009 13:36:14 -0700 (MST) Subject: [Freeswitch-users] mod_shout delay in trunk In-Reply-To: <15830211.131233952452225.JavaMail.root@zimbra> Message-ID: <28165623.151233952574612.JavaMail.root@zimbra> That worked great! I wanted to say just how awesome Freeswitch is, I have been doing voip related development with SIP since 2000 and this is by far the most well designed piece of voip software I have used or developed on. I currently have a homegrown sip server built on the NIST sip stack with Sun's JMF libraries for RTP processing. 95% of the code and complexity is handling the SIP and RTP sessions, the other 5% is the final application logic and what is most important to me . By letting freeswitch do whats its good at (call routing, sip and media handling) it allows me to focus on what I'm good at (what should we do with those streams, like record them). I have been bragging about this project to anybody who will listen! Dan- ----- Original Message ----- From: "Anthony Minessale" To: freeswitch-users at lists.freeswitch.org Sent: Friday, February 6, 2009 1:07:44 PM GMT -07:00 US/Canada Mountain Subject: Re: [Freeswitch-users] mod_shout delay in trunk edit switch_ivr_play_say.c line 423 comment the line out and recompile. Tell me if it helps you and i will consider making it configurable. On Fri, Feb 6, 2009 at 2:01 PM, < freeswitch-users at digitaldan.com > wrote: For me it is. For what I'm using it for I can tolerate around a second or two delay. I have the icecast server setup to only buffer 1K for their on-connect burst as well as my flash/flex player to only buffer 1k (yes I might as well not buffer at all, which I may end up doing). In 1.0.2 this worked very well. Is this buffer configurable? If not, where is it being set? Thanks Dan- ----- Original Message ----- From: "Brian West" < brian at freeswitch.org > To: freeswitch-users at lists.freeswitch.org Sent: Friday, February 6, 2009 12:47:53 PM GMT -07:00 US/Canada Mountain Subject: Re: [Freeswitch-users] mod_shout delay in trunk Let me clarify.. yes this is normal file buffering was added so we wouldn't thrash your hard drive with tiny bits of data when recording calls so now it buffers and writes larger chunks to disk. This is why you have this delay which is 100% normal.... is realtime a critical thing? It is shout cast so you know it doesn't have to be realtime.. in fact some clients will buffer a little bit anyway and add to it. /b On Feb 6, 2009, at 1:43 PM, freeswitch-users at digitaldan.com wrote: I have, do you know what would have changed between 1.0.2 and trunk that would cause the buffer to change? Also if its not in mod_shout.c (which I copied from 1.0.2 to trunk for testing with no luck), where else would fs be buffering? One thing I have noticed is that in 1.0.2 as soon as the dial plan hits my record statement I see mod_shout logging that it has connected to the icecast server, in trunk it takes about 5 seconds to see the same log mesage. Below is my current svn info Path: . _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/ PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090206/64231bc1/attachment.html From wasim at convergence.pk Fri Feb 6 12:40:11 2009 From: wasim at convergence.pk (Wasim Baig) Date: Sat, 7 Feb 2009 01:40:11 +0500 Subject: [Freeswitch-users] Transcoding G723 In-Reply-To: <6E8D2069C08AA84A83D336E996AE4C6702362F6BCE@mse17be1.mse17.exchange.ms> References: <1b46b4e80902051252o1b5a5c75l9f8d242cbaeec552@mail.gmail.com> <35b355e90902051301r4c2c53b6tb958774f4754e60f@mail.gmail.com> <35b355e90902060716v2dff2586t253a69c97ab1adac@mail.gmail.com> <498C5F0D.9040409@coppice.org> <35b355e90902060828g2552e05i882f8a8f1a05d37c@mail.gmail.com> <26923699-F29A-42AB-BACE-11921955F0C2@enterux.com> <6E8D2069C08AA84A83D336E996AE4C6702362F6BCE@mse17be1.mse17.exchange.ms> Message-ID: On Sat, Feb 7, 2009 at 1:30 AM, Michael Giagnocavo wrote: $22K would buy quite a few machines with many core Xeons. I just don't see > how it'd be effective at that price. Not to mention a yearly figure. G729 is roughly 25 MIPS (encode+decode), coppice, please correct as necessary. A dual quad-core xeon 3 ghz should do roughly 1k calls, assuming nothing else is being run. > The only thing to take into consideration would be the G729 licenses. But > in bulk, the price should be pretty effective, even figuring in hardware. Or places where licensing isn't enforced, like offshore g729 farms :) > (Not to mention what things like Larrabee will mean for encoding -- 32 > 1.5GHz "Pentium 4 x64" cores, each with 4 threads?) 32 x 1.5 GHZ = 48 Ghz or roughly 2000 channels ... > And... if you're running Asterisk, um, isn't 5000 channels a wee bit over > what you can handle anyways? ;) Yeh, but not when you have FS :) which is why talks of this capacity on HMP are becoming more and more relevant ... -- wasim h. baig | principal consultant | convergence pk | +92 300 8508070 | peace be upon you ... -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090207/b9a20c5d/attachment-0001.html From nik.middleton at noblesolutions.co.uk Fri Feb 6 12:40:51 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Fri, 6 Feb 2009 20:40:51 -0000 Subject: [Freeswitch-users] Call accounting - CDR's In-Reply-To: <030001c98880$9e660f90$db322eb0$@net> References: <498C64E5.3040103@lists.rupa.com> <030001c98880$9e660f90$db322eb0$@net> Message-ID: What about using a radius server, would that be more resilient? Regards -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Adam Long Sent: 06 February 2009 17:31 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Call accounting - CDR's Is mod_cdr_xml asynchronous ... by that I mean .. if there is a communication failure how does this effect load on the system or the call for that matter... it wouldn't hold up the progress or delay the turn up of the call or anything would it? I understand it would write to the error directory (but my thoughts are what is the impact of this beyond the obvious IO hit) I guess a good question is what is more load intensive??? 1.) mod_cdr_csv (with batch script that loads into DB somewhere) 2.) mod_cdr_xml (posting to lighttpd on remote host inserting into DB) I'm thinking about this for a system that would be handling in excess of 200-300 call setups per second. -Adam -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rupa Schomaker (lists) Sent: Friday, February 06, 2009 11:27 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Call accounting - CDR's Use the script in scripts/contrib/wasim/ as a starting point. Basically, you log to csv files, and then the script periodically picks them up and loads to your DB. This is using mysql as an example, but you can do the same with postgres as well. If you need realtime inserts, then use mod_cdr_xml and have those post to a script on a webserver that parses the xml and inserts into appropriate tables. This is what I use along with a rails app. Remember that if you do real time, you also need to periodically scrape the error directory and load those (mod_cdr_xml will save to error if it can't successfully post to your script). On 2/6/2009 10:09 AM, Nik Middleton wrote: > Hi Guys > > > > I'm looking for some pointers on how to collect CDR's and store in > mysql. Is there anything built in yet? > > > > I can rate the calls as a batch process, I simply need the call data. > > > > Regards > _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From anthony.minessale at gmail.com Fri Feb 6 12:48:31 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 6 Feb 2009 14:48:31 -0600 Subject: [Freeswitch-users] mod_shout delay in trunk In-Reply-To: <28165623.151233952574612.JavaMail.root@zimbra> References: <15830211.131233952452225.JavaMail.root@zimbra> <28165623.151233952574612.JavaMail.root@zimbra> Message-ID: <191c3a030902061248r3d307647le5574c1004c6da24@mail.gmail.com> Thanks, We appreciate the positive feedback! if you revert the change I suggested and update i added a new variable enable_file_write_buffering=false set this variable on the channel before you start recording it with the set application or in the dialstring in {} on outbound calls and it should skip the buffering. Could you test it for me and confirm it works? Thank you On Fri, Feb 6, 2009 at 2:36 PM, wrote: > That worked great! > > I wanted to say just how awesome Freeswitch is, I have been doing voip > related development with SIP since 2000 and this is by far the most well > designed piece of voip software I have used or developed on. I currently > have a homegrown sip server built on the NIST sip stack with Sun's JMF > libraries for RTP processing. 95% of the code and complexity is handling > the SIP and RTP sessions, the other 5% is the final application logic and > what is most important to me. By letting freeswitch do whats its good at > (call routing, sip and media handling) it allows me to focus on what I'm > good at (what should we do with those streams, like record them). I have > been bragging about this project to anybody who will listen! > > Dan- > ----- Original Message ----- > From: "Anthony Minessale" > To: freeswitch-users at lists.freeswitch.org > Sent: Friday, February 6, 2009 1:07:44 PM GMT -07:00 US/Canada Mountain > Subject: Re: [Freeswitch-users] mod_shout delay in trunk > > edit switch_ivr_play_say.c line 423 > > comment the line out and recompile. > Tell me if it helps you and i will consider making it configurable. > > > On Fri, Feb 6, 2009 at 2:01 PM, wrote: > >> For me it is. For what I'm using it for I can tolerate around a second or >> two delay. I have the icecast server setup to only buffer 1K for their >> on-connect burst as well as my flash/flex player to only buffer 1k (yes I >> might as well not buffer at all, which I may end up doing). In 1.0.2 this >> worked very well. Is this buffer configurable? If not, where is it being >> set? >> >> Thanks >> Dan- >> ----- Original Message ----- >> From: "Brian West" >> To: freeswitch-users at lists.freeswitch.org >> Sent: Friday, February 6, 2009 12:47:53 PM GMT -07:00 US/Canada Mountain >> Subject: Re: [Freeswitch-users] mod_shout delay in trunk >> >> Let me clarify.. yes this is normal file buffering was added so we >> wouldn't thrash your hard drive with tiny bits of data when recording calls >> so now it buffers and writes larger chunks to disk. This is why you have >> this delay which is 100% normal.... is realtime a critical thing? It is >> shout cast so you know it doesn't have to be realtime.. in fact some clients >> will buffer a little bit anyway and add to it. >> /b >> >> On Feb 6, 2009, at 1:43 PM, freeswitch-users at digitaldan.com wrote: >> >> I have, do you know what would have changed between 1.0.2 and trunk that >> would cause the buffer to change? Also if its not in mod_shout.c (which I >> copied from 1.0.2 to trunk for testing with no luck), where else would fs be >> buffering? One thing I have noticed is that in 1.0.2 as soon as the dial >> plan hits my record statement I see mod_shout logging that it has connected >> to the icecast server, in trunk it takes about 5 seconds to see the same log >> mesage. Below is my current svn info >> Path: . >> >> >> >> _______________________________________________ Freeswitch-users mailing >> list Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ Freeswitch-users mailing > list Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090206/50050217/attachment.html From nik.middleton at noblesolutions.co.uk Fri Feb 6 13:07:12 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Fri, 6 Feb 2009 21:07:12 -0000 Subject: [Freeswitch-users] Call accounting - CDR's In-Reply-To: References: <030001c98880$9e660f90$db322eb0$@net> Message-ID: So you're simply posting this file to a web server? How do you find the load on it at this rate of calls? BTW can anyone point me to resources discussing how to do this? (Using a web server to post data to a db) I've not this sort of thing before, and I'm not too sure what I should be goggling for Regards, -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice Sent: 06 February 2009 17:40 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Call accounting - CDR's Mod_xml_cdr will drop a file to the file system on failure to post. You can also leverage this drop a file to the file system and run a CDR processor locally. We handle call rates in the 500+ range using the local file system as a caching mechanism and a simple PHP script to rate the CDRs and load them into a pgsql db. > From: Adam Long > Reply-To: > Date: Fri, 6 Feb 2009 12:30:34 -0500 > To: > Subject: Re: [Freeswitch-users] Call accounting - CDR's > > Is mod_cdr_xml asynchronous ... by that I mean .. if there is a communication > failure how does this effect load on the system or the call for that > matter... it wouldn?t hold up the progress or delay the turn up of the call or > anything would it? I understand it would write to the error directory (but my > thoughts are what is the impact of this beyond the obvious IO hit) I guess a > good question is what is more load intensive??? 1.) mod_cdr_csv (with batch > script that loads into DB somewhere) 2.) mod_cdr_xml (posting to lighttpd on > remote host inserting into DB) I'm thinking about this for a system that > would be handling in excess of 200-300 call setups per > second. -Adam -----Original Message----- From: > freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rupa > Schomaker (lists) Sent: Friday, February 06, 2009 11:27 AM To: > freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Call > accounting - CDR's Use the script in scripts/contrib/wasim/ as a starting > point. Basically, you log to csv files, and then the script periodically > picks them up and loads to your DB. This is using mysql as an example, > but you can do the same with postgres as well. If you need realtime inserts, > then use mod_cdr_xml and have those post to a script on a webserver that > parses the xml and inserts into appropriate tables. This is what I use along > with a rails app. Remember that if you do real time, you also need to > periodically scrape the error directory and load those (mod_cdr_xml will save > to error if it can't successfully post to your script). On 2/6/2009 10:09 AM, > Nik Middleton wrote: > Hi Guys > > > > I?m looking for some pointers on > how to collect CDR?s and store in > mysql. Is there anything built in yet? > > > > > I can rate the calls as a batch process, I simply need the call > data. > > > > Regards > > _______________________________________________ Freeswitch-users mailing > list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman > /listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/opt > ions/freeswitch-users http://www.freeswitch.org ____________________________ > ___________________ Freeswitch-users mailing > list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman > /listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/opt > ions/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From mitul at enterux.com Fri Feb 6 15:02:13 2009 From: mitul at enterux.com (Mitul Limbani) Date: Fri, 6 Feb 2009 18:02:13 -0500 Subject: [Freeswitch-users] Transcoding G723 Message-ID: <49796.1233961333@enterux.com> BODY { font-family:Arial, Helvetica, sans-serif;font-size:12px; }Hello guys, Sorry it was my mistake, i re-read the entire proposal and it looks like their specialized 1U Hardware with custom CPU can handle close to 1500 simultaneous G729 encoding n transmission, also this hardware offers from any codec to any codec transcoding path. Also this box easily works over FreeSWITCH, Asterisk, RTPProxy, OpenSIPS are their claims. and their pricing are First year ?16,000 Second year ?13,000/annum recurring This is found slightly on higher side though, Thanks & Regards, Mitul Limbani, Founder & CEO, Enterux Solutions, The Enterprise Linux Company (TM), www.enterux.com +91-9820332422 On Sat 07/02/09 02:00 , Michael Giagnocavo mgg at giagnocavo.net sent: $22K would buy quite a few machines with many core Xeons. I just don't see how it'd be effective at that price. Not to mention a yearly figure. The only thing to take into consideration would be the G729 licenses. But in bulk, the price should be pretty effective, even figuring in hardware. (Not to mention what things like Larrabee will mean for encoding -- 32 1.5GHz "Pentium 4 x64" cores, each with 4 threads?) And... if you're running Asterisk, um, isn't 5000 channels a wee bit over what you can handle anyways? ;) -Michael -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org [1]] On Behalf Of Mitul Limbani Sent: Friday, February 06, 2009 12:28 PM To: freeswitch-users at lists.freeswitch.org [2] Subject: Re: [Freeswitch-users] Transcoding G723 Hi, I came across a company who was selling hardware which could do like 5000 G729 conversion simultanoeusly, I was like this sounds cool, they have support for asterisk, I haven't enquirer yet how they do this, but anyone wishes to buy it cost ?15000/year for the hrdware + support Any one on the list requires such high throughput calls can connect with me so that we can really test if it can work with freeswitch. Regards, Mitul Limbani, Founder & CEO, Enterux Solutions Pvt Ltd, The Enterprise Linux Company(r), http://www.enterux.com/ On 06-Feb-09, at 21:58, Shelby Ramsey wrote: > Steve, > > You definitely have a better grasp on this topic than me. But I > think it's a tough sell on the host based processing ... when you > look at products like what audio codes can do on a card (3 DS-3's > worth of transcoding) ... but I have had a couple of soft switch > vendors claim though that they could do 2,000 calls per host (but I > seriously doubt it). > > I agree that producing a card that does 120 channels is pretty > worthless ... but having something that could do say a 1000 would be > very helpful. > > G723 is a pretty big deal internationally ... I'm even seeing crazy > requests like AMR from folks trying to originate VoIP off of mobile > devices in Europe. > > But I agree just being able to g729 --> ulaw or ulaw --> g729 would > be a great first step (host based or otherwise). > > SDR > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org [4] > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org [5] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org [6] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ------------------------- Msg sent via Enterux Enterprise Email Server : http://www.enterux.com/ Links: ------ [1] mailto:freeswitch-users-bounces at lists.freeswitch.org [2] mailto:freeswitch-users at lists.freeswitch.org [3] mailto:sicfslist at gmail.com [4] mailto:Freeswitch-users at lists.freeswitch.org [5] mailto:Freeswitch-users at lists.freeswitch.org [6] mailto:Freeswitch-users at lists.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090206/cd736ff9/attachment-0001.html From freeswitch-users at digitaldan.com Fri Feb 6 13:13:06 2009 From: freeswitch-users at digitaldan.com (Dan) Date: Fri, 6 Feb 2009 14:13:06 -0700 (MST) Subject: [Freeswitch-users] mod_shout delay in trunk In-Reply-To: <191c3a030902061248r3d307647le5574c1004c6da24@mail.gmail.com> Message-ID: <15850979.181233954786201.JavaMail.root@zimbra> On line 424 I think it needs to be changed from if (!vval || !switch_true(vval)) { to if (!vval || switch_true(vval)) { Other wise it works, thanks! ----- Original Message ----- From: "Anthony Minessale" To: freeswitch-users at lists.freeswitch.org Sent: Friday, February 6, 2009 1:48:31 PM GMT -07:00 US/Canada Mountain Subject: Re: [Freeswitch-users] mod_shout delay in trunk Thanks, We appreciate the positive feedback! if you revert the change I suggested and update i added a new variable enable_file_write_buffering=false set this variable on the channel before you start recording it with the set application or in the dialstring in {} on outbound calls and it should skip the buffering. Could you test it for me and confirm it works? Thank you On Fri, Feb 6, 2009 at 2:36 PM, < freeswitch-users at digitaldan.com > wrote: That worked great! I wanted to say just how awesome Freeswitch is, I have been doing voip related development with SIP since 2000 and this is by far the most well designed piece of voip software I have used or developed on. I currently have a homegrown sip server built on the NIST sip stack with Sun's JMF libraries for RTP processing. 95% of the code and complexity is handling the SIP and RTP sessions, the other 5% is the final application logic and what is most important to me. By letting freeswitch do whats its good at (call routing, sip and media handling) it allows me to focus on what I'm good at (what should we do with those streams, like record them). I have been bragging about this project to anybody who will listen! Dan- ----- Original Message ----- From: "Anthony Minessale" < anthony.minessale at gmail.com > To: freeswitch-users at lists.freeswitch.org Sent: Friday, February 6, 2009 1:07:44 PM GMT -07:00 US/Canada Mountain Subject: Re: [Freeswitch-users] mod_shout delay in trunk edit switch_ivr_play_say.c line 423 comment the line out and recompile. Tell me if it helps you and i will consider making it configurable. On Fri, Feb 6, 2009 at 2:01 PM, < freeswitch-users at digitaldan.com > wrote: For me it is. For what I'm using it for I can tolerate around a second or two delay. I have the icecast server setup to only buffer 1K for their on-connect burst as well as my flash/flex player to only buffer 1k (yes I might as well not buffer at all, which I may end up doing). In 1.0.2 this worked very well. Is this buffer configurable? If not, where is it being set? Thanks Dan- ----- Original Message ----- From: "Brian West" < brian at freeswitch.org > To: freeswitch-users at lists.freeswitch.org Sent: Friday, February 6, 2009 12:47:53 PM GMT -07:00 US/Canada Mountain Subject: Re: [Freeswitch-users] mod_shout delay in trunk Let me clarify.. yes this is normal file buffering was added so we wouldn't thrash your hard drive with tiny bits of data when recording calls so now it buffers and writes larger chunks to disk. This is why you have this delay which is 100% normal.... is realtime a critical thing? It is shout cast so you know it doesn't have to be realtime.. in fact some clients will buffer a little bit anyway and add to it. /b On Feb 6, 2009, at 1:43 PM, freeswitch-users at digitaldan.com wrote: I have, do you know what would have changed between 1.0.2 and trunk that would cause the buffer to change? Also if its not in mod_shout.c (which I copied from 1.0.2 to trunk for testing with no luck), where else would fs be buffering? One thing I have noticed is that in 1.0.2 as soon as the dial plan hits my record statement I see mod_shout logging that it has connected to the icecast server, in trunk it takes about 5 seconds to see the same log mesage. Below is my current svn info Path: . _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/ PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/ PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090206/cf3646ae/attachment.html From msc at freeswitch.org Fri Feb 6 13:16:28 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 6 Feb 2009 13:16:28 -0800 Subject: [Freeswitch-users] Compiling FreeSWITCH for AstLinux In-Reply-To: <2d9149cd0902061220i11b87fd9se253109d7a39249a@mail.gmail.com> References: <2d9149cd0902061220i11b87fd9se253109d7a39249a@mail.gmail.com> Message-ID: <87f2f3b90902061316h5fe6e8afw253c95a55ddf3aa0@mail.gmail.com> > P.S. - Yes, yes I know "AstLinux" isn't the best name for a distro > with FreeSWITCH. Depending on my success here I have some other > ideas... > How about KickAstLinux? ;) -MC From krice at suspicious.org Fri Feb 6 13:18:39 2009 From: krice at suspicious.org (Ken Rice) Date: Fri, 06 Feb 2009 15:18:39 -0600 Subject: [Freeswitch-users] Call accounting - CDR's In-Reply-To: Message-ID: This is built into mod_xml_cdr and should be covered on its wiki page... I personally don't post the records to a web server as I think that's too much over head at this load... I use a script to scrape the directory and process the CDRs... This gives me the FileSystem as a buffer, and allows me to get a ton of info out of the xml cdrs like custom channel variables etc K > From: Nik Middleton > Reply-To: > Date: Fri, 6 Feb 2009 21:07:12 -0000 > To: > Subject: Re: [Freeswitch-users] Call accounting - CDR's > > So you're simply posting this file to a web server? How do you find the load > on it at this rate of calls? > > BTW can anyone point me to resources discussing how to do this? (Using a web > server to post data to a db) I've not this sort of thing before, and I'm not > too sure what I should be goggling for > > > Regards, > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice > Sent: 06 February 2009 17:40 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Call accounting - CDR's > > Mod_xml_cdr will drop a file to the file system on failure to post. You can > also leverage this drop a file to the file system and run a CDR processor > locally. > > We handle call rates in the 500+ range using the local file system as a > caching mechanism and a simple PHP script to rate the CDRs and load them > into a pgsql db. > > > > >> From: Adam Long >> Reply-To: >> Date: Fri, 6 Feb 2009 12:30:34 -0500 >> To: >> Subject: Re: [Freeswitch-users] Call accounting - CDR's >> >> Is mod_cdr_xml asynchronous ... by that I mean .. if there is a communication >> failure how > does this effect load on the system or the call for that >> matter... > it wouldn?t hold up the progress or delay the turn up of the call or >> anything would it? > > I understand it would write to the error directory (but my >> thoughts are what is the impact of > this beyond the obvious IO hit) > > I guess a >> good question is what is more load intensive??? > > 1.) mod_cdr_csv (with batch >> script that loads into DB somewhere) > 2.) mod_cdr_xml (posting to lighttpd on >> remote host inserting into DB) > > I'm thinking about this for a system that >> would be handling in excess of 200-300 call setups > per >> second. > > -Adam > > -----Original Message----- > From: >> freeswitch-users-bounces at lists.freeswitch.org >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rupa >> Schomaker (lists) > Sent: Friday, February 06, 2009 11:27 AM > To: >> freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Call >> accounting - CDR's > > Use the script in scripts/contrib/wasim/ as a starting >> point. > > Basically, you log to csv files, and then the script periodically >> picks > them up and loads to your DB. This is using mysql as an example, >> but > you can do the same with postgres as well. > > If you need realtime inserts, >> then use mod_cdr_xml and have those post > to a script on a webserver that >> parses the xml and inserts into > appropriate tables. This is what I use along >> with a rails app. > > Remember that if you do real time, you also need to >> periodically scrape > the error directory and load those (mod_cdr_xml will save >> to error if it > can't successfully post to your script). > > On 2/6/2009 10:09 AM, >> Nik Middleton wrote: >> Hi Guys >> >> >> >> I?m looking for some pointers on >> how to collect CDR?s and store in >> mysql. Is there anything built in yet? >> >> >> >> >> I can rate the calls as a batch process, I simply need the call >> data. >> >> >> >> Regards >> >> > > > _______________________________________________ > Freeswitch-users mailing >> list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman >> /listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/opt >> ions/freeswitch-users > http://www.freeswitch.org > > > ____________________________ >> ___________________ > Freeswitch-users mailing >> list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman >> /listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/opt >> ions/freeswitch-users > http://www.freeswitch.org > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From freeswitch-users at lists.rupa.com Fri Feb 6 13:24:59 2009 From: freeswitch-users at lists.rupa.com (Rupa Schomaker (lists)) Date: Fri, 06 Feb 2009 15:24:59 -0600 Subject: [Freeswitch-users] Call accounting - CDR's In-Reply-To: References: <030001c98880$9e660f90$db322eb0$@net> Message-ID: <498CAAAB.2040405@lists.rupa.com> I'm doing this on low volume pbx setup, so posting to the web server is fine with my load. If you are doing high load, then definitely write to files and batch process them. On 2/6/2009 3:07 PM, Nik Middleton wrote: > So you're simply posting this file to a web server? How do you find > the load on it at this rate of calls? > > BTW can anyone point me to resources discussing how to do this? > (Using a web server to post data to a db) I've not this sort of thing > before, and I'm not too sure what I should be goggling for > > > Regards, > > > -----Original Message----- From: > freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Ken Rice Sent: 06 February 2009 17:40 To: > freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] > Call accounting - CDR's > > Mod_xml_cdr will drop a file to the file system on failure to post. > You can also leverage this drop a file to the file system and run a > CDR processor locally. > > We handle call rates in the 500+ range using the local file system as > a caching mechanism and a simple PHP script to rate the CDRs and load > them into a pgsql db. > > > > >> From: Adam Long Reply-To: >> Date: Fri, 6 Feb 2009 >> 12:30:34 -0500 To: Subject: >> Re: [Freeswitch-users] Call accounting - CDR's >> >> Is mod_cdr_xml asynchronous ... by that I mean .. if there is a >> communication failure how > does this effect load on the system or the call for that >> matter... > it wouldn?t hold up the progress or delay the turn up of the call or >> anything would it? > > I understand it would write to the error directory (but my >> thoughts are what is the impact of > this beyond the obvious IO hit) > > I guess a >> good question is what is more load intensive??? > > 1.) mod_cdr_csv (with batch >> script that loads into DB somewhere) > 2.) mod_cdr_xml (posting to lighttpd on >> remote host inserting into DB) > > I'm thinking about this for a system that >> would be handling in excess of 200-300 call setups > per >> second. > > -Adam > > -----Original Message----- From: >> freeswitch-users-bounces at lists.freeswitch.org >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of >> Rupa Schomaker (lists) > Sent: Friday, February 06, 2009 11:27 AM To: >> freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Call >> accounting - CDR's > > Use the script in scripts/contrib/wasim/ as a starting >> point. > > Basically, you log to csv files, and then the script periodically >> picks > them up and loads to your DB. This is using mysql as an example, >> but > you can do the same with postgres as well. > > If you need realtime inserts, >> then use mod_cdr_xml and have those post > to a script on a webserver that >> parses the xml and inserts into > appropriate tables. This is what I use along >> with a rails app. > > Remember that if you do real time, you also need to >> periodically scrape > the error directory and load those (mod_cdr_xml will save >> to error if it > can't successfully post to your script). > > On 2/6/2009 10:09 AM, >> Nik Middleton wrote: Hi Guys >> >> >> >> I?m looking for some pointers on how to collect CDR?s and store in >> mysql. Is there anything built in yet? >> >> >> >> >> I can rate the calls as a batch process, I simply need the call >> data. >> >> >> >> Regards >> From sicfslist at gmail.com Fri Feb 6 13:30:13 2009 From: sicfslist at gmail.com (Shelby Ramsey) Date: Fri, 6 Feb 2009 15:30:13 -0600 Subject: [Freeswitch-users] Call accounting - CDR's In-Reply-To: References: Message-ID: <35b355e90902061330t34c22f27x11013de60b101c76@mail.gmail.com> Just out of curiosity ... You actually set the values in xml_cdr_conf.xml to an invalid value ... and then FS tries it and then dumps it into the err_dir? Nik, Just configure the xml_conf_cdr and it will post all of the channel variables to your web server ... you can look at the variables and see what you want. Or I actually like Ken's suggestion ... that makes a lot of sense ... same benefit of having all of the channel variables ... no overhead. SDR -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090206/d1bc6461/attachment.html From krice at freeswitch.org Fri Feb 6 13:33:13 2009 From: krice at freeswitch.org (Ken Rice) Date: Fri, 06 Feb 2009 15:33:13 -0600 Subject: [Freeswitch-users] Call accounting - CDR's In-Reply-To: <35b355e90902061330t34c22f27x11013de60b101c76@mail.gmail.com> Message-ID: Nope you comment out that line and it wont even attempt to post and will drop it into the log/xml_cdr directory From: Shelby Ramsey Reply-To: Date: Fri, 6 Feb 2009 15:30:13 -0600 To: Subject: Re: [Freeswitch-users] Call accounting - CDR's Just out of curiosity ... You actually set the values in xml_cdr_conf.xml to an invalid value ... and then FS tries it and then dumps it into the err_dir? Nik, Just configure the xml_conf_cdr and it will post all of the channel variables to your web server ... you can look at the variables and see what you want. Or I actually like Ken's suggestion ... that makes a lot of sense ... same benefit of having all of the channel variables ... no overhead. SDR _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090206/7cc7417b/attachment.html From jalsot at gmail.com Fri Feb 6 13:36:51 2009 From: jalsot at gmail.com (Tamas) Date: Fri, 06 Feb 2009 22:36:51 +0100 Subject: [Freeswitch-users] mod_shout delay in trunk In-Reply-To: <191c3a030902061248r3d307647le5574c1004c6da24@mail.gmail.com> References: <15830211.131233952452225.JavaMail.root@zimbra> <28165623.151233952574612.JavaMail.root@zimbra> <191c3a030902061248r3d307647le5574c1004c6da24@mail.gmail.com> Message-ID: <498CAD73.2040709@gmail.com> Hello, could this option be used to lower I/O load - to rather write more bytes at once rather than one by one - on file recording (record_session)? Regards, Tamas Anthony Minessale ?rta: > Thanks, > > We appreciate the positive feedback! > > if you revert the change I suggested and update i added a new variable > > enable_file_write_buffering=false > > set this variable on the channel before you start recording it with > the set application or in the dialstring in {} > on outbound calls and it should skip the buffering. > > Could you test it for me and confirm it works? > > Thank you > > > On Fri, Feb 6, 2009 at 2:36 PM, > wrote: > > That worked great! > > I wanted to say just how awesome Freeswitch is, I have been doing > voip related development with SIP since 2000 and this is by far > the most well designed piece of voip software I have used or > developed on. I currently have a homegrown sip server built on > the NIST sip stack with Sun's JMF libraries for RTP processing. > 95% of the code and complexity is handling the SIP and RTP > sessions, the other 5% is the final application logic and what is > most important to me. By letting freeswitch do whats its good at > (call routing, sip and media handling) it allows me to focus on > what I'm good at (what should we do with those streams, like > record them). I have been bragging about this project to anybody > who will listen! > > Dan- > > ----- Original Message ----- > From: "Anthony Minessale" > > To: freeswitch-users at lists.freeswitch.org > > Sent: Friday, February 6, 2009 1:07:44 PM GMT -07:00 US/Canada > Mountain > Subject: Re: [Freeswitch-users] mod_shout delay in trunk > > edit switch_ivr_play_say.c line 423 > > comment the line out and recompile. > Tell me if it helps you and i will consider making it configurable. > > > On Fri, Feb 6, 2009 at 2:01 PM, > wrote: > > For me it is. For what I'm using it for I can tolerate around > a second or two delay. I have the icecast server setup to > only buffer 1K for their on-connect burst as well as my > flash/flex player to only buffer 1k (yes I might as well not > buffer at all, which I may end up doing). In 1.0.2 this > worked very well. Is this buffer configurable? If not, where > is it being set? > > Thanks > Dan- > > ----- Original Message ----- > From: "Brian West" > > To: freeswitch-users at lists.freeswitch.org > > Sent: Friday, February 6, 2009 12:47:53 PM GMT -07:00 > US/Canada Mountain > Subject: Re: [Freeswitch-users] mod_shout delay in trunk > > Let me clarify.. yes this is normal file buffering was added > so we wouldn't thrash your hard drive with tiny bits of data > when recording calls so now it buffers and writes larger > chunks to disk. This is why you have this delay which is 100% > normal.... is realtime a critical thing? It is shout cast so > you know it doesn't have to be realtime.. in fact some clients > will buffer a little bit anyway and add to it. > > /b > > On Feb 6, 2009, at 1:43 PM, freeswitch-users at digitaldan.com > wrote: > > I have, do you know what would have changed between 1.0.2 > and trunk that would cause the buffer to change? Also if > its not in mod_shout.c (which I copied from 1.0.2 to trunk > for testing with no luck), where else would fs be > buffering? One thing I have noticed is that in 1.0.2 as > soon as the dial plan hits my record statement I see > mod_shout logging that it has connected to the icecast > server, in trunk it takes about 5 seconds to see the same > log mesage. Below is my current svn info > Path: . > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > > _______________________________________________ Freeswitch-users > mailing list Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From msc at freeswitch.org Fri Feb 6 13:36:39 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 6 Feb 2009 13:36:39 -0800 Subject: [Freeswitch-users] mod_shout delay in trunk In-Reply-To: <15850979.181233954786201.JavaMail.root@zimbra> References: <191c3a030902061248r3d307647le5574c1004c6da24@mail.gmail.com> <15850979.181233954786201.JavaMail.root@zimbra> Message-ID: <87f2f3b90902061336s5bbbdce3o921023b83590df88@mail.gmail.com> > if you revert the change I suggested and update i added a new variable > > enable_file_write_buffering=false > > set this variable on the channel before you start recording it with the set > application or in the dialstring in {} > on outbound calls and it should skip the buffering. > > Could you test it for me and confirm it works? > > Thank you Also, as payment for services rendered could you please add this variable and description to the wiki? http://wiki.freeswitch.org/wiki/Channel_Variables Thanks! -MC From mgg at giagnocavo.net Fri Feb 6 13:38:17 2009 From: mgg at giagnocavo.net (Michael Giagnocavo) Date: Fri, 6 Feb 2009 16:38:17 -0500 Subject: [Freeswitch-users] Transcoding G723 In-Reply-To: <49796.1233961333@enterux.com> References: <49796.1233961333@enterux.com> Message-ID: <6E8D2069C08AA84A83D336E996AE4C6702362F6C4C@mse17be1.mse17.exchange.ms> Sure ?Custom CPU? isn?t just a 4 socket Intel setup? -Michael From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mitul Limbani Sent: Friday, February 06, 2009 4:02 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Transcoding G723 Hello guys, Sorry it was my mistake, i re-read the entire proposal and it looks like their specialized 1U Hardware with custom CPU can handle close to 1500 simultaneous G729 encoding n transmission, also this hardware offers from any codec to any codec transcoding path. Also this box easily works over FreeSWITCH, Asterisk, RTPProxy, OpenSIPS are their claims. and their pricing are First year ?16,000 Second year ?13,000/annum recurring This is found slightly on higher side though, Thanks & Regards, Mitul Limbani, Founder & CEO, Enterux Solutions, The Enterprise Linux Company (TM), www.enterux.com +91-9820332422 On Sat 07/02/09 02:00 , Michael Giagnocavo mgg at giagnocavo.net sent: $22K would buy quite a few machines with many core Xeons. I just don't see how it'd be effective at that price. Not to mention a yearly figure. The only thing to take into consideration would be the G729 licenses. But in bulk, the price should be pretty effective, even figuring in hardware. (Not to mention what things like Larrabee will mean for encoding -- 32 1.5GHz "Pentium 4 x64" cores, each with 4 threads?) And... if you're running Asterisk, um, isn't 5000 channels a wee bit over what you can handle anyways? ;) -Michael -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mitul Limbani Sent: Friday, February 06, 2009 12:28 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Transcoding G723 Hi, I came across a company who was selling hardware which could do like 5000 G729 conversion simultanoeusly, I was like this sounds cool, they have support for asterisk, I haven't enquirer yet how they do this, but anyone wishes to buy it cost ?15000/year for the hrdware + support Any one on the list requires such high throughput calls can connect with me so that we can really test if it can work with freeswitch. Regards, Mitul Limbani, Founder & CEO, Enterux Solutions Pvt Ltd, The Enterprise Linux Company(r), http://www.enterux.com/ On 06-Feb-09, at 21:58, Shelby Ramsey > wrote: > Steve, > > You definitely have a better grasp on this topic than me. But I > think it's a tough sell on the host based processing ... when you > look at products like what audio codes can do on a card (3 DS-3's > worth of transcoding) ... but I have had a couple of soft switch > vendors claim though that they could do 2,000 calls per host (but I > seriously doubt it). > > I agree that producing a card that does 120 channels is pretty > worthless ... but having something that could do say a 1000 would be > very helpful. > > G723 is a pretty big deal internationally ... I'm even seeing crazy > requests like AMR from folks trying to originate VoIP off of mobile > devices in Europe. > > But I agree just being able to g729 --> ulaw or ulaw --> g729 would > be a great first step (host based or otherwise). > > SDR > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ________________________________ Msg sent via Enterux Enterprise Email Server : http://www.enterux.com/ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090206/adc47ed6/attachment-0001.html From sicfslist at gmail.com Fri Feb 6 13:39:02 2009 From: sicfslist at gmail.com (Shelby Ramsey) Date: Fri, 6 Feb 2009 15:39:02 -0600 Subject: [Freeswitch-users] Call accounting - CDR's In-Reply-To: References: <35b355e90902061330t34c22f27x11013de60b101c76@mail.gmail.com> Message-ID: <35b355e90902061339w2084ddfbvbe7ed95d11b438ea@mail.gmail.com> Even better ... Thanks Ken! SDR -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090206/7291a097/attachment.html From freeswitch-users at digitaldan.com Fri Feb 6 13:54:07 2009 From: freeswitch-users at digitaldan.com (Dan) Date: Fri, 6 Feb 2009 14:54:07 -0700 (MST) Subject: [Freeswitch-users] mod_shout delay in trunk In-Reply-To: <87f2f3b90902061336s5bbbdce3o921023b83590df88@mail.gmail.com> Message-ID: <9128420.211233957247696.JavaMail.root@zimbra> Done! ----- Original Message ----- From: "Michael Collins" To: freeswitch-users at lists.freeswitch.org Sent: Friday, February 6, 2009 2:36:39 PM GMT -07:00 US/Canada Mountain Subject: Re: [Freeswitch-users] mod_shout delay in trunk > if you revert the change I suggested and update i added a new variable > > enable_file_write_buffering=false > > set this variable on the channel before you start recording it with the set > application or in the dialstring in {} > on outbound calls and it should skip the buffering. > > Could you test it for me and confirm it works? > > Thank you Also, as payment for services rendered could you please add this variable and description to the wiki? http://wiki.freeswitch.org/wiki/Channel_Variables Thanks! -MC _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090206/50f4be53/attachment.html From msc at freeswitch.org Fri Feb 6 14:04:36 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 6 Feb 2009 14:04:36 -0800 Subject: [Freeswitch-users] mod_shout delay in trunk In-Reply-To: <9128420.211233957247696.JavaMail.root@zimbra> References: <87f2f3b90902061336s5bbbdce3o921023b83590df88@mail.gmail.com> <9128420.211233957247696.JavaMail.root@zimbra> Message-ID: <87f2f3b90902061404v74ff9552sbb8a0464c18c65bc@mail.gmail.com> > Done! Many thanks! From nik.middleton at noblesolutions.co.uk Fri Feb 6 14:28:36 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Fri, 6 Feb 2009 22:28:36 -0000 Subject: [Freeswitch-users] Call accounting - CDR's In-Reply-To: <35b355e90902061339w2084ddfbvbe7ed95d11b438ea@mail.gmail.com> References: <35b355e90902061330t34c22f27x11013de60b101c76@mail.gmail.com> <35b355e90902061339w2084ddfbvbe7ed95d11b438ea@mail.gmail.com> Message-ID: Guy's, Thanks for all the responses; it's truly refreshing to get so much valuable input. I'm reading the docs furiously, but I still don't know what I don't know yet. But given time I will return the favor to those that come later. Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090206/37e6dfb6/attachment.html From msc at freeswitch.org Fri Feb 6 14:42:25 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 6 Feb 2009 14:42:25 -0800 Subject: [Freeswitch-users] Call accounting - CDR's In-Reply-To: References: <35b355e90902061330t34c22f27x11013de60b101c76@mail.gmail.com> <35b355e90902061339w2084ddfbvbe7ed95d11b438ea@mail.gmail.com> Message-ID: <87f2f3b90902061442s2d1ef36fq5199a6766c083581@mail.gmail.com> > > Thanks for all the responses; it's truly refreshing to get so much valuable > input. I'm reading the docs furiously, but I still don't know what I don't > know yet. But given time I will return the favor to those that come later. Sounds good! If you feel up to doing any wiki documentation please let me know and I'll be happy to offer some pointers, etc. -MC From nik.middleton at noblesolutions.co.uk Fri Feb 6 15:20:07 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Fri, 6 Feb 2009 23:20:07 -0000 Subject: [Freeswitch-users] Call accounting - CDR's In-Reply-To: <87f2f3b90902061442s2d1ef36fq5199a6766c083581@mail.gmail.com> References: <35b355e90902061330t34c22f27x11013de60b101c76@mail.gmail.com><35b355e90902061339w2084ddfbvbe7ed95d11b438ea@mail.gmail.com> <87f2f3b90902061442s2d1ef36fq5199a6766c083581@mail.gmail.com> Message-ID: More than happy to add my 2 cents worth when I have something useful to say Question regarding the xml cdr's Let's say I have a cron job looking at these files and processing them. How does FS create them. Does a MV occur from some other DIR, as otherwise it's possible I might try and open an in progress record. Regards -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 06 February 2009 22:42 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Call accounting - CDR's > > Thanks for all the responses; it's truly refreshing to get so much valuable > input. I'm reading the docs furiously, but I still don't know what I don't > know yet. But given time I will return the favor to those that come later. Sounds good! If you feel up to doing any wiki documentation please let me know and I'll be happy to offer some pointers, etc. -MC _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From gkuri at ieee.org Fri Feb 6 15:19:49 2009 From: gkuri at ieee.org (Gabriel Kuri) Date: Fri, 06 Feb 2009 15:19:49 -0800 Subject: [Freeswitch-users] Separate NICs for Performance Message-ID: <498CC595.8070601@ieee.org> Hey Folks: For a FS box that's generally handling higher amounts of inbound/outbound call traffic (say 500 - 700 calls) and registrations (30 - 50 per second), is it recommended to split off the signaling and media traffic onto separate NICs for performance reasons? Or is it better to split all the traffic for the phones and outside traffic to carriers into separate profiles and split those profiles onto separate NICs? or perhaps none of the above and there's still a better solution I didn't mention ;) ? Thanks, Gabe From brian at freeswitch.org Fri Feb 6 15:28:36 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 6 Feb 2009 17:28:36 -0600 Subject: [Freeswitch-users] Separate NICs for Performance In-Reply-To: <498CC595.8070601@ieee.org> References: <498CC595.8070601@ieee.org> Message-ID: <70153680-883F-4A6D-9D78-83FFE65C239B@freeswitch.org> If the nic's have their own bus you could do that to improve network performance of sip signaling and media... now the neat thing would be to have an option to have say three network interfaces and have FreeSWITCH round robin them per call. I smell a bounty. /b On Feb 6, 2009, at 5:19 PM, Gabriel Kuri wrote: > Hey Folks: > > For a FS box that's generally handling higher amounts of > inbound/outbound call traffic (say 500 - 700 calls) and registrations > (30 - 50 per second), is it recommended to split off the signaling and > media traffic onto separate NICs for performance reasons? Or is it > better to split all the traffic for the phones and outside traffic to > carriers into separate profiles and split those profiles onto separate > NICs? or perhaps none of the above and there's still a better > solution I > didn't mention ;) ? > > Thanks, > > Gabe From msc at freeswitch.org Fri Feb 6 16:00:11 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 6 Feb 2009 16:00:11 -0800 Subject: [Freeswitch-users] Call accounting - CDR's In-Reply-To: References: <35b355e90902061330t34c22f27x11013de60b101c76@mail.gmail.com> <35b355e90902061339w2084ddfbvbe7ed95d11b438ea@mail.gmail.com> <87f2f3b90902061442s2d1ef36fq5199a6766c083581@mail.gmail.com> Message-ID: <87f2f3b90902061600i5e3ce93esfe883b36afa6cc1c@mail.gmail.com> > Question regarding the xml cdr's > > Let's say I have a cron job looking at these files and processing them. > How does FS create them. Does a MV occur from some other DIR, as > otherwise it's possible I might try and open an in progress record. No worries - the file isn't "available" until it's ready to go, just like being mv'd into the dir. -MC From kristian.kielhofner at gmail.com Fri Feb 6 16:00:50 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Fri, 6 Feb 2009 19:00:50 -0500 Subject: [Freeswitch-users] Separate NICs for Performance In-Reply-To: <70153680-883F-4A6D-9D78-83FFE65C239B@freeswitch.org> References: <498CC595.8070601@ieee.org> <70153680-883F-4A6D-9D78-83FFE65C239B@freeswitch.org> Message-ID: <2d9149cd0902061600k1515998t8adf3dcc84c1ce21@mail.gmail.com> Bonding? Intel ANS? On Fri, Feb 6, 2009 at 6:28 PM, Brian West wrote: > If the nic's have their own bus you could do that to improve network > performance of sip signaling and media... now the neat thing would be > to have an option to have say three network interfaces and have > FreeSWITCH round robin them per call. I smell a bounty. > > /b -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From brian at freeswitch.org Fri Feb 6 16:04:37 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 6 Feb 2009 18:04:37 -0600 Subject: [Freeswitch-users] Separate NICs for Performance In-Reply-To: <2d9149cd0902061600k1515998t8adf3dcc84c1ce21@mail.gmail.com> References: <498CC595.8070601@ieee.org> <70153680-883F-4A6D-9D78-83FFE65C239B@freeswitch.org> <2d9149cd0902061600k1515998t8adf3dcc84c1ce21@mail.gmail.com> Message-ID: <6BDE6A1E-673D-4B01-B8C0-99F2E7C771CC@freeswitch.org> That would work too I suspect. /b On Feb 6, 2009, at 6:00 PM, Kristian Kielhofner wrote: > Bonding? Intel ANS? > > On Fri, Feb 6, 2009 at 6:28 PM, Brian West > wrote: >> If the nic's have their own bus you could do that to improve network >> performance of sip signaling and media... now the neat thing would be >> to have an option to have say three network interfaces and have >> FreeSWITCH round robin them per call. I smell a bounty. >> >> /b > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090206/b8a9a8db/attachment-0001.html From kristian.kielhofner at gmail.com Fri Feb 6 16:06:56 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Fri, 6 Feb 2009 19:06:56 -0500 Subject: [Freeswitch-users] Separate NICs for Performance In-Reply-To: <6BDE6A1E-673D-4B01-B8C0-99F2E7C771CC@freeswitch.org> References: <498CC595.8070601@ieee.org> <70153680-883F-4A6D-9D78-83FFE65C239B@freeswitch.org> <2d9149cd0902061600k1515998t8adf3dcc84c1ce21@mail.gmail.com> <6BDE6A1E-673D-4B01-B8C0-99F2E7C771CC@freeswitch.org> Message-ID: <2d9149cd0902061606g61913108ree46ed95087c292e@mail.gmail.com> I think the big problem is still going to be interrupts and other networking stuff. On Fri, Feb 6, 2009 at 7:04 PM, Brian West wrote: > That would work too I suspect. > /b -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From brian at freeswitch.org Fri Feb 6 16:14:11 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 6 Feb 2009 18:14:11 -0600 Subject: [Freeswitch-users] Separate NICs for Performance In-Reply-To: <2d9149cd0902061606g61913108ree46ed95087c292e@mail.gmail.com> References: <498CC595.8070601@ieee.org> <70153680-883F-4A6D-9D78-83FFE65C239B@freeswitch.org> <2d9149cd0902061600k1515998t8adf3dcc84c1ce21@mail.gmail.com> <6BDE6A1E-673D-4B01-B8C0-99F2E7C771CC@freeswitch.org> <2d9149cd0902061606g61913108ree46ed95087c292e@mail.gmail.com> Message-ID: <483DA153-9427-4681-8872-770F4E0DF5BF@freeswitch.org> Wouldn't be a huge deal if each card has a dedicated bus then it wouldn't be fighting for bandwidth... its still going to be hitting a limit at some point but you might get more milage out of it. /b On Feb 6, 2009, at 6:06 PM, Kristian Kielhofner wrote: > I think the big problem is still going to be interrupts and other > networking stuff. From mkarp at securesilence.com Fri Feb 6 16:32:42 2009 From: mkarp at securesilence.com (Maxim Karp) Date: Fri, 6 Feb 2009 16:32:42 -0800 Subject: [Freeswitch-users] Voicemail prompts and playback speed Message-ID: <009301c988bb$99adb5d0$cd092170$@com> Hello all, Can anyone please let me know how I might be able to configure the voice mail prompts and their playback speed? Thanks, Maxim. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090206/d95402a0/attachment.html From mike at jerris.com Fri Feb 6 18:27:48 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 6 Feb 2009 21:27:48 -0500 Subject: [Freeswitch-users] Voicemail prompts and playback speed In-Reply-To: <009301c988bb$99adb5d0$cd092170$@com> References: <009301c988bb$99adb5d0$cd092170$@com> Message-ID: Not sure what you mean by playback speed. All the prompts for voicemail are defined in the phrase macros in the configuration. Mike On Feb 6, 2009, at 7:32 PM, "Maxim Karp" wrote: > Hello all, > > > > Can anyone please let me know how I might be able to configure the > voice mail prompts and their playback speed? > > > > Thanks, > > > > Maxim. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From msc at freeswitch.org Fri Feb 6 20:26:33 2009 From: msc at freeswitch.org (Michael S Collins) Date: Fri, 6 Feb 2009 20:26:33 -0800 Subject: [Freeswitch-users] Voicemail prompts and playback speed Message-ID: On Feb 6, 2009, at 6:27 PM, Michael Jerris wrote: > Not sure what you mean by playback speed. All the prompts for > voicemail are defined in the phrase macros in the configuration. Check out conf/lang/en/vm/sounds.xml -MC From uv at yuvalhertzog.com Sat Feb 7 01:39:57 2009 From: uv at yuvalhertzog.com (UV) Date: Sat, 7 Feb 2009 20:39:57 +1100 Subject: [Freeswitch-users] Outgoing registration expiry Message-ID: I?m trying to set up a default provider via the example.com.xml using the variables set in vars.xml. The provider has a registration expiry of 120 seconds and I?m trying to set it up to register every 60 seconds but when I change the ?expire-seconds? variable (in directory/default/example.com.xml), it doesn?t seem to have any effect. Actually, doesn?t matter how long I wait, it doesn?t seem to re-register at all? Any idea what I?m missing? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090207/ecee7ffe/attachment.html From uv at yuvalhertzog.com Sat Feb 7 01:40:09 2009 From: uv at yuvalhertzog.com (UV) Date: Sat, 7 Feb 2009 20:40:09 +1100 Subject: [Freeswitch-users] Global Variables forgotten through the public context? Message-ID: <0B4E2726927041D09D0425DA0242C805@UVix> Another question: When I try routing calls through the public context to the default context, global variables (set in vars.xml) seem to be ?forgotten? and appear blank. I?m trying a very simple scenario of an incoming call on the public context routed to a phone number which is rightfully captured at ?local.example.com? but all three variables ${outbound_caller_id_number}, ${outbound_caller_id_name} and ${default_gateway} appear to be blanked out (according to the console debug output). Again, any idea what am I doing wrong? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090207/3ca9c5a5/attachment.html From brian at freeswitch.org Sat Feb 7 01:50:30 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 7 Feb 2009 03:50:30 -0600 Subject: [Freeswitch-users] Outgoing registration expiry In-Reply-To: References: Message-ID: <2EE57DC0-CF8B-4E56-B383-9FBFAA60CD7F@freeswitch.org> You would have to reload and restart the profile for that to take effect. You can't change the global and have it magically start using the new value. /b On Feb 7, 2009, at 3:39 AM, UV wrote: > I?m trying to set up a default provider via the example.com.xml > using the variables set in vars.xml. > The provider has a registration expiry of 120 seconds and I?m trying > to set it up to register every 60 seconds but when I change the > ?expire-seconds? variable (in directory/default/example.com.xml), it > doesn?t seem to have any effect. Actually, doesn?t matter how long I > wait, it doesn?t seem to re-register at all? > > Any idea what I?m missing? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090207/d21260cc/attachment.html From brian at freeswitch.org Sat Feb 7 01:51:04 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 7 Feb 2009 03:51:04 -0600 Subject: [Freeswitch-users] Global Variables forgotten through the public context? In-Reply-To: <0B4E2726927041D09D0425DA0242C805@UVix> References: <0B4E2726927041D09D0425DA0242C805@UVix> Message-ID: <9D0CFA79-C120-4D66-B48C-7A5641B07AF4@freeswitch.org> No the vars are there can you provide more detail on what exactly you're doing? The default config uses the call_debug variable and its a global set in vars.xml. /b On Feb 7, 2009, at 3:40 AM, UV wrote: > Another question: > When I try routing calls through the public context to the default > context, global variables (set in vars.xml) seem to be ?forgotten? > and appear blank. > I?m trying a very simple scenario of an incoming call on the public > context routed to a phone number which is rightfully captured at > ?local.example.com? but all three variables $ > {outbound_caller_id_number}, ${outbound_caller_id_name} and $ > {default_gateway} appear to be blanked out (according to the console > debug output). > > Again, any idea what am I doing wrong? > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090207/d9014d47/attachment-0001.html From nik.middleton at noblesolutions.co.uk Sat Feb 7 02:07:34 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Sat, 7 Feb 2009 10:07:34 -0000 Subject: [Freeswitch-users] Call accounting - CDR's In-Reply-To: <87f2f3b90902061600i5e3ce93esfe883b36afa6cc1c@mail.gmail.com> References: <35b355e90902061330t34c22f27x11013de60b101c76@mail.gmail.com><35b355e90902061339w2084ddfbvbe7ed95d11b438ea@mail.gmail.com><87f2f3b90902061442s2d1ef36fq5199a6766c083581@mail.gmail.com> <87f2f3b90902061600i5e3ce93esfe883b36afa6cc1c@mail.gmail.com> Message-ID: Great, thanks for that. One of the big issues with Asterisk's way of billing is that if let's say a remote phone diverts a call to another number, say a mobile, because a local channel is created for the redirect, Asterisk loses critical information such as the account code and therefore cannot be billed for. I'm going to try this with FS today and see what happens. It would be awesome if it is accountable. That way I wouldn't have to force the user to do diverts from a web page Regards, -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 07 February 2009 00:00 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Call accounting - CDR's > Question regarding the xml cdr's > > Let's say I have a cron job looking at these files and processing them. > How does FS create them. Does a MV occur from some other DIR, as > otherwise it's possible I might try and open an in progress record. No worries - the file isn't "available" until it's ready to go, just like being mv'd into the dir. -MC _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From nik.middleton at noblesolutions.co.uk Sat Feb 7 03:38:15 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Sat, 7 Feb 2009 11:38:15 -0000 Subject: [Freeswitch-users] AMD Functionality Message-ID: Hi Guys, Is there any form of Answer phone detection in FS? A search hasn't really brought up anything Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090207/b52a1b5b/attachment.html From uv at yuvalhertzog.com Sat Feb 7 05:14:34 2009 From: uv at yuvalhertzog.com (UV) Date: Sun, 8 Feb 2009 00:14:34 +1100 Subject: [Freeswitch-users] Outgoing registration expiry In-Reply-To: <2EE57DC0-CF8B-4E56-B383-9FBFAA60CD7F@freeswitch.org> References: <2EE57DC0-CF8B-4E56-B383-9FBFAA60CD7F@freeswitch.org> Message-ID: <057607FE113648A0B6D0DA614BDD3244@UVix> Obviously. XML was reloaded, sofia profiles were reloaded, freeswitch app was shutdown and restarted and last but not least, I?ve rebooted the computer several times just to make sure :-) No, seriously, I?ve done everything to verify the settings are there ? it?s just not re-registering. My question is does the ?expire-seconds? really work on per second interval? Value 5 equals to gateway being registered every 5 seconds? Because it doesn?t? Am I the only one to have this problem? Am I using it wrong? _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Saturday, February 07, 2009 8:51 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Outgoing registration expiry You would have to reload and restart the profile for that to take effect. You can't change the global and have it magically start using the new value. /b On Feb 7, 2009, at 3:39 AM, UV wrote: I?m trying to set up a default provider via the example.com.xml using the variables set in vars.xml. The provider has a registration expiry of 120 seconds and I?m trying to set it up to register every 60 seconds but when I change the ?expire-seconds? variable (in directory/default/example.com.xml), it doesn?t seem to have any effect. Actually, doesn?t matter how long I wait, it doesn?t seem to re-register at all? Any idea what I?m missing? No virus found in this incoming message. Checked by AVG i www.avg.com Version: 8.0.233 / Virus Database: 270.10.18/1937 i Release Date: 02/06/09 11:31:00 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090208/0ad85243/attachment.html From uv at yuvalhertzog.com Sat Feb 7 05:30:46 2009 From: uv at yuvalhertzog.com (UV) Date: Sun, 8 Feb 2009 00:30:46 +1100 Subject: [Freeswitch-users] Global Variables forgotten through thepublic context? In-Reply-To: <9D0CFA79-C120-4D66-B48C-7A5641B07AF4@freeswitch.org> References: <0B4E2726927041D09D0425DA0242C805@UVix> <9D0CFA79-C120-4D66-B48C-7A5641B07AF4@freeswitch.org> Message-ID: <5B1ED1D4834B4719B309FAC03F1595C0@UVix> I took the out-of-the-box public context dialplan, added an entry to dial a 10 digit number through the default context and when it ran, I noticed in the console log that all the values below are either null or empty. To be more specific: In the public context I?ve added this extension in the beginning: If you want, I can pastebin the log for you to see. It would be easier for you to replicate it yourself and see. It?s quite easy to replicate. _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Saturday, February 07, 2009 8:51 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Global Variables forgotten through thepublic context? No the vars are there can you provide more detail on what exactly you're doing? The default config uses the call_debug variable and its a global set in vars.xml. /b On Feb 7, 2009, at 3:40 AM, UV wrote: Another question: When I try routing calls through the public context to the default context, global variables (set in vars.xml) seem to be ?forgotten? and appear blank. I?m trying a very simple scenario of an incoming call on the public context routed to a phone number which is rightfully captured at ?local.example.com? but all three variables ${outbound_caller_id_number}, ${outbound_caller_id_name} and ${default_gateway} appear to be blanked out (according to the console debug output). Again, any idea what am I doing wrong? No virus found in this incoming message. Checked by AVG i www.avg.com Version: 8.0.233 / Virus Database: 270.10.18/1937 i Release Date: 02/06/09 11:31:00 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090208/59771189/attachment-0001.html From brian at freeswitch.org Sat Feb 7 05:32:08 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 7 Feb 2009 07:32:08 -0600 Subject: [Freeswitch-users] Outgoing registration expiry In-Reply-To: <057607FE113648A0B6D0DA614BDD3244@UVix> References: <2EE57DC0-CF8B-4E56-B383-9FBFAA60CD7F@freeswitch.org> <057607FE113648A0B6D0DA614BDD3244@UVix> Message-ID: <465CCF29-33B5-4A93-A322-EE71270F8BE6@freeswitch.org> turn on the sofia debug. on the profile. I will suspect that the far end proxy is forcing your expire to a higher number. /b On Feb 7, 2009, at 7:14 AM, UV wrote: > Obviously. XML was reloaded, sofia profiles were reloaded, > freeswitch app was shutdown and restarted and last but not least, > I?ve rebooted the computer several times just to make sure J > No, seriously, I?ve done everything to verify the settings are there > ? it?s just not re-registering. My question is does the ?expire- > seconds? really work on per second interval? Value 5 equals to > gateway being registered every 5 seconds? > Because it doesn?t? Am I the only one to have this problem? Am I > using it wrong? > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090207/4a693178/attachment.html From brian at freeswitch.org Sat Feb 7 05:32:43 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 7 Feb 2009 07:32:43 -0600 Subject: [Freeswitch-users] Global Variables forgotten through thepublic context? In-Reply-To: <5B1ED1D4834B4719B309FAC03F1595C0@UVix> References: <0B4E2726927041D09D0425DA0242C805@UVix> <9D0CFA79-C120-4D66-B48C-7A5641B07AF4@freeswitch.org> <5B1ED1D4834B4719B309FAC03F1595C0@UVix> Message-ID: Please do... also make sure you have a context param on your gateway. /b On Feb 7, 2009, at 7:30 AM, UV wrote: > I took the out-of-the-box public context dialplan, added an entry to > dial a 10 digit number through the default context and when it ran, > I noticed in the console log that all the values below are either > null or empty. > > To be more specific: > In the public context I?ve added this extension in the beginning: > > > > > > > If you want, I can pastebin the log for you to see. It would be > easier for you to replicate it yourself and see. It?s quite easy to > replicate. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090207/def372e5/attachment.html From sicfslist at gmail.com Sat Feb 7 06:40:14 2009 From: sicfslist at gmail.com (Shelby Ramsey) Date: Sat, 7 Feb 2009 08:40:14 -0600 Subject: [Freeswitch-users] AMD Functionality In-Reply-To: References: Message-ID: <35b355e90902070640p4811b4fy88796dd62ee0c306@mail.gmail.com> Nik, Right now there is mod_vmd. It sets the channel variable vmd_detect if it detects a beep. If a beep is detected it will set vmd_detect=TRUE. If no beep is detected then it won't do anything. Example of usage as follows (with the outcome being hangup if answering machine is detected): And Ken Rice also has a mod that can be licensed that probably is a better solution and works much better. Hope this helps. SDR -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090207/5340510b/attachment.html From uv at yuvalhertzog.com Sat Feb 7 07:24:44 2009 From: uv at yuvalhertzog.com (UV) Date: Sun, 8 Feb 2009 02:24:44 +1100 Subject: [Freeswitch-users] Outgoing registration expiry In-Reply-To: <465CCF29-33B5-4A93-A322-EE71270F8BE6@freeswitch.org> References: <2EE57DC0-CF8B-4E56-B383-9FBFAA60CD7F@freeswitch.org><057607FE113648A0B6D0DA614BDD3244@UVix> <465CCF29-33B5-4A93-A322-EE71270F8BE6@freeswitch.org> Message-ID: ?. And correct you are! Far end proxy does force to a higher number? Thanks! _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Sunday, February 08, 2009 12:32 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Outgoing registration expiry turn on the sofia debug. on the profile. I will suspect that the far end proxy is forcing your expire to a higher number. /b On Feb 7, 2009, at 7:14 AM, UV wrote: Obviously. XML was reloaded, sofia profiles were reloaded, freeswitch app was shutdown and restarted and last but not least, I?ve rebooted the computer several times just to make sure :-) No, seriously, I?ve done everything to verify the settings are there ? it?s just not re-registering. My question is does the ?expire-seconds? really work on per second interval? Value 5 equals to gateway being registered every 5 seconds? Because it doesn?t? Am I the only one to have this problem? Am I using it wrong? No virus found in this incoming message. Checked by AVG i www.avg.com Version: 8.0.233 / Virus Database: 270.10.18/1937 i Release Date: 02/06/09 11:31:00 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090208/0a4cad18/attachment-0001.html From uv at yuvalhertzog.com Sat Feb 7 07:26:47 2009 From: uv at yuvalhertzog.com (UV) Date: Sun, 8 Feb 2009 02:26:47 +1100 Subject: [Freeswitch-users] Global Variables forgotten through thepubliccontext? In-Reply-To: References: <0B4E2726927041D09D0425DA0242C805@UVix><9D0CFA79-C120-4D66-B48C-7A5641B07AF4@freeswitch.org><5B1ED1D4834B4719B309FAC03F1595C0@UVix> Message-ID: Will do. Now I have a little problem delaying me as the latest build changed something with the sound file playing and now the FS can?t find any local file to play (adds mysterious \16000\ to the file location?). I?ll try to isolate this first then get back to this issue. _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Sunday, February 08, 2009 12:33 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Global Variables forgotten through thepubliccontext? Please do... also make sure you have a context param on your gateway. /b On Feb 7, 2009, at 7:30 AM, UV wrote: I took the out-of-the-box public context dialplan, added an entry to dial a 10 digit number through the default context and when it ran, I noticed in the console log that all the values below are either null or empty. To be more specific: In the public context I?ve added this extension in the beginning: If you want, I can pastebin the log for you to see. It would be easier for you to replicate it yourself and see. It?s quite easy to replicate. No virus found in this incoming message. Checked by AVG i www.avg.com Version: 8.0.233 / Virus Database: 270.10.18/1937 i Release Date: 02/06/09 11:31:00 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090208/609c14c9/attachment.html From juanbackson at gmail.com Sat Feb 7 08:44:16 2009 From: juanbackson at gmail.com (Juan Backson) Date: Sun, 8 Feb 2009 00:44:16 +0800 Subject: [Freeswitch-users] need suggestion on using mod_easyroute Message-ID: <27c25bc40902070844h5db40f22x26bcfce9ff21e852@mail.gmail.com> Hi, I notice that there is a newly-developed mod_easyroute model available. Has anyone used it with large amount of routes ( ex > 1M ) on a high traffic scenario? For that kind of scenario, would it be better to consider using out-going event socket to serve that purpose? I would greatly appreciate any recommendation. Thanks, JB -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090208/efd5c64b/attachment.html From msc at freeswitch.org Sat Feb 7 10:14:57 2009 From: msc at freeswitch.org (Michael Collins) Date: Sat, 7 Feb 2009 10:14:57 -0800 Subject: [Freeswitch-users] Call accounting - CDR's In-Reply-To: References: <35b355e90902061330t34c22f27x11013de60b101c76@mail.gmail.com> <35b355e90902061339w2084ddfbvbe7ed95d11b438ea@mail.gmail.com> <87f2f3b90902061442s2d1ef36fq5199a6766c083581@mail.gmail.com> <87f2f3b90902061600i5e3ce93esfe883b36afa6cc1c@mail.gmail.com> Message-ID: <87f2f3b90902071014v2dd31baay43d8d6df57d10ed9@mail.gmail.com> On Sat, Feb 7, 2009 at 2:07 AM, Nik Middleton wrote: > Great, thanks for that. > > One of the big issues with Asterisk's way of billing is that if let's > say a remote phone diverts a call to another number, say a mobile, > because a local channel is created for the redirect, Asterisk loses > critical information such as the account code and therefore cannot be > billed for. I'm going to try this with FS today and see what happens. > It would be awesome if it is accountable. That way I wouldn't have to > force the user to do diverts from a web page Let us know how it goes and if you have any questions. I'm sure that we can help you nail this one down. -MC From krice at freeswitch.org Sat Feb 7 10:26:59 2009 From: krice at freeswitch.org (Ken Rice) Date: Sat, 07 Feb 2009 12:26:59 -0600 Subject: [Freeswitch-users] need suggestion on using mod_easyroute In-Reply-To: <27c25bc40902070844h5db40f22x26bcfce9ff21e852@mail.gmail.com> Message-ID: Mod_easyroute can handle millions of numbers... It is NOT however an LCR module... There are other things for that... Look at mod_lcr or if you need an extremely high performance LCR contact me off list From: Juan Backson Reply-To: Date: Sun, 8 Feb 2009 00:44:16 +0800 To: Subject: [Freeswitch-users] need suggestion on using mod_easyroute Hi, I notice that there is a newly-developed mod_easyroute model available. Has anyone used it with large amount of routes ( ex > 1M ) on a high traffic scenario? For that kind of scenario, would it be better to consider using out-going event socket to serve that purpose? I would greatly appreciate any recommendation. Thanks, JB _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090207/d8278e7c/attachment.html From brian at freeswitch.org Sat Feb 7 10:30:31 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 7 Feb 2009 12:30:31 -0600 Subject: [Freeswitch-users] Global Variables forgotten through thepubliccontext? In-Reply-To: References: <0B4E2726927041D09D0425DA0242C805@UVix><9D0CFA79-C120-4D66-B48C-7A5641B07AF4@freeswitch.org><5B1ED1D4834B4719B309FAC03F1595C0@UVix> Message-ID: <03BD385F-E942-4B32-98B0-0A29E13BC5D5@freeswitch.org> Its trying to open the file to match the current channel rate... you can install the 16k files via "make hd-sounds-install hd-moh-install" /b On Feb 7, 2009, at 9:26 AM, UV wrote: > Will do. > Now I have a little problem delaying me as the latest build changed > something with the sound file playing and now the FS can?t find any > local file to play (adds mysterious \16000\ to the file location?). > I?ll try to isolate this first then get back to this issue. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090207/65fe3943/attachment.html From anthony.minessale at gmail.com Sat Feb 7 13:23:54 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sat, 7 Feb 2009 15:23:54 -0600 Subject: [Freeswitch-users] mod_shout delay in trunk In-Reply-To: <498CAD73.2040709@gmail.com> References: <15830211.131233952452225.JavaMail.root@zimbra> <28165623.151233952574612.JavaMail.root@zimbra> <191c3a030902061248r3d307647le5574c1004c6da24@mail.gmail.com> <498CAD73.2040709@gmail.com> Message-ID: <191c3a030902071323w562c1f2cg9a7aeebb42781c8f@mail.gmail.com> tamas, the opposite. The default is to not do one by one and setting the var to false makes it more i/o intensive but it would provide more real time recording when recording to streams. BTW the reversed logic is fixed in tree On Fri, Feb 6, 2009 at 3:36 PM, Tamas wrote: > Hello, > > could this option be used to lower I/O load - to rather write more bytes > at once rather than one by one - on file recording (record_session)? > > Regards, > Tamas > > Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090207/b3c6be65/attachment-0001.html From uv at yuvalhertzog.com Sat Feb 7 16:46:51 2009 From: uv at yuvalhertzog.com (UV) Date: Sun, 8 Feb 2009 11:46:51 +1100 Subject: [Freeswitch-users] Global Variables forgotten throughthepubliccontext? In-Reply-To: <03BD385F-E942-4B32-98B0-0A29E13BC5D5@freeswitch.org> References: <0B4E2726927041D09D0425DA0242C805@UVix><9D0CFA79-C120-4D66-B48C-7A5641B07AF4@freeswitch.org><5B1ED1D4834B4719B309FAC03F1595C0@UVix> <03BD385F-E942-4B32-98B0-0A29E13BC5D5@freeswitch.org> Message-ID: <9F926F21CCB64A49A6EFF4425B111F67@UVix> Yeah, I have all the sounds installed. I don?t think it?s that. I?m getting error messages such as ?[ERR] mod_sndfile.c:185 sndfile_file_open() Error Opening File [E:\FS/sounds/en/us/callie\voicemail/8000\16000\vm-goodbye.w] [System error : The system cannot find the path specified.]? all across the board. The only thing still working is MoH? This started from one of yesterday?s builds (r11665 ? 11678). _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Sunday, February 08, 2009 5:31 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Global Variables forgotten throughthepubliccontext? Its trying to open the file to match the current channel rate... you can install the 16k files via "make hd-sounds-install hd-moh-install" /b On Feb 7, 2009, at 9:26 AM, UV wrote: Will do. Now I have a little problem delaying me as the latest build changed something with the sound file playing and now the FS can?t find any local file to play (adds mysterious \16000\ to the file location?). I?ll try to isolate this first then get back to this issue. No virus found in this incoming message. Checked by AVG i www.avg.com Version: 8.0.233 / Virus Database: 270.10.18/1937 i Release Date: 02/06/09 11:31:00 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090208/0cab0873/attachment.html From brian at freeswitch.org Sat Feb 7 17:56:50 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 7 Feb 2009 19:56:50 -0600 Subject: [Freeswitch-users] Global Variables forgotten throughthepubliccontext? In-Reply-To: <9F926F21CCB64A49A6EFF4425B111F67@UVix> References: <0B4E2726927041D09D0425DA0242C805@UVix><9D0CFA79-C120-4D66-B48C-7A5641B07AF4@freeswitch.org><5B1ED1D4834B4719B309FAC03F1595C0@UVix> <03BD385F-E942-4B32-98B0-0A29E13BC5D5@freeswitch.org> <9F926F21CCB64A49A6EFF4425B111F67@UVix> Message-ID: <51C9B370-500D-4D95-A515-E9EEF1705014@freeswitch.org> You have a \ somewhere in your path... which doesn't make sense... you're on windows. Can you open a jira... I think this was the cause http://fisheye.freeswitch.org/browse/FreeSWITCH/src/mod/formats/mod_sndfile/mod_sndfile.c?r1=11090&r2=11601 /b On Feb 7, 2009, at 6:46 PM, UV wrote: > Yeah, I have all the sounds installed. I don?t think it?s that. > I?m getting error messages such as ?[ERR] mod_sndfile.c:185 > sndfile_file_open() Error Opening File [E:\FS/sounds/en/us/callie > \voicemail/8000\16000\vm-goodbye.w] [System error : The system > cannot find the path specified.]? all across the board. The only > thing still working is MoH? This started from one of yesterday?s > builds (r11665 ? 11678). -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090207/4553b646/attachment.html From woodydickson at gmail.com Sat Feb 7 18:24:19 2009 From: woodydickson at gmail.com (Woody Dickson) Date: Sun, 8 Feb 2009 10:24:19 +0800 Subject: [Freeswitch-users] Seeking opinion on shared disk space Message-ID: Hi, In my deployment scenario, I plan to have two redundant freeswitch servers running on two different boxes. Two key features I am leveraging on freeswitch are voicemail and call recording and playback., and as a result of that, a shared storage for playback of the recorded wav files is needed. When the user traffic is high, I am affraid that NFS or even GFS can't scale well. On the other hand, a real SAN hardware with optical-fabric is too expensive for us. We are therefore considering using iSCSI SAN to build a cheap SAN for that purpose. Does anyone have experience setting up a shared storage between multiple freeswitch servers and can share some inputs with me? Thanks for all your help. Woody From krice at freeswitch.org Sat Feb 7 18:34:52 2009 From: krice at freeswitch.org (Ken Rice) Date: Sat, 07 Feb 2009 20:34:52 -0600 Subject: [Freeswitch-users] Seeking opinion on shared disk space In-Reply-To: Message-ID: Whats wrong with NFS? As long as you put a reasonable disk subsystem you'll be fine... GFS sucks for voice anyway... It can take several seconds to get a lock... No matter what you use, you have to remember that you *MUST* have a cluster aware file system, simply mounting the same iscsi or SAN LUN on 2 different boxes running ext3 won't work since things aren't guaranteed to be flushed until a sync is called > From: Woody Dickson > Reply-To: > Date: Sun, 8 Feb 2009 10:24:19 +0800 > To: > Subject: [Freeswitch-users] Seeking opinion on shared disk space > > Hi, > > In my deployment scenario, I plan to have two redundant freeswitch > servers running on two different boxes. Two key features I am > leveraging on freeswitch are voicemail and call recording and > playback., and as a result of that, a shared storage for playback of > the recorded wav files is needed. When the user traffic is high, I am > affraid that NFS or even GFS can't scale well. On the other hand, a > real SAN hardware with optical-fabric is too expensive for us. We are > therefore considering using iSCSI SAN to build a cheap SAN for that > purpose. > > Does anyone have experience setting up a shared storage between > multiple freeswitch servers and can share some inputs with me? > > Thanks for all your help. > > Woody > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From john at whitesmiths.com Sat Feb 7 13:25:32 2009 From: john at whitesmiths.com (John O'Brien) Date: Sun, 8 Feb 2009 08:25:32 +1100 Subject: [Freeswitch-users] Newbie - point me in the right direction Message-ID: <008F70F6-818D-476B-9DE9-33DA018D37D9@whitesmiths.com> Hi, I am a real newbie. I have been building Asterisk based applications for a couple of years now. I am looking at migrating these apps to FreeSwitch - eventually. I want to do this gradually - I need to keep things running in the meantime. I have two Asterisk boxes, A1 & A2, each running a separate telephony app. We have an external SIP service with DID's NNNNN200 -> NNNNN299. We want to direct the incoming SIP calls so that the DID's NNNNN200 -> NNNNN219 go to Asterisk server A1 and NNNNN220 -> NNNNN299 to Asterisk server A2. Yes we really just want the calls switched on the DID. I'm struggling to know where to start - can someone point me in the right direction? Regards, John From pauld at versafon.com Sat Feb 7 21:03:59 2009 From: pauld at versafon.com (Paul D.) Date: Sun, 08 Feb 2009 00:03:59 -0500 Subject: [Freeswitch-users] Cannot choose Cepstral voice from dialplan Message-ID: <498E67BF.3060207@versafon.com> Followed Wiki to install and configure mod_cepstral. The problem is FS always defaults to one voice, which I installed first, and ignores others. I did define SWIFT_HOME and added swift lib path to /etc/ld.so.conf. After I restart FS I see on FS console after dialing my test extension: Failed to load library libceplex_us.so due to: libceplex_us.so: cannot open shared object file: No such file or directory Failed to load language / lexical libraries for Callie-8kHz I do have this voice installed. This message does not appear after subsequent calls to the test extension, until next FS restart. I use this in my dialplan: This is all under centos 5.2 64 bit. Any suggestions would be greatly appreciated. From nik.middleton at noblesolutions.co.uk Sun Feb 8 08:21:22 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Sun, 8 Feb 2009 16:21:22 -0000 Subject: [Freeswitch-users] Struggling with Originate Message-ID: Hi Guys, I'm placing calls ok by using the event socket. However, I need to modify the To: Sip header prior to the call going out for routing purposes. Is it possible to do this in the Originate action? If not, can someone explain if it's possible to trigger part of the dial plan externally? I can then modify the headers and then place the call/ Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090208/b1b9cea1/attachment-0001.html From dave at 3c.co.uk Sun Feb 8 09:09:09 2009 From: dave at 3c.co.uk (David Knell) Date: Sun, 8 Feb 2009 17:09:09 +0000 Subject: [Freeswitch-users] Struggling with Originate In-Reply-To: References: Message-ID: <4BFB326A-23AE-4768-9FBA-9774EC0FC34A@3c.co.uk> Hi Nik, How do you need to modify it? Cheers -- Dave > Hi Guys, > > I?m placing calls ok by using the event socket. However, I need to > modify the To: Sip header prior to the call going out for routing > purposes. Is it possible to do this in the Originate action? > > If not, can someone explain if it?s possible to trigger part of the > dial plan externally? I can then modify the headers and then place > the call/ > > > Regards, > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090208/6f980746/attachment.html From moises.silva at gmail.com Sun Feb 8 09:36:18 2009 From: moises.silva at gmail.com (Moises Silva) Date: Sun, 8 Feb 2009 11:36:18 -0600 Subject: [Freeswitch-users] Transcoding G723 In-Reply-To: <35b355e90902060716v2dff2586t253a69c97ab1adac@mail.gmail.com> References: <2FEEE7F7-DCE7-4919-A0FD-A0042BBB7EEE@freeswitch.org> <35b355e90902051226g12756d2fsf85fb996ec1ed188@mail.gmail.com> <11C3C72A-247D-4F8C-B106-52D49949D026@freeswitch.org> <1b46b4e80902051252o1b5a5c75l9f8d242cbaeec552@mail.gmail.com> <35b355e90902051301r4c2c53b6tb958774f4754e60f@mail.gmail.com> <35b355e90902060716v2dff2586t253a69c97ab1adac@mail.gmail.com> Message-ID: It seems I know the same you know, it was on the works then not, then back in the works. However I don't know the status on that. If you have contact with Doug he is a better person to ask to regarding new products coming out. Moy On Fri, Feb 6, 2009 at 9:16 AM, Shelby Ramsey wrote: > Thanks Moises. It looks like good work. When is Sangoma coming out with a > similar product ... Doug told me it was in the works, then not in the works, > then back in the works ... > The problem is this particular card is PCI only and it will only do 120 > channels .... > Thanks! > SDR > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- "I do not agree with what you have to say, but I'll defend to the death your right to say it." Voltaire From msc at freeswitch.org Sun Feb 8 14:11:00 2009 From: msc at freeswitch.org (Michael S Collins) Date: Sun, 8 Feb 2009 14:11:00 -0800 Subject: [Freeswitch-users] Cannot choose Cepstral voice from dialplan In-Reply-To: <498E67BF.3060207@versafon.com> References: <498E67BF.3060207@versafon.com> Message-ID: <8B964172-6EEB-4B5B-B4B1-D35BFEAEB460@freeswitch.org> Sent from my iPhone On Feb 7, 2009, at 9:03 PM, "Paul D." wrote: > Followed Wiki to install and configure mod_cepstral. The problem is FS > always defaults to one voice, which I installed first, and ignores > others. > I did define SWIFT_HOME and added swift lib path to /etc/ld.so.conf. > After I restart FS I see on FS console after dialing my test > extension: > > Failed to load library libceplex_us.so due to: libceplex_us.so: cannot > open shared object file: No such file or directory > Failed to load language / lexical libraries for Callie-8kHz > Look in /opt/swift for the dir that has the lib files. Most likely you'll find that they have the files are but have more specific names, perhaps with a version number. If you create symlinks to the files in question it should work. -MC > I do have this voice installed. This message does not appear after > subsequent calls to the test extension, until next FS restart. > I use this in my dialplan: > > > > > > > > > > This is all under centos 5.2 64 bit. Any suggestions would be greatly > appreciated. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From nik.middleton at noblesolutions.co.uk Sun Feb 8 14:31:35 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Sun, 8 Feb 2009 22:31:35 -0000 Subject: [Freeswitch-users] Problems passing arguments to lua Message-ID: Hi Guys, I'm having some issues passing an argument to an lua script. Basically in an originate command I pass the name of a .wav file If I hard code the file session:streamFile("myfile.wav"]); It works, However, using this: session:streamFile(argv[1]); causes this error 2009-02-08 22:09:07 [ERR] mod_lua.cpp:176 lua_parse_and_execute() cannot open /usr/local/freeswitch/scripts/helloworld.lua,myfile.wav: No such file or directory Any Ideas? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090208/c4f6b7b5/attachment.html From nik.middleton at noblesolutions.co.uk Sun Feb 8 14:41:29 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Sun, 8 Feb 2009 22:41:29 -0000 Subject: [Freeswitch-users] connecting to mysql using lua Message-ID: Hi Guys I want to access Mysql 5 from lua. The wiki is not too clear on this. Do I need to install lua and lua mysql? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090208/250993c5/attachment.html From pauld at versafon.com Sun Feb 8 15:14:25 2009 From: pauld at versafon.com (pauld) Date: Sun, 08 Feb 2009 18:14:25 -0500 Subject: [Freeswitch-users] Cannot choose Cepstral voice from dialplan In-Reply-To: <8B964172-6EEB-4B5B-B4B1-D35BFEAEB460@freeswitch.org> References: <498E67BF.3060207@versafon.com> <8B964172-6EEB-4B5B-B4B1-D35BFEAEB460@freeswitch.org> Message-ID: <498F6751.2000701@versafon.com> The libs are there with correct symlinks, see below. I tested both voices directly via swift command, works fine. Any other ideas? It's Cepstral 5.1, FS 1.0.2. ls -la /opt/swift/lib total 4352 drwxr-xr-x 2 root root 4096 Feb 7 22:59 . drwxr-xr-x 10 root root 4096 Feb 7 12:29 .. lrwxrwxrwx 1 999 20202 20 Feb 7 22:59 libceplang_en.so -> libceplang_en.so.5.1 lrwxrwxrwx 1 999 20202 20 Feb 7 22:59 libceplang_en.so.5 -> libceplang_en.so.5.1 -rwxrwxr-x 1 999 20202 412050 Jul 8 2008 libceplang_en.so.5.1 lrwxrwxrwx 1 999 20202 19 Feb 7 12:29 libceplex_uk.so -> libceplex_uk.so.5.1 lrwxrwxrwx 1 999 20202 19 Feb 7 12:29 libceplex_uk.so.5 -> libceplex_uk.so.5.1 -rwxrwxr-x 1 999 20202 904994 Jul 8 2008 libceplex_uk.so.5.1 lrwxrwxrwx 1 999 20202 19 Feb 7 22:59 libceplex_us.so -> libceplex_us.so.5.1 lrwxrwxrwx 1 999 20202 19 Feb 7 22:59 libceplex_us.so.5 -> libceplex_us.so.5.1 -rwxrwxr-x 1 999 20202 1009780 Jul 8 2008 libceplex_us.so.5.1 lrwxrwxrwx 1 999 20202 15 Feb 7 22:59 libswift.so -> libswift.so.5.1 lrwxrwxrwx 1 999 20202 15 Feb 7 22:59 libswift.so.5 -> libswift.so.5.1 -rwxrwxr-x 1 999 20202 2100418 Jul 8 2008 libswift.so.5.1 Michael S Collins wrote: > Sent from my iPhone > > On Feb 7, 2009, at 9:03 PM, "Paul D." wrote: > > >> Followed Wiki to install and configure mod_cepstral. The problem is FS >> always defaults to one voice, which I installed first, and ignores >> others. >> I did define SWIFT_HOME and added swift lib path to /etc/ld.so.conf. >> After I restart FS I see on FS console after dialing my test >> extension: >> >> Failed to load library libceplex_us.so due to: libceplex_us.so: cannot >> open shared object file: No such file or directory >> Failed to load language / lexical libraries for Callie-8kHz >> >> > > Look in /opt/swift for the dir that has the lib files. Most likely > you'll find that they have the files are but have more specific names, > perhaps with a version number. If you create symlinks to the files in > question it should work. > > -MC > > >> I do have this voice installed. This message does not appear after >> subsequent calls to the test extension, until next FS restart. >> I use this in my dialplan: >> >> >> >> >> >> >> >> >> >> This is all under centos 5.2 64 bit. Any suggestions would be greatly >> appreciated. >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From msc at freeswitch.org Sun Feb 8 15:20:50 2009 From: msc at freeswitch.org (Michael S Collins) Date: Sun, 8 Feb 2009 15:20:50 -0800 Subject: [Freeswitch-users] Problems passing arguments to lua In-Reply-To: References: Message-ID: <6E948AB3-1030-46C5-8E08-3AB74653F0AE@freeswitch.org> Print out the variable to make sure it is what you expect: io.write("argv is " .. argv[1] .. "\n"; Also, if you don't give the sound file an absolute path name then it will automatically use the sound dir path. -MC Sent from my iPhone On Feb 8, 2009, at 2:31 PM, "Nik Middleton" wrote: > Hi Guys, > > > > I?m having some issues passing an argument to an lua script. > > > > Basically in an originate command I pass the name of a .wav file > > > > If I hard code the file session:streamFile(?myfile.wav?]); > > > > It works, > > > > However, using this: > > > > session:streamFile(argv[1]); > > > > causes this error > > > > 2009-02-08 22:09:07 [ERR] mod_lua.cpp:176 lua_parse_and_execute() > cannot open /usr/local/freeswitch/scripts/helloworld.lua,myfile.wav: > No such file or directory > > > > Any Ideas? > > > > Regards > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090208/b7543fd6/attachment-0001.html From msc at freeswitch.org Sun Feb 8 15:27:19 2009 From: msc at freeswitch.org (Michael S Collins) Date: Sun, 8 Feb 2009 15:27:19 -0800 Subject: [Freeswitch-users] connecting to mysql using lua In-Reply-To: References: Message-ID: <78A54C9A-C807-445B-A3BA-0B09A3B27521@freeswitch.org> Nik, I see your point about the wiki entry regarding luasql. If someone could clarify then I will be happy to help get the wiki documentation updated appropriately. -MC Sent from my iPhone On Feb 8, 2009, at 2:41 PM, "Nik Middleton" wrote: > Hi Guys > > > > I want to access Mysql 5 from lua. The wiki is not too clear on > this. Do I need to install lua and lua mysql? > > > > Regards > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090208/9fa1eafc/attachment.html From nik.middleton at noblesolutions.co.uk Sun Feb 8 16:21:28 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Mon, 9 Feb 2009 00:21:28 -0000 Subject: [Freeswitch-users] Problems passing arguments to lua In-Reply-To: <6E948AB3-1030-46C5-8E08-3AB74653F0AE@freeswitch.org> References: <6E948AB3-1030-46C5-8E08-3AB74653F0AE@freeswitch.org> Message-ID: Done that, still doesn't work. My guess is "related Regards, ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael S Collins Sent: 08 February 2009 23:21 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Problems passing arguments to lua Print out the variable to make sure it is what you expect: io.write("argv is " .. argv[1] .. "\n"; Also, if you don't give the sound file an absolute path name then it will automatically use the sound dir path. -MC Sent from my iPhone On Feb 8, 2009, at 2:31 PM, "Nik Middleton" wrote: Hi Guys, I'm having some issues passing an argument to an lua script. Basically in an originate command I pass the name of a .wav file If I hard code the file session:streamFile("myfile.wav"]); It works, However, using this: session:streamFile(argv[1]); causes this error 2009-02-08 22:09:07 [ERR] mod_lua.cpp:176 lua_parse_and_execute() cannot open /usr/local/freeswitch/scripts/helloworld.lua,myfile.wav: No such file or directory Any Ideas? Regards _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090209/3aec6f2d/attachment.html From brian at freeswitch.org Sun Feb 8 16:54:46 2009 From: brian at freeswitch.org (Brian West) Date: Sun, 8 Feb 2009 18:54:46 -0600 Subject: [Freeswitch-users] Struggling with Originate In-Reply-To: <4BFB326A-23AE-4768-9FBA-9774EC0FC34A@3c.co.uk> References: <4BFB326A-23AE-4768-9FBA-9774EC0FC34A@3c.co.uk> Message-ID: If you're wanting to modify the invite domain you can do that via the sip_invite_domain variable either export it before the bridge or place it inside {} on the originate line... ie "{sip_invite_domain=example.com}sofia/blah/blah" /b On Feb 8, 2009, at 11:09 AM, David Knell wrote: > Hi Nik, > > How do you need to modify it? > > Cheers -- > > Dave From brian at freeswitch.org Sun Feb 8 16:55:40 2009 From: brian at freeswitch.org (Brian West) Date: Sun, 8 Feb 2009 18:55:40 -0600 Subject: [Freeswitch-users] Problems passing arguments to lua In-Reply-To: References: <6E948AB3-1030-46C5-8E08-3AB74653F0AE@freeswitch.org> Message-ID: <31920E97-233D-4331-9F07-B09E29B592CD@freeswitch.org> Looks like you put a , instead of a space when calling the script. /b On Feb 8, 2009, at 6:21 PM, Nik Middleton wrote: > cannot open /usr/local/freeswitch/scripts/helloworld.lua,myfile.wav -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090208/38935482/attachment.html From dougblackstone at gmail.com Sun Feb 8 19:21:49 2009 From: dougblackstone at gmail.com (Doug Blacksone) Date: Mon, 9 Feb 2009 11:21:49 +0800 Subject: [Freeswitch-users] Dynamic Dialplan Message-ID: <743112140902081921n2bb1bf49mfe36e5e2193c6b04@mail.gmail.com> Hi, Right now, I am working on getting freeswitch configured for our call-center with more than 1000 agents. There are several areas where we need the dialplan to be configurable based on some user detail in the database. Therefore, the dialplan needs to be some-what dynamic based-on inputs from the database. I would like to know from other implementation as to the most scalable way of doing high performance dynamic dialplan that is super scalable. There are three ways I can think of: 1. Static dialplan using customized freeswitch mod to access postgres for data pros: best performance cons: harder to program 2. Static dialplan using lua to access postgres for data pros: easy to program, maybe-performance is better than curl cons: need to search through all the extensions to find one dialplan, performance is slower than the first one. 3. curl-based dialplan using Java Servlet and HTTP pros: easy to program, freeswitch only gets one extension and no extension search cons: performance is slow than the other two Is this a correct analysis? If from a pure performance's perspective, how much performance can a customized mod gains in comparison to lua? For a production system that needs to be highly scalable, what do you recommend? Thank you very much for any input to our critical design decision. Doug -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090209/ec2253c8/attachment-0001.html From krice at freeswitch.org Sun Feb 8 19:32:18 2009 From: krice at freeswitch.org (Ken Rice) Date: Sun, 08 Feb 2009 21:32:18 -0600 Subject: [Freeswitch-users] Dynamic Dialplan In-Reply-To: <743112140902081921n2bb1bf49mfe36e5e2193c6b04@mail.gmail.com> Message-ID: If you want an extremely high performance you write your own dialplan module... Its not that hard... Or option 1 is the more high performance way to fo... Curl with a serverlet will scale to a point but I doubt it will get to where you need in the long run Static and do what you need, but how scalable is lua is unknown at this point (someone should post some info on that) and its known that java script doesn?t scale that well for large installs K From: Doug Blacksone Reply-To: Date: Mon, 9 Feb 2009 11:21:49 +0800 To: Subject: [Freeswitch-users] Dynamic Dialplan Hi, Right now, I am working on getting freeswitch configured for our call-center with more than 1000 agents. There are several areas where we need the dialplan to be configurable based on some user detail in the database. Therefore, the dialplan needs to be some-what dynamic based-on inputs from the database. I would like to know from other implementation as to the most scalable way of doing high performance dynamic dialplan that is super scalable. There are three ways I can think of: 1. Static dialplan using customized freeswitch mod to access postgres for data pros: best performance cons: harder to program 2. Static dialplan using lua to access postgres for data pros: easy to program, maybe-performance is better than curl cons: need to search through all the extensions to find one dialplan, performance is slower than the first one. 3. curl-based dialplan using Java Servlet and HTTP pros: easy to program, freeswitch only gets one extension and no extension search cons: performance is slow than the other two Is this a correct analysis? If from a pure performance's perspective, how much performance can a customized mod gains in comparison to lua? For a production system that needs to be highly scalable, what do you recommend? Thank you very much for any input to our critical design decision. Doug _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090208/6ffdb348/attachment.html From sicfslist at gmail.com Sun Feb 8 20:20:22 2009 From: sicfslist at gmail.com (Shelby Ramsey) Date: Sun, 8 Feb 2009 22:20:22 -0600 Subject: [Freeswitch-users] Dynamic Dialplan In-Reply-To: References: <743112140902081921n2bb1bf49mfe36e5e2193c6b04@mail.gmail.com> Message-ID: <35b355e90902082020s274fb805r7197a77e522b36f4@mail.gmail.com> Doug, Ken is right on this one. I know there are some guys on the list (like Ken) that could help you write a module. It's probably the best way to go (if you're going to have all agents running off of one or two boxes). If you're going to spread the agents / calls around on multiple boxes or use a combo of OpenSIPS / OpenSer / pick your flavor ... and FS then using xml_curl will work fine. We've got that working today and it's been acceptable. SDR -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090208/34593918/attachment.html From krice at suspicious.org Sun Feb 8 20:50:31 2009 From: krice at suspicious.org (Ken Rice) Date: Sun, 08 Feb 2009 22:50:31 -0600 Subject: [Freeswitch-users] Dynamic Dialplan In-Reply-To: <35b355e90902082020s274fb805r7197a77e522b36f4@mail.gmail.com> Message-ID: Also depending on what your Timeframe is like there is a distributed queue mechanism with skills based routing on the way... From: Shelby Ramsey Reply-To: Date: Sun, 8 Feb 2009 22:20:22 -0600 To: Subject: Re: [Freeswitch-users] Dynamic Dialplan Doug, Ken is right on this one. I know there are some guys on the list (like Ken) that could help you write a module. It's probably the best way to go (if you're going to have all agents running off of one or two boxes). If you're going to spread the agents / calls around on multiple boxes or use a combo of OpenSIPS / OpenSer / pick your flavor ... and FS then using xml_curl will work fine. We've got that working today and it's been acceptable. SDR _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090208/d0d1e752/attachment.html From andrew at hijacked.us Sun Feb 8 21:16:47 2009 From: andrew at hijacked.us (Andrew Thompson) Date: Mon, 9 Feb 2009 00:16:47 -0500 Subject: [Freeswitch-users] Dynamic Dialplan In-Reply-To: References: <35b355e90902082020s274fb805r7197a77e522b36f4@mail.gmail.com> Message-ID: <20090209051646.GD4963@hijacked.us> On Sun, Feb 08, 2009 at 10:50:31PM -0600, Ken Rice wrote: > Also depending on what your Timeframe is like there is a distributed queue > mechanism with skills based routing on the way... > It even managed to route 2 calls in a row this week ;) Still a ways off from anything production grade tho. Andrew From wasim at convergence.pk Sun Feb 8 21:14:41 2009 From: wasim at convergence.pk (Wasim Baig) Date: Mon, 9 Feb 2009 10:14:41 +0500 Subject: [Freeswitch-users] Dynamic Dialplan In-Reply-To: <20090209051646.GD4963@hijacked.us> References: <35b355e90902082020s274fb805r7197a77e522b36f4@mail.gmail.com> <20090209051646.GD4963@hijacked.us> Message-ID: On Mon, Feb 9, 2009 at 10:16 AM, Andrew Thompson wrote: > On Sun, Feb 08, 2009 at 10:50:31PM -0600, Ken Rice wrote: > > Also depending on what your Timeframe is like there is a distributed > queue > > mechanism with skills based routing on the way... > > > It even managed to route 2 calls in a row this week ;) Still a ways off > from anything production grade tho. Bravo, bravo! -- wasim h. baig | principal consultant | convergence pk | +92 300 8508070 | peace be upon you ... -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090209/2e57bf53/attachment.html From kristian.kielhofner at gmail.com Sun Feb 8 23:15:07 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Mon, 9 Feb 2009 02:15:07 -0500 Subject: [Freeswitch-users] Mod_native_file FreeSWITCH files (and script) available Message-ID: <2d9149cd0902082315l73703eb2t79d0bbb96845ff49@mail.gmail.com> Hello everyone, Now that I've got FreeSWITCH compiling under AstLinux I'm starting to look at ways to optimize FreeSWITCH. First things first: minimize transcoding. I hate transcoding. I modified a concept I came up with back in the day for Asterisk. I've created a script to convert WAV files to FreeSWITCH mod_native_file formats for various codecs. Of course I also updated the wiki: http://wiki.freeswitch.org/wiki/Mod_native_file#Script_to_convert_a_sound_file_to_specific_formats_to_avoid_transcoding As you can see I've also converted the current FreeSWITCH prompts and MOH into various file formats (including G.729 and G.723). These files have undergone very little testing (basically none) so I would appreciate some feedback on them. I'm also working on G.729, iLBC, speex (VBR?!?), and G.722. What about Siren, CELT, and some of the others? I don't have much experience with these and I could certainly use some help. P.S. - I'm currently using sox and Asterisk res_convert to do the conversion. I don't know how else to legitimately (and consistently) convert to these various formats. Without bringing up the usual G.729/G.723 discussions, I'm in the US and this is how I could legally do all of these conversions. -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From helmut.kuper at ewetel.de Sun Feb 8 23:48:13 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Mon, 09 Feb 2009 08:48:13 +0100 Subject: [Freeswitch-users] Openzap r654 doesn't compile. Error with ozmod_libpri Message-ID: <498FDFBD.8050401@ewetel.de> Hello, during some imporvements on q931toPcap as well as debugging my TDM PRI problem with loosing state sync after some time I updated the code to latest trunk (r654). After doing a bootstrap in FS trunk directory I tried to compile openzap in libs/openzap. I got this error: /bin/sh ./libtool --tag=CC --mode=link gcc -g -O2 -o ozmod_libpri.la -rpath /usr/local/openzap/mod -lm -lpthread -ldl libtool: link: libtool library `ozmod_libpri.la' must begin with `lib' Try `libtool --help --mode=link' for more information. make: *** [ozmod_libpri.la] Error 1 Additionally I wonder what ozmod_libpri is. Do I have consider it for Q931 capturing? regards Helmut From helmut.kuper at ewetel.de Sun Feb 8 23:54:04 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Mon, 09 Feb 2009 08:54:04 +0100 Subject: [Freeswitch-users] Can I force the A-leg codec in FS? Message-ID: <498FE11C.2070105@ewetel.de> Hello, in my reallife setup of FS all internal extensions use G.722 as preferred codec. Unfortunately when there is an outgoing TDM call I found that FS starts transcoding instead of forcing G.711 for A leg. So is there a way to force the A codec? regards helmut From krice at suspicious.org Mon Feb 9 00:00:18 2009 From: krice at suspicious.org (Ken Rice) Date: Mon, 09 Feb 2009 02:00:18 -0600 Subject: [Freeswitch-users] Can I force the A-leg codec in FS? In-Reply-To: <498FE11C.2070105@ewetel.de> Message-ID: Set the codec negotiation to greedy > From: Helmut Kuper > Reply-To: > Date: Mon, 09 Feb 2009 08:54:04 +0100 > To: > Subject: [Freeswitch-users] Can I force the A-leg codec in FS? > > Hello, > > in my reallife setup of FS all internal extensions use G.722 as > preferred codec. Unfortunately when there is an outgoing TDM call I > found that FS starts transcoding instead of forcing G.711 for A leg. So > is there a way to force the A codec? > > regards > helmut > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From helmut.kuper at ewetel.de Mon Feb 9 00:26:27 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Mon, 09 Feb 2009 09:26:27 +0100 Subject: [Freeswitch-users] Can I force the A-leg codec in FS? In-Reply-To: References: Message-ID: <498FE8B3.8040904@ewetel.de> Hi Ken, thx for the hint. It looks quite static, so I guess each call (also internal) are then forced to g711. I'm looking for a dynamic way depending on destination number. regards helmut On 09.02.2009 09:00, Ken Rice wrote: > Set the codec negotiation to greedy > > > From helmut.kuper at ewetel.de Mon Feb 9 00:27:33 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Mon, 09 Feb 2009 09:27:33 +0100 Subject: [Freeswitch-users] Openzap r654 doesn't compile. Error with ozmod_libpri In-Reply-To: <498FDFBD.8050401@ewetel.de> References: <498FDFBD.8050401@ewetel.de> Message-ID: <498FE8F5.9060206@ewetel.de> Hello, update, when I remove all ozmod_ from ozmod_libpri lines in Makefile, it compiles without errors. regards helmut On 09.02.2009 08:48, Helmut Kuper wrote: > Hello, > > during some imporvements on q931toPcap as well as debugging my TDM PRI > problem with loosing state sync after some time I updated the code to > latest trunk (r654). After doing a bootstrap in FS trunk directory I > tried to compile openzap in libs/openzap. I got this error: > > /bin/sh ./libtool --tag=CC --mode=link gcc -g -O2 -o > ozmod_libpri.la -rpath /usr/local/openzap/mod -lm -lpthread -ldl > libtool: link: libtool library `ozmod_libpri.la' must begin with `lib' > Try `libtool --help --mode=link' for more information. > make: *** [ozmod_libpri.la] Error 1 > > Additionally I wonder what ozmod_libpri is. Do I have consider it for > Q931 capturing? > > regards > Helmut > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > -- Mit freundlichen Gr??en Helmut Kuper Finanzdienstleistungen und Entwicklung Telefax: (0441) 8000-2799 mailto:helmut.kuper at ewetel.de ___________________________________ EWE TEL GmbH Cloppenburger Stra?e 310 26133 Oldenburg EWE TEL GmbH Handelsregister Amtsgericht Oldenburg HRB 3723 Vorsitzender des Aufsichtsrates: Heiko Harms Gesch?ftsf?hrung: Hans-Joachim Iken (Vorsitzender), Dr. Norbert Schulz, Dirk Thole Homepage: http://www.ewetel.de ___________________________________ From nik.middleton at noblesolutions.co.uk Mon Feb 9 00:56:00 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Mon, 9 Feb 2009 08:56:00 -0000 Subject: [Freeswitch-users] Problems passing arguments to lua In-Reply-To: <31920E97-233D-4331-9F07-B09E29B592CD@freeswitch.org> References: <6E948AB3-1030-46C5-8E08-3AB74653F0AE@freeswitch.org> <31920E97-233D-4331-9F07-B09E29B592CD@freeswitch.org> Message-ID: That and not enclosing in single quotes, thanks Regards, ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: 09 February 2009 00:56 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Problems passing arguments to lua Looks like you put a , instead of a space when calling the script. /b On Feb 8, 2009, at 6:21 PM, Nik Middleton wrote: cannot open /usr/local/freeswitch/scripts/helloworld.lua,myfile.wav -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090209/35d592a8/attachment.html From ahgindia308 at gmail.com Mon Feb 9 01:46:17 2009 From: ahgindia308 at gmail.com (Ankit Gandhi) Date: Mon, 9 Feb 2009 01:46:17 -0800 (PST) Subject: [Freeswitch-users] Freeswitch not processing calls Message-ID: <21909687.post@talk.nabble.com> Recently I noticed that fs (1.0.3 RC1) is not processing the calls. Previously about 50-60 calls were remaining active at any time. But then suddenly, only 5-6 active calls were there. So I checked sip INVITE's in ngrep, and I noticed that originator were sending CANCEL after INVITE. So I contacted them and they were telling that while sending calls, your gateways time outs after a call is sent, and so they send CANCEL after INVITE. My architecture is as follows : originator -> switch (our fs) -> terminator. CPU : Intel(R) Xeon(R) CPU X3220 @ 2.40GHz OS : Ubuntu 6.06.1 LTS 32-bit RAM : 4 GB We had also set ulimits on command prompt every time before starting fs with nc mode as follows : ulimit -c unlimited; ulimit -n 999999; ulimit -s 244; /usr/local/freeswitch/bin/freeswitch -nc In the ngrep trace, I noticed that fs was not sending INVITE to terminator in between the INVITE and CANCEL from originator. So what could be the reason for INVITE not being processed by fs and getting timeout from originator. Here is the sip trace for a sample call. Check the duration between INVITE and CANCEL from originator. -------------------------------------------------------------------- U 2009/02/07 13:16:44.354443 ori.ori.ori.ori:2000 -> fs.fs.fs.fs:5060 INVITE sip:yyyyyyyyyyyy at fs.fs.fs.fs:5060;user=phone SIP/2.0. Call-ID: 7008117790783130205-1234012604 at ori.ori.ori.ori. From: ;tag=13475. To: . Content-Type: application/sdp. CSeq: 1 INVITE. Via: SIP/2.0/UDP ori.ori.ori.ori:2000;branch=z9hG4bK-6141d60007b94a5d-4297d09d-1. Contact: sip:xxxxxxxxxxx at ori.ori.ori.ori:2000;user=phone. Allow: INVITE,CANCEL,BYE,ACK,REFER,UPDATE,INFO. Supported: timer,100rel. Max-Forwards: 70. Content-Length: 317. . v=0. o=MG4000|2.0 492612 492612 IN IP4 66.151.208.138. s=-. c=IN IP4 66.151.208.138. t=0 0. m=audio 38832 RTP/AVP 97 18 98 96 101 13. a=rtpmap:97 G.729b/8000. a=rtpmap:98 G.729a/8000. a=rtpmap:96 G.729ab/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=fmtp:18 annexb=yes. a=ptime:20. a=rtpmap:13 CN/8000. # U 2009/02/07 13:16:44.354633 fs.fs.fs.fs:5060 -> ori.ori.ori.ori:2000 SIP/2.0 100 Trying. Via: SIP/2.0/UDP ori.ori.ori.ori:2000;branch=z9hG4bK-6141d60007b94a5d-4297d09d-1. From: ;tag=13475. To: . Call-ID: 7008117790783130205-1234012604 at ori.ori.ori.ori. CSeq: 1 INVITE. User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-hacked. Content-Length: 0. U 2009/02/07 13:16:59.208008 ori.ori.ori.ori:2000 -> fs.fs.fs.fs:5060 CANCEL sip:yyyyyyyyyyyy at fs.fs.fs.fs:5060;user=phone SIP/2.0. Call-ID: 7008117790783130205-1234012604 at ori.ori.ori.ori. From: ;tag=13475. To: . CSeq: 1 CANCEL. Via: SIP/2.0/UDP ori.ori.ori.ori:2000;branch=z9hG4bK-6141d60007b94a5d-4297d09d-1. Supported: timer,100rel. Max-Forwards: 70. Content-Length: 0. . # U 2009/02/07 13:16:59.208095 fs.fs.fs.fs:5060 -> ori.ori.ori.ori:2000 SIP/2.0 200 OK. Via: SIP/2.0/UDP ori.ori.ori.ori:2000;branch=z9hG4bK-6141d60007b94a5d-4297d09d-1. From: ;tag=13475. To: ;tag=7B6p78S5SDF0j. Call-ID: 7008117790783130205-1234012604 at ori.ori.ori.ori. CSeq: 1 CANCEL. Content-Length: 0. . # U 2009/02/07 13:16:59.208134 fs.fs.fs.fs:5060 -> ori.ori.ori.ori:2000 SIP/2.0 487 Request Terminated. Via: SIP/2.0/UDP ori.ori.ori.ori:2000;branch=z9hG4bK-6141d60007b94a5d-4297d09d-1. From: ;tag=13475. To: ;tag=7B6p78S5SDF0j. Call-ID: 7008117790783130205-1234012604 at ori.ori.ori.ori. CSeq: 1 INVITE. User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-hacked. Accept: application/sdp. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. Supported: timer, precondition, path, replaces. Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Content-Length: 0. U 2009/02/07 13:16:59.253482 ori.ori.ori.ori:2000 -> fs.fs.fs.fs:5060 ACK sip:yyyyyyyyyyyy at fs.fs.fs.fs:5060;user=phone SIP/2.0. Call-ID: 7008117790783130205-1234012604 at ori.ori.ori.ori. From: ;tag=13475. To: ;tag=7B6p78S5SDF0j. CSeq: 1 ACK. Via: SIP/2.0/UDP ori.ori.ori.ori:2000;branch=z9hG4bK-6141d60007b94a5d-4297d09d-1. Max-Forwards: 70. Content-Length: 0. ------------------------------------------------------- -- View this message in context: http://www.nabble.com/Freeswitch-not-processing-calls-tp21909687p21909687.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From brian at freeswitch.org Mon Feb 9 02:14:44 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 9 Feb 2009 04:14:44 -0600 Subject: [Freeswitch-users] Freeswitch not processing calls In-Reply-To: <21909687.post@talk.nabble.com> References: <21909687.post@talk.nabble.com> Message-ID: I notice it offers 18 which is G729 but these listed below are 100% invalid. There is no such thing as G.729a, G.729b or G.729ab that are valid in the SDP. I suspect if you start FreeSWITCH and crank it up to debug level ("console loglevel debug") you'll clearly see why this is taking place. I think we talked to you on IRC about this and told this. If your termination gateway requires any of the above listed on 96,97 or 98 the call will fail. We can only do passthru on G729 which is the official name which is listed on payload 18. Your termination provider needs to be informed of this fact. The fmtp line is what controls annexb operation! /b On Feb 9, 2009, at 3:46 AM, Ankit Gandhi wrote: > a=rtpmap:97 G.729b/8000. > a=rtpmap:98 G.729a/8000. > a=rtpmap:96 G.729ab/8000. From ahgindia308 at gmail.com Mon Feb 9 03:48:12 2009 From: ahgindia308 at gmail.com (Ankit Gandhi) Date: Mon, 9 Feb 2009 03:48:12 -0800 (PST) Subject: [Freeswitch-users] Freeswitch not processing calls In-Reply-To: References: <21909687.post@talk.nabble.com> Message-ID: <21911300.post@talk.nabble.com> Hi Brian, But issue here is that, FS is not processing any such calls and not sending 488 to the caller. Also the sip trace I had provided was from the caller to fs. FS does not even bridge the call to terminator in between the INVITE and CANCEL from the caller. It just gives so many errors in log like following : 2009-02-09 11:31:04 [ERR] mod_g729.c:145 switch_g729_decode() This codec is only usable in passthrough mode! 2009-02-09 11:31:04 [ERR] mod_g729.c:145 switch_g729_decode() This codec is only usable in passthrough mode! 2009-02-09 11:31:04 [ERR] mod_g729.c:145 switch_g729_decode() This codec is only usable in passthrough mode! 2009-02-09 11:31:04 [ERR] mod_g729.c:145 switch_g729_decode() This codec is only usable in passthrough mode! 2009-02-09 11:31:04 [ERR] mod_g729.c:145 switch_g729_decode() This codec is only usable in passthrough mode! It seems odd that due to codec mismatch, fs does not process the call at all and just times out and the caller sends CANCEL to us. Waiting for your reply. Brian West-3 wrote: > > I notice it offers 18 which is G729 but these listed below are 100% > invalid. There is no such thing as G.729a, G.729b or G.729ab that are > valid in the SDP. I suspect if you start FreeSWITCH and crank it up > to debug level ("console loglevel debug") you'll clearly see why this > is taking place. I think we talked to you on IRC about this and told > this. If your termination gateway requires any of the above listed on > 96,97 or 98 the call will fail. We can only do passthru on G729 which > is the official name which is listed on payload 18. Your termination > provider needs to be informed of this fact. The fmtp line is what > controls annexb operation! > > /b > > On Feb 9, 2009, at 3:46 AM, Ankit Gandhi wrote: > >> a=rtpmap:97 G.729b/8000. >> a=rtpmap:98 G.729a/8000. >> a=rtpmap:96 G.729ab/8000. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/Freeswitch-not-processing-calls-tp21909687p21911300.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From ahgindia308 at gmail.com Mon Feb 9 04:06:24 2009 From: ahgindia308 at gmail.com (Ankit Gandhi) Date: Mon, 9 Feb 2009 04:06:24 -0800 (PST) Subject: [Freeswitch-users] Freeswitch not processing calls In-Reply-To: <21911300.post@talk.nabble.com> References: <21909687.post@talk.nabble.com> <21911300.post@talk.nabble.com> Message-ID: <21911561.post@talk.nabble.com> Here is the correct codec sent to fs, but it times out again. http://www.nabble.com/file/p21911561/correct_codec_with_cancel.txt correct_codec_with_cancel.txt Ankit Gandhi wrote: > > Hi Brian, > But issue here is that, FS is not processing any such calls and not > sending 488 to the caller. > Also the sip trace I had provided was from the caller to fs. FS does not > even bridge the call to terminator in between the INVITE and CANCEL from > the caller. > It just gives so many errors in log like following : > > 2009-02-09 11:31:04 [ERR] mod_g729.c:145 switch_g729_decode() This codec > is only usable in passthrough mode! > 2009-02-09 11:31:04 [ERR] mod_g729.c:145 switch_g729_decode() This codec > is only usable in passthrough mode! > 2009-02-09 11:31:04 [ERR] mod_g729.c:145 switch_g729_decode() This codec > is only usable in passthrough mode! > 2009-02-09 11:31:04 [ERR] mod_g729.c:145 switch_g729_decode() This codec > is only usable in passthrough mode! > 2009-02-09 11:31:04 [ERR] mod_g729.c:145 switch_g729_decode() This codec > is only usable in passthrough mode! > > It seems odd that due to codec mismatch, fs does not process the call at > all and just times out and the caller sends CANCEL to us. > Waiting for your reply. > > > Brian West-3 wrote: >> >> I notice it offers 18 which is G729 but these listed below are 100% >> invalid. There is no such thing as G.729a, G.729b or G.729ab that are >> valid in the SDP. I suspect if you start FreeSWITCH and crank it up >> to debug level ("console loglevel debug") you'll clearly see why this >> is taking place. I think we talked to you on IRC about this and told >> this. If your termination gateway requires any of the above listed on >> 96,97 or 98 the call will fail. We can only do passthru on G729 which >> is the official name which is listed on payload 18. Your termination >> provider needs to be informed of this fact. The fmtp line is what >> controls annexb operation! >> >> /b >> >> On Feb 9, 2009, at 3:46 AM, Ankit Gandhi wrote: >> >>> a=rtpmap:97 G.729b/8000. >>> a=rtpmap:98 G.729a/8000. >>> a=rtpmap:96 G.729ab/8000. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- View this message in context: http://www.nabble.com/Freeswitch-not-processing-calls-tp21909687p21911561.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From nik.middleton at noblesolutions.co.uk Mon Feb 9 05:30:35 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Mon, 9 Feb 2009 13:30:35 -0000 Subject: [Freeswitch-users] Making a system call with LUA Message-ID: In the absence of any directives on lua/mysql, is it possible to launch a PHP script from lua? All I need to do is pass some data to update a db record. I don't need to have any links to the call as I intend to call is in the hang-up callback Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090209/0a56c5c5/attachment.html From anthony.minessale at gmail.com Mon Feb 9 05:51:44 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 9 Feb 2009 07:51:44 -0600 Subject: [Freeswitch-users] Freeswitch not processing calls In-Reply-To: <21911561.post@talk.nabble.com> References: <21909687.post@talk.nabble.com> <21911300.post@talk.nabble.com> <21911561.post@talk.nabble.com> Message-ID: <191c3a030902090551w75a6d5e5hc736ba7ab2784b73@mail.gmail.com> 1) please do not report bugs on the mailing list. 2) please report the bug on jira http://jira.freeswitch.org according to the rules: http://wiki.freeswitch.org/wiki/Reporting_Bugs If you have an issue that you want us to correct you will have to try the latest SVN trunk (not a snapshot) issue "make current" from your RC1 directory. Attach the entire console log output from start of call to finish with console loglevel debug. On Mon, Feb 9, 2009 at 6:06 AM, Ankit Gandhi wrote: > > Here is the correct codec sent to fs, but it times out again. > http://www.nabble.com/file/p21911561/correct_codec_with_cancel.txt > correct_codec_with_cancel.txt > > > Ankit Gandhi wrote: > > > > Hi Brian, > > But issue here is that, FS is not processing any such calls and not > > sending 488 to the caller. > > Also the sip trace I had provided was from the caller to fs. FS does not > > even bridge the call to terminator in between the INVITE and CANCEL from > > the caller. > > It just gives so many errors in log like following : > > > > 2009-02-09 11:31:04 [ERR] mod_g729.c:145 switch_g729_decode() This codec > > is only usable in passthrough mode! > > 2009-02-09 11:31:04 [ERR] mod_g729.c:145 switch_g729_decode() This codec > > is only usable in passthrough mode! > > 2009-02-09 11:31:04 [ERR] mod_g729.c:145 switch_g729_decode() This codec > > is only usable in passthrough mode! > > 2009-02-09 11:31:04 [ERR] mod_g729.c:145 switch_g729_decode() This codec > > is only usable in passthrough mode! > > 2009-02-09 11:31:04 [ERR] mod_g729.c:145 switch_g729_decode() This codec > > is only usable in passthrough mode! > > > > It seems odd that due to codec mismatch, fs does not process the call at > > all and just times out and the caller sends CANCEL to us. > > Waiting for your reply. > > > > > > Brian West-3 wrote: > >> > >> I notice it offers 18 which is G729 but these listed below are 100% > >> invalid. There is no such thing as G.729a, G.729b or G.729ab that are > >> valid in the SDP. I suspect if you start FreeSWITCH and crank it up > >> to debug level ("console loglevel debug") you'll clearly see why this > >> is taking place. I think we talked to you on IRC about this and told > >> this. If your termination gateway requires any of the above listed on > >> 96,97 or 98 the call will fail. We can only do passthru on G729 which > >> is the official name which is listed on payload 18. Your termination > >> provider needs to be informed of this fact. The fmtp line is what > >> controls annexb operation! > >> > >> /b > >> > >> On Feb 9, 2009, at 3:46 AM, Ankit Gandhi wrote: > >> > >>> a=rtpmap:97 G.729b/8000. > >>> a=rtpmap:98 G.729a/8000. > >>> a=rtpmap:96 G.729ab/8000. > >> > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > >> > > > > > > -- > View this message in context: > http://www.nabble.com/Freeswitch-not-processing-calls-tp21909687p21911561.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090209/5fe6dfae/attachment.html From pauld at versafon.com Mon Feb 9 06:01:03 2009 From: pauld at versafon.com (pauld) Date: Mon, 09 Feb 2009 09:01:03 -0500 Subject: [Freeswitch-users] Dynamic Dialplan In-Reply-To: <743112140902081921n2bb1bf49mfe36e5e2193c6b04@mail.gmail.com> References: <743112140902081921n2bb1bf49mfe36e5e2193c6b04@mail.gmail.com> Message-ID: <4990371F.4050006@versafon.com> Option 3 does not have slow performance, Java apps can be highly scalable high performance when written right, this is a serious strong typed language unlike lua and javascript. I actually tested such solution against MySql cluster with 500 calls/m load script, scaled just fine. Contact me off list if you need professional help with that. Doug Blacksone wrote: > Hi, > > Right now, I am working on getting freeswitch configured for our > call-center with more than 1000 agents. There are several areas where > we need the dialplan to be configurable based on some user detail in > the database. Therefore, the dialplan needs to be some-what dynamic > based-on inputs from the database. > > I would like to know from other implementation as to the most scalable > way of doing high performance dynamic dialplan that is super scalable. > > There are three ways I can think of: > > 1. Static dialplan using customized freeswitch mod to access postgres > for data > pros: best performance > cons: harder to program > > 2. Static dialplan using lua to access postgres for data > pros: easy to program, maybe-performance is better than curl > cons: need to search through all the extensions to find one dialplan, > performance is slower than the first one. > > 3. curl-based dialplan using Java Servlet and HTTP > pros: easy to program, freeswitch only gets one extension and no > extension search > cons: performance is slow than the other two > > Is this a correct analysis? > If from a pure performance's perspective, how much performance can a > customized mod gains in comparison to lua? > > For a production system that needs to be highly scalable, what do you > recommend? > > > Thank you very much for any input to our critical design decision. > > Doug > > > > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From alex at sinapticode.ro Mon Feb 9 06:07:08 2009 From: alex at sinapticode.ro (Alexandru Nedelcu) Date: Mon, 09 Feb 2009 16:07:08 +0200 Subject: [Freeswitch-users] Making a system call with LUA In-Reply-To: References: Message-ID: <1234188428.6040.16.camel@gathern.lan> On Mon, 2009-02-09 at 13:30 +0000, Nik Middleton wrote: > In the absence of any directives on lua/mysql, is it possible to > launch a PHP script from lua? All I need to do is pass some data to > update a db record. I don?t need to have any links to the call as I > intend to call is in the hang-up callback I'm actually using Javascript, but os.execute should work: http://www.lua.org/pil/22.2.html From anthony.minessale at gmail.com Mon Feb 9 06:07:51 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 9 Feb 2009 08:07:51 -0600 Subject: [Freeswitch-users] Dynamic Dialplan In-Reply-To: <4990371F.4050006@versafon.com> References: <743112140902081921n2bb1bf49mfe36e5e2193c6b04@mail.gmail.com> <4990371F.4050006@versafon.com> Message-ID: <191c3a030902090607i3050187ay5c9047a36fb69a73@mail.gmail.com> hint: when you have "harder to program" under cons: that's usually how you find the right choice ;) On Mon, Feb 9, 2009 at 8:01 AM, pauld wrote: > Option 3 does not have slow performance, Java apps can be highly > scalable high performance when written right, this is a serious strong > typed language unlike lua and javascript. > I actually tested such solution against MySql cluster with 500 calls/m > load script, scaled just fine. > Contact me off list if you need professional help with that. > > Doug Blacksone wrote: > > Hi, > > > > Right now, I am working on getting freeswitch configured for our > > call-center with more than 1000 agents. There are several areas where > > we need the dialplan to be configurable based on some user detail in > > the database. Therefore, the dialplan needs to be some-what dynamic > > based-on inputs from the database. > > > > I would like to know from other implementation as to the most scalable > > way of doing high performance dynamic dialplan that is super scalable. > > > > There are three ways I can think of: > > > > 1. Static dialplan using customized freeswitch mod to access postgres > > for data > > pros: best performance > > cons: harder to program > > > > 2. Static dialplan using lua to access postgres for data > > pros: easy to program, maybe-performance is better than curl > > cons: need to search through all the extensions to find one dialplan, > > performance is slower than the first one. > > > > 3. curl-based dialplan using Java Servlet and HTTP > > pros: easy to program, freeswitch only gets one extension and no > > extension search > > cons: performance is slow than the other two > > > > Is this a correct analysis? > > If from a pure performance's perspective, how much performance can a > > customized mod gains in comparison to lua? > > > > For a production system that needs to be highly scalable, what do you > > recommend? > > > > > > Thank you very much for any input to our critical design decision. > > > > Doug > > > > > > > > > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090209/dee9c4d7/attachment-0001.html From jalsot at gmail.com Mon Feb 9 06:28:59 2009 From: jalsot at gmail.com (Tamas) Date: Mon, 09 Feb 2009 15:28:59 +0100 Subject: [Freeswitch-users] Questions about speex Message-ID: <49903DAB.3080706@gmail.com> Hello, I'm looking for the best codec/scenario for the last-mile and checked speex codec capabilities (http://www.speex.org/comparison/nb_codecs.png). This will be FS-FS interconnect (where one side uses portaudio, aka a simple softphone). As I see, it would be worth to use Speex in VBR mode and also to turn on DTX. Would it be possible with current mod_speex? As I saw, currently vad and vbr options are set to 0. Would setting these hardwired options to 1 make the trick? I was thinking about G.726-32 as it was suggested last week on irc, but it has still too high bandwidth requirements and Speex seems to have better MOS values over 10kbps (or 6kbps with VBR). Thanks in advance, Tamas From odermann at googlemail.com Mon Feb 9 06:59:53 2009 From: odermann at googlemail.com (Dennis) Date: Mon, 9 Feb 2009 15:59:53 +0100 Subject: [Freeswitch-users] DTMF: Mute sound for the other side? Message-ID: <5e414ed0902090659p18b970c9y8ba8f1bacda1f3ac@mail.gmail.com> hi, i am having a small problem with the dtmf-sounds... if i press a dtmf digit while i am bridged with another leg, the other side will hear the dtmf sound. this is very annoying and even worse in a conference, when multiple people can press dtmf digits (for (un-)muting themselves or using other functions). is there a way, to NOT let the other side hear the dtmf sound (but of course still make fs listening to it)? thanks for the help dennis From javieraristizabal at gmail.com Mon Feb 9 07:03:28 2009 From: javieraristizabal at gmail.com (=?ISO-8859-1?Q?Javier_Aristiz=E1bal?=) Date: Mon, 9 Feb 2009 10:03:28 -0500 Subject: [Freeswitch-users] connecting to mysql using lua In-Reply-To: <78A54C9A-C807-445B-A3BA-0B09A3B27521@freeswitch.org> References: <78A54C9A-C807-445B-A3BA-0B09A3B27521@freeswitch.org> Message-ID: Hi. Well you need to install luasql, and work only with lua 5.0 or major. You need a ODBC connection to MySQL. And there is an lua script example: ============================================================ #!/usr/local/bin/lua require "luasql.mysql" env = assert (luasql.mysql()) con = assert (env:connect("DB","user","password","localhost)) cur = assert (con:execute"SELECT id, name FROM test") row = cur:fetch ({}, "a") while row do print(string.format("Id: %s, Name: %s", row.id, row.name)) -- reusing the table of results row = cur:fetch (row, "a") end cur:close() con:close() env:close() ========================================================== regards javar -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090209/ab98f942/attachment.html From kerrada2003 at yahoo.com Mon Feb 9 07:08:21 2009 From: kerrada2003 at yahoo.com (Ali Al-Rubaie) Date: Mon, 9 Feb 2009 07:08:21 -0800 (PST) Subject: [Freeswitch-users] SIP Authentication In-Reply-To: Message-ID: <367769.62870.qm@web33705.mail.mud.yahoo.com> Thanks so much Anthony but I have one more question: I was checking the source file sofia_reg.c and it seems that the code had been written iin such a way that FreeSWITCH can authenticate SIP agents based on RFC2069 and RFC2617. Is that conclusion correct? Thanks in advance, Message: 2 Date: Thu, 5 Feb 2009 10:46:54 -0600 From: Anthony Minessale Subject: Re: [Freeswitch-users] SIP Authentication To: freeswitch-users at lists.freeswitch.org Message-ID: <191c3a030902050846o60047c30pa2890707eae386d6 at mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" It's optional for us but it's mandatory for the client if we exercise the option which we have opted to always do =D There is no way in the code to disable sending it because we prefer the more secure version of SIP auth. So again it's a bug in the client for not following the protocol. It would be considered a feature in FreeSWITCH to support limping for the sake of this broken client and we currently do not have any plans for implementing this feature. On Thu, Feb 5, 2009 at 10:34 AM, Ali Al-Rubaie wrote: > > We're using HelpCaster softphone. > > The issue here is that in Digest Authentication, if the server sends the > parameter "qop" in the challenge then the client should respond with the > "cnonce" parameter. The parameter "qop" is optional in Digest Auth. So the > question here is that, can we configure FreeSWITCH so that it will not send > "qop" in the challenge? > > Thanks! > > --- On *Wed, 2/4/09, freeswitch-users-request at lists.freeswitch.org < > freeswitch-users-request at lists.freeswitch.org>* wrote: > > From: freeswitch-users-request at lists.freeswitch.org < > freeswitch-users-request at lists.freeswitch.org> > Subject: Freeswitch-users Digest, Vol 32, Issue 39 > To: freeswitch-users at lists.freeswitch.org > Date: Wednesday, February 4, 2009, 2:05 PM > > Send Freeswitch-users mailing list submissions to > freeswitch-users at lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request at lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner at lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Freeswitch-users digest..." > > > Today's Topics: > > 1. Re: SIP Authentication (Brian West) > 2. Re: origainate through sofia gateway (Michael Collins) > 3. Recording background music and voice is out of sync (Daniel Liang) > 4. Re: Q931 decoding Update (Gopalakrishnan A.N) > 5. mod_limit (Chav Paskov) > 6. Re: mod_limit (Michael Collins) > 7. Re: mod_limit (Chav > Paskov) > 8. Re: mod_limit (Michael Collins) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Wed, 4 Feb 2009 10:52:45 -0600 > From: Brian West > Subject: Re: [Freeswitch-users] SIP Authentication > To: freeswitch-users at lists.freeswitch.org > Message-ID: <7DAC91F6-2FD3-464D-AA81-321EBCADC8C0 at freeswitch.org> > Content-Type: text/plain; charset=US-ASCII; format=flowed; delsp=yes > > What client is this? I also notice we receive port 3458 and reply to > port 1059... > > /b > > On Feb 4, 2009, at 10:17 AM, Ali Al-Rubaie wrote: > > > What I have noted is that the client does not send the values for > > "cnonce" and "nc" in the response. I'm not sure if > this is the > > reason, however how this problem can be solved? > > > > Thanks, > > > > Ali > > > > > ------------------------------ > > Message: > 2 > Date: Wed, 4 Feb 2009 09:41:07 -0800 > From: Michael Collins > Subject: Re: [Freeswitch-users] origainate through sofia gateway > To: freeswitch-users at lists.freeswitch.org > Message-ID: > <87f2f3b90902040941r61d669aaie949aa7cc8578a9a at mail.gmail.com> > Content-Type: text/plain; charset=ISO-8859-1 > > I'll make sure the substance of this is in the wiki and I'll look for > references to the deprecated way and remove those. > -MC > > On Wed, Feb 4, 2009 at 6:09 AM, Anthony Minessale > wrote: > > Where did you learn how to use js this way? > > session.originate is being misused here and is depricated and may be > > removed. > > > > the first arg to session.originate is either undefined or a *different* > > session (the a leg) > > > > session1 = new Session(); > > session1.originate(undefined, > > > "{ignore_early_media=true}user/1008 at 192.168.1.122"); > > > > > session1.setVariable("effective_caller_id_number","fixed0248b"); > > > > //once you have session1 when you originate session2 you pass session1 as > > the arg > > // the effective_caller_id is taken from session1 > > > > session2 = new Session(); > > session2.originate(session1, > "sofia/gateway/halonet/0225490317"); > > > > Anyway this whole code is depricated in favor of this: > > > > session1 = new > Session("{ignore_early_media=true}user/1008 at 192.168.1.122"); > > if (session1.ready()) { > > > session1.setVariable("effective_caller_id_number","fixed0248b"); > > session2 = new Session("sofia/gateway/halonet/0225490317", > session1); > > } > > > > and could be further refactored down to this: > > > > session1 = new > Session("{ignore_early_media=true}user/1008 at 192.168.1.122"); > > if > (session1.ready()) { > > > session1.setVariable("effective_caller_id_number","fixed0248b"); > > session1.execute("bridge", > "sofia/gateway/halonet/0225490317"); > > } > > > > or down to this one line of code that will setup the call detached from > the > > script and exit. > > > > var result = apiExecute("originate", > > > "{effective_caller_id_number=fixed0248b,origination_caller_id_number=1000,ignore_early_media=true}user/1008 at 192.168.1.122 > > bridge:sofia/gateway/halonet/0225490317 inline"); > > > > if you dont care about the result and want to exit even before the call is > > completed. > > > > var result = apiExecute("bgapi", "originate > > > {effective_caller_id_number=fixed0248b,origination_caller_id_number=1000,ignore_early_media=true}user/1008 at 192.168.1.122 > > bridge:sofia/gateway/halonet/0225490317 inline"); > > > > > > > > On Wed, Feb 4, 2009 at > 2:51 AM, Jacek Sokulski > > > wrote: > >> > >> We have tried setting both effective_caller_id_number and > >> origination_caller_id_number: > >> > >> > >> > session1.originate(session1,"{origination_caller_id_number=fixed0248b}sofia/gateway/halonet/0225490317",15); > >> but the problem still exists. The solution we have found for the case > >> when we originate two calls, local and external, is as follow: > >> > >> session1 = new Session(); > >> > session1.originate(session1,"user/1003 at 192.168.1.122",15);//local > >> if(session1.ready()) { > >> session1.execute("execute_extension","00930691688627 > XML > >> default");//external > >> } > >> > >> so the external call goes through the dialplan. > >> It does not work if both calls are external. One possible solution > could > >> > be > >> to pass the originating call through dialplan (loopback?) but we have > not > >> managed > >> to figure out how to do it. > >> > >> Thanks > >> Jacek > >> > >> Dnia 03-02-2009, wto o godzinie 14:31 -0300, Nicolas Brenner pisze: > >> > Oops! Well, fortunately I don't use that voip provider > anymore (nor the > >> > script). > >> > > >> > Thanks Brian. > >> > > >> > Nicolas > >> > > >> > On Tue, Feb 3, 2009 at 2:25 PM, Brian West > wrote: > >> > > YOU should NEVER use this method or call setCallerData at > all you > >> > > should use the correct methods to override the callerid. > >> > > > >> > > If its a B-Leg born from an A-Leg you use these on the on > the A-Leg: > >> > > > >> > > > >> > > > > http://wiki.freeswitch.org/wiki/Channel_Variables#effective_caller_id_name > >> > > > >> > > > http://wiki.freeswitch.org/wiki/Channel_Variables#effective_caller_id_number > >> > > > >> > > If you're originating you use this: > >> > > > >> > > > >> > > > http://wiki.freeswitch.org/wiki/Channel_Variables#origination_caller_id_name > >> > > > >> > > > http://wiki.freeswitch.org/wiki/Channel_Variables#origination_caller_id_number > >> > > > >> > > /b > >> > > >> > _______________________________________________ > >> > Freeswitch-users mailing list > >> > Freeswitch-users at lists.freeswitch.org > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> > http://www.freeswitch.org > >> > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > ------------------------------ > > Message: 3 > Date: Wed, 4 Feb 2009 09:43:10 -0800 > From: "Daniel Liang" > Subject: [Freeswitch-users] Recording background music and voice is > out of sync > To: > Message-ID: > <0B02E756F603CC409EB553879B090CC80A23EBB5 at HPEXCHVS01.exchange.airg> > Content-Type: text/plain; charset="us-ascii" > > What I did was the following: > > First, I sent the > playback command: > > call-command: execute > execute-app-name: playback > execute-app-arg: > > Then I send uuid_record (Sorry, it was not Record command): > > api uuid_record start 120 > > I also tried replacing the playback command with: > api uuid_displace start 0 mux > > But the end results are the same. The recorded user's voice is about 0.5 > second behind the expected result. > > Thanks, > Daniel > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Brian West > Sent: February 3, 2009 6:36 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Recording background music and voice is > outof sync > > Can you show us an example of how you're doing this? Playback and > Record aren't async so > you'll need to show us how you're doing > this. > > Also don't hijack threads you hit replay on the one "Re: [Freeswitch- > users] FreeSwitch setup as a "Dumb" SBC" as the subject, deleted > the > subject and started a new body. That hijacks the thread and that can > cause your problem to go ignored in some cases if people aren't > interested in the thread topic depending on how their reader threads the > emails. > > Please click new message and type freeswitch- users at lists.freeswitch.org > in and then input your subject and body to start a new thread. > > Thanks, > Brian West > FreeSWITCH.org > > > On Feb 3, 2009, at 8:21 PM, Daniel Liang wrote: > > > Hi, > > > > I was trying to record a background music with a user's voice at the > > same time. I did a playback and started recording. But the recorded > > user's voice and the background music is about 0.5 second out of sync. > > > I also tried > to use uuid_displace instead of playback, but I got the > > same result. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090204/2d1124e2/attachment-0001.html > > > ------------------------------ > > Message: 4 > Date: Wed, 4 Feb 2009 23:26:14 +0530 > From: "Gopalakrishnan A.N" > Subject: Re: [Freeswitch-users] Q931 decoding Update > To: freeswitch-users at lists.freeswitch.org > Message-ID: > > <2ea4d47e0902040956v75c5472foa4649c50b7340484 at mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > Hi, > Its a awesome. Can the packet capturing be done with event socket? > > -- > Thank you with regards, > Gopal, > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090204/74f6434a/attachment-0001.html > > > ------------------------------ > > Message: 5 > Date: Wed, 04 Feb 2009 09:59:48 -0800 > From: Chav Paskov > Subject: [Freeswitch-users] mod_limit > To: freeswitch-users at lists.freeswitch.org > Message-ID: <4989D794.1010805 at shaw.ca> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Hi , > is it possible to use mod_limit in case if the end point is not > registered / gateway for > example/. > Regards > Chav > > > > ------------------------------ > > Message: 6 > Date: Wed, 4 Feb 2009 10:06:52 -0800 > From: Michael Collins > Subject: Re: [Freeswitch-users] mod_limit > To: freeswitch-users at lists.freeswitch.org > Message-ID: > <87f2f3b90902041006v7d7d7ff2t9961274905d9d5b0 at mail.gmail.com> > Content-Type: text/plain; charset=ISO-8859-1 > > On Wed, Feb 4, 2009 at 9:59 AM, Chav Paskov wrote: > > Hi , > > is it possible to use mod_limit in case if the end point is not > > registered / gateway for example/. > > Could you add some detail to this question? What are you trying to do? > (mod_limit may or may not work, but there might be another solution > which is why I am asking.) > > -MC > > > Regards > > Chav > > > > _______________________________________________ > > Freeswitch-users mailing list > > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > ------------------------------ > > Message: 7 > Date: Wed, 04 Feb 2009 10:54:56 -0800 > From: Chav Paskov > Subject: Re: [Freeswitch-users] mod_limit > To: freeswitch-users at lists.freeswitch.org > Message-ID: <4989E480.1080105 at shaw.ca> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Michael Collins wrote: > > On Wed, Feb 4, 2009 at 9:59 AM, Chav Paskov > wrote: > > > >> Hi , > >> is it possible to use mod_limit in case if the end point is not > >> registered / gateway for example/. > >> > > > > Could you add some detail to this question? What are you trying to do? > > > (mod_limit may or may not work, but there might be another solution > > which is why I am asking.) > > > > -MC > > > > > >> Regards > >> Chav > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > >> > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > i have few gateways under my ACL that > are allowed to send calls to FS, > but i want to be able to enforce "capacity" policy on the traffic > coming from any one of them depending on total termination capacity on > my termination end. > Let say GW 1 has to be limited to make 10 simultaneous calls while GW2 > could make up to 30 and so on. > Regards > Chav > > > > ------------------------------ > > Message: 8 > Date: Wed, 4 Feb 2009 11:05:09 -0800 > From: Michael Collins > Subject: Re: [Freeswitch-users] mod_limit > To: freeswitch-users at lists.freeswitch.org > Message-ID: > <87f2f3b90902041105l50f51f08t230bab8d69eefb4e at mail.gmail.com> > Content-Type: text/plain; charset=ISO-8859-1 > > On Wed, Feb 4, 2009 at 10:54 AM, Chav Paskov wrote: > > Michael Collins wrote: > >> On Wed, Feb 4, 2009 at 9:59 AM, Chav Paskov > wrote: > >> > >>> Hi > , > >>> is it possible to use mod_limit in case if the end point is not > >>> registered / gateway for example/. > >>> > >> > >> Could you add some detail to this question? What are you trying to do? > >> (mod_limit may or may not work, but there might be another solution > >> which is why I am asking.) > >> > >> -MC > >> > >> > >>> Regards > >>> Chav > >>> > >>> _______________________________________________ > >>> Freeswitch-users mailing list > >>> Freeswitch-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> http://www.freeswitch.org > >>> > >>> > >> > >> _______________________________________________ > >> > Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > >> > > i have few gateways under my ACL that are allowed to send calls to FS, > > but i want to be able to enforce "capacity" policy on the > traffic > > coming from any one of them depending on total termination capacity on > > my termination end. > > Let say GW 1 has to be limited to make 10 simultaneous calls while GW2 > > could make up to 30 and so on. > > I'm sure that this is possible. I don't personally have a way to test > all of this but I know that a number of our users are doing things > like this currently. Can you hop on to the IRC channel? #freeswitch on > irc.freenode.net. A lot of people there can help with > this one. > > -MC (IRC: mercutioviz) > > > Regards > > Chav > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > ------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > End of Freeswitch-users Digest, Vol 32, Issue 39 > ************************************************ > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090205/61200d9e/attachment.html ------------------------------ _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org End of Freeswitch-users Digest, Vol 32, Issue 50 ************************************************ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090209/70114813/attachment-0001.html From anthony.minessale at gmail.com Mon Feb 9 07:09:53 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 9 Feb 2009 09:09:53 -0600 Subject: [Freeswitch-users] Compiling FreeSWITCH for AstLinux In-Reply-To: <87f2f3b90902061316h5fe6e8afw253c95a55ddf3aa0@mail.gmail.com> References: <2d9149cd0902061220i11b87fd9se253109d7a39249a@mail.gmail.com> <87f2f3b90902061316h5fe6e8afw253c95a55ddf3aa0@mail.gmail.com> Message-ID: <191c3a030902090709j7a96e42csfbac48b9de1409f8@mail.gmail.com> How about PBlx I even have the domain name ;) On Fri, Feb 6, 2009 at 3:16 PM, Michael Collins wrote: > > P.S. - Yes, yes I know "AstLinux" isn't the best name for a distro > > with FreeSWITCH. Depending on my success here I have some other > > ideas... > > > > How about KickAstLinux? ;) > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090209/34791ee1/attachment.html From odermann at googlemail.com Mon Feb 9 07:16:15 2009 From: odermann at googlemail.com (Dennis) Date: Mon, 9 Feb 2009 16:16:15 +0100 Subject: [Freeswitch-users] Socket-Event on "originate call"? Message-ID: <5e414ed0902090716y10f95f71o5760dfdf50d1c5cb@mail.gmail.com> hi, i am using socket outbound with fs. if i do an originate over the console, for starting an outbound call without having an inbound call, and send the originate directly to the socket, the socket is first started, if the call is in answer or ringing state. before this, i will not receive any event, because the socket was not started. therefore i will not know, if the target is "busy" (hangup, hangup cause: user busy). it would be very helpful, if the socket would start immediately on an event like "channel originate". thanks for the help dennis From kokoska.rokoska at post.cz Mon Feb 9 07:18:52 2009 From: kokoska.rokoska at post.cz (kokoska rokoska) Date: Mon, 09 Feb 2009 16:18:52 +0100 Subject: [Freeswitch-users] Dynamic Dialplan In-Reply-To: <743112140902081921n2bb1bf49mfe36e5e2193c6b04@mail.gmail.com> References: <743112140902081921n2bb1bf49mfe36e5e2193c6b04@mail.gmail.com> Message-ID: <4990495C.9010008@post.cz> Just my 2c: If you use curl with lighttpd and custom built fastcgi "C" responder (it is really simple with fcgi libs - even I can do it :-) performance could be not that bad. Like I wrote in the past it can handle about 2000 reguest per second (including SQL query wiht simple "postprocessing"). Best regards, kokoska.rokoska Doug Blacksone napsal(a): > Hi, > > Right now, I am working on getting freeswitch configured for our > call-center with more than 1000 agents. There are several areas where > we need the dialplan to be configurable based on some user detail in the > database. Therefore, the dialplan needs to be some-what dynamic > based-on inputs from the database. > > I would like to know from other implementation as to the most scalable > way of doing high performance dynamic dialplan that is super scalable. > > There are three ways I can think of: > > 1. Static dialplan using customized freeswitch mod to access postgres > for data > pros: best performance > cons: harder to program > > 2. Static dialplan using lua to access postgres for data > pros: easy to program, maybe-performance is better than curl > cons: need to search through all the extensions to find one dialplan, > performance is slower than the first one. > > 3. curl-based dialplan using Java Servlet and HTTP > pros: easy to program, freeswitch only gets one extension and no > extension search > cons: performance is slow than the other two > > Is this a correct analysis? > If from a pure performance's perspective, how much performance can a > customized mod gains in comparison to lua? > > For a production system that needs to be highly scalable, what do you > recommend? > > > Thank you very much for any input to our critical design decision. > > Doug > > > > > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From anthony.minessale at gmail.com Mon Feb 9 07:25:57 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 9 Feb 2009 09:25:57 -0600 Subject: [Freeswitch-users] DTMF: Mute sound for the other side? In-Reply-To: <5e414ed0902090659p18b970c9y8ba8f1bacda1f3ac@mail.gmail.com> References: <5e414ed0902090659p18b970c9y8ba8f1bacda1f3ac@mail.gmail.com> Message-ID: <191c3a030902090725q11c8b3eaka6ee5e8565b1a7ea@mail.gmail.com> 1) don't use inband tones for dtmf. 2) post a bounty to have FS clip the audio for x milliseconds when a tone is detected. (you will still hear faint clicks between the start of the tone and when the clipping activates) On Mon, Feb 9, 2009 at 8:59 AM, Dennis wrote: > hi, > > i am having a small problem with the dtmf-sounds... > > if i press a dtmf digit while i am bridged with another leg, the other > side will hear the dtmf sound. > this is very annoying and even worse in a conference, when multiple > people can press dtmf digits (for (un-)muting themselves or using > other functions). > > is there a way, to NOT let the other side hear the dtmf sound (but of > course still make fs listening to it)? > > > thanks for the help > dennis > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090209/bff5e75b/attachment.html From anthony.minessale at gmail.com Mon Feb 9 07:27:38 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 9 Feb 2009 09:27:38 -0600 Subject: [Freeswitch-users] Socket-Event on "originate call"? In-Reply-To: <5e414ed0902090716y10f95f71o5760dfdf50d1c5cb@mail.gmail.com> References: <5e414ed0902090716y10f95f71o5760dfdf50d1c5cb@mail.gmail.com> Message-ID: <191c3a030902090727n44eef7d7x83704b66e628081@mail.gmail.com> when an originate is unsuccessful the failure and the cause code is returned as the reply to the originate request. On Mon, Feb 9, 2009 at 9:16 AM, Dennis wrote: > hi, > > i am using socket outbound with fs. > > if i do an originate over the console, for starting an outbound call > without having an inbound call, and send the originate directly to the > socket, the socket is first started, if the call is in answer or > ringing state. > before this, i will not receive any event, because the socket was not > started. therefore i will not know, if the target is "busy" (hangup, > hangup cause: user busy). > > it would be very helpful, if the socket would start immediately on an > event like "channel originate". > > > thanks for the help > dennis > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090209/05a2c7f0/attachment.html From anthony.minessale at gmail.com Mon Feb 9 07:32:48 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 9 Feb 2009 09:32:48 -0600 Subject: [Freeswitch-users] Can I force the A-leg codec in FS? In-Reply-To: <498FE8B3.8040904@ewetel.de> References: <498FE8B3.8040904@ewetel.de> Message-ID: <191c3a030902090732v30f79c91qb06b3762e84b047@mail.gmail.com> 1) set late-negotation=true in the sofia profile 2) set absolute_codec_string channel variable to the exact codec you want as the first action in your dialplan. On Mon, Feb 9, 2009 at 2:26 AM, Helmut Kuper wrote: > Hi Ken, > > thx for the hint. It looks quite static, so I guess each call (also > internal) are then forced to g711. I'm looking for a dynamic way > depending on destination number. > > regards > helmut > > On 09.02.2009 09:00, Ken Rice wrote: > > Set the codec negotiation to greedy > > > > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090209/51042c84/attachment-0001.html From sprice at gmail.com Sun Feb 8 20:20:54 2009 From: sprice at gmail.com (SP) Date: Sun, 8 Feb 2009 22:20:54 -0600 Subject: [Freeswitch-users] Dynamic Dialplan In-Reply-To: <743112140902081921n2bb1bf49mfe36e5e2193c6b04@mail.gmail.com> References: <743112140902081921n2bb1bf49mfe36e5e2193c6b04@mail.gmail.com> Message-ID: <7e2ac3270902082020r3b8ec2f9of2721a6d1db0243c@mail.gmail.com> Everything you can do in a static dialplan you can do via curl as well. Multiple extensions, search/conditions are allowed. Don't sell the curl short, it's very powerful and can get the ball rolling. On Sun, Feb 8, 2009 at 21:21, Doug Blacksone wrote: > Hi, > > Right now, I am working on getting freeswitch configured for our call-center > with more than 1000 agents. There are several areas where we need the > dialplan to be configurable based on some user detail in the database. > Therefore, the dialplan needs to be some-what dynamic based-on inputs from > the database. > > I would like to know from other implementation as to the most scalable way > of doing high performance dynamic dialplan that is super scalable. > > There are three ways I can think of: > > 1. Static dialplan using customized freeswitch mod to access postgres for > data > pros: best performance > cons: harder to program > > 2. Static dialplan using lua to access postgres for data > pros: easy to program, maybe-performance is better than curl > cons: need to search through all the extensions to find one dialplan, > performance is slower than the first one. > > 3. curl-based dialplan using Java Servlet and HTTP > pros: easy to program, freeswitch only gets one extension and no extension > search > cons: performance is slow than the other two > > Is this a correct analysis? > If from a pure performance's perspective, how much performance can a > customized mod gains in comparison to lua? > > For a production system that needs to be highly scalable, what do you > recommend? > > > Thank you very much for any input to our critical design decision. > > Doug From anthony.minessale at gmail.com Mon Feb 9 07:37:44 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 9 Feb 2009 09:37:44 -0600 Subject: [Freeswitch-users] SIP Authentication In-Reply-To: <367769.62870.qm@web33705.mail.mud.yahoo.com> References: <367769.62870.qm@web33705.mail.mud.yahoo.com> Message-ID: <191c3a030902090737o61cbdf28i1e1ffb91198c504@mail.gmail.com> See this post: http://www.mail-archive.com/freeswitch-dev at lists.freeswitch.org/msg00926.html On Mon, Feb 9, 2009 at 9:08 AM, Ali Al-Rubaie wrote: > Thanks so much Anthony but I have one more question: > > I was checking the source file sofia_reg.c and it seems that the code had > been written iin such a way that FreeSWITCH can authenticate SIP agents > based on RFC2069 and RFC2617. Is that conclusion correct? > > Thanks in advance, > > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090209/ffef0342/attachment.html From nik.middleton at noblesolutions.co.uk Mon Feb 9 07:47:17 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Mon, 9 Feb 2009 15:47:17 -0000 Subject: [Freeswitch-users] Making a system call with LUA In-Reply-To: <1234188428.6040.16.camel@gathern.lan> References: <1234188428.6040.16.camel@gathern.lan> Message-ID: Can I assume that info/functions in lua are all available in the embedded lua in FS? Regards -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Alexandru Nedelcu Sent: 09 February 2009 14:07 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Making a system call with LUA On Mon, 2009-02-09 at 13:30 +0000, Nik Middleton wrote: > In the absence of any directives on lua/mysql, is it possible to > launch a PHP script from lua? All I need to do is pass some data to > update a db record. I don't need to have any links to the call as I > intend to call is in the hang-up callback I'm actually using Javascript, but os.execute should work: http://www.lua.org/pil/22.2.html _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From mike at jerris.com Mon Feb 9 07:59:33 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 9 Feb 2009 10:59:33 -0500 Subject: [Freeswitch-users] Openzap r654 doesn't compile. Error with ozmod_libpri In-Reply-To: <498FE8F5.9060206@ewetel.de> References: <498FDFBD.8050401@ewetel.de> <498FE8F5.9060206@ewetel.de> Message-ID: <14E5C011-BCC8-4F97-99E7-471BD203C14F@jerris.com> It sounds like your automake got screwed up with some new changes. I tried and was unable to reproduce this issue, can you test a fresh checkout and see if you still see this issue? Mike On Feb 9, 2009, at 3:27 AM, Helmut Kuper wrote: > Hello, > > update, when I remove all ozmod_ from ozmod_libpri lines in > Makefile, it > compiles without errors. > > regards > helmut > > On 09.02.2009 08:48, Helmut Kuper wrote: >> Hello, >> >> during some imporvements on q931toPcap as well as debugging my TDM >> PRI >> problem with loosing state sync after some time I updated the code to >> latest trunk (r654). After doing a bootstrap in FS trunk directory I >> tried to compile openzap in libs/openzap. I got this error: >> >> /bin/sh ./libtool --tag=CC --mode=link gcc -g -O2 -o >> ozmod_libpri.la -rpath /usr/local/openzap/mod -lm -lpthread -ldl >> libtool: link: libtool library `ozmod_libpri.la' must begin with >> `lib' >> Try `libtool --help --mode=link' for more information. >> make: *** [ozmod_libpri.la] Error 1 >> >> Additionally I wonder what ozmod_libpri is. Do I have consider it for >> Q931 capturing? >> >> regards >> Helmut >> From helmut.kuper at ewetel.de Mon Feb 9 08:06:55 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Mon, 09 Feb 2009 17:06:55 +0100 Subject: [Freeswitch-users] Openzap r654 doesn't compile. Error with ozmod_libpri In-Reply-To: <14E5C011-BCC8-4F97-99E7-471BD203C14F@jerris.com> References: <498FDFBD.8050401@ewetel.de> <498FE8F5.9060206@ewetel.de> <14E5C011-BCC8-4F97-99E7-471BD203C14F@jerris.com> Message-ID: <4990549F.50207@ewetel.de> Hi Mike, of course I can ... will do it tomorrow. regards helmut On 09.02.2009 16:59, Michael Jerris wrote: > It sounds like your automake got screwed up with some new changes. I > tried and was unable to reproduce this issue, can you test a fresh > checkout and see if you still see this issue? > > Mike > > On Feb 9, 2009, at 3:27 AM, Helmut Kuper wrote: > From intralanman at freeswitch.org Mon Feb 9 08:15:21 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Mon, 09 Feb 2009 11:15:21 -0500 Subject: [Freeswitch-users] Cannot choose Cepstral voice from dialplan In-Reply-To: <498F6751.2000701@versafon.com> References: <498E67BF.3060207@versafon.com> <8B964172-6EEB-4B5B-B4B1-D35BFEAEB460@freeswitch.org> <498F6751.2000701@versafon.com> Message-ID: <49905699.40605@freeswitch.org> pauld wrote: > The libs are there with correct symlinks, see below. I tested both > voices directly via swift command, works fine. > Any other ideas? > It's Cepstral 5.1, FS 1.0.2. > Unpredictable issues have been reported using cepstral 5 with FreeSWITCH. I'd suggest using their 4.x release. If you have a really good reason to only use 5, then you might entice someone to work on reliable Cepstral 5 integration with a bounty... upgrade to FreeSWITCH trunk first though. -Ray From msc at freeswitch.org Mon Feb 9 09:04:57 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 9 Feb 2009 09:04:57 -0800 Subject: [Freeswitch-users] Cannot choose Cepstral voice from dialplan In-Reply-To: <498F6751.2000701@versafon.com> References: <498E67BF.3060207@versafon.com> <8B964172-6EEB-4B5B-B4B1-D35BFEAEB460@freeswitch.org> <498F6751.2000701@versafon.com> Message-ID: <87f2f3b90902090904i72e37241hb5d4808eb31fcc18@mail.gmail.com> On Sun, Feb 8, 2009 at 3:14 PM, pauld wrote: > The libs are there with correct symlinks, see below. I tested both > voices directly via swift command, works fine. > Any other ideas? > It's Cepstral 5.1, FS 1.0.2. > Well, first I recommend getting on latest trunk if that's at all possible for you. The devs have made a ton of improvements in the last five weeks. Second, this might actually be an issue with FS looking in its own lib directory for these .so files. Try a symlink from /usr/local/freeswitch/lib to your /opt/swift/lib (or whatever the name is) dir for each .so file. However, I think Raymond is correct - some weirdness has been reported by some Cepstral users on 5.1. We'd definitely like to hear about your experiences if and when you get it running. -MC From msc at freeswitch.org Mon Feb 9 09:22:20 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 9 Feb 2009 09:22:20 -0800 Subject: [Freeswitch-users] Making a system call with LUA In-Reply-To: References: <1234188428.6040.16.camel@gathern.lan> Message-ID: <87f2f3b90902090922x2950912xe44f4d750d57d4c2@mail.gmail.com> On Mon, Feb 9, 2009 at 7:47 AM, Nik Middleton wrote: > Can I assume that info/functions in lua are all available in the > embedded lua in FS? > > Regards > Generally speaking that is a safe assumption. -MC From msc at freeswitch.org Mon Feb 9 09:28:30 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 9 Feb 2009 09:28:30 -0800 Subject: [Freeswitch-users] Newbie - point me in the right direction In-Reply-To: <008F70F6-818D-476B-9DE9-33DA018D37D9@whitesmiths.com> References: <008F70F6-818D-476B-9DE9-33DA018D37D9@whitesmiths.com> Message-ID: <87f2f3b90902090928hd683219od44f1f6b075112c8@mail.gmail.com> > I have two Asterisk boxes, A1 & A2, each running a separate telephony > app. > We have an external SIP service with DID's NNNNN200 -> NNNNN299. > We want to direct the incoming SIP calls so that the DID's NNNNN200 -> > NNNNN219 go to Asterisk server A1 and NNNNN220 -> NNNNN299 to Asterisk > server A2. > Yes we really just want the calls switched on the DID. > Are you thinking about using FreeSWITCH to direct these calls? Something like this? SIP Provider <--> FS <--+--> A1 +--> A2 I just want to make sure that we understand what you are trying to accomplish and why you might need FS in this scenario... -MC From intralanman at freeswitch.org Mon Feb 9 09:31:53 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Mon, 09 Feb 2009 12:31:53 -0500 Subject: [Freeswitch-users] Dynamic Dialplan In-Reply-To: <4990495C.9010008@post.cz> References: <743112140902081921n2bb1bf49mfe36e5e2193c6b04@mail.gmail.com> <4990495C.9010008@post.cz> Message-ID: <49906889.9030801@freeswitch.org> kokoska rokoska wrote: > Just my 2c: > > If you use curl with lighttpd and custom built fastcgi "C" responder (it > is really simple with fcgi libs - even I can do it :-) performance could > be not that bad. hmmm, mod_xml_curl using C, interesting thought.. all of the complexities of writing your own module without the nice structured FS API... although, as a benefit, i guess you do get a little extra latency ;-) -Ray From Daniell at airg.com Mon Feb 9 09:40:16 2009 From: Daniell at airg.com (Daniel Liang) Date: Mon, 9 Feb 2009 09:40:16 -0800 Subject: [Freeswitch-users] Recording background music and voice is outof sync In-Reply-To: <341AE5F8-20B2-4CF3-92EE-7311B3E71C7E@freeswitch.org> References: <019501c985ac$4f00ee60$ed02cb20$@net><4987E527.1040909@laposte.net><022001c9862f$efd4b7d0$cf7e2770$@net><0B02E756F603CC409EB553879B090CC80A23EB2F@HPEXCHVS01.exchange.airg> <341AE5F8-20B2-4CF3-92EE-7311B3E71C7E@freeswitch.org> Message-ID: <0B02E756F603CC409EB553879B090CC80A23F578@HPEXCHVS01.exchange.airg> Hi Brian, I have created a new thread regarding this issue a few days ago, you may have missed it. So, I am reposting the same content there: What I did was the following: First, I sent the playback command: call-command: execute execute-app-name: playback execute-app-arg: Then I sent uuid_record (Sorry, it was not Record command): api uuid_record start 120 I also tried replacing the playback command with: api uuid_displace start 0 mux But the end results are the same. The recorded user's voice is about 0.5 second behind the expected result. Thanks, Daniel -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: February 3, 2009 6:36 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Recording background music and voice is outof sync Can you show us an example of how you're doing this? Playback and Record aren't async so you'll need to show us how you're doing this. Also don't hijack threads you hit replay on the one "Re: [Freeswitch- users] FreeSwitch setup as a "Dumb" SBC" as the subject, deleted the subject and started a new body. That hijacks the thread and that can cause your problem to go ignored in some cases if people aren't interested in the thread topic depending on how their reader threads the emails. Please click new message and type freeswitch- users at lists.freeswitch.org in and then input your subject and body to start a new thread. Thanks, Brian West FreeSWITCH.org On Feb 3, 2009, at 8:21 PM, Daniel Liang wrote: > Hi, > > I was trying to record a background music with a user's voice at the > same time. I did a playback and started recording. But the recorded > user's voice and the background music is about 0.5 second out of sync. > I also tried to use uuid_displace instead of playback, but I got the > same result. _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From helmut.kuper at ewetel.de Mon Feb 9 09:58:52 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Mon, 09 Feb 2009 18:58:52 +0100 Subject: [Freeswitch-users] Openzap r654 doesn't compile. Error with ozmod_libpri In-Reply-To: <4990549F.50207@ewetel.de> References: <498FDFBD.8050401@ewetel.de> <498FE8F5.9060206@ewetel.de> <14E5C011-BCC8-4F97-99E7-471BD203C14F@jerris.com> <4990549F.50207@ewetel.de> Message-ID: <49906EDC.4050408@ewetel.de> Hello, well, tomorrow is today ;) and so I compiled a fresh truch checout of FS and all went well ... Any idea to get my old trunk dir clean again without doing a sure or current? I don't want to clean up my binary directory due to a "make sure" ... thx for your help. And again: FS is a really nice piece of software. The more I crawl into it the more I'm impressed :) regards Helmut From kokoska.rokoska at post.cz Mon Feb 9 10:03:18 2009 From: kokoska.rokoska at post.cz (kokoska rokoska) Date: Mon, 09 Feb 2009 19:03:18 +0100 Subject: [Freeswitch-users] Dynamic Dialplan In-Reply-To: <49906889.9030801@freeswitch.org> References: <743112140902081921n2bb1bf49mfe36e5e2193c6b04@mail.gmail.com> <4990495C.9010008@post.cz> <49906889.9030801@freeswitch.org> Message-ID: <49906FE6.8000008@post.cz> Raymond Chandler napsal(a): > kokoska rokoska wrote: >> Just my 2c: >> >> If you use curl with lighttpd and custom built fastcgi "C" responder (it >> is really simple with fcgi libs - even I can do it :-) performance could >> be not that bad. > hmmm, mod_xml_curl using C, interesting thought.. May be not the best way, but very simple. Well, it depends on what you have to do, but "directory" serving based on DB queries (this what I'm using it for) is very simple - just few lines of code. > all of the > complexities of writing your own module without the nice structured FS > API... I should say I have no idea how hard is to write custom FreeSWITCH module (may be I should try it :-), but the FS code is very nice! > although, as a benefit, i guess you do get a little extra latency ;-) > :-) Yes, you are right. And as a bonus some CPU utilization... Like I wrote above, I didn't say it is faster, but IMO it is very simple and not as slow as it looks (when using apache + php + apc). Best regards, kokoska.rokoska From helmut.kuper at ewetel.de Mon Feb 9 10:17:11 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Mon, 09 Feb 2009 19:17:11 +0100 Subject: [Freeswitch-users] mod_openzap stops working after some calls In-Reply-To: <191c3a030902060802r424f8a22tb4887dff7c6aead5@mail.gmail.com> References: <498C2DC3.40701@ewetel.de> <191c3a030902060802r424f8a22tb4887dff7c6aead5@mail.gmail.com> Message-ID: <49907327.6010703@ewetel.de> Hello Anthony, :D yes that's what I'm doing ... beneath some code changes in openzap ... So I found a real timestamp in pcap is quite usefull if you have more than one call at a time ... I added that function today. It uses "libapr-1" functions. Unfortunately I introduced a dependency to libs/apr to openzap by that. If it delivers micro seconds, maybe it's better to use zap_time_now(). Have to check that tomorrow. I agree there are some problems in maintaining channel states correctly. Once a day I have to restart FS. I get "TOMANYCALLS" errors, no matching channels on RELEASE, SETUP duplicates and "oz dump 1" shows more and more channels with states other than DOWN, even, when no current calls are there. I did some timebased changes in ozmod_isdn SETUP handling and hope it helps out until state timers a available. If it works I would like to upload it to trunk, if you allow. regards helmut On 06.02.2009 17:02, Anthony Minessale wrote: > I think we have some trouble surviving issues. > So when everything is ok we do fine but if something goes wrong we > don't recover. > We are still missing state timers in the q931. > > maybe you can use your new pcap thing to see what goes wrong =D From mike at jerris.com Mon Feb 9 10:28:33 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 9 Feb 2009 13:28:33 -0500 Subject: [Freeswitch-users] Openzap r654 doesn't compile. Error with ozmod_libpri In-Reply-To: <49906EDC.4050408@ewetel.de> References: <498FDFBD.8050401@ewetel.de> <498FE8F5.9060206@ewetel.de> <14E5C011-BCC8-4F97-99E7-471BD203C14F@jerris.com> <4990549F.50207@ewetel.de> <49906EDC.4050408@ewetel.de> Message-ID: a fresh bootstrap and configure should fix it ... it may do it for you now if you just update. Mike On Feb 9, 2009, at 12:58 PM, Helmut Kuper wrote: > Hello, > > well, tomorrow is today ;) and so I compiled a fresh truch checout > of FS > and all went well ... Any idea to get my old trunk dir clean again > without doing a sure or current? I don't want to clean up my binary > directory due to a "make sure" ... > > thx for your help. And again: FS is a really nice piece of software. > The > more I crawl into it the more I'm impressed :) From mike at jerris.com Mon Feb 9 10:30:35 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 9 Feb 2009 13:30:35 -0500 Subject: [Freeswitch-users] mod_openzap stops working after some calls In-Reply-To: <49907327.6010703@ewetel.de> References: <498C2DC3.40701@ewetel.de> <191c3a030902060802r424f8a22tb4887dff7c6aead5@mail.gmail.com> <49907327.6010703@ewetel.de> Message-ID: <7CEFE4F8-7461-4DF3-ABE0-C0B818466AD0@jerris.com> We can not add apr dependency in openzap, we should use the native openzap calls instead. If there is anything you NEED that you don't have, please let me know and we will try to add replacement functions. Mike On Feb 9, 2009, at 1:17 PM, Helmut Kuper wrote: > Hello Anthony, > > > :D yes that's what I'm doing ... beneath some code changes in openzap > ... So I found a real timestamp in pcap is quite usefull if you have > more than one call at a time ... I added that function today. It uses > "libapr-1" functions. Unfortunately I introduced a dependency to > libs/apr to openzap by that. If it delivers micro seconds, maybe it's > better to use zap_time_now(). Have to check that tomorrow. > > I agree there are some problems in maintaining channel states > correctly. > Once a day I have to restart FS. I get "TOMANYCALLS" errors, no > matching > channels on RELEASE, SETUP duplicates and "oz dump 1" shows more and > more channels with states other than DOWN, even, when no current calls > are there. I did some timebased changes in ozmod_isdn SETUP handling > and hope it helps out until state timers a available. If it works I > would like to upload it to trunk, if you allow. > > regards > helmut > > > On 06.02.2009 17:02, Anthony Minessale wrote: >> I think we have some trouble surviving issues. >> So when everything is ok we do fine but if something goes wrong we >> don't recover. >> We are still missing state timers in the q931. >> >> maybe you can use your new pcap thing to see what goes wrong =D From helmut.kuper at ewetel.de Mon Feb 9 10:40:27 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Mon, 09 Feb 2009 19:40:27 +0100 Subject: [Freeswitch-users] mod_openzap stops working after some calls In-Reply-To: <7CEFE4F8-7461-4DF3-ABE0-C0B818466AD0@jerris.com> References: <498C2DC3.40701@ewetel.de> <191c3a030902060802r424f8a22tb4887dff7c6aead5@mail.gmail.com> <49907327.6010703@ewetel.de> <7CEFE4F8-7461-4DF3-ABE0-C0B818466AD0@jerris.com> Message-ID: <4990789B.40405@ewetel.de> Hi Mike, I would like to have a function which gives current time in sec, usec since unix epoch. It's only for pcap timestamp. I found a zap_time_now() somewhere in openzap maybe it helps ... regards helmut From anthony.minessale at gmail.com Mon Feb 9 11:45:38 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 9 Feb 2009 13:45:38 -0600 Subject: [Freeswitch-users] Dynamic Dialplan In-Reply-To: <49906FE6.8000008@post.cz> References: <743112140902081921n2bb1bf49mfe36e5e2193c6b04@mail.gmail.com> <4990495C.9010008@post.cz> <49906889.9030801@freeswitch.org> <49906FE6.8000008@post.cz> Message-ID: <191c3a030902091145i1fffdbdcoe49e16366dc84acf@mail.gmail.com> That's why I chose mod_xml_curl as a demo for the xml_hook api. It's not only a demo, it's rather functional =D You have 2 choices other than using the stuff we already have in tree. 1) write a custom dialplan module, this module gets a single callback function a dialplan_hunt function that has the session and the caller profile. you can see from mod_enum or mod_dialplan_xml how this can be used to make your own module that looks in a db and returns instructions to FS on the fly. 2) write a custom xml_hook and use it with mod_dialplan_xml, this type of module embeds itself into the xml lookups so when something tries to find something in the xml registry, your function is called and you can do your db lookups and generate the xml returned as binary xml obj built from a result of the query. This is more powerfule because it allows you to pre-empt any xml lookups so you can deliver directory, config, dialplan, phrase macros, etc mod_xml_curl is an example of #2, it turns the xml_req into a url req and feeds the xml returned over the http socket into an xml object and returns it as the result in place of the static contents of the xml file. On Mon, Feb 9, 2009 at 12:03 PM, kokoska rokoska wrote: > > > > Raymond Chandler napsal(a): > > kokoska rokoska wrote: > >> Just my 2c: > >> > >> If you use curl with lighttpd and custom built fastcgi "C" responder (it > >> is really simple with fcgi libs - even I can do it :-) performance could > >> be not that bad. > > hmmm, mod_xml_curl using C, interesting thought.. > > May be not the best way, but very simple. > Well, it depends on what you have to do, but "directory" serving based > on DB queries (this what I'm using it for) is very simple - just few > lines of code. > > > all of the > > complexities of writing your own module without the nice structured FS > > API... > > I should say I have no idea how hard is to write custom FreeSWITCH > module (may be I should try it :-), but the FS code is very nice! > > > although, as a benefit, i guess you do get a little extra latency ;-) > > > > :-) Yes, you are right. And as a bonus some CPU utilization... > > Like I wrote above, I didn't say it is faster, but IMO it is very simple > and not as slow as it looks (when using apache + php + apc). > > Best regards, > > kokoska.rokoska > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090209/e232d7c1/attachment.html From anthony.minessale at gmail.com Mon Feb 9 12:03:34 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 9 Feb 2009 14:03:34 -0600 Subject: [Freeswitch-users] Global Variables forgotten throughthepubliccontext? In-Reply-To: <51C9B370-500D-4D95-A515-E9EEF1705014@freeswitch.org> References: <0B4E2726927041D09D0425DA0242C805@UVix> <9D0CFA79-C120-4D66-B48C-7A5641B07AF4@freeswitch.org> <5B1ED1D4834B4719B309FAC03F1595C0@UVix> <03BD385F-E942-4B32-98B0-0A29E13BC5D5@freeswitch.org> <9F926F21CCB64A49A6EFF4425B111F67@UVix> <51C9B370-500D-4D95-A515-E9EEF1705014@freeswitch.org> Message-ID: <191c3a030902091203j3accbda1o1be699d0025cf2da@mail.gmail.com> should be fixed in latest trunk On Sat, Feb 7, 2009 at 7:56 PM, Brian West wrote: > You have a \ somewhere in your path... which doesn't make sense... you're > on windows. > > Can you open a jira... I think this was the cause > http://fisheye.freeswitch.org/browse/FreeSWITCH/src/mod/formats/mod_sndfile/mod_sndfile.c?r1=11090&r2=11601 > > /b > > > On Feb 7, 2009, at 6:46 PM, UV wrote: > > Yeah, I have all the sounds installed. I don't think it's that. > I'm getting error messages such as "[ERR] mod_sndfile.c:185 > sndfile_file_open() Error Opening File [E:\FS/sounds/en/us/callie\voicemail/ > *8000\16000*\vm-goodbye.w] [System error : The system cannot find the path > specified.]" all across the board. The only thing still working is MoH? This > started from one of yesterday's builds (r11665 ? 11678). > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090209/c267bc68/attachment-0001.html From nik.middleton at noblesolutions.co.uk Mon Feb 9 12:10:15 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Mon, 9 Feb 2009 20:10:15 -0000 Subject: [Freeswitch-users] DTMF not being recognised Message-ID: Hi Guys, I have an IVR that's working fine on internal extensions, but when a call is via my sip GW, they're not being trapped. I have tried the following in the gw profile References: Message-ID: Further to this message, DTMF works with PMCU but not with PMCA which is the native format for this sip provider. Regards ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Nik Middleton Sent: 09 February 2009 20:10 To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] DTMF not being recognised Hi Guys, I have an IVR that's working fine on internal extensions, but when a call is via my sip GW, they're not being trapped. I have tried the following in the gw profile References: <0B4E2726927041D09D0425DA0242C805@UVix> <9D0CFA79-C120-4D66-B48C-7A5641B07AF4@freeswitch.org> <5B1ED1D4834B4719B309FAC03F1595C0@UVix> <03BD385F-E942-4B32-98B0-0A29E13BC5D5@freeswitch.org> <9F926F21CCB64A49A6EFF4425B111F67@UVix> <51C9B370-500D-4D95-A515-E9EEF1705014@freeswitch.org> <191c3a030902091203j3accbda1o1be699d0025cf2da@mail.gmail.com> Message-ID: <191c3a030902091318wf64698n77bb95c75338f059@mail.gmail.com> this was the wrong thread, i have no idea if this is fixed or is even a real issue. On Mon, Feb 9, 2009 at 2:03 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > should be fixed in latest trunk > > On Sat, Feb 7, 2009 at 7:56 PM, Brian West wrote: > >> You have a \ somewhere in your path... which doesn't make sense... you're >> on windows. >> >> Can you open a jira... I think this was the cause >> http://fisheye.freeswitch.org/browse/FreeSWITCH/src/mod/formats/mod_sndfile/mod_sndfile.c?r1=11090&r2=11601 >> >> /b >> >> >> On Feb 7, 2009, at 6:46 PM, UV wrote: >> >> Yeah, I have all the sounds installed. I don't think it's that. >> I'm getting error messages such as "[ERR] mod_sndfile.c:185 >> sndfile_file_open() Error Opening File [E:\FS/sounds/en/us/callie\voicemail/ >> *8000\16000*\vm-goodbye.w] [System error : The system cannot find the >> path specified.]" all across the board. The only thing still working is MoH? >> This started from one of yesterday's builds (r11665 ? 11678). >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090209/7c706b15/attachment.html From msc at freeswitch.org Mon Feb 9 13:26:31 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 9 Feb 2009 13:26:31 -0800 Subject: [Freeswitch-users] DTMF not being recognised In-Reply-To: References: Message-ID: <87f2f3b90902091326t34847d95qd763ce55efcbe9b3@mail.gmail.com> On Mon, Feb 9, 2009 at 12:21 PM, Nik Middleton wrote: > Further to this message, DTMF works with PMCU but not with PMCA which is the > native format for this sip provider. > Any chance you could get some debug information? I'm wondering what is actually being sent vs. what is actually being received. A pcap at the far end to compare with a pcap at the near end would be quite enlightening. -MC From nik.middleton at noblesolutions.co.uk Mon Feb 9 13:34:13 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Mon, 9 Feb 2009 21:34:13 -0000 Subject: [Freeswitch-users] DTMF not being recognized In-Reply-To: <87f2f3b90902091326t34847d95qd763ce55efcbe9b3@mail.gmail.com> References: <87f2f3b90902091326t34847d95qd763ce55efcbe9b3@mail.gmail.com> Message-ID: Forgive me, I'm not sure how I get that info with FS, can you enlighten me? DTMF also works with GSM and others, but not Alaw Regards, -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 09 February 2009 21:27 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] DTMF not being recognised On Mon, Feb 9, 2009 at 12:21 PM, Nik Middleton wrote: > Further to this message, DTMF works with PMCU but not with PMCA which is the > native format for this sip provider. > Any chance you could get some debug information? I'm wondering what is actually being sent vs. what is actually being received. A pcap at the far end to compare with a pcap at the near end would be quite enlightening. -MC _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From msc at freeswitch.org Mon Feb 9 13:38:23 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 9 Feb 2009 13:38:23 -0800 Subject: [Freeswitch-users] DTMF not being recognized In-Reply-To: References: <87f2f3b90902091326t34847d95qd763ce55efcbe9b3@mail.gmail.com> Message-ID: <87f2f3b90902091338k126a839dya747b34fe5806aad@mail.gmail.com> On Mon, Feb 9, 2009 at 1:34 PM, Nik Middleton wrote: > Forgive me, I'm not sure how I get that info with FS, can you enlighten > me? > I was thinking of something like Wireshark. You can also check out this: http://wiki.freeswitch.org/wiki/Reporting_Bugs#Capturing_RTP_With_tshark_.28Advanced.29 Being able to see what *actually* is going out over the wire (or coming in on the wire) can take much of the guess work out of debugging. -MC From blake.france at gmail.com Mon Feb 9 10:04:28 2009 From: blake.france at gmail.com (Blake France) Date: Mon, 09 Feb 2009 12:04:28 -0600 Subject: [Freeswitch-users] Recording play end of sound file again Message-ID: <4990702C.9090305@gmail.com> Whenever I try to record and IVR or Voicemail Greeting, it will record and playback, but playback does something like this. "Please leave a message" ... "Message" It plays the end of the sound file AGAIN after playing the sound file. I've tried leaving extra time before and after speaking, but it still does this. Anyone ran into this issue? From john at argv.co.uk Mon Feb 9 14:50:52 2009 From: john at argv.co.uk (John Daragon) Date: Mon, 09 Feb 2009 22:50:52 +0000 Subject: [Freeswitch-users] BT IPExchange Interoperability Testing In-Reply-To: References: Message-ID: <4990B34C.2070505@argv.co.uk> Hi; We're looking to set up a CP which will interact with BT's 21CN network using the IPX gateway. We're running through the test scenarios (which, unfortunately, we have under NDA) now. Just wondering if anyone out there has already passed the test suite with Freeswitch ? jd -- John Daragon argv[0] limited Lambs Lawn Cottage, Staple Fitzpaine, Taunton, TA3 5SL, UK Registered in England Company Number 02947782 v +44 (0) 1460 234068 f +44 (0) 1460 234069 m +44 (0) 7836 576127 From brian at freeswitch.org Mon Feb 9 15:52:10 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 9 Feb 2009 17:52:10 -0600 Subject: [Freeswitch-users] Recording play end of sound file again In-Reply-To: <4990702C.9090305@gmail.com> References: <4990702C.9090305@gmail.com> Message-ID: Can you tell me what SVN rev you're on? /b On Feb 9, 2009, at 12:04 PM, Blake France wrote: > Whenever I try to record and IVR or Voicemail Greeting, it will record > and playback, but playback does something like this. > > "Please leave a message" ... "Message" > > It plays the end of the sound file AGAIN after playing the sound file. > I've tried leaving extra time before and after speaking, but it still > does this. Anyone ran into this issue? From brian at freeswitch.org Mon Feb 9 15:54:48 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 9 Feb 2009 17:54:48 -0600 Subject: [Freeswitch-users] BT IPExchange Interoperability Testing In-Reply-To: <4990B34C.2070505@argv.co.uk> References: <4990B34C.2070505@argv.co.uk> Message-ID: Yes search the mailing list people have interoped with BT in record time. On another note you hijacked the "DTMF not being recognized" by clicking reply, deleting the text and changing the subject. Please try not to do that in the future, click "new message" input freeswitch-users at lists.freeswitch.org then type your subject and message then click send. Your email client echo's back the headers that causes the mailing list server and many email clients to thread the message properly. /b On Feb 9, 2009, at 4:50 PM, John Daragon wrote: > Hi; > > We're looking to set up a CP which will interact with BT's 21CN > network > using the IPX gateway. > > We're running through the test scenarios (which, unfortunately, we > have > under NDA) now. > > Just wondering if anyone out there has already passed the test suite > with Freeswitch ? > > jd > > -- > John Daragon argv[0] limited > Lambs Lawn Cottage, Staple Fitzpaine, Taunton, TA3 5SL, UK > Registered in England Company Number 02947782 > v +44 (0) 1460 234068 f +44 (0) 1460 234069 m +44 (0) 7836 576127 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From shannon at sacredhearts.us Mon Feb 9 16:04:10 2009 From: shannon at sacredhearts.us (Shannon) Date: Mon, 9 Feb 2009 18:04:10 -0600 Subject: [Freeswitch-users] BT IPExchange Interoperability Testing In-Reply-To: References: <4990B34C.2070505@argv.co.uk> Message-ID: <7e2ac3270902091604y5e765b04l4934aa9088516abe@mail.gmail.com> Test A - proper user list manners -FAIL :) On 2/9/09, Brian West wrote: > Yes search the mailing list people have interoped with BT in record > time. On another note you hijacked the "DTMF not being recognized" by > clicking reply, deleting the text and changing the subject. Please > try not to do that in the future, click "new message" input > freeswitch-users at lists.freeswitch.org > then type your subject and message then click send. Your email > client echo's back the headers that causes the mailing list server and > many email clients to thread the message properly. > > /b > > On Feb 9, 2009, at 4:50 PM, John Daragon wrote: > >> Hi; >> >> We're looking to set up a CP which will interact with BT's 21CN >> network >> using the IPX gateway. >> >> We're running through the test scenarios (which, unfortunately, we >> have >> under NDA) now. >> >> Just wondering if anyone out there has already passed the test suite >> with Freeswitch ? >> >> jd >> >> -- >> John Daragon argv[0] limited >> Lambs Lawn Cottage, Staple Fitzpaine, Taunton, TA3 5SL, UK >> Registered in England Company Number 02947782 >> v +44 (0) 1460 234068 f +44 (0) 1460 234069 m +44 (0) 7836 576127 >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Shannon From brian at freeswitch.org Mon Feb 9 16:13:51 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 9 Feb 2009 18:13:51 -0600 Subject: [Freeswitch-users] BT IPExchange Interoperability Testing In-Reply-To: <7e2ac3270902091604y5e765b04l4934aa9088516abe@mail.gmail.com> References: <4990B34C.2070505@argv.co.uk> <7e2ac3270902091604y5e765b04l4934aa9088516abe@mail.gmail.com> Message-ID: <7C6E76E6-4CCB-45DC-8A91-561F3D575C9E@freeswitch.org> John, Here is the post http://lists.freeswitch.org/pipermail/freeswitch-users/2007-December/001825.html Shannon, I want to make sure everyone knows that list etiquette is critical to keep the SNR low. ;) Anyway welcome to FreeSWITCH, sit back, relax and enjoy the ride... ;) /b On Feb 9, 2009, at 6:04 PM, Shannon wrote: > Test A - proper user list manners -FAIL :) > > > On 2/9/09, Brian West wrote: >> Yes search the mailing list people have interoped with BT in record >> time. On another note you hijacked the "DTMF not being recognized" >> by >> clicking reply, deleting the text and changing the subject. Please >> try not to do that in the future, click "new message" input >> freeswitch-users at lists.freeswitch.org >> then type your subject and message then click send. Your email >> client echo's back the headers that causes the mailing list server >> and >> many email clients to thread the message properly. >> >> /b From brian at freeswitch.org Mon Feb 9 16:15:47 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 9 Feb 2009 18:15:47 -0600 Subject: [Freeswitch-users] BT IPExchange Interoperability Testing In-Reply-To: <7e2ac3270902091604y5e765b04l4934aa9088516abe@mail.gmail.com> References: <4990B34C.2070505@argv.co.uk> <7e2ac3270902091604y5e765b04l4934aa9088516abe@mail.gmail.com> Message-ID: <691796D7-3778-411A-BCE1-6493EDA6D6FE@freeswitch.org> What is funny the post about this from David Knell was also a thread hijack :P /b On Feb 9, 2009, at 6:04 PM, Shannon wrote: > Test A - proper user list manners -FAIL :) > > > On 2/9/09, Brian West wrote: >> Yes search the mailing list people have interoped with BT in record >> time. On another note you hijacked the "DTMF not being recognized" >> by >> clicking reply, deleting the text and changing the subject. Please >> try not to do that in the future, click "new message" input >> freeswitch-users at lists.freeswitch.org >> then type your subject and message then click send. Your email >> client echo's back the headers that causes the mailing list server >> and >> many email clients to thread the message properly. >> >> /b -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090209/d8cac34e/attachment.html From dave at 3c.co.uk Mon Feb 9 23:04:57 2009 From: dave at 3c.co.uk (David Knell) Date: Tue, 10 Feb 2009 07:04:57 +0000 Subject: [Freeswitch-users] Thread hijacking and BT interop In-Reply-To: <691796D7-3778-411A-BCE1-6493EDA6D6FE@freeswitch.org> References: <4990B34C.2070505@argv.co.uk> <7e2ac3270902091604y5e765b04l4934aa9088516abe@mail.gmail.com> <691796D7-3778-411A-BCE1-6493EDA6D6FE@freeswitch.org> Message-ID: <9E97ECB0-825E-4622-85CE-D1961DE2A019@3c.co.uk> Oops - I did it again ;-) --Dave From dave at 3c.co.uk Mon Feb 9 23:19:00 2009 From: dave at 3c.co.uk (David Knell) Date: Tue, 10 Feb 2009 07:19:00 +0000 Subject: [Freeswitch-users] BT IPExchange Interoperability Testing In-Reply-To: <4990B34C.2070505@argv.co.uk> References: <4990B34C.2070505@argv.co.uk> Message-ID: Hi John, I think we had a chat at a show at Olympia(?) a couple of years back. We did an IPX interconnect some 12 months ago - it all went pretty well. I'll give you a call later on: depending on where you are in the process, we might be able to save you a pound or two. Cheers -- Dave > Hi; > > We're looking to set up a CP which will interact with BT's 21CN > network > using the IPX gateway. > > We're running through the test scenarios (which, unfortunately, we > have > under NDA) now. > > Just wondering if anyone out there has already passed the test suite > with Freeswitch ? > > jd > > -- > John Daragon argv[0] limited > Lambs Lawn Cottage, Staple Fitzpaine, Taunton, TA3 5SL, UK > Registered in England Company Number 02947782 > v +44 (0) 1460 234068 f +44 (0) 1460 234069 m +44 (0) 7836 576127 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From kristian.kielhofner at gmail.com Mon Feb 9 23:43:49 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Tue, 10 Feb 2009 02:43:49 -0500 Subject: [Freeswitch-users] FreeSWITCH uclibc segfault Message-ID: <2d9149cd0902092343k4109d132v9d974c77e7ac8de0@mail.gmail.com> I don't think this is worth filing a bug for (yet)... FS rev 11655 segfaults with AstLinux (uClibc). Backtrace: http://astbuild.star2star.com/astlinux-freeswitch-segfault.txt I'm sorry it doesn't have all the symbols... Everything except FS is stripped. Configure options: http://astlinux.svn.sourceforge.net/viewvc/astlinux/branches/astlinux-freeswitch/package/freeswitch/freeswitch.mk FreeSWITCH compiles cleanly. Are there any known issues with uclibc (couldn't find anything on Jira) or did I do something stupid? Thanks! -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From krice at suspicious.org Mon Feb 9 23:47:16 2009 From: krice at suspicious.org (Ken Rice) Date: Tue, 10 Feb 2009 01:47:16 -0600 Subject: [Freeswitch-users] FreeSWITCH uclibc segfault In-Reply-To: <2d9149cd0902092343k4109d132v9d974c77e7ac8de0@mail.gmail.com> Message-ID: There arent known issues cause I don't think anyone else has tried it hah > From: Kristian Kielhofner > Reply-To: > Date: Tue, 10 Feb 2009 02:43:49 -0500 > To: > Subject: [Freeswitch-users] FreeSWITCH uclibc segfault > > I don't think this is worth filing a bug for (yet)... > > FS rev 11655 segfaults with AstLinux (uClibc). > > Backtrace: > http://astbuild.star2star.com/astlinux-freeswitch-segfault.txt > > I'm sorry it doesn't have all the symbols... Everything except FS is > stripped. > > Configure options: > http://astlinux.svn.sourceforge.net/viewvc/astlinux/branches/astlinux-freeswit > ch/package/freeswitch/freeswitch.mk > > FreeSWITCH compiles cleanly. > > Are there any known issues with uclibc (couldn't find anything on > Jira) or did I do something stupid? > > Thanks! > > -- > Kristian Kielhofner > http://blog.krisk.org > http://www.submityoursip.com > http://www.astlinux.org > http://www.star2star.com > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From saigop at gmail.com Tue Feb 10 00:30:11 2009 From: saigop at gmail.com (Gopalakrishnan A.N) Date: Tue, 10 Feb 2009 14:00:11 +0530 Subject: [Freeswitch-users] javascript to get the status Message-ID: <2ea4d47e0902100030r7ca8fa94k32fb2004944ad939@mail.gmail.com> Hi, I am trying to execute the following script, its working fine for call origination, but cant able to get the status for dialed numbers, able to get only the last dialed number not for both the numbers. The script as follows, Javascript var array = [2]; array[0]="39841799874"; array[1]="39894929942"; for(var i=0;i References: <498FE8B3.8040904@ewetel.de> <191c3a030902090732v30f79c91qb06b3762e84b047@mail.gmail.com> Message-ID: <49914C75.7060104@ewetel.de> Hi Anthony, thanks, works perfectly :) regards helmut On 09.02.2009 16:32, Anthony Minessale wrote: > 1) set late-negotation=true in the sofia profile > 2) set absolute_codec_string channel variable to the exact codec you > want as the first action in your dialplan. From helmut.kuper at ewetel.de Tue Feb 10 02:12:14 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Tue, 10 Feb 2009 11:12:14 +0100 Subject: [Freeswitch-users] little problem with gateway registration Message-ID: <499152FE.7090909@ewetel.de> Hello, today I connected FS to my SIP-DDI SoftSwitch. It registered successfully, but not as expected at the register-proxy (sip2.ewetel.net), but at the proxy (proxy2.ewetel.net). My gateway config is this: The corresponding register request generated by FS is this: REGISTER sip:proxy2.ewetel.net;transport=udp SIP/2.0 Via: SIP/2.0/UDP 85.16.246.22;branch=z9hG4bK2FNpmSQHpZtye Via: SIP/2.0/UDP 85.16.246.6:5070;received=85.16.246.6;rport=5070;branch=z9hG4bK2FNpmSQHpZtye Max-Forwards: 69 From: ;tag=QrFtyDUrpg94N To: Call-ID: 8bd8f32c-f759-11dd-bb38-778d64a5265f CSeq: 111011668 REGISTER Contact: Expires: 60 User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-11698M Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: timer, precondition, path, replaces Authorization: Digest username="ippbx", realm="sip2.ewetel.net", nonce="4991527400000e269de527c4c699b9466bbf1e015cd0532c", cnonce="HE1PznH9Eiyb4QAUT+bjMA", algorithm=MD5, uri="sip:proxy2.ewetel.net;transport=udp", response="b5af17310919ec6b8f82acd4c89ffb9e", qop=auth, nc=00000001 Content-Length: 0 Shouldn't it register to register-proxy when it is given? regards Helmut From john at argv.co.uk Tue Feb 10 02:15:25 2009 From: john at argv.co.uk (John Daragon) Date: Tue, 10 Feb 2009 10:15:25 +0000 Subject: [Freeswitch-users] BT IPExchange Interoperability Testing In-Reply-To: References: <4990B34C.2070505@argv.co.uk> Message-ID: <499153BD.8010807@argv.co.uk> Brian West wrote: > Yes search the mailing list people have interoped with BT in record > time. On another note you hijacked the "DTMF not being recognized" by > clicking reply, deleting the text and changing the subject. Please > try not to do that in the future, click "new message" input freeswitch-users at lists.freeswitch.org > then type your subject and message then click send. Your email > client echo's back the headers that causes the mailing list server and > many email clients to thread the message properly. > Whoops, sorry! User IQ Error. jd -- John Daragon argv[0] limited Lambs Lawn Cottage, Staple Fitzpaine, Taunton, TA3 5SL, UK Registered in England Company Number 02947782 v +44 (0) 1460 234068 f +44 (0) 1460 234069 m +44 (0) 7836 576127 From blake.france at gmail.com Tue Feb 10 05:05:17 2009 From: blake.france at gmail.com (Blake France) Date: Tue, 10 Feb 2009 07:05:17 -0600 Subject: [Freeswitch-users] Recording play end of sound file again In-Reply-To: References: <4990702C.9090305@gmail.com> Message-ID: <49917B8D.3040409@gmail.com> Brian West wrote: > Can you tell me what SVN rev you're on? > /b > > On Feb 9, 2009, at 12:04 PM, Blake France wrote: > > >> Whenever I try to record and IVR or Voicemail Greeting, it will record >> and playback, but playback does something like this. >> >> "Please leave a message" ... "Message" >> >> It plays the end of the sound file AGAIN after playing the sound file. >> I've tried leaving extra time before and after speaking, but it still >> does this. Anyone ran into this issue? >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > Honestly I'm a noob and can't. I'm running the lastest release for PFSense. From anthony.minessale at gmail.com Tue Feb 10 06:13:17 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 10 Feb 2009 08:13:17 -0600 Subject: [Freeswitch-users] little problem with gateway registration In-Reply-To: <499152FE.7090909@ewetel.de> References: <499152FE.7090909@ewetel.de> Message-ID: <191c3a030902100613m1b9a8fc1w787b3e177a373557@mail.gmail.com> register-proxy is for where it actually sends the packet but it will still say the name of proxy in the packet. Did you check the destination address of the packet it should end up as the ip:port of that proxy. On Tue, Feb 10, 2009 at 4:12 AM, Helmut Kuper wrote: > Hello, > > today I connected FS to my SIP-DDI SoftSwitch. It registered > successfully, but not as expected at the register-proxy > (sip2.ewetel.net), but at the proxy (proxy2.ewetel.net). > > My gateway config is this: > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > The corresponding register request generated by FS is this: > > REGISTER sip:proxy2.ewetel.net;transport=udp SIP/2.0 > Via: SIP/2.0/UDP 85.16.246.22;branch=z9hG4bK2FNpmSQHpZtye > Via: SIP/2.0/UDP 85.16.246.6:5070 > ;received=85.16.246.6;rport=5070;branch=z9hG4bK2FNpmSQHpZtye > Max-Forwards: 69 > From: > ;transport=udp>;tag=QrFtyDUrpg94N > To: > ;transport=udp> > Call-ID: 8bd8f32c-f759-11dd-bb38-778d64a5265f > CSeq: 111011668 REGISTER > Contact: > Expires: 60 > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-11698M > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO > Supported: timer, precondition, path, replaces > Authorization: Digest username="ippbx", realm="sip2.ewetel.net", > nonce="4991527400000e269de527c4c699b9466bbf1e015cd0532c", > cnonce="HE1PznH9Eiyb4QAUT+bjMA", algorithm=MD5, uri="sip:proxy2.ewetel.net;transport=udp", > response="b5af17310919ec6b8f82acd4c89ffb9e", qop=auth, nc=00000001 > Content-Length: 0 > > > Shouldn't it register to register-proxy when it is given? > > regards > Helmut > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090210/a09f3d7d/attachment-0001.html From anthony.minessale at gmail.com Tue Feb 10 06:14:46 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 10 Feb 2009 08:14:46 -0600 Subject: [Freeswitch-users] Recording play end of sound file again In-Reply-To: <49917B8D.3040409@gmail.com> References: <4990702C.9090305@gmail.com> <49917B8D.3040409@gmail.com> Message-ID: <191c3a030902100614l51900772u2f88862159565aae@mail.gmail.com> your issue has already been fixed in FS. You will have to wait for pfsense to upgrade to get the fix. On Tue, Feb 10, 2009 at 7:05 AM, Blake France wrote: > Brian West wrote: > > Can you tell me what SVN rev you're on? > > /b > > > > On Feb 9, 2009, at 12:04 PM, Blake France wrote: > > > > > >> Whenever I try to record and IVR or Voicemail Greeting, it will record > >> and playback, but playback does something like this. > >> > >> "Please leave a message" ... "Message" > >> > >> It plays the end of the sound file AGAIN after playing the sound file. > >> I've tried leaving extra time before and after speaking, but it still > >> does this. Anyone ran into this issue? > >> > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > Honestly I'm a noob and can't. I'm running the lastest release for > PFSense. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090210/952b970d/attachment.html From blake.france at gmail.com Tue Feb 10 06:18:50 2009 From: blake.france at gmail.com (Blake France) Date: Tue, 10 Feb 2009 08:18:50 -0600 Subject: [Freeswitch-users] Recording play end of sound file again In-Reply-To: <191c3a030902100614l51900772u2f88862159565aae@mail.gmail.com> References: <4990702C.9090305@gmail.com> <49917B8D.3040409@gmail.com> <191c3a030902100614l51900772u2f88862159565aae@mail.gmail.com> Message-ID: <49918CCA.8020701@gmail.com> Anthony Minessale wrote: > your issue has already been fixed in FS. You will have to wait for > pfsense to upgrade to get the fix. > > > On Tue, Feb 10, 2009 at 7:05 AM, Blake France > wrote: > > Brian West wrote: > > Can you tell me what SVN rev you're on? > > /b > > > > On Feb 9, 2009, at 12:04 PM, Blake France wrote: > > > > > >> Whenever I try to record and IVR or Voicemail Greeting, it will > record > >> and playback, but playback does something like this. > >> > >> "Please leave a message" ... "Message" > >> > >> It plays the end of the sound file AGAIN after playing the > sound file. > >> I've tried leaving extra time before and after speaking, but it > still > >> does this. Anyone ran into this issue? > >> > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > Honestly I'm a noob and can't. I'm running the lastest release > for PFSense. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > Thank you. From odermann at googlemail.com Tue Feb 10 06:58:32 2009 From: odermann at googlemail.com (Dennis) Date: Tue, 10 Feb 2009 15:58:32 +0100 Subject: [Freeswitch-users] DTMF: Mute sound for the other side? In-Reply-To: <191c3a030902090725q11c8b3eaka6ee5e8565b1a7ea@mail.gmail.com> References: <5e414ed0902090659p18b970c9y8ba8f1bacda1f3ac@mail.gmail.com> <191c3a030902090725q11c8b3eaka6ee5e8565b1a7ea@mail.gmail.com> Message-ID: <5e414ed0902100658j2656c453mb74d4840bc256063@mail.gmail.com> we are not using inband tones. we are using rfc2833. is it still neccessary, to do some extra programming? if yes: isn't there a way for fs to recognize, that there is a rfc2833 and simply does not play it back for the others? 2009/2/9 Anthony Minessale : > 1) don't use inband tones for dtmf. > 2) post a bounty to have FS clip the audio for x milliseconds when a tone is > detected. (you will still hear faint clicks between the start of the tone > and when the clipping activates) > > > > On Mon, Feb 9, 2009 at 8:59 AM, Dennis wrote: >> >> hi, >> >> i am having a small problem with the dtmf-sounds... >> >> if i press a dtmf digit while i am bridged with another leg, the other >> side will hear the dtmf sound. >> this is very annoying and even worse in a conference, when multiple >> people can press dtmf digits (for (un-)muting themselves or using >> other functions). >> >> is there a way, to NOT let the other side hear the dtmf sound (but of >> course still make fs listening to it)? >> >> >> thanks for the help >> dennis >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From msc at freeswitch.org Tue Feb 10 06:59:42 2009 From: msc at freeswitch.org (Michael S Collins) Date: Tue, 10 Feb 2009 06:59:42 -0800 Subject: [Freeswitch-users] Thread hijacking and BT interop In-Reply-To: <9E97ECB0-825E-4622-85CE-D1961DE2A019@3c.co.uk> References: <4990B34C.2070505@argv.co.uk> <7e2ac3270902091604y5e765b04l4934aa9088516abe@mail.gmail.com> <691796D7-3778-411A-BCE1-6493EDA6D6FE@freeswitch.org> <9E97ECB0-825E-4622-85CE-D1961DE2A019@3c.co.uk> Message-ID: On Feb 9, 2009, at 11:04 PM, David Knell wrote: > Oops - I did it again ;-) > You Britney Spears wannabe!! :p -MC > --Dave > From kawarod at laposte.net Tue Feb 10 07:07:14 2009 From: kawarod at laposte.net (rod) Date: Tue, 10 Feb 2009 19:07:14 +0400 Subject: [Freeswitch-users] mod_fax and sending a fax Message-ID: <49919822.3030101@laposte.net> Hi all, I don't understand how to use the fax commands for sending a fax. In the wiki I saw this: my question is how to specify the gateway/profile that will handle the call. For a call I can use the bridge application like this, but for the txfax ?? regards, rod From odermann at googlemail.com Tue Feb 10 07:16:28 2009 From: odermann at googlemail.com (Dennis) Date: Tue, 10 Feb 2009 16:16:28 +0100 Subject: [Freeswitch-users] Socket-Event on "originate call"? In-Reply-To: <191c3a030902090727n44eef7d7x83704b66e628081@mail.gmail.com> References: <5e414ed0902090716y10f95f71o5760dfdf50d1c5cb@mail.gmail.com> <191c3a030902090727n44eef7d7x83704b66e628081@mail.gmail.com> Message-ID: <5e414ed0902100716s2c053a48w2174f7587dd0641e@mail.gmail.com> yes, you are right. we are receiving the reply. but, we are using socket outbound and manage all calls over this socket. we also measure the durations (like variable_duration and variable_billsec) and count all outgoing calls over the socket. but, if the originate (without an inbound call) will not start the socket, we can not count up, how many calls failed because of "user busy" or how long the platform was in use. a possible workarround: is it possible to trigger a dialplan over the cli (like our default dialplan, which starts the socket), so that the dialplan starts the originates? the basic problem for us, that, if we just want to make dialouts, we are missing the inbound call to start the socket. kind regards dennis 2009/2/9 Anthony Minessale : > when an originate is unsuccessful the failure and the cause code is returned > as the reply to the originate request. > > > On Mon, Feb 9, 2009 at 9:16 AM, Dennis wrote: >> >> hi, >> >> i am using socket outbound with fs. >> >> if i do an originate over the console, for starting an outbound call >> without having an inbound call, and send the originate directly to the >> socket, the socket is first started, if the call is in answer or >> ringing state. >> before this, i will not receive any event, because the socket was not >> started. therefore i will not know, if the target is "busy" (hangup, >> hangup cause: user busy). >> >> it would be very helpful, if the socket would start immediately on an >> event like "channel originate". >> >> >> thanks for the help >> dennis From kristian.kielhofner at gmail.com Tue Feb 10 07:34:32 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Tue, 10 Feb 2009 10:34:32 -0500 Subject: [Freeswitch-users] FreeSWITCH uclibc segfault In-Reply-To: References: <2d9149cd0902092343k4109d132v9d974c77e7ac8de0@mail.gmail.com> Message-ID: <2d9149cd0902100734w4a0483a2p57f9548d9eeec1fc@mail.gmail.com> Uh oh, I was afraid of that. I haven't had to work around uClibc issues in a while. Hopefully I still remember some of that stuff. ;) On Tue, Feb 10, 2009 at 2:47 AM, Ken Rice wrote: > There arent known issues cause I don't think anyone else has tried it hah > > -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From f.koliqi at gmail.com Tue Feb 10 07:21:58 2009 From: f.koliqi at gmail.com (Fadil Berisha) Date: Tue, 10 Feb 2009 10:21:58 -0500 Subject: [Freeswitch-users] FreeSWITCH uclibc segfault In-Reply-To: <2d9149cd0902092343k4109d132v9d974c77e7ac8de0@mail.gmail.com> References: <2d9149cd0902092343k4109d132v9d974c77e7ac8de0@mail.gmail.com> Message-ID: <5c7d82f20902100721y65b60f26x79af9c770918bad6@mail.gmail.com> In my uClibc system ( busybox + uClibc), FreeSWITCH compiles cleanly with: ./bootstrap.sh make make install but after starting, getting segfaults in mod_lua and and spidermonkey. Without those modules, FreeSWITCH running. Need more tests in order to confirm OK koliqi On Tue, Feb 10, 2009 at 2:43 AM, Kristian Kielhofner < kristian.kielhofner at gmail.com> wrote: > I don't think this is worth filing a bug for (yet)... > > FS rev 11655 segfaults with AstLinux (uClibc). > > Backtrace: > http://astbuild.star2star.com/astlinux-freeswitch-segfault.txt > > I'm sorry it doesn't have all the symbols... Everything except FS is > stripped. > > Configure options: > > http://astlinux.svn.sourceforge.net/viewvc/astlinux/branches/astlinux-freeswitch/package/freeswitch/freeswitch.mk > > FreeSWITCH compiles cleanly. > > Are there any known issues with uclibc (couldn't find anything on > Jira) or did I do something stupid? > > Thanks! > > -- > Kristian Kielhofner > http://blog.krisk.org > http://www.submityoursip.com > http://www.astlinux.org > http://www.star2star.com > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090210/f97b496f/attachment-0001.html From anthony.minessale at gmail.com Tue Feb 10 08:15:34 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 10 Feb 2009 10:15:34 -0600 Subject: [Freeswitch-users] FreeSWITCH uclibc segfault In-Reply-To: <2d9149cd0902100734w4a0483a2p57f9548d9eeec1fc@mail.gmail.com> References: <2d9149cd0902092343k4109d132v9d974c77e7ac8de0@mail.gmail.com> <2d9149cd0902100734w4a0483a2p57f9548d9eeec1fc@mail.gmail.com> Message-ID: <191c3a030902100815p33a6c3b7pd5f23c35a4644703@mail.gmail.com> hmm crashing in mutex lock, maybe the pthread lib is messed up. how did you trick it into compiling? Maybe some of the answers are wrong and apr is using the wrong thread abstraction? Some guy made this wiki page regarding cross compiling, did you see it ? http://wiki.freeswitch.org/wiki/Download_%26_Installation_Guide#Cross_Compiling_for_ARM_on_Linux I was able to build FS in scratchbox for arm11 before. I do remember one time when trying to get asterisk to work on wrt in the old days that the pthread lib was bad and I had to use a different version of uClibc runtime to get around it. On Tue, Feb 10, 2009 at 9:34 AM, Kristian Kielhofner < kristian.kielhofner at gmail.com> wrote: > Uh oh, I was afraid of that. > > I haven't had to work around uClibc issues in a while. Hopefully I > still remember some of that stuff. ;) > > On Tue, Feb 10, 2009 at 2:47 AM, Ken Rice wrote: > > There arent known issues cause I don't think anyone else has tried it hah > > > > > > -- > Kristian Kielhofner > http://blog.krisk.org > http://www.submityoursip.com > http://www.astlinux.org > http://www.star2star.com > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090210/f4bb36c4/attachment.html From mike at jerris.com Tue Feb 10 08:27:32 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 10 Feb 2009 11:27:32 -0500 Subject: [Freeswitch-users] DTMF: Mute sound for the other side? In-Reply-To: <5e414ed0902100658j2656c453mb74d4840bc256063@mail.gmail.com> References: <5e414ed0902090659p18b970c9y8ba8f1bacda1f3ac@mail.gmail.com> <191c3a030902090725q11c8b3eaka6ee5e8565b1a7ea@mail.gmail.com> <5e414ed0902100658j2656c453mb74d4840bc256063@mail.gmail.com> Message-ID: <174C0607-ADBE-4AB2-92C3-588A86B74EAE@jerris.com> If your in a conference and your hearing other people hitting dtmf digits that IS inband, it means that the place upstream that is doing inband to 2833 conversion is not properly clipping the dtmf, this probably needs to be fixed on that device. Mike On Feb 10, 2009, at 9:58 AM, Dennis wrote: > we are not using inband tones. we are using rfc2833. > > is it still neccessary, to do some extra programming? if yes: isn't > there a way for fs to recognize, that there is a rfc2833 and simply > does not play it back for the others? > > > 2009/2/9 Anthony Minessale : >> 1) don't use inband tones for dtmf. >> 2) post a bounty to have FS clip the audio for x milliseconds when >> a tone is >> detected. (you will still hear faint clicks between the start of >> the tone >> and when the clipping activates) >> >> >> >> On Mon, Feb 9, 2009 at 8:59 AM, Dennis >> wrote: >>> >>> hi, >>> >>> i am having a small problem with the dtmf-sounds... >>> >>> if i press a dtmf digit while i am bridged with another leg, the >>> other >>> side will hear the dtmf sound. >>> this is very annoying and even worse in a conference, when multiple >>> people can press dtmf digits (for (un-)muting themselves or using >>> other functions). >>> >>> is there a way, to NOT let the other side hear the dtmf sound (but >>> of >>> course still make fs listening to it)? >>> >>> >>> thanks for the help >>> dennis >>> From mike at jerris.com Tue Feb 10 08:34:13 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 10 Feb 2009 11:34:13 -0500 Subject: [Freeswitch-users] FreeSWITCH uclibc segfault In-Reply-To: <191c3a030902100815p33a6c3b7pd5f23c35a4644703@mail.gmail.com> References: <2d9149cd0902092343k4109d132v9d974c77e7ac8de0@mail.gmail.com> <2d9149cd0902100734w4a0483a2p57f9548d9eeec1fc@mail.gmail.com> <191c3a030902100815p33a6c3b7pd5f23c35a4644703@mail.gmail.com> Message-ID: <1AACA9A6-8228-4947-8B7D-75C599311EA0@jerris.com> From that link, these are the ones that are most likey causing the issue. I think the first one. Can you check your config.log and see what results it has for those checks? If you can compile native on the device I would suggest to on the device do a ./configure -C and then to use that config.cache file generated to get all your answers to feed your cross toolchain. You can probably just slim down that exact file and have everything you need. export apr_cv_mutex_recursive=yes; \ export ac_cv_func_pthread_rwlock_init=yes; \ export apr_cv_type_rwlock_t=yes; \ Mike On Feb 10, 2009, at 11:15 AM, Anthony Minessale wrote: > hmm crashing in mutex lock, maybe the pthread lib is messed up. > how did you trick it into compiling? Maybe some of the answers are > wrong and apr is using the wrong thread abstraction? > > Some guy made this wiki page regarding cross compiling, did you see > it ? > http://wiki.freeswitch.org/wiki/Download_%26_Installation_Guide#Cross_Compiling_for_ARM_on_Linux > > I was able to build FS in scratchbox for arm11 before. > > I do remember one time when trying to get asterisk to work on wrt in > the old days that the pthread lib > was bad and I had to use a different version of uClibc runtime to > get around it. > > > On Tue, Feb 10, 2009 at 9:34 AM, Kristian Kielhofner > wrote: > Uh oh, I was afraid of that. > > I haven't had to work around uClibc issues in a while. Hopefully I > still remember some of that stuff. ;) > > On Tue, Feb 10, 2009 at 2:47 AM, Ken Rice > wrote: > > There arent known issues cause I don't think anyone else has tried > it hah > > > > > > -- > Kristian Kielhofner > http://blog.krisk.org > http://www.submityoursip.com > http://www.astlinux.org > http://www.star2star.com > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090210/7a6639e4/attachment.html From kristian.kielhofner at gmail.com Tue Feb 10 09:25:05 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Tue, 10 Feb 2009 12:25:05 -0500 Subject: [Freeswitch-users] FreeSWITCH uclibc segfault In-Reply-To: <191c3a030902100815p33a6c3b7pd5f23c35a4644703@mail.gmail.com> References: <2d9149cd0902092343k4109d132v9d974c77e7ac8de0@mail.gmail.com> <2d9149cd0902100734w4a0483a2p57f9548d9eeec1fc@mail.gmail.com> <191c3a030902100815p33a6c3b7pd5f23c35a4644703@mail.gmail.com> Message-ID: <2d9149cd0902100925n1ea6c99bn6b0ada20edab0be9@mail.gmail.com> Tony, Thanks for looking at this. That just goes to show you how useless gdb is to me. Now that you say mutex lock I can think of some configure variables to try... :) I had read that section of the wiki but there is no mention he is using uclibc. Chances are he probably is but there's no way to be sure. I'll make sure to update the wiki with whatever I find. I only have a cross compiler. Some tests cannot be run at all and other guesses are incorrect. That's the fun part. I think pthread is pretty decent in this version of uClibc but I could be wrong. On Tue, Feb 10, 2009 at 11:15 AM, Anthony Minessale wrote: > hmm crashing in mutex lock, maybe the pthread lib is messed up. > how did you trick it into compiling? Maybe some of the answers are wrong > and apr is using the wrong thread abstraction? > > Some guy made this wiki page regarding cross compiling, did you see it ? > http://wiki.freeswitch.org/wiki/Download_%26_Installation_Guide#Cross_Compiling_for_ARM_on_Linux > > I was able to build FS in scratchbox for arm11 before. > > I do remember one time when trying to get asterisk to work on wrt in the old > days that the pthread lib > was bad and I had to use a different version of uClibc runtime to get around > it. > -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From kristian.kielhofner at gmail.com Tue Feb 10 09:26:09 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Tue, 10 Feb 2009 12:26:09 -0500 Subject: [Freeswitch-users] FreeSWITCH uclibc segfault In-Reply-To: <5c7d82f20902100721y65b60f26x79af9c770918bad6@mail.gmail.com> References: <2d9149cd0902092343k4109d132v9d974c77e7ac8de0@mail.gmail.com> <5c7d82f20902100721y65b60f26x79af9c770918bad6@mail.gmail.com> Message-ID: <2d9149cd0902100926h33858165vfde06a2d6e175686@mail.gmail.com> Thanks, it's good to know it's possible. Lua and spidermonkey wouldn't even compile for me; I'm going to look into that once FS starts. On Tue, Feb 10, 2009 at 10:21 AM, Fadil Berisha wrote: > In my uClibc system ( busybox + uClibc), FreeSWITCH compiles cleanly with: > > ./bootstrap.sh > make > make install > > but after starting, getting segfaults in mod_lua and and spidermonkey. > Without those modules, FreeSWITCH running. Need more tests in order to > confirm OK > > koliqi -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From kristian.kielhofner at gmail.com Tue Feb 10 09:28:11 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Tue, 10 Feb 2009 12:28:11 -0500 Subject: [Freeswitch-users] FreeSWITCH uclibc segfault In-Reply-To: <1AACA9A6-8228-4947-8B7D-75C599311EA0@jerris.com> References: <2d9149cd0902092343k4109d132v9d974c77e7ac8de0@mail.gmail.com> <2d9149cd0902100734w4a0483a2p57f9548d9eeec1fc@mail.gmail.com> <191c3a030902100815p33a6c3b7pd5f23c35a4644703@mail.gmail.com> <1AACA9A6-8228-4947-8B7D-75C599311EA0@jerris.com> Message-ID: <2d9149cd0902100928ocfbc858x48a3f77527c1b87@mail.gmail.com> Yep, now that Tony boiled down that gdb output to a mutex lock I agree. I'm trying a compile now with these values (cross compiling, no native compiler). Cross compiling is so much fun! On Tue, Feb 10, 2009 at 11:34 AM, Michael Jerris wrote: > From that link, these are the ones that are most likey causing the issue. I > think the first one. Can you check your config.log and see what results it > has for those checks? If you can compile native on the device I would > suggest to on the device do a ./configure -C and then to use that > config.cache file generated to get all your answers to feed your cross > toolchain. You can probably just slim down that exact file and have > everything you need. > export apr_cv_mutex_recursive=yes; \ > export ac_cv_func_pthread_rwlock_init=yes; \ > export apr_cv_type_rwlock_t=yes; \ > > Mike -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From kristian.kielhofner at gmail.com Tue Feb 10 10:02:22 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Tue, 10 Feb 2009 13:02:22 -0500 Subject: [Freeswitch-users] FreeSWITCH uclibc segfault In-Reply-To: <2d9149cd0902100928ocfbc858x48a3f77527c1b87@mail.gmail.com> References: <2d9149cd0902092343k4109d132v9d974c77e7ac8de0@mail.gmail.com> <2d9149cd0902100734w4a0483a2p57f9548d9eeec1fc@mail.gmail.com> <191c3a030902100815p33a6c3b7pd5f23c35a4644703@mail.gmail.com> <1AACA9A6-8228-4947-8B7D-75C599311EA0@jerris.com> <2d9149cd0902100928ocfbc858x48a3f77527c1b87@mail.gmail.com> Message-ID: <2d9149cd0902101002i4d992397uca1f6c471f66c80@mail.gmail.com> Replying to myself: That was it! Time to do some testing... Thanks!!! On Tue, Feb 10, 2009 at 12:28 PM, Kristian Kielhofner wrote: > Yep, now that Tony boiled down that gdb output to a mutex lock I agree. > > I'm trying a compile now with these values (cross compiling, no native > compiler). > > Cross compiling is so much fun! > > On Tue, Feb 10, 2009 at 11:34 AM, Michael Jerris wrote: >> From that link, these are the ones that are most likey causing the issue. I >> think the first one. Can you check your config.log and see what results it >> has for those checks? If you can compile native on the device I would >> suggest to on the device do a ./configure -C and then to use that >> config.cache file generated to get all your answers to feed your cross >> toolchain. You can probably just slim down that exact file and have >> everything you need. >> export apr_cv_mutex_recursive=yes; \ >> export ac_cv_func_pthread_rwlock_init=yes; \ >> export apr_cv_type_rwlock_t=yes; \ >> >> Mike > > -- > Kristian Kielhofner > http://blog.krisk.org > http://www.submityoursip.com > http://www.astlinux.org > http://www.star2star.com > -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From ajlong at worldlink.net Tue Feb 10 11:01:38 2009 From: ajlong at worldlink.net (Adam Long) Date: Tue, 10 Feb 2009 14:01:38 -0500 Subject: [Freeswitch-users] Dynamic Dialplan In-Reply-To: <49906FE6.8000008@post.cz> References: <743112140902081921n2bb1bf49mfe36e5e2193c6b04@mail.gmail.com> <4990495C.9010008@post.cz> <49906889.9030801@freeswitch.org> <49906FE6.8000008@post.cz> Message-ID: <000301c98bb2$00d66560$02833020$@net> What about a mod_perl XML binding like the example here? http://wiki.freeswitch.org/wiki/Mod_perl and http://wiki.freeswitch.org/wiki/Mod_perl_and_Generating_XML Would this be faster than a setup like mod_curl_xml -> lighttpd -> FastCGI/Perl ? I guess it would depend on if mod_perl in FreeSWITCH spawns new interpreter per request or if it uses one interpreter instance with multiple threads to execute pre-loaded perl. Anyone know if this is the case? Regards, -Adam -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of kokoska rokoska Sent: Monday, February 09, 2009 1:03 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Dynamic Dialplan Raymond Chandler napsal(a): > kokoska rokoska wrote: >> Just my 2c: >> >> If you use curl with lighttpd and custom built fastcgi "C" responder (it >> is really simple with fcgi libs - even I can do it :-) performance could >> be not that bad. > hmmm, mod_xml_curl using C, interesting thought.. May be not the best way, but very simple. Well, it depends on what you have to do, but "directory" serving based on DB queries (this what I'm using it for) is very simple - just few lines of code. > all of the > complexities of writing your own module without the nice structured FS > API... I should say I have no idea how hard is to write custom FreeSWITCH module (may be I should try it :-), but the FS code is very nice! > although, as a benefit, i guess you do get a little extra latency ;-) > :-) Yes, you are right. And as a bonus some CPU utilization... Like I wrote above, I didn't say it is faster, but IMO it is very simple and not as slow as it looks (when using apache + php + apc). Best regards, kokoska.rokoska _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From pbd at suspiria.net Tue Feb 10 10:41:37 2009 From: pbd at suspiria.net (Public Dump) Date: Tue, 10 Feb 2009 19:41:37 +0100 Subject: [Freeswitch-users] High CPU load after starting Message-ID: <13C421883438EB42B9E2C30069FD4AB766F9B1F840@crushinator.central.local> After starting FreeSwitch (1.0.2) on a 4 core server running Windows Server 2008, the CPU load (privileged time/kernel) for one of the cores goes to 50% and stays there. Stoping FreeSwitch stops the load. I have tried to disable all modules but the problem persists. Has anybody seen this problem, can it be fixed ? regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090210/f57d864c/attachment.html From brian at freeswitch.org Tue Feb 10 12:44:03 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 10 Feb 2009 14:44:03 -0600 Subject: [Freeswitch-users] High CPU load after starting In-Reply-To: <13C421883438EB42B9E2C30069FD4AB766F9B1F840@crushinator.central.local> References: <13C421883438EB42B9E2C30069FD4AB766F9B1F840@crushinator.central.local> Message-ID: <7A22A338-495E-4D95-8D0F-FD196838B0CD@freeswitch.org> Please update to SVN trunk or the latest Windows Build and this problem should go away. /b On Feb 10, 2009, at 12:41 PM, Public Dump wrote: > After starting FreeSwitch (1.0.2) on a 4 core server running > Windows Server 2008, the CPU load (privileged time/kernel) for one > of the cores goes to 50% and stays there. > Stoping FreeSwitch stops the load. I have tried to disable all > modules but the problem persists. > > Has anybody seen this problem, can it be fixed ? > > regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090210/2b4107ed/attachment.html From gmaruzz at celliax.org Tue Feb 10 13:03:52 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Tue, 10 Feb 2009 22:03:52 +0100 Subject: [Freeswitch-users] High CPU load after starting In-Reply-To: <7A22A338-495E-4D95-8D0F-FD196838B0CD@freeswitch.org> References: <13C421883438EB42B9E2C30069FD4AB766F9B1F840@crushinator.central.local> <7A22A338-495E-4D95-8D0F-FD196838B0CD@freeswitch.org> Message-ID: <7b197bef0902101303r664fc8f0h280ce6f31517e9a1@mail.gmail.com> I use often the Windows svn on Vista, no problem seen Sincerely, Giovanni Maruzzelli ========================================= www.celliax.org via Pierlombardo 9, 20135 Milano Italy gmaruzz at celliax dot org Cell : +39-347-2665618 Fax : +39-02-87390039 On Tue, Feb 10, 2009 at 9:44 PM, Brian West wrote: > Please update to SVN trunk or the latest Windows Build and this problem > should go away. > /b > > On Feb 10, 2009, at 12:41 PM, Public Dump wrote: > > After starting FreeSwitch (1.0.2) on a 4 core server running Windows > Server 2008, the CPU load (privileged time/kernel) for one of the cores goes > to 50% and stays there. > Stoping FreeSwitch stops the load. I have tried to disable all modules but > the problem persists. > > Has anybody seen this problem, can it be fixed ? > > regards > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090210/81ca2b0c/attachment.html From nik.middleton at noblesolutions.co.uk Tue Feb 10 13:04:21 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Tue, 10 Feb 2009 21:04:21 -0000 Subject: [Freeswitch-users] Strange error message Message-ID: Hi Guys, I'm baffled by this error. I'm updating the db on call hang-up If I comment out curs:close() no error, but I'm concerned about memory leaks. Can anyone tell me what FS is complaining about? The db gets updated in both cases Regards require "luasql.mysql" function myHangupHook(s, status, arg) freeswitch.consoleLog("info", " : They hung up on US!!!\n"); env = assert (luasql.mysql()) con = assert (env:connect("xxxxl","xxxxxxxxx","pass","192.168.3.205")) curs = assert (con:execute"UPDATE callers SET lastcall = 'BOB' WHERE id = 33292") curs:close() con:close() env:close() freeswitch.consoleLog("NOTICE", "myHangupHook: " .. status .. "\n"); --error() end 2009-02-10 20:53:20 [INFO] switch_cpp.cpp:1086 console_log() : They hung up on US!!! 2009-02-10 20:53:20 [ERR] mod_lua.cpp:176 lua_parse_and_execute() /usr/local/freeswitch/scripts/helloworld.lua:50: attempt to index global 'curs' (a number value) stack traceback: /usr/local/freeswitch/scripts/helloworld.lua:50: in function [C]: in function 'hangup' /usr/local/freeswitch/scripts/helloworld.lua:70: in main chunk 2009-02-10 20:53:20 [NOTICE] switch_core_session.c:957 switch_core_session_thread() Session 63 (sofia/internal/1001 at 192.168.3.206) Ended -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090210/82e04d3a/attachment-0001.html From jesse.peterson at exbiblio.com Tue Feb 10 12:49:37 2009 From: jesse.peterson at exbiblio.com (Jesse Peterson) Date: Tue, 10 Feb 2009 12:49:37 -0800 Subject: [Freeswitch-users] SIP registration/retry/authorization problem Message-ID: <86FB611F-47CE-4472-8CE7-E52F6F1AADF5@exbiblio.com> Hello, I seem to be experiencing the exact same issue as is documented here: http://lists.freeswitch.org/pipermail/freeswitch-users/2008-September/006272.html Like above a "sofia profile external restart" immediately resumes operation. Does anyone have an idea what this may be? Is there a debugging step I can take? If it were a predictable outage I could monitor the registration attempts and find out why there are [401][Unauthorized] errors suddenly (again after a 'sofia restart' all is well). This happens multiple times a day for us. Thanks, - Jesse From msc at freeswitch.org Tue Feb 10 13:18:02 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 10 Feb 2009 13:18:02 -0800 Subject: [Freeswitch-users] SIP registration/retry/authorization problem In-Reply-To: <86FB611F-47CE-4472-8CE7-E52F6F1AADF5@exbiblio.com> References: <86FB611F-47CE-4472-8CE7-E52F6F1AADF5@exbiblio.com> Message-ID: <87f2f3b90902101318x37bf045of2ec3bec5890d279@mail.gmail.com> On Tue, Feb 10, 2009 at 12:49 PM, Jesse Peterson wrote: > Hello, > > I seem to be experiencing the exact same issue as is documented here: > http://lists.freeswitch.org/pipermail/freeswitch-users/2008-September/006272.html > > Like above a "sofia profile external restart" immediately resumes > operation. Does anyone have an idea what this may be? > > Is there a debugging step I can take? If it were a predictable outage > I could monitor the registration attempts and find out why there are > [401][Unauthorized] errors suddenly (again after a 'sofia restart' > all is well). > > This happens multiple times a day for us. Which revision of FS are you running? If you can update to the latest trunk and reproduce the symptoms that would be helpful. -MC From c_cav_01 at yahoo.com Tue Feb 10 13:19:28 2009 From: c_cav_01 at yahoo.com (Chris) Date: Tue, 10 Feb 2009 13:19:28 -0800 (PST) Subject: [Freeswitch-users] Strange error message In-Reply-To: Message-ID: <77751.47799.qm@web55105.mail.re4.yahoo.com> Closing the connection will force the server to close any open transactions, as well as release recordsets in local memory that reference the connection. ? However curs is not a recordset.? An SQL update is going to return an integer (rows affected) or boolean depending on the which server you use since no recordset is actually requested. --- On Tue, 2/10/09, Nik Middleton wrote: From: Nik Middleton Subject: [Freeswitch-users] Strange error message To: freeswitch-users at lists.freeswitch.org Date: Tuesday, February 10, 2009, 2:04 PM Hi Guys, ? I?m baffled by this error.? I?m updating the db on call hang-up If I comment out curs:close() no error, but I?m concerned about memory leaks.? Can anyone tell me what FS is complaining about? ? The db gets updated in both cases ? Regards ? ? ? require "luasql.mysql" ? function myHangupHook(s, status, arg) ??????????? freeswitch.consoleLog("info", " : They hung up on US!!!\n"); ??? ??????? env = assert (luasql.mysql()) ??????????? con = assert (env:connect("xxxxl","xxxxxxxxx","pass","192.168.3.205")) ??????????? curs = assert (con:execute"UPDATE callers SET lastcall = 'BOB' WHERE id = 33292") ??????????? curs:close() ??????????? con:close() ??????????? env:close() ??????????? freeswitch.consoleLog("NOTICE", "myHangupHook: " .. status .. "\n"); ??? --error() end ? ? ? ? 2009-02-10 20:53:20 [INFO] switch_cpp.cpp:1086 console_log()? : They hung up on US!!! 2009-02-10 20:53:20 [ERR] mod_lua.cpp:176 lua_parse_and_execute() /usr/local/freeswitch/scripts/helloworld.lua:50: attempt to index global 'curs' (a number value) stack traceback: ??????? /usr/local/freeswitch/scripts/helloworld.lua:50: in function ??????? [C]: in function 'hangup' ??????? /usr/local/freeswitch/scripts/helloworld.lua:70: in main chunk 2009-02-10 20:53:20 [NOTICE] switch_core_session.c:957 switch_core_session_thread() Session 63 (sofia/internal/1001 at 192.168.3.206) Ended_______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090210/58f2bc72/attachment.html From brian at freeswitch.org Tue Feb 10 13:25:43 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 10 Feb 2009 15:25:43 -0600 Subject: [Freeswitch-users] High CPU load after starting In-Reply-To: <7b197bef0902101303r664fc8f0h280ce6f31517e9a1@mail.gmail.com> References: <13C421883438EB42B9E2C30069FD4AB766F9B1F840@crushinator.central.local> <7A22A338-495E-4D95-8D0F-FD196838B0CD@freeswitch.org> <7b197bef0902101303r664fc8f0h280ce6f31517e9a1@mail.gmail.com> Message-ID: <5E26356E-2E69-4828-84B3-A537436FE2AB@freeswitch.org> Yes but this was a problem that was confirmed and fixed long ago... if you recall we have 1.0.3RC1 out and svn trunk. /b On Feb 10, 2009, at 3:03 PM, Giovanni Maruzzelli wrote: > I use often the Windows svn on Vista, no problem seen > > Sincerely, From brian at freeswitch.org Tue Feb 10 13:27:35 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 10 Feb 2009 15:27:35 -0600 Subject: [Freeswitch-users] SIP registration/retry/authorization problem In-Reply-To: <86FB611F-47CE-4472-8CE7-E52F6F1AADF5@exbiblio.com> References: <86FB611F-47CE-4472-8CE7-E52F6F1AADF5@exbiblio.com> Message-ID: <286A20C5-4088-4CD9-8BAA-D294777AF931@freeswitch.org> Try this: In sofia.conf.xml /b On Feb 10, 2009, at 2:49 PM, Jesse Peterson wrote: > Hello, > > I seem to be experiencing the exact same issue as is documented here: > http://lists.freeswitch.org/pipermail/freeswitch-users/2008-September/006272.html > > Like above a "sofia profile external restart" immediately resumes > operation. Does anyone have an idea what this may be? > > Is there a debugging step I can take? If it were a predictable outage > I could monitor the registration attempts and find out why there are > [401][Unauthorized] errors suddenly (again after a 'sofia restart' > all is well). > > This happens multiple times a day for us. > > Thanks, > - Jesse -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090210/794cd661/attachment.html From kokoska.rokoska at post.cz Tue Feb 10 13:34:30 2009 From: kokoska.rokoska at post.cz (kokoska rokoska) Date: Tue, 10 Feb 2009 22:34:30 +0100 Subject: [Freeswitch-users] Dynamic Dialplan In-Reply-To: <191c3a030902091145i1fffdbdcoe49e16366dc84acf@mail.gmail.com> References: <743112140902081921n2bb1bf49mfe36e5e2193c6b04@mail.gmail.com> <4990495C.9010008@post.cz> <49906889.9030801@freeswitch.org> <49906FE6.8000008@post.cz> <191c3a030902091145i1fffdbdcoe49e16366dc84acf@mail.gmail.com> Message-ID: <4991F2E6.1020102@post.cz> Anthony Minessale napsal(a): > That's why I chose mod_xml_curl as a demo for the xml_hook api. It's > not only a demo, it's rather functional =D > :-)) > You have 2 choices other than using the stuff we already have in tree. > > 1) write a custom dialplan module, this module gets a single callback > function a dialplan_hunt function that has the session and the caller > profile. you can see from mod_enum or mod_dialplan_xml how this can be > used to make your own module that looks in a db and returns instructions > to FS on the fly. > I try to look at it. > 2) write a custom xml_hook and use it with mod_dialplan_xml, this type > of module embeds itself into the xml lookups so when something tries to > find something in the xml registry, your function is called and you can > do your db lookups and generate the xml returned as binary xml obj built > from a result of the query. This is more powerfule because it allows > you to pre-empt any xml lookups so you can deliver directory, config, > dialplan, phrase macros, etc > Super tip, Anthony! Thank you very much. In case of directory it can dramatically increase maximu of registers per second. > > mod_xml_curl is an example of #2, it turns the xml_req into a url req > and feeds the xml returned over the http socket into an xml object and > returns it as the result in place of the static contents of the xml file. > Well I will look deep into mod_xml_curl to see how it is done and than (may be :-) will introduce new module mod_db_directory :-) Thanks once more, Anthony, for very valuable info! Best regards, kokoska.rokoska From anthony.minessale at gmail.com Tue Feb 10 14:32:40 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 10 Feb 2009 16:32:40 -0600 Subject: [Freeswitch-users] Dynamic Dialplan In-Reply-To: <000301c98bb2$00d66560$02833020$@net> References: <743112140902081921n2bb1bf49mfe36e5e2193c6b04@mail.gmail.com> <4990495C.9010008@post.cz> <49906889.9030801@freeswitch.org> <49906FE6.8000008@post.cz> <000301c98bb2$00d66560$02833020$@net> Message-ID: <191c3a030902101432y5954389dje53f6dc05ce654cc@mail.gmail.com> oh yeah, i forgot about those too. python,perl and lua can all do that. On Tue, Feb 10, 2009 at 1:01 PM, Adam Long wrote: > What about a mod_perl XML binding like the example here? > http://wiki.freeswitch.org/wiki/Mod_perl and > http://wiki.freeswitch.org/wiki/Mod_perl_and_Generating_XML > > Would this be faster than a setup like > > mod_curl_xml -> lighttpd -> FastCGI/Perl ? > > I guess it would depend on if mod_perl in FreeSWITCH spawns new interpreter > per request > or if it uses one interpreter instance with multiple threads to execute > pre-loaded perl. > > Anyone know if this is the case? > > Regards, > -Adam > > > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of kokoska > rokoska > Sent: Monday, February 09, 2009 1:03 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Dynamic Dialplan > > > > > Raymond Chandler napsal(a): > > kokoska rokoska wrote: > >> Just my 2c: > >> > >> If you use curl with lighttpd and custom built fastcgi "C" responder (it > >> is really simple with fcgi libs - even I can do it :-) performance could > >> be not that bad. > > hmmm, mod_xml_curl using C, interesting thought.. > > May be not the best way, but very simple. > Well, it depends on what you have to do, but "directory" serving based > on DB queries (this what I'm using it for) is very simple - just few > lines of code. > > > all of the > > complexities of writing your own module without the nice structured FS > > API... > > I should say I have no idea how hard is to write custom FreeSWITCH > module (may be I should try it :-), but the FS code is very nice! > > > although, as a benefit, i guess you do get a little extra latency ;-) > > > > :-) Yes, you are right. And as a bonus some CPU utilization... > > Like I wrote above, I didn't say it is faster, but IMO it is very simple > and not as slow as it looks (when using apache + php + apc). > > Best regards, > > kokoska.rokoska > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090210/cac42683/attachment-0001.html From nik.middleton at noblesolutions.co.uk Tue Feb 10 15:22:00 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Tue, 10 Feb 2009 23:22:00 -0000 Subject: [Freeswitch-users] Strange error message In-Reply-To: <77751.47799.qm@web55105.mail.re4.yahoo.com> References: <77751.47799.qm@web55105.mail.re4.yahoo.com> Message-ID: So what you're saying is that I can comment out curs:close() as it's not needed? Regards, ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Chris Sent: 10 February 2009 21:19 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Strange error message Closing the connection will force the server to close any open transactions, as well as release recordsets in local memory that reference the connection. However curs is not a recordset. An SQL update is going to return an integer (rows affected) or boolean depending on the which server you use since no recordset is actually requested. --- On Tue, 2/10/09, Nik Middleton wrote: From: Nik Middleton Subject: [Freeswitch-users] Strange error message To: freeswitch-users at lists.freeswitch.org Date: Tuesday, February 10, 2009, 2:04 PM Hi Guys, I'm baffled by this error. I'm updating the db on call hang-up If I comment out curs:close() no error, but I'm concerned about memory leaks. Can anyone tell me what FS is complaining about? The db gets updated in both cases Regards require "luasql.mysql" function myHangupHook(s, status, arg) freeswitch.consoleLog("info", " : They hung up on US!!!\n"); env = assert (luasql.mysql()) con = assert (env:connect("xxxxl","xxxxxxxxx","pass","192.168.3.205")) curs = assert (con:execute"UPDATE callers SET lastcall = 'BOB' WHERE id = 33292") curs:close() con:close() env:close() freeswitch.consoleLog("NOTICE", "myHangupHook: " .. status .. "\n"); --error() end 2009-02-10 20:53:20 [INFO] switch_cpp.cpp:1086 console_log() : They hung up on US!!! 2009-02-10 20:53:20 [ERR] mod_lua.cpp:176 lua_parse_and_execute() /usr/local/freeswitch/scripts/helloworld.lua:50: attempt to index global 'curs' (a number value) stack traceback: /usr/local/freeswitch/scripts/helloworld.lua:50: in function [C]: in function 'hangup' /usr/local/freeswitch/scripts/helloworld.lua:70: in main chunk 2009-02-10 20:53:20 [NOTICE] switch_core_session.c:957 switch_core_session_thread() Session 63 (sofia/internal/1001 at 192.168.3.206) Ended _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090210/cb925ccd/attachment.html From jonathan at myogre.com Tue Feb 10 13:22:58 2009 From: jonathan at myogre.com (Jonathan Creasy) Date: Tue, 10 Feb 2009 15:22:58 -0600 Subject: [Freeswitch-users] dialplan question Message-ID: <18cd54110902101322g76cc3aedj6dd38a52ecdada81@mail.gmail.com> I'm trying to setup a dialstring to send a call to a user as a DID number not their user. I want to be able to do one of two things: One, I want the Request URI to match the contact from their REGISTER and To header to be "+1NXXNXXXXXX at contactdomain" Two, I want to make both the Request URI and the To header be "+1NXXNXXXXXX at contactdomain". I looked at "force-user" but that gives me the ability to do one DID to a user (by forcing the user to the DID). So, how do I need to setup the call for these scenarios? From jonathan at myogre.com Tue Feb 10 17:00:23 2009 From: jonathan at myogre.com (Jonathan Creasy) Date: Tue, 10 Feb 2009 19:00:23 -0600 Subject: [Freeswitch-users] dialplan question Message-ID: <18cd54110902101700r3f039a2ex19c1b0818c1a3af4@mail.gmail.com> I'm trying to setup an extension to send a call to a user as a DID number not their user. I want to be able to do one of two things: One, I want the Request URI to match the contact from their REGISTER and To header to be "+1NXXNXXXXXX at contactdomain" Two, I want to make both the Request URI and the To header be "+1NXXNXXXXXX at contactdomain". I looked at "force-user" but that gives me the ability to do one DID to a user (by forcing the user to the DID). So, how do I need to setup the call for these scenarios? From jesse.peterson at exbiblio.com Tue Feb 10 17:39:18 2009 From: jesse.peterson at exbiblio.com (Jesse Peterson) Date: Tue, 10 Feb 2009 17:39:18 -0800 Subject: [Freeswitch-users] SIP registration/retry/authorization problem In-Reply-To: <87f2f3b90902101318x37bf045of2ec3bec5890d279@mail.gmail.com> References: <86FB611F-47CE-4472-8CE7-E52F6F1AADF5@exbiblio.com> <87f2f3b90902101318x37bf045of2ec3bec5890d279@mail.gmail.com> Message-ID: On Feb 10, 2009, at 1:18 PM, Michael Collins wrote: > Which revision of FS are you running? If you can update to the latest > trunk and reproduce the symptoms that would be helpful. > -MC This is FreeSwitch 1.0 running from a twixswitch 0.4 installation. The cited user was using FreeSwitch 1.0.1 with his symptoms. I unfortunately do not currently have the ability to try a different version. I have seen a similar issue using r10558M in the past. Thanks, - Jesse From jesse.peterson at exbiblio.com Tue Feb 10 17:43:33 2009 From: jesse.peterson at exbiblio.com (Jesse Peterson) Date: Tue, 10 Feb 2009 17:43:33 -0800 Subject: [Freeswitch-users] SIP registration/retry/authorization problem In-Reply-To: <286A20C5-4088-4CD9-8BAA-D294777AF931@freeswitch.org> References: <86FB611F-47CE-4472-8CE7-E52F6F1AADF5@exbiblio.com> <286A20C5-4088-4CD9-8BAA-D294777AF931@freeswitch.org> Message-ID: <24DD6D23-C3EB-48A4-AC4F-587F7DD16361@exbiblio.com> On Feb 10, 2009, at 1:27 PM, Brian West wrote: > Try this: > > > > In sofia.conf.xml > > /b I'm not able to find any documentation on this setting. I think it may be newer than my version of FreeSwitch (1.0). What does it do? Thanks, - Jesse From brian at freeswitch.org Tue Feb 10 17:48:13 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 10 Feb 2009 19:48:13 -0600 Subject: [Freeswitch-users] SIP registration/retry/authorization problem In-Reply-To: <24DD6D23-C3EB-48A4-AC4F-587F7DD16361@exbiblio.com> References: <86FB611F-47CE-4472-8CE7-E52F6F1AADF5@exbiblio.com> <286A20C5-4088-4CD9-8BAA-D294777AF931@freeswitch.org> <24DD6D23-C3EB-48A4-AC4F-587F7DD16361@exbiblio.com> Message-ID: <980C8919-ECFF-47F4-824E-842EBB0293BF@freeswitch.org> I highly recommend you wipe the box/install and install from Scratch using SVN trunk /b On Feb 10, 2009, at 7:43 PM, Jesse Peterson wrote: > I'm not able to find any documentation on this setting. I think it may > be newer than my version of FreeSwitch (1.0). What does it do? > > Thanks, > - Jesse From ajlong at worldlink.net Tue Feb 10 18:22:31 2009 From: ajlong at worldlink.net (Adam Long) Date: Tue, 10 Feb 2009 21:22:31 -0500 Subject: [Freeswitch-users] Dynamic Dialplan In-Reply-To: <191c3a030902101432y5954389dje53f6dc05ce654cc@mail.gmail.com> References: <743112140902081921n2bb1bf49mfe36e5e2193c6b04@mail.gmail.com> <4990495C.9010008@post.cz> <49906889.9030801@freeswitch.org> <49906FE6.8000008@post.cz> <000301c98bb2$00d66560$02833020$@net> <191c3a030902101432y5954389dje53f6dc05ce654cc@mail.gmail.com> Message-ID: <006401c98bef$980e04f0$c82a0ed0$@net> There sure are lots of options J Does the perl implementation use a single persistent embedded interpreter or does it spawn a new interpreter per request? Things like persistent database handles come to mind. Performance would be drastically impacted if it were to spawn new interpreter per request. Any ideas? Regards, -Adam From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Tuesday, February 10, 2009 5:33 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Dynamic Dialplan oh yeah, i forgot about those too. python,perl and lua can all do that. On Tue, Feb 10, 2009 at 1:01 PM, Adam Long wrote: What about a mod_perl XML binding like the example here? http://wiki.freeswitch.org/wiki/Mod_perl and http://wiki.freeswitch.org/wiki/Mod_perl_and_Generating_XML Would this be faster than a setup like mod_curl_xml -> lighttpd -> FastCGI/Perl ? I guess it would depend on if mod_perl in FreeSWITCH spawns new interpreter per request or if it uses one interpreter instance with multiple threads to execute pre-loaded perl. Anyone know if this is the case? Regards, -Adam -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of kokoska rokoska Sent: Monday, February 09, 2009 1:03 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Dynamic Dialplan Raymond Chandler napsal(a): > kokoska rokoska wrote: >> Just my 2c: >> >> If you use curl with lighttpd and custom built fastcgi "C" responder (it >> is really simple with fcgi libs - even I can do it :-) performance could >> be not that bad. > hmmm, mod_xml_curl using C, interesting thought.. May be not the best way, but very simple. Well, it depends on what you have to do, but "directory" serving based on DB queries (this what I'm using it for) is very simple - just few lines of code. > all of the > complexities of writing your own module without the nice structured FS > API... I should say I have no idea how hard is to write custom FreeSWITCH module (may be I should try it :-), but the FS code is very nice! > although, as a benefit, i guess you do get a little extra latency ;-) > :-) Yes, you are right. And as a bonus some CPU utilization... Like I wrote above, I didn't say it is faster, but IMO it is very simple and not as slow as it looks (when using apache + php + apc). Best regards, kokoska.rokoska _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090210/024df86e/attachment.html From chavpaskov at shaw.ca Tue Feb 10 18:39:17 2009 From: chavpaskov at shaw.ca (Chav Paskov) Date: Tue, 10 Feb 2009 18:39:17 -0800 Subject: [Freeswitch-users] Dynamic Dialplan In-Reply-To: <000301c98bb2$00d66560$02833020$@net> References: <743112140902081921n2bb1bf49mfe36e5e2193c6b04@mail.gmail.com> <4990495C.9010008@post.cz> <49906889.9030801@freeswitch.org> <49906FE6.8000008@post.cz> <000301c98bb2$00d66560$02833020$@net> Message-ID: <49923A55.1070009@shaw.ca> Adam Long wrote: > What about a mod_perl XML binding like the example here? > http://wiki.freeswitch.org/wiki/Mod_perl and > http://wiki.freeswitch.org/wiki/Mod_perl_and_Generating_XML > > Would this be faster than a setup like > > mod_curl_xml -> lighttpd -> FastCGI/Perl ? > > I guess it would depend on if mod_perl in FreeSWITCH spawns new interpreter per request > or if it uses one interpreter instance with multiple threads to execute pre-loaded perl. > > Anyone know if this is the case? > > Regards, > -Adam > > > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of kokoska rokoska > Sent: Monday, February 09, 2009 1:03 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Dynamic Dialplan > > > > > Raymond Chandler napsal(a): > >> kokoska rokoska wrote: >> >>> Just my 2c: >>> >>> If you use curl with lighttpd and custom built fastcgi "C" responder (it >>> is really simple with fcgi libs - even I can do it :-) performance could >>> be not that bad. >>> >> hmmm, mod_xml_curl using C, interesting thought.. >> > > May be not the best way, but very simple. > Well, it depends on what you have to do, but "directory" serving based > on DB queries (this what I'm using it for) is very simple - just few > lines of code. > > >> all of the >> complexities of writing your own module without the nice structured FS >> API... >> > > I should say I have no idea how hard is to write custom FreeSWITCH > module (may be I should try it :-), but the FS code is very nice! > > >> although, as a benefit, i guess you do get a little extra latency ;-) >> >> > > :-) Yes, you are right. And as a bonus some CPU utilization... > > Like I wrote above, I didn't say it is faster, but IMO it is very simple > and not as slow as it looks (when using apache + php + apc). > > Best regards, > > kokoska.rokoska > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > I'm developing something similar and now on testing mode. xml works just fine. The key is FCGI and web server. i think this week i made a progress in the right direction. will report the progress promptly when done. If you need hand pls drop an e-mail i do not mind sharing. and if it works for you i'll come with kind of how to instructions if anybody else is interested. Regards Chav From chavpaskov at shaw.ca Tue Feb 10 18:41:36 2009 From: chavpaskov at shaw.ca (Chav Paskov) Date: Tue, 10 Feb 2009 18:41:36 -0800 Subject: [Freeswitch-users] Dynamic Dialplan In-Reply-To: <006401c98bef$980e04f0$c82a0ed0$@net> References: <743112140902081921n2bb1bf49mfe36e5e2193c6b04@mail.gmail.com> <4990495C.9010008@post.cz> <49906889.9030801@freeswitch.org> <49906FE6.8000008@post.cz> <000301c98bb2$00d66560$02833020$@net> <191c3a030902101432y5954389dje53f6dc05ce654cc@mail.gmail.com> <006401c98bef$980e04f0$c82a0ed0$@net> Message-ID: <49923AE0.1030802@shaw.ca> Adam Long wrote: > > There sure are lots of options J > > > > Does the perl implementation use a single persistent embedded > interpreter or does it spawn a new interpreter per request? > > > > Things like persistent database handles come to mind. > > Performance would be drastically impacted if it were to spawn new > interpreter per request. > > > > Any ideas? > > > > Regards, > > -Adam > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of > *Anthony Minessale > *Sent:* Tuesday, February 10, 2009 5:33 PM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Dynamic Dialplan > > > > oh yeah, i forgot about those too. > python,perl and lua can all do that. > > On Tue, Feb 10, 2009 at 1:01 PM, Adam Long > wrote: > > What about a mod_perl XML binding like the example here? > http://wiki.freeswitch.org/wiki/Mod_perl and > http://wiki.freeswitch.org/wiki/Mod_perl_and_Generating_XML > > Would this be faster than a setup like > > mod_curl_xml -> lighttpd -> FastCGI/Perl ? > > I guess it would depend on if mod_perl in FreeSWITCH spawns new > interpreter per request > or if it uses one interpreter instance with multiple threads to > execute pre-loaded perl. > > Anyone know if this is the case? > > Regards, > -Adam > > > > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > > [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of > kokoska rokoska > Sent: Monday, February 09, 2009 1:03 PM > To: freeswitch-users at lists.freeswitch.org > > Subject: Re: [Freeswitch-users] Dynamic Dialplan > > > > Raymond Chandler napsal(a): > > kokoska rokoska wrote: > >> Just my 2c: > >> > >> If you use curl with lighttpd and custom built fastcgi "C" > responder (it > >> is really simple with fcgi libs - even I can do it :-) performance > could > >> be not that bad. > > hmmm, mod_xml_curl using C, interesting thought.. > > May be not the best way, but very simple. > Well, it depends on what you have to do, but "directory" serving based > on DB queries (this what I'm using it for) is very simple - just few > lines of code. > > > all of the > > complexities of writing your own module without the nice structured FS > > API... > > I should say I have no idea how hard is to write custom FreeSWITCH > module (may be I should try it :-), but the FS code is very nice! > > > although, as a benefit, i guess you do get a little extra latency ;-) > > > > :-) Yes, you are right. And as a bonus some CPU utilization... > > Like I wrote above, I didn't say it is faster, but IMO it is very simple > and not as slow as it looks (when using apache + php + apc). > > Best regards, > > kokoska.rokoska > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > Perl is the language i'm using. and is working exactly as i expect. Chav From helmut.kuper at ewetel.de Wed Feb 11 02:19:17 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Wed, 11 Feb 2009 11:19:17 +0100 Subject: [Freeswitch-users] little problem with gateway registration In-Reply-To: <191c3a030902100613m1b9a8fc1w787b3e177a373557@mail.gmail.com> References: <499152FE.7090909@ewetel.de> <191c3a030902100613m1b9a8fc1w787b3e177a373557@mail.gmail.com> Message-ID: <4992A625.5080902@ewetel.de> Hi Anthony, well currently both ip addresses and ports (of proxy and registrar) are the same. And it works good as it is now. :) regards Helmut On 10.02.2009 15:13, Anthony Minessale wrote: > register-proxy is for where it actually sends the packet but it will > still say the name of proxy in the packet. > Did you check the destination address of the packet it should end up > as the ip:port of that proxy. > From helmut.kuper at ewetel.de Wed Feb 11 04:09:32 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Wed, 11 Feb 2009 13:09:32 +0100 Subject: [Freeswitch-users] segfault when shoutcast plays mp3 and extension hangs up Message-ID: <4992BFFC.50006@ewetel.de> Hello, today I tried to play a mp3. It works fine until extension hangs up. Then FS (FreeSWITCH Version 1.0.trunk (11698M)) crashed with segfault. The mp3 file was generated by MP3Splitter (http://www.codevisions.de/hp/upload/_files/mp3splitter20.zip) as a piece out of a complete mp3 song. There is a good chance that it generates corrupt mp3s. At least those mp3s are playable in winamp and media player. My dialplan: FS console output shows problems in mp3 file: freeswitch at ippbx-prod-node0> 2009-02-11 11:45:43 [DEBUG] Span:1 Q.931() Timer 0 of call 0 (CRV: 2, State: 0) timed out Note: Illegal Audio-MPEG-Header 0x00000000 at offset 0x10ec15. Note: Trying to resync... Note: Hit end of (available) data during resync. 2009-02-11 11:45:44 [DEBUG] switch_ivr_play_say.c:1261 switch_ivr_play_file() done playing file ./start_fs.sh: line 6: 27201 Segmentation fault (core dumped) bin/freeswitch $1 Here are the backtraces: (gdb) bt #0 0xabe0e408 in mpg123_delete at plt () from /opt/app/voip/ippbx.prod/mod/mod_shout.so #1 0xabe13325 in ?? () from /opt/app/voip/ippbx.prod/mod/mod_shout.so #2 0xb7dc04b9 in switch_core_file_seek (fh=0xa76fef28, cur_pos=0xa76fef88, samples=0, whence=1) at src/switch_core_file.c:345 #3 0xb7dfab82 in switch_ivr_play_file (session=0xa7700030, fh=0xa76fef28, file=0xa7711ed8 "/opt/app/voip/ippbx.prod/sounds/de/callie/qet.mp3", args=0xa76ff0d8) at src/switch_ivr_play_say.c:1262 #4 0xad71da25 in ?? () from /opt/app/voip/ippbx.prod/mod/mod_dptools.so #5 0xb7dc6c54 in switch_core_session_exec (session=0xa7700030, application_interface=0xb346b388, arg=0xb34fd790 "qet.mp3") at src/switch_core_session.c:1332 #6 0xb7dc70fe in switch_core_session_execute_application (session=0xa7700030, app=0xb34fd780 "playback", arg=0xb34fd790 "qet.mp3") at src/switch_core_session.c:1254 #7 0xb7dc93a4 in switch_core_session_run (session=0xa7700030) at src/switch_core_state_machine.c:155 #8 0xb7dc6725 in switch_core_session_thread (thread=0xb34fd3d0, obj=0xa7700030) at src/switch_core_session.c:940 #9 0xb7e30bf6 in dummy_worker (opaque=0xb34fd3d0) at threadproc/unix/thread.c:138 #10 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #11 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 Current language: auto; currently asm (gdb) bt full #0 0xabe0e408 in mpg123_delete at plt () from /opt/app/voip/ippbx.prod/mod/mod_shout.so No locals. #1 0xabe13325 in ?? () from /opt/app/voip/ippbx.prod/mod/mod_shout.so No locals. #2 0xb7dc04b9 in switch_core_file_seek (fh=0xa76fef28, cur_pos=0xa76fef88, samples=0, whence=1) at src/switch_core_file.c:345 status = 4294967295 __PRETTY_FUNCTION__ = "switch_core_file_seek" #3 0xb7dfab82 in switch_ivr_play_file (session=0xa7700030, fh=0xa76fef28, file=0xa7711ed8 "/opt/app/voip/ippbx.prod/sounds/de/callie/qet.mp3", args=0xa76ff0d8) at src/switch_ivr_play_say.c:1262 channel = (switch_channel_t *) 0xa7704598 dtmf = {digit = 0 '\0', duration = 0} interval = 2909916864 samples = 320 framelen = 640 sample_start = olen = 320 llen = 320 write_frame = {codec = 0xa76fefb8, source = 0x0, packet = 0x0, packetlen = 0, data = 0xa7743f30, datalen = 640, buflen = 32768, samples = 320, rate = 16000, payload = 0 '\0', timestamp = 0, seq = 0, ssrc = 0, m = SWITCH_FALSE, flags = 0} timer = {interval = 0, flags = 0, samples = 0, samplecount = 0, timer_interface = 0x0, memory_pool = 0x0, private_info = 0x0, diff = 0, tick = 0} codec = {codec_interface = 0x80cd8f8, implementation = 0x80cdf20, fmtp_in = 0x0, fmtp_out = 0x0, codec_settings = {quality = 0, complexity = 0, enhancement = 0, vad = 0, vbr = 0, vbr_quality = 0, abr = 0, dtx = 0, preproc = 0, pp_vad = 0, pp_agc = 0, pp_agc_level = 0, pp_denoise = 0, pp_dereverb = 0, pp_dereverb_decay = 0, pp_dereverb_level = 0}, flags = 3, memory_pool = 0xb34fbb38, private_info = 0x0, agreed_pt = 0 '\0', mutex = 0xa7711f10} pool = (switch_memory_pool_t *) 0xb34fbb38 status = SWITCH_STATUS_SUCCESS lfh = {file_interface = 0x8136258, flags = 3085, fd = 0x0, samples = 0, samplerate = 16000, native_rate = 16000, channels = 1 '\001', format = 0, sections = 0, seekable = 0, sample_count = 729088, speed = 0, memory_pool = 0xa7712040, prebuf = 0, interval = 0, private_info = 0xa7714048, handler = 0x0, pos = 0, audio_buffer = 0xb3476f88, sp_audio_buffer = 0x0, thresh = 0, silence_hits = 0, offset_pos = 364640, last_pos = 0, vol = 0, resampler = 0x0, buffer = 0x0, dbuf = 0x0, dbuflen = 0, pre_buffer = 0x0, pre_buffer_data = 0x0, pre_buffer_datalen = 0, file = 0xb7eb71f0 "src/switch_ivr_play_say.c", func = 0xb7eb794b "switch_ivr_play_file", line = 894} read_codec = (switch_codec_t *) 0xb34fcb90 p = 0xb7eaf8d4 "current_application" asis = 0 '\0' prefix = timer_name = 0x0 prebuf = eof = 1 bread = __func__ = "switch_ivr_play_file" __PRETTY_FUNCTION__ = "switch_ivr_play_file" #4 0xad71da25 in ?? () from /opt/app/voip/ippbx.prod/mod/mod_dptools.so No locals. #5 0xb7dc6c54 in switch_core_session_exec (session=0xa7700030, application_interface=0xb346b388, arg=0xb34fd790 "qet.mp3") at src/switch_core_session.c:1332 log = lp = event = (switch_event_t *) 0x0 var = channel = (switch_channel_t *) 0xa7704598 expanded = 0xb34fd790 "qet.mp3" app = 0xad725133 "playback" __PRETTY_FUNCTION__ = "switch_core_session_exec" __func__ = "switch_core_session_exec" #6 0xb7dc70fe in switch_core_session_execute_application (session=0xa7700030, app=0xb34fd780 "playback", arg=0xb34fd790 "qet.mp3") at src/switch_core_session.c:1254 application_interface = (switch_application_interface_t *) 0xb346b388 __func__ = "switch_core_session_execute_application" #7 0xb7dc93a4 in switch_core_session_run (session=0xa7700030) at src/switch_core_state_machine.c:155 proceed = global_proceed = do_extra_handlers = state = endstate = endpoint_interface = driver_state_handler = (const switch_state_handler_table_t *) 0xb331f860 application_state_handler = thread_id = 3084680427 env = {{__jmpbuf = {0, 0, 0, 0, 0, 0}, __mask_was_saved = 0, __saved_mask = {__val = {0 }}}} sig = __func__ = "switch_core_session_run" __PRETTY_FUNCTION__ = "switch_core_session_run" #8 0xb7dc6725 in switch_core_session_thread (thread=0xb34fd3d0, obj=0xa7700030) at src/switch_core_session.c:940 session = (switch_core_session_t *) 0xa7700030 event = event_str = 0x0 val = __func__ = "switch_core_session_thread" __PRETTY_FUNCTION__ = "switch_core_session_thread" #9 0xb7e30bf6 in dummy_worker (opaque=0xb34fd3d0) at threadproc/unix/thread.c:138 No locals. #10 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 No symbol table info available. #11 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. (gdb) thread apply all bt Thread 31 (process 27201): #0 0xb7f14410 in __kernel_vsyscall () #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 #2 0xb7e32799 in apr_sleep (t=1000000) at time/unix/time.c:246 #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 #4 0xb7db9154 in switch_console_loop () at src/switch_console.c:792 #5 0xb7dcedf0 in switch_core_runtime_loop (bg=0) at src/switch_core.c:659 #6 0x0804a36a in main (argc=1, argv=0xbffe65c4) at src/switch.c:666 Thread 30 (process 27202): #0 0xb7f14410 in __kernel_vsyscall () #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 #2 0xb7e32799 in apr_sleep (t=100000) at time/unix/time.c:246 #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 #4 0xb7dbe2a4 in pool_thread (thread=0xb7a07da8, obj=0x0) at src/switch_core_memory.c:421 #5 0xb7e30bf6 in dummy_worker (opaque=0xb7a07da8) at threadproc/unix/thread.c:138 #6 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #7 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 Thread 29 (process 27203): #0 0xb7f14410 in __kernel_vsyscall () #1 0xb7d2eaa5 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/tls/i686/cmov/libpthread.so.0 #2 0xb7e2acaa in apr_thread_cond_wait (cond=0xb6fb53b8, mutex=0xb6fb5388) at locks/unix/thread_cond.c:68 #3 0xb7e21a3c in apr_queue_pop (queue=0xb6fb5358, data=0xb6ec03a8) at misc/apr_queue.c:276 #4 0xb7dadd44 in switch_queue_pop (queue=0xb6fb5358, data=0xb6ec03a8) at src/switch_apr.c:879 #5 0xb7ddf879 in switch_event_dispatch_thread (thread=0x8068140, obj=0xb6fb5358) at src/switch_event.c:230 #6 0xb7e30bf6 in dummy_worker (opaque=0x8068140) at threadproc/unix/thread.c:138 #7 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #8 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 Thread 28 (process 27204): #0 0xb7f14410 in __kernel_vsyscall () #1 0xb7d2eaa5 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/tls/i686/cmov/libpthread.so.0 #2 0xb7e2acaa in apr_thread_cond_wait (cond=0x805e380, mutex=0x805e350) at locks/unix/thread_cond.c:68 #3 0xb7e21a3c in apr_queue_pop (queue=0x805e320, data=0xb66bf3a8) at misc/apr_queue.c:276 #4 0xb7dadd44 in switch_queue_pop (queue=0x805e320, data=0xb66bf3a8) at src/switch_apr.c:879 #5 0xb7ddec2d in switch_event_thread (thread=0x8068160, obj=0x805e320) at src/switch_event.c:273 #6 0xb7e30bf6 in dummy_worker (opaque=0x8068160) at threadproc/unix/thread.c:138 #7 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #8 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 Thread 27 (process 27205): #0 0xb7f14410 in __kernel_vsyscall () #1 0xb7d2eaa5 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/tls/i686/cmov/libpthread.so.0 #2 0xb7e2acaa in apr_thread_cond_wait (cond=0xb71d4b38, mutex=0xb71d4b08) at locks/unix/thread_cond.c:68 #3 0xb7e21a3c in apr_queue_pop (queue=0xb71d4ad8, data=0xb5ebe3a8) at misc/apr_queue.c:276 #4 0xb7dadd44 in switch_queue_pop (queue=0xb71d4ad8, data=0xb5ebe3a8) at src/switch_apr.c:879 #5 0xb7ddec2d in switch_event_thread (thread=0x8068180, obj=0xb71d4ad8) at src/switch_event.c:273 #6 0xb7e30bf6 in dummy_worker (opaque=0x8068180) at threadproc/unix/thread.c:138 #7 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 ---Type to continue, or q to quit--- #8 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 Thread 26 (process 27206): #0 0xb7f14410 in __kernel_vsyscall () #1 0xb7d2eaa5 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/tls/i686/cmov/libpthread.so.0 #2 0xb7e2acaa in apr_thread_cond_wait (cond=0xb7171b38, mutex=0xb7171b08) at locks/unix/thread_cond.c:68 #3 0xb7e21a3c in apr_queue_pop (queue=0xb7171ad8, data=0xb56bd3a8) at misc/apr_queue.c:276 #4 0xb7dadd44 in switch_queue_pop (queue=0xb7171ad8, data=0xb56bd3a8) at src/switch_apr.c:879 #5 0xb7ddec2d in switch_event_thread (thread=0x80681a0, obj=0xb7171ad8) at src/switch_event.c:273 #6 0xb7e30bf6 in dummy_worker (opaque=0x80681a0) at threadproc/unix/thread.c:138 #7 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #8 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 Thread 25 (process 27207): #0 0xb7f14410 in __kernel_vsyscall () #1 0xb7d2eaa5 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/tls/i686/cmov/libpthread.so.0 #2 0xb7e2acaa in apr_thread_cond_wait (cond=0xb6fb54a8, mutex=0xb6fb5478) at locks/unix/thread_cond.c:68 #3 0xb7e21a3c in apr_queue_pop (queue=0xb6fb5448, data=0xb4dce3a8) at misc/apr_queue.c:276 #4 0xb7dadd44 in switch_queue_pop (queue=0xb6fb5448, data=0xb4dce3a8) at src/switch_apr.c:879 #5 0xb7e082fd in log_thread (thread=0xb4e30ae0, obj=0x0) at src/switch_log.c:209 #6 0xb7e30bf6 in dummy_worker (opaque=0xb4e30ae0) at threadproc/unix/thread.c:138 #7 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #8 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 Thread 24 (process 27210): #0 0xb7f14410 in __kernel_vsyscall () #1 0xb7d2eaa5 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/tls/i686/cmov/libpthread.so.0 #2 0xb7e2acaa in apr_thread_cond_wait (cond=0x80c0c58, mutex=0x80c0c28) at locks/unix/thread_cond.c:68 #3 0xb7daed54 in switch_thread_cond_wait (cond=0x80c0c58, mutex=0x80c0c28) at src/switch_apr.c:359 #4 0xb7e11266 in switch_cond_next () at src/switch_time.c:203 #5 0xb7dc27a5 in switch_core_sql_thread (thread=0xb3567ae8, obj=0x0) at src/switch_core_sqldb.c:220 #6 0xb7e30bf6 in dummy_worker (opaque=0xb3567ae8) at threadproc/unix/thread.c:138 #7 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #8 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 Thread 23 (process 27211): #0 0xb7f14410 in __kernel_vsyscall () #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 #2 0xb7e32799 in apr_sleep (t=500000) at time/unix/time.c:246 #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 #4 0xb7dd2af4 in switch_scheduler_task_thread (thread=0x80baa90, obj=0x0) at src/switch_scheduler.c:171 #5 0xb7e30bf6 in dummy_worker (opaque=0x80baa90) at threadproc/unix/thread.c:138 #6 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #7 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 Thread 22 (process 27212): #0 0xb7f14410 in __kernel_vsyscall () #1 0xb7c7d676 in epoll_wait () from /lib/tls/i686/cmov/libc.so.6 #2 0xb32b7126 in su_epoll_port_wait_events (self=0x80c6340, tout=1000) at su_epoll_port.c:491 #3 0xb32bd9b8 in su_base_port_step (self=0x80c6340, tout=1000) at su_base_port.c:442 #4 0xb32b8551 in su_port_step (self=0x80c6340, tout=1000) at su_port.h:326 ---Type to continue, or q to quit--- #5 0xb32b8521 in su_root_step (self=0x80c68f0, tout=1000) at su_root.c:730 #6 0xb31e8f29 in sofia_profile_thread_run (thread=0x80d2aa8, obj=0x80d1e10) at sofia.c:831 #7 0xb7e30bf6 in dummy_worker (opaque=0x80d2aa8) at threadproc/unix/thread.c:138 #8 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #9 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 Thread 21 (process 27213): #0 0xb7f14410 in __kernel_vsyscall () #1 0xb7c7d676 in epoll_wait () from /lib/tls/i686/cmov/libc.so.6 #2 0xb32b7126 in su_epoll_port_wait_events (self=0x80dac30, tout=1000) at su_epoll_port.c:491 #3 0xb32bd89b in su_base_port_run (self=0x80dac30) at su_base_port.c:342 #4 0xb32b842b in su_port_run (self=0x80dac30) at su_port.h:312 #5 0xb32b8408 in su_root_run (self=0x80dacb0) at su_root.c:691 #6 0xb32a8b31 in su_pthread_port_clone_main (varg=0xb311c0a8) at su_pthread_port.c:321 #7 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #8 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 Thread 20 (process 27214): #0 0xb7f14410 in __kernel_vsyscall () #1 0xb7c7d676 in epoll_wait () from /lib/tls/i686/cmov/libc.so.6 #2 0xb32b7126 in su_epoll_port_wait_events (self=0x80da208, tout=1000) at su_epoll_port.c:491 #3 0xb32bd9b8 in su_base_port_step (self=0x80da208, tout=1000) at su_base_port.c:442 #4 0xb32b8551 in su_port_step (self=0x80da208, tout=1000) at su_port.h:326 #5 0xb32b8521 in su_root_step (self=0x80d7558, tout=1000) at su_root.c:730 #6 0xb31e8f29 in sofia_profile_thread_run (thread=0x80dde88, obj=0x80dd650) at sofia.c:831 #7 0xb7e30bf6 in dummy_worker (opaque=0x80dde88) at threadproc/unix/thread.c:138 #8 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #9 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 Thread 19 (process 27215): #0 0xb7f14410 in __kernel_vsyscall () #1 0xb7c7d676 in epoll_wait () from /lib/tls/i686/cmov/libc.so.6 #2 0xb32b7126 in su_epoll_port_wait_events (self=0x80e5228, tout=1000) at su_epoll_port.c:491 #3 0xb32bd89b in su_base_port_run (self=0x80e5228) at su_base_port.c:342 #4 0xb32b842b in su_port_run (self=0x80e5228) at su_port.h:312 #5 0xb32b8408 in su_root_run (self=0x80e3b40) at su_root.c:691 #6 0xb32a8b31 in su_pthread_port_clone_main (varg=0xb211a0a8) at su_pthread_port.c:321 #7 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #8 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 Thread 18 (process 27216): #0 0xb7f14410 in __kernel_vsyscall () #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 #2 0xb7e32799 in apr_sleep (t=10000) at time/unix/time.c:246 #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 #4 0xb31d4b37 in sofia_profile_worker_thread_run (thread=0x80d2b88, obj=0x80d1e10) at sofia.c:656 #5 0xb7e30bf6 in dummy_worker (opaque=0x80d2b88) at threadproc/unix/thread.c:138 #6 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #7 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 Thread 17 (process 27217): ---Type to continue, or q to quit--- #0 0xb7f14410 in __kernel_vsyscall () #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 #2 0xb7e32799 in apr_sleep (t=10000) at time/unix/time.c:246 #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 #4 0xb31d4b37 in sofia_profile_worker_thread_run (thread=0x80ddf68, obj=0x80dd650) at sofia.c:656 #5 0xb7e30bf6 in dummy_worker (opaque=0x80ddf68) at threadproc/unix/thread.c:138 #6 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #7 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 Thread 16 (process 27218): #0 0xb7f14410 in __kernel_vsyscall () #1 0xb7c7d676 in epoll_wait () from /lib/tls/i686/cmov/libc.so.6 #2 0xb32b7126 in su_epoll_port_wait_events (self=0x80e66d8, tout=1000) at su_epoll_port.c:491 #3 0xb32bd9b8 in su_base_port_step (self=0x80e66d8, tout=1000) at su_base_port.c:442 #4 0xb32b8551 in su_port_step (self=0x80e66d8, tout=1000) at su_port.h:326 #5 0xb32b8521 in su_root_step (self=0x80e49f0, tout=1000) at su_root.c:730 #6 0xb31e8f29 in sofia_profile_thread_run (thread=0x80e8710, obj=0x80e7e90) at sofia.c:831 #7 0xb7e30bf6 in dummy_worker (opaque=0x80e8710) at threadproc/unix/thread.c:138 #8 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #9 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 Thread 15 (process 27219): #0 0xb7f14410 in __kernel_vsyscall () #1 0xb7c7d676 in epoll_wait () from /lib/tls/i686/cmov/libc.so.6 #2 0xb32b7126 in su_epoll_port_wait_events (self=0x80ee648, tout=1000) at su_epoll_port.c:491 #3 0xb32bd89b in su_base_port_run (self=0x80ee648) at su_base_port.c:342 #4 0xb32b842b in su_port_run (self=0x80ee648) at su_port.h:312 #5 0xb32b8408 in su_root_run (self=0x80f0bd0) at su_root.c:691 #6 0xb32a8b31 in su_pthread_port_clone_main (varg=0xb00b20a8) at su_pthread_port.c:321 #7 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #8 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 Thread 14 (process 27220): #0 0xb7f14410 in __kernel_vsyscall () #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 #2 0xb7e32799 in apr_sleep (t=10000) at time/unix/time.c:246 #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 #4 0xb31d4b37 in sofia_profile_worker_thread_run (thread=0x80e87f0, obj=0x80e7e90) at sofia.c:656 #5 0xb7e30bf6 in dummy_worker (opaque=0x80e87f0) at threadproc/unix/thread.c:138 #6 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #7 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 Thread 13 (process 27221): #0 0xb7f14410 in __kernel_vsyscall () #1 0xb7c72c07 in poll () from /lib/tls/i686/cmov/libc.so.6 #2 0xb337af00 in wanpipe_wait (zchan=0xb3446128, flags=0xae751f80, to=100) at src/ozmod/ozmod_wanpipe/ozmod_wanpipe.c:255 #3 0xae837ae7 in zap_channel_wait (zchan=0x1, flags=0xae751f80, to=100) at src/zap_io.c:1479 #4 0xae80aee8 in zap_isdn_run (me=0xb3416528, obj=0xb341dbc8) at src/ozmod/ozmod_isdn/ozmod_isdn.c:1725 #5 0xae8421ba in thread_launch (args=0xb3416528) at src/zap_threadmutex.c:74 #6 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #7 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 ---Type to continue, or q to quit--- Thread 12 (process 27222): #0 0xb7f14410 in __kernel_vsyscall () #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 #2 0xb7e32799 in apr_sleep (t=100000) at time/unix/time.c:246 #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 #4 0xb31f929e in sofia_presence_event_thread_run (thread=0x80cf958, obj=0x0) at sofia_presence.c:664 #5 0xb7e30bf6 in dummy_worker (opaque=0x80cf958) at threadproc/unix/thread.c:138 #6 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #7 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 Thread 11 (process 27223): #0 0xb7f14410 in __kernel_vsyscall () #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 #2 0xb7e32799 in apr_sleep (t=1000000) at time/unix/time.c:246 #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 #4 0xad706ea6 in ?? () from /opt/app/voip/ippbx.prod/mod/mod_fifo.so #5 0xb7e30bf6 in dummy_worker (opaque=0xad386b90) at threadproc/unix/thread.c:138 #6 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #7 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 Thread 10 (process 27225): #0 0xb7f14410 in __kernel_vsyscall () #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 #2 0xb7e32799 in apr_sleep (t=1000000) at time/unix/time.c:246 #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 #4 0xabdf94ab in ?? () from /opt/app/voip/ippbx.prod/mod/mod_local_stream.so #5 0xb7e30bf6 in dummy_worker (opaque=0x81386e0) at threadproc/unix/thread.c:138 #6 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #7 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 Thread 9 (process 27226): #0 0xb7f14410 in __kernel_vsyscall () #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 #2 0xb7e32799 in apr_sleep (t=1000000) at time/unix/time.c:246 #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 #4 0xabdf94ab in ?? () from /opt/app/voip/ippbx.prod/mod/mod_local_stream.so #5 0xb7e30bf6 in dummy_worker (opaque=0x815e728) at threadproc/unix/thread.c:138 #6 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #7 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 Thread 8 (process 27227): #0 0xb7f14410 in __kernel_vsyscall () #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 #2 0xb7e32799 in apr_sleep (t=1000000) at time/unix/time.c:246 #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 #4 0xabdf94ab in ?? () from /opt/app/voip/ippbx.prod/mod/mod_local_stream.so #5 0xb7e30bf6 in dummy_worker (opaque=0x8184770) at threadproc/unix/thread.c:138 #6 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #7 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 ---Type to continue, or q to quit--- Thread 7 (process 27228): #0 0xb7f14410 in __kernel_vsyscall () #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 #2 0xb7e32799 in apr_sleep (t=1000) at time/unix/time.c:246 #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 #4 0xb7e12a59 in softtimer_runtime () at src/switch_time.c:459 #5 0xb7dd4fb3 in switch_loadable_module_exec (thread=0x80bf8d8, obj=0x80bf6c8) at src/switch_loadable_module.c:93 #6 0xb7e30bf6 in dummy_worker (opaque=0x80bf8d8) at threadproc/unix/thread.c:138 #7 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #8 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 Thread 6 (process 27229): #0 0xb7f14410 in __kernel_vsyscall () #1 0xb7d31bb8 in accept () from /lib/tls/i686/cmov/libpthread.so.0 #2 0xb7e2f90d in apr_socket_accept (new=0xaa2d434c, sock=0x81bbbb0, connection_context=0x81bdaa8) at network_io/unix/sockets.c:187 #3 0xb7dae3fb in switch_socket_accept (new_sock=0xaa2d434c, sock=0x81bbbb0, pool=0x81bdaa8) at src/switch_apr.c:664 #4 0xb33262f2 in mod_event_socket_runtime () at mod_event_socket.c:2134 #5 0xb7dd4fb3 in switch_loadable_module_exec (thread=0x80bfb40, obj=0x80bf930) at src/switch_loadable_module.c:93 #6 0xb7e30bf6 in dummy_worker (opaque=0x80bfb40) at threadproc/unix/thread.c:138 #7 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #8 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 Thread 5 (process 27230): #0 0xb7f14410 in __kernel_vsyscall () #1 0xb7c72c07 in poll () from /lib/tls/i686/cmov/libc.so.6 #2 0xb33ce86e in ?? () from /opt/app/voip/ippbx.prod/mod/mod_xml_rpc.so #3 0xb33c1464 in ChanSwitchAccept (chanSwitchP=0x81f5030, channelPP=0xa9ad30e0, channelInfoPP=0xa9ad30dc, errorP=0xa9ad30e4) at ../../../../libs/xmlrpc-c/lib/abyss/src/chanswitch.c:149 #4 0xb33cd37e in ServerRun (serverP=0xb33ffe4c) at ../../../../libs/xmlrpc-c/lib/abyss/src/server.c:908 #5 0xb33be832 in mod_xml_rpc_runtime () at mod_xml_rpc.c:837 #6 0xb7dd4fb3 in switch_loadable_module_exec (thread=0x80bfda8, obj=0x80bfb98) at src/switch_loadable_module.c:93 #7 0xb7e30bf6 in dummy_worker (opaque=0x80bfda8) at threadproc/unix/thread.c:138 #8 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #9 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 Thread 4 (process 27231): #0 0xb7f14410 in __kernel_vsyscall () #1 0xb7d3199b in read () from /lib/tls/i686/cmov/libpthread.so.0 #2 0xb7e9dfb3 in read_char (el=0x81c5040, cp=0xa92d235b "?|???`!\b\200a\"\b?#-?;\237?@P\034\b?#-?\a") at read.c:294 #3 0xb7e9da9c in el_getc (el=0x81c5040, cp=0xa92d235b "?|???`!\b\200a\"\b?#-?;\237?@P\034\b?#-?\a") at read.c:362 #4 0xb7e9dbdf in el_gets (el=0x81c5040, nread=0xa92d23a8) at read.c:241 #5 0xb7db9f3b in console_thread (thread=0x82102d0, obj=0x8210248) at src/switch_console.c:441 #6 0xb7e30bf6 in dummy_worker (opaque=0x82102d0) at threadproc/unix/thread.c:138 #7 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #8 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 Thread 3 (process 27235): #0 0xb7f14410 in __kernel_vsyscall () #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 #2 0xb7e32799 in apr_sleep (t=100000) at time/unix/time.c:246 #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 ---Type to continue, or q to quit--- #4 0xb7df019e in switch_ivr_originate (session=0xb34dd080, bleg=0xa8ad10c0, cause=0xa8ad10bc, bridgeto=0x823a528 "openzap/1/a/04855711", timelimit_sec=60, table=0xb7ecda20, cid_name_override=0x0, cid_num_override=0x0, caller_profile_override=0x0, ovars=0x0, flags=) at src/switch_ivr_originate.c:1793 #5 0xad7231af in ?? () from /opt/app/voip/ippbx.prod/mod/mod_dptools.so #6 0xb7dc6c54 in switch_core_session_exec (session=0xb34dd080, application_interface=0xb346ba10, arg=0x823a528 "openzap/1/a/04855711") at src/switch_core_session.c:1332 #7 0xb7dc70fe in switch_core_session_execute_application (session=0xb34dd080, app=0x823a520 "bridge", arg=0x823a528 "openzap/1/a/04855711") at src/switch_core_session.c:1254 #8 0xb7dc93a4 in switch_core_session_run (session=0xb34dd080) at src/switch_core_state_machine.c:155 #9 0xb7dc6725 in switch_core_session_thread (thread=0x823a050, obj=0xb34dd080) at src/switch_core_session.c:940 #10 0xb7e30bf6 in dummy_worker (opaque=0x823a050) at threadproc/unix/thread.c:138 #11 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #12 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 Thread 2 (process 27236): #0 0xb7f14410 in __kernel_vsyscall () #1 0xb7c72c07 in poll () from /lib/tls/i686/cmov/libc.so.6 #2 0xb337af00 in wanpipe_wait (zchan=0xb341f9d8, flags=0xa80fee60, to=40) at src/ozmod/ozmod_wanpipe/ozmod_wanpipe.c:255 #3 0xae837ae7 in zap_channel_wait (zchan=0x1, flags=0xa80fee60, to=40) at src/zap_io.c:1479 #4 0xae849059 in channel_read_frame (session=0xb34e5400, frame=0xa80ff170, flags=0, stream_id=0) at mod_openzap.c:593 #5 0xb7dcbe6f in switch_core_session_read_frame (session=0xb34e5400, frame=0xa80ff170, flags=0, stream_id=0) at src/switch_core_io.c:161 #6 0xb7e054ec in switch_ivr_sleep (session=0xb34e5400, ms=10, sync=SWITCH_FALSE, args=0x0) at src/switch_ivr.c:262 #7 0xb7deaf94 in originate_on_consume_media_transmit (session=0xb34e5400) at src/switch_ivr_originate.c:47 #8 0xb7dc8b74 in switch_core_session_run (session=0xb34e5400) at src/switch_core_state_machine.c:476 #9 0xb7dc6725 in switch_core_session_thread (thread=0xb34f1060, obj=0xb34e5400) at src/switch_core_session.c:940 #10 0xb7e30bf6 in dummy_worker (opaque=0xb34f1060) at threadproc/unix/thread.c:138 #11 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #12 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 Thread 1 (process 27234): #0 0xabe0e408 in mpg123_delete at plt () from /opt/app/voip/ippbx.prod/mod/mod_shout.so #1 0xabe13325 in ?? () from /opt/app/voip/ippbx.prod/mod/mod_shout.so #2 0xb7dc04b9 in switch_core_file_seek (fh=0xa76fef28, cur_pos=0xa76fef88, samples=0, whence=1) at src/switch_core_file.c:345 #3 0xb7dfab82 in switch_ivr_play_file (session=0xa7700030, fh=0xa76fef28, file=0xa7711ed8 "/opt/app/voip/ippbx.prod/sounds/de/callie/qet.mp3", args=0xa76ff0d8) at src/switch_ivr_play_say.c:1262 #4 0xad71da25 in ?? () from /opt/app/voip/ippbx.prod/mod/mod_dptools.so #5 0xb7dc6c54 in switch_core_session_exec (session=0xa7700030, application_interface=0xb346b388, arg=0xb34fd790 "qet.mp3") at src/switch_core_session.c:1332 #6 0xb7dc70fe in switch_core_session_execute_application (session=0xa7700030, app=0xb34fd780 "playback", arg=0xb34fd790 "qet.mp3") at src/switch_core_session.c:1254 #7 0xb7dc93a4 in switch_core_session_run (session=0xa7700030) at src/switch_core_state_machine.c:155 #8 0xb7dc6725 in switch_core_session_thread (thread=0xb34fd3d0, obj=0xa7700030) at src/switch_core_session.c:940 #9 0xb7e30bf6 in dummy_worker (opaque=0xb34fd3d0) at threadproc/unix/thread.c:138 #10 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #11 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 (gdb) thread apply all bt full Thread 31 (process 27201): #0 0xb7f14410 in __kernel_vsyscall () No symbol table info available. #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. #2 0xb7e32799 in apr_sleep (t=1000000) at time/unix/time.c:246 tv = {tv_sec = 0, tv_usec = 940000} #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 No locals. #4 0xb7db9154 in switch_console_loop () at src/switch_console.c:792 arg = 1 thread = (switch_thread_t *) 0x82102d0 thd_attr = (switch_threadattr_t *) 0x8210298 pool = (switch_memory_pool_t *) 0x8210248 __func__ = "switch_console_loop" __PRETTY_FUNCTION__ = "switch_console_loop" #5 0xb7dcedf0 in switch_core_runtime_loop (bg=0) at src/switch_core.c:659 No locals. #6 0x0804a36a in main (argc=1, argv=0xbffe65c4) at src/switch.c:666 pid_path = "/opt/app/voip/ippbx.prod/log/freeswitch.pid", '\0' pid_buffer = "27201", '\0' old_pid_buffer = "27150", '\0' pid_len = 5 old_pid_len = 5 err = 0x0 nf = 0 runas_user = 0x0 runas_group = 0x0 nc = 0 pid = 27201 x = 1111804576 die = 0 alt_dirs = 0 known_opt = -1208927888 high_prio = 0 flags = 1 ret = destroy_status = fd = (switch_file_t *) 0x80529b0 pool = (switch_memory_pool_t *) 0x80528f0 __PRETTY_FUNCTION__ = "main" Thread 30 (process 27202): #0 0xb7f14410 in __kernel_vsyscall () No symbol table info available. #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. #2 0xb7e32799 in apr_sleep (t=100000) at time/unix/time.c:246 tv = {tv_sec = 0, tv_usec = 100000} #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 ---Type to continue, or q to quit--- No locals. #4 0xb7dbe2a4 in pool_thread (thread=0xb7a07da8, obj=0x0) at src/switch_core_memory.c:421 No locals. #5 0xb7e30bf6 in dummy_worker (opaque=0xb7a07da8) at threadproc/unix/thread.c:138 No locals. #6 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 No symbol table info available. #7 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. Thread 29 (process 27203): #0 0xb7f14410 in __kernel_vsyscall () No symbol table info available. #1 0xb7d2eaa5 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/tls/i686/cmov/libpthread.so.0 No symbol table info available. #2 0xb7e2acaa in apr_thread_cond_wait (cond=0xb6fb53b8, mutex=0xb6fb5388) at locks/unix/thread_cond.c:68 rv = -512 #3 0xb7e21a3c in apr_queue_pop (queue=0xb6fb5358, data=0xb6ec03a8) at misc/apr_queue.c:276 rv = 0 #4 0xb7dadd44 in switch_queue_pop (queue=0xb6fb5358, data=0xb6ec03a8) at src/switch_apr.c:879 No locals. #5 0xb7ddf879 in switch_event_dispatch_thread (thread=0x8068140, obj=0xb6fb5358) at src/switch_event.c:230 pop = (void *) 0x0 event = (switch_event_t *) 0x0 my_id = 0 __func__ = "switch_event_dispatch_thread" #6 0xb7e30bf6 in dummy_worker (opaque=0x8068140) at threadproc/unix/thread.c:138 No locals. #7 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 No symbol table info available. #8 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. Thread 28 (process 27204): #0 0xb7f14410 in __kernel_vsyscall () No symbol table info available. #1 0xb7d2eaa5 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/tls/i686/cmov/libpthread.so.0 No symbol table info available. #2 0xb7e2acaa in apr_thread_cond_wait (cond=0x805e380, mutex=0x805e350) at locks/unix/thread_cond.c:68 rv = -512 #3 0xb7e21a3c in apr_queue_pop (queue=0x805e320, data=0xb66bf3a8) at misc/apr_queue.c:276 rv = 0 #4 0xb7dadd44 in switch_queue_pop (queue=0x805e320, data=0xb66bf3a8) at src/switch_apr.c:879 No locals. #5 0xb7ddec2d in switch_event_thread (thread=0x8068160, obj=0x805e320) at src/switch_event.c:273 pop = (void *) 0x0 index = 0 my_id = 0 __func__ = "switch_event_thread" #6 0xb7e30bf6 in dummy_worker (opaque=0x8068160) at threadproc/unix/thread.c:138 No locals. ---Type to continue, or q to quit--- #7 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 No symbol table info available. #8 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. Thread 27 (process 27205): #0 0xb7f14410 in __kernel_vsyscall () No symbol table info available. #1 0xb7d2eaa5 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/tls/i686/cmov/libpthread.so.0 No symbol table info available. #2 0xb7e2acaa in apr_thread_cond_wait (cond=0xb71d4b38, mutex=0xb71d4b08) at locks/unix/thread_cond.c:68 rv = -512 #3 0xb7e21a3c in apr_queue_pop (queue=0xb71d4ad8, data=0xb5ebe3a8) at misc/apr_queue.c:276 rv = 0 #4 0xb7dadd44 in switch_queue_pop (queue=0xb71d4ad8, data=0xb5ebe3a8) at src/switch_apr.c:879 No locals. #5 0xb7ddec2d in switch_event_thread (thread=0x8068180, obj=0xb71d4ad8) at src/switch_event.c:273 pop = (void *) 0x0 index = 0 my_id = 1 __func__ = "switch_event_thread" #6 0xb7e30bf6 in dummy_worker (opaque=0x8068180) at threadproc/unix/thread.c:138 No locals. #7 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 No symbol table info available. #8 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. Thread 26 (process 27206): #0 0xb7f14410 in __kernel_vsyscall () No symbol table info available. #1 0xb7d2eaa5 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/tls/i686/cmov/libpthread.so.0 No symbol table info available. #2 0xb7e2acaa in apr_thread_cond_wait (cond=0xb7171b38, mutex=0xb7171b08) at locks/unix/thread_cond.c:68 rv = -512 #3 0xb7e21a3c in apr_queue_pop (queue=0xb7171ad8, data=0xb56bd3a8) at misc/apr_queue.c:276 rv = 0 #4 0xb7dadd44 in switch_queue_pop (queue=0xb7171ad8, data=0xb56bd3a8) at src/switch_apr.c:879 No locals. #5 0xb7ddec2d in switch_event_thread (thread=0x80681a0, obj=0xb7171ad8) at src/switch_event.c:273 pop = (void *) 0x0 index = 0 my_id = 2 __func__ = "switch_event_thread" #6 0xb7e30bf6 in dummy_worker (opaque=0x80681a0) at threadproc/unix/thread.c:138 No locals. #7 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 No symbol table info available. #8 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. ---Type to continue, or q to quit--- Thread 25 (process 27207): #0 0xb7f14410 in __kernel_vsyscall () No symbol table info available. #1 0xb7d2eaa5 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/tls/i686/cmov/libpthread.so.0 No symbol table info available. #2 0xb7e2acaa in apr_thread_cond_wait (cond=0xb6fb54a8, mutex=0xb6fb5478) at locks/unix/thread_cond.c:68 rv = -512 #3 0xb7e21a3c in apr_queue_pop (queue=0xb6fb5448, data=0xb4dce3a8) at misc/apr_queue.c:276 rv = 0 #4 0xb7dadd44 in switch_queue_pop (queue=0xb6fb5448, data=0xb4dce3a8) at src/switch_apr.c:879 No locals. #5 0xb7e082fd in log_thread (thread=0xb4e30ae0, obj=0x0) at src/switch_log.c:209 pop = (void *) 0x0 binding = (switch_log_binding_t *) 0x0 __func__ = "log_thread" #6 0xb7e30bf6 in dummy_worker (opaque=0xb4e30ae0) at threadproc/unix/thread.c:138 No locals. #7 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 No symbol table info available. #8 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. Thread 24 (process 27210): #0 0xb7f14410 in __kernel_vsyscall () No symbol table info available. #1 0xb7d2eaa5 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/tls/i686/cmov/libpthread.so.0 No symbol table info available. #2 0xb7e2acaa in apr_thread_cond_wait (cond=0x80c0c58, mutex=0x80c0c28) at locks/unix/thread_cond.c:68 rv = -512 #3 0xb7daed54 in switch_thread_cond_wait (cond=0x80c0c58, mutex=0x80c0c28) at src/switch_apr.c:359 No locals. #4 0xb7e11266 in switch_cond_next () at src/switch_time.c:203 No locals. #5 0xb7dc27a5 in switch_core_sql_thread (thread=0xb3567ae8, obj=0x0) at src/switch_core_sqldb.c:220 pop = (void *) 0x811bf00 itterations = 0 trans = 0 '\0' nothing_in_queue = 1 '\001' len = 100 sql_len = 65536 sqlbuf = 0x80aa9a8 "" newlen = lc = 0 __PRETTY_FUNCTION__ = "switch_core_sql_thread" __func__ = "switch_core_sql_thread" #6 0xb7e30bf6 in dummy_worker (opaque=0xb3567ae8) at threadproc/unix/thread.c:138 No locals. #7 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 No symbol table info available. #8 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. ---Type to continue, or q to quit--- Thread 23 (process 27211): #0 0xb7f14410 in __kernel_vsyscall () No symbol table info available. #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. #2 0xb7e32799 in apr_sleep (t=500000) at time/unix/time.c:246 tv = {tv_sec = 0, tv_usec = 340000} #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 No locals. #4 0xb7dd2af4 in switch_scheduler_task_thread (thread=0x80baa90, obj=0x0) at src/switch_scheduler.c:171 __func__ = "switch_scheduler_task_thread" #5 0xb7e30bf6 in dummy_worker (opaque=0x80baa90) at threadproc/unix/thread.c:138 No locals. #6 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 No symbol table info available. #7 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. Thread 22 (process 27212): #0 0xb7f14410 in __kernel_vsyscall () No symbol table info available. #1 0xb7c7d676 in epoll_wait () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. #2 0xb32b7126 in su_epoll_port_wait_events (self=0x80c6340, tout=1000) at su_epoll_port.c:491 j = -1290679408 n = -1288576964 events = 0 index = -1288973717 version = 1 M = 4 ev = 0xb311c0e0 __PRETTY_FUNCTION__ = "su_epoll_port_wait_events" #3 0xb32bd9b8 in su_base_port_step (self=0x80c6340, tout=1000) at su_base_port.c:442 now = {tv_sec = 3443337943, tv_usec = 249063} __PRETTY_FUNCTION__ = "su_base_port_step" #4 0xb32b8551 in su_port_step (self=0x80c6340, tout=1000) at su_port.h:326 base = (su_virtual_port_t *) 0x80c6340 #5 0xb32b8521 in su_root_step (self=0x80c68f0, tout=1000) at su_root.c:730 __PRETTY_FUNCTION__ = "su_root_step" #6 0xb31e8f29 in sofia_profile_thread_run (thread=0x80d2aa8, obj=0x80d1e10) at sofia.c:831 pool = node = (sip_alias_node_t *) 0xb32f45dc s_event = (switch_event_t *) 0x0 sanity = __func__ = "sofia_profile_thread_run" __PRETTY_FUNCTION__ = "sofia_profile_thread_run" #7 0xb7e30bf6 in dummy_worker (opaque=0x80d2aa8) at threadproc/unix/thread.c:138 No locals. #8 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 No symbol table info available. ---Type to continue, or q to quit--- #9 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. Thread 21 (process 27213): #0 0xb7f14410 in __kernel_vsyscall () No symbol table info available. #1 0xb7c7d676 in epoll_wait () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. #2 0xb32b7126 in su_epoll_port_wait_events (self=0x80dac30, tout=1000) at su_epoll_port.c:491 j = -1299074328 n = 1 events = 0 index = 2 version = 4 M = 4 ev = 0xb291b260 __PRETTY_FUNCTION__ = "su_epoll_port_wait_events" #3 0xb32bd89b in su_base_port_run (self=0x80dac30) at su_base_port.c:342 tout = 1000 __PRETTY_FUNCTION__ = "su_base_port_run" #4 0xb32b842b in su_port_run (self=0x80dac30) at su_port.h:312 base = (su_virtual_port_t *) 0x80dac30 #5 0xb32b8408 in su_root_run (self=0x80dacb0) at su_root.c:691 __PRETTY_FUNCTION__ = "su_root_run" #6 0xb32a8b31 in su_pthread_port_clone_main (varg=0xb311c0a8) at su_pthread_port.c:321 arg = (struct clone_args *) 0x0 task = {{sut_port = 0x80dac30, sut_root = 0x80dacb0}} zap = 1 #7 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 No symbol table info available. #8 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. Thread 20 (process 27214): #0 0xb7f14410 in __kernel_vsyscall () No symbol table info available. #1 0xb7c7d676 in epoll_wait () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. #2 0xb32b7126 in su_epoll_port_wait_events (self=0x80da208, tout=1000) at su_epoll_port.c:491 j = -1307464816 n = -1288576964 events = 0 index = -1288973717 version = 1 M = 4 ev = 0xb211a0e0 __PRETTY_FUNCTION__ = "su_epoll_port_wait_events" #3 0xb32bd9b8 in su_base_port_step (self=0x80da208, tout=1000) at su_base_port.c:442 now = {tv_sec = 3443337943, tv_usec = 148997} __PRETTY_FUNCTION__ = "su_base_port_step" #4 0xb32b8551 in su_port_step (self=0x80da208, tout=1000) at su_port.h:326 ---Type to continue, or q to quit--- base = (su_virtual_port_t *) 0x80da208 #5 0xb32b8521 in su_root_step (self=0x80d7558, tout=1000) at su_root.c:730 __PRETTY_FUNCTION__ = "su_root_step" #6 0xb31e8f29 in sofia_profile_thread_run (thread=0x80dde88, obj=0x80dd650) at sofia.c:831 pool = node = (sip_alias_node_t *) 0xb32f45dc s_event = (switch_event_t *) 0x0 sanity = __func__ = "sofia_profile_thread_run" __PRETTY_FUNCTION__ = "sofia_profile_thread_run" #7 0xb7e30bf6 in dummy_worker (opaque=0x80dde88) at threadproc/unix/thread.c:138 No locals. #8 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 No symbol table info available. #9 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. Thread 19 (process 27215): #0 0xb7f14410 in __kernel_vsyscall () No symbol table info available. #1 0xb7c7d676 in epoll_wait () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. #2 0xb32b7126 in su_epoll_port_wait_events (self=0x80e5228, tout=1000) at su_epoll_port.c:491 j = -1315859736 n = 1 events = 0 index = 1 version = 3 M = 4 ev = 0xb1919260 __PRETTY_FUNCTION__ = "su_epoll_port_wait_events" #3 0xb32bd89b in su_base_port_run (self=0x80e5228) at su_base_port.c:342 tout = 1000 __PRETTY_FUNCTION__ = "su_base_port_run" #4 0xb32b842b in su_port_run (self=0x80e5228) at su_port.h:312 base = (su_virtual_port_t *) 0x80e5228 #5 0xb32b8408 in su_root_run (self=0x80e3b40) at su_root.c:691 __PRETTY_FUNCTION__ = "su_root_run" #6 0xb32a8b31 in su_pthread_port_clone_main (varg=0xb211a0a8) at su_pthread_port.c:321 arg = (struct clone_args *) 0x0 task = {{sut_port = 0x80e5228, sut_root = 0x80e3b40}} zap = 1 #7 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 No symbol table info available. #8 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. Thread 18 (process 27216): #0 0xb7f14410 in __kernel_vsyscall () No symbol table info available. #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 ---Type to continue, or q to quit--- No symbol table info available. #2 0xb7e32799 in apr_sleep (t=10000) at time/unix/time.c:246 tv = {tv_sec = 0, tv_usec = 10000} #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 No locals. #4 0xb31d4b37 in sofia_profile_worker_thread_run (thread=0x80d2b88, obj=0x80d1e10) at sofia.c:656 ireg_loops = 1 gateway_loops = 0 loops = 95 qsize = 4294966782 __PRETTY_FUNCTION__ = "sofia_profile_worker_thread_run" #5 0xb7e30bf6 in dummy_worker (opaque=0x80d2b88) at threadproc/unix/thread.c:138 No locals. #6 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 No symbol table info available. #7 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. Thread 17 (process 27217): #0 0xb7f14410 in __kernel_vsyscall () No symbol table info available. #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. #2 0xb7e32799 in apr_sleep (t=10000) at time/unix/time.c:246 tv = {tv_sec = 0, tv_usec = 10000} #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 No locals. #4 0xb31d4b37 in sofia_profile_worker_thread_run (thread=0x80ddf68, obj=0x80dd650) at sofia.c:656 ireg_loops = 1 gateway_loops = 0 loops = 95 qsize = 4294966782 __PRETTY_FUNCTION__ = "sofia_profile_worker_thread_run" #5 0xb7e30bf6 in dummy_worker (opaque=0x80ddf68) at threadproc/unix/thread.c:138 No locals. #6 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 No symbol table info available. #7 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. Thread 16 (process 27218): #0 0xb7f14410 in __kernel_vsyscall () No symbol table info available. #1 0xb7c7d676 in epoll_wait () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. #2 0xb32b7126 in su_epoll_port_wait_events (self=0x80e66d8, tout=1000) at su_epoll_port.c:491 j = -1341445232 n = -1288576964 events = 0 index = -1288973717 version = 1 ---Type to continue, or q to quit--- M = 4 ev = 0xb00b20e0 __PRETTY_FUNCTION__ = "su_epoll_port_wait_events" #3 0xb32bd9b8 in su_base_port_step (self=0x80e66d8, tout=1000) at su_base_port.c:442 now = {tv_sec = 3443337943, tv_usec = 560073} __PRETTY_FUNCTION__ = "su_base_port_step" #4 0xb32b8551 in su_port_step (self=0x80e66d8, tout=1000) at su_port.h:326 base = (su_virtual_port_t *) 0x80e66d8 #5 0xb32b8521 in su_root_step (self=0x80e49f0, tout=1000) at su_root.c:730 __PRETTY_FUNCTION__ = "su_root_step" #6 0xb31e8f29 in sofia_profile_thread_run (thread=0x80e8710, obj=0x80e7e90) at sofia.c:831 pool = node = (sip_alias_node_t *) 0xb32f45dc s_event = (switch_event_t *) 0x0 sanity = __func__ = "sofia_profile_thread_run" __PRETTY_FUNCTION__ = "sofia_profile_thread_run" #7 0xb7e30bf6 in dummy_worker (opaque=0x80e8710) at threadproc/unix/thread.c:138 No locals. #8 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 No symbol table info available. #9 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. Thread 15 (process 27219): #0 0xb7f14410 in __kernel_vsyscall () No symbol table info available. #1 0xb7c7d676 in epoll_wait () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. #2 0xb32b7126 in su_epoll_port_wait_events (self=0x80ee648, tout=1000) at su_epoll_port.c:491 j = -1349840152 n = 1 events = 0 index = 1 version = 3 M = 4 ev = 0xaf8b1260 __PRETTY_FUNCTION__ = "su_epoll_port_wait_events" #3 0xb32bd89b in su_base_port_run (self=0x80ee648) at su_base_port.c:342 tout = 1000 __PRETTY_FUNCTION__ = "su_base_port_run" #4 0xb32b842b in su_port_run (self=0x80ee648) at su_port.h:312 base = (su_virtual_port_t *) 0x80ee648 #5 0xb32b8408 in su_root_run (self=0x80f0bd0) at su_root.c:691 __PRETTY_FUNCTION__ = "su_root_run" #6 0xb32a8b31 in su_pthread_port_clone_main (varg=0xb00b20a8) at su_pthread_port.c:321 arg = (struct clone_args *) 0x0 task = {{sut_port = 0x80ee648, sut_root = 0x80f0bd0}} zap = 1 #7 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 No symbol table info available. ---Type to continue, or q to quit--- #8 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. Thread 14 (process 27220): #0 0xb7f14410 in __kernel_vsyscall () No symbol table info available. #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. #2 0xb7e32799 in apr_sleep (t=10000) at time/unix/time.c:246 tv = {tv_sec = 0, tv_usec = 0} #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 No locals. #4 0xb31d4b37 in sofia_profile_worker_thread_run (thread=0x80e87f0, obj=0x80e7e90) at sofia.c:656 ireg_loops = 0 gateway_loops = 0 loops = 85 qsize = 4294966782 __PRETTY_FUNCTION__ = "sofia_profile_worker_thread_run" #5 0xb7e30bf6 in dummy_worker (opaque=0x80e87f0) at threadproc/unix/thread.c:138 No locals. #6 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 No symbol table info available. #7 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. Thread 13 (process 27221): #0 0xb7f14410 in __kernel_vsyscall () No symbol table info available. #1 0xb7c72c07 in poll () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. #2 0xb337af00 in wanpipe_wait (zchan=0xb3446128, flags=0xae751f80, to=100) at src/ozmod/ozmod_wanpipe/ozmod_wanpipe.c:255 inflags = 1 result = #3 0xae837ae7 in zap_channel_wait (zchan=0x1, flags=0xae751f80, to=100) at src/zap_io.c:1479 __PRETTY_FUNCTION__ = "zap_channel_wait" #4 0xae80aee8 in zap_isdn_run (me=0xb3416528, obj=0xb341dbc8) at src/ozmod/ozmod_isdn/ozmod_isdn.c:1725 flags = ZAP_READ status = ZAP_TIMEOUT span = isdn_data = (zap_isdn_data_t *) 0xae753008 frame = "\002\001\002\002\b\002\200\002\001\036\002\201\210", '\0' len = 13 errs = 0 __func__ = "zap_isdn_run" #5 0xae8421ba in thread_launch (args=0xb3416528) at src/zap_threadmutex.c:74 exit_val = #6 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 No symbol table info available. #7 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. ---Type to continue, or q to quit--- Thread 12 (process 27222): #0 0xb7f14410 in __kernel_vsyscall () No symbol table info available. #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. #2 0xb7e32799 in apr_sleep (t=100000) at time/unix/time.c:246 tv = {tv_sec = 0, tv_usec = 100000} #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 No locals. #4 0xb31f929e in sofia_presence_event_thread_run (thread=0x80cf958, obj=0x0) at sofia_presence.c:664 count = 0 pop = (void *) 0xb3462cb8 __func__ = "sofia_presence_event_thread_run" #5 0xb7e30bf6 in dummy_worker (opaque=0x80cf958) at threadproc/unix/thread.c:138 No locals. #6 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 No symbol table info available. #7 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. Thread 11 (process 27223): #0 0xb7f14410 in __kernel_vsyscall () No symbol table info available. #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. #2 0xb7e32799 in apr_sleep (t=1000000) at time/unix/time.c:246 tv = {tv_sec = 0, tv_usec = 320000} #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 No locals. #4 0xad706ea6 in ?? () from /opt/app/voip/ippbx.prod/mod/mod_fifo.so No locals. #5 0xb7e30bf6 in dummy_worker (opaque=0xad386b90) at threadproc/unix/thread.c:138 No locals. #6 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 No symbol table info available. #7 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. Thread 10 (process 27225): #0 0xb7f14410 in __kernel_vsyscall () No symbol table info available. #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. #2 0xb7e32799 in apr_sleep (t=1000000) at time/unix/time.c:246 tv = {tv_sec = 0, tv_usec = 840000} #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 No locals. #4 0xabdf94ab in ?? () from /opt/app/voip/ippbx.prod/mod/mod_local_stream.so No locals. #5 0xb7e30bf6 in dummy_worker (opaque=0x81386e0) at threadproc/unix/thread.c:138 No locals. ---Type to continue, or q to quit--- #6 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 No symbol table info available. #7 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. Thread 9 (process 27226): #0 0xb7f14410 in __kernel_vsyscall () No symbol table info available. #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. #2 0xb7e32799 in apr_sleep (t=1000000) at time/unix/time.c:246 tv = {tv_sec = 0, tv_usec = 840000} #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 No locals. #4 0xabdf94ab in ?? () from /opt/app/voip/ippbx.prod/mod/mod_local_stream.so No locals. #5 0xb7e30bf6 in dummy_worker (opaque=0x815e728) at threadproc/unix/thread.c:138 No locals. #6 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 No symbol table info available. #7 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. Thread 8 (process 27227): #0 0xb7f14410 in __kernel_vsyscall () No symbol table info available. #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. #2 0xb7e32799 in apr_sleep (t=1000000) at time/unix/time.c:246 tv = {tv_sec = 0, tv_usec = 840000} #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 No locals. #4 0xabdf94ab in ?? () from /opt/app/voip/ippbx.prod/mod/mod_local_stream.so No locals. #5 0xb7e30bf6 in dummy_worker (opaque=0x8184770) at threadproc/unix/thread.c:138 No locals. #6 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 No symbol table info available. #7 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. Thread 7 (process 27228): #0 0xb7f14410 in __kernel_vsyscall () No symbol table info available. #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. #2 0xb7e32799 in apr_sleep (t=1000) at time/unix/time.c:246 tv = {tv_sec = 0, tv_usec = 1000} #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 No locals. #4 0xb7e12a59 in softtimer_runtime () at src/switch_time.c:459 ---Type to continue, or q to quit--- current_ms = 1544 x = tick = 144 ts = last = 1234349144129112 fwd_errs = 0 rev_errs = 0 __func__ = "softtimer_runtime" #5 0xb7dd4fb3 in switch_loadable_module_exec (thread=0x80bf8d8, obj=0x80bf6c8) at src/switch_loadable_module.c:93 status = module = (switch_loadable_module_t *) 0x80c0bc8 __PRETTY_FUNCTION__ = "switch_loadable_module_exec" __func__ = "switch_loadable_module_exec" #6 0xb7e30bf6 in dummy_worker (opaque=0x80bf8d8) at threadproc/unix/thread.c:138 No locals. #7 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 No symbol table info available. #8 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. Thread 6 (process 27229): #0 0xb7f14410 in __kernel_vsyscall () No symbol table info available. #1 0xb7d31bb8 in accept () from /lib/tls/i686/cmov/libpthread.so.0 No symbol table info available. #2 0xb7e2f90d in apr_socket_accept (new=0xaa2d434c, sock=0x81bbbb0, connection_context=0x81bdaa8) at network_io/unix/sockets.c:187 No locals. #3 0xb7dae3fb in switch_socket_accept (new_sock=0xaa2d434c, sock=0x81bbbb0, pool=0x81bdaa8) at src/switch_apr.c:664 No locals. #4 0xb33262f2 in mod_event_socket_runtime () at mod_event_socket.c:2134 pool = (switch_memory_pool_t *) 0x81bbaa0 listener_pool = (switch_memory_pool_t *) 0x81bdaa8 rv = sa = (switch_sockaddr_t *) 0x81bbaf8 inbound_socket = (switch_socket_t *) 0x81bdb00 listener = x = __func__ = "mod_event_socket_runtime" #5 0xb7dd4fb3 in switch_loadable_module_exec (thread=0x80bfb40, obj=0x80bf930) at src/switch_loadable_module.c:93 status = module = (switch_loadable_module_t *) 0xb340ec28 __PRETTY_FUNCTION__ = "switch_loadable_module_exec" __func__ = "switch_loadable_module_exec" #6 0xb7e30bf6 in dummy_worker (opaque=0x80bfb40) at threadproc/unix/thread.c:138 No locals. #7 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 No symbol table info available. #8 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. Thread 5 (process 27230): ---Type to continue, or q to quit--- #0 0xb7f14410 in __kernel_vsyscall () No symbol table info available. #1 0xb7c72c07 in poll () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. #2 0xb33ce86e in ?? () from /opt/app/voip/ippbx.prod/mod/mod_xml_rpc.so No locals. #3 0xb33c1464 in ChanSwitchAccept (chanSwitchP=0x81f5030, channelPP=0xa9ad30e0, channelInfoPP=0xa9ad30dc, errorP=0xa9ad30e4) at ../../../../libs/xmlrpc-c/lib/abyss/src/chanswitch.c:149 No locals. #4 0xb33cd37e in ServerRun (serverP=0xb33ffe4c) at ../../../../libs/xmlrpc-c/lib/abyss/src/server.c:908 srvP = (struct _TServer * const) 0x81f4fc8 #5 0xb33be832 in mod_xml_rpc_runtime () at mod_xml_rpc.c:837 registryP = (xmlrpc_registry *) 0x81bb960 env = {fault_occurred = 0, fault_code = 0, fault_string = 0x0} logfile = "/opt/app/voip/ippbx.prod/log/freeswitch_http.log", '\0' hi = var = (const void *) 0x80609b8 val = (void *) 0x805dc78 __func__ = "mod_xml_rpc_runtime" #6 0xb7dd4fb3 in switch_loadable_module_exec (thread=0x80bfda8, obj=0x80bfb98) at src/switch_loadable_module.c:93 status = module = (switch_loadable_module_t *) 0xb3405178 __PRETTY_FUNCTION__ = "switch_loadable_module_exec" __func__ = "switch_loadable_module_exec" #7 0xb7e30bf6 in dummy_worker (opaque=0x80bfda8) at threadproc/unix/thread.c:138 No locals. #8 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 No symbol table info available. #9 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. Thread 4 (process 27231): #0 0xb7f14410 in __kernel_vsyscall () No symbol table info available. #1 0xb7d3199b in read () from /lib/tls/i686/cmov/libpthread.so.0 No symbol table info available. #2 0xb7e9dfb3 in read_char (el=0x81c5040, cp=0xa92d235b "?|???`!\b\200a\"\b?#-?;\237?@P\034\b?#-?\a") at read.c:294 num_read = 136073656 tried = 0 #3 0xb7e9da9c in el_getc (el=0x81c5040, cp=0xa92d235b "?|???`!\b\200a\"\b?#-?;\237?@P\034\b?#-?\a") at read.c:362 num_read = -1456659621 ma = (c_macro_t *) 0x81c52e0 #4 0xb7e9dbdf in el_gets (el=0x81c5040, nread=0xa92d23a8) at read.c:241 cmdnum = 232 '?' num = 136405224 ch = -1 '?' #5 0xb7db9f3b in console_thread (thread=0x82102d0, obj=0x8210248) at src/switch_console.c:441 arg = 1 count = 7 line = 0x82160e8 "" pool = (switch_memory_pool_t *) 0x8210248 ---Type to continue, or q to quit--- __func__ = "console_thread" __PRETTY_FUNCTION__ = "console_thread" #6 0xb7e30bf6 in dummy_worker (opaque=0x82102d0) at threadproc/unix/thread.c:138 No locals. #7 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 No symbol table info available. #8 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. Thread 3 (process 27235): #0 0xb7f14410 in __kernel_vsyscall () No symbol table info available. #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. #2 0xb7e32799 in apr_sleep (t=100000) at time/unix/time.c:246 tv = {tv_sec = 0, tv_usec = 20000} #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 No locals. #4 0xb7df019e in switch_ivr_originate (session=0xb34dd080, bleg=0xa8ad10c0, cause=0xa8ad10bc, bridgeto=0x823a528 "openzap/1/a/04855711", timelimit_sec=60, table=0xb7ecda20, cid_name_override=0x0, cid_num_override=0x0, caller_profile_override=0x0, ovars=0x0, flags=) at src/switch_ivr_originate.c:1793 end = var_begin = originate_status = {{peer_session = 0xb34e5400, peer_channel = 0xb34e9968, caller_profile = 0xb34e1dc8, ring_ready = 0 '\0', early_media = 1 '\001', answered = 0 '\0', per_channel_timelimit_sec = 0, per_channel_progress_timelimit_sec = 0}, {peer_session = 0x0, peer_channel = 0x0, caller_profile = 0x0, ring_ready = 0 '\0', early_media = 0 '\0', answered = 0 '\0', per_channel_timelimit_sec = 0, per_channel_progress_timelimit_sec = 0} } dftflags = 0 myflags = 0 pipe_names = {0xa811de20 "openzap", 0x0 } data = status = SWITCH_STATUS_SUCCESS caller_channel = (switch_channel_t *) 0xb34e15e8 peer_names = {0xa811de20 "openzap", 0x0 } new_session = (switch_core_session_t *) 0xb34e5400 peer_session = new_profile = (switch_caller_profile_t *) 0xb34e1dc8 caller_caller_profile = chan_type = chan_data = peer_channel = ringback = {audio_buffer = 0x0, ts = {TONES = {{freqs = {0, 0, 0, 0, 0, 0}} }, channels = 0, rate = 0, duration = 0, wait = 0, tmp_duration = 0, tmp_wait = 0, loops = 0, LOOPS = 0, decay_factor = 0, decay_direction = 0, decay_step = 0, volume = 0, debug = 0, debug_stream = 0x0, user_data = 0x0, buffer = 0x0, datalen = 0, samples = 0, dynamic = 0, handler = 0}, fhb = { file_interface = 0x0, flags = 0, fd = 0x0, samples = 0, samplerate = 0, native_rate = 0, channels = 0 '\0', format = 0, sections = 0, seekable = 0, sample_count = 0, speed = 0, memory_pool = 0x0, prebuf = 0, interval = 0, private_info = 0x0, handler = 0x0, pos = 0, audio_buffer = 0x0, sp_audio_buffer = 0x0, thresh = 0, silence_hits = 0, offset_pos = 0, last_pos = 0, vol = 0, resampler = 0x0, buffer = 0x0, dbuf = 0x0, dbuflen = 0, pre_buffer = 0x0, pre_buffer_data = 0x0, pre_buffer_datalen = 0, file = 0x0, func = 0x0, line = 0}, fh = 0x0, silence = 0, asis = 0 '\0'} start = 1234349134 read_frame = (switch_frame_t *) 0x0 ---Type to continue, or q to quit--- pool = (switch_memory_pool_t *) 0x0 r = 0 i = -1286728216 and_argc = 1 or_argc = 1 sleep_ms = 1000 try = 0 retries = 1 write_codec = {codec_interface = 0x0, implementation = 0x0, fmtp_in = 0x0, fmtp_out = 0x0, codec_settings = {quality = 0, complexity = 0, enhancement = 0, vad = 0, vbr = 0, vbr_quality = 0, abr = 0, dtx = 0, preproc = 0, pp_vad = 0, pp_agc = 0, pp_agc_level = 0, pp_denoise = 0, pp_dereverb = 0, pp_dereverb_decay = 0, pp_dereverb_level = 0}, flags = 0, memory_pool = 0x0, private_info = 0x0, agreed_pt = 0 '\0', mutex = 0x0} write_frame = {codec = 0x0, source = 0x0, packet = 0x0, packetlen = 0, data = 0x821ec80, datalen = 0, buflen = 4096, samples = 0, rate = 0, payload = 0 '\0', timestamp = 0, seq = 0, ssrc = 0, m = SWITCH_FALSE, flags = 0} pass = 0 '\0' odata = 0xa811e160 "openzap/1/a/04855711" var = reason = SWITCH_CAUSE_SUCCESS to = 0 '\0' var_val = vars = 0x0 var_block_count = 0 e = ringback_data = 0x0 read_codec = message = (switch_core_session_message_t *) 0x0 var_event = (switch_event_t *) 0xb346cea8 fail_on_single_reject = 0 '\0' fail_on_single_reject_var = 0x0 loop_data = 0xa811de20 "openzap" progress_timelimit_sec = 60 oglobals = {session = 0xb34dd080, idx = -1, hups = 0, file = '\0' , key = '\0' , early_ok = 0 '\0', ring_ready = 1 '\001', sent_ring = 1 '\001', progress = 1 '\001', return_ring_ready = 0 '\0', monitor_early_media_ring = 0 '\0', monitor_early_media_fail = 0 '\0', gen_ringback = 0 '\0', ignore_early_media = 1 '\001', ignore_ring_ready = 0 '\0'} __PRETTY_FUNCTION__ = "switch_ivr_originate" __func__ = "switch_ivr_originate" #5 0xad7231af in ?? () from /opt/app/voip/ippbx.prod/mod/mod_dptools.so No locals. #6 0xb7dc6c54 in switch_core_session_exec (session=0xb34dd080, application_interface=0xb346ba10, arg=0x823a528 "openzap/1/a/04855711") at src/switch_core_session.c:1332 log = lp = event = (switch_event_t *) 0x0 var = channel = (switch_channel_t *) 0xb34e15e8 expanded = 0x823a528 "openzap/1/a/04855711" app = 0xad7253b0 "bridge" __PRETTY_FUNCTION__ = "switch_core_session_exec" __func__ = "switch_core_session_exec" #7 0xb7dc70fe in switch_core_session_execute_application (session=0xb34dd080, app=0x823a520 "bridge", arg=0x823a528 "openzap/1/a/04855711") ---Type to continue, or q to quit--- at src/switch_core_session.c:1254 application_interface = (switch_application_interface_t *) 0xb346ba10 __func__ = "switch_core_session_execute_application" #8 0xb7dc93a4 in switch_core_session_run (session=0xb34dd080) at src/switch_core_state_machine.c:155 proceed = global_proceed = do_extra_handlers = state = endstate = endpoint_interface = driver_state_handler = (const switch_state_handler_table_t *) 0xb331f860 application_state_handler = thread_id = 3084680427 env = {{__jmpbuf = {134603552, -1287115232, -1209317480, 27232, -1287651304, 24}, __mask_was_saved = -1210923727, __saved_mask = { __val = {2819623592, 2829914760, 3084048908, 2819620880, 27232, 1, 3008239368, 2829914744, 3085760892, 250000, 3084048908, 2829914760, 3085085008, 1, 3085760892, 2829914792, 3085760892, 0, 134564192, 3084043569, 3085085008, 134564244, 3085760892, 2829914824, 27232, 134564240, 3007790064, 3084573819, 3085760892, 3008213112, 3084048908, 2829914856}}}} sig = __func__ = "switch_core_session_run" __PRETTY_FUNCTION__ = "switch_core_session_run" #9 0xb7dc6725 in switch_core_session_thread (thread=0x823a050, obj=0xb34dd080) at src/switch_core_session.c:940 session = (switch_core_session_t *) 0xb34dd080 event = event_str = 0x0 val = __func__ = "switch_core_session_thread" __PRETTY_FUNCTION__ = "switch_core_session_thread" #10 0xb7e30bf6 in dummy_worker (opaque=0x823a050) at threadproc/unix/thread.c:138 No locals. #11 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 No symbol table info available. #12 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. Thread 2 (process 27236): #0 0xb7f14410 in __kernel_vsyscall () No symbol table info available. #1 0xb7c72c07 in poll () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. #2 0xb337af00 in wanpipe_wait (zchan=0xb341f9d8, flags=0xa80fee60, to=40) at src/ozmod/ozmod_wanpipe/ozmod_wanpipe.c:255 inflags = 1 result = #3 0xae837ae7 in zap_channel_wait (zchan=0x1, flags=0xa80fee60, to=40) at src/zap_io.c:1479 __PRETTY_FUNCTION__ = "zap_channel_wait" #4 0xae849059 in channel_read_frame (session=0xb34e5400, frame=0xa80ff170, flags=0, stream_id=0) at mod_openzap.c:593 channel = (switch_channel_t *) 0xb34e9968 len = wflags = ZAP_READ dtmf = '\0' status = total_to = 80 ---Type to continue, or q to quit--- chunk = 40 do_break = 0 __PRETTY_FUNCTION__ = "channel_read_frame" __func__ = "channel_read_frame" #5 0xb7dcbe6f in switch_core_session_read_frame (session=0xb34e5400, frame=0xa80ff170, flags=0, stream_id=0) at src/switch_core_io.c:161 ptr = status = SWITCH_STATUS_SUCCESS need_codec = perfect = do_bugs = -1212118646 do_resample = 0 is_cng = 0 flag = 0 __PRETTY_FUNCTION__ = "switch_core_session_read_frame" __func__ = "switch_core_session_read_frame" #6 0xb7e054ec in switch_ivr_sleep (session=0xb34e5400, ms=10, sync=SWITCH_FALSE, args=0x0) at src/switch_ivr.c:262 channel = (switch_channel_t *) 0xb34e9968 status = SWITCH_STATUS_SUCCESS start = 1234349144119246 now = 0 done = 1234349144129246 read_frame = (switch_frame_t *) 0x0 cng_frame = {codec = 0x0, source = 0x0, packet = 0x0, packetlen = 0, data = 0xa80ff176, datalen = 2, buflen = 2, samples = 0, rate = 0, payload = 0 '\0', timestamp = 0, seq = 0, ssrc = 0, m = SWITCH_FALSE, flags = 1} data = "\000" write_frame = {codec = 0x0, source = 0x0, packet = 0x0, packetlen = 0, data = 0x0, datalen = 0, buflen = 0, samples = 0, rate = 0, payload = 0 '\0', timestamp = 0, seq = 0, ssrc = 0, m = SWITCH_FALSE, flags = 0} abuf = (unsigned char *) 0x0 imp = {codec_type = SWITCH_CODEC_TYPE_AUDIO, ianacode = 0 '\0', iananame = 0x0, fmtp = 0x0, samples_per_second = 0, actual_samples_per_second = 0, bits_per_second = 0, microseconds_per_packet = 0, samples_per_packet = 0, decoded_bytes_per_packet = 0, encoded_bytes_per_packet = 0, number_of_channels = 0 '\0', codec_frames_per_packet = 0, init = 0, encode = 0, decode = 0, destroy = 0, codec_id = 0, next = 0x0} codec = {codec_interface = 0x0, implementation = 0x0, fmtp_in = 0x0, fmtp_out = 0x0, codec_settings = {quality = 0, complexity = 0, enhancement = 0, vad = 0, vbr = 0, vbr_quality = 0, abr = 0, dtx = 0, preproc = 0, pp_vad = 0, pp_agc = 0, pp_agc_level = 0, pp_denoise = 0, pp_dereverb = 0, pp_dereverb_decay = 0, pp_dereverb_level = 0}, flags = 0, memory_pool = 0x0, private_info = 0x0, agreed_pt = 0 '\0', mutex = 0x0} sval = 0 var = __func__ = "switch_ivr_sleep" __PRETTY_FUNCTION__ = "switch_ivr_sleep" #7 0xb7deaf94 in originate_on_consume_media_transmit (session=0xb34e5400) at src/switch_ivr_originate.c:47 channel = (switch_channel_t *) 0xb34e9968 #8 0xb7dc8b74 in switch_core_session_run (session=0xb34e5400) at src/switch_core_state_machine.c:476 proceed = 1 global_proceed = 0 do_extra_handlers = state = endstate = endpoint_interface = driver_state_handler = (const switch_state_handler_table_t *) 0xae8523c0 application_state_handler = ---Type to continue, or q to quit--- thread_id = 3084680427 env = {{__jmpbuf = {0, -1287122688, -1475349896, 27233, -1211162264, -1210902852}, __mask_was_saved = -1210923727, __saved_mask = { __val = {0, 3084099572, 3084048908, 3085631456, 27233, 1, 3008273032, 2819617400, 3085760892, 250000, 3084048908, 2819617416, 3085085008, 1, 3085760892, 2819617448, 3085760892, 0, 134564192, 3084043569, 3085085008, 134564244, 3085760892, 2819617480, 27233, 134564240, 135380168, 3084573819, 3085760892, 3008246776, 3084048908, 2819617512}}}} sig = __func__ = "switch_core_session_run" __PRETTY_FUNCTION__ = "switch_core_session_run" #9 0xb7dc6725 in switch_core_session_thread (thread=0xb34f1060, obj=0xb34e5400) at src/switch_core_session.c:940 session = (switch_core_session_t *) 0xb34e5400 event = event_str = 0x0 val = __func__ = "switch_core_session_thread" __PRETTY_FUNCTION__ = "switch_core_session_thread" #10 0xb7e30bf6 in dummy_worker (opaque=0xb34f1060) at threadproc/unix/thread.c:138 No locals. #11 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 No symbol table info available. #12 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. Thread 1 (process 27234): #0 0xabe0e408 in mpg123_delete at plt () from /opt/app/voip/ippbx.prod/mod/mod_shout.so No locals. #1 0xabe13325 in ?? () from /opt/app/voip/ippbx.prod/mod/mod_shout.so No locals. #2 0xb7dc04b9 in switch_core_file_seek (fh=0xa76fef28, cur_pos=0xa76fef88, samples=0, whence=1) at src/switch_core_file.c:345 status = 4294967295 __PRETTY_FUNCTION__ = "switch_core_file_seek" #3 0xb7dfab82 in switch_ivr_play_file (session=0xa7700030, fh=0xa76fef28, file=0xa7711ed8 "/opt/app/voip/ippbx.prod/sounds/de/callie/qet.mp3", args=0xa76ff0d8) at src/switch_ivr_play_say.c:1262 channel = (switch_channel_t *) 0xa7704598 dtmf = {digit = 0 '\0', duration = 0} interval = 2909916864 samples = 320 framelen = 640 sample_start = olen = 320 llen = 320 write_frame = {codec = 0xa76fefb8, source = 0x0, packet = 0x0, packetlen = 0, data = 0xa7743f30, datalen = 640, buflen = 32768, samples = 320, rate = 16000, payload = 0 '\0', timestamp = 0, seq = 0, ssrc = 0, m = SWITCH_FALSE, flags = 0} timer = {interval = 0, flags = 0, samples = 0, samplecount = 0, timer_interface = 0x0, memory_pool = 0x0, private_info = 0x0, diff = 0, tick = 0} codec = {codec_interface = 0x80cd8f8, implementation = 0x80cdf20, fmtp_in = 0x0, fmtp_out = 0x0, codec_settings = {quality = 0, complexity = 0, enhancement = 0, vad = 0, vbr = 0, vbr_quality = 0, abr = 0, dtx = 0, preproc = 0, pp_vad = 0, pp_agc = 0, pp_agc_level = 0, pp_denoise = 0, pp_dereverb = 0, pp_dereverb_decay = 0, pp_dereverb_level = 0}, flags = 3, memory_pool = 0xb34fbb38, private_info = 0x0, agreed_pt = 0 '\0', mutex = 0xa7711f10} pool = (switch_memory_pool_t *) 0xb34fbb38 status = SWITCH_STATUS_SUCCESS lfh = {file_interface = 0x8136258, flags = 3085, fd = 0x0, samples = 0, samplerate = 16000, native_rate = 16000, channels = 1 '\001', ---Type to continue, or q to quit--- format = 0, sections = 0, seekable = 0, sample_count = 729088, speed = 0, memory_pool = 0xa7712040, prebuf = 0, interval = 0, private_info = 0xa7714048, handler = 0x0, pos = 0, audio_buffer = 0xb3476f88, sp_audio_buffer = 0x0, thresh = 0, silence_hits = 0, offset_pos = 364640, last_pos = 0, vol = 0, resampler = 0x0, buffer = 0x0, dbuf = 0x0, dbuflen = 0, pre_buffer = 0x0, pre_buffer_data = 0x0, pre_buffer_datalen = 0, file = 0xb7eb71f0 "src/switch_ivr_play_say.c", func = 0xb7eb794b "switch_ivr_play_file", line = 894} read_codec = (switch_codec_t *) 0xb34fcb90 p = 0xb7eaf8d4 "current_application" asis = 0 '\0' prefix = timer_name = 0x0 prebuf = eof = 1 bread = __func__ = "switch_ivr_play_file" __PRETTY_FUNCTION__ = "switch_ivr_play_file" #4 0xad71da25 in ?? () from /opt/app/voip/ippbx.prod/mod/mod_dptools.so No locals. #5 0xb7dc6c54 in switch_core_session_exec (session=0xa7700030, application_interface=0xb346b388, arg=0xb34fd790 "qet.mp3") at src/switch_core_session.c:1332 log = lp = event = (switch_event_t *) 0x0 var = channel = (switch_channel_t *) 0xa7704598 expanded = 0xb34fd790 "qet.mp3" app = 0xad725133 "playback" __PRETTY_FUNCTION__ = "switch_core_session_exec" __func__ = "switch_core_session_exec" #6 0xb7dc70fe in switch_core_session_execute_application (session=0xa7700030, app=0xb34fd780 "playback", arg=0xb34fd790 "qet.mp3") at src/switch_core_session.c:1254 application_interface = (switch_application_interface_t *) 0xb346b388 __func__ = "switch_core_session_execute_application" #7 0xb7dc93a4 in switch_core_session_run (session=0xa7700030) at src/switch_core_state_machine.c:155 proceed = global_proceed = do_extra_handlers = state = endstate = endpoint_interface = driver_state_handler = (const switch_state_handler_table_t *) 0xb331f860 application_state_handler = thread_id = 3084680427 env = {{__jmpbuf = {0, 0, 0, 0, 0, 0}, __mask_was_saved = 0, __saved_mask = {__val = {0 }}}} sig = __func__ = "switch_core_session_run" __PRETTY_FUNCTION__ = "switch_core_session_run" #8 0xb7dc6725 in switch_core_session_thread (thread=0xb34fd3d0, obj=0xa7700030) at src/switch_core_session.c:940 session = (switch_core_session_t *) 0xa7700030 event = event_str = 0x0 val = ---Type to continue, or q to quit--- __func__ = "switch_core_session_thread" __PRETTY_FUNCTION__ = "switch_core_session_thread" #9 0xb7e30bf6 in dummy_worker (opaque=0xb34fd3d0) at threadproc/unix/thread.c:138 No locals. #10 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 No symbol table info available. #11 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. regards Helmut From brian at freeswitch.org Wed Feb 11 04:16:42 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 11 Feb 2009 06:16:42 -0600 Subject: [Freeswitch-users] segfault when shoutcast plays mp3 and extension hangs up In-Reply-To: <4992BFFC.50006@ewetel.de> References: <4992BFFC.50006@ewetel.de> Message-ID: The proper location to post this is on Jira... please in the future report ALL issues via Jira. They'll get lost if not done this way. Thanks, Brian On Feb 11, 2009, at 6:09 AM, Helmut Kuper wrote: > Hello, > > today I tried to play a mp3. It works fine until extension hangs up. > Then FS (FreeSWITCH Version 1.0.trunk (11698M)) crashed with segfault. > The mp3 file was generated by MP3Splitter > (http://www.codevisions.de/hp/upload/_files/mp3splitter20.zip) as a > piece out of a complete mp3 song. There is a good chance that it > generates corrupt mp3s. At least those mp3s are playable in winamp and > media player. > > My dialplan: > > expression="^9123$"> > data="absolute_codec_string=G722"/> > > > > > > > > FS console output shows problems in mp3 file: > > freeswitch at ippbx-prod-node0> 2009-02-11 11:45:43 [DEBUG] Span:1 Q. > 931() > Timer 0 of call 0 (CRV: 2, State: 0) timed out > Note: Illegal Audio-MPEG-Header 0x00000000 at offset 0x10ec15. > Note: Trying to resync... > Note: Hit end of (available) data during resync. > 2009-02-11 11:45:44 [DEBUG] switch_ivr_play_say.c:1261 > switch_ivr_play_file() done playing file > ./start_fs.sh: line 6: 27201 Segmentation fault (core dumped) > bin/freeswitch $1 > > > Here are the backtraces: > > (gdb) bt > #0 0xabe0e408 in mpg123_delete at plt () from > /opt/app/voip/ippbx.prod/mod/mod_shout.so > #1 0xabe13325 in ?? () from /opt/app/voip/ippbx.prod/mod/mod_shout.so > #2 0xb7dc04b9 in switch_core_file_seek (fh=0xa76fef28, > cur_pos=0xa76fef88, samples=0, whence=1) at src/switch_core_file.c:345 > #3 0xb7dfab82 in switch_ivr_play_file (session=0xa7700030, > fh=0xa76fef28, > file=0xa7711ed8 "/opt/app/voip/ippbx.prod/sounds/de/callie/ > qet.mp3", > args=0xa76ff0d8) at src/switch_ivr_play_say.c:1262 > #4 0xad71da25 in ?? () from /opt/app/voip/ippbx.prod/mod/ > mod_dptools.so > #5 0xb7dc6c54 in switch_core_session_exec (session=0xa7700030, > application_interface=0xb346b388, arg=0xb34fd790 "qet.mp3") > at src/switch_core_session.c:1332 > #6 0xb7dc70fe in switch_core_session_execute_application > (session=0xa7700030, app=0xb34fd780 "playback", arg=0xb34fd790 > "qet.mp3") > at src/switch_core_session.c:1254 > #7 0xb7dc93a4 in switch_core_session_run (session=0xa7700030) at > src/switch_core_state_machine.c:155 > #8 0xb7dc6725 in switch_core_session_thread (thread=0xb34fd3d0, > obj=0xa7700030) at src/switch_core_session.c:940 > #9 0xb7e30bf6 in dummy_worker (opaque=0xb34fd3d0) at > threadproc/unix/thread.c:138 > #10 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > #11 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > Current language: auto; currently asm > > > (gdb) bt full > #0 0xabe0e408 in mpg123_delete at plt () from > /opt/app/voip/ippbx.prod/mod/mod_shout.so > No locals. > #1 0xabe13325 in ?? () from /opt/app/voip/ippbx.prod/mod/mod_shout.so > No locals. > #2 0xb7dc04b9 in switch_core_file_seek (fh=0xa76fef28, > cur_pos=0xa76fef88, samples=0, whence=1) at src/switch_core_file.c:345 > status = 4294967295 > __PRETTY_FUNCTION__ = "switch_core_file_seek" > #3 0xb7dfab82 in switch_ivr_play_file (session=0xa7700030, > fh=0xa76fef28, > file=0xa7711ed8 "/opt/app/voip/ippbx.prod/sounds/de/callie/ > qet.mp3", > args=0xa76ff0d8) at src/switch_ivr_play_say.c:1262 > channel = (switch_channel_t *) 0xa7704598 > dtmf = {digit = 0 '\0', duration = 0} > interval = 2909916864 > samples = 320 > framelen = 640 > sample_start = > olen = 320 > llen = 320 > write_frame = {codec = 0xa76fefb8, source = 0x0, packet = 0x0, > packetlen = 0, data = 0xa7743f30, datalen = 640, buflen = 32768, > samples = 320, rate = 16000, payload = 0 '\0', timestamp = 0, seq = > 0, > ssrc = 0, m = SWITCH_FALSE, flags = 0} > timer = {interval = 0, flags = 0, samples = 0, samplecount = 0, > timer_interface = 0x0, memory_pool = 0x0, private_info = 0x0, > diff = 0, tick = 0} > codec = {codec_interface = 0x80cd8f8, implementation = > 0x80cdf20, fmtp_in = 0x0, fmtp_out = 0x0, codec_settings = {quality > = 0, > complexity = 0, enhancement = 0, vad = 0, vbr = 0, vbr_quality = 0, > abr = 0, dtx = 0, preproc = 0, pp_vad = 0, pp_agc = 0, > pp_agc_level = 0, pp_denoise = 0, pp_dereverb = 0, > pp_dereverb_decay > = 0, pp_dereverb_level = 0}, flags = 3, memory_pool = 0xb34fbb38, > private_info = 0x0, agreed_pt = 0 '\0', mutex = 0xa7711f10} > pool = (switch_memory_pool_t *) 0xb34fbb38 > status = SWITCH_STATUS_SUCCESS > lfh = {file_interface = 0x8136258, flags = 3085, fd = 0x0, > samples = 0, samplerate = 16000, native_rate = 16000, channels = 1 > '\001', > format = 0, sections = 0, seekable = 0, sample_count = 729088, > speed = > 0, memory_pool = 0xa7712040, prebuf = 0, interval = 0, > private_info = 0xa7714048, handler = 0x0, pos = 0, audio_buffer = > 0xb3476f88, sp_audio_buffer = 0x0, thresh = 0, silence_hits = 0, > offset_pos = 364640, last_pos = 0, vol = 0, resampler = 0x0, buffer = > 0x0, dbuf = 0x0, dbuflen = 0, pre_buffer = 0x0, > pre_buffer_data = 0x0, pre_buffer_datalen = 0, file = 0xb7eb71f0 > "src/switch_ivr_play_say.c", func = 0xb7eb794b "switch_ivr_play_file", > line = 894} > read_codec = (switch_codec_t *) 0xb34fcb90 > p = 0xb7eaf8d4 "current_application" > asis = 0 '\0' > prefix = > timer_name = 0x0 > prebuf = > eof = 1 > bread = > __func__ = "switch_ivr_play_file" > __PRETTY_FUNCTION__ = "switch_ivr_play_file" > #4 0xad71da25 in ?? () from /opt/app/voip/ippbx.prod/mod/ > mod_dptools.so > No locals. > #5 0xb7dc6c54 in switch_core_session_exec (session=0xa7700030, > application_interface=0xb346b388, arg=0xb34fd790 "qet.mp3") > at src/switch_core_session.c:1332 > log = > lp = > event = (switch_event_t *) 0x0 > var = > channel = (switch_channel_t *) 0xa7704598 > expanded = 0xb34fd790 "qet.mp3" > app = 0xad725133 "playback" > __PRETTY_FUNCTION__ = "switch_core_session_exec" > __func__ = "switch_core_session_exec" > #6 0xb7dc70fe in switch_core_session_execute_application > (session=0xa7700030, app=0xb34fd780 "playback", arg=0xb34fd790 > "qet.mp3") > at src/switch_core_session.c:1254 > application_interface = (switch_application_interface_t *) > 0xb346b388 > __func__ = "switch_core_session_execute_application" > #7 0xb7dc93a4 in switch_core_session_run (session=0xa7700030) at > src/switch_core_state_machine.c:155 > proceed = > global_proceed = > do_extra_handlers = > state = > endstate = > endpoint_interface = > driver_state_handler = (const switch_state_handler_table_t *) > 0xb331f860 > application_state_handler = > thread_id = 3084680427 > env = {{__jmpbuf = {0, 0, 0, 0, 0, 0}, __mask_was_saved = 0, > __saved_mask = {__val = {0 }}}} > sig = > __func__ = "switch_core_session_run" > __PRETTY_FUNCTION__ = "switch_core_session_run" > #8 0xb7dc6725 in switch_core_session_thread (thread=0xb34fd3d0, > obj=0xa7700030) at src/switch_core_session.c:940 > session = (switch_core_session_t *) 0xa7700030 > event = > event_str = 0x0 > val = > __func__ = "switch_core_session_thread" > __PRETTY_FUNCTION__ = "switch_core_session_thread" > #9 0xb7e30bf6 in dummy_worker (opaque=0xb34fd3d0) at > threadproc/unix/thread.c:138 > No locals. > #10 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > No symbol table info available. > #11 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > > > > > (gdb) thread apply all bt > > Thread 31 (process 27201): > #0 0xb7f14410 in __kernel_vsyscall () > #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 > #2 0xb7e32799 in apr_sleep (t=1000000) at time/unix/time.c:246 > #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 > #4 0xb7db9154 in switch_console_loop () at src/switch_console.c:792 > #5 0xb7dcedf0 in switch_core_runtime_loop (bg=0) at src/ > switch_core.c:659 > #6 0x0804a36a in main (argc=1, argv=0xbffe65c4) at src/switch.c:666 > > Thread 30 (process 27202): > #0 0xb7f14410 in __kernel_vsyscall () > #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 > #2 0xb7e32799 in apr_sleep (t=100000) at time/unix/time.c:246 > #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 > #4 0xb7dbe2a4 in pool_thread (thread=0xb7a07da8, obj=0x0) at > src/switch_core_memory.c:421 > #5 0xb7e30bf6 in dummy_worker (opaque=0xb7a07da8) at > threadproc/unix/thread.c:138 > #6 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > #7 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > > Thread 29 (process 27203): > #0 0xb7f14410 in __kernel_vsyscall () > #1 0xb7d2eaa5 in pthread_cond_wait@@GLIBC_2.3.2 () from > /lib/tls/i686/cmov/libpthread.so.0 > #2 0xb7e2acaa in apr_thread_cond_wait (cond=0xb6fb53b8, > mutex=0xb6fb5388) at locks/unix/thread_cond.c:68 > #3 0xb7e21a3c in apr_queue_pop (queue=0xb6fb5358, data=0xb6ec03a8) at > misc/apr_queue.c:276 > #4 0xb7dadd44 in switch_queue_pop (queue=0xb6fb5358, data=0xb6ec03a8) > at src/switch_apr.c:879 > #5 0xb7ddf879 in switch_event_dispatch_thread (thread=0x8068140, > obj=0xb6fb5358) at src/switch_event.c:230 > #6 0xb7e30bf6 in dummy_worker (opaque=0x8068140) at > threadproc/unix/thread.c:138 > #7 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > #8 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > > Thread 28 (process 27204): > #0 0xb7f14410 in __kernel_vsyscall () > #1 0xb7d2eaa5 in pthread_cond_wait@@GLIBC_2.3.2 () from > /lib/tls/i686/cmov/libpthread.so.0 > #2 0xb7e2acaa in apr_thread_cond_wait (cond=0x805e380, > mutex=0x805e350) > at locks/unix/thread_cond.c:68 > #3 0xb7e21a3c in apr_queue_pop (queue=0x805e320, data=0xb66bf3a8) at > misc/apr_queue.c:276 > #4 0xb7dadd44 in switch_queue_pop (queue=0x805e320, > data=0xb66bf3a8) at > src/switch_apr.c:879 > #5 0xb7ddec2d in switch_event_thread (thread=0x8068160, > obj=0x805e320) > at src/switch_event.c:273 > #6 0xb7e30bf6 in dummy_worker (opaque=0x8068160) at > threadproc/unix/thread.c:138 > #7 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > #8 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > > Thread 27 (process 27205): > #0 0xb7f14410 in __kernel_vsyscall () > #1 0xb7d2eaa5 in pthread_cond_wait@@GLIBC_2.3.2 () from > /lib/tls/i686/cmov/libpthread.so.0 > #2 0xb7e2acaa in apr_thread_cond_wait (cond=0xb71d4b38, > mutex=0xb71d4b08) at locks/unix/thread_cond.c:68 > #3 0xb7e21a3c in apr_queue_pop (queue=0xb71d4ad8, data=0xb5ebe3a8) at > misc/apr_queue.c:276 > #4 0xb7dadd44 in switch_queue_pop (queue=0xb71d4ad8, data=0xb5ebe3a8) > at src/switch_apr.c:879 > #5 0xb7ddec2d in switch_event_thread (thread=0x8068180, > obj=0xb71d4ad8) > at src/switch_event.c:273 > #6 0xb7e30bf6 in dummy_worker (opaque=0x8068180) at > threadproc/unix/thread.c:138 > #7 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > ---Type to continue, or q to quit--- > #8 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > > Thread 26 (process 27206): > #0 0xb7f14410 in __kernel_vsyscall () > #1 0xb7d2eaa5 in pthread_cond_wait@@GLIBC_2.3.2 () from > /lib/tls/i686/cmov/libpthread.so.0 > #2 0xb7e2acaa in apr_thread_cond_wait (cond=0xb7171b38, > mutex=0xb7171b08) at locks/unix/thread_cond.c:68 > #3 0xb7e21a3c in apr_queue_pop (queue=0xb7171ad8, data=0xb56bd3a8) at > misc/apr_queue.c:276 > #4 0xb7dadd44 in switch_queue_pop (queue=0xb7171ad8, data=0xb56bd3a8) > at src/switch_apr.c:879 > #5 0xb7ddec2d in switch_event_thread (thread=0x80681a0, > obj=0xb7171ad8) > at src/switch_event.c:273 > #6 0xb7e30bf6 in dummy_worker (opaque=0x80681a0) at > threadproc/unix/thread.c:138 > #7 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > #8 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > > Thread 25 (process 27207): > #0 0xb7f14410 in __kernel_vsyscall () > #1 0xb7d2eaa5 in pthread_cond_wait@@GLIBC_2.3.2 () from > /lib/tls/i686/cmov/libpthread.so.0 > #2 0xb7e2acaa in apr_thread_cond_wait (cond=0xb6fb54a8, > mutex=0xb6fb5478) at locks/unix/thread_cond.c:68 > #3 0xb7e21a3c in apr_queue_pop (queue=0xb6fb5448, data=0xb4dce3a8) at > misc/apr_queue.c:276 > #4 0xb7dadd44 in switch_queue_pop (queue=0xb6fb5448, data=0xb4dce3a8) > at src/switch_apr.c:879 > #5 0xb7e082fd in log_thread (thread=0xb4e30ae0, obj=0x0) at > src/switch_log.c:209 > #6 0xb7e30bf6 in dummy_worker (opaque=0xb4e30ae0) at > threadproc/unix/thread.c:138 > #7 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > #8 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > > Thread 24 (process 27210): > #0 0xb7f14410 in __kernel_vsyscall () > #1 0xb7d2eaa5 in pthread_cond_wait@@GLIBC_2.3.2 () from > /lib/tls/i686/cmov/libpthread.so.0 > #2 0xb7e2acaa in apr_thread_cond_wait (cond=0x80c0c58, > mutex=0x80c0c28) > at locks/unix/thread_cond.c:68 > #3 0xb7daed54 in switch_thread_cond_wait (cond=0x80c0c58, > mutex=0x80c0c28) at src/switch_apr.c:359 > #4 0xb7e11266 in switch_cond_next () at src/switch_time.c:203 > #5 0xb7dc27a5 in switch_core_sql_thread (thread=0xb3567ae8, > obj=0x0) at > src/switch_core_sqldb.c:220 > #6 0xb7e30bf6 in dummy_worker (opaque=0xb3567ae8) at > threadproc/unix/thread.c:138 > #7 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > #8 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > > Thread 23 (process 27211): > #0 0xb7f14410 in __kernel_vsyscall () > #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 > #2 0xb7e32799 in apr_sleep (t=500000) at time/unix/time.c:246 > #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 > #4 0xb7dd2af4 in switch_scheduler_task_thread (thread=0x80baa90, > obj=0x0) at src/switch_scheduler.c:171 > #5 0xb7e30bf6 in dummy_worker (opaque=0x80baa90) at > threadproc/unix/thread.c:138 > #6 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > #7 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > > Thread 22 (process 27212): > #0 0xb7f14410 in __kernel_vsyscall () > #1 0xb7c7d676 in epoll_wait () from /lib/tls/i686/cmov/libc.so.6 > #2 0xb32b7126 in su_epoll_port_wait_events (self=0x80c6340, > tout=1000) > at su_epoll_port.c:491 > #3 0xb32bd9b8 in su_base_port_step (self=0x80c6340, tout=1000) at > su_base_port.c:442 > #4 0xb32b8551 in su_port_step (self=0x80c6340, tout=1000) at > su_port.h:326 > ---Type to continue, or q to quit--- > #5 0xb32b8521 in su_root_step (self=0x80c68f0, tout=1000) at > su_root.c:730 > #6 0xb31e8f29 in sofia_profile_thread_run (thread=0x80d2aa8, > obj=0x80d1e10) at sofia.c:831 > #7 0xb7e30bf6 in dummy_worker (opaque=0x80d2aa8) at > threadproc/unix/thread.c:138 > #8 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > #9 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > > Thread 21 (process 27213): > #0 0xb7f14410 in __kernel_vsyscall () > #1 0xb7c7d676 in epoll_wait () from /lib/tls/i686/cmov/libc.so.6 > #2 0xb32b7126 in su_epoll_port_wait_events (self=0x80dac30, > tout=1000) > at su_epoll_port.c:491 > #3 0xb32bd89b in su_base_port_run (self=0x80dac30) at > su_base_port.c:342 > #4 0xb32b842b in su_port_run (self=0x80dac30) at su_port.h:312 > #5 0xb32b8408 in su_root_run (self=0x80dacb0) at su_root.c:691 > #6 0xb32a8b31 in su_pthread_port_clone_main (varg=0xb311c0a8) at > su_pthread_port.c:321 > #7 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > #8 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > > Thread 20 (process 27214): > #0 0xb7f14410 in __kernel_vsyscall () > #1 0xb7c7d676 in epoll_wait () from /lib/tls/i686/cmov/libc.so.6 > #2 0xb32b7126 in su_epoll_port_wait_events (self=0x80da208, > tout=1000) > at su_epoll_port.c:491 > #3 0xb32bd9b8 in su_base_port_step (self=0x80da208, tout=1000) at > su_base_port.c:442 > #4 0xb32b8551 in su_port_step (self=0x80da208, tout=1000) at > su_port.h:326 > #5 0xb32b8521 in su_root_step (self=0x80d7558, tout=1000) at > su_root.c:730 > #6 0xb31e8f29 in sofia_profile_thread_run (thread=0x80dde88, > obj=0x80dd650) at sofia.c:831 > #7 0xb7e30bf6 in dummy_worker (opaque=0x80dde88) at > threadproc/unix/thread.c:138 > #8 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > #9 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > > Thread 19 (process 27215): > #0 0xb7f14410 in __kernel_vsyscall () > #1 0xb7c7d676 in epoll_wait () from /lib/tls/i686/cmov/libc.so.6 > #2 0xb32b7126 in su_epoll_port_wait_events (self=0x80e5228, > tout=1000) > at su_epoll_port.c:491 > #3 0xb32bd89b in su_base_port_run (self=0x80e5228) at > su_base_port.c:342 > #4 0xb32b842b in su_port_run (self=0x80e5228) at su_port.h:312 > #5 0xb32b8408 in su_root_run (self=0x80e3b40) at su_root.c:691 > #6 0xb32a8b31 in su_pthread_port_clone_main (varg=0xb211a0a8) at > su_pthread_port.c:321 > #7 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > #8 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > > Thread 18 (process 27216): > #0 0xb7f14410 in __kernel_vsyscall () > #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 > #2 0xb7e32799 in apr_sleep (t=10000) at time/unix/time.c:246 > #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 > #4 0xb31d4b37 in sofia_profile_worker_thread_run (thread=0x80d2b88, > obj=0x80d1e10) at sofia.c:656 > #5 0xb7e30bf6 in dummy_worker (opaque=0x80d2b88) at > threadproc/unix/thread.c:138 > #6 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > #7 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > > Thread 17 (process 27217): > ---Type to continue, or q to quit--- > #0 0xb7f14410 in __kernel_vsyscall () > #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 > #2 0xb7e32799 in apr_sleep (t=10000) at time/unix/time.c:246 > #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 > #4 0xb31d4b37 in sofia_profile_worker_thread_run (thread=0x80ddf68, > obj=0x80dd650) at sofia.c:656 > #5 0xb7e30bf6 in dummy_worker (opaque=0x80ddf68) at > threadproc/unix/thread.c:138 > #6 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > #7 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > > Thread 16 (process 27218): > #0 0xb7f14410 in __kernel_vsyscall () > #1 0xb7c7d676 in epoll_wait () from /lib/tls/i686/cmov/libc.so.6 > #2 0xb32b7126 in su_epoll_port_wait_events (self=0x80e66d8, > tout=1000) > at su_epoll_port.c:491 > #3 0xb32bd9b8 in su_base_port_step (self=0x80e66d8, tout=1000) at > su_base_port.c:442 > #4 0xb32b8551 in su_port_step (self=0x80e66d8, tout=1000) at > su_port.h:326 > #5 0xb32b8521 in su_root_step (self=0x80e49f0, tout=1000) at > su_root.c:730 > #6 0xb31e8f29 in sofia_profile_thread_run (thread=0x80e8710, > obj=0x80e7e90) at sofia.c:831 > #7 0xb7e30bf6 in dummy_worker (opaque=0x80e8710) at > threadproc/unix/thread.c:138 > #8 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > #9 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > > Thread 15 (process 27219): > #0 0xb7f14410 in __kernel_vsyscall () > #1 0xb7c7d676 in epoll_wait () from /lib/tls/i686/cmov/libc.so.6 > #2 0xb32b7126 in su_epoll_port_wait_events (self=0x80ee648, > tout=1000) > at su_epoll_port.c:491 > #3 0xb32bd89b in su_base_port_run (self=0x80ee648) at > su_base_port.c:342 > #4 0xb32b842b in su_port_run (self=0x80ee648) at su_port.h:312 > #5 0xb32b8408 in su_root_run (self=0x80f0bd0) at su_root.c:691 > #6 0xb32a8b31 in su_pthread_port_clone_main (varg=0xb00b20a8) at > su_pthread_port.c:321 > #7 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > #8 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > > Thread 14 (process 27220): > #0 0xb7f14410 in __kernel_vsyscall () > #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 > #2 0xb7e32799 in apr_sleep (t=10000) at time/unix/time.c:246 > #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 > #4 0xb31d4b37 in sofia_profile_worker_thread_run (thread=0x80e87f0, > obj=0x80e7e90) at sofia.c:656 > #5 0xb7e30bf6 in dummy_worker (opaque=0x80e87f0) at > threadproc/unix/thread.c:138 > #6 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > #7 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > > Thread 13 (process 27221): > #0 0xb7f14410 in __kernel_vsyscall () > #1 0xb7c72c07 in poll () from /lib/tls/i686/cmov/libc.so.6 > #2 0xb337af00 in wanpipe_wait (zchan=0xb3446128, flags=0xae751f80, > to=100) at src/ozmod/ozmod_wanpipe/ozmod_wanpipe.c:255 > #3 0xae837ae7 in zap_channel_wait (zchan=0x1, flags=0xae751f80, > to=100) > at src/zap_io.c:1479 > #4 0xae80aee8 in zap_isdn_run (me=0xb3416528, obj=0xb341dbc8) at > src/ozmod/ozmod_isdn/ozmod_isdn.c:1725 > #5 0xae8421ba in thread_launch (args=0xb3416528) at > src/zap_threadmutex.c:74 > #6 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > #7 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > ---Type to continue, or q to quit--- > > Thread 12 (process 27222): > #0 0xb7f14410 in __kernel_vsyscall () > #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 > #2 0xb7e32799 in apr_sleep (t=100000) at time/unix/time.c:246 > #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 > #4 0xb31f929e in sofia_presence_event_thread_run (thread=0x80cf958, > obj=0x0) at sofia_presence.c:664 > #5 0xb7e30bf6 in dummy_worker (opaque=0x80cf958) at > threadproc/unix/thread.c:138 > #6 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > #7 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > > Thread 11 (process 27223): > #0 0xb7f14410 in __kernel_vsyscall () > #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 > #2 0xb7e32799 in apr_sleep (t=1000000) at time/unix/time.c:246 > #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 > #4 0xad706ea6 in ?? () from /opt/app/voip/ippbx.prod/mod/mod_fifo.so > #5 0xb7e30bf6 in dummy_worker (opaque=0xad386b90) at > threadproc/unix/thread.c:138 > #6 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > #7 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > > Thread 10 (process 27225): > #0 0xb7f14410 in __kernel_vsyscall () > #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 > #2 0xb7e32799 in apr_sleep (t=1000000) at time/unix/time.c:246 > #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 > #4 0xabdf94ab in ?? () from > /opt/app/voip/ippbx.prod/mod/mod_local_stream.so > #5 0xb7e30bf6 in dummy_worker (opaque=0x81386e0) at > threadproc/unix/thread.c:138 > #6 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > #7 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > > Thread 9 (process 27226): > #0 0xb7f14410 in __kernel_vsyscall () > #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 > #2 0xb7e32799 in apr_sleep (t=1000000) at time/unix/time.c:246 > #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 > #4 0xabdf94ab in ?? () from > /opt/app/voip/ippbx.prod/mod/mod_local_stream.so > #5 0xb7e30bf6 in dummy_worker (opaque=0x815e728) at > threadproc/unix/thread.c:138 > #6 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > #7 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > > Thread 8 (process 27227): > #0 0xb7f14410 in __kernel_vsyscall () > #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 > #2 0xb7e32799 in apr_sleep (t=1000000) at time/unix/time.c:246 > #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 > #4 0xabdf94ab in ?? () from > /opt/app/voip/ippbx.prod/mod/mod_local_stream.so > #5 0xb7e30bf6 in dummy_worker (opaque=0x8184770) at > threadproc/unix/thread.c:138 > #6 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > #7 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > > ---Type to continue, or q to quit--- > Thread 7 (process 27228): > #0 0xb7f14410 in __kernel_vsyscall () > #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 > #2 0xb7e32799 in apr_sleep (t=1000) at time/unix/time.c:246 > #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 > #4 0xb7e12a59 in softtimer_runtime () at src/switch_time.c:459 > #5 0xb7dd4fb3 in switch_loadable_module_exec (thread=0x80bf8d8, > obj=0x80bf6c8) at src/switch_loadable_module.c:93 > #6 0xb7e30bf6 in dummy_worker (opaque=0x80bf8d8) at > threadproc/unix/thread.c:138 > #7 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > #8 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > > Thread 6 (process 27229): > #0 0xb7f14410 in __kernel_vsyscall () > #1 0xb7d31bb8 in accept () from /lib/tls/i686/cmov/libpthread.so.0 > #2 0xb7e2f90d in apr_socket_accept (new=0xaa2d434c, sock=0x81bbbb0, > connection_context=0x81bdaa8) at network_io/unix/sockets.c:187 > #3 0xb7dae3fb in switch_socket_accept (new_sock=0xaa2d434c, > sock=0x81bbbb0, pool=0x81bdaa8) at src/switch_apr.c:664 > #4 0xb33262f2 in mod_event_socket_runtime () at mod_event_socket.c: > 2134 > #5 0xb7dd4fb3 in switch_loadable_module_exec (thread=0x80bfb40, > obj=0x80bf930) at src/switch_loadable_module.c:93 > #6 0xb7e30bf6 in dummy_worker (opaque=0x80bfb40) at > threadproc/unix/thread.c:138 > #7 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > #8 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > > Thread 5 (process 27230): > #0 0xb7f14410 in __kernel_vsyscall () > #1 0xb7c72c07 in poll () from /lib/tls/i686/cmov/libc.so.6 > #2 0xb33ce86e in ?? () from /opt/app/voip/ippbx.prod/mod/ > mod_xml_rpc.so > #3 0xb33c1464 in ChanSwitchAccept (chanSwitchP=0x81f5030, > channelPP=0xa9ad30e0, channelInfoPP=0xa9ad30dc, errorP=0xa9ad30e4) > at ../../../../libs/xmlrpc-c/lib/abyss/src/chanswitch.c:149 > #4 0xb33cd37e in ServerRun (serverP=0xb33ffe4c) at > ../../../../libs/xmlrpc-c/lib/abyss/src/server.c:908 > #5 0xb33be832 in mod_xml_rpc_runtime () at mod_xml_rpc.c:837 > #6 0xb7dd4fb3 in switch_loadable_module_exec (thread=0x80bfda8, > obj=0x80bfb98) at src/switch_loadable_module.c:93 > #7 0xb7e30bf6 in dummy_worker (opaque=0x80bfda8) at > threadproc/unix/thread.c:138 > #8 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > #9 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > > Thread 4 (process 27231): > #0 0xb7f14410 in __kernel_vsyscall () > #1 0xb7d3199b in read () from /lib/tls/i686/cmov/libpthread.so.0 > #2 0xb7e9dfb3 in read_char (el=0x81c5040, cp=0xa92d235b > "?|???`!\b\200a\"\b?#-?;\237?@P\034\b?#-?\a") at > read.c:294 > #3 0xb7e9da9c in el_getc (el=0x81c5040, cp=0xa92d235b > "?|???`!\b\200a\"\b?#-?;\237?@P\034\b?#-?\a") at > read.c:362 > #4 0xb7e9dbdf in el_gets (el=0x81c5040, nread=0xa92d23a8) at read.c: > 241 > #5 0xb7db9f3b in console_thread (thread=0x82102d0, obj=0x8210248) at > src/switch_console.c:441 > #6 0xb7e30bf6 in dummy_worker (opaque=0x82102d0) at > threadproc/unix/thread.c:138 > #7 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > #8 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > > Thread 3 (process 27235): > #0 0xb7f14410 in __kernel_vsyscall () > #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 > #2 0xb7e32799 in apr_sleep (t=100000) at time/unix/time.c:246 > #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 > ---Type to continue, or q to quit--- > #4 0xb7df019e in switch_ivr_originate (session=0xb34dd080, > bleg=0xa8ad10c0, cause=0xa8ad10bc, bridgeto=0x823a528 > "openzap/1/a/04855711", > timelimit_sec=60, table=0xb7ecda20, cid_name_override=0x0, > cid_num_override=0x0, caller_profile_override=0x0, ovars=0x0, > flags=) at src/switch_ivr_originate.c:1793 > #5 0xad7231af in ?? () from /opt/app/voip/ippbx.prod/mod/ > mod_dptools.so > #6 0xb7dc6c54 in switch_core_session_exec (session=0xb34dd080, > application_interface=0xb346ba10, arg=0x823a528 "openzap/1/a/ > 04855711") > at src/switch_core_session.c:1332 > #7 0xb7dc70fe in switch_core_session_execute_application > (session=0xb34dd080, app=0x823a520 "bridge", arg=0x823a528 > "openzap/1/a/04855711") > at src/switch_core_session.c:1254 > #8 0xb7dc93a4 in switch_core_session_run (session=0xb34dd080) at > src/switch_core_state_machine.c:155 > #9 0xb7dc6725 in switch_core_session_thread (thread=0x823a050, > obj=0xb34dd080) at src/switch_core_session.c:940 > #10 0xb7e30bf6 in dummy_worker (opaque=0x823a050) at > threadproc/unix/thread.c:138 > #11 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > #12 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > > Thread 2 (process 27236): > #0 0xb7f14410 in __kernel_vsyscall () > #1 0xb7c72c07 in poll () from /lib/tls/i686/cmov/libc.so.6 > #2 0xb337af00 in wanpipe_wait (zchan=0xb341f9d8, flags=0xa80fee60, > to=40) at src/ozmod/ozmod_wanpipe/ozmod_wanpipe.c:255 > #3 0xae837ae7 in zap_channel_wait (zchan=0x1, flags=0xa80fee60, > to=40) > at src/zap_io.c:1479 > #4 0xae849059 in channel_read_frame (session=0xb34e5400, > frame=0xa80ff170, flags=0, stream_id=0) at mod_openzap.c:593 > #5 0xb7dcbe6f in switch_core_session_read_frame (session=0xb34e5400, > frame=0xa80ff170, flags=0, stream_id=0) at src/switch_core_io.c:161 > #6 0xb7e054ec in switch_ivr_sleep (session=0xb34e5400, ms=10, > sync=SWITCH_FALSE, args=0x0) at src/switch_ivr.c:262 > #7 0xb7deaf94 in originate_on_consume_media_transmit > (session=0xb34e5400) at src/switch_ivr_originate.c:47 > #8 0xb7dc8b74 in switch_core_session_run (session=0xb34e5400) at > src/switch_core_state_machine.c:476 > #9 0xb7dc6725 in switch_core_session_thread (thread=0xb34f1060, > obj=0xb34e5400) at src/switch_core_session.c:940 > #10 0xb7e30bf6 in dummy_worker (opaque=0xb34f1060) at > threadproc/unix/thread.c:138 > #11 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > #12 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > > Thread 1 (process 27234): > #0 0xabe0e408 in mpg123_delete at plt () from > /opt/app/voip/ippbx.prod/mod/mod_shout.so > #1 0xabe13325 in ?? () from /opt/app/voip/ippbx.prod/mod/mod_shout.so > #2 0xb7dc04b9 in switch_core_file_seek (fh=0xa76fef28, > cur_pos=0xa76fef88, samples=0, whence=1) at src/switch_core_file.c:345 > #3 0xb7dfab82 in switch_ivr_play_file (session=0xa7700030, > fh=0xa76fef28, > file=0xa7711ed8 "/opt/app/voip/ippbx.prod/sounds/de/callie/ > qet.mp3", > args=0xa76ff0d8) at src/switch_ivr_play_say.c:1262 > #4 0xad71da25 in ?? () from /opt/app/voip/ippbx.prod/mod/ > mod_dptools.so > #5 0xb7dc6c54 in switch_core_session_exec (session=0xa7700030, > application_interface=0xb346b388, arg=0xb34fd790 "qet.mp3") > at src/switch_core_session.c:1332 > #6 0xb7dc70fe in switch_core_session_execute_application > (session=0xa7700030, app=0xb34fd780 "playback", arg=0xb34fd790 > "qet.mp3") > at src/switch_core_session.c:1254 > #7 0xb7dc93a4 in switch_core_session_run (session=0xa7700030) at > src/switch_core_state_machine.c:155 > #8 0xb7dc6725 in switch_core_session_thread (thread=0xb34fd3d0, > obj=0xa7700030) at src/switch_core_session.c:940 > #9 0xb7e30bf6 in dummy_worker (opaque=0xb34fd3d0) at > threadproc/unix/thread.c:138 > #10 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > #11 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > > > (gdb) thread apply all bt full > > Thread 31 (process 27201): > #0 0xb7f14410 in __kernel_vsyscall () > No symbol table info available. > #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > #2 0xb7e32799 in apr_sleep (t=1000000) at time/unix/time.c:246 > tv = {tv_sec = 0, tv_usec = 940000} > #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 > No locals. > #4 0xb7db9154 in switch_console_loop () at src/switch_console.c:792 > arg = 1 > thread = (switch_thread_t *) 0x82102d0 > thd_attr = (switch_threadattr_t *) 0x8210298 > pool = (switch_memory_pool_t *) 0x8210248 > __func__ = "switch_console_loop" > __PRETTY_FUNCTION__ = "switch_console_loop" > #5 0xb7dcedf0 in switch_core_runtime_loop (bg=0) at src/ > switch_core.c:659 > No locals. > #6 0x0804a36a in main (argc=1, argv=0xbffe65c4) at src/switch.c:666 > pid_path = "/opt/app/voip/ippbx.prod/log/freeswitch.pid", '\0' > > pid_buffer = "27201", '\0' > old_pid_buffer = "27150", '\0' > pid_len = 5 > old_pid_len = 5 > err = 0x0 > nf = 0 > runas_user = 0x0 > runas_group = 0x0 > nc = 0 > pid = 27201 > x = 1111804576 > die = 0 > alt_dirs = 0 > known_opt = -1208927888 > high_prio = 0 > flags = 1 > ret = > destroy_status = > fd = (switch_file_t *) 0x80529b0 > pool = (switch_memory_pool_t *) 0x80528f0 > __PRETTY_FUNCTION__ = "main" > > Thread 30 (process 27202): > #0 0xb7f14410 in __kernel_vsyscall () > No symbol table info available. > #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > #2 0xb7e32799 in apr_sleep (t=100000) at time/unix/time.c:246 > tv = {tv_sec = 0, tv_usec = 100000} > #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 > ---Type to continue, or q to quit--- > No locals. > #4 0xb7dbe2a4 in pool_thread (thread=0xb7a07da8, obj=0x0) at > src/switch_core_memory.c:421 > No locals. > #5 0xb7e30bf6 in dummy_worker (opaque=0xb7a07da8) at > threadproc/unix/thread.c:138 > No locals. > #6 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > No symbol table info available. > #7 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > > Thread 29 (process 27203): > #0 0xb7f14410 in __kernel_vsyscall () > No symbol table info available. > #1 0xb7d2eaa5 in pthread_cond_wait@@GLIBC_2.3.2 () from > /lib/tls/i686/cmov/libpthread.so.0 > No symbol table info available. > #2 0xb7e2acaa in apr_thread_cond_wait (cond=0xb6fb53b8, > mutex=0xb6fb5388) at locks/unix/thread_cond.c:68 > rv = -512 > #3 0xb7e21a3c in apr_queue_pop (queue=0xb6fb5358, data=0xb6ec03a8) at > misc/apr_queue.c:276 > rv = 0 > #4 0xb7dadd44 in switch_queue_pop (queue=0xb6fb5358, data=0xb6ec03a8) > at src/switch_apr.c:879 > No locals. > #5 0xb7ddf879 in switch_event_dispatch_thread (thread=0x8068140, > obj=0xb6fb5358) at src/switch_event.c:230 > pop = (void *) 0x0 > event = (switch_event_t *) 0x0 > my_id = 0 > __func__ = "switch_event_dispatch_thread" > #6 0xb7e30bf6 in dummy_worker (opaque=0x8068140) at > threadproc/unix/thread.c:138 > No locals. > #7 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > No symbol table info available. > #8 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > > Thread 28 (process 27204): > #0 0xb7f14410 in __kernel_vsyscall () > No symbol table info available. > #1 0xb7d2eaa5 in pthread_cond_wait@@GLIBC_2.3.2 () from > /lib/tls/i686/cmov/libpthread.so.0 > No symbol table info available. > #2 0xb7e2acaa in apr_thread_cond_wait (cond=0x805e380, > mutex=0x805e350) > at locks/unix/thread_cond.c:68 > rv = -512 > #3 0xb7e21a3c in apr_queue_pop (queue=0x805e320, data=0xb66bf3a8) at > misc/apr_queue.c:276 > rv = 0 > #4 0xb7dadd44 in switch_queue_pop (queue=0x805e320, > data=0xb66bf3a8) at > src/switch_apr.c:879 > No locals. > #5 0xb7ddec2d in switch_event_thread (thread=0x8068160, > obj=0x805e320) > at src/switch_event.c:273 > pop = (void *) 0x0 > index = 0 > my_id = 0 > __func__ = "switch_event_thread" > #6 0xb7e30bf6 in dummy_worker (opaque=0x8068160) at > threadproc/unix/thread.c:138 > No locals. > ---Type to continue, or q to quit--- > #7 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > No symbol table info available. > #8 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > > Thread 27 (process 27205): > #0 0xb7f14410 in __kernel_vsyscall () > No symbol table info available. > #1 0xb7d2eaa5 in pthread_cond_wait@@GLIBC_2.3.2 () from > /lib/tls/i686/cmov/libpthread.so.0 > No symbol table info available. > #2 0xb7e2acaa in apr_thread_cond_wait (cond=0xb71d4b38, > mutex=0xb71d4b08) at locks/unix/thread_cond.c:68 > rv = -512 > #3 0xb7e21a3c in apr_queue_pop (queue=0xb71d4ad8, data=0xb5ebe3a8) at > misc/apr_queue.c:276 > rv = 0 > #4 0xb7dadd44 in switch_queue_pop (queue=0xb71d4ad8, data=0xb5ebe3a8) > at src/switch_apr.c:879 > No locals. > #5 0xb7ddec2d in switch_event_thread (thread=0x8068180, > obj=0xb71d4ad8) > at src/switch_event.c:273 > pop = (void *) 0x0 > index = 0 > my_id = 1 > __func__ = "switch_event_thread" > #6 0xb7e30bf6 in dummy_worker (opaque=0x8068180) at > threadproc/unix/thread.c:138 > No locals. > #7 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > No symbol table info available. > #8 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > > Thread 26 (process 27206): > #0 0xb7f14410 in __kernel_vsyscall () > No symbol table info available. > #1 0xb7d2eaa5 in pthread_cond_wait@@GLIBC_2.3.2 () from > /lib/tls/i686/cmov/libpthread.so.0 > No symbol table info available. > #2 0xb7e2acaa in apr_thread_cond_wait (cond=0xb7171b38, > mutex=0xb7171b08) at locks/unix/thread_cond.c:68 > rv = -512 > #3 0xb7e21a3c in apr_queue_pop (queue=0xb7171ad8, data=0xb56bd3a8) at > misc/apr_queue.c:276 > rv = 0 > #4 0xb7dadd44 in switch_queue_pop (queue=0xb7171ad8, data=0xb56bd3a8) > at src/switch_apr.c:879 > No locals. > #5 0xb7ddec2d in switch_event_thread (thread=0x80681a0, > obj=0xb7171ad8) > at src/switch_event.c:273 > pop = (void *) 0x0 > index = 0 > my_id = 2 > __func__ = "switch_event_thread" > #6 0xb7e30bf6 in dummy_worker (opaque=0x80681a0) at > threadproc/unix/thread.c:138 > No locals. > #7 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > No symbol table info available. > #8 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > > ---Type to continue, or q to quit--- > Thread 25 (process 27207): > #0 0xb7f14410 in __kernel_vsyscall () > No symbol table info available. > #1 0xb7d2eaa5 in pthread_cond_wait@@GLIBC_2.3.2 () from > /lib/tls/i686/cmov/libpthread.so.0 > No symbol table info available. > #2 0xb7e2acaa in apr_thread_cond_wait (cond=0xb6fb54a8, > mutex=0xb6fb5478) at locks/unix/thread_cond.c:68 > rv = -512 > #3 0xb7e21a3c in apr_queue_pop (queue=0xb6fb5448, data=0xb4dce3a8) at > misc/apr_queue.c:276 > rv = 0 > #4 0xb7dadd44 in switch_queue_pop (queue=0xb6fb5448, data=0xb4dce3a8) > at src/switch_apr.c:879 > No locals. > #5 0xb7e082fd in log_thread (thread=0xb4e30ae0, obj=0x0) at > src/switch_log.c:209 > pop = (void *) 0x0 > binding = (switch_log_binding_t *) 0x0 > __func__ = "log_thread" > #6 0xb7e30bf6 in dummy_worker (opaque=0xb4e30ae0) at > threadproc/unix/thread.c:138 > No locals. > #7 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > No symbol table info available. > #8 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > > Thread 24 (process 27210): > #0 0xb7f14410 in __kernel_vsyscall () > No symbol table info available. > #1 0xb7d2eaa5 in pthread_cond_wait@@GLIBC_2.3.2 () from > /lib/tls/i686/cmov/libpthread.so.0 > No symbol table info available. > #2 0xb7e2acaa in apr_thread_cond_wait (cond=0x80c0c58, > mutex=0x80c0c28) > at locks/unix/thread_cond.c:68 > rv = -512 > #3 0xb7daed54 in switch_thread_cond_wait (cond=0x80c0c58, > mutex=0x80c0c28) at src/switch_apr.c:359 > No locals. > #4 0xb7e11266 in switch_cond_next () at src/switch_time.c:203 > No locals. > #5 0xb7dc27a5 in switch_core_sql_thread (thread=0xb3567ae8, > obj=0x0) at > src/switch_core_sqldb.c:220 > pop = (void *) 0x811bf00 > itterations = 0 > trans = 0 '\0' > nothing_in_queue = 1 '\001' > len = 100 > sql_len = 65536 > sqlbuf = 0x80aa9a8 "" > newlen = > lc = 0 > __PRETTY_FUNCTION__ = "switch_core_sql_thread" > __func__ = "switch_core_sql_thread" > #6 0xb7e30bf6 in dummy_worker (opaque=0xb3567ae8) at > threadproc/unix/thread.c:138 > No locals. > #7 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > No symbol table info available. > #8 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > ---Type to continue, or q to quit--- > > Thread 23 (process 27211): > #0 0xb7f14410 in __kernel_vsyscall () > No symbol table info available. > #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > #2 0xb7e32799 in apr_sleep (t=500000) at time/unix/time.c:246 > tv = {tv_sec = 0, tv_usec = 340000} > #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 > No locals. > #4 0xb7dd2af4 in switch_scheduler_task_thread (thread=0x80baa90, > obj=0x0) at src/switch_scheduler.c:171 > __func__ = "switch_scheduler_task_thread" > #5 0xb7e30bf6 in dummy_worker (opaque=0x80baa90) at > threadproc/unix/thread.c:138 > No locals. > #6 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > No symbol table info available. > #7 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > > Thread 22 (process 27212): > #0 0xb7f14410 in __kernel_vsyscall () > No symbol table info available. > #1 0xb7c7d676 in epoll_wait () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > #2 0xb32b7126 in su_epoll_port_wait_events (self=0x80c6340, > tout=1000) > at su_epoll_port.c:491 > j = -1290679408 > n = -1288576964 > events = 0 > index = -1288973717 > version = 1 > M = 4 > ev = 0xb311c0e0 > __PRETTY_FUNCTION__ = "su_epoll_port_wait_events" > #3 0xb32bd9b8 in su_base_port_step (self=0x80c6340, tout=1000) at > su_base_port.c:442 > now = {tv_sec = 3443337943, tv_usec = 249063} > __PRETTY_FUNCTION__ = "su_base_port_step" > #4 0xb32b8551 in su_port_step (self=0x80c6340, tout=1000) at > su_port.h:326 > base = (su_virtual_port_t *) 0x80c6340 > #5 0xb32b8521 in su_root_step (self=0x80c68f0, tout=1000) at > su_root.c:730 > __PRETTY_FUNCTION__ = "su_root_step" > #6 0xb31e8f29 in sofia_profile_thread_run (thread=0x80d2aa8, > obj=0x80d1e10) at sofia.c:831 > pool = > node = (sip_alias_node_t *) 0xb32f45dc > s_event = (switch_event_t *) 0x0 > sanity = > __func__ = "sofia_profile_thread_run" > __PRETTY_FUNCTION__ = "sofia_profile_thread_run" > #7 0xb7e30bf6 in dummy_worker (opaque=0x80d2aa8) at > threadproc/unix/thread.c:138 > No locals. > #8 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > No symbol table info available. > ---Type to continue, or q to quit--- > #9 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > > Thread 21 (process 27213): > #0 0xb7f14410 in __kernel_vsyscall () > No symbol table info available. > #1 0xb7c7d676 in epoll_wait () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > #2 0xb32b7126 in su_epoll_port_wait_events (self=0x80dac30, > tout=1000) > at su_epoll_port.c:491 > j = -1299074328 > n = 1 > events = 0 > index = 2 > version = 4 > M = 4 > ev = 0xb291b260 > __PRETTY_FUNCTION__ = "su_epoll_port_wait_events" > #3 0xb32bd89b in su_base_port_run (self=0x80dac30) at > su_base_port.c:342 > tout = 1000 > __PRETTY_FUNCTION__ = "su_base_port_run" > #4 0xb32b842b in su_port_run (self=0x80dac30) at su_port.h:312 > base = (su_virtual_port_t *) 0x80dac30 > #5 0xb32b8408 in su_root_run (self=0x80dacb0) at su_root.c:691 > __PRETTY_FUNCTION__ = "su_root_run" > #6 0xb32a8b31 in su_pthread_port_clone_main (varg=0xb311c0a8) at > su_pthread_port.c:321 > arg = (struct clone_args *) 0x0 > task = {{sut_port = 0x80dac30, sut_root = 0x80dacb0}} > zap = 1 > #7 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > No symbol table info available. > #8 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > > Thread 20 (process 27214): > #0 0xb7f14410 in __kernel_vsyscall () > No symbol table info available. > #1 0xb7c7d676 in epoll_wait () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > #2 0xb32b7126 in su_epoll_port_wait_events (self=0x80da208, > tout=1000) > at su_epoll_port.c:491 > j = -1307464816 > n = -1288576964 > events = 0 > index = -1288973717 > version = 1 > M = 4 > ev = 0xb211a0e0 > __PRETTY_FUNCTION__ = "su_epoll_port_wait_events" > #3 0xb32bd9b8 in su_base_port_step (self=0x80da208, tout=1000) at > su_base_port.c:442 > now = {tv_sec = 3443337943, tv_usec = 148997} > __PRETTY_FUNCTION__ = "su_base_port_step" > #4 0xb32b8551 in su_port_step (self=0x80da208, tout=1000) at > su_port.h:326 > ---Type to continue, or q to quit--- > base = (su_virtual_port_t *) 0x80da208 > #5 0xb32b8521 in su_root_step (self=0x80d7558, tout=1000) at > su_root.c:730 > __PRETTY_FUNCTION__ = "su_root_step" > #6 0xb31e8f29 in sofia_profile_thread_run (thread=0x80dde88, > obj=0x80dd650) at sofia.c:831 > pool = > node = (sip_alias_node_t *) 0xb32f45dc > s_event = (switch_event_t *) 0x0 > sanity = > __func__ = "sofia_profile_thread_run" > __PRETTY_FUNCTION__ = "sofia_profile_thread_run" > #7 0xb7e30bf6 in dummy_worker (opaque=0x80dde88) at > threadproc/unix/thread.c:138 > No locals. > #8 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > No symbol table info available. > #9 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > > Thread 19 (process 27215): > #0 0xb7f14410 in __kernel_vsyscall () > No symbol table info available. > #1 0xb7c7d676 in epoll_wait () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > #2 0xb32b7126 in su_epoll_port_wait_events (self=0x80e5228, > tout=1000) > at su_epoll_port.c:491 > j = -1315859736 > n = 1 > events = 0 > index = 1 > version = 3 > M = 4 > ev = 0xb1919260 > __PRETTY_FUNCTION__ = "su_epoll_port_wait_events" > #3 0xb32bd89b in su_base_port_run (self=0x80e5228) at > su_base_port.c:342 > tout = 1000 > __PRETTY_FUNCTION__ = "su_base_port_run" > #4 0xb32b842b in su_port_run (self=0x80e5228) at su_port.h:312 > base = (su_virtual_port_t *) 0x80e5228 > #5 0xb32b8408 in su_root_run (self=0x80e3b40) at su_root.c:691 > __PRETTY_FUNCTION__ = "su_root_run" > #6 0xb32a8b31 in su_pthread_port_clone_main (varg=0xb211a0a8) at > su_pthread_port.c:321 > arg = (struct clone_args *) 0x0 > task = {{sut_port = 0x80e5228, sut_root = 0x80e3b40}} > zap = 1 > #7 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > No symbol table info available. > #8 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > > Thread 18 (process 27216): > #0 0xb7f14410 in __kernel_vsyscall () > No symbol table info available. > #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 > ---Type to continue, or q to quit--- > No symbol table info available. > #2 0xb7e32799 in apr_sleep (t=10000) at time/unix/time.c:246 > tv = {tv_sec = 0, tv_usec = 10000} > #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 > No locals. > #4 0xb31d4b37 in sofia_profile_worker_thread_run (thread=0x80d2b88, > obj=0x80d1e10) at sofia.c:656 > ireg_loops = 1 > gateway_loops = 0 > loops = 95 > qsize = 4294966782 > __PRETTY_FUNCTION__ = "sofia_profile_worker_thread_run" > #5 0xb7e30bf6 in dummy_worker (opaque=0x80d2b88) at > threadproc/unix/thread.c:138 > No locals. > #6 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > No symbol table info available. > #7 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > > Thread 17 (process 27217): > #0 0xb7f14410 in __kernel_vsyscall () > No symbol table info available. > #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > #2 0xb7e32799 in apr_sleep (t=10000) at time/unix/time.c:246 > tv = {tv_sec = 0, tv_usec = 10000} > #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 > No locals. > #4 0xb31d4b37 in sofia_profile_worker_thread_run (thread=0x80ddf68, > obj=0x80dd650) at sofia.c:656 > ireg_loops = 1 > gateway_loops = 0 > loops = 95 > qsize = 4294966782 > __PRETTY_FUNCTION__ = "sofia_profile_worker_thread_run" > #5 0xb7e30bf6 in dummy_worker (opaque=0x80ddf68) at > threadproc/unix/thread.c:138 > No locals. > #6 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > No symbol table info available. > #7 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > > Thread 16 (process 27218): > #0 0xb7f14410 in __kernel_vsyscall () > No symbol table info available. > #1 0xb7c7d676 in epoll_wait () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > #2 0xb32b7126 in su_epoll_port_wait_events (self=0x80e66d8, > tout=1000) > at su_epoll_port.c:491 > j = -1341445232 > n = -1288576964 > events = 0 > index = -1288973717 > version = 1 > ---Type to continue, or q to quit--- > M = 4 > ev = 0xb00b20e0 > __PRETTY_FUNCTION__ = "su_epoll_port_wait_events" > #3 0xb32bd9b8 in su_base_port_step (self=0x80e66d8, tout=1000) at > su_base_port.c:442 > now = {tv_sec = 3443337943, tv_usec = 560073} > __PRETTY_FUNCTION__ = "su_base_port_step" > #4 0xb32b8551 in su_port_step (self=0x80e66d8, tout=1000) at > su_port.h:326 > base = (su_virtual_port_t *) 0x80e66d8 > #5 0xb32b8521 in su_root_step (self=0x80e49f0, tout=1000) at > su_root.c:730 > __PRETTY_FUNCTION__ = "su_root_step" > #6 0xb31e8f29 in sofia_profile_thread_run (thread=0x80e8710, > obj=0x80e7e90) at sofia.c:831 > pool = > node = (sip_alias_node_t *) 0xb32f45dc > s_event = (switch_event_t *) 0x0 > sanity = > __func__ = "sofia_profile_thread_run" > __PRETTY_FUNCTION__ = "sofia_profile_thread_run" > #7 0xb7e30bf6 in dummy_worker (opaque=0x80e8710) at > threadproc/unix/thread.c:138 > No locals. > #8 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > No symbol table info available. > #9 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > > Thread 15 (process 27219): > #0 0xb7f14410 in __kernel_vsyscall () > No symbol table info available. > #1 0xb7c7d676 in epoll_wait () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > #2 0xb32b7126 in su_epoll_port_wait_events (self=0x80ee648, > tout=1000) > at su_epoll_port.c:491 > j = -1349840152 > n = 1 > events = 0 > index = 1 > version = 3 > M = 4 > ev = 0xaf8b1260 > __PRETTY_FUNCTION__ = "su_epoll_port_wait_events" > #3 0xb32bd89b in su_base_port_run (self=0x80ee648) at > su_base_port.c:342 > tout = 1000 > __PRETTY_FUNCTION__ = "su_base_port_run" > #4 0xb32b842b in su_port_run (self=0x80ee648) at su_port.h:312 > base = (su_virtual_port_t *) 0x80ee648 > #5 0xb32b8408 in su_root_run (self=0x80f0bd0) at su_root.c:691 > __PRETTY_FUNCTION__ = "su_root_run" > #6 0xb32a8b31 in su_pthread_port_clone_main (varg=0xb00b20a8) at > su_pthread_port.c:321 > arg = (struct clone_args *) 0x0 > task = {{sut_port = 0x80ee648, sut_root = 0x80f0bd0}} > zap = 1 > #7 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > No symbol table info available. > ---Type to continue, or q to quit--- > #8 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > > Thread 14 (process 27220): > #0 0xb7f14410 in __kernel_vsyscall () > No symbol table info available. > #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > #2 0xb7e32799 in apr_sleep (t=10000) at time/unix/time.c:246 > tv = {tv_sec = 0, tv_usec = 0} > #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 > No locals. > #4 0xb31d4b37 in sofia_profile_worker_thread_run (thread=0x80e87f0, > obj=0x80e7e90) at sofia.c:656 > ireg_loops = 0 > gateway_loops = 0 > loops = 85 > qsize = 4294966782 > __PRETTY_FUNCTION__ = "sofia_profile_worker_thread_run" > #5 0xb7e30bf6 in dummy_worker (opaque=0x80e87f0) at > threadproc/unix/thread.c:138 > No locals. > #6 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > No symbol table info available. > #7 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > > Thread 13 (process 27221): > #0 0xb7f14410 in __kernel_vsyscall () > No symbol table info available. > #1 0xb7c72c07 in poll () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > #2 0xb337af00 in wanpipe_wait (zchan=0xb3446128, flags=0xae751f80, > to=100) at src/ozmod/ozmod_wanpipe/ozmod_wanpipe.c:255 > inflags = 1 > result = > #3 0xae837ae7 in zap_channel_wait (zchan=0x1, flags=0xae751f80, > to=100) > at src/zap_io.c:1479 > __PRETTY_FUNCTION__ = "zap_channel_wait" > #4 0xae80aee8 in zap_isdn_run (me=0xb3416528, obj=0xb341dbc8) at > src/ozmod/ozmod_isdn/ozmod_isdn.c:1725 > flags = ZAP_READ > status = ZAP_TIMEOUT > span = > isdn_data = (zap_isdn_data_t *) 0xae753008 > frame = "\002\001\002\002\b\002\200\002\001\036\002\201\210", > '\0' > len = 13 > errs = 0 > __func__ = "zap_isdn_run" > #5 0xae8421ba in thread_launch (args=0xb3416528) at > src/zap_threadmutex.c:74 > exit_val = > #6 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > No symbol table info available. > #7 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > > ---Type to continue, or q to quit--- > Thread 12 (process 27222): > #0 0xb7f14410 in __kernel_vsyscall () > No symbol table info available. > #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > #2 0xb7e32799 in apr_sleep (t=100000) at time/unix/time.c:246 > tv = {tv_sec = 0, tv_usec = 100000} > #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 > No locals. > #4 0xb31f929e in sofia_presence_event_thread_run (thread=0x80cf958, > obj=0x0) at sofia_presence.c:664 > count = 0 > pop = (void *) 0xb3462cb8 > __func__ = "sofia_presence_event_thread_run" > #5 0xb7e30bf6 in dummy_worker (opaque=0x80cf958) at > threadproc/unix/thread.c:138 > No locals. > #6 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > No symbol table info available. > #7 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > > Thread 11 (process 27223): > #0 0xb7f14410 in __kernel_vsyscall () > No symbol table info available. > #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > #2 0xb7e32799 in apr_sleep (t=1000000) at time/unix/time.c:246 > tv = {tv_sec = 0, tv_usec = 320000} > #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 > No locals. > #4 0xad706ea6 in ?? () from /opt/app/voip/ippbx.prod/mod/mod_fifo.so > No locals. > #5 0xb7e30bf6 in dummy_worker (opaque=0xad386b90) at > threadproc/unix/thread.c:138 > No locals. > #6 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > No symbol table info available. > #7 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > > Thread 10 (process 27225): > #0 0xb7f14410 in __kernel_vsyscall () > No symbol table info available. > #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > #2 0xb7e32799 in apr_sleep (t=1000000) at time/unix/time.c:246 > tv = {tv_sec = 0, tv_usec = 840000} > #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 > No locals. > #4 0xabdf94ab in ?? () from > /opt/app/voip/ippbx.prod/mod/mod_local_stream.so > No locals. > #5 0xb7e30bf6 in dummy_worker (opaque=0x81386e0) at > threadproc/unix/thread.c:138 > No locals. > ---Type to continue, or q to quit--- > #6 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > No symbol table info available. > #7 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > > Thread 9 (process 27226): > #0 0xb7f14410 in __kernel_vsyscall () > No symbol table info available. > #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > #2 0xb7e32799 in apr_sleep (t=1000000) at time/unix/time.c:246 > tv = {tv_sec = 0, tv_usec = 840000} > #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 > No locals. > #4 0xabdf94ab in ?? () from > /opt/app/voip/ippbx.prod/mod/mod_local_stream.so > No locals. > #5 0xb7e30bf6 in dummy_worker (opaque=0x815e728) at > threadproc/unix/thread.c:138 > No locals. > #6 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > No symbol table info available. > #7 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > > Thread 8 (process 27227): > #0 0xb7f14410 in __kernel_vsyscall () > No symbol table info available. > #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > #2 0xb7e32799 in apr_sleep (t=1000000) at time/unix/time.c:246 > tv = {tv_sec = 0, tv_usec = 840000} > #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 > No locals. > #4 0xabdf94ab in ?? () from > /opt/app/voip/ippbx.prod/mod/mod_local_stream.so > No locals. > #5 0xb7e30bf6 in dummy_worker (opaque=0x8184770) at > threadproc/unix/thread.c:138 > No locals. > #6 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > No symbol table info available. > #7 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > > Thread 7 (process 27228): > #0 0xb7f14410 in __kernel_vsyscall () > No symbol table info available. > #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > #2 0xb7e32799 in apr_sleep (t=1000) at time/unix/time.c:246 > tv = {tv_sec = 0, tv_usec = 1000} > #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 > No locals. > #4 0xb7e12a59 in softtimer_runtime () at src/switch_time.c:459 > ---Type to continue, or q to quit--- > current_ms = 1544 > x = > tick = 144 > ts = > last = 1234349144129112 > fwd_errs = 0 > rev_errs = 0 > __func__ = "softtimer_runtime" > #5 0xb7dd4fb3 in switch_loadable_module_exec (thread=0x80bf8d8, > obj=0x80bf6c8) at src/switch_loadable_module.c:93 > status = > module = (switch_loadable_module_t *) 0x80c0bc8 > __PRETTY_FUNCTION__ = "switch_loadable_module_exec" > __func__ = "switch_loadable_module_exec" > #6 0xb7e30bf6 in dummy_worker (opaque=0x80bf8d8) at > threadproc/unix/thread.c:138 > No locals. > #7 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > No symbol table info available. > #8 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > > Thread 6 (process 27229): > #0 0xb7f14410 in __kernel_vsyscall () > No symbol table info available. > #1 0xb7d31bb8 in accept () from /lib/tls/i686/cmov/libpthread.so.0 > No symbol table info available. > #2 0xb7e2f90d in apr_socket_accept (new=0xaa2d434c, sock=0x81bbbb0, > connection_context=0x81bdaa8) at network_io/unix/sockets.c:187 > No locals. > #3 0xb7dae3fb in switch_socket_accept (new_sock=0xaa2d434c, > sock=0x81bbbb0, pool=0x81bdaa8) at src/switch_apr.c:664 > No locals. > #4 0xb33262f2 in mod_event_socket_runtime () at mod_event_socket.c: > 2134 > pool = (switch_memory_pool_t *) 0x81bbaa0 > listener_pool = (switch_memory_pool_t *) 0x81bdaa8 > rv = > sa = (switch_sockaddr_t *) 0x81bbaf8 > inbound_socket = (switch_socket_t *) 0x81bdb00 > listener = > x = > __func__ = "mod_event_socket_runtime" > #5 0xb7dd4fb3 in switch_loadable_module_exec (thread=0x80bfb40, > obj=0x80bf930) at src/switch_loadable_module.c:93 > status = > module = (switch_loadable_module_t *) 0xb340ec28 > __PRETTY_FUNCTION__ = "switch_loadable_module_exec" > __func__ = "switch_loadable_module_exec" > #6 0xb7e30bf6 in dummy_worker (opaque=0x80bfb40) at > threadproc/unix/thread.c:138 > No locals. > #7 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > No symbol table info available. > #8 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > > Thread 5 (process 27230): > ---Type to continue, or q to quit--- > #0 0xb7f14410 in __kernel_vsyscall () > No symbol table info available. > #1 0xb7c72c07 in poll () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > #2 0xb33ce86e in ?? () from /opt/app/voip/ippbx.prod/mod/ > mod_xml_rpc.so > No locals. > #3 0xb33c1464 in ChanSwitchAccept (chanSwitchP=0x81f5030, > channelPP=0xa9ad30e0, channelInfoPP=0xa9ad30dc, errorP=0xa9ad30e4) > at ../../../../libs/xmlrpc-c/lib/abyss/src/chanswitch.c:149 > No locals. > #4 0xb33cd37e in ServerRun (serverP=0xb33ffe4c) at > ../../../../libs/xmlrpc-c/lib/abyss/src/server.c:908 > srvP = (struct _TServer * const) 0x81f4fc8 > #5 0xb33be832 in mod_xml_rpc_runtime () at mod_xml_rpc.c:837 > registryP = (xmlrpc_registry *) 0x81bb960 > env = {fault_occurred = 0, fault_code = 0, fault_string = 0x0} > logfile = "/opt/app/voip/ippbx.prod/log/freeswitch_http.log", > '\0' > hi = > var = (const void *) 0x80609b8 > val = (void *) 0x805dc78 > __func__ = "mod_xml_rpc_runtime" > #6 0xb7dd4fb3 in switch_loadable_module_exec (thread=0x80bfda8, > obj=0x80bfb98) at src/switch_loadable_module.c:93 > status = > module = (switch_loadable_module_t *) 0xb3405178 > __PRETTY_FUNCTION__ = "switch_loadable_module_exec" > __func__ = "switch_loadable_module_exec" > #7 0xb7e30bf6 in dummy_worker (opaque=0x80bfda8) at > threadproc/unix/thread.c:138 > No locals. > #8 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > No symbol table info available. > #9 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > > Thread 4 (process 27231): > #0 0xb7f14410 in __kernel_vsyscall () > No symbol table info available. > #1 0xb7d3199b in read () from /lib/tls/i686/cmov/libpthread.so.0 > No symbol table info available. > #2 0xb7e9dfb3 in read_char (el=0x81c5040, cp=0xa92d235b > "?|???`!\b\200a\"\b?#-?;\237?@P\034\b?#-?\a") at > read.c:294 > num_read = 136073656 > tried = 0 > #3 0xb7e9da9c in el_getc (el=0x81c5040, cp=0xa92d235b > "?|???`!\b\200a\"\b?#-?;\237?@P\034\b?#-?\a") at > read.c:362 > num_read = -1456659621 > ma = (c_macro_t *) 0x81c52e0 > #4 0xb7e9dbdf in el_gets (el=0x81c5040, nread=0xa92d23a8) at read.c: > 241 > cmdnum = 232 '?' > num = 136405224 > ch = -1 '?' > #5 0xb7db9f3b in console_thread (thread=0x82102d0, obj=0x8210248) at > src/switch_console.c:441 > arg = 1 > count = 7 > line = 0x82160e8 "" > pool = (switch_memory_pool_t *) 0x8210248 > ---Type to continue, or q to quit--- > __func__ = "console_thread" > __PRETTY_FUNCTION__ = "console_thread" > #6 0xb7e30bf6 in dummy_worker (opaque=0x82102d0) at > threadproc/unix/thread.c:138 > No locals. > #7 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > No symbol table info available. > #8 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > > Thread 3 (process 27235): > #0 0xb7f14410 in __kernel_vsyscall () > No symbol table info available. > #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > #2 0xb7e32799 in apr_sleep (t=100000) at time/unix/time.c:246 > tv = {tv_sec = 0, tv_usec = 20000} > #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 > No locals. > #4 0xb7df019e in switch_ivr_originate (session=0xb34dd080, > bleg=0xa8ad10c0, cause=0xa8ad10bc, bridgeto=0x823a528 > "openzap/1/a/04855711", > timelimit_sec=60, table=0xb7ecda20, cid_name_override=0x0, > cid_num_override=0x0, caller_profile_override=0x0, ovars=0x0, > flags=) at src/switch_ivr_originate.c:1793 > end = > var_begin = > originate_status = {{peer_session = 0xb34e5400, peer_channel = > 0xb34e9968, caller_profile = 0xb34e1dc8, ring_ready = 0 '\0', > early_media = 1 '\001', answered = 0 '\0', > per_channel_timelimit_sec > = 0, per_channel_progress_timelimit_sec = 0}, {peer_session = 0x0, > peer_channel = 0x0, caller_profile = 0x0, ring_ready = 0 '\0', > early_media = 0 '\0', answered = 0 '\0', per_channel_timelimit_sec = > 0, > per_channel_progress_timelimit_sec = 0} } > dftflags = 0 > myflags = 0 > pipe_names = {0xa811de20 "openzap", 0x0 } > data = > status = SWITCH_STATUS_SUCCESS > caller_channel = (switch_channel_t *) 0xb34e15e8 > peer_names = {0xa811de20 "openzap", 0x0 } > new_session = (switch_core_session_t *) 0xb34e5400 > peer_session = > new_profile = (switch_caller_profile_t *) 0xb34e1dc8 > caller_caller_profile = > chan_type = > chan_data = > peer_channel = > ringback = {audio_buffer = 0x0, ts = {TONES = {{freqs = {0, 0, > 0, 0, 0, 0}} }, channels = 0, rate = 0, > duration = 0, wait = 0, tmp_duration = 0, tmp_wait = 0, loops = 0, > LOOPS = 0, decay_factor = 0, decay_direction = 0, decay_step = 0, > volume = 0, debug = 0, debug_stream = 0x0, user_data = 0x0, > buffer = > 0x0, datalen = 0, samples = 0, dynamic = 0, handler = 0}, fhb = { > file_interface = 0x0, flags = 0, fd = 0x0, samples = 0, > samplerate = > 0, native_rate = 0, channels = 0 '\0', format = 0, sections = 0, > seekable = 0, sample_count = 0, speed = 0, memory_pool = 0x0, > prebuf > = 0, interval = 0, private_info = 0x0, handler = 0x0, pos = 0, > audio_buffer = 0x0, sp_audio_buffer = 0x0, thresh = 0, silence_hits > = 0, offset_pos = 0, last_pos = 0, vol = 0, resampler = 0x0, > buffer = 0x0, dbuf = 0x0, dbuflen = 0, pre_buffer = 0x0, > pre_buffer_data = 0x0, pre_buffer_datalen = 0, file = 0x0, func = 0x0, > line = 0}, fh = 0x0, silence = 0, asis = 0 '\0'} > start = 1234349134 > read_frame = (switch_frame_t *) 0x0 > ---Type to continue, or q to quit--- > pool = (switch_memory_pool_t *) 0x0 > r = 0 > i = -1286728216 > and_argc = 1 > or_argc = 1 > sleep_ms = 1000 > try = 0 > retries = 1 > write_codec = {codec_interface = 0x0, implementation = 0x0, > fmtp_in = 0x0, fmtp_out = 0x0, codec_settings = {quality = 0, > complexity = 0, enhancement = 0, vad = 0, vbr = 0, vbr_quality = 0, > abr = 0, dtx = 0, preproc = 0, pp_vad = 0, pp_agc = 0, > pp_agc_level = 0, pp_denoise = 0, pp_dereverb = 0, > pp_dereverb_decay > = 0, pp_dereverb_level = 0}, flags = 0, memory_pool = 0x0, > private_info = 0x0, agreed_pt = 0 '\0', mutex = 0x0} > write_frame = {codec = 0x0, source = 0x0, packet = 0x0, > packetlen = 0, data = 0x821ec80, datalen = 0, buflen = 4096, samples > = 0, > rate = 0, payload = 0 '\0', timestamp = 0, seq = 0, ssrc = 0, m = > SWITCH_FALSE, flags = 0} > pass = 0 '\0' > odata = 0xa811e160 "openzap/1/a/04855711" > var = > reason = SWITCH_CAUSE_SUCCESS > to = 0 '\0' > var_val = > vars = 0x0 > var_block_count = 0 > e = > ringback_data = 0x0 > read_codec = > message = (switch_core_session_message_t *) 0x0 > var_event = (switch_event_t *) 0xb346cea8 > fail_on_single_reject = 0 '\0' > fail_on_single_reject_var = 0x0 > loop_data = 0xa811de20 "openzap" > progress_timelimit_sec = 60 > oglobals = {session = 0xb34dd080, idx = -1, hups = 0, file = > '\0' , key = '\0' , > early_ok = 0 '\0', ring_ready = 1 '\001', sent_ring = 1 '\001', > progress = 1 '\001', return_ring_ready = 0 '\0', > monitor_early_media_ring = 0 '\0', monitor_early_media_fail = 0 '\0', > gen_ringback = 0 '\0', ignore_early_media = 1 '\001', > ignore_ring_ready = 0 '\0'} > __PRETTY_FUNCTION__ = "switch_ivr_originate" > __func__ = "switch_ivr_originate" > #5 0xad7231af in ?? () from /opt/app/voip/ippbx.prod/mod/ > mod_dptools.so > No locals. > #6 0xb7dc6c54 in switch_core_session_exec (session=0xb34dd080, > application_interface=0xb346ba10, arg=0x823a528 "openzap/1/a/ > 04855711") > at src/switch_core_session.c:1332 > log = > lp = > event = (switch_event_t *) 0x0 > var = > channel = (switch_channel_t *) 0xb34e15e8 > expanded = 0x823a528 "openzap/1/a/04855711" > app = 0xad7253b0 "bridge" > __PRETTY_FUNCTION__ = "switch_core_session_exec" > __func__ = "switch_core_session_exec" > #7 0xb7dc70fe in switch_core_session_execute_application > (session=0xb34dd080, app=0x823a520 "bridge", arg=0x823a528 > "openzap/1/a/04855711") > ---Type to continue, or q to quit--- > at src/switch_core_session.c:1254 > application_interface = (switch_application_interface_t *) > 0xb346ba10 > __func__ = "switch_core_session_execute_application" > #8 0xb7dc93a4 in switch_core_session_run (session=0xb34dd080) at > src/switch_core_state_machine.c:155 > proceed = > global_proceed = > do_extra_handlers = > state = > endstate = > endpoint_interface = > driver_state_handler = (const switch_state_handler_table_t *) > 0xb331f860 > application_state_handler = > thread_id = 3084680427 > env = {{__jmpbuf = {134603552, -1287115232, -1209317480, 27232, > -1287651304, 24}, __mask_was_saved = -1210923727, __saved_mask = { > __val = {2819623592, 2829914760, 3084048908, 2819620880, 27232, > 1, > 3008239368, 2829914744, 3085760892, 250000, 3084048908, 2829914760, > 3085085008, 1, 3085760892, 2829914792, 3085760892, 0, > 134564192, > 3084043569, 3085085008, 134564244, 3085760892, 2829914824, 27232, > 134564240, 3007790064, 3084573819, 3085760892, 3008213112, > 3084048908, 2829914856}}}} > sig = > __func__ = "switch_core_session_run" > __PRETTY_FUNCTION__ = "switch_core_session_run" > #9 0xb7dc6725 in switch_core_session_thread (thread=0x823a050, > obj=0xb34dd080) at src/switch_core_session.c:940 > session = (switch_core_session_t *) 0xb34dd080 > event = > event_str = 0x0 > val = > __func__ = "switch_core_session_thread" > __PRETTY_FUNCTION__ = "switch_core_session_thread" > #10 0xb7e30bf6 in dummy_worker (opaque=0x823a050) at > threadproc/unix/thread.c:138 > No locals. > #11 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > No symbol table info available. > #12 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > > Thread 2 (process 27236): > #0 0xb7f14410 in __kernel_vsyscall () > No symbol table info available. > #1 0xb7c72c07 in poll () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > #2 0xb337af00 in wanpipe_wait (zchan=0xb341f9d8, flags=0xa80fee60, > to=40) at src/ozmod/ozmod_wanpipe/ozmod_wanpipe.c:255 > inflags = 1 > result = > #3 0xae837ae7 in zap_channel_wait (zchan=0x1, flags=0xa80fee60, > to=40) > at src/zap_io.c:1479 > __PRETTY_FUNCTION__ = "zap_channel_wait" > #4 0xae849059 in channel_read_frame (session=0xb34e5400, > frame=0xa80ff170, flags=0, stream_id=0) at mod_openzap.c:593 > channel = (switch_channel_t *) 0xb34e9968 > len = > wflags = ZAP_READ > dtmf = '\0' > status = > total_to = 80 > ---Type to continue, or q to quit--- > chunk = 40 > do_break = 0 > __PRETTY_FUNCTION__ = "channel_read_frame" > __func__ = "channel_read_frame" > #5 0xb7dcbe6f in switch_core_session_read_frame (session=0xb34e5400, > frame=0xa80ff170, flags=0, stream_id=0) at src/switch_core_io.c:161 > ptr = > status = SWITCH_STATUS_SUCCESS > need_codec = > perfect = > do_bugs = -1212118646 > do_resample = 0 > is_cng = 0 > flag = 0 > __PRETTY_FUNCTION__ = "switch_core_session_read_frame" > __func__ = "switch_core_session_read_frame" > #6 0xb7e054ec in switch_ivr_sleep (session=0xb34e5400, ms=10, > sync=SWITCH_FALSE, args=0x0) at src/switch_ivr.c:262 > channel = (switch_channel_t *) 0xb34e9968 > status = SWITCH_STATUS_SUCCESS > start = 1234349144119246 > now = 0 > done = 1234349144129246 > read_frame = (switch_frame_t *) 0x0 > cng_frame = {codec = 0x0, source = 0x0, packet = 0x0, packetlen > = 0, data = 0xa80ff176, datalen = 2, buflen = 2, samples = 0, > rate = 0, payload = 0 '\0', timestamp = 0, seq = 0, ssrc = 0, m = > SWITCH_FALSE, flags = 1} > data = "\000" > write_frame = {codec = 0x0, source = 0x0, packet = 0x0, > packetlen = 0, data = 0x0, datalen = 0, buflen = 0, samples = 0, > rate = 0, > payload = 0 '\0', timestamp = 0, seq = 0, ssrc = 0, m = SWITCH_FALSE, > flags = 0} > abuf = (unsigned char *) 0x0 > imp = {codec_type = SWITCH_CODEC_TYPE_AUDIO, ianacode = 0 '\0', > iananame = 0x0, fmtp = 0x0, samples_per_second = 0, > actual_samples_per_second = 0, bits_per_second = 0, > microseconds_per_packet = 0, samples_per_packet = 0, > decoded_bytes_per_packet = 0, > encoded_bytes_per_packet = 0, number_of_channels = 0 '\0', > codec_frames_per_packet = 0, init = 0, encode = 0, decode = 0, > destroy = 0, > codec_id = 0, next = 0x0} > codec = {codec_interface = 0x0, implementation = 0x0, fmtp_in = > 0x0, fmtp_out = 0x0, codec_settings = {quality = 0, complexity = 0, > enhancement = 0, vad = 0, vbr = 0, vbr_quality = 0, abr = 0, dtx = > 0, preproc = 0, pp_vad = 0, pp_agc = 0, pp_agc_level = 0, > pp_denoise = 0, pp_dereverb = 0, pp_dereverb_decay = 0, > pp_dereverb_level = 0}, flags = 0, memory_pool = 0x0, private_info = > 0x0, > agreed_pt = 0 '\0', mutex = 0x0} > sval = 0 > var = > __func__ = "switch_ivr_sleep" > __PRETTY_FUNCTION__ = "switch_ivr_sleep" > #7 0xb7deaf94 in originate_on_consume_media_transmit > (session=0xb34e5400) at src/switch_ivr_originate.c:47 > channel = (switch_channel_t *) 0xb34e9968 > #8 0xb7dc8b74 in switch_core_session_run (session=0xb34e5400) at > src/switch_core_state_machine.c:476 > proceed = 1 > global_proceed = 0 > do_extra_handlers = > state = > endstate = > endpoint_interface = > driver_state_handler = (const switch_state_handler_table_t *) > 0xae8523c0 > application_state_handler = > ---Type to continue, or q to quit--- > thread_id = 3084680427 > env = {{__jmpbuf = {0, -1287122688, -1475349896, 27233, > -1211162264, -1210902852}, __mask_was_saved = -1210923727, > __saved_mask = { > __val = {0, 3084099572, 3084048908, 3085631456, 27233, 1, > 3008273032, 2819617400, 3085760892, 250000, 3084048908, 2819617416, > 3085085008, 1, 3085760892, 2819617448, 3085760892, 0, > 134564192, > 3084043569, 3085085008, 134564244, 3085760892, 2819617480, 27233, > 134564240, 135380168, 3084573819, 3085760892, 3008246776, > 3084048908, 2819617512}}}} > sig = > __func__ = "switch_core_session_run" > __PRETTY_FUNCTION__ = "switch_core_session_run" > #9 0xb7dc6725 in switch_core_session_thread (thread=0xb34f1060, > obj=0xb34e5400) at src/switch_core_session.c:940 > session = (switch_core_session_t *) 0xb34e5400 > event = > event_str = 0x0 > val = > __func__ = "switch_core_session_thread" > __PRETTY_FUNCTION__ = "switch_core_session_thread" > #10 0xb7e30bf6 in dummy_worker (opaque=0xb34f1060) at > threadproc/unix/thread.c:138 > No locals. > #11 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > No symbol table info available. > #12 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > > Thread 1 (process 27234): > #0 0xabe0e408 in mpg123_delete at plt () from > /opt/app/voip/ippbx.prod/mod/mod_shout.so > No locals. > #1 0xabe13325 in ?? () from /opt/app/voip/ippbx.prod/mod/mod_shout.so > No locals. > #2 0xb7dc04b9 in switch_core_file_seek (fh=0xa76fef28, > cur_pos=0xa76fef88, samples=0, whence=1) at src/switch_core_file.c:345 > status = 4294967295 > __PRETTY_FUNCTION__ = "switch_core_file_seek" > #3 0xb7dfab82 in switch_ivr_play_file (session=0xa7700030, > fh=0xa76fef28, > file=0xa7711ed8 "/opt/app/voip/ippbx.prod/sounds/de/callie/ > qet.mp3", > args=0xa76ff0d8) at src/switch_ivr_play_say.c:1262 > channel = (switch_channel_t *) 0xa7704598 > dtmf = {digit = 0 '\0', duration = 0} > interval = 2909916864 > samples = 320 > framelen = 640 > sample_start = > olen = 320 > llen = 320 > write_frame = {codec = 0xa76fefb8, source = 0x0, packet = 0x0, > packetlen = 0, data = 0xa7743f30, datalen = 640, buflen = 32768, > samples = 320, rate = 16000, payload = 0 '\0', timestamp = 0, seq = > 0, > ssrc = 0, m = SWITCH_FALSE, flags = 0} > timer = {interval = 0, flags = 0, samples = 0, samplecount = 0, > timer_interface = 0x0, memory_pool = 0x0, private_info = 0x0, > diff = 0, tick = 0} > codec = {codec_interface = 0x80cd8f8, implementation = > 0x80cdf20, fmtp_in = 0x0, fmtp_out = 0x0, codec_settings = {quality > = 0, > complexity = 0, enhancement = 0, vad = 0, vbr = 0, vbr_quality = 0, > abr = 0, dtx = 0, preproc = 0, pp_vad = 0, pp_agc = 0, > pp_agc_level = 0, pp_denoise = 0, pp_dereverb = 0, > pp_dereverb_decay > = 0, pp_dereverb_level = 0}, flags = 3, memory_pool = 0xb34fbb38, > private_info = 0x0, agreed_pt = 0 '\0', mutex = 0xa7711f10} > pool = (switch_memory_pool_t *) 0xb34fbb38 > status = SWITCH_STATUS_SUCCESS > lfh = {file_interface = 0x8136258, flags = 3085, fd = 0x0, > samples = 0, samplerate = 16000, native_rate = 16000, channels = 1 > '\001', > ---Type to continue, or q to quit--- > format = 0, sections = 0, seekable = 0, sample_count = 729088, > speed = > 0, memory_pool = 0xa7712040, prebuf = 0, interval = 0, > private_info = 0xa7714048, handler = 0x0, pos = 0, audio_buffer = > 0xb3476f88, sp_audio_buffer = 0x0, thresh = 0, silence_hits = 0, > offset_pos = 364640, last_pos = 0, vol = 0, resampler = 0x0, buffer = > 0x0, dbuf = 0x0, dbuflen = 0, pre_buffer = 0x0, > pre_buffer_data = 0x0, pre_buffer_datalen = 0, file = 0xb7eb71f0 > "src/switch_ivr_play_say.c", func = 0xb7eb794b "switch_ivr_play_file", > line = 894} > read_codec = (switch_codec_t *) 0xb34fcb90 > p = 0xb7eaf8d4 "current_application" > asis = 0 '\0' > prefix = > timer_name = 0x0 > prebuf = > eof = 1 > bread = > __func__ = "switch_ivr_play_file" > __PRETTY_FUNCTION__ = "switch_ivr_play_file" > #4 0xad71da25 in ?? () from /opt/app/voip/ippbx.prod/mod/ > mod_dptools.so > No locals. > #5 0xb7dc6c54 in switch_core_session_exec (session=0xa7700030, > application_interface=0xb346b388, arg=0xb34fd790 "qet.mp3") > at src/switch_core_session.c:1332 > log = > lp = > event = (switch_event_t *) 0x0 > var = > channel = (switch_channel_t *) 0xa7704598 > expanded = 0xb34fd790 "qet.mp3" > app = 0xad725133 "playback" > __PRETTY_FUNCTION__ = "switch_core_session_exec" > __func__ = "switch_core_session_exec" > #6 0xb7dc70fe in switch_core_session_execute_application > (session=0xa7700030, app=0xb34fd780 "playback", arg=0xb34fd790 > "qet.mp3") > at src/switch_core_session.c:1254 > application_interface = (switch_application_interface_t *) > 0xb346b388 > __func__ = "switch_core_session_execute_application" > #7 0xb7dc93a4 in switch_core_session_run (session=0xa7700030) at > src/switch_core_state_machine.c:155 > proceed = > global_proceed = > do_extra_handlers = > state = > endstate = > endpoint_interface = > driver_state_handler = (const switch_state_handler_table_t *) > 0xb331f860 > application_state_handler = > thread_id = 3084680427 > env = {{__jmpbuf = {0, 0, 0, 0, 0, 0}, __mask_was_saved = 0, > __saved_mask = {__val = {0 }}}} > sig = > __func__ = "switch_core_session_run" > __PRETTY_FUNCTION__ = "switch_core_session_run" > #8 0xb7dc6725 in switch_core_session_thread (thread=0xb34fd3d0, > obj=0xa7700030) at src/switch_core_session.c:940 > session = (switch_core_session_t *) 0xa7700030 > event = > event_str = 0x0 > val = > ---Type to continue, or q to quit--- > __func__ = "switch_core_session_thread" > __PRETTY_FUNCTION__ = "switch_core_session_thread" > #9 0xb7e30bf6 in dummy_worker (opaque=0xb34fd3d0) at > threadproc/unix/thread.c:138 > No locals. > #10 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > No symbol table info available. > #11 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > > > regards > Helmut > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From helmut.kuper at ewetel.de Wed Feb 11 04:41:22 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Wed, 11 Feb 2009 13:41:22 +0100 Subject: [Freeswitch-users] Question: SIP BYE authentication Message-ID: <4992C772.4090906@ewetel.de> Hello, my FS is connected to my SIP-DDI softswitch, which requires all SIP requests sent by a registered SIP account to be authenticated. I found that when FS sends a BYE FreeSWITCH ignores the authentication challenge (SIP/2.0 407) received from proxy and simply terminates the session. Is there a way to configure FS in that way that it react on auth challenges for BYEs ? regards Helmut From helmut.kuper at ewetel.de Wed Feb 11 04:43:45 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Wed, 11 Feb 2009 13:43:45 +0100 Subject: [Freeswitch-users] segfault when shoutcast plays mp3 and extension hangs up In-Reply-To: References: <4992BFFC.50006@ewetel.de> Message-ID: <4992C801.4020906@ewetel.de> Hi Brian, of course. Will do it as soon as I have a second FS plattform for testing SVN trunks. thx and regards Helmut On 11.02.2009 13:16, Brian West wrote: > The proper location to post this is on Jira... please in the future > report ALL issues via Jira. They'll get lost if not done this way. > > Thanks, > Brian > From edpimentl at gmail.com Wed Feb 11 04:58:47 2009 From: edpimentl at gmail.com (EdPimentl) Date: Wed, 11 Feb 2009 07:58:47 -0500 Subject: [Freeswitch-users] TODAY/URGENT: Stop IETF Enactment of Patented Standard for TLS In-Reply-To: <4992C6B7.A0699D98@RealMeasures.dyndns.org> References: <48804AEE.2231AA61@RealMeasures.dyndns.org> <49524E6F.BB09B1C4@RealMeasures.dyndns.org> <4992C6B7.A0699D98@RealMeasures.dyndns.org> Message-ID: <9dc4a1670902110458w7858da12v449052271521d8af@mail.gmail.com> Have you seen this? and how will impact FS in the future? -------------------------------------------------------------------------------------------- >From Seth: (Urgent. Send your note TODAY and CONFIRM the automatic reply from IETF. You can cc campaigns at fsf.org . Three links below, FSF's action page, Glyn Moody's blog, and the list announcement for TLS-AUTHZ at IETF. -- Seth) > http://www.fsf.org/news/reoppose-tls-authz-standard Send comments opposing TLS-authz standard by February 11 Last January, the Free Software Foundation issued an alert to efforts at the Internet Engineering Task Force (IETF) to sneak a patent-encumbered standard for "TLS authorization" through a back-door approval process that was referenced as "experimental" or "informational" (http://www.fsf.org/news/reoppose-tls-authz-standard/newsitem_view). The many comments sent to IETF at that time alerted committee members to this attempt and successfully prevented the standard gaining approval. Unfortunately, attempts to push through this standard have been renewed and become more of a threat. The proposal now at the IETF has a changed status from "experimental" to "proposed standard". The FSF is again issuing an alert and request for comments to be sent urgently and prior to the February 11 deadline to ietf at ietf.org. Please include us in your message by a CC to campaigns at fsf.org. You should also expect an automated reply from ietf at ietf.org, which you will need to answer to confirm your original message. That patent in question is claimed by RedPhone Security (https://datatracker.ietf.org/ipr/1026/). RedPhone has given a license to anyone who implements the protocol, but they still threaten to sue anyone that uses it. If our voice is strong enough, the IETF will not approve this standard on any level unless the patent threat is removed entirely with a royalty-free license for all users. Further background for your comment See the IETF summary: > http://www.ietf.org/mail-archive/web/ietf-announce/current/msg05617.html Much of the communication on the Internet happens between computers according to standards that define common languages. If we are going to live in a free world using free software, our software must be allowed to speak these languages. Unfortunately, discussions about possible new standards are tempting opportunities for people who would prefer to profit by extending proprietary control over our communities. If someone holds a software patent on a technique that a programmer or user has to use in order to make use of a standard, then no one is free without getting permission from and paying the patent holder (http://www.gnu.org/philosophy/fighting-software-patents.html). If we are not careful, standards can become major barriers to computer users having and exercising their freedom. We depend on organizations like the Internet Engineering Task Force (IETF) and the Internet Engineering Steering Group (IESG) to evaluate new proposals for standards and make sure that they are not encumbered by patents or any other sort of restriction that would prevent free software users and programmers from participating in the world they define. In February 2006, a standard for "TLS authorization" was introduced in the IETF for consideration (http://tools.ietf.org/wg/tls/draft-housley-tls-authz-extns-07.txt). Very late in the discussion, a company called RedPhone Security disclosed (this disclosure has subsequently been unpublished from the IETF website) that they applied for a patent which would need to be licensed to anyone wanting to practice the standard (https://datatracker.ietf.org/ipr/833/). After this disclosure, the proposal was rejected. Despite claims that RedPhone have offered a license for implementation of this protocol, users of this protocol would still be threatened by the patent. The IETF should continue to oppose this standard until RedPhone provide a royalty-free license for all users. Media Contacts Peter T. Brown Executive Director Free Software Foundation (617)542-5942 campaigns at fsf.org --- > http://www.computerworlduk.com/community/blogs/index.cfm?blogid=14&entryid=1845 Help Fight This Patent-Encumbered IETF Standard February 10, 2009 Posted by: Glyn Moody I've written numerous times about the importance of writing to governments about their hare-brained schemes, but this one is rather different. In this case, it's the normally sane Internet Engineering Task Force that wants to do something really daft. The FSF explains: Last January, the Free Software Foundation issued an alert to efforts at the Internet Engineering Task Force (IETF) to sneak a patent-encumbered standard for "TLS authorization" through a back-door approval process that was referenced as "experimental" or "informational". The many comments sent to IETF at that time alerted committee members to this attempt and successfully prevented the standard gaining approval. Unfortunately, attempts to push through this standard have been renewed and become more of a threat. The proposal now at the IETF has a changed status from "experimental" to "proposed standard". This is a throwback to the bad old days of sneaking patents into nominal standards. It is yet another reason why such patents should not be given in the first place. But until such time as the patent offices around the world come to their senses, the only option is to fight patent-encumbered standards on an individual basis. Here are the details for doing so: The FSF is again issuing an alert and request for comments to be sent urgently and prior to the February 11 deadline to ietf at ietf.org. Please include us in your message by a CC to campaigns at fsf.org. You should also expect an automated reply from ietf at ietf.org, which you will need to answer to confirm your original message. Here's what I've sent: I am writing to ask you not to approve the proposed patent-encumbered standard for TLS authorisation. To do so would fly in the face of the IETF's fundamental commitment to openness. It would weaken not just the standard itself, but the IETF's authority in this sphere. --- > http://www.ietf.org/mail-archive/web/ietf-announce/current/msg05617.html Fourth Last Call: draft-housley-tls-authz-extns * To: IETF-Announce * Subject: Fourth Last Call: draft-housley-tls-authz-extns * From: The IESG * Date: Wed, 14 Jan 2009 08:18:20 -0800 (PST) * List-archive: * Reply-to: ietf at ietf.org On June 27, 2006, the IESG approved "Transport Layer Security (TLS) Authorization Extensions," (draft-housley-tls-authz-extns) as a proposed standard. On November 29, 2006, Redphone Security (with whom Mark Brown, a co-author of the draft is affiliated) filed IETF IPR disclosure 767. Because of the timing of the IPR Disclosure, the IESG withdrew its approval of draft-housley-tls-authz-extns. A second IETF Last Call was initiated to determine whether the IETF community still had consensus to publish draft-housley-tls-authz-extns as a proposed standard given the IPR claimed. Consensus to publish as a standards track document was not demonstrated, and the document was withdrawn from IESG consideration. A third IETF Last Call was initiated to determine whether the IETF community had consensus to publish draft-housley-tls-authz-extns as an experimental track RFC with knowledge of the IPR disclosure from Redphone Security. Consensus to publish as experimental was not demonstrated; a substantial segment of the community objected to publication on any track in light of the IPR terms. Since the third Last Call, RedPhone Security filed IETF IPR disclosure 1026. This disclosure statement asserts in part that "the techniques for sending and receiving authorizations defined in TLS Authorizations Extensions (version draft-housley-tls-authz-extns-07.txt) do not infringe upon RedPhone Security's intellectual property rights". The full text of IPR disclosure 1026 is available at: https://datatracker.ietf.org/ipr/1026/ This Last Call is intended to determine whether the IETF community had consensus to publish draft-housley-tls-authz-extns as a proposed standard given IPR Disclosure 1026. The IESG is considering approving this draft as a standards track RFC. The IESG solicits final comments on whether the IETF community has consensus to publish draft-housley-tls-authz-extns as a proposed standard. Comments can be sent to ietf at ietf.org or exceptionally to iesg at ietf.org. Comments should be sent by 2009-02-11. A URL of this Internet-Draft is: http://www.ietf.org/internet-drafts/draft-housley-tls-authz-extns-07.txt _______________________________________________ IETF-Announce mailing list IETF-Announce at ietf.org https://www.ietf.org/mailman/listinfo/ietf-announce -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090211/23c6d0ab/attachment.html From anthony.minessale at gmail.com Wed Feb 11 05:58:56 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 11 Feb 2009 07:58:56 -0600 Subject: [Freeswitch-users] segfault when shoutcast plays mp3 and extension hangs up In-Reply-To: <4992C801.4020906@ewetel.de> References: <4992BFFC.50006@ewetel.de> <4992C801.4020906@ewetel.de> Message-ID: <191c3a030902110558i9702139g6b12aaf15f6d3aac@mail.gmail.com> Why do you need to wait? jira is just a website, just go there and file the bug and attach the file that causes the issue. On Wed, Feb 11, 2009 at 6:43 AM, Helmut Kuper wrote: > Hi Brian, > > of course. Will do it as soon as I have a second FS plattform for > testing SVN trunks. > > thx and regards > Helmut > > > On 11.02.2009 13:16, Brian West wrote: > > The proper location to post this is on Jira... please in the future > > report ALL issues via Jira. They'll get lost if not done this way. > > > > Thanks, > > Brian > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090211/2054327f/attachment-0001.html From helmut.kuper at ewetel.de Wed Feb 11 06:12:18 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Wed, 11 Feb 2009 15:12:18 +0100 Subject: [Freeswitch-users] segfault when shoutcast plays mp3 and extension hangs up In-Reply-To: <191c3a030902110558i9702139g6b12aaf15f6d3aac@mail.gmail.com> References: <4992BFFC.50006@ewetel.de> <4992C801.4020906@ewetel.de> <191c3a030902110558i9702139g6b12aaf15f6d3aac@mail.gmail.com> Message-ID: <4992DCC2.3020702@ewetel.de> Hi Anthony, On 11.02.2009 14:58, Anthony Minessale wrote: > Why do you need to wait? > jira is just a website, just go there and file the bug and attach the > file that causes the issue. Well, there is a question on jira, which makes sure that I have reproduced the bug on SVN trunk ... but I'm not on latest trunk currently. For Jira it's a mandatory field which has to be anwered with yes. Reproduced with SVN Trunk? is required. ^* Reproduced with SVN Trunk?: Have you tried to reproduce this with SVN Trunk? If not STOP, make current and try... if the problem still persists verify you're on the latest SVN rev. as of RIGHT NOW please continue with your issue report. regards Helmut -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090211/57c75022/attachment.html From anthony.minessale at gmail.com Wed Feb 11 06:17:39 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 11 Feb 2009 08:17:39 -0600 Subject: [Freeswitch-users] segfault when shoutcast plays mp3 and extension hangs up In-Reply-To: <4992DCC2.3020702@ewetel.de> References: <4992BFFC.50006@ewetel.de> <4992C801.4020906@ewetel.de> <191c3a030902110558i9702139g6b12aaf15f6d3aac@mail.gmail.com> <4992DCC2.3020702@ewetel.de> Message-ID: <191c3a030902110617n3dcd19cfr204c00792e1e87ed@mail.gmail.com> if the alternative is to post it to the mailing list, you have our permission this one time to answer "not yet" so you have somewhere to attach the bad file so we can reproduce it. On Wed, Feb 11, 2009 at 8:12 AM, Helmut Kuper wrote: > Hi Anthony, > > On 11.02.2009 14:58, Anthony Minessale wrote: > > Why do you need to wait? > jira is just a website, just go there and file the bug and attach the file > that causes the issue. > > Well, there is a question on jira, which makes sure that I have reproduced > the bug on SVN trunk ... but I'm not on latest trunk currently. For Jira > it's a mandatory field which has to be anwered with yes. > > > > Reproduced with SVN Trunk? is required. * Reproduced with SVN Trunk?: Have > you tried to reproduce this with SVN Trunk? If not STOP, make current and > try... if the problem still persists verify you're on the latest SVN rev. as > of RIGHT NOW please continue with your issue report. > > > regards > Helmut > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090211/73631d18/attachment.html From helmut.kuper at ewetel.de Wed Feb 11 06:48:57 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Wed, 11 Feb 2009 15:48:57 +0100 Subject: [Freeswitch-users] segfault when shoutcast plays mp3 and extension hangs up In-Reply-To: <191c3a030902110617n3dcd19cfr204c00792e1e87ed@mail.gmail.com> References: <4992BFFC.50006@ewetel.de> <4992C801.4020906@ewetel.de> <191c3a030902110558i9702139g6b12aaf15f6d3aac@mail.gmail.com> <4992DCC2.3020702@ewetel.de> <191c3a030902110617n3dcd19cfr204c00792e1e87ed@mail.gmail.com> Message-ID: <4992E559.6060506@ewetel.de> Hi Anthony, I quickly have setup a test server with current trunk. So I can now enter there a "YES" into that field. Current trunk crashed as well. But thx for stretching the jira rules a bit :) I attached the file on jira in http://jira.freeswitch.org/browse/MODFORM-24 Can you delete it asap because of copyright reasons, please? regards helmut On 11.02.2009 15:17, Anthony Minessale wrote: > if the alternative is to post it to the mailing list, you have our > permission this one time to answer "not yet" so you have somewhere to > attach the bad file so we can reproduce it. From c_cav_01 at yahoo.com Wed Feb 11 07:23:01 2009 From: c_cav_01 at yahoo.com (Chris) Date: Wed, 11 Feb 2009 07:23:01 -0800 (PST) Subject: [Freeswitch-users] Strange error message In-Reply-To: Message-ID: <226281.90672.qm@web55102.mail.re4.yahoo.com> In that particular instance, no, it's not needed.? It's assigned a numeric value and is just a variable that will disappear when it goes out of scope.? You could deallocate or release the variable if you wished to be tidy. ? In the case of a recordset being returned by the server (an SQL "SELECT") then it will be a recordset object and would need to be closed. ? If you look at your error, your trying to index a number.? e.g curs assigned numeric by the UPDATE returning the number of rows affected, then your trying to look for a :close method or property on a numeric var which isn't going to work and the compiler is interpreting that as an attempt to index it. ? In general, on most SQL servers, SQL insert, update and delete calls return numerics, only select returns recordsets. --- On Tue, 2/10/09, Nik Middleton wrote: From: Nik Middleton Subject: Re: [Freeswitch-users] Strange error message To: freeswitch-users at lists.freeswitch.org Date: Tuesday, February 10, 2009, 4:22 PM So what you?re saying is that I can comment out curs:close()? as it?s not needed? ? Regards, ? ? ? From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Chris Sent: 10 February 2009 21:19 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Strange error message ? Closing the connection will force the server to close any open transactions, as well as release recordsets in local memory that reference the connection. ? However curs is not a recordset.? An SQL update is going to return an integer (rows affected) or boolean depending on the which server you use since no recordset is actually requested. --- On Tue, 2/10/09, Nik Middleton wrote: From: Nik Middleton Subject: [Freeswitch-users] Strange error message To: freeswitch-users at lists.freeswitch.org Date: Tuesday, February 10, 2009, 2:04 PM Hi Guys, ? I?m baffled by this error.? I?m updating the db on call hang-up If I comment out curs:close() no error, but I?m concerned about memory leaks.? Can anyone tell me what FS is complaining about? ? The db gets updated in both cases ? Regards ? ? ? require "luasql.mysql" ? function myHangupHook(s, status, arg) ??????????? freeswitch.consoleLog("info", " : They hung up on US!!!\n"); ??? ??????? env = assert (luasql.mysql()) ??????????? con = assert (env:connect("xxxxl","xxxxxxxxx","pass","192.168.3.205")) ??????????? curs = assert (con:execute"UPDATE callers SET lastcall = 'BOB' WHERE id = 33292") ??????????? curs:close() ??????????? con:close() ??????????? env:close() ??????????? freeswitch.consoleLog("NOTICE", "myHangupHook: " .. status .. "\n"); ??? --error() end ? ? ? ? 2009-02-10 20:53:20 [INFO] switch_cpp.cpp:1086 console_log()? : They hung up on US!!! 2009-02-10 20:53:20 [ERR] mod_lua.cpp:176 lua_parse_and_execute() /usr/local/freeswitch/scripts/helloworld.lua:50: attempt to index global 'curs' (a number value) stack traceback: ??????? /usr/local/freeswitch/scripts/helloworld.lua:50: in function ??????? [C]: in function 'hangup' ??????? /usr/local/freeswitch/scripts/helloworld.lua:70: in main chunk 2009-02-10 20:53:20 [NOTICE] switch_core_session.c:957 switch_core_session_thread() Session 63 (sofia/internal/1001 at 192.168.3.206) Ended_______________________________________________Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org ?_______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090211/5773e5fb/attachment-0001.html From anthony.minessale at gmail.com Wed Feb 11 07:29:52 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 11 Feb 2009 09:29:52 -0600 Subject: [Freeswitch-users] segfault when shoutcast plays mp3 and extension hangs up In-Reply-To: <4992E559.6060506@ewetel.de> References: <4992BFFC.50006@ewetel.de> <4992C801.4020906@ewetel.de> <191c3a030902110558i9702139g6b12aaf15f6d3aac@mail.gmail.com> <4992DCC2.3020702@ewetel.de> <191c3a030902110617n3dcd19cfr204c00792e1e87ed@mail.gmail.com> <4992E559.6060506@ewetel.de> Message-ID: <191c3a030902110729g30b30d46gac0f56205f8d48cf@mail.gmail.com> I am highly suspicious of the ubuntu. you are using a prerelease of gcc that we have already found at least 1 bug. we tried the file on our box and it doesn't even say anything about the file being bad etc...... it plays and hangs up fine even 4 times at once. It would be a big help if you could try to reproduce it on CentOS 5 as a comparison. We have had 3 cases this week where doing so has fixed problems and i don't want to believe it so I would appropriate it if you could test it. On Wed, Feb 11, 2009 at 8:48 AM, Helmut Kuper wrote: > Hi Anthony, > > I quickly have setup a test server with current trunk. So I can now > enter there a "YES" into that field. Current trunk crashed as well. But > thx for stretching the jira rules a bit :) > > I attached the file on jira in > http://jira.freeswitch.org/browse/MODFORM-24 > > Can you delete it asap because of copyright reasons, please? > > regards > helmut > > > > On 11.02.2009 15:17, Anthony Minessale wrote: > > if the alternative is to post it to the mailing list, you have our > > permission this one time to answer "not yet" so you have somewhere to > > attach the bad file so we can reproduce it. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090211/278ae62f/attachment.html From helmut.kuper at ewetel.de Wed Feb 11 07:45:25 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Wed, 11 Feb 2009 16:45:25 +0100 Subject: [Freeswitch-users] mod_openzap stops working after some calls Update In-Reply-To: <4990789B.40405@ewetel.de> References: <498C2DC3.40701@ewetel.de> <191c3a030902060802r424f8a22tb4887dff7c6aead5@mail.gmail.com> <49907327.6010703@ewetel.de> <7CEFE4F8-7461-4DF3-ABE0-C0B818466AD0@jerris.com> <4990789B.40405@ewetel.de> Message-ID: <4992F295.4070809@ewetel.de> Hi Mike, I removed the apr dependency for timestamp and use the function openzap delivers. Works good so far. For unstabilities in openzap and Q931: On my side main problem seems to be, that channels for inbound traffic sometimes not be freed during runtime. Maybe our remote TDM end (AVAYA) simply doesn't release calls as it should, maybe openzap doesn't catch all q931 messages. I added a hack, which forces channels which are in TERMINATE or beyond *AND* this state is older than e.g. 500ms for inbound SETUP to DOWN. Openzap uses those down forced channels. I have the patch in production and I saw the first aid hack serveral times "InUse" channels freeing. "oz dump 1" shows channel states which hung in TERMINATE or above are down serveral minutes later :) This is just a quite brutal hack because it assumes that the remote TDM end is able to recover channels as well. regards helmut From helmut.kuper at ewetel.de Wed Feb 11 07:57:44 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Wed, 11 Feb 2009 16:57:44 +0100 Subject: [Freeswitch-users] segfault when shoutcast plays mp3 and extension hangs up In-Reply-To: <191c3a030902110729g30b30d46gac0f56205f8d48cf@mail.gmail.com> References: <4992BFFC.50006@ewetel.de> <4992C801.4020906@ewetel.de> <191c3a030902110558i9702139g6b12aaf15f6d3aac@mail.gmail.com> <4992DCC2.3020702@ewetel.de> <191c3a030902110617n3dcd19cfr204c00792e1e87ed@mail.gmail.com> <4992E559.6060506@ewetel.de> <191c3a030902110729g30b30d46gac0f56205f8d48cf@mail.gmail.com> Message-ID: <4992F578.6000401@ewetel.de> Hi Anthony, hm ... have to dig for a centos5 machine here ... I have one somewhere ... I will test it there as well. Concerning the prerelease of gcc ... My svn trunk FS was compiled by "gcc version 4.1.2 (Ubuntu 4.1.2-0ubuntu4)". In Jira I entered the gcc version of FS in production. regards Helmut On 11.02.2009 16:29, Anthony Minessale wrote: > I am highly suspicious of the ubuntu. > you are using a prerelease of gcc that we have already found at least > 1 bug. > > we tried the file on our box and it doesn't even say anything about > the file being bad etc...... it plays and hangs up fine even 4 times > at once. > It would be a big help if you could try to reproduce it on CentOS 5 as > a comparison. We have had 3 cases this week where doing so has fixed > problems and i don't want to believe it so I would appropriate it if > you could test it. > From saeedahmad1981 at gmail.com Wed Feb 11 08:31:37 2009 From: saeedahmad1981 at gmail.com (Saeed Ahmed) Date: Wed, 11 Feb 2009 17:31:37 +0100 Subject: [Freeswitch-users] FS + Call Center Solution Message-ID: <6309E7515E4F43159B9564800564B562@SaeedLaptop> Hi List, Is there any open source call center tool available which works with FS? Kind Regards Saeed Ahmed Tariq -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090211/579e5bf4/attachment.html From odermann at googlemail.com Wed Feb 11 08:42:51 2009 From: odermann at googlemail.com (Dennis) Date: Wed, 11 Feb 2009 17:42:51 +0100 Subject: [Freeswitch-users] DTMF: Mute sound for the other side? In-Reply-To: <174C0607-ADBE-4AB2-92C3-588A86B74EAE@jerris.com> References: <5e414ed0902090659p18b970c9y8ba8f1bacda1f3ac@mail.gmail.com> <191c3a030902090725q11c8b3eaka6ee5e8565b1a7ea@mail.gmail.com> <5e414ed0902100658j2656c453mb74d4840bc256063@mail.gmail.com> <174C0607-ADBE-4AB2-92C3-588A86B74EAE@jerris.com> Message-ID: <5e414ed0902110842q49e0e454v6b9401b759fd04f0@mail.gmail.com> that is interesting. we are receiving the dtmf digits over 2833. might it be possible, that we receive 2833 AND inband (we asked our carrier for 2833, because we had problems with inband and fs - and we got it)? is there something we can setup in fs or is it a problem wich only our carrier can solve? 2009/2/10 Michael Jerris : > If your in a conference and your hearing other people hitting dtmf > digits that IS inband, it means that the place upstream that is doing > inband to 2833 conversion is not properly clipping the dtmf, this > probably needs to be fixed on that device. From brian at freeswitch.org Wed Feb 11 08:56:58 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 11 Feb 2009 10:56:58 -0600 Subject: [Freeswitch-users] DTMF: Mute sound for the other side? In-Reply-To: <5e414ed0902110842q49e0e454v6b9401b759fd04f0@mail.gmail.com> References: <5e414ed0902090659p18b970c9y8ba8f1bacda1f3ac@mail.gmail.com> <191c3a030902090725q11c8b3eaka6ee5e8565b1a7ea@mail.gmail.com> <5e414ed0902100658j2656c453mb74d4840bc256063@mail.gmail.com> <174C0607-ADBE-4AB2-92C3-588A86B74EAE@jerris.com> <5e414ed0902110842q49e0e454v6b9401b759fd04f0@mail.gmail.com> Message-ID: <26CE501F-8F06-4E7D-A9E7-24BD34E22DFE@freeswitch.org> Well if they are sending both they are broken. I would call and yell at them and beat them with a cluebat. /b On Feb 11, 2009, at 10:42 AM, Dennis wrote: > that is interesting. we are receiving the dtmf digits over 2833. might > it be possible, that we receive 2833 AND inband (we asked our carrier > for 2833, because we had problems with inband and fs - and we got it)? > > is there something we can setup in fs or is it a problem wich only our > carrier can solve? From odermann at googlemail.com Wed Feb 11 09:14:32 2009 From: odermann at googlemail.com (Dennis) Date: Wed, 11 Feb 2009 18:14:32 +0100 Subject: [Freeswitch-users] DTMF: Mute sound for the other side? In-Reply-To: <26CE501F-8F06-4E7D-A9E7-24BD34E22DFE@freeswitch.org> References: <5e414ed0902090659p18b970c9y8ba8f1bacda1f3ac@mail.gmail.com> <191c3a030902090725q11c8b3eaka6ee5e8565b1a7ea@mail.gmail.com> <5e414ed0902100658j2656c453mb74d4840bc256063@mail.gmail.com> <174C0607-ADBE-4AB2-92C3-588A86B74EAE@jerris.com> <5e414ed0902110842q49e0e454v6b9401b759fd04f0@mail.gmail.com> <26CE501F-8F06-4E7D-A9E7-24BD34E22DFE@freeswitch.org> Message-ID: <5e414ed0902110914u78e13547pbb52bfea588bbc29@mail.gmail.com> i can't tell, if they are sending both, but it seems so. we get 2833 for sure. they were kind enough to give it to us, because inband seems to be quite unreliable over sip. how can in find out, if both are coming and is there a way to "block" inband to test? perhaps we need both: if we bridge an inbound with another ivr on the outbound side, which is not sip and does not understand 2833, we need to pass inband through or something like this. or am i wrong with this? 2009/2/11 Brian West : > Well if they are sending both they are broken. I would call and yell > at them and beat them with a cluebat. > > /b > > On Feb 11, 2009, at 10:42 AM, Dennis wrote: > >> that is interesting. we are receiving the dtmf digits over 2833. might >> it be possible, that we receive 2833 AND inband (we asked our carrier >> for 2833, because we had problems with inband and fs - and we got it)? >> >> is there something we can setup in fs or is it a problem wich only our >> carrier can solve? > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From brian at freeswitch.org Wed Feb 11 09:23:35 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 11 Feb 2009 11:23:35 -0600 Subject: [Freeswitch-users] DTMF: Mute sound for the other side? In-Reply-To: <5e414ed0902110914u78e13547pbb52bfea588bbc29@mail.gmail.com> References: <5e414ed0902090659p18b970c9y8ba8f1bacda1f3ac@mail.gmail.com> <191c3a030902090725q11c8b3eaka6ee5e8565b1a7ea@mail.gmail.com> <5e414ed0902100658j2656c453mb74d4840bc256063@mail.gmail.com> <174C0607-ADBE-4AB2-92C3-588A86B74EAE@jerris.com> <5e414ed0902110842q49e0e454v6b9401b759fd04f0@mail.gmail.com> <26CE501F-8F06-4E7D-A9E7-24BD34E22DFE@freeswitch.org> <5e414ed0902110914u78e13547pbb52bfea588bbc29@mail.gmail.com> Message-ID: <32609C6A-EF32-48BE-9FA1-188BF4E05282@freeswitch.org> turn on the start_dtmf app and dial digits from the outside.. if you get duplicate digits then they are sending both. /b On Feb 11, 2009, at 11:14 AM, Dennis wrote: > i can't tell, if they are sending both, but it seems so. we get 2833 > for sure. they were kind enough to give it to us, because inband seems > to be quite unreliable over sip. > > how can in find out, if both are coming and is there a way to "block" > inband to test? > > perhaps we need both: if we bridge an inbound with another ivr on the > outbound side, which is not sip and does not understand 2833, we need > to pass inband through or something like this. or am i wrong with > this? From msc at freeswitch.org Wed Feb 11 09:43:24 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 11 Feb 2009 09:43:24 -0800 Subject: [Freeswitch-users] mod_openzap stops working after some calls Update In-Reply-To: <4992F295.4070809@ewetel.de> References: <498C2DC3.40701@ewetel.de> <191c3a030902060802r424f8a22tb4887dff7c6aead5@mail.gmail.com> <49907327.6010703@ewetel.de> <7CEFE4F8-7461-4DF3-ABE0-C0B818466AD0@jerris.com> <4990789B.40405@ewetel.de> <4992F295.4070809@ewetel.de> Message-ID: <87f2f3b90902110943q12c550e4qf8e6ecae1ecc6bec@mail.gmail.com> > This is just a quite brutal hack because it assumes that the remote TDM > end is able to recover channels as well. Could you possibly modify your hack to be less brutal? For example, could it send a STATUS ENQ to the far end for the channel in question? Just curious. -MC From anthony.minessale at gmail.com Wed Feb 11 10:05:31 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 11 Feb 2009 12:05:31 -0600 Subject: [Freeswitch-users] mod_openzap stops working after some calls Update In-Reply-To: <87f2f3b90902110943q12c550e4qf8e6ecae1ecc6bec@mail.gmail.com> References: <498C2DC3.40701@ewetel.de> <191c3a030902060802r424f8a22tb4887dff7c6aead5@mail.gmail.com> <49907327.6010703@ewetel.de> <7CEFE4F8-7461-4DF3-ABE0-C0B818466AD0@jerris.com> <4990789B.40405@ewetel.de> <4992F295.4070809@ewetel.de> <87f2f3b90902110943q12c550e4qf8e6ecae1ecc6bec@mail.gmail.com> Message-ID: <191c3a030902111005q661aa154tbcf1e33fb6082055@mail.gmail.com> you should come into irc on irc.freenode.net and join #openzap Stefan Knoblich (stkn) in the channel is doing some work on implementation actual q931 timers which would solve the problem the real way. Maybe you could collaberate with him. On Wed, Feb 11, 2009 at 11:43 AM, Michael Collins wrote: > > This is just a quite brutal hack because it assumes that the remote TDM > > end is able to recover channels as well. > > Could you possibly modify your hack to be less brutal? For example, > could it send a STATUS ENQ to the far end for the channel in question? > Just curious. > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090211/ca25e702/attachment.html From odermann at googlemail.com Wed Feb 11 10:23:00 2009 From: odermann at googlemail.com (Dennis) Date: Wed, 11 Feb 2009 19:23:00 +0100 Subject: [Freeswitch-users] DTMF: Mute sound for the other side? In-Reply-To: <32609C6A-EF32-48BE-9FA1-188BF4E05282@freeswitch.org> References: <5e414ed0902090659p18b970c9y8ba8f1bacda1f3ac@mail.gmail.com> <191c3a030902090725q11c8b3eaka6ee5e8565b1a7ea@mail.gmail.com> <5e414ed0902100658j2656c453mb74d4840bc256063@mail.gmail.com> <174C0607-ADBE-4AB2-92C3-588A86B74EAE@jerris.com> <5e414ed0902110842q49e0e454v6b9401b759fd04f0@mail.gmail.com> <26CE501F-8F06-4E7D-A9E7-24BD34E22DFE@freeswitch.org> <5e414ed0902110914u78e13547pbb52bfea588bbc29@mail.gmail.com> <32609C6A-EF32-48BE-9FA1-188BF4E05282@freeswitch.org> Message-ID: <5e414ed0902111023q6353787ckf9a5af746b97a433@mail.gmail.com> ok, i will try this, but how can it be possible, that inband tones are audible in conference, when we do not even have start_dtmf activated? i just don't understand, why it must be dtmf inband, if the tones are audible and how they can be audible, if start_dtmf is not set. is it, because the carrier just sends them as normal sound, which is played as a tone, without beeing used for dtmf? 2009/2/11 Brian West : > turn on the start_dtmf app and dial digits from the outside.. if you > get duplicate digits then they are sending both. > > /b > > On Feb 11, 2009, at 11:14 AM, Dennis wrote: > >> i can't tell, if they are sending both, but it seems so. we get 2833 >> for sure. they were kind enough to give it to us, because inband seems >> to be quite unreliable over sip. >> >> how can in find out, if both are coming and is there a way to "block" >> inband to test? >> >> perhaps we need both: if we bridge an inbound with another ivr on the >> outbound side, which is not sip and does not understand 2833, we need >> to pass inband through or something like this. or am i wrong with >> this? > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From brian at freeswitch.org Wed Feb 11 10:36:11 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 11 Feb 2009 12:36:11 -0600 Subject: [Freeswitch-users] DTMF: Mute sound for the other side? In-Reply-To: <5e414ed0902111023q6353787ckf9a5af746b97a433@mail.gmail.com> References: <5e414ed0902090659p18b970c9y8ba8f1bacda1f3ac@mail.gmail.com> <191c3a030902090725q11c8b3eaka6ee5e8565b1a7ea@mail.gmail.com> <5e414ed0902100658j2656c453mb74d4840bc256063@mail.gmail.com> <174C0607-ADBE-4AB2-92C3-588A86B74EAE@jerris.com> <5e414ed0902110842q49e0e454v6b9401b759fd04f0@mail.gmail.com> <26CE501F-8F06-4E7D-A9E7-24BD34E22DFE@freeswitch.org> <5e414ed0902110914u78e13547pbb52bfea588bbc29@mail.gmail.com> <32609C6A-EF32-48BE-9FA1-188BF4E05282@freeswitch.org> <5e414ed0902111023q6353787ckf9a5af746b97a433@mail.gmail.com> Message-ID: <73433A3E-7641-4966-8FF1-73E6C62C5D60@freeswitch.org> On Feb 11, 2009, at 12:23 PM, Dennis wrote: > ok, i will try this, but how can it be possible, that inband tones are > audible in conference, when we do not even have start_dtmf activated? They aren't really sending 2833. > > > i just don't understand, why it must be dtmf inband, if the tones are > audible and how they can be audible, if start_dtmf is not set. > is it, because the carrier just sends them as normal sound, which is > played as a tone, without beeing used for dtmf? I bet they don't know how to config their switch to do 2833. /b From odermann at googlemail.com Wed Feb 11 10:37:51 2009 From: odermann at googlemail.com (Dennis) Date: Wed, 11 Feb 2009 19:37:51 +0100 Subject: [Freeswitch-users] Socket-Event on "originate call"? In-Reply-To: <5e414ed0902100716s2c053a48w2174f7587dd0641e@mail.gmail.com> References: <5e414ed0902090716y10f95f71o5760dfdf50d1c5cb@mail.gmail.com> <191c3a030902090727n44eef7d7x83704b66e628081@mail.gmail.com> <5e414ed0902100716s2c053a48w2174f7587dd0641e@mail.gmail.com> Message-ID: <5e414ed0902111037q2a6f4097sebd08cb712f042bf@mail.gmail.com> anthony, did you make the changes with "add {instant_ringback=true} to make ringback not wait for indication to generate ringback" for the described problem? we read something like this out of it, but we can not test it, because we get errors with the latest fs version (switch_ivr.c:674 switch_ivr_park() Cannot park channels that have no read codec.). 2009/2/10 Dennis : > yes, you are right. we are receiving the reply. > > but, we are using socket outbound and manage all calls over this > socket. we also measure the durations (like variable_duration and > variable_billsec) and count all outgoing calls over the socket. > but, if the originate (without an inbound call) will not start the > socket, we can not count up, how many calls failed because of "user > busy" or how long the platform was in use. > > a possible workarround: is it possible to trigger a dialplan over the > cli (like our default dialplan, which starts the socket), so that the > dialplan starts the originates? > > the basic problem for us, that, if we just want to make dialouts, we > are missing the inbound call to start the socket. > > > kind regards > dennis > > > > 2009/2/9 Anthony Minessale : >> when an originate is unsuccessful the failure and the cause code is returned >> as the reply to the originate request. >> >> >> On Mon, Feb 9, 2009 at 9:16 AM, Dennis wrote: >>> >>> hi, >>> >>> i am using socket outbound with fs. >>> >>> if i do an originate over the console, for starting an outbound call >>> without having an inbound call, and send the originate directly to the >>> socket, the socket is first started, if the call is in answer or >>> ringing state. >>> before this, i will not receive any event, because the socket was not >>> started. therefore i will not know, if the target is "busy" (hangup, >>> hangup cause: user busy). >>> >>> it would be very helpful, if the socket would start immediately on an >>> event like "channel originate". >>> >>> >>> thanks for the help >>> dennis > From brian at freeswitch.org Wed Feb 11 11:13:36 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 11 Feb 2009 13:13:36 -0600 Subject: [Freeswitch-users] Socket-Event on "originate call"? In-Reply-To: <5e414ed0902111037q2a6f4097sebd08cb712f042bf@mail.gmail.com> References: <5e414ed0902090716y10f95f71o5760dfdf50d1c5cb@mail.gmail.com> <191c3a030902090727n44eef7d7x83704b66e628081@mail.gmail.com> <5e414ed0902100716s2c053a48w2174f7587dd0641e@mail.gmail.com> <5e414ed0902111037q2a6f4097sebd08cb712f042bf@mail.gmail.com> Message-ID: <397EFED1-22E6-4B78-BDE4-3AE5F4204E5B@freeswitch.org> Try answer or pre_answer before park. /b On Feb 11, 2009, at 12:37 PM, Dennis wrote: > anthony, did you make the changes with "add {instant_ringback=true} to > make ringback not wait for indication to generate ringback" for the > described problem? > > we read something like this out of it, but we can not test it, because > we get errors with the latest fs version (switch_ivr.c:674 > switch_ivr_park() Cannot park channels that have no read codec.). From nik.middleton at noblesolutions.co.uk Wed Feb 11 11:15:34 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Wed, 11 Feb 2009 19:15:34 -0000 Subject: [Freeswitch-users] FS 1.0.2 Crash and burn Message-ID: I have a situation where FS aborts I'm running an lua script with mysql statements First time it runs, on hangup I get [CONSOLE] switch_core_memory.c:374 switch_core_memory_reclaim() Returning 4 recycled memory pool(s) If I run it again, FS exits. Should there be an error log somewhere that explains why FS dies? Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090211/8a185422/attachment.html From brian at freeswitch.org Wed Feb 11 11:17:40 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 11 Feb 2009 13:17:40 -0600 Subject: [Freeswitch-users] FS 1.0.2 Crash and burn In-Reply-To: References: Message-ID: <7EA294D9-389F-4543-8F24-ED4C08600BE7@freeswitch.org> Can you show us what you're doing? /b On Feb 11, 2009, at 1:15 PM, Nik Middleton wrote: > I have a situation where FS aborts > > I?m running an lua script with mysql statements > > First time it runs, on hangup I get > > [CONSOLE] switch_core_memory.c:374 switch_core_memory_reclaim() > Returning 4 recycled memory pool(s) > > If I run it again, FS exits. > > Should there be an error log somewhere that explains why FS dies? > > Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090211/5b672908/attachment-0001.html From nik.middleton at noblesolutions.co.uk Wed Feb 11 11:35:25 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Wed, 11 Feb 2009 19:35:25 -0000 Subject: [Freeswitch-users] FS 1.0.2 Crash and burn In-Reply-To: <7EA294D9-389F-4543-8F24-ED4C08600BE7@freeswitch.org> References: <7EA294D9-389F-4543-8F24-ED4C08600BE7@freeswitch.org> Message-ID: I was running in a screen session, so going back to the console it shows it's a seg fault 2009-02-11 19:27:53 [NOTICE] sofia.c:3090 sofia_handle_sip_i_state() Hangup sofia/internal/1001 at 192.168.3.206 [CS_EXECUTE] [NORMAL_CLEARING] Segmentation fault (core dumped) Seg fault occurs on hangup What seems to be causing the problem is an insert statement. Note I'm using the protected call function to trap on any sql error (script will abort on error otherwise) but even calling it unprotected, the result is the same. function updatecall() query = "INSERT INTO CONTACT phonenum, group values 0771111111111, " .. CALLER ; freeswitch.consoleLog("info", query.."\n"); res = assert (con:execute(query)); if unexpected_condition then error() end end if type == "dtmf" and obj['digit'] == '9' then CALL_STATUS = "ORDER"; pcall(updateDNC); session:streamFile("wait48.wav"); return "break"; end function myHangupHook(s, status, arg) freeswitch.consoleLog("info", " : They hung up on US!!!\n"); con:close() env:close() end ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: 11 February 2009 19:18 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] FS 1.0.2 Crash and burn Can you show us what you're doing? /b On Feb 11, 2009, at 1:15 PM, Nik Middleton wrote: I have a situation where FS aborts I'm running an lua script with mysql statements First time it runs, on hangup I get [CONSOLE] switch_core_memory.c:374 switch_core_memory_reclaim() Returning 4 recycled memory pool(s) If I run it again, FS exits. Should there be an error log somewhere that explains why FS dies? Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090211/ceb3bc31/attachment.html From pbd at suspiria.net Wed Feb 11 11:49:29 2009 From: pbd at suspiria.net (Public Dump) Date: Wed, 11 Feb 2009 20:49:29 +0100 Subject: [Freeswitch-users] Compile Freeswitch 64bit for Windows Message-ID: <13C421883438EB42B9E2C30069FD4AB767BF49AF0A@crushinator.central.local> ... did anybody succeed with this ? The solution for VS2008 does not seem to have a valid 64bit configuration. Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090211/45a1310e/attachment.html From brian at freeswitch.org Wed Feb 11 11:38:14 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 11 Feb 2009 13:38:14 -0600 Subject: [Freeswitch-users] FS 1.0.2 Crash and burn In-Reply-To: References: <7EA294D9-389F-4543-8F24-ED4C08600BE7@freeswitch.org> Message-ID: <75A42DD8-9BA0-46CC-8121-04CE9DB27A27@freeswitch.org> How about getting a backtrace of the core dump and opening a jira? http://wiki.freeswitch.org/wiki/Reporting_Bugs /b On Feb 11, 2009, at 1:35 PM, Nik Middleton wrote: > I was running in a screen session, so going back to the console it > shows it?s a seg fault > > 2009-02-11 19:27:53 [NOTICE] sofia.c:3090 sofia_handle_sip_i_state() > Hangup sofia/internal/1001 at 192.168.3.206 [CS_EXECUTE] > [NORMAL_CLEARING] > Segmentation fault (core dumped) > > Seg fault occurs on hangup > > What seems to be causing the problem is an insert statement. > > Note I?m using the protected call function to trap on any sql error > (script will abort on error otherwise) but even calling it > unprotected, the result is the same. > > function updatecall() > query = "INSERT INTO CONTACT phonenum, group values > 0771111111111, " .. CALLER ; > freeswitch.consoleLog("info", query.."\n"); > res = assert (con:execute(query)); > if unexpected_condition then error() end > > end > > > if type == "dtmf" and obj['digit'] == '9' then > CALL_STATUS = "ORDER"; > pcall(updateDNC); > session:streamFile("wait48.wav"); > return "break"; > end > > > function myHangupHook(s, status, arg) > freeswitch.consoleLog("info", " : They hung up on US!!! > \n"); > con:close() > env:close() > > end -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090211/8450db85/attachment-0001.html From mike at jerris.com Wed Feb 11 12:07:55 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 11 Feb 2009 15:07:55 -0500 Subject: [Freeswitch-users] Compile Freeswitch 64bit for Windows In-Reply-To: <13C421883438EB42B9E2C30069FD4AB767BF49AF0A@crushinator.central.local> References: <13C421883438EB42B9E2C30069FD4AB767BF49AF0A@crushinator.central.local> Message-ID: I don't have a 64 bit windows box/os to get this working. Someone with access to such a box would have to set this up and submit a patch. Mike On Feb 11, 2009, at 2:49 PM, Public Dump wrote: > ? did anybody succeed with this ? The solution for VS2008 does not > seem to have a valid 64bit configuration. > > Regards > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090211/54deb044/attachment.html From odermann at googlemail.com Wed Feb 11 12:11:11 2009 From: odermann at googlemail.com (Dennis) Date: Wed, 11 Feb 2009 21:11:11 +0100 Subject: [Freeswitch-users] Socket-Event on "originate call"? In-Reply-To: <397EFED1-22E6-4B78-BDE4-3AE5F4204E5B@freeswitch.org> References: <5e414ed0902090716y10f95f71o5760dfdf50d1c5cb@mail.gmail.com> <191c3a030902090727n44eef7d7x83704b66e628081@mail.gmail.com> <5e414ed0902100716s2c053a48w2174f7587dd0641e@mail.gmail.com> <5e414ed0902111037q2a6f4097sebd08cb712f042bf@mail.gmail.com> <397EFED1-22E6-4B78-BDE4-3AE5F4204E5B@freeswitch.org> Message-ID: <5e414ed0902111211y3456d5eaqc99baa126cb9dcad@mail.gmail.com> this does not help. we are using socket outbound and everything worked before the changes yesterday. we have the same error with other dialplans. > 2009/2/11 Brian West : > Try answer or pre_answer before park. > > /b > > On Feb 11, 2009, at 12:37 PM, Dennis wrote: > >> anthony, did you make the changes with "add {instant_ringback=true} to >> make ringback not wait for indication to generate ringback" for the >> described problem? >> >> we read something like this out of it, but we can not test it, because >> we get errors with the latest fs version (switch_ivr.c:674 >> switch_ivr_park() Cannot park channels that have no read codec.). From brian at freeswitch.org Wed Feb 11 12:17:30 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 11 Feb 2009 14:17:30 -0600 Subject: [Freeswitch-users] Socket-Event on "originate call"? In-Reply-To: <5e414ed0902111211y3456d5eaqc99baa126cb9dcad@mail.gmail.com> References: <5e414ed0902090716y10f95f71o5760dfdf50d1c5cb@mail.gmail.com> <191c3a030902090727n44eef7d7x83704b66e628081@mail.gmail.com> <5e414ed0902100716s2c053a48w2174f7587dd0641e@mail.gmail.com> <5e414ed0902111037q2a6f4097sebd08cb712f042bf@mail.gmail.com> <397EFED1-22E6-4B78-BDE4-3AE5F4204E5B@freeswitch.org> <5e414ed0902111211y3456d5eaqc99baa126cb9dcad@mail.gmail.com> Message-ID: Please collect the backtrace and report it on Jira. /b On Feb 11, 2009, at 2:11 PM, Dennis wrote: > this does not help. we are using socket outbound and everything worked > before the changes yesterday. > > we have the same error with other dialplans. From nik.middleton at noblesolutions.co.uk Wed Feb 11 12:20:08 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Wed, 11 Feb 2009 20:20:08 -0000 Subject: [Freeswitch-users] FS 1.0.2 Crash and burn In-Reply-To: <75A42DD8-9BA0-46CC-8121-04CE9DB27A27@freeswitch.org> References: <7EA294D9-389F-4543-8F24-ED4C08600BE7@freeswitch.org> <75A42DD8-9BA0-46CC-8121-04CE9DB27A27@freeswitch.org> Message-ID: Where is the core dump written? ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: 11 February 2009 19:38 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] FS 1.0.2 Crash and burn How about getting a backtrace of the core dump and opening a jira? http://wiki.freeswitch.org/wiki/Reporting_Bugs /b On Feb 11, 2009, at 1:35 PM, Nik Middleton wrote: I was running in a screen session, so going back to the console it shows it's a seg fault 2009-02-11 19:27:53 [NOTICE] sofia.c:3090 sofia_handle_sip_i_state() Hangup sofia/internal/1001 at 192.168.3.206 [CS_EXECUTE] [NORMAL_CLEARING] Segmentation fault (core dumped) Seg fault occurs on hangup What seems to be causing the problem is an insert statement. Note I'm using the protected call function to trap on any sql error (script will abort on error otherwise) but even calling it unprotected, the result is the same. function updatecall() query = "INSERT INTO CONTACT phonenum, group values 0771111111111, " .. CALLER ; freeswitch.consoleLog("info", query.."\n"); res = assert (con:execute(query)); if unexpected_condition then error() end end if type == "dtmf" and obj['digit'] == '9' then CALL_STATUS = "ORDER"; pcall(updateDNC); session:streamFile("wait48.wav"); return "break"; end function myHangupHook(s, status, arg) freeswitch.consoleLog("info", " : They hung up on US!!!\n"); con:close() env:close() end -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090211/a52bc528/attachment.html From nik.middleton at noblesolutions.co.uk Wed Feb 11 12:22:01 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Wed, 11 Feb 2009 20:22:01 -0000 Subject: [Freeswitch-users] FS 1.0.2 Crash and burn In-Reply-To: <75A42DD8-9BA0-46CC-8121-04CE9DB27A27@freeswitch.org> References: <7EA294D9-389F-4543-8F24-ED4C08600BE7@freeswitch.org> <75A42DD8-9BA0-46CC-8121-04CE9DB27A27@freeswitch.org> Message-ID: Forget my last, followed the link Regards, ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: 11 February 2009 19:38 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] FS 1.0.2 Crash and burn How about getting a backtrace of the core dump and opening a jira? http://wiki.freeswitch.org/wiki/Reporting_Bugs /b On Feb 11, 2009, at 1:35 PM, Nik Middleton wrote: I was running in a screen session, so going back to the console it shows it's a seg fault 2009-02-11 19:27:53 [NOTICE] sofia.c:3090 sofia_handle_sip_i_state() Hangup sofia/internal/1001 at 192.168.3.206 [CS_EXECUTE] [NORMAL_CLEARING] Segmentation fault (core dumped) Seg fault occurs on hangup What seems to be causing the problem is an insert statement. Note I'm using the protected call function to trap on any sql error (script will abort on error otherwise) but even calling it unprotected, the result is the same. function updatecall() query = "INSERT INTO CONTACT phonenum, group values 0771111111111, " .. CALLER ; freeswitch.consoleLog("info", query.."\n"); res = assert (con:execute(query)); if unexpected_condition then error() end end if type == "dtmf" and obj['digit'] == '9' then CALL_STATUS = "ORDER"; pcall(updateDNC); session:streamFile("wait48.wav"); return "break"; end function myHangupHook(s, status, arg) freeswitch.consoleLog("info", " : They hung up on US!!!\n"); con:close() env:close() end -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090211/6678be19/attachment-0001.html From brian at freeswitch.org Wed Feb 11 12:23:06 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 11 Feb 2009 14:23:06 -0600 Subject: [Freeswitch-users] FS 1.0.2 Crash and burn In-Reply-To: References: <7EA294D9-389F-4543-8F24-ED4C08600BE7@freeswitch.org> <75A42DD8-9BA0-46CC-8121-04CE9DB27A27@freeswitch.org> Message-ID: Try starting it from your /usr/local/freeswitch/bin... ./freeswitch it'll dump in the same folder. /b On Feb 11, 2009, at 2:20 PM, Nik Middleton wrote: > Where is the core dump written? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090211/52f62146/attachment.html From celam at arn.com Wed Feb 11 12:32:08 2009 From: celam at arn.com (Chris Elam) Date: Wed, 11 Feb 2009 15:32:08 -0500 Subject: [Freeswitch-users] Originate call from one ext to another from php? Message-ID: Hi all, I?m just starting playing around with FS and I?ve searched for the answer to what I think is an easy question but I can?t find it. I have FS running, 2 X-lite clients on 2 different computers connected using the preconfigured 1000 and 1001 extenstions. Both can call each other and everything is fine. I?m trying to figure out though how to originate the call from 1000 to 1001 via php. I?m using this script: http://wiki.freeswitch.org/wiki/PHP_Event_Socket Except that I?ve changed this line: $cmd = "api help"; To: $cmd = "api originate sofia/mydomain.com/1000 at 192.168.15.50 &bridge(sofia/mydomain.com/1001 at 192.168.15.50)"; The result I get is : -ERR DESTINATION_OUT_OF_ORDER PS, I literally have ?mydomain.com? in there as it looks like from the wiki this is the default. Any help is much appreciated, thanks all. This email may contain confidential information and is solely for the use of the intended recipient. Any review, distribution, disclosure or other use of this information by anyone other than the intended recipient is prohibited. If you have received this communication in error, please notify the sender immediately and delete this message from your system. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090211/72aae965/attachment.html From brian at freeswitch.org Wed Feb 11 12:48:25 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 11 Feb 2009 14:48:25 -0600 Subject: [Freeswitch-users] Originate call from one ext to another from php? In-Reply-To: References: Message-ID: show me "sofia status", Try changing the @ to a % but I really need to see the sofia status output. /b On Feb 11, 2009, at 2:32 PM, Chris Elam wrote: > $cmd = "api originate sofia/mydomain.com/1000 at 192.168.15.50 &bridge(sofia/mydomain.com/1001 at 192.168.15.50 > )"; > > The result I get is : -ERR DESTINATION_OUT_OF_ORDER -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090211/680faaee/attachment.html From celam at arn.com Wed Feb 11 12:54:15 2009 From: celam at arn.com (Chris Elam) Date: Wed, 11 Feb 2009 15:54:15 -0500 Subject: [Freeswitch-users] Originate call from one ext to another from php? In-Reply-To: Message-ID: The % gives the same error. Here is the sofia status output: API CALL [sofia(status)] output: Name Type Data State ============================================================================ ===================== external profile sip:mod_sofia at myoutsideip:5080 RUNNING (0) example.com gateway sip:joeuser at example.com NOREG internal profile sip:mod_sofia at myinsideip:5060 RUNNING (0) myinsideip alias internal ALIASED internal-ipv6 profile sip:mod_sofia@[::1]:5060 RUNNING (0) default alias internal ALIASED nat alias external ALIASED outbound alias external ALIASED ============================================================================ ===================== 3 profiles 4 aliases On 2/11/09 3:48 PM, "Brian West" wrote: > show me "sofia status", Try changing the @ to a % but I really need to see > the sofia status output. > > /b > > On Feb 11, 2009, at 2:32 PM, Chris Elam wrote: > >> $cmd = "api originate sofia/mydomain.com/1000 at 192.168.15.50 >> &bridge(sofia/mydomain.com/1001 at 192.168.15.50)"; >> >> The result I get is : -ERR DESTINATION_OUT_OF_ORDER >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org This email may contain confidential information and is solely for the use of the intended recipient. Any review, distribution, disclosure or other use of this information by anyone other than the intended recipient is prohibited. If you have received this communication in error, please notify the sender immediately and delete this message from your system. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090211/9d4af77f/attachment.html From brian at freeswitch.org Wed Feb 11 12:59:17 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 11 Feb 2009 14:59:17 -0600 Subject: [Freeswitch-users] Originate call from one ext to another from php? In-Reply-To: References: Message-ID: try sofia/myinsideip/1000 and sofia/myinsideip/1001 I sure hope it doesn't say myinsideip on there and you only tried to hide your IP. /b On Feb 11, 2009, at 2:54 PM, Chris Elam wrote: > The % gives the same error. Here is the sofia status output: > > API CALL [sofia(status)] output: > Name > Type Data State > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > ====================================================================== > external profile sip:mod_sofia at myoutsideip:5080 > RUNNING (0) > example.com gateway sip:joeuser at example.com > NOREG > internal profile sip:mod_sofia at myinsideip:5060 > RUNNING (0) > myinsideip alias > internal ALIASED > internal-ipv6 profile sip:mod_sofia@[:: > 1]:5060 RUNNING (0) > default alias > internal ALIASED > nat alias > external ALIASED > outbound alias > external ALIASED > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > ====================================================================== > 3 profiles 4 aliases From celam at arn.com Wed Feb 11 13:06:04 2009 From: celam at arn.com (Chris Elam) Date: Wed, 11 Feb 2009 16:06:04 -0500 Subject: [Freeswitch-users] Originate call from one ext to another from php? In-Reply-To: Message-ID: That's it, worked perfectly, thanks a bunch! On 2/11/09 3:59 PM, "Brian West" wrote: > try sofia/myinsideip/1000 and sofia/myinsideip/1001 > > I sure hope it doesn't say myinsideip on there and you only tried to > hide your IP. > > /b > > > On Feb 11, 2009, at 2:54 PM, Chris Elam wrote: > >> The % gives the same error. Here is the sofia status output: >> >> API CALL [sofia(status)] output: >> Name >> Type Data State >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> ====================================================================== >> external profile sip:mod_sofia at myoutsideip:5080 >> RUNNING (0) >> example.com gateway sip:joeuser at example.com >> NOREG >> internal profile sip:mod_sofia at myinsideip:5060 >> RUNNING (0) >> myinsideip alias >> internal ALIASED >> internal-ipv6 profile sip:mod_sofia@[:: >> 1]:5060 RUNNING (0) >> default alias >> internal ALIASED >> nat alias >> external ALIASED >> outbound alias >> external ALIASED >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> ====================================================================== >> 3 profiles 4 aliases > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org This email may contain confidential information and is solely for the use of the intended recipient. Any review, distribution, disclosure or other use of this information by anyone other than the intended recipient is prohibited. If you have received this communication in error, please notify the sender immediately and delete this message from your system. From pbd at suspiria.net Wed Feb 11 13:11:11 2009 From: pbd at suspiria.net (Public Dump) Date: Wed, 11 Feb 2009 22:11:11 +0100 Subject: [Freeswitch-users] High CPU load after starting Message-ID: <13C421883438EB42B9E2C30069FD4AB767BF49AF0C@crushinator.central.local> After reading you suggestions I deployed the version from SVN today, the problem persists. Regards Von: Public Dump Gesendet: Dienstag, 10. Februar 2009 19:42 An: 'freeswitch-users at lists.freeswitch.org' Betreff: High CPU load after starting After starting FreeSwitch (1.0.2) on a 4 core server running Windows Server 2008, the CPU load (privileged time/kernel) for one of the cores goes to 50% and stays there. Stoping FreeSwitch stops the load. I have tried to disable all modules but the problem persists. Has anybody seen this problem, can it be fixed ? regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090211/7ef613df/attachment.html From brian at freeswitch.org Wed Feb 11 13:13:32 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 11 Feb 2009 15:13:32 -0600 Subject: [Freeswitch-users] Originate call from one ext to another from php? In-Reply-To: References: Message-ID: remember its sofia/profilename/user%domain or sofia/domain/user the latter requires an alias on the profile for the domain the user registers with. /b On Feb 11, 2009, at 3:06 PM, Chris Elam wrote: > That's it, worked perfectly, thanks a bunch! > > > On 2/11/09 3:59 PM, "Brian West" wrote: > >> try sofia/myinsideip/1000 and sofia/myinsideip/1001 >> >> I sure hope it doesn't say myinsideip on there and you only tried to >> hide your IP. >> >> /b > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090211/97380167/attachment.html From brian at freeswitch.org Wed Feb 11 13:16:16 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 11 Feb 2009 15:16:16 -0600 Subject: [Freeswitch-users] High CPU load after starting In-Reply-To: <13C421883438EB42B9E2C30069FD4AB767BF49AF0C@crushinator.central.local> References: <13C421883438EB42B9E2C30069FD4AB767BF49AF0C@crushinator.central.local> Message-ID: <1A18CBF4-81F9-4E57-AF9D-54B6103B292B@freeswitch.org> Are you sure you rebuilt it clean? Are you doing anything special? Changing any configs? /b On Feb 11, 2009, at 3:11 PM, Public Dump wrote: > After reading you suggestions I deployed the version from SVN today, > the problem persists. > > Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090211/744b3a0a/attachment.html From pbd at suspiria.net Wed Feb 11 13:32:10 2009 From: pbd at suspiria.net (Public Dump) Date: Wed, 11 Feb 2009 22:32:10 +0100 Subject: [Freeswitch-users] Compile Freeswitch 64bit for Windows Message-ID: <13C421883438EB42B9E2C30069FD4AB767BF49AF0D@crushinator.central.local> For compiling x64 code you shouldn't need a 64bit system, but you couldn't run it of course. I don't have a 64 bit windows box/os to get this working. Someone with access to such a box would have to set this up and submit a patch. Mike -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090211/83da0d55/attachment.html From anthony.minessale at gmail.com Wed Feb 11 13:40:07 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 11 Feb 2009 15:40:07 -0600 Subject: [Freeswitch-users] FS 1.0.2 Crash and burn In-Reply-To: References: <7EA294D9-389F-4543-8F24-ED4C08600BE7@freeswitch.org> <75A42DD8-9BA0-46CC-8121-04CE9DB27A27@freeswitch.org> Message-ID: <191c3a030902111340s2a9de96cg2a1cfa8cfddad6b0@mail.gmail.com> and make sure it's svn trunk or at least a daily snapshot and not 1.0.2 On Wed, Feb 11, 2009 at 2:23 PM, Brian West wrote: > Try starting it from your /usr/local/freeswitch/bin... ./freeswitch it'll > dump in the same folder. > /b > > On Feb 11, 2009, at 2:20 PM, Nik Middleton wrote: > > Where is the core dump written? > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090211/e5efa5a8/attachment.html From jacredit at gmail.com Wed Feb 11 14:12:24 2009 From: jacredit at gmail.com (John Hyde) Date: Wed, 11 Feb 2009 14:12:24 -0800 Subject: [Freeswitch-users] does anyone have a working FS / aastra config In-Reply-To: <777d76f40902042206u7c448985i8690c6df4472f3b7@mail.gmail.com> References: <777d76f40902042206u7c448985i8690c6df4472f3b7@mail.gmail.com> Message-ID: <777d76f40902111412t38280ecdjf86f8a761d381414@mail.gmail.com> Figured out the phone was sending packets that were too large, and the receiving system was not reassembling the fragmented packet. This can be fixed on the Aastra by enabling basic codecs: Go to the phone web-UI -- global SIP -- Codec Preference List -- Codec 1 -- change all to basic, save settings and restart the phone. Or in cfg files for aastra set: sip use basic codecs: 1 regards- John On Wed, Feb 4, 2009 at 10:06 PM, John Hyde wrote: > I am having problems getting an Aastra 57i to make calls through FS. the > phone registers fine, but all calls fail. If i use xlite or a nokia sip > phone, i have no problems. > > Here is a packet capture of an attempted call: > > http://pastebin.freeswitch.org/7039 > > notice packet 9, it should have been a SIP INVITE, but it turned out to be > a Fragmented IP protocol > > The phone and FS are both on the same lan subnet, and the phone connects > fine with an asterisk server on the same subnet. > > Is there a known config for aastra phones that I can reference, or does > anyone know why I am having this issue? > > -- john > -- - j -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090211/b760d9d0/attachment-0001.html From pbd at suspiria.net Wed Feb 11 14:21:23 2009 From: pbd at suspiria.net (Public Dump) Date: Wed, 11 Feb 2009 23:21:23 +0100 Subject: [Freeswitch-users] Freeswitch-users Digest, Vol 32, Issue 98 In-Reply-To: References: Message-ID: <13C421883438EB42B9E2C30069FD4AB767BF49AF0E@crushinator.central.local> > Message: 3 > Date: Wed, 11 Feb 2009 15:16:16 -0600 > From: Brian West > Subject: Re: [Freeswitch-users] High CPU load after starting > To: freeswitch-users at lists.freeswitch.org > Message-ID: <1A18CBF4-81F9-4E57-AF9D-54B6103B292B at freeswitch.org> > Content-Type: text/plain; charset="us-ascii" Downloaded tarball, performed SVN update. Nothing special, just building it right out of the box. Already tired disabling all modules and loading all cores. Problem persists. What is really striking, is that the load is all privileged time (kernel), consumes exactly 50% of one core and the core in question is always the same one. The Server is running Hyper-V (Host OS) but this also means it is running under the control of the hypervisor. > Are you sure you rebuilt it clean? Are you doing anything special? > Changing any configs? > > /b > From anthony.minessale at gmail.com Wed Feb 11 14:25:37 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 11 Feb 2009 16:25:37 -0600 Subject: [Freeswitch-users] Freeswitch-users Digest, Vol 32, Issue 98 In-Reply-To: <13C421883438EB42B9E2C30069FD4AB767BF49AF0E@crushinator.central.local> References: <13C421883438EB42B9E2C30069FD4AB767BF49AF0E@crushinator.central.local> Message-ID: <191c3a030902111425p158d29b5gf68d2993d39d5e40@mail.gmail.com> it was a completely clean build? as in compleletely new and/or clean solution? Which tarball was it ? On Wed, Feb 11, 2009 at 4:21 PM, Public Dump wrote: > > Message: 3 > > Date: Wed, 11 Feb 2009 15:16:16 -0600 > > From: Brian West > > Subject: Re: [Freeswitch-users] High CPU load after starting > > To: freeswitch-users at lists.freeswitch.org > > Message-ID: <1A18CBF4-81F9-4E57-AF9D-54B6103B292B at freeswitch.org> > > Content-Type: text/plain; charset="us-ascii" > > Downloaded tarball, performed SVN update. > Nothing special, just building it right out of the box. Already tired > disabling all modules and loading all cores. > Problem persists. > > What is really striking, is that the load is all privileged time (kernel), > consumes exactly 50% of one core and the core in question is always the same > one. > > The Server is running Hyper-V (Host OS) but this also means it is running > under the control of the hypervisor. > > > Are you sure you rebuilt it clean? Are you doing anything special? > > Changing any configs? > > > > /b > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090211/8df90bb6/attachment.html From freeswitch at servercorps.com Wed Feb 11 14:34:47 2009 From: freeswitch at servercorps.com (Nik Martin) Date: Wed, 11 Feb 2009 16:34:47 -0600 Subject: [Freeswitch-users] FreeSWITCH VPSs Message-ID: <92e7d2090902111434l60afd3a8i634b4df688ed4011@mail.gmail.com> If any one needs a FreeSWITCH box with a public, static IP, I can provide them for you at a reasonable cost. I'm building a Virtualization platform for FreeSWITCH hosting, and have the first node complete. These are OpenVZ Virtual Engines with Centos 5.2, a full build environment, and the latest FreeSWITCH trunk. You get 1 static IP, no NAT, 256 mb ram, and 8 gb disk space, with 1 Megabit of bandwidth. Great for VOIP service providers, backup switch, testing, etc. You can contact me directly if you are interested. Nik From brian at freeswitch.org Wed Feb 11 14:38:31 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 11 Feb 2009 16:38:31 -0600 Subject: [Freeswitch-users] FreeSWITCH VPSs In-Reply-To: <92e7d2090902111434l60afd3a8i634b4df688ed4011@mail.gmail.com> References: <92e7d2090902111434l60afd3a8i634b4df688ed4011@mail.gmail.com> Message-ID: <40CC1F17-A4C3-4CA6-8DA0-C8B2A5346CCD@freeswitch.org> Quick note make sure you're 100% 64 bit.. if you need help with that I can show you how on CentOS 5.2 /b On Feb 11, 2009, at 4:34 PM, Nik Martin wrote: > If any one needs a FreeSWITCH box with a public, static IP, I can > provide them for you at a reasonable cost. I'm building a > Virtualization platform for FreeSWITCH hosting, and have the first > node complete. These are OpenVZ Virtual Engines with Centos 5.2, a > full build environment, and the latest FreeSWITCH trunk. You get 1 > static IP, no NAT, 256 mb ram, and 8 gb disk space, with 1 Megabit of > bandwidth. Great for VOIP service providers, backup switch, testing, > etc. You can contact me directly if you are interested. > > Nik From freeswitch at servercorps.com Wed Feb 11 14:47:23 2009 From: freeswitch at servercorps.com (Nik Martin) Date: Wed, 11 Feb 2009 16:47:23 -0600 Subject: [Freeswitch-users] FreeSWITCH VPSs In-Reply-To: <40CC1F17-A4C3-4CA6-8DA0-C8B2A5346CCD@freeswitch.org> References: <92e7d2090902111434l60afd3a8i634b4df688ed4011@mail.gmail.com> <40CC1F17-A4C3-4CA6-8DA0-C8B2A5346CCD@freeswitch.org> Message-ID: <92e7d2090902111447v1cbb00c0l8fc9f1c8b0c1a514@mail.gmail.com> On Wed, Feb 11, 2009 at 4:38 PM, Brian West wrote: > Quick note make sure you're 100% 64 bit.. if you need help with that I > can show you how on CentOS 5.2 > My hardware Node is running 64 bit Centos 5.2, with OpenVZ's kernel: 2.6.18-92.1.18.el5.028stab060.2 #1 SMP Tue Jan 13 11:38:36 MSK 2009 x86_64 x86_64 x86_64 GNU/Linux I think the VE I've built is too, but uname is a bit cryptic: 2.6.18-92.1.18.el5.028stab060.2 #1 SMP Tue Jan 13 11:38:36 MSK 2009 i686 i686 i386 GNU/Linux I can easily change it if FS will run better. Nik > /b > > On Feb 11, 2009, at 4:34 PM, Nik Martin wrote: > >> If any one needs a FreeSWITCH box with a public, static IP, I can >> provide them for you at a reasonable cost. I'm building a >> Virtualization platform for FreeSWITCH hosting, and have the first >> node complete. These are OpenVZ Virtual Engines with Centos 5.2, a >> full build environment, and the latest FreeSWITCH trunk. You get 1 >> static IP, no NAT, 256 mb ram, and 8 gb disk space, with 1 Megabit of >> bandwidth. Great for VOIP service providers, backup switch, testing, >> etc. You can contact me directly if you are interested. >> >> Nik > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From krice at freeswitch.org Wed Feb 11 14:50:23 2009 From: krice at freeswitch.org (Ken Rice) Date: Wed, 11 Feb 2009 16:50:23 -0600 Subject: [Freeswitch-users] FreeSWITCH VPSs In-Reply-To: <92e7d2090902111447v1cbb00c0l8fc9f1c8b0c1a514@mail.gmail.com> Message-ID: Be sure to make the Virt nodes 64bit too... FS works 100% better w/ 64bit! > From: Nik Martin > Reply-To: > Date: Wed, 11 Feb 2009 16:47:23 -0600 > To: > Subject: Re: [Freeswitch-users] FreeSWITCH VPSs > > On Wed, Feb 11, 2009 at 4:38 PM, Brian West wrote: >> Quick note make sure you're 100% 64 bit.. if you need help with that I >> can show you how on CentOS 5.2 >> > > My hardware Node is running 64 bit Centos 5.2, with OpenVZ's kernel: > 2.6.18-92.1.18.el5.028stab060.2 #1 SMP Tue Jan 13 11:38:36 MSK 2009 > x86_64 x86_64 x86_64 GNU/Linux > > I think the VE I've built is too, but uname is a bit cryptic: > 2.6.18-92.1.18.el5.028stab060.2 #1 SMP Tue Jan 13 11:38:36 MSK 2009 > i686 i686 i386 GNU/Linux > > I can easily change it if FS will run better. > > Nik > > > > >> /b >> >> On Feb 11, 2009, at 4:34 PM, Nik Martin wrote: >> >>> If any one needs a FreeSWITCH box with a public, static IP, I can >>> provide them for you at a reasonable cost. I'm building a >>> Virtualization platform for FreeSWITCH hosting, and have the first >>> node complete. These are OpenVZ Virtual Engines with Centos 5.2, a >>> full build environment, and the latest FreeSWITCH trunk. You get 1 >>> static IP, no NAT, 256 mb ram, and 8 gb disk space, with 1 Megabit of >>> bandwidth. Great for VOIP service providers, backup switch, testing, >>> etc. You can contact me directly if you are interested. >>> >>> Nik >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Wed Feb 11 14:54:50 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 11 Feb 2009 16:54:50 -0600 Subject: [Freeswitch-users] FreeSWITCH VPSs In-Reply-To: <92e7d2090902111447v1cbb00c0l8fc9f1c8b0c1a514@mail.gmail.com> References: <92e7d2090902111434l60afd3a8i634b4df688ed4011@mail.gmail.com> <40CC1F17-A4C3-4CA6-8DA0-C8B2A5346CCD@freeswitch.org> <92e7d2090902111447v1cbb00c0l8fc9f1c8b0c1a514@mail.gmail.com> Message-ID: <4285D10F-3B87-47B4-979A-14C3066B14E0@freeswitch.org> Your VE must be 64bit also. http://wiki.openvz.org/Install_OpenVZ_on_a_x86_64_system_Centos-Fedora If you need the set util listed on that page let me know I have a copy of it. /b On Feb 11, 2009, at 4:47 PM, Nik Martin wrote: > I think the VE I've built is too, but uname is a bit cryptic: > 2.6.18-92.1.18.el5.028stab060.2 #1 SMP Tue Jan 13 11:38:36 MSK 2009 > i686 i686 i386 GNU/Linux > > I can easily change it if FS will run better. From nicolas at medularis.com Wed Feb 11 14:55:11 2009 From: nicolas at medularis.com (Nicolas Brenner) Date: Wed, 11 Feb 2009 19:55:11 -0300 Subject: [Freeswitch-users] FreeSWITCH VPSs In-Reply-To: References: <92e7d2090902111447v1cbb00c0l8fc9f1c8b0c1a514@mail.gmail.com> Message-ID: <1b46b4e80902111455x5b2f87f6p109af34fa306fefc@mail.gmail.com> Also be sure to test it right. I had a mediatemple VPS (they use Virtuozzo I think, the paid version of OpenVZ) and FS would not work right, I had multiple problems, then I switched to a real server and all of that went away. FS would compile and run ok, but then calls wouldn't work or sound wouldn't go through... I never investigated what was the real problem, but switching made the difference. Best regards and good luck! Nicolas On Wed, Feb 11, 2009 at 7:50 PM, Ken Rice wrote: > Be sure to make the Virt nodes 64bit too... FS works 100% better w/ 64bit! > > >> From: Nik Martin >> Reply-To: >> Date: Wed, 11 Feb 2009 16:47:23 -0600 >> To: >> Subject: Re: [Freeswitch-users] FreeSWITCH VPSs >> >> On Wed, Feb 11, 2009 at 4:38 PM, Brian West wrote: >>> Quick note make sure you're 100% 64 bit.. if you need help with that I >>> can show you how on CentOS 5.2 >>> >> >> My hardware Node is running 64 bit Centos 5.2, with OpenVZ's kernel: >> 2.6.18-92.1.18.el5.028stab060.2 #1 SMP Tue Jan 13 11:38:36 MSK 2009 >> x86_64 x86_64 x86_64 GNU/Linux >> >> I think the VE I've built is too, but uname is a bit cryptic: >> 2.6.18-92.1.18.el5.028stab060.2 #1 SMP Tue Jan 13 11:38:36 MSK 2009 >> i686 i686 i386 GNU/Linux >> >> I can easily change it if FS will run better. >> >> Nik >> >> >> >> >>> /b >>> >>> On Feb 11, 2009, at 4:34 PM, Nik Martin wrote: >>> >>>> If any one needs a FreeSWITCH box with a public, static IP, I can >>>> provide them for you at a reasonable cost. I'm building a >>>> Virtualization platform for FreeSWITCH hosting, and have the first >>>> node complete. These are OpenVZ Virtual Engines with Centos 5.2, a >>>> full build environment, and the latest FreeSWITCH trunk. You get 1 >>>> static IP, no NAT, 256 mb ram, and 8 gb disk space, with 1 Megabit of >>>> bandwidth. Great for VOIP service providers, backup switch, testing, >>>> etc. You can contact me directly if you are interested. >>>> >>>> Nik >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From brian at freeswitch.org Wed Feb 11 14:57:35 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 11 Feb 2009 16:57:35 -0600 Subject: [Freeswitch-users] FreeSWITCH VPSs In-Reply-To: <1b46b4e80902111455x5b2f87f6p109af34fa306fefc@mail.gmail.com> References: <92e7d2090902111447v1cbb00c0l8fc9f1c8b0c1a514@mail.gmail.com> <1b46b4e80902111455x5b2f87f6p109af34fa306fefc@mail.gmail.com> Message-ID: <01AFA6BD-7B77-43CC-A22B-64E448E82F04@freeswitch.org> It runs fine under OpenVZ pure 64bit... /b On Feb 11, 2009, at 4:55 PM, Nicolas Brenner wrote: > Also be sure to test it right. I had a mediatemple VPS (they use > Virtuozzo I think, the paid version of OpenVZ) and FS would not work > right, I had multiple problems, then I switched to a real server and > all of that went away. FS would compile and run ok, but then calls > wouldn't work or sound wouldn't go through... I never investigated > what was the real problem, but switching made the difference. > > Best regards and good luck! > > Nicolas From freeswitch at servercorps.com Wed Feb 11 15:03:04 2009 From: freeswitch at servercorps.com (Nik Martin) Date: Wed, 11 Feb 2009 17:03:04 -0600 Subject: [Freeswitch-users] FreeSWITCH VPSs In-Reply-To: <1b46b4e80902111455x5b2f87f6p109af34fa306fefc@mail.gmail.com> References: <92e7d2090902111447v1cbb00c0l8fc9f1c8b0c1a514@mail.gmail.com> <1b46b4e80902111455x5b2f87f6p109af34fa306fefc@mail.gmail.com> Message-ID: <92e7d2090902111503s48b82b2apb2050dc8e33b7caa@mail.gmail.com> I've had a VE in (light) production for about 2 weeks, with no issues so far. I'm going to build a pure 64 bit VE container though, and will run in that for a while too. Brian, you sid you have a readme on that? On Wed, Feb 11, 2009 at 4:55 PM, Nicolas Brenner wrote: > Also be sure to test it right. I had a mediatemple VPS (they use > Virtuozzo I think, the paid version of OpenVZ) and FS would not work > right, I had multiple problems, then I switched to a real server and > all of that went away. FS would compile and run ok, but then calls > wouldn't work or sound wouldn't go through... I never investigated > what was the real problem, but switching made the difference. > > Best regards and good luck! > > Nicolas > > On Wed, Feb 11, 2009 at 7:50 PM, Ken Rice wrote: >> Be sure to make the Virt nodes 64bit too... FS works 100% better w/ 64bit! >> >> >>> From: Nik Martin >>> Reply-To: >>> Date: Wed, 11 Feb 2009 16:47:23 -0600 >>> To: >>> Subject: Re: [Freeswitch-users] FreeSWITCH VPSs >>> >>> On Wed, Feb 11, 2009 at 4:38 PM, Brian West wrote: >>>> Quick note make sure you're 100% 64 bit.. if you need help with that I >>>> can show you how on CentOS 5.2 >>>> >>> >>> My hardware Node is running 64 bit Centos 5.2, with OpenVZ's kernel: >>> 2.6.18-92.1.18.el5.028stab060.2 #1 SMP Tue Jan 13 11:38:36 MSK 2009 >>> x86_64 x86_64 x86_64 GNU/Linux >>> >>> I think the VE I've built is too, but uname is a bit cryptic: >>> 2.6.18-92.1.18.el5.028stab060.2 #1 SMP Tue Jan 13 11:38:36 MSK 2009 >>> i686 i686 i386 GNU/Linux >>> >>> I can easily change it if FS will run better. >>> >>> Nik >>> >>> >>> >>> >>>> /b >>>> >>>> On Feb 11, 2009, at 4:34 PM, Nik Martin wrote: >>>> >>>>> If any one needs a FreeSWITCH box with a public, static IP, I can >>>>> provide them for you at a reasonable cost. I'm building a >>>>> Virtualization platform for FreeSWITCH hosting, and have the first >>>>> node complete. These are OpenVZ Virtual Engines with Centos 5.2, a >>>>> full build environment, and the latest FreeSWITCH trunk. You get 1 >>>>> static IP, no NAT, 256 mb ram, and 8 gb disk space, with 1 Megabit of >>>>> bandwidth. Great for VOIP service providers, backup switch, testing, >>>>> etc. You can contact me directly if you are interested. >>>>> >>>>> Nik >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From brian at freeswitch.org Wed Feb 11 15:08:35 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 11 Feb 2009 17:08:35 -0600 Subject: [Freeswitch-users] FreeSWITCH VPSs In-Reply-To: <92e7d2090902111503s48b82b2apb2050dc8e33b7caa@mail.gmail.com> References: <92e7d2090902111447v1cbb00c0l8fc9f1c8b0c1a514@mail.gmail.com> <1b46b4e80902111455x5b2f87f6p109af34fa306fefc@mail.gmail.com> <92e7d2090902111503s48b82b2apb2050dc8e33b7caa@mail.gmail.com> Message-ID: http://wiki.openvz.org/Install_OpenVZ_on_a_x86_64_system_Centos-Fedora and http://linux.carreira.com.pt/ovzutils/setx86_64-0.3.tar.gz Will set it up for 64bit containers and patch everything to work correctly... /b On Feb 11, 2009, at 5:03 PM, Nik Martin wrote: > I've had a VE in (light) production for about 2 weeks, with no issues > so far. I'm going to build a pure 64 bit VE container though, and > will run in that for a while too. Brian, you sid you have a readme on > that? > From steveu at coppice.org Wed Feb 11 15:15:52 2009 From: steveu at coppice.org (Steve Underwood) Date: Thu, 12 Feb 2009 07:15:52 +0800 Subject: [Freeswitch-users] Skype as a path to wideband adoption Message-ID: <49935C28.8090008@coppice.org> Hi all, Consider the following: - FS has recently greatly enhanced its support for wide, wider and widest band telephony - That advantage is of no benefit when interworking with the PSTN - Skype has had wideband since day one, and just got super-ultra-duper-wideband. - FS is acquiring skype connectivity options (2, which hopefully will converge to one best in class option). One obvious conclusion is Skype offers the best possibility for having broad coverage with wideband voice. However, codec issues are certain to degrade quality. Lossy compressed codecs don't transcode well, and the codecs Skype uses are their own. FS is gaining a range of open and royalty free licence wideband codecs, but not the ones Skype uses. Skype choose widely supported narrowband codecs - G.711 and G.729 - but have not used a wideband codec with broad support. So.... the point of this note is "what can we do to optimise things?" Steve From nik.middleton at noblesolutions.co.uk Wed Feb 11 15:36:04 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Wed, 11 Feb 2009 23:36:04 -0000 Subject: [Freeswitch-users] FS 1.0.2 Crash and burn In-Reply-To: References: <7EA294D9-389F-4543-8F24-ED4C08600BE7@freeswitch.org><75A42DD8-9BA0-46CC-8121-04CE9DB27A27@freeswitch.org> Message-ID: I've abandoned LUA. All sorts of problems (DTMF etc). Also reports of memory leaks when using MYSQL driver. Looking on the WIKI, JavaScript seems very well supported; PLUS DTMF works just fine (pulling my hair out on LUA) Guess I'm going to follow the path of least resistance on this one and use JS and ODBC Regards, ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: 11 February 2009 20:23 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] FS 1.0.2 Crash and burn Try starting it from your /usr/local/freeswitch/bin... ./freeswitch it'll dump in the same folder. /b On Feb 11, 2009, at 2:20 PM, Nik Middleton wrote: Where is the core dump written? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090211/b1471e63/attachment.html From nik.middleton at noblesolutions.co.uk Wed Feb 11 15:41:48 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Wed, 11 Feb 2009 23:41:48 -0000 Subject: [Freeswitch-users] Setting outbound callerid using js Message-ID: Hi Guys I'm trying to set the outbound caller-id in js. The params seem to be acceptable, except I'm getting the default +000000000 caller-ID sent. Should the below work with js? session.originate(session,'{accountcode=54321,ignore_early_media=true,or igination_caller_id_number=07630600000,originate_timeout=25}sofia/gatewa y/mygw/01XXXXXXXXXXX'); (this works using lua BTW) regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090211/41695d60/attachment.html From brian at freeswitch.org Wed Feb 11 15:50:07 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 11 Feb 2009 17:50:07 -0600 Subject: [Freeswitch-users] FS 1.0.2 Crash and burn In-Reply-To: References: <7EA294D9-389F-4543-8F24-ED4C08600BE7@freeswitch.org><75A42DD8-9BA0-46CC-8121-04CE9DB27A27@freeswitch.org> Message-ID: <6C1B5368-931F-4FA0-87F1-F0EE011C35DA@freeswitch.org> Lua has known issues with MySQL you must use latest SVN builds of the luasql driver for that to avoid it.. and still its not stellar.. the unixODBC one on the other hand works fine. /b On Feb 11, 2009, at 5:36 PM, Nik Middleton wrote: > I?ve abandoned LUA. > > All sorts of problems (DTMF etc). Also reports of memory leaks when > using MYSQL driver. > > Looking on the WIKI, JavaScript seems very well supported; PLUS DTMF > works just fine (pulling my hair out on LUA) > > Guess I?m going to follow the path of least resistance on this one > and use JS and ODBC > > Regards, > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090211/0e081cdf/attachment-0001.html From freeswitch at servercorps.com Wed Feb 11 16:04:20 2009 From: freeswitch at servercorps.com (Nik Martin) Date: Wed, 11 Feb 2009 18:04:20 -0600 Subject: [Freeswitch-users] FreeSWITCH VPSs In-Reply-To: References: <92e7d2090902111447v1cbb00c0l8fc9f1c8b0c1a514@mail.gmail.com> <1b46b4e80902111455x5b2f87f6p109af34fa306fefc@mail.gmail.com> <92e7d2090902111503s48b82b2apb2050dc8e33b7caa@mail.gmail.com> Message-ID: <92e7d2090902111604y516b54c6ya8832cb1944a8a3a@mail.gmail.com> Great, thanks! Nik On Wed, Feb 11, 2009 at 5:08 PM, Brian West wrote: > http://wiki.openvz.org/Install_OpenVZ_on_a_x86_64_system_Centos-Fedora > and > http://linux.carreira.com.pt/ovzutils/setx86_64-0.3.tar.gz > > Will set it up for 64bit containers and patch everything to work > correctly... > > /b > > On Feb 11, 2009, at 5:03 PM, Nik Martin wrote: > >> I've had a VE in (light) production for about 2 weeks, with no issues >> so far. I'm going to build a pure 64 bit VE container though, and >> will run in that for a while too. Brian, you sid you have a readme on >> that? >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From edpimentl at gmail.com Wed Feb 11 16:08:14 2009 From: edpimentl at gmail.com (EdPimentl) Date: Wed, 11 Feb 2009 19:08:14 -0500 Subject: [Freeswitch-users] FreeSWITCH VPSs In-Reply-To: <92e7d2090902111434l60afd3a8i634b4df688ed4011@mail.gmail.com> References: <92e7d2090902111434l60afd3a8i634b4df688ed4011@mail.gmail.com> Message-ID: <9dc4a1670902111608h4d8ca43eq85800ee902113a1a@mail.gmail.com> 256 MB Ram ..... is this correct?... Does any VoIP provider to use this? -E -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090211/3afe87ef/attachment.html From nik.middleton at noblesolutions.co.uk Wed Feb 11 16:09:53 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Thu, 12 Feb 2009 00:09:53 -0000 Subject: [Freeswitch-users] Call accounting not working as expected Message-ID: I'm having an issue with call accounting If I initiate a call, and it is then transferred to an IVR menu. Person selects 1 to talk to someone. In js else if (data.digit == "5") { if (session.ready()) { var new_session = new Session(); new_session.originate(..... This Second call leg is not accounted for in either CSV or xml logs Am I doing something wrong? In the XML record is shows that I've diverted to the new number, but the time is all bundled with the initial call. This is exactly the same issue in Asterisk, which I was hoping to avoid. In Other words, why isn't a new call record created for the second leg? Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090212/262d461a/attachment.html From chavpaskov at shaw.ca Wed Feb 11 16:12:09 2009 From: chavpaskov at shaw.ca (Tchavdar Paskov) Date: Wed, 11 Feb 2009 16:12:09 -0800 Subject: [Freeswitch-users] How i can trigger action or application in case of sip 302 received Message-ID: Hi, Everybody i was wondering if anybody can give me a hint on how i can set a condition/action in? dial plan in case of SIP 302 being received. Regards Chav? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090211/4105d3cc/attachment.html From brian at freeswitch.org Wed Feb 11 16:12:52 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 11 Feb 2009 18:12:52 -0600 Subject: [Freeswitch-users] FreeSWITCH VPSs In-Reply-To: <9dc4a1670902111608h4d8ca43eq85800ee902113a1a@mail.gmail.com> References: <92e7d2090902111434l60afd3a8i634b4df688ed4011@mail.gmail.com> <9dc4a1670902111608h4d8ca43eq85800ee902113a1a@mail.gmail.com> Message-ID: <28F5713C-838B-409A-897B-F9E40D39B928@freeswitch.org> You can run a small SOHO operation on 256 megs /b On Feb 11, 2009, at 6:08 PM, EdPimentl wrote: > 256 MB Ram ..... is this correct?... Does any VoIP provider to use > this? > -E From brian at freeswitch.org Wed Feb 11 16:14:17 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 11 Feb 2009 18:14:17 -0600 Subject: [Freeswitch-users] Call accounting not working as expected In-Reply-To: References: Message-ID: first off don't use the session.originate var new_session = new Session({var=val}sofia/blah/blah); will do it all for you in one step. Also can you point me to where on the wiki that keeps talking about session.originate? I need to clean them off there. /b On Feb 11, 2009, at 6:09 PM, Nik Middleton wrote: > else if (data.digit == "5") { > if (session.ready()) { > var new_session = new Session(); > new_session.originate(?.. > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090211/c9627e52/attachment.html From brian at freeswitch.org Wed Feb 11 16:16:39 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 11 Feb 2009 18:16:39 -0600 Subject: [Freeswitch-users] How i can trigger action or application in case of sip 302 received In-Reply-To: References: Message-ID: <28FEEE0A-D0EA-4D78-BCB8-DDA0F47A1F1D@freeswitch.org> Please refer to the extension in public.xml and default.xml both will cause a deflect to be done so the 3 leg call gets turned back into a 2 leg call. In some cases it might be desired to do a 3 leg call so you can bill the party that caused the 302 and the original party also. /b On Feb 11, 2009, at 6:12 PM, Tchavdar Paskov wrote: > Hi, Everybody > > i was wondering if anybody can give me a hint on how i can set a > condition/action in dial plan in case of SIP 302 being received. > > Regards > > Chav From chavpaskov at shaw.ca Wed Feb 11 16:21:57 2009 From: chavpaskov at shaw.ca (Tchavdar Paskov) Date: Wed, 11 Feb 2009 16:21:57 -0800 Subject: [Freeswitch-users] How i can trigger action or application in case of sip 302 received In-Reply-To: <28FEEE0A-D0EA-4D78-BCB8-DDA0F47A1F1D@freeswitch.org> References: <28FEEE0A-D0EA-4D78-BCB8-DDA0F47A1F1D@freeswitch.org> Message-ID: Thank you Brian, is there any way to inspect? what exactly is sent in 302 message and if possible? to replace it? or remove it. Regards Chav ----- Original Message ----- From: Brian West Date: Wednesday, February 11, 2009 4:17 pm Subject: Re: [Freeswitch-users] How i can trigger action or application in case of sip 302 received To: freeswitch-users at lists.freeswitch.org > Please refer to the extension in > public.xml? > and default.xml? both will cause a deflect to be done so > the 3 leg? > call gets turned back into a 2 leg call.? In some cases it > might be? > desired to do a 3 leg call so you can bill the party that caused > the? > 302 and the original party also. > > /b > > On Feb 11, 2009, at 6:12 PM, Tchavdar Paskov wrote: > > > Hi, Everybody > > > > i was wondering if anybody can give me a hint on how i can set > a? > > condition/action in? dial plan in case of SIP 302 being > received.> > > Regards > > > > Chav > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090211/7380a6ea/attachment-0001.html From edpimentl at gmail.com Wed Feb 11 16:24:46 2009 From: edpimentl at gmail.com (EdPimentl) Date: Wed, 11 Feb 2009 19:24:46 -0500 Subject: [Freeswitch-users] FreeSWITCH VPSs In-Reply-To: <28F5713C-838B-409A-897B-F9E40D39B928@freeswitch.org> References: <92e7d2090902111434l60afd3a8i634b4df688ed4011@mail.gmail.com> <9dc4a1670902111608h4d8ca43eq85800ee902113a1a@mail.gmail.com> <28F5713C-838B-409A-897B-F9E40D39B928@freeswitch.org> Message-ID: <9dc4a1670902111624s456b7fbeg99c1e078fe8def3d@mail.gmail.com> Soho,,, yes of course... Voip (soho)Service Provider.... not convinced is possible to provide reliable QoS. My .02 cents -E -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090211/d6b45c24/attachment.html From msc at freeswitch.org Wed Feb 11 16:26:26 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 11 Feb 2009 16:26:26 -0800 Subject: [Freeswitch-users] Setting outbound callerid using js In-Reply-To: References: Message-ID: <87f2f3b90902111626x6db5dc61t68a0170a3460b42d@mail.gmail.com> session.originate(session,'{accountcode=54321,ignore_early_media=true,origination_caller_id_number=07630600000,originate_timeout=25}sofia/gateway/mygw/01XXXXXXXXXXX'); > > > > (this works using lua BTW) > hmmmm... how about using "effective_caller_id_number" instead? I think the JavaScript paradigm is a bit different than the Lua/Perl one. Let us know if that works or not. -MC From brian at freeswitch.org Wed Feb 11 16:28:03 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 11 Feb 2009 18:28:03 -0600 Subject: [Freeswitch-users] How i can trigger action or application in case of sip 302 received In-Reply-To: References: <28FEEE0A-D0EA-4D78-BCB8-DDA0F47A1F1D@freeswitch.org> Message-ID: <97D545F6-8972-41CB-8D1B-0865E62B680C@freeswitch.org> Nope its on auto pilot... we don't get passed the 302 from sofia. So what you have there is all you can get at. /b On Feb 11, 2009, at 6:21 PM, Tchavdar Paskov wrote: > Thank you Brian, > is there any way to inspect what exactly is sent in 302 message and > if possible to replace it or remove it. > > Regards > Chav From brian at freeswitch.org Wed Feb 11 16:28:56 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 11 Feb 2009 18:28:56 -0600 Subject: [Freeswitch-users] FreeSWITCH VPSs In-Reply-To: <9dc4a1670902111624s456b7fbeg99c1e078fe8def3d@mail.gmail.com> References: <92e7d2090902111434l60afd3a8i634b4df688ed4011@mail.gmail.com> <9dc4a1670902111608h4d8ca43eq85800ee902113a1a@mail.gmail.com> <28F5713C-838B-409A-897B-F9E40D39B928@freeswitch.org> <9dc4a1670902111624s456b7fbeg99c1e078fe8def3d@mail.gmail.com> Message-ID: <76543A10-18D7-450D-ACA1-E4CAE64F471F@freeswitch.org> Actually you can if you don't overload the machine like most VPS providers do... The advantage with OpenVZ in this case is that you can migrate the running FreeSWITCH instance between hardware nodes and not drop calls at this size. /b On Feb 11, 2009, at 6:24 PM, EdPimentl wrote: > Soho,,, yes of course... > Voip (soho)Service Provider.... not convinced is possible to provide > reliable QoS. > My .02 cents > -E From msc at freeswitch.org Wed Feb 11 16:31:05 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 11 Feb 2009 16:31:05 -0800 Subject: [Freeswitch-users] Call accounting not working as expected In-Reply-To: References: Message-ID: <87f2f3b90902111631q717facfctd4b1771d62f31207@mail.gmail.com> > > > This Second call leg is not accounted for in either CSV or xml logs > > > > Am I doing something wrong? In the XML record is shows that I've diverted > to the new number, but the time is all bundled with the initial call. > > > > This is exactly the same issue in Asterisk, which I was hoping to avoid. In > Other words, why isn't a new call record created for the second leg? Could you pastebin the xml cdr? I'm curious to see if it's anything like the ones I have. -MC From anthony.minessale at gmail.com Wed Feb 11 16:31:57 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 11 Feb 2009 18:31:57 -0600 Subject: [Freeswitch-users] Skype as a path to wideband adoption In-Reply-To: <49935C28.8090008@coppice.org> References: <49935C28.8090008@coppice.org> Message-ID: <191c3a030902111631r143ea868y8ee6164c86f3f071@mail.gmail.com> Good question, Does anybody have any contacts at Skype to open a discussion with them? Should we just call them anyway? They have chosen to interop directly with asterisk which has not completed it's attempt at wideband support. Maybe they are more interested in connecting to the PSTN but it's worth a try to ask them. On Wed, Feb 11, 2009 at 5:15 PM, Steve Underwood wrote: > Hi all, > > Consider the following: > > - FS has recently greatly enhanced its support for wide, wider and > widest band telephony > - That advantage is of no benefit when interworking with the PSTN > - Skype has had wideband since day one, and just got > super-ultra-duper-wideband. > - FS is acquiring skype connectivity options (2, which hopefully > will converge to one best in class option). > > One obvious conclusion is Skype offers the best possibility for having > broad coverage with wideband voice. > However, codec issues are certain to degrade quality. Lossy compressed > codecs don't transcode well, and the > codecs Skype uses are their own. FS is gaining a range of open and > royalty free licence wideband codecs, but > not the ones Skype uses. Skype choose widely supported narrowband codecs > - G.711 and G.729 - but have not used a wideband codec with broad support. > > So.... the point of this note is "what can we do to optimise things?" > > Steve > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090211/1b5b4a5d/attachment.html From nik.middleton at noblesolutions.co.uk Wed Feb 11 16:47:32 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Thu, 12 Feb 2009 00:47:32 -0000 Subject: [Freeswitch-users] Call accounting not working as expected In-Reply-To: References: Message-ID: Thanks, that cured the call accounting However, in the original originate, any ideas why {var=val} is not being processed? Regards, ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: 12 February 2009 00:14 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Call accounting not working as expected first off don't use the session.originate var new_session = new Session({var=val}sofia/blah/blah); will do it all for you in one step. Also can you point me to where on the wiki that keeps talking about session.originate? I need to clean them off there. /b On Feb 11, 2009, at 6:09 PM, Nik Middleton wrote: else if (data.digit == "5") { if (session.ready()) { var new_session = new Session(); new_session.originate(..... -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090212/2648c12d/attachment-0001.html From msc at freeswitch.org Wed Feb 11 16:59:10 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 11 Feb 2009 16:59:10 -0800 Subject: [Freeswitch-users] Skype as a path to wideband adoption In-Reply-To: <191c3a030902111631r143ea868y8ee6164c86f3f071@mail.gmail.com> References: <49935C28.8090008@coppice.org> <191c3a030902111631r143ea868y8ee6164c86f3f071@mail.gmail.com> Message-ID: <87f2f3b90902111659u71330393waf48c93d8564bfc9@mail.gmail.com> On Wed, Feb 11, 2009 at 4:31 PM, Anthony Minessale wrote: > Good question, > > Does anybody have any contacts at Skype to open a discussion with them? > Should we just call them anyway? > > They have chosen to interop directly with asterisk which has not completed > it's attempt at wideband support. > Maybe they are more interested in connecting to the PSTN but it's worth a > try to ask them. > If someone with some clout could call them that would be ideal. If no one cares to then I would be happy to contact them. Volunteers? :) -MC From nik.middleton at noblesolutions.co.uk Wed Feb 11 16:59:42 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Thu, 12 Feb 2009 00:59:42 -0000 Subject: [Freeswitch-users] Call accounting not working as expected In-Reply-To: References: Message-ID: Further to my last. It is kind of being processed, the account code is being set, XML cdr's are created and are correct, but csv cdr's for the account code are not Caller ID is not being set in the A leg but is in the B Leg ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Nik Middleton Sent: 12 February 2009 00:48 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Call accounting not working as expected Thanks, that cured the call accounting However, in the original originate, any ideas why {var=val} is not being processed? Regards, ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: 12 February 2009 00:14 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Call accounting not working as expected first off don't use the session.originate var new_session = new Session({var=val}sofia/blah/blah); will do it all for you in one step. Also can you point me to where on the wiki that keeps talking about session.originate? I need to clean them off there. /b On Feb 11, 2009, at 6:09 PM, Nik Middleton wrote: else if (data.digit == "5") { if (session.ready()) { var new_session = new Session(); new_session.originate(..... -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090212/71b67346/attachment.html From msc at freeswitch.org Wed Feb 11 17:01:16 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 11 Feb 2009 17:01:16 -0800 Subject: [Freeswitch-users] Call accounting not working as expected In-Reply-To: References: Message-ID: <87f2f3b90902111701s3f2f4801jc80f1a192264f203@mail.gmail.com> > However, in the original originate, any ideas why {var=val} is not being > processed? I think Brian's suggestion is the way to go: > first off don't use the session.originate > > > > var new_session = new Session({var=val}sofia/blah/blah); The above syntax is the clean way to do it. > > > > will do it all for you in one step. Also can you point me to where on the > wiki that keeps talking about session.originate? I need to clean them off > there. From msc at freeswitch.org Wed Feb 11 17:09:59 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 11 Feb 2009 17:09:59 -0800 Subject: [Freeswitch-users] Call accounting not working as expected In-Reply-To: References: Message-ID: <87f2f3b90902111709t2c6db8e2jf190b799eeca6ee8@mail.gmail.com> > It is kind of being processed, the account code is being set, XML cdr's are > created and are correct, but csv cdr's for the account code are not > > > > Caller ID is not being set in the A leg but is in the B Leg DING DING DING!!! We have a weener! Okay, that was the key piece of info. Most likely you are logging only the A leg in the CSV CDRs. Go to conf/autoload_configs/cdr_csv.conf.xml and look for these two lines: Most likely you need to use "b" or "ab" depending on your scenario. Try it each way and see how you like the results, then please report back. Thanks! -MC From edpimentl at gmail.com Wed Feb 11 17:22:37 2009 From: edpimentl at gmail.com (EdPimentl) Date: Wed, 11 Feb 2009 20:22:37 -0500 Subject: [Freeswitch-users] FreeSWITCH VPSs In-Reply-To: <76543A10-18D7-450D-ACA1-E4CAE64F471F@freeswitch.org> References: <92e7d2090902111434l60afd3a8i634b4df688ed4011@mail.gmail.com> <9dc4a1670902111608h4d8ca43eq85800ee902113a1a@mail.gmail.com> <28F5713C-838B-409A-897B-F9E40D39B928@freeswitch.org> <9dc4a1670902111624s456b7fbeg99c1e078fe8def3d@mail.gmail.com> <76543A10-18D7-450D-ACA1-E4CAE64F471F@freeswitch.org> Message-ID: <9dc4a1670902111722k6dfba174m302681f902e7ac4c@mail.gmail.com> Thanks and agree 100% and appreciated the added insight. My thinking of service provide grade deployment something along the line of Ken Rice or Michal B. Or a FS / TelcoBridges External service deployment.... Best regards, E -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090211/b551a9db/attachment.html From brian at freeswitch.org Wed Feb 11 17:29:21 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 11 Feb 2009 19:29:21 -0600 Subject: [Freeswitch-users] gateways not hitting right context now? Message-ID: <62B6AA3B-86E2-408D-8D5D-742E3B93A355@freeswitch.org> If you have outbound gateways registering make sure you set the context and extension param on the gateway so it'll go to the right spot. Recent changes made it work much smoother. /b From freeswitch at servercorps.com Wed Feb 11 17:59:50 2009 From: freeswitch at servercorps.com (Nik Martin) Date: Wed, 11 Feb 2009 19:59:50 -0600 Subject: [Freeswitch-users] FreeSWITCH VPSs In-Reply-To: <76543A10-18D7-450D-ACA1-E4CAE64F471F@freeswitch.org> References: <92e7d2090902111434l60afd3a8i634b4df688ed4011@mail.gmail.com> <9dc4a1670902111608h4d8ca43eq85800ee902113a1a@mail.gmail.com> <28F5713C-838B-409A-897B-F9E40D39B928@freeswitch.org> <9dc4a1670902111624s456b7fbeg99c1e078fe8def3d@mail.gmail.com> <76543A10-18D7-450D-ACA1-E4CAE64F471F@freeswitch.org> Message-ID: <92e7d2090902111759g629ca736k879103144d012155@mail.gmail.com> My goal is obviously not to provide carrier grade VOIP switch service, but a platform for sandbox testing of configurations, SOHO VOIP Switches, development of FS addons, backup switch capability, etc. Doing this stuff at home behind NAT and a consumer grade router is one reason Brian, Anthony, Mike, et al. are half crazy. I run my company's phone switch in a 32 bit OpenVZ VE with 256 Mb ram, and have no issues. When I goof around trying to transcode between 8 and 16 bit codecs and whatnot, sure it gets tight, but FS on an idle system keeps 23 mb of of ram resident, and rarely if ever hits a 256 mb bean counter (limit). Also, these limits are not hard, I just know what my hardware has, and am trying to offer as much value as I can for what I have in the systems. 128 Gb of ECC ram and Quad/Quad Core zeons are still pretty pricey! Nik On Wed, Feb 11, 2009 at 6:28 PM, Brian West wrote: > Actually you can if you don't overload the machine like most VPS > providers do... The advantage with OpenVZ in this case is that you can > migrate the running FreeSWITCH instance between hardware nodes and not > drop calls at this size. > > /b > > On Feb 11, 2009, at 6:24 PM, EdPimentl wrote: > >> Soho,,, yes of course... >> Voip (soho)Service Provider.... not convinced is possible to provide >> reliable QoS. >> My .02 cents >> -E > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From pbd at suspiria.net Wed Feb 11 17:59:56 2009 From: pbd at suspiria.net (Public Dump) Date: Thu, 12 Feb 2009 02:59:56 +0100 Subject: [Freeswitch-users] High CPU load after starting In-Reply-To: References: Message-ID: <13C421883438EB42B9E2C30069FD4AB767BF49AF13@crushinator.central.local> http://files.freeswitch.org/freeswitch-snapshot.tar.gz Extracted into empty directory, SVN update, compile. > it was a completely clean build? > as in compleletely new and/or clean solution? > > Which tarball was it ? > From brian at freeswitch.org Wed Feb 11 18:10:58 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 11 Feb 2009 20:10:58 -0600 Subject: [Freeswitch-users] High CPU load after starting In-Reply-To: <13C421883438EB42B9E2C30069FD4AB767BF49AF13@crushinator.central.local> References: <13C421883438EB42B9E2C30069FD4AB767BF49AF13@crushinator.central.local> Message-ID: <2D54170F-8539-4EAB-9297-8331DAA5E512@freeswitch.org> OK does it work now? We have tested this on various windows installs among the team here and not seeing this issue... it was a known issue back in Nov. or Dec. but thats long been fixed. /b On Feb 11, 2009, at 7:59 PM, Public Dump wrote: > http://files.freeswitch.org/freeswitch-snapshot.tar.gz > > Extracted into empty directory, SVN update, compile. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090211/a9abd53a/attachment.html From red.rain.seven at gmail.com Wed Feb 11 16:37:03 2009 From: red.rain.seven at gmail.com (Henry Huang) Date: Wed, 11 Feb 2009 16:37:03 -0800 Subject: [Freeswitch-users] FreeSWITCH VPSs In-Reply-To: <92e7d2090902111434l60afd3a8i634b4df688ed4011@mail.gmail.com> References: <92e7d2090902111434l60afd3a8i634b4df688ed4011@mail.gmail.com> Message-ID: <59ad9ca10902111637n1459d3cep5bc73dffe6bb474f@mail.gmail.com> Brian: I am also running my freeswitch on my own openVZ containers. Just how do you verify if the freeswitch is compiled as 64bit? I would assume if I compile it under a 64bit container, I would automatically get a 64bit freeswitch right? On Wed, Feb 11, 2009 at 2:34 PM, Nik Martin wrote: > If any one needs a FreeSWITCH box with a public, static IP, I can > provide them for you at a reasonable cost. I'm building a > Virtualization platform for FreeSWITCH hosting, and have the first > node complete. These are OpenVZ Virtual Engines with Centos 5.2, a > full build environment, and the latest FreeSWITCH trunk. You get 1 > static IP, no NAT, 256 mb ram, and 8 gb disk space, with 1 Megabit of > bandwidth. Great for VOIP service providers, backup switch, testing, > etc. You can contact me directly if you are interested. > > Nik > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Henry Huang UniC Solution - Communication Unified -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090211/8e3d8474/attachment.html From brian at freeswitch.org Wed Feb 11 18:55:36 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 11 Feb 2009 20:55:36 -0600 Subject: [Freeswitch-users] FreeSWITCH VPSs In-Reply-To: <59ad9ca10902111637n1459d3cep5bc73dffe6bb474f@mail.gmail.com> References: <92e7d2090902111434l60afd3a8i634b4df688ed4011@mail.gmail.com> <59ad9ca10902111637n1459d3cep5bc73dffe6bb474f@mail.gmail.com> Message-ID: ding ding ding .. yep! "file /usr/local/freeswitch/bin/freeswitch" will also confirm /b On Feb 11, 2009, at 6:37 PM, Henry Huang wrote: > Brian: > > I am also running my freeswitch on my own openVZ containers. Just > how do you verify if the freeswitch is compiled as 64bit? I would > assume if I compile it under a 64bit container, I would > automatically get a 64bit freeswitch right? From anthony.minessale at gmail.com Wed Feb 11 20:25:55 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 11 Feb 2009 22:25:55 -0600 Subject: [Freeswitch-users] FS 1.0.2 Crash and burn In-Reply-To: <6C1B5368-931F-4FA0-87F1-F0EE011C35DA@freeswitch.org> References: <7EA294D9-389F-4543-8F24-ED4C08600BE7@freeswitch.org> <75A42DD8-9BA0-46CC-8121-04CE9DB27A27@freeswitch.org> <6C1B5368-931F-4FA0-87F1-F0EE011C35DA@freeswitch.org> Message-ID: <191c3a030902112025l401c759dife416cf34d51d134@mail.gmail.com> There is always C, it's actually considered a high level language by many ;) On Wed, Feb 11, 2009 at 5:50 PM, Brian West wrote: > Lua has known issues with MySQL you must use latest SVN builds of the > luasql driver for that to avoid it.. and still its not stellar.. the > unixODBC one on the other hand works fine. > /b > > On Feb 11, 2009, at 5:36 PM, Nik Middleton wrote: > > I've abandoned LUA. > > All sorts of problems (DTMF etc). Also reports of memory leaks when using > MYSQL driver. > > Looking on the WIKI, JavaScript seems very well supported; PLUS DTMF works > just fine (pulling my hair out on LUA) > > Guess I'm going to follow the path of least resistance on this one and use > JS and ODBC > > Regards, > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090211/d102432c/attachment.html From pauld at versafon.com Wed Feb 11 20:38:48 2009 From: pauld at versafon.com (pauld) Date: Wed, 11 Feb 2009 23:38:48 -0500 Subject: [Freeswitch-users] Cannot choose Cepstral voice from dialplan In-Reply-To: <87f2f3b90902090904i72e37241hb5d4808eb31fcc18@mail.gmail.com> References: <498E67BF.3060207@versafon.com> <8B964172-6EEB-4B5B-B4B1-D35BFEAEB460@freeswitch.org> <498F6751.2000701@versafon.com> <87f2f3b90902090904i72e37241hb5d4808eb31fcc18@mail.gmail.com> Message-ID: <4993A7D8.1090004@versafon.com> The issue was resolved by creating symlinks to cepstral libs in FS lib directory. I tried that on 1.0.3, but most probably it would work on 1.0.2 as well. Thanks for help. BTW, without that FS would do a core dump (seg fault) on shutdown after TTS was invoked at least once. Looking at FS logs I see "TRANSCODING_NECESSARY" when executing dynamic text even with 8 kHz voice. Why would that be? Looks like it's PCMU/8000 what it's transcoding to what? Michael Collins wrote: > On Sun, Feb 8, 2009 at 3:14 PM, pauld wrote: > >> The libs are there with correct symlinks, see below. I tested both >> voices directly via swift command, works fine. >> Any other ideas? >> It's Cepstral 5.1, FS 1.0.2. >> >> > > Well, first I recommend getting on latest trunk if that's at all > possible for you. The devs have made a ton of improvements in the last > five weeks. Second, this might actually be an issue with FS looking in > its own lib directory for these .so files. Try a symlink from > /usr/local/freeswitch/lib to your /opt/swift/lib (or whatever the name > is) dir for each .so file. However, I think Raymond is correct - some > weirdness has been reported by some Cepstral users on 5.1. We'd > definitely like to hear about your experiences if and when you get it > running. > > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From brian at freeswitch.org Wed Feb 11 20:43:53 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 11 Feb 2009 22:43:53 -0600 Subject: [Freeswitch-users] Cannot choose Cepstral voice from dialplan In-Reply-To: <4993A7D8.1090004@versafon.com> References: <498E67BF.3060207@versafon.com> <8B964172-6EEB-4B5B-B4B1-D35BFEAEB460@freeswitch.org> <498F6751.2000701@versafon.com> <87f2f3b90902090904i72e37241hb5d4808eb31fcc18@mail.gmail.com> <4993A7D8.1090004@versafon.com> Message-ID: <10703C0D-0BA2-4FB3-B669-B87BF93912E0@freeswitch.org> This is normal. Are you using 5.0? can you include examples of how you're doing this? /b On Feb 11, 2009, at 10:38 PM, pauld wrote: > "TRANSCODING_NECESSARY" From anthony.minessale at gmail.com Wed Feb 11 20:48:26 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 11 Feb 2009 22:48:26 -0600 Subject: [Freeswitch-users] FreeSWITCH VPSs In-Reply-To: <92e7d2090902111759g629ca736k879103144d012155@mail.gmail.com> References: <92e7d2090902111434l60afd3a8i634b4df688ed4011@mail.gmail.com> <9dc4a1670902111608h4d8ca43eq85800ee902113a1a@mail.gmail.com> <28F5713C-838B-409A-897B-F9E40D39B928@freeswitch.org> <9dc4a1670902111624s456b7fbeg99c1e078fe8def3d@mail.gmail.com> <76543A10-18D7-450D-ACA1-E4CAE64F471F@freeswitch.org> <92e7d2090902111759g629ca736k879103144d012155@mail.gmail.com> Message-ID: <191c3a030902112048l1d362f79nf2d0fbb65f2c4984@mail.gmail.com> cool, However.... you say... "but a platform for sandbox testing of configurations, SOHO VOIP Switches, development of FS addons, backup switch capability, etc. Doing this stuff at home behind NAT and a consumer grade router is one reason Brian, Anthony, Mike, et al. are half crazy. " First of all, you have no idea on what platform and where we do our development. If you only saw how many terminals to random servers spanning the globe we have open...... You have the NAT part right, if you knew the hours it took from behind NAT to get all the code right for SIP interop or even had to deal with half the bullshit it takes to get SIP working, you'd be in the madhouse so don't you dare try using us as an ad slogan. Secondly, we are not half crazy, we are completely crazy and most of it comes from spending all day on this list tending to your never-ending threads while trying to help the other people in the community who actually give something back. If you want to use our list to advertise this service maybe you should find a way to contribute to the project rather than constantly asking for help in 5 separate emails in one day followed by another thread trying to sell something. A reasonable cost would be FREE just like everything else around here. On Wed, Feb 11, 2009 at 7:59 PM, Nik Martin wrote: > My goal is obviously not to provide carrier grade VOIP switch service, > but a platform for sandbox testing of configurations, SOHO VOIP > Switches, development of FS addons, backup switch capability, etc. > Doing this stuff at home behind NAT and a consumer grade router is one > reason Brian, Anthony, Mike, et al. are half crazy. I run my > company's phone switch in a 32 bit OpenVZ VE with 256 Mb ram, and have > no issues. When I goof around trying to transcode between 8 and 16 > bit codecs and whatnot, sure it gets tight, but FS on an idle system > keeps 23 mb of of ram resident, and rarely if ever hits a 256 mb bean > counter (limit). > > Also, these limits are not hard, I just know what my hardware has, and > am trying to offer as much value as I can for what I have in the > systems. 128 Gb of ECC ram and Quad/Quad Core zeons are still pretty > pricey! > > Nik > > > > > On Wed, Feb 11, 2009 at 6:28 PM, Brian West wrote: > > Actually you can if you don't overload the machine like most VPS > > providers do... The advantage with OpenVZ in this case is that you can > > migrate the running FreeSWITCH instance between hardware nodes and not > > drop calls at this size. > > > > /b > > > > On Feb 11, 2009, at 6:24 PM, EdPimentl wrote: > > > >> Soho,,, yes of course... > >> Voip (soho)Service Provider.... not convinced is possible to provide > >> reliable QoS. > >> My .02 cents > >> -E > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090211/123eae22/attachment-0001.html From odermann at googlemail.com Wed Feb 11 23:39:30 2009 From: odermann at googlemail.com (Dennis) Date: Thu, 12 Feb 2009 08:39:30 +0100 Subject: [Freeswitch-users] Socket-Event on "originate call"? In-Reply-To: References: <5e414ed0902090716y10f95f71o5760dfdf50d1c5cb@mail.gmail.com> <191c3a030902090727n44eef7d7x83704b66e628081@mail.gmail.com> <5e414ed0902100716s2c053a48w2174f7587dd0641e@mail.gmail.com> <5e414ed0902111037q2a6f4097sebd08cb712f042bf@mail.gmail.com> <397EFED1-22E6-4B78-BDE4-3AE5F4204E5B@freeswitch.org> <5e414ed0902111211y3456d5eaqc99baa126cb9dcad@mail.gmail.com> Message-ID: <5e414ed0902112339o6c1c499cxe30d0e255d6756f0@mail.gmail.com> the problem is fixed in the latest version of fs - at least it is working as before without any errors. but there is still the question, if the changes where made because of our problem with the not starting socket!? we can see in the cli, that the var is set, but it does not change anything regarding our problem. 2009/2/11 Brian West : > Please collect the backtrace and report it on Jira. > > /b > > On Feb 11, 2009, at 2:11 PM, Dennis wrote: > >> this does not help. we are using socket outbound and everything worked >> before the changes yesterday. >> >> we have the same error with other dialplans. From helmut.kuper at ewetel.de Thu Feb 12 00:07:38 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Thu, 12 Feb 2009 09:07:38 +0100 Subject: [Freeswitch-users] segfault when shoutcast plays mp3 and extension hangs up In-Reply-To: <191c3a030902110729g30b30d46gac0f56205f8d48cf@mail.gmail.com> References: <4992BFFC.50006@ewetel.de> <4992C801.4020906@ewetel.de> <191c3a030902110558i9702139g6b12aaf15f6d3aac@mail.gmail.com> <4992DCC2.3020702@ewetel.de> <191c3a030902110617n3dcd19cfr204c00792e1e87ed@mail.gmail.com> <4992E559.6060506@ewetel.de> <191c3a030902110729g30b30d46gac0f56205f8d48cf@mail.gmail.com> Message-ID: <4993D8CA.1010602@ewetel.de> Hi Anthony, hm... on centos5 it works fine. No problems, no warning, no crash. regards Helmut On 11.02.2009 16:29, Anthony Minessale wrote: > I am highly suspicious of the ubuntu. > you are using a prerelease of gcc that we have already found at least > 1 bug. > > we tried the file on our box and it doesn't even say anything about > the file being bad etc...... it plays and hangs up fine even 4 times > at once. > It would be a big help if you could try to reproduce it on CentOS 5 as > a comparison. We have had 3 cases this week where doing so has fixed > problems and i don't want to believe it so I would appropriate it if > you could test it. From helmut.kuper at ewetel.de Thu Feb 12 00:34:21 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Thu, 12 Feb 2009 09:34:21 +0100 Subject: [Freeswitch-users] Question: SIP BYE authentication In-Reply-To: <4992C772.4090906@ewetel.de> References: <4992C772.4090906@ewetel.de> Message-ID: <4993DF0D.9000403@ewetel.de> Hi, any ideas how to get FS's BYEs authenticated ? On 11.02.2009 13:41, Helmut Kuper wrote: > Hello, > > my FS is connected to my SIP-DDI softswitch, which requires all SIP > requests sent by a registered SIP account to be authenticated. I found > that when FS sends a BYE FreeSWITCH ignores the authentication > challenge (SIP/2.0 407) received from proxy and simply terminates the > session. > > Is there a way to configure FS in that way that it react on auth > challenges for BYEs ? > > regards > Helmut > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > From thomas.mangin at exa-networks.co.uk Thu Feb 12 00:59:33 2009 From: thomas.mangin at exa-networks.co.uk (Thomas Mangin) Date: Thu, 12 Feb 2009 08:59:33 +0000 Subject: [Freeswitch-users] FreeSWITCH VPSs In-Reply-To: <4285D10F-3B87-47B4-979A-14C3066B14E0@freeswitch.org> References: <92e7d2090902111434l60afd3a8i634b4df688ed4011@mail.gmail.com> <40CC1F17-A4C3-4CA6-8DA0-C8B2A5346CCD@freeswitch.org> <92e7d2090902111447v1cbb00c0l8fc9f1c8b0c1a514@mail.gmail.com> <4285D10F-3B87-47B4-979A-14C3066B14E0@freeswitch.org> Message-ID: <9C725DB2-CD87-49A3-998F-74B1A9030708@exa-networks.co.uk> Hello Brian / Everyone, Like Nik and Nicolas, I created a openvz box to test things in a 'near production' environment. The box does only take 'test' calls, ie it never saw more that a few calls at a time. The design was 100% openser/opensips/kamailio but I since replaced the pstn gateways with FS and it has been working perfectly and I am not looking back. I am now planning to use freeswitch as the registrar/voicemail/media servers to only keep openser as a proxy inbound proxy (as it is possible to program it to assign a RTP proxies topologically near the caller and you can use it to fix some really broken sip packets - like LLU operators cheap DSL routers badly NAT fixing the contact header). Following this thread I am wondering if I should/could expect some issues with my setup which is 32 bits or if your comments are only related to the performance/behaviour of FS once under load (in which case I need not to worry). voip-master:~# uname -a Linux voip-master 2.6.18-ovz-028stab053.5-smp #1 SMP Sat Mar 1 12:19:31 CET 2008 i686 GNU/Linux voip-master:~# vzlist VEID NPROC STATUS IP_ADDR HOSTNAME 1001 24 running A.B.C.A proxy1.sip (openser phone outbound proxy - accept REGISTER - range locked) 1002 24 running A.B.C.B in1.sip (openser incoming calls from the net - enum, no REGISTER - open) 1003 19 running A.B.C.C out1.sip (openser outgoing calls to the net - enum - to be FS) 1004 4 running A.B.C.D rtp1.nat (rtpproxy nat) 1005 3 running A.B.C.E media1 (was sems for voicemail/media) 1006 29 running A.B.C.F database1 1107 23 running A.B.C.G registrar1.sip (openser) 1108 26 running A.B.C.H registrar2.sip (FS) 1109 8 running A.B.C.I ns1(auth DNS for the tested zone with ENUM info) 1110 8 running A.B.C.J internal1.cache (cache with internal ENUM routing) 1111 8 running A.B.C.K external1.cache (normal DNS cache) 1112 21 running A.B.C.L pstn-out-1 (FS gateway out to pstn) 1113 21 running A.B.C.M pstn-in-1 (FS gateway in from pstn) voip-master:~# cat /proc/cpuinfo | grep model model : 4 model name : Intel(R) Xeon(TM) CPU 3.00GHz model : 4 model name : Intel(R) Xeon(TM) CPU 3.00GHz voip-master:~# free -m total used free shared buffers cached Mem: 2023 1902 120 0 431 1009 -/+ buffers/cache: 460 1562 Swap: 2588 3 2585 yep, memory is short :p) Regards, Thomas On 11 Feb 2009, at 22:54, Brian West wrote: > Your VE must be 64bit also. > > http://wiki.openvz.org/Install_OpenVZ_on_a_x86_64_system_Centos-Fedora > > If you need the set util listed on that page let me know I have a > copy of it. > > /b > > On Feb 11, 2009, at 4:47 PM, Nik Martin wrote: > >> I think the VE I've built is too, but uname is a bit cryptic: >> 2.6.18-92.1.18.el5.028stab060.2 #1 SMP Tue Jan 13 11:38:36 MSK 2009 >> i686 i686 i386 GNU/Linux >> >> I can easily change it if FS will run better. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From pbd at suspiria.net Thu Feb 12 01:23:18 2009 From: pbd at suspiria.net (Public Dump) Date: Thu, 12 Feb 2009 10:23:18 +0100 Subject: [Freeswitch-users] High CPU load after starting (Brian West) In-Reply-To: References: Message-ID: <13C421883438EB42B9E2C30069FD4AB767BF49AF15@crushinator.central.local> > OK does it work now? We have tested this on various windows installs > among the team here and not seeing this issue... it was a known issue > back in Nov. or Dec. but thats long been fixed. No, the problem is still there. I have tested it on a Core AMD 32bit AMD machine = everything is fine. On a 64bit 4 Core Intel Xeon machine = Problem is there. From r.pankratz at fh-wolfenbuettel.de Thu Feb 12 03:05:46 2009 From: r.pankratz at fh-wolfenbuettel.de (Rene Pankratz) Date: Thu, 12 Feb 2009 12:05:46 +0100 Subject: [Freeswitch-users] Change registration information for SIP-Registrar via XML/RPC? Message-ID: <4994028A.9090005@fh-wolfenbuettel.de> Hello, we want to use mod_pa as a softphone, that registers to a SIPregistrar. But the username and password need to be changed over time without restarting freeswitch. Currently we are using XML/RPC to control the call functions. So it would be best (if possible) to use it also for changing registration information. Is there any way to do this? Thanks in advance Ren? From helmut.kuper at ewetel.de Thu Feb 12 05:03:32 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Thu, 12 Feb 2009 14:03:32 +0100 Subject: [Freeswitch-users] mod_openzap stops working after some calls Update In-Reply-To: <87f2f3b90902110943q12c550e4qf8e6ecae1ecc6bec@mail.gmail.com> References: <498C2DC3.40701@ewetel.de> <191c3a030902060802r424f8a22tb4887dff7c6aead5@mail.gmail.com> <49907327.6010703@ewetel.de> <7CEFE4F8-7461-4DF3-ABE0-C0B818466AD0@jerris.com> <4990789B.40405@ewetel.de> <4992F295.4070809@ewetel.de> <87f2f3b90902110943q12c550e4qf8e6ecae1ecc6bec@mail.gmail.com> Message-ID: <49941E24.2070002@ewetel.de> Hi Mike, at least for incoming calls this shouldn't be too brutal, cause far end seems to know that the channel should be free otherwise it never would allocate it. By now the hack works at least for me quite good. Nobody from AVAYA side moaned about it, yet. But I have to wait one or two further days to be sure ... I guess I have to talk to stkn in irc to get an idea how long I have to use it. regards helmut On 11.02.2009 18:43, Michael Collins wrote: >> This is just a quite brutal hack because it assumes that the remote TDM >> end is able to recover channels as well. >> > > Could you possibly modify your hack to be less brutal? For example, > could it send a STATUS ENQ to the far end for the channel in question? > Just curious. > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090212/02632a8c/attachment.html From Thomas.Sluschny at siemens.com Thu Feb 12 06:00:51 2009 From: Thomas.Sluschny at siemens.com (Sluschny, Thomas) Date: Thu, 12 Feb 2009 15:00:51 +0100 Subject: [Freeswitch-users] stream a file multicast with mod_esf Message-ID: Hi, i want to stream a file per IP multicast with mod_esf. I can stream IP multicast with: pa call stream XML and in XML dialplan: and i can also play files with 'playback' app, BUT: how can put these 2 things together? May be its trivial, but i cant get it make working with 'originate' or 'uuid_broadcast' or 'bridge', 'transfer' and so on ... Thanks in advance, Thomas -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090212/8a8320b8/attachment-0001.html From pauld at versafon.com Thu Feb 12 06:08:58 2009 From: pauld at versafon.com (pauld) Date: Thu, 12 Feb 2009 09:08:58 -0500 Subject: [Freeswitch-users] Cannot choose Cepstral voice from dialplan In-Reply-To: <10703C0D-0BA2-4FB3-B669-B87BF93912E0@freeswitch.org> References: <498E67BF.3060207@versafon.com> <8B964172-6EEB-4B5B-B4B1-D35BFEAEB460@freeswitch.org> <498F6751.2000701@versafon.com> <87f2f3b90902090904i72e37241hb5d4808eb31fcc18@mail.gmail.com> <4993A7D8.1090004@versafon.com> <10703C0D-0BA2-4FB3-B669-B87BF93912E0@freeswitch.org> Message-ID: <49942D7A.2090503@versafon.com> Yes I am using 5.1, I haven't done anything special other than followed wiki and then the advice given here to create symlinks in FS lib dir to all cepstral libs. I have cepstral libs in a standard location /opt/swift/lib. I have given an example extension I used for testing earlier in this thread. Brian West wrote: > This is normal. Are you using 5.0? can you include examples of how > you're doing this? > > /b > > On Feb 11, 2009, at 10:38 PM, pauld wrote: > > >> "TRANSCODING_NECESSARY" >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From freeswitch at servercorps.com Thu Feb 12 07:01:40 2009 From: freeswitch at servercorps.com (Nik Martin) Date: Thu, 12 Feb 2009 09:01:40 -0600 Subject: [Freeswitch-users] FreeSWITCH VPSs In-Reply-To: <191c3a030902112048l1d362f79nf2d0fbb65f2c4984@mail.gmail.com> References: <92e7d2090902111434l60afd3a8i634b4df688ed4011@mail.gmail.com> <9dc4a1670902111608h4d8ca43eq85800ee902113a1a@mail.gmail.com> <28F5713C-838B-409A-897B-F9E40D39B928@freeswitch.org> <9dc4a1670902111624s456b7fbeg99c1e078fe8def3d@mail.gmail.com> <76543A10-18D7-450D-ACA1-E4CAE64F471F@freeswitch.org> <92e7d2090902111759g629ca736k879103144d012155@mail.gmail.com> <191c3a030902112048l1d362f79nf2d0fbb65f2c4984@mail.gmail.com> Message-ID: <92e7d2090902120701y15b0fffej34bb8c46c7aca0b1@mail.gmail.com> I didn't mean to touch a nerve there, I think you mis-interpreted 100% of my original post. You may also have me confused with some other Nik, as I contribute as much or more than I request . There is a nother Nik on here that are probably referring to. I am nikko from #freeswitch. I have made many edits and contributions to the wiki, patches and verified bug reports submitted to Jira. See my other responses inline. On Wed, Feb 11, 2009 at 10:48 PM, Anthony Minessale wrote: > cool, > > However.... you say... > > "but a platform for sandbox testing of configurations, SOHO VOIP > Switches, development of FS addons, backup switch capability, etc. > Doing this stuff at home behind NAT and a consumer grade router is one > reason Brian, Anthony, Mike, et al. are half crazy. " > > First of all, you have no idea on what platform and where we do our > development. > If you only saw how many terminals to random servers spanning the globe we > have open...... > You have the NAT part right, if you knew the hours it took from behind NAT > to get all the code right for SIP interop or even had to deal with half the > bullshit it takes to get SIP working, you'd be in the madhouse so don't you > dare try using us as an ad slogan. I was adressing all the support hours you and others waste trying to help people get FS running on their home servers behind NAT, when they should be testing in an environment that more closely matches what their production one will be. > > Secondly, we are not half crazy, we are completely crazy and most of it > comes from spending all day > on this list tending to your never-ending threads while trying to help the > other people in the community who actually give something back. > > If you want to use our list to advertise this service maybe you should find > a way to contribute to the project rather than constantly asking for help in > 5 separate emails in one day followed by another thread trying to sell > something. > Sorry, you are mistaking me with another Nik. I'm nikko from #freeswitch, and contribute PLENTY. > A reasonable cost would be FREE just like everything else around here. I'm just trying to cover costs, and suport a single NOC engineer, and get more people to adopt FreeSWITCH. If people like Paige and others that come and go had a reasonable environment to test and configure in, they would not be saying crap like "this works in asterisk, blah blah blah". Again, sorry to have upset you, but we had an email conversation an few days ago, and you and Brian were cool with my plans. > > > On Wed, Feb 11, 2009 at 7:59 PM, Nik Martin > wrote: >> >> My goal is obviously not to provide carrier grade VOIP switch service, >> but a platform for sandbox testing of configurations, SOHO VOIP >> Switches, development of FS addons, backup switch capability, etc. >> Doing this stuff at home behind NAT and a consumer grade router is one >> reason Brian, Anthony, Mike, et al. are half crazy. I run my >> company's phone switch in a 32 bit OpenVZ VE with 256 Mb ram, and have >> no issues. When I goof around trying to transcode between 8 and 16 >> bit codecs and whatnot, sure it gets tight, but FS on an idle system >> keeps 23 mb of of ram resident, and rarely if ever hits a 256 mb bean >> counter (limit). >> >> Also, these limits are not hard, I just know what my hardware has, and >> am trying to offer as much value as I can for what I have in the >> systems. 128 Gb of ECC ram and Quad/Quad Core zeons are still pretty >> pricey! >> >> Nik >> >> >> >> >> On Wed, Feb 11, 2009 at 6:28 PM, Brian West wrote: >> > Actually you can if you don't overload the machine like most VPS >> > providers do... The advantage with OpenVZ in this case is that you can >> > migrate the running FreeSWITCH instance between hardware nodes and not >> > drop calls at this size. >> > >> > /b >> > >> > On Feb 11, 2009, at 6:24 PM, EdPimentl wrote: >> > >> >> Soho,,, yes of course... >> >> Voip (soho)Service Provider.... not convinced is possible to provide >> >> reliable QoS. >> >> My .02 cents >> >> -E >> > >> > >> > _______________________________________________ >> > Freeswitch-users mailing list >> > Freeswitch-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From brian at freeswitch.org Thu Feb 12 07:07:36 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 12 Feb 2009 09:07:36 -0600 Subject: [Freeswitch-users] stream a file multicast with mod_esf In-Reply-To: References: Message-ID: <6F68FEDB-8976-44D4-9A67-F871EC7D5120@freeswitch.org> esf is for multi cast paging... it currently won't let you play files... we would have to create a multicast playback application. /b On Feb 12, 2009, at 8:00 AM, Sluschny, Thomas wrote: > Hi, > > i want to stream a file per IP multicast with mod_esf. > > I can stream IP multicast with: > pa call stream XML > and in XML dialplan: > > > > > > > > and i can also play files with 'playback' app, > > BUT: how can put these 2 things together? > > May be its trivial, but i cant get it make working with 'originate' > or 'uuid_broadcast' or 'bridge', 'transfer' and so on ... > > Thanks in advance, > Thomas -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090212/dd1bef26/attachment.html From brian at freeswitch.org Thu Feb 12 07:08:04 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 12 Feb 2009 09:08:04 -0600 Subject: [Freeswitch-users] Cannot choose Cepstral voice from dialplan In-Reply-To: <49942D7A.2090503@versafon.com> References: <498E67BF.3060207@versafon.com> <8B964172-6EEB-4B5B-B4B1-D35BFEAEB460@freeswitch.org> <498F6751.2000701@versafon.com> <87f2f3b90902090904i72e37241hb5d4808eb31fcc18@mail.gmail.com> <4993A7D8.1090004@versafon.com> <10703C0D-0BA2-4FB3-B669-B87BF93912E0@freeswitch.org> <49942D7A.2090503@versafon.com> Message-ID: <9268AE39-6841-4819-9A61-37806B48BEFF@freeswitch.org> You still didn't answer my question. How are you trying to do this from the dialplan. /b On Feb 12, 2009, at 8:08 AM, pauld wrote: > Yes I am using 5.1, I haven't done anything special other than > followed > wiki and then the advice given here to create symlinks in FS lib dir > to all > cepstral libs. I have cepstral libs in a standard location /opt/ > swift/lib. > I have given an example extension I used for testing earlier in this > thread From brian at freeswitch.org Thu Feb 12 07:10:51 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 12 Feb 2009 09:10:51 -0600 Subject: [Freeswitch-users] Change registration information for SIP-Registrar via XML/RPC? In-Reply-To: <4994028A.9090005@fh-wolfenbuettel.de> References: <4994028A.9090005@fh-wolfenbuettel.de> Message-ID: <2BCCBF51-82DF-4F88-AABB-8C4A44480D3B@freeswitch.org> You could store the data in globals and then restart the profiles via XML PRC. ie global_setvar, reloadxml, sofia profile blah restart. /b On Feb 12, 2009, at 5:05 AM, Rene Pankratz wrote: > Hello, > we want to use mod_pa as a softphone, that registers to a > SIPregistrar. > But the username and password need to be changed over time without > restarting freeswitch. > Currently we are using XML/RPC to control the call functions. So it > would be best (if possible) to use it also for changing registration > information. Is there any way to do this? > > Thanks in advance > Ren? From brian at freeswitch.org Thu Feb 12 07:11:12 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 12 Feb 2009 09:11:12 -0600 Subject: [Freeswitch-users] Question: SIP BYE authentication In-Reply-To: <4993DF0D.9000403@ewetel.de> References: <4992C772.4090906@ewetel.de> <4993DF0D.9000403@ewetel.de> Message-ID: <3154F1F1-9286-42BC-9922-1675E004E4A1@freeswitch.org> Are you calling via a gateway? /b On Feb 12, 2009, at 2:34 AM, Helmut Kuper wrote: > Hi, > > any ideas how to get FS's BYEs authenticated ? From anthony.minessale at gmail.com Thu Feb 12 07:29:03 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 12 Feb 2009 09:29:03 -0600 Subject: [Freeswitch-users] Socket-Event on "originate call"? In-Reply-To: <5e414ed0902112339o6c1c499cxe30d0e255d6756f0@mail.gmail.com> References: <5e414ed0902090716y10f95f71o5760dfdf50d1c5cb@mail.gmail.com> <191c3a030902090727n44eef7d7x83704b66e628081@mail.gmail.com> <5e414ed0902100716s2c053a48w2174f7587dd0641e@mail.gmail.com> <5e414ed0902111037q2a6f4097sebd08cb712f042bf@mail.gmail.com> <397EFED1-22E6-4B78-BDE4-3AE5F4204E5B@freeswitch.org> <5e414ed0902111211y3456d5eaqc99baa126cb9dcad@mail.gmail.com> <5e414ed0902112339o6c1c499cxe30d0e255d6756f0@mail.gmail.com> Message-ID: <191c3a030902120729p6da3868eo99a05a5152fd5fce@mail.gmail.com> No, I have not made any changes to reflect anything you asked about. instant_ringback=true is designed to send artificial ringback to the a leg while it's executing the bridge app. it will be meaningless to you if you do not use it with the bridge application On Thu, Feb 12, 2009 at 1:39 AM, Dennis wrote: > the problem is fixed in the latest version of fs - at least it is > working as before without any errors. > > but there is still the question, if the changes where made because of > our problem with the not starting socket!? > we can see in the cli, that the var is set, but it does not change > anything regarding our problem. > > > > 2009/2/11 Brian West : > > Please collect the backtrace and report it on Jira. > > > > /b > > > > On Feb 11, 2009, at 2:11 PM, Dennis wrote: > > > >> this does not help. we are using socket outbound and everything worked > >> before the changes yesterday. > >> > >> we have the same error with other dialplans. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090212/933b9df0/attachment-0001.html From ivdreg at gmail.com Thu Feb 12 06:06:16 2009 From: ivdreg at gmail.com (ivdreg ivdreg) Date: Thu, 12 Feb 2009 16:06:16 +0200 Subject: [Freeswitch-users] Codec negotiation questions Message-ID: Hi all, Can I ask 2 questions about codec negotiation: 1. Is it possible Freeswitch to work negotiate codecs between two phones as it is described below. INVITE from A with some codecs in SDP ---> Freeswitch rewrites codec preference according absolute_codec_string but exclude all codecs not offered by A ----> INVITE to B with rewrited SDP. example: from A SDP:PCMA,PCMU,SPEEX ----> absolute_codec_string=G722,PCMU,PCMA,GSM ----> to B SDP: PCMU,PCMA 2. Can I get codec list in INVITE with mod_perl for example or via xml_curl without processing SDP variable (switch_r_sdp). It will be useful to be in format that absolute_codec_string variable takes. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090212/a04dd7f3/attachment.html From codecomplete at free.fr Thu Feb 12 06:11:40 2009 From: codecomplete at free.fr (Fred) Date: Thu, 12 Feb 2009 15:11:40 +0100 Subject: [Freeswitch-users] Switching from Asterisk to Freeswitch? Message-ID: <7.0.1.0.2.20090212094510.02496ca0@fredshack.com> Hello I successfully used Asterisk to build a voice server for our SOHO business. I did read the article comparing Asterisk to Freeswitch, but I have a couple of questions: 1. What are the decisive reasons that would justify taking a look at Freeswitch? What makes it a better option? 2. I'd like to build an affordable solution based on Asus' EeeBox and (because it's too small to add a PCI card) Sangoma's USB device to connect the host to POTS. Has someone successfully used Freeswitch to work on this hardware? www.asus.com/products.aspx?l1=24&l2=165 http://wiki.sangoma.com/sangoma-wanpipe-usbfxo Thank you for your feedback. From Thomas.Sluschny at siemens.com Thu Feb 12 08:12:26 2009 From: Thomas.Sluschny at siemens.com (Sluschny, Thomas) Date: Thu, 12 Feb 2009 17:12:26 +0100 Subject: [Freeswitch-users] stream a file multicast with mod_esf In-Reply-To: References: Message-ID: Hi Brian, i thought if i can stream from portaudio it is almost the same with streaming from file, so it should working already now. Is this not the design idea of channels and media to do so? regards, thomas PS: sry for improper formatted mail, i cant reply at the moment and have to copy mail from archive :( ________________________________ >From brian at freeswitch.org Thu Feb 12 07:07:36 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 12 Feb 2009 09:07:36 -0600 Subject: [Freeswitch-users] stream a file multicast with mod_esf Message-ID: <6F68FEDB-8976-44D4-9A67-F871EC7D5120 at freeswitch.org > esf is for multi cast paging... it currently won't let you play files... we would have to create a multicast playback application. /b On Feb 12, 2009, at 8:00 AM, Sluschny, Thomas wrote: > Hi, > > i want to stream a file per IP multicast with mod_esf. > > I can stream IP multicast with: > pa call stream XML > and in XML dialplan: > > > > > > > > and i can also play files with 'playback' app, > > BUT: how can put these 2 things together? > > May be its trivial, but i cant get it make working with 'originate' > or 'uuid_broadcast' or 'bridge', 'transfer' and so on ... > > Thanks in advance, > Thomas -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090212/7538c00b/attachment.html From brian at freeswitch.org Thu Feb 12 08:25:15 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 12 Feb 2009 10:25:15 -0600 Subject: [Freeswitch-users] stream a file multicast with mod_esf In-Reply-To: References: Message-ID: <7C996136-963A-42D2-B0FE-D2729F968E52@freeswitch.org> You could but I think you want to stream RTP to a multicast it would be better off building an rtp format mod so you can record rtp:// x.x.x.x:5000 and play from rtp://y.y.y.y:5000 /b On Feb 12, 2009, at 10:12 AM, Sluschny, Thomas wrote: > Hi Brian, > > i thought if i can stream from portaudio it is almost the same with > streaming from file, > so it should working already now. > Is this not the design idea of channels and media to do so? > > regards, > thomas > > PS: sry for improper formatted mail, i cant reply at the moment and > have to copy mail from archive :( -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090212/4c3f4a37/attachment.html From anthony.minessale at gmail.com Thu Feb 12 08:52:10 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 12 Feb 2009 10:52:10 -0600 Subject: [Freeswitch-users] Switching from Asterisk to Freeswitch? In-Reply-To: <7.0.1.0.2.20090212094510.02496ca0@fredshack.com> References: <7.0.1.0.2.20090212094510.02496ca0@fredshack.com> Message-ID: <191c3a030902120852mfef78b1sef49eaec55277e94@mail.gmail.com> On Thu, Feb 12, 2009 at 8:11 AM, Fred wrote: > Hello > > I successfully used Asterisk to build a voice server for our SOHO > business. I did read the article comparing Asterisk to Freeswitch, > but I have a couple of questions: > > 1. What are the decisive reasons that would justify taking a look at > Freeswitch? What makes it a better option? > It's sort of a loaded question because we are likely to prefer FS having worked on it for some years. But anyway, since you asked, I prefer FS because it actually lets you do thing things you dreamed of doing when you first try Asterisk. Asterisk is somewhat like a mirage where you tend to see a pool of water and end up jumping into a sand pit. Asterisk doesn't lack at all in inspiration and possibilities but every time I tried to make something from Asterisk I ended up with a mouth full of sand. Keep in mind I spent 3 years as a core developer in Asterisk doing my best to contribute to its success so it was a pretty big challenge to have to start over with all that functionality right at my fingertips and put up with the claims it was "vaporware" but in 3 short years we have all the functionality back and it's more scalable and is reaching towards the future by supporting things like wideband and ultra wide band audio, resampling and im integration. The short answer to the question is because we are all perfectionists. > > 2. I'd like to build an affordable solution based on Asus' EeeBox and > (because it's too small to add a PCI card) Sangoma's USB device to > connect the host to POTS. Has someone successfully used Freeswitch to > work on this hardware? > www.asus.com/products.aspx?l1=24&l2=165 > http://wiki.sangoma.com/sangoma-wanpipe-usbfxo > > Thank you for your feedback. > We support Sangoma hardware so i am sure if it is not currently supported it will be in the near future. Sangoma has been a big proponent to FreeSWITCH and we have worked very closely over the years. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090212/3dbf297b/attachment.html From mike at jerris.com Thu Feb 12 09:01:24 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 12 Feb 2009 12:01:24 -0500 Subject: [Freeswitch-users] Question: SIP BYE authentication In-Reply-To: <4993DF0D.9000403@ewetel.de> References: <4992C772.4090906@ewetel.de> <4993DF0D.9000403@ewetel.de> Message-ID: If using gayeway it should already do this. On Feb 12, 2009, at 3:34 AM, Helmut Kuper wrote: > Hi, > > any ideas how to get FS's BYEs authenticated ? > > On 11.02.2009 13:41, Helmut Kuper wrote: >> Hello, >> >> my FS is connected to my SIP-DDI softswitch, which requires all SIP >> requests sent by a registered SIP account to be authenticated. I >> found >> that when FS sends a BYE FreeSWITCH ignores the authentication >> challenge (SIP/2.0 407) received from proxy and simply terminates the >> session. >> >> Is there a way to configure FS in that way that it react on auth >> challenges for BYEs ? >> >> regards >> Helmut >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mike at jerris.com Thu Feb 12 09:02:39 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 12 Feb 2009 12:02:39 -0500 Subject: [Freeswitch-users] High CPU load after starting (Brian West) In-Reply-To: <13C421883438EB42B9E2C30069FD4AB767BF49AF15@crushinator.central.local> References: <13C421883438EB42B9E2C30069FD4AB767BF49AF15@crushinator.central.local> Message-ID: Is this running on 64 bit os or 32? On Feb 12, 2009, at 4:23 AM, Public Dump wrote: >> OK does it work now? We have tested this on various windows installs >> among the team here and not seeing this issue... it was a known issue >> back in Nov. or Dec. but thats long been fixed. > > No, the problem is still there. > > I have tested it on a Core AMD 32bit AMD machine = everything is fine. > On a 64bit 4 Core Intel Xeon machine = Problem is there. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mike at jerris.com Thu Feb 12 09:04:31 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 12 Feb 2009 12:04:31 -0500 Subject: [Freeswitch-users] Change registration information for SIP-Registrar via XML/RPC? In-Reply-To: <4994028A.9090005@fh-wolfenbuettel.de> References: <4994028A.9090005@fh-wolfenbuettel.de> Message-ID: You can change the config files on disk and then issue reloadxml or use mod_XML_curl Mike On Feb 12, 2009, at 6:05 AM, Rene Pankratz wrote: > Hello, > we want to use mod_pa as a softphone, that registers to a > SIPregistrar. > But the username and password need to be changed over time without > restarting freeswitch. > Currently we are using XML/RPC to control the call functions. So it > would be best (if possible) to use it also for changing registration > information. Is there any way to do this? > > Thanks in advance > Ren? > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From msc at freeswitch.org Thu Feb 12 09:03:23 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 12 Feb 2009 09:03:23 -0800 Subject: [Freeswitch-users] Switching from Asterisk to Freeswitch? In-Reply-To: <7.0.1.0.2.20090212094510.02496ca0@fredshack.com> References: <7.0.1.0.2.20090212094510.02496ca0@fredshack.com> Message-ID: <87f2f3b90902120903r6d71a685i89a1ea81a9203960@mail.gmail.com> On Thu, Feb 12, 2009 at 6:11 AM, Fred wrote: > Hello > > I successfully used Asterisk to build a voice server for our SOHO > business. I did read the article comparing Asterisk to Freeswitch, > but I have a couple of questions: > > 1. What are the decisive reasons that would justify taking a look at > Freeswitch? What makes it a better option? The answer is, of course, "It depends." If all you need is a simple PBX for your small office then Asterisk actually isn't necessarily a bad choice. My personal experience from talking to people is that when it works, it works well. As to the question of what makes FS and better option than Asterisk it goes back to what you want to use it for. But in a nutshell FreeSWITCH does a lot of things better: FS is more scalable FS is more modular FS is much better written in terms of code quality - no voodoo > > 2. I'd like to build an affordable solution based on Asus' EeeBox and > (because it's too small to add a PCI card) Sangoma's USB device to > connect the host to POTS. Has someone successfully used Freeswitch to > work on this hardware? > www.asus.com/products.aspx?l1=24&l2=165 > http://wiki.sangoma.com/sangoma-wanpipe-usbfxo I personally have not but keep asking around. -MC From nik.middleton at noblesolutions.co.uk Thu Feb 12 09:30:39 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Thu, 12 Feb 2009 17:30:39 -0000 Subject: [Freeswitch-users] Call accounting not working as expected In-Reply-To: <87f2f3b90902111709t2c6db8e2jf190b799eeca6ee8@mail.gmail.com> References: <87f2f3b90902111709t2c6db8e2jf190b799eeca6ee8@mail.gmail.com> Message-ID: Bang on, Thanks -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 12 February 2009 01:10 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Call accounting not working as expected > It is kind of being processed, the account code is being set, XML cdr's are > created and are correct, but csv cdr's for the account code are not > > > > Caller ID is not being set in the A leg but is in the B Leg DING DING DING!!! We have a weener! Okay, that was the key piece of info. Most likely you are logging only the A leg in the CSV CDRs. Go to conf/autoload_configs/cdr_csv.conf.xml and look for these two lines: Most likely you need to use "b" or "ab" depending on your scenario. Try it each way and see how you like the results, then please report back. Thanks! -MC _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From ajlong at worldlink.net Thu Feb 12 09:46:57 2009 From: ajlong at worldlink.net (Adam Long) Date: Thu, 12 Feb 2009 12:46:57 -0500 Subject: [Freeswitch-users] RFC 4497 Originate Timeout / Progress Timeout .. No 100 Trying ... triggering 480 Response Code??? Message-ID: <018901c98d39$e6bd7cc0$b4387640$@net> Hi Guys, I've been experimenting with originate_timeout and progress_timeout as follows. However, shouldn't the timeout trigger a 408 Request Timeout instead of 480 Temporary Failure if no Provisional response received? Just curious, it seems to make sense to me.. but maybe SIP gods see differently. I have also tried using ${originate_disposition} after both bridge attempts to fetch the timeout disposition but instead this is set to NO_ANSWER (which would be correct for first attempt) As I understand it originate_disposition is reset for each bridge completed either successfully or unsuccessfully. Shouldn't the second attempt with no 100 Trying ever received trigger a NO_USER_REPONSE on timeout? According to RFC 4497 that would map to 408 Request Timeout For this test (please note progress_timeout set to low "2" value to test timeout) Node 10.200.1.11 is setup in such a way it responds with 100 Trying but never reaches 180 or 183 before 2 sec timer expires (as desired for this test) Node 10.200.1.12 (is disconnected and never even sends a provisional response, as desired) I have tried.
As well as .
Any thoughts, am I completely nuts and missing something in the spec? Regards, -Adam -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090212/b286db1e/attachment.html From nik.middleton at noblesolutions.co.uk Thu Feb 12 09:51:14 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Thu, 12 Feb 2009 17:51:14 -0000 Subject: [Freeswitch-users] FS equiv for waitforextension Message-ID: HI, Is there an equivalent function in FS to waitforexten ? Closest I've seen is collectInput? Right now I'm using stream file, which is ok if they hit a digit before stream ends, but I want them to have a certain period after the file is played to hit a button. Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090212/1a32b4e9/attachment.html From brian at freeswitch.org Thu Feb 12 09:55:10 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 12 Feb 2009 11:55:10 -0600 Subject: [Freeswitch-users] FS equiv for waitforextension In-Reply-To: References: Message-ID: <65691A20-8D66-43A0-A859-69AB7DE3056A@freeswitch.org> Dialplan or language method...btw if you're on IRC its better to ask there.. faster response... ;) /b On Feb 12, 2009, at 11:51 AM, Nik Middleton wrote: > HI, > > Is there an equivalent function in FS to waitforexten ? Closest > I?ve seen is collectInput? > > Right now I?m using stream file, which is ok if they hit a digit > before stream ends, but I want them to have a certain period after > the file is played to hit a button. > > Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090212/ef1d97a6/attachment-0001.html From sicfslist at gmail.com Thu Feb 12 10:01:12 2009 From: sicfslist at gmail.com (Shelby Ramsey) Date: Thu, 12 Feb 2009 12:01:12 -0600 Subject: [Freeswitch-users] FS equiv for waitforextension In-Reply-To: References: Message-ID: <35b355e90902121001i3a049252x852336b18846ad21@mail.gmail.com> Nik, I'm not sure if this is the right way ... but I use application="read" data="0 1 /path/silence.wav var 1000 # I'm sure there is a better way ... but this seems to work for me. SDR On Thu, Feb 12, 2009 at 11:51 AM, Nik Middleton < nik.middleton at noblesolutions.co.uk> wrote: > HI, > > > > Is there an equivalent function in FS to waitforexten ? Closest I've seen > is collectInput? > > > > Right now I'm using stream file, which is ok if they hit a digit before > stream ends, but I want them to have a certain period after the file is > played to hit a button. > > > > Regards, > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090212/92f65b4a/attachment.html From brian at freeswitch.org Thu Feb 12 10:07:40 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 12 Feb 2009 12:07:40 -0600 Subject: [Freeswitch-users] FS equiv for waitforextension In-Reply-To: <35b355e90902121001i3a049252x852336b18846ad21@mail.gmail.com> References: <35b355e90902121001i3a049252x852336b18846ad21@mail.gmail.com> Message-ID: <76D5D669-B367-4639-925F-DA213FC93539@freeswitch.org> Dialplan isn't for writing IVR's... doing so is against the design of FreeSWITCH.. you can do simple things in dialplan but more complex stuff needs to be in a language. /b On Feb 12, 2009, at 12:01 PM, Shelby Ramsey wrote: > Nik, > > I'm not sure if this is the right way ... but I use > application="read" data="0 1 /path/silence.wav var 1000 # > > I'm sure there is a better way ... but this seems to work for me. > > SDR From msc at freeswitch.org Thu Feb 12 10:14:56 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 12 Feb 2009 10:14:56 -0800 Subject: [Freeswitch-users] FS equiv for waitforextension In-Reply-To: <76D5D669-B367-4639-925F-DA213FC93539@freeswitch.org> References: <35b355e90902121001i3a049252x852336b18846ad21@mail.gmail.com> <76D5D669-B367-4639-925F-DA213FC93539@freeswitch.org> Message-ID: <87f2f3b90902121014t29ab0e25u129c4c824d63e11e@mail.gmail.com> On Thu, Feb 12, 2009 at 10:07 AM, Brian West wrote: > Dialplan isn't for writing IVR's... doing so is against the design of > FreeSWITCH.. you can do simple things in dialplan but more complex > stuff needs to be in a language. Or create an IVR and send the call there from the dialplan. You can do IVRs in Lua/JS/Perl or in XML. -MC From nik.middleton at noblesolutions.co.uk Thu Feb 12 10:15:11 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Thu, 12 Feb 2009 18:15:11 -0000 Subject: [Freeswitch-users] FS equiv for waitforextension In-Reply-To: <76D5D669-B367-4639-925F-DA213FC93539@freeswitch.org> References: <35b355e90902121001i3a049252x852336b18846ad21@mail.gmail.com> <76D5D669-B367-4639-925F-DA213FC93539@freeswitch.org> Message-ID: Sorry, should have said this was in js Regards, -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: 12 February 2009 18:08 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] FS equiv for waitforextension Dialplan isn't for writing IVR's... doing so is against the design of FreeSWITCH.. you can do simple things in dialplan but more complex stuff needs to be in a language. /b On Feb 12, 2009, at 12:01 PM, Shelby Ramsey wrote: > Nik, > > I'm not sure if this is the right way ... but I use > application="read" data="0 1 /path/silence.wav var 1000 # > > I'm sure there is a better way ... but this seems to work for me. > > SDR _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From pbd at suspiria.net Thu Feb 12 10:49:44 2009 From: pbd at suspiria.net (Public Dump) Date: Thu, 12 Feb 2009 19:49:44 +0100 Subject: [Freeswitch-users] Freeswitch-users Digest, Vol 32, Issue 108 In-Reply-To: References: Message-ID: <13C421883438EB42B9E2C30069FD4AB76AEA2B386F@crushinator.central.local> > > Is this running on 64 bit os or 32? A 64bit , Windows 2008 Server. From nik.middleton at noblesolutions.co.uk Thu Feb 12 11:36:52 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Thu, 12 Feb 2009 19:36:52 -0000 Subject: [Freeswitch-users] js and VMD Message-ID: Hi Guys, I'm trying to get VMD running in js, does anyone have an example of how it's called? If I try session:execute("vmd"); I get an error Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090212/0ebc0032/attachment.html From msc at freeswitch.org Thu Feb 12 12:02:24 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 12 Feb 2009 12:02:24 -0800 Subject: [Freeswitch-users] FS + Call Center Solution In-Reply-To: <6309E7515E4F43159B9564800564B562@SaeedLaptop> References: <6309E7515E4F43159B9564800564B562@SaeedLaptop> Message-ID: <87f2f3b90902121202m5652f0e3x42eabb6ced2fd646@mail.gmail.com> On Wed, Feb 11, 2009 at 8:31 AM, Saeed Ahmed wrote: > Hi List, > > Is there any open source call center tool available which works with FS? Check this out: http://opencsm.org/wiki/index.php/Spice_Telephony -MC From msc at freeswitch.org Thu Feb 12 11:59:57 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 12 Feb 2009 11:59:57 -0800 Subject: [Freeswitch-users] js and VMD In-Reply-To: References: Message-ID: <87f2f3b90902121159k570e0f4waa3afc2629ac5920@mail.gmail.com> > I'm trying to get VMD running in js, does anyone have an example of how it's > called? http://wiki.freeswitch.org/wiki/Mod_vmd You need to use the event socket because that is the way VMD is designed. If called from the dialplan it will set a channel variable but that isn't of much use in a real-time application. Using it as an API (or bgapi) will yield an event when VMD is detected. This makes sense because you don't know when (or even if) VMD will be detected, so using the event system is the best choice. -MC From red.rain.seven at gmail.com Thu Feb 12 12:13:47 2009 From: red.rain.seven at gmail.com (Henry Huang) Date: Thu, 12 Feb 2009 12:13:47 -0800 Subject: [Freeswitch-users] FreeSWITCH VPSs In-Reply-To: References: <92e7d2090902111434l60afd3a8i634b4df688ed4011@mail.gmail.com> <59ad9ca10902111637n1459d3cep5bc73dffe6bb474f@mail.gmail.com> Message-ID: <59ad9ca10902121213v24100f30qfcbb4e2fdb350806@mail.gmail.com> I run /usr/local/freeswitch/bin/freeswitch but I don't see a place where it says it's 32bit or 64bit. at the end of the initial script, I do see a version statement though. FreeSWITCH Version 1.0.trunk (exported) Started. Is there other ways to check if it's 32bit or 64bit? On Wed, Feb 11, 2009 at 6:55 PM, Brian West wrote: > ding ding ding .. yep! > > "file /usr/local/freeswitch/bin/freeswitch" will also confirm > > /b > > On Feb 11, 2009, at 6:37 PM, Henry Huang wrote: > > > Brian: > > > > I am also running my freeswitch on my own openVZ containers. Just > > how do you verify if the freeswitch is compiled as 64bit? I would > > assume if I compile it under a 64bit container, I would > > automatically get a 64bit freeswitch right? > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Henry Huang UniC Solution - Communication Unified -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090212/bc8d2934/attachment.html From william.suffill at gmail.com Thu Feb 12 12:18:45 2009 From: william.suffill at gmail.com (William Suffill) Date: Thu, 12 Feb 2009 15:18:45 -0500 Subject: [Freeswitch-users] FreeSWITCH VPSs In-Reply-To: <59ad9ca10902121213v24100f30qfcbb4e2fdb350806@mail.gmail.com> References: <92e7d2090902111434l60afd3a8i634b4df688ed4011@mail.gmail.com> <59ad9ca10902111637n1459d3cep5bc73dffe6bb474f@mail.gmail.com> <59ad9ca10902121213v24100f30qfcbb4e2fdb350806@mail.gmail.com> Message-ID: <6b65470d0902121218j247dc688gbc9e615f6c8af281@mail.gmail.com> If you run in your shell: file /usr/local/freeswitch/bin/freeswitch as Brian suggested it will return something like what I got below: /usr/local/freeswitch/bin/freeswitch: ELF 64-bit LSB executable, x86-64, version 1 (SYSV), for GNU/Linux 2.6.8, dynamically linked (uses shared libs), not stripped He didn't want you to start freeswitch but instead pass the path of the binary to the file command which will tell information about most files. -- W -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090212/aa1ddbf2/attachment-0001.html From brian at freeswitch.org Thu Feb 12 12:31:16 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 12 Feb 2009 14:31:16 -0600 Subject: [Freeswitch-users] FreeSWITCH VPSs In-Reply-To: <59ad9ca10902121213v24100f30qfcbb4e2fdb350806@mail.gmail.com> References: <92e7d2090902111434l60afd3a8i634b4df688ed4011@mail.gmail.com> <59ad9ca10902111637n1459d3cep5bc73dffe6bb474f@mail.gmail.com> <59ad9ca10902121213v24100f30qfcbb4e2fdb350806@mail.gmail.com> Message-ID: <21D9E0A3-29A7-442F-AB2B-B26FF2568BFD@freeswitch.org> Well when I do this: root at taz [Thu Feb 12 02:20 PM] /usr/src/freeswitch.trunk <13>:file /usr/local/freeswitch/bin/freeswitch /usr/local/freeswitch/bin/freeswitch: ELF 64-bit LSB executable, AMD x86-64, version 1 (SYSV), for GNU/Linux 2.6.9, dynamically linked (uses shared libs), for GNU/Linux 2.6.9, not stripped It should clearly tell you. run the "file" command on it. /b On Feb 12, 2009, at 2:13 PM, Henry Huang wrote: > I run /usr/local/freeswitch/bin/freeswitch > but I don't see a place where it says it's 32bit or 64bit. > at the end of the initial script, I do see a version statement though. > FreeSWITCH Version 1.0.trunk (exported) Started. > Is there other ways to check if it's 32bit or 64bit? From red.rain.seven at gmail.com Thu Feb 12 12:40:50 2009 From: red.rain.seven at gmail.com (Henry Huang) Date: Thu, 12 Feb 2009 12:40:50 -0800 Subject: [Freeswitch-users] FreeSWITCH VPSs In-Reply-To: <21D9E0A3-29A7-442F-AB2B-B26FF2568BFD@freeswitch.org> References: <92e7d2090902111434l60afd3a8i634b4df688ed4011@mail.gmail.com> <59ad9ca10902111637n1459d3cep5bc73dffe6bb474f@mail.gmail.com> <59ad9ca10902121213v24100f30qfcbb4e2fdb350806@mail.gmail.com> <21D9E0A3-29A7-442F-AB2B-B26FF2568BFD@freeswitch.org> Message-ID: <59ad9ca10902121240t3eec6913x2d83a03facab7387@mail.gmail.com> Thinak you, William and Brian I got it now, I didn't know file was a command before because it didn't come with my CentOS installation. Now I have installed the file package and able to see the file info. Thanks again On Thu, Feb 12, 2009 at 12:31 PM, Brian West wrote: > Well when I do this: > > > root at taz [Thu Feb 12 02:20 PM] /usr/src/freeswitch.trunk > <13>:file /usr/local/freeswitch/bin/freeswitch > /usr/local/freeswitch/bin/freeswitch: ELF 64-bit LSB executable, AMD > x86-64, version 1 (SYSV), for GNU/Linux 2.6.9, dynamically linked > (uses shared libs), for GNU/Linux 2.6.9, not stripped > > > It should clearly tell you. run the "file" command on it. > > /b > > > > On Feb 12, 2009, at 2:13 PM, Henry Huang wrote: > > > I run /usr/local/freeswitch/bin/freeswitch > > but I don't see a place where it says it's 32bit or 64bit. > > at the end of the initial script, I do see a version statement though. > > FreeSWITCH Version 1.0.trunk (exported) Started. > > Is there other ways to check if it's 32bit or 64bit? > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Henry Huang UniC Solution - Communication Unified -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090212/2d2ba4dc/attachment.html From jaugenstine at gmail.com Thu Feb 12 12:44:54 2009 From: jaugenstine at gmail.com (jonathan augenstine) Date: Thu, 12 Feb 2009 12:44:54 -0800 Subject: [Freeswitch-users] deflect issue Message-ID: <207e7a5e0902121244t157960ar7d775430c60baa66@mail.gmail.com> I am trying to use the deflect command to transfer an inbound call. The call is established and the command seems to complete successfully. If I bump up the sofia logging, I see the command executed in the LUA script and I see output from the console from sofia that seems to indicate the deflect refer has been initiated, but there are never any SIP messages sent to the gateway. See the console output below. The next SIP message I see is a BYE from the gateway when I hang up. Do I have something configured incorrectly? Jonathan 2009-02-12 12:36:52 [INFO] switch_cpp.cpp:1086 console_log() Awake Lua execute(deflect, sofia/external/6265551212 at aristotle.mn.maestroconference.com:5080:5080) nua: nua_refer: entering nua(0x937f0f0): sent signal r_refer nua: nua_stack_set_params: entering soa_set_params(static::0x93e65c0, ...) called nua: nua_application_event: entering nua: nua_handle_magic: entering -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090212/f0cff6fc/attachment.html From nik.middleton at noblesolutions.co.uk Thu Feb 12 12:49:45 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Thu, 12 Feb 2009 20:49:45 -0000 Subject: [Freeswitch-users] js and VMD In-Reply-To: <87f2f3b90902121159k570e0f4waa3afc2629ac5920@mail.gmail.com> References: <87f2f3b90902121159k570e0f4waa3afc2629ac5920@mail.gmail.com> Message-ID: That makes sense, though could it not have a call back mechanism similar to DTMF detect? I'm still not sure how I could use it even in an event socket. I plan to call my js IVR script using a socket, but that has the originate call in it which is nice and simple, but I'm unsure how I could abort it (js IVR. The functionality I'm looking for is really simple. I simply don't want to leave a voicemail message. So VMD looks just the ticket. Regards -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 12 February 2009 20:00 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] js and VMD > I'm trying to get VMD running in js, does anyone have an example of how it's > called? http://wiki.freeswitch.org/wiki/Mod_vmd You need to use the event socket because that is the way VMD is designed. If called from the dialplan it will set a channel variable but that isn't of much use in a real-time application. Using it as an API (or bgapi) will yield an event when VMD is detected. This makes sense because you don't know when (or even if) VMD will be detected, so using the event system is the best choice. -MC _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From brian at freeswitch.org Thu Feb 12 12:47:15 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 12 Feb 2009 14:47:15 -0600 Subject: [Freeswitch-users] deflect issue In-Reply-To: <207e7a5e0902121244t157960ar7d775430c60baa66@mail.gmail.com> References: <207e7a5e0902121244t157960ar7d775430c60baa66@mail.gmail.com> Message-ID: deflect takes one arg. and that isn't one. Try a SIP uri... not a sofia/ string. ie sip:blah at host:5080 /b On Feb 12, 2009, at 2:44 PM, jonathan augenstine wrote: > I am trying to use the deflect command to transfer an inbound call. > The call is established and the command seems to complete > successfully. If I bump up the sofia logging, I see the command > executed in the LUA script and I see output from the console from > sofia that seems to indicate the deflect refer has been initiated, > but there are never any SIP messages sent to the gateway. See the > console output below. The next SIP message I see is a BYE from the > gateway when I hang up. Do I have something configured incorrectly? > > Jonathan > > 2009-02-12 12:36:52 [INFO] switch_cpp.cpp:1086 console_log() Awake > Lua execute(deflect, sofia/external/6265551212 at aristotle.mn.maestroconference.com > :5080:5080) > nua: nua_refer: entering > nua(0x937f0f0): sent signal r_refer > nua: nua_stack_set_params: entering > soa_set_params(static::0x93e65c0, ...) called > nua: nua_application_event: entering > nua: nua_handle_magic: entering -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090212/91091bc8/attachment.html From anthony.minessale at gmail.com Thu Feb 12 12:59:57 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 12 Feb 2009 14:59:57 -0600 Subject: [Freeswitch-users] Cannot choose Cepstral voice from dialplan In-Reply-To: <4993A7D8.1090004@versafon.com> References: <498E67BF.3060207@versafon.com> <8B964172-6EEB-4B5B-B4B1-D35BFEAEB460@freeswitch.org> <498F6751.2000701@versafon.com> <87f2f3b90902090904i72e37241hb5d4808eb31fcc18@mail.gmail.com> <4993A7D8.1090004@versafon.com> Message-ID: <191c3a030902121259x41e03e50yc5c7c5dc00f509a7@mail.gmail.com> transcoding from PCMU (g711) to PCM (raw signed linear) the format that cepstral speaks. On Wed, Feb 11, 2009 at 10:38 PM, pauld wrote: > The issue was resolved by creating symlinks to cepstral libs in FS lib > directory. I tried that on 1.0.3, but most probably it would work on > 1.0.2 as well. Thanks for help. > BTW, without that FS would do a core dump (seg fault) on shutdown after > TTS was invoked at least once. > Looking at FS logs I see "TRANSCODING_NECESSARY" when executing dynamic > text even with 8 kHz voice. Why would that be? Looks like it's PCMU/8000 > what it's transcoding to what? > > > Michael Collins wrote: > > On Sun, Feb 8, 2009 at 3:14 PM, pauld wrote: > > > >> The libs are there with correct symlinks, see below. I tested both > >> voices directly via swift command, works fine. > >> Any other ideas? > >> It's Cepstral 5.1, FS 1.0.2. > >> > >> > > > > Well, first I recommend getting on latest trunk if that's at all > > possible for you. The devs have made a ton of improvements in the last > > five weeks. Second, this might actually be an issue with FS looking in > > its own lib directory for these .so files. Try a symlink from > > /usr/local/freeswitch/lib to your /opt/swift/lib (or whatever the name > > is) dir for each .so file. However, I think Raymond is correct - some > > weirdness has been reported by some Cepstral users on 5.1. We'd > > definitely like to hear about your experiences if and when you get it > > running. > > > > -MC > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090212/839f9df5/attachment-0001.html From anthony.minessale at gmail.com Thu Feb 12 13:06:15 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 12 Feb 2009 15:06:15 -0600 Subject: [Freeswitch-users] Codec negotiation questions In-Reply-To: References: Message-ID: <191c3a030902121306l1dae21a3m614f400703d15bd2@mail.gmail.com> the entire sdp is available as a variable (route the call to the info app to see the variables) so if you have inbound-late-negotiation set to true on the sip profile then you can use a regex or a script to set absolute_codec string before you answer. On Thu, Feb 12, 2009 at 8:06 AM, ivdreg ivdreg wrote: > Hi all, > > Can I ask 2 questions about codec negotiation: > > 1. Is it possible Freeswitch to work negotiate codecs between two phones as > it is described below. > INVITE from A with some codecs in SDP ---> Freeswitch rewrites codec > preference according absolute_codec_string but exclude all codecs not > offered by A ----> INVITE to B with rewrited SDP. > > example: > from A SDP:PCMA,PCMU,SPEEX ----> absolute_codec_string=G722,PCMU,PCMA,GSM > ----> to B SDP: PCMU,PCMA > > 2. Can I get codec list in INVITE with mod_perl for example or via xml_curl > without processing SDP variable (switch_r_sdp). It will be useful to be in > format that absolute_codec_string variable takes. > > Thanks > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090212/4f0f84cf/attachment.html From kerrada2003 at yahoo.com Thu Feb 12 13:41:29 2009 From: kerrada2003 at yahoo.com (Ali Al-Rubaie) Date: Thu, 12 Feb 2009 13:41:29 -0800 (PST) Subject: [Freeswitch-users] Realm value Message-ID: <268387.58846.qm@web33701.mail.mud.yahoo.com> Hi, ? How can the default value of "realm" be changed? I had changed the command: ? ? in the file internal.xml but FS still uses the server IP address as the challenge realm. ? Thanks in advance! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090212/9fecc8d4/attachment.html From msc at freeswitch.org Thu Feb 12 13:44:45 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 12 Feb 2009 13:44:45 -0800 Subject: [Freeswitch-users] js and VMD In-Reply-To: References: <87f2f3b90902121159k570e0f4waa3afc2629ac5920@mail.gmail.com> Message-ID: <87f2f3b90902121344h487cafeai724c8a56b9195201@mail.gmail.com> On Thu, Feb 12, 2009 at 12:49 PM, Nik Middleton wrote: > That makes sense, though could it not have a call back mechanism similar > to DTMF detect? > It probably could but the mod's author was using it exclusively from event socket. I personally added the channel variable code for the sake of testing. I'm sure this could be added but it's beyond my skills presently. I would recommend opening up a JIRA and requesting this functionality as an improvement. Perhaps the author, Eric Des Courtis, could add it or perhaps another skilled programmer could add this functionality. In the grand scheme of things it probably isn't that difficult and with a little time even I could figure it out. > I'm still not sure how I could use it even in an event socket. I plan > to call my js IVR script using a socket, but that has the originate call > in it which is nice and simple, but I'm unsure how I could abort it (js > IVR. As a proof of concept you could have your script loop and check the value of ${vmd_status} every 1000ms or so, and if it ever has the value "TRUE" then you know VMD was positive and you could hangup and do whatever other cleanup is necessary. That solution would be a temp fix even though it wouldn't actually scale very well. How are you handling answered calls now? Do you just start playing a message? I'm wondering how this would work even if there was a callback. Would you mind doing a pastebin of your script? I'd like to see the big picture. -MC > > The functionality I'm looking for is really simple. I simply don't want > to leave a voicemail message. So VMD looks just the ticket. > > Regards From lfurrea at gmail.com Thu Feb 12 13:46:23 2009 From: lfurrea at gmail.com (Luis F Urrea) Date: Thu, 12 Feb 2009 15:46:23 -0600 Subject: [Freeswitch-users] xml_cdr call flow Message-ID: Hi all, We are writing a xml_cdr parser to load CDRs in SQLite. We are interested in logging times for both A leg and B leg so that transfers are reported as individual calls with accurate timing. eg Inboud call to AA lasted 14 seconds then call to operator 20s and then call to actual extension 5min As of now we are using the tag with the "number" attribute to find out who did the A leg talk to, then we open the B leg files and get the times from each jump from the tag within the tag on the B leg file. Is this right or maybe someone could suggest a better way to do it. TIA Luis -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090212/4422ed95/attachment.html From brian at freeswitch.org Thu Feb 12 13:48:00 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 12 Feb 2009 15:48:00 -0600 Subject: [Freeswitch-users] Realm value In-Reply-To: <268387.58846.qm@web33701.mail.mud.yahoo.com> References: <268387.58846.qm@web33701.mail.mud.yahoo.com> Message-ID: What SVN rev? /b On Feb 12, 2009, at 3:41 PM, Ali Al-Rubaie wrote: > Hi, > > How can the default value of "realm" be changed? I had changed the > command: > > > > in the file internal.xml but FS still uses the server IP address as > the challenge realm. > > Thanks in advance! > From anthony.minessale at gmail.com Thu Feb 12 14:20:18 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 12 Feb 2009 16:20:18 -0600 Subject: [Freeswitch-users] segfault when shoutcast plays mp3 and extension hangs up In-Reply-To: <4993D8CA.1010602@ewetel.de> References: <4992BFFC.50006@ewetel.de> <4992C801.4020906@ewetel.de> <191c3a030902110558i9702139g6b12aaf15f6d3aac@mail.gmail.com> <4992DCC2.3020702@ewetel.de> <191c3a030902110617n3dcd19cfr204c00792e1e87ed@mail.gmail.com> <4992E559.6060506@ewetel.de> <191c3a030902110729g30b30d46gac0f56205f8d48cf@mail.gmail.com> <4993D8CA.1010602@ewetel.de> Message-ID: <191c3a030902121420o3b4ec7a8j1fa0059b849bd866@mail.gmail.com> That's scary.... So I wonder what about the distro you are using that makes the same exact code not work? maybe the GCC ? On Thu, Feb 12, 2009 at 2:07 AM, Helmut Kuper wrote: > Hi Anthony, > > hm... on centos5 it works fine. No problems, no warning, no crash. > > regards > Helmut > > On 11.02.2009 16:29, Anthony Minessale wrote: > > I am highly suspicious of the ubuntu. > > you are using a prerelease of gcc that we have already found at least > > 1 bug. > > > > we tried the file on our box and it doesn't even say anything about > > the file being bad etc...... it plays and hangs up fine even 4 times > > at once. > > It would be a big help if you could try to reproduce it on CentOS 5 as > > a comparison. We have had 3 cases this week where doing so has fixed > > problems and i don't want to believe it so I would appropriate it if > > you could test it. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090212/f8c65917/attachment.html From nik.middleton at noblesolutions.co.uk Thu Feb 12 14:26:17 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Thu, 12 Feb 2009 22:26:17 -0000 Subject: [Freeswitch-users] js and VMD In-Reply-To: <87f2f3b90902121344h487cafeai724c8a56b9195201@mail.gmail.com> References: <87f2f3b90902121159k570e0f4waa3afc2629ac5920@mail.gmail.com> <87f2f3b90902121344h487cafeai724c8a56b9195201@mail.gmail.com> Message-ID: Just been chatting to Ken Rice, his view (and he may be mistaken) is that it should fire the call back event in much the same way as DTMF does, however, it's not working. I used to develop with C/C++ for about 10 years, but that was 12 years ago. Very rusty. However, I'm going to look at the start_dtmf code and try to replicate the functionality in mod_vmd. Regarding your suggestion, that wouldn't really work as I'm streaming a file. However, if memory serves me well, there is a timer function in C that you can set to run that can call a function. There is a function in js called setTimeout(time_func, 500) but sadly it's not available in spidermonkey. BTW this function would resolve a bounty on call duration timeouts Regards -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 12 February 2009 21:45 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] js and VMD On Thu, Feb 12, 2009 at 12:49 PM, Nik Middleton wrote: > That makes sense, though could it not have a call back mechanism similar > to DTMF detect? > It probably could but the mod's author was using it exclusively from event socket. I personally added the channel variable code for the sake of testing. I'm sure this could be added but it's beyond my skills presently. I would recommend opening up a JIRA and requesting this functionality as an improvement. Perhaps the author, Eric Des Courtis, could add it or perhaps another skilled programmer could add this functionality. In the grand scheme of things it probably isn't that difficult and with a little time even I could figure it out. > I'm still not sure how I could use it even in an event socket. I plan > to call my js IVR script using a socket, but that has the originate call > in it which is nice and simple, but I'm unsure how I could abort it (js > IVR. As a proof of concept you could have your script loop and check the value of ${vmd_status} every 1000ms or so, and if it ever has the value "TRUE" then you know VMD was positive and you could hangup and do whatever other cleanup is necessary. That solution would be a temp fix even though it wouldn't actually scale very well. How are you handling answered calls now? Do you just start playing a message? I'm wondering how this would work even if there was a callback. Would you mind doing a pastebin of your script? I'd like to see the big picture. -MC > > The functionality I'm looking for is really simple. I simply don't want > to leave a voicemail message. So VMD looks just the ticket. > > Regards _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From woof at nortel.com Thu Feb 12 14:34:58 2009 From: woof at nortel.com (Andy Spitzer) Date: Thu, 12 Feb 2009 17:34:58 -0500 Subject: [Freeswitch-users] segfault when shoutcast plays mp3 and extension hangs up In-Reply-To: <191c3a030902121420o3b4ec7a8j1fa0059b849bd866@mail.gmail.com> References: <4992BFFC.50006@ewetel.de> <4992C801.4020906@ewetel.de> <191c3a030902110558i9702139g6b12aaf15f6d3aac@mail.gmail.com> <4992DCC2.3020702@ewetel.de> <191c3a030902110617n3dcd19cfr204c00792e1e87ed@mail.gmail.com> <4992E559.6060506@ewetel.de> <191c3a030902110729g30b30d46gac0f56205f8d48cf@mail.gmail.com> <4993D8CA.1010602@ewetel.de> <191c3a030902121420o3b4ec7a8j1fa0059b849bd866@mail.gmail.com> Message-ID: Woof! On Thu, 12 Feb 2009 17:20:18 -0500, Anthony Minessale wrote: > So I wonder what about the distro you are using that makes the same exact code not work? > maybe the GCC ? Possibly. A recent (last year?) GCC change caused some order of operations to change, and so code that inadvertently relied on the previous behavior doesn't work any more. The "c standard" doesn't define many of the orders, and some code may have side effects that depend on one way or the other without realizing it. I discovered this the hard way last year some time. --Woof! From brian at freeswitch.org Thu Feb 12 14:38:06 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 12 Feb 2009 16:38:06 -0600 Subject: [Freeswitch-users] segfault when shoutcast plays mp3 and extension hangs up In-Reply-To: References: <4992BFFC.50006@ewetel.de> <4992C801.4020906@ewetel.de> <191c3a030902110558i9702139g6b12aaf15f6d3aac@mail.gmail.com> <4992DCC2.3020702@ewetel.de> <191c3a030902110617n3dcd19cfr204c00792e1e87ed@mail.gmail.com> <4992E559.6060506@ewetel.de> <191c3a030902110729g30b30d46gac0f56205f8d48cf@mail.gmail.com> <4993D8CA.1010602@ewetel.de> <191c3a030902121420o3b4ec7a8j1fa0059b849bd866@mail.gmail.com> Message-ID: <9CF2F413-1DE1-49A7-A550-6423C3444D14@freeswitch.org> This is prob. why we don't see this crazy stuff on CentOS since the compiler is 4.1.2 /b On Feb 12, 2009, at 4:34 PM, Andy Spitzer wrote: > Possibly. A recent (last year?) GCC change caused some order of > operations to change, and so code that inadvertently relied on the > previous behavior doesn't work any more. The "c standard" doesn't > define many of the orders, and some code may have side effects that > depend on one way or the other without realizing it. > > I discovered this the hard way last year some time. > > --Woof! From msc at freeswitch.org Thu Feb 12 14:42:55 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 12 Feb 2009 14:42:55 -0800 Subject: [Freeswitch-users] js and VMD In-Reply-To: References: <87f2f3b90902121159k570e0f4waa3afc2629ac5920@mail.gmail.com> <87f2f3b90902121344h487cafeai724c8a56b9195201@mail.gmail.com> Message-ID: <87f2f3b90902121442t22d9e5b7l3d2b7711646d8ca2@mail.gmail.com> On Thu, Feb 12, 2009 at 2:26 PM, Nik Middleton wrote: > Just been chatting to Ken Rice, his view (and he may be mistaken) is > that it should fire the call back event in much the same way as DTMF > does, however, it's not working. I used to develop with C/C++ for about > 10 years, but that was 12 years ago. Very rusty. However, I'm going to > look at the start_dtmf code and try to replicate the functionality in > mod_vmd. That would be awesome. I think once you get into the code you'll realize that it isn't like walking through a warzone or tapdancing in a minefield like some other codebases. :) The code is really well-ordered so you can frequently copy and paste from other files and functions and create new functionality. Feel free to get in there and start mixing it up! It's kinda fun. > > Regarding your suggestion, that wouldn't really work as I'm streaming a > file. However, if memory serves me well, there is a timer function in C > that you can set to run that can call a function. There is a function > in js called setTimeout(time_func, 500) but sadly it's not available in > spidermonkey. BTW this function would resolve a bounty on call duration > timeouts Understood. The other thing you could do, at least to test the events for VMD, is to create a little daemon kind of program that sits there and listens for VMD hits and have it uuid_kill those channels for you. Crude, to be sure, but it would definitely let you confirm that the system is working. -MC From lfurrea at gmail.com Thu Feb 12 14:50:47 2009 From: lfurrea at gmail.com (Luis F Urrea) Date: Thu, 12 Feb 2009 16:50:47 -0600 Subject: [Freeswitch-users] xml_cdr call flow In-Reply-To: References: Message-ID: On our test calls we haven't been able to correlate times from the A leg with times from the B leg. I would expect something as A-leg(duration)= B-leg1(duration)+B-leg2(duration) Also the tag within tag does not seem to be in epoch microseconds. so it does not seem that's where i should be looking for that info. Here's an example of the tag for a test call on the A-Leg: 1. 2. 1233942283835696 3. 1233942283835696 4. 1233942283999716 5. 1233942283999716 6. 1233942287291931 7. 0 8. 1233942303240916 9. any hint is appreciated On Thu, Feb 12, 2009 at 3:46 PM, Luis F Urrea wrote: > Hi all, > > We are writing a xml_cdr parser to load CDRs in SQLite. We are interested > in logging times for both A leg and B leg so that transfers are reported as > individual calls with accurate timing. eg Inboud call to AA lasted 14 > seconds then call to operator 20s and then call to actual extension 5min > > As of now we are using the tag with the "number" attribute to > find out who did the A leg talk to, then we open the B leg files and get the > times from each jump from the tag within the tag on the B > leg file. > > Is this right or maybe someone could suggest a better way to do it. > > TIA > > Luis > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090212/2077fea2/attachment.html From nik.middleton at noblesolutions.co.uk Thu Feb 12 14:58:02 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Thu, 12 Feb 2009 22:58:02 -0000 Subject: [Freeswitch-users] FS equiv for waitforextension In-Reply-To: <87f2f3b90902121014t29ab0e25u129c4c824d63e11e@mail.gmail.com> References: <35b355e90902121001i3a049252x852336b18846ad21@mail.gmail.com><76D5D669-B367-4639-925F-DA213FC93539@freeswitch.org> <87f2f3b90902121014t29ab0e25u129c4c824d63e11e@mail.gmail.com> Message-ID: Hi, Not sure who updates the WIKI, but it's wrong on collectinput for the example. In the call, dtmf needs quotes, ie "dtmf" Correction is session.collectInput( mycb, "dtmf", 8000 ); Without it you get [ERR] voice.js:70 mod_spidermonkey() ReferenceError: dtmf is not defined if ( session.ready( ) ) { session.answer( ); session.streamFile( "sounds/typeSomeDigits.wav" ); session.collectInput( mycb, dtmf, 8000 ); console_log( "info", "Got " + dtmf.digits + "\n" ); session.streamFile( "sounds/thanksBye.wav" ); session.hangup( ); } -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 12 February 2009 18:15 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] FS equiv for waitforextension On Thu, Feb 12, 2009 at 10:07 AM, Brian West wrote: > Dialplan isn't for writing IVR's... doing so is against the design of > FreeSWITCH.. you can do simple things in dialplan but more complex > stuff needs to be in a language. Or create an IVR and send the call there from the dialplan. You can do IVRs in Lua/JS/Perl or in XML. -MC _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From msc at freeswitch.org Thu Feb 12 15:00:32 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 12 Feb 2009 15:00:32 -0800 Subject: [Freeswitch-users] xml_cdr call flow In-Reply-To: References: Message-ID: <87f2f3b90902121500m2b86efc2q722314688d532648@mail.gmail.com> Pastebin the whole file so that we can see it in context... -MC On Thu, Feb 12, 2009 at 2:50 PM, Luis F Urrea wrote: > On our test calls we haven't been able to correlate times from the A leg > with times from the B leg. > > I would expect something as A-leg(duration)= > B-leg1(duration)+B-leg2(duration) > > Also the tag within tag does not seem to be in epoch > microseconds. so it does not seem that's where i should be looking for that > info. > > Here's an example of the tag for a test call on the A-Leg: > > > 1233942283835696 > 1233942283835696 > 1233942283999716 > 1233942283999716 > 1233942287291931 > 0 > 1233942303240916 > > > any hint is appreciated > > > On Thu, Feb 12, 2009 at 3:46 PM, Luis F Urrea wrote: >> >> Hi all, >> >> We are writing a xml_cdr parser to load CDRs in SQLite. We are interested >> in logging times for both A leg and B leg so that transfers are reported as >> individual calls with accurate timing. eg Inboud call to AA lasted 14 >> seconds then call to operator 20s and then call to actual extension 5min >> >> As of now we are using the tag with the "number" attribute to >> find out who did the A leg talk to, then we open the B leg files and get the >> times from each jump from the tag within the tag on the B >> leg file. >> >> Is this right or maybe someone could suggest a better way to do it. >> >> TIA >> >> Luis > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From msc at freeswitch.org Thu Feb 12 15:01:43 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 12 Feb 2009 15:01:43 -0800 Subject: [Freeswitch-users] FS equiv for waitforextension In-Reply-To: References: <35b355e90902121001i3a049252x852336b18846ad21@mail.gmail.com> <76D5D669-B367-4639-925F-DA213FC93539@freeswitch.org> <87f2f3b90902121014t29ab0e25u129c4c824d63e11e@mail.gmail.com> Message-ID: <87f2f3b90902121501g2d0c69f9p52a7003880c42304@mail.gmail.com> On Thu, Feb 12, 2009 at 2:58 PM, Nik Middleton wrote: > Hi, > > Not sure who updates the WIKI, but it's wrong on collectinput for the > example. In the call, dtmf needs quotes, ie "dtmf" Thanks for the heads up. Actually, YOU can update the wiki. If you want me to do so I will be happy to. -MC From brian at freeswitch.org Thu Feb 12 15:01:08 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 12 Feb 2009 17:01:08 -0600 Subject: [Freeswitch-users] FS equiv for waitforextension In-Reply-To: References: <35b355e90902121001i3a049252x852336b18846ad21@mail.gmail.com><76D5D669-B367-4639-925F-DA213FC93539@freeswitch.org> <87f2f3b90902121014t29ab0e25u129c4c824d63e11e@mail.gmail.com> Message-ID: <7A3C8938-9231-4A61-93EB-7503D4778C85@freeswitch.org> YOU DO! ;) Its a user edited content portal. /b On Feb 12, 2009, at 4:58 PM, Nik Middleton wrote: > > Not sure who updates the WIKI, but it's wrong on collectinput for the > example. In the call, dtmf needs quotes, ie "dtmf" From nik.middleton at noblesolutions.co.uk Thu Feb 12 15:17:53 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Thu, 12 Feb 2009 23:17:53 -0000 Subject: [Freeswitch-users] FS equiv for waitforextension In-Reply-To: <7A3C8938-9231-4A61-93EB-7503D4778C85@freeswitch.org> References: <35b355e90902121001i3a049252x852336b18846ad21@mail.gmail.com><76D5D669-B367-4639-925F-DA213FC93539@freeswitch.org><87f2f3b90902121014t29ab0e25u129c4c824d63e11e@mail.gmail.com> <7A3C8938-9231-4A61-93EB-7503D4778C85@freeswitch.org> Message-ID: Done, that was easy, unlike FS :) -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: 12 February 2009 23:01 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] FS equiv for waitforextension YOU DO! ;) Its a user edited content portal. /b On Feb 12, 2009, at 4:58 PM, Nik Middleton wrote: > > Not sure who updates the WIKI, but it's wrong on collectinput for the > example. In the call, dtmf needs quotes, ie "dtmf" _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From lfurrea at gmail.com Thu Feb 12 15:31:16 2009 From: lfurrea at gmail.com (Luis F Urrea) Date: Thu, 12 Feb 2009 17:31:16 -0600 Subject: [Freeswitch-users] xml_cdr call flow In-Reply-To: <87f2f3b90902121500m2b86efc2q722314688d532648@mail.gmail.com> References: <87f2f3b90902121500m2b86efc2q722314688d532648@mail.gmail.com> Message-ID: Heres pastebin of the A-leg http://pastebin.com/m6731913d On Thu, Feb 12, 2009 at 5:00 PM, Michael Collins wrote: > Pastebin the whole file so that we can see it in context... > -MC > > On Thu, Feb 12, 2009 at 2:50 PM, Luis F Urrea wrote: > > On our test calls we haven't been able to correlate times from the A leg > > with times from the B leg. > > > > I would expect something as A-leg(duration)= > > B-leg1(duration)+B-leg2(duration) > > > > Also the tag within tag does not seem to be in epoch > > microseconds. so it does not seem that's where i should be looking for > that > > info. > > > > Here's an example of the tag for a test call on the A-Leg: > > > > > > 1233942283835696 > > 1233942283835696 > > 1233942283999716 > > 1233942283999716 > > 1233942287291931 > > 0 > > 1233942303240916 > > > > > > any hint is appreciated > > > > > > On Thu, Feb 12, 2009 at 3:46 PM, Luis F Urrea wrote: > >> > >> Hi all, > >> > >> We are writing a xml_cdr parser to load CDRs in SQLite. We are > interested > >> in logging times for both A leg and B leg so that transfers are reported > as > >> individual calls with accurate timing. eg Inboud call to AA lasted 14 > >> seconds then call to operator 20s and then call to actual extension 5min > >> > >> As of now we are using the tag with the "number" attribute to > >> find out who did the A leg talk to, then we open the B leg files and get > the > >> times from each jump from the tag within the tag on > the B > >> leg file. > >> > >> Is this right or maybe someone could suggest a better way to do it. > >> > >> TIA > >> > >> Luis > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090212/f15e763a/attachment-0001.html From msc at freeswitch.org Thu Feb 12 16:08:16 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 12 Feb 2009 16:08:16 -0800 Subject: [Freeswitch-users] xml_cdr call flow In-Reply-To: References: <87f2f3b90902121500m2b86efc2q722314688d532648@mail.gmail.com> Message-ID: <87f2f3b90902121608x217f682v289eae73012e27a5@mail.gmail.com> On Thu, Feb 12, 2009 at 3:31 PM, Luis F Urrea wrote: > Heres pastebin of the A-leg > > http://pastebin.com/m6731913d > > > On Thu, Feb 12, 2009 at 5:00 PM, Michael Collins wrote: >> >> Pastebin the whole file so that we can see it in context... >> -MC >> >> On Thu, Feb 12, 2009 at 2:50 PM, Luis F Urrea wrote: >> > On our test calls we haven't been able to correlate times from the A leg >> > with times from the B leg. >> > >> > I would expect something as A-leg(duration)= >> > B-leg1(duration)+B-leg2(duration) I don't know that this is necessarily true. Can you pastebin your dialplan entry (or whatever generated this call) so we can take a look? Also, please use our pastebin so that it's easier for us to find stuff: http://pastebin.freeswitch.org >> > >> > Also the tag within tag does not seem to be in epoch >> > microseconds. so it does not seem that's where i should be looking for >> > that >> > info. >> > >> > Here's an example of the tag for a test call on the A-Leg: >> > >> > >> > 1233942283835696 >> > 1233942283835696 >> > 1233942283999716 >> > 1233942283999716 >> > 1233942287291931 >> > 0 >> > 1233942303240916 >> > >> > >> > any hint is appreciated >> > Perhaps I'm missing something but they sure look like epoch microseconds to me. -MC From pauld at versafon.com Thu Feb 12 16:35:02 2009 From: pauld at versafon.com (pauld) Date: Thu, 12 Feb 2009 19:35:02 -0500 Subject: [Freeswitch-users] Cannot choose Cepstral voice from dialplan In-Reply-To: <9268AE39-6841-4819-9A61-37806B48BEFF@freeswitch.org> References: <498E67BF.3060207@versafon.com> <8B964172-6EEB-4B5B-B4B1-D35BFEAEB460@freeswitch.org> <498F6751.2000701@versafon.com> <87f2f3b90902090904i72e37241hb5d4808eb31fcc18@mail.gmail.com> <4993A7D8.1090004@versafon.com> <10703C0D-0BA2-4FB3-B669-B87BF93912E0@freeswitch.org> <49942D7A.2090503@versafon.com> <9268AE39-6841-4819-9A61-37806B48BEFF@freeswitch.org> Message-ID: <4994C036.30608@versafon.com> Brian West wrote: > You still didn't answer my question. How are you trying to do this > from the dialplan. > > /b > > On Feb 12, 2009, at 8:08 AM, pauld wrote: > > >> Yes I am using 5.1, I haven't done anything special other than >> followed >> wiki and then the advice given here to create symlinks in FS lib dir >> to all >> cepstral libs. I have cepstral libs in a standard location /opt/ >> swift/lib. >> I have given an example extension I used for testing earlier in this >> thread >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From helmut.kuper at ewetel.de Fri Feb 13 01:13:21 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Fri, 13 Feb 2009 10:13:21 +0100 Subject: [Freeswitch-users] segfault when shoutcast plays mp3 and extension hangs up In-Reply-To: <9CF2F413-1DE1-49A7-A550-6423C3444D14@freeswitch.org> References: <4992BFFC.50006@ewetel.de> <4992C801.4020906@ewetel.de> <191c3a030902110558i9702139g6b12aaf15f6d3aac@mail.gmail.com> <4992DCC2.3020702@ewetel.de> <191c3a030902110617n3dcd19cfr204c00792e1e87ed@mail.gmail.com> <4992E559.6060506@ewetel.de> <191c3a030902110729g30b30d46gac0f56205f8d48cf@mail.gmail.com> <4993D8CA.1010602@ewetel.de> <191c3a030902121420o3b4ec7a8j1fa0059b849bd866@mail.gmail.com> <9CF2F413-1DE1-49A7-A550-6423C3444D14@freeswitch.org> Message-ID: <499539B1.1000507@ewetel.de> Hello, it works now. I'm not really sure what it was, but I know what I did in what order: 1. update gcc to "gcc version 4.2.4 (Ubuntu 4.2.4-1ubuntu3)" 2. configure 3. make sure 4. make install 5 Tested it: FS still crash 6 did a gdb backtrace, last function call was mpg123_delete ... 7. delete lame directory an archive 8. delete mpg123 directory and archive 9. bootstrap.sh 10. configure ... 11. make sure 12. chmod 755 ./libs/libsndfile/src/create_symbols_file.py (execution permissions were missed) 13. autogen was missed, so I installed it 14. make all 15. make install 16. Tested it: SUCCESS - no crash anymore For me it seems to be caused by mpg123 and/or lame ... Maybe a simple recompile and reinstall helps as well. regards Helmut On 12.02.2009 23:38, Brian West wrote: > This is prob. why we don't see this crazy stuff on CentOS since the > compiler is 4.1.2 > > /b > From alex at sinapticode.ro Fri Feb 13 03:33:14 2009 From: alex at sinapticode.ro (Alexandru Nedelcu) Date: Fri, 13 Feb 2009 13:33:14 +0200 Subject: [Freeswitch-users] Problems with Originate Message-ID: <1234524794.4431.56.camel@gathern.lan> Hi all, I'm using "originate" to initiate calls, and streamFile to play audio files on the answered sessions. All the logic was encapsulated in a Javascript file. The problem with this setup is that origination_caller_id_number doesn't work from inside the JS file (when calling session.originate). My setup only works when doing a direct originate command, with the JS script attached as an application to it, i.e... originate {ignore_early_media=true}sofia/gateway/myprovider/873040711222222 '&javascript(dialer.js )' Now, my next problem ... this doesn't work properly because with "ignore_early_media=true" then "dialer.js" isn't executed on FAIL. And I need that. If "ignore_early_media" is not specified, then "dialer.js" executes, but the recording starts before the phone is answered. Can you give me any tips? Thanks, -- Alexandru Nedelcu Software Developer, Sinapticode From alex at sinapticode.ro Fri Feb 13 03:48:44 2009 From: alex at sinapticode.ro (Alexandru Nedelcu) Date: Fri, 13 Feb 2009 13:48:44 +0200 Subject: [Freeswitch-users] Problems with Originate In-Reply-To: <1234524794.4431.56.camel@gathern.lan> References: <1234524794.4431.56.camel@gathern.lan> Message-ID: <1234525724.4431.59.camel@gathern.lan> On Fri, 2009-02-13 at 13:33 +0200, Alexandru Nedelcu wrote: > The problem with this setup is that origination_caller_id_number doesn't > work from inside the JS file (when calling session.originate). I just discovered something interesting. When originating the call like this ... session = new Session("") instead of this ... session = new Session(); session.originate("") ... then it works. Is this some kind of bug, or what's the difference here? Thanks, -- Alexandru Nedelcu Software Developer, Sinapticode From nik.middleton at noblesolutions.co.uk Fri Feb 13 03:48:39 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Fri, 13 Feb 2009 11:48:39 -0000 Subject: [Freeswitch-users] Problems with Originate In-Reply-To: <1234524794.4431.56.camel@gathern.lan> References: <1234524794.4431.56.camel@gathern.lan> Message-ID: Use this method in js var session = new Session('{absolute_codec_string=PCMA,accountcode=54321,ignore_early_medi a=true,origination_caller_id_number=40711222222,originate_timeout=25}sof ia/gateway/myprovider/873040711222222); -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Alexandru Nedelcu Sent: 13 February 2009 11:33 To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Problems with Originate Hi all, I'm using "originate" to initiate calls, and streamFile to play audio files on the answered sessions. All the logic was encapsulated in a Javascript file. The problem with this setup is that origination_caller_id_number doesn't work from inside the JS file (when calling session.originate). My setup only works when doing a direct originate command, with the JS script attached as an application to it, i.e... originate {ignore_early_media=true}sofia/gateway/myprovider/873040711222222 '&javascript(dialer.js )' Now, my next problem ... this doesn't work properly because with "ignore_early_media=true" then "dialer.js" isn't executed on FAIL. And I need that. If "ignore_early_media" is not specified, then "dialer.js" executes, but the recording starts before the phone is answered. Can you give me any tips? Thanks, -- Alexandru Nedelcu Software Developer, Sinapticode _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From zolotov at altron.ua Fri Feb 13 04:10:01 2009 From: zolotov at altron.ua (Evgeniy Zolotov) Date: Fri, 13 Feb 2009 14:10:01 +0200 Subject: [Freeswitch-users] Sending channel variables Message-ID: <1234527001.5507.14.camel@opos20.altron.lan> Hello! I'm trying to make such scheme: ---> FS_A --> FS_B --> record Incoming calls to FS_A are redirected to FS_B with the help of this context: FS_B records them to the file: This works good. But I have a question - in what manner I can send back (from FS_B to FS_A) some channel variables? Thanks, Evgeniy. From anthony.minessale at gmail.com Fri Feb 13 05:54:59 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 13 Feb 2009 07:54:59 -0600 Subject: [Freeswitch-users] Problems with Originate In-Reply-To: <1234525724.4431.59.camel@gathern.lan> References: <1234524794.4431.56.camel@gathern.lan> <1234525724.4431.59.camel@gathern.lan> Message-ID: <191c3a030902130554w1dfe6b1dw9e5525cf6a210dc4@mail.gmail.com> The first way is deprecated and will be removed. The 2nd way is the correct way. On Fri, Feb 13, 2009 at 5:48 AM, Alexandru Nedelcu wrote: > On Fri, 2009-02-13 at 13:33 +0200, Alexandru Nedelcu wrote: > > The problem with this setup is that origination_caller_id_number doesn't > > work from inside the JS file (when calling session.originate). > > I just discovered something interesting. > > When originating the call like this ... > session = new Session("") > instead of this ... > session = new Session(); session.originate("") > > ... then it works. Is this some kind of bug, or what's the difference > here? > > Thanks, > > -- > Alexandru Nedelcu > Software Developer, Sinapticode > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090213/27f5aeaf/attachment.html From anthony.minessale at gmail.com Fri Feb 13 06:05:47 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 13 Feb 2009 08:05:47 -0600 Subject: [Freeswitch-users] Setting outbound callerid using js In-Reply-To: <87f2f3b90902111626x6db5dc61t68a0170a3460b42d@mail.gmail.com> References: <87f2f3b90902111626x6db5dc61t68a0170a3460b42d@mail.gmail.com> Message-ID: <191c3a030902130605m5e834e3dh27cc88387408b8b5@mail.gmail.com> 1) session.originate is depricated. 2) the first arg to session.originate is *another* session (not the same one) *or* undefined..... session.originate(undefined, ""); session.originate(a_leg_session, ""); session.originate(session, "") is asking the session to use itself as it's own a leg which makes no sense. This is perhaps the 4th time i have seen someone do this, can you point out where this is incorrectly documented? BTW effective_caller_id_name/number are variables you set on the A leg so when it's used to generate b legs that var is copied instead. a_leg_session.setVariable("effective_caller_id_number=1234"); b_leg_session = new Session(a_leg_session, ""); which is of course pointless because you never need to create the session if you just use the bridge application. session.execute("bridge", ""); even better just set the dest to a var and exit the script and use that var in your dialplan. --- contents of get_dest.js --- session.setVariable("dial_string", ""); -- dialplan -- the JavaScript paradigm is a bit different than the Lua/Perl one. Let > us know if that works or not. > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090213/d378128c/attachment-0001.html From nik.middleton at noblesolutions.co.uk Fri Feb 13 06:12:37 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Fri, 13 Feb 2009 14:12:37 -0000 Subject: [Freeswitch-users] Setting outbound callerid using js In-Reply-To: <191c3a030902130605m5e834e3dh27cc88387408b8b5@mail.gmail.com> References: <87f2f3b90902111626x6db5dc61t68a0170a3460b42d@mail.gmail.com> <191c3a030902130605m5e834e3dh27cc88387408b8b5@mail.gmail.com> Message-ID: I think this page (external) is the source http://alexn.org/docs/dialer.html Regards, ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: 13 February 2009 14:06 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Setting outbound callerid using js 1) session.originate is depricated. 2) the first arg to session.originate is *another* session (not the same one) *or* undefined..... session.originate(undefined, ""); session.originate(a_leg_session, ""); session.originate(session, "") is asking the session to use itself as it's own a leg which makes no sense. This is perhaps the 4th time i have seen someone do this, can you point out where this is incorrectly documented? BTW effective_caller_id_name/number are variables you set on the A leg so when it's used to generate b legs that var is copied instead. a_leg_session.setVariable("effective_caller_id_number=1234"); b_leg_session = new Session(a_leg_session, ""); which is of course pointless because you never need to create the session if you just use the bridge application. session.execute("bridge", ""); even better just set the dest to a var and exit the script and use that var in your dialplan. --- contents of get_dest.js --- session.setVariable("dial_string", ""); -- dialplan -- GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090213/a1900005/attachment.html From alex at sinapticode.ro Fri Feb 13 06:25:36 2009 From: alex at sinapticode.ro (Alexandru Nedelcu) Date: Fri, 13 Feb 2009 16:25:36 +0200 Subject: [Freeswitch-users] Setting outbound callerid using js In-Reply-To: References: <87f2f3b90902111626x6db5dc61t68a0170a3460b42d@mail.gmail.com> <191c3a030902130605m5e834e3dh27cc88387408b8b5@mail.gmail.com> Message-ID: <1234535136.4431.70.camel@gathern.lan> I wrote that document ... I can't remember from where I got the idea that you should specify the a-leg as being the same session. That document is a draft, but it got indexed by Google unfortunately :( On Fri, 2009-02-13 at 14:12 +0000, Nik Middleton wrote: > I think this page (external) is the source > > > > http://alexn.org/docs/dialer.html > > > > Regards, > > > > > ______________________________________________________________________ > From:freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Anthony Minessale > Sent: 13 February 2009 14:06 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Setting outbound callerid using js > > > > > 1) session.originate is depricated. > 2) the first arg to session.originate is *another* session (not the > same one) *or* undefined..... > session.originate(undefined, ""); > session.originate(a_leg_session, ""); > > session.originate(session, "") is asking the session to > use itself as it's own a leg which makes no sense. > > This is perhaps the 4th time i have seen someone do this, can you > point out where this is incorrectly documented? > > BTW > > effective_caller_id_name/number are variables you set on the A leg so > when it's used to generate b legs that var is copied instead. > > a_leg_session.setVariable("effective_caller_id_number=1234"); > b_leg_session = new Session(a_leg_session, ""); > > which is of course pointless because you never need to create the > session if you just use the bridge application. > > > session.execute("bridge", ""); > > even better just set the dest to a var and exit the script and use > that var in your dialplan. > > > --- contents of get_dest.js --- > session.setVariable("dial_string", ""); > > -- dialplan -- > > the JavaScript paradigm is a bit different than the Lua/Perl one. Let > us know if that works or not. > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From jan.kubr at gmail.com Fri Feb 13 06:27:17 2009 From: jan.kubr at gmail.com (Jan Kubr) Date: Fri, 13 Feb 2009 15:27:17 +0100 Subject: [Freeswitch-users] Ruby framework for event socket Message-ID: <698401620902130627l5e8dec56g5fd43bbeb8b9ac0f@mail.gmail.com> Hi all, I've created a simple framework in Ruby that you can use to talk to Freeswitch via even socket outbound. It won't suite your needs perfectly if you are doing anything non-trivial, but it might be a nice starting point. Check it out at http://github.com/jankubr/freec Cheers, Jan Kubr From lfurrea at gmail.com Fri Feb 13 06:42:11 2009 From: lfurrea at gmail.com (Luis F Urrea) Date: Fri, 13 Feb 2009 08:42:11 -0600 Subject: [Freeswitch-users] xml_cdr call flow In-Reply-To: <87f2f3b90902121608x217f682v289eae73012e27a5@mail.gmail.com> References: <87f2f3b90902121500m2b86efc2q722314688d532648@mail.gmail.com> <87f2f3b90902121608x217f682v289eae73012e27a5@mail.gmail.com> Message-ID: My mistake, they do seem to be microsecs. But still I cannot correlate times from the A-leg with the B-legs. I have included below the xml_cdr files generated for the test call. The test call was made using three registered extensions. Basically, Ext 201 calls ext 203 and they talk, then 203 blindly transfers to 202, 202 does not answer and call rolls to voicemail. A-leg: http://pastebin.freeswitch.org/7206 B-leg: http://pastebin.freeswitch.org/7204 B-leg: http://pastebin.freeswitch.org/7205 Thanks for your help On Thu, Feb 12, 2009 at 6:08 PM, Michael Collins wrote: > On Thu, Feb 12, 2009 at 3:31 PM, Luis F Urrea wrote: > > Heres pastebin of the A-leg > > > > http://pastebin.com/m6731913d > > > > > > On Thu, Feb 12, 2009 at 5:00 PM, Michael Collins > wrote: > >> > >> Pastebin the whole file so that we can see it in context... > >> -MC > >> > >> On Thu, Feb 12, 2009 at 2:50 PM, Luis F Urrea > wrote: > >> > On our test calls we haven't been able to correlate times from the A > leg > >> > with times from the B leg. > >> > > >> > I would expect something as A-leg(duration)= > >> > B-leg1(duration)+B-leg2(duration) > I don't know that this is necessarily true. Can you pastebin your > dialplan entry (or whatever generated this call) so we can take a > look? Also, please use our pastebin so that it's easier for us to find > stuff: > http://pastebin.freeswitch.org > > >> > > >> > Also the tag within tag does not seem to be in > epoch > >> > microseconds. so it does not seem that's where i should be looking for > >> > that > >> > info. > >> > > >> > Here's an example of the tag for a test call on the A-Leg: > >> > > >> > > >> > 1233942283835696 > >> > 1233942283835696 > >> > 1233942283999716 > >> > 1233942283999716 > >> > 1233942287291931 > >> > 0 > >> > 1233942303240916 > >> > > >> > > >> > any hint is appreciated > >> > > Perhaps I'm missing something but they sure look like epoch microseconds to > me. > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090213/51a2ae45/attachment.html From sicfslist at gmail.com Fri Feb 13 06:47:36 2009 From: sicfslist at gmail.com (Shelby Ramsey) Date: Fri, 13 Feb 2009 08:47:36 -0600 Subject: [Freeswitch-users] Sending channel variables In-Reply-To: <1234527001.5507.14.camel@opos20.altron.lan> References: <1234527001.5507.14.camel@opos20.altron.lan> Message-ID: <35b355e90902130647k1726383co4ea360f2fb3db628@mail.gmail.com> I'm assuming that you are saying these are 2 boxes .... if the protocol is a sip you can append a sip header ... _sip_h_X- .... This should be available as a channel variable on FS A. SDR On Fri, Feb 13, 2009 at 6:10 AM, Evgeniy Zolotov wrote: > Hello! > > I'm trying to make such scheme: > > ---> FS_A --> FS_B --> record > > Incoming calls to FS_A are redirected to FS_B with the help of this > context: > > > > > > data="sofia/outbound/$1 at 1.2.3.4:5080" /> > > > > > FS_B records them to the file: > > > > > > data="$${base_dir}/recordings/test/testrec.wav" /> > > > > This works good. But I have a question - in what manner I can send back > (from FS_B to FS_A) some channel variables? > > Thanks, Evgeniy. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090213/70717dde/attachment-0001.html From alex at sinapticode.ro Fri Feb 13 06:53:38 2009 From: alex at sinapticode.ro (Alexandru Nedelcu) Date: Fri, 13 Feb 2009 16:53:38 +0200 Subject: [Freeswitch-users] http://alexn.org/docs/dialer.html (was: Setting outbound callerid using js) In-Reply-To: <1234535136.4431.70.camel@gathern.lan> References: <87f2f3b90902111626x6db5dc61t68a0170a3460b42d@mail.gmail.com> <191c3a030902130605m5e834e3dh27cc88387408b8b5@mail.gmail.com> <1234535136.4431.70.camel@gathern.lan> Message-ID: <1234536818.4431.81.camel@gathern.lan> Btw ... I fixed the document. Sorry about that guys, I'm a rookie and I thought other people would find my setup useful. Can you guys read it and tell me if it contains other mistakes? My intention was to publish it on the wiki once it was ready, but I temporarily moved on to another project. http://alexn.org/docs/dialer.html Thanks, On Fri, 2009-02-13 at 16:25 +0200, Alexandru Nedelcu wrote: > I wrote that document ... I can't remember from where I got the idea > that you should specify the a-leg as being the same session. > > That document is a draft, but it got indexed by Google unfortunately :( > > > On Fri, 2009-02-13 at 14:12 +0000, Nik Middleton wrote: > > I think this page (external) is the source > > > > http://alexn.org/docs/dialer.html From ivdreg at gmail.com Fri Feb 13 06:57:48 2009 From: ivdreg at gmail.com (ivdreg ivdreg) Date: Fri, 13 Feb 2009 16:57:48 +0200 Subject: [Freeswitch-users] Codec negotiation questions In-Reply-To: <191c3a030902121306l1dae21a3m614f400703d15bd2@mail.gmail.com> References: <191c3a030902121306l1dae21a3m614f400703d15bd2@mail.gmail.com> Message-ID: Hi Anthony, Excuse me if I'm wrong but inbound-late-negotiation must be used proxy_media as I see in documentation. I don't want to proxy media because of some issues with MOH or 3-way conferencing. Also I want to exclude media codecs that are supported only in pass-trough mode. Let mi give you an example: SDP from caller v=0 o=- 1 2 IN IP4 192.168.20.193 s=CounterPath eyeBeam 1.5 c=IN IP4 192.168.40.81 t=0 0 m=audio 56888 RTP/AVP 100 106 97 105 98 3 101 a=fmtp:101 0-15 a=rtpmap:100 SPEEX/16000 a=rtpmap:106 SPEEX-FEC/16000 a=rtpmap:97 SPEEX/8000 a=rtpmap:105 SPEEX-FEC/8000 a=rtpmap:98 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv a=x-rtp-session-id:61940567309B49E8909127E1393A966E m=video 46378 RTP/AVP 125 115 34 a=fmtp:125 profile-level-id=42e015; max-br=4000; max-mbps=19800 a=fmtp:115 QCIF=1 MAXBR=4520 a=fmtp:34 QCIF=1 MAXBR=4520 a=rtpmap:125 H264/90000 a=rtpmap:115 H263-1998/90000 a=rtpmap:34 H263/90000 a=sendrecv a=x-rtp-session-id:E8244E608F65445BA183BBE641C5DF3C a=nortpproxy:yes SDP from Freeswitch to called v=0 o=FreeSWITCH 6228154995644782318 4633980766357417433 IN IP4 10.10.10.10 s=FreeSWITCH c=IN IP4 10.10.10.10 t=0 0 m=audio 26920 RTP/AVP 3 101 13 * a=rtpmap:3 GSM/8000* a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 So we offer *only first* in codec preference list from called wich is normal receives SIP/2.0 488 Not Acceptable Here *called suports - PCMA,PCMU,iLBC * Codec preference to this vars.xml we have witch is used in provile: also we have in profile: In dialplan I've set: About my second question: Why I should parse variable_switch_r_sdp: [v=0 o=- 6 2 IN IP4 192.168.20.193 s=CounterPath eyeBeam 1.5 c=IN IP4 192.168.40.81 t=0 0 m=audio 60642 RTP/AVP 100 106 97 105 98 3 101 a=rtpmap:100 SPEEX/16000 a=rtpmap:106 SPEEX-FEC/16000 a=rtpmap:97 SPEEX/8000 a=rtpmap:105 SPEEX-FEC/8000 a=rtpmap:98 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=x-rtp-session-id:0674790128EB43ACB8C7F55829BCFF14 m=video 44938 RTP/AVP 125 115 34 a=rtpmap:125 H264/90000 a=fmtp:125 profile-level-id=42e015; max-br=4000; max-mbps=19800 a=rtpmap:115 H263-1998/90000 a=fmtp:115 QCIF=1 MAXBR=4520 a=rtpmap:34 H263/90000 a=fmtp:34 QCIF=1 MAXBR=4520 a=x-rtp-session-id:D1EAE543055746AC9C03006B91ADE6DF a=nortpproxy:yes ] In FS core this parse is already done I'm sure in much more intelligent way. It can be exported as a variable like a absolute codec string I think. Thanks again. 2009/2/12 Anthony Minessale > the entire sdp is available as a variable (route the call to the info app > to see the variables) > so if you have inbound-late-negotiation set to true on the sip profile > then you can use a regex or a script to set absolute_codec string before > you answer. > > > On Thu, Feb 12, 2009 at 8:06 AM, ivdreg ivdreg wrote: > >> Hi all, >> >> Can I ask 2 questions about codec negotiation: >> >> 1. Is it possible Freeswitch to work negotiate codecs between two phones >> as it is described below. >> INVITE from A with some codecs in SDP ---> Freeswitch rewrites codec >> preference according absolute_codec_string but exclude all codecs not >> offered by A ----> INVITE to B with rewrited SDP. >> >> example: >> from A SDP:PCMA,PCMU,SPEEX ----> absolute_codec_string=G722,PCMU,PCMA,GSM >> ----> to B SDP: PCMU,PCMA >> >> 2. Can I get codec list in INVITE with mod_perl for example or via >> xml_curl without processing SDP variable (switch_r_sdp). It will be useful >> to be in format that absolute_codec_string variable takes. >> >> Thanks >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090213/c27eeee3/attachment.html From helmut.kuper at ewetel.de Fri Feb 13 07:00:02 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Fri, 13 Feb 2009 16:00:02 +0100 Subject: [Freeswitch-users] Question: SIP BYE authentication In-Reply-To: References: <4992C772.4090906@ewetel.de> <4993DF0D.9000403@ewetel.de> Message-ID: <49958AF2.4010407@ewetel.de> Hi, yes, I'm using gateway, but it ignores softswitch side challenges for BYE messages coming from FS. My dialplan: regards helmut On 12.02.2009 18:01, Michael Jerris wrote: > If using gayeway it should already do this. > > On Feb 12, 2009, at 3:34 AM, Helmut Kuper > wrote: > From anthony.minessale at gmail.com Fri Feb 13 07:27:29 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 13 Feb 2009 09:27:29 -0600 Subject: [Freeswitch-users] Setting outbound callerid using js In-Reply-To: <1234535136.4431.70.camel@gathern.lan> References: <87f2f3b90902111626x6db5dc61t68a0170a3460b42d@mail.gmail.com> <191c3a030902130605m5e834e3dh27cc88387408b8b5@mail.gmail.com> <1234535136.4431.70.camel@gathern.lan> Message-ID: <191c3a030902130727y132a55c2v8fb27e87576154da@mail.gmail.com> I made it an error now to do it this way which should clear things up. Doing it that way probably led to instability in js. On Fri, Feb 13, 2009 at 8:25 AM, Alexandru Nedelcu wrote: > I wrote that document ... I can't remember from where I got the idea > that you should specify the a-leg as being the same session. > > That document is a draft, but it got indexed by Google unfortunately :( > > > On Fri, 2009-02-13 at 14:12 +0000, Nik Middleton wrote: > > I think this page (external) is the source > > > > > > > > http://alexn.org/docs/dialer.html > > > > > > > > Regards, > > > > > > > > > > ______________________________________________________________________ > > From:freeswitch-users-bounces at lists.freeswitch.org > > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > > Anthony Minessale > > Sent: 13 February 2009 14:06 > > To: freeswitch-users at lists.freeswitch.org > > Subject: Re: [Freeswitch-users] Setting outbound callerid using js > > > > > > > > > > 1) session.originate is depricated. > > 2) the first arg to session.originate is *another* session (not the > > same one) *or* undefined..... > > session.originate(undefined, ""); > > session.originate(a_leg_session, ""); > > > > session.originate(session, "") is asking the session to > > use itself as it's own a leg which makes no sense. > > > > This is perhaps the 4th time i have seen someone do this, can you > > point out where this is incorrectly documented? > > > > BTW > > > > effective_caller_id_name/number are variables you set on the A leg so > > when it's used to generate b legs that var is copied instead. > > > > a_leg_session.setVariable("effective_caller_id_number=1234"); > > b_leg_session = new Session(a_leg_session, ""); > > > > which is of course pointless because you never need to create the > > session if you just use the bridge application. > > > > > > session.execute("bridge", ""); > > > > even better just set the dest to a var and exit the script and use > > that var in your dialplan. > > > > > > --- contents of get_dest.js --- > > session.setVariable("dial_string", ""); > > > > -- dialplan -- > > > > > the JavaScript paradigm is a bit different than the Lua/Perl one. Let > > us know if that works or not. > > -MC > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090213/7c2ec5e7/attachment-0001.html From saeedahmad1981 at gmail.com Fri Feb 13 07:36:50 2009 From: saeedahmad1981 at gmail.com (Saeed Ahmed) Date: Fri, 13 Feb 2009 16:36:50 +0100 Subject: [Freeswitch-users] FS + Call Center Solution In-Reply-To: <87f2f3b90902121202m5652f0e3x42eabb6ced2fd646@mail.gmail.com> References: <6309E7515E4F43159B9564800564B562@SaeedLaptop> <87f2f3b90902121202m5652f0e3x42eabb6ced2fd646@mail.gmail.com> Message-ID: <3BF8F4AE71434041994251CFC355785C@SaeedLaptop> Thanks -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Thursday, February 12, 2009 9:02 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] FS + Call Center Solution On Wed, Feb 11, 2009 at 8:31 AM, Saeed Ahmed wrote: > Hi List, > > Is there any open source call center tool available which works with FS? Check this out: http://opencsm.org/wiki/index.php/Spice_Telephony -MC _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From anthony.minessale at gmail.com Fri Feb 13 07:40:35 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 13 Feb 2009 09:40:35 -0600 Subject: [Freeswitch-users] Question: SIP BYE authentication In-Reply-To: <49958AF2.4010407@ewetel.de> References: <4992C772.4090906@ewetel.de> <4993DF0D.9000403@ewetel.de> <49958AF2.4010407@ewetel.de> Message-ID: <191c3a030902130740y201f1a5dr573e382f80e49111@mail.gmail.com> Turn up debug and look harder are you sure it does not say "no matching gateway" when it gets the challenge to bye? On Fri, Feb 13, 2009 at 9:00 AM, Helmut Kuper wrote: > Hi, > > yes, I'm using gateway, but it ignores softswitch side challenges for > BYE messages coming from FS. > > My dialplan: > > expression="^0([0-9]+|^940[0-9]+)" break="on-false"> > > > expression="^(4918|4919)$"> > > data="ignore_early_media=true"/> > data="effective_caller_id_name="/> > data="effective_caller_id_number=1234$1"/> > data="sofia/gateway/SIPDDI/${dialed}@sip2.ewetel.net > "/> > > > > regards > helmut > > > On 12.02.2009 18:01, Michael Jerris wrote: > > If using gayeway it should already do this. > > > > On Feb 12, 2009, at 3:34 AM, Helmut Kuper > > wrote: > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090213/075dc2db/attachment.html From anthony.minessale at gmail.com Fri Feb 13 07:46:15 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 13 Feb 2009 09:46:15 -0600 Subject: [Freeswitch-users] Codec negotiation questions In-Reply-To: References: <191c3a030902121306l1dae21a3m614f400703d15bd2@mail.gmail.com> Message-ID: <191c3a030902130746h23018e17n5e5c6c4900233097@mail.gmail.com> yes you are wrong. inbound-late-negotiation setting delays the codec negotiation until the instant audio is needed. It is not tied to inbound-proxy-media. This allows the call to come into the dialplan before any codec negotiation is done giving you a chance to look at the SDP before the negotiation takes place and insert an absolute_codec string essentially letting you chose unique codec preferences per inbound call. >>> Why I should parse variable_switch_r_sdp Well....you must parse it because it's you who cares about what it says, as described above it lets you peek at the sdp and enforce a unique set of codec prefs per call. On Fri, Feb 13, 2009 at 8:57 AM, ivdreg ivdreg wrote: > Hi Anthony, > > Excuse me if I'm wrong but inbound-late-negotiation must be used > proxy_media as I see in documentation. I don't want to proxy media because > of some issues with MOH or 3-way conferencing. Also I want to exclude media > codecs that are supported only in pass-trough mode. Let mi give you an > example: > > SDP from caller > > v=0 > o=- 1 2 IN IP4 192.168.20.193 > s=CounterPath eyeBeam 1.5 > c=IN IP4 192.168.40.81 > t=0 0 > m=audio 56888 RTP/AVP 100 106 97 105 98 3 101 > a=fmtp:101 0-15 > a=rtpmap:100 SPEEX/16000 > a=rtpmap:106 SPEEX-FEC/16000 > a=rtpmap:97 SPEEX/8000 > a=rtpmap:105 SPEEX-FEC/8000 > a=rtpmap:98 iLBC/8000 > a=rtpmap:101 telephone-event/8000 > a=sendrecv > a=x-rtp-session-id:61940567309B49E8909127E1393A966E > m=video 46378 RTP/AVP 125 115 34 > a=fmtp:125 profile-level-id=42e015; max-br=4000; max-mbps=19800 > a=fmtp:115 QCIF=1 MAXBR=4520 > a=fmtp:34 QCIF=1 MAXBR=4520 > a=rtpmap:125 H264/90000 > a=rtpmap:115 H263-1998/90000 > a=rtpmap:34 H263/90000 > a=sendrecv > a=x-rtp-session-id:E8244E608F65445BA183BBE641C5DF3C > a=nortpproxy:yes > > SDP from Freeswitch to called > > v=0 > o=FreeSWITCH 6228154995644782318 4633980766357417433 IN IP4 10.10.10.10 > s=FreeSWITCH > c=IN IP4 10.10.10.10 > t=0 0 > m=audio 26920 RTP/AVP 3 101 13 > * a=rtpmap:3 GSM/8000* > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=rtpmap:13 CN/8000 > a=ptime:20 > > So we offer *only first* in codec preference list > > from called wich is normal receives > SIP/2.0 488 Not Acceptable Here > *called suports - PCMA,PCMU,iLBC > * > Codec preference to this vars.xml we have witch is used in provile: > > also we have in profile: > > > In dialplan I've set: > > > > About my second question: > Why I should parse variable_switch_r_sdp: [v=0 > o=- 6 2 IN IP4 192.168.20.193 > s=CounterPath eyeBeam 1.5 > c=IN IP4 192.168.40.81 > t=0 0 > m=audio 60642 RTP/AVP 100 106 97 105 98 3 101 > a=rtpmap:100 SPEEX/16000 > a=rtpmap:106 SPEEX-FEC/16000 > a=rtpmap:97 SPEEX/8000 > a=rtpmap:105 SPEEX-FEC/8000 > a=rtpmap:98 iLBC/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=x-rtp-session-id:0674790128EB43ACB8C7F55829BCFF14 > m=video 44938 RTP/AVP 125 115 34 > a=rtpmap:125 H264/90000 > a=fmtp:125 profile-level-id=42e015; max-br=4000; max-mbps=19800 > a=rtpmap:115 H263-1998/90000 > a=fmtp:115 QCIF=1 MAXBR=4520 > a=rtpmap:34 H263/90000 > a=fmtp:34 QCIF=1 MAXBR=4520 > a=x-rtp-session-id:D1EAE543055746AC9C03006B91ADE6DF > a=nortpproxy:yes > ] > In FS core this parse is already done I'm sure in much more intelligent > way. It can be exported as a variable like a absolute codec string I think. > > Thanks again. > > 2009/2/12 Anthony Minessale > > the entire sdp is available as a variable (route the call to the info app >> to see the variables) >> so if you have inbound-late-negotiation set to true on the sip profile >> then you can use a regex or a script to set absolute_codec string before >> you answer. >> >> >> On Thu, Feb 12, 2009 at 8:06 AM, ivdreg ivdreg wrote: >> >>> Hi all, >>> >>> Can I ask 2 questions about codec negotiation: >>> >>> 1. Is it possible Freeswitch to work negotiate codecs between two phones >>> as it is described below. >>> INVITE from A with some codecs in SDP ---> Freeswitch rewrites codec >>> preference according absolute_codec_string but exclude all codecs not >>> offered by A ----> INVITE to B with rewrited SDP. >>> >>> example: >>> from A SDP:PCMA,PCMU,SPEEX ----> absolute_codec_string=G722,PCMU,PCMA,GSM >>> ----> to B SDP: PCMU,PCMA >>> >>> 2. Can I get codec list in INVITE with mod_perl for example or via >>> xml_curl without processing SDP variable (switch_r_sdp). It will be useful >>> to be in format that absolute_codec_string variable takes. >>> >>> Thanks >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:213-799-1400 >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090213/b4d736f6/attachment-0001.html From anthony.minessale at gmail.com Fri Feb 13 07:57:38 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 13 Feb 2009 09:57:38 -0600 Subject: [Freeswitch-users] xml_cdr call flow In-Reply-To: References: <87f2f3b90902121500m2b86efc2q722314688d532648@mail.gmail.com> <87f2f3b90902121608x217f682v289eae73012e27a5@mail.gmail.com> Message-ID: <191c3a030902130757v757dfa3foaaf35e64fbfb5c82@mail.gmail.com> each b leg call on the a leg shows up in a tag At the bottom of http://pastebin.freeswitch.org/7206 1233942283835696 1233942283835696 1233942283999716 1233942283999716 1233942287291931 0 1233942303240916 note transfer time, this is the time that the call was transferred to another extension. 1233942303240916 divide this number by one million to get epoch time 1233942303240916 / 1000000 == 1233942303 now look at http://pastebin.freeswitch.org/7204 at the bottom 1233942303325062 1233942303325062 1233942303368916 0 0 1233942333010768 0 This is the b leg, see the created_time: 1233942303325062 1233942303325062 / 1000000 == 1233942303 so as you can see the epoch time of your b leg cdr has a created_time that is the same one second window that corresponds to the transfer_time in the callflow tag in your a leg cdr On Fri, Feb 13, 2009 at 8:42 AM, Luis F Urrea wrote: > My mistake, they do seem to be microsecs. > > > But still I cannot correlate times from the A-leg with the B-legs. > > I have included below the xml_cdr files generated for the test call. > > The test call was made using three registered extensions. Basically, Ext > 201 calls ext 203 and they talk, then 203 blindly transfers to 202, 202 does > not answer and call rolls to voicemail. > > A-leg: > http://pastebin.freeswitch.org/7206 > > B-leg: > http://pastebin.freeswitch.org/7204 > > B-leg: > http://pastebin.freeswitch.org/7205 > > > Thanks for your help > > > On Thu, Feb 12, 2009 at 6:08 PM, Michael Collins wrote: > >> On Thu, Feb 12, 2009 at 3:31 PM, Luis F Urrea wrote: >> > Heres pastebin of the A-leg >> > >> > http://pastebin.com/m6731913d >> > >> > >> > On Thu, Feb 12, 2009 at 5:00 PM, Michael Collins >> wrote: >> >> >> >> Pastebin the whole file so that we can see it in context... >> >> -MC >> >> >> >> On Thu, Feb 12, 2009 at 2:50 PM, Luis F Urrea >> wrote: >> >> > On our test calls we haven't been able to correlate times from the A >> leg >> >> > with times from the B leg. >> >> > >> >> > I would expect something as A-leg(duration)= >> >> > B-leg1(duration)+B-leg2(duration) >> I don't know that this is necessarily true. Can you pastebin your >> dialplan entry (or whatever generated this call) so we can take a >> look? Also, please use our pastebin so that it's easier for us to find >> stuff: >> http://pastebin.freeswitch.org >> >> >> > >> >> > Also the tag within tag does not seem to be in >> epoch >> >> > microseconds. so it does not seem that's where i should be looking >> for >> >> > that >> >> > info. >> >> > >> >> > Here's an example of the tag for a test call on the A-Leg: >> >> > >> >> > >> >> > 1233942283835696 >> >> > 1233942283835696 >> >> > 1233942283999716 >> >> > 1233942283999716 >> >> > 1233942287291931 >> >> > 0 >> >> > 1233942303240916 >> >> > >> >> > >> >> > any hint is appreciated >> >> > >> Perhaps I'm missing something but they sure look like epoch microseconds >> to me. >> -MC >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090213/83c53fb0/attachment.html From helmut.kuper at ewetel.de Fri Feb 13 08:03:02 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Fri, 13 Feb 2009 17:03:02 +0100 Subject: [Freeswitch-users] Question: SIP BYE authentication In-Reply-To: <191c3a030902130740y201f1a5dr573e382f80e49111@mail.gmail.com> References: <4992C772.4090906@ewetel.de> <4993DF0D.9000403@ewetel.de> <49958AF2.4010407@ewetel.de> <191c3a030902130740y201f1a5dr573e382f80e49111@mail.gmail.com> Message-ID: <499599B6.2030106@ewetel.de> Hi Anthony, yes you are right, sorry. 2009-02-13 16:56:20 [ERR] sofia_reg.c:1358 sofia_reg_handle_sip_r_challenge() No Matching gateway found works now :) thx regards helmut On 13.02.2009 16:40, Anthony Minessale wrote: > Turn up debug and look harder are you sure it does not say "no > matching gateway" when it gets the challenge to bye? From ivdreg at gmail.com Fri Feb 13 08:50:00 2009 From: ivdreg at gmail.com (ivdreg ivdreg) Date: Fri, 13 Feb 2009 18:50:00 +0200 Subject: [Freeswitch-users] Codec negotiation questions In-Reply-To: <191c3a030902130746h23018e17n5e5c6c4900233097@mail.gmail.com> References: <191c3a030902121306l1dae21a3m614f400703d15bd2@mail.gmail.com> <191c3a030902130746h23018e17n5e5c6c4900233097@mail.gmail.com> Message-ID: Hi Anthony, I'm not sure that you understood the problem. As it shown bellow the offered codec in leg B contains only one codec (first matched in codec preference list for this profile). Is there way to offer in leg B not only first codec but all codecs that exists in INVITE in leg A that matches codec preference list. If not is the only way is to parse SDP and set absolute_codec_string manualy? Regards 2009/2/13 Anthony Minessale > yes you are wrong. > > inbound-late-negotiation setting delays the codec negotiation until the > instant audio is needed. > It is not tied to inbound-proxy-media. > > > This allows the call to come into the dialplan before any codec negotiation > is done giving you a chance to look at the SDP before the negotiation takes > place and insert an absolute_codec string essentially letting you chose > unique codec preferences per inbound call. > > >>> Why I should parse variable_switch_r_sdp > > Well....you must parse it because it's you who cares about what it says, as > described above it lets you peek at the sdp and enforce a unique set of > codec prefs per call. > > > > > On Fri, Feb 13, 2009 at 8:57 AM, ivdreg ivdreg wrote: > >> Hi Anthony, >> >> Excuse me if I'm wrong but inbound-late-negotiation must be used >> proxy_media as I see in documentation. I don't want to proxy media because >> of some issues with MOH or 3-way conferencing. Also I want to exclude media >> codecs that are supported only in pass-trough mode. Let mi give you an >> example: >> >> SDP from caller >> >> v=0 >> o=- 1 2 IN IP4 192.168.20.193 >> s=CounterPath eyeBeam 1.5 >> c=IN IP4 192.168.40.81 >> t=0 0 >> m=audio 56888 RTP/AVP 100 106 97 105 98 3 101 >> a=fmtp:101 0-15 >> a=rtpmap:100 SPEEX/16000 >> a=rtpmap:106 SPEEX-FEC/16000 >> a=rtpmap:97 SPEEX/8000 >> a=rtpmap:105 SPEEX-FEC/8000 >> a=rtpmap:98 iLBC/8000 >> a=rtpmap:101 telephone-event/8000 >> a=sendrecv >> a=x-rtp-session-id:61940567309B49E8909127E1393A966E >> m=video 46378 RTP/AVP 125 115 34 >> a=fmtp:125 profile-level-id=42e015; max-br=4000; max-mbps=19800 >> a=fmtp:115 QCIF=1 MAXBR=4520 >> a=fmtp:34 QCIF=1 MAXBR=4520 >> a=rtpmap:125 H264/90000 >> a=rtpmap:115 H263-1998/90000 >> a=rtpmap:34 H263/90000 >> a=sendrecv >> a=x-rtp-session-id:E8244E608F65445BA183BBE641C5DF3C >> a=nortpproxy:yes >> >> SDP from Freeswitch to called >> >> v=0 >> o=FreeSWITCH 6228154995644782318 4633980766357417433 IN IP4 10.10.10.10 >> s=FreeSWITCH >> c=IN IP4 10.10.10.10 >> t=0 0 >> m=audio 26920 RTP/AVP 3 101 13 >> * a=rtpmap:3 GSM/8000* >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> a=rtpmap:13 CN/8000 >> a=ptime:20 >> >> So we offer *only first* in codec preference list >> >> from called wich is normal receives >> SIP/2.0 488 Not Acceptable Here >> *called suports - PCMA,PCMU,iLBC >> * >> Codec preference to this vars.xml we have witch is used in provile: >> >> also we have in profile: >> >> >> In dialplan I've set: >> >> >> >> About my second question: >> Why I should parse variable_switch_r_sdp: [v=0 >> o=- 6 2 IN IP4 192.168.20.193 >> s=CounterPath eyeBeam 1.5 >> c=IN IP4 192.168.40.81 >> t=0 0 >> m=audio 60642 RTP/AVP 100 106 97 105 98 3 101 >> a=rtpmap:100 SPEEX/16000 >> a=rtpmap:106 SPEEX-FEC/16000 >> a=rtpmap:97 SPEEX/8000 >> a=rtpmap:105 SPEEX-FEC/8000 >> a=rtpmap:98 iLBC/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-15 >> a=x-rtp-session-id:0674790128EB43ACB8C7F55829BCFF14 >> m=video 44938 RTP/AVP 125 115 34 >> a=rtpmap:125 H264/90000 >> a=fmtp:125 profile-level-id=42e015; max-br=4000; max-mbps=19800 >> a=rtpmap:115 H263-1998/90000 >> a=fmtp:115 QCIF=1 MAXBR=4520 >> a=rtpmap:34 H263/90000 >> a=fmtp:34 QCIF=1 MAXBR=4520 >> a=x-rtp-session-id:D1EAE543055746AC9C03006B91ADE6DF >> a=nortpproxy:yes >> ] >> In FS core this parse is already done I'm sure in much more intelligent >> way. It can be exported as a variable like a absolute codec string I think. >> >> Thanks again. >> >> 2009/2/12 Anthony Minessale >> >> the entire sdp is available as a variable (route the call to the info app >>> to see the variables) >>> so if you have inbound-late-negotiation set to true on the sip profile >>> then you can use a regex or a script to set absolute_codec string before >>> you answer. >>> >>> >>> On Thu, Feb 12, 2009 at 8:06 AM, ivdreg ivdreg wrote: >>> >>>> Hi all, >>>> >>>> Can I ask 2 questions about codec negotiation: >>>> >>>> 1. Is it possible Freeswitch to work negotiate codecs between two phones >>>> as it is described below. >>>> INVITE from A with some codecs in SDP ---> Freeswitch rewrites codec >>>> preference according absolute_codec_string but exclude all codecs not >>>> offered by A ----> INVITE to B with rewrited SDP. >>>> >>>> example: >>>> from A SDP:PCMA,PCMU,SPEEX ----> absolute_codec_string=G722,PCMU,PCMA,GSM >>>> ----> to B SDP: PCMU,PCMA >>>> >>>> 2. Can I get codec list in INVITE with mod_perl for example or via >>>> xml_curl without processing SDP variable (switch_r_sdp). It will be useful >>>> to be in format that absolute_codec_string variable takes. >>>> >>>> Thanks >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> Anthony Minessale II >>> >>> FreeSWITCH http://www.freeswitch.org/ >>> ClueCon http://www.cluecon.com/ >>> >>> AIM: anthm >>> MSN:anthony_minessale at hotmail.com >>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>> IRC: irc.freenode.net #freeswitch >>> >>> FreeSWITCH Developer Conference >>> sip:888 at conference.freeswitch.org >>> iax:guest at conference.freeswitch.org/888 >>> googletalk:conf+888 at conference.freeswitch.org >>> pstn:213-799-1400 >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090213/edfd036b/attachment-0001.html From msc at freeswitch.org Fri Feb 13 08:57:41 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 13 Feb 2009 08:57:41 -0800 Subject: [Freeswitch-users] Ruby framework for event socket In-Reply-To: <698401620902130627l5e8dec56g5fd43bbeb8b9ac0f@mail.gmail.com> References: <698401620902130627l5e8dec56g5fd43bbeb8b9ac0f@mail.gmail.com> Message-ID: <87f2f3b90902130857g635bd8bfp37e67b96425e4bb@mail.gmail.com> On Fri, Feb 13, 2009 at 6:27 AM, Jan Kubr wrote: > Hi all, > I've created a simple framework in Ruby that you can use to talk to > Freeswitch via even socket outbound. It won't suite your needs > perfectly if you are doing anything non-trivial, but it might be a > nice starting point. > Check it out at http://github.com/jankubr/freec > Thanks for sharing your work with the community! It is appreciated. -MC From msc at freeswitch.org Fri Feb 13 09:00:08 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 13 Feb 2009 09:00:08 -0800 Subject: [Freeswitch-users] http://alexn.org/docs/dialer.html (was: Setting outbound callerid using js) In-Reply-To: <1234536818.4431.81.camel@gathern.lan> References: <87f2f3b90902111626x6db5dc61t68a0170a3460b42d@mail.gmail.com> <191c3a030902130605m5e834e3dh27cc88387408b8b5@mail.gmail.com> <1234535136.4431.70.camel@gathern.lan> <1234536818.4431.81.camel@gathern.lan> Message-ID: <87f2f3b90902130900v3f363055qafa5e28af19f71f5@mail.gmail.com> On Fri, Feb 13, 2009 at 6:53 AM, Alexandru Nedelcu wrote: > Btw ... I fixed the document. > Sorry about that guys, I'm a rookie and I thought other people would > find my setup useful. > > Can you guys read it and tell me if it contains other mistakes? My > intention was to publish it on the wiki once it was ready, but I > temporarily moved on to another project. I'll check it out. FYI, I recommend s/FreeSwitch/FreeSWITCH/g. :) -MC From anthony.minessale at gmail.com Fri Feb 13 09:21:25 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 13 Feb 2009 11:21:25 -0600 Subject: [Freeswitch-users] Codec negotiation questions In-Reply-To: References: <191c3a030902121306l1dae21a3m614f400703d15bd2@mail.gmail.com> <191c3a030902130746h23018e17n5e5c6c4900233097@mail.gmail.com> Message-ID: <191c3a030902130921o53e6752bxf90b5a3ebe65578f@mail.gmail.com> As i have already answered, no, it does not do what you want automaticly, the only way to influence codec negotiation is the way i have described. parsing the sdp string allows you to set absolute_codec_string going both ways. if you set it before you answer the channel with late negotiation enabled it will influence the codecs accepted on the inbound call. it you set it on the b leg either by using export instead of set on the a leg or putting it in {} in the dial string it controlls what codecs are offered in the outbound invite. On Fri, Feb 13, 2009 at 10:50 AM, ivdreg ivdreg wrote: > Hi Anthony, > > I'm not sure that you understood the problem. As it shown bellow the > offered codec in leg B contains only one codec (first matched in codec > preference list for this profile). Is there way to offer in leg B not only > first codec but all codecs that exists in INVITE in leg A that matches codec > preference list. If not is the only way is to parse SDP and set > absolute_codec_string manualy? > > Regards > > 2009/2/13 Anthony Minessale > > yes you are wrong. >> >> inbound-late-negotiation setting delays the codec negotiation until the >> instant audio is needed. >> It is not tied to inbound-proxy-media. >> >> >> This allows the call to come into the dialplan before any codec >> negotiation is done giving you a chance to look at the SDP before the >> negotiation takes place and insert an absolute_codec string essentially >> letting you chose unique codec preferences per inbound call. >> >> >>> Why I should parse variable_switch_r_sdp >> >> Well....you must parse it because it's you who cares about what it says, >> as described above it lets you peek at the sdp and enforce a unique set of >> codec prefs per call. >> >> >> >> >> On Fri, Feb 13, 2009 at 8:57 AM, ivdreg ivdreg wrote: >> >>> Hi Anthony, >>> >>> Excuse me if I'm wrong but inbound-late-negotiation must be used >>> proxy_media as I see in documentation. I don't want to proxy media because >>> of some issues with MOH or 3-way conferencing. Also I want to exclude media >>> codecs that are supported only in pass-trough mode. Let mi give you an >>> example: >>> >>> SDP from caller >>> >>> v=0 >>> o=- 1 2 IN IP4 192.168.20.193 >>> s=CounterPath eyeBeam 1.5 >>> c=IN IP4 192.168.40.81 >>> t=0 0 >>> m=audio 56888 RTP/AVP 100 106 97 105 98 3 101 >>> a=fmtp:101 0-15 >>> a=rtpmap:100 SPEEX/16000 >>> a=rtpmap:106 SPEEX-FEC/16000 >>> a=rtpmap:97 SPEEX/8000 >>> a=rtpmap:105 SPEEX-FEC/8000 >>> a=rtpmap:98 iLBC/8000 >>> a=rtpmap:101 telephone-event/8000 >>> a=sendrecv >>> a=x-rtp-session-id:61940567309B49E8909127E1393A966E >>> m=video 46378 RTP/AVP 125 115 34 >>> a=fmtp:125 profile-level-id=42e015; max-br=4000; max-mbps=19800 >>> a=fmtp:115 QCIF=1 MAXBR=4520 >>> a=fmtp:34 QCIF=1 MAXBR=4520 >>> a=rtpmap:125 H264/90000 >>> a=rtpmap:115 H263-1998/90000 >>> a=rtpmap:34 H263/90000 >>> a=sendrecv >>> a=x-rtp-session-id:E8244E608F65445BA183BBE641C5DF3C >>> a=nortpproxy:yes >>> >>> SDP from Freeswitch to called >>> >>> v=0 >>> o=FreeSWITCH 6228154995644782318 4633980766357417433 IN IP4 >>> 10.10.10.10 >>> s=FreeSWITCH >>> c=IN IP4 10.10.10.10 >>> t=0 0 >>> m=audio 26920 RTP/AVP 3 101 13 >>> * a=rtpmap:3 GSM/8000* >>> a=rtpmap:101 telephone-event/8000 >>> a=fmtp:101 0-16 >>> a=rtpmap:13 CN/8000 >>> a=ptime:20 >>> >>> So we offer *only first* in codec preference list >>> >>> from called wich is normal receives >>> SIP/2.0 488 Not Acceptable Here >>> *called suports - PCMA,PCMU,iLBC >>> * >>> Codec preference to this vars.xml we have witch is used in provile: >>> >>> also we have in profile: >>> >>> >>> In dialplan I've set: >>> >>> >>> >>> About my second question: >>> Why I should parse variable_switch_r_sdp: [v=0 >>> o=- 6 2 IN IP4 192.168.20.193 >>> s=CounterPath eyeBeam 1.5 >>> c=IN IP4 192.168.40.81 >>> t=0 0 >>> m=audio 60642 RTP/AVP 100 106 97 105 98 3 101 >>> a=rtpmap:100 SPEEX/16000 >>> a=rtpmap:106 SPEEX-FEC/16000 >>> a=rtpmap:97 SPEEX/8000 >>> a=rtpmap:105 SPEEX-FEC/8000 >>> a=rtpmap:98 iLBC/8000 >>> a=rtpmap:101 telephone-event/8000 >>> a=fmtp:101 0-15 >>> a=x-rtp-session-id:0674790128EB43ACB8C7F55829BCFF14 >>> m=video 44938 RTP/AVP 125 115 34 >>> a=rtpmap:125 H264/90000 >>> a=fmtp:125 profile-level-id=42e015; max-br=4000; max-mbps=19800 >>> a=rtpmap:115 H263-1998/90000 >>> a=fmtp:115 QCIF=1 MAXBR=4520 >>> a=rtpmap:34 H263/90000 >>> a=fmtp:34 QCIF=1 MAXBR=4520 >>> a=x-rtp-session-id:D1EAE543055746AC9C03006B91ADE6DF >>> a=nortpproxy:yes >>> ] >>> In FS core this parse is already done I'm sure in much more intelligent >>> way. It can be exported as a variable like a absolute codec string I think. >>> >>> Thanks again. >>> >>> 2009/2/12 Anthony Minessale >>> >>> the entire sdp is available as a variable (route the call to the info app >>>> to see the variables) >>>> so if you have inbound-late-negotiation set to true on the sip profile >>>> then you can use a regex or a script to set absolute_codec string before >>>> you answer. >>>> >>>> >>>> On Thu, Feb 12, 2009 at 8:06 AM, ivdreg ivdreg wrote: >>>> >>>>> Hi all, >>>>> >>>>> Can I ask 2 questions about codec negotiation: >>>>> >>>>> 1. Is it possible Freeswitch to work negotiate codecs between two >>>>> phones as it is described below. >>>>> INVITE from A with some codecs in SDP ---> Freeswitch rewrites codec >>>>> preference according absolute_codec_string but exclude all codecs not >>>>> offered by A ----> INVITE to B with rewrited SDP. >>>>> >>>>> example: >>>>> from A SDP:PCMA,PCMU,SPEEX ----> absolute_codec_string=G722,PCMU,PCMA,GSM >>>>> ----> to B SDP: PCMU,PCMA >>>>> >>>>> 2. Can I get codec list in INVITE with mod_perl for example or via >>>>> xml_curl without processing SDP variable (switch_r_sdp). It will be useful >>>>> to be in format that absolute_codec_string variable takes. >>>>> >>>>> Thanks >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> -- >>>> Anthony Minessale II >>>> >>>> FreeSWITCH http://www.freeswitch.org/ >>>> ClueCon http://www.cluecon.com/ >>>> >>>> AIM: anthm >>>> MSN:anthony_minessale at hotmail.com >>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>>> IRC: irc.freenode.net #freeswitch >>>> >>>> FreeSWITCH Developer Conference >>>> sip:888 at conference.freeswitch.org >>>> iax:guest at conference.freeswitch.org/888 >>>> googletalk:conf+888 at conference.freeswitch.org >>>> pstn:213-799-1400 >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:213-799-1400 >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090213/23d65dca/attachment-0001.html From msc at freeswitch.org Fri Feb 13 09:34:05 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 13 Feb 2009 09:34:05 -0800 Subject: [Freeswitch-users] Setting outbound callerid using js In-Reply-To: <191c3a030902130727y132a55c2v8fb27e87576154da@mail.gmail.com> References: <87f2f3b90902111626x6db5dc61t68a0170a3460b42d@mail.gmail.com> <191c3a030902130605m5e834e3dh27cc88387408b8b5@mail.gmail.com> <1234535136.4431.70.camel@gathern.lan> <191c3a030902130727y132a55c2v8fb27e87576154da@mail.gmail.com> Message-ID: <87f2f3b90902130934i25e52f07keff6d2f48067443b@mail.gmail.com> >> > This is perhaps the 4th time i have seen someone do this, can you >> > point out where this is incorrectly documented? FYI, I've updated this html file for the orig author and sent it to him for review. -MC From ivdreg at gmail.com Fri Feb 13 09:50:54 2009 From: ivdreg at gmail.com (ivdreg ivdreg) Date: Fri, 13 Feb 2009 19:50:54 +0200 Subject: [Freeswitch-users] Codec negotiation questions In-Reply-To: <191c3a030902130921o53e6752bxf90b5a3ebe65578f@mail.gmail.com> References: <191c3a030902121306l1dae21a3m614f400703d15bd2@mail.gmail.com> <191c3a030902130746h23018e17n5e5c6c4900233097@mail.gmail.com> <191c3a030902130921o53e6752bxf90b5a3ebe65578f@mail.gmail.com> Message-ID: Hi Antony, Can you tell me why you do codec negotiation like that. I'm just curious. If you do not have time do not reply me. Thanks a lot for your help. 2009/2/13 Anthony Minessale > As i have already answered, no, it does not do what you want automaticly, > the only way to influence codec negotiation is the way i have described. > > parsing the sdp string allows you to set absolute_codec_string going both > ways. > if you set it before you answer the channel with late negotiation enabled > it will influence the codecs accepted on the inbound call. > it you set it on the b leg either by using export instead of set on the a > leg or putting it in {} in the dial string it controlls what codecs are > offered in the outbound invite. > > > > > On Fri, Feb 13, 2009 at 10:50 AM, ivdreg ivdreg wrote: > >> Hi Anthony, >> >> I'm not sure that you understood the problem. As it shown bellow the >> offered codec in leg B contains only one codec (first matched in codec >> preference list for this profile). Is there way to offer in leg B not only >> first codec but all codecs that exists in INVITE in leg A that matches codec >> preference list. If not is the only way is to parse SDP and set >> absolute_codec_string manualy? >> >> Regards >> >> 2009/2/13 Anthony Minessale >> >> yes you are wrong. >>> >>> inbound-late-negotiation setting delays the codec negotiation until the >>> instant audio is needed. >>> It is not tied to inbound-proxy-media. >>> >>> >>> This allows the call to come into the dialplan before any codec >>> negotiation is done giving you a chance to look at the SDP before the >>> negotiation takes place and insert an absolute_codec string essentially >>> letting you chose unique codec preferences per inbound call. >>> >>> >>> Why I should parse variable_switch_r_sdp >>> >>> Well....you must parse it because it's you who cares about what it says, >>> as described above it lets you peek at the sdp and enforce a unique set of >>> codec prefs per call. >>> >>> >>> >>> >>> On Fri, Feb 13, 2009 at 8:57 AM, ivdreg ivdreg wrote: >>> >>>> Hi Anthony, >>>> >>>> Excuse me if I'm wrong but inbound-late-negotiation must be used >>>> proxy_media as I see in documentation. I don't want to proxy media because >>>> of some issues with MOH or 3-way conferencing. Also I want to exclude media >>>> codecs that are supported only in pass-trough mode. Let mi give you an >>>> example: >>>> >>>> SDP from caller >>>> >>>> v=0 >>>> o=- 1 2 IN IP4 192.168.20.193 >>>> s=CounterPath eyeBeam 1.5 >>>> c=IN IP4 192.168.40.81 >>>> t=0 0 >>>> m=audio 56888 RTP/AVP 100 106 97 105 98 3 101 >>>> a=fmtp:101 0-15 >>>> a=rtpmap:100 SPEEX/16000 >>>> a=rtpmap:106 SPEEX-FEC/16000 >>>> a=rtpmap:97 SPEEX/8000 >>>> a=rtpmap:105 SPEEX-FEC/8000 >>>> a=rtpmap:98 iLBC/8000 >>>> a=rtpmap:101 telephone-event/8000 >>>> a=sendrecv >>>> a=x-rtp-session-id:61940567309B49E8909127E1393A966E >>>> m=video 46378 RTP/AVP 125 115 34 >>>> a=fmtp:125 profile-level-id=42e015; max-br=4000; max-mbps=19800 >>>> a=fmtp:115 QCIF=1 MAXBR=4520 >>>> a=fmtp:34 QCIF=1 MAXBR=4520 >>>> a=rtpmap:125 H264/90000 >>>> a=rtpmap:115 H263-1998/90000 >>>> a=rtpmap:34 H263/90000 >>>> a=sendrecv >>>> a=x-rtp-session-id:E8244E608F65445BA183BBE641C5DF3C >>>> a=nortpproxy:yes >>>> >>>> SDP from Freeswitch to called >>>> >>>> v=0 >>>> o=FreeSWITCH 6228154995644782318 4633980766357417433 IN IP4 >>>> 10.10.10.10 >>>> s=FreeSWITCH >>>> c=IN IP4 10.10.10.10 >>>> t=0 0 >>>> m=audio 26920 RTP/AVP 3 101 13 >>>> * a=rtpmap:3 GSM/8000* >>>> a=rtpmap:101 telephone-event/8000 >>>> a=fmtp:101 0-16 >>>> a=rtpmap:13 CN/8000 >>>> a=ptime:20 >>>> >>>> So we offer *only first* in codec preference list >>>> >>>> from called wich is normal receives >>>> SIP/2.0 488 Not Acceptable Here >>>> *called suports - PCMA,PCMU,iLBC >>>> * >>>> Codec preference to this vars.xml we have witch is used in provile: >>>> >>>> also we have in profile: >>>> >>>> >>>> In dialplan I've set: >>>> >>>> >>>> >>>> About my second question: >>>> Why I should parse variable_switch_r_sdp: [v=0 >>>> o=- 6 2 IN IP4 192.168.20.193 >>>> s=CounterPath eyeBeam 1.5 >>>> c=IN IP4 192.168.40.81 >>>> t=0 0 >>>> m=audio 60642 RTP/AVP 100 106 97 105 98 3 101 >>>> a=rtpmap:100 SPEEX/16000 >>>> a=rtpmap:106 SPEEX-FEC/16000 >>>> a=rtpmap:97 SPEEX/8000 >>>> a=rtpmap:105 SPEEX-FEC/8000 >>>> a=rtpmap:98 iLBC/8000 >>>> a=rtpmap:101 telephone-event/8000 >>>> a=fmtp:101 0-15 >>>> a=x-rtp-session-id:0674790128EB43ACB8C7F55829BCFF14 >>>> m=video 44938 RTP/AVP 125 115 34 >>>> a=rtpmap:125 H264/90000 >>>> a=fmtp:125 profile-level-id=42e015; max-br=4000; max-mbps=19800 >>>> a=rtpmap:115 H263-1998/90000 >>>> a=fmtp:115 QCIF=1 MAXBR=4520 >>>> a=rtpmap:34 H263/90000 >>>> a=fmtp:34 QCIF=1 MAXBR=4520 >>>> a=x-rtp-session-id:D1EAE543055746AC9C03006B91ADE6DF >>>> a=nortpproxy:yes >>>> ] >>>> In FS core this parse is already done I'm sure in much more intelligent >>>> way. It can be exported as a variable like a absolute codec string I think. >>>> >>>> Thanks again. >>>> >>>> 2009/2/12 Anthony Minessale >>>> >>>> the entire sdp is available as a variable (route the call to the info >>>>> app to see the variables) >>>>> so if you have inbound-late-negotiation set to true on the sip profile >>>>> then you can use a regex or a script to set absolute_codec string >>>>> before you answer. >>>>> >>>>> >>>>> On Thu, Feb 12, 2009 at 8:06 AM, ivdreg ivdreg wrote: >>>>> >>>>>> Hi all, >>>>>> >>>>>> Can I ask 2 questions about codec negotiation: >>>>>> >>>>>> 1. Is it possible Freeswitch to work negotiate codecs between two >>>>>> phones as it is described below. >>>>>> INVITE from A with some codecs in SDP ---> Freeswitch rewrites codec >>>>>> preference according absolute_codec_string but exclude all codecs not >>>>>> offered by A ----> INVITE to B with rewrited SDP. >>>>>> >>>>>> example: >>>>>> from A SDP:PCMA,PCMU,SPEEX ----> absolute_codec_string=G722,PCMU,PCMA,GSM >>>>>> ----> to B SDP: PCMU,PCMA >>>>>> >>>>>> 2. Can I get codec list in INVITE with mod_perl for example or via >>>>>> xml_curl without processing SDP variable (switch_r_sdp). It will be useful >>>>>> to be in format that absolute_codec_string variable takes. >>>>>> >>>>>> Thanks >>>>>> >>>>>> _______________________________________________ >>>>>> Freeswitch-users mailing list >>>>>> Freeswitch-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>> >>>>> >>>>> -- >>>>> Anthony Minessale II >>>>> >>>>> FreeSWITCH http://www.freeswitch.org/ >>>>> ClueCon http://www.cluecon.com/ >>>>> >>>>> AIM: anthm >>>>> MSN:anthony_minessale at hotmail.com >>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>>>> IRC: irc.freenode.net #freeswitch >>>>> >>>>> FreeSWITCH Developer Conference >>>>> sip:888 at conference.freeswitch.org >>>>> iax:guest at conference.freeswitch.org/888 >>>>> googletalk:conf+888 at conference.freeswitch.org >>>>> pstn:213-799-1400 >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> Anthony Minessale II >>> >>> FreeSWITCH http://www.freeswitch.org/ >>> ClueCon http://www.cluecon.com/ >>> >>> AIM: anthm >>> MSN:anthony_minessale at hotmail.com >>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>> IRC: irc.freenode.net #freeswitch >>> >>> FreeSWITCH Developer Conference >>> sip:888 at conference.freeswitch.org >>> iax:guest at conference.freeswitch.org/888 >>> googletalk:conf+888 at conference.freeswitch.org >>> pstn:213-799-1400 >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090213/a251917a/attachment-0001.html From mike at jerris.com Fri Feb 13 10:11:21 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 13 Feb 2009 13:11:21 -0500 Subject: [Freeswitch-users] speex build issues in svn trunk. Message-ID: <0F5A9820-2EE1-4A72-BD29-D12C6B45C25C@jerris.com> I updated the version of the speex library we use in tree last night and it may cause some build issues for those with current working copies. To fix this issue you can type "make speex-reconf" MIke From gmaruzz at celliax.org Fri Feb 13 10:15:08 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Fri, 13 Feb 2009 19:15:08 +0100 Subject: [Freeswitch-users] Skypiax (Skype Network endpoint) for FreeSWITCH Message-ID: <7b197bef0902131015k456dee87x8f0f1b7059f1b49c@mail.gmail.com> Hello FreeSWITCHers, mod_skypiax is available for testing, feature requests, bug hunting. I would like to ask the help of you all to make Skypiax robust and feature full on FreeSWITCH, and particularly of Massimo Cetra (CtRiX on IRC), that has developed mod_airpe (another Skype endpoint). I've written a first documentation on Skypiax installation and usage at: http://wiki.freeswitch.org/wiki/Skypiax and there is a Jira module at: http://jira.freeswitch.org/browse/MODSKYPIAX So, please, test the software, edit the wiki page both for style and content, file bug reports and feature requests. FreeSWITCH is now the platform of first development for me, so the FreeSWITCH part of Skypiax is more tested (if any) and the code is more readable compared to the Asterisk part where lot of legacy from my other projects clutter the code. But Skypiax strive to be available as a Skype compatible endpoint for all the opensource telephony community, and in the near time the Asterisk part will be cleaned much more, and documented. As you will see, the code is made by skypiax_protocol.c (the interaction with Skype client), mod_skypiax.c (the interaction with FreeSWITCH), chan_skypiax.c (the interaction with Asterisk). Please consider me available for all infos, clarifications, discussions, etc. I would like to thanks all the peoples that helped me via mail and IRC (so bad to have different timezones, isn't?), the *very early adopters*, the testers, the patchers, and you all. Particularly Anthony Minessale, Michael Jerris, Brian West, Michael Collins, Ken Rice, Seven Du, Clif Cox, Hristo Trendev, Rehan Allah Wala, Jason Garland and Antonio Gallo. >From the wiki page (http://wiki.freeswitch.org/wiki/Skypiax): WHAT IS SKYPIAX This software (Skypiax) uses the Skype API but is not endorsed, certified or otherwise approved in any way by Skype. Skypiax is an endpoint (channel driver) that uses the Skype client as an interface to the Skype network, and allows incoming and outgoing Skype calls to/from FreeSWITCH (that can be bridged, originated, answered, etc. as in all other endpoints, e.g. sofia/SIP). Skypiax works in FreeSWITCH (FS) on both Linux and Windows, at both 8khz and 16khz (Skype client has 16khz audio I/O). Skypiax works on Asterisk too, at 8khz, on Linux and Windows (through CygWin). Think of Skypiax as similar to OpenZAP for analog lines. For each channel you need an interface (a Skype client). So, for example, two concurrent calls would need two channels, and therefor two Skype clients running on your FreeSWITCH server. If your Skype client(s) have Skype credits, then Skypiax works for SkypeOut calls as well. Sincerely, Giovanni Maruzzelli ========================================= www.celliax.org via Pierlombardo 9, 20135 Milano Italy gmaruzz at celliax dot org Cell : +39-347-2665618 Fax : +39-02-87390039 On Thu, Jan 15, 2009 at 6:39 PM, Giovanni Maruzzelli wrote: > Hi FreeSWITCH developers > > I would like to propose to the community my plans, so we can discuss > and coordinate efforts. > > I developed a couple of channel drivers for Asterisk in the past > (works on both Linux and Windows), and I would like to port them to FS > and further enhance them. > > The two endpoints are: > - Skypiax, Skype compatible, makes and receives calls to/from Skype > network and Skypeout service, using the Skype client as interface. > - Celliax, GSM and SMS endpoint, makes and receives voice calls and > SMSs to/from the GSM/CDMA network, using second hand cellphones and/or > embedded professional devices as interfaces > > My aims are: > > a) port both endpoints from Asterisk to FreeSWITCH > b) have both endpoints continue to support at least Linux and Windows on FS > c) I would like better having most of the endpoints code working for > both FreeSWITCH and Asterisk, maintaining separated the code that > interface with the GSM and Skype network, from the code that interface > with the core. > > Skypiax, the skype compatible endpoint, is a fork of celliax, the GSM > endpoint, and they share the same skeleton and logic, so porting > celliax after having ported skypiax will be easier and faster :-). > > Current situation and next steps: > 1) skypiax (http://wiki.freeswitch.org/wiki/Skypiax) is now available > for testing and debugging, needs to be polished and cleaned > 2) starting mid next week (I'll be back in office), I want to > integrate into skypiax the code and ideas from mod_airpe of Massimo > (ctrix), that has developed an alternative Skype compatible module, > and coordinate any future development with him and any other > interested developer > 3) begin the porting of celliax to FS, aiming at a pre-beta release > for Linux and Windows during February. > 4) coordinate further development of celliax with any other developer > interested in GSM, SMSs, CDMA, IDEN, AT commands, FBUS commands, > embedded devices, audio sampling > > I am gmaruzz on #freeswitch and #freeswitch-dev, you can find more > info at www.celliax.org. > > Sincerely, > > Giovanni Maruzzelli > ========================================= > www.celliax.org > via Pierlombardo 9, 20135 Milano > Italy > gmaruzz at celliax dot org > Cell : +39-347-2665618 > Fax : +39-02-87390039 > From anthony.minessale at gmail.com Fri Feb 13 10:29:31 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 13 Feb 2009 12:29:31 -0600 Subject: [Freeswitch-users] Codec negotiation questions In-Reply-To: References: <191c3a030902121306l1dae21a3m614f400703d15bd2@mail.gmail.com> <191c3a030902130746h23018e17n5e5c6c4900233097@mail.gmail.com> <191c3a030902130921o53e6752bxf90b5a3ebe65578f@mail.gmail.com> Message-ID: <191c3a030902131029g204582adxad6dd0f53be71f2c@mail.gmail.com> Do it like what? you are using FS as a b2bua, the default behavior is to make a list of codecs that are parsed on every inbound call as soon as the invite is received. if you bridge that inbound leg to an outbound leg it will again use that same list with the one used by the inbound leg as the first choice (this is a favor we do by passing the chosen codec on the a leg over to the outbound call on the b leg) if this is not satisfactory i have left you with the option to control the behaviour yourself in a script. On Fri, Feb 13, 2009 at 11:50 AM, ivdreg ivdreg wrote: > Hi Antony, > > Can you tell me why you do codec negotiation like that. I'm just curious. > If you do not have time do not reply me. > > Thanks a lot for your help. > > > 2009/2/13 Anthony Minessale > >> As i have already answered, no, it does not do what you want automaticly, >> the only way to influence codec negotiation is the way i have described. >> >> parsing the sdp string allows you to set absolute_codec_string going both >> ways. >> if you set it before you answer the channel with late negotiation enabled >> it will influence the codecs accepted on the inbound call. >> it you set it on the b leg either by using export instead of set on the a >> leg or putting it in {} in the dial string it controlls what codecs are >> offered in the outbound invite. >> >> >> >> >> On Fri, Feb 13, 2009 at 10:50 AM, ivdreg ivdreg wrote: >> >>> Hi Anthony, >>> >>> I'm not sure that you understood the problem. As it shown bellow the >>> offered codec in leg B contains only one codec (first matched in codec >>> preference list for this profile). Is there way to offer in leg B not only >>> first codec but all codecs that exists in INVITE in leg A that matches codec >>> preference list. If not is the only way is to parse SDP and set >>> absolute_codec_string manualy? >>> >>> Regards >>> >>> 2009/2/13 Anthony Minessale >>> >>> yes you are wrong. >>>> >>>> inbound-late-negotiation setting delays the codec negotiation until the >>>> instant audio is needed. >>>> It is not tied to inbound-proxy-media. >>>> >>>> >>>> This allows the call to come into the dialplan before any codec >>>> negotiation is done giving you a chance to look at the SDP before the >>>> negotiation takes place and insert an absolute_codec string essentially >>>> letting you chose unique codec preferences per inbound call. >>>> >>>> >>> Why I should parse variable_switch_r_sdp >>>> >>>> Well....you must parse it because it's you who cares about what it says, >>>> as described above it lets you peek at the sdp and enforce a unique set of >>>> codec prefs per call. >>>> >>>> >>>> >>>> >>>> On Fri, Feb 13, 2009 at 8:57 AM, ivdreg ivdreg wrote: >>>> >>>>> Hi Anthony, >>>>> >>>>> Excuse me if I'm wrong but inbound-late-negotiation must be used >>>>> proxy_media as I see in documentation. I don't want to proxy media because >>>>> of some issues with MOH or 3-way conferencing. Also I want to exclude media >>>>> codecs that are supported only in pass-trough mode. Let mi give you an >>>>> example: >>>>> >>>>> SDP from caller >>>>> >>>>> v=0 >>>>> o=- 1 2 IN IP4 192.168.20.193 >>>>> s=CounterPath eyeBeam 1.5 >>>>> c=IN IP4 192.168.40.81 >>>>> t=0 0 >>>>> m=audio 56888 RTP/AVP 100 106 97 105 98 3 101 >>>>> a=fmtp:101 0-15 >>>>> a=rtpmap:100 SPEEX/16000 >>>>> a=rtpmap:106 SPEEX-FEC/16000 >>>>> a=rtpmap:97 SPEEX/8000 >>>>> a=rtpmap:105 SPEEX-FEC/8000 >>>>> a=rtpmap:98 iLBC/8000 >>>>> a=rtpmap:101 telephone-event/8000 >>>>> a=sendrecv >>>>> a=x-rtp-session-id:61940567309B49E8909127E1393A966E >>>>> m=video 46378 RTP/AVP 125 115 34 >>>>> a=fmtp:125 profile-level-id=42e015; max-br=4000; max-mbps=19800 >>>>> a=fmtp:115 QCIF=1 MAXBR=4520 >>>>> a=fmtp:34 QCIF=1 MAXBR=4520 >>>>> a=rtpmap:125 H264/90000 >>>>> a=rtpmap:115 H263-1998/90000 >>>>> a=rtpmap:34 H263/90000 >>>>> a=sendrecv >>>>> a=x-rtp-session-id:E8244E608F65445BA183BBE641C5DF3C >>>>> a=nortpproxy:yes >>>>> >>>>> SDP from Freeswitch to called >>>>> >>>>> v=0 >>>>> o=FreeSWITCH 6228154995644782318 4633980766357417433 IN IP4 >>>>> 10.10.10.10 >>>>> s=FreeSWITCH >>>>> c=IN IP4 10.10.10.10 >>>>> t=0 0 >>>>> m=audio 26920 RTP/AVP 3 101 13 >>>>> * a=rtpmap:3 GSM/8000* >>>>> a=rtpmap:101 telephone-event/8000 >>>>> a=fmtp:101 0-16 >>>>> a=rtpmap:13 CN/8000 >>>>> a=ptime:20 >>>>> >>>>> So we offer *only first* in codec preference list >>>>> >>>>> from called wich is normal receives >>>>> SIP/2.0 488 Not Acceptable Here >>>>> *called suports - PCMA,PCMU,iLBC >>>>> * >>>>> Codec preference to this vars.xml we have witch is used in provile: >>>>> >>>>> also we have in profile: >>>>> >>>>> >>>>> In dialplan I've set: >>>>> >>>>> >>>>> >>>>> About my second question: >>>>> Why I should parse variable_switch_r_sdp: [v=0 >>>>> o=- 6 2 IN IP4 192.168.20.193 >>>>> s=CounterPath eyeBeam 1.5 >>>>> c=IN IP4 192.168.40.81 >>>>> t=0 0 >>>>> m=audio 60642 RTP/AVP 100 106 97 105 98 3 101 >>>>> a=rtpmap:100 SPEEX/16000 >>>>> a=rtpmap:106 SPEEX-FEC/16000 >>>>> a=rtpmap:97 SPEEX/8000 >>>>> a=rtpmap:105 SPEEX-FEC/8000 >>>>> a=rtpmap:98 iLBC/8000 >>>>> a=rtpmap:101 telephone-event/8000 >>>>> a=fmtp:101 0-15 >>>>> a=x-rtp-session-id:0674790128EB43ACB8C7F55829BCFF14 >>>>> m=video 44938 RTP/AVP 125 115 34 >>>>> a=rtpmap:125 H264/90000 >>>>> a=fmtp:125 profile-level-id=42e015; max-br=4000; max-mbps=19800 >>>>> a=rtpmap:115 H263-1998/90000 >>>>> a=fmtp:115 QCIF=1 MAXBR=4520 >>>>> a=rtpmap:34 H263/90000 >>>>> a=fmtp:34 QCIF=1 MAXBR=4520 >>>>> a=x-rtp-session-id:D1EAE543055746AC9C03006B91ADE6DF >>>>> a=nortpproxy:yes >>>>> ] >>>>> In FS core this parse is already done I'm sure in much more intelligent >>>>> way. It can be exported as a variable like a absolute codec string I think. >>>>> >>>>> Thanks again. >>>>> >>>>> 2009/2/12 Anthony Minessale >>>>> >>>>> the entire sdp is available as a variable (route the call to the info >>>>>> app to see the variables) >>>>>> so if you have inbound-late-negotiation set to true on the sip profile >>>>>> then you can use a regex or a script to set absolute_codec string >>>>>> before you answer. >>>>>> >>>>>> >>>>>> On Thu, Feb 12, 2009 at 8:06 AM, ivdreg ivdreg wrote: >>>>>> >>>>>>> Hi all, >>>>>>> >>>>>>> Can I ask 2 questions about codec negotiation: >>>>>>> >>>>>>> 1. Is it possible Freeswitch to work negotiate codecs between two >>>>>>> phones as it is described below. >>>>>>> INVITE from A with some codecs in SDP ---> Freeswitch rewrites codec >>>>>>> preference according absolute_codec_string but exclude all codecs not >>>>>>> offered by A ----> INVITE to B with rewrited SDP. >>>>>>> >>>>>>> example: >>>>>>> from A SDP:PCMA,PCMU,SPEEX ----> absolute_codec_string=G722,PCMU,PCMA,GSM >>>>>>> ----> to B SDP: PCMU,PCMA >>>>>>> >>>>>>> 2. Can I get codec list in INVITE with mod_perl for example or via >>>>>>> xml_curl without processing SDP variable (switch_r_sdp). It will be useful >>>>>>> to be in format that absolute_codec_string variable takes. >>>>>>> >>>>>>> Thanks >>>>>>> >>>>>>> _______________________________________________ >>>>>>> Freeswitch-users mailing list >>>>>>> Freeswitch-users at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE: >>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> -- >>>>>> Anthony Minessale II >>>>>> >>>>>> FreeSWITCH http://www.freeswitch.org/ >>>>>> ClueCon http://www.cluecon.com/ >>>>>> >>>>>> AIM: anthm >>>>>> MSN:anthony_minessale at hotmail.com >>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>>>>> IRC: irc.freenode.net #freeswitch >>>>>> >>>>>> FreeSWITCH Developer Conference >>>>>> sip:888 at conference.freeswitch.org >>>>>> iax:guest at conference.freeswitch.org/888 >>>>>> googletalk:conf+888 at conference.freeswitch.org >>>>>> pstn:213-799-1400 >>>>>> >>>>>> _______________________________________________ >>>>>> Freeswitch-users mailing list >>>>>> Freeswitch-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> -- >>>> Anthony Minessale II >>>> >>>> FreeSWITCH http://www.freeswitch.org/ >>>> ClueCon http://www.cluecon.com/ >>>> >>>> AIM: anthm >>>> MSN:anthony_minessale at hotmail.com >>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>>> IRC: irc.freenode.net #freeswitch >>>> >>>> FreeSWITCH Developer Conference >>>> sip:888 at conference.freeswitch.org >>>> iax:guest at conference.freeswitch.org/888 >>>> googletalk:conf+888 at conference.freeswitch.org >>>> pstn:213-799-1400 >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:213-799-1400 >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090213/9e43d6e6/attachment-0001.html From gmaruzz at celliax.org Fri Feb 13 11:55:35 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Fri, 13 Feb 2009 20:55:35 +0100 Subject: [Freeswitch-users] speex build issues in svn trunk. In-Reply-To: <0F5A9820-2EE1-4A72-BD29-D12C6B45C25C@jerris.com> References: <0F5A9820-2EE1-4A72-BD29-D12C6B45C25C@jerris.com> Message-ID: <7b197bef0902131155l13a598b4s7c4f39a1983964c6@mail.gmail.com> Yay for the new speex with good Acoustic Echo Cancellation. I'll put it to work when I'll port Celliax, the GSM endpoint, for cancelling the sidetone that certain interfaces give back. :-) Thanks MikeJ ! Sincerely, Giovanni Maruzzelli ========================================= www.celliax.org via Pierlombardo 9, 20135 Milano Italy gmaruzz at celliax dot org Cell : +39-347-2665618 Fax : +39-02-87390039 On Fri, Feb 13, 2009 at 7:11 PM, Michael Jerris wrote: > I updated the version of the speex library we use in tree last night > and it may cause some build issues for those with current working > copies. To fix this issue you can type "make speex-reconf" > > MIke > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From msc at freeswitch.org Fri Feb 13 13:38:20 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 13 Feb 2009 13:38:20 -0800 Subject: [Freeswitch-users] INFO: Some new content on main page Message-ID: <87f2f3b90902131338n671fd82fod862314960fa8aa8@mail.gmail.com> FYI, There are a few new items on the main page: www.freeswitch.org, just in case you haven't been there lately. :) -MC From nik.middleton at noblesolutions.co.uk Fri Feb 13 14:30:28 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Fri, 13 Feb 2009 22:30:28 -0000 Subject: [Freeswitch-users] Hangup hook in js is never called Message-ID: Can't figure this one out. I've enabled a hang-up hook in js to do some cleanup. I've followed the example on the wiki, but it would appear it's never called. http://wiki.freeswitch.org/wiki/Example_Hangup_hook Is the code in error? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090213/821459cd/attachment.html From msc at freeswitch.org Fri Feb 13 14:51:15 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 13 Feb 2009 14:51:15 -0800 Subject: [Freeswitch-users] Hangup hook in js is never called In-Reply-To: References: Message-ID: <87f2f3b90902131451t56f84286p13d7a0b815db0068@mail.gmail.com> > http://wiki.freeswitch.org/wiki/Example_Hangup_hook > > > > Is the code in error? > It might just be. I think you are better off using api_hangup_hook. What are you trying to do on hangup? The api_hangup_hook lets you call any API, including running a script. Here's an example that we played with today that I haven't even put on the wiki yet. I renames the wav file after the call is hung up. -MC From nik.middleton at noblesolutions.co.uk Fri Feb 13 15:04:16 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Fri, 13 Feb 2009 23:04:16 -0000 Subject: [Freeswitch-users] FS 1.0.2 Crash and burn In-Reply-To: <191c3a030902112025l401c759dife416cf34d51d134@mail.gmail.com> References: <7EA294D9-389F-4543-8F24-ED4C08600BE7@freeswitch.org><75A42DD8-9BA0-46CC-8121-04CE9DB27A27@freeswitch.org><6C1B5368-931F-4FA0-87F1-F0EE011C35DA@freeswitch.org> <191c3a030902112025l401c759dife416cf34d51d134@mail.gmail.com> Message-ID: Guess I'll have to dust off K&R. Having made the mindset leap to c++ I find C very procedural. Still would be like old times. Code I've looked at so far is very neat, but boy is there a lack of in-line comments. Haven't looked at the main source yet though. I always used to work on 3 lines of comments to 1 major line of code. Call me pedantic, but it aids maintenance. I've always thought of C as a low level language, just up from assembler, and nearly as efficient. An for me, low level is good not bad. Regards ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: 12 February 2009 04:26 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] FS 1.0.2 Crash and burn There is always C, it's actually considered a high level language by many ;) On Wed, Feb 11, 2009 at 5:50 PM, Brian West wrote: Lua has known issues with MySQL you must use latest SVN builds of the luasql driver for that to avoid it.. and still its not stellar.. the unixODBC one on the other hand works fine. /b On Feb 11, 2009, at 5:36 PM, Nik Middleton wrote: I've abandoned LUA. All sorts of problems (DTMF etc). Also reports of memory leaks when using MYSQL driver. Looking on the WIKI, JavaScript seems very well supported; PLUS DTMF works just fine (pulling my hair out on LUA) Guess I'm going to follow the path of least resistance on this one and use JS and ODBC Regards, _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090213/a98d82cb/attachment.html From nik.middleton at noblesolutions.co.uk Fri Feb 13 15:07:20 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Fri, 13 Feb 2009 23:07:20 -0000 Subject: [Freeswitch-users] Hangup hook in js is never called In-Reply-To: <87f2f3b90902131451t56f84286p13d7a0b815db0068@mail.gmail.com> References: <87f2f3b90902131451t56f84286p13d7a0b815db0068@mail.gmail.com> Message-ID: I'm trying to capture the hang-up reason and write it to the db (Was it busy etc). I also close the db in that function. That way I know I don't have any open connections. This is in JavaScript BTW -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 13 February 2009 22:51 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Hangup hook in js is never called > http://wiki.freeswitch.org/wiki/Example_Hangup_hook > > > > Is the code in error? > It might just be. I think you are better off using api_hangup_hook. What are you trying to do on hangup? The api_hangup_hook lets you call any API, including running a script. Here's an example that we played with today that I haven't even put on the wiki yet. I renames the wav file after the call is hung up. -MC _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From anthony.minessale at gmail.com Fri Feb 13 15:19:59 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 13 Feb 2009 17:19:59 -0600 Subject: [Freeswitch-users] FS 1.0.2 Crash and burn In-Reply-To: References: <7EA294D9-389F-4543-8F24-ED4C08600BE7@freeswitch.org> <75A42DD8-9BA0-46CC-8121-04CE9DB27A27@freeswitch.org> <6C1B5368-931F-4FA0-87F1-F0EE011C35DA@freeswitch.org> <191c3a030902112025l401c759dife416cf34d51d134@mail.gmail.com> Message-ID: <191c3a030902131519m808654eo268f56c4728adf12@mail.gmail.com> modules can be c++ too. See mod_opal , mod_python, mod_java, mod_soundtouch, mod_managed and mod_perl all use switch_cpp.cpp a wrapper used to bridge into scripting langs. On Fri, Feb 13, 2009 at 5:04 PM, Nik Middleton < nik.middleton at noblesolutions.co.uk> wrote: > Guess I'll have to dust off K&R. Having made the mindset leap to > c++ I find C very procedural. Still would be like old times. Code I've > looked at so far is very neat, but boy is there a lack of in-line comments. > Haven't looked at the main source yet though. I always used to work on 3 > lines of comments to 1 major line of code. Call me pedantic, but it aids > maintenance. > > > > I've always thought of C as a low level language, just up from assembler, > and nearly as efficient. An for me, low level is good not bad. > > > > Regards > > > > > ------------------------------ > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony > Minessale > *Sent:* 12 February 2009 04:26 > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] FS 1.0.2 Crash and burn > > > > There is always C, it's actually considered a high level language by many > ;) > > On Wed, Feb 11, 2009 at 5:50 PM, Brian West wrote: > > Lua has known issues with MySQL you must use latest SVN builds of the > luasql driver for that to avoid it.. and still its not stellar.. the > unixODBC one on the other hand works fine. > > > > /b > > > > On Feb 11, 2009, at 5:36 PM, Nik Middleton wrote: > > > > I've abandoned LUA. > > > > All sorts of problems (DTMF etc). Also reports of memory leaks when using > MYSQL driver. > > > > Looking on the WIKI, JavaScript seems very well supported; PLUS DTMF works > just fine (pulling my hair out on LUA) > > > > Guess I'm going to follow the path of least resistance on this one and use > JS and ODBC > > > > Regards, > > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090213/1cc7ead7/attachment.html From jason at jasonjgw.net Fri Feb 13 15:24:26 2009 From: jason at jasonjgw.net (Jason White) Date: Sat, 14 Feb 2009 10:24:26 +1100 Subject: [Freeswitch-users] FS 1.0.2 Crash and burn In-Reply-To: References: <191c3a030902112025l401c759dife416cf34d51d134@mail.gmail.com> Message-ID: <20090213232426.GA5549@jdc.jasonjgw.net> Nik Middleton wrote: > Code > I've looked at so far is very neat, but boy is there a lack of in-line > comments. Haven't looked at the main source yet though. I always used > to work on 3 lines of comments to 1 major line of code. Call me > pedantic, but it aids maintenance. I find the FreeSWITCH code quite readable. Public API functions have comments, which are all that we need, I think. From nik.middleton at noblesolutions.co.uk Fri Feb 13 15:40:32 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Fri, 13 Feb 2009 23:40:32 -0000 Subject: [Freeswitch-users] FS 1.0.2 Crash and burn In-Reply-To: <20090213232426.GA5549@jdc.jasonjgw.net> References: <191c3a030902112025l401c759dife416cf34d51d134@mail.gmail.com> <20090213232426.GA5549@jdc.jasonjgw.net> Message-ID: That would assume that the underlying code is perfect, which it probably isn't. Not knocking the efforts, but in my view, you can't have too much in line documentation. I hope to make a contribution shortly. Right now I'm updating the WIKI where appropriate. Top level examples should work with a cut and paste, if they don't you're going to alienate new entrants. Regards -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jason White Sent: 13 February 2009 23:24 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] FS 1.0.2 Crash and burn Nik Middleton wrote: > Code > I've looked at so far is very neat, but boy is there a lack of in-line > comments. Haven't looked at the main source yet though. I always used > to work on 3 lines of comments to 1 major line of code. Call me > pedantic, but it aids maintenance. I find the FreeSWITCH code quite readable. Public API functions have comments, which are all that we need, I think. _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From msc at freeswitch.org Fri Feb 13 16:08:44 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 13 Feb 2009 16:08:44 -0800 Subject: [Freeswitch-users] FS 1.0.2 Crash and burn In-Reply-To: References: <191c3a030902112025l401c759dife416cf34d51d134@mail.gmail.com> <20090213232426.GA5549@jdc.jasonjgw.net> Message-ID: <87f2f3b90902131608p5f8ead6fxecebac1b3baf8f52@mail.gmail.com> > Right now I'm updating the WIKI where appropriate. Top level examples > should work with a cut and paste, if they don't you're going to alienate > new entrants. This is a valid point. I will be happy to help with the wiki since documentation is kind of my bailiwick. My challenge is just being in a position to test everything. I have only so much equipment (and time) so it isn't always easy for me to set things up and do testing. But I will definitely do my best. -MC From mkarp at securesilence.com Fri Feb 13 18:17:25 2009 From: mkarp at securesilence.com (Maxim Karp) Date: Fri, 13 Feb 2009 18:17:25 -0800 Subject: [Freeswitch-users] Not getting a ring back for local extensions on a specific device Message-ID: <002901c98e4a$62ae3d40$280ab7c0$@com> Hello, I am using a SNOM 320 and Windows Mobile 6 VoIP softphone on two separate extensions. When dialing from the SNOM to the WM6 device I get ringback on the SNOM but when calling the SNOM from the WM6 device I don't get ringback though the call does complete and I get voice after the connection. Any ideas? Maxim. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090213/da97b2dc/attachment-0001.html From brian at freeswitch.org Fri Feb 13 19:21:09 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 13 Feb 2009 21:21:09 -0600 Subject: [Freeswitch-users] Not getting a ring back for local extensions on a specific device In-Reply-To: <002901c98e4a$62ae3d40$280ab7c0$@com> References: <002901c98e4a$62ae3d40$280ab7c0$@com> Message-ID: Would need a sip trace to know. TPORT_LOG=1 ./freeswitch /b On Feb 13, 2009, at 8:17 PM, Maxim Karp wrote: > Hello, > > I am using a SNOM 320 and Windows Mobile 6 VoIP softphone on two > separate extensions. When dialing from the SNOM to the WM6 device I > get ringback on the SNOM but when calling the SNOM from the WM6 > device I don?t get ringback though the call does complete and I get > voice after the connection. > > Any ideas? > > Maxim. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090213/5c30238f/attachment.html From woodydickson at gmail.com Fri Feb 13 19:41:28 2009 From: woodydickson at gmail.com (Woody Dickson) Date: Sat, 14 Feb 2009 11:41:28 +0800 Subject: [Freeswitch-users] No-media problem with opensips-freeswitch setup Message-ID: Hi, I tried to configure opensips as sip proxy and sip registrars and freeswitch as B2BUA. Everything works until I start to connect sip clients that are behind ADSL. Both freeswitch and opensips are on public IP and I am using external profile as well. Does anyone have experience in setting up opensips and freeswitch together and can share the configuration with me? Thank you very much in advance for any help. Regards, Woody From brian at freeswitch.org Fri Feb 13 19:55:28 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 13 Feb 2009 21:55:28 -0600 Subject: [Freeswitch-users] No-media problem with opensips-freeswitch setup In-Reply-To: References: Message-ID: You have let the names of the profiles confuse you. Chances are you're trying to hair pin the calls out and back into the same nat. That usually doesn't work. You will need to give me more details about your setup. /b On Feb 13, 2009, at 9:41 PM, Woody Dickson wrote: > Hi, > > I tried to configure opensips as sip proxy and sip registrars and > freeswitch as B2BUA. Everything works until I start to connect sip > clients that are behind ADSL. > > Both freeswitch and opensips are on public IP and I am using external > profile as well. > > Does anyone have experience in setting up opensips and freeswitch > together and can share the configuration with me? > > Thank you very much in advance for any help. > > Regards, > Woody From wiltingtree at gmail.com Fri Feb 13 21:45:30 2009 From: wiltingtree at gmail.com (Adam Wilt) Date: Sat, 14 Feb 2009 00:45:30 -0500 Subject: [Freeswitch-users] monitoring events in Python Message-ID: I'm trying to use custom events for a conference call in a Python script. I set-up the events in the conference.conf.xml file, and I send "bgapi event plain CUSTOM conference::maintenance" to enable them. But I don't know how to look for these events in my script. Does anybody have some example code, or maybe just point me in the right direction? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090214/c4b833cc/attachment.html From egghunt at gmail.com Sat Feb 14 02:27:17 2009 From: egghunt at gmail.com (Arnaldo de Moraes Pereira) Date: Sat, 14 Feb 2009 08:27:17 -0200 Subject: [Freeswitch-users] monitoring events in Python In-Reply-To: References: Message-ID: You might use freepy, which can be found in freeswitch tree, on dir scripts/socket/freepy. fseventlistener.py is a good example to follow. You'll want to pass your custom event to sniff_custom_events(). On Sat, Feb 14, 2009 at 3:45 AM, Adam Wilt wrote: > I'm trying to use custom events for a conference call in a Python script. > I set-up the events in the conference.conf.xml file, and I send "bgapi event > plain CUSTOM conference::maintenance" to enable them. But I don't know how > to look for these events in my script. Does anybody have some example code, > or maybe just point me in the right direction? Thanks. > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Arnaldo M Pereira ap at arnaldopereira.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090214/3bef0643/attachment.html From nik.middleton at noblesolutions.co.uk Sat Feb 14 02:26:54 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Sat, 14 Feb 2009 10:26:54 -0000 Subject: [Freeswitch-users] Hangup hook in js is never called [RESOLVED In-Reply-To: <87f2f3b90902131451t56f84286p13d7a0b815db0068@mail.gmail.com> References: <87f2f3b90902131451t56f84286p13d7a0b815db0068@mail.gmail.com> Message-ID: The JS hook does indeed work. New to js, I hadn't declared the function prior calling it. I can only guess that java scripts are processed sequentially and do not throw up errors if a call is made to a function that hasn't been processed yet Regards, -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 13 February 2009 22:51 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Hangup hook in js is never called > http://wiki.freeswitch.org/wiki/Example_Hangup_hook > > > > Is the code in error? > It might just be. I think you are better off using api_hangup_hook. What are you trying to do on hangup? The api_hangup_hook lets you call any API, including running a script. Here's an example that we played with today that I haven't even put on the wiki yet. I renames the wav file after the call is hung up. From nik.middleton at noblesolutions.co.uk Sat Feb 14 05:17:50 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Sat, 14 Feb 2009 13:17:50 -0000 Subject: [Freeswitch-users] Problems with Originate In-Reply-To: <191c3a030902130554w1dfe6b1dw9e5525cf6a210dc4@mail.gmail.com> References: <1234524794.4431.56.camel@gathern.lan><1234525724.4431.59.camel@gathern.lan> <191c3a030902130554w1dfe6b1dw9e5525cf6a210dc4@mail.gmail.com> Message-ID: Understood. However, using the second method, how can I trap on call failure? If I originate a call and the user is busy, the console reports this fact, but then the script continues to execute if (session.ready()) { console_log("notice","Session result=[" + session.cause + "] \n"); if (session.cause == "USER_BUSY") { Disposition = "BUSY"; session.Hangup(); } In this case session.cause reports 'NONE' and what's surprising is that even though the call failed (busy) session.ready returns a true value. ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: 13 February 2009 13:55 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Problems with Originate The first way is deprecated and will be removed. The 2nd way is the correct way. On Fri, Feb 13, 2009 at 5:48 AM, Alexandru Nedelcu wrote: On Fri, 2009-02-13 at 13:33 +0200, Alexandru Nedelcu wrote: > The problem with this setup is that origination_caller_id_number doesn't > work from inside the JS file (when calling session.originate). I just discovered something interesting. When originating the call like this ... session = new Session("") instead of this ... session = new Session(); session.originate("") ... then it works. Is this some kind of bug, or what's the difference here? Thanks, -- Alexandru Nedelcu Software Developer, Sinapticode _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090214/3674f880/attachment-0001.html From woodydickson at gmail.com Sat Feb 14 06:18:58 2009 From: woodydickson at gmail.com (Woody Dickson) Date: Sat, 14 Feb 2009 22:18:58 +0800 Subject: [Freeswitch-users] No-media problem with opensips-freeswitch setup In-Reply-To: References: Message-ID: Hi My external.xml is just the default configuration: In my opensips.cfg, all the nated traffic is sent to the external_sip_ip and external_rtp_port. Is there anything I should add or change to enable media for device behind nat? Regards, Woody From anthony.minessale at gmail.com Sat Feb 14 07:49:04 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sat, 14 Feb 2009 09:49:04 -0600 Subject: [Freeswitch-users] Not getting a ring back for local extensions on a specific device In-Reply-To: <002901c98e4a$62ae3d40$280ab7c0$@com> References: <002901c98e4a$62ae3d40$280ab7c0$@com> Message-ID: <191c3a030902140749o57172558xd0833bb0a0add7a5@mail.gmail.com> If the device doesn't support 183 early media you may not hear the ringback. The example dialplan uses generated ringback tones. you could edit the default extension and comment out the lines that set the ringback variable and put in one that says "ring_ready" that would send a 180. most devices who implement sip have no idea what they have gotten themselves into and only end up implementing their one test case and not all the possible nightmares that result from trying to "interop" On Fri, Feb 13, 2009 at 8:17 PM, Maxim Karp wrote: > Hello, > > > > I am using a SNOM 320 and Windows Mobile 6 VoIP softphone on two separate > extensions. When dialing from the SNOM to the WM6 device I get ringback on > the SNOM but when calling the SNOM from the WM6 device I don't get ringback > though the call does complete and I get voice after the connection. > > > > Any ideas? > > > > Maxim. > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090214/c08965c0/attachment.html From mike at yes.net.ua Sat Feb 14 02:16:28 2009 From: mike at yes.net.ua (Mike Tkachuk) Date: Sat, 14 Feb 2009 12:16:28 +0200 Subject: [Freeswitch-users] Transcoding G723 Message-ID: <102435003.20090214121628@yes.net.ua> Hello Freeswitch-users, Check this one: http://freehg.org/u/deepwalker/fs_g729/ G.729 is not G.723 but may be interesting. IPP have also g.723 implementation, not too hard to port. That code is working OK for me on development servers. -- Mike From mike at jerris.com Sat Feb 14 09:07:21 2009 From: mike at jerris.com (Michael Jerris) Date: Sat, 14 Feb 2009 12:07:21 -0500 Subject: [Freeswitch-users] monitoring events in Python In-Reply-To: References: Message-ID: <37F78651-0690-4599-B08C-BDBC24562D17@jerris.com> its just "event plain CUSTOM conference::maintenance" no "bgapi" see: http://wiki.freeswitch.org/wiki/Event_Socket for more info. MIke On Feb 14, 2009, at 12:45 AM, Adam Wilt wrote: > I'm trying to use custom events for a conference call in a Python > script. I set-up the events in the conference.conf.xml file, and I > send "bgapi event plain CUSTOM conference::maintenance" to enable > them. But I don't know how to look for these events in my script. > Does anybody have some example code, or maybe just point me in the > right direction? Thanks. From mkarp at securesilence.com Sat Feb 14 09:16:25 2009 From: mkarp at securesilence.com (Maxim Karp) Date: Sat, 14 Feb 2009 09:16:25 -0800 Subject: [Freeswitch-users] Not getting a ring back for local extensions on a specific device In-Reply-To: <191c3a030902140749o57172558xd0833bb0a0add7a5@mail.gmail.com> References: <002901c98e4a$62ae3d40$280ab7c0$@com> <191c3a030902140749o57172558xd0833bb0a0add7a5@mail.gmail.com> Message-ID: <004c01c98ec7$f9163cb0$eb42b610$@com> Hi Anthony, Thanks for the suggestion. In which config file do I change the ringback variable and what should the exact syntax be? Maxim. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Saturday, February 14, 2009 7:49 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Not getting a ring back for local extensions on a specific device If the device doesn't support 183 early media you may not hear the ringback. The example dialplan uses generated ringback tones. you could edit the default extension and comment out the lines that set the ringback variable and put in one that says "ring_ready" that would send a 180. most devices who implement sip have no idea what they have gotten themselves into and only end up implementing their one test case and not all the possible nightmares that result from trying to "interop" On Fri, Feb 13, 2009 at 8:17 PM, Maxim Karp wrote: Hello, I am using a SNOM 320 and Windows Mobile 6 VoIP softphone on two separate extensions. When dialing from the SNOM to the WM6 device I get ringback on the SNOM but when calling the SNOM from the WM6 device I don't get ringback though the call does complete and I get voice after the connection. Any ideas? Maxim. _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090214/67e9a64f/attachment.html From anthony.minessale at gmail.com Sat Feb 14 09:19:50 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sat, 14 Feb 2009 11:19:50 -0600 Subject: [Freeswitch-users] Problems with Originate In-Reply-To: References: <1234524794.4431.56.camel@gathern.lan> <1234525724.4431.59.camel@gathern.lan> <191c3a030902130554w1dfe6b1dw9e5525cf6a210dc4@mail.gmail.com> Message-ID: <191c3a030902140919w806923ex2dc44c2734010251@mail.gmail.com> are you running this as a dialplan app? session is a reserved variable name for the session you executed the app on. are you using an alternate name for your new session like my_session etc....? this works for me, try it yourself. var my_session = new Session("sofia/external/7003 at conference.freeswitch.org "); consoleLog("err", "ready: " + my_session.ready() + "\n"); On Sat, Feb 14, 2009 at 7:17 AM, Nik Middleton < nik.middleton at noblesolutions.co.uk> wrote: > Understood. > > > > However, using the second method, how can I trap on call failure? > > > > If I originate a call and the user is busy, the console reports this fact, > but then the script continues to execute > > > > if (session.ready()) { > > console_log("notice","Session result=[" + > session.cause + "] \n"); > > if (session.cause == "USER_BUSY") { > > Disposition = > "BUSY"; > > session.Hangup(); > > } > > In this case session.cause reports 'NONE' and what's surprising is that > even though the call failed (busy) session.ready returns a true value. > ------------------------------ > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony > Minessale > *Sent:* 13 February 2009 13:55 > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Problems with Originate > > > > The first way is deprecated and will be removed. > The 2nd way is the correct way. > > On Fri, Feb 13, 2009 at 5:48 AM, Alexandru Nedelcu > wrote: > > On Fri, 2009-02-13 at 13:33 +0200, Alexandru Nedelcu wrote: > > The problem with this setup is that origination_caller_id_number doesn't > > work from inside the JS file (when calling session.originate). > > I just discovered something interesting. > > When originating the call like this ... > session = new Session("") > instead of this ... > session = new Session(); session.originate("") > > ... then it works. Is this some kind of bug, or what's the difference > here? > > > Thanks, > > -- > Alexandru Nedelcu > Software Developer, Sinapticode > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090214/4592692a/attachment-0001.html From anthony.minessale at gmail.com Sat Feb 14 09:20:23 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sat, 14 Feb 2009 11:20:23 -0600 Subject: [Freeswitch-users] Not getting a ring back for local extensions on a specific device In-Reply-To: <004c01c98ec7$f9163cb0$eb42b610$@com> References: <002901c98e4a$62ae3d40$280ab7c0$@com> <191c3a030902140749o57172558xd0833bb0a0add7a5@mail.gmail.com> <004c01c98ec7$f9163cb0$eb42b610$@com> Message-ID: <191c3a030902140920t6e808796ue0c0419855fea326@mail.gmail.com> dialplan/defualt.xml in the conf directory look for the word ringback On Sat, Feb 14, 2009 at 11:16 AM, Maxim Karp wrote: > Hi Anthony, > > > > Thanks for the suggestion. In which config file do I change the ringback > variable and what should the exact syntax be? > > > > Maxim. > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony > Minessale > *Sent:* Saturday, February 14, 2009 7:49 AM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Not getting a ring back for local > extensions on a specific device > > > > If the device doesn't support 183 early media you may not hear the > ringback. > The example dialplan uses generated ringback tones. > > you could edit the default extension and comment out the lines that set the > ringback variable > and put in one that says "ring_ready" that would send a 180. > > most devices who implement sip have no idea what they have gotten > themselves into and only end up > implementing their one test case and not all the possible nightmares that > result from trying to "interop" > > On Fri, Feb 13, 2009 at 8:17 PM, Maxim Karp > wrote: > > Hello, > > > > I am using a SNOM 320 and Windows Mobile 6 VoIP softphone on two separate > extensions. When dialing from the SNOM to the WM6 device I get ringback on > the SNOM but when calling the SNOM from the WM6 device I don't get ringback > though the call does complete and I get voice after the connection. > > > > Any ideas? > > > > Maxim. > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090214/c9b3793e/attachment.html From anthony.minessale at gmail.com Sat Feb 14 09:28:11 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sat, 14 Feb 2009 11:28:11 -0600 Subject: [Freeswitch-users] FS 1.0.2 Crash and burn In-Reply-To: <87f2f3b90902131608p5f8ead6fxecebac1b3baf8f52@mail.gmail.com> References: <191c3a030902112025l401c759dife416cf34d51d134@mail.gmail.com> <20090213232426.GA5549@jdc.jasonjgw.net> <87f2f3b90902131608p5f8ead6fxecebac1b3baf8f52@mail.gmail.com> Message-ID: <191c3a030902140928p4f6dada0y3c7f7b3dfb9b8fde@mail.gmail.com> Maybe you can start a new thread then. On Fri, Feb 13, 2009 at 6:08 PM, Michael Collins wrote: > > Right now I'm updating the WIKI where appropriate. Top level examples > > should work with a cut and paste, if they don't you're going to alienate > > new entrants. > > This is a valid point. I will be happy to help with the wiki since > documentation is kind of my bailiwick. My challenge is just being in a > position to test everything. I have only so much equipment (and time) > so it isn't always easy for me to set things up and do testing. But I > will definitely do my best. > > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090214/32a8b7af/attachment.html From nik.middleton at noblesolutions.co.uk Sat Feb 14 11:47:20 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Sat, 14 Feb 2009 19:47:20 -0000 Subject: [Freeswitch-users] Problems with Originate In-Reply-To: <191c3a030902140919w806923ex2dc44c2734010251@mail.gmail.com> References: <1234524794.4431.56.camel@gathern.lan><1234525724.4431.59.camel@gathern.lan><191c3a030902130554w1dfe6b1dw9e5525cf6a210dc4@mail.gmail.com> <191c3a030902140919w806923ex2dc44c2734010251@mail.gmail.com> Message-ID: Nope, Still not working. Here's my little test javascript var new_session = new Session('{ignore_early_media=true,}sofia/internal/1001 at 192.168.3.206'); //set the on_hangup function to be called when this session is hungup new_session.setHangupHook(on_hangup,"hup"); var on_hangup = function(hup_session, how) { console_log("err","In hangup section\n"); //exit here would end the script so you could cleanup and just be done exit(); } if (new_session.ready()) { new_session.answer( ); new_session.sleep(1500); new_session.streamFile("female2.wav"); } And this is the output [NOTICE] sofia.c:3090 sofia_handle_sip_i_state() Hangup sofia/internal/sip:1001 at 192.168.0.29:5060 [CS_CONSUME_MEDIA] [USER_BUSY] [ERR] switch_ivr_originate.c:1116 switch_ivr_originate() Cannot create outgoing channel of type [user] cause: [USER_BUSY] [INFO] mod_dptools.c:1909 audio_bridge_function() Originate Failed. Cause: USER_BUSY [NOTICE] switch_core_session.c:957 switch_core_session_thread() Session 95 (sofia/internal/sip:1001 at 192.168.0.29:5060) Ended [NOTICE] switch_core_session.c:959 switch_core_session_thread() Close Channel sofia/internal/sip:1001 at 192.168.0.29:5060 [CS_HANGUP] [NOTICE] mod_dptools.c:600 answer_function() Channel [sofia/internal/0000000000 at 192.168.3.206] has been answered ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: 14 February 2009 17:20 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Problems with Originate are you running this as a dialplan app? session is a reserved variable name for the session you executed the app on. are you using an alternate name for your new session like my_session etc....? this works for me, try it yourself. var my_session = new Session("sofia/external/7003 at conference.freeswitch.org"); consoleLog("err", "ready: " + my_session.ready() + "\n"); On Sat, Feb 14, 2009 at 7:17 AM, Nik Middleton wrote: Understood. However, using the second method, how can I trap on call failure? If I originate a call and the user is busy, the console reports this fact, but then the script continues to execute if (session.ready()) { console_log("notice","Session result=[" + session.cause + "] \n"); if (session.cause == "USER_BUSY") { Disposition = "BUSY"; session.Hangup(); } In this case session.cause reports 'NONE' and what's surprising is that even though the call failed (busy) session.ready returns a true value. ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: 13 February 2009 13:55 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Problems with Originate The first way is deprecated and will be removed. The 2nd way is the correct way. On Fri, Feb 13, 2009 at 5:48 AM, Alexandru Nedelcu wrote: On Fri, 2009-02-13 at 13:33 +0200, Alexandru Nedelcu wrote: > The problem with this setup is that origination_caller_id_number doesn't > work from inside the JS file (when calling session.originate). I just discovered something interesting. When originating the call like this ... session = new Session("") instead of this ... session = new Session(); session.originate("") ... then it works. Is this some kind of bug, or what's the difference here? Thanks, -- Alexandru Nedelcu Software Developer, Sinapticode _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090214/939795c9/attachment-0001.html From anthony.minessale at gmail.com Sat Feb 14 14:45:13 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sat, 14 Feb 2009 16:45:13 -0600 Subject: [Freeswitch-users] Problems with Originate In-Reply-To: References: <1234524794.4431.56.camel@gathern.lan> <1234525724.4431.59.camel@gathern.lan> <191c3a030902130554w1dfe6b1dw9e5525cf6a210dc4@mail.gmail.com> <191c3a030902140919w806923ex2dc44c2734010251@mail.gmail.com> Message-ID: <191c3a030902141445q7ffb1febm242c6785a9afef7c@mail.gmail.com> What do you suggest is not working? the call failed and it *did not* run the code inside if (new_session.ready()) The call was never established therefore it would not run the hangup hook either. In order to trigger the hangup hook the session would need to exist. If the session could not originate the new_session obj is an empty shell with no actual session inside. var new_session = new Session(, ); if (!new_session.ready()) { // the call never was established. } I gave you real code to try in my last email that you completely ignored..... and finally you are declaring the function wrong declare it at the top. function on_hangup(hup_session, how) { ..... } I sense you seem to think things are going to magically transform into however you are thinking they should work instead of you perhaps learning how they actually work. On Sat, Feb 14, 2009 at 1:47 PM, Nik Middleton < nik.middleton at noblesolutions.co.uk> wrote: > Nope, > > > > Still not working. Here's my little test javascript > > > > var new_session = new Session('{ignore_early_media=true,}sofia/internal/ > 1001 at 192.168.3.206'); > > > > //set the on_hangup function to be called when this session is > hungup > > > new_session.setHangupHook(on_hangup,"hup"); > > > > var on_hangup = function(hup_session, how) > { > > > console_log("err","In hangup > section\n"); > > //exit here would end the script so you > could cleanup and just be > done > > > > exit(); > > > } > > > > > > if (new_session.ready()) { > > new_session.answer( ); > > new_session.sleep(1500); > > new_session.streamFile("female2.wav"); > > } > > > > And this is the output > > > > [NOTICE] sofia.c:3090 sofia_handle_sip_i_state() Hangup sofia/internal/ > sip:1001 at 192.168.0.29:5060 [CS_CONSUME_MEDIA] [USER_BUSY] > > [ERR] switch_ivr_originate.c:1116 switch_ivr_originate() Cannot create > outgoing channel of type [user] cause: [USER_BUSY] > > [INFO] mod_dptools.c:1909 audio_bridge_function() Originate Failed. Cause: > USER_BUSY > > [NOTICE] switch_core_session.c:957 switch_core_session_thread() Session 95 > (sofia/internal/sip:1001 at 192.168.0.29:5060) Ended > > [NOTICE] switch_core_session.c:959 switch_core_session_thread() Close > Channel sofia/internal/sip:1001 at 192.168.0.29:5060 [CS_HANGUP] > > [NOTICE] mod_dptools.c:600 answer_function() Channel [sofia/internal/ > 0000000000 at 192.168.3.206] has been answered > > > ------------------------------ > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony > Minessale > *Sent:* 14 February 2009 17:20 > > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Problems with Originate > > > > are you running this as a dialplan app? > session is a reserved variable name for the session you executed the app > on. > > are you using an alternate name for your new session like my_session > etc....? > > this works for me, try it yourself. > > var my_session = new Session("sofia/external/ > 7003 at conference.freeswitch.org"); > consoleLog("err", "ready: " + my_session.ready() + "\n"); > > > > On Sat, Feb 14, 2009 at 7:17 AM, Nik Middleton < > nik.middleton at noblesolutions.co.uk> wrote: > > Understood. > > > > However, using the second method, how can I trap on call failure? > > > > If I originate a call and the user is busy, the console reports this fact, > but then the script continues to execute > > > > if (session.ready()) { > > console_log("notice","Session result=[" + > session.cause + "] \n"); > > if (session.cause == "USER_BUSY") { > > Disposition = > "BUSY"; > > session.Hangup(); > > } > > In this case session.cause reports 'NONE' and what's surprising is that > even though the call failed (busy) session.ready returns a true value. > ------------------------------ > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony > Minessale > *Sent:* 13 February 2009 13:55 > > > *To:* freeswitch-users at lists.freeswitch.org > > *Subject:* Re: [Freeswitch-users] Problems with Originate > > > > The first way is deprecated and will be removed. > The 2nd way is the correct way. > > On Fri, Feb 13, 2009 at 5:48 AM, Alexandru Nedelcu > wrote: > > On Fri, 2009-02-13 at 13:33 +0200, Alexandru Nedelcu wrote: > > The problem with this setup is that origination_caller_id_number doesn't > > work from inside the JS file (when calling session.originate). > > I just discovered something interesting. > > When originating the call like this ... > session = new Session("") > instead of this ... > session = new Session(); session.originate("") > > ... then it works. Is this some kind of bug, or what's the difference > here? > > > Thanks, > > -- > Alexandru Nedelcu > Software Developer, Sinapticode > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090214/acf28ded/attachment-0001.html From rinorhoxha at hotmail.com Sat Feb 14 15:04:01 2009 From: rinorhoxha at hotmail.com (JCATS) Date: Sat, 14 Feb 2009 15:04:01 -0800 (PST) Subject: [Freeswitch-users] [ANN] Spice Telephony - an open source FreeSWITCH/Erlang callcenter platform Message-ID: <22015518.post@talk.nabble.com> Have you planned any predictive dialer features ( like VICIDIAL )? -- View this message in context: http://www.nabble.com/-ANN--Spice-Telephony---an-open-source-FreeSWITCH-Erlang-callcenter-platform-tp21384907p22015518.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From xspambox at gmail.com Sat Feb 14 16:02:16 2009 From: xspambox at gmail.com (xs) Date: Sun, 15 Feb 2009 01:02:16 +0100 Subject: [Freeswitch-users] [newbie] Clean start with a simple configuration Message-ID: <70ad8fdb0902141602w5829a059p3fe9b4159cba499a@mail.gmail.com> Hi, I installed Freeswitch with this http://wiki.freeswitch.org/wiki/Quick_Startmanual and it works! But it has al sorts of options enabled: 5000,9995, 9996, 9999, 80+[group], 30+[conf], voicemail, something about pizza (00_pizza_demo.xml) and a lot of other xml files and directories (default.xml and default/) etc. etc. That is not what i want. When i install Asterisk (the only PBX that i have experience with) and i clean out sip.conf and extensions.conf and replace them with this: ---------------------------------------- sip.conf [general] [1001] username=1001 secret=password1001 type=friend context=phones host=dynamic qualify=yes disallow=all allow=g726 [1002] username=1002 secret=password1002 type=friend context=phones host=dynamic qualify=yes disallow=all allow=ilbc ---------------------------------------- extensions.conf ---------------------------------------- [phones] exten => _[1001-1002],1,Dial(SIP/${EXTEN},60) exten => _[1001-1002],n,Hangup() ---------------------------------------- then i can simply call from one (local) sip phone to another and force transcoding between them. Nothing else. I want that also with Freeswitch. It is a good starting point but i am fiddeling with it for a couple of days now, read the docs but i can't get Freeswitch to do just this. So just calling between a few local sip phones with transcoding and _everything_ else disabled. Can anyone help me a little? Thanks in advance! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090215/80256b70/attachment.html From brian at freeswitch.org Sat Feb 14 18:10:30 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 14 Feb 2009 20:10:30 -0600 Subject: [Freeswitch-users] [newbie] Clean start with a simple configuration In-Reply-To: <70ad8fdb0902141602w5829a059p3fe9b4159cba499a@mail.gmail.com> References: <70ad8fdb0902141602w5829a059p3fe9b4159cba499a@mail.gmail.com> Message-ID: <13479F65-7CE3-4C41-9845-6D92A647C946@freeswitch.org> FreeSWITCH default config already has this feature. Register two phones... 1000 and 1001 both with password of 1234 then you can call between them. That will work exactly as you want out of the box. Expect more simplified configs to show up after 1.0.3. /b On Feb 14, 2009, at 6:02 PM, xs wrote: > I want that also with Freeswitch. It is a good starting point but i > am fiddeling with it for a couple of days now, read the docs but i > can't get Freeswitch to do just this. So just calling between a few > local sip phones with transcoding and _everything_ else disabled. From brian at freeswitch.org Sat Feb 14 18:12:02 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 14 Feb 2009 20:12:02 -0600 Subject: [Freeswitch-users] [ANN] Spice Telephony - an open source FreeSWITCH/Erlang callcenter platform In-Reply-To: <22015518.post@talk.nabble.com> References: <22015518.post@talk.nabble.com> Message-ID: <1F887BD4-3B6B-4A87-89C9-6E6A823996FE@freeswitch.org> And are you planning on contributing anything useful back to the community? Or just take take take? /b On Feb 14, 2009, at 5:04 PM, JCATS wrote: > > Have you planned any predictive dialer features ( like VICIDIAL )? From krice at freeswitch.org Sat Feb 14 18:16:45 2009 From: krice at freeswitch.org (Ken Rice) Date: Sat, 14 Feb 2009 20:16:45 -0600 Subject: [Freeswitch-users] [ANN] Spice Telephony - an open source FreeSWITCH/Erlang callcenter platform In-Reply-To: <22015518.post@talk.nabble.com> Message-ID: Vicidial is not a predictive dialer... At the best its a power dialer... And spice is not an outbound thing its an inbound queue manager... If you need a predictive dialer contact me offlist or contact one of the other freeswitch developers directly we're be happy to consult for you on building a dialer. > From: JCATS > Reply-To: > Date: Sat, 14 Feb 2009 15:04:01 -0800 (PST) > To: > Subject: Re: [Freeswitch-users] [ANN] Spice Telephony - an open source > FreeSWITCH/Erlang callcenter platform > > > Have you planned any predictive dialer features ( like VICIDIAL )? > > -- > View this message in context: > http://www.nabble.com/-ANN--Spice-Telephony---an-open-source-FreeSWITCH-Erlang > -callcenter-platform-tp21384907p22015518.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From tim at marocon.com Sat Feb 14 18:20:13 2009 From: tim at marocon.com (Tim Mattison) Date: Sat, 14 Feb 2009 21:20:13 -0500 Subject: [Freeswitch-users] [newbie] Clean start with a simple configuration In-Reply-To: <13479F65-7CE3-4C41-9845-6D92A647C946@freeswitch.org> References: <70ad8fdb0902141602w5829a059p3fe9b4159cba499a@mail.gmail.com> <13479F65-7CE3-4C41-9845-6D92A647C946@freeswitch.org> Message-ID: <2FA0A303-9D53-453A-89A3-517D4B7FBC19@marocon.com> I think the crux of the matter is "I can't get Freeswitch to do _just_ this" (emphasis added). Users that are relatively new to FreeSWITCH but Asterisk veterans want to know how to build a simple, lean configuration. I'm trying to figure it out myself right now. If simplified configs are only going to be posted after 1.0.3 what's the best place to post my findings and configs? Is there an explicit, sanctioned place for this kind of thing on the Wiki already? Tim On Feb 14, 2009, at 9:10 PM, Brian West wrote: > FreeSWITCH default config already has this feature. Register two > phones... 1000 and 1001 both with password of 1234 then you can call > between them. That will work exactly as you want out of the box. > Expect more simplified configs to show up after 1.0.3. > > /b > > On Feb 14, 2009, at 6:02 PM, xs wrote: > >> I want that also with Freeswitch. It is a good starting point but i >> am fiddeling with it for a couple of days now, read the docs but i >> can't get Freeswitch to do just this. So just calling between a few >> local sip phones with transcoding and _everything_ else disabled. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090214/3dc7dd89/attachment.html From brian at freeswitch.org Sat Feb 14 18:28:45 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 14 Feb 2009 20:28:45 -0600 Subject: [Freeswitch-users] [newbie] Clean start with a simple configuration In-Reply-To: <2FA0A303-9D53-453A-89A3-517D4B7FBC19@marocon.com> References: <70ad8fdb0902141602w5829a059p3fe9b4159cba499a@mail.gmail.com> <13479F65-7CE3-4C41-9845-6D92A647C946@freeswitch.org> <2FA0A303-9D53-453A-89A3-517D4B7FBC19@marocon.com> Message-ID: <2598EA3B-E07B-448A-8915-F974A7F1594F@freeswitch.org> Well the config itself is easy to follow. The problem is coming from Asterisk you have the wrong mindset to approach most things in FreeSWITCH. http://svn.freeswitch.org/svn/configs/ (softphone is the best small config to look at) The default config is easy to strip down once you take a few moments to look at it and maybe hop on IRC and ask questions. I wrote the default and the softphone config. So if you have questions I'm just the person to ask. More config sets will appear as I have time to write them. /b On Feb 14, 2009, at 8:20 PM, Tim Mattison wrote: > I think the crux of the matter is "I can't get Freeswitch to do > _just_ this" (emphasis added). Users that are relatively new to > FreeSWITCH but Asterisk veterans want to know how to build a simple, > lean configuration. I'm trying to figure it out myself right now. > > If simplified configs are only going to be posted after 1.0.3 what's > the best place to post my findings and configs? Is there an > explicit, sanctioned place for this kind of thing on the Wiki already? > > Tim From pauld at versafon.com Sat Feb 14 18:37:49 2009 From: pauld at versafon.com (Paul D.) Date: Sat, 14 Feb 2009 21:37:49 -0500 Subject: [Freeswitch-users] FS SIP audio quality? Message-ID: <49977FFD.1020002@versafon.com> Comparing FS 1.0.3 audio quality vs * 1.4.2, simple Sip-to-Sip call, or call to VM prompt, or call via gateway to PSTN - FS audio volume level (should I say gain?) seems noticeably lower than on *, this may be a reason that FS audio seems to be subpar, more noise less clear. Test calls made using PCMU codec from X-Lite and Linksys 2002. Is there anything can be tweaked in FS to correct that? Same issue was with 1.0.2. From brian at freeswitch.org Sat Feb 14 18:43:40 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 14 Feb 2009 20:43:40 -0600 Subject: [Freeswitch-users] FS SIP audio quality? In-Reply-To: <49977FFD.1020002@versafon.com> References: <49977FFD.1020002@versafon.com> Message-ID: I haven't ever experienced this issue can you maybe elaborate on the issue a little more? We usually hear that the audio quality is much better... have you tried latest SVN trunk? If resampling was involved it might cause some audio issues but those were usually gain issue and that has since been fixed in SVN trunk as of yesterday. /b On Feb 14, 2009, at 8:37 PM, Paul D. wrote: > Comparing FS 1.0.3 audio quality vs * 1.4.2, simple Sip-to-Sip call, > or > call to VM prompt, or call via gateway to PSTN - FS audio volume > level > (should I say gain?) seems noticeably lower than on *, this may be a > reason that FS audio seems to be subpar, more noise less clear. Test > calls made using PCMU codec from X-Lite and Linksys 2002. > Is there anything can be tweaked in FS to correct that? Same issue was > with 1.0.2. From pauld at versafon.com Sat Feb 14 19:02:54 2009 From: pauld at versafon.com (Paul D.) Date: Sat, 14 Feb 2009 22:02:54 -0500 Subject: [Freeswitch-users] FS SIP audio quality? In-Reply-To: References: <49977FFD.1020002@versafon.com> Message-ID: <499785DE.3010805@versafon.com> I am not sure what else I can add to that, I would love to elaborate more if you ask anything specific. I haven't tried the latest trunk, but since there's no difference between 1.0.2 and 1.0.3RC1 in audio quality I don't think it make sense trying. From what I see in FS logs there's no resampling involved, and that looks like true since I specifically restricted codecs in my test SIP equipment. But the fact is I tried different boxes, same OS centos 5.2 x64, and I had to bring audio volume and mic level all the way up in X-Lite to compensate for the difference to * audio, and in * such volume level sounds like way too high. FS installed cleanly from scratch, mostly default settings, except some dialplan/directory additions. Brian West wrote: > I haven't ever experienced this issue can you maybe elaborate on the > issue a little more? We usually hear that the audio quality is much > better... have you tried latest SVN trunk? If resampling was involved > it might cause some audio issues but those were usually gain issue and > that has since been fixed in SVN trunk as of yesterday. > > /b > > On Feb 14, 2009, at 8:37 PM, Paul D. wrote: > > >> Comparing FS 1.0.3 audio quality vs * 1.4.2, simple Sip-to-Sip call, >> or >> call to VM prompt, or call via gateway to PSTN - FS audio volume >> level >> (should I say gain?) seems noticeably lower than on *, this may be a >> reason that FS audio seems to be subpar, more noise less clear. Test >> calls made using PCMU codec from X-Lite and Linksys 2002. >> Is there anything can be tweaked in FS to correct that? Same issue was >> with 1.0.2. >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From brian at freeswitch.org Sat Feb 14 19:10:41 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 14 Feb 2009 21:10:41 -0600 Subject: [Freeswitch-users] FS SIP audio quality? In-Reply-To: <499785DE.3010805@versafon.com> References: <49977FFD.1020002@versafon.com> <499785DE.3010805@versafon.com> Message-ID: <034A0C6C-0C28-4C9C-8ECA-379E090387F6@freeswitch.org> What is odd some people have reported the same issue with Asterisk. I would like to get to the bottom of it but nobody can provide any more detail on what might be going on and I haven't experienced this issue with the 30 or so phones I have on my desk .... I highly recommend you try SVN trunk. Let me know how that goes. ;) /b On Feb 14, 2009, at 9:02 PM, Paul D. wrote: > I am not sure what else I can add to that, I would love to elaborate > more if you ask anything specific. > I haven't tried the latest trunk, but since there's no difference > between 1.0.2 and 1.0.3RC1 in audio quality I don't think > it make sense trying. From what I see in FS logs there's no resampling > involved, and that looks like true since I specifically restricted > codecs in my test SIP equipment. > But the fact is I tried different boxes, same OS centos 5.2 x64, and I > had to bring audio volume and mic level all the way up in X-Lite to > compensate for the difference to * audio, > and in * such volume level sounds like way too high. > FS installed cleanly from scratch, mostly default settings, except > some > dialplan/directory additions. From jason at jasonjgw.net Sat Feb 14 19:18:53 2009 From: jason at jasonjgw.net (Jason White) Date: Sun, 15 Feb 2009 14:18:53 +1100 Subject: [Freeswitch-users] FS SIP audio quality? In-Reply-To: <034A0C6C-0C28-4C9C-8ECA-379E090387F6@freeswitch.org> References: <49977FFD.1020002@versafon.com> <499785DE.3010805@versafon.com> <034A0C6C-0C28-4C9C-8ECA-379E090387F6@freeswitch.org> Message-ID: <20090215031853.GA4488@jdc.jasonjgw.net> Brian West wrote: > What is odd some people have reported the same issue with Asterisk. I > would like to get to the bottom of it but nobody can provide any more > detail on what might be going on and I haven't experienced this issue > with the 30 or so phones I have on my desk .... I highly recommend you A data point that may or may not be helpful: if I set up PortAudio on FreeSWITCH and call an Asterisk conference from there, the audio is significantly louder than a comparable SIP call with another FreeSWITCH box at the other end. > try SVN trunk. Let me know how that goes. ;) I'll recreate the above scenario with SVN trunk (I've just built rev. 12018), and report if there is still a problem. I sometimes get audio distortion in the above situation if anyone speaks too loudly. I suspect clipping somewhere in the audio processing. From brian at freeswitch.org Sat Feb 14 19:27:06 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 14 Feb 2009 21:27:06 -0600 Subject: [Freeswitch-users] FS SIP audio quality? In-Reply-To: <20090215031853.GA4488@jdc.jasonjgw.net> References: <49977FFD.1020002@versafon.com> <499785DE.3010805@versafon.com> <034A0C6C-0C28-4C9C-8ECA-379E090387F6@freeswitch.org> <20090215031853.GA4488@jdc.jasonjgw.net> Message-ID: <0DF0A788-0AA7-4FA6-A0FC-DABC19E0A148@freeswitch.org> This was a problem with the resampler which was replaced... we use the resampler in Speex now which will not exhibit the problem. /b On Feb 14, 2009, at 9:18 PM, Jason White wrote: > I sometimes get audio distortion in the above situation if anyone > speaks too > loudly. I suspect clipping somewhere in the audio processing. From jason at jasonjgw.net Sat Feb 14 23:02:16 2009 From: jason at jasonjgw.net (Jason White) Date: Sun, 15 Feb 2009 18:02:16 +1100 Subject: [Freeswitch-users] Build and Sofia issues with recent svn trunk revisions Message-ID: <20090215070216.GA20246@jdc.jasonjgw.net> Following the resampling discussion, I tried upgrading to revision 12018. which compiled cleanly, but then failed to load my internal SIP profile: [ERR] sofia.c:739 sofia_profile_thread_run() Error Creating SIP UA for profile: internal The same configuration works under revision 11488, to which I've temporarily downgraded. Either something has changed in FreeSWITCH that requires modifications to my SIP configuration, or this is a regression. I decided to try rev. 12027, which, on the same machine (Debian Sid) fails to build with the following error: x86_64-linux-gnu-gcc -c -ggdb -I. ./dftables.c In file included from ./dftables.c:50: ./pcre_internal.h:239:2: error: #error LINK_SIZE must be either 2, 3, or 4 make[2]: *** [dftables.o] Error 1 I suspect that recent changes to the build system are responsible for the latter. I'm sure these are minor matters that will be sorted out soon. Thanks once again to the developers for a great project! From alex at sinapticode.ro Sun Feb 15 00:47:50 2009 From: alex at sinapticode.ro (Alexandru Nedelcu) Date: Sun, 15 Feb 2009 10:47:50 +0200 Subject: [Freeswitch-users] Hangup hook in js is never called In-Reply-To: References: <87f2f3b90902131451t56f84286p13d7a0b815db0068@mail.gmail.com> Message-ID: <1234687670.4604.5.camel@gathern.lan> I'm using the CDR logs for that, because I need other info as well. To make the connection between a log-file and a DB record, I'm passing a custom channel variable on originate. I've written a short article about it, but as the other one, it's a draft: http://alexn.org/docs/dialer_part_2.html On Fri, 2009-02-13 at 23:07 +0000, Nik Middleton wrote: > I'm trying to capture the hang-up reason and write it to the db (Was it > busy etc). I also close the db in that function. That way I know I > don't have any open connections. This is in JavaScript BTW > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Michael Collins > Sent: 13 February 2009 22:51 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Hangup hook in js is never called > > > http://wiki.freeswitch.org/wiki/Example_Hangup_hook > > > > > > > > Is the code in error? > > > > It might just be. I think you are better off using api_hangup_hook. > What are you trying to do on hangup? The api_hangup_hook lets you call > any API, including running a script. Here's an example that we played > with today that I haven't even put on the wiki yet. I renames the wav > file after the call is hung up. > > > > > > > > > > > > > > > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From nik.middleton at noblesolutions.co.uk Sun Feb 15 03:39:12 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Sun, 15 Feb 2009 11:39:12 -0000 Subject: [Freeswitch-users] Problems with Originate In-Reply-To: <191c3a030902141445q7ffb1febm242c6785a9afef7c@mail.gmail.com> References: <1234524794.4431.56.camel@gathern.lan><1234525724.4431.59.camel@gathern.lan><191c3a030902130554w1dfe6b1dw9e5525cf6a210dc4@mail.gmail.com><191c3a030902140919w806923ex2dc44c2734010251@mail.gmail.com> <191c3a030902141445q7ffb1febm242c6785a9afef7c@mail.gmail.com> Message-ID: You are indeed correct. I still had my asterisk hat on, and was expecting a hang-up event to be fired with the call outcome. I was explicitly testing for call failure. I've not modified the code to test the result of the originate and it works as expected. I will add some words to the wiki explaining this for those converting from Asterisk Regards, ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: 14 February 2009 22:45 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Problems with Originate What do you suggest is not working? the call failed and it *did not* run the code inside if (new_session.ready()) The call was never established therefore it would not run the hangup hook either. In order to trigger the hangup hook the session would need to exist. If the session could not originate the new_session obj is an empty shell with no actual session inside. var new_session = new Session(, ); if (!new_session.ready()) { // the call never was established. } I gave you real code to try in my last email that you completely ignored..... and finally you are declaring the function wrong declare it at the top. function on_hangup(hup_session, how) { ..... } I sense you seem to think things are going to magically transform into however you are thinking they should work instead of you perhaps learning how they actually work. On Sat, Feb 14, 2009 at 1:47 PM, Nik Middleton wrote: Nope, Still not working. Here's my little test javascript var new_session = new Session('{ignore_early_media=true,}sofia/internal/1001 at 192.168.3.206'); //set the on_hangup function to be called when this session is hungup new_session.setHangupHook(on_hangup,"hup"); var on_hangup = function(hup_session, how) { console_log("err","In hangup section\n"); //exit here would end the script so you could cleanup and just be done exit(); } if (new_session.ready()) { new_session.answer( ); new_session.sleep(1500); new_session.streamFile("female2.wav"); } And this is the output [NOTICE] sofia.c:3090 sofia_handle_sip_i_state() Hangup sofia/internal/sip:1001 at 192.168.0.29:5060 [CS_CONSUME_MEDIA] [USER_BUSY] [ERR] switch_ivr_originate.c:1116 switch_ivr_originate() Cannot create outgoing channel of type [user] cause: [USER_BUSY] [INFO] mod_dptools.c:1909 audio_bridge_function() Originate Failed. Cause: USER_BUSY [NOTICE] switch_core_session.c:957 switch_core_session_thread() Session 95 (sofia/internal/sip:1001 at 192.168.0.29:5060) Ended [NOTICE] switch_core_session.c:959 switch_core_session_thread() Close Channel sofia/internal/sip:1001 at 192.168.0.29:5060 [CS_HANGUP] [NOTICE] mod_dptools.c:600 answer_function() Channel [sofia/internal/0000000000 at 192.168.3.206] has been answered ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: 14 February 2009 17:20 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Problems with Originate are you running this as a dialplan app? session is a reserved variable name for the session you executed the app on. are you using an alternate name for your new session like my_session etc....? this works for me, try it yourself. var my_session = new Session("sofia/external/7003 at conference.freeswitch.org"); consoleLog("err", "ready: " + my_session.ready() + "\n"); On Sat, Feb 14, 2009 at 7:17 AM, Nik Middleton wrote: Understood. However, using the second method, how can I trap on call failure? If I originate a call and the user is busy, the console reports this fact, but then the script continues to execute if (session.ready()) { console_log("notice","Session result=[" + session.cause + "] \n"); if (session.cause == "USER_BUSY") { Disposition = "BUSY"; session.Hangup(); } In this case session.cause reports 'NONE' and what's surprising is that even though the call failed (busy) session.ready returns a true value. ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: 13 February 2009 13:55 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Problems with Originate The first way is deprecated and will be removed. The 2nd way is the correct way. On Fri, Feb 13, 2009 at 5:48 AM, Alexandru Nedelcu wrote: On Fri, 2009-02-13 at 13:33 +0200, Alexandru Nedelcu wrote: > The problem with this setup is that origination_caller_id_number doesn't > work from inside the JS file (when calling session.originate). I just discovered something interesting. When originating the call like this ... session = new Session("") instead of this ... session = new Session(); session.originate("") ... then it works. Is this some kind of bug, or what's the difference here? Thanks, -- Alexandru Nedelcu Software Developer, Sinapticode _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090215/590620a3/attachment-0001.html From daletrub at gmail.com Sat Feb 14 21:15:04 2009 From: daletrub at gmail.com (Dale Trub) Date: Sat, 14 Feb 2009 21:15:04 -0800 Subject: [Freeswitch-users] help! DTMFs disappearing in mod_conference Message-ID: Hey folks, I'm having a very odd issue and I'm wondering if anyone else has seen this, or if there's a setting to change etc. I should mention that if anyone by chance helps THIS WEEKEND, it could SAVE my butt. We are doing an important demo monday morning and honestly this stops us in our tracks. We are listening for DTMFs from mod_conference and passing that via the socket on to a separate display layer (in development). It works perfectly, but at a certain point in a conference, it seems the switch stops sensing the DTMFs on most (but not all) lines. FYI, we saw this before with FS 1.0 running on a VPS slice and thought maybe it was somehow related to that box, or that DID provider. We've now switched to a full server and a different DID provider, and are getting the exact same behavior. Today, here was the deal: - 10 people called in (practice walkthrough of our demo this monday) - all lines: DTMFs displayed - tried them several times - 6= mute/unmute also works (doesn't go through our display layer) - about 30 minutes in, again asked everyone to hit 1 (which again we pass to display layer) - and now most lines do not pass DTMFs - a couple lines still do pass them - (the "6" which we trap within FS as "mute/unmute" also stops working on those lines that stopped passing others) - the FS logs STOP reflecting DTMFs from the lines where we don't see them - so, we know it's FS and not our application - some time passes - keep trying the working ones -- eventually they stop working - one caller (with DTMFs non functional) hangs up and calls back - that caller now does have DTMFs working - we hung up and called back in - this time DTMFs worked ~100 times, and then again stopped - switched logs from INFO to DEBUG - below are some log file entries We're on CENT-OS and FS 1.0.2 Besides the obvious question ("how do I fix this") Non-obvious Questions: - Is there any way to tell if the DID provider is trapping the DTMFs and sending them out of band, or is sending them in-band? - Is there any reasonably easy way to get in and see/sniff/visualize/measure the SIP packets to see what is coming in? - Could this be related to this? http://wiki.freeswitch.org/wiki/RTP_Issues - Any other thoughts on how to debug? Thanks!! -Dale Here's the last working DTMF, and then some events I don't know ... through a place where this definitely wasn't working. 2009-02-14 22:26:03 [DEBUG] switch_rtp.c:1701 switch_rtp_dequeue_dtmf() RTP RECV DTMF 5:2000 2009-02-14 22:37:06 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state() Channel sofi a/external/xxphonenumxx at 172.16.250.4 entering state [received] 2009-02-14 22:37:06 [DEBUG] sofia.c:2546 sofia_handle_sip_i_state() Remote SDP: v=0 o=FreeSWITCH 8044373728746667485 7321340529655007764 IN IP4 172.16.250.4 s=FreeSWITCH c=IN IP4 172.16.3.13 t=0 0 m=audio 33440 RTP/AVP 0 101 a=rtpmap:101 telephone-event/8000 a=ptime:20 2009-02-14 22:37:06 [DEBUG] sofia_glue.c:2447 sofia_glue_negotiate_sdp() Our exi sting sdp is still good [PCMU 172.16.3.13:33440], let's keep it. 2009-02-14 22:37:06 [DEBUG] sofia_glue.c:2473 sofia_glue_negotiate_sdp() Set 283 3 dtmf payload to 101 2009-02-14 22:37:06 [DEBUG] sofia_glue.c:1880 sofia_glue_activate_rtp() Audio pa rams are unchanged for sofia/external/xxphonenumxx at 172.16.250.4. 2009-02-14 22:37:06 [DEBUG] sofia.c:2896 sofia_handle_sip_i_state() Processing R einvite 2009-02-14 22:37:06 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state() Channel sofi a/external/xxphonenumxx at 172.16.250.4 entering state [completed] 2009-02-14 22:37:06 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state() Channel sofi a/external/xxphonenumxx at 172.16.250.4 entering state [ready] 2009-02-14 22:38:34 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state() Channel sofi a/external/xxphonenumxx2 at 172.16.250.4 entering state [received] 2009-02-14 22:38:34 [DEBUG] sofia.c:2546 sofia_handle_sip_i_state() Remote SDP: v=0 o=FreeSWITCH 934104982290142318 4836750446264379897 IN IP4 172.16.250.4 s=FreeSWITCH c=IN IP4 172.16.1.21 t=0 0 m=audio 35356 RTP/AVP 0 101 a=rtpmap:101 telephone-event/8000 a=ptime:20 2009-02-14 22:38:34 [DEBUG] sofia_glue.c:2447 sofia_glue_negotiate_sdp() Our exi sting sdp is still good [PCMU 172.16.1.21:35356], let's keep it. 2009-02-14 22:38:34 [DEBUG] sofia_glue.c:2473 sofia_glue_negotiate_sdp() Set 283 3 dtmf payload to 101 2009-02-14 22:38:34 [DEBUG] sofia_glue.c:1880 sofia_glue_activate_rtp() Audio pa rams are unchanged for sofia/external/xxphonenumxx2 at 172.16.250.4. 2009-02-14 22:38:34 [DEBUG] sofia_glue.c:1880 sofia_glue_activate_rtp() Audio pa rams are unchanged for sofia/external/xxphonenumxx2 at 172.16.250.4. 2009-02-14 22:38:34 [DEBUG] sofia.c:2896 sofia_handle_sip_i_state() Processing R einvite 2009-02-14 22:38:34 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state() Channel sofi a/external/xxphonenumxx2 at 172.16.250.4 entering state [completed] 2009-02-14 22:38:34 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state() Channel sofi a/external/xxphonenumxx2 at 172.16.250.4 entering state [ready] -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090214/5051ba09/attachment-0001.html From shahal at jajah.com Sat Feb 14 23:47:10 2009 From: shahal at jajah.com (Shahal Hazan) Date: Sun, 15 Feb 2009 09:47:10 +0200 Subject: [Freeswitch-users] Adding a third person to an ongoing conversation by dialing DTMF Message-ID: I would to add a third person to an ongoing conversation (between two SIP callers for example) by dialing a DTMF. The DTMF can be as simple as 1 or 2 or 3 to add a predefined person: person1 or person2 or person3 respectively. What is the best way to accomplish that? 1) Receiving the DTMF: After I added: I wasn't able to receive the DTMF on a non IVR call (I only got the IVR examples to work) Can I capture the DTMF in JS? 2) Adding the third person: a. Creating a conference on the fly and adding the third person? b. Bridging the third person? c. Using an API to "originate" a new call added to the current call? d. Creating a predefined group and adding members from that group per caller's choice? Thanks, Shahal Hazan This mail was sent via Mail-SeCure System. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090215/e93e747d/attachment-0001.html From daletrub at gmail.com Sun Feb 15 08:29:59 2009 From: daletrub at gmail.com (Dale Trub) Date: Sun, 15 Feb 2009 08:29:59 -0800 Subject: [Freeswitch-users] help! DTMFs disappearing in mod_conference In-Reply-To: References: Message-ID: The bug I describe sure looks a lot like: http://jira.freeswitch.org/browse/FSCORE-266 We have a direct Metaswitch-> FS connection, and both machines in the same LAN/location. It's 64-bit CentOS btw. Bug also occurred on a 32-bit CentOS dev machine. On Sat, Feb 14, 2009 at 9:15 PM, Dale Trub wrote: > Hey folks, > I'm having a very odd issue and I'm wondering if anyone else has seen this, > or if there's a setting to change etc. > > I should mention that if anyone by chance helps THIS WEEKEND, it could SAVE > my butt. We are doing an important demo monday morning and honestly this > stops us in our tracks. > > We are listening for DTMFs from mod_conference and passing that via the > socket on to a separate display layer (in development). > > It works perfectly, but at a certain point in a conference, it seems the > switch stops sensing the DTMFs on most (but not all) lines. > > FYI, we saw this before with FS 1.0 running on a VPS slice and thought > maybe it was somehow related to that box, or that DID provider. We've now > switched to a full server and a different DID provider, and are getting the > exact same behavior. > > Today, here was the deal: > > - 10 people called in (practice walkthrough of our demo this monday) > - all lines: DTMFs displayed - tried them several times > - 6= mute/unmute also works (doesn't go through our display layer) > - about 30 minutes in, again asked everyone to hit 1 (which again we > pass to display layer) > - and now most lines do not pass DTMFs > - a couple lines still do pass them > - (the "6" which we trap within FS as "mute/unmute" also stops > working on those lines that stopped passing others) > - the FS logs STOP reflecting DTMFs from the lines where we don't > see them > - so, we know it's FS and not our application > - some time passes > - keep trying the working ones -- eventually they stop working > - one caller (with DTMFs non functional) hangs up and calls back > - that caller now does have DTMFs working > - we hung up and called back in > - this time DTMFs worked ~100 times, and then again stopped > - switched logs from INFO to DEBUG > - below are some log file entries > > > We're on CENT-OS and FS 1.0.2 > > Besides the obvious question ("how do I fix this") > > Non-obvious Questions: > > - Is there any way to tell if the DID provider is trapping the DTMFs > and sending them out of band, or is sending them in-band? > - Is there any reasonably easy way to get in and > see/sniff/visualize/measure the SIP packets to see what is coming in? > - Could this be related to this? > http://wiki.freeswitch.org/wiki/RTP_Issues > - Any other thoughts on how to debug? > > Thanks!! > > -Dale > > Here's the last working DTMF, and then some events I don't know ... through > a place where this definitely wasn't working. > > > 2009-02-14 22:26:03 [DEBUG] switch_rtp.c:1701 switch_rtp_dequeue_dtmf() RTP > RECV > DTMF 5:2000 > 2009-02-14 22:37:06 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state() Channel > sofi > a/external/xxphonenumxx at 172.16.250.4 entering state [received] > 2009-02-14 22:37:06 [DEBUG] sofia.c:2546 sofia_handle_sip_i_state() Remote > SDP: > v=0 > o=FreeSWITCH 8044373728746667485 7321340529655007764 IN IP4 172.16.250.4 > s=FreeSWITCH > c=IN IP4 172.16.3.13 > t=0 0 > m=audio 33440 RTP/AVP 0 101 > a=rtpmap:101 telephone-event/8000 > a=ptime:20 > > 2009-02-14 22:37:06 [DEBUG] sofia_glue.c:2447 sofia_glue_negotiate_sdp() > Our exi > sting sdp is still good [PCMU 172.16.3.13:33440], let's keep it. > 2009-02-14 22:37:06 [DEBUG] sofia_glue.c:2473 sofia_glue_negotiate_sdp() > Set 283 > 3 dtmf payload to 101 > 2009-02-14 22:37:06 [DEBUG] sofia_glue.c:1880 sofia_glue_activate_rtp() > Audio pa > rams are unchanged for sofia/external/xxphonenumxx at 172.16.250.4. > 2009-02-14 22:37:06 [DEBUG] sofia.c:2896 sofia_handle_sip_i_state() > Processing R > einvite > 2009-02-14 22:37:06 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state() Channel > sofi > a/external/xxphonenumxx at 172.16.250.4 entering state [completed] > 2009-02-14 22:37:06 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state() Channel > sofi > a/external/xxphonenumxx at 172.16.250.4 entering state [ready] > 2009-02-14 22:38:34 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state() Channel > sofi > a/external/xxphonenumxx2 at 172.16.250.4 entering state [received] > 2009-02-14 22:38:34 [DEBUG] sofia.c:2546 sofia_handle_sip_i_state() Remote > SDP: > v=0 > o=FreeSWITCH 934104982290142318 4836750446264379897 IN IP4 172.16.250.4 > s=FreeSWITCH > c=IN IP4 172.16.1.21 > t=0 0 > m=audio 35356 RTP/AVP 0 101 > a=rtpmap:101 telephone-event/8000 > a=ptime:20 > > 2009-02-14 22:38:34 [DEBUG] sofia_glue.c:2447 sofia_glue_negotiate_sdp() > Our exi > sting sdp is still good [PCMU 172.16.1.21:35356], let's keep it. > 2009-02-14 22:38:34 [DEBUG] sofia_glue.c:2473 sofia_glue_negotiate_sdp() > Set 283 > 3 dtmf payload to 101 > 2009-02-14 22:38:34 [DEBUG] sofia_glue.c:1880 sofia_glue_activate_rtp() > Audio pa > rams are unchanged for sofia/external/xxphonenumxx2 at 172.16.250.4. > > 2009-02-14 22:38:34 [DEBUG] sofia_glue.c:1880 sofia_glue_activate_rtp() > Audio pa > rams are unchanged for sofia/external/xxphonenumxx2 at 172.16.250.4. > 2009-02-14 22:38:34 [DEBUG] sofia.c:2896 sofia_handle_sip_i_state() > Processing R > einvite > 2009-02-14 22:38:34 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state() Channel > sofi > a/external/xxphonenumxx2 at 172.16.250.4 entering state [completed] > 2009-02-14 22:38:34 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state() Channel > sofi > a/external/xxphonenumxx2 at 172.16.250.4 entering state [ready] > > > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090215/05fe60da/attachment-0001.html From pbd at suspiria.net Sun Feb 15 11:16:25 2009 From: pbd at suspiria.net (Public Dump) Date: Sun, 15 Feb 2009 20:16:25 +0100 Subject: [Freeswitch-users] High CPU load after starting Message-ID: <13C421883438EB42B9E2C30069FD4AB76AEA2B38A2@crushinator.central.local> So, no ideas left how to fix this problem ? Von: Public Dump Gesendet: Dienstag, 10. Februar 2009 19:42 An: 'freeswitch-users at lists.freeswitch.org' Betreff: High CPU load after starting After starting FreeSwitch (1.0.2) on a 4 core server running Windows Server 2008, the CPU load (privileged time/kernel) for one of the cores goes to 50% and stays there. Stoping FreeSwitch stops the load. I have tried to disable all modules but the problem persists. Has anybody seen this problem, can it be fixed ? regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090215/9615846e/attachment.html From brian at freeswitch.org Sun Feb 15 12:09:50 2009 From: brian at freeswitch.org (Brian West) Date: Sun, 15 Feb 2009 14:09:50 -0600 Subject: [Freeswitch-users] help! DTMFs disappearing in mod_conference In-Reply-To: References: Message-ID: Yes this issue has already been fixed in SVN Trunk. I recommend you update. /b On Feb 15, 2009, at 10:29 AM, Dale Trub wrote: > The bug I describe sure looks a lot like: > http://jira.freeswitch.org/browse/FSCORE-266 > > We have a direct Metaswitch-> FS connection, and both machines in > the same LAN/location. > > It's 64-bit CentOS btw. Bug also occurred on a 32-bit CentOS dev > machine. > > On Sat, Feb 14, 2009 at 9:15 PM, Dale Trub wrote: > Hey folks, > > I'm having a very odd issue and I'm wondering if anyone else has > seen this, or if there's a setting to change etc. > > I should mention that if anyone by chance helps THIS WEEKEND, it > could SAVE my butt. We are doing an important demo monday morning > and honestly this stops us in our tracks. > > We are listening for DTMFs from mod_conference and passing that via > the socket on to a separate display layer (in development). > > It works perfectly, but at a certain point in a conference, it seems > the switch stops sensing the DTMFs on most (but not all) lines. > > FYI, we saw this before with FS 1.0 running on a VPS slice and > thought maybe it was somehow related to that box, or that DID > provider. We've now switched to a full server and a different DID > provider, and are getting the exact same behavior. > > Today, here was the deal: > 10 people called in (practice walkthrough of our demo this monday) > all lines: DTMFs displayed - tried them several times > 6= mute/unmute also works (doesn't go through our display layer) > about 30 minutes in, again asked everyone to hit 1 (which again we > pass to display layer) > and now most lines do not pass DTMFs > a couple lines still do pass them > (the "6" which we trap within FS as "mute/unmute" also stops working > on those lines that stopped passing others) > the FS logs STOP reflecting DTMFs from the lines where we don't see > them > so, we know it's FS and not our application > some time passes > keep trying the working ones -- eventually they stop working > one caller (with DTMFs non functional) hangs up and calls back > that caller now does have DTMFs working > we hung up and called back in > this time DTMFs worked ~100 times, and then again stopped > switched logs from INFO to DEBUG > below are some log file entries > > We're on CENT-OS and FS 1.0.2 > > Besides the obvious question ("how do I fix this") > > Non-obvious Questions: > Is there any way to tell if the DID provider is trapping the DTMFs > and sending them out of band, or is sending them in-band? > Is there any reasonably easy way to get in and see/sniff/visualize/ > measure the SIP packets to see what is coming in? > Could this be related to this? http://wiki.freeswitch.org/wiki/RTP_Issues > Any other thoughts on how to debug? > Thanks!! > > -Dale > > Here's the last working DTMF, and then some events I don't know ... > through a place where this definitely wasn't working. > > > 2009-02-14 22:26:03 [DEBUG] switch_rtp.c:1701 > switch_rtp_dequeue_dtmf() RTP RECV > DTMF 5:2000 > 2009-02-14 22:37:06 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state() > Channel sofi > a/external/xxphonenumxx at 172.16.250.4 entering state [received] > 2009-02-14 22:37:06 [DEBUG] sofia.c:2546 sofia_handle_sip_i_state() > Remote SDP: > v=0 > o=FreeSWITCH 8044373728746667485 7321340529655007764 IN IP4 > 172.16.250.4 > s=FreeSWITCH > c=IN IP4 172.16.3.13 > t=0 0 > m=audio 33440 RTP/AVP 0 101 > a=rtpmap:101 telephone-event/8000 > a=ptime:20 > > 2009-02-14 22:37:06 [DEBUG] sofia_glue.c:2447 > sofia_glue_negotiate_sdp() Our exi > sting sdp is still good [PCMU 172.16.3.13:33440], let's keep it. > 2009-02-14 22:37:06 [DEBUG] sofia_glue.c:2473 > sofia_glue_negotiate_sdp() Set 283 > 3 dtmf payload to 101 > 2009-02-14 22:37:06 [DEBUG] sofia_glue.c:1880 > sofia_glue_activate_rtp() Audio pa > rams are unchanged for sofia/external/xxphonenumxx at 172.16.250.4. > 2009-02-14 22:37:06 [DEBUG] sofia.c:2896 sofia_handle_sip_i_state() > Processing R > einvite > 2009-02-14 22:37:06 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state() > Channel sofi > a/external/xxphonenumxx at 172.16.250.4 entering state [completed] > 2009-02-14 22:37:06 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state() > Channel sofi > a/external/xxphonenumxx at 172.16.250.4 entering state [ready] > 2009-02-14 22:38:34 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state() > Channel sofi > a/external/xxphonenumxx2 at 172.16.250.4 entering state [received] > 2009-02-14 22:38:34 [DEBUG] sofia.c:2546 sofia_handle_sip_i_state() > Remote SDP: > v=0 > o=FreeSWITCH 934104982290142318 4836750446264379897 IN IP4 > 172.16.250.4 > s=FreeSWITCH > c=IN IP4 172.16.1.21 > t=0 0 > m=audio 35356 RTP/AVP 0 101 > a=rtpmap:101 telephone-event/8000 > a=ptime:20 > > 2009-02-14 22:38:34 [DEBUG] sofia_glue.c:2447 > sofia_glue_negotiate_sdp() Our exi > sting sdp is still good [PCMU 172.16.1.21:35356], let's keep it. > 2009-02-14 22:38:34 [DEBUG] sofia_glue.c:2473 > sofia_glue_negotiate_sdp() Set 283 > 3 dtmf payload to 101 > 2009-02-14 22:38:34 [DEBUG] sofia_glue.c:1880 > sofia_glue_activate_rtp() Audio pa > rams are unchanged for sofia/external/xxphonenumxx2 at 172.16.250.4. > > 2009-02-14 22:38:34 [DEBUG] sofia_glue.c:1880 > sofia_glue_activate_rtp() Audio pa > rams are unchanged for sofia/external/xxphonenumxx2 at 172.16.250.4. > 2009-02-14 22:38:34 [DEBUG] sofia.c:2896 sofia_handle_sip_i_state() > Processing R > einvite > 2009-02-14 22:38:34 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state() > Channel sofi > a/external/xxphonenumxx2 at 172.16.250.4 entering state [completed] > 2009-02-14 22:38:34 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state() > Channel sofi > a/external/xxphonenumxx2 at 172.16.250.4 entering state [ready] > > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090215/4dafd255/attachment.html From brian at freeswitch.org Sun Feb 15 12:10:23 2009 From: brian at freeswitch.org (Brian West) Date: Sun, 15 Feb 2009 14:10:23 -0600 Subject: [Freeswitch-users] High CPU load after starting In-Reply-To: <13C421883438EB42B9E2C30069FD4AB76AEA2B38A2@crushinator.central.local> References: <13C421883438EB42B9E2C30069FD4AB76AEA2B38A2@crushinator.central.local> Message-ID: <16A96D11-7880-4B03-A729-D50C0538C0EB@freeswitch.org> Since nobody can reproduce it... not sure how we can proceed... have you done a fresh checkout from SVN trunk and tried again? /b On Feb 15, 2009, at 1:16 PM, Public Dump wrote: > So, no ideas left how to fix this problem ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090215/520708ba/attachment.html From nik.middleton at noblesolutions.co.uk Sun Feb 15 14:18:56 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Sun, 15 Feb 2009 22:18:56 -0000 Subject: [Freeswitch-users] Getting current call count Message-ID: Hi guys, I'd like to get the number of calls on the system so that I can manage the load. >From the cli, I've tried the following: Show channels This along with the call detail shows me the correct number of calls Show calls count This delivers a value of zero. I should add that I'm placing an outbound call from a JavaScript. If I originate another call within the script and bridge it with the first, it then shows 1 call It's as if it's only counting calls between two end points Status Shows correct number of sessions, BUT... shows 2/200 (200 is the value set in call setups/sec. I've set maximum calls to 1000, so I'm hoping that this is a typo) Am I missing something here? Is there another way of doing this? Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090215/ddd70464/attachment-0001.html From nik.middleton at noblesolutions.co.uk Sun Feb 15 15:21:17 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Sun, 15 Feb 2009 23:21:17 -0000 Subject: [Freeswitch-users] [newbie] Clean start with asimple configuration In-Reply-To: <2FA0A303-9D53-453A-89A3-517D4B7FBC19@marocon.com> References: <70ad8fdb0902141602w5829a059p3fe9b4159cba499a@mail.gmail.com><13479F65-7CE3-4C41-9845-6D92A647C946@freeswitch.org> <2FA0A303-9D53-453A-89A3-517D4B7FBC19@marocon.com> Message-ID: I'm in the same boat, finding the transition from Asterisk to FS very frustrating. Something I can do in Asterisk in 10 minutes is taking me a day with FS. Do I think it's worth it? Absolutely, but it's incredibly painful at times. What I've done is to create some WIKI pages to help those familiar with Asterisk to understand the nuances of FS. I posted them in the user pages. Hopefully when there are enough contributions we can have a section on the main WIKI entitled 'Asterisk conversion' or something. Asterisk is very forgiving and takes a lot of the pain away from doing simple tasks. FS on the other hand is less forgiving, but you have more control. Being a control freak I like that. I kind of liken Asterisk to the early versions of basic. Each command had a line number. You could be up and running basic apps in a few hours. Then jump to C, and you'll spend ages doing a simple task, but once you've mastered it, you'll never go back. (7 Day FS veteran :-)) Regards, ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Tim Mattison Sent: 15 February 2009 02:20 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] [newbie] Clean start with asimple configuration I think the crux of the matter is "I can't get Freeswitch to do _just_ this" (emphasis added). Users that are relatively new to FreeSWITCH but Asterisk veterans want to know how to build a simple, lean configuration. I'm trying to figure it out myself right now. If simplified configs are only going to be posted after 1.0.3 what's the best place to post my findings and configs? Is there an explicit, sanctioned place for this kind of thing on the Wiki already? Tim On Feb 14, 2009, at 9:10 PM, Brian West wrote: FreeSWITCH default config already has this feature. Register two phones... 1000 and 1001 both with password of 1234 then you can call between them. That will work exactly as you want out of the box. Expect more simplified configs to show up after 1.0.3. /b On Feb 14, 2009, at 6:02 PM, xs wrote: I want that also with Freeswitch. It is a good starting point but i am fiddeling with it for a couple of days now, read the docs but i can't get Freeswitch to do just this. So just calling between a few local sip phones with transcoding and _everything_ else disabled. _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090215/0b3f19cb/attachment.html From krice at freeswitch.org Sun Feb 15 15:40:19 2009 From: krice at freeswitch.org (Ken Rice) Date: Sun, 15 Feb 2009 17:40:19 -0600 Subject: [Freeswitch-users] Getting current call count In-Reply-To: Message-ID: 2/200 is number of sessions per second/MAX number of sessions per second... If you do a ?fsctl max_sessions? that will show you max number of sessions Status outputs something like this API CALL [status()] output: UP 0 years, 0 days, 17 hours, 18 minutes, 36 seconds, 913 milliseconds, 565 microseconds 1574 session(s) since startup 4 session(s) 0/30 Uptime Total Number of session(s) since startup Active session(s) sessions per sec/max sessions per sec From: Nik Middleton Reply-To: Date: Sun, 15 Feb 2009 22:18:56 -0000 To: Subject: [Freeswitch-users] Getting current call count Hi guys, I?d like to get the number of calls on the system so that I can manage the load. >From the cli, I?ve tried the following: Show channels This along with the call detail shows me the correct number of calls Show calls count This delivers a value of zero. I should add that I?m placing an outbound call from a JavaScript. If I originate another call within the script and bridge it with the first, it then shows 1 call It?s as if it?s only counting calls between two end points Status Shows correct number of sessions, BUT? shows 2/200 (200 is the value set in call setups/sec. I?ve set maximum calls to 1000, so I?m hoping that this is a typo) Am I missing something here? Is there another way of doing this? Regards, _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090215/969f8e75/attachment.html From anthony.minessale at gmail.com Sun Feb 15 16:43:31 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sun, 15 Feb 2009 18:43:31 -0600 Subject: [Freeswitch-users] FS SIP audio quality? In-Reply-To: <49977FFD.1020002@versafon.com> References: <49977FFD.1020002@versafon.com> Message-ID: <191c3a030902151643n62c2b1a0k2c341eb79c1393ea@mail.gmail.com> it's digital audio. The only thing doing sampling and reconstruction of the signal are the phones. The audio files have been captured long ago from the microphone in the studio. We do nothing to alter the volume of the audio signal or manipulate it in any way unless you are transcoding between sample rates or codecs which you are not because you mentioned it was PCMU. If you are making a call from x-lite to a linksys using just PCMU there is no transcoding going on at all and it would not be any more or less loud than if the devices were exchanging media directly because all we would be doing is passing the digital packets across. I believe you are somehow mistaken in your explanation. There is a good chance that your x-lite has the gain set lower when you are testing FS since that's the only device in your whole scenario that is capable of adjusting the gain. If you wish, please get a complete packet capture of a completed call in both situations. On Sat, Feb 14, 2009 at 8:37 PM, Paul D. wrote: > Comparing FS 1.0.3 audio quality vs * 1.4.2, simple Sip-to-Sip call, or > call to VM prompt, or call via gateway to PSTN - FS audio volume level > (should I say gain?) seems noticeably lower than on *, this may be a > reason that FS audio seems to be subpar, more noise less clear. Test > calls made using PCMU codec from X-Lite and Linksys 2002. > Is there anything can be tweaked in FS to correct that? Same issue was > with 1.0.2. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090215/413477e6/attachment.html From anthony.minessale at gmail.com Sun Feb 15 16:51:59 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sun, 15 Feb 2009 18:51:59 -0600 Subject: [Freeswitch-users] [newbie] Clean start with asimple configuration In-Reply-To: References: <70ad8fdb0902141602w5829a059p3fe9b4159cba499a@mail.gmail.com> <13479F65-7CE3-4C41-9845-6D92A647C946@freeswitch.org> <2FA0A303-9D53-453A-89A3-517D4B7FBC19@marocon.com> Message-ID: <191c3a030902151651k4108d9a0he12fa555ac6ceab1@mail.gmail.com> We do have mod_dialplan_asterisk if you miss that. You do realize you did not do it in 10 minutes with asterisk when you first started using it only once you learned how to work it. Your problem is more with the paradigm shift than the complexity. This is common so it's good that you are adding some wiki pages from you perspective. I deputize you in charge of training all new users with an asterisk background. It's harder for us to explain it, however, We did all come from asterisk too ;) There are 2 distinct camps of new users. *) Those who try to make it work like asterisk and take month to learn how the rain in Spain falls gently on the plain. *) Those who never heard of asterisk before and understand everything instantly. We need more people to step up and contribute more to the documentation from each perspective , keep up the good work! On Sun, Feb 15, 2009 at 5:21 PM, Nik Middleton < nik.middleton at noblesolutions.co.uk> wrote: > I'm in the same boat, finding the transition from Asterisk to FS very > frustrating. Something I can do in Asterisk in 10 minutes is taking me a > day with FS. > > > > Do I think it's worth it? Absolutely, but it's incredibly painful at > times. > > > > What I've done is to create some WIKI pages to help those familiar with > Asterisk to understand the nuances of FS. I posted them in the user pages. > Hopefully when there are enough contributions we can have a section on the > main WIKI entitled 'Asterisk conversion' or something. > > > > Asterisk is very forgiving and takes a lot of the pain away from doing > simple tasks. FS on the other hand is less forgiving, but you have more > control. Being a control freak I like that. > > > > I kind of liken Asterisk to the early versions of basic. Each command had > a line number. You could be up and running basic apps in a few hours. Then > jump to C, and you'll spend ages doing a simple task, but once you've > mastered it, you'll never go back. > > > > (7 Day FS veteran J) > > > > Regards, > > > ------------------------------ > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Tim Mattison > *Sent:* 15 February 2009 02:20 > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] [newbie] Clean start with asimple > configuration > > > > I think the crux of the matter is "I can't get Freeswitch to do _just_ > this" (emphasis added). Users that are relatively new to FreeSWITCH but > Asterisk veterans want to know how to build a simple, lean configuration. > I'm trying to figure it out myself right now. > > > > If simplified configs are only going to be posted after 1.0.3 what's the > best place to post my findings and configs? Is there an explicit, > sanctioned place for this kind of thing on the Wiki already? > > > > Tim > > > > On Feb 14, 2009, at 9:10 PM, Brian West wrote: > > > > FreeSWITCH default config already has this feature. Register two > phones... 1000 and 1001 both with password of 1234 then you can call > between them. That will work exactly as you want out of the box. > Expect more simplified configs to show up after 1.0.3. > > /b > > On Feb 14, 2009, at 6:02 PM, xs wrote: > > > I want that also with Freeswitch. It is a good starting point but i > > am fiddeling with it for a couple of days now, read the docs but i > > can't get Freeswitch to do just this. So just calling between a few > > local sip phones with transcoding and _everything_ else disabled. > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090215/642e0e5d/attachment-0001.html From anthony.minessale at gmail.com Sun Feb 15 17:14:49 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sun, 15 Feb 2009 19:14:49 -0600 Subject: [Freeswitch-users] Problems with Originate In-Reply-To: References: <1234524794.4431.56.camel@gathern.lan> <1234525724.4431.59.camel@gathern.lan> <191c3a030902130554w1dfe6b1dw9e5525cf6a210dc4@mail.gmail.com> <191c3a030902140919w806923ex2dc44c2734010251@mail.gmail.com> <191c3a030902141445q7ffb1febm242c6785a9afef7c@mail.gmail.com> Message-ID: <191c3a030902151714u4be8e9dakd01547b5e49b9dec@mail.gmail.com> btw on the failed session you can still access the attribute session.cause and session.causecode to see why it failed to setup On Sun, Feb 15, 2009 at 5:39 AM, Nik Middleton < nik.middleton at noblesolutions.co.uk> wrote: > You are indeed correct. I still had my asterisk hat on, and was > expecting a hang-up event to be fired with the call outcome. I was > explicitly testing for call failure. I've not modified the code to test the > result of the originate and it works as expected. > > > > I will add some words to the wiki explaining this for those converting from > Asterisk > > > > Regards, > > > ------------------------------ > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony > Minessale > *Sent:* 14 February 2009 22:45 > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Problems with Originate > > > > > What do you suggest is not working? > > the call failed and it *did not* run the code inside if > (new_session.ready()) > The call was never established therefore it would not run the hangup hook > either. > In order to trigger the hangup hook the session would need to exist. If > the session could not originate the new_session obj > is an empty shell with no actual session inside. > > var new_session = new Session(, ); > if (!new_session.ready()) { > // the call never was established. > } > > I gave you real code to try in my last email that you completely > ignored..... > > > and finally you are declaring the function wrong > > declare it at the top. > > function on_hangup(hup_session, how) > { > ..... > } > > I sense you seem to think things are going to magically transform into > however you are thinking they should work > instead of you perhaps learning how they actually work. > > > On Sat, Feb 14, 2009 at 1:47 PM, Nik Middleton < > nik.middleton at noblesolutions.co.uk> wrote: > > Nope, > > > > Still not working. Here's my little test javascript > > > > var new_session = new Session('{ignore_early_media=true,}sofia/internal/ > 1001 at 192.168.3.206'); > > > > //set the on_hangup function to be called when this session is > hungup > > > new_session.setHangupHook(on_hangup,"hup"); > > > > var on_hangup = function(hup_session, how) > { > > > console_log("err","In hangup > section\n"); > > //exit here would end the script so you > could cleanup and just be > done > > > > exit(); > > > } > > > > > > if (new_session.ready()) { > > new_session.answer( ); > > new_session.sleep(1500); > > new_session.streamFile("female2.wav"); > > } > > > > And this is the output > > > > [NOTICE] sofia.c:3090 sofia_handle_sip_i_state() Hangup sofia/internal/ > sip:1001 at 192.168.0.29:5060 [CS_CONSUME_MEDIA] [USER_BUSY] > > [ERR] switch_ivr_originate.c:1116 switch_ivr_originate() Cannot create > outgoing channel of type [user] cause: [USER_BUSY] > > [INFO] mod_dptools.c:1909 audio_bridge_function() Originate Failed. Cause: > USER_BUSY > > [NOTICE] switch_core_session.c:957 switch_core_session_thread() Session 95 > (sofia/internal/sip:1001 at 192.168.0.29:5060) Ended > > [NOTICE] switch_core_session.c:959 switch_core_session_thread() Close > Channel sofia/internal/sip:1001 at 192.168.0.29:5060 [CS_HANGUP] > > [NOTICE] mod_dptools.c:600 answer_function() Channel [sofia/internal/ > 0000000000 at 192.168.3.206] has been answered > > > ------------------------------ > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony > Minessale > *Sent:* 14 February 2009 17:20 > > > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Problems with Originate > > > > are you running this as a dialplan app? > session is a reserved variable name for the session you executed the app > on. > > are you using an alternate name for your new session like my_session > etc....? > > this works for me, try it yourself. > > var my_session = new Session("sofia/external/ > 7003 at conference.freeswitch.org"); > consoleLog("err", "ready: " + my_session.ready() + "\n"); > > > On Sat, Feb 14, 2009 at 7:17 AM, Nik Middleton < > nik.middleton at noblesolutions.co.uk> wrote: > > Understood. > > > > However, using the second method, how can I trap on call failure? > > > > If I originate a call and the user is busy, the console reports this fact, > but then the script continues to execute > > > > if (session.ready()) { > > console_log("notice","Session result=[" + > session.cause + "] \n"); > > if (session.cause == "USER_BUSY") { > > Disposition = > "BUSY"; > > session.Hangup(); > > } > > In this case session.cause reports 'NONE' and what's surprising is that > even though the call failed (busy) session.ready returns a true value. > ------------------------------ > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony > Minessale > *Sent:* 13 February 2009 13:55 > > > *To:* freeswitch-users at lists.freeswitch.org > > *Subject:* Re: [Freeswitch-users] Problems with Originate > > > > The first way is deprecated and will be removed. > The 2nd way is the correct way. > > On Fri, Feb 13, 2009 at 5:48 AM, Alexandru Nedelcu > wrote: > > On Fri, 2009-02-13 at 13:33 +0200, Alexandru Nedelcu wrote: > > The problem with this setup is that origination_caller_id_number doesn't > > work from inside the JS file (when calling session.originate). > > I just discovered something interesting. > > When originating the call like this ... > session = new Session("") > instead of this ... > session = new Session(); session.originate("") > > ... then it works. Is this some kind of bug, or what's the difference > here? > > > Thanks, > > -- > Alexandru Nedelcu > Software Developer, Sinapticode > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090215/837fa4a5/attachment-0001.html From anthony.minessale at gmail.com Sun Feb 15 17:20:31 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sun, 15 Feb 2009 19:20:31 -0600 Subject: [Freeswitch-users] Adding a third person to an ongoing conversation by dialing DTMF In-Reply-To: References: Message-ID: <191c3a030902151720u79c06acw671d3e9d86ab5954@mail.gmail.com> bind-meta-app + (three_way app or transfer app with -both to a conference) On Sun, Feb 15, 2009 at 1:47 AM, Shahal Hazan wrote: > I would to add a third person to an ongoing conversation (between two SIP > callers for example) by dialing a DTMF. > > The DTMF can be as simple as 1 or 2 or 3 to add a predefined person: > person1 or person2 or person3 respectively. > > What is the best way to accomplish that? > > 1) Receiving the DTMF: > > After I added: > > ** > > I wasn't able to receive the DTMF on a non IVR call (I only got the IVR > examples to work) > > Can I capture the DTMF in JS? > > > > 2) Adding the third person: > > a. Creating a conference on the fly and adding the third person? > > b. Bridging the third person? > > c. Using an API to "originate" a new call added to the current call? > > d. Creating a predefined group and adding members from that group per > caller's choice? > > > > Thanks, > > Shahal Hazan > > > This mail was sent via Mail-SeCure System. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090215/989d4127/attachment.html From anthony.minessale at gmail.com Sun Feb 15 17:21:19 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sun, 15 Feb 2009 19:21:19 -0600 Subject: [Freeswitch-users] Build and Sofia issues with recent svn trunk revisions In-Reply-To: <20090215070216.GA20246@jdc.jasonjgw.net> References: <20090215070216.GA20246@jdc.jasonjgw.net> Message-ID: <191c3a030902151721pae57190g1ede358ee717962@mail.gmail.com> [ERR] sofia.c:739 sofia_profile_thread_run() Error Creating SIP UA for profile: internal 9/10 times means something is already running and listening to the sip port. On Sun, Feb 15, 2009 at 1:02 AM, Jason White wrote: > Following the resampling discussion, I tried upgrading to revision 12018. > which compiled cleanly, but then failed to load my internal SIP profile: > > [ERR] sofia.c:739 sofia_profile_thread_run() Error Creating SIP UA for > profile: internal > > The same configuration works under revision 11488, to which I've > temporarily > downgraded. Either something has changed in FreeSWITCH that requires > modifications to my SIP configuration, or this is a regression. > > I decided to try rev. 12027, which, on the same machine (Debian Sid) fails > to > build with the following error: > x86_64-linux-gnu-gcc -c -ggdb -I. ./dftables.c > In file included from ./dftables.c:50: > ./pcre_internal.h:239:2: error: #error LINK_SIZE must be either 2, 3, or 4 > make[2]: *** [dftables.o] Error 1 > > I suspect that recent changes to the build system are responsible for the > latter. > > I'm sure these are minor matters that will be sorted out soon. Thanks once > again to the developers for a great project! > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090215/784db508/attachment.html From jason at jasonjgw.net Sun Feb 15 17:44:38 2009 From: jason at jasonjgw.net (Jason White) Date: Mon, 16 Feb 2009 12:44:38 +1100 Subject: [Freeswitch-users] [newbie] Clean start with asimple configuration In-Reply-To: <191c3a030902151651k4108d9a0he12fa555ac6ceab1@mail.gmail.com> References: <70ad8fdb0902141602w5829a059p3fe9b4159cba499a@mail.gmail.com> <13479F65-7CE3-4C41-9845-6D92A647C946@freeswitch.org> <2FA0A303-9D53-453A-89A3-517D4B7FBC19@marocon.com> <191c3a030902151651k4108d9a0he12fa555ac6ceab1@mail.gmail.com> Message-ID: <20090216014438.GA4590@jdc.jasonjgw.net> Anthony Minessale wrote: > > There are 2 distinct camps of new users. > > *) Those who try to make it work like asterisk and take month to learn how > the rain in Spain falls gently on the plain. > *) Those who never heard of asterisk before and understand everything > instantly. I must be one of those rare users who stand in the middle: I had used Asterisk before, but I didn't try to apply my Asterisk knowledge to learning FreeSWITCH, other than to make sure that all of the desirable features of my Asterisk configuration eventually had counterparts in my FreeSWITCH configuration. From pauld at versafon.com Sun Feb 15 18:04:14 2009 From: pauld at versafon.com (Paul D.) Date: Sun, 15 Feb 2009 21:04:14 -0500 Subject: [Freeswitch-users] FS SIP audio quality? In-Reply-To: <191c3a030902151643n62c2b1a0k2c341eb79c1393ea@mail.gmail.com> References: <49977FFD.1020002@versafon.com> <191c3a030902151643n62c2b1a0k2c341eb79c1393ea@mail.gmail.com> Message-ID: <4998C99E.9060706@versafon.com> Well, I tried several call scenarios: 1. Call from X-Lite or Linksys to VM. 2. Call from X-Lite or Linksys to a conference. 3. Call from X-Lite or Linksys to a PSTN number via Gafachi and CallWithUs. I have now * 1.6.5 and FS 1.0.3RC1 installed on the same enterprise grade Intel server. So just comparing audio in the call scenarios above * somehow does noticeably better job, sounds clearer and volume is at the right level. I am not changing any phone settings of course when switching between * and FS. I am not biased towards FS or * at the moment, though FS seems to have a better designed configuration options and community. Just wanted to share my experience, and hear some opinions. Unfortunately I cannot spend whole amount of time investigating this case now, capturing packets etc., but I will try to do that once I have time. Meanwhile I will have to stick to * for prod. Anthony Minessale wrote: > it's digital audio. The only thing doing sampling and reconstruction > of the signal are the phones. The audio files have been captured long > ago from the microphone in the studio. > We do nothing to alter the volume of the audio signal or manipulate it > in any way unless you are transcoding between sample rates or codecs > which you are not because you mentioned it was PCMU. > > If you are making a call from x-lite to a linksys using just PCMU > there is no transcoding going on at all and it would not be any more > or less loud than if the > devices were exchanging media directly because all we would be doing > is passing the digital packets across. > > I believe you are somehow mistaken in your explanation. There is a > good chance that your x-lite has the gain set lower when you are > testing FS since that's the only device > in your whole scenario that is capable of adjusting the gain. > > If you wish, please get a complete packet capture of a completed call > in both situations. > > > On Sat, Feb 14, 2009 at 8:37 PM, Paul D. > wrote: > > Comparing FS 1.0.3 audio quality vs * 1.4.2, simple Sip-to-Sip > call, or > call to VM prompt, or call via gateway to PSTN - FS audio volume > level > (should I say gain?) seems noticeably lower than on *, this may be a > reason that FS audio seems to be subpar, more noise less clear. Test > calls made using PCMU codec from X-Lite and Linksys 2002. > Is there anything can be tweaked in FS to correct that? Same issue was > with 1.0.2. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From jason at jasonjgw.net Sun Feb 15 18:09:09 2009 From: jason at jasonjgw.net (Jason White) Date: Mon, 16 Feb 2009 13:09:09 +1100 Subject: [Freeswitch-users] Location of config files - Debian packaging issue? Message-ID: <20090216020909.GA5212@jdc.jasonjgw.net> I've found the cause of my problem: As of the 12018 build, FreeSWITCH is searching for its configuration files in /etc/freeswitch rather than /opt/freeswitch/conf. I am using Debian packages built from a copy of the repository. If this is a deliberate change, it's fine, but if it isn't deliberate then something is amiss with the packaging. From brian at freeswitch.org Sun Feb 15 18:11:05 2009 From: brian at freeswitch.org (Brian West) Date: Sun, 15 Feb 2009 20:11:05 -0600 Subject: [Freeswitch-users] FS SIP audio quality? In-Reply-To: <4998C99E.9060706@versafon.com> References: <49977FFD.1020002@versafon.com> <191c3a030902151643n62c2b1a0k2c341eb79c1393ea@mail.gmail.com> <4998C99E.9060706@versafon.com> Message-ID: I'm not able to reproduce this issue.. can you verify the codecs are what you think they are on both Asterisk and FreeSWITCH. /b On Feb 15, 2009, at 8:04 PM, Paul D. wrote: > Well, I tried several call scenarios: > 1. Call from X-Lite or Linksys to VM. > 2. Call from X-Lite or Linksys to a conference. > 3. Call from X-Lite or Linksys to a PSTN number via Gafachi and > CallWithUs. > > I have now * 1.6.5 and FS 1.0.3RC1 installed on the same enterprise > grade Intel server. So just comparing audio in the call scenarios > above > * somehow does noticeably better job, sounds clearer and volume is at > the right level. I am not changing any phone settings of course when > switching between * and FS. > I am not biased towards FS or * at the moment, though FS seems to > have a > better designed configuration options and community. > Just wanted to share my experience, and hear some opinions. > Unfortunately I cannot spend whole amount of time investigating this > case now, capturing packets etc., but I will try to do that once I > have > time. Meanwhile I will have to stick to * for prod. From brian at freeswitch.org Sun Feb 15 18:12:21 2009 From: brian at freeswitch.org (Brian West) Date: Sun, 15 Feb 2009 20:12:21 -0600 Subject: [Freeswitch-users] FS SIP audio quality? In-Reply-To: <4998C99E.9060706@versafon.com> References: <49977FFD.1020002@versafon.com> <191c3a030902151643n62c2b1a0k2c341eb79c1393ea@mail.gmail.com> <4998C99E.9060706@versafon.com> Message-ID: <4DA657EE-2150-435D-BD15-3A5605A0A10F@freeswitch.org> Also you didn't try SVN Trunk? /b On Feb 15, 2009, at 8:04 PM, Paul D. wrote: > I have now * 1.6.5 and FS 1.0.3RC1 From brian at freeswitch.org Sun Feb 15 18:13:57 2009 From: brian at freeswitch.org (Brian West) Date: Sun, 15 Feb 2009 20:13:57 -0600 Subject: [Freeswitch-users] Location of config files - Debian packaging issue? In-Reply-To: <20090216020909.GA5212@jdc.jasonjgw.net> References: <20090216020909.GA5212@jdc.jasonjgw.net> Message-ID: <083B5AE6-D74E-416F-930E-20CEC06DAB0F@freeswitch.org> I think this is in the process of getting corrected to beh the "debian" way. Please join on IRC and interact with everyone related to this. /b On Feb 15, 2009, at 8:09 PM, Jason White wrote: > I've found the cause of my problem: > As of the 12018 build, FreeSWITCH is searching for its configuration > files in > /etc/freeswitch rather than /opt/freeswitch/conf. I am using Debian > packages > built from a copy of the repository. > > If this is a deliberate change, it's fine, but if it isn't > deliberate then > something is amiss with the packaging. From krice at freeswitch.org Sun Feb 15 18:25:10 2009 From: krice at freeswitch.org (Ken Rice) Date: Sun, 15 Feb 2009 20:25:10 -0600 Subject: [Freeswitch-users] FS SIP audio quality? In-Reply-To: <4998C99E.9060706@versafon.com> Message-ID: Paul, If you are truly having a problem please do get us a full packet trace including the RTP... As one of the largest FS users, I can tell you we have not seen this issue and we interconnect with dozens of different endpoint manufacturers using FreeSWITCH. (I run tollfreegateway.com an Open SIP to North American Tollfree TDM termination gateway) If this problem was wide spread I would suspect that users of several ITSPs would be complaining and their ITSPs would be be complaining to me. Now that being said, you're post really smells of a troll. If it is meant as an honest problem please do get us the trace and we'll be more than happy to look at it. Also, as was stated earlier if you are running 1.0.3RC1 then you might see a re-sampling problem in a trans-coding scenario, this has been resolved and you were advised to run trunk to get this fix. As far as your comment on spending too much time to investigate this, all we have asked for is a simple packet trace... This is something that can be done in 5 minutes K > From: "Paul D." > Reply-To: > Date: Sun, 15 Feb 2009 21:04:14 -0500 > To: > Subject: Re: [Freeswitch-users] FS SIP audio quality? > > Well, I tried several call scenarios: > 1. Call from X-Lite or Linksys to VM. > 2. Call from X-Lite or Linksys to a conference. > 3. Call from X-Lite or Linksys to a PSTN number via Gafachi and CallWithUs. > > I have now * 1.6.5 and FS 1.0.3RC1 installed on the same enterprise > grade Intel server. So just comparing audio in the call scenarios above > * somehow does noticeably better job, sounds clearer and volume is at > the right level. I am not changing any phone settings of course when > switching between * and FS. > I am not biased towards FS or * at the moment, though FS seems to have a > better designed configuration options and community. > Just wanted to share my experience, and hear some opinions. > Unfortunately I cannot spend whole amount of time investigating this > case now, capturing packets etc., but I will try to do that once I have > time. Meanwhile I will have to stick to * for prod. > > > Anthony Minessale wrote: >> it's digital audio. The only thing doing sampling and reconstruction >> of the signal are the phones. The audio files have been captured long >> ago from the microphone in the studio. >> We do nothing to alter the volume of the audio signal or manipulate it >> in any way unless you are transcoding between sample rates or codecs >> which you are not because you mentioned it was PCMU. >> >> If you are making a call from x-lite to a linksys using just PCMU >> there is no transcoding going on at all and it would not be any more >> or less loud than if the >> devices were exchanging media directly because all we would be doing >> is passing the digital packets across. >> >> I believe you are somehow mistaken in your explanation. There is a >> good chance that your x-lite has the gain set lower when you are >> testing FS since that's the only device >> in your whole scenario that is capable of adjusting the gain. >> >> If you wish, please get a complete packet capture of a completed call >> in both situations. >> >> >> On Sat, Feb 14, 2009 at 8:37 PM, Paul D. > > wrote: >> >> Comparing FS 1.0.3 audio quality vs * 1.4.2, simple Sip-to-Sip >> call, or >> call to VM prompt, or call via gateway to PSTN - FS audio volume >> level >> (should I say gain?) seems noticeably lower than on *, this may be a >> reason that FS audio seems to be subpar, more noise less clear. Test >> calls made using PCMU codec from X-Lite and Linksys 2002. >> Is there anything can be tweaked in FS to correct that? Same issue was >> with 1.0.2. >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> >> iax:guest at conference.freeswitch.org/888 >> >> googletalk:conf+888 at conference.freeswitch.org >> >> pstn:213-799-1400 >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From anthony.minessale at gmail.com Sun Feb 15 18:43:25 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sun, 15 Feb 2009 20:43:25 -0600 Subject: [Freeswitch-users] FS SIP audio quality? In-Reply-To: <191c3a030902151842n30590accrd6c7df69e546fb49@mail.gmail.com> References: <49977FFD.1020002@versafon.com> <191c3a030902151643n62c2b1a0k2c341eb79c1393ea@mail.gmail.com> <4998C99E.9060706@versafon.com> <191c3a030902151842n30590accrd6c7df69e546fb49@mail.gmail.com> Message-ID: <191c3a030902151843p3fdd1903x990ae6cc991724b2@mail.gmail.com> The typing it takes to start a pcap of each call and email them is less than you have typed thusfar. Please just take the captures and send them to us to examine. That's all. If you have a real issue we would like to address it. On Feb 15, 2009 8:06 PM, "Paul D." wrote: Well, I tried several call scenarios: 1. Call from X-Lite or Linksys to VM. 2. Call from X-Lite or Linksys to a conference. 3. Call from X-Lite or Linksys to a PSTN number via Gafachi and CallWithUs. I have now * 1.6.5 and FS 1.0.3RC1 installed on the same enterprise grade Intel server. So just comparing audio in the call scenarios above * somehow does noticeably better job, sounds clearer and volume is at the right level. I am not changing any phone settings of course when switching between * and FS. I am not biased towards FS or * at the moment, though FS seems to have a better designed configuration options and community. Just wanted to share my experience, and hear some opinions. Unfortunately I cannot spend whole amount of time investigating this case now, capturing packets etc., but I will try to do that once I have time. Meanwhile I will have to stick to * for prod. Anthony Minessale wrote: > it's digital audio. The only thing doing sampling and reconstruction ... > > wrote: > > Comparing FS 1.0.3 audio quality vs * 1.4.2, simple Si... > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.f... > > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > > pstn:213-799-1400 > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-use... -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090215/81a80269/attachment.html From jaybinks at gmail.com Sun Feb 15 18:51:09 2009 From: jaybinks at gmail.com (jay binks) Date: Mon, 16 Feb 2009 12:51:09 +1000 Subject: [Freeswitch-users] FS SIP audio quality? In-Reply-To: <4998C99E.9060706@versafon.com> References: <49977FFD.1020002@versafon.com> <191c3a030902151643n62c2b1a0k2c341eb79c1393ea@mail.gmail.com> <4998C99E.9060706@versafon.com> Message-ID: another thing to try here... is to put FS in RTP proxy and bypass mode. http://wiki.freeswitch.org/wiki/Bypass_Media it would be interesting to see if your still experiencing this problem in either of those 2 modes. Jay On Mon, Feb 16, 2009 at 12:04 PM, Paul D. wrote: > Well, I tried several call scenarios: > 1. Call from X-Lite or Linksys to VM. > 2. Call from X-Lite or Linksys to a conference. > 3. Call from X-Lite or Linksys to a PSTN number via Gafachi and CallWithUs. > > I have now * 1.6.5 and FS 1.0.3RC1 installed on the same enterprise > grade Intel server. So just comparing audio in the call scenarios above > * somehow does noticeably better job, sounds clearer and volume is at > the right level. I am not changing any phone settings of course when > switching between * and FS. > I am not biased towards FS or * at the moment, though FS seems to have a > better designed configuration options and community. > Just wanted to share my experience, and hear some opinions. > Unfortunately I cannot spend whole amount of time investigating this > case now, capturing packets etc., but I will try to do that once I have > time. Meanwhile I will have to stick to * for prod. > > > Anthony Minessale wrote: > > it's digital audio. The only thing doing sampling and reconstruction > > of the signal are the phones. The audio files have been captured long > > ago from the microphone in the studio. > > We do nothing to alter the volume of the audio signal or manipulate it > > in any way unless you are transcoding between sample rates or codecs > > which you are not because you mentioned it was PCMU. > > > > If you are making a call from x-lite to a linksys using just PCMU > > there is no transcoding going on at all and it would not be any more > > or less loud than if the > > devices were exchanging media directly because all we would be doing > > is passing the digital packets across. > > > > I believe you are somehow mistaken in your explanation. There is a > > good chance that your x-lite has the gain set lower when you are > > testing FS since that's the only device > > in your whole scenario that is capable of adjusting the gain. > > > > If you wish, please get a complete packet capture of a completed call > > in both situations. > > > > > > On Sat, Feb 14, 2009 at 8:37 PM, Paul D. > > wrote: > > > > Comparing FS 1.0.3 audio quality vs * 1.4.2, simple Sip-to-Sip > > call, or > > call to VM prompt, or call via gateway to PSTN - FS audio volume > > level > > (should I say gain?) seems noticeably lower than on *, this may be a > > reason that FS audio seems to be subpar, more noise less clear. Test > > calls made using PCMU codec from X-Lite and Linksys 2002. > > Is there anything can be tweaked in FS to correct that? Same issue > was > > with 1.0.2. > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com > > > > > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > > > > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > > > > > iax:guest at conference.freeswitch.org/888 > > > > googletalk:conf+888 at conference.freeswitch.org > > > > > > pstn:213-799-1400 > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Sincerely Jay -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090216/dd111844/attachment-0001.html From jason at jasonjgw.net Sun Feb 15 19:02:52 2009 From: jason at jasonjgw.net (Jason White) Date: Mon, 16 Feb 2009 14:02:52 +1100 Subject: [Freeswitch-users] Build and Sofia issues with recent svn trunk revisions In-Reply-To: <20090215070216.GA20246@jdc.jasonjgw.net> References: <20090215070216.GA20246@jdc.jasonjgw.net> Message-ID: <20090216030252.GA2229@jdc.jasonjgw.net> Jason White wrote: > I decided to try rev. 12027, which, on the same machine (Debian Sid) fails to > build with the following error: > x86_64-linux-gnu-gcc -c -ggdb -I. ./dftables.c > In file included from ./dftables.c:50: > ./pcre_internal.h:239:2: error: #error LINK_SIZE must be either 2, 3, or 4 > make[2]: *** [dftables.o] Error 1 This error occurred with a freshly exported copy of the sources; there were no pre-existing makefiles or autoconf-generated scripts lying around. From brian at freeswitch.org Sun Feb 15 19:14:30 2009 From: brian at freeswitch.org (Brian West) Date: Sun, 15 Feb 2009 21:14:30 -0600 Subject: [Freeswitch-users] Build and Sofia issues with recent svn trunk revisions In-Reply-To: <20090216030252.GA2229@jdc.jasonjgw.net> References: <20090215070216.GA20246@jdc.jasonjgw.net> <20090216030252.GA2229@jdc.jasonjgw.net> Message-ID: Please open a jira http://jira.freeswitch.org /b On Feb 15, 2009, at 9:02 PM, Jason White wrote: > Jason White wrote: > >> I decided to try rev. 12027, which, on the same machine (Debian >> Sid) fails to >> build with the following error: >> x86_64-linux-gnu-gcc -c -ggdb -I. ./dftables.c >> In file included from ./dftables.c:50: >> ./pcre_internal.h:239:2: error: #error LINK_SIZE must be either 2, >> 3, or 4 >> make[2]: *** [dftables.o] Error 1 > > This error occurred with a freshly exported copy of the sources; > there were no > pre-existing makefiles or autoconf-generated scripts lying around. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090215/35d58aa1/attachment.html From jason at jasonjgw.net Sun Feb 15 20:15:25 2009 From: jason at jasonjgw.net (Jason White) Date: Mon, 16 Feb 2009 15:15:25 +1100 Subject: [Freeswitch-users] Build and Sofia issues with recent svn trunk revisions In-Reply-To: References: <20090215070216.GA20246@jdc.jasonjgw.net> <20090216030252.GA2229@jdc.jasonjgw.net> Message-ID: <20090216041525.GA18456@jdc.jasonjgw.net> Brian West wrote: > Please open a jira http://jira.freeswitch.org Has anyone succeeded in doing this with a text-based Web browser such as Lynx or Elinks? Jira keeps complaining that I haven't selected a valid project. There are reasons why I would rather avoid a graphical browser under X at the moment, including X bugs that cause the X server to crash on this machine. From codecomplete at free.fr Sun Feb 15 20:29:45 2009 From: codecomplete at free.fr (Fred) Date: Mon, 16 Feb 2009 05:29:45 +0100 Subject: [Freeswitch-users] Switching from Asterisk to Freeswitch? In-Reply-To: References: Message-ID: <7.0.1.0.2.20090216052901.06a91d28@free.fr> Thanks guys for the input. I'll download FS and give it a shot. From mike at jerris.com Sun Feb 15 22:06:17 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 16 Feb 2009 01:06:17 -0500 Subject: [Freeswitch-users] Location of config files - Debian packaging issue? In-Reply-To: <083B5AE6-D74E-416F-930E-20CEC06DAB0F@freeswitch.org> References: <20090216020909.GA5212@jdc.jasonjgw.net> <083B5AE6-D74E-416F-930E-20CEC06DAB0F@freeswitch.org> Message-ID: <3B7025A0-DAB9-498C-AEF1-2587845C7DE1@jerris.com> This patch was incorrect and was supposed to be reverted. I will correct this error. Mike On Feb 15, 2009, at 9:13 PM, Brian West wrote: > I think this is in the process of getting corrected to beh the > "debian" way. Please join on IRC and interact with everyone related > to this. > > /b > > On Feb 15, 2009, at 8:09 PM, Jason White wrote: > >> I've found the cause of my problem: >> As of the 12018 build, FreeSWITCH is searching for its configuration >> files in >> /etc/freeswitch rather than /opt/freeswitch/conf. I am using Debian >> packages >> built from a copy of the repository. >> >> If this is a deliberate change, it's fine, but if it isn't >> deliberate then >> something is amiss with the packaging. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From woodydickson at gmail.com Mon Feb 16 04:14:43 2009 From: woodydickson at gmail.com (Woody Dickson) Date: Mon, 16 Feb 2009 20:14:43 +0800 Subject: [Freeswitch-users] dynamically add ip to an ACL Message-ID: Hi, Is it possible to dynamically add entries to an ACL without having to go through the xml file? Can it be done via command line or api? Thanks, Woody From leon at scarlet-internet.nl Mon Feb 16 04:49:46 2009 From: leon at scarlet-internet.nl (Leon de Rooij) Date: Mon, 16 Feb 2009 13:49:46 +0100 Subject: [Freeswitch-users] ipauth - directory In-Reply-To: <496E068B.6050404@kinetix.gr> References: <496E0187.8090805@kinetix.gr> <23DEEC73-4DC7-4056-902F-D74DD6644385@freeswitch.org> <496E068B.6050404@kinetix.gr> Message-ID: Hi all, I'd really like to know more about this too. Currently, I have two sip_profiles: - residential (where users can do authenticated registers and invites) - transit (where other users can do un-authenticated invites) Right now, FS is not aware of *who* is accessing the transit profile except for an acl that is set on this profile so unauthorized use is not possible. But what should I do when I want to allow multiple parties (from different IP addresses) to send their invites to the transit profile, and still be able to differentiate between them ? I'd like to set some variables, like an accountcode for example, on the basis of what IP address the INVITE originates from. So, is it possible to not use digest authentication, but still use a dialplan-directory user with IP= field or some such ? thanks a lot & kind regards, Leon de Rooij On Jan 14, 2009, at 4:36 PM, Apostolos Pantsiopoulos wrote: > Yes I know that. But what does the "ip=" setting do? > > Brian West wrote: >> >> cidr= and the domains acl in acl.conf.xml then apply that ACL to the >> sofia profile. >> >> /b >> >> On Jan 14, 2009, at 9:15 AM, Apostolos Pantsiopoulos wrote: >> >> >>> I noticed an "ip=" setting in the brian.xml sample file. >>> The comments state that this is used for ipauth (IP based >>> authentication?) >>> >>> What exactly is this setting. I cannot find anything in the wiki >>> about it. >>> Does it replace the use of the >>> >>> + ACL >>> >>> mechanism for IP authentication? >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > -- > ------------------------------------------- > Apostolos Pantsiopoulos > Kinetix Tele.com R & D > email: regs at kinetix.gr > ------------------------------------------- > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090216/5823e3be/attachment.html From anthony.minessale at gmail.com Mon Feb 16 05:49:12 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 16 Feb 2009 07:49:12 -0600 Subject: [Freeswitch-users] dynamically add ip to an ACL In-Reply-To: References: Message-ID: <191c3a030902160549k4081d4c2s29a3a86bd267febf@mail.gmail.com> no, it's not possible. On Mon, Feb 16, 2009 at 6:14 AM, Woody Dickson wrote: > Hi, > > Is it possible to dynamically add entries to an ACL without having to > go through the xml file? Can it be done via command line or api? > > Thanks, > Woody > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090216/26304763/attachment-0001.html From anthony.minessale at gmail.com Mon Feb 16 06:04:06 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 16 Feb 2009 08:04:06 -0600 Subject: [Freeswitch-users] ipauth - directory In-Reply-To: References: <496E0187.8090805@kinetix.gr> <23DEEC73-4DC7-4056-902F-D74DD6644385@freeswitch.org> <496E068B.6050404@kinetix.gr> Message-ID: <191c3a030902160604r77089a6s5b3b9f3d07914218@mail.gmail.com> you have 3 options. on authenticated users, every tag in his account will be set on each call from that authenticated user. 1) make them register, this sets the variables automatically 2) use the ACL list with cidr= this has the same effect with no auth needed. 3) use some other way to differentiate the user and use the set_user application in the dialplan to inherit that user's variables. On Mon, Feb 16, 2009 at 6:49 AM, Leon de Rooij wrote: > Hi all, > > I'd really like to know more about this too. > > Currently, I have two sip_profiles: > > - residential (where users can do authenticated registers and invites) > - transit (where other users can do un-authenticated invites) > > Right now, FS is not aware of *who* is accessing the transit profile except > for an acl that is set on this profile so unauthorized use is not possible. > > But what should I do when I want to allow multiple parties (from different > IP addresses) to send their invites to the transit profile, and still be > able to differentiate between them ? > > I'd like to set some variables, like an accountcode for example, on the > basis of what IP address the INVITE originates from. > > So, is it possible to not use digest authentication, but still use a > dialplan-directory user with IP= field or some such ? > > thanks a lot & kind regards, > > Leon de Rooij > > > > On Jan 14, 2009, at 4:36 PM, Apostolos Pantsiopoulos wrote: > > Yes I know that. But what does the "ip=" setting do? > > Brian West wrote: > > cidr= and the domains acl in acl.conf.xml then apply that ACL to the > sofia profile. > > /b > > On Jan 14, 2009, at 9:15 AM, Apostolos Pantsiopoulos wrote: > > > > I noticed an "ip=" setting in the brian.xml sample file. > The comments state that this is used for ipauth (IP based > authentication?) > > What exactly is this setting. I cannot find anything in the wiki > about it. > Does it replace the use of the > > + ACL > > mechanism for IP authentication? > > > _______________________________________________ > Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > -- > ------------------------------------------- > Apostolos Pantsiopoulos > Kinetix Tele.com R & D > email: regs at kinetix.gr > ------------------------------------------- > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090216/9bdb02e6/attachment.html From fanatikneo at gmx.de Sun Feb 15 23:59:47 2009 From: fanatikneo at gmx.de (Jan Fricke) Date: Mon, 16 Feb 2009 08:59:47 +0100 Subject: [Freeswitch-users] mod_pa - pa list - call states Message-ID: <49991CF3.5060906@gmx.de> Hello, I'm using Freeswitch (1.0.trunk) as a softphone with mod_pa. My GUI communicates with freeswitch via xml-rpc and fetches calls with "pa list". If somebody is calling, the state of the call is hold. When the call is answered with "pa answer" it is active. If someone calls me while I'm in call the state of the second call is hold. So far so good. But if the first call ends, the second is marked as active although it is still ringing and should be "hold". Is this the intended behavior of "pa list" or did I missunderstand the command? Best regards Jan From anthony.minessale at gmail.com Mon Feb 16 07:10:53 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 16 Feb 2009 09:10:53 -0600 Subject: [Freeswitch-users] mod_pa - pa list - call states In-Reply-To: <49991CF3.5060906@gmx.de> References: <49991CF3.5060906@gmx.de> Message-ID: <191c3a030902160710g464d2210w40fe46e4e126374b@mail.gmail.com> I recommend you use events instead of polling On Mon, Feb 16, 2009 at 1:59 AM, Jan Fricke wrote: > Hello, > I'm using Freeswitch (1.0.trunk) as a softphone with mod_pa. My GUI > communicates with freeswitch via xml-rpc and fetches calls with "pa list". > If somebody is calling, the state of the call is hold. When the call is > answered with "pa answer" it is active. If someone calls me while I'm in > call the state of the second call is hold. So far so good. > But if the first call ends, the second is marked as active although it > is still ringing and should be "hold". > Is this the intended behavior of "pa list" or did I missunderstand the > command? > > Best regards > > Jan > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090216/b09f52fb/attachment.html From brian at freeswitch.org Mon Feb 16 07:05:52 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 16 Feb 2009 09:05:52 -0600 Subject: [Freeswitch-users] mod_pa - pa list - call states In-Reply-To: <49991CF3.5060906@gmx.de> References: <49991CF3.5060906@gmx.de> Message-ID: <366DB45A-6DD3-4309-97A3-498907B5FADE@freeswitch.org> Try "pa switch x", x being the call number. /b On Feb 16, 2009, at 1:59 AM, Jan Fricke wrote: > Hello, > I'm using Freeswitch (1.0.trunk) as a softphone with mod_pa. My GUI > communicates with freeswitch via xml-rpc and fetches calls with "pa > list". > If somebody is calling, the state of the call is hold. When the call > is > answered with "pa answer" it is active. If someone calls me while > I'm in > call the state of the second call is hold. So far so good. > But if the first call ends, the second is marked as active although it > is still ringing and should be "hold". > Is this the intended behavior of "pa list" or did I missunderstand the > command? > > Best regards > > Jan From ajlong at worldlink.net Mon Feb 16 07:18:23 2009 From: ajlong at worldlink.net (Adam Long) Date: Mon, 16 Feb 2009 10:18:23 -0500 Subject: [Freeswitch-users] dynamically add ip to an ACL In-Reply-To: <191c3a030902160549k4081d4c2s29a3a86bd267febf@mail.gmail.com> References: <191c3a030902160549k4081d4c2s29a3a86bd267febf@mail.gmail.com> Message-ID: <014001c99049$cf0edd40$6d2c97c0$@net> Hi Anthony, could he use mod_xml_curl for this to serve up a dynamic acl.conf.xml? Or would reloadacl have to be called somehow? Regards, -Adam From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Monday, February 16, 2009 8:49 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] dynamically add ip to an ACL no, it's not possible. On Mon, Feb 16, 2009 at 6:14 AM, Woody Dickson wrote: Hi, Is it possible to dynamically add entries to an ACL without having to go through the xml file? Can it be done via command line or api? Thanks, Woody _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090216/3786e362/attachment-0001.html From brian at freeswitch.org Mon Feb 16 07:21:20 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 16 Feb 2009 09:21:20 -0600 Subject: [Freeswitch-users] dynamically add ip to an ACL In-Reply-To: <014001c99049$cf0edd40$6d2c97c0$@net> References: <191c3a030902160549k4081d4c2s29a3a86bd267febf@mail.gmail.com> <014001c99049$cf0edd40$6d2c97c0$@net> Message-ID: <6950B56E-25ED-4E99-8495-7966116B2C28@freeswitch.org> Reloadacl would have to be called in either case. /b On Feb 16, 2009, at 9:18 AM, Adam Long wrote: > Hi Anthony, could he use mod_xml_curl for this to serve up a dynamic > acl.conf.xml? > > Or would reloadacl have to be called somehow? > > Regards, > -Adam > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090216/0e5b3402/attachment.html From asannucci at gmail.com Mon Feb 16 07:24:30 2009 From: asannucci at gmail.com (Andrea) Date: Mon, 16 Feb 2009 10:24:30 -0500 Subject: [Freeswitch-users] Perl error when compiling Message-ID: <39B5704908BC4C5BAF066DE0AC8AD32C@quos> Hi, when y try to compiling perl module on freeswitch (from tarball 1.0.3RC1) I have this error: making all mod_perl Creating mod_perl.so... /usr/bin/ld: cannot find -ldb collect2: ld returned 1 exit status Any idea? Thank you - Andrea - From brian at freeswitch.org Mon Feb 16 07:28:19 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 16 Feb 2009 09:28:19 -0600 Subject: [Freeswitch-users] Perl error when compiling In-Reply-To: <39B5704908BC4C5BAF066DE0AC8AD32C@quos> References: <39B5704908BC4C5BAF066DE0AC8AD32C@quos> Message-ID: <8728518C-0AB3-4DE4-BBBB-C85150ACBD31@freeswitch.org> install gdbm-devel and db4-devel. /b On Feb 16, 2009, at 9:24 AM, Andrea wrote: > > making all mod_perl > Creating mod_perl.so... > /usr/bin/ld: cannot find -ldb > collect2: ld returned 1 exit status From hochlehnert at hotmail.com Mon Feb 16 07:28:27 2009 From: hochlehnert at hotmail.com (Klaus Hochlehnert) Date: Mon, 16 Feb 2009 16:28:27 +0100 Subject: [Freeswitch-users] Perl error when compiling In-Reply-To: <39B5704908BC4C5BAF066DE0AC8AD32C@quos> References: <39B5704908BC4C5BAF066DE0AC8AD32C@quos> Message-ID: Hi, install libdb development files, e.g. for Ubuntu/Debian: aptitude install libdb-dev Regards, Klaus> From: asannucci at gmail.com> To: freeswitch-users at lists.freeswitch.org> Date: Mon, 16 Feb 2009 10:24:30 -0500> Subject: [Freeswitch-users] Perl error when compiling> > Hi,> > when y try to compiling perl module on freeswitch (from tarball 1.0.3RC1) I> have this error:> > making all mod_perl> Creating mod_perl.so...> /usr/bin/ld: cannot find -ldb> collect2: ld returned 1 exit status> > Any idea?> > Thank you> > - Andrea -> > > _______________________________________________> Freeswitch-users mailing list> Freeswitch-users at 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://redirect.gimas.net/?n=M0902xHMMobile Nie wieder eine Mail verpassen mit Hotmail f?rs Handy! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090216/eec0c086/attachment.html From mike at jerris.com Mon Feb 16 08:04:59 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 16 Feb 2009 11:04:59 -0500 Subject: [Freeswitch-users] Location of config files - Debian packaging issue? In-Reply-To: <3B7025A0-DAB9-498C-AEF1-2587845C7DE1@jerris.com> References: <20090216020909.GA5212@jdc.jasonjgw.net> <083B5AE6-D74E-416F-930E-20CEC06DAB0F@freeswitch.org> <3B7025A0-DAB9-498C-AEF1-2587845C7DE1@jerris.com> Message-ID: I have reverted this patch, it should be in /opt/freeswitch/conf in trunk now properly. Mike On Feb 16, 2009, at 1:06 AM, Michael Jerris wrote: > This patch was incorrect and was supposed to be reverted. I will > correct this error. > > Mike > > On Feb 15, 2009, at 9:13 PM, Brian West wrote: > >> I think this is in the process of getting corrected to beh the >> "debian" way. Please join on IRC and interact with everyone related >> to this. >> >> /b >> >> On Feb 15, 2009, at 8:09 PM, Jason White wrote: >> >>> I've found the cause of my problem: >>> As of the 12018 build, FreeSWITCH is searching for its configuration >>> files in >>> /etc/freeswitch rather than /opt/freeswitch/conf. I am using Debian >>> packages >>> built from a copy of the repository. >>> >>> If this is a deliberate change, it's fine, but if it isn't >>> deliberate then >>> something is amiss with the packaging. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > From asannucci at gmail.com Mon Feb 16 08:57:27 2009 From: asannucci at gmail.com (Andrea) Date: Mon, 16 Feb 2009 11:57:27 -0500 Subject: [Freeswitch-users] Perl error when compiling References: <39B5704908BC4C5BAF066DE0AC8AD32C@quos> <8728518C-0AB3-4DE4-BBBB-C85150ACBD31@freeswitch.org> Message-ID: <243DB2D99A1149CB8022A4B43C5B1358@quos> Thank you. Now work fine Now when i unload and load mod_java i receive this error: -freeswitch@ unload mod_java 2009-02-16 11:51:36 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'java' 2009-02-16 11:51:36 [CONSOLE] switch_loadable_module.c:1231 do_shutdown() Stopping: mod_java API CALL [unload(mod_java)] output: +OK 2009-02-16 11:51:37 [CONSOLE] switch_loadable_module.c:1244 do_shutdown() mod_java unloaded. freeswitch@ load mod_java API CALL [load(mod_java)] output: -ERR [module load file routine returned an error] freeswitch@ 2009-02-16 11:51:41 [NOTICE] modjava.c:244 mod_java_load() Java Framework Loading... 2009-02-16 11:51:41 [ERR] modjava.c:222 create_java_vm() Error creating Java VM! 2009-02-16 11:51:41 [CRIT] switch_loadable_module.c:839 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_java.so **Module load routine returned an error** This is my java.conf.xml Y have configured the PATH for jdk to point to right directory and i run ./configure with the java path options I hope you understand. I'm new on this :) Regards - Andrea - From msc at freeswitch.org Mon Feb 16 13:14:37 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 16 Feb 2009 13:14:37 -0800 Subject: [Freeswitch-users] Getting current call count In-Reply-To: References: Message-ID: <87f2f3b90902161314y3b8387cayfb7eb3c5c883c736@mail.gmail.com> > Show calls count > > > > This delivers a value of zero. Remember that a "call" is defined as "two or more bridged channels" so you could have lots of channels that aren't bridged and therefore many channels but zero calls. -MC From simon at airg.com Mon Feb 16 14:26:23 2009 From: simon at airg.com (Simon Tang) Date: Mon, 16 Feb 2009 14:26:23 -0800 Subject: [Freeswitch-users] How to keep leg A from hanging up from a bridge when Leg B hangs up (and vice versa)? Message-ID: <872970CF4A55BF42A5337D570860209F018189D8@HPEXCHVS01.exchange.airg> Hello, I'm using event socket outbound and have a framework that does stuff based on the events that come back (this includes my own IVR). What I have now is an IVR system that allows 2 users to bridge to one another at will, and to 'unbridge' at will by catching DTMF events. I have 2 requirements: 1. When one leg hangs up during a bridge, the other leg is presented with the IVR 2. After a bridge, when one leg sends a DTMF tone, both legs will be presented with the IVR and no longer be bridged (they can bridge with other sessions again after this point if they desire) I have done multiple experiments by using netcat and 2 sessions. Here is what I have found: * The hangup_after_bridge variable does nothing for me. I've set it on both legs, but whenever one leg hangs up after a uuid_bridge, the other leg will automatically hang up * I've tried setting "park_after_bridge=true" on both legs, and this works to a certain extent. If one leg hangs up, the other leg will be parked, and I can present that user with my IVR. This meets requirement #1. However, requirement #2 won't be met because: o If I set "park_after_bridge=true" and one leg sends a DTMF tone to signal an unbridge, I will "unbridge" the legs by "parking" both legs and I am able to present them both with an IVR. If they decide to bridge with each other again (by selecting an option in the IVR), I will attempt to do a uuid_bridge and this will FAIL! (both parties do not hear each other.) In the simplest terms, I can't do "uuid_bridge uuidA uuidB", "park", "uuid_bridge uuidA uuidB". * With "park_after_bridge=false" (default), I can do "uuid_bridge uuidA uuidB", "park", "uuid_bridge uuidA uuidB" with no issues, meeting requirement #2. However, this will not meet requirement #1, because when one leg hangs up, it will trigger a hangup on the other. Please help. How can I meet both of my requirements? Thanks. Simon Tang Lead, Server Team Suite 706, 1155 Robson Street Vancouver, B.C. Canada V6E 1B5 T: +1.604.408.2228 Ext. 116 F: +1.866.874.8136 E: simon at airg.com W: www.airg.com The information transmitted is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material communicated under NDA. Any review, retransmission, dissemination or other use of, or taking of any action in reliance upon, this information by persons or entities other than the intended recipient is prohibited. If you received this in error, please contact the sender and delete the material from any computer. -------------- next part -------------- An HTML attachment was scrubbed... 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Name: not available Type: image/jpeg Size: 1010 bytes Desc: image001.jpg Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090216/392db3da/attachment-0001.jpe From msc at freeswitch.org Mon Feb 16 14:32:51 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 16 Feb 2009 14:32:51 -0800 Subject: [Freeswitch-users] [newbie] Clean start with asimple configuration In-Reply-To: <20090216014438.GA4590@jdc.jasonjgw.net> References: <70ad8fdb0902141602w5829a059p3fe9b4159cba499a@mail.gmail.com> <13479F65-7CE3-4C41-9845-6D92A647C946@freeswitch.org> <2FA0A303-9D53-453A-89A3-517D4B7FBC19@marocon.com> <191c3a030902151651k4108d9a0he12fa555ac6ceab1@mail.gmail.com> <20090216014438.GA4590@jdc.jasonjgw.net> Message-ID: <87f2f3b90902161432o66faf2c0pc32d1d906db4c22@mail.gmail.com> > I must be one of those rare users who stand in the middle: I had used Asterisk > before, but I didn't try to apply my Asterisk knowledge to learning > FreeSWITCH, other than to make sure that all of the desirable features of my > Asterisk configuration eventually had counterparts in my FreeSWITCH > configuration. > The key for people coming from Asterisk is to learn the difference between the WHAT and the HOW. It's admittedly difficult, so any who've overcome the challenges and can help other new ones to make the transition are welcome to document their steps. My advice to Asterisk users who want their FS box to do something that their Asterisk box can do is this: separate the WHAT from the HOW. In other words, start at the top (WHAT) and work your way down, not at the bottom (WHAT). The top-down view is like this: Asterisk allows to SIP phones to talk to each other. The bottom-up viewpoint is this: I edited sip.conf, created these PEER entries, then I edited extensions.conf and added exten => foo... See the difference? Start at the top: ask yourself, WHAT does my Ast box do? Now, ask yourself, HOW does FS implement that feature? As much as I like the Rosetta Stone page (because I started it) I don't think that it is the first place to go when you are migrating from Asterisk. (Go there *after* you've played around with FS for a bit.) -MC From jason at jasonjgw.net Mon Feb 16 14:44:52 2009 From: jason at jasonjgw.net (Jason White) Date: Tue, 17 Feb 2009 09:44:52 +1100 Subject: [Freeswitch-users] Location of config files - Debian packaging issue? In-Reply-To: References: <20090216020909.GA5212@jdc.jasonjgw.net> <083B5AE6-D74E-416F-930E-20CEC06DAB0F@freeswitch.org> <3B7025A0-DAB9-498C-AEF1-2587845C7DE1@jerris.com> Message-ID: <20090216224452.GA6177@jdc.jasonjgw.net> Michael Jerris wrote: > I have reverted this patch, it should be in /opt/freeswitch/conf in > trunk now properly. Thank you for the excellent work. I will upgrade to it as soon as this pcre build problem is fixed - it's still failing with latest trunk, by the way. From msc at freeswitch.org Mon Feb 16 14:49:31 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 16 Feb 2009 14:49:31 -0800 Subject: [Freeswitch-users] How to keep leg A from hanging up from a bridge when Leg B hangs up (and vice versa)? In-Reply-To: <872970CF4A55BF42A5337D570860209F018189D8@HPEXCHVS01.exchange.airg> References: <872970CF4A55BF42A5337D570860209F018189D8@HPEXCHVS01.exchange.airg> Message-ID: <87f2f3b90902161449s26a2f63u55454bd4ac9b338b@mail.gmail.com> On Mon, Feb 16, 2009 at 2:26 PM, Simon Tang wrote: > Hello, > > > > I'm using event socket outbound and have a framework that does stuff based > on the events that come back (this includes my own IVR). What I have now is > an IVR system that allows 2 users to bridge to one another at will, and to > 'unbridge' at will by catching DTMF events. Two questions: What version of FS? Preferably latest SVN Are you using the default config, the one created with "make samples"? -MC From simon at airg.com Mon Feb 16 15:04:05 2009 From: simon at airg.com (Simon Tang) Date: Mon, 16 Feb 2009 15:04:05 -0800 Subject: [Freeswitch-users] How to keep leg A from hanging up from abridge when Leg B hangs up (and vice versa)? In-Reply-To: <87f2f3b90902161449s26a2f63u55454bd4ac9b338b@mail.gmail.com> References: <872970CF4A55BF42A5337D570860209F018189D8@HPEXCHVS01.exchange.airg> <87f2f3b90902161449s26a2f63u55454bd4ac9b338b@mail.gmail.com> Message-ID: <872970CF4A55BF42A5337D570860209F018189E9@HPEXCHVS01.exchange.airg> Revision 10626. Default config. I haven't tried latest svn yet, because my framework breaks with it (probably due to some event formatting changes). Can you explain what you mean by "make samples" and what default config? I don't recall ever doing a "make samples". -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: February 16, 2009 2:50 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] How to keep leg A from hanging up from abridge when Leg B hangs up (and vice versa)? On Mon, Feb 16, 2009 at 2:26 PM, Simon Tang wrote: > Hello, > > > > I'm using event socket outbound and have a framework that does stuff based > on the events that come back (this includes my own IVR). What I have now is > an IVR system that allows 2 users to bridge to one another at will, and to > 'unbridge' at will by catching DTMF events. Two questions: What version of FS? Preferably latest SVN Are you using the default config, the one created with "make samples"? -MC _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From msc at freeswitch.org Mon Feb 16 15:30:22 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 16 Feb 2009 15:30:22 -0800 Subject: [Freeswitch-users] How to keep leg A from hanging up from abridge when Leg B hangs up (and vice versa)? In-Reply-To: <872970CF4A55BF42A5337D570860209F018189E9@HPEXCHVS01.exchange.airg> References: <872970CF4A55BF42A5337D570860209F018189D8@HPEXCHVS01.exchange.airg> <87f2f3b90902161449s26a2f63u55454bd4ac9b338b@mail.gmail.com> <872970CF4A55BF42A5337D570860209F018189E9@HPEXCHVS01.exchange.airg> Message-ID: <87f2f3b90902161530u32ab7035v615d4171df4e6407@mail.gmail.com> On Mon, Feb 16, 2009 at 3:04 PM, Simon Tang wrote: > Revision 10626. Default config. I haven't tried latest svn yet, > because my framework breaks with it (probably due to some event > formatting changes). > > Can you explain what you mean by "make samples" and what default config? > I don't recall ever doing a "make samples". > I suppose I should have asked what your platform is! :) Is this Windows, Linux, Unix, or Mac ? -MC From jason at jasonjgw.net Mon Feb 16 15:46:05 2009 From: jason at jasonjgw.net (Jason White) Date: Tue, 17 Feb 2009 10:46:05 +1100 Subject: [Freeswitch-users] Build and Sofia issues with recent svn trunk revisions In-Reply-To: <20090216030252.GA2229@jdc.jasonjgw.net> References: <20090215070216.GA20246@jdc.jasonjgw.net> <20090216030252.GA2229@jdc.jasonjgw.net> Message-ID: <20090216234605.GA26164@jdc.jasonjgw.net> A fresh checkout from svn trunk fixed my problem, giving me a working build. Interestingly, running svn export and building from a separate directly wasn't enough; a full svn checkout proved necessary to fix this. From lfurrea at gmail.com Mon Feb 16 15:52:47 2009 From: lfurrea at gmail.com (Luis F Urrea) Date: Mon, 16 Feb 2009 17:52:47 -0600 Subject: [Freeswitch-users] xml_cdr Call Flow for attended transfer Message-ID: Hi all, I am trying to understand xml_cdr for an attended (consultative) transfer, I was thinking that the A-leg that initially originated the call would remain untouched but I see that it's global tags get replaced. I have a test call that goes as follows: 201 originates a call and talks to 203 -----> A-leg(1) and B-leg(1) 203 puts 201 on hold and calls 202 (attended) ------> A-leg(2) and B-leg(2) 203 transfers the call 201 and 202 are talking ------> A-leg(1) w/ B-leg(2) ??? Here are the relevant captures: A-leg(1) http://pastebin.freeswitch.org/7253 B-leg(1) http://pastebin.freeswitch.org/7254 A-leg(2) http://pastebin.freeswitch.org/7252 B-leg(2) http://pastebin.freeswitch.org/7255 I was expecting A-leg(1) to have corresponding to 201 which is the original A-leg but it seems that on the transfer, it reverts and 202 appears as the A-leg and 201 as the B-leg. Can someone shed some light on how that transfer gets logged in terms of A-leg and B-leg? TIA -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090216/ced47935/attachment.html From simon at airg.com Mon Feb 16 15:58:03 2009 From: simon at airg.com (Simon Tang) Date: Mon, 16 Feb 2009 15:58:03 -0800 Subject: [Freeswitch-users] How to keep leg A from hanging up fromabridge when Leg B hangs up (and vice versa)? In-Reply-To: <87f2f3b90902161530u32ab7035v615d4171df4e6407@mail.gmail.com> References: <872970CF4A55BF42A5337D570860209F018189D8@HPEXCHVS01.exchange.airg><87f2f3b90902161449s26a2f63u55454bd4ac9b338b@mail.gmail.com><872970CF4A55BF42A5337D570860209F018189E9@HPEXCHVS01.exchange.airg> <87f2f3b90902161530u32ab7035v615d4171df4e6407@mail.gmail.com> Message-ID: <872970CF4A55BF42A5337D570860209F018189FE@HPEXCHVS01.exchange.airg> Linux :). I've mucked around in the conf files, sip_profiles and dialplans to customize it for my use, but that's as far as I've done in playing with configs. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: February 16, 2009 3:30 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] How to keep leg A from hanging up fromabridge when Leg B hangs up (and vice versa)? On Mon, Feb 16, 2009 at 3:04 PM, Simon Tang wrote: > Revision 10626. Default config. I haven't tried latest svn yet, > because my framework breaks with it (probably due to some event > formatting changes). > > Can you explain what you mean by "make samples" and what default config? > I don't recall ever doing a "make samples". > I suppose I should have asked what your platform is! :) Is this Windows, Linux, Unix, or Mac ? -MC _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From pauld at versafon.com Mon Feb 16 16:06:18 2009 From: pauld at versafon.com (Paul D.) Date: Mon, 16 Feb 2009 19:06:18 -0500 Subject: [Freeswitch-users] FS SIP audio quality? In-Reply-To: References: Message-ID: <4999FF7A.6080704@versafon.com> Eh?? Ken Rice wrote: > Paul, > > > > Now that being said, you're post really smells of a troll. > > > From pauld at versafon.com Mon Feb 16 16:08:28 2009 From: pauld at versafon.com (Paul D.) Date: Mon, 16 Feb 2009 19:08:28 -0500 Subject: [Freeswitch-users] FS SIP audio quality? In-Reply-To: <191c3a030902151843p3fdd1903x990ae6cc991724b2@mail.gmail.com> References: <49977FFD.1020002@versafon.com> <191c3a030902151643n62c2b1a0k2c341eb79c1393ea@mail.gmail.com> <4998C99E.9060706@versafon.com> <191c3a030902151842n30590accrd6c7df69e546fb49@mail.gmail.com> <191c3a030902151843p3fdd1903x990ae6cc991724b2@mail.gmail.com> Message-ID: <4999FFFC.10504@versafon.com> I was trying to send tcp dumps today, but the message was rejected because of its size (zipped). How do I send them? Anthony Minessale wrote: > > The typing it takes to start a pcap of each call and email them is > less than you have typed thusfar. > Please just take the captures and send them to us to examine. That's > all. If you have a real issue we would like to address it. > From brian at freeswitch.org Mon Feb 16 16:15:57 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 16 Feb 2009 18:15:57 -0600 Subject: [Freeswitch-users] FS SIP audio quality? In-Reply-To: <4999FFFC.10504@versafon.com> References: <49977FFD.1020002@versafon.com> <191c3a030902151643n62c2b1a0k2c341eb79c1393ea@mail.gmail.com> <4998C99E.9060706@versafon.com> <191c3a030902151842n30590accrd6c7df69e546fb49@mail.gmail.com> <191c3a030902151843p3fdd1903x990ae6cc991724b2@mail.gmail.com> <4999FFFC.10504@versafon.com> Message-ID: <92BA7612-7F23-4079-BE13-755823F6EB49@freeswitch.org> You can send them directly to me brian at freeswitch.org Thanks, /b On Feb 16, 2009, at 6:08 PM, Paul D. wrote: > I was trying to send tcp dumps today, but the message was rejected > because of its size (zipped). How do I send them? > > > Anthony Minessale wrote: >> >> The typing it takes to start a pcap of each call and email them is >> less than you have typed thusfar. >> Please just take the captures and send them to us to examine. That's >> all. If you have a real issue we would like to address it. >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From msc at freeswitch.org Mon Feb 16 16:18:25 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 16 Feb 2009 16:18:25 -0800 Subject: [Freeswitch-users] FS SIP audio quality? In-Reply-To: <4999FFFC.10504@versafon.com> References: <49977FFD.1020002@versafon.com> <191c3a030902151643n62c2b1a0k2c341eb79c1393ea@mail.gmail.com> <4998C99E.9060706@versafon.com> <191c3a030902151842n30590accrd6c7df69e546fb49@mail.gmail.com> <191c3a030902151843p3fdd1903x990ae6cc991724b2@mail.gmail.com> <4999FFFC.10504@versafon.com> Message-ID: <87f2f3b90902161618r19413bb9p238ef00f0f7f10f2@mail.gmail.com> On Mon, Feb 16, 2009 at 4:08 PM, Paul D. wrote: > I was trying to send tcp dumps today, but the message was rejected > because of its size (zipped). How do I send them? Can you put them on a server where the devs can use wget or a browser to download them? -MC From chavpaskov at shaw.ca Mon Feb 16 17:40:04 2009 From: chavpaskov at shaw.ca (Tchavdar Paskov) Date: Mon, 16 Feb 2009 17:40:04 -0800 Subject: [Freeswitch-users] Passing Caller_ID to MOD_LCR Message-ID: Hi, Is there a way to pass Caller Id to mod lcr and somehow to include it in a custom? sql. currently? my dialplan looks like this: ?? ????? ???????? ?????? ????? ????? ??? i guess that Caller_ID is already passed but i was thinking? about making some LCR decisions based on Destination number and Caller_ID /Interstate,Intrastate for example/ Thanks for your time Regards Chav -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090216/a4ee4ffd/attachment.html From krice at suspicious.org Mon Feb 16 18:21:13 2009 From: krice at suspicious.org (Ken Rice) Date: Mon, 16 Feb 2009 20:21:13 -0600 Subject: [Freeswitch-users] FS SIP audio quality? In-Reply-To: <4999FF7A.6080704@versafon.com> Message-ID: If it wasn't meant as a troll I personally, publically, and directly apologize to you... What I see all the time is people want everything for free and then think its the developers responsibility to give away free tech support on this software which is free in the first place. Tony and his crew work on FreeSWITCH as much to feed their families as they do to have an open platform that anyone can use K > From: "Paul D." > Reply-To: > Date: Mon, 16 Feb 2009 19:06:18 -0500 > To: > Subject: Re: [Freeswitch-users] FS SIP audio quality? > > Eh?? > > Ken Rice wrote: >> Paul, >> >> >> >> Now that being said, you're post really smells of a troll. >> >> >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From pauld at versafon.com Mon Feb 16 19:33:26 2009 From: pauld at versafon.com (Paul D.) Date: Mon, 16 Feb 2009 22:33:26 -0500 Subject: [Freeswitch-users] FS SIP audio quality? In-Reply-To: References: <49977FFD.1020002@versafon.com> <191c3a030902151643n62c2b1a0k2c341eb79c1393ea@mail.gmail.com> <4998C99E.9060706@versafon.com> Message-ID: <499A3006.6070208@versafon.com> I re-tested calls to VM replacing some of FS prompts with * ones, and it appears that * sounds were recorded with a better quality/higher volume, so FS itself has nothing to do with that. That's solved. :-) I am going to double check all the equipment we used for tests, like headphones, telephone sets, cables since I am almost convinced that there's nothing in FS which can produce effects I observe. I will post back if I find anything wrong, appreciate everybody's help with this. Brian West wrote: > I'm not able to reproduce this issue.. can you verify the codecs are > what you think they are on both Asterisk and FreeSWITCH. > > /b > > On Feb 15, 2009, at 8:04 PM, Paul D. wrote: > > >> Well, I tried several call scenarios: >> 1. Call from X-Lite or Linksys to VM. >> 2. Call from X-Lite or Linksys to a conference. >> 3. Call from X-Lite or Linksys to a PSTN number via Gafachi and >> CallWithUs. >> >> I have now * 1.6.5 and FS 1.0.3RC1 installed on the same enterprise >> grade Intel server. So just comparing audio in the call scenarios >> above >> * somehow does noticeably better job, sounds clearer and volume is at >> the right level. I am not changing any phone settings of course when >> switching between * and FS. >> I am not biased towards FS or * at the moment, though FS seems to >> have a >> better designed configuration options and community. >> Just wanted to share my experience, and hear some opinions. >> Unfortunately I cannot spend whole amount of time investigating this >> case now, capturing packets etc., but I will try to do that once I >> have >> time. Meanwhile I will have to stick to * for prod. >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From freeswitch-users at lists.rupa.com Mon Feb 16 21:01:11 2009 From: freeswitch-users at lists.rupa.com (Rupa Schomaker (lists)) Date: Mon, 16 Feb 2009 23:01:11 -0600 Subject: [Freeswitch-users] Passing Caller_ID to MOD_LCR In-Reply-To: References: Message-ID: <499A4497.9080001@lists.rupa.com> On 2/16/2009 7:40 PM, Tchavdar Paskov wrote: > > Hi, > > Is there a way to pass Caller Id to mod lcr and somehow to include it in > > a custom sql. > > currently my dialplan looks like this: > > > > > > > break="never"> > > > > > > > > > > > > i guess that Caller_ID is already passed but i was thinking about > > making some LCR decisions based on Destination number and Caller_ID > > /Interstate,Intrastate for example/ > > > > Thanks for your time mod_lcr doesn't make any decisions based on caller id. Probably the best way to handle this would be to use profiles. You could extract the areacode and use that to determine which profile to use. It would be awkward if you want to handle all area codes -- but for a smaller set of area codes it might be sufficient. > > Regards > > Chav > > From cesar at auronix.com Mon Feb 16 21:33:19 2009 From: cesar at auronix.com (Cesar Cepeda) Date: Mon, 16 Feb 2009 23:33:19 -0600 Subject: [Freeswitch-users] Playing a G729 file as ringback Message-ID: <088801c990c1$3e139b50$ba3ad1f0$@com> Hi, I'm using FS with g279 on passthrough mode and I'm trying to play a g729 file as ringback to the A-leg while bridging a call. As far as I understand it should go something like this: . originate {channel_vars}dialstring . set some combination of values on 'ringback', 'transfer_ringback', 'instant_ringback' . bridge The bridge works correctly, but no matter what combination of values I try for the 'xxxringback' vars I never hear the ringback on the A-leg. Can you tell me what I'm missing? Thanks. Cesar Cepeda. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090216/4b920469/attachment.html From chavpaskov at shaw.ca Mon Feb 16 22:44:01 2009 From: chavpaskov at shaw.ca (Tchavdar Paskov) Date: Mon, 16 Feb 2009 22:44:01 -0800 Subject: [Freeswitch-users] Passing Caller_ID to MOD_LCR In-Reply-To: <499A4497.9080001@lists.rupa.com> References: <499A4497.9080001@lists.rupa.com> Message-ID: i was thinking more in direction of building custom sql that deals with both Caller_id and destination number. I'm aware that the current? off the box mod_lcr? has no such ability? and that's why i asked if there is a way from? dial plan to pass the caller_id_number? channel variable to? mod_lcr. Thanks? ----- Original Message ----- From: "Rupa Schomaker (lists)" Date: Monday, February 16, 2009 9:01 pm Subject: Re: [Freeswitch-users] Passing Caller_ID to MOD_LCR To: freeswitch-users at lists.freeswitch.org > On 2/16/2009 7:40 PM, Tchavdar Paskov wrote: > > > Hi, > > > Is there a way to pass Caller Id to mod lcr and somehow to > include it in > > > a custom? sql. > > > currently? my dialplan looks like this: > > > > > >??? > > >?????? field="destination_number" expression="^1(\d+)$" > > > break="never"> > > >????????? > > > >??????? application="lcr" data="$1"/> > > >?????? application="bridge" data="${lcr_auto_route}"/> > > >?????? > > >???? > > > i guess that Caller_ID is already passed but i was > thinking? about > > > making some LCR decisions based on Destination number and > Caller_ID> > /Interstate,Intrastate for example/ > > > > > > Thanks for your time > > mod_lcr doesn't make any decisions based on caller id.? > Probably the > best way to handle this would be to use profiles.? You > could extract the > areacode and use that to determine which profile to use.? > It would be > awkward if you want to handle all area codes -- but for a > smaller set of > area codes it might be sufficient. > > > > > Regards > > > Chav > > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090216/2ac5de86/attachment-0001.html From chavpaskov at shaw.ca Mon Feb 16 23:18:47 2009 From: chavpaskov at shaw.ca (Tchavdar Paskov) Date: Mon, 16 Feb 2009 23:18:47 -0800 Subject: [Freeswitch-users] Passing Caller_ID to MOD_LCR In-Reply-To: References: <499A4497.9080001@lists.rupa.com> Message-ID: I also have? another question. When i insert my custom query? it looks like the profile is loaded successfully? but then when i place a call or use ? lcr? ##########? default? /which is where i defined the sql query/? and check the console output? turns out that the switch is using? the default? sql guery . How i can make sure that? the custom sql is the only one that is to be executed. /All required fields in the custom sql are in accordance with the requirements - it returns the exact required names and returned field number is also correct/. Thank you agai for your time Chav ----- Original Message ----- From: Tchavdar Paskov Date: Monday, February 16, 2009 10:44 pm Subject: Re: [Freeswitch-users] Passing Caller_ID to MOD_LCR To: freeswitch-users at lists.freeswitch.org > i was thinking more in direction of building custom sql that > deals with both Caller_id and destination number. I'm aware that > the current? off the box mod_lcr? has no such ability? and > that's why i asked if there is a way from? dial plan to pass the > caller_id_number? channel variable to? mod_lcr. > Thanks? > > ----- Original Message ----- > From: "Rupa Schomaker (lists)" > Date: Monday, February 16, 2009 9:01 pm > Subject: Re: [Freeswitch-users] Passing Caller_ID to MOD_LCR > To: freeswitch-users at lists.freeswitch.org > > > On 2/16/2009 7:40 PM, Tchavdar Paskov wrote: > > > > Hi, > > > > Is there a way to pass Caller Id to mod lcr and somehow to > > include it in > > > > a custom? sql. > > > > currently? my dialplan looks like this: > > > > > > > >??? > > > >?????? > field="destination_number" expression="^1(\d+)$" > > > > break="never"> > > > >????????? > > > > > >??????? > application="lcr" data="$1"/> > > > >?????? > application="bridge" data="${lcr_auto_route}"/> > > > >?????? > > > >???? > > > > i guess that Caller_ID is already passed but i was > > thinking? about > > > > making some LCR decisions based on Destination number and > > Caller_ID> > /Interstate,Intrastate for example/ > > > > > > > > Thanks for your time > > > > mod_lcr doesn't make any decisions based on caller id.? > > Probably the > > best way to handle this would be to use profiles.? You > > could extract the > > areacode and use that to determine which profile to use.? > > It would be > > awkward if you want to handle all area codes -- but for a > > smaller set of > > area codes it might be sufficient. > > > > > > > > Regards > > > > Chav > > > > > > > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090216/20a911a0/attachment.html From nik.middleton at noblesolutions.co.uk Tue Feb 17 02:05:12 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Tue, 17 Feb 2009 10:05:12 -0000 Subject: [Freeswitch-users] FS SIP audio quality? In-Reply-To: <499A3006.6070208@versafon.com> References: <49977FFD.1020002@versafon.com> <191c3a030902151643n62c2b1a0k2c341eb79c1393ea@mail.gmail.com> <4998C99E.9060706@versafon.com> <499A3006.6070208@versafon.com> Message-ID: For what it's worth, using Asterisk recordings, I found FS to be better than when played on an Asterisk system. I came to the same conclusion early on that the included prompts with FS were of a relatively poor nature. Not volunteering to record new ones, but they do let the product down, as they lead to discussions as below. Regards, -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Paul D. Sent: 17 February 2009 03:33 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] FS SIP audio quality? I re-tested calls to VM replacing some of FS prompts with * ones, and it appears that * sounds were recorded with a better quality/higher volume, so FS itself has nothing to do with that. That's solved. :-) I am going to double check all the equipment we used for tests, like headphones, telephone sets, cables since I am almost convinced that there's nothing in FS which can produce effects I observe. I will post back if I find anything wrong, appreciate everybody's help with this. Brian West wrote: > I'm not able to reproduce this issue.. can you verify the codecs are > what you think they are on both Asterisk and FreeSWITCH. > > /b > > On Feb 15, 2009, at 8:04 PM, Paul D. wrote: > > >> Well, I tried several call scenarios: >> 1. Call from X-Lite or Linksys to VM. >> 2. Call from X-Lite or Linksys to a conference. >> 3. Call from X-Lite or Linksys to a PSTN number via Gafachi and >> CallWithUs. >> >> I have now * 1.6.5 and FS 1.0.3RC1 installed on the same enterprise >> grade Intel server. So just comparing audio in the call scenarios >> above >> * somehow does noticeably better job, sounds clearer and volume is at >> the right level. I am not changing any phone settings of course when >> switching between * and FS. >> I am not biased towards FS or * at the moment, though FS seems to >> have a >> better designed configuration options and community. >> Just wanted to share my experience, and hear some opinions. >> Unfortunately I cannot spend whole amount of time investigating this >> case now, capturing packets etc., but I will try to do that once I >> have >> time. Meanwhile I will have to stick to * for prod. >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From gmaruzz at celliax.org Tue Feb 17 02:19:30 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Tue, 17 Feb 2009 11:19:30 +0100 Subject: [Freeswitch-users] FS SIP audio quality? In-Reply-To: References: <49977FFD.1020002@versafon.com> <191c3a030902151643n62c2b1a0k2c341eb79c1393ea@mail.gmail.com> <4998C99E.9060706@versafon.com> <499A3006.6070208@versafon.com> Message-ID: <7b197bef0902170219l217522aal4fff04b5e0e51d06@mail.gmail.com> There is also another side to make mimd to: the Asterisk sounds you hear more often (the demo ones) are very long ones. The ones of the FS demo are very very short (many times just one word) and concatenated with the insertion of sleeps. That is probably someway altering the equation between user experiences Sincerely, Giovanni Maruzzelli ========================================= www.celliax.org via Pierlombardo 9, 20135 Milano Italy gmaruzz at celliax dot org Cell : +39-347-2665618 Fax : +39-02-87390039 On Tue, Feb 17, 2009 at 11:05 AM, Nik Middleton wrote: > For what it's worth, using Asterisk recordings, I found FS to be better > than when played on an Asterisk system. > > I came to the same conclusion early on that the included prompts with FS > were of a relatively poor nature. Not volunteering to record new ones, > but they do let the product down, as they lead to discussions as below. > > > Regards, > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Paul > D. > Sent: 17 February 2009 03:33 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] FS SIP audio quality? > > I re-tested calls to VM replacing some of FS prompts with * ones, and it > > appears that * sounds were recorded with a better quality/higher volume, > > so FS itself has nothing to do with that. That's solved. :-) > I am going to double check all the equipment we used for tests, like > headphones, telephone sets, cables since I am almost convinced that > there's nothing in FS which can produce effects I observe. > I will post back if I find anything wrong, appreciate everybody's help > with this. > > Brian West wrote: >> I'm not able to reproduce this issue.. can you verify the codecs are >> what you think they are on both Asterisk and FreeSWITCH. >> >> /b >> >> On Feb 15, 2009, at 8:04 PM, Paul D. wrote: >> >> >>> Well, I tried several call scenarios: >>> 1. Call from X-Lite or Linksys to VM. >>> 2. Call from X-Lite or Linksys to a conference. >>> 3. Call from X-Lite or Linksys to a PSTN number via Gafachi and >>> CallWithUs. >>> >>> I have now * 1.6.5 and FS 1.0.3RC1 installed on the same enterprise >>> grade Intel server. So just comparing audio in the call scenarios >>> above >>> * somehow does noticeably better job, sounds clearer and volume is at >>> the right level. I am not changing any phone settings of course when >>> switching between * and FS. >>> I am not biased towards FS or * at the moment, though FS seems to >>> have a >>> better designed configuration options and community. >>> Just wanted to share my experience, and hear some opinions. >>> Unfortunately I cannot spend whole amount of time investigating this >>> case now, capturing packets etc., but I will try to do that once I >>> have >>> time. Meanwhile I will have to stick to * for prod. >>> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From jaybinks at gmail.com Tue Feb 17 02:27:42 2009 From: jaybinks at gmail.com (jay binks) Date: Tue, 17 Feb 2009 20:27:42 +1000 Subject: [Freeswitch-users] FS SIP audio quality? In-Reply-To: <7b197bef0902170219l217522aal4fff04b5e0e51d06@mail.gmail.com> References: <49977FFD.1020002@versafon.com> <191c3a030902151643n62c2b1a0k2c341eb79c1393ea@mail.gmail.com> <4998C99E.9060706@versafon.com> <499A3006.6070208@versafon.com> <7b197bef0902170219l217522aal4fff04b5e0e51d06@mail.gmail.com> Message-ID: Back in November, Brian ( BKW ) was raising money to get new sounds recorded ... intending to have them for the 1.0.2 release.. I wonder if they made it in, or if they are still coming ... Jay On Tue, Feb 17, 2009 at 8:19 PM, Giovanni Maruzzelli wrote: > There is also another side to make mimd to: the Asterisk sounds you > hear more often (the demo ones) are very long ones. > > The ones of the FS demo are very very short (many times just one word) > and concatenated with the insertion of sleeps. > > That is probably someway altering the equation between user experiences > > > Sincerely, > > Giovanni Maruzzelli > ========================================= > www.celliax.org > via Pierlombardo 9, 20135 Milano > Italy > gmaruzz at celliax dot org > Cell : +39-347-2665618 > Fax : +39-02-87390039 > > > > > On Tue, Feb 17, 2009 at 11:05 AM, Nik Middleton > wrote: > > For what it's worth, using Asterisk recordings, I found FS to be better > > than when played on an Asterisk system. > > > > I came to the same conclusion early on that the included prompts with FS > > were of a relatively poor nature. Not volunteering to record new ones, > > but they do let the product down, as they lead to discussions as below. > > > > > > Regards, > > > > -----Original Message----- > > From: freeswitch-users-bounces at lists.freeswitch.org > > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Paul > > D. > > Sent: 17 February 2009 03:33 > > To: freeswitch-users at lists.freeswitch.org > > Subject: Re: [Freeswitch-users] FS SIP audio quality? > > > > I re-tested calls to VM replacing some of FS prompts with * ones, and it > > > > appears that * sounds were recorded with a better quality/higher volume, > > > > so FS itself has nothing to do with that. That's solved. :-) > > I am going to double check all the equipment we used for tests, like > > headphones, telephone sets, cables since I am almost convinced that > > there's nothing in FS which can produce effects I observe. > > I will post back if I find anything wrong, appreciate everybody's help > > with this. > > > > Brian West wrote: > >> I'm not able to reproduce this issue.. can you verify the codecs are > >> what you think they are on both Asterisk and FreeSWITCH. > >> > >> /b > >> > >> On Feb 15, 2009, at 8:04 PM, Paul D. wrote: > >> > >> > >>> Well, I tried several call scenarios: > >>> 1. Call from X-Lite or Linksys to VM. > >>> 2. Call from X-Lite or Linksys to a conference. > >>> 3. Call from X-Lite or Linksys to a PSTN number via Gafachi and > >>> CallWithUs. > >>> > >>> I have now * 1.6.5 and FS 1.0.3RC1 installed on the same enterprise > >>> grade Intel server. So just comparing audio in the call scenarios > >>> above > >>> * somehow does noticeably better job, sounds clearer and volume is at > >>> the right level. I am not changing any phone settings of course when > >>> switching between * and FS. > >>> I am not biased towards FS or * at the moment, though FS seems to > >>> have a > >>> better designed configuration options and community. > >>> Just wanted to share my experience, and hear some opinions. > >>> Unfortunately I cannot spend whole amount of time investigating this > >>> case now, capturing packets etc., but I will try to do that once I > >>> have > >>> time. Meanwhile I will have to stick to * for prod. > >>> > >> > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > >> > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Sincerely Jay -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090217/9c1d8430/attachment-0001.html From jason at jasonjgw.net Tue Feb 17 02:35:10 2009 From: jason at jasonjgw.net (Jason White) Date: Tue, 17 Feb 2009 21:35:10 +1100 Subject: [Freeswitch-users] FS SIP audio quality? In-Reply-To: References: <49977FFD.1020002@versafon.com> <191c3a030902151643n62c2b1a0k2c341eb79c1393ea@mail.gmail.com> <4998C99E.9060706@versafon.com> <499A3006.6070208@versafon.com> <7b197bef0902170219l217522aal4fff04b5e0e51d06@mail.gmail.com> Message-ID: <20090217103510.GA29891@jdc.jasonjgw.net> jay binks wrote: > Back in November, Brian ( BKW ) was raising money to get new sounds recorded > ... > intending to have them for the 1.0.2 release.. > > I wonder if they made it in, or if they are still coming ... Release 1.0.7 of the sound files was made available soon thereafter, which I understand includes the new material that was recorded. I might be wrong, though. From dave at 3c.co.uk Tue Feb 17 04:35:09 2009 From: dave at 3c.co.uk (David Knell) Date: Tue, 17 Feb 2009 12:35:09 +0000 Subject: [Freeswitch-users] FS SIP audio quality? In-Reply-To: <499A3006.6070208@versafon.com> References: <49977FFD.1020002@versafon.com> <191c3a030902151643n62c2b1a0k2c341eb79c1393ea@mail.gmail.com> <4998C99E.9060706@versafon.com> <499A3006.6070208@versafon.com> Message-ID: <499AAEFD.3050307@3c.co.uk> Paul D. wrote: > I re-tested calls to VM replacing some of FS prompts with * ones, and it > appears that * sounds were recorded with a better quality/higher volume, > so FS itself has nothing to do with that. That's solved. :-) > There's a long history of people in A/B listening tests reporting louder as sounding better on the same source material - even if the additional volume isn't detectable as such. Which, I guess, explains my 25 years of going to Motorhead gigs. --Dave -- David Knell, Director, 3C Limited T: 020 8114 5002 F: 020 3002 7257 M: 07773 800623 http://www.3c.co.uk From pablosaro at gmail.com Tue Feb 17 05:29:00 2009 From: pablosaro at gmail.com (Pablo Hernan Saro) Date: Tue, 17 Feb 2009 11:29:00 -0200 Subject: [Freeswitch-users] Options for configure script Message-ID: <247f8100902170529v79ed65a4p7df7241b4c11cb2d@mail.gmail.com> Hi there, Anyone knows how to configure FS in order to have log folder in /var/log/freeswitch ? I did ./configure --prefix=/opt/freeswitch and after install I got logs, pid file and cdrs under /opt/freeswitch/log Thanks in advance. Pablo From kokoska.rokoska at post.cz Tue Feb 17 05:42:31 2009 From: kokoska.rokoska at post.cz (kokoska rokoska) Date: Tue, 17 Feb 2009 14:42:31 +0100 Subject: [Freeswitch-users] FS SIP audio quality? In-Reply-To: <499AAEFD.3050307@3c.co.uk> References: <49977FFD.1020002@versafon.com> <191c3a030902151643n62c2b1a0k2c341eb79c1393ea@mail.gmail.com> <4998C99E.9060706@versafon.com> <499A3006.6070208@versafon.com> <499AAEFD.3050307@3c.co.uk> Message-ID: <499ABEC7.2040703@post.cz> David Knell napsal(a): > There's a long history of people in A/B listening tests reporting louder > as sounding > better on the same source material - even if the additional volume isn't > detectable > as such. > Yes, you are right :-) And therefor a lot of (nearly all of) European TelCo operator (TDM, not VoIP) normalize their messages to -3 dB instead of -6 dB. And one of them (yes, you guess it right, it is Telefonica O2 :-) normalize recordings to 0 dB. And it is VERY loud :-) Best regards, kokoska.rokoska From edpimentl at gmail.com Tue Feb 17 05:50:50 2009 From: edpimentl at gmail.com (EdPimentl) Date: Tue, 17 Feb 2009 08:50:50 -0500 Subject: [Freeswitch-users] Anyone running FS from a Thumb Flash USB? Message-ID: <9dc4a1670902170550r3071f65dn76112fdbebaeb6ba@mail.gmail.com> Hello FS Members, Are there any example of FS running on a Thumb Flash USB? Thanks in advance, -E -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090217/44ace346/attachment.html From anthony.minessale at gmail.com Tue Feb 17 05:54:51 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 17 Feb 2009 07:54:51 -0600 Subject: [Freeswitch-users] FS SIP audio quality? In-Reply-To: <499AAEFD.3050307@3c.co.uk> References: <49977FFD.1020002@versafon.com> <191c3a030902151643n62c2b1a0k2c341eb79c1393ea@mail.gmail.com> <4998C99E.9060706@versafon.com> <499A3006.6070208@versafon.com> <499AAEFD.3050307@3c.co.uk> Message-ID: <191c3a030902170554g20dc665bq2c4933ef2ec1becb@mail.gmail.com> Maybe the sox script brian uses to downsample the files has a problem. What if you download the 48k package (original) and listen to that? On Tue, Feb 17, 2009 at 6:35 AM, David Knell wrote: > Paul D. wrote: > > I re-tested calls to VM replacing some of FS prompts with * ones, and it > > appears that * sounds were recorded with a better quality/higher volume, > > so FS itself has nothing to do with that. That's solved. :-) > > > There's a long history of people in A/B listening tests reporting louder > as sounding > better on the same source material - even if the additional volume isn't > detectable > as such. > > Which, I guess, explains my 25 years of going to Motorhead gigs. > > --Dave > > -- > David Knell, Director, 3C Limited > T: 020 8114 5002 F: 020 3002 7257 M: 07773 800623 > http://www.3c.co.uk > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090217/90747390/attachment.html From krice at freeswitch.org Tue Feb 17 06:00:49 2009 From: krice at freeswitch.org (Ken Rice) Date: Tue, 17 Feb 2009 08:00:49 -0600 Subject: [Freeswitch-users] Anyone running FS from a Thumb Flash USB? In-Reply-To: <9dc4a1670902170550r3071f65dn76112fdbebaeb6ba@mail.gmail.com> Message-ID: Could be done ed... FS itself isnt that big From: EdPimentl Reply-To: Date: Tue, 17 Feb 2009 08:50:50 -0500 To: Subject: [Freeswitch-users] Anyone running FS from a Thumb Flash USB? Hello FS Members, Are there any example of FS running on a Thumb Flash USB? Thanks in advance, -E _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090217/e58da0bf/attachment.html From edpimentl at gmail.com Tue Feb 17 06:10:30 2009 From: edpimentl at gmail.com (EdPimentl) Date: Tue, 17 Feb 2009 09:10:30 -0500 Subject: [Freeswitch-users] Anyone running FS from a Thumb Flash USB? In-Reply-To: References: <9dc4a1670902170550r3071f65dn76112fdbebaeb6ba@mail.gmail.com> Message-ID: <9dc4a1670902170610p2e3d94c4u3796958f52f5b187@mail.gmail.com> I would be glad to put a bounty for a FS(and Skypiax/softphone) running on Flash Thumb Drive. Project would be documented on Wiki for others to use and improve. -E -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090217/753a1f40/attachment.html From freeswitch-users at lists.rupa.com Tue Feb 17 06:11:38 2009 From: freeswitch-users at lists.rupa.com (Rupa Schomaker (lists)) Date: Tue, 17 Feb 2009 08:11:38 -0600 Subject: [Freeswitch-users] Passing Caller_ID to MOD_LCR In-Reply-To: References: <499A4497.9080001@lists.rupa.com> Message-ID: <499AC59A.4050209@lists.rupa.com> On 2/17/2009 1:18 AM, Tchavdar Paskov wrote: > I also have another question. > When i insert my custom query it looks like the profile is loaded > successfully but then when i place a call or use > > lcr ########## default /which is where i defined the sql query/ and > check the console output turns out that the switch is using the > default sql guery . lcr_admin show profiles should show you the profiles loaded and the profile's settings. "default" is a reserved profile name -- I should probably prevent that from loading. > How i can make sure that the custom sql is the only one that is to be > executed. > /All required fields in the custom sql are in accordance with the > requirements - it returns the exact required names and returned field > number is also correct/. > > Thank you agai for your time > Chav Try running with debug logging turned on (f8 on the console). This will show the sql being passed to the database. == Regarding passing the callerid to the custom sql, let me see what I can come up with... From kerrada2003 at yahoo.com Tue Feb 17 06:51:52 2009 From: kerrada2003 at yahoo.com (Ali Al-Rubaie) Date: Tue, 17 Feb 2009 06:51:52 -0800 (PST) Subject: [Freeswitch-users] Realm Value In-Reply-To: Message-ID: <350067.9862.qm@web33701.mail.mud.yahoo.com> Thanks Brian, Actually we're using freeswitch ver 1.0.2. Regards, Message: 5 Date: Thu, 12 Feb 2009 15:48:00 -0600 From: Brian West Subject: Re: [Freeswitch-users] Realm value To: freeswitch-users at lists.freeswitch.org Message-ID: Content-Type: text/plain; charset=US-ASCII; format=flowed; delsp=yes What SVN rev? /b On Feb 12, 2009, at 3:41 PM, Ali Al-Rubaie wrote: > Hi, > > How can the default value of "realm" be changed? I had changed the > command: > > > > in the file internal.xml but FS still uses the server IP address as > the challenge realm. > > Thanks in advance! > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090217/af260081/attachment-0001.html From brian at freeswitch.org Tue Feb 17 07:02:54 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 17 Feb 2009 09:02:54 -0600 Subject: [Freeswitch-users] Playing a G729 file as ringback In-Reply-To: <088801c990c1$3e139b50$ba3ad1f0$@com> References: <088801c990c1$3e139b50$ba3ad1f0$@com> Message-ID: <8730408C-CD63-4CF5-B505-11CE3AB03310@freeswitch.org> Its currently not possible. /b On Feb 16, 2009, at 11:33 PM, Cesar Cepeda wrote: > Hi, > > I?m using FS with g279 on passthrough mode and I?m trying to play a > g729 file as ringback to the A-leg while bridging a call. As far as > I understand it should go something like this: > > ? originate {channel_vars}dialstring > ? set some combination of values on ?ringback?, > ?transfer_ringback?, ?instant_ringback? > ? bridge > > The bridge works correctly, but no matter what combination of values > I try for the ?xxxringback? vars I never hear the ringback on the A- > leg. > > Can you tell me what I?m missing? > > Thanks. > > Cesar Cepeda. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090217/3349148f/attachment.html From mrene_lists at avgs.ca Tue Feb 17 07:05:07 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Tue, 17 Feb 2009 10:05:07 -0500 Subject: [Freeswitch-users] Options for configure script In-Reply-To: <247f8100902170529v79ed65a4p7df7241b4c11cb2d@mail.gmail.com> References: <247f8100902170529v79ed65a4p7df7241b4c11cb2d@mail.gmail.com> Message-ID: You can specify those as runtime argument. $ ./freeswitch -h these are the optional arguments you can pass to freeswitch -nf -- no forking -u [user] -- specify user to switch to -g [group] -- specify group to switch to -help -- this message -core -- dump cores -hp -- enable high priority settings -vg -- run under valgrind -nosql -- disable internal sql scoreboard -stop -- stop freeswitch -nc -- do not output to a console and background -c -- output to a console and stay in the foreground -conf [confdir] -- specify an alternate config dir -log [logdir] -- specify an alternate log dir -db [dbdir] -- specify an alternate db dir -mod [moddir] -- specify an alternate mod dir -htdocs [htdocsdir] -- specify an alternate htdocs dir -scripts [scriptsdir] -- specify an alternate scripts dir But in my opinion, you should keep all files at the same place and make a symbolic link if you really want stuf to be accessible from /var/log Mathieu -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090217/7b35685e/attachment.html From pablosaro at gmail.com Tue Feb 17 08:54:01 2009 From: pablosaro at gmail.com (Pablo Hernan Saro) Date: Tue, 17 Feb 2009 14:54:01 -0200 Subject: [Freeswitch-users] Options for configure script In-Reply-To: References: <247f8100902170529v79ed65a4p7df7241b4c11cb2d@mail.gmail.com> Message-ID: <247f8100902170854w291a5ed8ja1090c8a573a5179@mail.gmail.com> Thanks Mathieu. So, it is not possible to set this at build time by passing a parameter to the configure script... IMHO, a symbolic link is not a good idea in production environments. BR Pablo On Tue, Feb 17, 2009 at 1:05 PM, Mathieu Rene wrote: > You can specify those as runtime argument. > > $ ./freeswitch -h > these are the optional arguments you can pass to freeswitch > -nf -- no forking > -u [user] -- specify user to switch to > -g [group] -- specify group to switch to > -help -- this message > -core -- dump cores > -hp -- enable high priority settings > -vg -- run under valgrind > -nosql -- disable internal sql scoreboard > -stop -- stop freeswitch > -nc -- do not output to a console and background > -c -- output to a console and stay in the foreground > -conf [confdir] -- specify an alternate config dir > -log [logdir] -- specify an alternate log dir > -db [dbdir] -- specify an alternate db dir > -mod [moddir] -- specify an alternate mod dir > -htdocs [htdocsdir] -- specify an alternate htdocs dir > -scripts [scriptsdir] -- specify an alternate scripts dir > > But in my opinion, you should keep all files at the same place and make a > symbolic link if you really want stuf to be accessible from /var/log > > > Mathieu > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From mkarp at securesilence.com Tue Feb 17 09:09:41 2009 From: mkarp at securesilence.com (Maxim Karp) Date: Tue, 17 Feb 2009 09:09:41 -0800 Subject: [Freeswitch-users] Voicemail prompts playback too quickly Message-ID: <007301c99122$86438fa0$92caaee0$@com> Hello, Our voicemail prompts playback much too quickly on FS v1.0.1. Any suggestions to slow them down? Thanks, Maxim. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090217/886edbca/attachment.html From cjonesmo at gmail.com Mon Feb 16 18:24:46 2009 From: cjonesmo at gmail.com (Chris Jones) Date: Mon, 16 Feb 2009 20:24:46 -0600 Subject: [Freeswitch-users] SIPX/FS Auto attendant Message-ID: I'm having a problem that calls to the auto-attendant won't transfer. I know this has been a problem in the past but thought that it was fixed. Whenever I enter an extension (or press a key to transfer me to one), the call just hangs up. I ran Freeswitch on the console, but all I see happening is this: 2009-02-16 20:22:51 [NOTICE] mod_sofia.c:1338 sofia_receive_message() Ring-Ready sofia/papayaecs.bluewavecorp.local/8323160383 at 64.2.142.116! 2009-02-16 20:22:51 [NOTICE] mod_sofia.c:1338 sofia_receive_message() Pre-Answer sofia/papayaecs.bluewavecorp.local/8323160383 at 64.2.142.116! 2009-02-16 20:22:51 [NOTICE] mod_dptools.c:600 answer_function() Channel [sofia/papayaecs.bluewavecorp.local/8323160383 at 64.2.142.116] has been answered 2009-02-16 20:23:07 [NOTICE] sofia.c:3179 sofia_handle_sip_i_state() Hangup sofia/papayaecs.bluewavecorp.local/8323160383 at 64.2.142.116 [CS_EXECUTE] [NORMAL_UNSPECIFIED] 2009-02-16 20:23:07 [NOTICE] switch_core_session.c:960 switch_core_session_thread() Session 1 ( sofia/papayaecs.bluewavecorp.local/8323160383 at 64.2.142.116) Ended 2009-02-16 20:23:07 [NOTICE] switch_core_session.c:962 switch_core_session_thread() Close Channel sofia/papayaecs.bluewavecorp.local/8323160383 at 64.2.142.116 [CS_HANGUP] Does anyone have an idea as to how to fix this? The call is coming from a SIP Trunk on vitelity. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090216/e33b9e70/attachment.html From msc at freeswitch.org Tue Feb 17 09:18:51 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 17 Feb 2009 09:18:51 -0800 Subject: [Freeswitch-users] Voicemail prompts playback too quickly In-Reply-To: <007301c99122$86438fa0$92caaee0$@com> References: <007301c99122$86438fa0$92caaee0$@com> Message-ID: <87f2f3b90902170918oc86f4dm4548a68085565ddb@mail.gmail.com> > Our voicemail prompts playback much too quickly on FS v1.0.1. Any > suggestions to slow them down? At this point the best thing for you to do is to update to the latest trunk. We are 99.99% ready to tag 1.0.3RC2 which has significant improvements over 1.0.1. -MC P.S. - you might find this page useful: http://wiki.freeswitch.org/wiki/Reporting_Bugs If you see anything missing from this page please let me know. I'm trying to make it as friendly and easy to use as possible. From brian at freeswitch.org Tue Feb 17 09:21:10 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 17 Feb 2009 11:21:10 -0600 Subject: [Freeswitch-users] SIPX/FS Auto attendant In-Reply-To: References: Message-ID: You're missing some key information to help us answer your question. First off we will need to know the SVN rev, Then you might want to press F8 and check out the debug log. Chances are it'll tell you exactly why. What concerns me is the fact that a .local domain is in there. I wonder if maybe it can't resolve this at some point... again just me guessing here. Please crank up the debug and lets see if we can see any details in that log. /b On Feb 16, 2009, at 8:24 PM, Chris Jones wrote: > I'm having a problem that calls to the auto-attendant won't > transfer. I know this has been a problem in the past but thought > that it was fixed. Whenever I enter an extension (or press a key to > transfer me to one), the call just hangs up. I ran Freeswitch on the > console, but all I see happening is this: > > 2009-02-16 20:22:51 [NOTICE] mod_sofia.c:1338 > sofia_receive_message() Ring-Ready sofia/papayaecs.bluewavecorp.local/8323160383 at 64.2.142.116 > ! > 2009-02-16 20:22:51 [NOTICE] mod_sofia.c:1338 > sofia_receive_message() Pre-Answer sofia/papayaecs.bluewavecorp.local/8323160383 at 64.2.142.116 > ! > 2009-02-16 20:22:51 [NOTICE] mod_dptools.c:600 answer_function() > Channel [sofia/papayaecs.bluewavecorp.local/8323160383 at 64.2.142.116] > has been answered > 2009-02-16 20:23:07 [NOTICE] sofia.c:3179 sofia_handle_sip_i_state() > Hangup sofia/papayaecs.bluewavecorp.local/8323160383 at 64.2.142.116 > [CS_EXECUTE] [NORMAL_UNSPECIFIED] > 2009-02-16 20:23:07 [NOTICE] switch_core_session.c:960 > switch_core_session_thread() Session 1 (sofia/papayaecs.bluewavecorp.local/8323160383 at 64.2.142.116 > ) Ended > 2009-02-16 20:23:07 [NOTICE] switch_core_session.c:962 > switch_core_session_thread() Close Channel sofia/papayaecs.bluewavecorp.local/8323160383 at 64.2.142.116 > [CS_HANGUP] > Does anyone have an idea as to how to fix this? The call is coming > from a SIP Trunk on vitelity. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090217/5f187060/attachment-0001.html From msc at freeswitch.org Tue Feb 17 09:34:39 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 17 Feb 2009 09:34:39 -0800 Subject: [Freeswitch-users] Options for configure script In-Reply-To: <247f8100902170854w291a5ed8ja1090c8a573a5179@mail.gmail.com> References: <247f8100902170529v79ed65a4p7df7241b4c11cb2d@mail.gmail.com> <247f8100902170854w291a5ed8ja1090c8a573a5179@mail.gmail.com> Message-ID: <87f2f3b90902170934w6853809anb7c06b9fc92bf32e@mail.gmail.com> On Tue, Feb 17, 2009 at 8:54 AM, Pablo Hernan Saro wrote: > Thanks Mathieu. > So, it is not possible to set this at build time by passing a > parameter to the configure script... > IMHO, a symbolic link is not a good idea in production environments. > BR > You can also specify these in the config files. You can put the logs and/or CDR files wherever you'd like. If you open up conf/autoload_configs/logfile.conf.xml you'll see that there is a line commented out - this line will let you choose the default log file path. In fact, I think you'll approve of the location already specified on that line. :) -MC From brian at freeswitch.org Tue Feb 17 09:39:58 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 17 Feb 2009 11:39:58 -0600 Subject: [Freeswitch-users] Realm Value In-Reply-To: <350067.9862.qm@web33701.mail.mud.yahoo.com> References: <350067.9862.qm@web33701.mail.mud.yahoo.com> Message-ID: <2ADB8FA0-3E87-4C99-9A6F-F09C6E0F5DB8@freeswitch.org> I'm trying to get a clear picture of what you're trying to accomplish. Why would you need/want to set a static realm? Anyway can you collect sip traces? /b On Feb 17, 2009, at 8:51 AM, Ali Al-Rubaie wrote: > Thanks Brian, > > Actually we're using freeswitch ver 1.0.2. > > > Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090217/e801928a/attachment.html From pablosaro at gmail.com Tue Feb 17 09:59:09 2009 From: pablosaro at gmail.com (Pablo Hernan Saro) Date: Tue, 17 Feb 2009 15:59:09 -0200 Subject: [Freeswitch-users] Options for configure script In-Reply-To: <87f2f3b90902170934w6853809anb7c06b9fc92bf32e@mail.gmail.com> References: <247f8100902170529v79ed65a4p7df7241b4c11cb2d@mail.gmail.com> <247f8100902170854w291a5ed8ja1090c8a573a5179@mail.gmail.com> <87f2f3b90902170934w6853809anb7c06b9fc92bf32e@mail.gmail.com> Message-ID: <247f8100902170959m30c321e9md1979d328dcaf846@mail.gmail.com> Hi Michael, Thank you very much for your help. It meets my needs. I was thinking in something like: ./configure --prefix=/opt/freeswitch --localstatedir=/var/log/freeswitch But I can get same results changing the default configuration of conf/autoload_configs/logfile.conf.xml and conf/autoload_configs/xml_cdr.conf.xml. Regards, Pablo On Tue, Feb 17, 2009 at 3:34 PM, Michael Collins wrote: > On Tue, Feb 17, 2009 at 8:54 AM, Pablo Hernan Saro wrote: >> Thanks Mathieu. >> So, it is not possible to set this at build time by passing a >> parameter to the configure script... >> IMHO, a symbolic link is not a good idea in production environments. >> BR >> > > You can also specify these in the config files. You can put the logs > and/or CDR files wherever you'd like. If you open up > conf/autoload_configs/logfile.conf.xml you'll see that there is a line > commented out - this line will let you choose the default log file > path. In fact, I think you'll approve of the location already > specified on that line. :) > > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From mike at jerris.com Tue Feb 17 10:24:03 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 17 Feb 2009 13:24:03 -0500 Subject: [Freeswitch-users] Options for configure script In-Reply-To: <247f8100902170959m30c321e9md1979d328dcaf846@mail.gmail.com> References: <247f8100902170529v79ed65a4p7df7241b4c11cb2d@mail.gmail.com> <247f8100902170854w291a5ed8ja1090c8a573a5179@mail.gmail.com> <87f2f3b90902170934w6853809anb7c06b9fc92bf32e@mail.gmail.com> <247f8100902170959m30c321e9md1979d328dcaf846@mail.gmail.com> Message-ID: We don't yet support localstatedir configure option. I expect we will soon. Mike On Feb 17, 2009, at 12:59 PM, Pablo Hernan Saro wrote: > Hi Michael, > > Thank you very much for your help. It meets my needs. > I was thinking in something like: > ./configure --prefix=/opt/freeswitch --localstatedir=/var/log/ > freeswitch > But I can get same results changing the default configuration of > conf/autoload_configs/logfile.conf.xml and > conf/autoload_configs/xml_cdr.conf.xml. > Regards, > > Pablo From nik.middleton at noblesolutions.co.uk Tue Feb 17 10:23:35 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Tue, 17 Feb 2009 18:23:35 -0000 Subject: [Freeswitch-users] Big delays in playing audio files Message-ID: Having spent the last week developing a small js app, I ran some tests today. With just 5 calls going on, I'm seeing huge delays from when the call is answered to when the audio file is played. Sometimes it doesn't even play at all!! Example 3 calls and the matching playbacks 2009-02-17 15:41:04 [NOTICE] voice.js:1 console_log() ready: Start DTMF 2009-02-17 15:41:08 [NOTICE] voice.js:1 console_log() ready: Start DTMF 2009-02-17 15:41:22 [NOTICE] voice.js:1 console_log() ready: Start DTMF 2009-02-17 15:41:26 [NOTICE] voice.js:1 console_log() Playing message: message.wav 2009-02-17 15:41:26 [NOTICE] voice.js:1 console_log() Playing message: message.wav 2009-02-17 15:41:26 [NOTICE] voice.js:1 console_log() Playing message: message.wav That's 22 seconds for the first one!! Anyone any ideas as to what's going on here? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090217/b4c93f4f/attachment.html From anthony.minessale at gmail.com Tue Feb 17 10:34:00 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 17 Feb 2009 12:34:00 -0600 Subject: [Freeswitch-users] Big delays in playing audio files In-Reply-To: References: Message-ID: <191c3a030902171034o4ebd68f8w690092763019ced@mail.gmail.com> we would need to see your script. On Tue, Feb 17, 2009 at 12:23 PM, Nik Middleton < nik.middleton at noblesolutions.co.uk> wrote: > Having spent the last week developing a small js app, I ran some tests > today. With just 5 calls going on, I'm seeing huge delays from when the call > is answered to when the audio file is played. Sometimes it doesn't even > play at all!! > > > > Example 3 calls and the matching playbacks > > > > 2009-02-17 15:41:04 [NOTICE] voice.js:1 console_log() ready: Start DTMF > > 2009-02-17 15:41:08 [NOTICE] voice.js:1 console_log() ready: Start DTMF > > 2009-02-17 15:41:22 [NOTICE] voice.js:1 console_log() ready: Start DTMF > > > > 2009-02-17 15:41:26 [NOTICE] voice.js:1 console_log() Playing message: > message.wav > > 2009-02-17 15:41:26 [NOTICE] voice.js:1 console_log() Playing message: > message.wav > > 2009-02-17 15:41:26 [NOTICE] voice.js:1 console_log() Playing message: > message.wav > > > > That's 22 seconds for the first one!! > > > > Anyone any ideas as to what's going on here? > > > > Regards > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090217/d5651da8/attachment.html From nik.middleton at noblesolutions.co.uk Tue Feb 17 11:05:10 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Tue, 17 Feb 2009 19:05:10 -0000 Subject: [Freeswitch-users] Big delays in playing audio files In-Reply-To: <191c3a030902171034o4ebd68f8w690092763019ced@mail.gmail.com> References: <191c3a030902171034o4ebd68f8w690092763019ced@mail.gmail.com> Message-ID: if (first_session.ready()) { console_log("notice","Session state=[" + first_session.state + "] \n"); consoleLog("NOTICE", "ready: Start DTMF\n"); first_session.execute("start_dtmf"); first_session.answer( ); Disposition = "ANS"; first_session.sleep(1500); console_log("notice", "Playing message: " + recording + "\n"); first_session.streamFile(recording, on_event); if (first_session.ready()) { consoleLog("err", "ready: Waiting for input\n"); first_session.streamFile("4.wav",on_event, "dtmf"); consoleLog("err", "ready: Timeout on input\n"); first_session.execute("stop_tone_detect"); //disp_call() first_session.hangup() first_session.execute("sleep", "2000"); consoleLog("NOTICE", "EXITING\n"); exit(); } } ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: 17 February 2009 18:34 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Big delays in playing audio files we would need to see your script. On Tue, Feb 17, 2009 at 12:23 PM, Nik Middleton wrote: Having spent the last week developing a small js app, I ran some tests today. With just 5 calls going on, I'm seeing huge delays from when the call is answered to when the audio file is played. Sometimes it doesn't even play at all!! Example 3 calls and the matching playbacks 2009-02-17 15:41:04 [NOTICE] voice.js:1 console_log() ready: Start DTMF 2009-02-17 15:41:08 [NOTICE] voice.js:1 console_log() ready: Start DTMF 2009-02-17 15:41:22 [NOTICE] voice.js:1 console_log() ready: Start DTMF 2009-02-17 15:41:26 [NOTICE] voice.js:1 console_log() Playing message: message.wav 2009-02-17 15:41:26 [NOTICE] voice.js:1 console_log() Playing message: message.wav 2009-02-17 15:41:26 [NOTICE] voice.js:1 console_log() Playing message: message.wav That's 22 seconds for the first one!! Anyone any ideas as to what's going on here? Regards _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090217/3bb675ae/attachment-0001.html From msc at freeswitch.org Tue Feb 17 11:24:42 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 17 Feb 2009 11:24:42 -0800 Subject: [Freeswitch-users] Big delays in playing audio files In-Reply-To: References: <191c3a030902171034o4ebd68f8w690092763019ced@mail.gmail.com> Message-ID: <87f2f3b90902171124h5a5d4448mb872ca4aded94efa@mail.gmail.com> Is this the entire script?! -MC On Tue, Feb 17, 2009 at 11:05 AM, Nik Middleton wrote: > if (first_session.ready()) { > > console_log("notice","Session state=[" + > first_session.state + "] \n"); > > > > consoleLog("NOTICE", "ready: Start DTMF\n"); > > > > first_session.execute("start_dtmf"); > > first_session.answer( ); > > > > Disposition = "ANS"; > > > > first_session.sleep(1500); > > console_log("notice", "Playing message: " + > recording + "\n"); > > first_session.streamFile(recording, on_event); > > > > if (first_session.ready()) { > > consoleLog("err", "ready: Waiting for input\n"); > > first_session.streamFile("4.wav",on_event, "dtmf"); > > consoleLog("err", "ready: Timeout on input\n"); > > first_session.execute("stop_tone_detect"); > > > > //disp_call() > > first_session.hangup() > > first_session.execute("sleep", "2000"); > > consoleLog("NOTICE", "EXITING\n"); > > exit(); > > } > > } > > > > ________________________________ > > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony > Minessale > Sent: 17 February 2009 18:34 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Big delays in playing audio files > > > > we would need to see your script. > > On Tue, Feb 17, 2009 at 12:23 PM, Nik Middleton > wrote: > > Having spent the last week developing a small js app, I ran some tests > today. With just 5 calls going on, I'm seeing huge delays from when the call > is answered to when the audio file is played. Sometimes it doesn't even > play at all!! > > > > Example 3 calls and the matching playbacks > > > > 2009-02-17 15:41:04 [NOTICE] voice.js:1 console_log() ready: Start DTMF > > 2009-02-17 15:41:08 [NOTICE] voice.js:1 console_log() ready: Start DTMF > > 2009-02-17 15:41:22 [NOTICE] voice.js:1 console_log() ready: Start DTMF > > > > 2009-02-17 15:41:26 [NOTICE] voice.js:1 console_log() Playing message: > message.wav > > 2009-02-17 15:41:26 [NOTICE] voice.js:1 console_log() Playing message: > message.wav > > 2009-02-17 15:41:26 [NOTICE] voice.js:1 console_log() Playing message: > message.wav > > > > That's 22 seconds for the first one!! > > > > Anyone any ideas as to what's going on here? > > > > Regards > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From nik.middleton at noblesolutions.co.uk Tue Feb 17 12:11:19 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Tue, 17 Feb 2009 20:11:19 -0000 Subject: [Freeswitch-users] Big delays in playing audio files In-Reply-To: <87f2f3b90902171124h5a5d4448mb872ca4aded94efa@mail.gmail.com> References: <191c3a030902171034o4ebd68f8w690092763019ced@mail.gmail.com> <87f2f3b90902171124h5a5d4448mb872ca4aded94efa@mail.gmail.com> Message-ID: Pretty much I haven't included the on-event hooks as it never gets to the point where they're called. Only other thing is the dial it's self, attached below. However, I notice in the default dial plan, if I call extension 1001 from 1000 it takes about 2-3 seconds for the phone to ring. Is that normal? //build dial string var dial_string = "{absolute_codec_string=PCMA," + "accountcode=" + account_code + ",ignore_early_media=true" + " ,origination_caller_id_number=" + caller_id + ",originate_timeout=25}" + "sofia/gateway/" + "mygateway/" + dial_num + "' " var first_session = new Session(dial_string); // Trap for call failure if (!first_session.ready()) { consoleLog("err", "Disposition: " + first_session.cause + "\n"); if (first_session.cause == "USER_BUSY") { Disposition = "BUSY"; } else if (first_session.cause == "NO_ROUTE_DESTINATION") { Disposition = "DCN"; } else if (first_session.cause == "NO_ANSWER") { Disposition = "NA"; } disp_call() exit(); } //set the on_hangup function to be called when this session is hungup first_session.setHangupHook(on_hangup,"hup"); -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 17 February 2009 19:25 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Big delays in playing audio files Is this the entire script?! -MC On Tue, Feb 17, 2009 at 11:05 AM, Nik Middleton wrote: > if (first_session.ready()) { > > console_log("notice","Session state=[" + > first_session.state + "] \n"); > > > > consoleLog("NOTICE", "ready: Start DTMF\n"); > > > > first_session.execute("start_dtmf"); > > first_session.answer( ); > > > > Disposition = "ANS"; > > > > first_session.sleep(1500); > > console_log("notice", "Playing message: " + > recording + "\n"); > > first_session.streamFile(recording, on_event); > > > > if (first_session.ready()) { > > consoleLog("err", "ready: Waiting for input\n"); > > first_session.streamFile("4.wav",on_event, "dtmf"); > > consoleLog("err", "ready: Timeout on input\n"); > > first_session.execute("stop_tone_detect"); > > > > //disp_call() > > first_session.hangup() > > first_session.execute("sleep", "2000"); > > consoleLog("NOTICE", "EXITING\n"); > > exit(); > > } > > } > > > > ________________________________ > > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony > Minessale > Sent: 17 February 2009 18:34 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Big delays in playing audio files > > > > we would need to see your script. > > On Tue, Feb 17, 2009 at 12:23 PM, Nik Middleton > wrote: > > Having spent the last week developing a small js app, I ran some tests > today. With just 5 calls going on, I'm seeing huge delays from when the call > is answered to when the audio file is played. Sometimes it doesn't even > play at all!! > > > > Example 3 calls and the matching playbacks > > > > 2009-02-17 15:41:04 [NOTICE] voice.js:1 console_log() ready: Start DTMF > > 2009-02-17 15:41:08 [NOTICE] voice.js:1 console_log() ready: Start DTMF > > 2009-02-17 15:41:22 [NOTICE] voice.js:1 console_log() ready: Start DTMF > > > > 2009-02-17 15:41:26 [NOTICE] voice.js:1 console_log() Playing message: > message.wav > > 2009-02-17 15:41:26 [NOTICE] voice.js:1 console_log() Playing message: > message.wav > > 2009-02-17 15:41:26 [NOTICE] voice.js:1 console_log() Playing message: > message.wav > > > > That's 22 seconds for the first one!! > > > > Anyone any ideas as to what's going on here? > > > > Regards > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From kokoska.rokoska at post.cz Tue Feb 17 12:17:06 2009 From: kokoska.rokoska at post.cz (kokoska rokoska) Date: Tue, 17 Feb 2009 21:17:06 +0100 Subject: [Freeswitch-users] call goes to wrong context Message-ID: <499B1B42.7090000@post.cz> Hi all, I have just "upgraded" to current trunk (before an hour or so), configuration remain the same (served through mod_xml_curl), but something has changed and I don'nt know "where", "what" and "why" :-) What's going on: I have few sofia profiles and each of them has its own context. When call arrives, http POST is made by FreeSWITCH to get the dialplan, but Caller-Context and Hunt-Context variables are always set to "default" regardless what contex I set in sofia profile the call comes in through. May be I miss something, but really have no idea where :-) Any hint is very appreciated. Best regards, kokoska.rokoska From brian at freeswitch.org Tue Feb 17 12:19:36 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 17 Feb 2009 14:19:36 -0600 Subject: [Freeswitch-users] call goes to wrong context In-Reply-To: <499B1B42.7090000@post.cz> References: <499B1B42.7090000@post.cz> Message-ID: <4432BDD8-67A4-45B4-A013-7BB7C7E35D7F@freeswitch.org> Make sure on outbound registrations/gateways you have the context and extension params set. /b On Feb 17, 2009, at 2:17 PM, kokoska rokoska wrote: > > Hi all, > > I have just "upgraded" to current trunk (before an hour or so), > configuration remain the same (served through mod_xml_curl), but > something has changed and I don'nt know "where", "what" and "why" :-) > > What's going on: > I have few sofia profiles and each of them has its own context. When > call arrives, http POST is made by FreeSWITCH to get the dialplan, but > Caller-Context and Hunt-Context variables are always set to "default" > regardless what contex I set in sofia profile the call comes in > through. > > May be I miss something, but really have no idea where :-) > > Any hint is very appreciated. > > Best regards, > > kokoska.rokoska From kerrada2003 at yahoo.com Tue Feb 17 12:21:39 2009 From: kerrada2003 at yahoo.com (Ali Al-Rubaie) Date: Tue, 17 Feb 2009 12:21:39 -0800 (PST) Subject: [Freeswitch-users] Realm Value In-Reply-To: Message-ID: <326635.86195.qm@web33706.mail.mud.yahoo.com> I have to use a specific softphone, HelpCaster, but it can not pass the authentication stage. However it can authenticate with OpenSips server! What I had noticed is that it uses static realm with OpenSips therefore I'm trying to do the same. recv 292 bytes from udp/[209.82.10.250]:3458 at 16:35:24.758862: ?? ------------------------------------------------------------------------ ?? REGISTER sip:209.82.10.235 SIP/2.0 ?? Via: SIP/2.0/UDP 209.82.10.250:1059 ?? From: sip:1001 at 209.82.10.235 ?? To: sip:1001 at 209.82.10.235 ?? Contact: sip:1001 at 209.82.10.250:1059 ?? Call-ID: fc383368-e3e0-4c32-b87b-c3127d0cc2d8 at 192.168.10.23 ?? CSeq: 597775306 REGISTER ?? Content-Length: 0 ?? Expires: 3600 ? ?? ------------------------------------------------------------------------ send 582 bytes to udp/[209.82.10.250]:1059 at 16:35:24.763948: ?? ------------------------------------------------------------------------ ?? SIP/2.0 401 Unauthorized ?? Via: SIP/2.0/UDP 209.82.10.250:1059 ?? From: sip:1001 at 209.82.10.235 ?? To: ;tag=7yam2F01ZH3vH ?? Call-ID: fc383368-e3e0-4c32-b87b-c3127d0cc2d8 at 192.168.10.23 ?? CSeq: 597775306 REGISTER ?? User-Agent: FreeSWITCH-mod_sofia/1.0.2-hacked ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH ?? Supported: timer, precondition, path, replaces ?? WWW-Authenticate: Digest realm="209.82.10.235", nonce="40b63193-85c2-4ed9-874e-c03f81be313d", algorithm=MD5, qop="auth" ?? Content-Length: 0 ? ?? ------------------------------------------------------------------------ recv 466 bytes from udp/[209.82.10.250]:3458 at 16:35:24.772834: ?? ------------------------------------------------------------------------ ?? REGISTER sip:209.82.10.235 SIP/2.0 ?? Via: SIP/2.0/UDP 209.82.10.250:1059 ?? From: sip:1001 at 209.82.10.235 ?? To: sip:1001 at 209.82.10.235 ?? Contact: sip:1001 at 209.82.10.250:1059 ?? Call-ID: fc383368-e3e0-4c32-b87b-c3127d0cc2d8 at 192.168.10.23 ?? CSeq: 597775307 REGISTER ?? Content-Length: 0 ?? Expires: 3600 ?? Authorization: Digest username="1001",realm="209.82.10.235",nonce="40b63193-85c2-4ed9-874e-c03f81be313d",response="eebe0ea43319e82cc5f6dba5877de706",uri="sip:209.82.10.235" ? ?? ------------------------------------------------------------------------ send 458 bytes to udp/[209.82.10.250]:1059 at 16:35:24.774354: ?? ------------------------------------------------------------------------ ?? SIP/2.0 403 Forbidden ?? Via: SIP/2.0/UDP 209.82.10.250:1059 ?? From: sip:1001 at 209.82.10.235 ?? To: ;tag=873c4aH5vtSFD ?? Call-ID: fc383368-e3e0-4c32-b87b-c3127d0cc2d8 at 192.168.10.23 ?? CSeq: 597775307 REGISTER ?? User-Agent: FreeSWITCH-mod_sofia/1.0.2-hacked ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH ?? Supported: timer, precondition, path, replaces ?? Content-Length: 0 ? ?? ------------------------------------------------------------------------ Thanks! ? Message: 2 Date: Tue, 17 Feb 2009 11:39:58 -0600 From: Brian West Subject: Re: [Freeswitch-users] Realm Value To: freeswitch-users at lists.freeswitch.org Message-ID: <2ADB8FA0-3E87-4C99-9A6F-F09C6E0F5DB8 at freeswitch.org> Content-Type: text/plain; charset="us-ascii" I'm trying to get a clear picture of what you're trying to accomplish. Why would you need/want to set a static realm? Anyway can you collect sip traces? /b On Feb 17, 2009, at 8:51 AM, Ali Al-Rubaie wrote: > Thanks Brian, > > Actually we're using freeswitch ver 1.0.2. > > > Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090217/e801928a/attachment-0001.html -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090217/8a59c8fc/attachment-0001.html From msc at freeswitch.org Tue Feb 17 12:28:05 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 17 Feb 2009 12:28:05 -0800 Subject: [Freeswitch-users] Big delays in playing audio files In-Reply-To: References: <191c3a030902171034o4ebd68f8w690092763019ced@mail.gmail.com> <87f2f3b90902171124h5a5d4448mb872ca4aded94efa@mail.gmail.com> Message-ID: <87f2f3b90902171228u7ea87a57h7dab1738370c0970@mail.gmail.com> On Tue, Feb 17, 2009 at 12:11 PM, Nik Middleton wrote: > Pretty much > > I haven't included the on-event hooks as it never gets to the point > where they're called. > > Only other thing is the dial it's self, attached below. However, I > notice in the default dial plan, if I call extension 1001 from 1000 it > takes about 2-3 seconds for the phone to ring. Is that normal? By "ring" do you mean caller hears ringback tone or target phone audibly rings? I haven't noticed a long delay when dialing but I haven't been looking for one either. > > > //build dial string > var dial_string = "{absolute_codec_string=PCMA," + > "accountcode=" + account_code > + > ",ignore_early_media=true" > + > " > ,origination_caller_id_number=" + > caller_id > + > ",originate_timeout=25}" > + > "sofia/gateway/" > + > "mygateway/" > + > dial_num + "' " > > var first_session = new Session(dial_string); > > // Trap for call failure > if (!first_session.ready()) { > consoleLog("err", "Disposition: " + first_session.cause > + "\n"); > if (first_session.cause == "USER_BUSY") { > Disposition = "BUSY"; > } > else if (first_session.cause == > "NO_ROUTE_DESTINATION") { > Disposition = "DCN"; > } > > else if (first_session.cause == "NO_ANSWER") { > Disposition = "NA"; > } > > disp_call() > exit(); > } > > > //set the on_hangup function to be called when this session is > hungup > first_session.setHangupHook(on_hangup,"hup"); > > Can I just say how much I hate JavaScript for stuff like this? :) I can't test this right now but I will lab it up as soon as I can. In the meantime I would have you turn on debugging and capture the results from start to finish for a successful call. Save that for future reference. Then try to recreate the symptoms, also with debug turned on, capturing output. It will be A LOT of output, so you might want to consider rolling log files. (see the "Reporting Bugs" wiki page for hints on how to do that and more) Has anybody else out there used js for something like this, or otherwise have any input on why js seems to be acting up in this case? -MC From nik.middleton at noblesolutions.co.uk Tue Feb 17 12:30:02 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Tue, 17 Feb 2009 20:30:02 -0000 Subject: [Freeswitch-users] Big delays in playing audio files In-Reply-To: References: <191c3a030902171034o4ebd68f8w690092763019ced@mail.gmail.com><87f2f3b90902171124h5a5d4448mb872ca4aded94efa@mail.gmail.com> Message-ID: I'm starting to think it's a thread/DTMF issue. Ran 15 lines to my office number (using latest trunk) 2009-02-17 20:19:38 [CRIT] switch_core_state_machine.c:259 handle_fatality() Caught signal 11 for unmapped thread!Aborted (core dumped) Also then I had tone detect on, I'd often get this freeswitch: src/switch_ivr_async.c:1328: switch_ivr_tone_detect_session: Assertion `read_codec != ((void *)0)' failed. Hardware, HP DL360 G4. Centos 5.2, 4 GB ram. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Nik Middleton Sent: 17 February 2009 20:11 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Big delays in playing audio files Pretty much I haven't included the on-event hooks as it never gets to the point where they're called. Only other thing is the dial it's self, attached below. However, I notice in the default dial plan, if I call extension 1001 from 1000 it takes about 2-3 seconds for the phone to ring. Is that normal? //build dial string var dial_string = "{absolute_codec_string=PCMA," + "accountcode=" + account_code + ",ignore_early_media=true" + " ,origination_caller_id_number=" + caller_id + ",originate_timeout=25}" + "sofia/gateway/" + "mygateway/" + dial_num + "' " var first_session = new Session(dial_string); // Trap for call failure if (!first_session.ready()) { consoleLog("err", "Disposition: " + first_session.cause + "\n"); if (first_session.cause == "USER_BUSY") { Disposition = "BUSY"; } else if (first_session.cause == "NO_ROUTE_DESTINATION") { Disposition = "DCN"; } else if (first_session.cause == "NO_ANSWER") { Disposition = "NA"; } disp_call() exit(); } //set the on_hangup function to be called when this session is hungup first_session.setHangupHook(on_hangup,"hup"); -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 17 February 2009 19:25 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Big delays in playing audio files Is this the entire script?! -MC On Tue, Feb 17, 2009 at 11:05 AM, Nik Middleton wrote: > if (first_session.ready()) { > > console_log("notice","Session state=[" + > first_session.state + "] \n"); > > > > consoleLog("NOTICE", "ready: Start DTMF\n"); > > > > first_session.execute("start_dtmf"); > > first_session.answer( ); > > > > Disposition = "ANS"; > > > > first_session.sleep(1500); > > console_log("notice", "Playing message: " + > recording + "\n"); > > first_session.streamFile(recording, on_event); > > > > if (first_session.ready()) { > > consoleLog("err", "ready: Waiting for input\n"); > > first_session.streamFile("4.wav",on_event, "dtmf"); > > consoleLog("err", "ready: Timeout on input\n"); > > first_session.execute("stop_tone_detect"); > > > > //disp_call() > > first_session.hangup() > > first_session.execute("sleep", "2000"); > > consoleLog("NOTICE", "EXITING\n"); > > exit(); > > } > > } > > > > ________________________________ > > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony > Minessale > Sent: 17 February 2009 18:34 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Big delays in playing audio files > > > > we would need to see your script. > > On Tue, Feb 17, 2009 at 12:23 PM, Nik Middleton > wrote: > > Having spent the last week developing a small js app, I ran some tests > today. With just 5 calls going on, I'm seeing huge delays from when the call > is answered to when the audio file is played. Sometimes it doesn't even > play at all!! > > > > Example 3 calls and the matching playbacks > > > > 2009-02-17 15:41:04 [NOTICE] voice.js:1 console_log() ready: Start DTMF > > 2009-02-17 15:41:08 [NOTICE] voice.js:1 console_log() ready: Start DTMF > > 2009-02-17 15:41:22 [NOTICE] voice.js:1 console_log() ready: Start DTMF > > > > 2009-02-17 15:41:26 [NOTICE] voice.js:1 console_log() Playing message: > message.wav > > 2009-02-17 15:41:26 [NOTICE] voice.js:1 console_log() Playing message: > message.wav > > 2009-02-17 15:41:26 [NOTICE] voice.js:1 console_log() Playing message: > message.wav > > > > That's 22 seconds for the first one!! > > > > Anyone any ideas as to what's going on here? > > > > Regards > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From mark at markehle.net Tue Feb 17 12:33:21 2009 From: mark at markehle.net (Mark) Date: Tue, 17 Feb 2009 15:33:21 -0500 Subject: [Freeswitch-users] (OT) SPA-922 unlock In-Reply-To: References: <191c3a030902171034o4ebd68f8w690092763019ced@mail.gmail.com> <87f2f3b90902171124h5a5d4448mb872ca4aded94efa@mail.gmail.com> Message-ID: <20090217153321.16395204b8l8b604@markehle.net> Hello, folks - I hope that I can reach someone who knows the answer to this one: I bought 2 Linksys SPA-922 phones from a guy on ebay. The phones are locked by Webnet global Communications. From what I can tell, this company went bankrupt, and the ebay seller bought the phones from a bankruptcy auction. He does not know the admin username or password. Nowhere on the linksys site is there a solution to how to unlock these phones. Is there a way, or did I buy 2 interesting looking doorstops? Other than the password thing, they function fine. Thanks - Library Mark From msc at freeswitch.org Tue Feb 17 12:35:25 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 17 Feb 2009 12:35:25 -0800 Subject: [Freeswitch-users] Big delays in playing audio files In-Reply-To: References: <191c3a030902171034o4ebd68f8w690092763019ced@mail.gmail.com> <87f2f3b90902171124h5a5d4448mb872ca4aded94efa@mail.gmail.com> Message-ID: <87f2f3b90902171235k56ff0357vd4d0da45613f68ae@mail.gmail.com> okay, can you do the usual stuff and report a bug on jira? Not sure if it's really bug but having you collect all of the data and submit a bug report will assist us greatly. -MC On Tue, Feb 17, 2009 at 12:30 PM, Nik Middleton wrote: > I'm starting to think it's a thread/DTMF issue. Ran 15 lines to my > office number (using latest trunk) > > 2009-02-17 20:19:38 [CRIT] switch_core_state_machine.c:259 > handle_fatality() Caught signal 11 for unmapped thread!Aborted (core > dumped) > > Also then I had tone detect on, I'd often get this > > freeswitch: src/switch_ivr_async.c:1328: switch_ivr_tone_detect_session: > Assertion `read_codec != ((void *)0)' failed. > > Hardware, HP DL360 G4. Centos 5.2, 4 GB ram. > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Nik > Middleton > Sent: 17 February 2009 20:11 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Big delays in playing audio files > > Pretty much > > I haven't included the on-event hooks as it never gets to the point > where they're called. > > Only other thing is the dial it's self, attached below. However, I > notice in the default dial plan, if I call extension 1001 from 1000 it > takes about 2-3 seconds for the phone to ring. Is that normal? > > > //build dial string > var dial_string = "{absolute_codec_string=PCMA," + > "accountcode=" + account_code > + > ",ignore_early_media=true" > + > " > ,origination_caller_id_number=" + > caller_id > + > ",originate_timeout=25}" > + > "sofia/gateway/" > + > "mygateway/" > + > dial_num + "' " > > var first_session = new Session(dial_string); > > // Trap for call failure > if (!first_session.ready()) { > consoleLog("err", "Disposition: " + first_session.cause > + "\n"); > if (first_session.cause == "USER_BUSY") { > Disposition = "BUSY"; > } > else if (first_session.cause == > "NO_ROUTE_DESTINATION") { > Disposition = "DCN"; > } > > else if (first_session.cause == "NO_ANSWER") { > Disposition = "NA"; > } > > disp_call() > exit(); > } > > > //set the on_hangup function to be called when this session is > hungup > first_session.setHangupHook(on_hangup,"hup"); > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Michael Collins > Sent: 17 February 2009 19:25 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Big delays in playing audio files > > Is this the entire script?! > -MC > > On Tue, Feb 17, 2009 at 11:05 AM, Nik Middleton > wrote: >> if (first_session.ready()) { >> >> console_log("notice","Session state=[" + >> first_session.state + "] \n"); >> >> >> >> consoleLog("NOTICE", "ready: Start DTMF\n"); >> >> >> >> first_session.execute("start_dtmf"); >> >> first_session.answer( ); >> >> >> >> Disposition = "ANS"; >> >> >> >> first_session.sleep(1500); >> >> console_log("notice", "Playing message: " + >> recording + "\n"); >> >> first_session.streamFile(recording, on_event); >> >> >> >> if (first_session.ready()) { >> >> consoleLog("err", "ready: Waiting for > input\n"); >> >> first_session.streamFile("4.wav",on_event, > "dtmf"); >> >> consoleLog("err", "ready: Timeout on > input\n"); >> >> first_session.execute("stop_tone_detect"); >> >> >> >> //disp_call() >> >> first_session.hangup() >> >> first_session.execute("sleep", "2000"); >> >> consoleLog("NOTICE", "EXITING\n"); >> >> exit(); >> >> } >> >> } >> >> >> >> ________________________________ >> >> From: freeswitch-users-bounces at lists.freeswitch.org >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Anthony >> Minessale >> Sent: 17 February 2009 18:34 >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] Big delays in playing audio files >> >> >> >> we would need to see your script. >> >> On Tue, Feb 17, 2009 at 12:23 PM, Nik Middleton >> wrote: >> >> Having spent the last week developing a small js app, I ran some tests >> today. With just 5 calls going on, I'm seeing huge delays from when > the call >> is answered to when the audio file is played. Sometimes it doesn't > even >> play at all!! >> >> >> >> Example 3 calls and the matching playbacks >> >> >> >> 2009-02-17 15:41:04 [NOTICE] voice.js:1 console_log() ready: Start > DTMF >> >> 2009-02-17 15:41:08 [NOTICE] voice.js:1 console_log() ready: Start > DTMF >> >> 2009-02-17 15:41:22 [NOTICE] voice.js:1 console_log() ready: Start > DTMF >> >> >> >> 2009-02-17 15:41:26 [NOTICE] voice.js:1 console_log() Playing message: >> message.wav >> >> 2009-02-17 15:41:26 [NOTICE] voice.js:1 console_log() Playing message: >> message.wav >> >> 2009-02-17 15:41:26 [NOTICE] voice.js:1 console_log() Playing message: >> message.wav >> >> >> >> That's 22 seconds for the first one!! >> >> >> >> Anyone any ideas as to what's going on here? >> >> >> >> Regards >> >> >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:213-799-1400 >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From brian at freeswitch.org Tue Feb 17 12:36:51 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 17 Feb 2009 14:36:51 -0600 Subject: [Freeswitch-users] Realm Value In-Reply-To: <326635.86195.qm@web33706.mail.mud.yahoo.com> References: <326635.86195.qm@web33706.mail.mud.yahoo.com> Message-ID: <090382A2-AF83-4635-90CF-35749F50E0FA@freeswitch.org> Very sorry to hear you have to use Broken Software. But some good has come of this if you update to rev 12113 or great you'll be 100% OK. /b On Feb 17, 2009, at 2:21 PM, Ali Al-Rubaie wrote: > > I have to use a specific softphone, HelpCaster, but it can not pass > the authentication stage. However it can authenticate with OpenSips > server! What I had noticed is that it uses static realm with > OpenSips therefore I'm trying to do the same. From anthony.minessale at gmail.com Tue Feb 17 12:37:44 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 17 Feb 2009 14:37:44 -0600 Subject: [Freeswitch-users] Big delays in playing audio files In-Reply-To: References: <191c3a030902171034o4ebd68f8w690092763019ced@mail.gmail.com> <87f2f3b90902171124h5a5d4448mb872ca4aded94efa@mail.gmail.com> Message-ID: <191c3a030902171237w4dff4c45s4a10aa0deb0ece56@mail.gmail.com> 1) turn off crash protection. 2) you cant manipulate more that one call per script, design the script to be run from the application interface so you originate the call with the api interface and transfer the call to the script so each one has it's own copy of the script. On Tue, Feb 17, 2009 at 2:30 PM, Nik Middleton < nik.middleton at noblesolutions.co.uk> wrote: > I'm starting to think it's a thread/DTMF issue. Ran 15 lines to my > office number (using latest trunk) > > 2009-02-17 20:19:38 [CRIT] switch_core_state_machine.c:259 > handle_fatality() Caught signal 11 for unmapped thread!Aborted (core > dumped) > > Also then I had tone detect on, I'd often get this > > freeswitch: src/switch_ivr_async.c:1328: switch_ivr_tone_detect_session: > Assertion `read_codec != ((void *)0)' failed. > > Hardware, HP DL360 G4. Centos 5.2, 4 GB ram. > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Nik > Middleton > Sent: 17 February 2009 20:11 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Big delays in playing audio files > > Pretty much > > I haven't included the on-event hooks as it never gets to the point > where they're called. > > Only other thing is the dial it's self, attached below. However, I > notice in the default dial plan, if I call extension 1001 from 1000 it > takes about 2-3 seconds for the phone to ring. Is that normal? > > > //build dial string > var dial_string = "{absolute_codec_string=PCMA," + > "accountcode=" + account_code > + > ",ignore_early_media=true" > + > " > ,origination_caller_id_number=" + > caller_id > + > ",originate_timeout=25}" > + > "sofia/gateway/" > + > "mygateway/" > + > dial_num + "' " > > var first_session = new Session(dial_string); > > // Trap for call failure > if (!first_session.ready()) { > consoleLog("err", "Disposition: " + first_session.cause > + "\n"); > if (first_session.cause == "USER_BUSY") { > Disposition = "BUSY"; > } > else if (first_session.cause == > "NO_ROUTE_DESTINATION") { > Disposition = "DCN"; > } > > else if (first_session.cause == "NO_ANSWER") { > Disposition = "NA"; > } > > disp_call() > exit(); > } > > > //set the on_hangup function to be called when this session is > hungup > first_session.setHangupHook(on_hangup,"hup"); > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Michael Collins > Sent: 17 February 2009 19:25 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Big delays in playing audio files > > Is this the entire script?! > -MC > > On Tue, Feb 17, 2009 at 11:05 AM, Nik Middleton > wrote: > > if (first_session.ready()) { > > > > console_log("notice","Session state=[" + > > first_session.state + "] \n"); > > > > > > > > consoleLog("NOTICE", "ready: Start DTMF\n"); > > > > > > > > first_session.execute("start_dtmf"); > > > > first_session.answer( ); > > > > > > > > Disposition = "ANS"; > > > > > > > > first_session.sleep(1500); > > > > console_log("notice", "Playing message: " + > > recording + "\n"); > > > > first_session.streamFile(recording, on_event); > > > > > > > > if (first_session.ready()) { > > > > consoleLog("err", "ready: Waiting for > input\n"); > > > > first_session.streamFile("4.wav",on_event, > "dtmf"); > > > > consoleLog("err", "ready: Timeout on > input\n"); > > > > first_session.execute("stop_tone_detect"); > > > > > > > > //disp_call() > > > > first_session.hangup() > > > > first_session.execute("sleep", "2000"); > > > > consoleLog("NOTICE", "EXITING\n"); > > > > exit(); > > > > } > > > > } > > > > > > > > ________________________________ > > > > From: freeswitch-users-bounces at lists.freeswitch.org > > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Anthony > > Minessale > > Sent: 17 February 2009 18:34 > > To: freeswitch-users at lists.freeswitch.org > > Subject: Re: [Freeswitch-users] Big delays in playing audio files > > > > > > > > we would need to see your script. > > > > On Tue, Feb 17, 2009 at 12:23 PM, Nik Middleton > > wrote: > > > > Having spent the last week developing a small js app, I ran some tests > > today. With just 5 calls going on, I'm seeing huge delays from when > the call > > is answered to when the audio file is played. Sometimes it doesn't > even > > play at all!! > > > > > > > > Example 3 calls and the matching playbacks > > > > > > > > 2009-02-17 15:41:04 [NOTICE] voice.js:1 console_log() ready: Start > DTMF > > > > 2009-02-17 15:41:08 [NOTICE] voice.js:1 console_log() ready: Start > DTMF > > > > 2009-02-17 15:41:22 [NOTICE] voice.js:1 console_log() ready: Start > DTMF > > > > > > > > 2009-02-17 15:41:26 [NOTICE] voice.js:1 console_log() Playing message: > > message.wav > > > > 2009-02-17 15:41:26 [NOTICE] voice.js:1 console_log() Playing message: > > message.wav > > > > 2009-02-17 15:41:26 [NOTICE] voice.js:1 console_log() Playing message: > > message.wav > > > > > > > > That's 22 seconds for the first one!! > > > > > > > > Anyone any ideas as to what's going on here? > > > > > > > > Regards > > > > > > > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090217/fc8b7947/attachment-0001.html From nik.middleton at noblesolutions.co.uk Tue Feb 17 12:38:00 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Tue, 17 Feb 2009 20:38:00 -0000 Subject: [Freeswitch-users] Big delays in playing audio files In-Reply-To: <87f2f3b90902171228u7ea87a57h7dab1738370c0970@mail.gmail.com> References: <191c3a030902171034o4ebd68f8w690092763019ced@mail.gmail.com><87f2f3b90902171124h5a5d4448mb872ca4aded94efa@mail.gmail.com> <87f2f3b90902171228u7ea87a57h7dab1738370c0970@mail.gmail.com> Message-ID: I'm talking of the time when I hit the dial button to the phone 3 ft away starting to ring Regards, -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 17 February 2009 20:28 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Big delays in playing audio files On Tue, Feb 17, 2009 at 12:11 PM, Nik Middleton wrote: > Pretty much > > I haven't included the on-event hooks as it never gets to the point > where they're called. > > Only other thing is the dial it's self, attached below. However, I > notice in the default dial plan, if I call extension 1001 from 1000 it > takes about 2-3 seconds for the phone to ring. Is that normal? By "ring" do you mean caller hears ringback tone or target phone audibly rings? I haven't noticed a long delay when dialing but I haven't been looking for one either. > > > //build dial string > var dial_string = "{absolute_codec_string=PCMA," + > "accountcode=" + account_code > + > ",ignore_early_media=true" > + > " > ,origination_caller_id_number=" + > caller_id > + > ",originate_timeout=25}" > + > "sofia/gateway/" > + > "mygateway/" > + > dial_num + "' " > > var first_session = new Session(dial_string); > > // Trap for call failure > if (!first_session.ready()) { > consoleLog("err", "Disposition: " + first_session.cause > + "\n"); > if (first_session.cause == "USER_BUSY") { > Disposition = "BUSY"; > } > else if (first_session.cause == > "NO_ROUTE_DESTINATION") { > Disposition = "DCN"; > } > > else if (first_session.cause == "NO_ANSWER") { > Disposition = "NA"; > } > > disp_call() > exit(); > } > > > //set the on_hangup function to be called when this session is > hungup > first_session.setHangupHook(on_hangup,"hup"); > > Can I just say how much I hate JavaScript for stuff like this? :) I can't test this right now but I will lab it up as soon as I can. In the meantime I would have you turn on debugging and capture the results from start to finish for a successful call. Save that for future reference. Then try to recreate the symptoms, also with debug turned on, capturing output. It will be A LOT of output, so you might want to consider rolling log files. (see the "Reporting Bugs" wiki page for hints on how to do that and more) Has anybody else out there used js for something like this, or otherwise have any input on why js seems to be acting up in this case? -MC _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From brian at freeswitch.org Tue Feb 17 12:40:14 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 17 Feb 2009 14:40:14 -0600 Subject: [Freeswitch-users] (OT) SPA-922 unlock In-Reply-To: <20090217153321.16395204b8l8b604@markehle.net> References: <191c3a030902171034o4ebd68f8w690092763019ced@mail.gmail.com> <87f2f3b90902171124h5a5d4448mb872ca4aded94efa@mail.gmail.com> <20090217153321.16395204b8l8b604@markehle.net> Message-ID: <0A1321E0-8EDA-434C-BB47-009E0B8A0BBD@freeswitch.org> Mark, Sorry to say but I think you're pretty much SOL. I would check voipsupply or the like for a replacement. On a side note you hijacked the "Big delays in playing audio files" thread by clicking reply on one of those messages then changing the subject and body... in the future please try not to do this as it can cause your request to be overlooked depending on how the reader threads the messages. Maybe someone else has more input on your SPA's... Good Luck, /b On Feb 17, 2009, at 2:33 PM, Mark wrote: > Hello, folks - I hope that I can reach someone who knows the answer to > this one: > > I bought 2 Linksys SPA-922 phones from a guy on ebay. The phones are > locked by Webnet global Communications. From what I can tell, this > company went bankrupt, and the ebay seller bought the phones from a > bankruptcy auction. He does not know the admin username or password. > Nowhere on the linksys site is there a solution to how to unlock these > phones. > > Is there a way, or did I buy 2 interesting looking doorstops? Other > than the password thing, they function fine. > > Thanks - > > Library Mark From gkuri at ieee.org Tue Feb 17 12:41:42 2009 From: gkuri at ieee.org (Gabriel Kuri) Date: Tue, 17 Feb 2009 12:41:42 -0800 Subject: [Freeswitch-users] (OT) SPA-922 unlock In-Reply-To: <20090217153321.16395204b8l8b604@markehle.net> References: <191c3a030902171034o4ebd68f8w690092763019ced@mail.gmail.com> <87f2f3b90902171124h5a5d4448mb872ca4aded94efa@mail.gmail.com> <20090217153321.16395204b8l8b604@markehle.net> Message-ID: <499B2106.20102@ieee.org> Have you tried resetting the phone via the built-in IVR menu? Pick up the handset and dial ****73738# This should reset the phone to factory defaults, assuming that company didn't lock this feature out. Gabe Mark wrote: > Hello, folks - I hope that I can reach someone who knows the answer to > this one: > > I bought 2 Linksys SPA-922 phones from a guy on ebay. The phones are > locked by Webnet global Communications. From what I can tell, this > company went bankrupt, and the ebay seller bought the phones from a > bankruptcy auction. He does not know the admin username or password. > Nowhere on the linksys site is there a solution to how to unlock these > phones. > > Is there a way, or did I buy 2 interesting looking doorstops? Other > than the password thing, they function fine. > > Thanks - > > Library Mark > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mark at markehle.net Tue Feb 17 12:47:56 2009 From: mark at markehle.net (Mark) Date: Tue, 17 Feb 2009 15:47:56 -0500 Subject: [Freeswitch-users] (OT) SPA-922 unlock In-Reply-To: <499B2106.20102@ieee.org> References: <191c3a030902171034o4ebd68f8w690092763019ced@mail.gmail.com> <87f2f3b90902171124h5a5d4448mb872ca4aded94efa@mail.gmail.com> <20090217153321.16395204b8l8b604@markehle.net> <499B2106.20102@ieee.org> Message-ID: <20090217154756.744658d4l2acjnok@markehle.net> Sadly, ****73738# does not work. Is there a jumper on the board or some other hardware fix for this? Quoting "Gabriel Kuri" : > Have you tried resetting the phone via the built-in IVR menu? > > Pick up the handset and dial ****73738# > > This should reset the phone to factory defaults, assuming that company > didn't lock this feature out. > > Gabe > > > > Mark wrote: >> Hello, folks - I hope that I can reach someone who knows the answer to >> this one: >> >> I bought 2 Linksys SPA-922 phones from a guy on ebay. The phones are >> locked by Webnet global Communications. From what I can tell, this >> company went bankrupt, and the ebay seller bought the phones from a >> bankruptcy auction. He does not know the admin username or password. >> Nowhere on the linksys site is there a solution to how to unlock these >> phones. >> >> Is there a way, or did I buy 2 interesting looking doorstops? Other >> than the password thing, they function fine. >> >> Thanks - >> >> Library Mark >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From nik.middleton at noblesolutions.co.uk Tue Feb 17 12:48:51 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Tue, 17 Feb 2009 20:48:51 -0000 Subject: [Freeswitch-users] Big delays in playing audio files In-Reply-To: <191c3a030902171237w4dff4c45s4a10aa0deb0ece56@mail.gmail.com> References: <191c3a030902171034o4ebd68f8w690092763019ced@mail.gmail.com><87f2f3b90902171124h5a5d4448mb872ca4aded94efa@mail.gmail.com> <191c3a030902171237w4dff4c45s4a10aa0deb0ece56@mail.gmail.com> Message-ID: 1, OK, 2. Right now I have a php script calling bgapi via and event socket with the call parameters. Is that what you mean? If not, can you give me a pointer? I had assumed that every time I called bgapi it with the script in it, it would get it's own copy. Regards, ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: 17 February 2009 20:38 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Big delays in playing audio files 1) turn off crash protection. 2) you cant manipulate more that one call per script, design the script to be run from the application interface so you originate the call with the api interface and transfer the call to the script so each one has it's own copy of the script. On Tue, Feb 17, 2009 at 2:30 PM, Nik Middleton wrote: I'm starting to think it's a thread/DTMF issue. Ran 15 lines to my office number (using latest trunk) 2009-02-17 20:19:38 [CRIT] switch_core_state_machine.c:259 handle_fatality() Caught signal 11 for unmapped thread!Aborted (core dumped) Also then I had tone detect on, I'd often get this freeswitch: src/switch_ivr_async.c:1328: switch_ivr_tone_detect_session: Assertion `read_codec != ((void *)0)' failed. Hardware, HP DL360 G4. Centos 5.2, 4 GB ram. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Nik Middleton Sent: 17 February 2009 20:11 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Big delays in playing audio files Pretty much I haven't included the on-event hooks as it never gets to the point where they're called. Only other thing is the dial it's self, attached below. However, I notice in the default dial plan, if I call extension 1001 from 1000 it takes about 2-3 seconds for the phone to ring. Is that normal? //build dial string var dial_string = "{absolute_codec_string=PCMA," + "accountcode=" + account_code + ",ignore_early_media=true" + " ,origination_caller_id_number=" + caller_id + ",originate_timeout=25}" + "sofia/gateway/" + "mygateway/" + dial_num + "' " var first_session = new Session(dial_string); // Trap for call failure if (!first_session.ready()) { consoleLog("err", "Disposition: " + first_session.cause + "\n"); if (first_session.cause == "USER_BUSY") { Disposition = "BUSY"; } else if (first_session.cause == "NO_ROUTE_DESTINATION") { Disposition = "DCN"; } else if (first_session.cause == "NO_ANSWER") { Disposition = "NA"; } disp_call() exit(); } //set the on_hangup function to be called when this session is hungup first_session.setHangupHook(on_hangup,"hup"); -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 17 February 2009 19:25 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Big delays in playing audio files Is this the entire script?! -MC On Tue, Feb 17, 2009 at 11:05 AM, Nik Middleton wrote: > if (first_session.ready()) { > > console_log("notice","Session state=[" + > first_session.state + "] \n"); > > > > consoleLog("NOTICE", "ready: Start DTMF\n"); > > > > first_session.execute("start_dtmf"); > > first_session.answer( ); > > > > Disposition = "ANS"; > > > > first_session.sleep(1500); > > console_log("notice", "Playing message: " + > recording + "\n"); > > first_session.streamFile(recording, on_event); > > > > if (first_session.ready()) { > > consoleLog("err", "ready: Waiting for input\n"); > > first_session.streamFile("4.wav",on_event, "dtmf"); > > consoleLog("err", "ready: Timeout on input\n"); > > first_session.execute("stop_tone_detect"); > > > > //disp_call() > > first_session.hangup() > > first_session.execute("sleep", "2000"); > > consoleLog("NOTICE", "EXITING\n"); > > exit(); > > } > > } > > > > ________________________________ > > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony > Minessale > Sent: 17 February 2009 18:34 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Big delays in playing audio files > > > > we would need to see your script. > > On Tue, Feb 17, 2009 at 12:23 PM, Nik Middleton > wrote: > > Having spent the last week developing a small js app, I ran some tests > today. With just 5 calls going on, I'm seeing huge delays from when the call > is answered to when the audio file is played. Sometimes it doesn't even > play at all!! > > > > Example 3 calls and the matching playbacks > > > > 2009-02-17 15:41:04 [NOTICE] voice.js:1 console_log() ready: Start DTMF > > 2009-02-17 15:41:08 [NOTICE] voice.js:1 console_log() ready: Start DTMF > > 2009-02-17 15:41:22 [NOTICE] voice.js:1 console_log() ready: Start DTMF > > > > 2009-02-17 15:41:26 [NOTICE] voice.js:1 console_log() Playing message: > message.wav > > 2009-02-17 15:41:26 [NOTICE] voice.js:1 console_log() Playing message: > message.wav > > 2009-02-17 15:41:26 [NOTICE] voice.js:1 console_log() Playing message: > message.wav > > > > That's 22 seconds for the first one!! > > > > Anyone any ideas as to what's going on here? > > > > Regards > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090217/622f060e/attachment-0001.html From msc at freeswitch.org Tue Feb 17 12:56:41 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 17 Feb 2009 12:56:41 -0800 Subject: [Freeswitch-users] Big delays in playing audio files In-Reply-To: References: <191c3a030902171034o4ebd68f8w690092763019ced@mail.gmail.com> <87f2f3b90902171124h5a5d4448mb872ca4aded94efa@mail.gmail.com> <191c3a030902171237w4dff4c45s4a10aa0deb0ece56@mail.gmail.com> Message-ID: <87f2f3b90902171256m3d4da7efo9189794d29a87aa3@mail.gmail.com> On Tue, Feb 17, 2009 at 12:48 PM, Nik Middleton wrote: > 1, OK, > > > > 2. Right now I have a php script calling bgapi via and event socket with the > call parameters. Is that what you mean? If not, can you give me a pointer? > I had assumed that every time I called bgapi it with the script in it, it > would get it's own copy. yes, bgapi counts as an API call. Ken Rice thinks this might be related to a spidermonky concurrency issue... -MC From gkuri at ieee.org Tue Feb 17 12:58:08 2009 From: gkuri at ieee.org (Gabriel Kuri) Date: Tue, 17 Feb 2009 12:58:08 -0800 Subject: [Freeswitch-users] (OT) SPA-922 unlock In-Reply-To: <20090217154756.744658d4l2acjnok@markehle.net> References: <191c3a030902171034o4ebd68f8w690092763019ced@mail.gmail.com> <87f2f3b90902171124h5a5d4448mb872ca4aded94efa@mail.gmail.com> <20090217153321.16395204b8l8b604@markehle.net> <499B2106.20102@ieee.org> <20090217154756.744658d4l2acjnok@markehle.net> Message-ID: <499B24E0.4080204@ieee.org> I believe you need to make sure the Ethernet cable is unplugged from the phone when trying to dial that string. Now I've never tried this, but it should theoretically be possible ... Sniff the traffic of the phone and see where it's attempting to pickup the config file. Then setup a local network with your own DNS server, and re-direct the phone (via DNS) to your own web server (assuming it's picking up the config via http) and have a config file on the web server with a username and password you specify to reset the config and get into the phone. Let's hope they didn't setup the phone to provision via https, otherwise you're really SOL If you need help generating a config for the phone, with Linksys' special config tool, contact me offlist. Gabe Mark wrote: > Sadly, ****73738# does not work. > > Is there a jumper on the board or some other hardware fix for this? > > Quoting "Gabriel Kuri" : > >> Have you tried resetting the phone via the built-in IVR menu? >> >> Pick up the handset and dial ****73738# >> >> This should reset the phone to factory defaults, assuming that company >> didn't lock this feature out. >> >> Gabe >> >> >> >> Mark wrote: >>> Hello, folks - I hope that I can reach someone who knows the answer to >>> this one: >>> >>> I bought 2 Linksys SPA-922 phones from a guy on ebay. The phones are >>> locked by Webnet global Communications. From what I can tell, this >>> company went bankrupt, and the ebay seller bought the phones from a >>> bankruptcy auction. He does not know the admin username or password. >>> Nowhere on the linksys site is there a solution to how to unlock these >>> phones. >>> >>> Is there a way, or did I buy 2 interesting looking doorstops? Other >>> than the password thing, they function fine. >>> >>> Thanks - >>> >>> Library Mark >>> >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Tue Feb 17 13:00:20 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 17 Feb 2009 15:00:20 -0600 Subject: [Freeswitch-users] (OT) SPA-922 unlock In-Reply-To: <499B24E0.4080204@ieee.org> References: <191c3a030902171034o4ebd68f8w690092763019ced@mail.gmail.com> <87f2f3b90902171124h5a5d4448mb872ca4aded94efa@mail.gmail.com> <20090217153321.16395204b8l8b604@markehle.net> <499B2106.20102@ieee.org> <20090217154756.744658d4l2acjnok@markehle.net> <499B24E0.4080204@ieee.org> Message-ID: <8D48CE5C-DEF9-4B49-9B2F-BE189C9F95EA@freeswitch.org> Don't they cryptographically sign the config also? /b On Feb 17, 2009, at 2:58 PM, Gabriel Kuri wrote: > If you need help generating a config for the phone, with Linksys' > special config tool, contact me offlist. > > Gabe From kokoska.rokoska at post.cz Tue Feb 17 13:05:21 2009 From: kokoska.rokoska at post.cz (kokoska.rokoska) Date: Tue, 17 Feb 2009 22:05:21 +0100 Subject: [Freeswitch-users] call goes to wrong context In-Reply-To: <4432BDD8-67A4-45B4-A013-7BB7C7E35D7F@freeswitch.org> References: <499B1B42.7090000@post.cz> <4432BDD8-67A4-45B4-A013-7BB7C7E35D7F@freeswitch.org> Message-ID: <499B2691.70003@post.cz> Brian West napsal(a): > Make sure on outbound registrations/gateways you have the context and > extension params set. > Thank you very much, Brian, for your suggestion! I had context defined on all sofia profiles, but I didn't have extension param set on gateways (but it works till I upgraded to current FS). So I add the extension param to all gateways, but I doesn't help. mod_xml_curl still asks for the "default" context instead of the context defined in sofia profile... If you have more hints, I be very happy :-) Best regards, kokoska.rokoska From brian at freeswitch.org Tue Feb 17 13:09:18 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 17 Feb 2009 15:09:18 -0600 Subject: [Freeswitch-users] call goes to wrong context In-Reply-To: <499B2691.70003@post.cz> References: <499B1B42.7090000@post.cz> <4432BDD8-67A4-45B4-A013-7BB7C7E35D7F@freeswitch.org> <499B2691.70003@post.cz> Message-ID: No problem. Just join us on IRC.. things move faster on there. /b On Feb 17, 2009, at 3:05 PM, kokoska.rokoska wrote: > If you have more hints, I be very happy :-) From mark at markehle.net Tue Feb 17 13:10:22 2009 From: mark at markehle.net (Mark) Date: Tue, 17 Feb 2009 16:10:22 -0500 Subject: [Freeswitch-users] (OT) SPA-922 unlock In-Reply-To: <499B24E0.4080204@ieee.org> References: <191c3a030902171034o4ebd68f8w690092763019ced@mail.gmail.com> <87f2f3b90902171124h5a5d4448mb872ca4aded94efa@mail.gmail.com> <20090217153321.16395204b8l8b604@markehle.net> <499B2106.20102@ieee.org> <20090217154756.744658d4l2acjnok@markehle.net> <499B24E0.4080204@ieee.org> Message-ID: <20090217161022.13903dpm9s716484@markehle.net> I did unplug the ethernet cable. I have never been able to make the IVR work on any of the Linksys phones that I have. I must be doing something wrong. I will try to sniff the traffic on the phone when I start it up. I will report back when I do. Thanks so much - Library Mark Quoting "Gabriel Kuri" : > I believe you need to make sure the Ethernet cable is unplugged from the > phone when trying to dial that string. > > Now I've never tried this, but it should theoretically be possible ... > > Sniff the traffic of the phone and see where it's attempting to pickup > the config file. Then setup a local network with your own DNS server, > and re-direct the phone (via DNS) to your own web server (assuming it's > picking up the config via http) and have a config file on the web server > with a username and password you specify to reset the config and get > into the phone. Let's hope they didn't setup the phone to provision via > https, otherwise you're really SOL > > If you need help generating a config for the phone, with Linksys' > special config tool, contact me offlist. > > Gabe > > Mark wrote: >> Sadly, ****73738# does not work. >> >> Is there a jumper on the board or some other hardware fix for this? >> >> Quoting "Gabriel Kuri" : >> >>> Have you tried resetting the phone via the built-in IVR menu? >>> >>> Pick up the handset and dial ****73738# >>> >>> This should reset the phone to factory defaults, assuming that company >>> didn't lock this feature out. >>> >>> Gabe >>> >>> >>> >>> Mark wrote: >>>> Hello, folks - I hope that I can reach someone who knows the answer to >>>> this one: >>>> >>>> I bought 2 Linksys SPA-922 phones from a guy on ebay. The phones are >>>> locked by Webnet global Communications. From what I can tell, this >>>> company went bankrupt, and the ebay seller bought the phones from a >>>> bankruptcy auction. He does not know the admin username or password. >>>> Nowhere on the linksys site is there a solution to how to unlock these >>>> phones. >>>> >>>> Is there a way, or did I buy 2 interesting looking doorstops? Other >>>> than the password thing, they function fine. >>>> >>>> Thanks - >>>> >>>> Library Mark >>>> >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From ivan at myrvold.org Tue Feb 17 13:19:58 2009 From: ivan at myrvold.org (Ivan C Myrvold) Date: Tue, 17 Feb 2009 22:19:58 +0100 Subject: [Freeswitch-users] Skypiax on OS X Message-ID: s it possible to run Skypiax on OS X? The wiki says Linux and Windows, but says nothing about OS X. I have been running FreeSWITCH on OS X for a couple of years now, and love it. Adding Skype gateway would be really sweet. Are there any plans for adding Skypiax to trunk, or do we have to build it only from svn branch? Ivan From nik.middleton at noblesolutions.co.uk Tue Feb 17 13:24:23 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Tue, 17 Feb 2009 21:24:23 -0000 Subject: [Freeswitch-users] Big delays in playing audio files In-Reply-To: <87f2f3b90902171256m3d4da7efo9189794d29a87aa3@mail.gmail.com> References: <191c3a030902171034o4ebd68f8w690092763019ced@mail.gmail.com><87f2f3b90902171124h5a5d4448mb872ca4aded94efa@mail.gmail.com><191c3a030902171237w4dff4c45s4a10aa0deb0ece56@mail.gmail.com> <87f2f3b90902171256m3d4da7efo9189794d29a87aa3@mail.gmail.com> Message-ID: > yes, bgapi counts as an API call. Ken Rice thinks this might be > related to a spidermonky concurrency issue... Well that kinda fits, as I see the audio files stacking up, seems like they're being queued. Question is, what's my alternative, lua? -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 17 February 2009 20:57 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Big delays in playing audio files On Tue, Feb 17, 2009 at 12:48 PM, Nik Middleton wrote: > 1, OK, > > > > 2. Right now I have a php script calling bgapi via and event socket with the > call parameters. Is that what you mean? If not, can you give me a pointer? > I had assumed that every time I called bgapi it with the script in it, it > would get it's own copy. From lfurrea at gmail.com Tue Feb 17 13:26:35 2009 From: lfurrea at gmail.com (Luis F Urrea) Date: Tue, 17 Feb 2009 15:26:35 -0600 Subject: [Freeswitch-users] xml_cdr Call Flow for attended transfer In-Reply-To: References: Message-ID: any hints? what would be the best way to report CDRs for attended transfers?? We are using C with libxml to create a binary that can be called from a script to rotate xml_cdrs and insert them on SQLite and would gladly submit the code to your revision, advice and maybe even potential use. I appreciate your advice. On Mon, Feb 16, 2009 at 5:52 PM, Luis F Urrea wrote: > Hi all, > > > I am trying to understand xml_cdr for an attended (consultative) transfer, > I was thinking that the A-leg that initially > originated the call would remain untouched but I see that it's global tags > get replaced. > > I have a test call that goes as follows: > > 201 originates a call and talks to 203 -----> A-leg(1) and > B-leg(1) > > 203 puts 201 on hold and calls 202 (attended) ------> A-leg(2) and > B-leg(2) > > 203 transfers the call > > 201 and 202 are talking ------> > A-leg(1) w/ B-leg(2) ??? > > Here are the relevant captures: > > A-leg(1) > http://pastebin.freeswitch.org/7253 > > B-leg(1) > http://pastebin.freeswitch.org/7254 > > A-leg(2) > http://pastebin.freeswitch.org/7252 > > B-leg(2) > http://pastebin.freeswitch.org/7255 > > I was expecting A-leg(1) to have corresponding to 201 which is > the original A-leg but it seems that on the transfer, it reverts and 202 > appears as the A-leg and 201 as the B-leg. > > Can someone shed some light on how that transfer gets logged in terms of > A-leg and B-leg? > > TIA > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090217/940eb515/attachment-0001.html From intralanman at freeswitch.org Tue Feb 17 13:27:04 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Tue, 17 Feb 2009 16:27:04 -0500 Subject: [Freeswitch-users] (OT) SPA-922 unlock In-Reply-To: <8D48CE5C-DEF9-4B49-9B2F-BE189C9F95EA@freeswitch.org> References: <191c3a030902171034o4ebd68f8w690092763019ced@mail.gmail.com> <87f2f3b90902171124h5a5d4448mb872ca4aded94efa@mail.gmail.com> <20090217153321.16395204b8l8b604@markehle.net> <499B2106.20102@ieee.org> <20090217154756.744658d4l2acjnok@markehle.net> <499B24E0.4080204@ieee.org> <8D48CE5C-DEF9-4B49-9B2F-BE189C9F95EA@freeswitch.org> Message-ID: <499B2BA8.1070509@freeswitch.org> Brian West wrote: > Don't they cryptographically sign the config also? > > it's an option in the device... some providers do, some don't. but it shouldn't matter too much if they're using https or not, as long as the ata doesn't authenticate via certificate or something. -Ray From msc at freeswitch.org Tue Feb 17 13:30:10 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 17 Feb 2009 13:30:10 -0800 Subject: [Freeswitch-users] Big delays in playing audio files In-Reply-To: References: <191c3a030902171034o4ebd68f8w690092763019ced@mail.gmail.com> <87f2f3b90902171124h5a5d4448mb872ca4aded94efa@mail.gmail.com> <191c3a030902171237w4dff4c45s4a10aa0deb0ece56@mail.gmail.com> <87f2f3b90902171256m3d4da7efo9189794d29a87aa3@mail.gmail.com> Message-ID: <87f2f3b90902171330v181cd5dej79c1e7b95fb599a5@mail.gmail.com> >> yes, bgapi counts as an API call. Ken Rice thinks this might be >> related to a spidermonky concurrency issue... > > Well that kinda fits, as I see the audio files stacking up, seems like > they're being queued. > > Question is, what's my alternative, lua? > Lua or C/C++, but Lua is the consensus for quick and easy. If you join IRC you can ask user "hmmhesays" about his experiences. He's got like 400+ concurrent calls with Lua scripts. -MC From brian at freeswitch.org Tue Feb 17 13:31:26 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 17 Feb 2009 15:31:26 -0600 Subject: [Freeswitch-users] Skypiax on OS X In-Reply-To: References: Message-ID: I think Ctrix is working on mod_airpe in his branch for OS X. /b On Feb 17, 2009, at 3:19 PM, Ivan C Myrvold wrote: > s it possible to run Skypiax on OS X? The wiki says Linux and Windows, > but says nothing about OS X. > I have been running FreeSWITCH on OS X for a couple of years now, and > love it. Adding Skype gateway would be really sweet. > > Are there any plans for adding Skypiax to trunk, or do we have to > build it only from svn branch? > > Ivan From gkuri at ieee.org Tue Feb 17 13:45:36 2009 From: gkuri at ieee.org (Gabriel Kuri) Date: Tue, 17 Feb 2009 13:45:36 -0800 Subject: [Freeswitch-users] (OT) SPA-922 unlock In-Reply-To: <8D48CE5C-DEF9-4B49-9B2F-BE189C9F95EA@freeswitch.org> References: <191c3a030902171034o4ebd68f8w690092763019ced@mail.gmail.com> <87f2f3b90902171124h5a5d4448mb872ca4aded94efa@mail.gmail.com> <20090217153321.16395204b8l8b604@markehle.net> <499B2106.20102@ieee.org> <20090217154756.744658d4l2acjnok@markehle.net> <499B24E0.4080204@ieee.org> <8D48CE5C-DEF9-4B49-9B2F-BE189C9F95EA@freeswitch.org> Message-ID: <499B3000.10306@ieee.org> It depends on the whether you pass the option to the Linksys/Cisco Profile Compiler to generate the config file. In any case, that shouldn't be an issue. Gabe Brian West wrote: > Don't they cryptographically sign the config also? > > /b > > On Feb 17, 2009, at 2:58 PM, Gabriel Kuri wrote: > >> If you need help generating a config for the phone, with Linksys' >> special config tool, contact me offlist. >> >> Gabe > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From gkuri at ieee.org Tue Feb 17 13:51:09 2009 From: gkuri at ieee.org (Gabriel Kuri) Date: Tue, 17 Feb 2009 13:51:09 -0800 Subject: [Freeswitch-users] (OT) SPA-922 unlock In-Reply-To: <20090217161022.13903dpm9s716484@markehle.net> References: <191c3a030902171034o4ebd68f8w690092763019ced@mail.gmail.com> <87f2f3b90902171124h5a5d4448mb872ca4aded94efa@mail.gmail.com> <20090217153321.16395204b8l8b604@markehle.net> <499B2106.20102@ieee.org> <20090217154756.744658d4l2acjnok@markehle.net> <499B24E0.4080204@ieee.org> <20090217161022.13903dpm9s716484@markehle.net> Message-ID: <499B314D.2030302@ieee.org> On the slight chance they're not doing remote provisioning and the phone is just simply locked with a username/password, you'll need to feed the phone a TFTP server via DHCP Option 66 and setup a config file on that tftp server with the name spa922.cfg. Contact me off list about generating a config file for the phone. Gabe Mark wrote: > I did unplug the ethernet cable. I have never been able to make the > IVR work on any of the Linksys phones that I have. I must be doing > something wrong. > > I will try to sniff the traffic on the phone when I start it up. I > will report back when I do. > > Thanks so much - > > Library Mark > > Quoting "Gabriel Kuri" : > >> I believe you need to make sure the Ethernet cable is unplugged from the >> phone when trying to dial that string. >> >> Now I've never tried this, but it should theoretically be possible ... >> >> Sniff the traffic of the phone and see where it's attempting to pickup >> the config file. Then setup a local network with your own DNS server, >> and re-direct the phone (via DNS) to your own web server (assuming it's >> picking up the config via http) and have a config file on the web server >> with a username and password you specify to reset the config and get >> into the phone. Let's hope they didn't setup the phone to provision via >> https, otherwise you're really SOL >> >> If you need help generating a config for the phone, with Linksys' >> special config tool, contact me offlist. >> >> Gabe >> >> Mark wrote: >>> Sadly, ****73738# does not work. >>> >>> Is there a jumper on the board or some other hardware fix for this? >>> >>> Quoting "Gabriel Kuri" : >>> >>>> Have you tried resetting the phone via the built-in IVR menu? >>>> >>>> Pick up the handset and dial ****73738# >>>> >>>> This should reset the phone to factory defaults, assuming that company >>>> didn't lock this feature out. >>>> >>>> Gabe >>>> >>>> >>>> >>>> Mark wrote: >>>>> Hello, folks - I hope that I can reach someone who knows the answer to >>>>> this one: >>>>> >>>>> I bought 2 Linksys SPA-922 phones from a guy on ebay. The phones are >>>>> locked by Webnet global Communications. From what I can tell, this >>>>> company went bankrupt, and the ebay seller bought the phones from a >>>>> bankruptcy auction. He does not know the admin username or password. >>>>> Nowhere on the linksys site is there a solution to how to unlock these >>>>> phones. >>>>> >>>>> Is there a way, or did I buy 2 interesting looking doorstops? Other >>>>> than the password thing, they function fine. >>>>> >>>>> Thanks - >>>>> >>>>> Library Mark >>>>> >>>>> >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From gkuri at ieee.org Tue Feb 17 13:55:25 2009 From: gkuri at ieee.org (Gabriel Kuri) Date: Tue, 17 Feb 2009 13:55:25 -0800 Subject: [Freeswitch-users] (OT) SPA-922 unlock In-Reply-To: <499B2BA8.1070509@freeswitch.org> References: <191c3a030902171034o4ebd68f8w690092763019ced@mail.gmail.com> <87f2f3b90902171124h5a5d4448mb872ca4aded94efa@mail.gmail.com> <20090217153321.16395204b8l8b604@markehle.net> <499B2106.20102@ieee.org> <20090217154756.744658d4l2acjnok@markehle.net> <499B24E0.4080204@ieee.org> <8D48CE5C-DEF9-4B49-9B2F-BE189C9F95EA@freeswitch.org> <499B2BA8.1070509@freeswitch.org> Message-ID: <499B324D.8010603@ieee.org> I'm about 99% positive that if https is enabled for remote provisioning, the web server needs an SSL certificate signed by the Linksys Enterprise CA, otherwise the phone will reject it. Gabe > but it shouldn't matter too much if they're using https or not, as long > as the ata doesn't authenticate via certificate or something. From freeswitch-users at digitaldan.com Tue Feb 17 14:40:06 2009 From: freeswitch-users at digitaldan.com (Dan) Date: Tue, 17 Feb 2009 15:40:06 -0700 (MST) Subject: [Freeswitch-users] Debian rules Message-ID: <1496906.31234910406670.JavaMail.root@zimbra> Hi guys, I noticed that the debian build is missing lines for shout.conf.xml and does not install mod_flite (if its built) . This can be fixed by adding the following lines to debian/freeswitch.install opt/freeswitch/mod/mod_flite* opt/freeswitch/conf/autoload_configs/shout.conf.xml and the following lines to debian/freeswitch.conffiles /opt/freeswitch/conf/autoload_configs/shout.conf.xml thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090217/60597ae6/attachment.html From brian at freeswitch.org Tue Feb 17 14:51:37 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 17 Feb 2009 16:51:37 -0600 Subject: [Freeswitch-users] Debian rules In-Reply-To: <1496906.31234910406670.JavaMail.root@zimbra> References: <1496906.31234910406670.JavaMail.root@zimbra> Message-ID: Please submit all patches and changes via jira if possible http://jira.freeswitch.org Thanks, Brian On Feb 17, 2009, at 4:40 PM, Dan wrote: > Hi guys, > I noticed that the debian build is missing lines for shout.conf.xml > and does not install mod_flite (if its built) . This can be fixed > by adding the following lines to debian/freeswitch.install > > opt/freeswitch/mod/mod_flite* > opt/freeswitch/conf/autoload_configs/shout.conf.xml > > and the following lines to debian/freeswitch.conffiles > > /opt/freeswitch/conf/autoload_configs/shout.conf.xml > > thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090217/844b6886/attachment.html From msc at freeswitch.org Tue Feb 17 14:57:54 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 17 Feb 2009 14:57:54 -0800 Subject: [Freeswitch-users] Debian rules In-Reply-To: <1496906.31234910406670.JavaMail.root@zimbra> References: <1496906.31234910406670.JavaMail.root@zimbra> Message-ID: <87f2f3b90902171457h37c4790crf8fc6ea78d06f0d3@mail.gmail.com> Oh, and thanks for the info! -MC From nik.middleton at noblesolutions.co.uk Tue Feb 17 15:11:48 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Tue, 17 Feb 2009 23:11:48 -0000 Subject: [Freeswitch-users] AddBody to events in lua Message-ID: Hi Guys, I'm having real problems doing something trivial, and there doesn't seem to be any docs on this issue In js I do this //Disposition = disp; //Create Custom event custom_msg = "call_disposition: " + Disposition + "\n" + "called_number: " + dial_num + "\n" ; e = new Event("custom", "dialer::dialer-result"); e.addBody(custom_msg); e.fire(); And it works In lua I try this --Disposition = disp; --Create Custom event custom_msg = "call_disposition: " .. Disposition .. "\n" .. "called_number: " .. dial_num .."\n" ; local e = freeswitch.Event("custom", "dialer::dialer-result"); e.addBody(custom_msg); e:fire(e); This doesn't work, I get an error : Error in addBody expected 2..2 args, got 1 What are the arguments? It seems to be looking for a pointer for the first one, but there's nothing on the wiki on this. Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090217/223bd566/attachment.html From freeswitch-users at digitaldan.com Tue Feb 17 15:17:52 2009 From: freeswitch-users at digitaldan.com (Dan) Date: Tue, 17 Feb 2009 16:17:52 -0700 (MST) Subject: [Freeswitch-users] Debian rules In-Reply-To: <87f2f3b90902171457h37c4790crf8fc6ea78d06f0d3@mail.gmail.com> Message-ID: <1141554.61234912672229.JavaMail.root@zimbra> no problem, its now in jira, key fsbuild-124. D- ----- Original Message ----- From: "Michael Collins" To: freeswitch-users at lists.freeswitch.org Sent: Tuesday, February 17, 2009 3:57:54 PM GMT -07:00 US/Canada Mountain Subject: Re: [Freeswitch-users] Debian rules Oh, and thanks for the info! -MC _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090217/06fdfeb2/attachment.html From msc at freeswitch.org Tue Feb 17 15:23:13 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 17 Feb 2009 15:23:13 -0800 Subject: [Freeswitch-users] AddBody to events in lua In-Reply-To: References: Message-ID: <87f2f3b90902171523o5e014ce9w4fae9f05f378b861@mail.gmail.com> > local e = freeswitch.Event("custom", > "dialer::dialer-result"); > > e.addBody(custom_msg); > > e:fire(e); The wiki page (http://wiki.freeswitch.org/wiki/Lua#event:fire) shows that you fire thusly: e:fire(); --No "e" in the parens. Can you try it and report back? -MC From anthony.minessale at gmail.com Tue Feb 17 15:25:11 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 17 Feb 2009 17:25:11 -0600 Subject: [Freeswitch-users] AddBody to events in lua In-Reply-To: References: Message-ID: <191c3a030902171525y17166787ud37ab5243c287008@mail.gmail.com> in lua you call methods with a colon : e:addBody(blah); calling with a . implies you are going to supply the obj too e.addBody(e, blah); On Tue, Feb 17, 2009 at 5:11 PM, Nik Middleton < nik.middleton at noblesolutions.co.uk> wrote: > Hi Guys, > > > > I'm having real problems doing something trivial, and there doesn't seem to > be any docs on this issue > > > > In js I do this > > > > //Disposition = disp; > > //Create Custom event > > > > custom_msg = > > "call_disposition: " + Disposition + > "\n" + > > "called_number: " + dial_num + "\n" > ; > > > > e = new Event("custom", > "dialer::dialer-result"); > > e.addBody(custom_msg); > > e.fire(); > > > > And it works > > > > In lua I try this > > > > --Disposition = disp; > > --Create Custom event > > > > custom_msg = "call_disposition: " .. Disposition .. "\n" .. > > "called_number: " .. dial_num > .."\n" ; > > > > local e = freeswitch.Event("custom", > "dialer::dialer-result"); > > e.addBody(custom_msg); > > e:fire(e); > > > > This doesn't work, I get an error : Error in addBody expected 2..2 args, > got 1 > > > > What are the arguments? It seems to be looking for a pointer for the first > one, but there's nothing on the wiki on this. > > > > > > Regards, > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090217/6e79ecb9/attachment-0001.html From msc at freeswitch.org Tue Feb 17 15:29:48 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 17 Feb 2009 15:29:48 -0800 Subject: [Freeswitch-users] AddBody to events in lua In-Reply-To: <191c3a030902171525y17166787ud37ab5243c287008@mail.gmail.com> References: <191c3a030902171525y17166787ud37ab5243c287008@mail.gmail.com> Message-ID: <87f2f3b90902171529w1d5c32e5ydc6e6479d03ecab6@mail.gmail.com> On Tue, Feb 17, 2009 at 3:25 PM, Anthony Minessale wrote: > in lua you call methods with a colon : > > e:addBody(blah); > > calling with a . implies you are going to supply the obj too > > e.addBody(e, blah); > Also, there is an explicit example here: http://wiki.freeswitch.org/wiki/Lua#event:addBody It looks exactly like what you're trying to do. -MC From nik.middleton at noblesolutions.co.uk Tue Feb 17 15:30:48 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Tue, 17 Feb 2009 23:30:48 -0000 Subject: [Freeswitch-users] AddBody to events in lua In-Reply-To: <87f2f3b90902171523o5e014ce9w4fae9f05f378b861@mail.gmail.com> References: <87f2f3b90902171523o5e014ce9w4fae9f05f378b861@mail.gmail.com> Message-ID: I've got it working now thanks I've also added a working example to the Wiki (lua/addBody) which was empty Regards, -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 17 February 2009 23:23 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] AddBody to events in lua > local e = freeswitch.Event("custom", > "dialer::dialer-result"); > > e.addBody(custom_msg); > > e:fire(e); The wiki page (http://wiki.freeswitch.org/wiki/Lua#event:fire) shows that you fire thusly: e:fire(); --No "e" in the parens. Can you try it and report back? -MC _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From nik.middleton at noblesolutions.co.uk Tue Feb 17 15:33:46 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Tue, 17 Feb 2009 23:33:46 -0000 Subject: [Freeswitch-users] AddBody to events in lua In-Reply-To: <87f2f3b90902171529w1d5c32e5ydc6e6479d03ecab6@mail.gmail.com> References: <191c3a030902171525y17166787ud37ab5243c287008@mail.gmail.com> <87f2f3b90902171529w1d5c32e5ydc6e6479d03ecab6@mail.gmail.com> Message-ID: Err, that's what I just posted :) Regards, -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 17 February 2009 23:30 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] AddBody to events in lua On Tue, Feb 17, 2009 at 3:25 PM, Anthony Minessale wrote: > in lua you call methods with a colon : > > e:addBody(blah); > > calling with a . implies you are going to supply the obj too > > e.addBody(e, blah); > Also, there is an explicit example here: http://wiki.freeswitch.org/wiki/Lua#event:addBody It looks exactly like what you're trying to do. -MC _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From brian at freeswitch.org Tue Feb 17 15:35:54 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 17 Feb 2009 17:35:54 -0600 Subject: [Freeswitch-users] AddBody to events in lua In-Reply-To: References: <191c3a030902171525y17166787ud37ab5243c287008@mail.gmail.com> <87f2f3b90902171529w1d5c32e5ydc6e6479d03ecab6@mail.gmail.com> Message-ID: Good... keep up the good work adding more docs. ;) /b On Feb 17, 2009, at 5:33 PM, Nik Middleton wrote: > Err, that's what I just posted :) > > Regards, From msc at freeswitch.org Tue Feb 17 15:39:49 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 17 Feb 2009 15:39:49 -0800 Subject: [Freeswitch-users] AddBody to events in lua In-Reply-To: References: <191c3a030902171525y17166787ud37ab5243c287008@mail.gmail.com> <87f2f3b90902171529w1d5c32e5ydc6e6479d03ecab6@mail.gmail.com> Message-ID: <87f2f3b90902171539u29954bf7q2777a96c581a5b88@mail.gmail.com> On Tue, Feb 17, 2009 at 3:33 PM, Nik Middleton wrote: > Err, that's what I just posted :) > oops, hehe, that would explain why I thought my browser cache was messing with me. Nice work. Please definitely add to the Lua wiki page as you gain experience. Hopefully your pain will be other Lua users' gain. ;) -MC From kristian.kielhofner at gmail.com Tue Feb 17 15:43:31 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Tue, 17 Feb 2009 18:43:31 -0500 Subject: [Freeswitch-users] Anyone running FS from a Thumb Flash USB? In-Reply-To: <9dc4a1670902170550r3071f65dn76112fdbebaeb6ba@mail.gmail.com> References: <9dc4a1670902170550r3071f65dn76112fdbebaeb6ba@mail.gmail.com> Message-ID: <2d9149cd0902171543t1d1a5c9dqb23210ab8a2f3eeb@mail.gmail.com> FreeSWITCH now compiles in AsLinux: http://www.astlinux.org AstLinux with the new bootloader Runnix (or you could just use syslinux) boots from flash. It also boots from PXE, ISO, disk, etc. Pretty much anything :). FreeSWITCH (compiled against uClibc) is about 3.3MB and as Tony pointed out, mod_sofia is 1.2MB of that. The sample configs and sounds are much larger. Luckily the sounds compress well with something like squashfs. Put it this way: - Default AstLinux install (quite a bit of stuff these days) - FreeSWITCH (default mods + mod_xml_curl, -spidermonkey, but w/ lua, snom, vmd, and others) - Sample configs (pretty big too but also compress well) - 8k Sounds (HUGE, but compress well) - Native sounds (G723, G729, GSM, PCMU, PCMA all 8K obviously) Results in a squashfs disk image of about 41MB. You could run off a 64MB flash drive and have plenty left over for your union filesystem (configs, etc). :) Be aware that if you are going to run from such a config we recommend the default (which is to run AstLinux from RAM). Otherwise you take quite a hit reading audio from a squashfs filesystem. If you want an ISO to boot on a generic machine (VMware, virtualbox, etc work too) let me know. On Tue, Feb 17, 2009 at 8:50 AM, EdPimentl wrote: > Hello FS Members, > > Are there any example of FS running on a Thumb Flash USB? > Thanks in advance, > -E > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From brian at freeswitch.org Tue Feb 17 15:48:22 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 17 Feb 2009 17:48:22 -0600 Subject: [Freeswitch-users] Anyone running FS from a Thumb Flash USB? In-Reply-To: <2d9149cd0902171543t1d1a5c9dqb23210ab8a2f3eeb@mail.gmail.com> References: <9dc4a1670902170550r3071f65dn76112fdbebaeb6ba@mail.gmail.com> <2d9149cd0902171543t1d1a5c9dqb23210ab8a2f3eeb@mail.gmail.com> Message-ID: Great news!!! Good Job! /b On Feb 17, 2009, at 5:43 PM, Kristian Kielhofner wrote: > FreeSWITCH now compiles in AsLinux: > > http://www.astlinux.org > > AstLinux with the new bootloader Runnix (or you could just use > syslinux) boots from flash. It also boots from PXE, ISO, disk, etc. > Pretty much anything :). > > FreeSWITCH (compiled against uClibc) is about 3.3MB and as Tony > pointed out, mod_sofia is 1.2MB of that. The sample configs and > sounds are much larger. Luckily the sounds compress well with > something like squashfs. > > Put it this way: > > - Default AstLinux install (quite a bit of stuff these days) > - FreeSWITCH (default mods + mod_xml_curl, -spidermonkey, but w/ lua, > snom, vmd, and others) > - Sample configs (pretty big too but also compress well) > - 8k Sounds (HUGE, but compress well) > - Native sounds (G723, G729, GSM, PCMU, PCMA all 8K obviously) > > Results in a squashfs disk image of about 41MB. You could run off a > 64MB flash drive and have plenty left over for your union filesystem > (configs, etc). :) > > Be aware that if you are going to run from such a config we > recommend the default (which is to run AstLinux from RAM). Otherwise > you take quite a hit reading audio from a squashfs filesystem. > > If you want an ISO to boot on a generic machine (VMware, virtualbox, > etc work too) let me know. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090217/5bc53bff/attachment.html From jaybinks at gmail.com Tue Feb 17 15:58:17 2009 From: jaybinks at gmail.com (jay binks) Date: Wed, 18 Feb 2009 09:58:17 +1000 Subject: [Freeswitch-users] Anyone running FS from a Thumb Flash USB? In-Reply-To: <2d9149cd0902171543t1d1a5c9dqb23210ab8a2f3eeb@mail.gmail.com> References: <9dc4a1670902170550r3071f65dn76112fdbebaeb6ba@mail.gmail.com> <2d9149cd0902171543t1d1a5c9dqb23210ab8a2f3eeb@mail.gmail.com> Message-ID: wow... this is awesome ! good job mate. On Wed, Feb 18, 2009 at 9:43 AM, Kristian Kielhofner < kristian.kielhofner at gmail.com> wrote: > FreeSWITCH now compiles in AsLinux: > > http://www.astlinux.org > > AstLinux with the new bootloader Runnix (or you could just use > syslinux) boots from flash. It also boots from PXE, ISO, disk, etc. > Pretty much anything :). > > FreeSWITCH (compiled against uClibc) is about 3.3MB and as Tony > pointed out, mod_sofia is 1.2MB of that. The sample configs and > sounds are much larger. Luckily the sounds compress well with > something like squashfs. > > Put it this way: > > - Default AstLinux install (quite a bit of stuff these days) > - FreeSWITCH (default mods + mod_xml_curl, -spidermonkey, but w/ lua, > snom, vmd, and others) > - Sample configs (pretty big too but also compress well) > - 8k Sounds (HUGE, but compress well) > - Native sounds (G723, G729, GSM, PCMU, PCMA all 8K obviously) > > Results in a squashfs disk image of about 41MB. You could run off a > 64MB flash drive and have plenty left over for your union filesystem > (configs, etc). :) > > Be aware that if you are going to run from such a config we > recommend the default (which is to run AstLinux from RAM). Otherwise > you take quite a hit reading audio from a squashfs filesystem. > > If you want an ISO to boot on a generic machine (VMware, virtualbox, > etc work too) let me know. > > On Tue, Feb 17, 2009 at 8:50 AM, EdPimentl wrote: > > Hello FS Members, > > > > Are there any example of FS running on a Thumb Flash USB? > > Thanks in advance, > > -E > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > -- > Kristian Kielhofner > http://blog.krisk.org > http://www.submityoursip.com > http://www.astlinux.org > http://www.star2star.com > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Sincerely Jay -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090218/8c9e4a6d/attachment-0001.html From nik.middleton at noblesolutions.co.uk Tue Feb 17 16:07:05 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Wed, 18 Feb 2009 00:07:05 -0000 Subject: [Freeswitch-users] AddBody to events in lua In-Reply-To: References: <191c3a030902171525y17166787ud37ab5243c287008@mail.gmail.com><87f2f3b90902171529w1d5c32e5ydc6e6479d03ecab6@mail.gmail.com> Message-ID: I'll shortly post some docs on the php fs_sock. There's also a couple of bugs in it that I've fixed. I ran 10,000 events, which completed in around 20 seconds, all received and processed flawlessly. A new one on me was arrayshift. To think that I messed around in C for ages with circular buffers, this is so simple. Regards, -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: 17 February 2009 23:36 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] AddBody to events in lua Good... keep up the good work adding more docs. ;) /b On Feb 17, 2009, at 5:33 PM, Nik Middleton wrote: > Err, that's what I just posted :) > > Regards, _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From msc at freeswitch.org Tue Feb 17 16:10:17 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 17 Feb 2009 16:10:17 -0800 Subject: [Freeswitch-users] Anyone running FS from a Thumb Flash USB? In-Reply-To: <2d9149cd0902171543t1d1a5c9dqb23210ab8a2f3eeb@mail.gmail.com> References: <9dc4a1670902170550r3071f65dn76112fdbebaeb6ba@mail.gmail.com> <2d9149cd0902171543t1d1a5c9dqb23210ab8a2f3eeb@mail.gmail.com> Message-ID: <87f2f3b90902171610r37679769h2b4f1c033e7b3e94@mail.gmail.com> On Tue, Feb 17, 2009 at 3:43 PM, Kristian Kielhofner wrote: > FreeSWITCH now compiles in AsLinux: Nice work! I'll go tell our friends over in the Yahoo financial forums - I'm sure they're dying to hear about it! ;) -MC From msc at freeswitch.org Tue Feb 17 16:12:28 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 17 Feb 2009 16:12:28 -0800 Subject: [Freeswitch-users] AddBody to events in lua In-Reply-To: References: <191c3a030902171525y17166787ud37ab5243c287008@mail.gmail.com> <87f2f3b90902171529w1d5c32e5ydc6e6479d03ecab6@mail.gmail.com> Message-ID: <87f2f3b90902171612p11dfd5e1u92287e76e264e10d@mail.gmail.com> > I ran 10,000 events, which completed in around 20 seconds, all received > and processed flawlessly. A new one on me was arrayshift. To think that > I messed around in C for ages with circular buffers, this is so simple. Excellent! You're officially deputized to add any Lua examples you create. We can use examples on the wiki and sample scripts in the contrib directory. Nice work. -MC From brian at freeswitch.org Tue Feb 17 16:14:39 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 17 Feb 2009 18:14:39 -0600 Subject: [Freeswitch-users] AddBody to events in lua In-Reply-To: References: <191c3a030902171525y17166787ud37ab5243c287008@mail.gmail.com><87f2f3b90902171529w1d5c32e5ydc6e6479d03ecab6@mail.gmail.com> Message-ID: <677AB3FE-E0B9-46A8-8808-74FE1A6D5C3D@freeswitch.org> And you ran this in lua? /b On Feb 17, 2009, at 6:07 PM, Nik Middleton wrote: > > I ran 10,000 events, which completed in around 20 seconds, all > received > and processed flawlessly. A new one on me was arrayshift. To think > that > I messed around in C for ages with circular buffers, this is so > simple. From gkuri at ieee.org Tue Feb 17 16:16:05 2009 From: gkuri at ieee.org (Gabriel Kuri) Date: Tue, 17 Feb 2009 16:16:05 -0800 Subject: [Freeswitch-users] Anyone running FS from a Thumb Flash USB? In-Reply-To: <2d9149cd0902171543t1d1a5c9dqb23210ab8a2f3eeb@mail.gmail.com> References: <9dc4a1670902170550r3071f65dn76112fdbebaeb6ba@mail.gmail.com> <2d9149cd0902171543t1d1a5c9dqb23210ab8a2f3eeb@mail.gmail.com> Message-ID: <499B5345.1010701@ieee.org> awesome work! on a slightly related [embedded] note, do you know if any work has been done to port FS to any of the Analog Blackfin MCUs? I'd be interested in hearing if anyone has had any such luck. Gabe Kristian Kielhofner wrote: > FreeSWITCH now compiles in AsLinux: > > http://www.astlinux.org > > AstLinux with the new bootloader Runnix (or you could just use > syslinux) boots from flash. It also boots from PXE, ISO, disk, etc. > Pretty much anything :). > > FreeSWITCH (compiled against uClibc) is about 3.3MB and as Tony > pointed out, mod_sofia is 1.2MB of that. The sample configs and > sounds are much larger. Luckily the sounds compress well with > something like squashfs. > > Put it this way: > > - Default AstLinux install (quite a bit of stuff these days) > - FreeSWITCH (default mods + mod_xml_curl, -spidermonkey, but w/ lua, > snom, vmd, and others) > - Sample configs (pretty big too but also compress well) > - 8k Sounds (HUGE, but compress well) > - Native sounds (G723, G729, GSM, PCMU, PCMA all 8K obviously) > > Results in a squashfs disk image of about 41MB. You could run off a > 64MB flash drive and have plenty left over for your union filesystem > (configs, etc). :) > > Be aware that if you are going to run from such a config we > recommend the default (which is to run AstLinux from RAM). Otherwise > you take quite a hit reading audio from a squashfs filesystem. > > If you want an ISO to boot on a generic machine (VMware, virtualbox, > etc work too) let me know. > > On Tue, Feb 17, 2009 at 8:50 AM, EdPimentl wrote: >> Hello FS Members, >> >> Are there any example of FS running on a Thumb Flash USB? >> Thanks in advance, >> -E >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > From kristian.kielhofner at gmail.com Tue Feb 17 16:18:35 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Tue, 17 Feb 2009 19:18:35 -0500 Subject: [Freeswitch-users] Anyone running FS from a Thumb Flash USB? In-Reply-To: <87f2f3b90902171610r37679769h2b4f1c033e7b3e94@mail.gmail.com> References: <9dc4a1670902170550r3071f65dn76112fdbebaeb6ba@mail.gmail.com> <2d9149cd0902171543t1d1a5c9dqb23210ab8a2f3eeb@mail.gmail.com> <87f2f3b90902171610r37679769h2b4f1c033e7b3e94@mail.gmail.com> Message-ID: <2d9149cd0902171618s47640fb2vff3cc139b74c8f64@mail.gmail.com> Ah yes, the "pinnacle of online discussion"! ;) On Tue, Feb 17, 2009 at 7:10 PM, Michael Collins wrote: > On Tue, Feb 17, 2009 at 3:43 PM, Kristian Kielhofner > wrote: >> FreeSWITCH now compiles in AsLinux: > > Nice work! I'll go tell our friends over in the Yahoo financial forums > - I'm sure they're dying to hear about it! ;) > -MC > -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From kristian.kielhofner at gmail.com Tue Feb 17 16:20:01 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Tue, 17 Feb 2009 19:20:01 -0500 Subject: [Freeswitch-users] Anyone running FS from a Thumb Flash USB? In-Reply-To: <499B5345.1010701@ieee.org> References: <9dc4a1670902170550r3071f65dn76112fdbebaeb6ba@mail.gmail.com> <2d9149cd0902171543t1d1a5c9dqb23210ab8a2f3eeb@mail.gmail.com> <499B5345.1010701@ieee.org> Message-ID: <2d9149cd0902171620m28f82962sbdc087c0adbca70e@mail.gmail.com> I don't think so but something tells me that FreeSWITCH won't do too well without an MMU and the external libs and modules could cause quite a problem. Not that it is impossible but the uh, performance, would be interesting... Can anyone call me out on this assumption? On Tue, Feb 17, 2009 at 7:16 PM, Gabriel Kuri wrote: > awesome work! on a slightly related [embedded] note, do you know if any > work has been done to port FS to any of the Analog Blackfin MCUs? I'd be > interested in hearing if anyone has had any such luck. > > Gabe > -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From raul at etellicom.com Tue Feb 17 16:20:55 2009 From: raul at etellicom.com (Raul Fragoso) Date: Tue, 17 Feb 2009 21:20:55 -0300 Subject: [Freeswitch-users] Deployment information and use cases Message-ID: <1234916455.16581.49.camel@raul-laptop> Hello FreeSWITCHERS, My company is currently creating a suite of applications which uses FreeSWITCH as the back-end for an IP-PBX solution. We currently have a prospect to have our first customer installation - a governmental department. That is a tender to have an IP-PBX installation to connect their four office branches, each one with about 300 users - which I am sure FreeSWITCH is able to handle. Since this is an official tender, it's part of their protocol to ask about real sites using the product. Having said that, would you mind sharing some information about your experience with FreeSWITCH deployments ? No need to give many details, but a short summary with company name (if possible), when it was deployed, server equipment, number of users, number of concurrent calls, what kind of functions and services are used and overall capacity of the system. I would really appreciate if you can share that information. And if you guys agree (and explicitly manifest your agreement), I can compile the information in the FreeSWITCH wiki under a "Use Cases" page so it can serve as a common reference as well. Kind regards, Raul Fragoso From nik.middleton at noblesolutions.co.uk Tue Feb 17 16:21:09 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Wed, 18 Feb 2009 00:21:09 -0000 Subject: [Freeswitch-users] AddBody to events in lua In-Reply-To: <677AB3FE-E0B9-46A8-8808-74FE1A6D5C3D@freeswitch.org> References: <191c3a030902171525y17166787ud37ab5243c287008@mail.gmail.com><87f2f3b90902171529w1d5c32e5ydc6e6479d03ecab6@mail.gmail.com> <677AB3FE-E0B9-46A8-8808-74FE1A6D5C3D@freeswitch.org> Message-ID: No, js, I was trying to break the fs_sock.php, though I found the time was dependant on how much I echoed to the screen. I expect lua to be even faster Regards, -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: 18 February 2009 00:15 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] AddBody to events in lua And you ran this in lua? /b On Feb 17, 2009, at 6:07 PM, Nik Middleton wrote: > > I ran 10,000 events, which completed in around 20 seconds, all > received > and processed flawlessly. A new one on me was arrayshift. To think > that > I messed around in C for ages with circular buffers, this is so > simple. _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From nik.middleton at noblesolutions.co.uk Tue Feb 17 16:25:26 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Wed, 18 Feb 2009 00:25:26 -0000 Subject: [Freeswitch-users] Anyone running FS from a Thumb Flash USB? In-Reply-To: <2d9149cd0902171618s47640fb2vff3cc139b74c8f64@mail.gmail.com> References: <9dc4a1670902170550r3071f65dn76112fdbebaeb6ba@mail.gmail.com><2d9149cd0902171543t1d1a5c9dqb23210ab8a2f3eeb@mail.gmail.com><87f2f3b90902171610r37679769h2b4f1c033e7b3e94@mail.gmail.com> <2d9149cd0902171618s47640fb2vff3cc139b74c8f64@mail.gmail.com> Message-ID: Kristian, You're my hero, if I hadn't come across astlinux 3 years ago, I wouldn't be doing this stuff right now. Not too sure if that's a good thing though ;) -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Kristian Kielhofner Sent: 18 February 2009 00:19 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Anyone running FS from a Thumb Flash USB? Ah yes, the "pinnacle of online discussion"! ;) On Tue, Feb 17, 2009 at 7:10 PM, Michael Collins wrote: > On Tue, Feb 17, 2009 at 3:43 PM, Kristian Kielhofner > wrote: >> FreeSWITCH now compiles in AsLinux: > > Nice work! I'll go tell our friends over in the Yahoo financial forums > - I'm sure they're dying to hear about it! ;) > -MC > -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From fax at virgintechnologies.com Tue Feb 17 15:51:30 2009 From: fax at virgintechnologies.com (fax at virgintechnologies.com) Date: Tue, 17 Feb 2009 23:51:30 +0000 Subject: [Freeswitch-users] Sending media streams to a media gateway Message-ID: I have Freeswitch running successfully with a fairly basic config. Nat traversal is working well on both the client and server side. I want to start running all RTP streams through a media gateway, and use Freeswitch for SIP registrations and signalling only. I believe that I need to have Freeswitch invite the SIP phone to send the RTP stream directly to the media gateway when a call starts. Where can I start with this? Does anyone have any example configs? Justin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090217/3d53695c/attachment-0001.html From anthony.minessale at gmail.com Tue Feb 17 16:34:54 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 17 Feb 2009 18:34:54 -0600 Subject: [Freeswitch-users] Sending media streams to a media gateway In-Reply-To: References: Message-ID: <191c3a030902171634t4dc2e15etfa02d118e8fe3322@mail.gmail.com> you could set the variable bypass_media to true before you call bridge that will negotiate a point to point media connection between the caller and callee On Tue, Feb 17, 2009 at 5:51 PM, wrote: > I have Freeswitch running successfully with a fairly basic config. Nat > traversal is working well on both the client and server side. I want to > start running all RTP streams through a media gateway, and use Freeswitch > for SIP registrations and signalling only. > I believe that I need to have Freeswitch invite the SIP phone to send the > RTP stream directly to the media gateway when a call starts. Where can I > start with this? Does anyone have any example configs? > Justin > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090217/3688e75e/attachment.html From gkuri at ieee.org Tue Feb 17 16:44:10 2009 From: gkuri at ieee.org (Gabriel Kuri) Date: Tue, 17 Feb 2009 16:44:10 -0800 Subject: [Freeswitch-users] Anyone running FS from a Thumb Flash USB? In-Reply-To: <2d9149cd0902171620m28f82962sbdc087c0adbca70e@mail.gmail.com> References: <9dc4a1670902170550r3071f65dn76112fdbebaeb6ba@mail.gmail.com> <2d9149cd0902171543t1d1a5c9dqb23210ab8a2f3eeb@mail.gmail.com> <499B5345.1010701@ieee.org> <2d9149cd0902171620m28f82962sbdc087c0adbca70e@mail.gmail.com> Message-ID: <499B59DA.3000806@ieee.org> That and the lack of an FPU, I'm curious how that would affect FS, especially with transcoding. At one point I was interested in building a little embedded PBX running FS on the Blackfin MCU. Since Analog Devices seems fairly open with the Blackfin, I thought it might be a good choice, but I'm not sure how the lack of an MMU or FPU would affect FS. Gabe Kristian Kielhofner wrote: > I don't think so but something tells me that FreeSWITCH won't do too > well without an MMU and the external libs and modules could cause > quite a problem. Not that it is impossible but the uh, performance, > would be interesting... > > Can anyone call me out on this assumption? > > On Tue, Feb 17, 2009 at 7:16 PM, Gabriel Kuri wrote: >> awesome work! on a slightly related [embedded] note, do you know if any >> work has been done to port FS to any of the Analog Blackfin MCUs? I'd be >> interested in hearing if anyone has had any such luck. >> >> Gabe >> > From fax at virgintechnologies.com Tue Feb 17 16:45:05 2009 From: fax at virgintechnologies.com (fax at virgintechnologies.com) Date: Wed, 18 Feb 2009 00:45:05 +0000 Subject: [Freeswitch-users] Sending media streams to a media gateway Message-ID: I looked at that, but I think that will cause issues with the NAT traversal. Our phones will all be in external networks. I forgot to mention that. -----Original Message----- From: Anthony Minessale [mailto:anthony.minessale at gmail.com] Sent: Tuesday, February 17, 2009 05:34 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Sending media streams to a media gateway you could set the variable bypass_media to true before you call bridge that will negotiate a point to point media connection between the caller and callee On Tue, Feb 17, 2009 at 5:51 PM, wrote: I have Freeswitch running successfully with a fairly basic config. Nat traversal is working well on both the client and server side. I want to start running all RTP streams through a media gateway, and use Freeswitch for SIP registrations and signalling only. I believe that I need to have Freeswitch invite the SIP phone to send the RTP stream directly to the media gateway when a call starts. Where can I start with this? Does anyone have any example configs? Justin _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm mailto:MSN%3Aanthony_minessale at hotmail.com GTALK/JABBER/mailto:PAYPAL%3Aanthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference mailto:sip%3A888 at conference.freeswitch.org http://iax:guest at conference.freeswitch.org/888 mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090218/b2b679b1/attachment.html From brian at freeswitch.org Tue Feb 17 17:36:39 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 17 Feb 2009 19:36:39 -0600 Subject: [Freeswitch-users] Anyone running FS from a Thumb Flash USB? In-Reply-To: <2d9149cd0902171620m28f82962sbdc087c0adbca70e@mail.gmail.com> References: <9dc4a1670902170550r3071f65dn76112fdbebaeb6ba@mail.gmail.com> <2d9149cd0902171543t1d1a5c9dqb23210ab8a2f3eeb@mail.gmail.com> <499B5345.1010701@ieee.org> <2d9149cd0902171620m28f82962sbdc087c0adbca70e@mail.gmail.com> Message-ID: So you sticking with the Astlinux name? or switching it to something more general? /b On Feb 17, 2009, at 6:20 PM, Kristian Kielhofner wrote: > I don't think so but something tells me that FreeSWITCH won't do too > well without an MMU and the external libs and modules could cause > quite a problem. Not that it is impossible but the uh, performance, > would be interesting... From kristian.kielhofner at gmail.com Tue Feb 17 17:41:56 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Tue, 17 Feb 2009 20:41:56 -0500 Subject: [Freeswitch-users] AstLinux-FreeSWITCH ISO available for download Message-ID: <2d9149cd0902171741h709cae3ckc9ff6ea00e268710@mail.gmail.com> I really need to work on that name but in the meantime it seems like people are interested. Check it out: http://www.astlinux.org/node/41 It's just a little ISO, download it and give it a shot! ;) -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From krice at freeswitch.org Tue Feb 17 17:48:08 2009 From: krice at freeswitch.org (Ken Rice) Date: Tue, 17 Feb 2009 19:48:08 -0600 Subject: [Freeswitch-users] Happy B-Day BKW! Message-ID: Ok Tomorrow is BKW's b-day... Lets as a community see if we cant pull together and get him something nice off his Amazon wish list... Visit his wish list here... http://www.amazon.com/gp/registry/wishlist/1BWDJUX5LYQE0 If you want to chip in contact me on IRC (I'm SwK) or send something via the donate link at http://www.tollfreegateway.com/bailmeout.html be sure to put "For BKW" in the comment or drop me an off list email... If you want to send something via paypal direct to brian his paypal ID is brian at freeswitch.org Come on guys he works day and night on FS to help bring us arguably the best OpenSource Telephony platform out there!! Happy B'Day Brian!! Ken From sprice at gmail.com Tue Feb 17 17:55:20 2009 From: sprice at gmail.com (SP) Date: Tue, 17 Feb 2009 19:55:20 -0600 Subject: [Freeswitch-users] Happy B-Day BKW! In-Reply-To: References: Message-ID: <7e2ac3270902171755g7774d04fhc659962ba1610d6a@mail.gmail.com> We could ban him from IRC for the day... that would be a gift :) On Tue, Feb 17, 2009 at 19:48, Ken Rice wrote: > Ok Tomorrow is BKW's b-day... > > Lets as a community see if we cant pull together and get him something nice > off his Amazon wish list... > > Visit his wish list here... > http://www.amazon.com/gp/registry/wishlist/1BWDJUX5LYQE0 > > If you want to chip in contact me on IRC (I'm SwK) or send something via the > donate link at http://www.tollfreegateway.com/bailmeout.html be sure to put > "For BKW" in the comment or drop me an off list email... > > If you want to send something via paypal direct to brian his paypal ID is > brian at freeswitch.org > > Come on guys he works day and night on FS to help bring us arguably the best > OpenSource Telephony platform out there!! > > Happy B'Day Brian!! > > Ken > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Shannon From jaybinks at gmail.com Tue Feb 17 18:04:05 2009 From: jaybinks at gmail.com (jay binks) Date: Wed, 18 Feb 2009 12:04:05 +1000 Subject: [Freeswitch-users] AstLinux-FreeSWITCH ISO available for download In-Reply-To: <2d9149cd0902171741h709cae3ckc9ff6ea00e268710@mail.gmail.com> References: <2d9149cd0902171741h709cae3ckc9ff6ea00e268710@mail.gmail.com> Message-ID: awwww man.... geez im interested in this .. I hope it ends up kicking ass ! :) Congrats, you are awesome. Jay On Wed, Feb 18, 2009 at 11:41 AM, Kristian Kielhofner < kristian.kielhofner at gmail.com> wrote: > I really need to work on that name but in the meantime it seems like > people are interested. Check it out: > > http://www.astlinux.org/node/41 > > It's just a little ISO, download it and give it a shot! ;) > > -- > Kristian Kielhofner > http://blog.krisk.org > http://www.submityoursip.com > http://www.astlinux.org > http://www.star2star.com > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Sincerely Jay -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090218/7f861c91/attachment-0001.html From kristian.kielhofner at gmail.com Tue Feb 17 18:15:08 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Tue, 17 Feb 2009 21:15:08 -0500 Subject: [Freeswitch-users] Anyone running FS from a Thumb Flash USB? In-Reply-To: References: <9dc4a1670902170550r3071f65dn76112fdbebaeb6ba@mail.gmail.com> <2d9149cd0902171543t1d1a5c9dqb23210ab8a2f3eeb@mail.gmail.com> <499B5345.1010701@ieee.org> <2d9149cd0902171620m28f82962sbdc087c0adbca70e@mail.gmail.com> Message-ID: <2d9149cd0902171815q4a25b42n447efffb251574f@mail.gmail.com> Brian, We'll figure something out... On Tue, Feb 17, 2009 at 8:36 PM, Brian West wrote: > So you sticking with the Astlinux name? or switching it to something > more general? > > /b > -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From intralanman at freeswitch.org Tue Feb 17 18:41:22 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Tue, 17 Feb 2009 21:41:22 -0500 Subject: [Freeswitch-users] AddBody to events in lua In-Reply-To: References: <191c3a030902171525y17166787ud37ab5243c287008@mail.gmail.com><87f2f3b90902171529w1d5c32e5ydc6e6479d03ecab6@mail.gmail.com> Message-ID: <499B7552.3010104@freeswitch.org> Nik Middleton wrote: > I'll shortly post some docs on the php fs_sock. don't waste your time... There's a php .so for ESL now, and i'll probably be removing the fs_sock from tree sometime very soon... maybe replacing it with some specific api classes... i'm not sure on that part yet. -Ray From mike at jerris.com Tue Feb 17 20:16:22 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 17 Feb 2009 23:16:22 -0500 Subject: [Freeswitch-users] Anyone running FS from a Thumb Flash USB? In-Reply-To: <499B59DA.3000806@ieee.org> References: <9dc4a1670902170550r3071f65dn76112fdbebaeb6ba@mail.gmail.com> <2d9149cd0902171543t1d1a5c9dqb23210ab8a2f3eeb@mail.gmail.com> <499B5345.1010701@ieee.org> <2d9149cd0902171620m28f82962sbdc087c0adbca70e@mail.gmail.com> <499B59DA.3000806@ieee.org> Message-ID: <007ACD53-6B53-4E4D-BB44-38845B8E6B26@jerris.com> Pretty much all the codecs (mod_voipcodecs, mod_speex, mod_ilbc, mod_g722_1, mod_celt) and the resampler all have fixed point implementations (in tree) as well. Mike On Feb 17, 2009, at 7:44 PM, Gabriel Kuri wrote: > That and the lack of an FPU, I'm curious how that would affect FS, > especially with transcoding. At one point I was interested in > building a > little embedded PBX running FS on the Blackfin MCU. Since Analog > Devices > seems fairly open with the Blackfin, I thought it might be a good > choice, but I'm not sure how the lack of an MMU or FPU would affect > FS. > > Gabe > > Kristian Kielhofner wrote: >> I don't think so but something tells me that FreeSWITCH won't do too >> well without an MMU and the external libs and modules could cause >> quite a problem. Not that it is impossible but the uh, performance, >> would be interesting... >> >> Can anyone call me out on this assumption? >> >> On Tue, Feb 17, 2009 at 7:16 PM, Gabriel Kuri wrote: >>> awesome work! on a slightly related [embedded] note, do you know >>> if any >>> work has been done to port FS to any of the Analog Blackfin MCUs? >>> I'd be >>> interested in hearing if anyone has had any such luck. >>> >>> Gabe >>> From kristian.kielhofner at gmail.com Tue Feb 17 23:09:58 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Wed, 18 Feb 2009 02:09:58 -0500 Subject: [Freeswitch-users] Anyone running FS from a Thumb Flash USB? In-Reply-To: References: <9dc4a1670902170550r3071f65dn76112fdbebaeb6ba@mail.gmail.com> <2d9149cd0902171543t1d1a5c9dqb23210ab8a2f3eeb@mail.gmail.com> <87f2f3b90902171610r37679769h2b4f1c033e7b3e94@mail.gmail.com> <2d9149cd0902171618s47640fb2vff3cc139b74c8f64@mail.gmail.com> Message-ID: <2d9149cd0902172309i1aa93fcej8fce68133028ee93@mail.gmail.com> Nik, Thanks but I'm not sure I want to take the credit (blame?) for that! ;) On Tue, Feb 17, 2009 at 7:25 PM, Nik Middleton wrote: > Kristian, > > You're my hero, if I hadn't come across astlinux 3 years ago, I wouldn't > be doing this stuff right now. Not too sure if that's a good thing > though ;) > -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From uv at yuvalhertzog.com Wed Feb 18 03:55:35 2009 From: uv at yuvalhertzog.com (UV) Date: Wed, 18 Feb 2009 22:55:35 +1100 Subject: [Freeswitch-users] Anyone running FS from a Thumb Flash USB? In-Reply-To: <2d9149cd0902172309i1aa93fcej8fce68133028ee93@mail.gmail.com> References: <9dc4a1670902170550r3071f65dn76112fdbebaeb6ba@mail.gmail.com><2d9149cd0902171543t1d1a5c9dqb23210ab8a2f3eeb@mail.gmail.com><87f2f3b90902171610r37679769h2b4f1c033e7b3e94@mail.gmail.com><2d9149cd0902171618s47640fb2vff3cc139b74c8f64@mail.gmail.com> <2d9149cd0902172309i1aa93fcej8fce68133028ee93@mail.gmail.com> Message-ID: <05B5115F398B44C28AFDB2DE0E1A9281@UVix> Awesome work, Kristian! And very much needed for the Freeswitch platform (to me, at least). A suggestion: if the FS team doesn't mind (after getting over the naming issue), it would be a good idea to put Kristian's latest blog entry on the FS Wiki. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Kristian Kielhofner Sent: Wednesday, February 18, 2009 6:10 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Anyone running FS from a Thumb Flash USB? Nik, Thanks but I'm not sure I want to take the credit (blame?) for that! ;) On Tue, Feb 17, 2009 at 7:25 PM, Nik Middleton wrote: > Kristian, > > You're my hero, if I hadn't come across astlinux 3 years ago, I wouldn't > be doing this stuff right now. Not too sure if that's a good thing > though ;) > -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org No virus found in this incoming message. Checked by AVG - www.avg.com Version: 8.0.237 / Virus Database: 270.10.25/1956 - Release Date: 02/17/09 07:07:00 No virus found in this outgoing message. Checked by AVG - www.avg.com Version: 8.0.237 / Virus Database: 270.10.25/1956 - Release Date: 02/17/09 07:07:00 From gmaruzz at celliax.org Wed Feb 18 03:58:12 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Wed, 18 Feb 2009 12:58:12 +0100 Subject: [Freeswitch-users] Skypiax on OS X In-Reply-To: References: Message-ID: <7b197bef0902180358s7c3ca246kf354beef2e72f4e6@mail.gmail.com> > On Feb 17, 2009, at 3:19 PM, Ivan C Myrvold wrote: >> Is it possible to run Skypiax on OS X? The wiki says Linux and Windows, >> but says nothing about OS X. On Tue, Feb 17, 2009 at 10:31 PM, Brian West wrote: > I think Ctrix is working on mod_airpe in his branch for OS X. At the moment is not possible to run Skypiax on OSX. I have no OSX machines at the moment, sorry. I would like to add OSX support to Skypiax tough. I'll have someone lend me a machine in the future, if nobody else sends patches. In the mean time, as bkw wrote, you can try the mod_airpe from Ctrix. Sincerely, Giovanni Maruzzelli ========================================= www.celliax.org via Pierlombardo 9, 20135 Milano Italy gmaruzz at celliax dot org Cell : +39-347-2665618 Fax : +39-02-87390039 On Tue, Feb 17, 2009 at 10:31 PM, Brian West wrote: > I think Ctrix is working on mod_airpe in his branch for OS X. > > /b > > On Feb 17, 2009, at 3:19 PM, Ivan C Myrvold wrote: > >> s it possible to run Skypiax on OS X? The wiki says Linux and Windows, >> but says nothing about OS X. >> I have been running FreeSWITCH on OS X for a couple of years now, and >> love it. Adding Skype gateway would be really sweet. >> >> Are there any plans for adding Skypiax to trunk, or do we have to >> build it only from svn branch? >> >> Ivan > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From leon at scarlet-internet.nl Wed Feb 18 06:03:52 2009 From: leon at scarlet-internet.nl (Leon de Rooij) Date: Wed, 18 Feb 2009 15:03:52 +0100 Subject: [Freeswitch-users] ipauth - directory In-Reply-To: <191c3a030902160604r77089a6s5b3b9f3d07914218@mail.gmail.com> References: <496E0187.8090805@kinetix.gr> <23DEEC73-4DC7-4056-902F-D74DD6644385@freeswitch.org> <496E068B.6050404@kinetix.gr> <191c3a030902160604r77089a6s5b3b9f3d07914218@mail.gmail.com> Message-ID: Hi Anthony, I tried your second option, but how does it work with xml-curl then ? As far as I understand it, this doesn't work by doing a user-directory xml lookup at INVITE time, or does it ? Or does it want to generate an ACL at FS startup and filling up all the allow-nodes by polling the entire domain, filtering out all users with CIDR entry and putting those in the ACL itself ? If so, is that the reason why FS tries (at startup) to POST to the webserver with: hostname = test §ion = directory &tag_name = domain &key_name=name&key_value=test.com&domain=test.com&purpose=network-list ? Thanks & regards, Leon On Feb 16, 2009, at 3:04 PM, Anthony Minessale wrote: > you have 3 options. > on authenticated users, every tag in his account will be > set on each call from that authenticated user. > > 1) make them register, this sets the variables automatically > 2) use the ACL list with cidr= from> this has the same effect with no auth needed. > 3) use some other way to differentiate the user and use the set_user > application in the dialplan to inherit that user's variables. > > > > On Mon, Feb 16, 2009 at 6:49 AM, Leon de Rooij > wrote: > Hi all, > > I'd really like to know more about this too. > > Currently, I have two sip_profiles: > > - residential (where users can do authenticated registers and invites) > - transit (where other users can do un-authenticated invites) > > Right now, FS is not aware of *who* is accessing the transit profile > except for an acl that is set on this profile so unauthorized use is > not possible. > > But what should I do when I want to allow multiple parties (from > different IP addresses) to send their invites to the transit > profile, and still be able to differentiate between them ? > > I'd like to set some variables, like an accountcode for example, on > the basis of what IP address the INVITE originates from. > > So, is it possible to not use digest authentication, but still use a > dialplan-directory user with IP= field or some such ? > > thanks a lot & kind regards, > > Leon de Rooij > > > > On Jan 14, 2009, at 4:36 PM, Apostolos Pantsiopoulos wrote: > >> Yes I know that. But what does the "ip=" setting do? >> >> Brian West wrote: >>> >>> cidr= and the domains acl in acl.conf.xml then apply that ACL to the >>> sofia profile. >>> >>> /b >>> >>> On Jan 14, 2009, at 9:15 AM, Apostolos Pantsiopoulos wrote: >>> >>> >>>> I noticed an "ip=" setting in the brian.xml sample file. >>>> The comments state that this is used for ipauth (IP based >>>> authentication?) >>>> >>>> What exactly is this setting. I cannot find anything in the wiki >>>> about it. >>>> Does it replace the use of the >>>> >>>> + ACL >>>> >>>> mechanism for IP authentication? >>>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> -- >> ------------------------------------------- >> Apostolos Pantsiopoulos >> Kinetix Tele.com R & D >> email: regs at kinetix.gr >> ------------------------------------------- >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090218/17eecf9a/attachment-0001.html From anthony.minessale at gmail.com Wed Feb 18 06:09:27 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 18 Feb 2009 08:09:27 -0600 Subject: [Freeswitch-users] call goes to wrong context In-Reply-To: References: <499B1B42.7090000@post.cz> <4432BDD8-67A4-45B4-A013-7BB7C7E35D7F@freeswitch.org> <499B2691.70003@post.cz> Message-ID: <191c3a030902180609m52ad6b2es2aa84f87b994e3e2@mail.gmail.com> if the inbound calls are coming from a registration to a provider you will have to set a context param in the gateway itself. All inbound calls from a gateway registration are now associated with the gateway they were registered with and inherit the context from there. Maybe i'll change the default context of a gateway to be the default context of it's host profile to avoid this issue. On Tue, Feb 17, 2009 at 3:09 PM, Brian West wrote: > No problem. Just join us on IRC.. things move faster on there. > > /b > > On Feb 17, 2009, at 3:05 PM, kokoska.rokoska wrote: > > > If you have more hints, I be very happy :-) > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090218/d7e1aaeb/attachment.html From kawarod at laposte.net Wed Feb 18 06:10:14 2009 From: kawarod at laposte.net (rod) Date: Wed, 18 Feb 2009 18:10:14 +0400 Subject: [Freeswitch-users] mod_fax and sending a fax In-Reply-To: <49919822.3030101@laposte.net> References: <49919822.3030101@laposte.net> Message-ID: <499C16C6.1000006@laposte.net> Hi all, I'm able to receive a fax with mod_fax, but I still don't understand how to send fax. I don't understand how to send the fax through a specific profile/IP. Is mod_fax limited to an openzap interface?? when I dial to the extension with tx_fax, I get a tone then hangup, but what I'd like to do is send the fax call through a specific peer (as the bridge application). For those who'd like to trigger an event on received fax (convert to pdf, send a mail...), I think a tool like this could help: http://projects.l3ib.org/trac/fsniper I did not try it at this time, if I'm successful I will update the wiki. regards, rod. rod wrote: > Hi all, > > I don't understand how to use the fax commands for sending a fax. In the > wiki I saw this: > > > > > > > > > > my question is how to specify the gateway/profile that will handle the call. > For a call I can use the bridge application like this, but for the txfax ?? > data="sofia/external/${destination_number}@10.10.10.10"/> > > regards, > rod > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > From anthony.minessale at gmail.com Wed Feb 18 06:13:32 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 18 Feb 2009 08:13:32 -0600 Subject: [Freeswitch-users] call goes to wrong context In-Reply-To: <191c3a030902180609m52ad6b2es2aa84f87b994e3e2@mail.gmail.com> References: <499B1B42.7090000@post.cz> <4432BDD8-67A4-45B4-A013-7BB7C7E35D7F@freeswitch.org> <499B2691.70003@post.cz> <191c3a030902180609m52ad6b2es2aa84f87b994e3e2@mail.gmail.com> Message-ID: <191c3a030902180613x2f3b66av336f1bbcedf41703@mail.gmail.com> done, r12138 should give you the correct behavior On Wed, Feb 18, 2009 at 8:09 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > if the inbound calls are coming from a registration to a provider you will > have to set a context param in the gateway itself. > All inbound calls from a gateway registration are now associated with the > gateway they were registered with and inherit > the context from there. > > Maybe i'll change the default context of a gateway to be the default > context of it's host profile to avoid this issue. > > > > > On Tue, Feb 17, 2009 at 3:09 PM, Brian West wrote: > >> No problem. Just join us on IRC.. things move faster on there. >> >> /b >> >> On Feb 17, 2009, at 3:05 PM, kokoska.rokoska wrote: >> >> > If you have more hints, I be very happy :-) >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090218/b49a6ef7/attachment.html From anthony.minessale at gmail.com Wed Feb 18 06:14:20 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 18 Feb 2009 08:14:20 -0600 Subject: [Freeswitch-users] ipauth - directory In-Reply-To: References: <496E0187.8090805@kinetix.gr> <23DEEC73-4DC7-4056-902F-D74DD6644385@freeswitch.org> <496E068B.6050404@kinetix.gr> <191c3a030902160604r77089a6s5b3b9f3d07914218@mail.gmail.com> Message-ID: <191c3a030902180614p5de79174r56b21fe52cb0fe33@mail.gmail.com> yes that is correct. On Wed, Feb 18, 2009 at 8:03 AM, Leon de Rooij wrote: > Hi Anthony, > I tried your second option, but how does it work with xml-curl then ? As > far as I understand it, this doesn't work by doing a user-directory xml > lookup at INVITE time, or does it ? > > Or does it want to generate an ACL at FS startup and filling up all the > allow-nodes by polling the entire domain, filtering out all users with CIDR > entry and putting those in the ACL itself ? > > If so, is that the reason why FS tries (at startup) to POST to the > webserver with: > hostname=test§ion=directory&tag_name=domain&key_name=name&key_value= > test.com&domain=test.com&purpose=network-list > > ? > > Thanks & regards, > > Leon > > > On Feb 16, 2009, at 3:04 PM, Anthony Minessale wrote: > > you have 3 options. > on authenticated users, every tag in his account will be set on > each call from that authenticated user. > > 1) make them register, this sets the variables automatically > 2) use the ACL list with cidr= this > has the same effect with no auth needed. > 3) use some other way to differentiate the user and use the set_user > application in the dialplan to inherit that user's variables. > > > > On Mon, Feb 16, 2009 at 6:49 AM, Leon de Rooij wrote: > >> Hi all, >> >> I'd really like to know more about this too. >> >> Currently, I have two sip_profiles: >> >> - residential (where users can do authenticated registers and invites) >> - transit (where other users can do un-authenticated invites) >> >> Right now, FS is not aware of *who* is accessing the transit profile >> except for an acl that is set on this profile so unauthorized use is not >> possible. >> >> But what should I do when I want to allow multiple parties (from different >> IP addresses) to send their invites to the transit profile, and still be >> able to differentiate between them ? >> >> I'd like to set some variables, like an accountcode for example, on the >> basis of what IP address the INVITE originates from. >> >> So, is it possible to not use digest authentication, but still use a >> dialplan-directory user with IP= field or some such ? >> >> thanks a lot & kind regards, >> >> Leon de Rooij >> >> >> >> On Jan 14, 2009, at 4:36 PM, Apostolos Pantsiopoulos wrote: >> >> Yes I know that. But what does the "ip=" setting do? >> >> Brian West wrote: >> >> cidr= and the domains acl in acl.conf.xml then apply that ACL to the >> sofia profile. >> >> /b >> >> On Jan 14, 2009, at 9:15 AM, Apostolos Pantsiopoulos wrote: >> >> >> >> I noticed an "ip=" setting in the brian.xml sample file. >> The comments state that this is used for ipauth (IP based >> authentication?) >> >> What exactly is this setting. I cannot find anything in the wiki >> about it. >> Does it replace the use of the >> >> + ACL >> >> mechanism for IP authentication? >> >> >> _______________________________________________ >> Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >> >> >> >> -- >> ------------------------------------------- >> Apostolos Pantsiopoulos >> Kinetix Tele.com R & D >> email: regs at kinetix.gr >> ------------------------------------------- >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090218/b0c289a9/attachment-0001.html From edpimentl at gmail.com Wed Feb 18 06:21:24 2009 From: edpimentl at gmail.com (EdPimentl) Date: Wed, 18 Feb 2009 09:21:24 -0500 Subject: [Freeswitch-users] Skypiax on OS X In-Reply-To: <7b197bef0902180358s7c3ca246kf354beef2e72f4e6@mail.gmail.com> References: <7b197bef0902180358s7c3ca246kf354beef2e72f4e6@mail.gmail.com> Message-ID: <9dc4a1670902180621u350ed70bt6aa92731790014d0@mail.gmail.com> OSX can be loaded on any new Intel machines.. -E http://TwiTR.Me -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090218/1a6798e7/attachment.html From kokoska.rokoska at post.cz Wed Feb 18 06:24:51 2009 From: kokoska.rokoska at post.cz (kokoska rokoska) Date: Wed, 18 Feb 2009 15:24:51 +0100 Subject: [Freeswitch-users] call goes to wrong context In-Reply-To: <191c3a030902180609m52ad6b2es2aa84f87b994e3e2@mail.gmail.com> References: <499B1B42.7090000@post.cz> <4432BDD8-67A4-45B4-A013-7BB7C7E35D7F@freeswitch.org> <499B2691.70003@post.cz> <191c3a030902180609m52ad6b2es2aa84f87b994e3e2@mail.gmail.com> Message-ID: <499C1A33.8070002@post.cz> Anthony Minessale napsal(a): > if the inbound calls are coming from a registration to a provider you > will have to set a context param in the gateway itself. > All inbound calls from a gateway registration are now associated with > the gateway they were registered with and inherit > the context from there. > Thank you very much, Anthony, for your explanation! I got advice to setup context in gateway definition from Brian on IRC channel and it works, so I asume the the reason is some internal FS change :-) > Maybe i'll change the default context of a gateway to be the default > context of it's host profile to avoid this issue. > You'll be very glad :-) Because it is how it works in the past (at about 3-4 weeks old FS svn trunk) and even more - it is a little bit strange if xml_curl looks for "default" context which I don't have either... BTW: Is "default" context mandatory for FreeSWITCH (hardcoded somewhere in the code) or it is up to users decision how to name contexts? Tahnks once more, Anthony! Best regards, kokoska.rokoska From kokoska.rokoska at post.cz Wed Feb 18 06:26:45 2009 From: kokoska.rokoska at post.cz (kokoska rokoska) Date: Wed, 18 Feb 2009 15:26:45 +0100 Subject: [Freeswitch-users] call goes to wrong context In-Reply-To: <191c3a030902180613x2f3b66av336f1bbcedf41703@mail.gmail.com> References: <499B1B42.7090000@post.cz> <4432BDD8-67A4-45B4-A013-7BB7C7E35D7F@freeswitch.org> <499B2691.70003@post.cz> <191c3a030902180609m52ad6b2es2aa84f87b994e3e2@mail.gmail.com> <191c3a030902180613x2f3b66av336f1bbcedf41703@mail.gmail.com> Message-ID: <499C1AA5.5080807@post.cz> Anthony Minessale napsal(a): > done, > > r12138 should give you the correct behavior > Thank you very much, Anthony! Incredible speed :-) Best regards, kokoska.rokoska From moizchinoy at gmail.com Wed Feb 18 07:09:47 2009 From: moizchinoy at gmail.com (Moiz Chinoy) Date: Wed, 18 Feb 2009 07:09:47 -0800 (PST) Subject: [Freeswitch-users] FreeSwitch - PcoketSphinx Prompt Playback & Recognition Issue... Message-ID: <22080033.post@talk.nabble.com> Hi, I have downloaded and build the Freeswitch from http://files.freeswitch.org/freeswitch-1.0.3RC1.tar.gz on Windows XP. Everything built successfully. Then I configured PocketSphinx as described at http://wiki.freeswitch.org/wiki/Mod_pocketsphinx. The problem is that prompts (wave files) are not being played properly while testing the Pizza demo i.e it plays, stops then start playing again... I am using heaset for testing. Moreover, the recognition seems to be very poor. Any clue what might be the issue? Moiz Chinoy. Following is snippet from log: 2009-02-18 16:59:59 [DEBUG] switch_core_session.c:513 switch_core_session_perform_receive_message() Send signal sofia/internal/1000 at 192.168.16.63 [BREAK] 2009-02-18 16:59:59 [NOTICE] mod_spidermonkey.c:2041 session_answer() Channel [sofia/internal/1000 at 192.168.16.63] has been answered 2009-02-18 16:59:59 [DEBUG] switch_channel.c:179 switch_channel_audio_sync() sofia/internal/1000 at 192.168.16.63 receive message [AUDIO_SYNC] 2009-02-18 16:59:59 [DEBUG] sofia.c:2672 sofia_handle_sip_i_state() Channel sofia/internal/1000 at 192.168.16.63 entering state [completed] 2009-02-18 16:59:59 [DEBUG] sofia.c:2672 sofia_handle_sip_i_state() Channel sofia/internal/1000 at 192.168.16.63 entering state [ready] 2009-02-18 16:59:59 [DEBUG] switch_ivr_play_say.c:968 switch_ivr_play_file() Codec Activated L16 at 8000hz 1 channels 20ms 2009-02-18 16:59:59 [DEBUG] switch_core_io.c:654 switch_core_session_write_frame() sofia/internal/1000 at 192.168.16.63 receive message [TRANSCODING_NECESSARY] 2009-02-18 17:00:00 [INFO] switch_rtp.c:1422 rtp_common_read() Auto Changing port from 127.0.0.1:49194 to 192.168.16.63:49194 2009-02-18 17:00:02 [DEBUG] switch_ivr_play_say.c:1258 switch_ivr_play_file() done playing file 2009-02-18 17:00:11 [WARNING] switch_scheduler.c:114 task_thread_loop() Task was executed late by 2 seconds 1 heartbeat (core) 2009-02-18 17:00:18 [DEBUG] switch_core_media_bug.c:284 switch_core_media_bug_add() Attaching BUG to sofia/internal/1000 at 192.168.16.63 2009-02-18 17:00:18 [DEBUG] switch_core_io.c:234 switch_core_session_read_frame() sofia/internal/1000 at 192.168.16.63 receive message [TRANSCODING_NECESSARY] 2009-02-18 17:00:51 [WARNING] switch_scheduler.c:114 task_thread_loop() Task was executed late by 20 seconds 1 heartbeat (core) 2009-02-18 17:00:51 [DEBUG] switch_ivr_play_say.c:968 switch_ivr_play_file() Codec Activated L16 at 8000hz 1 channels 20ms 2009-02-18 17:00:58 [DEBUG] switch_ivr_play_say.c:1258 switch_ivr_play_file() done playing file 2009-02-18 17:00:58 [DEBUG] switch_core_io.c:234 switch_core_session_read_frame() sofia/internal/1000 at 192.168.16.63 receive message [TRANSCODING_NECESSARY] 2009-02-18 17:01:00 [DEBUG] mod_pocketsphinx.c:386 pocketsphinx_asr_get_results() Recognized: ????????????????, Score: 100 2009-02-18 17:01:00 [DEBUG] js_modules/SpeechTools.jm:150 console_log() ----XML: ???????????????? ???????????????? 2009-02-18 17:01:00 [DEBUG] js_modules/SpeechTools.jm:150 console_log() ----Heard [????????????????] 2009-02-18 17:01:00 [DEBUG] js_modules/SpeechTools.jm:150 console_log() ----Hit score 100/40/70 2009-02-18 17:01:00 [INFO] js_modules/SpeechTools.jm:150 console_log() ----???????????????? 2009-02-18 17:01:00 [DEBUG] js_modules/SpeechTools.jm:365 console_log() ----Testing [0] [0] ???????????????? =~ [Delivery:::Delivery] 2009-02-18 17:01:00 [DEBUG] js_modules/SpeechTools.jm:365 console_log() ----Testing [0] [1] ???????????????? =~ [Takeout:::Pickup] 2009-02-18 17:01:00 [DEBUG] js_modules/SpeechTools.jm:365 console_log() ----Testing [0] [2] ???????????????? =~ [Pickup:::Pickup] 2009-02-18 17:01:02 [DEBUG] mod_pocketsphinx.c:342 pocketsphinx_asr_resume() Manually Resuming 2009-02-18 17:01:02 [DEBUG] switch_ivr_play_say.c:968 switch_ivr_play_file() Codec Activated L16 at 8000hz 1 channels 20ms 2009-02-18 17:01:02 [DEBUG] switch_ivr_play_say.c:1258 switch_ivr_play_file() done playing file 2009-02-18 17:01:02 [DEBUG] switch_core_io.c:234 switch_core_session_read_frame() sofia/internal/1000 at 192.168.16.63 receive message [TRANSCODING_NECESSARY] 2009-02-18 17:01:03 [DEBUG] switch_ivr_play_say.c:968 switch_ivr_play_file() Codec Activated L16 at 8000hz 1 channels 20ms 2009-02-18 17:01:06 [DEBUG] switch_ivr_play_say.c:1258 switch_ivr_play_file() done playing file -- View this message in context: http://www.nabble.com/FreeSwitch---PcoketSphinx-Prompt-Playback---Recognition-Issue...-tp22080033p22080033.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From moizchinoy at gmail.com Wed Feb 18 07:17:03 2009 From: moizchinoy at gmail.com (Moiz Chinoy) Date: Wed, 18 Feb 2009 07:17:03 -0800 (PST) Subject: [Freeswitch-users] FreeSwitch - PcoketSphinx Prompt Playback & Recognition Issue... In-Reply-To: <22080033.post@talk.nabble.com> References: <22080033.post@talk.nabble.com> Message-ID: <22080857.post@talk.nabble.com> And one more thing... As soon as it recognizez TAKEOUT, freeswitch crashes... 2009-02-18 19:03:51 [DEBUG] js_modules/SpeechTools.jm:150 console_log() ----Heard [????] 2009-02-18 19:03:51 [DEBUG] js_modules/SpeechTools.jm:150 console_log() ----Hit score 98/40/70 2009-02-18 19:03:51 [INFO] js_modules/SpeechTools.jm:150 console_log() ----???? 2009-02-18 19:03:51 [DEBUG] switch_ivr_play_say.c:1258 switch_ivr_play_file() done playing file 2009-02-18 19:03:51 [DEBUG] js_modules/SpeechTools.jm:365 console_log() ----Testing [0] [0] ???? =~ [Delivery:::Delivery] 2009-02-18 19:03:51 [DEBUG] js_modules/SpeechTools.jm:365 console_log() ----Testing [0] [1] ???? =~ [Takeout:::Pickup] 2009-02-18 19:03:51 [DEBUG] js_modules/SpeechTools.jm:365 console_log() ----Testing [0] [2] ???? =~ [Pickup:::Pickup] 2009-02-18 19:03:51 [DEBUG] switch_core_io.c:234 switch_core_session_read_frame() sofia/internal/1000 at 192.168.16.63 receive message [TRANSCODING_NECESSARY] 2009-02-18 19:03:53 [DEBUG] mod_pocketsphinx.c:342 pocketsphinx_asr_resume() Manually Resuming 2009-02-18 19:03:53 [DEBUG] switch_ivr_play_say.c:968 switch_ivr_play_file() Codec Activated L16 at 8000hz 1 channels 20ms 2009-02-18 19:03:58 [DEBUG] switch_ivr_play_say.c:1258 switch_ivr_play_file() done playing file 2009-02-18 19:03:58 [DEBUG] switch_core_io.c:234 switch_core_session_read_frame() sofia/internal/1000 at 192.168.16.63 receive message [TRANSCODING_NECESSARY] 2009-02-18 19:03:59 [DEBUG] switch_ivr_play_say.c:968 switch_ivr_play_file() Codec Activated L16 at 8000hz 1 channels 20ms 2009-02-18 19:04:02 [DEBUG] switch_ivr_play_say.c:1258 switch_ivr_play_file() done playing file 2009-02-18 19:04:02 [DEBUG] switch_core_io.c:234 switch_core_session_read_frame() sofia/internal/1000 at 192.168.16.63 receive message [TRANSCODING_NECESSARY] 2009-02-18 19:04:09 [DEBUG] mod_pocketsphinx.c:342 pocketsphinx_asr_resume() Manually Resuming 2009-02-18 19:04:09 [DEBUG] switch_ivr_play_say.c:968 switch_ivr_play_file() Codec Activated L16 at 8000hz 1 channels 20ms 2009-02-18 19:04:14 [DEBUG] switch_ivr_play_say.c:1258 switch_ivr_play_file() done playing file 2009-02-18 19:04:14 [DEBUG] switch_core_io.c:234 switch_core_session_read_frame() sofia/internal/1000 at 192.168.16.63 receive message [TRANSCODING_NECESSARY] 2009-02-18 19:04:15 [DEBUG] switch_ivr_play_say.c:968 switch_ivr_play_file() Codec Activated L16 at 8000hz 1 channels 20ms 2009-02-18 19:04:16 [DEBUG] mod_pocketsphinx.c:386 pocketsphinx_asr_get_results() Recognized: ?????|, Score: 100 2009-02-18 19:04:16 [DEBUG] js_modules/SpeechTools.jm:150 console_log() ----XML: ?????| ?????| 2009-02-18 19:04:16 [DEBUG] js_modules/SpeechTools.jm:150 console_log() ----Heard [?????|] 2009-02-18 19:04:16 [DEBUG] js_modules/SpeechTools.jm:150 console_log() ----Hit score 100/40/70 2009-02-18 19:04:16 [INFO] js_modules/SpeechTools.jm:150 console_log() ----?????| 2009-02-18 19:04:16 [DEBUG] switch_ivr_play_say.c:1258 switch_ivr_play_file() done playing file 2009-02-18 19:04:16 [DEBUG] js_modules/SpeechTools.jm:365 console_log() ----Testing [0] [0] ?????| =~ [Delivery:::Delivery] 2009-02-18 19:04:16 [DEBUG] js_modules/SpeechTools.jm:365 console_log() ----Testing [0] [1] ?????| =~ [Takeout:::Pickup] 2009-02-18 19:04:16 [DEBUG] js_modules/SpeechTools.jm:365 console_log() ----Testing [0] [2] ?????| =~ [Pickup:::Pickup] 2009-02-18 19:04:16 [DEBUG] switch_core_io.c:234 switch_core_session_read_frame() sofia/internal/1000 at 192.168.16.63 receive message [TRANSCODING_NECESSARY] 2009-02-18 19:04:18 [DEBUG] mod_pocketsphinx.c:342 pocketsphinx_asr_resume() Manually Resuming 2009-02-18 19:04:18 [DEBUG] switch_ivr_play_say.c:968 switch_ivr_play_file() Codec Activated L16 at 8000hz 1 channels 20ms 2009-02-18 19:04:18 [DEBUG] switch_ivr_play_say.c:1258 switch_ivr_play_file() done playing file 2009-02-18 19:04:18 [DEBUG] switch_core_io.c:234 switch_core_session_read_frame() sofia/internal/1000 at 192.168.16.63 receive message [TRANSCODING_NECESSARY] 2009-02-18 19:04:19 [DEBUG] switch_ivr_play_say.c:968 switch_ivr_play_file() Codec Activated L16 at 8000hz 1 channels 20ms 2009-02-18 19:04:20 [DEBUG] mod_pocketsphinx.c:386 pocketsphinx_asr_get_results() Recognized: TAKEOUT, Score: 72 2009-02-18 19:04:20 [DEBUG] js_modules/SpeechTools.jm:150 console_log() ----XML: TAKEOUT TAKEOUT 2009-02-18 19:04:20 [DEBUG] js_modules/SpeechTools.jm:150 console_log() ----Heard [TAKEOUT] 2009-02-18 19:04:20 [DEBUG] js_modules/SpeechTools.jm:150 console_log() ----Hit score 72/40/70 2009-02-18 19:04:20 [INFO] js_modules/SpeechTools.jm:150 console_log() ----TAKEOUT 2009-02-18 19:04:20 [DEBUG] switch_ivr_play_say.c:1258 switch_ivr_play_file() done playing file 2009-02-18 19:04:20 [DEBUG] js_modules/SpeechTools.jm:365 console_log() ----Testing [0] [0] TAKEOUT =~ [Delivery:::Delivery] 2009-02-18 19:04:20 [DEBUG] js_modules/SpeechTools.jm:365 console_log() ----Testing [0] [1] TAKEOUT =~ [Takeout:::Pickup] 2009-02-18 19:04:20 [DEBUG] js_modules/SpeechTools.jm:365 console_log() ----Adding Pickup 2009-02-18 19:04:20 [DEBUG] js_modules/SpeechTools.jm:365 console_log() ----Testing [0] [2] TAKEOUT =~ [Pickup:::Pickup] 2009-02-18 19:04:20 [DEBUG] switch_core_io.c:234 switch_core_session_read_frame() sofia/internal/1000 at 192.168.16.63 receive message [TRANSCODING_NECESSARY] 2009-02-18 19:04:21 [DEBUG] js_modules/SpeechTools.jm:109 console_log() Unloading grammar pizza_order Moiz Chinoy. Moiz Chinoy wrote: > > Hi, > > I have downloaded and build the Freeswitch from > http://files.freeswitch.org/freeswitch-1.0.3RC1.tar.gz on Windows XP. > > Everything built successfully. > Then I configured PocketSphinx as described at > http://wiki.freeswitch.org/wiki/Mod_pocketsphinx. > The problem is that prompts (wave files) are not being played properly > while testing the Pizza demo i.e it plays, stops then start playing > again... > > I am using heaset for testing. > > Moreover, the recognition seems to be very poor. > > Any clue what might be the issue? > > Moiz Chinoy. > > Following is snippet from log: > > 2009-02-18 16:59:59 [DEBUG] switch_core_session.c:513 > switch_core_session_perform_receive_message() Send signal > sofia/internal/1000 at 192.168.16.63 [BREAK] > 2009-02-18 16:59:59 [NOTICE] mod_spidermonkey.c:2041 session_answer() > Channel [sofia/internal/1000 at 192.168.16.63] has been answered > 2009-02-18 16:59:59 [DEBUG] switch_channel.c:179 > switch_channel_audio_sync() sofia/internal/1000 at 192.168.16.63 receive > message [AUDIO_SYNC] > 2009-02-18 16:59:59 [DEBUG] sofia.c:2672 sofia_handle_sip_i_state() > Channel sofia/internal/1000 at 192.168.16.63 entering state [completed] > 2009-02-18 16:59:59 [DEBUG] sofia.c:2672 sofia_handle_sip_i_state() > Channel sofia/internal/1000 at 192.168.16.63 entering state [ready] > 2009-02-18 16:59:59 [DEBUG] switch_ivr_play_say.c:968 > switch_ivr_play_file() Codec Activated L16 at 8000hz 1 channels 20ms > 2009-02-18 16:59:59 [DEBUG] switch_core_io.c:654 > switch_core_session_write_frame() sofia/internal/1000 at 192.168.16.63 > receive message [TRANSCODING_NECESSARY] > 2009-02-18 17:00:00 [INFO] switch_rtp.c:1422 rtp_common_read() Auto > Changing port from 127.0.0.1:49194 to 192.168.16.63:49194 > 2009-02-18 17:00:02 [DEBUG] switch_ivr_play_say.c:1258 > switch_ivr_play_file() done playing file > 2009-02-18 17:00:11 [WARNING] switch_scheduler.c:114 task_thread_loop() > Task was executed late by 2 seconds 1 heartbeat (core) > 2009-02-18 17:00:18 [DEBUG] switch_core_media_bug.c:284 > switch_core_media_bug_add() Attaching BUG to > sofia/internal/1000 at 192.168.16.63 > 2009-02-18 17:00:18 [DEBUG] switch_core_io.c:234 > switch_core_session_read_frame() sofia/internal/1000 at 192.168.16.63 receive > message [TRANSCODING_NECESSARY] > 2009-02-18 17:00:51 [WARNING] switch_scheduler.c:114 task_thread_loop() > Task was executed late by 20 seconds 1 heartbeat (core) > 2009-02-18 17:00:51 [DEBUG] switch_ivr_play_say.c:968 > switch_ivr_play_file() Codec Activated L16 at 8000hz 1 channels 20ms > 2009-02-18 17:00:58 [DEBUG] switch_ivr_play_say.c:1258 > switch_ivr_play_file() done playing file > 2009-02-18 17:00:58 [DEBUG] switch_core_io.c:234 > switch_core_session_read_frame() sofia/internal/1000 at 192.168.16.63 receive > message [TRANSCODING_NECESSARY] > 2009-02-18 17:01:00 [DEBUG] mod_pocketsphinx.c:386 > pocketsphinx_asr_get_results() Recognized: ????????????????, Score: 100 > 2009-02-18 17:01:00 [DEBUG] js_modules/SpeechTools.jm:150 console_log() > ----XML: > > ???????????????? > ???????????????? > > 2009-02-18 17:01:00 [DEBUG] js_modules/SpeechTools.jm:150 console_log() > ----Heard [????????????????] > 2009-02-18 17:01:00 [DEBUG] js_modules/SpeechTools.jm:150 console_log() > ----Hit score 100/40/70 > 2009-02-18 17:01:00 [INFO] js_modules/SpeechTools.jm:150 console_log() > ----???????????????? > 2009-02-18 17:01:00 [DEBUG] js_modules/SpeechTools.jm:365 console_log() > ----Testing [0] [0] ???????????????? =~ [Delivery:::Delivery] > 2009-02-18 17:01:00 [DEBUG] js_modules/SpeechTools.jm:365 console_log() > ----Testing [0] [1] ???????????????? =~ [Takeout:::Pickup] > 2009-02-18 17:01:00 [DEBUG] js_modules/SpeechTools.jm:365 console_log() > ----Testing [0] [2] ???????????????? =~ [Pickup:::Pickup] > 2009-02-18 17:01:02 [DEBUG] mod_pocketsphinx.c:342 > pocketsphinx_asr_resume() Manually Resuming > 2009-02-18 17:01:02 [DEBUG] switch_ivr_play_say.c:968 > switch_ivr_play_file() Codec Activated L16 at 8000hz 1 channels 20ms > 2009-02-18 17:01:02 [DEBUG] switch_ivr_play_say.c:1258 > switch_ivr_play_file() done playing file > 2009-02-18 17:01:02 [DEBUG] switch_core_io.c:234 > switch_core_session_read_frame() sofia/internal/1000 at 192.168.16.63 receive > message [TRANSCODING_NECESSARY] > 2009-02-18 17:01:03 [DEBUG] switch_ivr_play_say.c:968 > switch_ivr_play_file() Codec Activated L16 at 8000hz 1 channels 20ms > 2009-02-18 17:01:06 [DEBUG] switch_ivr_play_say.c:1258 > switch_ivr_play_file() done playing file > -- View this message in context: http://www.nabble.com/FreeSwitch---PcoketSphinx-Prompt-Playback---Recognition-Issue...-tp22080033p22080857.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From brian at freeswitch.org Wed Feb 18 07:31:32 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 18 Feb 2009 09:31:32 -0600 Subject: [Freeswitch-users] FreeSwitch - PcoketSphinx Prompt Playback & Recognition Issue... In-Reply-To: <22080033.post@talk.nabble.com> References: <22080033.post@talk.nabble.com> Message-ID: Please update to SVN Trunk and try again... what are the specs on your machine? I have been testing PocketSphinx the past couple of days on linux again and its fine. /b On Feb 18, 2009, at 9:09 AM, Moiz Chinoy wrote: > > Hi, > > I have downloaded and build the Freeswitch from > http://files.freeswitch.org/freeswitch-1.0.3RC1.tar.gz on Windows XP. > > Everything built successfully. > Then I configured PocketSphinx as described at > http://wiki.freeswitch.org/wiki/Mod_pocketsphinx. > The problem is that prompts (wave files) are not being played > properly while > testing the Pizza demo i.e it plays, stops then start playing again... > > I am using heaset for testing. > > Moreover, the recognition seems to be very poor. > > Any clue what might be the issue? > > Moiz Chinoy. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090218/1e214424/attachment-0001.html From moizchinoy at gmail.com Wed Feb 18 07:49:29 2009 From: moizchinoy at gmail.com (Moiz Chinoy) Date: Wed, 18 Feb 2009 07:49:29 -0800 (PST) Subject: [Freeswitch-users] FreeSwitch - PcoketSphinx Prompt Playback & Recognition Issue... In-Reply-To: References: <22080033.post@talk.nabble.com> Message-ID: <22081606.post@talk.nabble.com> System specs: - Intel Core 2 Duo - 2.00 GHZ CPU - 1 Gb Ram I will download the latest from here http://files.freeswitch.org/freeswitch-snapshot.tar.gz and will try it. Moiz Chinoy. Brian West-3 wrote: > > Please update to SVN Trunk and try again... what are the specs on your > machine? > > I have been testing PocketSphinx the past couple of days on linux > again and its fine. > > /b > > > On Feb 18, 2009, at 9:09 AM, Moiz Chinoy wrote: > >> >> Hi, >> >> I have downloaded and build the Freeswitch from >> http://files.freeswitch.org/freeswitch-1.0.3RC1.tar.gz on Windows XP. >> >> Everything built successfully. >> Then I configured PocketSphinx as described at >> http://wiki.freeswitch.org/wiki/Mod_pocketsphinx. >> The problem is that prompts (wave files) are not being played >> properly while >> testing the Pizza demo i.e it plays, stops then start playing again... >> >> I am using heaset for testing. >> >> Moreover, the recognition seems to be very poor. >> >> Any clue what might be the issue? >> >> Moiz Chinoy. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/FreeSwitch---PcoketSphinx-Prompt-Playback---Recognition-Issue...-tp22080033p22081606.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From brian at freeswitch.org Wed Feb 18 07:55:11 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 18 Feb 2009 09:55:11 -0600 Subject: [Freeswitch-users] FreeSwitch - PcoketSphinx Prompt Playback & Recognition Issue... In-Reply-To: <22081606.post@talk.nabble.com> References: <22080033.post@talk.nabble.com> <22081606.post@talk.nabble.com> Message-ID: <3FEB93CB-9E5B-40FA-8DF7-1CEC8A364732@freeswitch.org> Please go get an SVN client for windows... svn update vs downloading the tarball every day will save bandwidth. ;) /b On Feb 18, 2009, at 9:49 AM, Moiz Chinoy wrote: > > System specs: > > - Intel Core 2 Duo > - 2.00 GHZ CPU > - 1 Gb Ram > > I will download the latest from here > http://files.freeswitch.org/freeswitch-snapshot.tar.gz and will try > it. > > Moiz Chinoy. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090218/8a40aff2/attachment.html From msc at freeswitch.org Wed Feb 18 07:58:10 2009 From: msc at freeswitch.org (Michael S Collins) Date: Wed, 18 Feb 2009 07:58:10 -0800 Subject: [Freeswitch-users] Anyone running FS from a Thumb Flash USB? In-Reply-To: <05B5115F398B44C28AFDB2DE0E1A9281@UVix> References: <9dc4a1670902170550r3071f65dn76112fdbebaeb6ba@mail.gmail.com><2d9149cd0902171543t1d1a5c9dqb23210ab8a2f3eeb@mail.gmail.com><87f2f3b90902171610r37679769h2b4f1c033e7b3e94@mail.gmail.com><2d9149cd0902171618s47640fb2vff3cc139b74c8f64@mail.gmail.com> <2d9149cd0902172309i1aa93fcej8fce68133028ee93@mail.gmail.com> <05B5115F398B44C28AFDB2DE0E1A9281@UVix> Message-ID: <45834C32-CF49-4973-9C7C-637D92C44EF4@freeswitch.org> Sent from my iPhone On Feb 18, 2009, at 3:55 AM, "UV" wrote: > Awesome work, Kristian! > And very much needed for the Freeswitch platform (to me, at least). > > A suggestion: if the FS team doesn't mind (after getting over the > naming > issue), it would be a good idea to put Kristian's latest blog entry > on the > FS Wiki. > Not a problem at all. We already link to Kristian's blog from our main page. I will give KK's new ISO a test drive and then put some directions on the wiki. -MC > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Kristian > Kielhofner > Sent: Wednesday, February 18, 2009 6:10 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Anyone running FS from a Thumb Flash > USB? > > Nik, > > Thanks but I'm not sure I want to take the credit (blame?) for > that! ;) > > On Tue, Feb 17, 2009 at 7:25 PM, Nik Middleton > wrote: >> Kristian, >> >> You're my hero, if I hadn't come across astlinux 3 years ago, I >> wouldn't >> be doing this stuff right now. Not too sure if that's a good thing >> though ;) >> > > -- > Kristian Kielhofner > http://blog.krisk.org > http://www.submityoursip.com > http://www.astlinux.org > http://www.star2star.com > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > No virus found in this incoming message. > Checked by AVG - www.avg.com > Version: 8.0.237 / Virus Database: 270.10.25/1956 - Release Date: > 02/17/09 > 07:07:00 > > No virus found in this outgoing message. > Checked by AVG - www.avg.com > Version: 8.0.237 / Virus Database: 270.10.25/1956 - Release Date: > 02/17/09 > 07:07:00 > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From gmaruzz at celliax.org Wed Feb 18 08:29:07 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Wed, 18 Feb 2009 17:29:07 +0100 Subject: [Freeswitch-users] Skypiax on OS X In-Reply-To: <9dc4a1670902180621u350ed70bt6aa92731790014d0@mail.gmail.com> References: <7b197bef0902180358s7c3ca246kf354beef2e72f4e6@mail.gmail.com> <9dc4a1670902180621u350ed70bt6aa92731790014d0@mail.gmail.com> Message-ID: <7b197bef0902180829h1f7c7c88od2c6f6d4a8f7aa29@mail.gmail.com> On Wed, Feb 18, 2009 at 3:21 PM, EdPimentl wrote: > OSX can be loaded on any new Intel machines.. > -E That's nice! How I can do it, I mean, in an easy way that will let me develop on it? It's just like installing a distro, or involves black magic? -gm On Wed, Feb 18, 2009 at 3:21 PM, EdPimentl wrote: > OSX can be loaded on any new Intel machines.. > -E > http://TwiTR.Me > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From javieraristizabal at gmail.com Wed Feb 18 09:35:39 2009 From: javieraristizabal at gmail.com (=?ISO-8859-1?Q?Javier_Aristiz=E1bal?=) Date: Wed, 18 Feb 2009 12:35:39 -0500 Subject: [Freeswitch-users] mod_fax and sending a fax In-Reply-To: <499C16C6.1000006@laposte.net> References: <49919822.3030101@laposte.net> <499C16C6.1000006@laposte.net> Message-ID: Hi Rod, i just play with rx_fax and work for me. I didn't work with tx_fax but i understand, that you need a .tiff file to send passthrough the rx_fax. Maybe that can help you regards javar -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090218/fa28bcb2/attachment.html From red.rain.seven at gmail.com Wed Feb 18 10:09:33 2009 From: red.rain.seven at gmail.com (Henry Huang) Date: Wed, 18 Feb 2009 10:09:33 -0800 Subject: [Freeswitch-users] Deployment information and use cases In-Reply-To: <1234916455.16581.49.camel@raul-laptop> References: <1234916455.16581.49.camel@raul-laptop> Message-ID: <59ad9ca10902181009p7cbc0fd1s9860aacde7f3a30c@mail.gmail.com> bandwidth.com has a service called phonebooth which is developed upon freeswitch. On Tue, Feb 17, 2009 at 4:20 PM, Raul Fragoso wrote: > Hello FreeSWITCHERS, > > My company is currently creating a suite of applications which uses > FreeSWITCH as the back-end for an IP-PBX solution. We currently have a > prospect to have our first customer installation - a governmental > department. That is a tender to have an IP-PBX installation to connect > their four office branches, each one with about 300 users - which I am > sure FreeSWITCH is able to handle. Since this is an official tender, > it's part of their protocol to ask about real sites using the product. > > Having said that, would you mind sharing some information about your > experience with FreeSWITCH deployments ? > > No need to give many details, but a short summary with company name (if > possible), when it was deployed, server equipment, number of users, > number of concurrent calls, what kind of functions and services are used > and overall capacity of the system. > > I would really appreciate if you can share that information. And if you > guys agree (and explicitly manifest your agreement), I can compile the > information in the FreeSWITCH wiki under a "Use Cases" page so it can > serve as a common reference as well. > > Kind regards, > > Raul Fragoso > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Henry Huang UniC Solution - Communication Unified VoIP & Open Source software Consultant -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090218/a17c9b49/attachment.html From pablosaro at gmail.com Wed Feb 18 10:19:37 2009 From: pablosaro at gmail.com (Pablo Hernan Saro) Date: Wed, 18 Feb 2009 16:19:37 -0200 Subject: [Freeswitch-users] Deployment information and use cases In-Reply-To: <59ad9ca10902181009p7cbc0fd1s9860aacde7f3a30c@mail.gmail.com> References: <1234916455.16581.49.camel@raul-laptop> <59ad9ca10902181009p7cbc0fd1s9860aacde7f3a30c@mail.gmail.com> Message-ID: <247f8100902181019m61ce803m5ed504d9b198dfb6@mail.gmail.com> Hi Raul, In my company (http://www.globant.com) we're using FreeSWITCH for high quality conference services, integrated with OpenSIPS (http://www.opensips.org) and Asterisk. Its performance is pretty good. Pablo On Wed, Feb 18, 2009 at 4:09 PM, Henry Huang wrote: > bandwidth.com has a service called phonebooth which is developed upon > freeswitch. > > > On Tue, Feb 17, 2009 at 4:20 PM, Raul Fragoso wrote: >> >> Hello FreeSWITCHERS, >> >> My company is currently creating a suite of applications which uses >> FreeSWITCH as the back-end for an IP-PBX solution. We currently have a >> prospect to have our first customer installation - a governmental >> department. That is a tender to have an IP-PBX installation to connect >> their four office branches, each one with about 300 users - which I am >> sure FreeSWITCH is able to handle. Since this is an official tender, >> it's part of their protocol to ask about real sites using the product. >> >> Having said that, would you mind sharing some information about your >> experience with FreeSWITCH deployments ? >> >> No need to give many details, but a short summary with company name (if >> possible), when it was deployed, server equipment, number of users, >> number of concurrent calls, what kind of functions and services are used >> and overall capacity of the system. >> >> I would really appreciate if you can share that information. And if you >> guys agree (and explicitly manifest your agreement), I can compile the >> information in the FreeSWITCH wiki under a "Use Cases" page so it can >> serve as a common reference as well. >> >> Kind regards, >> >> Raul Fragoso >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > -- > Henry Huang > UniC Solution - Communication Unified > VoIP & Open Source software Consultant > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From nik.middleton at noblesolutions.co.uk Wed Feb 18 11:26:48 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Wed, 18 Feb 2009 19:26:48 -0000 Subject: [Freeswitch-users] Originate and bridge with lua Message-ID: Hi Guys, It's not clear from the docs how I can originate a call from within an lua script This what works in js, Question. How do I instantiate a new session, do I use the execute to dial, and same for bridge? Regards, if (!first_session.ready()) var new_session = new Session(tdial-string); if (!first_session.ready()) { disp_call(DROP) exit(); new_session.answer(); if (new_session.ready()) { bridge(first_session, new_session); } -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090218/999668e0/attachment-0001.html From msc at freeswitch.org Wed Feb 18 11:41:07 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 18 Feb 2009 11:41:07 -0800 Subject: [Freeswitch-users] Originate and bridge with lua In-Reply-To: References: Message-ID: <87f2f3b90902181141q180680d5i4b54272eec4bf3de@mail.gmail.com> Nik, What are you building? I'm wondering if this is the correct approach for your application. You might be better off using the even socket and controlling your calls from a central point. -MC On Wed, Feb 18, 2009 at 11:26 AM, Nik Middleton wrote: > Hi Guys, > > > > It's not clear from the docs how I can originate a call from within an lua > script > > > > This what works in js, > > > > Question. How do I instantiate a new session, do I use the execute to dial, > and same for bridge? > > > > Regards, > > > > if (!first_session.ready()) > > > > var new_session = new Session(tdial-string); > > > > if (!first_session.ready()) { > > disp_call(DROP) > > exit(); > > > > > > > > new_session.answer(); > > > > if (new_session.ready()) { > > bridge(first_session, new_session); > > } > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From fax at virgintechnologies.com Wed Feb 18 11:46:04 2009 From: fax at virgintechnologies.com (Justin Miller) Date: Wed, 18 Feb 2009 19:46:04 +0000 Subject: [Freeswitch-users] Sending media streams to a media gateway Message-ID: Anyone else have any ideas on this? -----Original Message----- From: fax at virgintechnologies.com [mailto:fax at virgintechnologies.com] Sent: Tuesday, February 17, 2009 05:45 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Sending media streams to a media gateway I looked at that, but I think that will cause issues with the NAT traversal. Our phones will all be in external networks. I forgot to mention that. -----Original Message----- From: Anthony Minessale [mailto:anthony.minessale at gmail.com] Sent: Tuesday, February 17, 2009 05:34 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Sending media streams to a media gateway you could set the variable bypass_media to true before you call bridge that will negotiate a point to point media connection between the caller and callee On Tue, Feb 17, 2009 at 5:51 PM, wrote: I have Freeswitch running successfully with a fairly basic config. Nat traversal is working well on both the client and server side. I want to start running all RTP streams through a media gateway, and use Freeswitch for SIP registrations and signalling only. I believe that I need to have Freeswitch invite the SIP phone to send the RTP stream directly to the media gateway when a call starts. Where can I start with this? Does anyone have any example configs? Justin _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm mailto:MSN%3Aanthony_minessale at hotmail.com GTALK/JABBER/mailto:PAYPAL%3Aanthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference mailto:sip%3A888 at conference.freeswitch.org http://iax:guest at conference.freeswitch.org/888 mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090218/dc5ccdf2/attachment.html From msc at freeswitch.org Wed Feb 18 11:48:05 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 18 Feb 2009 11:48:05 -0800 Subject: [Freeswitch-users] FreeSwitch - PcoketSphinx Prompt Playback & Recognition Issue... In-Reply-To: <3FEB93CB-9E5B-40FA-8DF7-1CEC8A364732@freeswitch.org> References: <22080033.post@talk.nabble.com> <22081606.post@talk.nabble.com> <3FEB93CB-9E5B-40FA-8DF7-1CEC8A364732@freeswitch.org> Message-ID: <87f2f3b90902181148r798c5ed5n4ab30877a8e06c00@mail.gmail.com> On Wed, Feb 18, 2009 at 7:55 AM, Brian West wrote: > Please go get an SVN client for windows... svn update vs downloading the > tarball every day will save bandwidth. ;) > /b Use this for windows: http://tortoisesvn.tigris.org/ -MC From anthony.minessale at gmail.com Wed Feb 18 11:51:05 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 18 Feb 2009 13:51:05 -0600 Subject: [Freeswitch-users] Sending media streams to a media gateway In-Reply-To: References: Message-ID: <191c3a030902181151o4c635d04qd7e1adebea390e2a@mail.gmail.com> In that case you would need a sip proxy in place to rewrite the packets for the nat issue. There's nothing else we can really do. We have a way to do what you want but you are using it under circumstances we can't control. On Wed, Feb 18, 2009 at 1:46 PM, Justin Miller wrote: > Anyone else have any ideas on this? > > -----Original Message----- > *From:* fax at virgintechnologies.com [mailto:fax at virgintechnologies.com] > *Sent:* Tuesday, February 17, 2009 05:45 PM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Sending media streams to a media gateway > > I looked at that, but I think that will cause issues with the NAT > traversal. Our phones will all be in external networks. I forgot to > mention that. > > -----Original Message----- > *From:* Anthony Minessale [mailto:anthony.minessale at gmail.com] > *Sent:* Tuesday, February 17, 2009 05:34 PM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Sending media streams to a media gateway > > you could set the variable bypass_media to true before you call bridge > > > > that will negotiate a point to point media connection between the caller > and callee > > > On Tue, Feb 17, 2009 at 5:51 PM, wrote: > >> I have Freeswitch running successfully with a fairly basic config. Nat >> traversal is working well on both the client and server side. I want to >> start running all RTP streams through a media gateway, and use Freeswitch >> for SIP registrations and signalling only. >> I believe that I need to have Freeswitch invite the SIP phone to send the >> RTP stream directly to the media gateway when a call starts. Where can I >> start with this? Does anyone have any example configs? >> Justin >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > mailto:MSN%3Aanthony_minessale at hotmail.com > GTALK/JABBER/mailto:PAYPAL%3Aanthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > mailto:sip%3A888 at conference.freeswitch.org > http://iax:guest at conference.freeswitch.org/888 > mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org > pstn:213-799-1400 > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090218/9c0e6d83/attachment.html From nik.middleton at noblesolutions.co.uk Wed Feb 18 11:53:51 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Wed, 18 Feb 2009 19:53:51 -0000 Subject: [Freeswitch-users] Originate and bridge with lua In-Reply-To: <87f2f3b90902181141q180680d5i4b54272eec4bf3de@mail.gmail.com> References: <87f2f3b90902181141q180680d5i4b54272eec4bf3de@mail.gmail.com> Message-ID: I'm trying to build an emergency broadcasting solution. So I place a call, and have ivr in the lua script. But I also want to give them the option of speaking to someone. If they hit the option to speak to someone, while I can fire an event to originate a call, I'm not sure how I could bridge the 2 call legs. Regards, -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 18 February 2009 19:41 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Originate and bridge with lua Nik, What are you building? I'm wondering if this is the correct approach for your application. You might be better off using the even socket and controlling your calls from a central point. -MC On Wed, Feb 18, 2009 at 11:26 AM, Nik Middleton wrote: > Hi Guys, > > > > It's not clear from the docs how I can originate a call from within an lua > script > > > > This what works in js, > > > > Question. How do I instantiate a new session, do I use the execute to dial, > and same for bridge? > > > > Regards, > > > > if (!first_session.ready()) > > > > var new_session = new Session(tdial-string); > > > > if (!first_session.ready()) { > > disp_call(DROP) > > exit(); > > > > > > > > new_session.answer(); > > > > if (new_session.ready()) { > > bridge(first_session, new_session); > > } > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From msc at freeswitch.org Wed Feb 18 13:09:26 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 18 Feb 2009 13:09:26 -0800 Subject: [Freeswitch-users] Originate and bridge with lua In-Reply-To: References: <87f2f3b90902181141q180680d5i4b54272eec4bf3de@mail.gmail.com> Message-ID: <87f2f3b90902181309r38e0c480u5e3ba64b838f8985@mail.gmail.com> On Wed, Feb 18, 2009 at 11:53 AM, Nik Middleton wrote: > I'm trying to build an emergency broadcasting solution. > > So I place a call, and have ivr in the lua script. But I also want to > give them the option of speaking to someone. > > If they hit the option to speak to someone, while I can fire an event to > originate a call, I'm not sure how I could bridge the 2 call legs. > > Regards, So really, it's just an outbound IVR, no? Just for a specific purpose. I would recommend using the event socket and bgapi originate commands from a central program/script/controller thingy. Generate the calls and then drop them into a dialplan or script that controls them. I like to use the dialpan but it really does not matter. Using a script lets you make changes without doing a reloadxml command. In any case, your originate commands could be something like this: bgapi originate {myvar='myval',myvar2='myval2'}sofia/gateway/mygateway/user at domain 5555 Have extension 5555 do the gruntwork of confirming that you actually had a successful call, got a human on the line, etc. It can also handle failures that are not handled by the originate itself. (Depends on whether or not you ignore early media.) In any case, you've got a single dp entry that handles the mundane call handling. Then, if there is a human on the line, you can do something like this: Now you can write a plain Lua script that only has to handle the delivery of the message. You can handle a DTMF event and the callback function could use session:execute("bridge","agent") to connect the called party with your agent. Hope that helps. -MC From nik.middleton at noblesolutions.co.uk Wed Feb 18 13:27:54 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Wed, 18 Feb 2009 21:27:54 -0000 Subject: [Freeswitch-users] Originate and bridge with lua In-Reply-To: <87f2f3b90902181309r38e0c480u5e3ba64b838f8985@mail.gmail.com> References: <87f2f3b90902181141q180680d5i4b54272eec4bf3de@mail.gmail.com> <87f2f3b90902181309r38e0c480u5e3ba64b838f8985@mail.gmail.com> Message-ID: Hi Michael, Yes that's exactly what it boils down to, an outbound ivr. Everything is working perfectly, except the bridge to another number. Because of the nature of the beast the bridge needs to dial an external number (ie sofia/gateway/Mygateway/num) What I'm getting is: attempt to perform arithmetic on global 'sofia' (a nil value) regards -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 18 February 2009 21:09 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Originate and bridge with lua On Wed, Feb 18, 2009 at 11:53 AM, Nik Middleton wrote: > I'm trying to build an emergency broadcasting solution. > > So I place a call, and have ivr in the lua script. But I also want to > give them the option of speaking to someone. > > If they hit the option to speak to someone, while I can fire an event to > originate a call, I'm not sure how I could bridge the 2 call legs. > > Regards, So really, it's just an outbound IVR, no? Just for a specific purpose. I would recommend using the event socket and bgapi originate commands from a central program/script/controller thingy. Generate the calls and then drop them into a dialplan or script that controls them. I like to use the dialpan but it really does not matter. Using a script lets you make changes without doing a reloadxml command. In any case, your originate commands could be something like this: bgapi originate {myvar='myval',myvar2='myval2'}sofia/gateway/mygateway/user at domain 5555 Have extension 5555 do the gruntwork of confirming that you actually had a successful call, got a human on the line, etc. It can also handle failures that are not handled by the originate itself. (Depends on whether or not you ignore early media.) In any case, you've got a single dp entry that handles the mundane call handling. Then, if there is a human on the line, you can do something like this: Now you can write a plain Lua script that only has to handle the delivery of the message. You can handle a DTMF event and the callback function could use session:execute("bridge","agent") to connect the called party with your agent. Hope that helps. -MC _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From msc at freeswitch.org Wed Feb 18 13:43:18 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 18 Feb 2009 13:43:18 -0800 Subject: [Freeswitch-users] Originate and bridge with lua In-Reply-To: References: <87f2f3b90902181141q180680d5i4b54272eec4bf3de@mail.gmail.com> <87f2f3b90902181309r38e0c480u5e3ba64b838f8985@mail.gmail.com> Message-ID: <87f2f3b90902181343m6e807edfv6280cf01a70af9a8@mail.gmail.com> > Everything is working perfectly, except the bridge to another number. > Because of the nature of the beast the bridge needs to dial an external > number (ie sofia/gateway/Mygateway/num) What I'm getting is: > > attempt to perform arithmetic on global 'sofia' (a nil value) > Can you pastebin your Lua script? -MC From edpimentl at gmail.com Wed Feb 18 13:53:06 2009 From: edpimentl at gmail.com (EdPimentl) Date: Wed, 18 Feb 2009 16:53:06 -0500 Subject: [Freeswitch-users] Skypiax on OS X In-Reply-To: <7b197bef0902180829h1f7c7c88od2c6f6d4a8f7aa29@mail.gmail.com> References: <7b197bef0902180358s7c3ca246kf354beef2e72f4e6@mail.gmail.com> <9dc4a1670902180621u350ed70bt6aa92731790014d0@mail.gmail.com> <7b197bef0902180829h1f7c7c88od2c6f6d4a8f7aa29@mail.gmail.com> Message-ID: <9dc4a1670902181353s271e82a8u3f54b6e51a4c6f2d@mail.gmail.com> http://www.tech-recipes.com/rx/964/install_osx_tiger_on_intel_usb_drives_windows/ -E -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090218/43dac7bc/attachment.html From nik.middleton at noblesolutions.co.uk Wed Feb 18 13:56:12 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Wed, 18 Feb 2009 21:56:12 -0000 Subject: [Freeswitch-users] Originate and bridge with lua In-Reply-To: <87f2f3b90902181343m6e807edfv6280cf01a70af9a8@mail.gmail.com> References: <87f2f3b90902181141q180680d5i4b54272eec4bf3de@mail.gmail.com><87f2f3b90902181309r38e0c480u5e3ba64b838f8985@mail.gmail.com> <87f2f3b90902181343m6e807edfv6280cf01a70af9a8@mail.gmail.com> Message-ID: Sorted now thanks, it needed to be in the format session:execute("bridge", "{params}sofia/gateway/Mygateway/number"); key change was '"' Now I've converted my js script to lua going to run some tests tomorrow. I sincerely hope it'll handle more than the 10 calls js would break at. Here's my current setup External prog generates bgapi calls via socket and calls originate with name of lua script also passed. Lua does IVR and then bridges where required. It also fires back an event to show result of call. Astererisk happily does around 200 calls, I'm hoping FS will do better or I've just been wasting my time. Is there a more efficient way of doing this? Regards, -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 18 February 2009 21:43 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Originate and bridge with lua > Everything is working perfectly, except the bridge to another number. > Because of the nature of the beast the bridge needs to dial an external > number (ie sofia/gateway/Mygateway/num) What I'm getting is: > > attempt to perform arithmetic on global 'sofia' (a nil value) > Can you pastebin your Lua script? -MC _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From brian at freeswitch.org Wed Feb 18 14:02:01 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 18 Feb 2009 16:02:01 -0600 Subject: [Freeswitch-users] Originate and bridge with lua In-Reply-To: References: <87f2f3b90902181141q180680d5i4b54272eec4bf3de@mail.gmail.com><87f2f3b90902181309r38e0c480u5e3ba64b838f8985@mail.gmail.com> <87f2f3b90902181343m6e807edfv6280cf01a70af9a8@mail.gmail.com> Message-ID: <3BEB0C42-9F44-497E-9CEB-F9049BD7E147@freeswitch.org> Learn C and write it all in C. /b On Feb 18, 2009, at 3:56 PM, Nik Middleton wrote: > Astererisk happily does around 200 calls, I'm hoping FS will do better > or I've just been wasting my time. Is there a more efficient way of > doing this? From anthony.minessale at gmail.com Wed Feb 18 14:06:33 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 18 Feb 2009 16:06:33 -0600 Subject: [Freeswitch-users] Originate and bridge with lua In-Reply-To: References: <87f2f3b90902181141q180680d5i4b54272eec4bf3de@mail.gmail.com> <87f2f3b90902181309r38e0c480u5e3ba64b838f8985@mail.gmail.com> <87f2f3b90902181343m6e807edfv6280cf01a70af9a8@mail.gmail.com> Message-ID: <191c3a030902181406g659a51d9j89a3dcf4502d9b40@mail.gmail.com> You want to make it even more efficient? when they press 1, session:execute("transfer", ""); Then, put an extension in your dialplan to match and do the bridge. Then you can exit the script and only run the script when you need it. Your problem with js was the same issue, you should have been doing something similar there too. BTW, If you make another comparison to asterisk comment, I will never answer another email from you again I don't have time for that crap. On Wed, Feb 18, 2009 at 3:56 PM, Nik Middleton < nik.middleton at noblesolutions.co.uk> wrote: > Sorted now thanks, it needed to be in the format > > session:execute("bridge", "{params}sofia/gateway/Mygateway/number"); > > key change was '"' > > Now I've converted my js script to lua going to run some tests tomorrow. > > I sincerely hope it'll handle more than the 10 calls js would break at. > > > Here's my current setup > > External prog generates bgapi calls via socket and calls originate with > name of lua script also passed. > > Lua does IVR and then bridges where required. It also fires back an > event to show result of call. > > Astererisk happily does around 200 calls, I'm hoping FS will do better > or I've just been wasting my time. Is there a more efficient way of > doing this? > > > Regards, > > > > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Michael Collins > Sent: 18 February 2009 21:43 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Originate and bridge with lua > > > Everything is working perfectly, except the bridge to another number. > > Because of the nature of the beast the bridge needs to dial an > external > > number (ie sofia/gateway/Mygateway/num) What I'm getting is: > > > > attempt to perform arithmetic on global 'sofia' (a nil value) > > > Can you pastebin your Lua script? > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090218/dcb3e7bb/attachment.html From anthony.minessale at gmail.com Wed Feb 18 15:39:14 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 18 Feb 2009 17:39:14 -0600 Subject: [Freeswitch-users] Originate and bridge with lua In-Reply-To: <191c3a030902181406g659a51d9j89a3dcf4502d9b40@mail.gmail.com> References: <87f2f3b90902181141q180680d5i4b54272eec4bf3de@mail.gmail.com> <87f2f3b90902181309r38e0c480u5e3ba64b838f8985@mail.gmail.com> <87f2f3b90902181343m6e807edfv6280cf01a70af9a8@mail.gmail.com> <191c3a030902181406g659a51d9j89a3dcf4502d9b40@mail.gmail.com> Message-ID: <191c3a030902181539g637b01fke2582e830f602033@mail.gmail.com> i replied to your last private message and it was returned as undeliverable. overzealous spam server? Can you add my account to your whitelist? On Wed, Feb 18, 2009 at 4:06 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > You want to make it even more efficient? > when they press 1, > session:execute("transfer", ""); > > Then, put an extension in your dialplan to match and do the > bridge. > Then you can exit the script and only run the script when you need it. > > Your problem with js was the same issue, you should have been doing > something similar there too. > > BTW, > If you make another comparison to asterisk comment, I will never answer > another email from you again I don't have time for that crap. > > > > > > On Wed, Feb 18, 2009 at 3:56 PM, Nik Middleton < > nik.middleton at noblesolutions.co.uk> wrote: > >> Sorted now thanks, it needed to be in the format >> >> session:execute("bridge", "{params}sofia/gateway/Mygateway/number"); >> >> key change was '"' >> >> Now I've converted my js script to lua going to run some tests tomorrow. >> >> I sincerely hope it'll handle more than the 10 calls js would break at. >> >> >> Here's my current setup >> >> External prog generates bgapi calls via socket and calls originate with >> name of lua script also passed. >> >> Lua does IVR and then bridges where required. It also fires back an >> event to show result of call. >> >> Astererisk happily does around 200 calls, I'm hoping FS will do better >> or I've just been wasting my time. Is there a more efficient way of >> doing this? >> >> >> Regards, >> >> >> >> >> >> -----Original Message----- >> From: freeswitch-users-bounces at lists.freeswitch.org >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of >> Michael Collins >> Sent: 18 February 2009 21:43 >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] Originate and bridge with lua >> >> > Everything is working perfectly, except the bridge to another number. >> > Because of the nature of the beast the bridge needs to dial an >> external >> > number (ie sofia/gateway/Mygateway/num) What I'm getting is: >> > >> > attempt to perform arithmetic on global 'sofia' (a nil value) >> > >> Can you pastebin your Lua script? >> -MC >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090218/6f945f3a/attachment-0001.html From nik.middleton at noblesolutions.co.uk Wed Feb 18 15:56:53 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Wed, 18 Feb 2009 23:56:53 -0000 Subject: [Freeswitch-users] Originate and bridge with lua In-Reply-To: <191c3a030902181539g637b01fke2582e830f602033@mail.gmail.com> References: <87f2f3b90902181141q180680d5i4b54272eec4bf3de@mail.gmail.com><87f2f3b90902181309r38e0c480u5e3ba64b838f8985@mail.gmail.com><87f2f3b90902181343m6e807edfv6280cf01a70af9a8@mail.gmail.com><191c3a030902181406g659a51d9j89a3dcf4502d9b40@mail.gmail.com> <191c3a030902181539g637b01fke2582e830f602033@mail.gmail.com> Message-ID: Done Seems it had a spam score of 2 for some reason Regards, ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: 18 February 2009 23:39 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Originate and bridge with lua i replied to your last private message and it was returned as undeliverable. overzealous spam server? Can you add my account to your whitelist? On Wed, Feb 18, 2009 at 4:06 PM, Anthony Minessale wrote: You want to make it even more efficient? when they press 1, session:execute("transfer", ""); Then, put an extension in your dialplan to match and do the bridge. Then you can exit the script and only run the script when you need it. Your problem with js was the same issue, you should have been doing something similar there too. BTW, If you make another comparison to asterisk comment, I will never answer another email from you again I don't have time for that crap. On Wed, Feb 18, 2009 at 3:56 PM, Nik Middleton wrote: Sorted now thanks, it needed to be in the format session:execute("bridge", "{params}sofia/gateway/Mygateway/number"); key change was '"' Now I've converted my js script to lua going to run some tests tomorrow. I sincerely hope it'll handle more than the 10 calls js would break at. Here's my current setup External prog generates bgapi calls via socket and calls originate with name of lua script also passed. Lua does IVR and then bridges where required. It also fires back an event to show result of call. Astererisk happily does around 200 calls, I'm hoping FS will do better or I've just been wasting my time. Is there a more efficient way of doing this? Regards, -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 18 February 2009 21:43 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Originate and bridge with lua > Everything is working perfectly, except the bridge to another number. > Because of the nature of the beast the bridge needs to dial an external > number (ie sofia/gateway/Mygateway/num) What I'm getting is: > > attempt to perform arithmetic on global 'sofia' (a nil value) > Can you pastebin your Lua script? -MC _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090218/cfc7b385/attachment.html From philip.patterson at gmail.com Wed Feb 18 18:00:07 2009 From: philip.patterson at gmail.com (Philip Patterson) Date: Wed, 18 Feb 2009 22:00:07 -0400 Subject: [Freeswitch-users] Missing file for 1.0.3 Message-ID: Hi All. Have a fresh server and going to install FS on it. Went to the download page (http://wiki.freeswitch.org/wiki/Installation_Guide) and tried to download the "Phoenix" build, which is supposed to be found at http://files.freeswitch.org/freeswitch-1.0.3.tar.gz but that file is nowhere to be found. Did the Wiki get updated before the file was uploaded, or is there something else going on? Philip -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090218/5a36e6cc/attachment.html From carlos.talbot at gmail.com Wed Feb 18 18:53:29 2009 From: carlos.talbot at gmail.com (Carlos Talbot) Date: Wed, 18 Feb 2009 20:53:29 -0600 Subject: [Freeswitch-users] Skypiax (Skype Network endpoint) for FreeSWITCH In-Reply-To: <7b197bef0902131015k456dee87x8f0f1b7059f1b49c@mail.gmail.com> References: <7b197bef0902131015k456dee87x8f0f1b7059f1b49c@mail.gmail.com> Message-ID: <5800526b0902181853h32de5535l6c0f9d7067a69cf0@mail.gmail.com> Giovannia, great work on mod_skypiax. I've been testing it under Windows and it sounds great including PSTN calls. I plan to include it as part of the Windows MSI build. One question I have, is ringback suppose to work with mod_skypiax? Whenever I dial a number I get a few seconds of dead air before the call is answered. I've tried adding ringback and transfer_ringback into the dialplan just before the bridge command but no go. Am I missing something? Thanks. regards, Carlos -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090218/f4740bd6/attachment.html From brian at freeswitch.org Wed Feb 18 18:55:50 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 18 Feb 2009 20:55:50 -0600 Subject: [Freeswitch-users] Missing file for 1.0.3 In-Reply-To: References: Message-ID: Looks like someone jumped the gun... just get SVN trunk... we are in the process of release right now. /b On Feb 18, 2009, at 8:00 PM, Philip Patterson wrote: > Hi All. > > Have a fresh server and going to install FS on it. Went to the > download page (http://wiki.freeswitch.org/wiki/Installation_Guide) > and tried to download the "Phoenix" build, which is supposed to be > found at http://files.freeswitch.org/freeswitch-1.0.3.tar.gz but > that file is nowhere to be found. Did the Wiki get updated before > the file was uploaded, or is there something else going on? > > Philip -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090218/54efa178/attachment.html From brian at freeswitch.org Wed Feb 18 18:57:37 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 18 Feb 2009 20:57:37 -0600 Subject: [Freeswitch-users] Skypiax (Skype Network endpoint) for FreeSWITCH In-Reply-To: <5800526b0902181853h32de5535l6c0f9d7067a69cf0@mail.gmail.com> References: <7b197bef0902131015k456dee87x8f0f1b7059f1b49c@mail.gmail.com> <5800526b0902181853h32de5535l6c0f9d7067a69cf0@mail.gmail.com> Message-ID: <9D0D74A5-7E6C-41FD-9A52-0430101C909E@freeswitch.org> Thats one I think Anthm will need to chime in on... maybe skypiax isn't sending the right indications to cause the core to trigger the ringback. /b On Feb 18, 2009, at 8:53 PM, Carlos Talbot wrote: > Giovannia, > > great work on mod_skypiax. I've been testing it under Windows and it > sounds great including PSTN calls. I plan to include it as part of > the Windows MSI build. > > One question I have, is ringback suppose to work with mod_skypiax? > Whenever I dial a number I get a few seconds of dead air before the > call is answered. I've tried adding ringback and transfer_ringback > into the dialplan just before the bridge command but no go. Am I > missing something? Thanks. > > regards, > > Carlos > > From raul at etellicom.com Wed Feb 18 20:13:13 2009 From: raul at etellicom.com (Raul Fragoso) Date: Thu, 19 Feb 2009 01:13:13 -0300 Subject: [Freeswitch-users] Deployment information and use cases In-Reply-To: <247f8100902181019m61ce803m5ed504d9b198dfb6@mail.gmail.com> References: <1234916455.16581.49.camel@raul-laptop> <59ad9ca10902181009p7cbc0fd1s9860aacde7f3a30c@mail.gmail.com> <247f8100902181019m61ce803m5ed504d9b198dfb6@mail.gmail.com> Message-ID: <1235016793.22050.0.camel@raul-laptop> Thanks guys, this is very useful information. Anyone else willing to share your experience ? Regards, Raul On Wed, 2009-02-18 at 16:19 -0200, Pablo Hernan Saro wrote: > Hi Raul, > > In my company (http://www.globant.com) we're using FreeSWITCH for high > quality conference services, integrated with OpenSIPS > (http://www.opensips.org) and Asterisk. Its performance is pretty > good. > > Pablo > > On Wed, Feb 18, 2009 at 4:09 PM, Henry Huang wrote: > > bandwidth.com has a service called phonebooth which is developed upon > > freeswitch. > > > > > > On Tue, Feb 17, 2009 at 4:20 PM, Raul Fragoso wrote: > >> > >> Hello FreeSWITCHERS, > >> > >> My company is currently creating a suite of applications which uses > >> FreeSWITCH as the back-end for an IP-PBX solution. We currently have a > >> prospect to have our first customer installation - a governmental > >> department. That is a tender to have an IP-PBX installation to connect > >> their four office branches, each one with about 300 users - which I am > >> sure FreeSWITCH is able to handle. Since this is an official tender, > >> it's part of their protocol to ask about real sites using the product. > >> > >> Having said that, would you mind sharing some information about your > >> experience with FreeSWITCH deployments ? > >> > >> No need to give many details, but a short summary with company name (if > >> possible), when it was deployed, server equipment, number of users, > >> number of concurrent calls, what kind of functions and services are used > >> and overall capacity of the system. > >> > >> I would really appreciate if you can share that information. And if you > >> guys agree (and explicitly manifest your agreement), I can compile the > >> information in the FreeSWITCH wiki under a "Use Cases" page so it can > >> serve as a common reference as well. > >> > >> Kind regards, > >> > >> Raul Fragoso > >> > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > > > -- > > Henry Huang > > UniC Solution - Communication Unified > > VoIP & Open Source software Consultant > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From msc at freeswitch.org Wed Feb 18 20:37:49 2009 From: msc at freeswitch.org (Michael S Collins) Date: Wed, 18 Feb 2009 20:37:49 -0800 Subject: [Freeswitch-users] Missing file for 1.0.3 Message-ID: <9384F444-E628-4769-A507-3693C06BB985@freeswitch.org> Sent from my iPhone On Feb 18, 2009, at 6:00 PM, Philip Patterson wrote: > Hi All. > > Have a fresh server and going to install FS on it. Went to the > download page (http://wiki.freeswitch.org/wiki/Installation_Guide) > and tried to download the "Phoenix" build, which is supposed to be > found at http://files.freeswitch.org/freeswitch-1.0.3.tar.gz but > that file is nowhere to be found. Did the Wiki get updated before > the file was uploaded, or is there something else going on? Oops, my bad. That's exactly what happened. The file is actually 1.0.3RC1.tar.gz, although 1.0.3 should hit the server in the next day or so. Stay tuned! -MC > > > Philip > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090218/c914b016/attachment.html From msc at freeswitch.org Wed Feb 18 21:39:29 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 18 Feb 2009 21:39:29 -0800 Subject: [Freeswitch-users] Skypiax (Skype Network endpoint) for FreeSWITCH In-Reply-To: <9D0D74A5-7E6C-41FD-9A52-0430101C909E@freeswitch.org> References: <7b197bef0902131015k456dee87x8f0f1b7059f1b49c@mail.gmail.com> <5800526b0902181853h32de5535l6c0f9d7067a69cf0@mail.gmail.com> <9D0D74A5-7E6C-41FD-9A52-0430101C909E@freeswitch.org> Message-ID: <87f2f3b90902182139q66393955r1cd6ef12bb496291@mail.gmail.com> On Wed, Feb 18, 2009 at 6:57 PM, Brian West wrote: > Thats one I think Anthm will need to chime in on... maybe skypiax > isn't sending the right indications to cause the core to trigger the > ringback. > > /b > Out of curiosity, you might try this trick: See also: http://wiki.freeswitch.org/wiki/Channel_Variables#instant_ringback I'm curious to know how that works with your setup. -MC From carlos.talbot at gmail.com Wed Feb 18 22:00:00 2009 From: carlos.talbot at gmail.com (Carlos Talbot) Date: Thu, 19 Feb 2009 00:00:00 -0600 Subject: [Freeswitch-users] Skypiax (Skype Network endpoint) for FreeSWITCH In-Reply-To: <87f2f3b90902182139q66393955r1cd6ef12bb496291@mail.gmail.com> References: <7b197bef0902131015k456dee87x8f0f1b7059f1b49c@mail.gmail.com> <5800526b0902181853h32de5535l6c0f9d7067a69cf0@mail.gmail.com> <9D0D74A5-7E6C-41FD-9A52-0430101C909E@freeswitch.org> <87f2f3b90902182139q66393955r1cd6ef12bb496291@mail.gmail.com> Message-ID: <5800526b0902182200l4da4be2dyfebfc3542aef67b4@mail.gmail.com> That did it! I had to add both lines below in order for it to work: Now, suppose I call a number that's busy...do I hear a ringback followed by a busy signal? On Wed, Feb 18, 2009 at 11:39 PM, Michael Collins wrote: > On Wed, Feb 18, 2009 at 6:57 PM, Brian West wrote: > > Thats one I think Anthm will need to chime in on... maybe skypiax > > isn't sending the right indications to cause the core to trigger the > > ringback. > > > > /b > > > Out of curiosity, you might try this trick: > > See also: > http://wiki.freeswitch.org/wiki/Channel_Variables#instant_ringback > > I'm curious to know how that works with your setup. > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090219/b081dea4/attachment.html From gmaruzz at celliax.org Wed Feb 18 22:07:36 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Thu, 19 Feb 2009 07:07:36 +0100 Subject: [Freeswitch-users] Skypiax (Skype Network endpoint) for FreeSWITCH In-Reply-To: <87f2f3b90902182139q66393955r1cd6ef12bb496291@mail.gmail.com> References: <7b197bef0902131015k456dee87x8f0f1b7059f1b49c@mail.gmail.com> <5800526b0902181853h32de5535l6c0f9d7067a69cf0@mail.gmail.com> <9D0D74A5-7E6C-41FD-9A52-0430101C909E@freeswitch.org> <87f2f3b90902182139q66393955r1cd6ef12bb496291@mail.gmail.com> Message-ID: <7b197bef0902182207q2b8a1b3bj1abac4aad768aa7@mail.gmail.com> Carlos, Maybe the solution Michael is suggesting could work. For sure you are not missing anything' Brian is right: rimgback and early media are to be added to skypiax. They're on the TODO section of the wiki :-) I'll be all day at a customer's premise, I'll add it this evening, late afternoon for you. Would be *very* nice to have skypiax in MSI, thank you! On 2/19/09, Michael Collins wrote: > On Wed, Feb 18, 2009 at 6:57 PM, Brian West wrote: >> Thats one I think Anthm will need to chime in on... maybe skypiax >> isn't sending the right indications to cause the core to trigger the >> ringback. >> >> /b >> > Out of curiosity, you might try this trick: > > See also: > http://wiki.freeswitch.org/wiki/Channel_Variables#instant_ringback > > I'm curious to know how that works with your setup. > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Sent from my mobile device Sincerely, Giovanni Maruzzelli ========================================= www.celliax.org via Pierlombardo 9, 20135 Milano Italy gmaruzz at celliax dot org Cell : +39-347-2665618 Fax : +39-02-87390039 From brian at freeswitch.org Wed Feb 18 22:42:37 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 19 Feb 2009 00:42:37 -0600 Subject: [Freeswitch-users] Missing file for 1.0.3 In-Reply-To: <9384F444-E628-4769-A507-3693C06BB985@freeswitch.org> References: <9384F444-E628-4769-A507-3693C06BB985@freeswitch.org> Message-ID: <72414A49-FB9B-43A1-99AB-5937D8721F25@freeswitch.org> go try now! ;) /b From brian at freeswitch.org Wed Feb 18 22:52:15 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 19 Feb 2009 00:52:15 -0600 Subject: [Freeswitch-users] Skypiax (Skype Network endpoint) for FreeSWITCH In-Reply-To: <7b197bef0902182207q2b8a1b3bj1abac4aad768aa7@mail.gmail.com> References: <7b197bef0902131015k456dee87x8f0f1b7059f1b49c@mail.gmail.com> <5800526b0902181853h32de5535l6c0f9d7067a69cf0@mail.gmail.com> <9D0D74A5-7E6C-41FD-9A52-0430101C909E@freeswitch.org> <87f2f3b90902182139q66393955r1cd6ef12bb496291@mail.gmail.com> <7b197bef0902182207q2b8a1b3bj1abac4aad768aa7@mail.gmail.com> Message-ID: <0E2F27BF-565D-42F5-B453-6A34D0EBEEB2@freeswitch.org> It has to be in trunk to be in the MSI... I don't want to cause confusion ... Now that 1.0.3 is tagged we can put it in trunk? /b On Feb 19, 2009, at 12:07 AM, Giovanni Maruzzelli wrote: > Would be *very* nice to have skypiax in MSI, thank you! From moizchinoy at gmail.com Wed Feb 18 23:29:41 2009 From: moizchinoy at gmail.com (Moiz Chinoy) Date: Thu, 19 Feb 2009 11:29:41 +0400 Subject: [Freeswitch-users] FreeSwitch - PcoketSphinx Prompt Playback & Recognition Issue... In-Reply-To: <87f2f3b90902181148r798c5ed5n4ab30877a8e06c00@mail.gmail.com> References: <22080033.post@talk.nabble.com> <22081606.post@talk.nabble.com> <3FEB93CB-9E5B-40FA-8DF7-1CEC8A364732@freeswitch.org> <87f2f3b90902181148r798c5ed5n4ab30877a8e06c00@mail.gmail.com> Message-ID: <29b888f80902182329o578a66d5x66454355d7575f48@mail.gmail.com> Thanks for your help.... I have downloaded the latest build and tried... Often in the log I see cryptic characters in the XML part returned by ASR. Is it silence or nose?? If yes is there any way we can control it? Prompt playback problem is still there... So far I am only able to get TAKEOUT and YES recognized and then the application crashes with Windows error: AppName: freeswitch.exe AppVer: 0.0.0.0 ModName: sphinxbase.dll ModVer: 0.0.0.0 Offset: 00053791 Below is log snippet: o=FreeSWITCH 1235002217 1235002218 IN IP4 192.168.16.63 s=FreeSWITCH c=IN IP4 192.168.16.63 t=0 0 m=audio 25190 RTP/AVP 3 101 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 2009-02-19 11:10:07 [DEBUG] switch_core_session.c:523 switch_core_session_perform_receive_message() Send signal sofia/internal/1000 at 192.168.16.63 [BREAK] 2009-02-19 11:10:07 [NOTICE] mod_spidermonkey.c:2041 session_answer() Channel [sofia/internal/1000 at 192.168.16.63] has been answered 2009-02-19 11:10:07 [DEBUG] switch_channel.c:179 switch_channel_audio_sync() sofia/internal/1000 at 192.168.16.63 receive message [AUDIO_SYNC] 2009-02-19 11:10:07 [DEBUG] switch_ivr_play_say.c:989 switch_ivr_play_file() Codec Activated L16 at 8000hz 1 channels 20ms 2009-02-19 11:10:07 [DEBUG] switch_core_io.c:652 switch_core_session_write_frame() sofia/internal/1000 at 192.168.16.63 receive message [TRANSCODING_NECESSARY] 2009-02-19 11:10:07 [DEBUG] sofia.c:2725 sofia_handle_sip_i_state() Channel sofia/internal/1000 at 192.168.16.63 entering state [completed] 2009-02-19 11:10:07 [DEBUG] sofia.c:2725 sofia_handle_sip_i_state() Channel sofia/internal/1000 at 192.168.16.63 entering state [ready] 2009-02-19 11:10:07 [INFO] switch_rtp.c:1422 rtp_common_read() Auto Changing port from 127.0.0.1:49166 to 192.168.16.63:49166 2009-02-19 11:10:09 [DEBUG] switch_ivr_play_say.c:1279 switch_ivr_play_file() done playing file 2009-02-19 11:10:12 [DEBUG] switch_core_media_bug.c:297 switch_core_media_bug_add() Attaching BUG to sofia/internal/1000 at 192.168.16.63 2009-02-19 11:10:12 [DEBUG] switch_core_io.c:234 switch_core_session_read_frame() sofia/internal/1000 at 192.168.16.63 receive message [TRANSCODING_NECESSARY] 2009-02-19 11:10:12 [DEBUG] switch_ivr_play_say.c:989 switch_ivr_play_file() Codec Activated L16 at 8000hz 1 channels 20ms 2009-02-19 11:10:15 [DEBUG] switch_ivr_play_say.c:1279 switch_ivr_play_file() done playing file 2009-02-19 11:10:15 [DEBUG] switch_core_io.c:234 switch_core_session_read_frame() sofia/internal/1000 at 192.168.16.63 receive message [TRANSCODING_NECESSARY] 2009-02-19 11:10:18 [DEBUG] mod_pocketsphinx.c:387 pocketsphinx_asr_get_results() Recognized: TAKEOUT, Score: 62 2009-02-19 11:10:18 [DEBUG] js_modules/SpeechTools.jm:150 console_log() ----XML: TAKEOUT TAKEOUT 2009-02-19 11:10:18 [DEBUG] js_modules/SpeechTools.jm:150 console_log() ----Heard [TAKEOUT] 2009-02-19 11:10:18 [DEBUG] js_modules/SpeechTools.jm:150 console_log() ----Hit score 62/40/70 2009-02-19 11:10:18 [INFO] js_modules/SpeechTools.jm:150 console_log() ----TAKEOUT 2009-02-19 11:10:18 [DEBUG] js_modules/SpeechTools.jm:365 console_log() ----We need to confirm this one 2009-02-19 11:10:18 [DEBUG] js_modules/SpeechTools.jm:365 console_log() ----Testing [1] [0] TAKEOUT =~ [Delivery:::Delivery] 2009-02-19 11:10:18 [DEBUG] js_modules/SpeechTools.jm:365 console_log() ----Testing [1] [1] TAKEOUT =~ [Takeout:::Pickup] 2009-02-19 11:10:18 [DEBUG] js_modules/SpeechTools.jm:365 console_log() ----Adding Pickup 2009-02-19 11:10:18 [DEBUG] js_modules/SpeechTools.jm:365 console_log() ----Testing [1] [2] TAKEOUT =~ [Pickup:::Pickup] 2009-02-19 11:10:19 [DEBUG] js_modules/SpeechTools.jm:109 console_log() Unloading grammar pizza_order 2009-02-19 11:10:21 [DEBUG] switch_ivr_play_say.c:989 switch_ivr_play_file() Codec Activated L16 at 8000hz 1 channels 20ms 2009-02-19 11:10:22 [DEBUG] mod_pocketsphinx.c:387 pocketsphinx_asr_get_results() Recognized: ????, Score: 100 2009-02-19 11:10:22 [DEBUG] js_modules/SpeechTools.jm:150 console_log() ----XML: ???? ???? 2009-02-19 11:10:22 [DEBUG] js_modules/SpeechTools.jm:150 console_log() ----Heard [????] 2009-02-19 11:10:22 [DEBUG] js_modules/SpeechTools.jm:150 console_log() ----Hit score 100/40/20 2009-02-19 11:10:22 [INFO] js_modules/SpeechTools.jm:150 console_log() ----???? 2009-02-19 11:10:22 [DEBUG] switch_ivr_play_say.c:1279 switch_ivr_play_file() done playing file 2009-02-19 11:10:22 [DEBUG] js_modules/SpeechTools.jm:365 console_log() ----Testing [0] [0] ???? =~ [^yes:::yes] 2009-02-19 11:10:22 [DEBUG] js_modules/SpeechTools.jm:365 console_log() ----Testing [0] [1] ???? =~ [^correct:::yes] 2009-02-19 11:10:22 [DEBUG] js_modules/SpeechTools.jm:365 console_log() ----Testing [0] [2] ???? =~ [^no:::no] 2009-02-19 11:10:22 [DEBUG] switch_core_io.c:234 switch_core_session_read_frame() sofia/internal/1000 at 192.168.16.63 receive message [TRANSCODING_NECESSARY] 2009-02-19 11:10:23 [DEBUG] mod_pocketsphinx.c:343 pocketsphinx_asr_resume() Manually Resuming 2009-02-19 11:10:23 [DEBUG] switch_ivr_play_say.c:989 switch_ivr_play_file() Codec Activated L16 at 8000hz 1 channels 20ms 2009-02-19 11:10:25 [DEBUG] switch_ivr_play_say.c:1279 switch_ivr_play_file() done playing file 2009-02-19 11:10:25 [DEBUG] switch_core_io.c:234 switch_core_session_read_frame() sofia/internal/1000 at 192.168.16.63 receive message [TRANSCODING_NECESSARY] 2009-02-19 11:10:26 [DEBUG] switch_ivr_play_say.c:989 switch_ivr_play_file() Codec Activated L16 at 8000hz 1 channels 20ms 2009-02-19 11:10:29 [DEBUG] switch_ivr_play_say.c:1279 switch_ivr_play_file() done playing file 2009-02-19 11:10:29 [DEBUG] switch_core_io.c:234 switch_core_session_read_frame() sofia/internal/1000 at 192.168.16.63 receive message [TRANSCODING_NECESSARY] 2009-02-19 11:10:31 [DEBUG] switch_ivr_play_say.c:989 switch_ivr_play_file() Codec Activated L16 at 8000hz 1 channels 20ms 2009-02-19 11:10:32 [DEBUG] mod_pocketsphinx.c:387 pocketsphinx_asr_get_results() Recognized: YES, Score: 100 2009-02-19 11:10:32 [DEBUG] js_modules/SpeechTools.jm:150 console_log() ----XML: YES YES 2009-02-19 11:10:32 [DEBUG] js_modules/SpeechTools.jm:150 console_log() ----Heard [YES] 2009-02-19 11:10:32 [DEBUG] js_modules/SpeechTools.jm:150 console_log() ----Hit score 100/40/20 2009-02-19 11:10:32 [INFO] js_modules/SpeechTools.jm:150 console_log() ----YES 2009-02-19 11:10:32 [DEBUG] switch_ivr_play_say.c:1279 switch_ivr_play_file() done playing file 2009-02-19 11:10:32 [DEBUG] js_modules/SpeechTools.jm:365 console_log() ----Testing [0] [0] YES =~ [^yes:::yes] 2009-02-19 11:10:32 [DEBUG] js_modules/SpeechTools.jm:365 console_log() ----Adding yes 2009-02-19 11:10:32 [DEBUG] js_modules/SpeechTools.jm:365 console_log() ----Testing [0] [1] YES =~ [^correct:::yes] 2009-02-19 11:10:32 [DEBUG] js_modules/SpeechTools.jm:365 console_log() ----Testing [0] [2] YES =~ [^no:::no] 2009-02-19 11:10:32 [DEBUG] switch_core_io.c:234 switch_core_session_read_frame() sofia/internal/1000 at 192.168.16.63 receive message [TRANSCODING_NECESSARY] 2009-02-19 11:10:33 [DEBUG] js_modules/SpeechTools.jm:109 console_log() Unloading grammar pizza_yesno On Wed, Feb 18, 2009 at 11:48 PM, Michael Collins wrote: > On Wed, Feb 18, 2009 at 7:55 AM, Brian West wrote: >> Please go get an SVN client for windows... svn update vs downloading the >> tarball every day will save bandwidth. ;) >> /b > > Use this for windows: > http://tortoisesvn.tigris.org/ > > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Regards, Moiz Chinoy. Mobile: 055-8527492 From gmaruzz at celliax.org Wed Feb 18 23:35:37 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Thu, 19 Feb 2009 08:35:37 +0100 Subject: [Freeswitch-users] Skypiax (Skype Network endpoint) for FreeSWITCH In-Reply-To: <0E2F27BF-565D-42F5-B453-6A34D0EBEEB2@freeswitch.org> References: <7b197bef0902131015k456dee87x8f0f1b7059f1b49c@mail.gmail.com> <5800526b0902181853h32de5535l6c0f9d7067a69cf0@mail.gmail.com> <9D0D74A5-7E6C-41FD-9A52-0430101C909E@freeswitch.org> <87f2f3b90902182139q66393955r1cd6ef12bb496291@mail.gmail.com> <7b197bef0902182207q2b8a1b3bj1abac4aad768aa7@mail.gmail.com> <0E2F27BF-565D-42F5-B453-6A34D0EBEEB2@freeswitch.org> Message-ID: <7b197bef0902182335h6faa1753w18fd59643039b01a@mail.gmail.com> Yes, I'd like it in trunk. There are still some rough edges, but I'll iron out in the trunk. Sincerely, Giovanni Maruzzelli ========================================= www.celliax.org via Pierlombardo 9, 20135 Milano Italy gmaruzz at celliax dot org Cell : +39-347-2665618 Fax : +39-02-87390039 On Thu, Feb 19, 2009 at 7:52 AM, Brian West wrote: > It has to be in trunk to be in the MSI... I don't want to cause > confusion ... Now that 1.0.3 is tagged we can put it in trunk? > > /b > > On Feb 19, 2009, at 12:07 AM, Giovanni Maruzzelli wrote: > >> Would be *very* nice to have skypiax in MSI, thank you! > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From brian at freeswitch.org Wed Feb 18 23:43:54 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 19 Feb 2009 01:43:54 -0600 Subject: [Freeswitch-users] FreeSwitch - PcoketSphinx Prompt Playback & Recognition Issue... In-Reply-To: <29b888f80902182329o578a66d5x66454355d7575f48@mail.gmail.com> References: <22080033.post@talk.nabble.com> <22081606.post@talk.nabble.com> <3FEB93CB-9E5B-40FA-8DF7-1CEC8A364732@freeswitch.org> <87f2f3b90902181148r798c5ed5n4ab30877a8e06c00@mail.gmail.com> <29b888f80902182329o578a66d5x66454355d7575f48@mail.gmail.com> Message-ID: <590097F8-ABB3-49B6-8574-E3A3B7ADD134@freeswitch.org> No clue, I haven't ever seen that behavior on linux. Maybe you can try to narrow it down and report it on jira.. chances are its a bug in the pocketsphinx libs. /b On Feb 19, 2009, at 1:29 AM, Moiz Chinoy wrote: > > Often in the log I see cryptic characters in the XML part returned by > ASR. Is it silence or nose?? > If yes is there any way we can control it? From gcd at i.ph Thu Feb 19 00:59:06 2009 From: gcd at i.ph (Nandy Dagondon) Date: Thu, 19 Feb 2009 16:59:06 +0800 Subject: [Freeswitch-users] Default IVR action Message-ID: <7d0bfd8c0902190059y8262895h3a470ccfa4f6c602@mail.gmail.com> hi everybody, i'm looking for a default action in an IVR if the caller doesn't press any key. for example, the caller will be transferred to the operator (or fifo) if no key is received after, let's say 5 seconds. is this available in the IVR? pls show a sample. tks, -nandy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090219/2a081326/attachment.html From moizchinoy at gmail.com Thu Feb 19 01:29:06 2009 From: moizchinoy at gmail.com (Moiz Chinoy) Date: Thu, 19 Feb 2009 13:29:06 +0400 Subject: [Freeswitch-users] FreeSwitch - PcoketSphinx Prompt Playback & Recognition Issue... In-Reply-To: <590097F8-ABB3-49B6-8574-E3A3B7ADD134@freeswitch.org> References: <22080033.post@talk.nabble.com> <22081606.post@talk.nabble.com> <3FEB93CB-9E5B-40FA-8DF7-1CEC8A364732@freeswitch.org> <87f2f3b90902181148r798c5ed5n4ab30877a8e06c00@mail.gmail.com> <29b888f80902182329o578a66d5x66454355d7575f48@mail.gmail.com> <590097F8-ABB3-49B6-8574-E3A3B7ADD134@freeswitch.org> Message-ID: <29b888f80902190129k377cd710v1fe525659ac68fcb@mail.gmail.com> Can anyone please explain the following fields from pocketsphinx.conf.xml: Moiz Chinoy. On Thu, Feb 19, 2009 at 11:43 AM, Brian West wrote: > No clue, I haven't ever seen that behavior on linux. Maybe you can > try to narrow it down and report it on jira.. chances are its a bug in > the pocketsphinx libs. > > /b > > On Feb 19, 2009, at 1:29 AM, Moiz Chinoy wrote: > >> >> Often in the log I see cryptic characters in the XML part returned by >> ASR. Is it silence or nose?? >> If yes is there any way we can control it? > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Regards, Moiz Chinoy. Mobile: 055-8527492 From Claudio.Cavalera at italtel.it Thu Feb 19 02:02:30 2009 From: Claudio.Cavalera at italtel.it (Cavalera Claudio Luigi) Date: Thu, 19 Feb 2009 11:02:30 +0100 Subject: [Freeswitch-users] howto master logging in fs_cli Message-ID: Hello, I'm trying to clarify behaviour of fs_cli http://wiki.freeswitch.org/wiki/Fs_cli After some experiments I'm still not sure on how to deal with logging. For example: --- root at lallobox:/usr/local/freeswitch/bin# ./fs_cli -d 6 _____ ____ ____ _ ___ | ___/ ___| / ___| | |_ _| | |_ \___ \ | | | | | | | _| ___) | | |___| |___ | | |_| |____/ \____|_____|___| ***************************************************** * Anthony Minessale II, Ken Rice, Michael Jerris * * FreeSWITCH (http://www.freeswitch.org) * * Brought to you by ClueCon http://www.cluecon.com/ * ***************************************************** Type /help to see a list of commands [INFO] libs/esl/fs_cli.c:726 main() FS CLI Ready. enter /help for a list of commands. freeswitch at internal> --- So I assume now I'm logging at level 6 in fs_cli, instead I still see debug messages like this as in loglevel 7: 2009-02-19 10:53:29 [DEBUG] mod_event_socket.c:1856 listener_run() Connection Open from 127.0.0.1:48400 2009-02-19 10:53:29 [DEBUG] mod_event_socket.c:1979 listener_run() Session complete, waiting for children To achieve a info loglevel i have to start fs_cli like this ./fs_cli -l info or type /log info in the fs_cli console So could you please someone more expert with me clarify the difference between -d and -l options so that I can update the wiki which is now wrong/incomplete ? -l, --loglevel=command Log Level -q, --quiet Disable logging -d, --debug=level Debug Level (0 - 7) Thanks, Claudio Internet Email Confidentiality Footer ----------------------------------------------------------------------------------------------------- La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. ----------------------------------------------------------------------------------------------------- From jason at jasonjgw.net Thu Feb 19 02:17:29 2009 From: jason at jasonjgw.net (Jason White) Date: Thu, 19 Feb 2009 21:17:29 +1100 Subject: [Freeswitch-users] howto master logging in fs_cli In-Reply-To: References: Message-ID: <20090219101729.GA1868@jdc.jasonjgw.net> Cavalera Claudio Luigi wrote: > So could you please someone more expert with me clarify the difference > between -d and -l options so that I can update the wiki which is now > wrong/incomplete ? > > -l, --loglevel=command Log Level > -q, --quiet Disable logging > -d, --debug=level Debug Level (0 - 7) The difference is that -d controls the level of debugging output generated by fs_cli itself. The log level controls which log messages from your running FreeSWITCH daemon are printed to the fs_cli console. From alex at sinapticode.ro Thu Feb 19 02:36:08 2009 From: alex at sinapticode.ro (Alexandru Nedelcu) Date: Thu, 19 Feb 2009 12:36:08 +0200 Subject: [Freeswitch-users] origination_caller_id_number used instead of effective_caller_id_number Message-ID: <1235039768.4537.17.camel@gathern.lan> I don't know what I'm doing wrong. origination_caller_id_number is used on the B-leg of the bridge, although I also specify the effective_caller_id_number. The thing is that it originally worked in my tests, and I can't figure out what changed in the meantime. The code is something like this ... session = new Session("{originate_retry_sleep_ms=30000,ignore_early_media=true,is_callcenter=0,origination_caller_id_number=+40722333444,effective_caller_id_number=+40711222333}sofia/gateway/provider/"); if (session.ready()) { new_session = new Session("sofia/gateway/provider/", session); if (new_session.ready()) bridge(session, new_session); } Thanks, From alex at sinapticode.ro Thu Feb 19 03:03:09 2009 From: alex at sinapticode.ro (Alexandru Nedelcu) Date: Thu, 19 Feb 2009 13:03:09 +0200 Subject: [Freeswitch-users] origination_caller_id_number used instead of effective_caller_id_number In-Reply-To: <1235039768.4537.17.camel@gathern.lan> References: <1235039768.4537.17.camel@gathern.lan> Message-ID: <1235041389.4537.22.camel@gathern.lan> Just a follow up, I execute the following command: originate {effective_caller_id_number=40722333444}sofia/gateway/provider/ &bridge('sofia/gateway/provider/') And it worked, but when I add origination_caller_id_number ... it overrides effective_caller_id_number. My provider's setup is nothing fancy, something like: On Thu, 2009-02-19 at 12:36 +0200, Alexandru Nedelcu wrote: > I don't know what I'm doing wrong. origination_caller_id_number is used > on the B-leg of the bridge, although I also specify the > effective_caller_id_number. > > The thing is that it originally worked in my tests, and I can't figure > out what changed in the meantime. > > The code is something like this ... > > session = new > Session("{originate_retry_sleep_ms=30000,ignore_early_media=true,is_callcenter=0,origination_caller_id_number=+40722333444,effective_caller_id_number=+40711222333}sofia/gateway/provider/"); > > if (session.ready()) { > > new_session = new Session("sofia/gateway/provider/", > session); > if (new_session.ready()) > bridge(session, new_session); > } > > Thanks, From Claudio.Cavalera at italtel.it Thu Feb 19 03:07:59 2009 From: Claudio.Cavalera at italtel.it (Cavalera Claudio Luigi) Date: Thu, 19 Feb 2009 12:07:59 +0100 Subject: [Freeswitch-users] Random problems with cepstral text to speech Message-ID: Hello list, sometimes when issue a say command for Cepstral TTS in a conference I get this error: [CRIT] mod_local_stream.c:237 read_stream_thread() Leaking stream handle! [conference_play_file() mod_conference.c:2431] and no audio is played to the conference. I have to restart fs to make it work again, any hint on what it could be? BRs, Claudio Internet Email Confidentiality Footer ----------------------------------------------------------------------------------------------------- La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. ----------------------------------------------------------------------------------------------------- From Claudio.Cavalera at italtel.it Thu Feb 19 03:12:25 2009 From: Claudio.Cavalera at italtel.it (Cavalera Claudio Luigi) Date: Thu, 19 Feb 2009 12:12:25 +0100 Subject: [Freeswitch-users] howto master logging in fs_cli In-Reply-To: <20090219101729.GA1868@jdc.jasonjgw.net> Message-ID: freeswitch-users-bounces at lists.freeswitch.org wrote: > Cavalera Claudio Luigi wrote: >> So could you please someone more expert with me clarify the >> difference between -d and -l options so that I can update the wiki >> which is now wrong/incomplete ? >> >> -l, --loglevel=command Log Level >> -q, --quiet Disable logging >> -d, --debug=level Debug Level (0 - 7) > > The difference is that -d controls the level of debugging output > generated by fs_cli itself. The log level controls which log messages > from your running FreeSWITCH daemon are printed to the fs_cli console. > Ah thanks, it was a bit confusing for me also because /log expects a key word such as "info" instead of the number "6". Claudio Internet Email Confidentiality Footer ----------------------------------------------------------------------------------------------------- La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. ----------------------------------------------------------------------------------------------------- From moizchinoy at gmail.com Thu Feb 19 04:50:40 2009 From: moizchinoy at gmail.com (Moiz Chinoy) Date: Thu, 19 Feb 2009 16:50:40 +0400 Subject: [Freeswitch-users] FreeSwitch - PcoketSphinx Prompt Playback & Recognition Issue... In-Reply-To: <29b888f80902190129k377cd710v1fe525659ac68fcb@mail.gmail.com> References: <22080033.post@talk.nabble.com> <22081606.post@talk.nabble.com> <3FEB93CB-9E5B-40FA-8DF7-1CEC8A364732@freeswitch.org> <87f2f3b90902181148r798c5ed5n4ab30877a8e06c00@mail.gmail.com> <29b888f80902182329o578a66d5x66454355d7575f48@mail.gmail.com> <590097F8-ABB3-49B6-8574-E3A3B7ADD134@freeswitch.org> <29b888f80902190129k377cd710v1fe525659ac68fcb@mail.gmail.com> Message-ID: <29b888f80902190450u29b383c5i46a555c0fed82bbb@mail.gmail.com> Can anyone please help me sort it out. Below is the call stack when sphinxbase.dll crashes: sphinxbase.dll!logmath_get_base(logmath_s * lmath=0x00000000) Line 370 + 0x3 bytes C sphinxbase.dll!ngram_model_set_init(cmd_ln_s * config=0x01f01068, ngram_model_s * * models=0x04a9e1dc, char * * names=0x047601ac, const float * weights=0x00000000, int n_models=1) Line 140 + 0x12 bytes C pocketsphinx.dll!ngram_search_init(cmd_ln_s * config=0x01f01068, acmod_s * acmod=0x035add20, dict_s * dict=0x0373ed08) Line 209 + 0x19 bytes C pocketsphinx.dll!ps_reinit(ps_decoder_s * ps=0x03587490, cmd_ln_s * config=0x01f01068) Line 179 + 0x17 bytes C mod_pocketsphinx.dll!pocketsphinx_asr_load_grammar(switch_asr_handle * ah=0x0365a3a0, const char * grammar=0x036d1428, const char * path=0x038622b0) Line 168 + 0x15 bytes C FreeSwitch.dll!switch_core_asr_load_grammar(switch_asr_handle * ah=0x0365a3a0, const char * grammar=0x036d1428, const char * path=0x038622b0) Line 94 + 0x18 bytes C FreeSwitch.dll!switch_ivr_detect_speech_load_grammar(switch_core_session * session=0x0353a308, char * grammar=0x036d1428, char * path=0x00000000) Line 1973 + 0x14 bytes C mod_dptools.dll!detect_speech_function(switch_core_session * session=0x0353a308, const char * data=0x036fa6b8) Line 97 + 0x14 bytes C FreeSwitch.dll!switch_core_session_exec(switch_core_session * session=0x0353a308, const switch_application_interface * application_interface=0x01f0c6c8, const char * arg=0x036fa6b8) Line 1342 + 0x12 bytes C mod_spidermonkey.dll!session_execute(JSContext * cx=0x035433b0, JSObject * obj=0x035b0138, unsigned int argc=2, long * argv=0x03627dd8, long * rval=0x04a9e610) Line 2232 + 0x16 bytes C js32.dll!js_Invoke(JSContext * cx=0x035433b0, unsigned int argc=2, unsigned int flags=0) Line 1181 + 0x20 bytes C js32.dll!js_Interpret(JSContext * cx=0x035433b0, unsigned char * pc=0x03630e56, long * result=0x04a9f014) Line 3571 + 0xf bytes C js32.dll!js_Execute(JSContext * cx=0x035433b0, JSObject * chain=0x035ae7c8, JSScript * script=0x03626fe8, JSStackFrame * down=0x00000000, unsigned int flags=0, long * result=0x04a9f0e8) Line 1427 + 0x13 bytes C js32.dll!JS_ExecuteScript(JSContext * cx=0x035433b0, JSObject * obj=0x035ae7c8, JSScript * script=0x03626fe8, long * rval=0x04a9f0e8) Line 4035 + 0x19 bytes C mod_spidermonkey.dll!eval_some_js(const char * code=0x035421d8, JSContext * cx=0x035433b0, JSObject * obj=0x035ae7c8, long * rval=0x04a9f0e8) Line 103 + 0x15 bytes C mod_spidermonkey.dll!js_parse_and_execute(switch_core_session * session=0x0353a308, const char * input_code=0x035421d8, request_obj * ro=0x00000000) Line 3582 + 0x1e bytes C mod_spidermonkey.dll!js_dp_function(switch_core_session * session=0x0353a308, const char * data=0x035421d8) Line 3591 + 0xf bytes C FreeSwitch.dll!switch_core_session_exec(switch_core_session * session=0x0353a308, const switch_application_interface * application_interface=0x020713f0, const char * arg=0x035421d8) Line 1342 + 0x12 bytes C FreeSwitch.dll!switch_core_session_execute_application(switch_core_session * session=0x0353a308, const char * app=0x035421c8, const char * arg=0x035421d8) Line 1266 C FreeSwitch.dll!switch_core_standard_on_execute(switch_core_session * session=0x0353a308) Line 157 + 0x16 bytes C FreeSwitch.dll!switch_core_session_run(switch_core_session * session=0x0353a308) Line 464 + 0x204 bytes C FreeSwitch.dll!switch_core_session_thread(apr_thread_t * thread=0x020778d0, void * obj=0x0353a308) Line 951 C libapr.dll!dummy_worker(void * opaque=0x020778d0) Line 80 C msvcr90d.dll!1023dfd3() [Frames below may be incorrect and/or missing, no symbols loaded for msvcr90d.dll] msvcr90d.dll!1023df69() kernel32.dll!7c80b683() Moiz Chinoy. On Thu, Feb 19, 2009 at 1:29 PM, Moiz Chinoy wrote: > Can anyone please explain the following fields from pocketsphinx.conf.xml: > > > > > > > > > Moiz Chinoy. > > On Thu, Feb 19, 2009 at 11:43 AM, Brian West wrote: >> No clue, I haven't ever seen that behavior on linux. Maybe you can >> try to narrow it down and report it on jira.. chances are its a bug in >> the pocketsphinx libs. >> >> /b >> >> On Feb 19, 2009, at 1:29 AM, Moiz Chinoy wrote: >> >>> >>> Often in the log I see cryptic characters in the XML part returned by >>> ASR. Is it silence or nose?? >>> If yes is there any way we can control it? >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Regards, > Moiz Chinoy. > Mobile: 055-8527492 > -- Regards, Moiz Chinoy. Mobile: 055-8527492 From frank at impactfax.com Thu Feb 19 04:58:25 2009 From: frank at impactfax.com (Frank @ Impact) Date: Thu, 19 Feb 2009 07:58:25 -0500 Subject: [Freeswitch-users] Voice pattern detection Message-ID: <20b801c99291$c0dc5c30$33014c0a@ws4> I am trying to detect if a caller is an automated greeting voice. And if so, take an action. I have samples of the caller recording that I am looking to match. So this is like a really complex tone detection I guess. It would work like this. - Call comes in - We answer/bridge the call - We start to listen for about 10 seconds. - During this time we are trying to match a snippet of a sound sample (say 2 or 3 seconds worth) we have recorded on a file on the server. We are trying to match this sound sample to the caller side only. - If we hear a match, we take the action. Or if we don't hear the match, we might take a different action. Any of this sound doable? Any guidance on how to accomplish this? -Frank -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090219/f5440595/attachment.html From r.pankratz at fh-wolfenbuettel.de Thu Feb 19 05:04:11 2009 From: r.pankratz at fh-wolfenbuettel.de (Rene Pankratz) Date: Thu, 19 Feb 2009 14:04:11 +0100 Subject: [Freeswitch-users] mod_portaudio: Do not accept next call after Hangup Message-ID: <499D58CB.9080405@fh-wolfenbuettel.de> Hello, when hanging up a call with portaudio automatically the next call that is incoming or held is accepted. Is it possible to configure PA that way, that after hanging up (doesn't matter whether caller or callee) no call is activated automatically? I want to choose if I accept the next call or not. Thanks in advance Ren? From mrene_lists at avgs.ca Thu Feb 19 05:42:33 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Thu, 19 Feb 2009 08:42:33 -0500 Subject: [Freeswitch-users] origination_caller_id_number used instead of effective_caller_id_number In-Reply-To: <1235041389.4537.22.camel@gathern.lan> References: <1235039768.4537.17.camel@gathern.lan> <1235041389.4537.22.camel@gathern.lan> Message-ID: Clarification on the 2 vars....you set origination_caller_id_number on the call leg directly. you set effective_caller_id_number on any leg that will get bridged to something else. Internally, the core will look if effective_caller_id_number is set on the A-leg to see if it can use it. This said: originate {origination_caller_id_number=12223334444,effective_caller_id_number=13334445555}sofia/gateway/blah/12345 &bridge(sofia/gateway/blah/54321) 12345 is called and sees 12223334444 as callerid, then the call is bridged to 54321 which sees 13334445555 because bridge looks up the variable in the a-leg. It would be the same as doing originate {origination_caller_id_number=12223334444}sofia/gateway/blah/12345 &bridge({origination_caller_id_number=13334445555}sofia/gateway/blah/54321) Having it as effective_caller_id_number only saves you the extra work of setting it on all B-legs Mathieu On Thu, Feb 19, 2009 at 6:03 AM, Alexandru Nedelcu wrote: > Just a follow up, I execute the following command: > > originate > {effective_caller_id_number=40722333444}sofia/gateway/provider/ > &bridge('sofia/gateway/provider/') > > And it worked, but when I add origination_caller_id_number ... it > overrides effective_caller_id_number. > > My provider's setup is nothing fancy, something like: > > > > > > > > > > > > > On Thu, 2009-02-19 at 12:36 +0200, Alexandru Nedelcu wrote: > > I don't know what I'm doing wrong. origination_caller_id_number is used > > on the B-leg of the bridge, although I also specify the > > effective_caller_id_number. > > > > The thing is that it originally worked in my tests, and I can't figure > > out what changed in the meantime. > > > > The code is something like this ... > > > > session = new > > > Session("{originate_retry_sleep_ms=30000,ignore_early_media=true,is_callcenter=0,origination_caller_id_number=+40722333444,effective_caller_id_number=+40711222333}sofia/gateway/provider/"); > > > > if (session.ready()) { > > > > new_session = new Session("sofia/gateway/provider/", > > session); > > if (new_session.ready()) > > bridge(session, new_session); > > } > > > > Thanks, > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090219/524241a5/attachment.html From Tim.Meade at millicorp.com Thu Feb 19 04:23:49 2009 From: Tim.Meade at millicorp.com (Tim Meade) Date: Thu, 19 Feb 2009 07:23:49 -0500 Subject: [Freeswitch-users] xml_cdr setup and use questions. Message-ID: <7832D4F0FEC057488FD3BA68CA25A6FD0CD1E05C@postman.millicorp.com> Greetings all. I'm fairly new to freeswitch, but I've been watching here for a while. We've finally decided to take the plunge and try to integrate this into our working environments, so I've spent the last couple days playing around with it. My immediate question has to do with xml_cdr. I want to have it contact our web system so that I can put away the specifics of each call. This should be perfect for what I need to do. Here is my current configuration. The call to the web server is being executed twice. Any ideas why? One has more fields than the other. Both have what seems to be identical , but one has several other sections as well. Thanks in advance. Tim -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090219/ed0dfadc/attachment-0001.html From krice at suspicious.org Thu Feb 19 05:48:42 2009 From: krice at suspicious.org (Ken Rice) Date: Thu, 19 Feb 2009 07:48:42 -0600 Subject: [Freeswitch-users] xml_cdr setup and use questions. In-Reply-To: <7832D4F0FEC057488FD3BA68CA25A6FD0CD1E05C@postman.millicorp.com> Message-ID: A-Leg and B-leg? From: Tim Meade Reply-To: Date: Thu, 19 Feb 2009 07:23:49 -0500 To: "freeswitch-users at lists.freeswitch.org" Subject: [Freeswitch-users] xml_cdr setup and use questions. Greetings all. I'm fairly new to freeswitch, but I've been watching here for a while. We've finally decided to take the plunge and try to integrate this into our working environments, so I've spent the last couple days playing around with it. My immediate question has to do with xml_cdr. I want to have it contact our web system so that I can put away the specifics of each call. This should be perfect for what I need to do. Here is my current configuration. The call to the web server is being executed twice. Any ideas why? One has more fields than the other. Both have what seems to be identical , but one has several other sections as well. Thanks in advance. Tim _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090219/49dd952d/attachment.html From Tim.Meade at millicorp.com Thu Feb 19 06:01:29 2009 From: Tim.Meade at millicorp.com (Tim Meade) Date: Thu, 19 Feb 2009 09:01:29 -0500 Subject: [Freeswitch-users] xml_cdr setup and use questions. In-Reply-To: References: <7832D4F0FEC057488FD3BA68CA25A6FD0CD1E05C@postman.millicorp.com> Message-ID: <7832D4F0FEC057488FD3BA68CA25A6FD0E001FCD@postman.millicorp.com> Not real certain on that. If I setup for it to drop a log file, it only drops a single file, but it seems to be calling the http twice. I've got the web page setup to send an email with the form vars and I'm getting two of them. One just one with that and several other sections. Thanks ... From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice Sent: Thursday, February 19, 2009 8:49 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] xml_cdr setup and use questions. A-Leg and B-leg? ________________________________ From: Tim Meade Reply-To: Date: Thu, 19 Feb 2009 07:23:49 -0500 To: "freeswitch-users at lists.freeswitch.org" Subject: [Freeswitch-users] xml_cdr setup and use questions. Greetings all. I'm fairly new to freeswitch, but I've been watching here for a while. We've finally decided to take the plunge and try to integrate this into our working environments, so I've spent the last couple days playing around with it. My immediate question has to do with xml_cdr. I want to have it contact our web system so that I can put away the specifics of each call. This should be perfect for what I need to do. Here is my current configuration. The call to the web server is being executed twice. Any ideas why? One has more fields than the other. Both have what seems to be identical , but one has several other sections as well. Thanks in advance. Tim ________________________________ _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090219/979cde41/attachment.html From Claudio.Cavalera at italtel.it Thu Feb 19 05:59:02 2009 From: Claudio.Cavalera at italtel.it (Cavalera Claudio Luigi) Date: Thu, 19 Feb 2009 14:59:02 +0100 Subject: [Freeswitch-users] Strange error load testing mod_xml_rpc Message-ID: Hello, doing a bit of load and stress I'm sending request like originate &bridge originate &playback to mod_xml_rpc and I get these errors in freeswitch_http.log ?? getpeername() failed. errno=107 (Transport endpoint is not connected) - no_user - [19/Feb/2009:14:45:25 -0100] "GET" 200 150 ?? getpeername() failed. errno=107 (Transport endpoint is not connected) - no_user - [19/Feb/2009:14:45:31 -0100] "GET" 200 0 Any idea on what it means? Could it be because of the seagull load runner I'm using? I have found this old JIRA with a similar message http://jira.freeswitch.org/browse/MDXMLINT-28 but I'm not sure it's related. BRs, Claudio Internet Email Confidentiality Footer ----------------------------------------------------------------------------------------------------- La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. ----------------------------------------------------------------------------------------------------- From krice at suspicious.org Thu Feb 19 06:11:27 2009 From: krice at suspicious.org (Ken Rice) Date: Thu, 19 Feb 2009 08:11:27 -0600 Subject: [Freeswitch-users] xml_cdr setup and use questions. In-Reply-To: <7832D4F0FEC057488FD3BA68CA25A6FD0E001FCD@postman.millicorp.com> Message-ID: Tim, Try this param and see if it helps in you xml_cdr.conf file From: Tim Meade Reply-To: Date: Thu, 19 Feb 2009 09:01:29 -0500 To: "freeswitch-users at lists.freeswitch.org" Subject: Re: [Freeswitch-users] xml_cdr setup and use questions. Not real certain on that. If I setup for it to drop a log file, it only drops a single file, but it seems to be calling the http twice. I've got the web page setup to send an email with the form vars and I'm getting two of them. One just one with that and several other sections. Thanks ... From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice Sent: Thursday, February 19, 2009 8:49 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] xml_cdr setup and use questions. A-Leg and B-leg? From: Tim Meade Reply-To: Date: Thu, 19 Feb 2009 07:23:49 -0500 To: "freeswitch-users at lists.freeswitch.org" Subject: [Freeswitch-users] xml_cdr setup and use questions. Greetings all. I'm fairly new to freeswitch, but I've been watching here for a while. We've finally decided to take the plunge and try to integrate this into our working environments, so I've spent the last couple days playing around with it. My immediate question has to do with xml_cdr. I want to have it contact our web system so that I can put away the specifics of each call. This should be perfect for what I need to do. Here is my current configuration. The call to the web server is being executed twice. Any ideas why? One has more fields than the other. Both have what seems to be identical , but one has several other sections as well. Thanks in advance. Tim _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090219/89c3b5a7/attachment-0001.html From anthony.minessale at gmail.com Thu Feb 19 06:16:42 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 19 Feb 2009 08:16:42 -0600 Subject: [Freeswitch-users] Random problems with cepstral text to speech In-Reply-To: References: Message-ID: <191c3a030902190616pae78f47x757702108c760264@mail.gmail.com> Are you using cepstral 5.1? There is a known issue with that release and it's closed source so we cannot do much about it. Cepstral 4.x works fine. 2009/2/19 Cavalera Claudio Luigi > Hello list, > sometimes when issue a say command for Cepstral TTS in a conference I > get this error: > [CRIT] mod_local_stream.c:237 read_stream_thread() Leaking stream > handle! [conference_play_file() mod_conference.c:2431] > > and no audio is played to the conference. > > I have to restart fs to make it work again, any hint on what it could > be? > > BRs, > Claudio > > > Internet Email Confidentiality Footer > > ----------------------------------------------------------------------------------------------------- > La presente comunicazione, con le informazioni in essa contenute e ogni > documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' > indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete > i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, > comunicazione, divulgazione o simili basate sul contenuto di tali > informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., > D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se > avete ricevuto questa comunicazione per errore, vi preghiamo di darne > immediata notizia al mittente e di distruggere il messaggio originale e ogni > file allegato senza farne copia alcuna o riprodurne in alcun modo il > contenuto. > > This e-mail and its attachments are intended for the addressee(s) only and > are confidential and/or may contain legally privileged information. If you > have received this message by mistake or are not one of the addressees > above, you may take no action based on it, and you may not copy or show it > to anyone; please reply to this e-mail and point out the error which has > occurred. > > ----------------------------------------------------------------------------------------------------- > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090219/e27f3a83/attachment.html From anthony.minessale at gmail.com Thu Feb 19 06:19:27 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 19 Feb 2009 08:19:27 -0600 Subject: [Freeswitch-users] Strange error load testing mod_xml_rpc In-Reply-To: References: Message-ID: <191c3a030902190619y176f62bdtd0e3329c0c9b1337@mail.gmail.com> Yes we already see you have reported this issue in jira and we are working on it. you do not need to report it here as well. You may want to make yourself available on irc or some other im today so we can contact you for more details. 2009/2/19 Cavalera Claudio Luigi > Hello, > doing a bit of load and stress I'm sending request like > > originate &bridge > originate &playback > > to mod_xml_rpc and I get these errors in freeswitch_http.log > > ?? getpeername() failed. errno=107 (Transport endpoint is not > connected) - no_user - [19/Feb/2009:14:45:25 -0100] "GET" 200 150 > ?? getpeername() failed. errno=107 (Transport endpoint is not > connected) - no_user - [19/Feb/2009:14:45:31 -0100] "GET" 200 0 > > Any idea on what it means? > Could it be because of the seagull load runner I'm using? > > I have found this old JIRA with a similar message > http://jira.freeswitch.org/browse/MDXMLINT-28 > but I'm not sure it's related. > > BRs, > Claudio > > > Internet Email Confidentiality Footer > > ----------------------------------------------------------------------------------------------------- > La presente comunicazione, con le informazioni in essa contenute e ogni > documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' > indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete > i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, > comunicazione, divulgazione o simili basate sul contenuto di tali > informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., > D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se > avete ricevuto questa comunicazione per errore, vi preghiamo di darne > immediata notizia al mittente e di distruggere il messaggio originale e ogni > file allegato senza farne copia alcuna o riprodurne in alcun modo il > contenuto. > > This e-mail and its attachments are intended for the addressee(s) only and > are confidential and/or may contain legally privileged information. If you > have received this message by mistake or are not one of the addressees > above, you may take no action based on it, and you may not copy or show it > to anyone; please reply to this e-mail and point out the error which has > occurred. > > ----------------------------------------------------------------------------------------------------- > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090219/96d52df1/attachment.html From alex at sinapticode.ro Thu Feb 19 06:19:45 2009 From: alex at sinapticode.ro (Alexandru Nedelcu) Date: Thu, 19 Feb 2009 16:19:45 +0200 Subject: [Freeswitch-users] origination_caller_id_number used instead of effective_caller_id_number In-Reply-To: References: <1235039768.4537.17.camel@gathern.lan> <1235041389.4537.22.camel@gathern.lan> Message-ID: <1235053185.4537.59.camel@gathern.lan> OK, so effective_caller_id_number is the same as origination_caller_id_number set on the B-leg (cool). Unfortunately origination_caller_id_number on the A-leg overides the origination_caller_id_number on the B-leg (I tried setting it with both effective_caller_id on the A-leg and origination_caller_id on the B-leg). It doesn't work. If I'm setting caller-id on the B-leg only, then it works. Has this something to do with my SIP provider maybe? On Thu, 2009-02-19 at 08:42 -0500, Mathieu Rene wrote: > Clarification on the 2 vars.... > you set origination_caller_id_number on the call leg directly. > you set effective_caller_id_number on any leg that will get bridged to > something else. > > > Internally, the core will look if effective_caller_id_number is set on > the A-leg to see if it can use it. > > > This said: originate > {origination_caller_id_number=12223334444,effective_caller_id_number=13334445555}sofia/gateway/blah/12345 &bridge(sofia/gateway/blah/54321) > > > 12345 is called and sees 12223334444 as callerid, then the call is > bridged to 54321 which sees 13334445555 because bridge looks up the > variable in the a-leg. > > > It would be the same as doing > originate > {origination_caller_id_number=12223334444}sofia/gateway/blah/12345 > &bridge({origination_caller_id_number=13334445555}sofia/gateway/blah/54321) > > > Having it as effective_caller_id_number only saves you the extra work > of setting it on all B-legs > > > Mathieu From Tim.Meade at millicorp.com Thu Feb 19 06:23:02 2009 From: Tim.Meade at millicorp.com (Tim Meade) Date: Thu, 19 Feb 2009 09:23:02 -0500 Subject: [Freeswitch-users] xml_cdr setup and use questions. In-Reply-To: References: <7832D4F0FEC057488FD3BA68CA25A6FD0E001FCD@postman.millicorp.com> Message-ID: <7832D4F0FEC057488FD3BA68CA25A6FD0E001FCE@postman.millicorp.com> That seemed to do it.... Noob question: reloadXML didn't reload the change. I still got the two emails. So to be sure, I shutdown and restarted. Now I'm only getting the one email. Shouldn't the reloadXML reload that module also? Or do I have to reload the modules directly as they are "outside" fs? Tim From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice Sent: Thursday, February 19, 2009 9:11 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] xml_cdr setup and use questions. Tim, Try this param and see if it helps in you xml_cdr.conf file ________________________________ From: Tim Meade Reply-To: Date: Thu, 19 Feb 2009 09:01:29 -0500 To: "freeswitch-users at lists.freeswitch.org" Subject: Re: [Freeswitch-users] xml_cdr setup and use questions. Not real certain on that. If I setup for it to drop a log file, it only drops a single file, but it seems to be calling the http twice. I've got the web page setup to send an email with the form vars and I'm getting two of them. One just one with that and several other sections. Thanks ... From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice Sent: Thursday, February 19, 2009 8:49 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] xml_cdr setup and use questions. A-Leg and B-leg? ________________________________ From: Tim Meade Reply-To: Date: Thu, 19 Feb 2009 07:23:49 -0500 To: "freeswitch-users at lists.freeswitch.org" Subject: [Freeswitch-users] xml_cdr setup and use questions. Greetings all. I'm fairly new to freeswitch, but I've been watching here for a while. We've finally decided to take the plunge and try to integrate this into our working environments, so I've spent the last couple days playing around with it. My immediate question has to do with xml_cdr. I want to have it contact our web system so that I can put away the specifics of each call. This should be perfect for what I need to do. Here is my current configuration. The call to the web server is being executed twice. Any ideas why? One has more fields than the other. Both have what seems to be identical , but one has several other sections as well. Thanks in advance. Tim ________________________________ _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ________________________________ _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090219/d7148c35/attachment-0001.html From krice at freeswitch.org Thu Feb 19 06:27:32 2009 From: krice at freeswitch.org (Ken Rice) Date: Thu, 19 Feb 2009 08:27:32 -0600 Subject: [Freeswitch-users] xml_cdr setup and use questions. In-Reply-To: <7832D4F0FEC057488FD3BA68CA25A6FD0E001FCE@postman.millicorp.com> Message-ID: You reloadxml just reparses the config it does not automattically tell specific modules to reload their configuration From: Tim Meade Reply-To: Date: Thu, 19 Feb 2009 09:23:02 -0500 To: "freeswitch-users at lists.freeswitch.org" Subject: Re: [Freeswitch-users] xml_cdr setup and use questions. That seemed to do it.... Noob question: reloadXML didn't reload the change. I still got the two emails. So to be sure, I shutdown and restarted. Now I'm only getting the one email. Shouldn't the reloadXML reload that module also? Or do I have to reload the modules directly as they are "outside" fs? Tim From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice Sent: Thursday, February 19, 2009 9:11 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] xml_cdr setup and use questions. Tim, Try this param and see if it helps in you xml_cdr.conf file From: Tim Meade Reply-To: Date: Thu, 19 Feb 2009 09:01:29 -0500 To: "freeswitch-users at lists.freeswitch.org" Subject: Re: [Freeswitch-users] xml_cdr setup and use questions. Not real certain on that. If I setup for it to drop a log file, it only drops a single file, but it seems to be calling the http twice. I've got the web page setup to send an email with the form vars and I'm getting two of them. One just one with that and several other sections. Thanks ... From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice Sent: Thursday, February 19, 2009 8:49 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] xml_cdr setup and use questions. A-Leg and B-leg? From: Tim Meade Reply-To: Date: Thu, 19 Feb 2009 07:23:49 -0500 To: "freeswitch-users at lists.freeswitch.org" Subject: [Freeswitch-users] xml_cdr setup and use questions. Greetings all. I'm fairly new to freeswitch, but I've been watching here for a while. We've finally decided to take the plunge and try to integrate this into our working environments, so I've spent the last couple days playing around with it. My immediate question has to do with xml_cdr. I want to have it contact our web system so that I can put away the specifics of each call. This should be perfect for what I need to do. Here is my current configuration. The call to the web server is being executed twice. Any ideas why? One has more fields than the other. Both have what seems to be identical , but one has several other sections as well. Thanks in advance. Tim _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090219/2acdb419/attachment.html From Tim.Meade at millicorp.com Thu Feb 19 07:01:41 2009 From: Tim.Meade at millicorp.com (Tim Meade) Date: Thu, 19 Feb 2009 10:01:41 -0500 Subject: [Freeswitch-users] xml_cdr setup and use questions. In-Reply-To: References: <7832D4F0FEC057488FD3BA68CA25A6FD0E001FCE@postman.millicorp.com> Message-ID: <7832D4F0FEC057488FD3BA68CA25A6FD0E001FD0@postman.millicorp.com> Thanks Ken. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice Sent: Thursday, February 19, 2009 9:28 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] xml_cdr setup and use questions. You reloadxml just reparses the config it does not automattically tell specific modules to reload their configuration ________________________________ From: Tim Meade Reply-To: Date: Thu, 19 Feb 2009 09:23:02 -0500 To: "freeswitch-users at lists.freeswitch.org" Subject: Re: [Freeswitch-users] xml_cdr setup and use questions. That seemed to do it.... Noob question: reloadXML didn't reload the change. I still got the two emails. So to be sure, I shutdown and restarted. Now I'm only getting the one email. Shouldn't the reloadXML reload that module also? Or do I have to reload the modules directly as they are "outside" fs? Tim From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice Sent: Thursday, February 19, 2009 9:11 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] xml_cdr setup and use questions. Tim, Try this param and see if it helps in you xml_cdr.conf file ________________________________ From: Tim Meade Reply-To: Date: Thu, 19 Feb 2009 09:01:29 -0500 To: "freeswitch-users at lists.freeswitch.org" Subject: Re: [Freeswitch-users] xml_cdr setup and use questions. Not real certain on that. If I setup for it to drop a log file, it only drops a single file, but it seems to be calling the http twice. I've got the web page setup to send an email with the form vars and I'm getting two of them. One just one with that and several other sections. Thanks ... From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice Sent: Thursday, February 19, 2009 8:49 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] xml_cdr setup and use questions. A-Leg and B-leg? ________________________________ From: Tim Meade Reply-To: Date: Thu, 19 Feb 2009 07:23:49 -0500 To: "freeswitch-users at lists.freeswitch.org" Subject: [Freeswitch-users] xml_cdr setup and use questions. Greetings all. I'm fairly new to freeswitch, but I've been watching here for a while. We've finally decided to take the plunge and try to integrate this into our working environments, so I've spent the last couple days playing around with it. My immediate question has to do with xml_cdr. I want to have it contact our web system so that I can put away the specifics of each call. This should be perfect for what I need to do. Here is my current configuration. The call to the web server is being executed twice. Any ideas why? One has more fields than the other. Both have what seems to be identical , but one has several other sections as well. Thanks in advance. Tim ________________________________ _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ________________________________ _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ________________________________ _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090219/3d9b6cc7/attachment-0001.html From mike at jerris.com Thu Feb 19 07:03:35 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 19 Feb 2009 10:03:35 -0500 Subject: [Freeswitch-users] origination_caller_id_number used instead of effective_caller_id_number In-Reply-To: <1235053185.4537.59.camel@gathern.lan> References: <1235039768.4537.17.camel@gathern.lan> <1235041389.4537.22.camel@gathern.lan> <1235053185.4537.59.camel@gathern.lan> Message-ID: <2CEAC5BC-D5F8-4253-A710-C5E60DAC7EB8@jerris.com> Can you re-test this with current svn trunk. I believe this was fixed yesterday. Mike On Feb 19, 2009, at 9:19 AM, Alexandru Nedelcu wrote: > OK, so effective_caller_id_number is the same as > origination_caller_id_number set on the B-leg (cool). > > Unfortunately origination_caller_id_number on the A-leg overides the > origination_caller_id_number on the B-leg (I tried setting it with > both > effective_caller_id on the A-leg and origination_caller_id on the > B-leg). It doesn't work. > > If I'm setting caller-id on the B-leg only, then it works. > > Has this something to do with my SIP provider maybe? > > > On Thu, 2009-02-19 at 08:42 -0500, Mathieu Rene wrote: >> Clarification on the 2 vars.... >> you set origination_caller_id_number on the call leg directly. >> you set effective_caller_id_number on any leg that will get bridged >> to >> something else. >> >> >> Internally, the core will look if effective_caller_id_number is set >> on >> the A-leg to see if it can use it. >> >> >> This said: originate >> {origination_caller_id_number >> =12223334444,effective_caller_id_number=13334445555}sofia/gateway/ >> blah/12345 &bridge(sofia/gateway/blah/54321) >> >> >> 12345 is called and sees 12223334444 as callerid, then the call is >> bridged to 54321 which sees 13334445555 because bridge looks up the >> variable in the a-leg. >> >> >> It would be the same as doing >> originate >> {origination_caller_id_number=12223334444}sofia/gateway/blah/12345 >> &bridge({origination_caller_id_number=13334445555}sofia/gateway/ >> blah/54321) >> >> >> Having it as effective_caller_id_number only saves you the extra work >> of setting it on all B-legs >> >> >> Mathieu From kerrada2003 at yahoo.com Thu Feb 19 07:05:05 2009 From: kerrada2003 at yahoo.com (Ali Al-Rubaie) Date: Thu, 19 Feb 2009 07:05:05 -0800 (PST) Subject: [Freeswitch-users] Realm Value In-Reply-To: Message-ID: <472895.39120.qm@web33708.mail.mud.yahoo.com> Thanks Brian, Sorry if the question looks primitive but in which file I can find the rvn? Is there any tarball with the latest revisions? Thanks, --- On Tue, 2/17/09, freeswitch-users-request at lists.freeswitch.org wrote: Message: 5 Date: Tue, 17 Feb 2009 14:36:51 -0600 From: Brian West Subject: Re: [Freeswitch-users] Realm Value To: freeswitch-users at lists.freeswitch.org Message-ID: <090382A2-AF83-4635-90CF-35749F50E0FA at freeswitch.org> Content-Type: text/plain; charset=US-ASCII; format=flowed; delsp=yes Very sorry to hear you have to use Broken Software. But some good has come of this if you update to rev 12113 or great you'll be 100% OK. /b On Feb 17, 2009, at 2:21 PM, Ali Al-Rubaie wrote: > > I have to use a specific softphone, HelpCaster, but it can not pass > the authentication stage. However it can authenticate with OpenSips > server! What I had noticed is that it uses static realm with > OpenSips therefore I'm trying to do the same. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090219/07b6169f/attachment.html From intralanman at freeswitch.org Thu Feb 19 07:26:15 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Thu, 19 Feb 2009 10:26:15 -0500 Subject: [Freeswitch-users] Realm Value In-Reply-To: <472895.39120.qm@web33708.mail.mud.yahoo.com> References: <472895.39120.qm@web33708.mail.mud.yahoo.com> Message-ID: <499D7A17.7030202@freeswitch.org> Ali Al-Rubaie wrote: > Thanks Brian, > > Sorry if the question looks primitive but in which file I can find the > rvn? > you can find the revision from typing "version" at the fs_cli > Is there any tarball with the latest revisions? > the 1.0.3 tarball was just rolled yesterday, give it a shot -Ray -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090219/89abc2b0/attachment.html From Claudio.Cavalera at italtel.it Thu Feb 19 07:47:27 2009 From: Claudio.Cavalera at italtel.it (Cavalera Claudio Luigi) Date: Thu, 19 Feb 2009 16:47:27 +0100 Subject: [Freeswitch-users] Random problems with cepstral text to speech In-Reply-To: <191c3a030902190616pae78f47x757702108c760264@mail.gmail.com> Message-ID: From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale > Are you using cepstral 5.1? > There is a known issue with that release and it's closed source so we cannot do much about it. > Cepstral 4.x works fine. Yes 5.1, my fault. I have added an initial warning here on the wiki http://wiki.freeswitch.org/wiki/Mod_cepstral although it also speaks about 5.1 and Ubuntu... Thanks, Claudio Internet E. Mail Confidentiality Footer ----------------------------------------------------------------------------------------------------- La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. From Claudio.Cavalera at italtel.it Thu Feb 19 08:00:44 2009 From: Claudio.Cavalera at italtel.it (Cavalera Claudio Luigi) Date: Thu, 19 Feb 2009 17:00:44 +0100 Subject: [Freeswitch-users] Strange error load testing mod_xml_rpc In-Reply-To: <191c3a030902190619y176f62bdtd0e3329c0c9b1337@mail.gmail.com> Message-ID: From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale > Yes we already see you have reported this issue in jira and we are working on it. > you do not need to report it here as well. > You may want to make yourself available on irc or some other im today so we can contact you for more details. I'm sorry I reported this message here because I did not think it's related with my JIRA reports. They are about two segmentation faults, while this message is about load testing not going very well, although it comes out during the same load tests :-) I'm starting to think that most people are using the sofia SIP stack in fs to get rate of 100cps while the other interfaces such as event socket and mod_xml_rpc are not well "engineered" yet. I'm going deeper into this and I hope my results will be of help for the community. At the moment I'm trying to understand the cause of TCP retransmissions I see in the snoop, it seems that fs lags a little bit in sending the TCP Ack back to the load runner, even with the machine doing nothing and a really tiny amount of load. BRs, Claudio Internet Email Confidentiality Footer ----------------------------------------------------------------------------------------------------- La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. ----------------------------------------------------------------------------------------------------- From Claudio.Cavalera at italtel.it Thu Feb 19 09:00:19 2009 From: Claudio.Cavalera at italtel.it (Cavalera Claudio Luigi) Date: Thu, 19 Feb 2009 18:00:19 +0100 Subject: [Freeswitch-users] Strange error load testing mod_xml_rpc In-Reply-To: Message-ID: Here is a snoop: http://pastebin.freeswitch.org/7351 thx, cla Internet Email Confidentiality Footer ----------------------------------------------------------------------------------------------------- La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. ----------------------------------------------------------------------------------------------------- From anthony.minessale at gmail.com Thu Feb 19 09:30:01 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 19 Feb 2009 11:30:01 -0600 Subject: [Freeswitch-users] Strange error load testing mod_xml_rpc In-Reply-To: References: Message-ID: <191c3a030902190930x121eba94jb644c25922cbdb7b@mail.gmail.com> api calls by default are blocking, it will not return until the result of the originate is determined. you must pre-empt the originate command with the bgapi api command which is similar to the event_socket bgapi command so that it tells the task to run in a dedicated thread. also the web interface was meant to use an xml rpc client. I have generated at least 400cps on event socket before. 2009/2/19 Cavalera Claudio Luigi > Here is a snoop: > http://pastebin.freeswitch.org/7351 > thx, > cla > > > Internet Email Confidentiality Footer > > ----------------------------------------------------------------------------------------------------- > La presente comunicazione, con le informazioni in essa contenute e ogni > documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' > indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete > i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, > comunicazione, divulgazione o simili basate sul contenuto di tali > informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., > D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se > avete ricevuto questa comunicazione per errore, vi preghiamo di darne > immediata notizia al mittente e di distruggere il messaggio originale e ogni > file allegato senza farne copia alcuna o riprodurne in alcun modo il > contenuto. > > This e-mail and its attachments are intended for the addressee(s) only and > are confidential and/or may contain legally privileged information. If you > have received this message by mistake or are not one of the addressees > above, you may take no action based on it, and you may not copy or show it > to anyone; please reply to this e-mail and point out the error which has > occurred. > > ----------------------------------------------------------------------------------------------------- > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090219/0c42b1d8/attachment-0001.html From brian at freeswitch.org Thu Feb 19 10:44:22 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 19 Feb 2009 12:44:22 -0600 Subject: [Freeswitch-users] ESL Message-ID: FreeSWITCHers, Not sure anyone is paying attention or not but Anthony wrapped the ESL library up so you can use it from Perl, Python, Lua, Ruby and PHP. What I'm requesting from our community is to help flex it out.. write examples and populate the Wiki page with information about it. http://wiki.freeswitch.org/wiki/Esl Collins and I are going to start filling in the page but I want someone thats good with Ruby, Python, PHP to help in those areas.. kick in some lua and perl if you like. It works with OES and IES... (Outbound Event Socket and Inbound Event Socket) Not sure those names are official but we have been calling them that ;) Thanks, Brian West From kerrada2003 at yahoo.com Thu Feb 19 12:06:54 2009 From: kerrada2003 at yahoo.com (Ali Al-Rubaie) Date: Thu, 19 Feb 2009 12:06:54 -0800 (PST) Subject: [Freeswitch-users] Realm Value In-Reply-To: Message-ID: <97318.60826.qm@web33708.mail.mud.yahoo.com> Thanks Ray but unfortunately the 1.0.3 tarball compilation results in the following errors: Compiling src/switch_odbc.c ... In file included from src/switch_odbc.c:33: ./src/include/switch_odbc.h:36:17: error: sql.h: No such file or directory ./src/include/switch_odbc.h:43:20: error: sqlext.h: No such file or directory ./src/include/switch_odbc.h:45:22: error: sqltypes.h: No such file or directory In file included from src/switch_odbc.c:33: ./src/include/switch_odbc.h:66: error: expected declaration specifiers or '...' before 'SQLHSTMT' ./src/include/switch_odbc.h:96: error: expected declaration specifiers or '...' before 'SQLHSTMT' src/switch_odbc.c:43: error: expected specifier-qualifier-list before 'SQLHENV' src/switch_odbc.c: In function 'switch_odbc_handle_new': src/switch_odbc.c:76: error: 'switch_odbc_handle_t' has no member named 'env' src/switch_odbc.c:76: error: 'SQL_NULL_HANDLE' undeclared (first use in this function) src/switch_odbc.c:76: error: (Each undeclared identifier is reported only once src/switch_odbc.c:76: error: for each function it appears in.) src/switch_odbc.c:77: error: 'switch_odbc_handle_t' has no member named 'state' src/switch_odbc.c: In function 'switch_odbc_handle_disconnect': src/switch_odbc.c:96: error: 'switch_odbc_handle_t' has no member named 'state' cc1: warnings being treated as errors src/switch_odbc.c:97: warning: implicit declaration of function 'SQLDisconnect' src/switch_odbc.c:97: error: 'switch_odbc_handle_t' has no member named 'con' src/switch_odbc.c:105: error: 'switch_odbc_handle_t' has no member named 'state' src/switch_odbc.c: In function 'switch_odbc_handle_connect': src/switch_odbc.c:113: error: 'SQLINTEGER' undeclared (first use in this function) src/switch_odbc.c:113: error: expected ';' before 'err' src/switch_odbc.c:116: error: 'SQLSMALLINT' undeclared (first use in this function) src/switch_odbc.c:116: error: expected ';' before 'valueLength' src/switch_odbc.c:119: error: 'switch_odbc_handle_t' has no member named 'env' src/switch_odbc.c:119: error: 'SQL_NULL_HANDLE' undeclared (first use in this function) src/switch_odbc.c:120: warning: implicit declaration of function 'SQLAllocHandle' src/switch_odbc.c:120: error: 'SQL_HANDLE_ENV' undeclared (first use in this function) src/switch_odbc.c:120: error: 'switch_odbc_handle_t' has no member named 'env' src/switch_odbc.c:122: error: 'SQL_SUCCESS' undeclared (first use in this function) src/switch_odbc.c:122: error: 'SQL_SUCCESS_WITH_INFO' undeclared (first use in this function) src/switch_odbc.c:127: warning: implicit declaration of function 'SQLSetEnvAttr' src/switch_odbc.c:127: error: 'switch_odbc_handle_t' has no member named 'env' src/switch_odbc.c:127: error: 'SQL_ATTR_ODBC_VERSION' undeclared (first use in this function) src/switch_odbc.c:127: error: 'SQL_OV_ODBC3' undeclared (first use in this function) src/switch_odbc.c:131: warning: implicit declaration of function 'SQLFreeHandle' src/switch_odbc.c:131: error: 'switch_odbc_handle_t' has no member named 'env' src/switch_odbc.c:135: error: 'SQL_HANDLE_DBC' undeclared (first use in this function) src/switch_odbc.c:135: error: 'switch_odbc_handle_t' has no member named 'env' src/switch_odbc.c:135: error: 'switch_odbc_handle_t' has no member named 'con' src/switch_odbc.c:139: error: 'switch_odbc_handle_t' has no member named 'env' src/switch_odbc.c:142: warning: implicit declaration of function 'SQLSetConnectAttr' src/switch_odbc.c:142: error: 'switch_odbc_handle_t' has no member named 'con' src/switch_odbc.c:142: error: 'SQL_LOGIN_TIMEOUT' undeclared (first use in this function) src/switch_odbc.c:142: error: 'SQLPOINTER' undeclared (first use in this function) src/switch_odbc.c:142: error: expected expression before ')' token src/switch_odbc.c:144: error: 'switch_odbc_handle_t' has no member named 'state' src/switch_odbc.c:152: warning: implicit declaration of function 'SQLConnect' src/switch_odbc.c:152: error: 'switch_odbc_handle_t' has no member named 'con' src/switch_odbc.c:152: error: 'SQLCHAR' undeclared (first use in this function) src/switch_odbc.c:152: error: expected expression before ')' token src/switch_odbc.c:154: error: expected ';' before 'outstr' src/switch_odbc.c:155: error: expected ';' before 'outstrlen' src/switch_odbc.c:157: warning: implicit declaration of function 'SQLDriverConnect' src/switch_odbc.c:157: error: 'switch_odbc_handle_t' has no member named 'con' src/switch_odbc.c:157: error: expected expression before ')' token src/switch_odbc.c:163: error: too many arguments to function 'switch_odbc_handle_get_error' src/switch_odbc.c:167: warning: implicit declaration of function 'SQLGetDiagRec' src/switch_odbc.c:167: error: 'switch_odbc_handle_t' has no member named 'con' src/switch_odbc.c:167: error: 'err' undeclared (first use in this function) src/switch_odbc.c:170: error: 'switch_odbc_handle_t' has no member named 'env' src/switch_odbc.c:174: warning: implicit declaration of function 'SQLGetInfo' src/switch_odbc.c:174: error: 'switch_odbc_handle_t' has no member named 'con' src/switch_odbc.c:174: error: 'SQL_DRIVER_NAME' undeclared (first use in this function) src/switch_odbc.c:174: error: expected expression before ')' token src/switch_odbc.c:176: error: 'valueLength' undeclared (first use in this function) src/switch_odbc.c:177: error: 'switch_odbc_handle_t' has no member named 'odbc_driver' src/switch_odbc.c:177: error: 'switch_odbc_handle_t' has no member named 'odbc_driver' src/switch_odbc.c:177: error: 'switch_odbc_handle_t' has no member named 'odbc_driver' src/switch_odbc.c:177: error: 'switch_odbc_handle_t' has no member named 'odbc_driver' src/switch_odbc.c:177: error: 'switch_odbc_handle_t' has no member named 'odbc_driver' src/switch_odbc.c:177: error: 'switch_odbc_handle_t' has no member named 'odbc_driver' src/switch_odbc.c:180: error: 'switch_odbc_handle_t' has no member named 'odbc_driver' src/switch_odbc.c:180: error: 'switch_odbc_handle_t' has no member named 'odbc_driver' src/switch_odbc.c:180: error: 'switch_odbc_handle_t' has no member named 'odbc_driver' src/switch_odbc.c:181: error: 'switch_odbc_handle_t' has no member named 'is_firebird' src/switch_odbc.c:183: error: 'switch_odbc_handle_t' has no member named 'is_firebird' src/switch_odbc.c:187: error: 'switch_odbc_handle_t' has no member named 'state' src/switch_odbc.c: In function 'db_is_up': src/switch_odbc.c:194: error: 'SQLHSTMT' undeclared (first use in this function) src/switch_odbc.c:194: error: expected ';' before 'stmt' src/switch_odbc.c:195: error: 'SQLLEN' undeclared (first use in this function) src/switch_odbc.c:195: error: expected ';' before 'm' src/switch_odbc.c:200: error: 'SQLCHAR' undeclared (first use in this function) src/switch_odbc.c:200: error: expected ';' before 'sql' src/switch_odbc.c:203: error: 'SQLRETURN' undeclared (first use in this function) src/switch_odbc.c:203: error: expected ';' before 'rc' src/switch_odbc.c:204: error: 'SQLSMALLINT' undeclared (first use in this function) src/switch_odbc.c:204: error: expected ';' before 'nresultcols' src/switch_odbc.c:213: error: 'switch_odbc_handle_t' has no member named 'is_firebird' src/switch_odbc.c:214: error: 'sql' undeclared (first use in this function) src/switch_odbc.c:219: error: 'SQL_HANDLE_STMT' undeclared (first use in this function) src/switch_odbc.c:219: error: 'switch_odbc_handle_t' has no member named 'con' src/switch_odbc.c:219: error: 'stmt' undeclared (first use in this function) src/switch_odbc.c:219: error: 'SQL_SUCCESS' undeclared (first use in this function) src/switch_odbc.c:223: warning: implicit declaration of function 'SQLPrepare' src/switch_odbc.c:223: error: 'SQL_NTS' undeclared (first use in this function) src/switch_odbc.c:227: warning: implicit declaration of function 'SQLExecute' src/switch_odbc.c:229: error: 'SQL_SUCCESS_WITH_INFO' undeclared (first use in this function) src/switch_odbc.c:233: warning: implicit declaration of function 'SQLRowCount' src/switch_odbc.c:233: error: 'm' undeclared (first use in this function) src/switch_odbc.c:234: error: 'rc' undeclared (first use in this function) src/switch_odbc.c:234: warning: implicit declaration of function 'SQLNumResultCols' src/switch_odbc.c:234: error: 'nresultcols' undeclared (first use in this function) src/switch_odbc.c:248: error: too many arguments to function 'switch_odbc_handle_get_error' src/switch_odbc.c: At top level: src/switch_odbc.c:293: error: expected declaration specifiers or '...' before 'SQLHSTMT' src/switch_odbc.c: In function 'switch_odbc_handle_exec': src/switch_odbc.c:295: error: 'SQLHSTMT' undeclared (first use in this function) src/switch_odbc.c:295: error: expected ';' before 'stmt' src/switch_odbc.c:302: error: 'SQL_HANDLE_STMT' undeclared (first use in this function) src/switch_odbc.c:302: error: 'switch_odbc_handle_t' has no member named 'con' src/switch_odbc.c:302: error: 'stmt' undeclared (first use in this function) src/switch_odbc.c:302: error: 'SQL_SUCCESS' undeclared (first use in this function) src/switch_odbc.c:306: error: 'SQL_NTS' undeclared (first use in this function) src/switch_odbc.c:312: error: 'SQL_SUCCESS_WITH_INFO' undeclared (first use in this function) src/switch_odbc.c:316: error: 'rstmt' undeclared (first use in this function) src/switch_odbc.c: In function 'switch_odbc_handle_callback_exec_detailed': src/switch_odbc.c:337: error: 'SQLHSTMT' undeclared (first use in this function) src/switch_odbc.c:337: error: expected ';' before 'stmt' src/switch_odbc.c:338: error: 'SQLSMALLINT' undeclared (first use in this function) src/switch_odbc.c:338: error: expected ';' before 'c' src/switch_odbc.c:339: error: 'SQLLEN' undeclared (first use in this function) src/switch_odbc.c:339: error: expected ';' before 'm' src/switch_odbc.c:350: error: 'SQL_HANDLE_STMT' undeclared (first use in this function) src/switch_odbc.c:350: error: 'switch_odbc_handle_t' has no member named 'con' src/switch_odbc.c:350: error: 'stmt' undeclared (first use in this function) src/switch_odbc.c:350: error: 'SQL_SUCCESS' undeclared (first use in this function) src/switch_odbc.c:355: error: 'SQL_NTS' undeclared (first use in this function) src/switch_odbc.c:362: error: 'SQL_SUCCESS_WITH_INFO' undeclared (first use in this function) src/switch_odbc.c:366: error: 'c' undeclared (first use in this function) src/switch_odbc.c:367: error: 'm' undeclared (first use in this function) src/switch_odbc.c:369: error: 't' undeclared (first use in this function) src/switch_odbc.c:376: warning: implicit declaration of function 'SQLFetch' src/switch_odbc.c:390: error: 'x' undeclared (first use in this function) src/switch_odbc.c:391: error: expected ';' before 'NameLength' src/switch_odbc.c:392: error: 'SQLULEN' undeclared (first use in this function) src/switch_odbc.c:392: error: expected ';' before 'ColumnSize' src/switch_odbc.c:396: warning: implicit declaration of function 'SQLDescribeCol' src/switch_odbc.c:396: error: 'SQLCHAR' undeclared (first use in this function) src/switch_odbc.c:396: error: expected expression before ')' token src/switch_odbc.c:397: error: 'ColumnSize' undeclared (first use in this function) src/switch_odbc.c:401: warning: implicit declaration of function 'SQLGetData' src/switch_odbc.c:401: error: 'SQL_C_CHAR' undeclared (first use in this function) src/switch_odbc.c:401: error: expected expression before ')' token src/switch_odbc.c:436: error: too many arguments to function 'switch_odbc_handle_get_error' src/switch_odbc.c: In function 'switch_odbc_handle_destroy': src/switch_odbc.c:459: error: 'SQL_HANDLE_DBC' undeclared (first use in this function) src/switch_odbc.c:459: error: 'switch_odbc_handle_t' has no member named 'con' src/switch_odbc.c:460: error: 'SQL_HANDLE_ENV' undeclared (first use in this function) src/switch_odbc.c:460: error: 'switch_odbc_handle_t' has no member named 'env' src/switch_odbc.c: In function 'switch_odbc_handle_get_state': src/switch_odbc.c:471: error: 'switch_odbc_handle_t' has no member named 'state' src/switch_odbc.c: At top level: src/switch_odbc.c:474: error: expected declaration specifiers or '...' before 'SQLHSTMT' src/switch_odbc.c: In function 'switch_odbc_handle_get_error': src/switch_odbc.c:476: error: 'SQL_MAX_MESSAGE_LENGTH' undeclared (first use in this function) src/switch_odbc.c:477: error: 'SQL_SQLSTATE_SIZE' undeclared (first use in this function) src/switch_odbc.c:478: error: 'SQLINTEGER' undeclared (first use in this function) src/switch_odbc.c:478: error: expected ';' before 'sqlcode' src/switch_odbc.c:479: error: 'SQLSMALLINT' undeclared (first use in this function) src/switch_odbc.c:479: error: expected ';' before 'length' src/switch_odbc.c:482: warning: implicit declaration of function 'SQLError' src/switch_odbc.c:482: error: 'switch_odbc_handle_t' has no member named 'env' src/switch_odbc.c:482: error: 'switch_odbc_handle_t' has no member named 'con' src/switch_odbc.c:482: error: 'stmt' undeclared (first use in this function) src/switch_odbc.c:482: error: 'SQLCHAR' undeclared (first use in this function) src/switch_odbc.c:482: error: expected expression before ')' token src/switch_odbc.c:482: error: 'SQL_SUCCESS' undeclared (first use in this function) src/switch_odbc.c:483: error: 'sqlcode' undeclared (first use in this function) src/switch_odbc.c:477: warning: unused variable 'sqlstate' src/switch_odbc.c:476: warning: unused variable 'buffer' make[2]: *** [libfreeswitch_la-switch_odbc.lo] Error 1 Making all in src Making all in mod making all mod_amr make[5]: *** No rule to make target `/usr/src/FreeSwitch/freeswitch-1.0.3/libfreeswitch.la', needed by `mod_amr.so'.? Stop. make[4]: *** [all] Error 1 make[3]: *** [mod_amr-all] Error 1 make[2]: *** [all-recursive] Error 1 Making all in build ?+-------- FreeSWITCH Build Complete -----------+ ?+ FreeSWITCH has been successfully built.????? + ?+ Install by running:????????????????????????? + ?+????????????????????????????????????????????? + ?+?????????????? make install?????????????????? + ?+----------------------------------------------+ make[1]: *** [all-recursive] Error 1 make: *** [all] Error 2 --- On Thu, 2/19/09, freeswitch-users-request at lists.freeswitch.org wrote: Message: 3 Date: Thu, 19 Feb 2009 10:26:15 -0500 From: Raymond Chandler Subject: Re: [Freeswitch-users] Realm Value To: freeswitch-users at lists.freeswitch.org Message-ID: <499D7A17.7030202 at freeswitch.org> Content-Type: text/plain; charset="iso-8859-1" Ali Al-Rubaie wrote: > Thanks Brian, > > Sorry if the question looks primitive but in which file I can find the > rvn? > you can find the revision from typing "version" at the fs_cli > Is there any tarball with the latest revisions? > the 1.0.3 tarball was just rolled yesterday, give it a shot -Ray -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090219/89abc2b0/attachment-0001.html ------------------------------ Message: 4 Date: Thu, 19 Feb 2009 16:47:27 +0100 From: "Cavalera Claudio Luigi" Subject: Re: [Freeswitch-users] Random problems with cepstral text to speech To: Message-ID: Content-Type: text/plain; charset="us-ascii" From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale > Are you using cepstral 5.1? > There is a known issue with that release and it's closed source so we cannot do much about it. > Cepstral 4.x works fine. Yes 5.1, my fault. I have added an initial warning here on the wiki http://wiki.freeswitch.org/wiki/Mod_cepstral although it also speaks about 5.1 and Ubuntu... Thanks, Claudio Internet E. Mail Confidentiality Footer ----------------------------------------------------------------------------------------------------- La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. ------------------------------ Message: 5 Date: Thu, 19 Feb 2009 17:00:44 +0100 From: "Cavalera Claudio Luigi" Subject: Re: [Freeswitch-users] Strange error load testing mod_xml_rpc To: Message-ID: Content-Type: text/plain; charset="us-ascii" From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale > Yes we already see you have reported this issue in jira and we are working on it. > you do not need to report it here as well. > You may want to make yourself available on irc or some other im today so we can contact you for more details. I'm sorry I reported this message here because I did not think it's related with my JIRA reports. They are about two segmentation faults, while this message is about load testing not going very well, although it comes out during the same load tests :-) I'm starting to think that most people are using the sofia SIP stack in fs to get rate of 100cps while the other interfaces such as event socket and mod_xml_rpc are not well "engineered" yet. I'm going deeper into this and I hope my results will be of help for the community. At the moment I'm trying to understand the cause of TCP retransmissions I see in the snoop, it seems that fs lags a little bit in sending the TCP Ack back to the load runner, even with the machine doing nothing and a really tiny amount of load. BRs, Claudio Internet Email Confidentiality Footer ----------------------------------------------------------------------------------------------------- La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. ----------------------------------------------------------------------------------------------------- ------------------------------ Message: 6 Date: Thu, 19 Feb 2009 18:00:19 +0100 From: "Cavalera Claudio Luigi" Subject: Re: [Freeswitch-users] Strange error load testing mod_xml_rpc To: Message-ID: Content-Type: text/plain; charset="us-ascii" Here is a snoop: http://pastebin.freeswitch.org/7351 thx, cla Internet Email Confidentiality Footer ----------------------------------------------------------------------------------------------------- La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. ----------------------------------------------------------------------------------------------------- ------------------------------ Message: 7 Date: Thu, 19 Feb 2009 11:30:01 -0600 From: Anthony Minessale Subject: Re: [Freeswitch-users] Strange error load testing mod_xml_rpc To: freeswitch-users at lists.freeswitch.org Message-ID: <191c3a030902190930x121eba94jb644c25922cbdb7b at mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" api calls by default are blocking, it will not return until the result of the originate is determined. you must pre-empt the originate command with the bgapi api command which is similar to the event_socket bgapi command so that it tells the task to run in a dedicated thread. also the web interface was meant to use an xml rpc client. I have generated at least 400cps on event socket before. 2009/2/19 Cavalera Claudio Luigi > Here is a snoop: > http://pastebin.freeswitch.org/7351 > thx, > cla > > > Internet Email Confidentiality Footer > > ----------------------------------------------------------------------------------------------------- > La presente comunicazione, con le informazioni in essa contenute e ogni > documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' > indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete > i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, > comunicazione, divulgazione o simili basate sul contenuto di tali > informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., > D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se > avete ricevuto questa comunicazione per errore, vi preghiamo di darne > immediata notizia al mittente e di distruggere il messaggio originale e ogni > file allegato senza farne copia alcuna o riprodurne in alcun modo il > contenuto. > > This e-mail and its attachments are intended for the addressee(s) only and > are confidential and/or may contain legally privileged information. If you > have received this message by mistake or are not one of the addressees > above, you may take no action based on it, and you may not copy or show it > to anyone; please reply to this e-mail and point out the error which has > occurred. > > ----------------------------------------------------------------------------------------------------- > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090219/0c42b1d8/attachment.html ------------------------------ _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org End of Freeswitch-users Digest, Vol 32, Issue 166 ************************************************* -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090219/67854f71/attachment-0001.html From intralanman at freeswitch.org Thu Feb 19 12:17:21 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Thu, 19 Feb 2009 15:17:21 -0500 Subject: [Freeswitch-users] Realm Value In-Reply-To: <97318.60826.qm@web33708.mail.mud.yahoo.com> References: <97318.60826.qm@web33708.mail.mud.yahoo.com> Message-ID: <499DBE51.5010700@freeswitch.org> did you ./configure --enable-core-odbc-suport... those errors reek of that flag with no unixODBC-devel package installed -Ray From msc at freeswitch.org Thu Feb 19 13:47:09 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 19 Feb 2009 13:47:09 -0800 Subject: [Freeswitch-users] Voice pattern detection In-Reply-To: <20b801c99291$c0dc5c30$33014c0a@ws4> References: <20b801c99291$c0dc5c30$33014c0a@ws4> Message-ID: <87f2f3b90902191347g1750a132k4a71159c44ef54b1@mail.gmail.com> This is really advanced stuff. You're going to need to pay someone who really understands DSP and programming. You might want to start with consulting at freeswitch.org. -MC On Thu, Feb 19, 2009 at 4:58 AM, Frank @ Impact wrote: > I am trying to detect if a caller is an automated greeting voice. And if > so, take an action. > > > > I have samples of the caller recording that I am looking to match. So > this is like a really complex tone detection I guess. > > > > It would work like this. > > - Call comes in > > - We answer/bridge the call > > - We start to listen for about 10 seconds. > > - During this time we are trying to match a snippet of a sound sample (say 2 > or 3 seconds worth) we have recorded on a file on the server. We are trying > to match this sound sample to the caller side only. > > - If we hear a match, we take the action. Or if we don't hear the match, we > might take a different action. > > > > Any of this sound doable? Any guidance on how to accomplish this? > > > > -Frank > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From BenHoltsclaw at averyschools.net Thu Feb 19 12:23:26 2009 From: BenHoltsclaw at averyschools.net (Ben Holtsclaw) Date: Thu, 19 Feb 2009 15:23:26 -0500 Subject: [Freeswitch-users] Deployment information and use cases In-Reply-To: <1235016793.22050.0.camel@raul-laptop> References: <1234916455.16581.49.camel@raul-laptop> <59ad9ca10902181009p7cbc0fd1s9860aacde7f3a30c@mail.gmail.com> <247f8100902181019m61ce803m5ed504d9b198dfb6@mail.gmail.com> <1235016793.22050.0.camel@raul-laptop> Message-ID: <499D796E.45B7.0079.0@averyschools.net> Raul, I am in the process of rolling out a FreeSWITCH IP PBX solution similar to what you describe. When I was trying to procure funds for a FreeSWITCH solution, I looked for the same information you're after, but came up with little. I'll briefly describe what we're trying to accomplish, and the tools I'm using to do it. This is probably more information than what you are looking for, but maybe it will also benefit someone else. We had several schools with aging or dying PBX's or KSU's. Each site had something different system, and was supported by a different VAR. Of course, the VAR's charged some outlandish fee to make onsite repair visits. Some number of Centrex lines supplied each school's dial tone. All in all, we had a very outdated and financially draining mess. Our district's long term goal had been to move to a more unified phone system. That made sense for many reasons, the chief of which was cost. We already had a strong fiber WAN in place. Why not use that for trunking and eliminate the monthly cost of the Centrex lines? That's the path we started down. Being a public entity, we had to be sure to explore all possible avenues. We looked at everything from traditional PBX's with IP add-on modules for trunking to a full blown Cisco CallManager solution. With third party proprietary systems, we were just never able to find the sweet spot between required feature set and cost. Would Cisco have been a workable solution? Absolutely. Could our small, rural, K12 public school district afford that? Not in a million years. I looked at several software packages -- some open source, some not -- but always came back to FreeSWITCH. The scalability and active development community were major factors for us. Server Hardware. Each of our five sites has a dedicated FreeSWITCH server. For hardware, we went with Dell PowerEdge 1950's with dual quad core Xeon 2.33 GHz processors, and 4GB of RAM. I have mirrored disks set up with enough space to accommodate users' voicemail. Each server will average only about 60 voicemail boxes, and we're storing sound as MP3. Disk space shouldn't be an issue. We have always been a Novell shop, so SLES is naturally our Linux distribution of choice. We chose to go with server hardware at each site so that in the event of a WAN outage, we would still at least have intra-building and emergency communication (see below). Telephony Hardware. Each of our servers includes Sangoma hardware. We actually looked at doing IP trunking to a carrier from our network core, but decided to use telco provided PRI's instead. Presently, we have two PRI's that connect to a FreeSWITCH server at the center of our network via a Sangoma A102 dual port telephony card. All calls to and from the PSTN traverse this primary server. Servers at each remote site include one of Sangoma's A200 analog cards. Emergency calls to 911 route out over this analog card through one of at least two POTS lines that remain connected at each site. Not only does this provide some redundancy in the event of a WAN outage, but it ensures proper caller location is delivered to the 911 dispatcher. Granted, there are some other solutions for the latter, but this seemed to be the most cost effective solution for us. Telephone Desksets. We chose to go with Aastra for the telephones. The standard phone that we will place in each classroom and office is the 9143i. This is an attractive phone with an adequate feature set at a price we can afford. The person that is primarily responsible for answering the phone at each site will have an Aastra 57i and some number of 560M expansion modules. We have purchased roughly 300 Aastra desksets. Logical Layout. As new sites come online, their primary phone number is being moved from the Centrex to our PRI group. All inbound calls hit our primary server, and then FreeSWITCH bridges to the appropriate secondary server based on the DID it received. On the reverse, each servers dial plan is set up to route outbound calls (save 911) to the primary server where FreeSWITCH bridges with Openzap. Site to site calls, accomplished via four digit dialing, do not hit the primary server. Outbound calls to the PSTN deliver the site's DID as the calling number. In other words, if a user from site two calls my cell phone, I see site two's published telephone number on my caller ID. Our dial plans are set up so that receptionists at each site still answer all outside calls. If not answered, the call fails over to an IVR. Should we ever decide to do so, we are now perfectly positioned to have all inbound calls to the district answered by one operator or IVR. "Welcome, and thank you for calling Avery County Schools." Stumbling Blocks. Problems we've faced so far have primarily surrounded Openzap and the Sangoma Wanpipe driver. FreeSWITCH developers won't mind telling you that this is an area that is currently not well "funded" and not 100% complete. There is some known issue where voice channels on the PRI get stuck in the wrong state and become unusable. We have experienced this a couple of times and have not been able to make or receive calls. Bouncing the Wanpipe driver has fixed this each time. We have also had trouble with DTMF detection across the PRI. If a user hits the IVR, it is oftentimes difficult to get it to properly recognize the digits that are being keyed in by the caller. This can be very, very frustrating to a caller that doesn't want to deal with an IVR anyway. The developers have suggested to me that this is a problem with the Sangoma's echo cancellation goofing up Openzap's ability to interpret the DTMF. The Sangoma hardware does have its own DTMF decoder and API, but the Openzap code currently does not make use of it. I have created a patch that makes use of the hardware decoder. We have been running it in production for a couple of weeks, and that does seem to have helped the problem. The problem hasn't gone away altogether. Those have been our two biggest issues, but we haven't let them hold us up. Conclusion. Of the five sites that will be on this system, one is fully functional with calls inbound and outbound from the PSTN. A second site is up and running with full outbound PSTN access. Their inbound DID is scheduled to move over to the PRI in one week. The server has been worked up for a third site, and the phones are starting to roll out. Sites four and five should come online by the end of April. Currently, I don't have numbers compiled for things like concurrent calls. At this point in my project, it is just not important. I really don't think our implementation will ever push FreeSWITCH's abilities in that regard. I base that statement primarily on other users' benchmarks, and what I've heard some are doing in carrier class environments. FreeSWITCH has made our project viable. An open source solution was the only way we could meet all of the project goals and stay within our budget. FreeSWITCH has proven to have all the features we require in a district wide phone system. It has not locked us into annual support contracts with third party vendors. I could go on with the accolades. However, I'll end this terribly lengthy post by saying that, overall, we have been very pleased with our choice to go with FreeSWITCH. The information in this email will seem very elementary to most people on this list, but having a message of this nature in hand would have made me feel much more confident the first time I ever went to my supervisor to mention something called FreeSWITCH. :-) Thanks Tony, Brian, and Mike for a great product! Ben Holtsclaw Network Engineer Avery County Schools Ph: 828.733.3567 x2301 >>> On 2/18/2009 at 11:13 PM, Raul Fragoso wrote: Thanks guys, this is very useful information. Anyone else willing to share your experience ? Regards, Raul On Wed, 2009-02-18 at 16:19 -0200, Pablo Hernan Saro wrote: > Hi Raul, > > In my company (http://www.globant.com) we're using FreeSWITCH for high > quality conference services, integrated with OpenSIPS > (http://www.opensips.org) and Asterisk. Its performance is pretty > good. > > Pablo > > On Wed, Feb 18, 2009 at 4:09 PM, Henry Huang wrote: > > bandwidth.com has a service called phonebooth which is developed upon > > freeswitch. > > > > > > On Tue, Feb 17, 2009 at 4:20 PM, Raul Fragoso wrote: > >> > >> Hello FreeSWITCHERS, > >> > >> My company is currently creating a suite of applications which uses > >> FreeSWITCH as the back-end for an IP-PBX solution. We currently have a > >> prospect to have our first customer installation - a governmental > >> department. That is a tender to have an IP-PBX installation to connect > >> their four office branches, each one with about 300 users - which I am > >> sure FreeSWITCH is able to handle. Since this is an official tender, > >> it's part of their protocol to ask about real sites using the product. > >> > >> Having said that, would you mind sharing some information about your > >> experience with FreeSWITCH deployments ? > >> > >> No need to give many details, but a short summary with company name (if > >> possible), when it was deployed, server equipment, number of users, > >> number of concurrent calls, what kind of functions and services are used > >> and overall capacity of the system. > >> > >> I would really appreciate if you can share that information. And if you > >> guys agree (and explicitly manifest your agreement), I can compile the > >> information in the FreeSWITCH wiki under a "Use Cases" page so it can > >> serve as a common reference as well. > >> > >> Kind regards, > >> > >> Raul Fragoso > >> > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > > > -- > > Henry Huang > > UniC Solution - Communication Unified > > VoIP & Open Source software Consultant > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090219/e0bf6b3e/attachment.html From brian at freeswitch.org Thu Feb 19 13:50:34 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 19 Feb 2009 15:50:34 -0600 Subject: [Freeswitch-users] Voice pattern detection In-Reply-To: <87f2f3b90902191347g1750a132k4a71159c44ef54b1@mail.gmail.com> References: <20b801c99291$c0dc5c30$33014c0a@ws4> <87f2f3b90902191347g1750a132k4a71159c44ef54b1@mail.gmail.com> Message-ID: Plus its not an exact science in the first place. /b On Feb 19, 2009, at 3:47 PM, Michael Collins wrote: > This is really advanced stuff. You're going to need to pay someone who > really understands DSP and programming. You might want to start with > consulting at freeswitch.org. > > -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090219/5a405300/attachment-0001.html From frank at impactfax.com Thu Feb 19 14:12:44 2009 From: frank at impactfax.com (Frank @ Impact) Date: Thu, 19 Feb 2009 17:12:44 -0500 Subject: [Freeswitch-users] Voice pattern detection In-Reply-To: <87f2f3b90902191347g1750a132k4a71159c44ef54b1@mail.gmail.com> Message-ID: <253f01c992df$30d5a580$33014c0a@ws4> Ok. Maybe it is more like answering machine detection in reverse? Detection on the caller leg instead of the called leg. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Thursday, February 19, 2009 4:47 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Voice pattern detection This is really advanced stuff. You're going to need to pay someone who really understands DSP and programming. You might want to start with consulting at freeswitch.org. From jaugenstine at gmail.com Thu Feb 19 14:13:39 2009 From: jaugenstine at gmail.com (jonathan augenstine) Date: Thu, 19 Feb 2009 14:13:39 -0800 Subject: [Freeswitch-users] Pika development Message-ID: <207e7a5e0902191413j4e7ac724uf6abc4ee94f2d45b@mail.gmail.com> I have heard a rumor that Pika support was being developed for Freeswitch. Is that still going on? Can someone tell me if the rumor is true or not, and if so, what is the status of the development? Thank you. Jonathan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090219/e66cc465/attachment.html From brian at freeswitch.org Thu Feb 19 14:19:53 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 19 Feb 2009 16:19:53 -0600 Subject: [Freeswitch-users] Pika development In-Reply-To: <207e7a5e0902191413j4e7ac724uf6abc4ee94f2d45b@mail.gmail.com> References: <207e7a5e0902191413j4e7ac724uf6abc4ee94f2d45b@mail.gmail.com> Message-ID: <3C32E446-8E6B-4738-AAE5-4422CD782979@freeswitch.org> Pika hardware already works with OpenZAP. /b On Feb 19, 2009, at 4:13 PM, jonathan augenstine wrote: > I have heard a rumor that Pika support was being developed for > Freeswitch. Is that still going on? Can someone tell me if the > rumor is true or not, and if so, what is the status of the > development? > > Thank you. > Jonathan From msc at freeswitch.org Thu Feb 19 14:22:31 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 19 Feb 2009 14:22:31 -0800 Subject: [Freeswitch-users] Pika development In-Reply-To: <207e7a5e0902191413j4e7ac724uf6abc4ee94f2d45b@mail.gmail.com> References: <207e7a5e0902191413j4e7ac724uf6abc4ee94f2d45b@mail.gmail.com> Message-ID: <87f2f3b90902191422w66834fe5m6764508ae08e2dd8@mail.gmail.com> On Thu, Feb 19, 2009 at 2:13 PM, jonathan augenstine wrote: > I have heard a rumor that Pika support was being developed for Freeswitch. > Is that still going on? Can someone tell me if the rumor is true or not, > and if so, what is the status of the development? Well, the PIKA cards work with FS and they have an appliance they were showing off at ClueCon last year... not sure what else is in the pipeline. -MC From msc at freeswitch.org Thu Feb 19 14:35:29 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 19 Feb 2009 14:35:29 -0800 Subject: [Freeswitch-users] Realm Value In-Reply-To: <499DBE51.5010700@freeswitch.org> References: <97318.60826.qm@web33708.mail.mud.yahoo.com> <499DBE51.5010700@freeswitch.org> Message-ID: <87f2f3b90902191435p1c9c03aend3303dfb013495b1@mail.gmail.com> On Thu, Feb 19, 2009 at 12:17 PM, Raymond Chandler wrote: > did you ./configure --enable-core-odbc-suport... those errors reek of > that flag with no unixODBC-devel package installed > > -Ray > Anthony described this as a false positive on detecting ODBC. If you are in Linux you can install the ODBC-devel package and be done with it. -MC From anthony.minessale at gmail.com Thu Feb 19 16:56:44 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 19 Feb 2009 18:56:44 -0600 Subject: [Freeswitch-users] Pika development In-Reply-To: <87f2f3b90902191422w66834fe5m6764508ae08e2dd8@mail.gmail.com> References: <207e7a5e0902191413j4e7ac724uf6abc4ee94f2d45b@mail.gmail.com> <87f2f3b90902191422w66834fe5m6764508ae08e2dd8@mail.gmail.com> Message-ID: <191c3a030902191656l788940c8uabe8077e3c81ad17@mail.gmail.com> If you get a pika card to play with on FS, please inform them that it is for that purpose so they can help make sure you get it going. On Thu, Feb 19, 2009 at 4:22 PM, Michael Collins wrote: > On Thu, Feb 19, 2009 at 2:13 PM, jonathan augenstine > wrote: > > I have heard a rumor that Pika support was being developed for > Freeswitch. > > Is that still going on? Can someone tell me if the rumor is true or not, > > and if so, what is the status of the development? > > Well, the PIKA cards work with FS and they have an appliance they were > showing off at ClueCon last year... not sure what else is in the > pipeline. > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090219/4be497a9/attachment.html From gmaruzz at celliax.org Thu Feb 19 17:32:46 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Fri, 20 Feb 2009 02:32:46 +0100 Subject: [Freeswitch-users] Skypiax (Skype Network endpoint) for FreeSWITCH In-Reply-To: <5800526b0902181853h32de5535l6c0f9d7067a69cf0@mail.gmail.com> References: <7b197bef0902131015k456dee87x8f0f1b7059f1b49c@mail.gmail.com> <5800526b0902181853h32de5535l6c0f9d7067a69cf0@mail.gmail.com> Message-ID: <7b197bef0902191732i6fead849uace0ac906a9437b0@mail.gmail.com> On Thu, Feb 19, 2009 at 3:53 AM, Carlos Talbot wrote: > One question I have, is ringback suppose to work with mod_skypiax? Whenever > I dial a number I get a few seconds of dead air before the call is answered. > I've tried adding ringback and transfer_ringback into the dialplan just > before the bridge command but no go. Am I missing something? Thanks. Carlos, ringback now works without tricks, and Skypiax is in trunk. Both remote ringing and early media are treated as remote ringing right now (eg: no early media, just ringing). I'll add early media support in the near future. Thanks a lot for testing and exercising skypiax, and please let me know any hint, suggestion, feature request, etc Sincerely, Giovanni Maruzzelli ========================================= www.celliax.org via Pierlombardo 9, 20135 Milano Italy gmaruzz at celliax dot org Cell : +39-347-2665618 Fax : +39-02-87390039 On Thu, Feb 19, 2009 at 3:53 AM, Carlos Talbot wrote: > Giovannia, > > great work on mod_skypiax. I've been testing it under Windows and it sounds > great including PSTN calls. I plan to include it as part of the Windows MSI > build. > > One question I have, is ringback suppose to work with mod_skypiax? Whenever > I dial a number I get a few seconds of dead air before the call is answered. > I've tried adding ringback and transfer_ringback into the dialplan just > before the bridge command but no go. Am I missing something? Thanks. > > regards, > > Carlos > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From jaugenstine at gmail.com Thu Feb 19 18:29:33 2009 From: jaugenstine at gmail.com (jonathan augenstine) Date: Thu, 19 Feb 2009 18:29:33 -0800 Subject: [Freeswitch-users] Pika development In-Reply-To: <191c3a030902191656l788940c8uabe8077e3c81ad17@mail.gmail.com> References: <207e7a5e0902191413j4e7ac724uf6abc4ee94f2d45b@mail.gmail.com> <87f2f3b90902191422w66834fe5m6764508ae08e2dd8@mail.gmail.com> <191c3a030902191656l788940c8uabe8077e3c81ad17@mail.gmail.com> Message-ID: <207e7a5e0902191829o6f23e2dej6362461f212ad939@mail.gmail.com> Anthony/Michael/Brian, Thank you for all the input. I appreciate the responses. I will certainly make sure they are aware of the application if I get the green light. Jonathan On Thu, Feb 19, 2009 at 4:56 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > If you get a pika card to play with on FS, please inform them that it is > for that purpose so they > can help make sure you get it going. > > > > On Thu, Feb 19, 2009 at 4:22 PM, Michael Collins wrote: > >> On Thu, Feb 19, 2009 at 2:13 PM, jonathan augenstine >> wrote: >> > I have heard a rumor that Pika support was being developed for >> Freeswitch. >> > Is that still going on? Can someone tell me if the rumor is true or >> not, >> > and if so, what is the status of the development? >> >> Well, the PIKA cards work with FS and they have an appliance they were >> showing off at ClueCon last year... not sure what else is in the >> pipeline. >> -MC >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090219/0d8df332/attachment.html From cesar.bermudez at gmail.com Thu Feb 19 18:38:31 2009 From: cesar.bermudez at gmail.com (Cesar Bermudez) Date: Fri, 20 Feb 2009 03:38:31 +0100 Subject: [Freeswitch-users] Pika development In-Reply-To: <207e7a5e0902191829o6f23e2dej6362461f212ad939@mail.gmail.com> References: <207e7a5e0902191413j4e7ac724uf6abc4ee94f2d45b@mail.gmail.com> <87f2f3b90902191422w66834fe5m6764508ae08e2dd8@mail.gmail.com> <191c3a030902191656l788940c8uabe8077e3c81ad17@mail.gmail.com> <207e7a5e0902191829o6f23e2dej6362461f212ad939@mail.gmail.com> Message-ID: this is for the pika warp http://svn.pikatech.com/pads/distro/branches/freeswitch-1.0.0/ On Fri, Feb 20, 2009 at 3:29 AM, jonathan augenstine wrote: > Anthony/Michael/Brian, > > Thank you for all the input. I appreciate the responses. I will certainly > make sure they are aware of the application if I get the green light. > > Jonathan > > > On Thu, Feb 19, 2009 at 4:56 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> If you get a pika card to play with on FS, please inform them that it is >> for that purpose so they >> can help make sure you get it going. >> >> >> >> On Thu, Feb 19, 2009 at 4:22 PM, Michael Collins wrote: >> >>> On Thu, Feb 19, 2009 at 2:13 PM, jonathan augenstine >>> wrote: >>> > I have heard a rumor that Pika support was being developed for >>> Freeswitch. >>> > Is that still going on? Can someone tell me if the rumor is true or >>> not, >>> > and if so, what is the status of the development? >>> >>> Well, the PIKA cards work with FS and they have an appliance they were >>> showing off at ClueCon last year... not sure what else is in the >>> pipeline. >>> -MC >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:213-799-1400 >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090220/6b95ee8e/attachment-0001.html From jason at jasonjgw.net Thu Feb 19 18:52:09 2009 From: jason at jasonjgw.net (Jason White) Date: Fri, 20 Feb 2009 13:52:09 +1100 Subject: [Freeswitch-users] Calling an IPv6 host with a SIP URI from portAudio Message-ID: <20090220025209.GA12844@jdc.jasonjgw.net> I notice that pa call sip:nnnn at host fails if the host in question only has an IPv6 address (i.e., an AAAA record in DNS). The logs show that FreeSWITCH is trying to use the internal profile, and failing. If I write a dialplan extension that accesses the same address using the internal-ipv6 profile, it succeeds. In the supplied default.xml dial plan, the SIP URI is processed thus: I can't find any documentation of use_profile on the wiki, but clearly it takes the value "internal" in this case. What would be the best way to fix this so that it will work regardless of whether the host is reachable over IPv4 or IPv6, or both? I could rewrite the extension to try multiple SIP profiles, but there could be a better way - hence the question. From brian at freeswitch.org Thu Feb 19 18:55:39 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 19 Feb 2009 20:55:39 -0600 Subject: [Freeswitch-users] Calling an IPv6 host with a SIP URI from portAudio In-Reply-To: <20090220025209.GA12844@jdc.jasonjgw.net> References: <20090220025209.GA12844@jdc.jasonjgw.net> Message-ID: <4F757A97-34E5-48A0-94D2-07883BFE23C3@freeswitch.org> Just make sure you send the call out an ipv6 profile and it'll work. /b On Feb 19, 2009, at 8:52 PM, Jason White wrote: > > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090219/aa71b4bd/attachment.html From jason at jasonjgw.net Thu Feb 19 19:53:17 2009 From: jason at jasonjgw.net (Jason White) Date: Fri, 20 Feb 2009 14:53:17 +1100 Subject: [Freeswitch-users] Calling an IPv6 host with a SIP URI from portAudio In-Reply-To: <4F757A97-34E5-48A0-94D2-07883BFE23C3@freeswitch.org> References: <20090220025209.GA12844@jdc.jasonjgw.net> <4F757A97-34E5-48A0-94D2-07883BFE23C3@freeswitch.org> Message-ID: <20090220035317.GA13566@jdc.jasonjgw.net> I have it working now. The relevant changes were as follows. From raul at etellicom.com Thu Feb 19 20:18:00 2009 From: raul at etellicom.com (Raul Fragoso) Date: Fri, 20 Feb 2009 01:18:00 -0300 Subject: [Freeswitch-users] Deployment information and use cases In-Reply-To: <499D796E.45B7.0079.0@averyschools.net> References: <1234916455.16581.49.camel@raul-laptop> <59ad9ca10902181009p7cbc0fd1s9860aacde7f3a30c@mail.gmail.com> <247f8100902181019m61ce803m5ed504d9b198dfb6@mail.gmail.com> <1235016793.22050.0.camel@raul-laptop> <499D796E.45B7.0079.0@averyschools.net> Message-ID: <1235103480.30511.346.camel@raul-laptop> Ben, Wow !!! Thank you very much for such descriptive and detailed information ! Indeed, this is really more than I expected, and once again I thank you for your collaboration. It's very cheering and inspiring to hear such successful story regarding FreeSWITCH. Kind regards, Raul On Thu, 2009-02-19 at 15:23 -0500, Ben Holtsclaw wrote: > Raul, > > I am in the process of rolling out a FreeSWITCH IP PBX solution > similar to what you describe. When I was trying to procure funds for a > FreeSWITCH solution, I looked for the same information you're after, > but came up with little. I'll briefly describe what we're trying to > accomplish, and the tools I'm using to do it. This is probably more > information than what you are looking for, but maybe it will also > benefit someone else. > > We had several schools with aging or dying PBX's or KSU's. Each site > had something different system, and was supported by a different > VAR. Of course, the VAR's charged some outlandish fee to make onsite > repair visits. Some number of Centrex lines supplied each school's > dial tone. All in all, we had a very outdated and financially draining > mess. Our district's long term goal had been to move to a more unified > phone system. That made sense for many reasons, the chief of which was > cost. We already had a strong fiber WAN in place. Why not use that for > trunking and eliminate the monthly cost of the Centrex lines? That's > the path we started down. > > Being a public entity, we had to be sure to explore all possible > avenues. We looked at everything from traditional PBX's with IP add-on > modules for trunking to a full blown Cisco CallManager solution. With > third party proprietary systems, we were just never able to find the > sweet spot between required feature set and cost. Would Cisco have > been a workable solution? Absolutely. Could our small, rural, K12 > public school district afford that? Not in a million years. I looked > at several software packages -- some open source, some not -- but > always came back to FreeSWITCH. The scalability and active development > community were major factors for us. > > Server Hardware. Each of our five sites has a dedicated FreeSWITCH > server. For hardware, we went with Dell PowerEdge 1950's with dual > quad core Xeon 2.33 GHz processors, and 4GB of RAM. I have mirrored > disks set up with enough space to accommodate users' voicemail. Each > server will average only about 60 voicemail boxes, and we're storing > sound as MP3. Disk space shouldn't be an issue. We have always been a > Novell shop, so SLES is naturally our Linux distribution of choice. We > chose to go with server hardware at each site so that in the event of > a WAN outage, we would still at least have intra-building and > emergency communication (see below). > > Telephony Hardware. Each of our servers includes Sangoma hardware. We > actually looked at doing IP trunking to a carrier from our network > core, but decided to use telco provided PRI's instead. Presently, we > have two PRI's that connect to a FreeSWITCH server at the center of > our network via a Sangoma A102 dual port telephony card. All calls to > and from the PSTN traverse this primary server. Servers at each remote > site include one of Sangoma's A200 analog cards. Emergency calls to > 911 route out over this analog card through one of at least two POTS > lines that remain connected at each site. Not only does this provide > some redundancy in the event of a WAN outage, but it ensures proper > caller location is delivered to the 911 dispatcher. Granted, there are > some other solutions for the latter, but this seemed to be the most > cost effective solution for us. > > Telephone Desksets. We chose to go with Aastra for the telephones. The > standard phone that we will place in each classroom and office is the > 9143i. This is an attractive phone with an adequate feature set at a > price we can afford. The person that is primarily responsible for > answering the phone at each site will have an Aastra 57i and some > number of 560M expansion modules. We have purchased roughly 300 Aastra > desksets. > > Logical Layout. As new sites come online, their primary phone number > is being moved from the Centrex to our PRI group. All inbound calls > hit our primary server, and then FreeSWITCH bridges to the appropriate > secondary server based on the DID it received. On the reverse, each > servers dial plan is set up to route outbound calls (save 911) to the > primary server where FreeSWITCH bridges with Openzap. Site to site > calls, accomplished via four digit dialing, do not hit the primary > server. Outbound calls to the PSTN deliver the site's DID as the > calling number. In other words, if a user from site two calls my cell > phone, I see site two's published telephone number on my caller ID. > Our dial plans are set up so that receptionists at each site still > answer all outside calls. If not answered, the call fails over to an > IVR. Should we ever decide to do so, we are now perfectly positioned > to have all inbound calls to the district answered by one operator or > IVR. "Welcome, and thank you for calling Avery County Schools." > > Stumbling Blocks. Problems we've faced so far have primarily > surrounded Openzap and the Sangoma Wanpipe driver. FreeSWITCH > developers won't mind telling you that this is an area that is > currently not well "funded" and not 100% complete. There is some known > issue where voice channels on the PRI get stuck in the wrong state and > become unusable. We have experienced this a couple of times and have > not been able to make or receive calls. Bouncing the Wanpipe driver > has fixed this each time. We have also had trouble with DTMF detection > across the PRI. If a user hits the IVR, it is oftentimes difficult to > get it to properly recognize the digits that are being keyed in by the > caller. This can be very, very frustrating to a caller that doesn't > want to deal with an IVR anyway. The developers have suggested to me > that this is a problem with the Sangoma's echo cancellation goofing up > Openzap's ability to interpret the DTMF. The Sangoma hardware does > have its own DTMF decoder and API, but the Openzap code currently does > not make use of it. I have created a patch that makes use of the > hardware decoder. We have been running it in production for a couple > of weeks, and that does seem to have helped the problem. The problem > hasn't gone away altogether. Those have been our two biggest issues, > but we haven't let them hold us up. > > Conclusion. Of the five sites that will be on this system, one is > fully functional with calls inbound and outbound from the PSTN. A > second site is up and running with full outbound PSTN access. Their > inbound DID is scheduled to move over to the PRI in one week. The > server has been worked up for a third site, and the phones are > starting to roll out. Sites four and five should come online by the > end of April. Currently, I don't have numbers compiled for things like > concurrent calls. At this point in my project, it is just not > important. I really don't think our implementation will ever push > FreeSWITCH's abilities in that regard. I base that statement primarily > on other users' benchmarks, and what I've heard some are doing in > carrier class environments. > > FreeSWITCH has made our project viable. An open source solution was > the only way we could meet all of the project goals and stay within > our budget. FreeSWITCH has proven to have all the features we require > in a district wide phone system. It has not locked us into annual > support contracts with third party vendors. I could go on with the > accolades. However, I'll end this terribly lengthy post by saying > that, overall, we have been very pleased with our choice to go with > FreeSWITCH. > > The information in this email will seem very elementary to most people > on this list, but having a message of this nature in hand would have > made me feel much more confident the first time I ever went to my > supervisor to mention something called FreeSWITCH. :-) Thanks Tony, > Brian, and Mike for a great product! > > > Ben Holtsclaw > Network Engineer > Avery County Schools > Ph: 828.733.3567 x2301 > > > >>> On 2/18/2009 at 11:13 PM, Raul Fragoso wrote: > > Thanks guys, this is very useful information. > > Anyone else willing to share your experience ? > > Regards, > > Raul > > On Wed, 2009-02-18 at 16:19 -0200, Pablo Hernan Saro wrote: > > Hi Raul, > > > > In my company (http://www.globant.com) we're using FreeSWITCH for > high > > quality conference services, integrated with OpenSIPS > > (http://www.opensips.org) and Asterisk. Its performance is pretty > > good. > > > > Pablo > > > > On Wed, Feb 18, 2009 at 4:09 PM, Henry Huang > wrote: > > > bandwidth.com has a service called phonebooth which is developed > upon > > > freeswitch. > > > > > > > > > On Tue, Feb 17, 2009 at 4:20 PM, Raul Fragoso > wrote: > > >> > > >> Hello FreeSWITCHERS, > > >> > > >> My company is currently creating a suite of applications which > uses > > >> FreeSWITCH as the back-end for an IP-PBX solution. We currently > have a > > >> prospect to have our first customer installation - a governmental > > >> department. That is a tender to have an IP-PBX installation to > connect > > >> their four office branches, each one with about 300 users - which > I am > > >> sure FreeSWITCH is able to handle. Since this is an official > tender, > > >> it's part of their protocol to ask about real sites using the > product. > > >> > > >> Having said that, would you mind sharing some information about > your > > >> experience with FreeSWITCH deployments ? > > >> > > >> No need to give many details, but a short summary with company > name (if > > >> possible), when it was deployed, server equipment, number of > users, > > >> number of concurrent calls, what kind of functions and services > are used > > >> and overall capacity of the system. > > >> > > >> I would really appreciate if you can share that information. And > if you > > >> guys agree (and explicitly manifest your agreement), I can > compile the > > >> information in the FreeSWITCH wiki under a "Use Cases" page so it > can > > >> serve as a common reference as well. > > >> > > >> Kind regards, > > >> > > >> Raul Fragoso > > >> > > >> > > >> _______________________________________________ > > >> Freeswitch-users mailing list > > >> Freeswitch-users at lists.freeswitch.org > > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > >> http://www.freeswitch.org > > > > > > > > > > > > -- > > > Henry Huang > > > UniC Solution - Communication Unified > > > VoIP & Open Source software Consultant > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From stevecrozz at gmail.com Thu Feb 19 21:27:05 2009 From: stevecrozz at gmail.com (Stephen Crosby) Date: Thu, 19 Feb 2009 21:27:05 -0800 Subject: [Freeswitch-users] js session.streamFile() interrupt slowness Message-ID: <11990ade0902192127l414ecfeaia04fb80f626c35d5@mail.gmail.com> I have a few scripts that use the javascript session.streamFile('somefile.wav', onDtmf); where onDtmf is a function that returns false to interrupt the streaming file. There is a short delay between the time when I press a key and the time the file stops playing. Is there anything I can adjust that would affect that? It's only maybe 2-3 seconds, but it "feels" too long to me. --Stephen From msc at freeswitch.org Thu Feb 19 22:22:19 2009 From: msc at freeswitch.org (Michael S Collins) Date: Thu, 19 Feb 2009 22:22:19 -0800 Subject: [Freeswitch-users] Calling an IPv6 host with a SIP URI from portAudio In-Reply-To: <20090220035317.GA13566@jdc.jasonjgw.net> References: <20090220025209.GA12844@jdc.jasonjgw.net> <4F757A97-34E5-48A0-94D2-07883BFE23C3@freeswitch.org> <20090220035317.GA13566@jdc.jasonjgw.net> Message-ID: <9426570E-EE76-4BAD-9E44-1AA5622FFA8E@freeswitch.org> On Feb 19, 2009, at 7:53 PM, Jason White wrote: > I have it working now. The relevant changes were as follows. > > > > > > > Jason, I like this approach. Good use of the many Dialplan tools. -MC From pmhshz at gmail.com Fri Feb 20 02:18:29 2009 From: pmhshz at gmail.com (shehzad p) Date: Fri, 20 Feb 2009 02:18:29 -0800 (PST) Subject: [Freeswitch-users] Suggestion for xml_curl performance Message-ID: <22118122.post@talk.nabble.com> Hi all, Recently I faced some performance bottleneck by using Javascript. Now I am testing xml_curl for next setup, In terms of performance and stability will any body give me some information, what are the pros & cons of using xml_curl, what is precaution for using it, or any other recommendations... Thanks, msp -- View this message in context: http://www.nabble.com/Suggestion-for-xml_curl-performance-tp22118122p22118122.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From pmhshz at gmail.com Fri Feb 20 02:59:24 2009 From: pmhshz at gmail.com (shehzad p) Date: Fri, 20 Feb 2009 02:59:24 -0800 (PST) Subject: [Freeswitch-users] Suggestion for xml_curl performance In-Reply-To: <22118122.post@talk.nabble.com> References: <22118122.post@talk.nabble.com> Message-ID: <22118614.post@talk.nabble.com> My setup of the system is like: When calls come from Originator gateway I route the call to Terminator Gateway based on database lookup. FS works as switching platform. In previous setup using JavaScript, JavaScript caused the performance bottleneck when call traffic increases. Now I am testing xml_curl so asking for any suggestion, if some one has experienced... shehzad p wrote: > > Hi all, > > Recently I faced some performance bottleneck by using Javascript. > > Now I am testing xml_curl for next setup, > In terms of performance and stability will any body give me some > information, what are the pros & cons of using xml_curl, what is > precaution for using it, or any other recommendations... > > Thanks, > msp > -- View this message in context: http://www.nabble.com/Suggestion-for-xml_curl-performance-tp22118122p22118614.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From kokoska.rokoska at post.cz Fri Feb 20 03:24:20 2009 From: kokoska.rokoska at post.cz (kokoska rokoska) Date: Fri, 20 Feb 2009 12:24:20 +0100 Subject: [Freeswitch-users] SIP dump to DB Message-ID: <499E92E4.5010503@post.cz> Hi all, I'm facing the problem I need all SIP messages "going thru" FreeSWITCH (I know FS i B2BUA - so, better to say just "all SIP messages") logged somewhere and this log have to be "searchable" (by call-id etc) and I should be able to simply delete "old" messages... And more over - it should be done on not trivial SIP messages amount - say hundreds messages per second. My questions are: 1. Do you have any suggestion how to do it with FreeSWITCH? 2. Or it is not possible now, and "bounty" is necessary? :-) 3. How hard it will be to implement? ---------- FYI: I think something like SER/Kamailio/OpenSIPS siptrace is what I'am (probably) looking for: http://www.kamailio.org/docs/modules/1.4.x/siptrace.html It could go (based on my tests) up to 7-10.000 messages per second till MySQL "dies"... ---------- Thanks for your time :-) Best regards, kokoska.rokoska From anthony.minessale at gmail.com Fri Feb 20 05:54:59 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 20 Feb 2009 07:54:59 -0600 Subject: [Freeswitch-users] SIP dump to DB In-Reply-To: <499E92E4.5010503@post.cz> References: <499E92E4.5010503@post.cz> Message-ID: <191c3a030902200554s4abda94g76ba4e2975fffa98@mail.gmail.com> what exact info do you need? That's likely to be a challenge with any database to store at that speed. On Fri, Feb 20, 2009 at 5:24 AM, kokoska rokoska wrote: > > Hi all, > > I'm facing the problem I need all SIP messages "going thru" FreeSWITCH > (I know FS i B2BUA - so, better to say just "all SIP messages") logged > somewhere and this log have to be "searchable" (by call-id etc) and I > should be able to simply delete "old" messages... > And more over - it should be done on not trivial SIP messages amount - > say hundreds messages per second. > > My questions are: > 1. Do you have any suggestion how to do it with FreeSWITCH? > 2. Or it is not possible now, and "bounty" is necessary? :-) > 3. How hard it will be to implement? > > ---------- > FYI: > I think something like SER/Kamailio/OpenSIPS siptrace is what I'am > (probably) looking for: > http://www.kamailio.org/docs/modules/1.4.x/siptrace.html > It could go (based on my tests) up to 7-10.000 messages per second till > MySQL "dies"... > ---------- > > Thanks for your time :-) > > Best regards, > > kokoska.rokoska > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090220/af47481d/attachment.html From anthony.minessale at gmail.com Fri Feb 20 06:21:08 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 20 Feb 2009 08:21:08 -0600 Subject: [Freeswitch-users] Calling an IPv6 host with a SIP URI from portAudio In-Reply-To: <9426570E-EE76-4BAD-9E44-1AA5622FFA8E@freeswitch.org> References: <20090220025209.GA12844@jdc.jasonjgw.net> <4F757A97-34E5-48A0-94D2-07883BFE23C3@freeswitch.org> <20090220035317.GA13566@jdc.jasonjgw.net> <9426570E-EE76-4BAD-9E44-1AA5622FFA8E@freeswitch.org> Message-ID: <191c3a030902200621i37e61312r36998c73164dc7da@mail.gmail.com> another way would be to make the original condition have no actions then use another condition under that with an ip6 specific regex and use action and anti-action to differentiate On Fri, Feb 20, 2009 at 12:22 AM, Michael S Collins wrote: > > On Feb 19, 2009, at 7:53 PM, Jason White wrote: > > > I have it working now. The relevant changes were as follows. > > > > > > > > > > > > > > > > Jason, > > I like this approach. Good use of the many Dialplan tools. > > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090220/415adecb/attachment.html From brian at freeswitch.org Fri Feb 20 06:22:54 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 20 Feb 2009 08:22:54 -0600 Subject: [Freeswitch-users] Suggestion for xml_curl performance In-Reply-To: <22118122.post@talk.nabble.com> References: <22118122.post@talk.nabble.com> Message-ID: it all depends on what you're doing.. can you elaborate? /b On Feb 20, 2009, at 4:18 AM, shehzad p wrote: > Recently I faced some performance bottleneck by using Javascript. From ajlong at worldlink.net Fri Feb 20 06:31:52 2009 From: ajlong at worldlink.net (Adam Long) Date: Fri, 20 Feb 2009 09:31:52 -0500 Subject: [Freeswitch-users] SIP dump to DB In-Reply-To: <191c3a030902200554s4abda94g76ba4e2975fffa98@mail.gmail.com> References: <499E92E4.5010503@post.cz> <191c3a030902200554s4abda94g76ba4e2975fffa98@mail.gmail.com> Message-ID: <03a601c99367$f92f48f0$eb8ddad0$@net> MySQL MEMORY/HEAP table might be ideal for this. This data is prob not critical and is probably being used for diagnosing peer connectivity issues anyway. If it is critical well then... there are always trade offs right :) I think in general what he is speaking of is just some sort of temporary SIP trace setup that can log that can be controlled or filtered. So its not just all or nothing. I wonder is it possible to enable the current sip trace functionality via a variable. For example something like this. This of course only helps for bridged B2BUA calls. But if inbound sip tracing is required a user param like sip_trace could address that, yes? That could be a good starting point, then perhaps I could help roll a mod_xml_siptrace module based on the mod_xml_cdr design/concept that could somehow link into the existing sip trace logging. Just some thoughts, no idea how much of this exists today. Regards, -Adam From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Friday, February 20, 2009 8:55 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] SIP dump to DB what exact info do you need? That's likely to be a challenge with any database to store at that speed. On Fri, Feb 20, 2009 at 5:24 AM, kokoska rokoska wrote: Hi all, I'm facing the problem I need all SIP messages "going thru" FreeSWITCH (I know FS i B2BUA - so, better to say just "all SIP messages") logged somewhere and this log have to be "searchable" (by call-id etc) and I should be able to simply delete "old" messages... And more over - it should be done on not trivial SIP messages amount - say hundreds messages per second. My questions are: 1. Do you have any suggestion how to do it with FreeSWITCH? 2. Or it is not possible now, and "bounty" is necessary? :-) 3. How hard it will be to implement? ---------- FYI: I think something like SER/Kamailio/OpenSIPS siptrace is what I'am (probably) looking for: http://www.kamailio.org/docs/modules/1.4.x/siptrace.html It could go (based on my tests) up to 7-10.000 messages per second till MySQL "dies"... ---------- Thanks for your time :-) Best regards, kokoska.rokoska _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 From kokoska.rokoska at post.cz Fri Feb 20 06:39:04 2009 From: kokoska.rokoska at post.cz (kokoska rokoska) Date: Fri, 20 Feb 2009 15:39:04 +0100 Subject: [Freeswitch-users] SIP dump to DB In-Reply-To: <191c3a030902200554s4abda94g76ba4e2975fffa98@mail.gmail.com> References: <499E92E4.5010503@post.cz> <191c3a030902200554s4abda94g76ba4e2975fffa98@mail.gmail.com> Message-ID: <499EC088.8080900@post.cz> Anthony Minessale napsal(a): > what exact info do you need? That's likely to be a challenge with any > database to store at that speed. > Thank you very much, Anthony, for your reply! I should say: Personally I don't need it (I see preformance penalty), but few people around me need to store somewhere ALL sip messages going through the server. And ALL means "really all" (well, it will be very helpful if I can skip OPTIONS and other nat-keep-alive related messages). So I need something like "sipgrep dump" but I should be able to simply corelate messages to user (in case of MESSAGE, REGISTER etc.) and to user+call (in case of INVITE, BYE, CANCEL etc.). ------------- BTW: I'm sure it will be challenge for DB - and thus I made some tests with mentioned siptrace. One SIP call is about 18-21 SIP messages, so for 100 cps I should make a little bit over 2.000 INSERTs per second. 900 REGISTERs per second will generate about 5.500 INSERTs per second => I need to fire cca 8.000 INSERTs per second... ------------- But, may be, better solution exists - without DB. Any hint is very appreciated :-) Best regards, kokoska.rokoska > On Fri, Feb 20, 2009 at 5:24 AM, kokoska rokoska > > wrote: > > > Hi all, > > I'm facing the problem I need all SIP messages "going thru" FreeSWITCH > (I know FS i B2BUA - so, better to say just "all SIP messages") logged > somewhere and this log have to be "searchable" (by call-id etc) and I > should be able to simply delete "old" messages... > And more over - it should be done on not trivial SIP messages amount - > say hundreds messages per second. > > My questions are: > 1. Do you have any suggestion how to do it with FreeSWITCH? > 2. Or it is not possible now, and "bounty" is necessary? :-) > 3. How hard it will be to implement? > > ---------- > FYI: > I think something like SER/Kamailio/OpenSIPS siptrace is what I'am > (probably) looking for: > http://www.kamailio.org/docs/modules/1.4.x/siptrace.html > It could go (based on my tests) up to 7-10.000 messages per second till > MySQL "dies"... > ---------- > > Thanks for your time :-) > > Best regards, > > kokoska.rokoska > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From kokoska.rokoska at post.cz Fri Feb 20 06:51:06 2009 From: kokoska.rokoska at post.cz (kokoska rokoska) Date: Fri, 20 Feb 2009 15:51:06 +0100 Subject: [Freeswitch-users] SIP dump to DB In-Reply-To: <03a601c99367$f92f48f0$eb8ddad0$@net> References: <499E92E4.5010503@post.cz> <191c3a030902200554s4abda94g76ba4e2975fffa98@mail.gmail.com> <03a601c99367$f92f48f0$eb8ddad0$@net> Message-ID: <499EC35A.1060808@post.cz> Adam Long napsal(a): > MySQL MEMORY/HEAP table might be ideal for this. This data is prob not > critical and is probably > being used for diagnosing peer connectivity issues anyway. > If it is critical well then... there are always trade offs right :) > Thank you very much, Adam, for interest! You are right - it is for diagnosis. And while the data is not critical, I can't leave them in MySQL MEMORY table. Based on my counts I have to store and maintain about 70-90 GiB of SIP messages and I simply don't have enough RAM :-) > I think in general what he is speaking of is just some sort of temporary SIP > trace setup that can log > that can be controlled or filtered. So its not just all or nothing. > Exactly! :-) > I wonder is it possible to enable the current sip trace functionality via a > variable. > For example something like this. > > data="{sip_trace=on}sofia/public/XXXXXXXXXX at 10.10.10.1" /> > This of course only helps for bridged B2BUA calls. > It will be very helpful... > But if inbound sip tracing is required a user param like sip_trace could > address that, yes? > Yes, I need all SIP packets except nat-keep-alives... BTW: How to recognize them? I know how could I do it on proxy (SER like), but what about on FreeSWITCH? Anyway - for me it shouldn't be an issue, because I filter then on loadbalancer. > That could be a good starting point, then perhaps I could help roll a > mod_xml_siptrace module > based on the mod_xml_cdr design/concept that could somehow link into the > existing sip trace logging. > It will be very powerfull, but I'm affraid it can't scale to thousands request per second. > Just some thoughts, no idea how much of this exists today. > Thanks once more, Adam! Best regards, kokoska.rokoska From sicfslist at gmail.com Fri Feb 20 06:58:36 2009 From: sicfslist at gmail.com (Shelby Ramsey) Date: Fri, 20 Feb 2009 08:58:36 -0600 Subject: [Freeswitch-users] SIP dump to DB In-Reply-To: <499EC35A.1060808@post.cz> References: <499E92E4.5010503@post.cz> <191c3a030902200554s4abda94g76ba4e2975fffa98@mail.gmail.com> <03a601c99367$f92f48f0$eb8ddad0$@net> <499EC35A.1060808@post.cz> Message-ID: <35b355e90902200658k48ba850ele3ede371ebe96631@mail.gmail.com> Why not just use NGREP and then dump the packets at a more reasonable pace? You aren't going to be able to analysis in real time anyway. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090220/75b5878e/attachment.html From kokoska.rokoska at post.cz Fri Feb 20 07:27:46 2009 From: kokoska.rokoska at post.cz (kokoska rokoska) Date: Fri, 20 Feb 2009 16:27:46 +0100 Subject: [Freeswitch-users] SIP dump to DB In-Reply-To: <35b355e90902200658k48ba850ele3ede371ebe96631@mail.gmail.com> References: <499E92E4.5010503@post.cz> <191c3a030902200554s4abda94g76ba4e2975fffa98@mail.gmail.com> <03a601c99367$f92f48f0$eb8ddad0$@net> <499EC35A.1060808@post.cz> <35b355e90902200658k48ba850ele3ede371ebe96631@mail.gmail.com> Message-ID: <499ECBF2.4030606@post.cz> Shelby Ramsey napsal(a): > Why not just use NGREP and then dump the packets at a more reasonable > pace? Thank you very much, Shelby, for your interest! I can't use ngrep because the dump is not searchable IMO :-) See below, please. You aren't going to be able to analysis in real time anyway. > Some basic diagnosis should by doable realtime. For example: 1. find whole SIP trace for selected call (both a-leg, b-leg) 2. find all succeful user REGISTERs for given period ... And I have no idea how to do it on 50-100 GiB plain text file. Best regards, kokoska.rokoska From anthony.minessale at gmail.com Fri Feb 20 07:33:52 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 20 Feb 2009 09:33:52 -0600 Subject: [Freeswitch-users] SIP dump to DB In-Reply-To: <499ECBF2.4030606@post.cz> References: <499E92E4.5010503@post.cz> <191c3a030902200554s4abda94g76ba4e2975fffa98@mail.gmail.com> <03a601c99367$f92f48f0$eb8ddad0$@net> <499EC35A.1060808@post.cz> <35b355e90902200658k48ba850ele3ede371ebe96631@mail.gmail.com> <499ECBF2.4030606@post.cz> Message-ID: <191c3a030902200733y795626edq8eef6f06bcd70571@mail.gmail.com> you could try sippcapdump but i hear it needs work but it snoops the wire and tries to make individual files out of each call. On Fri, Feb 20, 2009 at 9:27 AM, kokoska rokoska wrote: > > > > Shelby Ramsey napsal(a): > > Why not just use NGREP and then dump the packets at a more reasonable > > pace? > > Thank you very much, Shelby, for your interest! > > I can't use ngrep because the dump is not searchable IMO :-) > See below, please. > > You aren't going to be able to analysis in real time anyway. > > > > Some basic diagnosis should by doable realtime. For example: > 1. find whole SIP trace for selected call (both a-leg, b-leg) > 2. find all succeful user REGISTERs for given period > ... > > And I have no idea how to do it on 50-100 GiB plain text file. > > Best regards, > > kokoska.rokoska > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090220/f8086de7/attachment.html From sicfslist at gmail.com Fri Feb 20 07:38:17 2009 From: sicfslist at gmail.com (Shelby Ramsey) Date: Fri, 20 Feb 2009 09:38:17 -0600 Subject: [Freeswitch-users] SIP dump to DB In-Reply-To: <499ECBF2.4030606@post.cz> References: <499E92E4.5010503@post.cz> <191c3a030902200554s4abda94g76ba4e2975fffa98@mail.gmail.com> <03a601c99367$f92f48f0$eb8ddad0$@net> <499EC35A.1060808@post.cz> <35b355e90902200658k48ba850ele3ede371ebe96631@mail.gmail.com> <499ECBF2.4030606@post.cz> Message-ID: <35b355e90902200738n3f21dfdn50ccc038326ae5f1@mail.gmail.com> Sorry .. I didn't give enough detail. My point was to dump it via NGREP ... parse it using something else to get it into a database where it would be usable. Then you can match calls from the CDR (using the UUID) to the database. The benefit is that you don't have to put the burden on your FS boxes to do it ... Just monitor from another device and then dump it into the database. Of course you better have a beast of database if you want to do 10,000 writes per second :) or be running something like NDB that scales well. There are other tools like scapy as well that can be quite useful in fact. Shelby -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090220/628b6fca/attachment.html From jaugenstine at gmail.com Fri Feb 20 07:50:22 2009 From: jaugenstine at gmail.com (jonathan augenstine) Date: Fri, 20 Feb 2009 07:50:22 -0800 Subject: [Freeswitch-users] SIP dump to DB In-Reply-To: <191c3a030902200733y795626edq8eef6f06bcd70571@mail.gmail.com> References: <499E92E4.5010503@post.cz> <191c3a030902200554s4abda94g76ba4e2975fffa98@mail.gmail.com> <03a601c99367$f92f48f0$eb8ddad0$@net> <499EC35A.1060808@post.cz> <35b355e90902200658k48ba850ele3ede371ebe96631@mail.gmail.com> <499ECBF2.4030606@post.cz> <191c3a030902200733y795626edq8eef6f06bcd70571@mail.gmail.com> Message-ID: <207e7a5e0902200750t2ac9f21avfb5ae88791e4efa8@mail.gmail.com> You can tcpdump and then use wireshark to graph the calls. When the dump is displayed in wireshark, select 'Statistics' -> VoIP Calls. You will see a display of all VoIP calls. Select the one you want graphed, or select them all and you will see REINVITE and REFER interaction as well as RTP streams. Jonathan On Fri, Feb 20, 2009 at 7:33 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > you could try sippcapdump but i hear it needs work but it snoops the wire > and tries to make individual files out of each call. > > > > On Fri, Feb 20, 2009 at 9:27 AM, kokoska rokoska wrote: > >> >> >> >> Shelby Ramsey napsal(a): >> > Why not just use NGREP and then dump the packets at a more reasonable >> > pace? >> >> Thank you very much, Shelby, for your interest! >> >> I can't use ngrep because the dump is not searchable IMO :-) >> See below, please. >> >> You aren't going to be able to analysis in real time anyway. >> > >> >> Some basic diagnosis should by doable realtime. For example: >> 1. find whole SIP trace for selected call (both a-leg, b-leg) >> 2. find all succeful user REGISTERs for given period >> ... >> >> And I have no idea how to do it on 50-100 GiB plain text file. >> >> Best regards, >> >> kokoska.rokoska >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090220/d0089afd/attachment.html From kokoska.rokoska at post.cz Fri Feb 20 07:57:15 2009 From: kokoska.rokoska at post.cz (kokoska rokoska) Date: Fri, 20 Feb 2009 16:57:15 +0100 Subject: [Freeswitch-users] SIP dump to DB In-Reply-To: <191c3a030902200733y795626edq8eef6f06bcd70571@mail.gmail.com> References: <499E92E4.5010503@post.cz> <191c3a030902200554s4abda94g76ba4e2975fffa98@mail.gmail.com> <03a601c99367$f92f48f0$eb8ddad0$@net> <499EC35A.1060808@post.cz> <35b355e90902200658k48ba850ele3ede371ebe96631@mail.gmail.com> <499ECBF2.4030606@post.cz> <191c3a030902200733y795626edq8eef6f06bcd70571@mail.gmail.com> Message-ID: <499ED2DB.90807@post.cz> Anthony Minessale napsal(a): > you could try sippcapdump but i hear it needs work but it snoops the > wire and tries to make individual files out of each call. > > Thank you very much, Anthony, for the suggestion! I will look at it. Best regards, kokoska.rokoska From kokoska.rokoska at post.cz Fri Feb 20 08:09:56 2009 From: kokoska.rokoska at post.cz (kokoska rokoska) Date: Fri, 20 Feb 2009 17:09:56 +0100 Subject: [Freeswitch-users] SIP dump to DB In-Reply-To: <35b355e90902200738n3f21dfdn50ccc038326ae5f1@mail.gmail.com> References: <499E92E4.5010503@post.cz> <191c3a030902200554s4abda94g76ba4e2975fffa98@mail.gmail.com> <03a601c99367$f92f48f0$eb8ddad0$@net> <499EC35A.1060808@post.cz> <35b355e90902200658k48ba850ele3ede371ebe96631@mail.gmail.com> <499ECBF2.4030606@post.cz> <35b355e90902200738n3f21dfdn50ccc038326ae5f1@mail.gmail.com> Message-ID: <499ED5D4.5010603@post.cz> Shelby Ramsey napsal(a): > Sorry .. I didn't give enough detail. My point was to dump it via NGREP > ... parse it using something else to get it into a database where it > would be usable. This is good point! Thank you very much, Shelby! > Then you can match calls from the CDR (using the UUID) > to the database. This is exactly what I try to accomplish :-) > The benefit is that you don't have to put the burden > on your FS boxes to do it ... Just monitor from another device and then > dump it into the database. I think of 2 possibilities: 1. On router replicate whole traffic to dedicated machine and process dump there (but it probably kills that machine, because it gets all RTP traffic) 2. Modify FreeSWITCH/Sofia (I have no idea how hard it will be, or if it is even possible) to duplicate all SIP messages to given URI - main benefit of this scenario is that I have only SIP messages on logging machine and that I can use SERlike proxy to parse messages and store them to DB. > Of course you better have a beast of > database if you want to do 10,000 writes per second :) or be running > something like NDB that scales well. > NDB is overkill for logging :-) What I think of is some kind of "caching" on FreeSWITCH side. I.e. store mesasges to DB in bigger chukns (whole call etc.) It will significatnly reduce DB utilization... > There are other tools like scapy as well that can be quite useful in fact. > I will look at it, thank you Shelby! Best regards, kokoska.roksoka From kokoska.rokoska at post.cz Fri Feb 20 08:15:51 2009 From: kokoska.rokoska at post.cz (kokoska rokoska) Date: Fri, 20 Feb 2009 17:15:51 +0100 Subject: [Freeswitch-users] SIP dump to DB In-Reply-To: <207e7a5e0902200750t2ac9f21avfb5ae88791e4efa8@mail.gmail.com> References: <499E92E4.5010503@post.cz> <191c3a030902200554s4abda94g76ba4e2975fffa98@mail.gmail.com> <03a601c99367$f92f48f0$eb8ddad0$@net> <499EC35A.1060808@post.cz> <35b355e90902200658k48ba850ele3ede371ebe96631@mail.gmail.com> <499ECBF2.4030606@post.cz> <191c3a030902200733y795626edq8eef6f06bcd70571@mail.gmail.com> <207e7a5e0902200750t2ac9f21avfb5ae88791e4efa8@mail.gmail.com> Message-ID: <499ED737.3020401@post.cz> jonathan augenstine napsal(a): > You can tcpdump and then use wireshark to graph the calls. When the > dump is displayed in wireshark, select 'Statistics' -> VoIP Calls. You > will see a display of all VoIP calls. Select the one you want graphed, > or select them all and you will see REINVITE and REFER interaction as > well as RTP streams. > Thank you very much, jonathan, for your interest! I use ngrep+wireshark many times a day, but I'm affraid it is not suitable for that amount of data. Even with few hundreds MiBs of pcap file wireshark becoms very slow and I can't imagine how to load 50-100 GiB file with milions of calls and try to search for one of them :-) And, even worse, I should "rotate" the file and, don't end with call divided to multiple files... Best regards, kokoska.rokoska From leon at scarlet-internet.nl Fri Feb 20 08:19:25 2009 From: leon at scarlet-internet.nl (Leon de Rooij) Date: Fri, 20 Feb 2009 17:19:25 +0100 Subject: [Freeswitch-users] mod_erlang_event compile problem In-Reply-To: <20090109230650.GF5210@hijacked.us> References: <20090109230650.GF5210@hijacked.us> Message-ID: <4A496EA6-D036-4BB5-9347-5536AD5FD0FC@scarlet-internet.nl> Hi, I wanted to try out the mod_erlang_event module. I have Erlang R12B5 compiled and it's in the same location as the Makefile specifies (/usr/ local/lib/erlang/...), but running make in the src/mod/event_handlers/ mod_erlang_event goes wrong: Compiling handle_msg.c... cc1: warnings being treated as errors handle_msg.c: In function 'handle_msg_sendmsg': handle_msg.c:429: warning: the address of 'uuid' will always evaluate as 'true' handle_msg.c: In function 'handle_msg_handlecall': handle_msg.c:541: warning: the address of 'uuid_str' will always evaluate as 'true' make[1]: *** [handle_msg.o] Error 1 make: *** [all] Error 1 At line 429 in handle_msg.c it says: if (!switch_strlen_zero(uuid) && (session = switch_core_session_locate(uuid))) { (at line 541 is the same problem) Is this a bug ? I tried removing the first part "! switch_strlen_zero(uuid) &&" after which it compiles fine, but since I don't fully understand what's going on, I'm sure this is not the solution.. Also, after this, FS goes haywire after loading the module and spews out these messages continuously: 2009-02-20 14:15:48 [ERR] mod_erlang_event.c:1417 mod_erlang_event_runtime() Failed to start empd manually 2009-02-20 14:15:48 [DEBUG] mod_erlang_event.c:1401 mod_erlang_event_runtime() Socket up listening on 127.0.0.1:8031 2009-02-20 14:15:48 [WARNING] mod_erlang_event.c:1415 mod_erlang_event_runtime() Failed to publish port to empd, trying to start empd manually 2009-02-20 14:15:48 [ERR] mod_erlang_event.c:1417 mod_erlang_event_runtime() Failed to start empd manually 2009-02-20 14:15:48 [DEBUG] mod_erlang_event.c:1401 mod_erlang_event_runtime() Socket up listening on 127.0.0.1:8031 2009-02-20 14:15:48 [WARNING] mod_erlang_event.c:1415 mod_erlang_event_runtime() Failed to publish port to empd, trying to start empd manually etc.. Can someone help me and point out what's wrong ? thanks & kind regards, Leon de Rooij From msc at freeswitch.org Fri Feb 20 08:36:29 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 20 Feb 2009 08:36:29 -0800 Subject: [Freeswitch-users] js session.streamFile() interrupt slowness In-Reply-To: <11990ade0902192127l414ecfeaia04fb80f626c35d5@mail.gmail.com> References: <11990ade0902192127l414ecfeaia04fb80f626c35d5@mail.gmail.com> Message-ID: <87f2f3b90902200836o638fa345qec10888f39e46414@mail.gmail.com> On Thu, Feb 19, 2009 at 9:27 PM, Stephen Crosby wrote: > I have a few scripts that use the javascript > session.streamFile('somefile.wav', onDtmf); where onDtmf is a function > that returns false to interrupt the streaming file. There is a short > delay between the time when I press a key and the time the file stops > playing. Is there anything I can adjust that would affect that? It's > only maybe 2-3 seconds, but it "feels" too long to me. > > --Stephen Could you pastebin your entire script plus the relevant dialplan entry? Also, could you tell us which operating system and FS revision? -MC From andrew at hijacked.us Fri Feb 20 11:08:11 2009 From: andrew at hijacked.us (Andrew Thompson) Date: Fri, 20 Feb 2009 14:08:11 -0500 Subject: [Freeswitch-users] mod_erlang_event compile problem In-Reply-To: <4A496EA6-D036-4BB5-9347-5536AD5FD0FC@scarlet-internet.nl> References: <20090109230650.GF5210@hijacked.us> <4A496EA6-D036-4BB5-9347-5536AD5FD0FC@scarlet-internet.nl> Message-ID: <20090220190811.GC29511@hijacked.us> On Fri, Feb 20, 2009 at 05:19:25PM +0100, Leon de Rooij wrote: > Hi, > > I wanted to try out the mod_erlang_event module. I have Erlang R12B5 > compiled and it's in the same location as the Makefile specifies (/usr/ > local/lib/erlang/...), but running make in the src/mod/event_handlers/ > mod_erlang_event goes wrong: > Yeah, this was a gcc4 thing, I've done most of my testing on gcc3 so it didn't show up for me. Thanks to MikeJ for the fix suggestion. > Also, after this, FS goes haywire after loading the module > and spews out these messages continuously: > You don't have the erlang port mapper daemon running (epmd). mod_erlang_event needs it to be running in order to be able to register itself as an erlang node. On your system; epmd isn't in $PATH so my system() call that tries to start it fails. I've made the module init system fail properly instead of looping indefinitely as well as print a slightly more helpful error message now. Let me know if you have any better luck :) The fix is in-tree as of r12192. Thanks again for the bug report. Andrew From andrew at hijacked.us Fri Feb 20 11:19:11 2009 From: andrew at hijacked.us (Andrew Thompson) Date: Fri, 20 Feb 2009 14:19:11 -0500 Subject: [Freeswitch-users] [ANN] Spice Telephony - an open source FreeSWITCH/Erlang callcenter platform In-Reply-To: <22015518.post@talk.nabble.com> References: <22015518.post@talk.nabble.com> Message-ID: <20090220191911.GD29511@hijacked.us> On Sat, Feb 14, 2009 at 03:04:01PM -0800, JCATS wrote: > > Have you planned any predictive dialer features ( like VICIDIAL )? > As Ken Rice mentioned, this isn't really the focus of the project - it's more for inbound and directed outbound (calling campaigns to specific people/businesses - not everyone in the phonebook). Primary focus is inbound (multi brand, skill based routing, dynamic wrapup times, etc). Expect a new release sometime soonish that actually does something useful (accepts and routes inbound calls from FreeSWITCH to an agent). Also; public source control. There's just some additional corporate nonsense that I have to sort out (again) before that can go live. Andrew From freeswitch-users at lists.rupa.com Fri Feb 20 14:08:40 2009 From: freeswitch-users at lists.rupa.com (Rupa Schomaker (lists)) Date: Fri, 20 Feb 2009 16:08:40 -0600 Subject: [Freeswitch-users] Passing Caller_ID to MOD_LCR In-Reply-To: <499AC59A.4050209@lists.rupa.com> References: <499A4497.9080001@lists.rupa.com> <499AC59A.4050209@lists.rupa.com> Message-ID: <499F29E8.4020100@lists.rupa.com> > "default" is a reserved profile name -- I should probably prevent that > from loading. correction, default is not reserved... > Regarding passing the callerid to the custom sql, let me see what I can > come up with... You can now specify channel variables in your custom sql. So, you should be able to pass CID or a subset of CID to your custom sql. I've updated the wiki with info on this. beware: channel vars only work when called in the context of a session. Using a profile that uses a custom sql with channel variables from the commandline will result in an error. -Rupa From leon at scarlet-internet.nl Fri Feb 20 15:28:08 2009 From: leon at scarlet-internet.nl (Leon de Rooij) Date: Sat, 21 Feb 2009 00:28:08 +0100 Subject: [Freeswitch-users] mod_erlang_event compile problem In-Reply-To: <20090220190811.GC29511@hijacked.us> References: <20090109230650.GF5210@hijacked.us> <4A496EA6-D036-4BB5-9347-5536AD5FD0FC@scarlet-internet.nl> <20090220190811.GC29511@hijacked.us> Message-ID: <8D43B57D-27ED-4517-BAD1-00EE526006A3@scarlet-internet.nl> Hi Andrew, Thanks! it compiles fine now.. Also thanks for the tip about empd, got it running without errors now :-) regards, Leon On Feb 20, 2009, at 8:08 PM, Andrew Thompson wrote: > On Fri, Feb 20, 2009 at 05:19:25PM +0100, Leon de Rooij wrote: >> Hi, >> >> I wanted to try out the mod_erlang_event module. I have Erlang R12B5 >> compiled and it's in the same location as the Makefile specifies (/ >> usr/ >> local/lib/erlang/...), but running make in the src/mod/ >> event_handlers/ >> mod_erlang_event goes wrong: >> > > Yeah, this was a gcc4 thing, I've done most of my testing on gcc3 so > it > didn't show up for me. Thanks to MikeJ for the fix suggestion. > >> Also, after this, FS goes haywire after loading the module >> and spews out these messages continuously: >> > > You don't have the erlang port mapper daemon running (epmd). > mod_erlang_event needs it to be running in order to be able to > register > itself as an erlang node. On your system; epmd isn't in $PATH so my > system() call that tries to start it fails. I've made the module init > system fail properly instead of looping indefinitely as well as > print a > slightly more helpful error message now. Let me know if you have any > better luck :) The fix is in-tree as of r12192. > > Thanks again for the bug report. > > Andrew > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From pmhshz at gmail.com Fri Feb 20 22:16:36 2009 From: pmhshz at gmail.com (shehzad p) Date: Fri, 20 Feb 2009 22:16:36 -0800 (PST) Subject: [Freeswitch-users] Suggestion for xml_curl performance In-Reply-To: References: <22118122.post@talk.nabble.com> Message-ID: <22133185.post@talk.nabble.com> Hi Brian, My setup is to use FS as basic calls routing. 1. Calls are coming to FS from more than one customer Gateways, and I need to authenticate them and check for enough balance based on database, [Caller Gateways] ===> [FreeSWITCH] ===> [Provider Gateways] 2. After knowing that Caller Gateways is valid, then based on dialed number it search in database for Provider Gateway and bridge the call there. 3. After call finish CDR is inserted back into database. My old setup was using Javascript which works fine in traffic of 10 to 20 calls, but then increase of traffic causes many problems. Now I eliminate use of any of the script (javascript or any other) for call routing, and route calls directly from dialplan, So I have setup test system using xml-curl to generate dynamic dialplan, I used below xml_curl PHP example as reference: http://wiki.freeswitch.org/wiki/Mod_xml_curl_PHP_example For CDR processing I used xml_cdr, with help of the example in FS source :scripts/contrib/trixter/xml-cdr. Waiting for any better suggestions, any comments... thanks msp. Brian West-3 wrote: > > it all depends on what you're doing.. can you elaborate? > > /b > > On Feb 20, 2009, at 4:18 AM, shehzad p wrote: > >> Recently I faced some performance bottleneck by using Javascript. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/Suggestion-for-xml_curl-performance-tp22118122p22133185.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From jmesquita at gmail.com Sat Feb 21 05:56:41 2009 From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Sat, 21 Feb 2009 11:56:41 -0200 Subject: [Freeswitch-users] Deployment information and use cases In-Reply-To: <499D796E.45B7.0079.0@averyschools.net> References: <1234916455.16581.49.camel@raul-laptop> <59ad9ca10902181009p7cbc0fd1s9860aacde7f3a30c@mail.gmail.com> <247f8100902181019m61ce803m5ed504d9b198dfb6@mail.gmail.com> <1235016793.22050.0.camel@raul-laptop> <499D796E.45B7.0079.0@averyschools.net> Message-ID: <7405656F-1AAA-4FF7-9DC3-4CE694D7B0AB@gmail.com> Ben, thank you for your story. I would very much like to add this to the wiki if you don't mind and everyone else agrees. What do you think guys? Use cases are _ALWAYS_ a good thing for new users. Mesquita On Feb 19, 2009, at 6:23 PM, Ben Holtsclaw wrote: > Raul, > > I am in the process of rolling out a FreeSWITCH IP PBX solution > similar to what you describe. When I was trying to procure funds for > a FreeSWITCH solution, I looked for the same information you're > after, but came up with little. I'll briefly describe what we're > trying to accomplish, and the tools I'm using to do it. This is > probably more information than what you are looking for, but maybe > it will also benefit someone else. > > We had several schools with aging or dying PBX's or KSU's. Each site > had something different system, and was supported by a different > VAR. Of course, the VAR's charged some outlandish fee to make onsite > repair visits. Some number of Centrex lines supplied each school's > dial tone. All in all, we had a very outdated and financially > draining mess. Our district's long term goal had been to move to a > more unified phone system. That made sense for many reasons, the > chief of which was cost. We already had a strong fiber WAN in place. > Why not use that for trunking and eliminate the monthly cost of the > Centrex lines? That's the path we started down. > > Being a public entity, we had to be sure to explore all possible > avenues. We looked at everything from traditional PBX's with IP add- > on modules for trunking to a full blown Cisco CallManager solution. > With third party proprietary systems, we were just never able to > find the sweet spot between required feature set and cost. Would > Cisco have been a workable solution? Absolutely. Could our small, > rural, K12 public school district afford that? Not in a million > years. I looked at several software packages -- some open source, > some not -- but always came back to FreeSWITCH. The scalability and > active development community were major factors for us. > > Server Hardware. Each of our five sites has a dedicated FreeSWITCH > server. For hardware, we went with Dell PowerEdge 1950's with dual > quad core Xeon 2.33 GHz processors, and 4GB of RAM. I have mirrored > disks set up with enough space to accommodate users' voicemail. Each > server will average only about 60 voicemail boxes, and we're storing > sound as MP3. Disk space shouldn't be an issue. We have always been > a Novell shop, so SLES is naturally our Linux distribution of > choice. We chose to go with server hardware at each site so that in > the event of a WAN outage, we would still at least have intra- > building and emergency communication (see below). > > Telephony Hardware. Each of our servers includes Sangoma hardware. > We actually looked at doing IP trunking to a carrier from our > network core, but decided to use telco provided PRI's instead. > Presently, we have two PRI's that connect to a FreeSWITCH server at > the center of our network via a Sangoma A102 dual port telephony > card. All calls to and from the PSTN traverse this primary server. > Servers at each remote site include one of Sangoma's A200 analog > cards. Emergency calls to 911 route out over this analog card > through one of at least two POTS lines that remain connected at each > site. Not only does this provide some redundancy in the event of a > WAN outage, but it ensures proper caller location is delivered to > the 911 dispatcher. Granted, there are some other solutions for the > latter, but this seemed to be the most cost effective solution for us. > > Telephone Desksets. We chose to go with Aastra for the telephones. > The standard phone that we will place in each classroom and office > is the 9143i. This is an attractive phone with an adequate feature > set at a price we can afford. The person that is primarily > responsible for answering the phone at each site will have an Aastra > 57i and some number of 560M expansion modules. We have purchased > roughly 300 Aastra desksets. > > Logical Layout. As new sites come online, their primary phone number > is being moved from the Centrex to our PRI group. All inbound calls > hit our primary server, and then FreeSWITCH bridges to the > appropriate secondary server based on the DID it received. On the > reverse, each servers dial plan is set up to route outbound calls > (save 911) to the primary server where FreeSWITCH bridges with > Openzap. Site to site calls, accomplished via four digit dialing, do > not hit the primary server. Outbound calls to the PSTN deliver the > site's DID as the calling number. In other words, if a user from > site two calls my cell phone, I see site two's published telephone > number on my caller ID. Our dial plans are set up so that > receptionists at each site still answer all outside calls. If not > answered, the call fails over to an IVR. Should we ever decide to do > so, we are now perfectly positioned to have all inbound calls to the > district answered by one operator or IVR. "Welcome, and thank you > for calling Avery County Schools." > > Stumbling Blocks. Problems we've faced so far have primarily > surrounded Openzap and the Sangoma Wanpipe driver. FreeSWITCH > developers won't mind telling you that this is an area that is > currently not well "funded" and not 100% complete. There is some > known issue where voice channels on the PRI get stuck in the wrong > state and become unusable. We have experienced this a couple of > times and have not been able to make or receive calls. Bouncing the > Wanpipe driver has fixed this each time. We have also had trouble > with DTMF detection across the PRI. If a user hits the IVR, it is > oftentimes difficult to get it to properly recognize the digits that > are being keyed in by the caller. This can be very, very frustrating > to a caller that doesn't want to deal with an IVR anyway. The > developers have suggested to me that this is a problem with the > Sangoma's echo cancellation goofing up Openzap's ability to > interpret the DTMF. The Sangoma hardware does have its own DTMF > decoder and API, but the Openzap code currently does not make use of > it. I have created a patch that makes use of the hardware decoder. > We have been running it in production for a couple of weeks, and > that does seem to have helped the problem. The problem hasn't gone > away altogether. Those have been our two biggest issues, but we > haven't let them hold us up. > > Conclusion. Of the five sites that will be on this system, one is > fully functional with calls inbound and outbound from the PSTN. A > second site is up and running with full outbound PSTN access. Their > inbound DID is scheduled to move over to the PRI in one week. The > server has been worked up for a third site, and the phones are > starting to roll out. Sites four and five should come online by the > end of April. Currently, I don't have numbers compiled for things > like concurrent calls. At this point in my project, it is just not > important. I really don't think our implementation will ever push > FreeSWITCH's abilities in that regard. I base that statement > primarily on other users' benchmarks, and what I've heard some are > doing in carrier class environments. > > FreeSWITCH has made our project viable. An open source solution was > the only way we could meet all of the project goals and stay within > our budget. FreeSWITCH has proven to have all the features we > require in a district wide phone system. It has not locked us into > annual support contracts with third party vendors. I could go on > with the accolades. However, I'll end this terribly lengthy post by > saying that, overall, we have been very pleased with our choice to > go with FreeSWITCH. > > The information in this email will seem very elementary to most > people on this list, but having a message of this nature in hand > would have made me feel much more confident the first time I ever > went to my supervisor to mention something called FreeSWITCH. :-) > Thanks Tony, Brian, and Mike for a great product! > > > Ben Holtsclaw > Network Engineer > Avery County Schools > Ph: 828.733.3567 x2301 > > >>> On 2/18/2009 at 11:13 PM, Raul Fragoso wrote: > Thanks guys, this is very useful information. > > Anyone else willing to share your experience ? > > Regards, > > Raul > > On Wed, 2009-02-18 at 16:19 -0200, Pablo Hernan Saro wrote: > > Hi Raul, > > > > In my company (http://www.globant.com) we're using FreeSWITCH for > high > > quality conference services, integrated with OpenSIPS > > (http://www.opensips.org) and Asterisk. Its performance is pretty > > good. > > > > Pablo > > > > On Wed, Feb 18, 2009 at 4:09 PM, Henry Huang > wrote: > > > bandwidth.com has a service called phonebooth which is developed > upon > > > freeswitch. > > > > > > > > > On Tue, Feb 17, 2009 at 4:20 PM, Raul Fragoso > wrote: > > >> > > >> Hello FreeSWITCHERS, > > >> > > >> My company is currently creating a suite of applications which > uses > > >> FreeSWITCH as the back-end for an IP-PBX solution. We currently > have a > > >> prospect to have our first customer installation - a governmental > > >> department. That is a tender to have an IP-PBX installation to > connect > > >> their four office branches, each one with about 300 users - > which I am > > >> sure FreeSWITCH is able to handle. Since this is an official > tender, > > >> it's part of their protocol to ask about real sites using the > product. > > >> > > >> Having said that, would you mind sharing some information about > your > > >> experience with FreeSWITCH deployments ? > > >> > > >> No need to give many details, but a short summary with company > name (if > > >> possible), when it was deployed, server equipment, number of > users, > > >> number of concurrent calls, what kind of functions and services > are used > > >> and overall capacity of the system. > > >> > > >> I would really appreciate if you can share that information. > And if you > > >> guys agree (and explicitly manifest your agreement), I can > compile the > > >> information in the FreeSWITCH wiki under a "Use Cases" page so > it can > > >> serve as a common reference as well. > > >> > > >> Kind regards, > > >> > > >> Raul Fragoso > > >> > > >> > > >> _______________________________________________ > > >> Freeswitch-users mailing list > > >> Freeswitch-users at lists.freeswitch.org > > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > >> http://www.freeswitch.org > > > > > > > > > > > > -- > > > Henry Huang > > > UniC Solution - Communication Unified > > > VoIP & Open Source software Consultant > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090221/3cdd410f/attachment-0001.html From leon at scarlet-internet.nl Sat Feb 21 06:42:24 2009 From: leon at scarlet-internet.nl (Leon de Rooij) Date: Sat, 21 Feb 2009 15:42:24 +0100 Subject: [Freeswitch-users] mod_erlang_event compile problem In-Reply-To: <20090220190811.GC29511@hijacked.us> References: <20090109230650.GF5210@hijacked.us> <4A496EA6-D036-4BB5-9347-5536AD5FD0FC@scarlet-internet.nl> <20090220190811.GC29511@hijacked.us> Message-ID: <90DDD253-BA8D-4A79-9ACA-1560F2E18688@scarlet-internet.nl> Hi Andrew, Thanks for your help so far, I hope you can help me a bit further as I don't get any reply from the FS erlang node, or so it seems.. Here is what I've done: - The erlang_event.conf.xml is unchanged: - mod_erlang_event is not loaded in FS. - First I start "epmd -d -d" epmd: Sat Feb 21 13:12:56 2009: epmd running - daemon = 0 epmd: Sat Feb 21 13:12:56 2009: try to initiate listening port 4369 epmd: Sat Feb 21 13:12:56 2009: starting epmd: Sat Feb 21 13:12:56 2009: entering the main select() loop - After that I "load mod_erlang_event" in FS: 2009-02-21 13:13:36 [DEBUG] mod_erlang_event.c:1324 mod_erlang_event_load() sections 16 2009-02-21 13:13:36 [CONSOLE] switch_loadable_module.c:858 switch_loadable_module_load_file() Successfully Loaded [mod_erlang_event] 2009-02-21 13:13:36 [NOTICE] switch_loadable_module.c:240 switch_loadable_module_process() Adding Application 'erlang' 2009-02-21 13:13:36 [NOTICE] switch_loadable_module.c:260 switch_loadable_module_process() Adding API Function 'erlang' 2009-02-21 13:13:36 [DEBUG] mod_erlang_event.c:1401 mod_erlang_event_runtime() Socket up listening on 127.0.0.1:8031 2009-02-21 13:13:36 [DEBUG] mod_erlang_event.c:1426 mod_erlang_event_runtime() Connected and published erlang cnode at freeswitch at erlyfs - For which epmd gives the following output: epmd: Sat Feb 21 13:13:36 2009: opening connection on file descriptor 4 epmd: Sat Feb 21 13:13:36 2009: got 25 bytes ***** 00000000 00 17 78 1f 5f 68 00 00 05 00 01 00 0a 66 72 65 |..x._h.......fre| ***** 00000010 65 73 77 69 74 63 68 00 00 | eswitch..| epmd: Sat Feb 21 13:13:36 2009: ** got ALIVE2_REQ epmd: Sat Feb 21 13:13:36 2009: registering 'freeswitch:1', port 8031 epmd: Sat Feb 21 13:13:36 2009: type 104 proto 0 highvsn 5 lowvsn 1 epmd: Sat Feb 21 13:13:36 2009: got 4 bytes ***** 00000000 79 00 00 01 |y...| epmd: Sat Feb 21 13:13:36 2009: ** sent ALIVE2_RESP for "freeswitch" - Then I start an erl shell on that same machine with "erl -sname ldr - setcookie ClueCon". Output of epmd: epmd: Sat Feb 21 13:16:24 2009: opening connection on file descriptor 5 epmd: Sat Feb 21 13:16:24 2009: got 18 bytes ***** 00000000 00 10 78 8e 2c 4d 00 00 05 00 05 00 03 6c 64 72 |..x.,M.......ldr| ***** 00000010 00 00 |..| epmd: Sat Feb 21 13:16:24 2009: ** got ALIVE2_REQ epmd: Sat Feb 21 13:16:24 2009: registering 'ldr:1', port 36396 epmd: Sat Feb 21 13:16:24 2009: type 77 proto 0 highvsn 5 lowvsn 5 epmd: Sat Feb 21 13:16:24 2009: got 4 bytes ***** 00000000 79 00 00 01 |y...| epmd: Sat Feb 21 13:16:24 2009: ** sent ALIVE2_RESP for "ldr" As far as I understand the freeswitch at erlyfs node cannot be seen with nodes() ? So does that mean that I also cannot net_adm:ping() it ? Anyway, I tried sending some tuples as is shown on the wiki, but I get no reply: (ldr at erlyfs)1> {foo, freeswitch at erlyfs} ! {api, status, ""}, receive X -> X after 1000 -> timeout end. timeout (ldr at erlyfs)2> - Epmd gives some logs: epmd: Sat Feb 21 13:19:09 2009: opening connection on file descriptor 6 epmd: Sat Feb 21 13:19:09 2009: got 13 bytes ***** 00000000 00 0b 7a 66 72 65 65 73 77 69 74 63 68 |..zfreeswitch| epmd: Sat Feb 21 13:19:09 2009: ** got PORT2_REQ epmd: Sat Feb 21 13:19:09 2009: got 23 bytes ***** 00000000 77 00 1f 5f 68 00 00 05 00 01 00 0a 66 72 65 65 | w.._h.......free| ***** 00000010 73 77 69 74 63 68 00 | switch.| epmd: Sat Feb 21 13:19:09 2009: ** sent PORT2_RESP (ok) for "freeswitch" epmd: Sat Feb 21 13:19:09 2009: closing connection on file descriptor 6 - And in tcpdump on lo, I see that epmd is contacted after which some traffic was sent to FS: 13:19:09.535293 IP 172.31.0.13.34678 > 172.31.0.13.4369: S 2875169966:2875169966(0) win 32792 ... 13:19:09.536834 IP 172.31.0.13.4369 > 172.31.0.13.34678: . ack 15 win 512 13:19:09.536923 IP 172.31.0.13.47054 > 172.31.0.13.8031: S 2868322908:2868322908(0) win 32792 13:19:09.536935 IP 172.31.0.13.8031 > 172.31.0.13.47054: R 0:0(0) ack 2868322909 win 0 Shouldn't FS then send a message back to the process in my erl shell ? I tried logging all events in fs_cli, by entering "/event plain all", but I see no events at all coming from erlang, just some heartbeats.. Also, I recompiled the module with EI_DEBUG defined as suggested on the wiki. Still I don't see anything in the CLI when set to debug logging. Thanks again, Leon On Feb 20, 2009, at 8:08 PM, Andrew Thompson wrote: > On Fri, Feb 20, 2009 at 05:19:25PM +0100, Leon de Rooij wrote: >> Hi, >> >> I wanted to try out the mod_erlang_event module. I have Erlang R12B5 >> compiled and it's in the same location as the Makefile specifies (/ >> usr/ >> local/lib/erlang/...), but running make in the src/mod/ >> event_handlers/ >> mod_erlang_event goes wrong: >> > > Yeah, this was a gcc4 thing, I've done most of my testing on gcc3 so > it > didn't show up for me. Thanks to MikeJ for the fix suggestion. > >> Also, after this, FS goes haywire after loading the module >> and spews out these messages continuously: >> > > You don't have the erlang port mapper daemon running (epmd). > mod_erlang_event needs it to be running in order to be able to > register > itself as an erlang node. On your system; epmd isn't in $PATH so my > system() call that tries to start it fails. I've made the module init > system fail properly instead of looping indefinitely as well as > print a > slightly more helpful error message now. Let me know if you have any > better luck :) The fix is in-tree as of r12192. > > Thanks again for the bug report. > > Andrew > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From shannon at sacredhearts.us Sat Feb 21 07:11:00 2009 From: shannon at sacredhearts.us (Shannon) Date: Sat, 21 Feb 2009 09:11:00 -0600 Subject: [Freeswitch-users] Suggestion for xml_curl performance In-Reply-To: <22133185.post@talk.nabble.com> References: <22118122.post@talk.nabble.com> <22133185.post@talk.nabble.com> Message-ID: <7e2ac3270902210711u3ea54b79j5868aef6b59d4800@mail.gmail.com> I'd recommend having a look at fastcgi as well. On 2/21/09, shehzad p wrote: > > Hi Brian, > > My setup is to use FS as basic calls routing. > 1. Calls are coming to FS from more than one customer Gateways, and I need > to authenticate them and check for enough balance based on database, > [Caller Gateways] ===> [FreeSWITCH] ===> > [Provider Gateways] > 2. After knowing that Caller Gateways is valid, then based on dialed number > it search in database for Provider Gateway and bridge the call there. > 3. After call finish CDR is inserted back into database. > > My old setup was using Javascript which works fine in traffic of 10 to 20 > calls, but then increase of traffic causes many problems. > > Now I eliminate use of any of the script (javascript or any other) for call > routing, and route calls directly from dialplan, > So I have setup test system using xml-curl to generate dynamic dialplan, > I used below xml_curl PHP example as reference: > http://wiki.freeswitch.org/wiki/Mod_xml_curl_PHP_example > For CDR processing I used xml_cdr, with help of the example in FS source > :scripts/contrib/trixter/xml-cdr. > > > Waiting for any better suggestions, any comments... > > thanks > msp. > > Brian West-3 wrote: >> >> it all depends on what you're doing.. can you elaborate? >> >> /b >> >> On Feb 20, 2009, at 4:18 AM, shehzad p wrote: >> >>> Recently I faced some performance bottleneck by using Javascript. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > View this message in context: > http://www.nabble.com/Suggestion-for-xml_curl-performance-tp22118122p22133185.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Shannon From freeswitch at servercorps.com Sat Feb 21 07:57:08 2009 From: freeswitch at servercorps.com (Addison Martin) Date: Sat, 21 Feb 2009 09:57:08 -0600 Subject: [Freeswitch-users] Suggestion for xml_curl performance In-Reply-To: <7e2ac3270902210711u3ea54b79j5868aef6b59d4800@mail.gmail.com> References: <22118122.post@talk.nabble.com> <22133185.post@talk.nabble.com> <7e2ac3270902210711u3ea54b79j5868aef6b59d4800@mail.gmail.com> Message-ID: <92e7d2090902210757u5ec034bew53353b7838c01b78@mail.gmail.com> Is this a good application for the new ESL (Event Socket Library)interface? -anm On Sat, Feb 21, 2009 at 9:11 AM, Shannon wrote: > I'd recommend having a look at fastcgi as well. > > On 2/21/09, shehzad p wrote: >> >> Hi Brian, >> >> My setup is to use FS as basic calls routing. >> 1. Calls are coming to FS from more than one customer Gateways, and I need >> to authenticate them and check for enough balance based on database, >> [Caller Gateways] ===> [FreeSWITCH] ===> >> [Provider Gateways] >> 2. After knowing that Caller Gateways is valid, then based on dialed number >> it search in database for Provider Gateway and bridge the call there. >> 3. After call finish CDR is inserted back into database. >> >> My old setup was using Javascript which works fine in traffic of 10 to 20 >> calls, but then increase of traffic causes many problems. >> >> Now I eliminate use of any of the script (javascript or any other) for call >> routing, and route calls directly from dialplan, >> So I have setup test system using xml-curl to generate dynamic dialplan, >> I used below xml_curl PHP example as reference: >> http://wiki.freeswitch.org/wiki/Mod_xml_curl_PHP_example >> For CDR processing I used xml_cdr, with help of the example in FS source >> :scripts/contrib/trixter/xml-cdr. >> >> >> Waiting for any better suggestions, any comments... >> >> thanks >> msp. >> >> Brian West-3 wrote: >>> >>> it all depends on what you're doing.. can you elaborate? >>> >>> /b >>> >>> On Feb 20, 2009, at 4:18 AM, shehzad p wrote: >>> >>>> Recently I faced some performance bottleneck by using Javascript. >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> -- >> View this message in context: >> http://www.nabble.com/Suggestion-for-xml_curl-performance-tp22118122p22133185.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > -- > Shannon > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From freeswitch at servercorps.com Sat Feb 21 07:59:49 2009 From: freeswitch at servercorps.com (Addison Martin) Date: Sat, 21 Feb 2009 09:59:49 -0600 Subject: [Freeswitch-users] ESL In-Reply-To: References: Message-ID: <92e7d2090902210759v6f934421y9873a3cb526dd245@mail.gmail.com> I have ported all the Perl samples to python, and they appear to be working fine. They are available in svn rev 12210 and > -anm On Thu, Feb 19, 2009 at 12:44 PM, Brian West wrote: > FreeSWITCHers, > Not sure anyone is paying attention or not but Anthony wrapped the > ESL library up so you can use it from Perl, Python, Lua, Ruby and > PHP. What I'm requesting from our community is to help flex it out.. > write examples and populate the Wiki page with information about it. > > http://wiki.freeswitch.org/wiki/Esl > > Collins and I are going to start filling in the page but I want > someone thats good with Ruby, Python, PHP to help in those areas.. > kick in some lua and perl if you like. > > It works with OES and IES... (Outbound Event Socket and Inbound Event > Socket) Not sure those names are official but we have been calling > them that ;) > > Thanks, > Brian West > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From freeswitch at servercorps.com Sat Feb 21 09:00:39 2009 From: freeswitch at servercorps.com (Addison Martin) Date: Sat, 21 Feb 2009 11:00:39 -0600 Subject: [Freeswitch-users] Deployment information and use cases In-Reply-To: <1234916455.16581.49.camel@raul-laptop> References: <1234916455.16581.49.camel@raul-laptop> Message-ID: <92e7d2090902210900u7da015c3n7a815ab4beaf3b00@mail.gmail.com> We're still in the construction and design phase, but my company is building a multi-tenant Freeswitch based PBX for a Research Park in South Alabama. We expect to handle about 120 concurrent calls, and 6-700 registered UAs. The system will be based on commodity house-built SuperMicro servers, with mod_xml_curl handling all configuration. We will have PRIs for fax and 911, and SIP trunks to upstream ITSPs for most call volume. -anm On Tue, Feb 17, 2009 at 6:20 PM, Raul Fragoso wrote: > Hello FreeSWITCHERS, > > My company is currently creating a suite of applications which uses > FreeSWITCH as the back-end for an IP-PBX solution. We currently have a > prospect to have our first customer installation - a governmental > department. That is a tender to have an IP-PBX installation to connect > their four office branches, each one with about 300 users - which I am > sure FreeSWITCH is able to handle. Since this is an official tender, > it's part of their protocol to ask about real sites using the product. > > Having said that, would you mind sharing some information about your > experience with FreeSWITCH deployments ? > > No need to give many details, but a short summary with company name (if > possible), when it was deployed, server equipment, number of users, > number of concurrent calls, what kind of functions and services are used > and overall capacity of the system. > > I would really appreciate if you can share that information. And if you > guys agree (and explicitly manifest your agreement), I can compile the > information in the FreeSWITCH wiki under a "Use Cases" page so it can > serve as a common reference as well. > > Kind regards, > > Raul Fragoso > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From stevecrozz at gmail.com Sat Feb 21 11:31:19 2009 From: stevecrozz at gmail.com (Stephen Crosby) Date: Sat, 21 Feb 2009 11:31:19 -0800 Subject: [Freeswitch-users] js session.streamFile() interrupt slowness In-Reply-To: <87f2f3b90902200836o638fa345qec10888f39e46414@mail.gmail.com> References: <11990ade0902192127l414ecfeaia04fb80f626c35d5@mail.gmail.com> <87f2f3b90902200836o638fa345qec10888f39e46414@mail.gmail.com> Message-ID: <11990ade0902211131i134b892q71f18626b7ae54b5@mail.gmail.com> Sure, I've stripped down the script somewhat to something smaller that still produces this effect and you can see it at: http://pastebin.freeswitch.org/7388 The file sound 'VR1' continues to play for a short time after I interrupt it with a DTMF event. It does interrupt, but it sounds a little awkward because of the delay. I was probably wrong in my estimate of the delay which seems to be about a full second, not two or three. I'm hoping I can adjust it somehow to feel more immediate. Any ideas? --Stephen On Fri, Feb 20, 2009 at 8:36 AM, Michael Collins wrote: > On Thu, Feb 19, 2009 at 9:27 PM, Stephen Crosby wrote: >> I have a few scripts that use the javascript >> session.streamFile('somefile.wav', onDtmf); where onDtmf is a function >> that returns false to interrupt the streaming file. There is a short >> delay between the time when I press a key and the time the file stops >> playing. Is there anything I can adjust that would affect that? It's >> only maybe 2-3 seconds, but it "feels" too long to me. >> >> --Stephen > > Could you pastebin your entire script plus the relevant dialplan > entry? Also, could you tell us which operating system and FS revision? > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From stevecrozz at gmail.com Sat Feb 21 11:33:11 2009 From: stevecrozz at gmail.com (Stephen Crosby) Date: Sat, 21 Feb 2009 11:33:11 -0800 Subject: [Freeswitch-users] js session.streamFile() interrupt slowness In-Reply-To: <11990ade0902211131i134b892q71f18626b7ae54b5@mail.gmail.com> References: <11990ade0902192127l414ecfeaia04fb80f626c35d5@mail.gmail.com> <87f2f3b90902200836o638fa345qec10888f39e46414@mail.gmail.com> <11990ade0902211131i134b892q71f18626b7ae54b5@mail.gmail.com> Message-ID: <11990ade0902211133n5fba7a27k4ae9b2e90978abf1@mail.gmail.com> There was a small error in that last script I sent, please test using this version: http://pastebin.freeswitch.org/7388 Thanks. On Sat, Feb 21, 2009 at 11:31 AM, Stephen Crosby wrote: > Sure, I've stripped down the script somewhat to something smaller that > still produces this effect and you can see it at: > http://pastebin.freeswitch.org/7388 > > The file sound 'VR1' continues to play for a short time after I > interrupt it with a DTMF event. It does interrupt, but it sounds a > little awkward because of the delay. I was probably wrong in my > estimate of the delay which seems to be about a full second, not two > or three. I'm hoping I can adjust it somehow to feel more immediate. > Any ideas? > > --Stephen > > On Fri, Feb 20, 2009 at 8:36 AM, Michael Collins wrote: >> On Thu, Feb 19, 2009 at 9:27 PM, Stephen Crosby wrote: >>> I have a few scripts that use the javascript >>> session.streamFile('somefile.wav', onDtmf); where onDtmf is a function >>> that returns false to interrupt the streaming file. There is a short >>> delay between the time when I press a key and the time the file stops >>> playing. Is there anything I can adjust that would affect that? It's >>> only maybe 2-3 seconds, but it "feels" too long to me. >>> >>> --Stephen >> >> Could you pastebin your entire script plus the relevant dialplan >> entry? Also, could you tell us which operating system and FS revision? >> -MC >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > From stevecrozz at gmail.com Sat Feb 21 11:35:07 2009 From: stevecrozz at gmail.com (Stephen Crosby) Date: Sat, 21 Feb 2009 11:35:07 -0800 Subject: [Freeswitch-users] js session.streamFile() interrupt slowness In-Reply-To: <11990ade0902211133n5fba7a27k4ae9b2e90978abf1@mail.gmail.com> References: <11990ade0902192127l414ecfeaia04fb80f626c35d5@mail.gmail.com> <87f2f3b90902200836o638fa345qec10888f39e46414@mail.gmail.com> <11990ade0902211131i134b892q71f18626b7ae54b5@mail.gmail.com> <11990ade0902211133n5fba7a27k4ae9b2e90978abf1@mail.gmail.com> Message-ID: <11990ade0902211135o7180f06fj3d9c1fa6cdbb59b8@mail.gmail.com> Verry sorry for the list spam, this is the link to the corrected script: http://pastebin.freeswitch.org/7389 --Stephen On Sat, Feb 21, 2009 at 11:33 AM, Stephen Crosby wrote: > There was a small error in that last script I sent, please test using > this version: http://pastebin.freeswitch.org/7388 > > Thanks. > > On Sat, Feb 21, 2009 at 11:31 AM, Stephen Crosby wrote: >> Sure, I've stripped down the script somewhat to something smaller that >> still produces this effect and you can see it at: >> http://pastebin.freeswitch.org/7388 >> >> The file sound 'VR1' continues to play for a short time after I >> interrupt it with a DTMF event. It does interrupt, but it sounds a >> little awkward because of the delay. I was probably wrong in my >> estimate of the delay which seems to be about a full second, not two >> or three. I'm hoping I can adjust it somehow to feel more immediate. >> Any ideas? >> >> --Stephen >> >> On Fri, Feb 20, 2009 at 8:36 AM, Michael Collins wrote: >>> On Thu, Feb 19, 2009 at 9:27 PM, Stephen Crosby wrote: >>>> I have a few scripts that use the javascript >>>> session.streamFile('somefile.wav', onDtmf); where onDtmf is a function >>>> that returns false to interrupt the streaming file. There is a short >>>> delay between the time when I press a key and the time the file stops >>>> playing. Is there anything I can adjust that would affect that? It's >>>> only maybe 2-3 seconds, but it "feels" too long to me. >>>> >>>> --Stephen >>> >>> Could you pastebin your entire script plus the relevant dialplan >>> entry? Also, could you tell us which operating system and FS revision? >>> -MC >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> > From chavpaskov at shaw.ca Sat Feb 21 11:40:20 2009 From: chavpaskov at shaw.ca (Chav Paskov) Date: Sat, 21 Feb 2009 11:40:20 -0800 Subject: [Freeswitch-users] Passing Caller_ID to MOD_LCR In-Reply-To: <499F29E8.4020100@lists.rupa.com> References: <499A4497.9080001@lists.rupa.com> <499AC59A.4050209@lists.rupa.com> <499F29E8.4020100@lists.rupa.com> Message-ID: <49A058A4.7000609@shaw.ca> Rupa Schomaker (lists) wrote: >> "default" is a reserved profile name -- I should probably prevent that >> from loading. >> > > correction, default is not reserved... > > >> Regarding passing the callerid to the custom sql, let me see what I can >> come up with... >> > > You can now specify channel variables in your custom sql. So, you > should be able to pass CID or a subset of CID to your custom sql. I've > updated the wiki with info on this. > > beware: channel vars only work when called in the context of a session. > Using a profile that uses a custom sql with channel variables from the > commandline will result in an error. > > -Rupa > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > do i have to upgrade to the latest trunk in order to pass channel variables to mod_lcr? Currently the version used is 1.0.trunk (12134M) Regards Chav From freeswitch-users at lists.rupa.com Sat Feb 21 13:46:02 2009 From: freeswitch-users at lists.rupa.com (Rupa Schomaker (lists)) Date: Sat, 21 Feb 2009 15:46:02 -0600 Subject: [Freeswitch-users] Passing Caller_ID to MOD_LCR In-Reply-To: <49A058A4.7000609@shaw.ca> References: <499A4497.9080001@lists.rupa.com> <499AC59A.4050209@lists.rupa.com> <499F29E8.4020100@lists.rupa.com> <49A058A4.7000609@shaw.ca> Message-ID: <49A0761A.8090101@lists.rupa.com> > do i have to upgrade to the latest trunk in order to pass channel > variables to mod_lcr? > Currently the version used is 1.0.trunk (12134M) > Regards > Chav > You need at least 12204. From chavpaskov at shaw.ca Sat Feb 21 14:53:27 2009 From: chavpaskov at shaw.ca (Chav Paskov) Date: Sat, 21 Feb 2009 14:53:27 -0800 Subject: [Freeswitch-users] Passing Caller_ID to MOD_LCR In-Reply-To: <49A0761A.8090101@lists.rupa.com> References: <499A4497.9080001@lists.rupa.com> <499AC59A.4050209@lists.rupa.com> <499F29E8.4020100@lists.rupa.com> <49A058A4.7000609@shaw.ca> <49A0761A.8090101@lists.rupa.com> Message-ID: <49A085E7.3000506@shaw.ca> Rupa Schomaker (lists) wrote: >> do i have to upgrade to the latest trunk in order to pass channel >> variables to mod_lcr? >> Currently the version used is 1.0.trunk (12134M) >> Regards >> Chav >> >> > > You need at least 12204. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > Thank you very much. I'll try it and will keep you postged. Chav From anthony.minessale at gmail.com Sat Feb 21 15:29:50 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sat, 21 Feb 2009 17:29:50 -0600 Subject: [Freeswitch-users] js session.streamFile() interrupt slowness In-Reply-To: <11990ade0902211135o7180f06fj3d9c1fa6cdbb59b8@mail.gmail.com> References: <11990ade0902192127l414ecfeaia04fb80f626c35d5@mail.gmail.com> <87f2f3b90902200836o638fa345qec10888f39e46414@mail.gmail.com> <11990ade0902211131i134b892q71f18626b7ae54b5@mail.gmail.com> <11990ade0902211133n5fba7a27k4ae9b2e90978abf1@mail.gmail.com> <11990ade0902211135o7180f06fj3d9c1fa6cdbb59b8@mail.gmail.com> Message-ID: <191c3a030902211529j2637494dqc4c476a461e8cc5f@mail.gmail.com> Try this one http://pastebin.freeswitch.org/7391 I just tested this on latest trunk and it stopped instantly. On Sat, Feb 21, 2009 at 1:35 PM, Stephen Crosby wrote: > Verry sorry for the list spam, this is the link to the corrected script: > http://pastebin.freeswitch.org/7389 > > --Stephen > > On Sat, Feb 21, 2009 at 11:33 AM, Stephen Crosby > wrote: > > There was a small error in that last script I sent, please test using > > this version: http://pastebin.freeswitch.org/7388 > > > > Thanks. > > > > On Sat, Feb 21, 2009 at 11:31 AM, Stephen Crosby > wrote: > >> Sure, I've stripped down the script somewhat to something smaller that > >> still produces this effect and you can see it at: > >> http://pastebin.freeswitch.org/7388 > >> > >> The file sound 'VR1' continues to play for a short time after I > >> interrupt it with a DTMF event. It does interrupt, but it sounds a > >> little awkward because of the delay. I was probably wrong in my > >> estimate of the delay which seems to be about a full second, not two > >> or three. I'm hoping I can adjust it somehow to feel more immediate. > >> Any ideas? > >> > >> --Stephen > >> > >> On Fri, Feb 20, 2009 at 8:36 AM, Michael Collins > wrote: > >>> On Thu, Feb 19, 2009 at 9:27 PM, Stephen Crosby > wrote: > >>>> I have a few scripts that use the javascript > >>>> session.streamFile('somefile.wav', onDtmf); where onDtmf is a function > >>>> that returns false to interrupt the streaming file. There is a short > >>>> delay between the time when I press a key and the time the file stops > >>>> playing. Is there anything I can adjust that would affect that? It's > >>>> only maybe 2-3 seconds, but it "feels" too long to me. > >>>> > >>>> --Stephen > >>> > >>> Could you pastebin your entire script plus the relevant dialplan > >>> entry? Also, could you tell us which operating system and FS revision? > >>> -MC > >>> > >>> _______________________________________________ > >>> Freeswitch-users mailing list > >>> Freeswitch-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> http://www.freeswitch.org > >>> > >> > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090221/8aa4340b/attachment.html From chavpaskov at shaw.ca Sat Feb 21 15:37:26 2009 From: chavpaskov at shaw.ca (Chav Paskov) Date: Sat, 21 Feb 2009 15:37:26 -0800 Subject: [Freeswitch-users] recommended settings for max-proceeding param Message-ID: <49A09036.2080905@shaw.ca> Hi Everybody, if it is not too much of a trouble can somebody point to a recommended value for max-proceeding in sofia.conf.xml ? If there is no recommended value what should be taken under consideration in order to determine one. I dug into the archives and discovered a thread called "Freeswitch freezes under increased call load" and there together with session per sec and max allowed sessions was recommended max-proceeding under sofia.conf.xml to be changed. I've just installed 1.0.3 version and checked the sofia.conf.xml file . I was not able to find a default setting for max-proceeding , when i added it i started getting cored umps. Regards Chav From alex at sinapticode.ro Sun Feb 22 07:30:37 2009 From: alex at sinapticode.ro (Alexandru Nedelcu) Date: Sun, 22 Feb 2009 17:30:37 +0200 Subject: [Freeswitch-users] origination_caller_id_number used instead of effective_caller_id_number In-Reply-To: <2CEAC5BC-D5F8-4253-A710-C5E60DAC7EB8@jerris.com> References: <1235039768.4537.17.camel@gathern.lan> <1235041389.4537.22.camel@gathern.lan> <1235053185.4537.59.camel@gathern.lan> <2CEAC5BC-D5F8-4253-A710-C5E60DAC7EB8@jerris.com> Message-ID: Yeap, it's fixed. On Thu, Feb 19, 2009 at 5:03 PM, Michael Jerris wrote: > Can you re-test this with current svn trunk. I believe this was fixed > yesterday. > > Mike > -- Alexandru Nedelcu Software Developer, Sinapticode From sicfslist at gmail.com Sun Feb 22 10:30:00 2009 From: sicfslist at gmail.com (Shelby Ramsey) Date: Sun, 22 Feb 2009 12:30:00 -0600 Subject: [Freeswitch-users] Compile Errors ... Message-ID: <35b355e90902221030n589cf95bxc33f5a57ab921046@mail.gmail.com> Hello, I'm getting this all over the place today: make[5]: *** No rule to make target `/usr/src/freeswitch/libfreeswitch.la', needed by `mod_commands.so'. Stop. make[4]: *** [all] Error 1 make[3]: *** [mod_commands-all] Error 1 make[2]: *** [all-recursive] Error 1 I normally do this: svn checkout http://svn.freeswitch.org/svn/freeswitch/trunk freeswitch.trunk ./bootstrap.sh ./configure make make install Thx! SDR -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090222/bfda4658/attachment-0001.html From mike at jerris.com Sun Feb 22 10:58:12 2009 From: mike at jerris.com (Michael Jerris) Date: Sun, 22 Feb 2009 13:58:12 -0500 Subject: [Freeswitch-users] Compile Errors ... In-Reply-To: <35b355e90902221030n589cf95bxc33f5a57ab921046@mail.gmail.com> References: <35b355e90902221030n589cf95bxc33f5a57ab921046@mail.gmail.com> Message-ID: <53236046-2332-440E-9B1C-DA693464836B@jerris.com> Look up furthur, there will be an error around where it builds or links the core. Try typing make core. Mike On Feb 22, 2009, at 1:30 PM, Shelby Ramsey wrote: > Hello, > > I'm getting this all over the place today: > > make[5]: *** No rule to make target `/usr/src/freeswitch/ > libfreeswitch.la', needed by `mod_commands.so'. Stop. > make[4]: *** [all] Error 1 > make[3]: *** [mod_commands-all] Error 1 > make[2]: *** [all-recursive] Error 1 > > I normally do this: > > svn checkout http://svn.freeswitch.org/svn/freeswitch/trunk > freeswitch.trunk > ./bootstrap.sh > ./configure > make > make install > > Thx! > > SDR > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090222/949901e3/attachment.html From sicfslist at gmail.com Sun Feb 22 11:26:21 2009 From: sicfslist at gmail.com (Shelby Ramsey) Date: Sun, 22 Feb 2009 13:26:21 -0600 Subject: [Freeswitch-users] Compile Errors ... In-Reply-To: <53236046-2332-440E-9B1C-DA693464836B@jerris.com> References: <35b355e90902221030n589cf95bxc33f5a57ab921046@mail.gmail.com> <53236046-2332-440E-9B1C-DA693464836B@jerris.com> Message-ID: <35b355e90902221126r4ac44a00k690499546c04ae7b@mail.gmail.com> Mike, This is what I get when I run make core (after I did a checkout on latest svn trunk, ./bootstrap.sh, ./configure): src/switch_console.c:35:28: error: switch_version.h: No such file or directory src/switch_console.c: In function 'switch_console_process': src/switch_console.c:233: error: 'SWITCH_VERSION_FULL' undeclared (first use in this function) src/switch_console.c:233: error: (Each undeclared identifier is reported only once src/switch_console.c:233: error: for each function it appears in.) make[1]: *** [libfreeswitch_la-switch_console.lo] Error 1 make: *** [core] Error 2 Thanks! SDR -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090222/f19c1c28/attachment.html From mike at jerris.com Sun Feb 22 11:55:28 2009 From: mike at jerris.com (Michael Jerris) Date: Sun, 22 Feb 2009 14:55:28 -0500 Subject: [Freeswitch-users] Compile Errors ... In-Reply-To: <35b355e90902221126r4ac44a00k690499546c04ae7b@mail.gmail.com> References: <35b355e90902221030n589cf95bxc33f5a57ab921046@mail.gmail.com> <53236046-2332-440E-9B1C-DA693464836B@jerris.com> <35b355e90902221126r4ac44a00k690499546c04ae7b@mail.gmail.com> Message-ID: There should be more errors above that? Are you cutting off some of them in your paste? On Feb 22, 2009, at 2:26 PM, Shelby Ramsey wrote: > Mike, > > This is what I get when I run make core (after I did a checkout on > latest svn trunk, ./bootstrap.sh, ./configure): > > src/switch_console.c:35:28: error: switch_version.h: No such file or > directory > src/switch_console.c: In function 'switch_console_process': > src/switch_console.c:233: error: 'SWITCH_VERSION_FULL' undeclared > (first use in this function) > src/switch_console.c:233: error: (Each undeclared identifier is > reported only once > src/switch_console.c:233: error: for each function it appears in.) > make[1]: *** [libfreeswitch_la-switch_console.lo] Error 1 > make: *** [core] Error 2 > > Thanks! > > SDR > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From sicfslist at gmail.com Sun Feb 22 12:09:45 2009 From: sicfslist at gmail.com (Shelby Ramsey) Date: Sun, 22 Feb 2009 14:09:45 -0600 Subject: [Freeswitch-users] Compile Errors ... In-Reply-To: References: <35b355e90902221030n589cf95bxc33f5a57ab921046@mail.gmail.com> <53236046-2332-440E-9B1C-DA693464836B@jerris.com> <35b355e90902221126r4ac44a00k690499546c04ae7b@mail.gmail.com> Message-ID: <35b355e90902221209t329e6819oae727031ce78a502@mail.gmail.com> Mike, the entire output can be seen @ http://www.sipinterchange.com/downloads/fs_compile_err.txt I don't see anything else. It's really odd ... SDR -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090222/4e49b439/attachment.html From gkuri at ieee.org Sun Feb 22 13:01:41 2009 From: gkuri at ieee.org (Gabriel Kuri) Date: Sun, 22 Feb 2009 13:01:41 -0800 Subject: [Freeswitch-users] Compile Errors ... In-Reply-To: <35b355e90902221209t329e6819oae727031ce78a502@mail.gmail.com> References: <35b355e90902221030n589cf95bxc33f5a57ab921046@mail.gmail.com> <53236046-2332-440E-9B1C-DA693464836B@jerris.com> <35b355e90902221126r4ac44a00k690499546c04ae7b@mail.gmail.com> <35b355e90902221209t329e6819oae727031ce78a502@mail.gmail.com> Message-ID: <49A1BD35.4000308@ieee.org> It sounds like your build environment is whacked, is this a fresh checkout of trunk or did you overwrite an existing directory? You might want to scrap that directory and try a fresh checkout. I just freshly checked out a copy of trunk into a new dir and ran ./bootstrap, ./configure, and make without a problem. Gabe Shelby Ramsey wrote: > Mike, > > the entire output can be seen @ > http://www.sipinterchange.com/downloads/fs_compile_err.txt > > I don't see anything else. It's really odd ... > > SDR > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From sicfslist at gmail.com Sun Feb 22 13:13:01 2009 From: sicfslist at gmail.com (Shelby Ramsey) Date: Sun, 22 Feb 2009 15:13:01 -0600 Subject: [Freeswitch-users] Compile Errors ... In-Reply-To: <49A1BD35.4000308@ieee.org> References: <35b355e90902221030n589cf95bxc33f5a57ab921046@mail.gmail.com> <53236046-2332-440E-9B1C-DA693464836B@jerris.com> <35b355e90902221126r4ac44a00k690499546c04ae7b@mail.gmail.com> <35b355e90902221209t329e6819oae727031ce78a502@mail.gmail.com> <49A1BD35.4000308@ieee.org> Message-ID: <35b355e90902221313u6af1ad75kd751050457b51ad3@mail.gmail.com> Yeah ... that's what I did when I first got the error. I'll try it again. Thanks for the help! SDR -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090222/feb8f6f7/attachment.html From sicfslist at gmail.com Sun Feb 22 14:52:43 2009 From: sicfslist at gmail.com (Shelby Ramsey) Date: Sun, 22 Feb 2009 16:52:43 -0600 Subject: [Freeswitch-users] Compile Errors ... In-Reply-To: <35b355e90902221313u6af1ad75kd751050457b51ad3@mail.gmail.com> References: <35b355e90902221030n589cf95bxc33f5a57ab921046@mail.gmail.com> <53236046-2332-440E-9B1C-DA693464836B@jerris.com> <35b355e90902221126r4ac44a00k690499546c04ae7b@mail.gmail.com> <35b355e90902221209t329e6819oae727031ce78a502@mail.gmail.com> <49A1BD35.4000308@ieee.org> <35b355e90902221313u6af1ad75kd751050457b51ad3@mail.gmail.com> Message-ID: <35b355e90902221452g3a9f705bnfb76f4c3095949ab@mail.gmail.com> Did this (to make sure I started from scratch): rm -rf /usr/src/freeswitch.trunk rm -rf /usr/local/freeswitch svn checkout http://svn.freeswitch.org/svn/freeswitch/trunk freeswitch.trunk cd freeswitch.trunk/ ./bootstrap.sh ./configure make ... and then it pukes all over the place with errors compiling odbc support ... but I thought that odbc was disabled by default. Errors look like this: Compiling src/switch_odbc.c ... In file included from src/switch_odbc.c:33: ./src/include/switch_odbc.h:36:17: error: sql.h: No such file or directory ./src/include/switch_odbc.h:43:20: error: sqlext.h: No such file or directory ./src/include/switch_odbc.h:45:22: error: sqltypes.h: No such file or directory In file included from src/switch_odbc.c:33: ./src/include/switch_odbc.h:66: error: expected declaration specifiers or '...' before 'SQLHSTMT' ./src/include/switch_odbc.h:96: error: expected declaration specifiers or '...' before 'SQLHSTMT' src/switch_odbc.c:43: error: expected specifier-qualifier-list before 'SQLHENV' src/switch_odbc.c: In function 'switch_odbc_handle_new': src/switch_odbc.c:76: error: 'switch_odbc_handle_t' has no member named 'env' src/switch_odbc.c:76: error: 'SQL_NULL_HANDLE' undeclared (first use in this function) src/switch_odbc.c:76: error: (Each undeclared identifier is reported only once src/switch_odbc.c:76: error: for each function it appears in.) src/switch_odbc.c:77: error: 'switch_odbc_handle_t' has no member named 'state' src/switch_odbc.c: In function 'switch_odbc_handle_disconnect': src/switch_odbc.c:96: error: 'switch_odbc_handle_t' has no member named 'state' cc1: warnings being treated as errors src/switch_odbc.c:97: warning: implicit declaration of function 'SQLDisconnect' src/switch_odbc.c:97: error: 'switch_odbc_handle_t' has no member named 'con' src/switch_odbc.c:105: error: 'switch_odbc_handle_t' has no member named 'state' src/switch_odbc.c: In function 'switch_odbc_handle_connect': src/switch_odbc.c:113: error: 'SQLINTEGER' undeclared (first use in this function) src/switch_odbc.c:113: error: expected ';' before 'err' src/switch_odbc.c:116: error: 'SQLSMALLINT' undeclared (first use in this function) src/switch_odbc.c:116: error: expected ';' before 'valueLength' SDR -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090222/13e59eb6/attachment.html From gkuri at ieee.org Sun Feb 22 15:12:49 2009 From: gkuri at ieee.org (Gabriel Kuri) Date: Sun, 22 Feb 2009 15:12:49 -0800 Subject: [Freeswitch-users] Compile Errors ... In-Reply-To: <35b355e90902221452g3a9f705bnfb76f4c3095949ab@mail.gmail.com> References: <35b355e90902221030n589cf95bxc33f5a57ab921046@mail.gmail.com> <53236046-2332-440E-9B1C-DA693464836B@jerris.com> <35b355e90902221126r4ac44a00k690499546c04ae7b@mail.gmail.com> <35b355e90902221209t329e6819oae727031ce78a502@mail.gmail.com> <49A1BD35.4000308@ieee.org> <35b355e90902221313u6af1ad75kd751050457b51ad3@mail.gmail.com> <35b355e90902221452g3a9f705bnfb76f4c3095949ab@mail.gmail.com> Message-ID: <49A1DBF1.8000702@ieee.org> Someone can problem correct me if I'm wrong, however I believe a recent change was made to the configure script to try and autodetect ODBC. The configure script may be hitting a false positive as described in this similar thread from a few days ago ... http://lists.freeswitch.org/pipermail/freeswitch-users/2009-February/011762.html Gabe Shelby Ramsey wrote: > Did this (to make sure I started from scratch): > > rm -rf /usr/src/freeswitch.trunk > rm -rf /usr/local/freeswitch > svn checkout http://svn.freeswitch.org/svn/freeswitch/trunk freeswitch.trunk > cd freeswitch.trunk/ > ./bootstrap.sh > ./configure > make > > ... and then it pukes all over the place with errors compiling odbc > support ... but I thought that odbc was disabled by default. > > Errors look like this: > > Compiling src/switch_odbc.c ... > In file included from src/switch_odbc.c:33: > ./src/include/switch_odbc.h:36:17: error: sql.h: No such file or directory > ./src/include/switch_odbc.h:43:20: error: sqlext.h: No such file or > directory > ./src/include/switch_odbc.h:45:22: error: sqltypes.h: No such file or > directory > In file included from src/switch_odbc.c:33: > ./src/include/switch_odbc.h:66: error: expected declaration specifiers > or '...' before 'SQLHSTMT' > ./src/include/switch_odbc.h:96: error: expected declaration specifiers > or '...' before 'SQLHSTMT' > src/switch_odbc.c:43: error: expected specifier-qualifier-list before > 'SQLHENV' > src/switch_odbc.c: In function 'switch_odbc_handle_new': > src/switch_odbc.c:76: error: 'switch_odbc_handle_t' has no member named > 'env' > src/switch_odbc.c:76: error: 'SQL_NULL_HANDLE' undeclared (first use in > this function) > src/switch_odbc.c:76: error: (Each undeclared identifier is reported > only once > src/switch_odbc.c:76: error: for each function it appears in.) > src/switch_odbc.c:77: error: 'switch_odbc_handle_t' has no member named > 'state' > src/switch_odbc.c: In function 'switch_odbc_handle_disconnect': > src/switch_odbc.c:96: error: 'switch_odbc_handle_t' has no member named > 'state' > cc1: warnings being treated as errors > src/switch_odbc.c:97: warning: implicit declaration of function > 'SQLDisconnect' > src/switch_odbc.c:97: error: 'switch_odbc_handle_t' has no member named > 'con' > src/switch_odbc.c:105: error: 'switch_odbc_handle_t' has no member named > 'state' > src/switch_odbc.c: In function 'switch_odbc_handle_connect': > src/switch_odbc.c:113: error: 'SQLINTEGER' undeclared (first use in this > function) > src/switch_odbc.c:113: error: expected ';' before 'err' > src/switch_odbc.c:116: error: 'SQLSMALLINT' undeclared (first use in > this function) > src/switch_odbc.c:116: error: expected ';' before 'valueLength' > > > SDR > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From andrew at hijacked.us Sun Feb 22 15:32:16 2009 From: andrew at hijacked.us (Andrew Thompson) Date: Sun, 22 Feb 2009 18:32:16 -0500 Subject: [Freeswitch-users] mod_erlang_event compile problem In-Reply-To: <90DDD253-BA8D-4A79-9ACA-1560F2E18688@scarlet-internet.nl> References: <20090109230650.GF5210@hijacked.us> <4A496EA6-D036-4BB5-9347-5536AD5FD0FC@scarlet-internet.nl> <20090220190811.GC29511@hijacked.us> <90DDD253-BA8D-4A79-9ACA-1560F2E18688@scarlet-internet.nl> Message-ID: <20090222233215.GB13957@hijacked.us> On Sat, Feb 21, 2009 at 03:42:24PM +0100, Leon de Rooij wrote: > Hi Andrew, > > Thanks for your help so far, I hope you can help me a bit further as I > don't get any reply from the FS erlang node, or so it seems.. Here is > what I've done: > > - The erlang_event.conf.xml is unchanged: > > > > > > > > > > > You actually installed this to the right place? It's not installed by default... The defaults *should* be sane anyway, but I'm just checking. > > - mod_erlang_event is not loaded in FS. > > - First I start "epmd -d -d" > > epmd: Sat Feb 21 13:12:56 2009: epmd running - daemon = 0 > epmd: Sat Feb 21 13:12:56 2009: try to initiate listening port 4369 > epmd: Sat Feb 21 13:12:56 2009: starting > epmd: Sat Feb 21 13:12:56 2009: entering the main select() loop > > - After that I "load mod_erlang_event" in FS: > > 2009-02-21 13:13:36 [DEBUG] mod_erlang_event.c:1324 > mod_erlang_event_load() sections 16 > 2009-02-21 13:13:36 [CONSOLE] switch_loadable_module.c:858 > switch_loadable_module_load_file() Successfully Loaded > [mod_erlang_event] > 2009-02-21 13:13:36 [NOTICE] switch_loadable_module.c:240 > switch_loadable_module_process() Adding Application 'erlang' > 2009-02-21 13:13:36 [NOTICE] switch_loadable_module.c:260 > switch_loadable_module_process() Adding API Function 'erlang' > 2009-02-21 13:13:36 [DEBUG] mod_erlang_event.c:1401 > mod_erlang_event_runtime() Socket up listening on 127.0.0.1:8031 > 2009-02-21 13:13:36 [DEBUG] mod_erlang_event.c:1426 > mod_erlang_event_runtime() Connected and published erlang cnode at > freeswitch at erlyfs > > - For which epmd gives the following output: > > epmd: Sat Feb 21 13:13:36 2009: opening connection on file descriptor 4 > epmd: Sat Feb 21 13:13:36 2009: got 25 bytes > ***** 00000000 00 17 78 1f 5f 68 00 00 05 00 01 00 0a 66 72 65 > |..x._h.......fre| > ***** 00000010 65 73 77 69 74 63 68 00 00 | > eswitch..| > epmd: Sat Feb 21 13:13:36 2009: ** got ALIVE2_REQ > epmd: Sat Feb 21 13:13:36 2009: registering 'freeswitch:1', port 8031 > epmd: Sat Feb 21 13:13:36 2009: type 104 proto 0 highvsn 5 lowvsn 1 > epmd: Sat Feb 21 13:13:36 2009: got 4 bytes > ***** 00000000 79 00 00 01 |y...| > epmd: Sat Feb 21 13:13:36 2009: ** sent ALIVE2_RESP for "freeswitch" > > - Then I start an erl shell on that same machine with "erl -sname ldr - > setcookie ClueCon". Output of epmd: > > epmd: Sat Feb 21 13:16:24 2009: opening connection on file descriptor 5 > epmd: Sat Feb 21 13:16:24 2009: got 18 bytes > ***** 00000000 00 10 78 8e 2c 4d 00 00 05 00 05 00 03 6c 64 72 > |..x.,M.......ldr| > ***** 00000010 00 00 |..| > epmd: Sat Feb 21 13:16:24 2009: ** got ALIVE2_REQ > epmd: Sat Feb 21 13:16:24 2009: registering 'ldr:1', port 36396 > epmd: Sat Feb 21 13:16:24 2009: type 77 proto 0 highvsn 5 lowvsn 5 > epmd: Sat Feb 21 13:16:24 2009: got 4 bytes > ***** 00000000 79 00 00 01 |y...| > epmd: Sat Feb 21 13:16:24 2009: ** sent ALIVE2_RESP for "ldr" > > As far as I understand the freeswitch at erlyfs node cannot be seen with > nodes() ? So does that mean that I also cannot net_adm:ping() it ? Yes, it's a 'hidden' node, as all non-erlang nodes are. However, it should be visible in the output of epmd -names. > > Anyway, I tried sending some tuples as is shown on the wiki, but I get > no reply: > > (ldr at erlyfs)1> {foo, freeswitch at erlyfs} ! {api, status, ""}, receive X > -> X after 1000 -> timeout end. > timeout > (ldr at erlyfs)2> > > - Epmd gives some logs: > > epmd: Sat Feb 21 13:19:09 2009: opening connection on file descriptor 6 > epmd: Sat Feb 21 13:19:09 2009: got 13 bytes > ***** 00000000 00 0b 7a 66 72 65 65 73 77 69 74 63 68 > |..zfreeswitch| > epmd: Sat Feb 21 13:19:09 2009: ** got PORT2_REQ > epmd: Sat Feb 21 13:19:09 2009: got 23 bytes > ***** 00000000 77 00 1f 5f 68 00 00 05 00 01 00 0a 66 72 65 65 | > w.._h.......free| > ***** 00000010 73 77 69 74 63 68 00 | > switch.| > epmd: Sat Feb 21 13:19:09 2009: ** sent PORT2_RESP (ok) for "freeswitch" > epmd: Sat Feb 21 13:19:09 2009: closing connection on file descriptor 6 > > - And in tcpdump on lo, I see that epmd is contacted after which some > traffic was sent to FS: > > 13:19:09.535293 IP 172.31.0.13.34678 > 172.31.0.13.4369: S > 2875169966:2875169966(0) win 32792 17946545 0,nop,wscale 6> > ... > 13:19:09.536834 IP 172.31.0.13.4369 > 172.31.0.13.34678: . ack 15 win > 512 > 13:19:09.536923 IP 172.31.0.13.47054 > 172.31.0.13.8031: S > 2868322908:2868322908(0) win 32792 17946546 0,nop,wscale 6> > 13:19:09.536935 IP 172.31.0.13.8031 > 172.31.0.13.47054: R 0:0(0) ack > 2868322909 win 0 > > Shouldn't FS then send a message back to the process in my erl shell ? > > I tried logging all events in fs_cli, by entering "/event plain all", > but I see no events at all coming from erlang, just some heartbeats.. > > Also, I recompiled the module with EI_DEBUG defined as suggested on > the wiki. Still I don't see anything in the CLI when set to debug > logging. This is the part that's confusing me, you should be seeing *something*, especially with EI_DEBUG on, because in that case you see *everything* that is sent to or received by the erlang module. Let me to a 'make current' and doublecheck something didn't get broken along the way. Andrew From andrew at hijacked.us Sun Feb 22 16:12:12 2009 From: andrew at hijacked.us (Andrew Thompson) Date: Sun, 22 Feb 2009 19:12:12 -0500 Subject: [Freeswitch-users] mod_erlang_event compile problem In-Reply-To: <90DDD253-BA8D-4A79-9ACA-1560F2E18688@scarlet-internet.nl> References: <20090109230650.GF5210@hijacked.us> <4A496EA6-D036-4BB5-9347-5536AD5FD0FC@scarlet-internet.nl> <20090220190811.GC29511@hijacked.us> <90DDD253-BA8D-4A79-9ACA-1560F2E18688@scarlet-internet.nl> Message-ID: <20090223001211.GC13957@hijacked.us> Leon, I can't replicate your issue, at the very least I'd expect you to see the "Ignorable error in ei_accept - probable bad client version, bad cookie or bad nodename" warning. What OS/Erlang version are you using? Andrew From hochlehnert at hotmail.com Sun Feb 22 16:04:35 2009 From: hochlehnert at hotmail.com (Klaus Hochlehnert) Date: Mon, 23 Feb 2009 01:04:35 +0100 Subject: [Freeswitch-users] Question about BLF... Message-ID: Hi, I'm just playing around with FreeSWITCH and I have 2 questions about BLF (with SNOM phones): - When I played around with the sample dial plan I found out that BLF works better than Asterisk, but not 100% right: > When phone 1000 gets a call the BLF lamp on phone 1001 blinks and after phone 1000 takes the call the lamp on phone 1001 is on > But when phone 1000 gets a second call, takes it and hangs up the lamp on phone 1001 turns off even if the first call is still active > Is that a problem or did I do something wrong??? - Second question is how can I set up BLF if I want to have my dial plan completely in a perl script (no XML besides calling the perl script)? Thanks, Klaus From sicfslist at gmail.com Sun Feb 22 16:46:04 2009 From: sicfslist at gmail.com (Shelby Ramsey) Date: Sun, 22 Feb 2009 18:46:04 -0600 Subject: [Freeswitch-users] Compile Errors ... In-Reply-To: <49A1DBF1.8000702@ieee.org> References: <35b355e90902221030n589cf95bxc33f5a57ab921046@mail.gmail.com> <53236046-2332-440E-9B1C-DA693464836B@jerris.com> <35b355e90902221126r4ac44a00k690499546c04ae7b@mail.gmail.com> <35b355e90902221209t329e6819oae727031ce78a502@mail.gmail.com> <49A1BD35.4000308@ieee.org> <35b355e90902221313u6af1ad75kd751050457b51ad3@mail.gmail.com> <35b355e90902221452g3a9f705bnfb76f4c3095949ab@mail.gmail.com> <49A1DBF1.8000702@ieee.org> Message-ID: <35b355e90902221646m5f1cd40u9037d7755eea8f48@mail.gmail.com> Thanks for the help. That did the trick. SDR -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090222/0add7f9c/attachment.html From mike at jerris.com Sun Feb 22 20:45:47 2009 From: mike at jerris.com (Michael Jerris) Date: Sun, 22 Feb 2009 23:45:47 -0500 Subject: [Freeswitch-users] Compile Errors ... In-Reply-To: <35b355e90902221646m5f1cd40u9037d7755eea8f48@mail.gmail.com> References: <35b355e90902221030n589cf95bxc33f5a57ab921046@mail.gmail.com> <53236046-2332-440E-9B1C-DA693464836B@jerris.com> <35b355e90902221126r4ac44a00k690499546c04ae7b@mail.gmail.com> <35b355e90902221209t329e6819oae727031ce78a502@mail.gmail.com> <49A1BD35.4000308@ieee.org> <35b355e90902221313u6af1ad75kd751050457b51ad3@mail.gmail.com> <35b355e90902221452g3a9f705bnfb76f4c3095949ab@mail.gmail.com> <49A1DBF1.8000702@ieee.org> <35b355e90902221646m5f1cd40u9037d7755eea8f48@mail.gmail.com> Message-ID: <10314107-A6F5-4E16-B8F4-CF604B4A0F45@jerris.com> can you please contact me off list and get me information to access your box so I can try to correct this for good in tree. Thanks Mike On Feb 22, 2009, at 7:46 PM, Shelby Ramsey wrote: > Thanks for the help. That did the trick. > > SDR From leon at scarlet-internet.nl Mon Feb 23 02:09:28 2009 From: leon at scarlet-internet.nl (Leon de Rooij) Date: Mon, 23 Feb 2009 11:09:28 +0100 Subject: [Freeswitch-users] mod_erlang_event compile problem In-Reply-To: <20090223001211.GC13957@hijacked.us> References: <20090109230650.GF5210@hijacked.us> <4A496EA6-D036-4BB5-9347-5536AD5FD0FC@scarlet-internet.nl> <20090220190811.GC29511@hijacked.us> <90DDD253-BA8D-4A79-9ACA-1560F2E18688@scarlet-internet.nl> <20090223001211.GC13957@hijacked.us> Message-ID: <0DEC394E-4739-4055-B524-E4217593DD3C@scarlet-internet.nl> Hi Andrew, Everything is running on an Ubuntu Hardy Xen domu with kernel 2.6.24-23-xen. Erlang is version R12B5 and was compiled from source with options -- enable-hipe, --enable-smp-support en --enable-threads. FS is trunk version 12197. I did copy the configuration file to ~freeswitch/conf/autoload_configs Also, I just checked the 'empd -names', after both FS and an erl shell have been started: root at erlyfs:~# epmd -names epmd: up and running on port 4369 with data: name ldr at port 57114 name freeswitch at port 8031 So that should be fine.. I also tried loading mod_erlang_event from modules.conf, and starting FS as root, but - not surprisingly - that didn't make any difference. I've been looking in wireshark, what exactly is going over the line, and the strange thing is, that erl opens a TCP connection, a SYN packet is sent to FS, after which FS immediately returns an RST/ACK packet and thus closes the connection.. I still don't see anything in the FS CLI. Is there anything I can do to get more verbose output from FS - esp info about why the connection was closed ? thanks, Leon On Feb 23, 2009, at 1:12 AM, Andrew Thompson wrote: > Leon, > > I can't replicate your issue, at the very least I'd expect you to see > the "Ignorable error in ei_accept - probable bad client version, bad > cookie or bad nodename" warning. What OS/Erlang version are you using? > > Andrew > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From Claudio.Cavalera at italtel.it Mon Feb 23 03:09:26 2009 From: Claudio.Cavalera at italtel.it (Cavalera Claudio Luigi) Date: Mon, 23 Feb 2009 12:09:26 +0100 Subject: [Freeswitch-users] Random problems with cepstral text to speech In-Reply-To: Message-ID: freeswitch-users-bounces at lists.freeswitch.org wrote: > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Anthony Minessale > >> Are you using cepstral 5.1? >> There is a known issue with that release and it's closed > source so we > cannot do much about it. >> Cepstral 4.x works fine. > > Yes 5.1, my fault. > I have added an initial warning here on the wiki > http://wiki.freeswitch.org/wiki/Mod_cepstral > although it also speaks about 5.1 and Ubuntu... Hello, if Cepstral 4.x is the way to go does anybody know where to get the demo version? BRs, Claudio Internet Email Confidentiality Footer ----------------------------------------------------------------------------------------------------- La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. ----------------------------------------------------------------------------------------------------- From pekkis50 at gmail.com Mon Feb 23 04:28:29 2009 From: pekkis50 at gmail.com (Pekka Kurki) Date: Mon, 23 Feb 2009 13:28:29 +0100 Subject: [Freeswitch-users] undefined symbol: krb5_auth_con_getrcache** Message-ID: <49A2966D.7010004@gmail.com> got this error when starting freeswitch -latest svn 2009-02-23 13:22:50 [CRIT] switch_loadable_module.c:840 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_sofia.so **/usr/local/freeswitch/mod/mod_sofia.so: undefined symbol: krb5_auth_con_getrcache** no ccompile errors, krb5libs installed, config with and w/o libcurl. br /pekka From kerrada2003 at yahoo.com Mon Feb 23 06:42:12 2009 From: kerrada2003 at yahoo.com (Ali Al-Rubaie) Date: Mon, 23 Feb 2009 06:42:12 -0800 (PST) Subject: [Freeswitch-users] Realm Value In-Reply-To: Message-ID: <711428.49972.qm@web33708.mail.mud.yahoo.com> I could compile and install FS 1.0.2 successsfully so, do I need to install ODBC-devel package for 1.0.3 version? Thanks, Message: 5 Date: Thu, 19 Feb 2009 14:35:29 -0800 From: Michael Collins Subject: Re: [Freeswitch-users] Realm Value To: freeswitch-users at lists.freeswitch.org Message-ID: <87f2f3b90902191435p1c9c03aend3303dfb013495b1 at mail.gmail.com> Content-Type: text/plain; charset=ISO-8859-1 On Thu, Feb 19, 2009 at 12:17 PM, Raymond Chandler wrote: > did you ./configure --enable-core-odbc-suport... those errors reek of > that flag with no unixODBC-devel package installed > > -Ray > Anthony described this as a false positive on detecting ODBC. If you are in Linux you can install the ODBC-devel package and be done with it. -MC ********************** -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090223/72ca12ae/attachment-0001.html From helmut.kuper at ewetel.de Mon Feb 23 06:44:45 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Mon, 23 Feb 2009 15:44:45 +0100 Subject: [Freeswitch-users] OpenZAP codec Question: Why L16@8000 codec for incoming calls Message-ID: <49A2B65D.2080003@ewetel.de> Hello, today I found in FS logfile lines like this: 2009-02-23 15:27:12 [DEBUG] switch_ivr_originate.c:1605 switch_ivr_originate() Raw Codec Activation Success L16 at 8000hz 1 channel 20ms It looks like L16 codec is used for incoming calls: 2009-02-23 15:27:03 [DEBUG] switch_core_session.c:523 switch_core_session_perform_receive_message() Send signal OpenZAP/1:18/2799 [BREAK] 2009-02-23 15:27:03 [NOTICE] switch_ivr_originate.c:1588 switch_ivr_originate() Pre-Answer OpenZAP/1:18/2799! 2009-02-23 15:27:03 [DEBUG] switch_ivr_originate.c:1605 switch_ivr_originate() Raw Codec Activation Success L16 at 8000hz 1 channel 20ms 2009-02-23 15:27:03 [DEBUG] switch_ivr_originate.c:1664 switch_ivr_originate() Play Ringback Tone [%(1000, 4000, 425.0, 0)] 2009-02-23 15:27:03 [DEBUG] sofia.c:2725 sofia_handle_sip_i_state() Channel sofia/internal/sip:2799 at 85.16.245.254:5060;user=phone;transport=udp entering state [proceeding] 2009-02-23 15:27:03 [NOTICE] sofia.c:2779 sofia_handle_sip_i_state() Ring-Ready sofia/internal/sip:2799 at 85.16.245.254:5060;user=phone;transport=udp! 2009-02-23 15:27:03 [DEBUG] switch_core_io.c:652 switch_core_session_write_frame() OpenZAP/1:18/2799 receive message [TRANSCODING_NECESSARY] 2009-02-23 15:27:07 [DEBUG] Span:1 Q.931() Timer 0 of call 6 (CRV: 61, State: 0) timed out 2009-02-23 15:27:12 [DEBUG] sofia.c:2725 sofia_handle_sip_i_state() Channel sofia/internal/sip:2799 at 85.16.245.254:5060;user=phone;transport=udp entering state [ready] 2009-02-23 15:27:12 [DEBUG] sofia.c:2729 sofia_handle_sip_i_state() Remote SDP: v=0^M o=2799 121183017 121183017 IN IP4 85.16.245.254^M s=ATA186 Call^M c=IN IP4 85.16.245.254^M t=0 0^M m=audio 16384 RTP/AVP 8 101^M a=rtpmap:8 PCMA/8000/1^M a=rtpmap:101 telephone-event/8000^M a=fmtp:101 0-15^M 2009-02-23 15:27:12 [DEBUG] sofia_glue.c:2549 sofia_glue_negotiate_sdp() Audio Codec Compare [PCMA:8:8000:0]/[PCMA:8:8000:20] 2009-02-23 15:27:12 [DEBUG] sofia_glue.c:1684 sofia_glue_tech_set_codec() Set Codec sofia/internal/sip:2799 at 85.16.245.254:5060;user=phone;transport=udp PCMA/8000 20 ms 160 samples The audio codec compare function finds slightly different codecs for A and B party. The dialplan for incoming calls via openzap is this. I set the codec to use in extensions "bridge" line: In my vars.xml config I have these codecs configured: So where can I disable the L16 codec, or why is a transcoding necessary? regards Helmut From brian at freeswitch.org Mon Feb 23 07:16:30 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 23 Feb 2009 09:16:30 -0600 Subject: [Freeswitch-users] Realm Value In-Reply-To: <711428.49972.qm@web33708.mail.mud.yahoo.com> References: <711428.49972.qm@web33708.mail.mud.yahoo.com> Message-ID: You have something on your system thats causing the audio detect to see you have odbc installed.. easiest way to get around this is to just install the devel headers. http://lists.freeswitch.org/pipermail/freeswitch-users/2009-February/011762.html /b On Feb 23, 2009, at 8:42 AM, Ali Al-Rubaie wrote: > > I could compile and install FS 1.0.2 successsfully so, do I need to > install ODBC-devel package for 1.0.3 version? > > Thanks, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090223/694a3fed/attachment.html From mike at jerris.com Mon Feb 23 07:40:25 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 23 Feb 2009 10:40:25 -0500 Subject: [Freeswitch-users] OpenZAP codec Question: Why L16@8000 codec for incoming calls In-Reply-To: <49A2B65D.2080003@ewetel.de> References: <49A2B65D.2080003@ewetel.de> Message-ID: <6352F00A-EC49-4A12-91ED-2EA56C2A09C5@jerris.com> On Feb 23, 2009, at 9:44 AM, Helmut Kuper wrote: > Hello, > > today I found in FS logfile lines like this: > > 2009-02-23 15:27:12 [DEBUG] switch_ivr_originate.c:1605 > switch_ivr_originate() Raw Codec Activation Success L16 at 8000hz 1 > channel > 20ms > > > It looks like L16 codec is used for incoming calls: > > 2009-02-23 15:27:03 [DEBUG] switch_core_session.c:523 > switch_core_session_perform_receive_message() Send signal > OpenZAP/1:18/2799 [BREAK] > 2009-02-23 15:27:03 [NOTICE] switch_ivr_originate.c:1588 > switch_ivr_originate() Pre-Answer OpenZAP/1:18/2799! > 2009-02-23 15:27:03 [DEBUG] switch_ivr_originate.c:1605 > switch_ivr_originate() Raw Codec Activation Success L16 at 8000hz 1 > channel > 20ms > 2009-02-23 15:27:03 [DEBUG] switch_ivr_originate.c:1664 > switch_ivr_originate() Play Ringback Tone [%(1000, 4000, 425.0, 0)] > 2009-02-23 15:27:03 [DEBUG] sofia.c:2725 sofia_handle_sip_i_state() > Channel > sofia/internal/sip:2799 at 85.16.245.254:5060;user=phone;transport=udp > entering state [proceeding] > 2009-02-23 15:27:03 [NOTICE] sofia.c:2779 sofia_handle_sip_i_state() > Ring-Ready > sofia/internal/sip:2799 at 85.16.245.254:5060;user=phone;transport=udp! > 2009-02-23 15:27:03 [DEBUG] switch_core_io.c:652 > switch_core_session_write_frame() OpenZAP/1:18/2799 receive message > [TRANSCODING_NECESSARY] > 2009-02-23 15:27:07 [DEBUG] Span:1 Q.931() Timer 0 of call 6 (CRV: 61, > State: 0) timed out > 2009-02-23 15:27:12 [DEBUG] sofia.c:2725 sofia_handle_sip_i_state() > Channel > sofia/internal/sip:2799 at 85.16.245.254:5060;user=phone;transport=udp > entering state [ready] > 2009-02-23 15:27:12 [DEBUG] sofia.c:2729 sofia_handle_sip_i_state() > Remote SDP: > v=0^M > o=2799 121183017 121183017 IN IP4 85.16.245.254^M > s=ATA186 Call^M > c=IN IP4 85.16.245.254^M > t=0 0^M > m=audio 16384 RTP/AVP 8 101^M > a=rtpmap:8 PCMA/8000/1^M > a=rtpmap:101 telephone-event/8000^M > a=fmtp:101 0-15^M > > 2009-02-23 15:27:12 [DEBUG] sofia_glue.c:2549 > sofia_glue_negotiate_sdp() > Audio Codec Compare [PCMA:8:8000:0]/[PCMA:8:8000:20] > 2009-02-23 15:27:12 [DEBUG] sofia_glue.c:1684 > sofia_glue_tech_set_codec() Set Codec > sofia/internal/sip:2799 at 85.16.245.254:5060;user=phone;transport=udp > PCMA/8000 20 ms 160 samples > > The audio codec compare function finds slightly different codecs for A > and B party. > > The dialplan for incoming calls via openzap is this. I set the codec > to > use in extensions "bridge" line: > > > expression="(491[0-9]|492[0-8])$"> > > > data="nolocal:sip_secure_media=${user_data(${dialed_extension}@$ > {domain_name} > var sip_secure_media)}"/> > data="{absolute_codec_string=PCMA}user/$1@$${domain}"/> > > > > > In my vars.xml config I have these codecs configured: > > > > > So where can I disable the L16 codec, or why is a transcoding > necessary? Your playing a tone, we need to encode that tone into the codec of the channel. You could make it stop transcoding by not providing ringback but we are still doing some transcoding for the tone detection in openzap that you won't see via log messages. Why is this transcoding a problem? Mike From mike at jerris.com Mon Feb 23 07:41:38 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 23 Feb 2009 10:41:38 -0500 Subject: [Freeswitch-users] Realm Value In-Reply-To: References: <711428.49972.qm@web33708.mail.mud.yahoo.com> Message-ID: <2A6B9260-50A4-4DDE-808C-1F68AF2EBF45@jerris.com> I need someone with this issue to provide me ssh access to their box so I can fix this problem for everyone. No one has done so yet. Please find me on irc if you can provide access. Mike On Feb 23, 2009, at 10:16 AM, Brian West wrote: > You have something on your system thats causing the audio detect to > see you have odbc installed.. easiest way to get around this is to > just install the devel headers. > > http://lists.freeswitch.org/pipermail/freeswitch-users/2009-February/011762.html > > /b > > > On Feb 23, 2009, at 8:42 AM, Ali Al-Rubaie wrote: > >> >> I could compile and install FS 1.0.2 successsfully so, do I need to >> install ODBC-devel package for 1.0.3 version? >> >> Thanks, > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090223/30f0c6ab/attachment.html From carlos.talbot at gmail.com Mon Feb 23 08:26:54 2009 From: carlos.talbot at gmail.com (Carlos Talbot) Date: Mon, 23 Feb 2009 10:26:54 -0600 Subject: [Freeswitch-users] Skypiax (Skype Network endpoint) for FreeSWITCH In-Reply-To: <7b197bef0902191732i6fead849uace0ac906a9437b0@mail.gmail.com> References: <7b197bef0902131015k456dee87x8f0f1b7059f1b49c@mail.gmail.com> <5800526b0902181853h32de5535l6c0f9d7067a69cf0@mail.gmail.com> <7b197bef0902191732i6fead849uace0ac906a9437b0@mail.gmail.com> Message-ID: <5800526b0902230826m255e0f4fmeeece95ed44e8cb4@mail.gmail.com> Thanks Giovanni. Were you planning to check in the sample skype.conf.xml into the default FreeSWITCH conf folder? If so, just be aware the default config causes freeswitch to hang right after a "load mod_skypiax" (if you do not have skype running or specify a nonexistant skype user). regards, Carlos On Thu, Feb 19, 2009 at 7:32 PM, Giovanni Maruzzelli wrote: > On Thu, Feb 19, 2009 at 3:53 AM, Carlos Talbot > wrote: > > > One question I have, is ringback suppose to work with mod_skypiax? > Whenever > > I dial a number I get a few seconds of dead air before the call is > answered. > > I've tried adding ringback and transfer_ringback into the dialplan just > > before the bridge command but no go. Am I missing something? Thanks. > > Carlos, > > ringback now works without tricks, and Skypiax is in trunk. > > Both remote ringing and early media are treated as remote ringing > right now (eg: no early media, just ringing). > > I'll add early media support in the near future. > > Thanks a lot for testing and exercising skypiax, and please let me > know any hint, suggestion, feature request, etc > > > > Sincerely, > > Giovanni Maruzzelli > ========================================= > www.celliax.org > via Pierlombardo 9, 20135 Milano > Italy > gmaruzz at celliax dot org > Cell : +39-347-2665618 > Fax : +39-02-87390039 > > > > > On Thu, Feb 19, 2009 at 3:53 AM, Carlos Talbot > wrote: > > Giovannia, > > > > great work on mod_skypiax. I've been testing it under Windows and it > sounds > > great including PSTN calls. I plan to include it as part of the Windows > MSI > > build. > > > > One question I have, is ringback suppose to work with mod_skypiax? > Whenever > > I dial a number I get a few seconds of dead air before the call is > answered. > > I've tried adding ringback and transfer_ringback into the dialplan just > > before the bridge command but no go. Am I missing something? Thanks. > > > > regards, > > > > Carlos > > > > > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090223/dfd925ef/attachment.html From andrew at hijacked.us Mon Feb 23 09:13:41 2009 From: andrew at hijacked.us (Andrew Thompson) Date: Mon, 23 Feb 2009 12:13:41 -0500 Subject: [Freeswitch-users] mod_erlang_event compile problem In-Reply-To: <0DEC394E-4739-4055-B524-E4217593DD3C@scarlet-internet.nl> References: <20090109230650.GF5210@hijacked.us> <4A496EA6-D036-4BB5-9347-5536AD5FD0FC@scarlet-internet.nl> <20090220190811.GC29511@hijacked.us> <90DDD253-BA8D-4A79-9ACA-1560F2E18688@scarlet-internet.nl> <20090223001211.GC13957@hijacked.us> <0DEC394E-4739-4055-B524-E4217593DD3C@scarlet-internet.nl> Message-ID: <20090223171340.GD13957@hijacked.us> On Mon, Feb 23, 2009 at 11:09:28AM +0100, Leon de Rooij wrote: > Everything is running on an Ubuntu Hardy Xen domu with kernel > 2.6.24-23-xen. Oh, this might explain some things.. > > Erlang is version R12B5 and was compiled from source with options -- > enable-hipe, --enable-smp-support en --enable-threads. > I'm running this too. > FS is trunk version 12197. > Fine too. > I did copy the configuration file to ~freeswitch/conf/autoload_configs > > Also, I just checked the 'empd -names', after both FS and an erl shell > have been started: > > root at erlyfs:~# epmd -names > epmd: up and running on port 4369 with data: > name ldr at port 57114 > name freeswitch at port 8031 > > So that should be fine.. > Yes, that's correct. > I also tried loading mod_erlang_event from modules.conf, and > starting FS as root, but - not surprisingly - that didn't make any > difference. > > I've been looking in wireshark, what exactly is going over the line, > and the strange thing is, that erl opens a TCP connection, a SYN > packet is sent to FS, after which FS immediately returns an RST/ACK > packet and thus closes the connection.. I still don't see anything in > the FS CLI. > > Is there anything I can do to get more verbose output from FS - esp > info about why the connection was closed ? > It looks like ei_accept_tmo is the one resetting the connection, not my code. I can't even get an error out of it when, for example, I telnet to port 8031, it just closes the connection instantly with no error to the console. Is it possible that something is screwy with the loopback device in a xen guest? Can you get normal erlang nodes on that host to net_adm:ping each other? Andrew From helmut.kuper at ewetel.de Mon Feb 23 09:28:53 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Mon, 23 Feb 2009 18:28:53 +0100 Subject: [Freeswitch-users] OpenZAP codec Question: Why L16@8000 codec for incoming calls In-Reply-To: <6352F00A-EC49-4A12-91ED-2EA56C2A09C5@jerris.com> References: <49A2B65D.2080003@ewetel.de> <6352F00A-EC49-4A12-91ED-2EA56C2A09C5@jerris.com> Message-ID: <49A2DCD5.5090105@ewetel.de> Hi Mike, thx. Today we had some failing test fax sessions (g711/PCMA) and my first thought was that it could be caused by FS during transcoding. So I looked into FS logfile and found those hints about transcoding. But ringback shouldn't be the problem. Fax path was from FAX device (source) through a voip cpe, through a SBC through a SS7 PSTN device into PSTN. Then from PSTN through AVAYA PBX trough sangoma A104d into freeswitch. From there RTP goes to a Cisco ATA to be converted to TDM and consumed by a FAX device (Target). So we have a lot of points to look at ... regrads Helmut > Your playing a tone, we need to encode that tone into the codec of the > channel. You could make it stop transcoding by not providing ringback > but we are still doing some transcoding for the tone detection in > openzap that you won't see via log messages. Why is this transcoding > a problem? > > Mike > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > -- Mit freundlichen Gr??en Helmut Kuper Finanzdienstleistungen und Entwicklung Telefax: (0441) 8000-2799 mailto:helmut.kuper at ewetel.de ___________________________________ EWE TEL GmbH Cloppenburger Stra?e 310 26133 Oldenburg EWE TEL GmbH Handelsregister Amtsgericht Oldenburg HRB 3723 Vorsitzender des Aufsichtsrates: Heiko Harms Gesch?ftsf?hrung: Hans-Joachim Iken (Vorsitzender), Dr. Norbert Schulz, Dirk Thole Homepage: http://www.ewetel.de ___________________________________ From helmut.kuper at ewetel.de Mon Feb 23 09:29:04 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Mon, 23 Feb 2009 18:29:04 +0100 Subject: [Freeswitch-users] OpenZAP codec Question: Why L16@8000 codec for incoming calls In-Reply-To: <6352F00A-EC49-4A12-91ED-2EA56C2A09C5@jerris.com> References: <49A2B65D.2080003@ewetel.de> <6352F00A-EC49-4A12-91ED-2EA56C2A09C5@jerris.com> Message-ID: <49A2DCE0.7090204@ewetel.de> Hi Mike, thx. Today we had some failing test fax sessions (g711/PCMA) and my first thought was that it could be caused by FS during transcoding. So I looked into FS logfile and found those hints about transcoding. But ringback shouldn't be the problem. Fax path was from FAX device (source) through a voip cpe, through a SBC through a SS7 PSTN device into PSTN. Then from PSTN through AVAYA PBX trough sangoma A104d into freeswitch. From there RTP goes to a Cisco ATA to be converted to TDM and consumed by a FAX device (Target). So we have a lot of points to look at ... regrads Helmut > Your playing a tone, we need to encode that tone into the codec of the > channel. You could make it stop transcoding by not providing ringback > but we are still doing some transcoding for the tone detection in > openzap that you won't see via log messages. Why is this transcoding > a problem? > > Mike > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > From BenHoltsclaw at averyschools.net Mon Feb 23 09:52:11 2009 From: BenHoltsclaw at averyschools.net (Ben Holtsclaw) Date: Mon, 23 Feb 2009 12:52:11 -0500 Subject: [Freeswitch-users] Deployment information and use cases In-Reply-To: <7405656F-1AAA-4FF7-9DC3-4CE694D7B0AB@gmail.com> References: <1234916455.16581.49.camel@raul-laptop> <59ad9ca10902181009p7cbc0fd1s9860aacde7f3a30c@mail.gmail.com> <247f8100902181019m61ce803m5ed504d9b198dfb6@mail.gmail.com> <1235016793.22050.0.camel@raul-laptop> <499D796E.45B7.0079.0@averyschools.net> <7405656F-1AAA-4FF7-9DC3-4CE694D7B0AB@gmail.com> Message-ID: <49A29BFB.45B7.0079.0@averyschools.net> Mesquita, Relatively speaking, I feel like we are near the end of our project roll out. Perhaps the case would be stronger once everything is completed. At that time, I will be very glad to share the story on the wiki -- and hopefully elsewhere! Ben >>> On 2/21/2009 at 8:56 AM, Jo?o Mesquita wrote: Ben, thank you for your story. I would very much like to add this to the wiki if you don't mind and everyone else agrees. What do you think guys? Use cases are _ALWAYS_ a good thing for new users. Mesquita On Feb 19, 2009, at 6:23 PM, Ben Holtsclaw wrote: Raul, I am in the process of rolling out a FreeSWITCH IP PBX solution similar to what you describe. When I was trying to procure funds for a FreeSWITCH solution, I looked for the same information you're after, but came up with little. I'll briefly describe what we're trying to accomplish, and the tools I'm using to do it. This is probably more information than what you are looking for, but maybe it will also benefit someone else. We had several schools with aging or dying PBX's or KSU's. Each site had something different system, and was supported by a different VAR. Of course, the VAR's charged some outlandish fee to make onsite repair visits. Some number of Centrex lines supplied each school's dial tone. All in all, we had a very outdated and financially draining mess. Our district's long term goal had been to move to a more unified phone system. That made sense for many reasons, the chief of which was cost. We already had a strong fiber WAN in place. Why not use that for trunking and eliminate the monthly cost of the Centrex lines? That's the path we started down. Being a public entity, we had to be sure to explore all possible avenues. We looked at everything from traditional PBX's with IP add-on modules for trunking to a full blown Cisco CallManager solution. With third party proprietary systems, we were just never able to find the sweet spot between required feature set and cost. Would Cisco have been a workable solution? Absolutely. Could our small, rural, K12 public school district afford that? Not in a million years. I looked at several software packages -- some open source, some not -- but always came back to FreeSWITCH. The scalability and active development community were major factors for us. Server Hardware. Each of our five sites has a dedicated FreeSWITCH server. For hardware, we went with Dell PowerEdge 1950's with dual quad core Xeon 2.33 GHz processors, and 4GB of RAM. I have mirrored disks set up with enough space to accommodate users' voicemail. Each server will average only about 60 voicemail boxes, and we're storing sound as MP3. Disk space shouldn't be an issue. We have always been a Novell shop, so SLES is naturally our Linux distribution of choice. We chose to go with server hardware at each site so that in the event of a WAN outage, we would still at least have intra-building and emergency communication (see below). Telephony Hardware. Each of our servers includes Sangoma hardware. We actually looked at doing IP trunking to a carrier from our network core, but decided to use telco provided PRI's instead. Presently, we have two PRI's that connect to a FreeSWITCH server at the center of our network via a Sangoma A102 dual port telephony card. All calls to and from the PSTN traverse this primary server. Servers at each remote site include one of Sangoma's A200 analog cards. Emergency calls to 911 route out over this analog card through one of at least two POTS lines that remain connected at each site. Not only does this provide some redundancy in the event of a WAN outage, but it ensures proper caller location is delivered to the 911 dispatcher. Granted, there are some other solutions for the latter, but this seemed to be the most cost effective solution for us. Telephone Desksets. We chose to go with Aastra for the telephones. The standard phone that we will place in each classroom and office is the 9143i. This is an attractive phone with an adequate feature set at a price we can afford. The person that is primarily responsible for answering the phone at each site will have an Aastra 57i and some number of 560M expansion modules. We have purchased roughly 300 Aastra desksets. Logical Layout. As new sites come online, their primary phone number is being moved from the Centrex to our PRI group. All inbound calls hit our primary server, and then FreeSWITCH bridges to the appropriate secondary server based on the DID it received. On the reverse, each servers dial plan is set up to route outbound calls (save 911) to the primary server where FreeSWITCH bridges with Openzap. Site to site calls, accomplished via four digit dialing, do not hit the primary server. Outbound calls to the PSTN deliver the site's DID as the calling number. In other words, if a user from site two calls my cell phone, I see site two's published telephone number on my caller ID. Our dial plans are set up so that receptionists at each site still answer all outside calls. If not answered, the call fails over to an IVR. Should we ever decide to do so, we are now perfectly positioned to have all inbound calls to the district answered by one operator or IVR. "Welcome, and thank you for calling Avery County Schools." Stumbling Blocks. Problems we've faced so far have primarily surrounded Openzap and the Sangoma Wanpipe driver. FreeSWITCH developers won't mind telling you that this is an area that is currently not well "funded" and not 100% complete. There is some known issue where voice channels on the PRI get stuck in the wrong state and become unusable. We have experienced this a couple of times and have not been able to make or receive calls. Bouncing the Wanpipe driver has fixed this each time. We have also had trouble with DTMF detection across the PRI. If a user hits the IVR, it is oftentimes difficult to get it to properly recognize the digits that are being keyed in by the caller. This can be very, very frustrating to a caller that doesn't want to deal with an IVR anyway. The developers have suggested to me that this is a problem with the Sangoma's echo cancellation goofing up Openzap's ability to interpret the DTMF. The Sangoma hardware does have its own DTMF decoder and API, but the Openzap code currently does not make use of it. I have created a patch that makes use of the hardware decoder. We have been running it in production for a couple of weeks, and that does seem to have helped the problem. The problem hasn't gone away altogether. Those have been our two biggest issues, but we haven't let them hold us up. Conclusion. Of the five sites that will be on this system, one is fully functional with calls inbound and outbound from the PSTN. A second site is up and running with full outbound PSTN access. Their inbound DID is scheduled to move over to the PRI in one week. The server has been worked up for a third site, and the phones are starting to roll out. Sites four and five should come online by the end of April. Currently, I don't have numbers compiled for things like concurrent calls. At this point in my project, it is just not important. I really don't think our implementation will ever push FreeSWITCH's abilities in that regard. I base that statement primarily on other users' benchmarks, and what I've heard some are doing in carrier class environments. FreeSWITCH has made our project viable. An open source solution was the only way we could meet all of the project goals and stay within our budget. FreeSWITCH has proven to have all the features we require in a district wide phone system. It has not locked us into annual support contracts with third party vendors. I could go on with the accolades. However, I'll end this terribly lengthy post by saying that, overall, we have been very pleased with our choice to go with FreeSWITCH. The information in this email will seem very elementary to most people on this list, but having a message of this nature in hand would ha ve made me feel much more confident the first time I ever went to my supervisor to mention something called FreeSWITCH. :-) Thanks Tony, Brian, and Mike for a great product! Ben Holtsclaw Network Engineer Avery County Schools Ph: 828.733.3567 x2301 >>> On 2/18/2009 at 11:13 PM, Raul Fragoso wrote: Thanks guys, this is very useful information. Anyone else willing to share your experience ? Regards, Raul On Wed, 2009-02-18 at 16:19 -0200, Pablo Hernan Saro wrote: > Hi Raul, > > In my company (http://www.globant.com) we're using FreeSWITCH for high > quality conference services, integrated with OpenSIPS > (http://www.opensips.org) and Asterisk. Its performance is pretty > good. > > Pablo > > On Wed, Feb 18, 2009 at 4:09 PM, Henry Huang wrote: > > bandwidth.com has a service called phonebooth which is developed upon > > freeswitch. > > > > > > On Tue, Feb 17, 2009 at 4:20 PM, Raul Fragoso wrote: > >> > >> Hello FreeSWITCHERS, > >> > >> My company is currently creating a suite of applications which uses > >> FreeSWITCH as the back-end for an IP-PBX solution. We currently have a > >> prospect to have our first customer installation - a governmental > >> department. That is a tender to have an IP-PBX installation to connect > >> their four office branches, each one with about 300 users - which I am > >> sure FreeSWITCH is able to handle. Since this is an official tender, > >> it's part of their protocol to ask about real sites using the product. > >> > >> Having said that, would you mind sharing some information about your > >> experience with FreeSWITCH deployments ? > >> > >> No need to give many details, but a short summary with company name (if > >> possible), when it was deployed, server equipment, number of users, > >> number of concurrent calls, what kind of functions and services are used > >> and overall capacity of the system. > >> > >> I would really appreciate if you can share that information. And if you > >> guys agree (and explicitly manifest your agreement), I can compile the > >> information in the FreeSWITCH wiki under a "Use Cases" page so it can > >> serve as a common reference as well. > >> > >> Kind regards, > >> > >> Raul Fragoso > >> > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > > > -- > > Henry Huang > > UniC Solution - Communication Unified > > VoIP & Open Source software Consultant > > _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090223/232e3701/attachment-0001.html From msc at freeswitch.org Mon Feb 23 10:08:01 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 23 Feb 2009 10:08:01 -0800 Subject: [Freeswitch-users] mod_portaudio: Do not accept next call after Hangup In-Reply-To: <499D58CB.9080405@fh-wolfenbuettel.de> References: <499D58CB.9080405@fh-wolfenbuettel.de> Message-ID: <87f2f3b90902231008n73ad33fbvb05d7c87f5aa2a0c@mail.gmail.com> On Thu, Feb 19, 2009 at 5:04 AM, Rene Pankratz wrote: > Hello, > when hanging up a call with portaudio automatically the next call that > is incoming or held is accepted. > Is it possible to configure PA that way, that after hanging up (doesn't > matter whether caller or callee) no call is activated automatically? I > want to choose if I accept the next call or not. > > Thanks in advance > Ren? > Just following up - did this get resolved? -MC From msc at freeswitch.org Mon Feb 23 10:23:47 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 23 Feb 2009 10:23:47 -0800 Subject: [Freeswitch-users] Random problems with cepstral text to speech In-Reply-To: References: Message-ID: <87f2f3b90902231023t3578cc4dg515dc604fc2f3eae@mail.gmail.com> > Hello, > if Cepstral 4.x is the way to go does anybody know where to get the demo > version? > > BRs, > Claudio I think you'll have to contact Cepstral on this one. I've tried to find older revisions on their site and I can't find any way to get any voices prior to 5.1. -MC From oseslija at gmail.com Mon Feb 23 10:55:19 2009 From: oseslija at gmail.com (Ognjen Seslija) Date: Mon, 23 Feb 2009 19:55:19 +0100 Subject: [Freeswitch-users] Deployment information and use cases In-Reply-To: <1234916455.16581.49.camel@raul-laptop> References: <1234916455.16581.49.camel@raul-laptop> Message-ID: <4468a6770902231055o7b235245tb41529b55cfb76c4@mail.gmail.com> Hello, I run FreeSWITCH as a PBX solution for several companies, all sharing a single server in a "vritual pbx" deployment. Dialplans and user directories are all separate and handled per domains. Currently, there is about 250 phones set to use it, about 200 more will be migrated soon from Asterisk (I'm still using it as a PSTN PRI gateway). Everything is designed per domain, so it's easy to add more servers, add more sites into a company's dialplan, lcr etc. I really love FS as it saved me a lot of trouble I had with Asterisk. Ognjen (sekil) On Wed, Feb 18, 2009 at 1:20 AM, Raul Fragoso wrote: > Hello FreeSWITCHERS, > > My company is currently creating a suite of applications which uses > FreeSWITCH as the back-end for an IP-PBX solution. We currently have a > prospect to have our first customer installation - a governmental > department. That is a tender to have an IP-PBX installation to connect > their four office branches, each one with about 300 users - which I am > sure FreeSWITCH is able to handle. Since this is an official tender, > it's part of their protocol to ask about real sites using the product. > > Having said that, would you mind sharing some information about your > experience with FreeSWITCH deployments ? > > No need to give many details, but a short summary with company name (if > possible), when it was deployed, server equipment, number of users, > number of concurrent calls, what kind of functions and services are used > and overall capacity of the system. > > I would really appreciate if you can share that information. And if you > guys agree (and explicitly manifest your agreement), I can compile the > information in the FreeSWITCH wiki under a "Use Cases" page so it can > serve as a common reference as well. > > Kind regards, > > Raul Fragoso > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090223/6b46967f/attachment.html From nik.middleton at noblesolutions.co.uk Mon Feb 23 10:58:28 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Mon, 23 Feb 2009 18:58:28 -0000 Subject: [Freeswitch-users] Help debuging core dump Message-ID: Hi Guys I'm having problems with seg faults about every 10 mins with call loads > 200. I've processed the core dump (http://pastebin.freeswitch.org/7436) but I'm unsure what I should be looking for. I don't see the point where the crash occurred. Can someone point me to where I should be looking? FreeSWITCH Version 1.0.trunk (12246) Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090223/9ec17c37/attachment.html From anthony.minessale at gmail.com Mon Feb 23 11:38:33 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 23 Feb 2009 13:38:33 -0600 Subject: [Freeswitch-users] Help debuging core dump In-Reply-To: References: Message-ID: <191c3a030902231138l56b751acrbfa06ec2b2a2b8cf@mail.gmail.com> It looks like a file rewind operation. does the lua script use the input callback to rewind a file? It maybe be a race in some other thread can you paste a "thread apply all bt" from the same core to look at the other threads. On Mon, Feb 23, 2009 at 12:58 PM, Nik Middleton < nik.middleton at noblesolutions.co.uk> wrote: > Hi Guys > > > > I'm having problems with seg faults about every 10 mins with call loads > > 200. I've processed the core dump (http://pastebin.freeswitch.org/7436) > but I'm unsure what I should be looking for. I don't see the point where the > crash occurred. Can someone point me to where I should be looking? > > > > > > FreeSWITCH Version 1.0.trunk (12246) > > > > Regards, > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090223/e7fd79f1/attachment.html From josephbajin at gmail.com Mon Feb 23 12:10:51 2009 From: josephbajin at gmail.com (Joseph Bajin) Date: Mon, 23 Feb 2009 15:10:51 -0500 Subject: [Freeswitch-users] SIP dump to DB In-Reply-To: <499ED737.3020401@post.cz> References: <499E92E4.5010503@post.cz> <191c3a030902200554s4abda94g76ba4e2975fffa98@mail.gmail.com> <03a601c99367$f92f48f0$eb8ddad0$@net> <499EC35A.1060808@post.cz> <35b355e90902200658k48ba850ele3ede371ebe96631@mail.gmail.com> <499ECBF2.4030606@post.cz> <191c3a030902200733y795626edq8eef6f06bcd70571@mail.gmail.com> <207e7a5e0902200750t2ac9f21avfb5ae88791e4efa8@mail.gmail.com> <499ED737.3020401@post.cz> Message-ID: <1dce11f20902231210u4c128b4ch491b49c75e6e58af@mail.gmail.com> Basically, you are trying to build what Empirix has with their Hammer tool. You can create an application that is basically a mix of tshark and a database feeder. You sniff with tshark and going to basically pipe it to another application that will read the pcap file, parse it, and load it into the db for you. There are plenty of modules out there that will read pcap for you. On Fri, Feb 20, 2009 at 11:15 AM, kokoska rokoska wrote: > > > > jonathan augenstine napsal(a): > > You can tcpdump and then use wireshark to graph the calls. When the > > dump is displayed in wireshark, select 'Statistics' -> VoIP Calls. You > > will see a display of all VoIP calls. Select the one you want graphed, > > or select them all and you will see REINVITE and REFER interaction as > > well as RTP streams. > > > > Thank you very much, jonathan, for your interest! > > I use ngrep+wireshark many times a day, but I'm affraid it is not > suitable for that amount of data. > > Even with few hundreds MiBs of pcap file wireshark becoms very slow and > I can't imagine how to load 50-100 GiB file with milions of calls and > try to search for one of them :-) > > And, even worse, I should "rotate" the file and, don't end with call > divided to multiple files... > > > Best regards, > > kokoska.rokoska > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090223/c21f50fa/attachment.html From nik.middleton at noblesolutions.co.uk Mon Feb 23 12:14:10 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Mon, 23 Feb 2009 20:14:10 -0000 Subject: [Freeswitch-users] Help debuging core dump In-Reply-To: <191c3a030902231138l56b751acrbfa06ec2b2a2b8cf@mail.gmail.com> References: <191c3a030902231138l56b751acrbfa06ec2b2a2b8cf@mail.gmail.com> Message-ID: There's 160 threads, but I don't want to post it on the pastebin as it has real phone numbers. I'm sending as an attachment Regards, ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: 23 February 2009 19:39 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Help debuging core dump It looks like a file rewind operation. does the lua script use the input callback to rewind a file? It maybe be a race in some other thread can you paste a "thread apply all bt" from the same core to look at the other threads. On Mon, Feb 23, 2009 at 12:58 PM, Nik Middleton < nik.middleton at noblesolutions.co.uk> wrote: Hi Guys I'm having problems with seg faults about every 10 mins with call loads > 200. I've processed the core dump ( http://pastebin.freeswitch.org/7436) but I'm unsure what I should be looking for. I don't see the point where the crash occurred. Can someone point me to where I should be looking? FreeSWITCH Version 1.0.trunk (12246) Regards, _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... 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Name: threads.txt Url: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090223/47eb6d4d/attachment-0001.txt From kokoska.rokoska at post.cz Mon Feb 23 14:32:26 2009 From: kokoska.rokoska at post.cz (kokoska.rokoska) Date: Mon, 23 Feb 2009 23:32:26 +0100 Subject: [Freeswitch-users] SIP dump to DB In-Reply-To: <1dce11f20902231210u4c128b4ch491b49c75e6e58af@mail.gmail.com> References: <499E92E4.5010503@post.cz> <191c3a030902200554s4abda94g76ba4e2975fffa98@mail.gmail.com> <03a601c99367$f92f48f0$eb8ddad0$@net> <499EC35A.1060808@post.cz> <35b355e90902200658k48ba850ele3ede371ebe96631@mail.gmail.com> <499ECBF2.4030606@post.cz> <191c3a030902200733y795626edq8eef6f06bcd70571@mail.gmail.com> <207e7a5e0902200750t2ac9f21avfb5ae88791e4efa8@mail.gmail.com> <499ED737.3020401@post.cz> <1dce11f20902231210u4c128b4ch491b49c75e6e58af@mail.gmail.com> Message-ID: <49A323FA.8000802@post.cz> Joseph Bajin napsal(a): > Basically, you are trying to build what Empirix has with their Hammer tool. > Thank you very much, Joseph, for your interest! I have never heard about Empirix (I'll look at it), but what I'm trying to build is something like SER/Kamailio/OpenSIPS sip_trace module. > You can create an application that is basically a mix of tshark and a > database feeder. > You sniff with tshark and going to basically pipe it to another > application that will read the pcap file, parse it, and load it into the > db for you. There are plenty of modules out there that will read pcap > for you. > Thank you once more, Joseph, for suggestion! I think about it - it will be challenge for me to write robust and still fast enough (thousands messages per second) SIP parser + DB feeder :-) Best regards, kokoska.rokoska From swalker at SONASEARCH.com Mon Feb 23 14:47:13 2009 From: swalker at SONASEARCH.com (Stephen Walker) Date: Mon, 23 Feb 2009 14:47:13 -0800 Subject: [Freeswitch-users] FREESwitch on Windows Server 2003 Message-ID: <3B93E0500B57D04CBAE85520B750CFF04CA6CE@exchange.sonasearch.com> Hello: I have successfully loaded the Windows implementation (SVN 11602 - 02/02/09) from your site and it runs fine. I configured a Linksys SPA 2102 and have acquired dial tone and the '999X' tests work. I have not been able to establish connection with either FreeWorldDialup or Broadvoice as of yet. Which files do I need to edit and what are the proper entries to enable connection to FreeWorldDialup and Broadvoice? Example files and where they reside in the file structure would be very much appreciated. Thank you All the Best, Steve Steve Walker President SONASEARCH, INC 425/883-1984 NOTICE: The information contained in this document is intended by Sonasearch, Inc. or one of its subsidiaries for the use of the named individuals or entities to which it is addressed and may contain information that is privileged or otherwise confidential. It is not intended for transmission to, or receipt by, any individual or entity other than the named addressee (or a person authorized to deliver it to the named addressee) except as otherwise expressly permitted in this document. If you have received this document in error, please destroy it without copying or forwarding it, and notify the sender of the error by calling Sonasearch at (425) 883-1984. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090223/7d514817/attachment.html From andrew at hijacked.us Mon Feb 23 16:22:08 2009 From: andrew at hijacked.us (Andrew Thompson) Date: Mon, 23 Feb 2009 19:22:08 -0500 Subject: [Freeswitch-users] mod_erlang_event compile problem In-Reply-To: <0DEC394E-4739-4055-B524-E4217593DD3C@scarlet-internet.nl> References: <20090109230650.GF5210@hijacked.us> <4A496EA6-D036-4BB5-9347-5536AD5FD0FC@scarlet-internet.nl> <20090220190811.GC29511@hijacked.us> <90DDD253-BA8D-4A79-9ACA-1560F2E18688@scarlet-internet.nl> <20090223001211.GC13957@hijacked.us> <0DEC394E-4739-4055-B524-E4217593DD3C@scarlet-internet.nl> Message-ID: <20090224002207.GF13957@hijacked.us> Leon, I think I found the problem. I shouldn't have been defaulting to binding to 127.0.0.1, instead the default should be 0.0.0.0. I've patched the module to actually bind to 0.0.0.0 correctly and made it the default in the config file. Erlang nodes by default bind to 0.0.0.0, so I decided to make mod_erlang_event follow suit. Please give that a shot and see if it fixes things. Andrew From carlos.talbot at gmail.com Mon Feb 23 18:20:20 2009 From: carlos.talbot at gmail.com (Carlos Talbot) Date: Mon, 23 Feb 2009 20:20:20 -0600 Subject: [Freeswitch-users] FREESwitch on Windows Server 2003 In-Reply-To: <3B93E0500B57D04CBAE85520B750CFF04CA6CE@exchange.sonasearch.com> References: <3B93E0500B57D04CBAE85520B750CFF04CA6CE@exchange.sonasearch.com> Message-ID: <5800526b0902231820u468908c6ia11191ccf8e37767@mail.gmail.com> On Mon, Feb 23, 2009 at 4:47 PM, Stephen Walker wrote: > > Which files do I need to edit and what are the proper entries to enable > connection to FreeWorldDialup and Broadvoice? Example files and where they > reside in the file structure would be very much appreciated. > You'll need to place a gateway configuration for Broadvoice in conf/sip_profiles/external similar to this example: http://wiki.freeswitch.org/wiki/SIP_Provider_Examples#Broadvoice The same applies to FWD. http://wiki.freeswitch.org/wiki/SIP_Provider_Examples#Free_World_Dialup_.28FWD.29 Once the gateways are configured you'll need to modify the default dial plan to recognize these gateways: http://wiki.freeswitch.org/wiki/Getting_Started_Guide#Dialing_out_via_Gatewayfor dialing out and http://wiki.freeswitch.org/wiki/Getting_Started_Guide#Receiving_an_inbound_call_from_a_Gatewayfor incoming. Most of this is actually covered here: http://wiki.freeswitch.org/wiki/Installation_Guide#Windows_quick_start regards, Carlos -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090223/9bef760f/attachment.html From josephbajin at gmail.com Mon Feb 23 20:44:04 2009 From: josephbajin at gmail.com (Joseph Bajin) Date: Mon, 23 Feb 2009 23:44:04 -0500 Subject: [Freeswitch-users] SIP dump to DB In-Reply-To: <49A323FA.8000802@post.cz> References: <499E92E4.5010503@post.cz> <03a601c99367$f92f48f0$eb8ddad0$@net> <499EC35A.1060808@post.cz> <35b355e90902200658k48ba850ele3ede371ebe96631@mail.gmail.com> <499ECBF2.4030606@post.cz> <191c3a030902200733y795626edq8eef6f06bcd70571@mail.gmail.com> <207e7a5e0902200750t2ac9f21avfb5ae88791e4efa8@mail.gmail.com> <499ED737.3020401@post.cz> <1dce11f20902231210u4c128b4ch491b49c75e6e58af@mail.gmail.com> <49A323FA.8000802@post.cz> Message-ID: <1dce11f20902232044u85259f4hf369da49ce00b46b@mail.gmail.com> If you write it correctly it will work just fine. That is how most of all the other correlation engines work. Your setup is not going to be bigger than some of the large telecoms that use these systems today. On 2/23/09, kokoska.rokoska wrote: > Joseph Bajin napsal(a): >> Basically, you are trying to build what Empirix has with their Hammer >> tool. >> > > Thank you very much, Joseph, for your interest! > > I have never heard about Empirix (I'll look at it), but what I'm trying > to build is something like SER/Kamailio/OpenSIPS sip_trace module. > >> You can create an application that is basically a mix of tshark and a >> database feeder. >> You sniff with tshark and going to basically pipe it to another >> application that will read the pcap file, parse it, and load it into the >> db for you. There are plenty of modules out there that will read pcap >> for you. >> > > Thank you once more, Joseph, for suggestion! > I think about it - it will be challenge for me to write robust and still > fast enough (thousands messages per second) SIP parser + DB feeder :-) > > Best regards, > > kokoska.rokoska > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Sent from my mobile device --Joe From kokoska.rokoska at post.cz Mon Feb 23 22:13:52 2009 From: kokoska.rokoska at post.cz (kokoska.rokoska) Date: Tue, 24 Feb 2009 07:13:52 +0100 Subject: [Freeswitch-users] SIP dump to DB In-Reply-To: <1dce11f20902232044u85259f4hf369da49ce00b46b@mail.gmail.com> References: <499E92E4.5010503@post.cz> <03a601c99367$f92f48f0$eb8ddad0$@net> <499EC35A.1060808@post.cz> <35b355e90902200658k48ba850ele3ede371ebe96631@mail.gmail.com> <499ECBF2.4030606@post.cz> <191c3a030902200733y795626edq8eef6f06bcd70571@mail.gmail.com> <207e7a5e0902200750t2ac9f21avfb5ae88791e4efa8@mail.gmail.com> <499ED737.3020401@post.cz> <1dce11f20902231210u4c128b4ch491b49c75e6e58af@mail.gmail.com> <49A323FA.8000802@post.cz> <1dce11f20902232044u85259f4hf369da49ce00b46b@mail.gmail.com> Message-ID: <49A39020.3020808@post.cz> Joseph Bajin napsal(a): > If you write it correctly it will work just fine. Yes, this is challenge I have talked about :-) > That is how most of > all the other correlation engines work. I don't have enough informations but from what I heard from friendly "competitors" they are usualy log (SIP|ISUP) messages after they are parsed by their "routing" servers and not run separate tshark+parser+logger. Or they duplicate (just) SIP messages to separate machine and parse and log them there (SERlike server + sip_trace). > Your setup is not going to be > bigger than some of the large telecoms that use these systems today. > I hope so :-) Thanks once more, Joseph, for your info! Best regards, kokoska.rokoska From r.pankratz at fh-wolfenbuettel.de Mon Feb 23 23:27:02 2009 From: r.pankratz at fh-wolfenbuettel.de (Rene Pankratz) Date: Tue, 24 Feb 2009 08:27:02 +0100 Subject: [Freeswitch-users] mod_portaudio: Do not accept next call after Hangup In-Reply-To: <87f2f3b90902231008n73ad33fbvb05d7c87f5aa2a0c@mail.gmail.com> References: <499D58CB.9080405@fh-wolfenbuettel.de> <87f2f3b90902231008n73ad33fbvb05d7c87f5aa2a0c@mail.gmail.com> Message-ID: <49A3A146.8050001@fh-wolfenbuettel.de> No, unfortunately the problem still persists. Portaudio still automatically accepts/takes the next call. Ren? > On Thu, Feb 19, 2009 at 5:04 AM, Rene Pankratz > wrote: > >> Hello, >> when hanging up a call with portaudio automatically the next call that >> is incoming or held is accepted. >> Is it possible to configure PA that way, that after hanging up (doesn't >> matter whether caller or callee) no call is activated automatically? I >> want to choose if I accept the next call or not. >> >> Thanks in advance >> Ren? >> >> > Just following up - did this get resolved? > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From helmut.kuper at ewetel.de Tue Feb 24 00:33:42 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Tue, 24 Feb 2009 09:33:42 +0100 Subject: [Freeswitch-users] Patch for openzap concerning finding a free channel. Message-ID: <49A3B0E6.80408@ewetel.de> Hello, today I uploaded a little patch for openzap into trunk (r667). It marks now inbound channels as "inUse" which is conform with outbound channel handling. This should solve some problems finding a free channel in ozmod_isdn.c for inbound and outbound calls. regards Helmut From helmut.kuper at ewetel.de Tue Feb 24 01:03:29 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Tue, 24 Feb 2009 10:03:29 +0100 Subject: [Freeswitch-users] mod_openzap stops working after some calls Update In-Reply-To: <49941E24.2070002@ewetel.de> References: <498C2DC3.40701@ewetel.de> <191c3a030902060802r424f8a22tb4887dff7c6aead5@mail.gmail.com> <49907327.6010703@ewetel.de> <7CEFE4F8-7461-4DF3-ABE0-C0B818466AD0@jerris.com> <4990789B.40405@ewetel.de> <4992F295.4070809@ewetel.de> <87f2f3b90902110943q12c550e4qf8e6ecae1ecc6bec@mail.gmail.com> <49941E24.2070002@ewetel.de> Message-ID: <49A3B7E1.1080009@ewetel.de> Hello, just to keep you informed about this problem. As mentioned I added a hack to free allocated channels depending on last event time. I enhanced oz dump as well to display "last event time" and "InUse"-Flag. What I found is this: 1. InUse channel flag wasn't set for inbound calls. I fixed that as far as I understood the openzap code ;) and I tested the patch successfully for 7 days now... 2. In my setup (AVAYA as remote end for a E1) channels tend to hang in a state <> DOWN after terminating a call. Then I found TOMANYCALLS entries in FS log. I had to restart FS resp. openzap module. The hack I added is in production and works for 7 days now. No channels hanging anymore. Of course, the hack is not the final solution, but it seems to solve at least my problems in production until openzap has state timers. If the board wants, I can upload the hack as well. regards Helmut On 12.02.2009 14:03, Helmut Kuper wrote: > Hi Mike, > > at least for incoming calls this shouldn't be too brutal, cause far > end seems to know that the channel should be free otherwise it never > would allocate it. By now the hack works at least for me quite good. > Nobody from AVAYA side moaned about it, yet. But I have to wait one or > two further days to be sure ... I guess I have to talk to stkn in irc > to get an idea how long I have to use it. > > regards > helmut -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090224/bc1e80bd/attachment.html From leon at scarlet-internet.nl Tue Feb 24 01:38:29 2009 From: leon at scarlet-internet.nl (Leon de Rooij) Date: Tue, 24 Feb 2009 10:38:29 +0100 Subject: [Freeswitch-users] mod_erlang_event compile problem In-Reply-To: <20090224002207.GF13957@hijacked.us> References: <20090109230650.GF5210@hijacked.us> <4A496EA6-D036-4BB5-9347-5536AD5FD0FC@scarlet-internet.nl> <20090220190811.GC29511@hijacked.us> <90DDD253-BA8D-4A79-9ACA-1560F2E18688@scarlet-internet.nl> <20090223001211.GC13957@hijacked.us> <0DEC394E-4739-4055-B524-E4217593DD3C@scarlet-internet.nl> <20090224002207.GF13957@hijacked.us> Message-ID: <4BD8A505-CC3D-45C0-9C1D-37983657DFC1@scarlet-internet.nl> Andrew, I think you're right, packets are indeed sent to 172.31.0.13 while mod_erlang_event is listening at 127.0.0.1 ! Why didn't I see that ! ;-) Will test it now and let you know how it goes.. regards, Leon On Feb 24, 2009, at 1:22 AM, Andrew Thompson wrote: > Leon, > > I think I found the problem. I shouldn't have been defaulting to > binding > to 127.0.0.1, instead the default should be 0.0.0.0. I've patched the > module to actually bind to 0.0.0.0 correctly and made it the default > in > the config file. Erlang nodes by default bind to 0.0.0.0, so I decided > to make mod_erlang_event follow suit. > > Please give that a shot and see if it fixes things. > > Andrew > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From leon at scarlet-internet.nl Tue Feb 24 01:49:24 2009 From: leon at scarlet-internet.nl (Leon de Rooij) Date: Tue, 24 Feb 2009 10:49:24 +0100 Subject: [Freeswitch-users] mod_erlang_event compile problem In-Reply-To: <20090224002207.GF13957@hijacked.us> References: <20090109230650.GF5210@hijacked.us> <4A496EA6-D036-4BB5-9347-5536AD5FD0FC@scarlet-internet.nl> <20090220190811.GC29511@hijacked.us> <90DDD253-BA8D-4A79-9ACA-1560F2E18688@scarlet-internet.nl> <20090223001211.GC13957@hijacked.us> <0DEC394E-4739-4055-B524-E4217593DD3C@scarlet-internet.nl> <20090224002207.GF13957@hijacked.us> Message-ID: <714AD975-7224-43F8-A8D2-3381379237D3@scarlet-internet.nl> Well, this works, I feel a bit stupid now :-] Now it's time to play with it.. Thanks a lot ! kind regards, Leon On Feb 24, 2009, at 1:22 AM, Andrew Thompson wrote: > Leon, > > I think I found the problem. I shouldn't have been defaulting to > binding > to 127.0.0.1, instead the default should be 0.0.0.0. I've patched the > module to actually bind to 0.0.0.0 correctly and made it the default > in > the config file. Erlang nodes by default bind to 0.0.0.0, so I decided > to make mod_erlang_event follow suit. > > Please give that a shot and see if it fixes things. > > Andrew > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From kawarod at laposte.net Tue Feb 24 06:05:53 2009 From: kawarod at laposte.net (rod) Date: Tue, 24 Feb 2009 18:05:53 +0400 Subject: [Freeswitch-users] mod_fax and sending a fax In-Reply-To: References: <49919822.3030101@laposte.net> <499C16C6.1000006@laposte.net> Message-ID: <49A3FEC1.5090300@laposte.net> Hi, the clue for sending fax is to use the originate command in the CLI: originate sofia/example/100 at 10.10.10.10 &txfax(/path_to_fax_file) this command will send the fax file via profile example to fax machine 100 reachable via 10.10.10.10 Hope this could help others :p regards, rod. Javier Aristiz?bal wrote: > Hi Rod, i just play with rx_fax and work for me. I didn't work with > tx_fax but i understand, that you need a .tiff file to send > passthrough the rx_fax. Maybe that can help you > > regards > javar > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From eric at rf.com Tue Feb 24 01:33:08 2009 From: eric at rf.com (Eric Chamberlain) Date: Tue, 24 Feb 2009 01:33:08 -0800 Subject: [Freeswitch-users] Skypiax, same skype user, multiple channels Message-ID: I was reading through the Skypiax documentation and saw the comment that it's not possible to run multiple skype clients on the same linux machine, all using the same skype user account. It's possible to run multiple skype clients with the same skype user account, as long as the skype clients are not accessing the same Skype dbpath. We use runuser to run multiple skype clients. All the clients use the same skype user, but each instance uses a different home directory, each with its own .Skype folder. In such a configuration, will Skypiax support multiple channels using the same skype username? -- Eric Chamberlain, Founder RF.com - http://RF.com/ From mike at jerris.com Tue Feb 24 06:47:27 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 24 Feb 2009 09:47:27 -0500 Subject: [Freeswitch-users] mod_portaudio: Do not accept next call after Hangup In-Reply-To: <49A3A146.8050001@fh-wolfenbuettel.de> References: <499D58CB.9080405@fh-wolfenbuettel.de> <87f2f3b90902231008n73ad33fbvb05d7c87f5aa2a0c@mail.gmail.com> <49A3A146.8050001@fh-wolfenbuettel.de> Message-ID: Please report this bug to jira.freeswitch.org. On Feb 24, 2009, at 2:27 AM, Rene Pankratz wrote: > No, unfortunately the problem still persists. Portaudio still > automatically accepts/takes the next call. > > Ren? >> On Thu, Feb 19, 2009 at 5:04 AM, Rene Pankratz >> wrote: >> >>> Hello, >>> when hanging up a call with portaudio automatically the next call >>> that >>> is incoming or held is accepted. >>> Is it possible to configure PA that way, that after hanging up >>> (doesn't >>> matter whether caller or callee) no call is activated >>> automatically? I >>> want to choose if I accept the next call or not. >>> >>> Thanks in advance >>> Ren? >>> >>> >> Just following up - did this get resolved? >> -MC >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From anthony.minessale at gmail.com Tue Feb 24 07:22:40 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 24 Feb 2009 09:22:40 -0600 Subject: [Freeswitch-users] mod_portaudio: Do not accept next call after Hangup In-Reply-To: <49A3A146.8050001@fh-wolfenbuettel.de> References: <499D58CB.9080405@fh-wolfenbuettel.de> <87f2f3b90902231008n73ad33fbvb05d7c87f5aa2a0c@mail.gmail.com> <49A3A146.8050001@fh-wolfenbuettel.de> Message-ID: <191c3a030902240722q30bae77cmd60cdea825011fb6@mail.gmail.com> What direction is the original call? Are you sure you do not have the auto_answer enabled? On Tue, Feb 24, 2009 at 1:27 AM, Rene Pankratz < r.pankratz at fh-wolfenbuettel.de> wrote: > No, unfortunately the problem still persists. Portaudio still > automatically accepts/takes the next call. > > Ren? > > On Thu, Feb 19, 2009 at 5:04 AM, Rene Pankratz > > wrote: > > > >> Hello, > >> when hanging up a call with portaudio automatically the next call that > >> is incoming or held is accepted. > >> Is it possible to configure PA that way, that after hanging up (doesn't > >> matter whether caller or callee) no call is activated automatically? I > >> want to choose if I accept the next call or not. > >> > >> Thanks in advance > >> Ren? > >> > >> > > Just following up - did this get resolved? > > -MC > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090224/007b9339/attachment.html From andrew at hijacked.us Tue Feb 24 09:33:00 2009 From: andrew at hijacked.us (Andrew Thompson) Date: Tue, 24 Feb 2009 12:33:00 -0500 Subject: [Freeswitch-users] mod_erlang_event compile problem In-Reply-To: <714AD975-7224-43F8-A8D2-3381379237D3@scarlet-internet.nl> References: <20090109230650.GF5210@hijacked.us> <4A496EA6-D036-4BB5-9347-5536AD5FD0FC@scarlet-internet.nl> <20090220190811.GC29511@hijacked.us> <90DDD253-BA8D-4A79-9ACA-1560F2E18688@scarlet-internet.nl> <20090223001211.GC13957@hijacked.us> <0DEC394E-4739-4055-B524-E4217593DD3C@scarlet-internet.nl> <20090224002207.GF13957@hijacked.us> <714AD975-7224-43F8-A8D2-3381379237D3@scarlet-internet.nl> Message-ID: <20090224173259.GH13957@hijacked.us> On Tue, Feb 24, 2009 at 10:49:24AM +0100, Leon de Rooij wrote: > Well, this works, I feel a bit stupid now :-] Now it's time to play > with it.. > Nah, bad choice of defaults on my part. Defaulting to 0.0.0.0 is much more consistant and compatible. For some reason I was trying to emulate the event socket, not an erlang node. Thanks for finally making me solve the problem instead of just working around it. Andrew From kerrada2003 at yahoo.com Tue Feb 24 09:24:07 2009 From: kerrada2003 at yahoo.com (Ali Al-Rubaie) Date: Tue, 24 Feb 2009 09:24:07 -0800 (PST) Subject: [Freeswitch-users] file directory.conf.xml In-Reply-To: Message-ID: <57884.44323.qm@web33703.mail.mud.yahoo.com> Hi, The file directory.conf.xml had been mentioned in the documentation many times but there is not such file in the conf folder. Do you mean default.xml in directory folder? Thanks! --- On Tue, 2/24/09, freeswitch-users-request at lists.freeswitch.org wrote: From: freeswitch-users-request at lists.freeswitch.org Subject: Freeswitch-users Digest, Vol 32, Issue 181 To: freeswitch-users at lists.freeswitch.org Date: Tuesday, February 24, 2009, 3:34 AM Send Freeswitch-users mailing list submissions to freeswitch-users at lists.freeswitch.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.freeswitch.org/mailman/listinfo/freeswitch-users or, via email, send a message with subject or body 'help' to freeswitch-users-request at lists.freeswitch.org You can reach the person managing the list at freeswitch-users-owner at lists.freeswitch.org When replying, please edit your Subject line so it is more specific than "Re: Contents of Freeswitch-users digest..." Today's Topics: 1. Re: SIP dump to DB (kokoska.rokoska) 2. FREESwitch on Windows Server 2003 (Stephen Walker) 3. Re: mod_erlang_event compile problem (Andrew Thompson) 4. Re: FREESwitch on Windows Server 2003 (Carlos Talbot) 5. Re: SIP dump to DB (Joseph Bajin) 6. Re: SIP dump to DB (kokoska.rokoska) 7. Re: mod_portaudio: Do not accept next call after Hangup (Rene Pankratz) 8. Patch for openzap concerning finding a free channel. (Helmut Kuper) ---------------------------------------------------------------------- Message: 1 Date: Mon, 23 Feb 2009 23:32:26 +0100 From: "kokoska.rokoska" Subject: Re: [Freeswitch-users] SIP dump to DB To: freeswitch-users at lists.freeswitch.org Message-ID: <49A323FA.8000802 at post.cz> Content-Type: text/plain; charset=ISO-8859-1 Joseph Bajin napsal(a): > Basically, you are trying to build what Empirix has with their Hammer tool. > Thank you very much, Joseph, for your interest! I have never heard about Empirix (I'll look at it), but what I'm trying to build is something like SER/Kamailio/OpenSIPS sip_trace module. > You can create an application that is basically a mix of tshark and a > database feeder. > You sniff with tshark and going to basically pipe it to another > application that will read the pcap file, parse it, and load it into the > db for you. There are plenty of modules out there that will read pcap > for you. > Thank you once more, Joseph, for suggestion! I think about it - it will be challenge for me to write robust and still fast enough (thousands messages per second) SIP parser + DB feeder :-) Best regards, kokoska.rokoska ------------------------------ Message: 2 Date: Mon, 23 Feb 2009 14:47:13 -0800 From: "Stephen Walker" Subject: [Freeswitch-users] FREESwitch on Windows Server 2003 To: Message-ID: <3B93E0500B57D04CBAE85520B750CFF04CA6CE at exchange.sonasearch.com> Content-Type: text/plain; charset="us-ascii" Hello: I have successfully loaded the Windows implementation (SVN 11602 - 02/02/09) from your site and it runs fine. I configured a Linksys SPA 2102 and have acquired dial tone and the '999X' tests work. I have not been able to establish connection with either FreeWorldDialup or Broadvoice as of yet. Which files do I need to edit and what are the proper entries to enable connection to FreeWorldDialup and Broadvoice? Example files and where they reside in the file structure would be very much appreciated. Thank you All the Best, Steve Steve Walker President SONASEARCH, INC 425/883-1984 NOTICE: The information contained in this document is intended by Sonasearch, Inc. or one of its subsidiaries for the use of the named individuals or entities to which it is addressed and may contain information that is privileged or otherwise confidential. It is not intended for transmission to, or receipt by, any individual or entity other than the named addressee (or a person authorized to deliver it to the named addressee) except as otherwise expressly permitted in this document. If you have received this document in error, please destroy it without copying or forwarding it, and notify the sender of the error by calling Sonasearch at (425) 883-1984. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090223/7d514817/attachment-0001.html ------------------------------ Message: 3 Date: Mon, 23 Feb 2009 19:22:08 -0500 From: Andrew Thompson Subject: Re: [Freeswitch-users] mod_erlang_event compile problem To: freeswitch-users at lists.freeswitch.org Message-ID: <20090224002207.GF13957 at hijacked.us> Content-Type: text/plain; charset=us-ascii Leon, I think I found the problem. I shouldn't have been defaulting to binding to 127.0.0.1, instead the default should be 0.0.0.0. I've patched the module to actually bind to 0.0.0.0 correctly and made it the default in the config file. Erlang nodes by default bind to 0.0.0.0, so I decided to make mod_erlang_event follow suit. Please give that a shot and see if it fixes things. Andrew ------------------------------ Message: 4 Date: Mon, 23 Feb 2009 20:20:20 -0600 From: Carlos Talbot Subject: Re: [Freeswitch-users] FREESwitch on Windows Server 2003 To: freeswitch-users at lists.freeswitch.org Message-ID: <5800526b0902231820u468908c6ia11191ccf8e37767 at mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" On Mon, Feb 23, 2009 at 4:47 PM, Stephen Walker wrote: > > Which files do I need to edit and what are the proper entries to enable > connection to FreeWorldDialup and Broadvoice? Example files and where they > reside in the file structure would be very much appreciated. > You'll need to place a gateway configuration for Broadvoice in conf/sip_profiles/external similar to this example: http://wiki.freeswitch.org/wiki/SIP_Provider_Examples#Broadvoice The same applies to FWD. http://wiki.freeswitch.org/wiki/SIP_Provider_Examples#Free_World_Dialup_.28FWD.29 Once the gateways are configured you'll need to modify the default dial plan to recognize these gateways: http://wiki.freeswitch.org/wiki/Getting_Started_Guide#Dialing_out_via_Gatewayfor dialing out and http://wiki.freeswitch.org/wiki/Getting_Started_Guide#Receiving_an_inbound_call_from_a_Gatewayfor incoming. Most of this is actually covered here: http://wiki.freeswitch.org/wiki/Installation_Guide#Windows_quick_start regards, Carlos -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090223/9bef760f/attachment-0001.html ------------------------------ Message: 5 Date: Mon, 23 Feb 2009 23:44:04 -0500 From: Joseph Bajin Subject: Re: [Freeswitch-users] SIP dump to DB To: freeswitch-users at lists.freeswitch.org Message-ID: <1dce11f20902232044u85259f4hf369da49ce00b46b at mail.gmail.com> Content-Type: text/plain; charset=ISO-8859-1 If you write it correctly it will work just fine. That is how most of all the other correlation engines work. Your setup is not going to be bigger than some of the large telecoms that use these systems today. On 2/23/09, kokoska.rokoska wrote: > Joseph Bajin napsal(a): >> Basically, you are trying to build what Empirix has with their Hammer >> tool. >> > > Thank you very much, Joseph, for your interest! > > I have never heard about Empirix (I'll look at it), but what I'm trying > to build is something like SER/Kamailio/OpenSIPS sip_trace module. > >> You can create an application that is basically a mix of tshark and a >> database feeder. >> You sniff with tshark and going to basically pipe it to another >> application that will read the pcap file, parse it, and load it into the >> db for you. There are plenty of modules out there that will read pcap >> for you. >> > > Thank you once more, Joseph, for suggestion! > I think about it - it will be challenge for me to write robust and still > fast enough (thousands messages per second) SIP parser + DB feeder :-) > > Best regards, > > kokoska.rokoska > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Sent from my mobile device --Joe ------------------------------ Message: 6 Date: Tue, 24 Feb 2009 07:13:52 +0100 From: "kokoska.rokoska" Subject: Re: [Freeswitch-users] SIP dump to DB To: freeswitch-users at lists.freeswitch.org Message-ID: <49A39020.3020808 at post.cz> Content-Type: text/plain; charset=ISO-8859-1 Joseph Bajin napsal(a): > If you write it correctly it will work just fine. Yes, this is challenge I have talked about :-) > That is how most of > all the other correlation engines work. I don't have enough informations but from what I heard from friendly "competitors" they are usualy log (SIP|ISUP) messages after they are parsed by their "routing" servers and not run separate tshark+parser+logger. Or they duplicate (just) SIP messages to separate machine and parse and log them there (SERlike server + sip_trace). > Your setup is not going to be > bigger than some of the large telecoms that use these systems today. > I hope so :-) Thanks once more, Joseph, for your info! Best regards, kokoska.rokoska ------------------------------ Message: 7 Date: Tue, 24 Feb 2009 08:27:02 +0100 From: Rene Pankratz Subject: Re: [Freeswitch-users] mod_portaudio: Do not accept next call after Hangup To: freeswitch-users at lists.freeswitch.org Message-ID: <49A3A146.8050001 at fh-wolfenbuettel.de> Content-Type: text/plain; charset=ISO-8859-1; format=flowed No, unfortunately the problem still persists. Portaudio still automatically accepts/takes the next call. Ren? > On Thu, Feb 19, 2009 at 5:04 AM, Rene Pankratz > wrote: > >> Hello, >> when hanging up a call with portaudio automatically the next call that >> is incoming or held is accepted. >> Is it possible to configure PA that way, that after hanging up (doesn't >> matter whether caller or callee) no call is activated automatically? I >> want to choose if I accept the next call or not. >> >> Thanks in advance >> Ren? >> >> > Just following up - did this get resolved? > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > ------------------------------ Message: 8 Date: Tue, 24 Feb 2009 09:33:42 +0100 From: Helmut Kuper Subject: [Freeswitch-users] Patch for openzap concerning finding a free channel. To: freeswitch-users at lists.freeswitch.org Message-ID: <49A3B0E6.80408 at ewetel.de> Content-Type: text/plain; charset=ISO-8859-1 Hello, today I uploaded a little patch for openzap into trunk (r667). It marks now inbound channels as "inUse" which is conform with outbound channel handling. This should solve some problems finding a free channel in ozmod_isdn.c for inbound and outbound calls. regards Helmut ------------------------------ _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org End of Freeswitch-users Digest, Vol 32, Issue 181 ************************************************* -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090224/cf3cb945/attachment.html From alex.gusak at gmail.com Tue Feb 24 09:28:29 2009 From: alex.gusak at gmail.com (Alex Gusak) Date: Tue, 24 Feb 2009 19:28:29 +0200 Subject: [Freeswitch-users] new ilbc lib Message-ID: Hello. After upgrade to version 1.0.3 we have a problem with the codec iLBC (I think that this is due to the transition to a new ilbc libs 1 week ago). Very poor quality for calls to the codec iLBC mode=20 (crack in the dynamic). iLBC mode=30 works well. Tested with phones and Zoiper SJPhone. After a rollback to the old version of FreeSWITCH 1.0.2 this is not a problem, iLBC works fine in both modes (mode = 20 and mode = 30). What could be the problem? -- Alex Gusak From brian at freeswitch.org Tue Feb 24 09:32:38 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 24 Feb 2009 11:32:38 -0600 Subject: [Freeswitch-users] new ilbc lib In-Reply-To: References: Message-ID: The problem comes up that the default is 30... the chances are that your phone doesn't set the mode= line so we default to 30 when this takes place. Not setting the mode= line in the FMTP usually means 30ms... which is the default. So to force this always to 30 you can allow iLBC at 30i, because if you invite to me with 20 and I 200 ok you 30.. you are to use 30 no exceptions. Most phones do not obey this rule. /b On Feb 24, 2009, at 11:28 AM, Alex Gusak wrote: > Hello. > > After upgrade to version 1.0.3 we have a problem with the codec iLBC > (I think that this is due to the transition to a new ilbc libs 1 week > ago). > Very poor quality for calls to the codec iLBC mode=20 (crack in the > dynamic). iLBC mode=30 works well. > Tested with phones and Zoiper SJPhone. > > After a rollback to the old version of FreeSWITCH 1.0.2 this is not a > problem, iLBC works fine in both modes (mode = 20 and mode = 30). > > What could be the problem? From freeswitch-users at digitaldan.com Tue Feb 24 09:49:51 2009 From: freeswitch-users at digitaldan.com (Dan) Date: Tue, 24 Feb 2009 10:49:51 -0700 (MST) Subject: [Freeswitch-users] Recording and outbound rtp Message-ID: <12581186.4501235497785860.JavaMail.daniel@osxlaptop> Hi, I have a small javascript application that accepts a call, retrieves some dtmf digits and then records the call to an icecast server. This works great. The problem I'm having is that when the call is being recorded freeswitch is no longer sending rtp packets back to the originating caller, in my case a Cisco 5300 with a bunch of T1 voice circuits in it. This makes sense, since no voice data back is being generated. Unfortunately my Cisco gear has rtp inactivity timers set up to hang up a call after 3 minutes of no incoming rtp packets, this is a global setting that cannot be configured for a single dial peer. Does anyone have a suggestion to generate rtp packets every once in a while? I tried setting comfort noise which did not seem to send anything. I could try playing a empty/short wav file every minute or so but the javascript call session.record is blocking, would a traditional javascript timer and callback to play a wav file be my best bet or is there a better approach? I'm using FreeSWITCH Version 1.0.trunk (12108M) on debian etch. Thanks! Dan- -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090224/edcd3f58/attachment.html From msc at freeswitch.org Tue Feb 24 10:16:44 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 24 Feb 2009 10:16:44 -0800 Subject: [Freeswitch-users] file directory.conf.xml In-Reply-To: <57884.44323.qm@web33703.mail.mud.yahoo.com> References: <57884.44323.qm@web33703.mail.mud.yahoo.com> Message-ID: <87f2f3b90902241016w5f31675bmf9cf20d13e552650@mail.gmail.com> On Tue, Feb 24, 2009 at 9:24 AM, Ali Al-Rubaie wrote: > Hi, > > The file directory.conf.xml had been mentioned in the documentation many > times but there is not such file in the conf folder. Do you mean default.xml > in directory folder? > > Thanks! Can you tell me where you see that file name listed? It's possible that it should be "dialplan_directory.conf.xml" but I don't know for sure. I will check it out. -MC From anthony.minessale at gmail.com Tue Feb 24 11:05:33 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 24 Feb 2009 13:05:33 -0600 Subject: [Freeswitch-users] Recording and outbound rtp In-Reply-To: <12581186.4501235497785860.JavaMail.daniel@osxlaptop> References: <12581186.4501235497785860.JavaMail.daniel@osxlaptop> Message-ID: <191c3a030902241105t5d57b1abt17555c68faf16263@mail.gmail.com> is it during a bridged call? On Tue, Feb 24, 2009 at 11:49 AM, Dan wrote: > Hi, > > I have a small javascript application that accepts a call, retrieves some > dtmf digits and then records the call to an icecast server. This works > great. > > The problem I'm having is that when the call is being recorded freeswitch > is no longer sending rtp packets back to the originating caller, in my case > a Cisco 5300 with a bunch of T1 voice circuits in it. This makes sense, > since no voice data back is being generated. Unfortunately my Cisco gear > has rtp inactivity timers set up to hang up a call after 3 minutes of no > incoming rtp packets, this is a global setting that cannot be configured for > a single dial peer. Does anyone have a suggestion to generate rtp packets > every once in a while? I tried setting comfort noise which did not seem to > send anything. I could try playing a empty/short wav file every minute or > so but the javascript call session.record is blocking, would a traditional > javascript timer and callback to play a wav file be my best bet or is there > a better approach? I'm using FreeSWITCH Version 1.0.trunk (12108M) on debian > etch. > > Thanks! > Dan- > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090224/82514870/attachment.html From codecomplete at free.fr Tue Feb 24 11:08:06 2009 From: codecomplete at free.fr (Fred) Date: Tue, 24 Feb 2009 20:08:06 +0100 Subject: [Freeswitch-users] Web-based forum? Message-ID: <7.0.1.0.2.20090224200706.02496ca0@fredshack.com> Hello Maybe this question has been raised before, but if not: There's so much traffic in this mailing list that I was wondering if adding a web-based forum on the site was in the works? Cheers, From mike at jerris.com Tue Feb 24 11:19:33 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 24 Feb 2009 14:19:33 -0500 Subject: [Freeswitch-users] Web-based forum? In-Reply-To: <7.0.1.0.2.20090224200706.02496ca0@fredshack.com> References: <7.0.1.0.2.20090224200706.02496ca0@fredshack.com> Message-ID: <2836579B-EA7A-4CC8-859E-3C0A439176C9@jerris.com> The web version of this list is available at: http://www.nabble.com/Freeswitch-users-f32209.html Mike On Feb 24, 2009, at 2:08 PM, Fred wrote: > Hello > > Maybe this question has been raised before, but if not: There's so > much traffic in this mailing list that I was wondering if adding a > web-based forum on the site was in the works? > > Cheers, > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From msc at freeswitch.org Tue Feb 24 11:19:44 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 24 Feb 2009 11:19:44 -0800 Subject: [Freeswitch-users] Web-based forum? In-Reply-To: <7.0.1.0.2.20090224200706.02496ca0@fredshack.com> References: <7.0.1.0.2.20090224200706.02496ca0@fredshack.com> Message-ID: <87f2f3b90902241119x5fe2f1cek57b4ee42c6b47525@mail.gmail.com> > Maybe this question has been raised before, but if not: There's so > much traffic in this mailing list that I was wondering if adding a > web-based forum on the site was in the works? We are upgrading the freeswitch.org site soon to drupal 6.9. We are considering turning on the forum feature there. No definitive decision has been made but this request has come in several times. However, we are trying to make it so that the devs don't have yet another place to have to monitor for user questions, etc. so we will need to figure out a way to make it easy to use for the experts... -MC From mszlazak at aol.com Tue Feb 24 11:36:44 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Tue, 24 Feb 2009 14:36:44 -0500 Subject: [Freeswitch-users] New build gives error message for default grammar file?? Message-ID: <8CB64CE5BC62C9B-8D0-13DC@WEBMAIL-MC11.sysops.aol.com> I'm getting this error message trying out the pizza demo in FS 1.0.3: ?"Can't open dictionary C:\Program Files\FreeSWITCH\grammar\default.dic" I didn't have this before where there was no default.dic file. Is there some place a path has to be set now? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090224/eb925b37/attachment.html From brian at freeswitch.org Tue Feb 24 11:45:11 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 24 Feb 2009 13:45:11 -0600 Subject: [Freeswitch-users] New build gives error message for default grammar file?? In-Reply-To: <8CB64CE5BC62C9B-8D0-13DC@WEBMAIL-MC11.sysops.aol.com> References: <8CB64CE5BC62C9B-8D0-13DC@WEBMAIL-MC11.sysops.aol.com> Message-ID: <89CDF21E-B363-49B4-BB71-22D97F2CF89F@freeswitch.org> You're not in 1.0.3 you're in SVN trunk... The reason I know this is that wasn't changed till AFTER 1.0.3 was tagged and a new file set was used... please go into your libs dir and wipe out pocketsphinx and sphinx base.. then let it redownload them. I'll make you a new tarball of the new grammar files which are in the jsgf format. An example was added to scripts yes_no.gram ... word of warning the new pocketsphinx loads the entire dictionary on port open which on a machine of any speed should load the entire thing in 2 seconds or less... but here is the WARNING WARNING WARNING... If you happen to use words that aren't in the dictionary pocketsphinx WILL CRASH.. I am already on this with the dev's to work out a fix for this its been a long standing bug apparently in the lib. /b On Feb 24, 2009, at 1:36 PM, mszlazak at aol.com wrote: > I'm getting this error message trying out the pizza demo in FS 1.0.3: > > "Can't open dictionary C:\Program Files\FreeSWITCH\grammar > \default.dic" > > I didn't have this before where there was no default.dic file. > > Is there some place a path has to be set now? > > Thanks. > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090224/455dad39/attachment-0001.html From brian at freeswitch.org Tue Feb 24 11:46:34 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 24 Feb 2009 13:46:34 -0600 Subject: [Freeswitch-users] New build gives error message for default grammar file?? In-Reply-To: <8CB64CE5BC62C9B-8D0-13DC@WEBMAIL-MC11.sysops.aol.com> References: <8CB64CE5BC62C9B-8D0-13DC@WEBMAIL-MC11.sysops.aol.com> Message-ID: <730B80E0-6DD8-4A74-9EF2-04E1C851815A@freeswitch.org> http://www.bkw.org/pizza_gram.tar.gz /b On Feb 24, 2009, at 1:36 PM, mszlazak at aol.com wrote: > I'm getting this error message trying out the pizza demo in FS 1.0.3: > > "Can't open dictionary C:\Program Files\FreeSWITCH\grammar > \default.dic" > > I didn't have this before where there was no default.dic file. > > Is there some place a path has to be set now? > > Thanks. > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090224/6dc0d4ce/attachment.html From freeswitch-users at digitaldan.com Tue Feb 24 12:02:19 2009 From: freeswitch-users at digitaldan.com (freeswitch-users at digitaldan.com) Date: Tue, 24 Feb 2009 13:02:19 -0700 (MST) Subject: [Freeswitch-users] Recording and outbound rtp In-Reply-To: <17338844.9091235504936529.JavaMail.daniel@radio> Message-ID: <611594.9131235505720885.JavaMail.daniel@radio> no, I'm matching the incoming sip call via the destination number in my public context and executing the javascript appliaction. This app directly answers the call and records it until the user hangs up. D- ----- Original Message ----- From: "Anthony Minessale" To: freeswitch-users at lists.freeswitch.org Sent: Tuesday, February 24, 2009 12:05:33 PM GMT -07:00 US/Canada Mountain Subject: Re: [Freeswitch-users] Recording and outbound rtp is it during a bridged call? On Tue, Feb 24, 2009 at 11:49 AM, Dan < freeswitch-users at digitaldan.com > wrote: Hi, I have a small javascript application that accepts a call, retrieves some dtmf digits and then records the call to an icecast server. This works great. The problem I'm having is that when the call is being recorded freeswitch is no longer sending rtp packets back to the originating caller, in my case a Cisco 5300 with a bunch of T1 voice circuits in it. This makes sense, since no voice data back is being generated. Unfortunately my Cisco gear has rtp inactivity timers set up to hang up a call after 3 minutes of no incoming rtp packets, this is a global setting that cannot be configured for a single dial peer. Does anyone have a suggestion to generate rtp packets every once in a while? I tried setting comfort noise which did not seem to send anything. I could try playing a empty/short wav file every minute or so but the javascript call session.record is blocking, would a traditional javascript timer and callback to play a wav file be my best bet or is there a better approach? I'm using FreeSWITCH Version 1.0.trunk (12108M) on debian etch. Thanks! Dan- _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/ PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090224/e8956712/attachment.html From mszlazak at aol.com Tue Feb 24 12:18:54 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Tue, 24 Feb 2009 15:18:54 -0500 Subject: [Freeswitch-users] New build gives error message for default grammar file?? In-Reply-To: <89CDF21E-B363-49B4-BB71-22D97F2CF89F@freeswitch.org> References: <8CB64CE5BC62C9B-8D0-13DC@WEBMAIL-MC11.sysops.aol.com> <89CDF21E-B363-49B4-BB71-22D97F2CF89F@freeswitch.org> Message-ID: <8CB64D43FA7B70B-8D0-16EF@WEBMAIL-MC11.sysops.aol.com> Hi Brian, It sounds like I'd be better off with 1.0.3 than SVN and will waiting for the fix? But thanks for the files and info. Mark. -----Original Message----- From: Brian West To: freeswitch-users at lists.freeswitch.org Sent: Tue, 24 Feb 2009 11:45 am Subject: Re: [Freeswitch-users] New build gives error message for default grammar file?? You're not in 1.0.3 you're in SVN trunk... The reason I know this is that wasn't changed till AFTER 1.0.3 was tagged and a new file set was used... please go into your libs dir and wipe out pocketsphinx and sphinx base.. then let it redownload them. ?I'll make you a new tarball of the new grammar files which are in the jsgf format. ?An example was added to scripts yes_no.gram ... word of warning the new pocketsphinx loads the entire dictionary on port open which on a machine of any speed should load the entire thing in 2 seconds or less... but here is the WARNING WARNING WARNING... If you happen to use words that aren't in the dictionary pocketsphinx WILL CRASH.. I am already on this with the dev's to work out a fix for this its been a long standing bug apparently in the lib. /b On Feb 24, 2009, at 1:36 PM, mszlazak at aol.com wrote: I'm getting this error message trying out the pizza demo in FS 1.0.3: ?"Can't open dictionary C:\Program Files\FreeSWITCH\grammar\default.dic" I didn't have this before where there was no default.dic file. Is there some place a path has to be set now? Thanks. = _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090224/0a673569/attachment.html From egghunt at gmail.com Tue Feb 24 12:40:17 2009 From: egghunt at gmail.com (Arnaldo de Moraes Pereira) Date: Tue, 24 Feb 2009 17:40:17 -0300 Subject: [Freeswitch-users] Web-based forum? In-Reply-To: <87f2f3b90902241119x5fe2f1cek57b4ee42c6b47525@mail.gmail.com> References: <7.0.1.0.2.20090224200706.02496ca0@fredshack.com> <87f2f3b90902241119x5fe2f1cek57b4ee42c6b47525@mail.gmail.com> Message-ID: On Tue, Feb 24, 2009 at 4:19 PM, Michael Collins wrote: > > Maybe this question has been raised before, but if not: There's so > > much traffic in this mailing list that I was wondering if adding a > > web-based forum on the site was in the works? > > We are upgrading the freeswitch.org site soon to drupal 6.9. We are > considering turning on the forum feature there. No definitive decision > has been made but this request has come in several times. However, we > are trying to make it so that the devs don't have yet another place to > have to monitor for user questions, etc. so we will need to figure out > a way to make it easy to use for the experts... -1 for a forum. > > > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Arnaldo M Pereira -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090224/5caa564a/attachment.html From alexander at degreiff.com Tue Feb 24 13:09:50 2009 From: alexander at degreiff.com (Alexander de Greiff) Date: Tue, 24 Feb 2009 22:09:50 +0100 (CET) Subject: [Freeswitch-users] howto originate fs call from webapp (python) Message-ID: <13600111.821235509735422.JavaMail.alexanderdegreiff@AdG-Mac> hi all, i come from asterisk an i am new to freeswitch. after my with days with freeswitch i am very excited! but trying to migrate our deployment i have three challenges. one of them is: i need to call freeswitch from a webapp (e.g. python) and pass number1 and number2. i then need freeswitch to call number1. as soon as it is picked up say a short confirmaton text, call number2 and bridge the two. my first approach was to call via xml_rpc like described in the wiki but when i call like server.freeswitch.api("originate","sofia/gategay/gateway1/{number1} &bridge(sofia/gateway/gateway2/{number2})") but in this case both numbers are called in parallel and the first number to pick up gets a ringback tone until the other number picks up. how can i get the sequence described above? thanks for your help alex From freeswitch-users at lists.rupa.com Tue Feb 24 13:28:05 2009 From: freeswitch-users at lists.rupa.com (Rupa Schomaker (lists)) Date: Tue, 24 Feb 2009 15:28:05 -0600 Subject: [Freeswitch-users] howto originate fs call from webapp (python) In-Reply-To: <13600111.821235509735422.JavaMail.alexanderdegreiff@AdG-Mac> References: <13600111.821235509735422.JavaMail.alexanderdegreiff@AdG-Mac> Message-ID: <49A46665.5030904@lists.rupa.com> > my first approach was to call via xml_rpc like described in the wiki > but when i call like > > server.freeswitch.api("originate","sofia/gategay/gateway1/{number1} > &bridge(sofia/gateway/gateway2/{number2})") > > but in this case both numbers are called in parallel and the first > number to pick up gets a ringback tone until the other number picks > up. how can i get the sequence described above? > > thanks for your help alex You are probably getting early media when dialing number 1. Try : server.freeswitch.api("originate","{ignore_early_media=true}sofia/gategay/gateway1/{number1} &bridge(sofia/gateway/gateway2/{number2})") From msc at freeswitch.org Tue Feb 24 13:43:11 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 24 Feb 2009 13:43:11 -0800 Subject: [Freeswitch-users] howto originate fs call from webapp (python) In-Reply-To: <13600111.821235509735422.JavaMail.alexanderdegreiff@AdG-Mac> References: <13600111.821235509735422.JavaMail.alexanderdegreiff@AdG-Mac> Message-ID: <87f2f3b90902241343w68078c38oc54ffa4a88ad32de@mail.gmail.com> On Tue, Feb 24, 2009 at 1:09 PM, Alexander de Greiff wrote: > hi all, > > i come from asterisk an i am new to freeswitch. after my with days with freeswitch i am very excited! Welcome to FreeSWITCH! > > but trying to migrate our deployment i have three challenges. one of them is: > > i need to call freeswitch from a webapp (e.g. python) and pass number1 and number2. i then need freeswitch to call number1. as soon as it is picked up say a short confirmaton text, call number2 and bridge the two. > > my first approach was to call via xml_rpc like described in the wiki but when i call like > > ?server.freeswitch.api("originate","sofia/gategay/gateway1/{number1} &bridge(sofia/gateway/gateway2/{number2})") > > but in this case both numbers are called in parallel and the first number to pick up gets a ringback tone until the other number picks up. how can i get the sequence described above? > > thanks for your help > alex Do you have any other requirements? For example, what happens if the first bridge fails? Does your Python app need to "do anything"? Just curious. Thanks, MC From mszlazak at aol.com Tue Feb 24 22:51:51 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Wed, 25 Feb 2009 01:51:51 -0500 Subject: [Freeswitch-users] New build gives error message for default grammar file?? In-Reply-To: <89CDF21E-B363-49B4-BB71-22D97F2CF89F@freeswitch.org> References: <8CB64CE5BC62C9B-8D0-13DC@WEBMAIL-MC11.sysops.aol.com> <89CDF21E-B363-49B4-BB71-22D97F2CF89F@freeswitch.org> Message-ID: <8CB652CABEF943F-DE0-3729@WEBMAIL-DF13.sysops.aol.com> Hey Brian, Where abouts do you keep the Window MSI 1.0.3 build that isn't in SVN trunk. Installing from the wiki installation page gets me a build with the same error. Thanks. Mark. -----Original Message----- From: Brian West To: freeswitch-users at lists.freeswitch.org Sent: Tue, 24 Feb 2009 11:45 am Subject: Re: [Freeswitch-users] New build gives error message for default grammar file?? You're not in 1.0.3 you're in SVN trunk... The reason I know this is that wasn't changed till AFTER 1.0.3 was tagged and a new file set was used... please go into your libs dir and wipe out pocketsphinx and sphinx base.. then let it redownload them. ?I'll make you a new tarball of the new grammar files which are in the jsgf format. ?An example was added to scripts yes_no.gram ... word of warning the new pocketsphinx loads the entire dictionary on port open which on a machine of any speed should load the entire thing in 2 seconds or less... but here is the WARNING WARNING WARNING... If you happen to use words that aren't in the dictionary pocketsphinx WILL CRASH.. I am already on this with the dev's to work out a fix for this its been a long standing bug apparently in the lib. /b On Feb 24, 2009, at 1:36 PM, mszlazak at aol.com wrote: I'm getting this error message trying out the pizza demo in FS 1.0.3: ?"Can't open dictionary C:\Program Files\FreeSWITCH\grammar\default.dic" I didn't have this before where there was no default.dic file. Is there some place a path has to be set now? Thanks. = _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090225/fc339b8d/attachment.html From gmaruzz at celliax.org Tue Feb 24 23:49:52 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Wed, 25 Feb 2009 08:49:52 +0100 Subject: [Freeswitch-users] Skypiax (Skype Network endpoint) for FreeSWITCH In-Reply-To: <5800526b0902230826m255e0f4fmeeece95ed44e8cb4@mail.gmail.com> References: <7b197bef0902131015k456dee87x8f0f1b7059f1b49c@mail.gmail.com> <5800526b0902181853h32de5535l6c0f9d7067a69cf0@mail.gmail.com> <7b197bef0902191732i6fead849uace0ac906a9437b0@mail.gmail.com> <5800526b0902230826m255e0f4fmeeece95ed44e8cb4@mail.gmail.com> Message-ID: <7b197bef0902242349o36f153a4y583c5c76685a95e0@mail.gmail.com> On Mon, Feb 23, 2009 at 5:26 PM, Carlos Talbot wrote: > Were you planning to check in the sample skype.conf.xml into the default > FreeSWITCH conf folder? If so, just be aware the default config causes > freeswitch to hang right after a "load mod_skypiax" (if you do not have > skype running or specify a nonexistant skype user). Carlos, many thanks for reporting! I'll fix this this evening, if you have time to file a Jira for it would be wonderful. ciao for now, giovanni > > regards, > > > Carlos > On Thu, Feb 19, 2009 at 7:32 PM, Giovanni Maruzzelli > wrote: >> >> On Thu, Feb 19, 2009 at 3:53 AM, Carlos Talbot >> wrote: >> >> > One question I have, is ringback suppose to work with mod_skypiax? >> > Whenever >> > I dial a number I get a few seconds of dead air before the call is >> > answered. >> > I've tried adding ringback and transfer_ringback into the dialplan just >> > before the bridge command but no go. Am I missing something? Thanks. >> >> Carlos, >> >> ringback now works without tricks, and Skypiax is in trunk. >> >> Both remote ringing and early media are treated as remote ringing >> right now (eg: no early media, just ringing). >> >> I'll add early media support in the near future. >> >> Thanks a lot for testing and exercising skypiax, and please let me >> know any hint, suggestion, feature request, etc >> >> >> >> Sincerely, >> >> Giovanni Maruzzelli >> ========================================= >> www.celliax.org >> via Pierlombardo 9, 20135 Milano >> Italy >> gmaruzz at celliax dot org >> Cell : +39-347-2665618 >> Fax : +39-02-87390039 >> >> >> >> >> On Thu, Feb 19, 2009 at 3:53 AM, Carlos Talbot >> wrote: >> > Giovannia, >> > >> > great work on mod_skypiax. I've been testing it under Windows and it >> > sounds >> > great including PSTN calls. I plan to include it as part of the Windows >> > MSI >> > build. >> > >> > One question I have, is ringback suppose to work with mod_skypiax? >> > Whenever >> > I dial a number I get a few seconds of dead air before the call is >> > answered. >> > I've tried adding ringback and transfer_ringback into the dialplan just >> > before the bridge command but no go. Am I missing something? Thanks. >> > >> > regards, >> > >> > Carlos >> > >> > >> > >> > >> > _______________________________________________ >> > Freeswitch-users mailing list >> > Freeswitch-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From gmaruzz at celliax.org Tue Feb 24 23:55:23 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Wed, 25 Feb 2009 08:55:23 +0100 Subject: [Freeswitch-users] Skypiax, same skype user, multiple channels In-Reply-To: References: Message-ID: <7b197bef0902242355p1475be00l98e6a5e0596fe4ca@mail.gmail.com> On Tue, Feb 24, 2009 at 10:33 AM, Eric Chamberlain wrote: > I was reading through the Skypiax documentation and saw the comment > that it's not possible to run multiple skype clients on the same linux > machine, all using the same skype user account. > > It's possible to run multiple skype clients with the same skype user > account, as long as the skype clients are not accessing the same Skype > dbpath. > > We use runuser to run multiple skype clients. ?All the clients use the > same skype user, but each instance uses a different home directory, > each with its own .Skype folder. > > In such a configuration, will Skypiax support multiple channels using > the same skype username? Hi Eric, yes, definitely yes. If you give me more details I would like to integrate this use case both in the docs and in my testings. BTW: I'm about to move on your previous *very useful* suggestions and feature requests, please continue to send it :-) Sincerely, Giovanni Maruzzelli ========================================= www.celliax.org via Pierlombardo 9, 20135 Milano Italy gmaruzz at celliax dot org Cell : +39-347-2665618 Fax : +39-02-87390039 > > -- > Eric Chamberlain, Founder > RF.com - http://RF.com/ > > > > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From alexander at degreiff.com Wed Feb 25 00:30:47 2009 From: alexander at degreiff.com (Alexander de Greiff) Date: Wed, 25 Feb 2009 09:30:47 +0100 (CET) Subject: [Freeswitch-users] howto originate fs call from webapp (python) In-Reply-To: <29008165.8511235550562928.JavaMail.root@h1376493.stratoserver.net> Message-ID: <26996569.8531235550647802.JavaMail.root@h1376493.stratoserver.net> hi, oops, i must have been very tired when i wrote my first mail to the list... thanks for your replies. {ignore_early_media=true} really worked for me. i try very hard to "unlearn" asterisk. with asterisk i did not do much more with the python script, but i would like the pthon script to interact more with freeswitch like: - call number1 - say a welcome message with cepstral voice - call number2 - bridge other scenario: enter telephone number in webapp python script have fs to call number say "please enter the pin code from the website" validate dtmf code pass back to webapp: correct or not correct unfortunately just from reading the wiki i don't know how to do it in my python script. can you share your experience? thanks alex From alexander at degreiff.com Wed Feb 25 02:35:27 2009 From: alexander at degreiff.com (Alexander de Greiff) Date: Wed, 25 Feb 2009 11:35:27 +0100 (CET) Subject: [Freeswitch-users] problem speaking with cepstral voice in xml ivr menu In-Reply-To: <16711643.881235557974513.JavaMail.alexanderdegreiff@AdG-Mac.local> Message-ID: <3236119.901235558070484.JavaMail.alexanderdegreiff@AdG-Mac.local> hi all, here is my second problem trying to migrate from * to fs: i can speak with cepstral voices from my dialplan, but when i implement an ivr menu with cepstral voices like this: i get the following errors: [ERR] mod_native_file.c:68 native_file_file_open() Error opening /usr/local/freeswitch/sounds/en/us/callie/say:text to speak.GSM can you point me in the right direction? thanks alex --- freeswitch 1.0.3 build 12166 From anthony.minessale at gmail.com Wed Feb 25 06:09:02 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 25 Feb 2009 08:09:02 -0600 Subject: [Freeswitch-users] howto originate fs call from webapp (python) In-Reply-To: <26996569.8531235550647802.JavaMail.root@h1376493.stratoserver.net> References: <29008165.8511235550562928.JavaMail.root@h1376493.stratoserver.net> <26996569.8531235550647802.JavaMail.root@h1376493.stratoserver.net> Message-ID: <191c3a030902250609j777fqb4464c994dfa953@mail.gmail.com> from the freeswitch build root cd libs/esl if you have python-devel or the equiv make pymod from there if you cd python you will see a python module you can use to control freeswitch. On Wed, Feb 25, 2009 at 2:30 AM, Alexander de Greiff wrote: > hi, > > oops, i must have been very tired when i wrote my first mail to the list... > > thanks for your replies. {ignore_early_media=true} really worked for me. > > i try very hard to "unlearn" asterisk. > > with asterisk i did not do much more with the python script, but i would > like the pthon script to interact more with freeswitch like: > > - call number1 > - say a welcome message with cepstral voice > - call number2 > - bridge > > > other scenario: > > enter telephone number in webapp > python script have fs to call number > say "please enter the pin code from the website" > validate dtmf code > pass back to webapp: correct or not correct > > unfortunately just from reading the wiki i don't know how to do it in my > python script. > > can you share your experience? > > thanks > alex > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090225/cf04cd3d/attachment-0001.html From anthony.minessale at gmail.com Wed Feb 25 06:19:50 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 25 Feb 2009 08:19:50 -0600 Subject: [Freeswitch-users] Recording and outbound rtp In-Reply-To: <611594.9131235505720885.JavaMail.daniel@radio> References: <17338844.9091235504936529.JavaMail.daniel@radio> <611594.9131235505720885.JavaMail.daniel@radio> Message-ID: <191c3a030902250619i5496929ap1e0d5c0d40c2b6a@mail.gmail.com> We would have to code in a feature to purposely write silence back during a recording that does not currently exist. You could perhaps post it on the bounty section in jira. On Tue, Feb 24, 2009 at 2:02 PM, wrote: > no, I'm matching the incoming sip call via the destination number in my > public context and executing the javascript appliaction. This app directly > answers the call and records it until the user hangs up. > D- > > > ----- Original Message ----- > From: "Anthony Minessale" > To: freeswitch-users at lists.freeswitch.org > Sent: Tuesday, February 24, 2009 12:05:33 PM GMT -07:00 US/Canada Mountain > Subject: Re: [Freeswitch-users] Recording and outbound rtp > > is it during a bridged call? > > > On Tue, Feb 24, 2009 at 11:49 AM, Dan wrote: > >> Hi, >> >> I have a small javascript application that accepts a call, retrieves some >> dtmf digits and then records the call to an icecast server. This works >> great. >> >> The problem I'm having is that when the call is being recorded freeswitch >> is no longer sending rtp packets back to the originating caller, in my case >> a Cisco 5300 with a bunch of T1 voice circuits in it. This makes sense, >> since no voice data back is being generated. Unfortunately my Cisco gear >> has rtp inactivity timers set up to hang up a call after 3 minutes of no >> incoming rtp packets, this is a global setting that cannot be configured for >> a single dial peer. Does anyone have a suggestion to generate rtp packets >> every once in a while? I tried setting comfort noise which did not seem to >> send anything. I could try playing a empty/short wav file every minute or >> so but the javascript call session.record is blocking, would a traditional >> javascript timer and callback to play a wav file be my best bet or is there >> a better approach? I'm using FreeSWITCH Version 1.0.trunk (12108M) on debian >> etch. >> >> Thanks! >> Dan- >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ Freeswitch-users mailing > list Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090225/133784e9/attachment.html From jforman at wcgltd.com Wed Feb 25 07:22:24 2009 From: jforman at wcgltd.com (Josh Forman) Date: Wed, 25 Feb 2009 10:22:24 -0500 Subject: [Freeswitch-users] Adding an info digit to sip from header Message-ID: <63C69E8D-3ED4-4FEC-8F21-2738A1A194DC@wcgltd.com> I'm trying to edit the sip headers to make the from field look like this: From: ;tag=gK0a00d6ea. I know that to read that data on an incoming sip message it is in $ {sip_from_params}, but how can I add the ;isup-oli=27 part on an outgoing message? Thanks Josh From brian at freeswitch.org Wed Feb 25 07:29:26 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 25 Feb 2009 09:29:26 -0600 Subject: [Freeswitch-users] Adding an info digit to sip from header In-Reply-To: <63C69E8D-3ED4-4FEC-8F21-2738A1A194DC@wcgltd.com> References: <63C69E8D-3ED4-4FEC-8F21-2738A1A194DC@wcgltd.com> Message-ID: <0E1E9F62-64C5-4B70-9C67-C7B5728DB111@freeswitch.org> You can do something like this "sofia/blah/somenumber at someip: 5060;this=rocks" /b On Feb 25, 2009, at 9:22 AM, Josh Forman wrote: > I'm trying to edit the sip headers to make the from field look like > this: > > From: ;tag=gK0a00d6ea. > > I know that to read that data on an incoming sip message it is in $ > {sip_from_params}, but how can I add the ;isup-oli=27 part on an > outgoing message? > > Thanks > > Josh -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090225/13b8feaf/attachment.html From freeswitch-users at digitaldan.com Wed Feb 25 08:00:45 2009 From: freeswitch-users at digitaldan.com (Dan) Date: Wed, 25 Feb 2009 09:00:45 -0700 (MST) Subject: [Freeswitch-users] Recording and outbound rtp In-Reply-To: <191c3a030902250619i5496929ap1e0d5c0d40c2b6a@mail.gmail.com> Message-ID: <18165784.9951235577626769.JavaMail.daniel@radio> Thanks, I will look around and see if I can come up with a solution. I'll post back here and on the wiki if I find one. D- ----- Original Message ----- From: "Anthony Minessale" To: freeswitch-users at lists.freeswitch.org Sent: Wednesday, February 25, 2009 7:19:50 AM GMT -07:00 US/Canada Mountain Subject: Re: [Freeswitch-users] Recording and outbound rtp We would have to code in a feature to purposely write silence back during a recording that does not currently exist. You could perhaps post it on the bounty section in jira. On Tue, Feb 24, 2009 at 2:02 PM, < freeswitch-users at digitaldan.com > wrote: no, I'm matching the incoming sip call via the destination number in my public context and executing the javascript appliaction. This app directly answers the call and records it until the user hangs up. D- ----- Original Message ----- From: "Anthony Minessale" < anthony.minessale at gmail.com > To: freeswitch-users at lists.freeswitch.org Sent: Tuesday, February 24, 2009 12:05:33 PM GMT -07:00 US/Canada Mountain Subject: Re: [Freeswitch-users] Recording and outbound rtp is it during a bridged call? On Tue, Feb 24, 2009 at 11:49 AM, Dan < freeswitch-users at digitaldan.com > wrote: Hi, I have a small javascript application that accepts a call, retrieves some dtmf digits and then records the call to an icecast server. This works great. The problem I'm having is that when the call is being recorded freeswitch is no longer sending rtp packets back to the originating caller, in my case a Cisco 5300 with a bunch of T1 voice circuits in it. This makes sense, since no voice data back is being generated. Unfortunately my Cisco gear has rtp inactivity timers set up to hang up a call after 3 minutes of no incoming rtp packets, this is a global setting that cannot be configured for a single dial peer. Does anyone have a suggestion to generate rtp packets every once in a while? I tried setting comfort noise which did not seem to send anything. I could try playing a empty/short wav file every minute or so but the javascript call session.record is blocking, would a traditional javascript timer and callback to play a wav file be my best bet or is there a better approach? I'm using FreeSWITCH Version 1.0.trunk (12108M) on debian etch. Thanks! Dan- _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/ PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/ PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090225/4a33ace5/attachment-0001.html From jforman at wcgltd.com Wed Feb 25 08:34:08 2009 From: jforman at wcgltd.com (Josh Forman) Date: Wed, 25 Feb 2009 11:34:08 -0500 Subject: [Freeswitch-users] Adding an info digit to sip from header In-Reply-To: References: Message-ID: <3E49EAB9-0333-4E99-AC99-9BAE34CAC500@wcgltd.com> Which variable would I need to set via the dialplan to do this though? Your example looks like it would be the dialstring for the bridge application but if that works it would probably be added to the To header instead of the From, right? I can't be sure since nothing I've tried has had any affect. Between looking at the wiki and random experimenting I haven't found anything that works thus far. On Feb 25, 2009, at 11:01 AM, freeswitch-users-request at lists.freeswitch.org wrote: > Message: 3 > Date: Wed, 25 Feb 2009 09:29:26 -0600 > From: Brian West > Subject: Re: [Freeswitch-users] Adding an info digit to sip from > header > To: freeswitch-users at lists.freeswitch.org > Message-ID: <0E1E9F62-64C5-4B70-9C67-C7B5728DB111 at freeswitch.org> > Content-Type: text/plain; charset="us-ascii" > > You can do something like this "sofia/blah/somenumber at someip: > 5060;this=rocks" > > /b > > On Feb 25, 2009, at 9:22 AM, Josh Forman wrote: > >> I'm trying to edit the sip headers to make the from field look like >> this: >> >> From: ;tag=gK0a00d6ea. >> >> I know that to read that data on an incoming sip message it is in $ >> {sip_from_params}, but how can I add the ;isup-oli=27 part on an >> outgoing message? >> >> Thanks >> >> Josh From brian at freeswitch.org Wed Feb 25 08:50:17 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 25 Feb 2009 10:50:17 -0600 Subject: [Freeswitch-users] problem speaking with cepstral voice in xml ivr menu In-Reply-To: <3236119.901235558070484.JavaMail.alexanderdegreiff@AdG-Mac.local> References: <3236119.901235558070484.JavaMail.alexanderdegreiff@AdG-Mac.local> Message-ID: Alex, If you want to update to svn trunk the tts-engine and tts-voice are now valid options on the menu. They were not before (But the wiki said they were). So to cut confusion I made them work... if you do not wish to upgrade you'll need to set the tts_engine and tts_voice variables before you call the IVR application and it will work with the code you already have. I highly recommend a "make curret" ;) Committed revision 12278. /b On Feb 25, 2009, at 4:35 AM, Alexander de Greiff wrote: > hi all, > > here is my second problem trying to migrate from * to fs: > > i can speak with cepstral voices from my dialplan, but when i > implement an ivr menu with cepstral voices like this: > > greet-long="say:text to speak" > greet-short="say:main menu" > invalid-sound="say:invalid entry" > exit-sound="say:goodbye" > timeout ="10000" > max-failures="3" > tts-engine="cepstral" > tts-voice="allison" > phrase_lang="en"> > > > > > > i get the following errors: > > [ERR] mod_native_file.c:68 native_file_file_open() Error opening / > usr/local/freeswitch/sounds/en/us/callie/say:text to speak.GSM > > > can you point me in the right direction? > > thanks > alex > > > --- > freeswitch 1.0.3 build 12166 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Wed Feb 25 08:53:39 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 25 Feb 2009 10:53:39 -0600 Subject: [Freeswitch-users] Adding an info digit to sip from header In-Reply-To: <3E49EAB9-0333-4E99-AC99-9BAE34CAC500@wcgltd.com> References: <3E49EAB9-0333-4E99-AC99-9BAE34CAC500@wcgltd.com> Message-ID: It will actually add it to both places INVITE sip:1235 at conference.freeswitch.org;this=rocks SIP/2.0 Via: SIP/2.0/UDP 99.185.85.3;rport;branch=z9hG4bK0Kaa1322U42eK Max-Forwards: 69 From: "1004" ;tag=1SparjgraS69m To: I verified it does indeed add it in both places. /b On Feb 25, 2009, at 10:34 AM, Josh Forman wrote: > Which variable would I need to set via the dialplan to do this > though? Your example looks like it would be the dialstring for the > bridge application but if that works it would probably be added to the > To header instead of the From, right? I can't be sure since nothing > I've tried has had any affect. > Between looking at the wiki and random experimenting I haven't found > anything that works thus far. From msc at freeswitch.org Wed Feb 25 10:11:30 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 25 Feb 2009 10:11:30 -0800 Subject: [Freeswitch-users] howto originate fs call from webapp (python) In-Reply-To: <26996569.8531235550647802.JavaMail.root@h1376493.stratoserver.net> References: <29008165.8511235550562928.JavaMail.root@h1376493.stratoserver.net> <26996569.8531235550647802.JavaMail.root@h1376493.stratoserver.net> Message-ID: <87f2f3b90902251011l3bc1a806gdcc75a781e090ec9@mail.gmail.com> > enter telephone number in webapp > python script have fs to call number > say "please enter the pin code from the website" > validate dtmf code > pass back to webapp: correct or not correct > > unfortunately just from reading the wiki i don't know how to do it in my python script. > > can you share your experience? You definitely need to become familiar with the event socket. However, to become familiar with the event socket you need also to become familiar with some of the basic FreeSWITCH API functions, like "bgapi" and "originate" as well as what kinds of events come over the event socket. Here is some recommended reading: #1 - The reporting bugs page on the wiki. It may sound crazy, but I promise you that if you at least skim over it then it will save you time when you start having to debug things. http://wiki.freeswitch.org/wiki/Reporting_Bugs #2 - The event socket page on the wiki: http://wiki.freeswitch.org/wiki/Mod_event_socket #3 - The commands page on the wiki. Pay special attention to the "originate," "bridge," and "bgapi" commands because they will be extremely useful to you in your application: http://wiki.freeswitch.org/wiki/Mod_commands #4 - The Asterisk/FreeSWITCH Rosetta Stone wiki page. In some cases you can leverage your Asterisk knowledge. This page gives you some tips on how to do stuff in FS that you already know how to do with Asterisk: http://wiki.freeswitch.org/wiki/Rosetta_stone You have lots of reading to do! :) You will also need to start doing test phone calls. Make test calls and see how things work. Watch the debug information on the CLI to see what FS is doing with each call. It's very interesting. Join us on IRC when you have questions and want to talk in real-time. -MC (IRC: mercutioviz) From alexander at degreiff.com Wed Feb 25 10:27:34 2009 From: alexander at degreiff.com (Alexander de Greiff) Date: Wed, 25 Feb 2009 19:27:34 +0100 (CET) Subject: [Freeswitch-users] problem speaking with cepstral voice in xml ivr menu In-Reply-To: Message-ID: <7349336.921235586397898.JavaMail.alexanderdegreiff@AdG-Mac> brian, thanks for your help. i really apreciate the active support. upgrading to the current trunk solved the problem. i can hear the cepstal voices in the ivr menus now. (with the current trunk i have all sorts of other compile problems (mod_fax, python) but i will work this out following the build instructions again). i only wonder because these worked with my last version of last week. so far i am a happy camper with freeswitch. this is a different snack bracket than asterisk... kind regards alex ----- Urspr?ngliche Mail ----- Alex, If you want to update to svn trunk the tts-engine and tts-voice are now valid options on the menu. They were not before (But the wiki said they were). So to cut confusion I made them work... if you do not wish to upgrade you'll need to set the tts_engine and tts_voice variables before you call the IVR application and it will work with the code you already have. I highly recommend a "make curret" ;) Committed revision 12278. /b On Feb 25, 2009, at 4:35 AM, Alexander de Greiff wrote: > hi all, > > here is my second problem trying to migrate from * to fs: > > i can speak with cepstral voices from my dialplan, but when i > implement an ivr menu with cepstral voices like this: > > greet-long="say:text to speak" > greet-short="say:main menu" > invalid-sound="say:invalid entry" > exit-sound="say:goodbye" > timeout ="10000" > max-failures="3" > tts-engine="cepstral" > tts-voice="allison" > phrase_lang="en"> > > > > > > i get the following errors: > > [ERR] mod_native_file.c:68 native_file_file_open() Error opening / > usr/local/freeswitch/sounds/en/us/callie/say:text to speak.GSM > > > can you point me in the right direction? > > thanks > alex > > > --- > freeswitch 1.0.3 build 12166 > From msc at freeswitch.org Wed Feb 25 10:28:00 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 25 Feb 2009 10:28:00 -0800 Subject: [Freeswitch-users] Question about BLF... In-Reply-To: References: Message-ID: <87f2f3b90902251028k1079bdb8s436e81a856c6256e@mail.gmail.com> Klaus, Can you update us on where you are with this? -MC On Sun, Feb 22, 2009 at 4:04 PM, Klaus Hochlehnert wrote: > Hi, > > I'm just playing around with FreeSWITCH and I have 2 questions about BLF > (with SNOM phones): > > - When I played around with the sample dial plan I found out that BLF works > better than Asterisk, but not 100% right: > ?> When phone 1000 gets a call the BLF lamp on phone 1001 blinks and after > phone 1000 takes the call the lamp on phone 1001 is on > ?> But when phone 1000 gets a second call, takes it and hangs up the lamp > on phone 1001 turns off even if the first call is still active > ?> Is that a problem or did I do something wrong??? > > > - Second question is how can I set up BLF if I want to have my dial plan > completely in a perl script (no XML besides calling the perl script)? > > Thanks, Klaus > From msc at freeswitch.org Wed Feb 25 10:31:24 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 25 Feb 2009 10:31:24 -0800 Subject: [Freeswitch-users] problem speaking with cepstral voice in xml ivr menu In-Reply-To: <7349336.921235586397898.JavaMail.alexanderdegreiff@AdG-Mac> References: <7349336.921235586397898.JavaMail.alexanderdegreiff@AdG-Mac> Message-ID: <87f2f3b90902251031t627f3822w7fa8733cfb52f7b8@mail.gmail.com> > so far i am a happy camper with freeswitch. this is a different snack bracket than asterisk... If you don't mind telling us, where did you hear about FS and what made you decide to try it? Are you unhappy with Asterisk or are you simply looking for something a bit different? Just curious. Thanks, MC From brian at freeswitch.org Wed Feb 25 11:10:09 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 25 Feb 2009 13:10:09 -0600 Subject: [Freeswitch-users] problem speaking with cepstral voice in xml ivr menu In-Reply-To: <7349336.921235586397898.JavaMail.alexanderdegreiff@AdG-Mac> References: <7349336.921235586397898.JavaMail.alexanderdegreiff@AdG-Mac> Message-ID: Can you report these issues in jira? http://jira.freeswitch.org /b On Feb 25, 2009, at 12:27 PM, Alexander de Greiff wrote: > (with the current trunk i have all sorts of other compile problems > (mod_fax, python) but i will work this out following the build > instructions again). From rex.alex345 at yahoo.com Wed Feb 25 10:42:31 2009 From: rex.alex345 at yahoo.com (Rex Alex) Date: Wed, 25 Feb 2009 10:42:31 -0800 (PST) Subject: [Freeswitch-users] ESL Wrapper Message-ID: <558004.60211.qm@web59511.mail.ac4.yahoo.com> Hi All, I am new to freeswitch but installed(freeswitch version 1.0.3), configured and tested successfully. Now I want to do the dialling funtions through a php script. Read about Event Socket Library(ESL). How to implement the same in freeswitch. Please assist. Thanks, Rex From mrene_lists at avgs.ca Wed Feb 25 11:31:37 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Wed, 25 Feb 2009 14:31:37 -0500 Subject: [Freeswitch-users] ESL Wrapper In-Reply-To: <558004.60211.qm@web59511.mail.ac4.yahoo.com> References: <558004.60211.qm@web59511.mail.ac4.yahoo.com> Message-ID: <78A818D3-6C4F-420D-A922-751A69E7E080@avgs.ca> FreeSWITCH will listen on a socket allowing clients to send commands / receive events. ESL is a library to ease the creation of applications connecting to that socket. To install the php ESL module, cd into libs/esl and type "make phpmod" A sample php file is included in the libs/esl/php directory. Mathieu On 25-Feb-09, at 1:42 PM, Rex Alex wrote: > > Hi All, > > I am new to freeswitch but installed(freeswitch version 1.0.3), > configured and tested successfully. Now I want to do the dialling > funtions through a php script. Read about Event Socket Library(ESL). > How to implement the same in freeswitch. > > Please assist. > > Thanks, > Rex > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Wed Feb 25 11:34:59 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 25 Feb 2009 13:34:59 -0600 Subject: [Freeswitch-users] ESL Wrapper In-Reply-To: <78A818D3-6C4F-420D-A922-751A69E7E080@avgs.ca> References: <558004.60211.qm@web59511.mail.ac4.yahoo.com> <78A818D3-6C4F-420D-A922-751A69E7E080@avgs.ca> Message-ID: <6CE58813-715A-43DB-877B-638B5CE7E6E9@freeswitch.org> If he's on 1.0.3 I don't think it has php in it.. /b On Feb 25, 2009, at 1:31 PM, Mathieu Rene wrote: > FreeSWITCH will listen on a socket allowing clients to send commands / > receive events. > ESL is a library to ease the creation of applications connecting to > that socket. > > To install the php ESL module, cd into libs/esl and type "make phpmod" > > A sample php file is included in the libs/esl/php directory. > > Mathieu From jforman at wcgltd.com Wed Feb 25 11:51:00 2009 From: jforman at wcgltd.com (Josh Forman) Date: Wed, 25 Feb 2009 14:51:00 -0500 Subject: [Freeswitch-users] Adding an info digit to sip from In-Reply-To: References: Message-ID: The problem here is that what you are showing me produces: From: "1004" ;tag=1SparjgraS69m To: when what I need to output would look like this: From: "1004" ;tag=1SparjgraS69m To: with the "this=rocks" in the FROM field, not the TO field. I know that you can change parts of the from field by setting effective_caller_id_name and effective_caller_id_number, but I don't know how I would add that bit of data to the end of the SIP URI inside the < > Is there a variable that I could set or perhaps some method similar to overwriting the To header shown at http://wiki.freeswitch.org/wiki/Sofia#Modifying_the_To :_header that can be used to accomplish this? Thanks Josh On Feb 25, 2009, at 2:35 PM, freeswitch-users-request at lists.freeswitch.org wrote: >> Message: 3 >> Date: Wed, 25 Feb 2009 09:29:26 -0600 >> From: Brian West >> Subject: Re: [Freeswitch-users] Adding an info digit to sip from >> header >> To: freeswitch-users at lists.freeswitch.org >> Message-ID: <0E1E9F62-64C5-4B70-9C67-C7B5728DB111 at freeswitch.org> >> Content-Type: text/plain; charset="us-ascii" >> >> You can do something like this "sofia/blah/somenumber at someip: >> 5060;this=rocks" >> >> /b >> >> On Feb 25, 2009, at 9:22 AM, Josh Forman wrote: >> >>> I'm trying to edit the sip headers to make the from field look like >>> this: >>> >>> From: ;tag=gK0a00d6ea. >>> >>> I know that to read that data on an incoming sip message it is in $ >>> {sip_from_params}, but how can I add the ;isup-oli=27 part on an >>> outgoing message? >>> >>> Thanks >>> >>> Josh > > > > ------------------------------ > > Message: 3 > Date: Wed, 25 Feb 2009 10:53:39 -0600 > From: Brian West > Subject: Re: [Freeswitch-users] Adding an info digit to sip from > header > To: freeswitch-users at lists.freeswitch.org > Message-ID: > Content-Type: text/plain; charset=US-ASCII; format=flowed > > It will actually add it to both places > > INVITE sip:1235 at conference.freeswitch.org;this=rocks SIP/2.0 > Via: SIP/2.0/UDP 99.185.85.3;rport;branch=z9hG4bK0Kaa1322U42eK > Max-Forwards: 69 > From: "1004" ;tag=1SparjgraS69m > To: > > I verified it does indeed add it in both places. > > /b > > > > On Feb 25, 2009, at 10:34 AM, Josh Forman wrote: > >> Which variable would I need to set via the dialplan to do this >> though? Your example looks like it would be the dialstring for the >> bridge application but if that works it would probably be added to >> the >> To header instead of the From, right? I can't be sure since nothing >> I've tried has had any affect. >> Between looking at the wiki and random experimenting I haven't found >> anything that works thus far. > From alexander at degreiff.com Wed Feb 25 12:12:03 2009 From: alexander at degreiff.com (Alexander de Greiff) Date: Wed, 25 Feb 2009 21:12:03 +0100 (CET) Subject: [Freeswitch-users] problem speaking with cepstral voice in xml ivr menu In-Reply-To: <1262470.981235592644063.JavaMail.alexanderdegreiff@AdG-Mac> Message-ID: <9100461.1001235592664872.JavaMail.alexanderdegreiff@AdG-Mac> michael, i googled for "asterisk alternative" and voila... the trigger was that every now and then i renew servers in the infrastructure and the one with asterisk was overdue. i wasn't really unhappy with asterisk, but these things bothered me (maybe i am not up to date): - dialplan gets messy - no conferences without hardware (rented remote server!) - ivr with cepstral voices: sometimes get hickups so far i like the fs approach very much. stable sip channels, no hickups with voices. kind regards alex ----- Urspr?ngliche Mail ----- Von: "Michael Collins" An: freeswitch-users at lists.freeswitch.org Gesendet: Mittwoch, 25. Februar 2009 19:31:24 GMT +01:00 Amsterdam/Berlin/Bern/Rom/Stockholm/Wien Betreff: Re: [Freeswitch-users] problem speaking with cepstral voice in xml ivr menu > so far i am a happy camper with freeswitch. this is a different snack bracket than asterisk... If you don't mind telling us, where did you hear about FS and what made you decide to try it? Are you unhappy with Asterisk or are you simply looking for something a bit different? Just curious. Thanks, MC _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From sprice at gmail.com Wed Feb 25 12:13:53 2009 From: sprice at gmail.com (SP) Date: Wed, 25 Feb 2009 14:13:53 -0600 Subject: [Freeswitch-users] Adding an info digit to sip from In-Reply-To: References: Message-ID: <7e2ac3270902251213j39bd8d9bme98172fe44305303@mail.gmail.com> On Wed, Feb 25, 2009 at 13:51, Josh Forman wrote: > The problem here is that what you are showing me produces: > > From: "1004" ;tag=1SparjgraS69m > To: > > when what I need to output would look like this: > > From: "1004" ;tag=1SparjgraS69m > To: > > with the "this=rocks" in the FROM field, not the TO field. > I know that you can change parts of the from field by setting > effective_caller_id_name and effective_caller_id_number, but I don't > know how I would add that bit of data to the end of the SIP URI inside > the < > > Is there a variable that I could set or perhaps some method similar to > overwriting the To header shown at http://wiki.freeswitch.org/wiki/Sofia#Modifying_the_To > :_header that can be used to accomplish this? > > Thanks > Josh > > On Feb 25, 2009, at 2:35 PM, freeswitch-users-request at lists.freeswitch.org > ?wrote: > >>> Message: 3 >>> Date: Wed, 25 Feb 2009 09:29:26 -0600 >>> From: Brian West >>> Subject: Re: [Freeswitch-users] Adding an info digit to sip from >>> ? ? ? header >>> To: freeswitch-users at lists.freeswitch.org >>> Message-ID: <0E1E9F62-64C5-4B70-9C67-C7B5728DB111 at freeswitch.org> >>> Content-Type: text/plain; charset="us-ascii" >>> >>> You can do something like this "sofia/blah/somenumber at someip: >>> 5060;this=rocks" >>> >>> /b >>> >>> On Feb 25, 2009, at 9:22 AM, Josh Forman wrote: >>> >>>> I'm trying to edit the sip headers to make the from field look like >>>> this: >>>> >>>> From: ;tag=gK0a00d6ea. >>>> >>>> I know that to read that data on an incoming sip message it is in $ >>>> {sip_from_params}, but how can I add the ;isup-oli=27 part on an >>>> outgoing message? >>>> >>>> Thanks >>>> >>>> Josh >> >> >> >> ------------------------------ >> >> Message: 3 >> Date: Wed, 25 Feb 2009 10:53:39 -0600 >> From: Brian West >> Subject: Re: [Freeswitch-users] Adding an info digit to sip from >> ? ? ? ?header >> To: freeswitch-users at lists.freeswitch.org >> Message-ID: >> Content-Type: text/plain; charset=US-ASCII; format=flowed >> >> It will actually add it to both places >> >> INVITE sip:1235 at conference.freeswitch.org;this=rocks SIP/2.0 >> Via: SIP/2.0/UDP 99.185.85.3;rport;branch=z9hG4bK0Kaa1322U42eK >> Max-Forwards: 69 >> From: "1004" ;tag=1SparjgraS69m >> To: >> >> I verified it does indeed add it in both places. >> >> /b >> >> >> >> On Feb 25, 2009, at 10:34 AM, Josh Forman wrote: >> >>> Which variable would I need to set via the dialplan to do this >>> though? ?Your example looks like it would be the dialstring for the >>> bridge application but if that works it would probably be added to >>> the >>> To header instead of the From, right? ?I can't be sure since nothing >>> I've tried has had any affect. >>> Between looking at the wiki and random experimenting I haven't found >>> anything that works thus far. >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Shannon From brian at freeswitch.org Wed Feb 25 12:27:12 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 25 Feb 2009 14:27:12 -0600 Subject: [Freeswitch-users] Adding an info digit to sip from In-Reply-To: <7e2ac3270902251213j39bd8d9bme98172fe44305303@mail.gmail.com> References: <7e2ac3270902251213j39bd8d9bme98172fe44305303@mail.gmail.com> Message-ID: SP, That won't go into the from. You can't add params to the from unless you have svn rev 12287 or higher. I added the ability to set "sip_invite_params, sip_invite_to_params, sip_invite_from_params" to sofia_glue.c, I added two lines and changed two lines to make this possible. So to be clear: sip_invite_params will set params on the request URI, sip_invite_to_params will set params on the to URI, sip_invite_from_params will set params on the from URI. (someone wiki this) http://fisheye.freeswitch.org/browse/FreeSWITCH/src/mod/endpoints/mod_sofia/sofia_glue.c?r1=12235&r2=12287 You brought up a good point that it wasn't possible but when I looked at the code it was fairly simple to add support for it so I did. Please check out that Donate button on the home page! ;) /b PS: Its MikeJ's birthday today! On Feb 25, 2009, at 2:13 PM, SP wrote: > data="sip_invite_domain=some.domain;this=rocks"/> > From brian at freeswitch.org Wed Feb 25 12:34:14 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 25 Feb 2009 14:34:14 -0600 Subject: [Freeswitch-users] Adding an info digit to sip from In-Reply-To: <7e2ac3270902251213j39bd8d9bme98172fe44305303@mail.gmail.com> References: <7e2ac3270902251213j39bd8d9bme98172fe44305303@mail.gmail.com> Message-ID: <5E93A503-391B-4B4D-B5D1-27CA1A147A33@freeswitch.org> I also realized I broke backwards compatibility for anyone using sip_invite_params so I corrected that in rev 12288 /b On Feb 25, 2009, at 2:13 PM, SP wrote: > data="sip_invite_domain=some.domain;this=rocks"/> From kristian.kielhofner at gmail.com Wed Feb 25 13:00:55 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Wed, 25 Feb 2009 16:00:55 -0500 Subject: [Freeswitch-users] SheevaPlug Development Kit Message-ID: <2d9149cd0902251300y336e5e35hb2b9a1f9d30d6f3f@mail.gmail.com> Hello everyone, I just ordered one of these: http://www.marvell.com/products/embedded_processors/developer/kirkwood/sheevaplug.jsp Just over $110 with shipping but they are expecting the price to come down quite a bit (perhaps as low as $50): - 1.2Ghz ARM5 - 512MB RAM - Multiple flash storage options - Gigabit ethernet - USB 2.0 - 5 watt power usage They probably won't be shipping until late March but I thought I'd get my order in early. Who knows how practical it will be but needless to say I'm going to get FreeSWITCH to run on it! -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From brian at freeswitch.org Wed Feb 25 13:07:44 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 25 Feb 2009 15:07:44 -0600 Subject: [Freeswitch-users] SheevaPlug Development Kit In-Reply-To: <2d9149cd0902251300y336e5e35hb2b9a1f9d30d6f3f@mail.gmail.com> References: <2d9149cd0902251300y336e5e35hb2b9a1f9d30d6f3f@mail.gmail.com> Message-ID: I seen that yesterday... looks interesting. /b On Feb 25, 2009, at 3:00 PM, Kristian Kielhofner wrote: > Hello everyone, > > I just ordered one of these: > > http://www.marvell.com/products/embedded_processors/developer/kirkwood/sheevaplug.jsp > > Just over $110 with shipping but they are expecting the price to > come down quite a bit (perhaps as low as $50): > > - 1.2Ghz ARM5 > - 512MB RAM > - Multiple flash storage options > - Gigabit ethernet > - USB 2.0 > - 5 watt power usage > > They probably won't be shipping until late March but I thought I'd > get my order in early. > > Who knows how practical it will be but needless to say I'm going to > get FreeSWITCH to run on it! > > -- > Kristian Kielhofner > http://blog.krisk.org > http://www.submityoursip.com > http://www.astlinux.org > http://www.star2star.com > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From erik at erikwickstrom.com Wed Feb 25 13:11:23 2009 From: erik at erikwickstrom.com (Erik Wickstrom) Date: Wed, 25 Feb 2009 13:11:23 -0800 Subject: [Freeswitch-users] Thin Client VOIP setup? Message-ID: <3d381e170902251311i3d3a4205j117d472228c30219@mail.gmail.com> Hi, I've deployed Freeswitch as our phone system at work. We now want to use our new phonesystem in a phone room with thin clients (Terminal Server, possibly LTSP) for each agent. Ideally, we'd like to use x-lite or another softphone for each agent. The desired workflow for the agents is as follows: 1) A web based CRM with click to dial. (and customer data card etc) 2) Agent clicks dial button and is connected to customer 3) Interact with CRM... >From what I've read so far, there are some challenges that need to be overcome in deploying softphones over thin clients. Has anyone here had any success in setting up a system like this? I'm I asking for trouble trying to use softphones with thin clients (should I just use hardware phones? Do they support click to dial?) Thanks! Erik -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090225/a80bad4b/attachment.html From msc at freeswitch.org Wed Feb 25 13:42:14 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 25 Feb 2009 13:42:14 -0800 Subject: [Freeswitch-users] ESL Wrapper In-Reply-To: <6CE58813-715A-43DB-877B-638B5CE7E6E9@freeswitch.org> References: <558004.60211.qm@web59511.mail.ac4.yahoo.com> <78A818D3-6C4F-420D-A922-751A69E7E080@avgs.ca> <6CE58813-715A-43DB-877B-638B5CE7E6E9@freeswitch.org> Message-ID: <87f2f3b90902251342q1e393127ha7fbdbebf6d83dac@mail.gmail.com> On Wed, Feb 25, 2009 at 11:34 AM, Brian West wrote: > If he's on 1.0.3 I don't think it has php in it.. Can't he do the whole bootstrap process? svn up && ./bootstrap.sh && ./configure && make install And then do Mathieu's suggestion? -MC From mashudiflexi at telkom.co.id Wed Feb 25 20:21:37 2009 From: mashudiflexi at telkom.co.id (mashudi) Date: Thu, 26 Feb 2009 11:21:37 +0700 Subject: [Freeswitch-users] Session Timer Message-ID: <49A618D1.2070900@telkom.co.id> Hi Folks, in case of FreeSwitch sip message response for UPDATE message wih SIP/2.0 200 OK, how to change the session timer value from 120 to 300 ? Session-Expires: 120;refresher=uac. thank's for help and suggestion. mashudi ***************************************** Sekarang Gratis Nelpon SLJJ Flexi diperluas ke Yogya ***************************************** From mrene_lists at avgs.ca Wed Feb 25 20:10:23 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Wed, 25 Feb 2009 23:10:23 -0500 Subject: [Freeswitch-users] Session Timer In-Reply-To: <49A618D1.2070900@telkom.co.id> References: <49A618D1.2070900@telkom.co.id> Message-ID: <21585264-712C-4673-BD91-43D63435330A@avgs.ca> In the sip profile: Math On 25-Feb-09, at 11:21 PM, mashudi wrote: > Hi Folks, > > in case of FreeSwitch sip message response for UPDATE message wih > SIP/2.0 200 OK, how to change the session timer value from 120 to > 300 ? > > Session-Expires: 120;refresher=uac. > > thank's for help and suggestion. > > mashudi > > > ***************************************** > Sekarang Gratis Nelpon SLJJ Flexi diperluas ke > Yogya > ***************************************** > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mashudiflexi at telkom.co.id Wed Feb 25 20:43:28 2009 From: mashudiflexi at telkom.co.id (mashudi) Date: Thu, 26 Feb 2009 11:43:28 +0700 Subject: [Freeswitch-users] Session Timer In-Reply-To: <21585264-712C-4673-BD91-43D63435330A@avgs.ca> References: <49A618D1.2070900@telkom.co.id> <21585264-712C-4673-BD91-43D63435330A@avgs.ca> Message-ID: <49A61DF0.8070204@telkom.co.id> Dear Mathieu Rene, Thanks for you response. I already change the value in the sip profile to 300 as prerquisite by our external gateway, for responds to INVITE message it's work, but no for response to UPDATE message, the session-timer still use default value, namely 120. Mathieu Rene wrote: > In the sip profile: > > > > Math > > On 25-Feb-09, at 11:21 PM, mashudi wrote: > > >> Hi Folks, >> >> in case of FreeSwitch sip message response for UPDATE message wih >> SIP/2.0 200 OK, how to change the session timer value from 120 to >> 300 ? >> >> Session-Expires: 120;refresher=uac. >> >> thank's for help and suggestion. >> >> mashudi >> >> >> ***************************************** >> Sekarang Gratis Nelpon SLJJ Flexi diperluas ke >> Yogya >> ***************************************** >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > ***************************************** > Sekarang Gratis Nelpon SLJJ Flexi diperluas ke > Yogya > ***************************************** > > ***************************************** Sekarang Gratis Nelpon SLJJ Flexi diperluas ke Yogya ***************************************** From rex.alex345 at yahoo.com Thu Feb 26 03:25:34 2009 From: rex.alex345 at yahoo.com (Rex_Alex) Date: Thu, 26 Feb 2009 03:25:34 -0800 (PST) Subject: [Freeswitch-users] ESL Wrapper In-Reply-To: <87f2f3b90902251342q1e393127ha7fbdbebf6d83dac@mail.gmail.com> References: <558004.60211.qm@web59511.mail.ac4.yahoo.com> <78A818D3-6C4F-420D-A922-751A69E7E080@avgs.ca> <6CE58813-715A-43DB-877B-638B5CE7E6E9@freeswitch.org> <87f2f3b90902251342q1e393127ha7fbdbebf6d83dac@mail.gmail.com> Message-ID: <1235647534150-2389093.post@n2.nabble.com> Hi All, I tried svn up && ./bootstrap.sh && ./configure && make install and did Mathieu's suggestion but getting error as below, [root at server esl]# make phpmod make MYLIB="../libesl.a" SOLINK="-shared -Xlinker -x" CFLAGS="-I/root/freeswitch-1.0.3/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2 -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes" CXXFLAGS="-I/root/freeswitch-1.0.3/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC" CXX_CFLAGS="" -C php make[1]: php-config: Command not found make[1]: Entering directory `/root/freeswitch-1.0.3/libs/esl/php' g++ -I/root/freeswitch-1.0.3/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -c esl_wrap.cpp -o esl_wrap.o esl_wrap.cpp:717:18: error: zend.h: No such file or directory esl_wrap.cpp:718:22: error: zend_API.h: No such file or directory esl_wrap.cpp:719:17: error: php.h: No such file or directory esl_wrap.cpp:973:21: error: php_ini.h: No such file or directory esl_wrap.cpp:974:31: error: ext/standard/info.h: No such file or directory esl_wrap.cpp:767: error: ?E_ERROR? was not declared in this scope esl_wrap.cpp:788: error: ISO C++ forbids declaration of ?ZEND_RSRC_DTOR_FUNC? with no type esl_wrap.cpp:788: error: ?SWIG_landfill? was not declared in this scope esl_wrap.cpp:788: error: expected ?,? or ?;? before ?{? token esl_wrap.cpp:793: error: variable or field ?SWIG_ZTS_SetPointerZval? declared void esl_wrap.cpp:793: error: ?zval? was not declared in this scope esl_wrap.cpp:793: error: ?z? was not declared in this scope esl_wrap.cpp:793: error: expected primary-expression before ?void? esl_wrap.cpp:793: error: expected primary-expression before ?*? token esl_wrap.cpp:793: error: ?type? was not declared in this scope esl_wrap.cpp:793: error: expected primary-expression before ?int? esl_wrap.cpp:793: error: initializer expression list treated as compound expression esl_wrap.cpp:793: error: expected ?,? or ?;? before ?{? token make[1]: *** [esl_wrap.o] Error 1 make[1]: Leaving directory `/root/freeswitch-1.0.3/libs/esl/php' make: *** [phpmod] Error 2 [root at server esl]# Please tell me where am i wrong? Thanks, Rex mercutioviz wrote: > > On Wed, Feb 25, 2009 at 11:34 AM, Brian West wrote: >> If he's on 1.0.3 I don't think it has php in it.. > > Can't he do the whole bootstrap process? > svn up && ./bootstrap.sh && ./configure && make install > > And then do Mathieu's suggestion? > > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://n2.nabble.com/ESL-Wrapper-tp2385651p2389093.html Sent from the freeswitch-users mailing list archive at Nabble.com. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090226/d53b1775/attachment.html From yudha2008 at gmail.com Thu Feb 26 03:45:06 2009 From: yudha2008 at gmail.com (Baskar) Date: Thu, 26 Feb 2009 17:15:06 +0530 Subject: [Freeswitch-users] Cant get Disposition status in Javascript Message-ID: Hi, I am using javascript to store uuid, phone_no, endpoint_disposition and hangup cause in my MYSQL. I can get the session UUID , Phone_no, endpoint disposition but i cant get the originate disposition. Javascript : session.setVariable("session.uuid", "ses_uuid: " + session.uuid); session.setVariable("phone", "phone_no: " +argv[0]); result = session.getVariable("endpoint_disposition") hangup = session.getVariable("originate_disposition") OUTPUT: S_UUID PHONE_NO RESULT HANGUP_STATE f579cb15-5145-4eb1-a080-03b9e53b90f739841799874 39841799874 ANSWER for "originate_disposition" i did not get any value stored in the Table. So how can get the originate_disposition ???? -- Warm Regards, N.Baskar -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090226/8505d026/attachment.html From yudha2008 at gmail.com Thu Feb 26 04:44:50 2009 From: yudha2008 at gmail.com (Baskar) Date: Thu, 26 Feb 2009 18:14:50 +0530 Subject: [Freeswitch-users] Cant get Disposition status in Javascript In-Reply-To: References: Message-ID: Hi, One thing i forget to tell i can able to get this Disconnection causeand Disconnection code in Freeswitch console but when i set in the variable i did not get the cause or cause code in javascript. I use these line in Javascript i get the output in the Freeswitch console. console_log("notice", "Disconnect cause: " + session.cause + "\n"); console_log("notice", "Disconnect cause: " + session.causecode + "\n"); OUTPUT: (For the Above line) 2009-02-26 18:08:57 [NOTICE] odbc1.js:1 console_log() Disconnect cause: NORMAL_CLEARING 2009-02-26 18:08:57 [NOTICE] odbc1.js:1 console_log() Disconnect cause: 16 But same session cause and code if i set in the variable i did not get output session.setVariable("session.causecode", "discause: " + session.causecode+ "\n); session.setVariable("notice", "Disconnect cause: " + session.cause + "\n"); OUTPUT: variable_session.causecode: [discause: 0] variable_notice: [Disconnect cause: NONE] Correct me where i am wrong how can i get the disconnection cause in variable. Please help to solve the problem. -- Warm Regards, N.Baskar -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090226/5764d9c5/attachment-0001.html From mrene_lists at avgs.ca Thu Feb 26 05:49:31 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Thu, 26 Feb 2009 08:49:31 -0500 Subject: [Freeswitch-users] ESL Wrapper In-Reply-To: <1235647534150-2389093.post@n2.nabble.com> References: <558004.60211.qm@web59511.mail.ac4.yahoo.com> <78A818D3-6C4F-420D-A922-751A69E7E080@avgs.ca> <6CE58813-715A-43DB-877B-638B5CE7E6E9@freeswitch.org> <87f2f3b90902251342q1e393127ha7fbdbebf6d83dac@mail.gmail.com> <1235647534150-2389093.post@n2.nabble.com> Message-ID: You need your distro's php dev pakage. On 26-Feb-09, at 6:25 AM, Rex_Alex wrote: > Hi All, I tried svn up && ./bootstrap.sh && ./configure && make > install and did Mathieu's suggestion but getting error as below, > [root at server esl]# make phpmod make MYLIB="../libesl.a" SOLINK="- > shared -Xlinker -x" CFLAGS="-I/root/freeswitch-1.0.3/libs/esl/src/ > include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2 > -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings -Wstrict- > prototypes -Wmissing-prototypes" CXXFLAGS="-I/root/freeswitch-1.0.3/ > libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/ > src/ -fPIC" CXX_CFLAGS="" -C php make[1]: php-config: Command not > found make[1]: Entering directory `/root/freeswitch-1.0.3/libs/esl/ > php' g++ -I/root/freeswitch-1.0.3/libs/esl/src/include - > DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -c > esl_wrap.cpp -o esl_wrap.o esl_wrap.cpp:717:18: error: zend.h: No > such file or directory esl_wrap.cpp:718:22: error: zend_API.h: No > such file or directory esl_wrap.cpp:719:17: error: php.h: No such > file or directory esl_wrap.cpp:973:21: error: php_ini.h: No such > file or directory esl_wrap.cpp:974:31: error: ext/standard/info.h: > No such file or directory esl_wrap.cpp:767: error: ?E_ERROR? was not > declared in this scope esl_wrap.cpp:788: error: ISO C++ forbids > declaration of ?ZEND_RSRC_DTOR_FUNC? with no type esl_wrap.cpp:788: > error: ?SWIG_landfill? was not declared in this scope esl_wrap.cpp: > 788: error: expected ?,? or ?;? before ?{? token esl_wrap.cpp:793: > error: variable or field ?SWIG_ZTS_SetPointerZval? declared void > esl_wrap.cpp:793: error: ?zval? was not declared in this scope > esl_wrap.cpp:793: error: ?z? was not declared in this scope > esl_wrap.cpp:793: error: expected primary-expression before ?void? > esl_wrap.cpp:793: error: expected primary-expression before ?*? > token esl_wrap.cpp:793: error: ?type? was not declared in this scope > esl_wrap.cpp:793: error: expected primary-expression before ?int? > esl_wrap.cpp:793: error: initializer expression list treated as > compound expression esl_wrap.cpp:793: error: expected ?,? or ?;? > before ?{? token make[1]: *** [esl_wrap.o] Error 1 make[1]: Leaving > directory `/root/freeswitch-1.0.3/libs/esl/php' make: *** [phpmod] > Error 2 [root at server esl]# Please tell me where am i wrong? Thanks, > Rex > mercutioviz wrote: > On Wed, Feb 25, 2009 at 11:34 AM, Brian West wrote: > If he's on > 1.0.3 I don't think it has php in it.. Can't he do the whole > bootstrap process? svn up && ./bootstrap.sh && ./configure && make > install And then do Mathieu's suggestion? -MC > _______________________________________________ Freeswitch-users > mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > View this message in context: Re: ESL Wrapper > Sent from the freeswitch-users mailing list archive at Nabble.com. > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090226/8003a394/attachment.html From anthony.minessale at gmail.com Thu Feb 26 06:11:55 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 26 Feb 2009 08:11:55 -0600 Subject: [Freeswitch-users] Cant get Disposition status in Javascript In-Reply-To: References: Message-ID: <191c3a030902260611p5ee885e1ud422b8c0bcd17502@mail.gmail.com> both of your set lines are wrong: session.cause and session.causecode are attributes session.setVariable("session.causecode", "discause: " + session.causecode+ "\n); session.setVariable("notice", "Disconnect cause: " + session.cause + "\n"); session.setVariable("cause_code", session.causecode); session.setVariable("cause_name", session.cause); would be more appropriate. Also originate_disposition is only on outgoing legs. Doing this sort of thing in the same script is not a good plan. You should really be doing it in the CDR engine where you can get records for both legs of the call in a relaxed environment. On Thu, Feb 26, 2009 at 6:44 AM, Baskar wrote: > Hi, > One thing i forget to tell i can able to get this Disconnection causeand Disconnection > code in Freeswitch console but when i set in the variable i did not get > the cause or cause code in javascript. > > I use these line in Javascript i get the output in the Freeswitch console. > > console_log("notice", "Disconnect cause: " + session.cause + "\n"); > console_log("notice", "Disconnect cause: " + session.causecode + "\n"); > > OUTPUT: (For the Above line) > > 2009-02-26 18:08:57 [NOTICE] odbc1.js:1 console_log() Disconnect cause: > NORMAL_CLEARING > 2009-02-26 18:08:57 [NOTICE] odbc1.js:1 console_log() Disconnect cause: 16 > > But same session cause and code if i set in the variable i did not get > output > > session.setVariable("session.causecode", "discause: " + session.causecode+ > "\n); > session.setVariable("notice", "Disconnect cause: " + session.cause + "\n"); > > OUTPUT: > > variable_session.causecode: [discause: 0] > variable_notice: [Disconnect cause: NONE] > > Correct me where i am wrong how can i get the disconnection cause in > variable. > > Please help to solve the problem. > > > > -- > Warm Regards, > N.Baskar > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090226/75a71d97/attachment.html From alexander at degreiff.com Thu Feb 26 06:56:07 2009 From: alexander at degreiff.com (Alexander de Greiff) Date: Thu, 26 Feb 2009 15:56:07 +0100 (CET) Subject: [Freeswitch-users] dialplan condition regex question Message-ID: <1520981.41235660110667.JavaMail.alexanderdegreiff@AdG-Mac.local> hi all, i am dialing the number 123456789 (example) reaching fs via inbound sip gateway and hitting following dialplan: ... ... via info i can see that the variable my_dialed_extension is populated ok with 789 but somehow the second condition is not met. when i change that to match (.*) the actions gets executed and the my_dialed_extension inside is correct. any suggestions? kind regards alex From yudha2008 at gmail.com Thu Feb 26 06:55:49 2009 From: yudha2008 at gmail.com (Baskar) Date: Thu, 26 Feb 2009 20:25:49 +0530 Subject: [Freeswitch-users] Cant get Disposition status in Javascript In-Reply-To: <191c3a030902260611p5ee885e1ud422b8c0bcd17502@mail.gmail.com> References: <191c3a030902260611p5ee885e1ud422b8c0bcd17502@mail.gmail.com> Message-ID: Hi Anthony Minessale, I have added these lines in my javascript with your *guidance. *But still i did not get any status like busy , no answer, etc . session.setVariable("cause_code", session.causecode); session.setVariable("cause_name", session.cause); I Get this output only for all the call: variable_cause_code: [0] variable_cause_name: [NONE] -- Warm Regards, N.Baskar -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090226/295d9234/attachment.html From saigop at gmail.com Thu Feb 26 07:11:03 2009 From: saigop at gmail.com (Gopalakrishnan A.N) Date: Thu, 26 Feb 2009 20:41:03 +0530 Subject: [Freeswitch-users] session orignate freeswitch 1.0.3 segmentation error Message-ID: <2ea4d47e0902260711m21fe581ge3a12288808359fa@mail.gmail.com> Hi, I have installed Freeswitch 1.0.3. I am using event socket with Javascript. When I try to dial the script with below command, the call is not going thru it seems to be idle. and segmentation fault core dump error, (freeswitch hangs).....[?] new_session = new Session.originate(session, "sofia/default/@foo.com"); bridge(session, new_session); I saw in the wiki http://wiki.freeswitch.org/wiki/FreeSwitch_Javascript_Session that the session is depreciated, earlier I was using like this in Freeswitch 1.0.2, it works fine....:) session = new Session(); session.originate(session, "{ignore_early_media=true}sofia/default/@foo.com"); So something I am missing, please let me know where I am wrong? -- Thank you with regards, Gopal, PeopleTech Systems Private Limited www.peopletech.co.in -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090226/ab99f74e/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/gif Size: 100 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090226/ab99f74e/attachment-0001.gif From alexander at degreiff.com Thu Feb 26 07:19:33 2009 From: alexander at degreiff.com (Alexander de Greiff) Date: Thu, 26 Feb 2009 16:19:33 +0100 (CET) Subject: [Freeswitch-users] switch voices in ivr menus Message-ID: <15479302.61235661515638.JavaMail.alexanderdegreiff@AdG-Mac.local> another question: - in the ivr application i have cepstral voice matthias read the main menu ok. - i select submenu1 and cepstral voice katrin reads the submenu1 correctly. - i go back to the main menu and the voice is not switched back to the specified voice matthias. each voice is explicitly specified in each menu. any suggestions? kind regards alex From brian at freeswitch.org Thu Feb 26 07:31:00 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 26 Feb 2009 09:31:00 -0600 Subject: [Freeswitch-users] dialplan condition regex question In-Reply-To: <1520981.41235660110667.JavaMail.alexanderdegreiff@AdG-Mac.local> References: <1520981.41235660110667.JavaMail.alexanderdegreiff@AdG-Mac.local> Message-ID: Read this http://wiki.freeswitch.org/wiki/Dialplan_XML#About_Dialplan_Variables You can't condition on a variable you set in the same extension because the set happens later. Thus its not possible to do what you're doing... Once you understand the dialplan is just a list of instructions that is compiled before its installed on the session and sent into execute. You're trying to have soup before the chicken has hatched. /b On Feb 26, 2009, at 8:56 AM, Alexander de Greiff wrote: > hi all, > > i am dialing the number 123456789 (example) reaching fs via inbound > sip gateway and hitting following dialplan: > > > > > > > ... > data="{ignore_early_media=true}user/${my_dialed_extension}@$$ > {domain}"/> > ... > > > via info i can see that the variable my_dialed_extension is > populated ok with 789 but somehow the second condition is not met. > when i change that to match (.*) the actions gets executed and the > my_dialed_extension inside is correct. > > any suggestions? From brian at freeswitch.org Thu Feb 26 07:35:00 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 26 Feb 2009 09:35:00 -0600 Subject: [Freeswitch-users] session orignate freeswitch 1.0.3 segmentation error In-Reply-To: <2ea4d47e0902260711m21fe581ge3a12288808359fa@mail.gmail.com> References: <2ea4d47e0902260711m21fe581ge3a12288808359fa@mail.gmail.com> Message-ID: can you include the backtrace? We might have already fixed this one. http://wiki.freeswitch.org/wiki/Reporting_Bugs /b On Feb 26, 2009, at 9:11 AM, Gopalakrishnan A.N wrote: > Hi, > > I have installed Freeswitch 1.0.3. I am using event socket with > Javascript. When I try to dial the script with below command, the > call is not going thru it seems to be idle. and segmentation fault > core dump error, (freeswitch hangs).....<323.gif> > > > new_session = new Session.originate(session, "sofia/default/ > @foo.com"); > bridge(session, new_session); > > I saw in the wiki http://wiki.freeswitch.org/wiki/FreeSwitch_Javascript_Session > that the session is depreciated, earlier I was using like this in > Freeswitch 1.0.2, it works fine....:) > > session = new Session(); > session.originate(session, "{ignore_early_media=true}sofia/default/ > @foo.com"); > > So something I am missing, please let me know where I am wrong? > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090226/e5d368be/attachment.html From brian at freeswitch.org Thu Feb 26 07:36:19 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 26 Feb 2009 09:36:19 -0600 Subject: [Freeswitch-users] switch voices in ivr menus In-Reply-To: <15479302.61235661515638.JavaMail.alexanderdegreiff@AdG-Mac.local> References: <15479302.61235661515638.JavaMail.alexanderdegreiff@AdG-Mac.local> Message-ID: <6A44C656-4352-4246-9CF1-816E86DACB54@freeswitch.org> What I would need is a debug log... ie press F8, attach that info to a jira http://jira.freeswitch.org and please assign it to me "brian" is the user. /b PS: do not paste logs in the comment box.. Attach them instead. On Feb 26, 2009, at 9:19 AM, Alexander de Greiff wrote: > another question: > > - in the ivr application i have cepstral voice matthias read the > main menu ok. > - i select submenu1 and cepstral voice katrin reads the submenu1 > correctly. > - i go back to the main menu and the voice is not switched back to > the specified voice matthias. > > each voice is explicitly specified in each menu. > > any suggestions? > > kind regards > alex From sunil.d.admin at gmail.com Thu Feb 26 07:36:18 2009 From: sunil.d.admin at gmail.com (Sunil Singh) Date: Thu, 26 Feb 2009 21:06:18 +0530 Subject: [Freeswitch-users] [ANN] Spice Telephony - an open source FreeSWITCH/Erlang callcenter platform In-Reply-To: <20090220191911.GD29511@hijacked.us> References: <22015518.post@talk.nabble.com> <20090220191911.GD29511@hijacked.us> Message-ID: Suggest Try C-Zentrix (from tvtworld.com). It has a very powerful predictive dialer and its also ranked 10th just 2 places below freewsitch. It supports 100 agents on a single box with predictive dialer,IVR, logger and CRM. I don't know how they are doing it but they are running successfully with almost 100% uptime. Try it out . On Sat, Feb 21, 2009 at 12:49 AM, Andrew Thompson wrote: > On Sat, Feb 14, 2009 at 03:04:01PM -0800, JCATS wrote: > > > > Have you planned any predictive dialer features ( like VICIDIAL )? > > > > As Ken Rice mentioned, this isn't really the focus of the project - it's > more for inbound and directed outbound (calling campaigns to specific > people/businesses - not everyone in the phonebook). Primary focus is > inbound (multi brand, skill based routing, dynamic wrapup times, etc). > > Expect a new release sometime soonish that actually does something > useful (accepts and routes inbound calls from FreeSWITCH to an agent). > Also; public source control. There's just some additional corporate > nonsense that I have to sort out (again) before that can go live. > > Andrew > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090226/0635f4d8/attachment.html From brian at freeswitch.org Thu Feb 26 07:57:53 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 26 Feb 2009 09:57:53 -0600 Subject: [Freeswitch-users] switch voices in ivr menus In-Reply-To: <15479302.61235661515638.JavaMail.alexanderdegreiff@AdG-Mac.local> References: <15479302.61235661515638.JavaMail.alexanderdegreiff@AdG-Mac.local> Message-ID: <539F17DE-96C8-45C1-8BEB-273D0AA83A3D@freeswitch.org> Alex, Mine changes voices every time.. can you post your ivr.conf.xml along with the report? /b From alexander at degreiff.com Thu Feb 26 08:52:31 2009 From: alexander at degreiff.com (Alexander de Greiff) Date: Thu, 26 Feb 2009 17:52:31 +0100 (CET) Subject: [Freeswitch-users] switch voices in ivr menus In-Reply-To: <4222480.81235667066314.JavaMail.alexanderdegreiff@AdG-Mac> Message-ID: <7219632.101235667091863.JavaMail.alexanderdegreiff@AdG-Mac> brian, demo3 is the main menu. all submenus change voices correctly. when i go to the main menu via menu-sub (7) then the voice is changes correctly. only when i menu-top (9) to main menu the voice is not changed. how do i produce the debug log? in the cli? this i a remote terminal. f8 is not an option. here is the part of my ivr.conf.xml: From brian at freeswitch.org Thu Feb 26 09:01:33 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 26 Feb 2009 11:01:33 -0600 Subject: [Freeswitch-users] switch voices in ivr menus In-Reply-To: <7219632.101235667091863.JavaMail.alexanderdegreiff@AdG-Mac> References: <7219632.101235667091863.JavaMail.alexanderdegreiff@AdG-Mac> Message-ID: Yes F8 will work on a remote teminal... thats how I do it :P /b On Feb 26, 2009, at 10:52 AM, Alexander de Greiff wrote: > how do i produce the debug log? in the cli? this i a remote > terminal. f8 is not an option. From brian at freeswitch.org Thu Feb 26 09:18:01 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 26 Feb 2009 11:18:01 -0600 Subject: [Freeswitch-users] switch voices in ivr menus In-Reply-To: <7219632.101235667091863.JavaMail.alexanderdegreiff@AdG-Mac> References: <7219632.101235667091863.JavaMail.alexanderdegreiff@AdG-Mac> Message-ID: :P fixed in 12297 /b On Feb 26, 2009, at 10:52 AM, Alexander de Greiff wrote: > brian, > > demo3 is the main menu. all submenus change voices correctly. when i > go to the main menu via menu-sub (7) then the voice is changes > correctly. only when i menu-top (9) to main menu the voice is not > changed. > > how do i produce the debug log? in the cli? this i a remote > terminal. f8 is not an option. > > here is the part of my ivr.conf.xml: From intralanman at freeswitch.org Thu Feb 26 10:39:23 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Thu, 26 Feb 2009 13:39:23 -0500 Subject: [Freeswitch-users] ESL Wrapper In-Reply-To: References: <558004.60211.qm@web59511.mail.ac4.yahoo.com> <78A818D3-6C4F-420D-A922-751A69E7E080@avgs.ca> <6CE58813-715A-43DB-877B-638B5CE7E6E9@freeswitch.org> <87f2f3b90902251342q1e393127ha7fbdbebf6d83dac@mail.gmail.com> <1235647534150-2389093.post@n2.nabble.com> Message-ID: <49A6E1DB.3070806@freeswitch.org> and it will probably be a good idea to do make phpmod-install so that the .so and .php files gets into the correct place to be included -Ray Mathieu Rene wrote: > > You need your distro's php dev pakage. > On 26-Feb-09, at 6:25 AM, Rex_Alex wrote: > >> Hi All, I tried svn up && ./bootstrap.sh && ./configure && make >> install and did Mathieu's suggestion but getting error as below, >> [root at server esl]# make phpmod make MYLIB="../libesl.a" >> SOLINK="-shared -Xlinker -x" >> CFLAGS="-I/root/freeswitch-1.0.3/libs/esl/src/include -DHAVE_EDITLINE >> -g -ggdb -I../../libs/libedit/src/ -fPIC -O2 -ffast-math -Wall >> -Werror -Wunused-variable -Wwrite-strings -Wstrict-prototypes >> -Wmissing-prototypes" >> CXXFLAGS="-I/root/freeswitch-1.0.3/libs/esl/src/include >> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC" >> CXX_CFLAGS="" -C php make[1]: php-config: Command not found make[1]: >> Entering directory `/root/freeswitch-1.0.3/libs/esl/php' g++ >> -I/root/freeswitch-1.0.3/libs/esl/src/include -DHAVE_EDITLINE -g >> -ggdb -I../../libs/libedit/src/ -fPIC -c esl_wrap.cpp -o esl_wrap.o >> esl_wrap.cpp:717:18: error: zend.h: No such file or directory >> esl_wrap.cpp:718:22: error: zend_API.h: No such file or directory >> esl_wrap.cpp:719:17: error: php.h: No such file or directory >> esl_wrap.cpp:973:21: error: php_ini.h: No such file or directory >> esl_wrap.cpp:974:31: error: ext/standard/info.h: No such file or >> directory esl_wrap.cpp:767: error: ?E_ERROR? was not declared in this >> scope esl_wrap.cpp:788: error: ISO C++ forbids declaration of >> ?ZEND_RSRC_DTOR_FUNC? with no type esl_wrap.cpp:788: error: >> ?SWIG_landfill? was not declared in this scope esl_wrap.cpp:788: >> error: expected ?,? or ?;? before ?{? token esl_wrap.cpp:793: error: >> variable or field ?SWIG_ZTS_SetPointerZval? declared void >> esl_wrap.cpp:793: error: ?zval? was not declared in this scope >> esl_wrap.cpp:793: error: ?z? was not declared in this scope >> esl_wrap.cpp:793: error: expected primary-expression before ?void? >> esl_wrap.cpp:793: error: expected primary-expression before ?*? token >> esl_wrap.cpp:793: error: ?type? was not declared in this scope >> esl_wrap.cpp:793: error: expected primary-expression before ?int? >> esl_wrap.cpp:793: error: initializer expression list treated as >> compound expression esl_wrap.cpp:793: error: expected ?,? or ?;? >> before ?{? token make[1]: *** [esl_wrap.o] Error 1 make[1]: Leaving >> directory `/root/freeswitch-1.0.3/libs/esl/php' make: *** [phpmod] >> Error 2 [root at server esl]# Please tell me where am i wrong? Thanks, Rex >> >> mercutioviz wrote: >> On Wed, Feb 25, 2009 at 11:34 AM, Brian West wrote: > If he's on >> 1.0.3 I don't think it has php in it.. Can't he do the whole >> bootstrap process? svn up && ./bootstrap.sh && ./configure && >> make install And then do Mathieu's suggestion? -MC >> _______________________________________________ Freeswitch-users >> mailing list Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> ------------------------------------------------------------------------ >> View this message in context: Re: ESL Wrapper >> >> Sent from the freeswitch-users mailing list archive >> at Nabble.com. >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090226/0e5cecd6/attachment.html From rex.alex345 at yahoo.com Thu Feb 26 10:51:01 2009 From: rex.alex345 at yahoo.com (Rex_Alex) Date: Thu, 26 Feb 2009 10:51:01 -0800 (PST) Subject: [Freeswitch-users] ESL Wrapper In-Reply-To: References: <558004.60211.qm@web59511.mail.ac4.yahoo.com> <78A818D3-6C4F-420D-A922-751A69E7E080@avgs.ca> <6CE58813-715A-43DB-877B-638B5CE7E6E9@freeswitch.org> <87f2f3b90902251342q1e393127ha7fbdbebf6d83dac@mail.gmail.com> <1235647534150-2389093.post@n2.nabble.com> Message-ID: <1235674261134-2391480.post@n2.nabble.com> Hi Mathieu, But other php scripts are working fine. Only when I am tring Single_Command.php with ESP.php, it's not working. Rex. Mathieu Rene wrote: > > > You need your distro's php dev pakage. > On 26-Feb-09, at 6:25 AM, Rex_Alex wrote: > >> Hi All, I tried svn up && ./bootstrap.sh && ./configure && make >> install and did Mathieu's suggestion but getting error as below, >> [root at server esl]# make phpmod make MYLIB="../libesl.a" SOLINK="- >> shared -Xlinker -x" CFLAGS="-I/root/freeswitch-1.0.3/libs/esl/src/ >> include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2 >> -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings -Wstrict- >> prototypes -Wmissing-prototypes" CXXFLAGS="-I/root/freeswitch-1.0.3/ >> libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/ >> src/ -fPIC" CXX_CFLAGS="" -C php make[1]: php-config: Command not >> found make[1]: Entering directory `/root/freeswitch-1.0.3/libs/esl/ >> php' g++ -I/root/freeswitch-1.0.3/libs/esl/src/include - >> DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -c >> esl_wrap.cpp -o esl_wrap.o esl_wrap.cpp:717:18: error: zend.h: No >> such file or directory esl_wrap.cpp:718:22: error: zend_API.h: No >> such file or directory esl_wrap.cpp:719:17: error: php.h: No such >> file or directory esl_wrap.cpp:973:21: error: php_ini.h: No such >> file or directory esl_wrap.cpp:974:31: error: ext/standard/info.h: >> No such file or directory esl_wrap.cpp:767: error: ?E_ERROR? was not >> declared in this scope esl_wrap.cpp:788: error: ISO C++ forbids >> declaration of ?ZEND_RSRC_DTOR_FUNC? with no type esl_wrap.cpp:788: >> error: ?SWIG_landfill? was not declared in this scope esl_wrap.cpp: >> 788: error: expected ?,? or ?;? before ?{? token esl_wrap.cpp:793: >> error: variable or field ?SWIG_ZTS_SetPointerZval? declared void >> esl_wrap.cpp:793: error: ?zval? was not declared in this scope >> esl_wrap.cpp:793: error: ?z? was not declared in this scope >> esl_wrap.cpp:793: error: expected primary-expression before ?void? >> esl_wrap.cpp:793: error: expected primary-expression before ?*? >> token esl_wrap.cpp:793: error: ?type? was not declared in this scope >> esl_wrap.cpp:793: error: expected primary-expression before ?int? >> esl_wrap.cpp:793: error: initializer expression list treated as >> compound expression esl_wrap.cpp:793: error: expected ?,? or ?;? >> before ?{? token make[1]: *** [esl_wrap.o] Error 1 make[1]: Leaving >> directory `/root/freeswitch-1.0.3/libs/esl/php' make: *** [phpmod] >> Error 2 [root at server esl]# Please tell me where am i wrong? Thanks, >> Rex >> mercutioviz wrote: >> On Wed, Feb 25, 2009 at 11:34 AM, Brian West wrote: > If he's on >> 1.0.3 I don't think it has php in it.. Can't he do the whole >> bootstrap process? svn up && ./bootstrap.sh && ./configure && make >> install And then do Mathieu's suggestion? -MC >> _______________________________________________ Freeswitch-users >> mailing list Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> View this message in context: Re: ESL Wrapper >> Sent from the freeswitch-users mailing list archive at Nabble.com. >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://n2.nabble.com/ESL-Wrapper-tp2385651p2391480.html Sent from the freeswitch-users mailing list archive at Nabble.com. From anthony.minessale at gmail.com Thu Feb 26 11:04:44 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 26 Feb 2009 13:04:44 -0600 Subject: [Freeswitch-users] ESL Wrapper In-Reply-To: <1235674261134-2391480.post@n2.nabble.com> References: <558004.60211.qm@web59511.mail.ac4.yahoo.com> <78A818D3-6C4F-420D-A922-751A69E7E080@avgs.ca> <6CE58813-715A-43DB-877B-638B5CE7E6E9@freeswitch.org> <87f2f3b90902251342q1e393127ha7fbdbebf6d83dac@mail.gmail.com> <1235647534150-2389093.post@n2.nabble.com> <1235674261134-2391480.post@n2.nabble.com> Message-ID: <191c3a030902261104n560475acqa924491afd07bd42@mail.gmail.com> the esl php mod is a binary mod which must be compiled so it requires the devel add on for php On Thu, Feb 26, 2009 at 12:51 PM, Rex_Alex wrote: > > Hi Mathieu, > > But other php scripts are working fine. Only when I am tring > Single_Command.php with ESP.php, it's not working. > > Rex. > > Mathieu Rene wrote: > > > > > > You need your distro's php dev pakage. > > On 26-Feb-09, at 6:25 AM, Rex_Alex wrote: > > > >> Hi All, I tried svn up && ./bootstrap.sh && ./configure && make > >> install and did Mathieu's suggestion but getting error as below, > >> [root at server esl]# make phpmod make MYLIB="../libesl.a" SOLINK="- > >> shared -Xlinker -x" CFLAGS="-I/root/freeswitch-1.0.3/libs/esl/src/ > >> include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2 > >> -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings -Wstrict- > >> prototypes -Wmissing-prototypes" CXXFLAGS="-I/root/freeswitch-1.0.3/ > >> libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/ > >> src/ -fPIC" CXX_CFLAGS="" -C php make[1]: php-config: Command not > >> found make[1]: Entering directory `/root/freeswitch-1.0.3/libs/esl/ > >> php' g++ -I/root/freeswitch-1.0.3/libs/esl/src/include - > >> DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -c > >> esl_wrap.cpp -o esl_wrap.o esl_wrap.cpp:717:18: error: zend.h: No > >> such file or directory esl_wrap.cpp:718:22: error: zend_API.h: No > >> such file or directory esl_wrap.cpp:719:17: error: php.h: No such > >> file or directory esl_wrap.cpp:973:21: error: php_ini.h: No such > >> file or directory esl_wrap.cpp:974:31: error: ext/standard/info.h: > >> No such file or directory esl_wrap.cpp:767: error: ?E_ERROR? was not > >> declared in this scope esl_wrap.cpp:788: error: ISO C++ forbids > >> declaration of ?ZEND_RSRC_DTOR_FUNC? with no type esl_wrap.cpp:788: > >> error: ?SWIG_landfill? was not declared in this scope esl_wrap.cpp: > >> 788: error: expected ?,? or ?;? before ?{? token esl_wrap.cpp:793: > >> error: variable or field ?SWIG_ZTS_SetPointerZval? declared void > >> esl_wrap.cpp:793: error: ?zval? was not declared in this scope > >> esl_wrap.cpp:793: error: ?z? was not declared in this scope > >> esl_wrap.cpp:793: error: expected primary-expression before ?void? > >> esl_wrap.cpp:793: error: expected primary-expression before ?*? > >> token esl_wrap.cpp:793: error: ?type? was not declared in this scope > >> esl_wrap.cpp:793: error: expected primary-expression before ?int? > >> esl_wrap.cpp:793: error: initializer expression list treated as > >> compound expression esl_wrap.cpp:793: error: expected ?,? or ?;? > >> before ?{? token make[1]: *** [esl_wrap.o] Error 1 make[1]: Leaving > >> directory `/root/freeswitch-1.0.3/libs/esl/php' make: *** [phpmod] > >> Error 2 [root at server esl]# Please tell me where am i wrong? Thanks, > >> Rex > >> mercutioviz wrote: > >> On Wed, Feb 25, 2009 at 11:34 AM, Brian West wrote: > If he's on > >> 1.0.3 I don't think it has php in it.. Can't he do the whole > >> bootstrap process? svn up && ./bootstrap.sh && ./configure && make > >> install And then do Mathieu's suggestion? -MC > >> _______________________________________________ Freeswitch-users > >> mailing list Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > >> View this message in context: Re: ESL Wrapper > >> Sent from the freeswitch-users mailing list archive at Nabble.com. > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > View this message in context: > http://n2.nabble.com/ESL-Wrapper-tp2385651p2391480.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090226/6bef0e9d/attachment-0001.html From msc at freeswitch.org Thu Feb 26 11:06:22 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 26 Feb 2009 11:06:22 -0800 Subject: [Freeswitch-users] ESL Wrapper In-Reply-To: <1235674261134-2391480.post@n2.nabble.com> References: <558004.60211.qm@web59511.mail.ac4.yahoo.com> <78A818D3-6C4F-420D-A922-751A69E7E080@avgs.ca> <6CE58813-715A-43DB-877B-638B5CE7E6E9@freeswitch.org> <87f2f3b90902251342q1e393127ha7fbdbebf6d83dac@mail.gmail.com> <1235647534150-2389093.post@n2.nabble.com> <1235674261134-2391480.post@n2.nabble.com> Message-ID: <87f2f3b90902261106l6925b035o1a5e19d881ff1584@mail.gmail.com> On Thu, Feb 26, 2009 at 10:51 AM, Rex_Alex wrote: > > Hi Mathieu, > > But other php scripts are working fine. Only when I am tring > Single_Command.php with ESP.php, it's not working. > > Rex. > That may be true but the php-devel package is necessary for building the ESL wrapper for php. The php-devel package is NOT necessary simply to run most php scripts. What OS is it? If it's CentOS or similar then you can just do this: yum install -y php-devel -MC From rex.alex345 at yahoo.com Thu Feb 26 11:20:38 2009 From: rex.alex345 at yahoo.com (Rex_Alex) Date: Thu, 26 Feb 2009 11:20:38 -0800 (PST) Subject: [Freeswitch-users] ESL Wrapper In-Reply-To: <87f2f3b90902261106l6925b035o1a5e19d881ff1584@mail.gmail.com> References: <558004.60211.qm@web59511.mail.ac4.yahoo.com> <78A818D3-6C4F-420D-A922-751A69E7E080@avgs.ca> <6CE58813-715A-43DB-877B-638B5CE7E6E9@freeswitch.org> <87f2f3b90902251342q1e393127ha7fbdbebf6d83dac@mail.gmail.com> <1235647534150-2389093.post@n2.nabble.com> <1235674261134-2391480.post@n2.nabble.com> <87f2f3b90902261106l6925b035o1a5e19d881ff1584@mail.gmail.com> Message-ID: <1235676038843-2391645.post@n2.nabble.com> Hi, Yea, you are right it's CentOS version 5.2. Let me try the same and then I will reply you with status. Thanks, Rex. mercutioviz wrote: > > On Thu, Feb 26, 2009 at 10:51 AM, Rex_Alex wrote: >> >> Hi Mathieu, >> >> But other php scripts are working fine. Only when I am tring >> Single_Command.php with ESP.php, it's not working. >> >> Rex. >> > > That may be true but the php-devel package is necessary for building > the ESL wrapper for php. The php-devel package is NOT necessary simply > to run most php scripts. What OS is it? If it's CentOS or similar then > you can just do this: > yum install -y php-devel > > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://n2.nabble.com/ESL-Wrapper-tp2385651p2391645.html Sent from the freeswitch-users mailing list archive at Nabble.com. From chavpaskov at shaw.ca Thu Feb 26 12:56:10 2009 From: chavpaskov at shaw.ca (Tchavdar Paskov) Date: Thu, 26 Feb 2009 12:56:10 -0800 Subject: [Freeswitch-users] Variables from failed call to be exported to a a new B leg Message-ID: Hi Everybody, this is what i' trying to do / unsuccessfully / so far: ?? ????? ???????? ???????? ???????? ???????? ? -> at this point i'd like to collect some sip Vars from the failed call ???????? - from what i red in wiki i think this is the way to export the var to the Bleg ???????? ????? ??? the gw_2? does not? seem to receive the? sip_hangup_phrase. pls? help me to figure out what i'm doing wrong. thank you in advance. regards Chav -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090226/fdc264ff/attachment.html From msc at freeswitch.org Thu Feb 26 13:06:31 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 26 Feb 2009 13:06:31 -0800 Subject: [Freeswitch-users] Variables from failed call to be exported to a a new B leg In-Reply-To: References: Message-ID: <87f2f3b90902261306t1efc6440y41bacef2a100b9ce@mail.gmail.com> > ???????? Why are you using $0 here? Is that a typo? -MC From nik.middleton at noblesolutions.co.uk Thu Feb 26 13:09:46 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Thu, 26 Feb 2009 21:09:46 -0000 Subject: [Freeswitch-users] Cant get Disposition status in Javascript In-Reply-To: References: <191c3a030902260611p5ee885e1ud422b8c0bcd17502@mail.gmail.com> Message-ID: Works for me, see snippet below var first_session = new Session(dial_string); // Trap for call failure if (!first_session.ready()) { consoleLog("err", "Disposition: " + first_session.cause + "\n"); if (first_session.cause == "USER_BUSY") { Disposition = "BUSY"; } else if (first_session.cause == "NO_ROUTE_DESTINATION") { Disposition = "DCN"; } else if (first_session.cause == "NO_ANSWER") { Disposition = "NA"; } exit(); } ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Baskar Sent: 26 February 2009 14:56 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Cant get Disposition status in Javascript Hi Anthony Minessale, I have added these lines in my javascript with your guidance. But still i did not get any status like busy , no answer, etc . session.setVariable("cause_code", session.causecode); session.setVariable("cause_name", session.cause); I Get this output only for all the call: variable_cause_code: [0] variable_cause_name: [NONE] -- Warm Regards, N.Baskar -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090226/0cd4cb79/attachment-0001.html From chavpaskov at shaw.ca Thu Feb 26 13:09:55 2009 From: chavpaskov at shaw.ca (Tchavdar Paskov) Date: Thu, 26 Feb 2009 13:09:55 -0800 Subject: [Freeswitch-users] Variables from failed call to be exported to a a new B leg In-Reply-To: <87f2f3b90902261306t1efc6440y41bacef2a100b9ce@mail.gmail.com> References: <87f2f3b90902261306t1efc6440y41bacef2a100b9ce@mail.gmail.com> Message-ID: Yes it was typo.My Bad Chav ----- Original Message ----- From: Michael Collins Date: Thursday, February 26, 2009 1:07 pm Subject: Re: [Freeswitch-users] Variables from failed call to be exported to a a new B leg To: freeswitch-users at lists.freeswitch.org > > ???????? data="sofia/gateway/gw_2/$0" /> > > Why are you using $0 here? Is that a typo? > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090226/d4a4583f/attachment.html From brian at freeswitch.org Thu Feb 26 13:14:39 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 26 Feb 2009 15:14:39 -0600 Subject: [Freeswitch-users] Variables from failed call to be exported to a a new B leg In-Reply-To: References: Message-ID: You might also want to actually SET a value to the variable. /b On Feb 26, 2009, at 2:56 PM, Tchavdar Paskov wrote: > From chavpaskov at shaw.ca Thu Feb 26 13:18:36 2009 From: chavpaskov at shaw.ca (Tchavdar Paskov) Date: Thu, 26 Feb 2009 13:18:36 -0800 Subject: [Freeswitch-users] Variables from failed call to be exported to a a new B leg In-Reply-To: References: Message-ID: isn't this already done because of the info application called before ? Chav ----- Original Message ----- From: Brian West Date: Thursday, February 26, 2009 1:14 pm Subject: Re: [Freeswitch-users] Variables from failed call to be exported to a a new B leg To: freeswitch-users at lists.freeswitch.org > You might also want to actually SET a value to the variable. > > /b > > On Feb 26, 2009, at 2:56 PM, Tchavdar Paskov wrote: > > > data="nolocal:sip_hangup_phrase" /> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090226/3d4b890c/attachment.html From msc at freeswitch.org Thu Feb 26 13:35:25 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 26 Feb 2009 13:35:25 -0800 Subject: [Freeswitch-users] Variables from failed call to be exported to a a new B leg In-Reply-To: References: Message-ID: <87f2f3b90902261335g337d1bb5w6191cb97b3e90602@mail.gmail.com> On Thu, Feb 26, 2009 at 1:18 PM, Tchavdar Paskov wrote: > isn't this already done because of the info application called before ? > Chav > to make sure that there is indeed a value and that it gets exported to the second b-leg try this: see if my_var is populated on the new b-leg. -MC From brian at freeswitch.org Thu Feb 26 13:47:02 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 26 Feb 2009 15:47:02 -0600 Subject: [Freeswitch-users] Variables from failed call to be exported to a a new B leg In-Reply-To: References: Message-ID: <797DE9FB-7B89-4FD1-A90B-2A09EDB96831@freeswitch.org> You then don't use the export command... you set the variable /b On Feb 26, 2009, at 3:18 PM, Tchavdar Paskov wrote: > isn't this already done because of the info application called > before ? > Chav From mrene_lists at avgs.ca Thu Feb 26 13:48:01 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Thu, 26 Feb 2009 16:48:01 -0500 Subject: [Freeswitch-users] Variables from failed call to be exported to a a new B leg In-Reply-To: References: Message-ID: <07CCDD39-4836-4114-8966-DE81C629D70F@avgs.ca> On 26-Feb-09, at 3:56 PM, Tchavdar Paskov wrote: > Hi Everybody, > > this is what i' trying to do / unsuccessfully / so far: > > > > break="never"> > > > > -> at this point i'd like to > collect some sip Vars from the failed call > data="nolocal:sip_hangup_phrase" /> - from what i red in wiki i > think this is the way to export the var to the Bleg > > > > > the gw_2 does not seem to receive the sip_hangup_phrase. > pls help me to figure out what i'm doing wrong. > > thank you in advance. > regards > Chav > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Thu Feb 26 13:48:32 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 26 Feb 2009 15:48:32 -0600 Subject: [Freeswitch-users] Variables from failed call to be exported to a a new B leg In-Reply-To: <87f2f3b90902261335g337d1bb5w6191cb97b3e90602@mail.gmail.com> References: <87f2f3b90902261335g337d1bb5w6191cb97b3e90602@mail.gmail.com> Message-ID: <5D84AF46-1C77-45D3-96C4-E670A1AFBE8C@freeswitch.org> You can also do it like this: I don't think what MC pointed out works. I'll have to double check. /b On Feb 26, 2009, at 3:35 PM, Michael Collins wrote: > > > to make sure that there is indeed a value and that it gets exported to > the second b-leg try this: > > > > see if my_var is populated on the new b-leg. > -MC From mrene_lists at avgs.ca Thu Feb 26 13:51:26 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Thu, 26 Feb 2009 16:51:26 -0500 Subject: [Freeswitch-users] Variables from failed call to be exported to a a new B leg In-Reply-To: References: Message-ID: <9FA332A7-0566-489B-9C1B-9E5A2CC04A5A@avgs.ca> Oh I just remembered something, You can set the "failed_xml_cdr_prefix" variable (on the A-leg). And it will copy ALL variables from the B-leg if the call fails. Then you should have providerA_hangup_cause, providerB_hangup_cause. Mathieu On 26-Feb-09, at 3:56 PM, Tchavdar Paskov wrote: > Hi Everybody, > > this is what i' trying to do / unsuccessfully / so far: > > > > break="never"> > > > > -> at this point i'd like to > collect some sip Vars from the failed call > data="nolocal:sip_hangup_phrase" /> - from what i red in wiki i > think this is the way to export the var to the Bleg > > > > > the gw_2 does not seem to receive the sip_hangup_phrase. > pls help me to figure out what i'm doing wrong. > > thank you in advance. > regards > Chav > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Thu Feb 26 13:52:29 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 26 Feb 2009 15:52:29 -0600 Subject: [Freeswitch-users] Variables from failed call to be exported to a a new B leg In-Reply-To: <07CCDD39-4836-4114-8966-DE81C629D70F@avgs.ca> References: <07CCDD39-4836-4114-8966-DE81C629D70F@avgs.ca> Message-ID: <8F49C3FE-3F8A-4BA2-8F67-09150DCBFC99@freeswitch.org> OK I think we have covered BOTH directions now ;) /b On Feb 26, 2009, at 3:48 PM, Mathieu Rene wrote: > > > > From msc at freeswitch.org Thu Feb 26 14:03:38 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 26 Feb 2009 14:03:38 -0800 Subject: [Freeswitch-users] Variables from failed call to be exported to a a new B leg In-Reply-To: <87f2f3b90902261335g337d1bb5w6191cb97b3e90602@mail.gmail.com> References: <87f2f3b90902261335g337d1bb5w6191cb97b3e90602@mail.gmail.com> Message-ID: <87f2f3b90902261403v50b3a5cfrf352115d40511cb0@mail.gmail.com> > > to make sure that there is indeed a value and that it gets exported to > the second b-leg try this: > oops, that set line should have been: That "nolocal:" was extraneous from a lazy copy & paste > > > see if my_var is populated on the new b-leg. > -MC > The most elegant solution is the one Brian gave: So use it, please, and forget what I wrote. :) -MC From brian at freeswitch.org Thu Feb 26 14:32:53 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 26 Feb 2009 16:32:53 -0600 Subject: [Freeswitch-users] Variables from failed call to be exported to a a new B leg In-Reply-To: <87f2f3b90902261403v50b3a5cfrf352115d40511cb0@mail.gmail.com> References: <87f2f3b90902261335g337d1bb5w6191cb97b3e90602@mail.gmail.com> <87f2f3b90902261403v50b3a5cfrf352115d40511cb0@mail.gmail.com> Message-ID: <60652253-9B46-4CBF-94E2-5E61BCC996D2@freeswitch.org> You can use nolocal: with export. Just not set. /b On Feb 26, 2009, at 4:03 PM, Michael Collins wrote: > That "nolocal:" was extraneous from a lazy copy & paste From anthony.minessale at gmail.com Thu Feb 26 14:34:53 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 26 Feb 2009 16:34:53 -0600 Subject: [Freeswitch-users] Variables from failed call to be exported to a a new B leg In-Reply-To: <87f2f3b90902261403v50b3a5cfrf352115d40511cb0@mail.gmail.com> References: <87f2f3b90902261335g337d1bb5w6191cb97b3e90602@mail.gmail.com> <87f2f3b90902261403v50b3a5cfrf352115d40511cb0@mail.gmail.com> Message-ID: <191c3a030902261434k325de0bcm4b30dcb960d6fd75@mail.gmail.com> set the var failed_xml_cdr_prefix=foo before you call bridge and the failed calls will have a complete xml cdr saved in foo_X where X is an incrementing number from 1 upwards. On Thu, Feb 26, 2009 at 4:03 PM, Michael Collins wrote: > > > > to make sure that there is indeed a value and that it gets exported to > > the second b-leg try this: > > > > oops, that set line should have been: > > That "nolocal:" was extraneous from a lazy copy & paste > > > > > > > see if my_var is populated on the new b-leg. > > -MC > > > > The most elegant solution is the one Brian gave: > > > So use it, please, and forget what I wrote. :) > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090226/a9b5c3d1/attachment.html From nik.middleton at noblesolutions.co.uk Thu Feb 26 14:55:22 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Thu, 26 Feb 2009 22:55:22 -0000 Subject: [Freeswitch-users] Console messages Message-ID: Hi Guys, Is there a way of displaying a console message not related to a log level? I've got the console only reporting errors now, but it would be nice to be able to display a message when a given condition exists. Yes, I could set it as an error level message, but I'd rather not do that. Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090226/d60d04b0/attachment.html From b_ball_henry at hotmail.com Thu Feb 26 16:32:30 2009 From: b_ball_henry at hotmail.com (Henry Huang) Date: Thu, 26 Feb 2009 16:32:30 -0800 Subject: [Freeswitch-users] snd_dummy setting for skype Message-ID: <59ad9ca10902261632s8a50903gba865a093c892bd9@mail.gmail.com> I went through the wiki on mod_skypiax and see there should be a script to make skype work without sound card in linux. Does anyone know where to obtain that script to make sound work "without sound card"? I am currently creating a /etc/asound.conf for skype to load the "fake" sound driver. I do hear sound, but it's not perfect, it's very choppy and it gives me error message when starting skype. The following is my asound.conf setting. Hopefully someone can shed some light : pcm.plugfile{ type plug slave { pcm infile format S16_LE channels 1 rate 16000 } } pcm.infile { type file slave { pcm null } file /dev/dsp infile /dev/dsp } by using this configuration. skype spit out error messages as follow but still works: ALSA lib control.c:909:(snd_ctl_open_noupdate) Invalid CTL plugfile ALSA lib control.c:909:(snd_ctl_open_noupdate) Invalid CTL infile -- Henry Huang UniC Solution - Communication Unified VoIP & Open Source software Consultant -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090226/a6524f53/attachment.html From msc at freeswitch.org Thu Feb 26 16:41:25 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 26 Feb 2009 16:41:25 -0800 Subject: [Freeswitch-users] Console messages In-Reply-To: References: Message-ID: <87f2f3b90902261641s30f2036bg6800e0b49222b514@mail.gmail.com> > Is there a way of displaying a console message not related to a log level? > I?ve got the console only reporting errors now, but it would be nice to be > able to display a message when a given condition exists.? Yes, I could set > it as an error level message, but I?d rather not do that. What is the condition? That will probably determine how you proceed. -MC From anthony.minessale at gmail.com Thu Feb 26 16:49:54 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 26 Feb 2009 18:49:54 -0600 Subject: [Freeswitch-users] Console messages In-Reply-To: References: Message-ID: <191c3a030902261649q7bc16ddaxfef5e2e57ff5efa6@mail.gmail.com> You could use level "console" which will always print or use "err" or "crit". On Thu, Feb 26, 2009 at 4:55 PM, Nik Middleton < nik.middleton at noblesolutions.co.uk> wrote: > Hi Guys, > > > > Is there a way of displaying a console message not related to a log level? > I?ve got the console only reporting errors now, but it would be nice to be > able to display a message when a given condition exists. Yes, I could set > it as an error level message, but I?d rather not do that. > > > > Regards, > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090226/68a33e98/attachment.html From mike at jerris.com Thu Feb 26 18:29:22 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 26 Feb 2009 21:29:22 -0500 Subject: [Freeswitch-users] Console messages In-Reply-To: <87f2f3b90902261641s30f2036bg6800e0b49222b514@mail.gmail.com> References: <87f2f3b90902261641s30f2036bg6800e0b49222b514@mail.gmail.com> Message-ID: <416BE849-131D-4A28-8381-1107AC9C19BF@jerris.com> You should be able to do loglevel of console Mike On Feb 26, 2009, at 7:41 PM, Michael Collins wrote: >> Is there a way of displaying a console message not related to a log >> level? >> I?ve got the console only reporting errors now, but it would be ni >> ce to be >> able to display a message when a given condition exists. Yes, I >> could set >> it as an error level message, but I?d rather not do that. > > What is the condition? That will probably determine how you proceed. > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mashudiflexi at telkom.co.id Thu Feb 26 20:52:24 2009 From: mashudiflexi at telkom.co.id (mashudi) Date: Fri, 27 Feb 2009 11:52:24 +0700 Subject: [Freeswitch-users] change session-timer value Message-ID: <49A77188.4070105@telkom.co.id> Hi Folks , I have problem to change the session timer value that automatically created by Freeswitch for 120, I can change the value of session-timer in /sip_profiles/internal.xml become and it work only for INVITE message response, but for UPDATE message response still use default value of 120, please help me to solved this problem. Thank you in advanced. regards mashudi pls help me to figure out what i'm doing wrong. > ***************************************** Sekarang Gratis Nelpon SLJJ Flexi diperluas ke Yogya ***************************************** From pmhshz at gmail.com Thu Feb 26 22:01:12 2009 From: pmhshz at gmail.com (shehzad p) Date: Thu, 26 Feb 2009 22:01:12 -0800 (PST) Subject: [Freeswitch-users] Suggestion for xml_curl performance In-Reply-To: <7e2ac3270902210711u3ea54b79j5868aef6b59d4800@mail.gmail.com> References: <22118122.post@talk.nabble.com> <22133185.post@talk.nabble.com> <7e2ac3270902210711u3ea54b79j5868aef6b59d4800@mail.gmail.com> Message-ID: <22240044.post@talk.nabble.com> Thanks for your response, I am studying FastCGI as Shannon recommend. Is there any other possibility of making this setup better, OR making it better requires change in architecture, If yes please anybody comment and point me out to that direction. Thanks msp Shannon-27 wrote: > > I'd recommend having a look at fastcgi as well. > > On 2/21/09, shehzad p wrote: >> >> Hi Brian, >> >> My setup is to use FS as basic calls routing. >> 1. Calls are coming to FS from more than one customer Gateways, and I >> need >> to authenticate them and check for enough balance based on database, >> [Caller Gateways] ===> [FreeSWITCH] ===> >> [Provider Gateways] >> 2. After knowing that Caller Gateways is valid, then based on dialed >> number >> it search in database for Provider Gateway and bridge the call there. >> 3. After call finish CDR is inserted back into database. >> >> My old setup was using Javascript which works fine in traffic of 10 to 20 >> calls, but then increase of traffic causes many problems. >> >> Now I eliminate use of any of the script (javascript or any other) for >> call >> routing, and route calls directly from dialplan, >> So I have setup test system using xml-curl to generate dynamic dialplan, >> I used below xml_curl PHP example as reference: >> http://wiki.freeswitch.org/wiki/Mod_xml_curl_PHP_example >> For CDR processing I used xml_cdr, with help of the example in FS source >> :scripts/contrib/trixter/xml-cdr. >> >> >> Waiting for any better suggestions, any comments... >> >> thanks >> msp. >> >> Brian West-3 wrote: >>> >>> it all depends on what you're doing.. can you elaborate? >>> >>> /b >>> >>> On Feb 20, 2009, at 4:18 AM, shehzad p wrote: >>> >>>> Recently I faced some performance bottleneck by using Javascript. >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> -- >> View this message in context: >> http://www.nabble.com/Suggestion-for-xml_curl-performance-tp22118122p22133185.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > -- > Shannon > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/Suggestion-for-xml_curl-performance-tp22118122p22240044.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From gopal2krishnan at gmail.com Thu Feb 26 22:11:23 2009 From: gopal2krishnan at gmail.com (Gopal krishnan) Date: Fri, 27 Feb 2009 11:41:23 +0530 Subject: [Freeswitch-users] session orignate freeswitch 1.0.3 segmentation error In-Reply-To: References: <2ea4d47e0902260711m21fe581ge3a12288808359fa@mail.gmail.com> Message-ID: <2ea4d47e0902262211y2e4b243ey52eb7bda04329dc7@mail.gmail.com> Hi Brian, Please find the attached backtrace files attached. And 1. SVN revision number (or binary file) - FreeSWITCH Version 1.0.3 (exported) 2. Operating System and revision - CentOS 5.2 3. Hardware information - 32 bit with 512 MB RAM 4. I am using Event socket 5. Language - Javascript One more thing in the same machine earlier I was using freeswitch 1.0.2. when the segmentation fault happens the core file was generated in the older version freeswitch bin. In 1.0.3 there is no bin directory. Is that could be the prob? On Thu, Feb 26, 2009 at 9:05 PM, Brian West wrote: > can you include the backtrace? We might have already fixed this one. > http://wiki.freeswitch.org/wiki/Reporting_Bugs > > /b > > > On Feb 26, 2009, at 9:11 AM, Gopalakrishnan A.N wrote: > > Hi, > > I have installed Freeswitch 1.0.3. I am using event socket with > Javascript. When I try to dial the script with below command, the call is > not going thru it seems to be idle. and segmentation fault core dump error, > (freeswitch hangs).....<323.gif> > > > new_session = new Session.originate(session, > "sofia/default/@foo.com"); > bridge(session, new_session); > > I saw in the wiki > http://wiki.freeswitch.org/wiki/FreeSwitch_Javascript_Session > that the session is depreciated, earlier I was using like this in > Freeswitch 1.0.2, it works fine....:) > > session = new Session(); > session.originate(session, > "{ignore_early_media=true}sofia/default/@foo.com"); > > So something I am missing, please let me know where I am wrong? > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Thank you with regards, Gopal, PeopleTech Systems Private Limited www.peopletech.co.in -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090227/6daf4869/attachment-0001.html -------------- next part -------------- (gdb) bt #0 session_originate (cx=0x856e988, obj=0x85d59b0, argc=2, argv=0x85e44c8, rval=0xaad1a25c) at mod_spidermonkey.c:2855 #1 0x00806231 in js_Invoke () from /usr/local/freeswitch/lib/libjs.so.1 #2 0x007f9418 in js_Interpret () from /usr/local/freeswitch/lib/libjs.so.1 #3 0x00805976 in js_Execute () from /usr/local/freeswitch/lib/libjs.so.1 #4 0x007c503a in JS_ExecuteScript () from /usr/local/freeswitch/lib/libjs.so.1 #5 0x0070bfe4 in eval_some_js (code=0x84c3656 "new1.js", cx=0x856e988, obj=0x85d3fc8, rval=0xaad1b278) at mod_spidermonkey.h:103 #6 0x0070c592 in js_parse_and_execute (session=0x0, input_code=0x84c3656 "new1.js", ro=0xaad1b2a0) at mod_spidermonkey.c:3583 #7 0x0070c8b1 in jsapi_function (cmd=0x84c3656 "new1.js", session=0x0, stream=0xaad1b328) at mod_spidermonkey.c:3663 #8 0x00edb91d in switch_api_execute (cmd=0x84c3650 "jsapi", arg=0x84c3656 "new1.js", session=0x0, stream=0xaad1b328) at src/switch_loadable_module.c:1524 #9 0x00ec0c06 in switch_console_process (cmd=0x84c3650 "jsapi", rec=0) at src/switch_console.c:254 #10 0x00ec0e3a in console_thread (thread=0x8535980, obj=0x85358f8) at src/switch_console.c:454 #11 0x00f370d6 in dummy_worker (opaque=0x8535980) at threadproc/unix/thread.c:138 #12 0x00caf462 in start_thread () from /lib/i686/nosegneg/libpthread.so.0 #13 0x00c062ce in clone () from /lib/i686/nosegneg/libc.so.6 -------------- next part -------------- An embedded and charset-unspecified text was scrubbed... Name: bt_full.txt Url: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090227/6daf4869/attachment-0001.txt -------------- next part -------------- [root at localhost bin]# cat /proc/cpuinfo processor : 0 vendor_id : GenuineIntel cpu family : 15 model : 4 model name : Intel(R) Celeron(R) CPU 2.66GHz stepping : 9 cpu MHz : 2659.202 cache size : 256 KB fdiv_bug : no hlt_bug : no f00f_bug : no coma_bug : no fpu : yes fpu_exception : yes cpuid level : 5 wp : yes flags : fpu tsc msr pae mce cx8 apic mtrr mca cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe nx lm constant_tsc up pni monitor ds_cpl tm2 cid cx16 xtpr lahf_lm bogomips : 6651.17 [root at localhost bin]# -------------- next part -------------- [root at localhost bin]# uname -a Linux localhost.localdomain 2.6.18-53.el5xen #1 SMP Mon Nov 12 03:26:12 EST 2007 i686 i686 i386 GNU/Linux [root at localhost bin]# From gmaruzz at celliax.org Fri Feb 27 04:15:47 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Fri, 27 Feb 2009 13:15:47 +0100 Subject: [Freeswitch-users] snd_dummy setting for skype In-Reply-To: <59ad9ca10902261632s8a50903gba865a093c892bd9@mail.gmail.com> References: <59ad9ca10902261632s8a50903gba865a093c892bd9@mail.gmail.com> Message-ID: <7b197bef0902270415g6bd06a96sa3dc9c7b694babb5@mail.gmail.com> On Fri, Feb 27, 2009 at 1:32 AM, Henry Huang wrote: > I went through the wiki on mod_skypiax and see there should be a script to > make skype work without sound card in linux. Does anyone know where to > obtain that script to make sound work "without sound card"? Dear Henry, I apologize if the wiki page was not clear. snd-dummy is an ALSA driver (loadable module for the linux kernel) that you load like the other ALSA modules using the 'modprobe' command, no need at all to create an asound.conf file. You can find an example on how to load snd-dummy in the first lines of the script mod_skypiax/configs/startskype.sh I modified the wiki page, could you check is now clear? Thanks for reporting this, please continue to help us! Sincerely, Giovanni Maruzzelli ========================================= www.celliax.org via Pierlombardo 9, 20135 Milano Italy gmaruzz at celliax dot org Cell : +39-347-2665618 Fax : +39-02-87390039 On Fri, Feb 27, 2009 at 1:32 AM, Henry Huang wrote: > I went through the wiki on mod_skypiax and see there should be a script to > make skype work without sound card in linux. Does anyone know where to > obtain that script to make sound work "without sound card"? > > I am currently creating a /etc/asound.conf for skype to load the "fake" > sound driver. I do hear sound, but it's not perfect, it's very choppy and it > gives me error message when starting skype. The following is my asound.conf > setting. Hopefully someone can shed some light : > pcm.plugfile{ > ??? type plug > ??? slave { > ??????? pcm infile > ??????? format S16_LE > ??????? channels 1 > ??????? rate 16000 > ??? } > } > > pcm.infile { > ??? type file > ??? slave { > ??????? pcm null > ??? } > ??? file /dev/dsp > ??? infile /dev/dsp > } > > by using this configuration. skype spit out error messages as follow but > still works: > ALSA lib control.c:909:(snd_ctl_open_noupdate) Invalid CTL plugfile > ALSA lib control.c:909:(snd_ctl_open_noupdate) Invalid CTL infile > > > > > -- > Henry Huang > UniC Solution - Communication Unified > VoIP & Open Source software Consultant > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From rex.alex345 at yahoo.com Fri Feb 27 05:13:12 2009 From: rex.alex345 at yahoo.com (Rex_Alex) Date: Fri, 27 Feb 2009 05:13:12 -0800 (PST) Subject: [Freeswitch-users] ESL Wrapper In-Reply-To: <49A6E1DB.3070806@freeswitch.org> References: <558004.60211.qm@web59511.mail.ac4.yahoo.com> <78A818D3-6C4F-420D-A922-751A69E7E080@avgs.ca> <6CE58813-715A-43DB-877B-638B5CE7E6E9@freeswitch.org> <87f2f3b90902251342q1e393127ha7fbdbebf6d83dac@mail.gmail.com> <1235647534150-2389093.post@n2.nabble.com> <49A6E1DB.3070806@freeswitch.org> Message-ID: <1235740392995-2395557.post@n2.nabble.com> Hi All, I did what you have all suggested. Now its working perfectly. Thanks a lot for all your assistance. Rex. Raymond Chandler wrote: > > and it will probably be a good idea to do > make phpmod-install > so that the .so and .php files gets into the correct place to be included > > -Ray > > Mathieu Rene wrote: >> >> You need your distro's php dev pakage. >> On 26-Feb-09, at 6:25 AM, Rex_Alex wrote: >> >>> Hi All, I tried svn up && ./bootstrap.sh && ./configure && make >>> install and did Mathieu's suggestion but getting error as below, >>> [root at server esl]# make phpmod make MYLIB="../libesl.a" >>> SOLINK="-shared -Xlinker -x" >>> CFLAGS="-I/root/freeswitch-1.0.3/libs/esl/src/include -DHAVE_EDITLINE >>> -g -ggdb -I../../libs/libedit/src/ -fPIC -O2 -ffast-math -Wall >>> -Werror -Wunused-variable -Wwrite-strings -Wstrict-prototypes >>> -Wmissing-prototypes" >>> CXXFLAGS="-I/root/freeswitch-1.0.3/libs/esl/src/include >>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC" >>> CXX_CFLAGS="" -C php make[1]: php-config: Command not found make[1]: >>> Entering directory `/root/freeswitch-1.0.3/libs/esl/php' g++ >>> -I/root/freeswitch-1.0.3/libs/esl/src/include -DHAVE_EDITLINE -g >>> -ggdb -I../../libs/libedit/src/ -fPIC -c esl_wrap.cpp -o esl_wrap.o >>> esl_wrap.cpp:717:18: error: zend.h: No such file or directory >>> esl_wrap.cpp:718:22: error: zend_API.h: No such file or directory >>> esl_wrap.cpp:719:17: error: php.h: No such file or directory >>> esl_wrap.cpp:973:21: error: php_ini.h: No such file or directory >>> esl_wrap.cpp:974:31: error: ext/standard/info.h: No such file or >>> directory esl_wrap.cpp:767: error: ?E_ERROR? was not declared in this >>> scope esl_wrap.cpp:788: error: ISO C++ forbids declaration of >>> ?ZEND_RSRC_DTOR_FUNC? with no type esl_wrap.cpp:788: error: >>> ?SWIG_landfill? was not declared in this scope esl_wrap.cpp:788: >>> error: expected ?,? or ?;? before ?{? token esl_wrap.cpp:793: error: >>> variable or field ?SWIG_ZTS_SetPointerZval? declared void >>> esl_wrap.cpp:793: error: ?zval? was not declared in this scope >>> esl_wrap.cpp:793: error: ?z? was not declared in this scope >>> esl_wrap.cpp:793: error: expected primary-expression before ?void? >>> esl_wrap.cpp:793: error: expected primary-expression before ?*? token >>> esl_wrap.cpp:793: error: ?type? was not declared in this scope >>> esl_wrap.cpp:793: error: expected primary-expression before ?int? >>> esl_wrap.cpp:793: error: initializer expression list treated as >>> compound expression esl_wrap.cpp:793: error: expected ?,? or ?;? >>> before ?{? token make[1]: *** [esl_wrap.o] Error 1 make[1]: Leaving >>> directory `/root/freeswitch-1.0.3/libs/esl/php' make: *** [phpmod] >>> Error 2 [root at server esl]# Please tell me where am i wrong? Thanks, Rex >>> >>> mercutioviz wrote: >>> On Wed, Feb 25, 2009 at 11:34 AM, Brian West wrote: > If he's on >>> 1.0.3 I don't think it has php in it.. Can't he do the whole >>> bootstrap process? svn up && ./bootstrap.sh && ./configure && >>> make install And then do Mathieu's suggestion? -MC >>> _______________________________________________ Freeswitch-users >>> mailing list Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> ------------------------------------------------------------------------ >>> View this message in context: Re: ESL Wrapper >>> >>> Sent from the freeswitch-users mailing list archive >>> at Nabble.com. >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://n2.nabble.com/ESL-Wrapper-tp2385651p2395557.html Sent from the freeswitch-users mailing list archive at Nabble.com. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090227/c650d3a7/attachment.html From anthony.minessale at gmail.com Fri Feb 27 05:58:29 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 27 Feb 2009 07:58:29 -0600 Subject: [Freeswitch-users] change session-timer value In-Reply-To: <49A77188.4070105@telkom.co.id> References: <49A77188.4070105@telkom.co.id> Message-ID: <191c3a030902270558q2a6e1f55v3b6d118d45ec00e1@mail.gmail.com> Didn't you already start a thread with this same question yesterday? If it doesn't work open a jira http://jira.freeswitch.org under the sofia sip category and we will give it to the sofia developer to look at. On Thu, Feb 26, 2009 at 10:52 PM, mashudi wrote: > Hi Folks , > I have problem to change the session timer value that automatically > created by Freeswitch for 120, I can change the value of session-timer > in /sip_profiles/internal.xml > > > > become > > > > and it work only for INVITE message response, > but for UPDATE message response still use default value of 120, > please help me to solved this problem. > Thank you in advanced. > regards > > mashudi > > pls help me to figure out what i'm doing wrong. > > > > > > > > ***************************************** > Sekarang Gratis Nelpon SLJJ Flexi diperluas ke > Yogya > ***************************************** > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090227/3948338f/attachment.html From helmut.kuper at ewetel.de Fri Feb 27 06:07:15 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Fri, 27 Feb 2009 15:07:15 +0100 Subject: [Freeswitch-users] Problems with record_stereo Message-ID: <49A7F393.6080406@ewetel.de> Hello, I play around with record_session and would like to have caller and callee separated on left and right channel. I found record_stereo is used for this. Unfortunately it doesn't work. A and B leg are still mixed. Additionally I found that B leg is significant louder than A leg, but both legs were local extensions. My Dialplan looks like this: regards helmut From anthony.minessale at gmail.com Fri Feb 27 06:08:54 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 27 Feb 2009 08:08:54 -0600 Subject: [Freeswitch-users] session orignate freeswitch 1.0.3 segmentation error In-Reply-To: <2ea4d47e0902262211y2e4b243ey52eb7bda04329dc7@mail.gmail.com> References: <2ea4d47e0902260711m21fe581ge3a12288808359fa@mail.gmail.com> <2ea4d47e0902262211y2e4b243ey52eb7bda04329dc7@mail.gmail.com> Message-ID: <191c3a030902270608i2e6988d1ld53aa3e3de1b9bd8@mail.gmail.com> why are we doing this on the mailing list. This info belongs in a jira ticket. Please reproduce this issue with SVN trunk and if it persists, report it on http://jira.freeswitch.org 2009/2/27 Gopal krishnan > Hi Brian, > Please find the attached backtrace files attached. > And > 1. SVN revision number (or binary file) - FreeSWITCH Version 1.0.3 > (exported) > 2. Operating System and revision - CentOS 5.2 > 3. Hardware information - 32 bit with 512 MB RAM > 4. I am using Event socket > 5. Language - Javascript > > One more thing in the same machine earlier I was using freeswitch 1.0.2. > when the segmentation fault happens the core file was generated in the older > version freeswitch bin. In 1.0.3 there is no bin directory. Is that could be > the prob? > > On Thu, Feb 26, 2009 at 9:05 PM, Brian West wrote: > >> can you include the backtrace? We might have already fixed this one. >> http://wiki.freeswitch.org/wiki/Reporting_Bugs >> >> /b >> >> >> On Feb 26, 2009, at 9:11 AM, Gopalakrishnan A.N wrote: >> >> Hi, >> >> I have installed Freeswitch 1.0.3. I am using event socket with >> Javascript. When I try to dial the script with below command, the call is >> not going thru it seems to be idle. and segmentation fault core dump error, >> (freeswitch hangs).....<323.gif> >> >> >> new_session = new Session.originate(session, >> "sofia/default/@foo.com"); >> bridge(session, new_session); >> >> I saw in the wiki >> http://wiki.freeswitch.org/wiki/FreeSwitch_Javascript_Session >> that the session is depreciated, earlier I was using like this in >> Freeswitch 1.0.2, it works fine....:) >> >> session = new Session(); >> session.originate(session, >> "{ignore_early_media=true}sofia/default/@foo.com"); >> >> So something I am missing, please let me know where I am wrong? >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Thank you with regards, > Gopal, > PeopleTech Systems Private Limited > www.peopletech.co.in > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090227/5b47f62b/attachment-0001.html From mashudiflexi at telkom.co.id Fri Feb 27 08:29:50 2009 From: mashudiflexi at telkom.co.id (mashudi) Date: Fri, 27 Feb 2009 23:29:50 +0700 Subject: [Freeswitch-users] change session-timer value In-Reply-To: <191c3a030902270558q2a6e1f55v3b6d118d45ec00e1@mail.gmail.com> References: <49A77188.4070105@telkom.co.id> <191c3a030902270558q2a6e1f55v3b6d118d45ec00e1@mail.gmail.com> Message-ID: <49A814FE.402@telkom.co.id> Dear Anthony Minessale, As information, I try install to Freeswitch as inbound conference server and integrated with the Huawei MSCe, I got trouble with the session-timer as response message UPDATE that send by Huawei MSC. Because the session timer value below the MSCe specification, MSCe send bye message after the message UPDATE response from Freeswitch. could you give me the guidance how to solve this? thank you in advanced for your kind support, regards, mashudi Anthony Minessale wrote: > Didn't you already start a thread with this same question yesterday? > > If it doesn't work open a jira http://jira.freeswitch.org under the > sofia sip category and we will > give it to the sofia developer to look at. > > > On Thu, Feb 26, 2009 at 10:52 PM, mashudi > wrote: > > Hi Folks , > I have problem to change the session timer value that automatically > created by Freeswitch for 120, I can change the value of session-timer > in /sip_profiles/internal.xml > > > > become > > > > and it work only for INVITE message response, > but for UPDATE message response still use default value of 120, > please help me to solved this problem. > Thank you in advanced. > regards > > mashudi > > pls help me to figure out what i'm doing wrong. > > > > > > > > ***************************************** > Sekarang Gratis Nelpon SLJJ Flexi diperluas ke > Yogya > ***************************************** > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > ***************************************** > Sekarang Gratis Nelpon SLJJ Flexi diperluas ke > Yogya > ***************************************** > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > ***************************************** Sekarang Gratis Nelpon SLJJ Flexi diperluas ke Yogya ***************************************** From brian at freeswitch.org Fri Feb 27 08:29:08 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 27 Feb 2009 10:29:08 -0600 Subject: [Freeswitch-users] change session-timer value In-Reply-To: <49A814FE.402@telkom.co.id> References: <49A77188.4070105@telkom.co.id> <191c3a030902270558q2a6e1f55v3b6d118d45ec00e1@mail.gmail.com> <49A814FE.402@telkom.co.id> Message-ID: Please open a ticket on http://jira.freeswitch.org, as per his last email. Attach all the information to explain the bug along with sip traces. Attach the info, DO NOT paste the logs or traces in the comment box. Attachments are much easier to download and read for us. /b On Feb 27, 2009, at 10:29 AM, mashudi wrote: > Dear Anthony Minessale, > As information, I try install to Freeswitch as inbound conference > server > and integrated with the Huawei MSCe, I got trouble with the > session-timer as response message UPDATE that send by Huawei MSC. > Because the session timer value below the MSCe specification, MSCe > send > bye message after the message UPDATE response from Freeswitch. > > could you give me the guidance how to solve this? > thank you in advanced for your kind support, > regards, > > mashudi From freeswitch at servercorps.com Fri Feb 27 09:17:04 2009 From: freeswitch at servercorps.com (Addison Martin) Date: Fri, 27 Feb 2009 11:17:04 -0600 Subject: [Freeswitch-users] Console messages In-Reply-To: References: Message-ID: <92e7d2090902270917t1f7865f9r8245b56934b0d70c@mail.gmail.com> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_log I'd set the log level to CONSOLE (0), then do: On Thu, Feb 26, 2009 at 4:55 PM, Nik Middleton wrote: > Hi Guys, > > > > Is there a way of displaying a console message not related to a log level? > I?ve got the console only reporting errors now, but it would be nice to be > able to display a message when a given condition exists.? Yes, I could set > it as an error level message, but I?d rather not do that. > > > > Regards, > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From freeswitch at servercorps.com Fri Feb 27 09:41:59 2009 From: freeswitch at servercorps.com (Addison Martin) Date: Fri, 27 Feb 2009 11:41:59 -0600 Subject: [Freeswitch-users] Thin Client VOIP setup? In-Reply-To: <3d381e170902251311i3d3a4205j117d472228c30219@mail.gmail.com> References: <3d381e170902251311i3d3a4205j117d472228c30219@mail.gmail.com> Message-ID: <92e7d2090902270941o76e68288q246009167323e7ac@mail.gmail.com> With a little bit of scripting + web programming, click to call is readily available with FreeSWITCH. Check out call.php in scripts/ under the freeswitch source directory. The code in that php file can be rolled into your CRM. You'll then need to set a variable based on the user's login that is his/her extension. When he clicks the link, the script will dial the user that clicked, and the number he clicked, and connect the two together. Hope this helps. nik On Wed, Feb 25, 2009 at 3:11 PM, Erik Wickstrom wrote: > Hi, > > I've deployed Freeswitch as our phone system at work.? We now want to use > our new phonesystem in a phone room with thin clients (Terminal Server, > possibly LTSP) for each agent.? Ideally, we'd like to use x-lite or another > softphone for each agent. > > The desired workflow for the agents is as follows: > 1) A web based CRM with click to dial. (and customer data card etc) > 2) Agent clicks dial button and is connected to customer > 3) Interact with CRM... > > From what I've read so far, there are some challenges that need to be > overcome in deploying softphones over thin clients. > > Has anyone here had any success in setting up a system like this?? I'm I > asking for trouble trying to use softphones with thin clients (should I just > use hardware phones?? Do they support click to dial?) > > Thanks! > Erik > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From freeswitch at servercorps.com Fri Feb 27 11:20:33 2009 From: freeswitch at servercorps.com (Addison Martin) Date: Fri, 27 Feb 2009 13:20:33 -0600 Subject: [Freeswitch-users] Thin Client VOIP setup? In-Reply-To: <92e7d2090902270941o76e68288q246009167323e7ac@mail.gmail.com> References: <3d381e170902251311i3d3a4205j117d472228c30219@mail.gmail.com> <92e7d2090902270941o76e68288q246009167323e7ac@mail.gmail.com> Message-ID: <92e7d2090902271120nbe5688di71736b30c96c40dc@mail.gmail.com> I forgot to add, this will work for ANY SIP UA, not just soft phones. nik On Fri, Feb 27, 2009 at 11:41 AM, Addison Martin wrote: > With a little bit of scripting + web programming, click to call is > readily available with FreeSWITCH. ?Check out call.php in scripts/ > under the freeswitch source directory. The code in that php file can > be rolled into your CRM. ? You'll then need to set a variable based on > the user's login that is his/her extension. ?When he clicks the link, > the script will dial the user that clicked, and the number he clicked, > and connect the two together. > > Hope this helps. > > nik > > > On Wed, Feb 25, 2009 at 3:11 PM, Erik Wickstrom wrote: >> Hi, >> >> I've deployed Freeswitch as our phone system at work.? We now want to use >> our new phonesystem in a phone room with thin clients (Terminal Server, >> possibly LTSP) for each agent.? Ideally, we'd like to use x-lite or another >> softphone for each agent. >> >> The desired workflow for the agents is as follows: >> 1) A web based CRM with click to dial. (and customer data card etc) >> 2) Agent clicks dial button and is connected to customer >> 3) Interact with CRM... >> >> From what I've read so far, there are some challenges that need to be >> overcome in deploying softphones over thin clients. >> >> Has anyone here had any success in setting up a system like this?? I'm I >> asking for trouble trying to use softphones with thin clients (should I just >> use hardware phones?? Do they support click to dial?) >> >> Thanks! >> Erik >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > From chavpaskov at shaw.ca Fri Feb 27 14:27:08 2009 From: chavpaskov at shaw.ca (Tchavdar Paskov) Date: Fri, 27 Feb 2009 14:27:08 -0800 Subject: [Freeswitch-users] Variables from failed call to be exported to a a new B leg In-Reply-To: <797DE9FB-7B89-4FD1-A90B-2A09EDB96831@freeswitch.org> References: <797DE9FB-7B89-4FD1-A90B-2A09EDB96831@freeswitch.org> Message-ID: Worked like charm. Thanks everyone for your support. Regards Chav ----- Original Message ----- From: Brian West Date: Thursday, February 26, 2009 1:47 pm Subject: Re: [Freeswitch-users] Variables from failed call to be exported to a a new B leg To: freeswitch-users at lists.freeswitch.org > You then don't use the export command...? you set the > variable application="set" data=""export_vars=sip_hangup_phrase"/> > > /b > > On Feb 26, 2009, at 3:18 PM, Tchavdar Paskov wrote: > > > isn't this already done because of the info application > called? > > before ? > > Chav > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090227/891e76e1/attachment.html From brian at freeswitch.org Fri Feb 27 14:30:58 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 27 Feb 2009 16:30:58 -0600 Subject: [Freeswitch-users] Variables from failed call to be exported to a a new B leg In-Reply-To: References: <797DE9FB-7B89-4FD1-A90B-2A09EDB96831@freeswitch.org> Message-ID: <755E463C-C0DE-4F0F-9F7E-7F3C582AC25E@freeswitch.org> Don't run off so fast. ;) You should join us on IRC... I would like to see the IRC numbers over 200 soon ;) /b On Feb 27, 2009, at 4:27 PM, Tchavdar Paskov wrote: > Worked like charm. > Thanks everyone for your support. > Regards > Chav From chavpaskov at shaw.ca Fri Feb 27 14:34:42 2009 From: chavpaskov at shaw.ca (Tchavdar Paskov) Date: Fri, 27 Feb 2009 14:34:42 -0800 Subject: [Freeswitch-users] Use XML dialplan and mod_perl Message-ID: Hi, i have a quick question. is it possible? to use both? XML dial plan? and mod_perl? together. examlpe: default.xml - used as default context mod_perl? - used to generate the public context If it is possible how i have to set perl.conf.xml? and especially? xml-handler-bindings ? is it possible? in value="dialplan"? to specify the? name of the context? Regards Chav -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090227/1e3f09c1/attachment-0001.html From chavpaskov at shaw.ca Fri Feb 27 14:36:19 2009 From: chavpaskov at shaw.ca (Tchavdar Paskov) Date: Fri, 27 Feb 2009 14:36:19 -0800 Subject: [Freeswitch-users] Variables from failed call to be exported to a a new B leg In-Reply-To: <755E463C-C0DE-4F0F-9F7E-7F3C582AC25E@freeswitch.org> References: <797DE9FB-7B89-4FD1-A90B-2A09EDB96831@freeswitch.org> <755E463C-C0DE-4F0F-9F7E-7F3C582AC25E@freeswitch.org> Message-ID: Just logging into IRC Regards Chav ----- Original Message ----- From: Brian West Date: Friday, February 27, 2009 2:31 pm Subject: Re: [Freeswitch-users] Variables from failed call to be exported to a a new B leg To: freeswitch-users at lists.freeswitch.org > Don't run off so fast.? ;)? You should join us on > IRC... I would like? > to see the IRC numbers over 200 soon ;) > > /b > > On Feb 27, 2009, at 4:27 PM, Tchavdar Paskov wrote: > > > Worked like charm. > > Thanks everyone for your support. > > Regards > > Chav > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090227/965f1f01/attachment.html From msc at freeswitch.org Fri Feb 27 16:24:22 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 27 Feb 2009 16:24:22 -0800 Subject: [Freeswitch-users] Use XML dialplan and mod_perl In-Reply-To: References: Message-ID: <87f2f3b90902271624n25fee20ere016fd04372f3e5e@mail.gmail.com> > Hi, > i have a quick question. > is it possible? to use both? XML dial plan? and mod_perl? together. > > examlpe: > > default.xml - used as default context > mod_perl? - used to generate the public context > > If it is possible how i have to set perl.conf.xml? and especially > xml-handler-bindings ? > > is it possible? in value="dialplan"? to specify the? name of the context? > > Regards > Chav What's your IRC nick? We can discuss it more there. -MC From Prometheus001 at gmx.net Sat Feb 28 04:18:54 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Sat, 28 Feb 2009 13:18:54 +0100 Subject: [Freeswitch-users] pocketsphinx and event socket Message-ID: <49A92BAE.4090907@gmx.net> Hello, I have tried the pizza demo and did it get to work so far. However I would like to use pocketsphinx through event socket. I saw in the wiki that there is a chapter for Speech Synthesis Commands: http://wiki.freeswitch.org/wiki/Mod_commands#Speech_Synthesis_Commands However this is empty. Also http://wiki.freeswitch.org/wiki/ASR didn't give me a hint. As I am not a Java programmer, it's hard for me to determine how the pizza demo actually works. Anybody has a sample how he did it e.g. in Php/Perl or so? (I am working with Ruby) Or back to the basics: Is it possible to use pocketsphinx through event socket? Best regards Peter From codecomplete at free.fr Sat Feb 28 07:47:46 2009 From: codecomplete at free.fr (Fred) Date: Sat, 28 Feb 2009 16:47:46 +0100 Subject: [Freeswitch-users] SIP server? PBX vs. softswitch? Message-ID: <7.0.1.0.2.20090228120132.0285f8f8@fredshack.com> Hello Even though I successfully set up an Asterisk voice server, I'm no telecom expert, and would like some clarification about the following things: - What is an SIP server as opposed to a IP PBX? - What is the different between a PBX like Asterisk and a softswitch? Thank you. From e.schmidbauer at gmail.com Sat Feb 28 08:01:36 2009 From: e.schmidbauer at gmail.com (e schmidbauer) Date: Sat, 28 Feb 2009 11:01:36 -0500 Subject: [Freeswitch-users] freeswitch with celt on windows Message-ID: <2cef777b0902280801o7380d81kf00be14f43c304a9@mail.gmail.com> Hi. I was wondering if it is possible to compile freeswitch with the celt codec on windows? I have been able to compile celt for windows but when i compiled freeswitch i did not see the mod_celt in the solution explorer. If this is possible could someone point me in the right direction. Thank you. From brian at freeswitch.org Sat Feb 28 08:10:57 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 28 Feb 2009 10:10:57 -0600 Subject: [Freeswitch-users] pocketsphinx and event socket In-Reply-To: <49A92BAE.4090907@gmx.net> References: <49A92BAE.4090907@gmx.net> Message-ID: You can accomplish this .... here is an example using ESL in perl http://fisheye.freeswitch.org/browse/FreeSWITCH/libs/esl/perl/server3.pl?r=12344 /b On Feb 28, 2009, at 6:18 AM, Peter P GMX wrote: > Or back to the basics: Is it possible to use pocketsphinx through > event > socket? From codecomplete at free.fr Sat Feb 28 08:30:07 2009 From: codecomplete at free.fr (Fred) Date: Sat, 28 Feb 2009 17:30:07 +0100 Subject: [Freeswitch-users] Using OpenZAP + FXO card just to get CID info? Message-ID: <7.0.1.0.2.20090228172749.027ad548@fredshack.com> Hello I'd like to write a single-host CRM application, so I need to get the CallerID information when a call comes in. I don't actually need a PBX/softswitch. The user will have the FXO cards and a phoneset connected on the same line, and will pick up the phone once the CRM application has picked up the CID info and popped up a dialog box, etc. Is it possible to just use OpenZAP to get this information, or must I install Freeswitch and provide an IP phone as well? Thank you. From mrene_lists at avgs.ca Sat Feb 28 08:32:18 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Sat, 28 Feb 2009 11:32:18 -0500 Subject: [Freeswitch-users] Using OpenZAP + FXO card just to get CID info? In-Reply-To: <7.0.1.0.2.20090228172749.027ad548@fredshack.com> References: <7.0.1.0.2.20090228172749.027ad548@fredshack.com> Message-ID: OpenZAP is just a module accessing the card, you need to use it within freeswitch. Then, you can use event socket to get the callerid. On 28-Feb-09, at 11:30 AM, Fred wrote: > Hello > > I'd like to write a single-host CRM application, so I need to get the > CallerID information when a call comes in. I don't actually need a > PBX/softswitch. The user will have the FXO cards and a phoneset > connected on the same line, and will pick up the phone once the CRM > application has picked up the CID info and popped up a dialog box, > etc. > > Is it possible to just use OpenZAP to get this information, or must I > install Freeswitch and provide an IP phone as well? > > Thank you. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Sat Feb 28 08:34:16 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 28 Feb 2009 10:34:16 -0600 Subject: [Freeswitch-users] freeswitch with celt on windows In-Reply-To: <2cef777b0902280801o7380d81kf00be14f43c304a9@mail.gmail.com> References: <2cef777b0902280801o7380d81kf00be14f43c304a9@mail.gmail.com> Message-ID: <5006814D-09D1-40EB-92C4-CD76AD574ECE@freeswitch.org> Someone just needs to do the work of adding it to the build... I thought Carlos did this already... Are you on SVN Trunk? /b On Feb 28, 2009, at 10:01 AM, e schmidbauer wrote: > Hi. I was wondering if it is possible to compile freeswitch with the > celt codec on windows? I have been able to compile celt for windows > but when i compiled freeswitch i did not see the mod_celt in the > solution explorer. If this is possible could someone point me in the > right direction. Thank you. From brian at freeswitch.org Sat Feb 28 08:37:07 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 28 Feb 2009 10:37:07 -0600 Subject: [Freeswitch-users] SIP server? PBX vs. softswitch? In-Reply-To: <7.0.1.0.2.20090228120132.0285f8f8@fredshack.com> References: <7.0.1.0.2.20090228120132.0285f8f8@fredshack.com> Message-ID: It depends on how you look at it... most will say there is no difference... but last I checked you usually don't run heavy apps on a softswitch. FreeSWITCH can be everything from softphone to softswitch and everything in between including PBX. The default config comes configured as a PBX. /b On Feb 28, 2009, at 9:47 AM, Fred wrote: > Hello > > Even though I successfully set up an Asterisk voice server, I'm no > telecom expert, and would like some clarification about the > following things: > - What is an SIP server as opposed to a IP PBX? > - What is the different between a PBX like Asterisk and a softswitch? > > Thank you. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090228/9aa8feab/attachment.html From e.schmidbauer at gmail.com Sat Feb 28 08:55:25 2009 From: e.schmidbauer at gmail.com (e schmidbauer) Date: Sat, 28 Feb 2009 11:55:25 -0500 Subject: [Freeswitch-users] freeswitch with celt on windows In-Reply-To: <5006814D-09D1-40EB-92C4-CD76AD574ECE@freeswitch.org> References: <2cef777b0902280801o7380d81kf00be14f43c304a9@mail.gmail.com> <5006814D-09D1-40EB-92C4-CD76AD574ECE@freeswitch.org> Message-ID: <2cef777b0902280855sa804542g71d8d6d46e879f10@mail.gmail.com> i used the nightly snapshot... http://files.freeswitch.org/freeswitch-snapshot.tar.gz On Sat, Feb 28, 2009 at 11:34 AM, Brian West wrote: > Someone just needs to do the work of adding it to the build... I > thought Carlos did this already... Are you on SVN Trunk? > > /b > > On Feb 28, 2009, at 10:01 AM, e schmidbauer wrote: > >> Hi. I was wondering if it is possible to compile freeswitch with the >> celt codec on windows? I have been able to compile celt for windows >> but when i compiled freeswitch i did not see the mod_celt in the >> solution explorer. If this is possible could someone point me in the >> right direction. Thank you. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From dyfet at gnutelephony.org Sat Feb 28 09:20:53 2009 From: dyfet at gnutelephony.org (David Sugar) Date: Sat, 28 Feb 2009 12:20:53 -0500 Subject: [Freeswitch-users] SIP server? PBX vs. softswitch? In-Reply-To: References: <7.0.1.0.2.20090228120132.0285f8f8@fredshack.com> Message-ID: <49A97275.6000007@gnutelephony.org> Where this is distinguished, it is not directly at the level that user's experience the end result. In the case of what is called a "softswitch", one answer is found in organizations like the ISC (International Softswitch Consortium) and vendors who built products around their architecture recommendations. These systems tend to be very complex and componetized, where basic functionality operates in self-contained components that then interact with the whole through defined open standards and network protocols, such as SIP. The primary reason for ISC-style architectures is a result of proprietary development, where code and internal operations cannot be shared or modified. Hence, by breaking up functionality into subcomponents, it is possible to replace a component subsystem as a whole while retaining the interfaces. A perfect example is call forwarding. In a "traditional" proprietary (ISC-model) softswitch, call forwarding would be an entirely separate self-contained proprietary "feature" server interacting over SIP. If someone wants to create a different call forwarding behavior, one slips in an alternate server. By contrast, it is far easier in an open source/free software PBX to simply modify the feature code that implements call forwarding directly to create new and specialized versions of that feature. Hence, you do not find or have need for micro-services for tiny features in pbx software that originated as open source and free software or that did not follow the path of proprietary architectures, such as Bayonne, Asterisk, or FreeSwitch. A perfect example of a traditional "softswitch" architecture is SipX, which originated as a proprietary VoIP pbx codebase. However, even at this point, such distinctions I think are still somewhat artificial, as Brian suggests. What does distinguish architectures that may be relevant to end users is whether a IP-PBX solution operates as a B2BUA (back-to-back user agent) or not. A pure B2BUA solution is one where all media as well as signalling goes directly through the central PBX switch. A perfect example of this is how Asterisk traditionally works. This makes it very easy to adapt and connect multi-protocol endpoints, to convert media formats for endpoints who do not have common codecs, etc, since all media endpoints talk to the switch rather than each other. However, since all media goes through a central point, the scalability of such systems can often become "compute-bound", and extra latency is induced. A "pure" network solution by contrast has all media connect directly peer to peer by the user agent endpoints, and the "pbx" really only handles and coordinate independently operating endpoints through signalling. This often requires separate servers for gateways to the PSTN or other protocols. But it does offer better latency and scalability, and the ability to provide end-to-end media security, such as when using ZRTP. This difference, between B2BUA and non-B2BUA, is I think far more relevant today than traditional classifications such as IP-PBX, softswitch, "SIP Server", etc. Brian West wrote: > It depends on how you look at it... most will say there is no > difference... but last I checked you usually don't run heavy apps on a > softswitch. > > FreeSWITCH can be everything from softphone to softswitch and everything > in between including PBX. The default config comes configured as a PBX. > > /b > > On Feb 28, 2009, at 9:47 AM, Fred wrote: > >> Hello >> >> Even though I successfully set up an Asterisk voice server, I'm no >> telecom expert, and would like some clarification about the following >> things: >> - What is an SIP server as opposed to a IP PBX? >> - What is the different between a PBX like Asterisk and a softswitch? >> >> Thank you. > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- A non-text attachment was scrubbed... Name: dyfet.vcf Type: text/x-vcard Size: 177 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090228/dd828db8/attachment.vcf From nik.middleton at noblesolutions.co.uk Sat Feb 28 14:49:39 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Sat, 28 Feb 2009 22:49:39 -0000 Subject: [Freeswitch-users] Orginate: getting status of call fail Message-ID: Hi Guys, I've been running a test script written in lua which now works very well thanks to Anthony's fix to stream file. Right now I'm using an event socket to initiate the call and passing the name of the script along with originate thus: $dialstring = "originate {ignore_early_media=true,origination,originate_timeout=25}sofia/gateway/ Mygw/phonenum '&lua(helloworld.lua )'"; $result = $obj ->bgapi_command($dialstring); The script gets fired (it would appear) on answer. However, if the number is invalid , timed out or was busy, I'm not sure the script gets executed or am I wrong? I want to be able to fire an event back on what happed to the call in the event that it failed for whatever reason. I know I can simply call the originate and pass the number as an argument and execute the dial within the script but I'm led to believe that's not very efficient, or am I completely wrong? Looking for the most FS friendly way here Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090228/3b813f25/attachment.html From davidwdan at gmail.com Sat Feb 28 11:33:53 2009 From: davidwdan at gmail.com (David Dan) Date: Sat, 28 Feb 2009 14:33:53 -0500 Subject: [Freeswitch-users] Problems loading mod_spidermonkey_curl Message-ID: <65bd1c9f0902281133t2d79806bm63461288e4ca0c0f@mail.gmail.com> I'm getting the following error when I try to load the mod_spidermonkey_curl module. I didn't get any errors when I compiled it. I also tried --without-libcurl but I got the same result. Any help would be appreciated. freeswitch at internal> load mod_spidermonkey_curl -ERR [module load file routine returned an error] 2009-02-28 14:17:54 [CRIT] switch_loadable_module.c:839 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_spidermonkey_curl.so **/usr/local/freeswitch/mod/mod_spidermonkey_curl.so: undefined symbol: mod_spidermonkey_curl_module_interface** freeswitch at internal> version FreeSWITCH Version 1.0.3 (exported) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090228/68f593fc/attachment.html From brian at freeswitch.org Sat Feb 28 17:57:40 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 28 Feb 2009 19:57:40 -0600 Subject: [Freeswitch-users] Problems loading mod_spidermonkey_curl In-Reply-To: <65bd1c9f0902281133t2d79806bm63461288e4ca0c0f@mail.gmail.com> References: <65bd1c9f0902281133t2d79806bm63461288e4ca0c0f@mail.gmail.com> Message-ID: <885EAAC5-FDF0-4DA3-A835-EC652EDC2368@freeswitch.org> please open up spidermonkey.conf.xml and add it to the load there... its a sub module of mod_spidermonkey so you can't load it at the CLI /b On Feb 28, 2009, at 1:33 PM, David Dan wrote: > I'm getting the following error when I try to load the > mod_spidermonkey_curl module. I didn't get any errors when I > compiled it. I also tried --without-libcurl but I got the same > result. Any help would be appreciated. > > freeswitch at internal> load mod_spidermonkey_curl > -ERR [module load file routine returned an error] > > 2009-02-28 14:17:54 [CRIT] switch_loadable_module.c:839 > switch_loadable_module_load_file() Error Loading module /usr/local/ > freeswitch/mod/mod_spidermonkey_curl.so > **/usr/local/freeswitch/mod/mod_spidermonkey_curl.so: undefined > symbol: mod_spidermonkey_curl_module_interface** From davidwdan at gmail.com Sat Feb 28 20:08:46 2009 From: davidwdan at gmail.com (David Dan) Date: Sat, 28 Feb 2009 23:08:46 -0500 Subject: [Freeswitch-users] Problems loading mod_spidermonkey_curl In-Reply-To: <885EAAC5-FDF0-4DA3-A835-EC652EDC2368@freeswitch.org> References: <65bd1c9f0902281133t2d79806bm63461288e4ca0c0f@mail.gmail.com> <885EAAC5-FDF0-4DA3-A835-EC652EDC2368@freeswitch.org> Message-ID: <65bd1c9f0902282008o3d6b721bu4735c26d21e61d59@mail.gmail.com> That did it. Thank you On 2/28/09, Brian West wrote: > please open up spidermonkey.conf.xml and add it to the load there... > its a sub module of mod_spidermonkey so you can't load it at the CLI > > /b > > On Feb 28, 2009, at 1:33 PM, David Dan wrote: > >> I'm getting the following error when I try to load the >> mod_spidermonkey_curl module. I didn't get any errors when I >> compiled it. I also tried --without-libcurl but I got the same >> result. Any help would be appreciated. >> >> freeswitch at internal> load mod_spidermonkey_curl >> -ERR [module load file routine returned an error] >> >> 2009-02-28 14:17:54 [CRIT] switch_loadable_module.c:839 >> switch_loadable_module_load_file() Error Loading module /usr/local/ >> freeswitch/mod/mod_spidermonkey_curl.so >> **/usr/local/freeswitch/mod/mod_spidermonkey_curl.so: undefined >> symbol: mod_spidermonkey_curl_module_interface** > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Sent from my mobile device From gcd at i.ph Sun Feb 1 00:30:39 2009 From: gcd at i.ph (Nandy Dagondon) Date: Sun, 1 Feb 2009 16:30:39 +0800 Subject: [Freeswitch-users] TDM400 FXO can dialout only once In-Reply-To: <7d0bfd8c0901261639x7c53f256yabf7347b5d22e7e4@mail.gmail.com> References: <7d0bfd8c0901250021q2969fe26u5ea79c3957989ffd@mail.gmail.com> <191c3a030901251818g284054ben60c0bfa61157824@mail.gmail.com> <7d0bfd8c0901252305v5a33fa02w4bc02deaa0c1b14b@mail.gmail.com> <87f2f3b90901260201r2e4d42a3x30367eee64b0e13b@mail.gmail.com> <7d0bfd8c0901260532g50deb359v3c92f5bb372f11a4@mail.gmail.com> <191c3a030901260833l2e07bfffkca9400d33a5483fe@mail.gmail.com> <7d0bfd8c0901261639x7c53f256yabf7347b5d22e7e4@mail.gmail.com> Message-ID: <7d0bfd8c0902010030n6b5d3c35k7539fac38e07300d@mail.gmail.com> hi everybody, i created [ph] tone definition per raul's suggestion and changed /etc/zaptel.conf entries to: tonezone=ph defaultzone=ph but it didn't solve the problem. i captured the console log during start-up and shutdown. i noticed openzap related errors during shutdown. here's the snippet of the log: STARTUP --------------- 2009-02-01 15:58:10 [NOTICE] zap_io.c:2517 zap_global_init() Modules configured: 1 2009-02-01 15:58:10 [INFO] zap_io.c:2341 zap_load_module() Loading IO from /opt/freeswitch/mod/ozmod_zt.so 2009-02-01 15:58:10 [INFO] zap_io.c:2127 load_config() auto-loaded 'zt' 2009-02-01 15:58:10 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 1 as OpenZAP device 1:1 fd:39 2009-02-01 15:58:10 [INFO] ozmod_zt.c:186 zt_open_range() configuring device /dev/zap/channel channel 2 as OpenZAP device 2:1 fd:40 2009-02-01 15:58:10 [INFO] zap_io.c:2265 load_config() Configured 2 channel(s) 2009-02-01 15:58:10 [INFO] zap_io.c:2358 zap_load_module() Loading SIG from /opt/freeswitch/mod/ozmod_analog.so 2009-02-01 15:58:10 [INFO] zap_io.c:2474 zap_configure_span() auto-loaded 'analog' 2009-02-01 15:58:10 [CONSOLE] switch_loadable_module.c:857 switch_loadable_module_load_file() Successfully Loaded [mod_openzap] --- DIDN'T MAKE ANY CALL --- SHUTDOWN ------------------ 2009-02-01 15:59:07 [NOTICE] switch_loadable_module.c:536 switch_loadable_module_unprocess() Deleting API Function 'oz' 2009-02-01 15:59:07 [CONSOLE] switch_loadable_module.c:1231 do_shutdown() Stopping: mod_openzap 2009-02-01 15:59:07 [INFO] zap_io.c:256 zap_channel_destroy() Closing channel zt:1:1 fd:39 2009-02-01 15:59:07 [INFO] zap_io.c:256 zap_channel_destroy() Closing channel zt:2:1 fd:40 2009-02-01 15:59:08 [ERR] ozmod_analog.c:899 zap_analog_run() Failure Polling event! [no matching descriptor] 2009-02-01 15:59:08 [ERR] ozmod_analog.c:899 zap_analog_run() Failure Polling event! [no matching descriptor] 2009-02-01 15:59:08 [INFO] zap_io.c:2456 zap_unload_modules() Unloading /opt/freeswitch/mod/ozmod_analog.so 2009-02-01 15:59:08 [INFO] zap_io.c:2441 zap_unload_modules() Unloading IO zt 2009-02-01 15:59:08 [INFO] zap_io.c:2456 zap_unload_modules() Unloading /opt/freeswitch/mod/ozmod_zt.so i also notice the same ERR flag during shutdown after making test calls. any suggestion what to do next? tks for your assistance. rgds, -nandy On Tue, Jan 27, 2009 at 8:39 AM, Nandy Dagondon wrote: > i tested the SVN trunk version. still the same behaviour. > -nandy > > > On Tue, Jan 27, 2009 at 12:33 AM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> The unhanded type message just means that mod_openzap does not do anything >> with the TONE_DETECTED event that was passed >> up from the ozmod_analog. >> >> On Mon, Jan 26, 2009 at 7:32 AM, Nandy Dagondon wrote: >> >>> that's great. yes, i'm in the philippines. there's a difference in >>> dialtone - it's 425 Hz. >>> -nandy >>> >>> >>> >>> On Mon, Jan 26, 2009 at 6:01 PM, Michael Collins wrote: >>> >>>> I have a TDM400 clone and I will see if I can reproduce these >>>> symptoms. BTW, are you in the Philippines? Is there any difference in >>>> the dial tone there than in the US? >>>> -MC >>>> >>>> On Sun, Jan 25, 2009 at 11:05 PM, Nandy Dagondon wrote: >>>> > i monitored the line using another phone. there's indeed dialtone in >>>> all >>>> > attempts. >>>> > i see TONE_DETECTED in the first call but i wonder there's a WARNING >>>> message >>>> > immediately following [WARNING] mod_openzap.c:1196 on_fxo_signal() >>>> Unhandled >>>> > type for channel 2:1. >>>> > the dialtone freq should be okay since it's detected in the first >>>> call.could >>>> > the WARNING message gives us a hint of a possible problem other than >>>> the >>>> > dialtone freq? >>>> > >>>> > okay, i'll try the SVN version next. >>>> > >>>> > >>>> > On Mon, Jan 26, 2009 at 10:18 AM, Anthony Minessale >>>> > wrote: >>>> >> >>>> >> Its not detecting a dial tone on the failure case. >>>> >> Before dialing it waits until it picks up dialtone. >>>> >> Try the svn trunk version to see if it works any better or verify >>>> there is >>>> >> a dialtone on the line. >>>> >> >>>> >> On Jan 25, 2009 6:19 PM, "Nandy Dagondon" wrote: >>>> >> >>>> >> hi everybody, >>>> >> >>>> >> i installed FS 1.0.2 to an Atom mobo (Intel D945GCLF2). it's working >>>> using >>>> >> IP phones, softphones and digium FXS port. but there's a problem in >>>> dialing >>>> >> out to PSTN using digium tdm400 fxo - it works fine on the first >>>> attempt >>>> >> (after starting FS) but it fails on the subsequent attempts. i tested >>>> to >>>> >> call using the FXS port and IP phone. same problem. >>>> >> >>>> >> before i place any call, i checked >oz dump 2 1 (show current state >>>> = >>>> >> DOWN, last state = DOWN) >>>> >> >>>> >> in the first call, there's this message: >>>> >> [WARNING] mod_openzap.c:1196 on_fxo_signal() Unhandled type for >>>> channel >>>> >> 2:1 >>>> >> but >>>> >> >>>> >> then i hangup. checked >oz dump 21 (show current state=DOWN, last >>>> >> state=HANGUP) >>>> >> >>>> >> in the 2nd (and subsequent) attempts, the fxo just goes off-hook but >>>> >> doesn't send the dtmf tones. >>>> >> >oz dump 2 1 (shows current state = DIALING, last state = DOWN) >>>> >> >>>> >> has anyone encountered this problem before? i appreciate for any help >>>> to >>>> >> correct this problem. >>>> >> >>>> >> tks, >>>> >> nandy >>>> >> >>>> >> >>>> >> Environment: >>>> >> ================== >>>> >> kernel 2.6.18-92.1.22.el5 >>>> >> FS 1.0.2 >>>> >> zaptel 1.4.11 >>>> >> oslec >>>> >> digium TDM400P Rev. I (1-FXS, 2-FXO,3-4 vacant) >>>> >> >>>> >> zaptel.conf >>>> >> ======== >>>> >> loadzone = us >>>> >> defaultzone=us >>>> >> channels=1-2 >>>> >> alaw=1-4 >>>> >> fxsks=2 >>>> >> fxoks=1 >>>> >> >>>> >> >>>> >> openzap.conf.xml: >>>> >> =============== >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> openzap.conf >>>> >> ========== >>>> >> [span zt] >>>> >> name => OpenZAP FXS >>>> >> number => 1 >>>> >> fxs-channel => 1 >>>> >> >>>> >> [span zt] >>>> >> name => OpenZAP FXO >>>> >> number => 2 >>>> >> fxo-channel => 2 >>>> >> >>>> >> tones.conf (the dialtone and ring tone is set to Philipping tones) >>>> >> ======== >>>> >> [us] >>>> >> generate-dial => v=-7;%(1000,0,425) >>>> >> detect-dial => 425 >>>> >> >>>> >> generate-ring => v=-7;%(1000,4000,425,480) >>>> >> detect-ring => 425,480 >>>> >> >>>> >> generate-busy => v=-7;%(500,500,480,620) >>>> >> detect-busy => 480,620 >>>> >> >>>> >> generate-attn => v=0;%(200,300,1400,1800) >>>> >> detect-attn => 1400,1800 >>>> >> >>>> >> generate-callwaiting-sas => v=0;%(300,10000,440) >>>> >> detect-callwaiting-sas => 440 >>>> >> >>>> >> generate-callwaiting-cas => v=0;%(80,0,2750,2130) >>>> >> detect-callwaiting-cas => 2750,2130 >>>> >> >>>> >> detect-fail1 => 913.8 >>>> >> detect-fail2 => 1370.6 >>>> >> detect-fail3 => 776.7 >>>> >> >>>> >> LOG OF FIRST CALL (OK) >>>> >> ==================== >>>> >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:152 >>>> >> switch_core_standard_on_execute() OpenZAP/1:1/93400534 Execute >>>> >> bridge(openzap/2/1/3400534) >>>> >> 2009-01-25 10:35:58 [DEBUG] mod_openzap.c:340 tech_init() Set codec >>>> PCMU >>>> >> 20ms >>>> >> 2009-01-25 10:35:58 [DEBUG] mod_openzap.c:1017 >>>> channel_outgoing_channel() >>>> >> Connect outbound channel OpenZAP/2:1/3400534 >>>> >> 2009-01-25 10:35:58 [NOTICE] switch_channel.c:565 >>>> >> switch_channel_set_name() New Channel OpenZAP/2:1/3400534 >>>> >> [e5f12114-ea88-11dd-9f5c-290fb4a527a4] >>>> >> 2009-01-25 10:35:58 [DEBUG] mod_openzap.c:1026 >>>> channel_outgoing_channel() >>>> >> (OpenZAP/2:1/3400534) State Change CS_NEW -> CS_INIT >>>> >> 2009-01-25 10:35:58 [DEBUG] switch_core_session.c:807 >>>> >> switch_core_session_signal_state_change() Send signal >>>> OpenZAP/2:1/3400534 >>>> >> [BREAK] >>>> >> 2009-01-25 10:35:58 [DEBUG] ozmod_analog.c:52 >>>> analog_fxo_outgoing_call() >>>> >> Changing state on 2:1 from DOWN to DIALING >>>> >> 2009-01-25 10:35:58 [DEBUG] ozmod_analog.c:239 >>>> zap_analog_channel_run() >>>> >> ANALOG CHANNEL thread starting. >>>> >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:379 >>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change >>>> CS_INIT >>>> >> 2009-01-25 10:35:58 [DEBUG] ozmod_analog.c:410 >>>> zap_analog_channel_run() >>>> >> Executing state handler on 2:1 for DIALING >>>> >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:444 >>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State INIT >>>> >> 2009-01-25 10:35:58 [DEBUG] mod_openzap.c:364 channel_on_init() >>>> >> (OpenZAP/2:1/3400534) State Change CS_INIT -> CS_ROUTING >>>> >> 2009-01-25 10:35:58 [DEBUG] switch_core_session.c:807 >>>> >> switch_core_session_signal_state_change() Send signal >>>> OpenZAP/2:1/3400534 >>>> >> [BREAK] >>>> >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:444 >>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State INIT going to >>>> sleep >>>> >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:379 >>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change >>>> >> CS_ROUTING >>>> >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:447 >>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State ROUTING >>>> >> 2009-01-25 10:35:58 [DEBUG] mod_openzap.c:387 channel_on_routing() >>>> >> OpenZAP/2:1/3400534 CHANNEL ROUTING >>>> >> 2009-01-25 10:35:58 [DEBUG] switch_ivr_originate.c:58 >>>> >> originate_on_routing() (OpenZAP/2:1/3400534) State Change CS_ROUTING >>>> -> >>>> >> CS_CONSUME_MEDIA >>>> >> 2009-01-25 10:35:58 [DEBUG] switch_core_session.c:807 >>>> >> switch_core_session_signal_state_change() Send signal >>>> OpenZAP/2:1/3400534 >>>> >> [BREAK] >>>> >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:447 >>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State ROUTING going >>>> to sleep >>>> >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:379 >>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change >>>> >> CS_CONSUME_MEDIA >>>> >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:466 >>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State CONSUME_MEDIA >>>> >> 2009-01-25 10:35:59 [DEBUG] ozmod_analog.c:615 >>>> zap_analog_channel_run() >>>> >> Detected tone DIAL on 2:1 >>>> >> 2009-01-25 10:35:59 [DEBUG] mod_openzap.c:1153 on_fxo_signal() got >>>> FXO sig >>>> >> 2:1 [TONE_DETECTED] >>>> >> 2009-01-25 10:35:59 [WARNING] mod_openzap.c:1196 on_fxo_signal() >>>> Unhandled >>>> >> type for channel 2:1 >>>> >> 2009-01-25 10:35:59 [DEBUG] zap_io.c:1179 >>>> zchan_activate_dtmf_buffer() >>>> >> Created DTMF Buffer! >>>> >> 2009-01-25 10:35:59 [DEBUG] zap_io.c:1715 handle_dtmf() 2:1 GENERATE >>>> DTMF >>>> >> [3400534] >>>> >> 2009-01-25 10:36:03 [DEBUG] ozmod_analog.c:302 >>>> zap_analog_channel_run() >>>> >> Changing state on 2:1 from DIALING to UP >>>> >> 2009-01-25 10:36:03 [DEBUG] ozmod_analog.c:410 >>>> zap_analog_channel_run() >>>> >> Executing state handler on 2:1 for UP >>>> >> 2009-01-25 10:36:03 [DEBUG] mod_openzap.c:1153 on_fxo_signal() got >>>> FXO sig >>>> >> 2:1 [UP] >>>> >> 2009-01-25 10:36:03 [DEBUG] switch_channel.c:1710 >>>> >> switch_channel_perform_mark_answered() Send signal >>>> OpenZAP/1:1/93400534 >>>> >> [BREAK] >>>> >> 2009-01-25 10:36:03 [NOTICE] mod_openzap.c:1180 on_fxo_signal() >>>> Channel >>>> >> [OpenZAP/2:1/3400534] has been answered >>>> >> 2009-01-25 10:36:03 [DEBUG] switch_channel.c:177 >>>> >> switch_channel_audio_sync() OpenZAP/2:1/3400534 receive message >>>> [AUDIO_SYNC] >>>> >> 2009-01-25 10:36:03 [DEBUG] switch_channel.c:1768 >>>> >> switch_channel_perform_answer() OpenZAP/1:1/93400534 receive message >>>> >> [ANSWER] >>>> >> 2009-01-25 10:36:03 [DEBUG] mod_openzap.c:813 >>>> >> channel_receive_message_fxs() Changing state on 1:1 from IDLE to UP >>>> >> 2009-01-25 10:36:03 [NOTICE] mod_openzap.c:814 >>>> >> channel_receive_message_fxs() Channel [OpenZAP/1:1/93400534] has been >>>> >> answered >>>> >> 2009-01-25 10:36:03 [DEBUG] switch_channel.c:177 >>>> >> switch_channel_audio_sync() OpenZAP/1:1/93400534 receive message >>>> >> [AUDIO_SYNC] >>>> >> 2009-01-25 10:36:03 [DEBUG] switch_core_session.c:511 >>>> >> switch_core_session_perform_receive_message() Send signal >>>> >> OpenZAP/1:1/93400534 [BREAK] >>>> >> 2009-01-25 10:36:03 [DEBUG] switch_ivr_originate.c:1627 >>>> >> switch_ivr_originate() Originate Resulted in Success: >>>> [OpenZAP/2:1/3400534] >>>> >> 2009-01-25 10:36:03 [DEBUG] switch_channel.c:177 >>>> >> switch_channel_audio_sync() OpenZAP/2:1/3400534 receive message >>>> [AUDIO_SYNC] >>>> >> 2009-01-25 10:36:03 [DEBUG] switch_channel.c:177 >>>> >> switch_channel_audio_sync() OpenZAP/1:1/93400534 receive message >>>> >> [AUDIO_SYNC] >>>> >> 2009-01-25 10:36:03 [DEBUG] switch_ivr_bridge.c:862 >>>> >> switch_ivr_multi_threaded_bridge() OpenZAP/2:1/3400534 receive >>>> message >>>> >> [BRIDGE] >>>> >> 2009-01-25 10:36:03 [DEBUG] switch_core_session.c:511 >>>> >> switch_core_session_perform_receive_message() Send signal >>>> >> OpenZAP/2:1/3400534 [BREAK] >>>> >> 2009-01-25 10:36:03 [DEBUG] switch_ivr_bridge.c:869 >>>> >> switch_ivr_multi_threaded_bridge() OpenZAP/1:1/93400534 receive >>>> message >>>> >> [BRIDGE] >>>> >> 2009-01-25 10:36:03 [DEBUG] switch_core_session.c:511 >>>> >> switch_core_session_perform_receive_message() Send signal >>>> >> OpenZAP/1:1/93400534 [BREAK] >>>> >> 2009-01-25 10:36:03 [DEBUG] switch_ivr_bridge.c:913 >>>> >> switch_ivr_multi_threaded_bridge() (OpenZAP/2:1/3400534) State Change >>>> >> CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA >>>> >> 2009-01-25 10:36:03 [DEBUG] switch_core_session.c:807 >>>> >> switch_core_session_signal_state_change() Send signal >>>> OpenZAP/2:1/3400534 >>>> >> [BREAK] >>>> >> 2009-01-25 10:36:03 [DEBUG] ozmod_analog.c:410 >>>> zap_analog_channel_run() >>>> >> Executing state handler on 1:1 for UP >>>> >> 2009-01-25 10:36:03 [DEBUG] mod_openzap.c:1212 on_fxs_signal() got >>>> FXS sig >>>> >> [UP] >>>> >> 2009-01-25 10:36:03 [DEBUG] switch_core_state_machine.c:466 >>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State CONSUME_MEDIA >>>> going to >>>> >> sleep >>>> >> 2009-01-25 10:36:03 [DEBUG] switch_core_state_machine.c:379 >>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change >>>> >> CS_EXCHANGE_MEDIA >>>> >> 2009-01-25 10:36:03 [DEBUG] switch_core_state_machine.c:457 >>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State EXCHANGE_MEDIA >>>> >> 2009-01-25 10:36:03 [DEBUG] mod_openzap.c:511 >>>> channel_on_exchange_media() >>>> >> CHANNEL EXCHANGE_MEDIA >>>> >> 2009-01-25 10:36:16 [DEBUG] ozmod_analog.c:744 process_event() EVENT >>>> >> [ONHOOK][1:1] STATE [UP] >>>> >> 2009-01-25 10:36:16 [DEBUG] ozmod_analog.c:780 process_event() >>>> Changing >>>> >> state on 1:1 from UP to DOWN >>>> >> 2009-01-25 10:36:16 [DEBUG] ozmod_analog.c:410 >>>> zap_analog_channel_run() >>>> >> Executing state handler on 1:1 for DOWN >>>> >> 2009-01-25 10:36:16 [DEBUG] mod_openzap.c:1212 on_fxs_signal() got >>>> FXS sig >>>> >> [STOP] >>>> >> 2009-01-25 10:36:16 [NOTICE] mod_openzap.c:1300 on_fxs_signal() >>>> Hangup >>>> >> OpenZAP/1:1/93400534 [CS_EXECUTE] [NORMAL_CLEARING] >>>> >> 2009-01-25 10:36:16 [DEBUG] switch_channel.c:1494 >>>> >> switch_channel_perform_hangup() Send signal OpenZAP/1:1/93400534 >>>> [KILL] >>>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_session.c:807 >>>> >> switch_core_session_signal_state_change() Send signal >>>> OpenZAP/1:1/93400534 >>>> >> [BREAK] >>>> >> 2009-01-25 10:36:16 [DEBUG] zap_io.c:1125 zap_channel_done() channel >>>> done >>>> >> 1:1 >>>> >> 2009-01-25 10:36:16 [DEBUG] ozmod_analog.c:726 >>>> zap_analog_channel_run() >>>> >> ANALOG CHANNEL 1:1 thread ended. >>>> >> 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:360 >>>> audio_bridge_thread() >>>> >> OpenZAP/1:1/93400534 ending bridge by request from read function >>>> >> 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:435 >>>> audio_bridge_thread() >>>> >> BRIDGE THREAD DONE [OpenZAP/1:1/93400534] >>>> >> 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:439 >>>> audio_bridge_thread() >>>> >> Send signal OpenZAP/2:1/3400534 [BREAK] >>>> >> 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:354 >>>> audio_bridge_thread() >>>> >> OpenZAP/1:1/93400534 ending bridge by request from write function >>>> >> 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:409 >>>> audio_bridge_thread() >>>> >> OpenZAP/2:1/3400534 receive message [UNBRIDGE] >>>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_session.c:511 >>>> >> switch_core_session_perform_receive_message() Send signal >>>> >> OpenZAP/2:1/3400534 [BREAK] >>>> >> 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:435 >>>> audio_bridge_thread() >>>> >> BRIDGE THREAD DONE [OpenZAP/2:1/3400534] >>>> >> 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:439 >>>> audio_bridge_thread() >>>> >> Send signal OpenZAP/1:1/93400534 [BREAK] >>>> >> 2009-01-25 10:36:16 [NOTICE] switch_ivr_bridge.c:470 >>>> >> audio_bridge_on_exchange_media() Hangup OpenZAP/2:1/3400534 >>>> >> [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] >>>> >> 2009-01-25 10:36:16 [DEBUG] switch_channel.c:1494 >>>> >> switch_channel_perform_hangup() Send signal OpenZAP/2:1/3400534 >>>> [KILL] >>>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_session.c:807 >>>> >> switch_core_session_signal_state_change() Send signal >>>> OpenZAP/2:1/3400534 >>>> >> [BREAK] >>>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:457 >>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State EXCHANGE_MEDIA >>>> going >>>> >> to sleep >>>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:379 >>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change >>>> >> CS_HANGUP >>>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:410 >>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State HANGUP >>>> >> 2009-01-25 10:36:16 [DEBUG] mod_openzap.c:429 channel_on_hangup() >>>> Changing >>>> >> state on 2:1 from UP to HANGUP >>>> >> 2009-01-25 10:36:16 [DEBUG] mod_openzap.c:472 channel_on_hangup() >>>> >> OpenZAP/2:1/3400534 CHANNEL HANGUP >>>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:46 >>>> >> switch_core_standard_on_hangup() OpenZAP/2:1/3400534 Standard HANGUP, >>>> cause: >>>> >> NORMAL_CLEARING >>>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:410 >>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State HANGUP going to >>>> sleep >>>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_session.c:939 >>>> >> switch_core_session_thread() Session 2 (OpenZAP/2:1/3400534) Locked, >>>> Waiting >>>> >> on external entities >>>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:454 >>>> >> switch_core_session_run() (OpenZAP/1:1/93400534) State EXECUTE going >>>> to >>>> >> sleep >>>> >> 2009-01-25 10:36:16 [NOTICE] switch_core_session.c:957 >>>> >> switch_core_session_thread() Session 2 (OpenZAP/2:1/3400534) Ended >>>> >> 2009-01-25 10:36:16 [NOTICE] switch_core_session.c:959 >>>> >> switch_core_session_thread() Close Channel OpenZAP/2:1/3400534 >>>> [CS_HANGUP] >>>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:379 >>>> >> switch_core_session_run() (OpenZAP/1:1/93400534) Running State Change >>>> >> CS_HANGUP >>>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:410 >>>> >> switch_core_session_run() (OpenZAP/1:1/93400534) State HANGUP >>>> >> 2009-01-25 10:36:16 [DEBUG] mod_openzap.c:472 channel_on_hangup() >>>> >> OpenZAP/1:1/93400534 CHANNEL HANGUP >>>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:46 >>>> >> switch_core_standard_on_hangup() OpenZAP/1:1/93400534 Standard >>>> HANGUP, >>>> >> cause: NORMAL_CLEARING >>>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:410 >>>> >> switch_core_session_run() (OpenZAP/1:1/93400534) State HANGUP going >>>> to sleep >>>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_session.c:939 >>>> >> switch_core_session_thread() Session 1 (OpenZAP/1:1/93400534) Locked, >>>> >> Waiting on external entities >>>> >> 2009-01-25 10:36:16 [NOTICE] switch_core_session.c:957 >>>> >> switch_core_session_thread() Session 1 (OpenZAP/1:1/93400534) Ended >>>> >> 2009-01-25 10:36:16 [NOTICE] switch_core_session.c:959 >>>> >> switch_core_session_thread() Close Channel OpenZAP/1:1/93400534 >>>> [CS_HANGUP] >>>> >> 2009-01-25 10:36:16 [DEBUG] ozmod_analog.c:410 >>>> zap_analog_channel_run() >>>> >> Executing state handler on 2:1 for HANGUP >>>> >> 2009-01-25 10:36:17 [DEBUG] ozmod_analog.c:351 >>>> zap_analog_channel_run() >>>> >> Changing state on 2:1 from HANGUP to DOWN >>>> >> 2009-01-25 10:36:17 [DEBUG] ozmod_analog.c:410 >>>> zap_analog_channel_run() >>>> >> Executing state handler on 2:1 for DOWN >>>> >> 2009-01-25 10:36:17 [DEBUG] mod_openzap.c:1153 on_fxo_signal() got >>>> FXO sig >>>> >> 2:1 [STOP] >>>> >> 2009-01-25 10:36:17 [DEBUG] zap_io.c:1125 zap_channel_done() channel >>>> done >>>> >> 2:1 >>>> >> 2009-01-25 10:36:17 [DEBUG] ozmod_analog.c:726 >>>> zap_analog_channel_run() >>>> >> ANALOG CHANNEL 2:1 thread ended. >>>> >> >>>> >> LOG OF FAILED CALLS >>>> >> ================== >>>> >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:152 >>>> >> switch_core_standard_on_execute() OpenZAP/1:1/93400534 Execute >>>> >> bridge(openzap/2/1/3400534) >>>> >> 2009-01-25 10:36:55 [DEBUG] mod_openzap.c:340 tech_init() Set codec >>>> PCMU >>>> >> 20ms >>>> >> 2009-01-25 10:36:55 [DEBUG] mod_openzap.c:1017 >>>> channel_outgoing_channel() >>>> >> Connect outbound channel OpenZAP/2:1/3400534 >>>> >> 2009-01-25 10:36:55 [NOTICE] switch_channel.c:565 >>>> >> switch_channel_set_name() New Channel OpenZAP/2:1/3400534 >>>> >> [079f5420-ea89-11dd-9f5c-290fb4a527a4] >>>> >> 2009-01-25 10:36:55 [DEBUG] mod_openzap.c:1026 >>>> channel_outgoing_channel() >>>> >> (OpenZAP/2:1/3400534) State Change CS_NEW -> CS_INIT >>>> >> 2009-01-25 10:36:55 [DEBUG] switch_core_session.c:807 >>>> >> switch_core_session_signal_state_change() Send signal >>>> OpenZAP/2:1/3400534 >>>> >> [BREAK] >>>> >> 2009-01-25 10:36:55 [DEBUG] ozmod_analog.c:52 >>>> analog_fxo_outgoing_call() >>>> >> Changing state on 2:1 from DOWN to DIALING >>>> >> 2009-01-25 10:36:55 [DEBUG] ozmod_analog.c:239 >>>> zap_analog_channel_run() >>>> >> ANALOG CHANNEL thread starting. >>>> >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:379 >>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change >>>> CS_INIT >>>> >> 2009-01-25 10:36:55 [DEBUG] ozmod_analog.c:410 >>>> zap_analog_channel_run() >>>> >> Executing state handler on 2:1 for DIALING >>>> >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:444 >>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State INIT >>>> >> 2009-01-25 10:36:55 [DEBUG] mod_openzap.c:364 channel_on_init() >>>> >> (OpenZAP/2:1/3400534) State Change CS_INIT -> CS_ROUTING >>>> >> 2009-01-25 10:36:55 [DEBUG] switch_core_session.c:807 >>>> >> switch_core_session_signal_state_change() Send signal >>>> OpenZAP/2:1/3400534 >>>> >> [BREAK] >>>> >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:444 >>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State INIT going to >>>> sleep >>>> >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:379 >>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change >>>> >> CS_ROUTING >>>> >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:447 >>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State ROUTING >>>> >> 2009-01-25 10:36:55 [DEBUG] mod_openzap.c:387 channel_on_routing() >>>> >> OpenZAP/2:1/3400534 CHANNEL ROUTING >>>> >> 2009-01-25 10:36:55 [DEBUG] switch_ivr_originate.c:58 >>>> >> originate_on_routing() (OpenZAP/2:1/3400534) State Change CS_ROUTING >>>> -> >>>> >> CS_CONSUME_MEDIA >>>> >> 2009-01-25 10:36:55 [DEBUG] switch_core_session.c:807 >>>> >> switch_core_session_signal_state_change() Send signal >>>> OpenZAP/2:1/3400534 >>>> >> [BREAK] >>>> >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:447 >>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State ROUTING going >>>> to sleep >>>> >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:379 >>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change >>>> >> CS_CONSUME_MEDIA >>>> >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:466 >>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State CONSUME_MEDIA >>>> >> 2009-01-25 10:37:08 [DEBUG] ozmod_analog.c:744 process_event() EVENT >>>> >> [ONHOOK][1:1] STATE [IDLE] >>>> >> 2009-01-25 10:37:08 [DEBUG] ozmod_analog.c:780 process_event() >>>> Changing >>>> >> state on 1:1 from IDLE to DOWN >>>> >> 2009-01-25 10:37:08 [DEBUG] ozmod_analog.c:410 >>>> zap_analog_channel_run() >>>> >> Executing state handler on 1:1 for DOWN >>>> >> 2009-01-25 10:37:08 [DEBUG] mod_openzap.c:1212 on_fxs_signal() got >>>> FXS sig >>>> >> [STOP] >>>> >> 2009-01-25 10:37:08 [NOTICE] mod_openzap.c:1300 on_fxs_signal() >>>> Hangup >>>> >> OpenZAP/1:1/93400534 [CS_EXECUTE] [NORMAL_CLEARING] >>>> >> 2009-01-25 10:37:08 [DEBUG] switch_channel.c:1494 >>>> >> switch_channel_perform_hangup() Send signal OpenZAP/1:1/93400534 >>>> [KILL] >>>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_session.c:807 >>>> >> switch_core_session_signal_state_change() Send signal >>>> OpenZAP/1:1/93400534 >>>> >> [BREAK] >>>> >> 2009-01-25 10:37:08 [DEBUG] zap_io.c:1125 zap_channel_done() channel >>>> done >>>> >> 1:1 >>>> >> 2009-01-25 10:37:08 [DEBUG] ozmod_analog.c:726 >>>> zap_analog_channel_run() >>>> >> ANALOG CHANNEL 1:1 thread ended. >>>> >> 2009-01-25 10:37:08 [NOTICE] switch_ivr_originate.c:1566 >>>> >> switch_ivr_originate() Hangup OpenZAP/2:1/3400534 [CS_CONSUME_MEDIA] >>>> >> [ORIGINATOR_CANCEL] >>>> >> 2009-01-25 10:37:08 [DEBUG] switch_channel.c:1494 >>>> >> switch_channel_perform_hangup() Send signal OpenZAP/2:1/3400534 >>>> [KILL] >>>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_session.c:807 >>>> >> switch_core_session_signal_state_change() Send signal >>>> OpenZAP/2:1/3400534 >>>> >> [BREAK] >>>> >> 2009-01-25 10:37:08 [DEBUG] switch_ivr_originate.c:1691 >>>> >> switch_ivr_originate() Originate Cancelled by originator termination >>>> Cause: >>>> >> 487 [ORIGINATOR_CANCEL] >>>> >> 2009-01-25 10:37:08 [INFO] mod_dptools.c:1909 audio_bridge_function() >>>> >> Originate Failed. Cause: ORIGINATOR_CANCEL >>>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:454 >>>> >> switch_core_session_run() (OpenZAP/1:1/93400534) State EXECUTE going >>>> to >>>> >> sleep >>>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:379 >>>> >> switch_core_session_run() (OpenZAP/1:1/93400534) Running State Change >>>> >> CS_HANGUP >>>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:410 >>>> >> switch_core_session_run() (OpenZAP/1:1/93400534) State HANGUP >>>> >> 2009-01-25 10:37:08 [DEBUG] mod_openzap.c:472 channel_on_hangup() >>>> >> OpenZAP/1:1/93400534 CHANNEL HANGUP >>>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:46 >>>> >> switch_core_standard_on_hangup() OpenZAP/1:1/93400534 Standard >>>> HANGUP, >>>> >> cause: NORMAL_CLEARING >>>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:410 >>>> >> switch_core_session_run() (OpenZAP/1:1/93400534) State HANGUP going >>>> to sleep >>>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_session.c:939 >>>> >> switch_core_session_thread() Session 3 (OpenZAP/1:1/93400534) Locked, >>>> >> Waiting on external entities >>>> >> 2009-01-25 10:37:08 [NOTICE] switch_core_session.c:957 >>>> >> switch_core_session_thread() Session 3 (OpenZAP/1:1/93400534) Ended >>>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:466 >>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State CONSUME_MEDIA >>>> going to >>>> >> sleep >>>> >> 2009-01-25 10:37:08 [NOTICE] switch_core_session.c:959 >>>> >> switch_core_session_thread() Close Channel OpenZAP/1:1/93400534 >>>> [CS_HANGUP] >>>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:379 >>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change >>>> >> CS_HANGUP >>>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:410 >>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State HANGUP >>>> >> 2009-01-25 10:37:08 [DEBUG] mod_openzap.c:429 channel_on_hangup() >>>> Changing >>>> >> state on 2:1 from DIALING to HANGUP >>>> >> 2009-01-25 10:37:08 [DEBUG] mod_openzap.c:472 channel_on_hangup() >>>> >> OpenZAP/2:1/3400534 CHANNEL HANGUP >>>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:46 >>>> >> switch_core_standard_on_hangup() OpenZAP/2:1/3400534 Standard HANGUP, >>>> cause: >>>> >> ORIGINATOR_CANCEL >>>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:410 >>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State HANGUP going to >>>> sleep >>>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_session.c:939 >>>> >> switch_core_session_thread() Session 4 (OpenZAP/2:1/3400534) Locked, >>>> Waiting >>>> >> on external entities >>>> >> 2009-01-25 10:37:08 [NOTICE] switch_core_session.c:957 >>>> >> switch_core_session_thread() Session 4 (OpenZAP/2:1/3400534) Ended >>>> >> 2009-01-25 10:37:08 [NOTICE] switch_core_session.c:959 >>>> >> switch_core_session_thread() Close Channel OpenZAP/2:1/3400534 >>>> [CS_HANGUP] >>>> >> 2009-01-25 10:37:08 [DEBUG] ozmod_analog.c:410 >>>> zap_analog_channel_run() >>>> >> Executing state handler on 2:1 for HANGUP >>>> >> 2009-01-25 10:37:09 [DEBUG] ozmod_analog.c:351 >>>> zap_analog_channel_run() >>>> >> Changing state on 2:1 from HANGUP to DOWN >>>> >> 2009-01-25 10:37:09 [DEBUG] ozmod_analog.c:410 >>>> zap_analog_channel_run() >>>> >> Executing state handler on 2:1 for DOWN >>>> >> 2009-01-25 10:37:09 [DEBUG] mod_openzap.c:1153 on_fxo_signal() got >>>> FXO sig >>>> >> 2:1 [STOP] >>>> >> 2009-01-25 10:37:09 [DEBUG] zap_io.c:1125 zap_channel_done() channel >>>> done >>>> >> 2:1 >>>> >> 2009-01-25 10:37:09 [DEBUG] ozmod_analog.c:726 >>>> zap_analog_channel_run() >>>> >> ANALOG CHANNEL 2:1 thread ended. >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> _______________________________________________ >>>> >> Freeswitch-users mailing list >>>> >> Freeswitch-users at lists.freeswitch.org >>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> >> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> >> http://www.freeswitch.org >>>> >> >>>> >> >>>> >> _______________________________________________ >>>> >> Freeswitch-users mailing list >>>> >> Freeswitch-users at lists.freeswitch.org >>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> >> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> >> http://www.freeswitch.org >>>> >> >>>> > >>>> > >>>> > _______________________________________________ >>>> > Freeswitch-users mailing list >>>> > Freeswitch-users at lists.freeswitch.org >>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> > UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> > http://www.freeswitch.org >>>> > >>>> > >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:213-799-1400 >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090201/c925d46d/attachment-0002.html From gcd at i.ph Sun Feb 1 01:17:39 2009 From: gcd at i.ph (Nandy Dagondon) Date: Sun, 1 Feb 2009 17:17:39 +0800 Subject: [Freeswitch-users] TDM400 FXO can dialout only once In-Reply-To: <7d0bfd8c0902010030n6b5d3c35k7539fac38e07300d@mail.gmail.com> References: <7d0bfd8c0901250021q2969fe26u5ea79c3957989ffd@mail.gmail.com> <191c3a030901251818g284054ben60c0bfa61157824@mail.gmail.com> <7d0bfd8c0901252305v5a33fa02w4bc02deaa0c1b14b@mail.gmail.com> <87f2f3b90901260201r2e4d42a3x30367eee64b0e13b@mail.gmail.com> <7d0bfd8c0901260532g50deb359v3c92f5bb372f11a4@mail.gmail.com> <191c3a030901260833l2e07bfffkca9400d33a5483fe@mail.gmail.com> <7d0bfd8c0901261639x7c53f256yabf7347b5d22e7e4@mail.gmail.com> <7d0bfd8c0902010030n6b5d3c35k7539fac38e07300d@mail.gmail.com> Message-ID: <7d0bfd8c0902010117h7e26a3a6x2112225cb5951cd8@mail.gmail.com> hi, i found a major one. this time i deliberately set the dialtone freq to US std on my PH definition. i expect FS wont dial at all. but to my surprise, the problem is gone!! i checked the log. it indicates successful detection of DIALTONE. going on further. i noticed FXO wont hangup on busy tone. one possibility is the volume settings. the default is -7. how many dBm is this? and what is the dB equivalent per increment? tks n rgds, nandy On Sun, Feb 1, 2009 at 4:30 PM, Nandy Dagondon wrote: > hi everybody, > > i created [ph] tone definition per raul's suggestion and changed > /etc/zaptel.conf entries to: > tonezone=ph > defaultzone=ph > > but it didn't solve the problem. > i captured the console log during start-up and shutdown. i noticed openzap > related errors during shutdown. here's the snippet of the log: > > STARTUP > --------------- > 2009-02-01 15:58:10 [NOTICE] zap_io.c:2517 zap_global_init() Modules > configured: 1 > 2009-02-01 15:58:10 [INFO] zap_io.c:2341 zap_load_module() Loading IO from > /opt/freeswitch/mod/ozmod_zt.so > 2009-02-01 15:58:10 [INFO] zap_io.c:2127 load_config() auto-loaded 'zt' > 2009-02-01 15:58:10 [INFO] ozmod_zt.c:186 zt_open_range() configuring > device /dev/zap/channel channel 1 as OpenZAP device 1:1 fd:39 > 2009-02-01 15:58:10 [INFO] ozmod_zt.c:186 zt_open_range() configuring > device /dev/zap/channel channel 2 as OpenZAP device 2:1 fd:40 > 2009-02-01 15:58:10 [INFO] zap_io.c:2265 load_config() Configured 2 > channel(s) > 2009-02-01 15:58:10 [INFO] zap_io.c:2358 zap_load_module() Loading SIG from > /opt/freeswitch/mod/ozmod_analog.so > 2009-02-01 15:58:10 [INFO] zap_io.c:2474 zap_configure_span() auto-loaded > 'analog' > 2009-02-01 15:58:10 [CONSOLE] switch_loadable_module.c:857 > switch_loadable_module_load_file() Successfully Loaded [mod_openzap] > > --- DIDN'T MAKE ANY CALL --- > > SHUTDOWN > ------------------ > 2009-02-01 15:59:07 [NOTICE] switch_loadable_module.c:536 > switch_loadable_module_unprocess() Deleting API Function 'oz' > 2009-02-01 15:59:07 [CONSOLE] switch_loadable_module.c:1231 do_shutdown() > Stopping: mod_openzap > 2009-02-01 15:59:07 [INFO] zap_io.c:256 zap_channel_destroy() Closing > channel zt:1:1 fd:39 > 2009-02-01 15:59:07 [INFO] zap_io.c:256 zap_channel_destroy() Closing > channel zt:2:1 fd:40 > 2009-02-01 15:59:08 [ERR] ozmod_analog.c:899 zap_analog_run() Failure > Polling event! [no matching descriptor] > 2009-02-01 15:59:08 [ERR] ozmod_analog.c:899 zap_analog_run() Failure > Polling event! [no matching descriptor] > 2009-02-01 15:59:08 [INFO] zap_io.c:2456 zap_unload_modules() Unloading > /opt/freeswitch/mod/ozmod_analog.so > 2009-02-01 15:59:08 [INFO] zap_io.c:2441 zap_unload_modules() Unloading IO > zt > 2009-02-01 15:59:08 [INFO] zap_io.c:2456 zap_unload_modules() Unloading > /opt/freeswitch/mod/ozmod_zt.so > > i also notice the same ERR flag during shutdown after making test calls. > > any suggestion what to do next? > > tks for your assistance. > > rgds, > -nandy > > > On Tue, Jan 27, 2009 at 8:39 AM, Nandy Dagondon wrote: > >> i tested the SVN trunk version. still the same behaviour. >> -nandy >> >> >> On Tue, Jan 27, 2009 at 12:33 AM, Anthony Minessale < >> anthony.minessale at gmail.com> wrote: >> >>> The unhanded type message just means that mod_openzap does not do >>> anything with the TONE_DETECTED event that was passed >>> up from the ozmod_analog. >>> >>> On Mon, Jan 26, 2009 at 7:32 AM, Nandy Dagondon wrote: >>> >>>> that's great. yes, i'm in the philippines. there's a difference in >>>> dialtone - it's 425 Hz. >>>> -nandy >>>> >>>> >>>> >>>> On Mon, Jan 26, 2009 at 6:01 PM, Michael Collins wrote: >>>> >>>>> I have a TDM400 clone and I will see if I can reproduce these >>>>> symptoms. BTW, are you in the Philippines? Is there any difference in >>>>> the dial tone there than in the US? >>>>> -MC >>>>> >>>>> On Sun, Jan 25, 2009 at 11:05 PM, Nandy Dagondon wrote: >>>>> > i monitored the line using another phone. there's indeed dialtone in >>>>> all >>>>> > attempts. >>>>> > i see TONE_DETECTED in the first call but i wonder there's a WARNING >>>>> message >>>>> > immediately following [WARNING] mod_openzap.c:1196 on_fxo_signal() >>>>> Unhandled >>>>> > type for channel 2:1. >>>>> > the dialtone freq should be okay since it's detected in the first >>>>> call.could >>>>> > the WARNING message gives us a hint of a possible problem other than >>>>> the >>>>> > dialtone freq? >>>>> > >>>>> > okay, i'll try the SVN version next. >>>>> > >>>>> > >>>>> > On Mon, Jan 26, 2009 at 10:18 AM, Anthony Minessale >>>>> > wrote: >>>>> >> >>>>> >> Its not detecting a dial tone on the failure case. >>>>> >> Before dialing it waits until it picks up dialtone. >>>>> >> Try the svn trunk version to see if it works any better or verify >>>>> there is >>>>> >> a dialtone on the line. >>>>> >> >>>>> >> On Jan 25, 2009 6:19 PM, "Nandy Dagondon" wrote: >>>>> >> >>>>> >> hi everybody, >>>>> >> >>>>> >> i installed FS 1.0.2 to an Atom mobo (Intel D945GCLF2). it's working >>>>> using >>>>> >> IP phones, softphones and digium FXS port. but there's a problem in >>>>> dialing >>>>> >> out to PSTN using digium tdm400 fxo - it works fine on the first >>>>> attempt >>>>> >> (after starting FS) but it fails on the subsequent attempts. i >>>>> tested to >>>>> >> call using the FXS port and IP phone. same problem. >>>>> >> >>>>> >> before i place any call, i checked >oz dump 2 1 (show current state >>>>> = >>>>> >> DOWN, last state = DOWN) >>>>> >> >>>>> >> in the first call, there's this message: >>>>> >> [WARNING] mod_openzap.c:1196 on_fxo_signal() Unhandled type for >>>>> channel >>>>> >> 2:1 >>>>> >> but >>>>> >> >>>>> >> then i hangup. checked >oz dump 21 (show current state=DOWN, last >>>>> >> state=HANGUP) >>>>> >> >>>>> >> in the 2nd (and subsequent) attempts, the fxo just goes off-hook but >>>>> >> doesn't send the dtmf tones. >>>>> >> >oz dump 2 1 (shows current state = DIALING, last state = DOWN) >>>>> >> >>>>> >> has anyone encountered this problem before? i appreciate for any >>>>> help to >>>>> >> correct this problem. >>>>> >> >>>>> >> tks, >>>>> >> nandy >>>>> >> >>>>> >> >>>>> >> Environment: >>>>> >> ================== >>>>> >> kernel 2.6.18-92.1.22.el5 >>>>> >> FS 1.0.2 >>>>> >> zaptel 1.4.11 >>>>> >> oslec >>>>> >> digium TDM400P Rev. I (1-FXS, 2-FXO,3-4 vacant) >>>>> >> >>>>> >> zaptel.conf >>>>> >> ======== >>>>> >> loadzone = us >>>>> >> defaultzone=us >>>>> >> channels=1-2 >>>>> >> alaw=1-4 >>>>> >> fxsks=2 >>>>> >> fxoks=1 >>>>> >> >>>>> >> >>>>> >> openzap.conf.xml: >>>>> >> =============== >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> openzap.conf >>>>> >> ========== >>>>> >> [span zt] >>>>> >> name => OpenZAP FXS >>>>> >> number => 1 >>>>> >> fxs-channel => 1 >>>>> >> >>>>> >> [span zt] >>>>> >> name => OpenZAP FXO >>>>> >> number => 2 >>>>> >> fxo-channel => 2 >>>>> >> >>>>> >> tones.conf (the dialtone and ring tone is set to Philipping tones) >>>>> >> ======== >>>>> >> [us] >>>>> >> generate-dial => v=-7;%(1000,0,425) >>>>> >> detect-dial => 425 >>>>> >> >>>>> >> generate-ring => v=-7;%(1000,4000,425,480) >>>>> >> detect-ring => 425,480 >>>>> >> >>>>> >> generate-busy => v=-7;%(500,500,480,620) >>>>> >> detect-busy => 480,620 >>>>> >> >>>>> >> generate-attn => v=0;%(200,300,1400,1800) >>>>> >> detect-attn => 1400,1800 >>>>> >> >>>>> >> generate-callwaiting-sas => v=0;%(300,10000,440) >>>>> >> detect-callwaiting-sas => 440 >>>>> >> >>>>> >> generate-callwaiting-cas => v=0;%(80,0,2750,2130) >>>>> >> detect-callwaiting-cas => 2750,2130 >>>>> >> >>>>> >> detect-fail1 => 913.8 >>>>> >> detect-fail2 => 1370.6 >>>>> >> detect-fail3 => 776.7 >>>>> >> >>>>> >> LOG OF FIRST CALL (OK) >>>>> >> ==================== >>>>> >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:152 >>>>> >> switch_core_standard_on_execute() OpenZAP/1:1/93400534 Execute >>>>> >> bridge(openzap/2/1/3400534) >>>>> >> 2009-01-25 10:35:58 [DEBUG] mod_openzap.c:340 tech_init() Set codec >>>>> PCMU >>>>> >> 20ms >>>>> >> 2009-01-25 10:35:58 [DEBUG] mod_openzap.c:1017 >>>>> channel_outgoing_channel() >>>>> >> Connect outbound channel OpenZAP/2:1/3400534 >>>>> >> 2009-01-25 10:35:58 [NOTICE] switch_channel.c:565 >>>>> >> switch_channel_set_name() New Channel OpenZAP/2:1/3400534 >>>>> >> [e5f12114-ea88-11dd-9f5c-290fb4a527a4] >>>>> >> 2009-01-25 10:35:58 [DEBUG] mod_openzap.c:1026 >>>>> channel_outgoing_channel() >>>>> >> (OpenZAP/2:1/3400534) State Change CS_NEW -> CS_INIT >>>>> >> 2009-01-25 10:35:58 [DEBUG] switch_core_session.c:807 >>>>> >> switch_core_session_signal_state_change() Send signal >>>>> OpenZAP/2:1/3400534 >>>>> >> [BREAK] >>>>> >> 2009-01-25 10:35:58 [DEBUG] ozmod_analog.c:52 >>>>> analog_fxo_outgoing_call() >>>>> >> Changing state on 2:1 from DOWN to DIALING >>>>> >> 2009-01-25 10:35:58 [DEBUG] ozmod_analog.c:239 >>>>> zap_analog_channel_run() >>>>> >> ANALOG CHANNEL thread starting. >>>>> >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:379 >>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change >>>>> CS_INIT >>>>> >> 2009-01-25 10:35:58 [DEBUG] ozmod_analog.c:410 >>>>> zap_analog_channel_run() >>>>> >> Executing state handler on 2:1 for DIALING >>>>> >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:444 >>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State INIT >>>>> >> 2009-01-25 10:35:58 [DEBUG] mod_openzap.c:364 channel_on_init() >>>>> >> (OpenZAP/2:1/3400534) State Change CS_INIT -> CS_ROUTING >>>>> >> 2009-01-25 10:35:58 [DEBUG] switch_core_session.c:807 >>>>> >> switch_core_session_signal_state_change() Send signal >>>>> OpenZAP/2:1/3400534 >>>>> >> [BREAK] >>>>> >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:444 >>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State INIT going to >>>>> sleep >>>>> >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:379 >>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change >>>>> >> CS_ROUTING >>>>> >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:447 >>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State ROUTING >>>>> >> 2009-01-25 10:35:58 [DEBUG] mod_openzap.c:387 channel_on_routing() >>>>> >> OpenZAP/2:1/3400534 CHANNEL ROUTING >>>>> >> 2009-01-25 10:35:58 [DEBUG] switch_ivr_originate.c:58 >>>>> >> originate_on_routing() (OpenZAP/2:1/3400534) State Change CS_ROUTING >>>>> -> >>>>> >> CS_CONSUME_MEDIA >>>>> >> 2009-01-25 10:35:58 [DEBUG] switch_core_session.c:807 >>>>> >> switch_core_session_signal_state_change() Send signal >>>>> OpenZAP/2:1/3400534 >>>>> >> [BREAK] >>>>> >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:447 >>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State ROUTING going >>>>> to sleep >>>>> >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:379 >>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change >>>>> >> CS_CONSUME_MEDIA >>>>> >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:466 >>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State CONSUME_MEDIA >>>>> >> 2009-01-25 10:35:59 [DEBUG] ozmod_analog.c:615 >>>>> zap_analog_channel_run() >>>>> >> Detected tone DIAL on 2:1 >>>>> >> 2009-01-25 10:35:59 [DEBUG] mod_openzap.c:1153 on_fxo_signal() got >>>>> FXO sig >>>>> >> 2:1 [TONE_DETECTED] >>>>> >> 2009-01-25 10:35:59 [WARNING] mod_openzap.c:1196 on_fxo_signal() >>>>> Unhandled >>>>> >> type for channel 2:1 >>>>> >> 2009-01-25 10:35:59 [DEBUG] zap_io.c:1179 >>>>> zchan_activate_dtmf_buffer() >>>>> >> Created DTMF Buffer! >>>>> >> 2009-01-25 10:35:59 [DEBUG] zap_io.c:1715 handle_dtmf() 2:1 GENERATE >>>>> DTMF >>>>> >> [3400534] >>>>> >> 2009-01-25 10:36:03 [DEBUG] ozmod_analog.c:302 >>>>> zap_analog_channel_run() >>>>> >> Changing state on 2:1 from DIALING to UP >>>>> >> 2009-01-25 10:36:03 [DEBUG] ozmod_analog.c:410 >>>>> zap_analog_channel_run() >>>>> >> Executing state handler on 2:1 for UP >>>>> >> 2009-01-25 10:36:03 [DEBUG] mod_openzap.c:1153 on_fxo_signal() got >>>>> FXO sig >>>>> >> 2:1 [UP] >>>>> >> 2009-01-25 10:36:03 [DEBUG] switch_channel.c:1710 >>>>> >> switch_channel_perform_mark_answered() Send signal >>>>> OpenZAP/1:1/93400534 >>>>> >> [BREAK] >>>>> >> 2009-01-25 10:36:03 [NOTICE] mod_openzap.c:1180 on_fxo_signal() >>>>> Channel >>>>> >> [OpenZAP/2:1/3400534] has been answered >>>>> >> 2009-01-25 10:36:03 [DEBUG] switch_channel.c:177 >>>>> >> switch_channel_audio_sync() OpenZAP/2:1/3400534 receive message >>>>> [AUDIO_SYNC] >>>>> >> 2009-01-25 10:36:03 [DEBUG] switch_channel.c:1768 >>>>> >> switch_channel_perform_answer() OpenZAP/1:1/93400534 receive message >>>>> >> [ANSWER] >>>>> >> 2009-01-25 10:36:03 [DEBUG] mod_openzap.c:813 >>>>> >> channel_receive_message_fxs() Changing state on 1:1 from IDLE to UP >>>>> >> 2009-01-25 10:36:03 [NOTICE] mod_openzap.c:814 >>>>> >> channel_receive_message_fxs() Channel [OpenZAP/1:1/93400534] has >>>>> been >>>>> >> answered >>>>> >> 2009-01-25 10:36:03 [DEBUG] switch_channel.c:177 >>>>> >> switch_channel_audio_sync() OpenZAP/1:1/93400534 receive message >>>>> >> [AUDIO_SYNC] >>>>> >> 2009-01-25 10:36:03 [DEBUG] switch_core_session.c:511 >>>>> >> switch_core_session_perform_receive_message() Send signal >>>>> >> OpenZAP/1:1/93400534 [BREAK] >>>>> >> 2009-01-25 10:36:03 [DEBUG] switch_ivr_originate.c:1627 >>>>> >> switch_ivr_originate() Originate Resulted in Success: >>>>> [OpenZAP/2:1/3400534] >>>>> >> 2009-01-25 10:36:03 [DEBUG] switch_channel.c:177 >>>>> >> switch_channel_audio_sync() OpenZAP/2:1/3400534 receive message >>>>> [AUDIO_SYNC] >>>>> >> 2009-01-25 10:36:03 [DEBUG] switch_channel.c:177 >>>>> >> switch_channel_audio_sync() OpenZAP/1:1/93400534 receive message >>>>> >> [AUDIO_SYNC] >>>>> >> 2009-01-25 10:36:03 [DEBUG] switch_ivr_bridge.c:862 >>>>> >> switch_ivr_multi_threaded_bridge() OpenZAP/2:1/3400534 receive >>>>> message >>>>> >> [BRIDGE] >>>>> >> 2009-01-25 10:36:03 [DEBUG] switch_core_session.c:511 >>>>> >> switch_core_session_perform_receive_message() Send signal >>>>> >> OpenZAP/2:1/3400534 [BREAK] >>>>> >> 2009-01-25 10:36:03 [DEBUG] switch_ivr_bridge.c:869 >>>>> >> switch_ivr_multi_threaded_bridge() OpenZAP/1:1/93400534 receive >>>>> message >>>>> >> [BRIDGE] >>>>> >> 2009-01-25 10:36:03 [DEBUG] switch_core_session.c:511 >>>>> >> switch_core_session_perform_receive_message() Send signal >>>>> >> OpenZAP/1:1/93400534 [BREAK] >>>>> >> 2009-01-25 10:36:03 [DEBUG] switch_ivr_bridge.c:913 >>>>> >> switch_ivr_multi_threaded_bridge() (OpenZAP/2:1/3400534) State >>>>> Change >>>>> >> CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA >>>>> >> 2009-01-25 10:36:03 [DEBUG] switch_core_session.c:807 >>>>> >> switch_core_session_signal_state_change() Send signal >>>>> OpenZAP/2:1/3400534 >>>>> >> [BREAK] >>>>> >> 2009-01-25 10:36:03 [DEBUG] ozmod_analog.c:410 >>>>> zap_analog_channel_run() >>>>> >> Executing state handler on 1:1 for UP >>>>> >> 2009-01-25 10:36:03 [DEBUG] mod_openzap.c:1212 on_fxs_signal() got >>>>> FXS sig >>>>> >> [UP] >>>>> >> 2009-01-25 10:36:03 [DEBUG] switch_core_state_machine.c:466 >>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State CONSUME_MEDIA >>>>> going to >>>>> >> sleep >>>>> >> 2009-01-25 10:36:03 [DEBUG] switch_core_state_machine.c:379 >>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change >>>>> >> CS_EXCHANGE_MEDIA >>>>> >> 2009-01-25 10:36:03 [DEBUG] switch_core_state_machine.c:457 >>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State EXCHANGE_MEDIA >>>>> >> 2009-01-25 10:36:03 [DEBUG] mod_openzap.c:511 >>>>> channel_on_exchange_media() >>>>> >> CHANNEL EXCHANGE_MEDIA >>>>> >> 2009-01-25 10:36:16 [DEBUG] ozmod_analog.c:744 process_event() EVENT >>>>> >> [ONHOOK][1:1] STATE [UP] >>>>> >> 2009-01-25 10:36:16 [DEBUG] ozmod_analog.c:780 process_event() >>>>> Changing >>>>> >> state on 1:1 from UP to DOWN >>>>> >> 2009-01-25 10:36:16 [DEBUG] ozmod_analog.c:410 >>>>> zap_analog_channel_run() >>>>> >> Executing state handler on 1:1 for DOWN >>>>> >> 2009-01-25 10:36:16 [DEBUG] mod_openzap.c:1212 on_fxs_signal() got >>>>> FXS sig >>>>> >> [STOP] >>>>> >> 2009-01-25 10:36:16 [NOTICE] mod_openzap.c:1300 on_fxs_signal() >>>>> Hangup >>>>> >> OpenZAP/1:1/93400534 [CS_EXECUTE] [NORMAL_CLEARING] >>>>> >> 2009-01-25 10:36:16 [DEBUG] switch_channel.c:1494 >>>>> >> switch_channel_perform_hangup() Send signal OpenZAP/1:1/93400534 >>>>> [KILL] >>>>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_session.c:807 >>>>> >> switch_core_session_signal_state_change() Send signal >>>>> OpenZAP/1:1/93400534 >>>>> >> [BREAK] >>>>> >> 2009-01-25 10:36:16 [DEBUG] zap_io.c:1125 zap_channel_done() channel >>>>> done >>>>> >> 1:1 >>>>> >> 2009-01-25 10:36:16 [DEBUG] ozmod_analog.c:726 >>>>> zap_analog_channel_run() >>>>> >> ANALOG CHANNEL 1:1 thread ended. >>>>> >> 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:360 >>>>> audio_bridge_thread() >>>>> >> OpenZAP/1:1/93400534 ending bridge by request from read function >>>>> >> 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:435 >>>>> audio_bridge_thread() >>>>> >> BRIDGE THREAD DONE [OpenZAP/1:1/93400534] >>>>> >> 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:439 >>>>> audio_bridge_thread() >>>>> >> Send signal OpenZAP/2:1/3400534 [BREAK] >>>>> >> 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:354 >>>>> audio_bridge_thread() >>>>> >> OpenZAP/1:1/93400534 ending bridge by request from write function >>>>> >> 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:409 >>>>> audio_bridge_thread() >>>>> >> OpenZAP/2:1/3400534 receive message [UNBRIDGE] >>>>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_session.c:511 >>>>> >> switch_core_session_perform_receive_message() Send signal >>>>> >> OpenZAP/2:1/3400534 [BREAK] >>>>> >> 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:435 >>>>> audio_bridge_thread() >>>>> >> BRIDGE THREAD DONE [OpenZAP/2:1/3400534] >>>>> >> 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:439 >>>>> audio_bridge_thread() >>>>> >> Send signal OpenZAP/1:1/93400534 [BREAK] >>>>> >> 2009-01-25 10:36:16 [NOTICE] switch_ivr_bridge.c:470 >>>>> >> audio_bridge_on_exchange_media() Hangup OpenZAP/2:1/3400534 >>>>> >> [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] >>>>> >> 2009-01-25 10:36:16 [DEBUG] switch_channel.c:1494 >>>>> >> switch_channel_perform_hangup() Send signal OpenZAP/2:1/3400534 >>>>> [KILL] >>>>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_session.c:807 >>>>> >> switch_core_session_signal_state_change() Send signal >>>>> OpenZAP/2:1/3400534 >>>>> >> [BREAK] >>>>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:457 >>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State EXCHANGE_MEDIA >>>>> going >>>>> >> to sleep >>>>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:379 >>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change >>>>> >> CS_HANGUP >>>>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:410 >>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State HANGUP >>>>> >> 2009-01-25 10:36:16 [DEBUG] mod_openzap.c:429 channel_on_hangup() >>>>> Changing >>>>> >> state on 2:1 from UP to HANGUP >>>>> >> 2009-01-25 10:36:16 [DEBUG] mod_openzap.c:472 channel_on_hangup() >>>>> >> OpenZAP/2:1/3400534 CHANNEL HANGUP >>>>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:46 >>>>> >> switch_core_standard_on_hangup() OpenZAP/2:1/3400534 Standard >>>>> HANGUP, cause: >>>>> >> NORMAL_CLEARING >>>>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:410 >>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State HANGUP going >>>>> to sleep >>>>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_session.c:939 >>>>> >> switch_core_session_thread() Session 2 (OpenZAP/2:1/3400534) Locked, >>>>> Waiting >>>>> >> on external entities >>>>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:454 >>>>> >> switch_core_session_run() (OpenZAP/1:1/93400534) State EXECUTE going >>>>> to >>>>> >> sleep >>>>> >> 2009-01-25 10:36:16 [NOTICE] switch_core_session.c:957 >>>>> >> switch_core_session_thread() Session 2 (OpenZAP/2:1/3400534) Ended >>>>> >> 2009-01-25 10:36:16 [NOTICE] switch_core_session.c:959 >>>>> >> switch_core_session_thread() Close Channel OpenZAP/2:1/3400534 >>>>> [CS_HANGUP] >>>>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:379 >>>>> >> switch_core_session_run() (OpenZAP/1:1/93400534) Running State >>>>> Change >>>>> >> CS_HANGUP >>>>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:410 >>>>> >> switch_core_session_run() (OpenZAP/1:1/93400534) State HANGUP >>>>> >> 2009-01-25 10:36:16 [DEBUG] mod_openzap.c:472 channel_on_hangup() >>>>> >> OpenZAP/1:1/93400534 CHANNEL HANGUP >>>>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:46 >>>>> >> switch_core_standard_on_hangup() OpenZAP/1:1/93400534 Standard >>>>> HANGUP, >>>>> >> cause: NORMAL_CLEARING >>>>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:410 >>>>> >> switch_core_session_run() (OpenZAP/1:1/93400534) State HANGUP going >>>>> to sleep >>>>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_session.c:939 >>>>> >> switch_core_session_thread() Session 1 (OpenZAP/1:1/93400534) >>>>> Locked, >>>>> >> Waiting on external entities >>>>> >> 2009-01-25 10:36:16 [NOTICE] switch_core_session.c:957 >>>>> >> switch_core_session_thread() Session 1 (OpenZAP/1:1/93400534) Ended >>>>> >> 2009-01-25 10:36:16 [NOTICE] switch_core_session.c:959 >>>>> >> switch_core_session_thread() Close Channel OpenZAP/1:1/93400534 >>>>> [CS_HANGUP] >>>>> >> 2009-01-25 10:36:16 [DEBUG] ozmod_analog.c:410 >>>>> zap_analog_channel_run() >>>>> >> Executing state handler on 2:1 for HANGUP >>>>> >> 2009-01-25 10:36:17 [DEBUG] ozmod_analog.c:351 >>>>> zap_analog_channel_run() >>>>> >> Changing state on 2:1 from HANGUP to DOWN >>>>> >> 2009-01-25 10:36:17 [DEBUG] ozmod_analog.c:410 >>>>> zap_analog_channel_run() >>>>> >> Executing state handler on 2:1 for DOWN >>>>> >> 2009-01-25 10:36:17 [DEBUG] mod_openzap.c:1153 on_fxo_signal() got >>>>> FXO sig >>>>> >> 2:1 [STOP] >>>>> >> 2009-01-25 10:36:17 [DEBUG] zap_io.c:1125 zap_channel_done() channel >>>>> done >>>>> >> 2:1 >>>>> >> 2009-01-25 10:36:17 [DEBUG] ozmod_analog.c:726 >>>>> zap_analog_channel_run() >>>>> >> ANALOG CHANNEL 2:1 thread ended. >>>>> >> >>>>> >> LOG OF FAILED CALLS >>>>> >> ================== >>>>> >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:152 >>>>> >> switch_core_standard_on_execute() OpenZAP/1:1/93400534 Execute >>>>> >> bridge(openzap/2/1/3400534) >>>>> >> 2009-01-25 10:36:55 [DEBUG] mod_openzap.c:340 tech_init() Set codec >>>>> PCMU >>>>> >> 20ms >>>>> >> 2009-01-25 10:36:55 [DEBUG] mod_openzap.c:1017 >>>>> channel_outgoing_channel() >>>>> >> Connect outbound channel OpenZAP/2:1/3400534 >>>>> >> 2009-01-25 10:36:55 [NOTICE] switch_channel.c:565 >>>>> >> switch_channel_set_name() New Channel OpenZAP/2:1/3400534 >>>>> >> [079f5420-ea89-11dd-9f5c-290fb4a527a4] >>>>> >> 2009-01-25 10:36:55 [DEBUG] mod_openzap.c:1026 >>>>> channel_outgoing_channel() >>>>> >> (OpenZAP/2:1/3400534) State Change CS_NEW -> CS_INIT >>>>> >> 2009-01-25 10:36:55 [DEBUG] switch_core_session.c:807 >>>>> >> switch_core_session_signal_state_change() Send signal >>>>> OpenZAP/2:1/3400534 >>>>> >> [BREAK] >>>>> >> 2009-01-25 10:36:55 [DEBUG] ozmod_analog.c:52 >>>>> analog_fxo_outgoing_call() >>>>> >> Changing state on 2:1 from DOWN to DIALING >>>>> >> 2009-01-25 10:36:55 [DEBUG] ozmod_analog.c:239 >>>>> zap_analog_channel_run() >>>>> >> ANALOG CHANNEL thread starting. >>>>> >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:379 >>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change >>>>> CS_INIT >>>>> >> 2009-01-25 10:36:55 [DEBUG] ozmod_analog.c:410 >>>>> zap_analog_channel_run() >>>>> >> Executing state handler on 2:1 for DIALING >>>>> >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:444 >>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State INIT >>>>> >> 2009-01-25 10:36:55 [DEBUG] mod_openzap.c:364 channel_on_init() >>>>> >> (OpenZAP/2:1/3400534) State Change CS_INIT -> CS_ROUTING >>>>> >> 2009-01-25 10:36:55 [DEBUG] switch_core_session.c:807 >>>>> >> switch_core_session_signal_state_change() Send signal >>>>> OpenZAP/2:1/3400534 >>>>> >> [BREAK] >>>>> >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:444 >>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State INIT going to >>>>> sleep >>>>> >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:379 >>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change >>>>> >> CS_ROUTING >>>>> >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:447 >>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State ROUTING >>>>> >> 2009-01-25 10:36:55 [DEBUG] mod_openzap.c:387 channel_on_routing() >>>>> >> OpenZAP/2:1/3400534 CHANNEL ROUTING >>>>> >> 2009-01-25 10:36:55 [DEBUG] switch_ivr_originate.c:58 >>>>> >> originate_on_routing() (OpenZAP/2:1/3400534) State Change CS_ROUTING >>>>> -> >>>>> >> CS_CONSUME_MEDIA >>>>> >> 2009-01-25 10:36:55 [DEBUG] switch_core_session.c:807 >>>>> >> switch_core_session_signal_state_change() Send signal >>>>> OpenZAP/2:1/3400534 >>>>> >> [BREAK] >>>>> >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:447 >>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State ROUTING going >>>>> to sleep >>>>> >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:379 >>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change >>>>> >> CS_CONSUME_MEDIA >>>>> >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:466 >>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State CONSUME_MEDIA >>>>> >> 2009-01-25 10:37:08 [DEBUG] ozmod_analog.c:744 process_event() EVENT >>>>> >> [ONHOOK][1:1] STATE [IDLE] >>>>> >> 2009-01-25 10:37:08 [DEBUG] ozmod_analog.c:780 process_event() >>>>> Changing >>>>> >> state on 1:1 from IDLE to DOWN >>>>> >> 2009-01-25 10:37:08 [DEBUG] ozmod_analog.c:410 >>>>> zap_analog_channel_run() >>>>> >> Executing state handler on 1:1 for DOWN >>>>> >> 2009-01-25 10:37:08 [DEBUG] mod_openzap.c:1212 on_fxs_signal() got >>>>> FXS sig >>>>> >> [STOP] >>>>> >> 2009-01-25 10:37:08 [NOTICE] mod_openzap.c:1300 on_fxs_signal() >>>>> Hangup >>>>> >> OpenZAP/1:1/93400534 [CS_EXECUTE] [NORMAL_CLEARING] >>>>> >> 2009-01-25 10:37:08 [DEBUG] switch_channel.c:1494 >>>>> >> switch_channel_perform_hangup() Send signal OpenZAP/1:1/93400534 >>>>> [KILL] >>>>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_session.c:807 >>>>> >> switch_core_session_signal_state_change() Send signal >>>>> OpenZAP/1:1/93400534 >>>>> >> [BREAK] >>>>> >> 2009-01-25 10:37:08 [DEBUG] zap_io.c:1125 zap_channel_done() channel >>>>> done >>>>> >> 1:1 >>>>> >> 2009-01-25 10:37:08 [DEBUG] ozmod_analog.c:726 >>>>> zap_analog_channel_run() >>>>> >> ANALOG CHANNEL 1:1 thread ended. >>>>> >> 2009-01-25 10:37:08 [NOTICE] switch_ivr_originate.c:1566 >>>>> >> switch_ivr_originate() Hangup OpenZAP/2:1/3400534 [CS_CONSUME_MEDIA] >>>>> >> [ORIGINATOR_CANCEL] >>>>> >> 2009-01-25 10:37:08 [DEBUG] switch_channel.c:1494 >>>>> >> switch_channel_perform_hangup() Send signal OpenZAP/2:1/3400534 >>>>> [KILL] >>>>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_session.c:807 >>>>> >> switch_core_session_signal_state_change() Send signal >>>>> OpenZAP/2:1/3400534 >>>>> >> [BREAK] >>>>> >> 2009-01-25 10:37:08 [DEBUG] switch_ivr_originate.c:1691 >>>>> >> switch_ivr_originate() Originate Cancelled by originator termination >>>>> Cause: >>>>> >> 487 [ORIGINATOR_CANCEL] >>>>> >> 2009-01-25 10:37:08 [INFO] mod_dptools.c:1909 >>>>> audio_bridge_function() >>>>> >> Originate Failed. Cause: ORIGINATOR_CANCEL >>>>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:454 >>>>> >> switch_core_session_run() (OpenZAP/1:1/93400534) State EXECUTE going >>>>> to >>>>> >> sleep >>>>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:379 >>>>> >> switch_core_session_run() (OpenZAP/1:1/93400534) Running State >>>>> Change >>>>> >> CS_HANGUP >>>>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:410 >>>>> >> switch_core_session_run() (OpenZAP/1:1/93400534) State HANGUP >>>>> >> 2009-01-25 10:37:08 [DEBUG] mod_openzap.c:472 channel_on_hangup() >>>>> >> OpenZAP/1:1/93400534 CHANNEL HANGUP >>>>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:46 >>>>> >> switch_core_standard_on_hangup() OpenZAP/1:1/93400534 Standard >>>>> HANGUP, >>>>> >> cause: NORMAL_CLEARING >>>>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:410 >>>>> >> switch_core_session_run() (OpenZAP/1:1/93400534) State HANGUP going >>>>> to sleep >>>>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_session.c:939 >>>>> >> switch_core_session_thread() Session 3 (OpenZAP/1:1/93400534) >>>>> Locked, >>>>> >> Waiting on external entities >>>>> >> 2009-01-25 10:37:08 [NOTICE] switch_core_session.c:957 >>>>> >> switch_core_session_thread() Session 3 (OpenZAP/1:1/93400534) Ended >>>>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:466 >>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State CONSUME_MEDIA >>>>> going to >>>>> >> sleep >>>>> >> 2009-01-25 10:37:08 [NOTICE] switch_core_session.c:959 >>>>> >> switch_core_session_thread() Close Channel OpenZAP/1:1/93400534 >>>>> [CS_HANGUP] >>>>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:379 >>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) Running State Change >>>>> >> CS_HANGUP >>>>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:410 >>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State HANGUP >>>>> >> 2009-01-25 10:37:08 [DEBUG] mod_openzap.c:429 channel_on_hangup() >>>>> Changing >>>>> >> state on 2:1 from DIALING to HANGUP >>>>> >> 2009-01-25 10:37:08 [DEBUG] mod_openzap.c:472 channel_on_hangup() >>>>> >> OpenZAP/2:1/3400534 CHANNEL HANGUP >>>>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:46 >>>>> >> switch_core_standard_on_hangup() OpenZAP/2:1/3400534 Standard >>>>> HANGUP, cause: >>>>> >> ORIGINATOR_CANCEL >>>>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:410 >>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State HANGUP going >>>>> to sleep >>>>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_session.c:939 >>>>> >> switch_core_session_thread() Session 4 (OpenZAP/2:1/3400534) Locked, >>>>> Waiting >>>>> >> on external entities >>>>> >> 2009-01-25 10:37:08 [NOTICE] switch_core_session.c:957 >>>>> >> switch_core_session_thread() Session 4 (OpenZAP/2:1/3400534) Ended >>>>> >> 2009-01-25 10:37:08 [NOTICE] switch_core_session.c:959 >>>>> >> switch_core_session_thread() Close Channel OpenZAP/2:1/3400534 >>>>> [CS_HANGUP] >>>>> >> 2009-01-25 10:37:08 [DEBUG] ozmod_analog.c:410 >>>>> zap_analog_channel_run() >>>>> >> Executing state handler on 2:1 for HANGUP >>>>> >> 2009-01-25 10:37:09 [DEBUG] ozmod_analog.c:351 >>>>> zap_analog_channel_run() >>>>> >> Changing state on 2:1 from HANGUP to DOWN >>>>> >> 2009-01-25 10:37:09 [DEBUG] ozmod_analog.c:410 >>>>> zap_analog_channel_run() >>>>> >> Executing state handler on 2:1 for DOWN >>>>> >> 2009-01-25 10:37:09 [DEBUG] mod_openzap.c:1153 on_fxo_signal() got >>>>> FXO sig >>>>> >> 2:1 [STOP] >>>>> >> 2009-01-25 10:37:09 [DEBUG] zap_io.c:1125 zap_channel_done() channel >>>>> done >>>>> >> 2:1 >>>>> >> 2009-01-25 10:37:09 [DEBUG] ozmod_analog.c:726 >>>>> zap_analog_channel_run() >>>>> >> ANALOG CHANNEL 2:1 thread ended. >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> >> _______________________________________________ >>>>> >> Freeswitch-users mailing list >>>>> >> Freeswitch-users at lists.freeswitch.org >>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> >> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> >> http://www.freeswitch.org >>>>> >> >>>>> >> >>>>> >> _______________________________________________ >>>>> >> Freeswitch-users mailing list >>>>> >> Freeswitch-users at lists.freeswitch.org >>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> >> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> >> http://www.freeswitch.org >>>>> >> >>>>> > >>>>> > >>>>> > _______________________________________________ >>>>> > Freeswitch-users mailing list >>>>> > Freeswitch-users at lists.freeswitch.org >>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> > UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> > http://www.freeswitch.org >>>>> > >>>>> > >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> Anthony Minessale II >>> >>> FreeSWITCH http://www.freeswitch.org/ >>> ClueCon http://www.cluecon.com/ >>> >>> AIM: anthm >>> MSN:anthony_minessale at hotmail.com >>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>> IRC: irc.freenode.net #freeswitch >>> >>> FreeSWITCH Developer Conference >>> sip:888 at conference.freeswitch.org >>> iax:guest at conference.freeswitch.org/888 >>> googletalk:conf+888 at conference.freeswitch.org >>> pstn:213-799-1400 >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090201/88e279e2/attachment-0002.html From pmhshz at gmail.com Sun Feb 1 02:02:04 2009 From: pmhshz at gmail.com (shehzad p) Date: Sun, 1 Feb 2009 02:02:04 -0800 (PST) Subject: [Freeswitch-users] Freeswitch freezes on increasing call traffic In-Reply-To: <191c3a030901310850s553f8c11oc4e559a710def6d2@mail.gmail.com> References: <21701744.post@talk.nabble.com> <191c3a030901280601t3e185ec2m76295110e4888837@mail.gmail.com> <21729863.post@talk.nabble.com> <191c3a030901290831y2ff329b3maee6c1aafa3c68e2@mail.gmail.com> <21745312.post@talk.nabble.com> <21749375.post@talk.nabble.com> <191c3a030901300818i5d85a7aawa0c53b4e54892f40@mail.gmail.com> <21761523.post@talk.nabble.com> <191c3a030901310850s553f8c11oc4e559a710def6d2@mail.gmail.com> Message-ID: <21773300.post@talk.nabble.com> Hi Anthony, How can I handle Garbage collector related things. Is there any way to look out where js variables are causing some kind of fault related to Garbage collector? There are some loops but they finish in hardly two or three iterations. I didn't check session.ready() before/after fetching values from database (odbc), which might block until response from server comes back. might this cause fault as you said? Thanks for help... Anthony Minessale-2 wrote: > > Clearly you have an issue with your javascript code. > > You have the Garbage collector blocking in every thread. > > Are you doing any endless loops in your code where you do not check > session.ready() as a condition for > continuing the script? > > any time session.ready() fails you must immediately exit. > > Are you using session.execute to execute long blocking operations like > bridging many calls or entering a conference? > You should avoid doing this as all the collective scripts on the system > share a common Garbage Collector provided by the > JS engine and it can lead to the exact issues you describe if the code is > not properly designed. > > What else does you script do that are things provided by FS such as > playing > files and executing applications. > > > > On Sat, Jan 31, 2009 at 3:44 AM, shehzad p wrote: > >> >> Hi Anthony, >> >> Freeswitch 1.0.2 was crashed on last test again... >> BT is on http://pastebin.freeswitch.org/6979. >> >> I tried to use scripts/freeswitch-gcore, to capture resident memory, but >> before the command complete the process system hanged up so only half >> output >> was captured.< http://www.nabble.com/file/p21761523/gcore-fs.txt >> gcore-fs.txt > >> >> >> Now I have checkout from trunk and will post back if any new thing >> found... >> >> Thanks, >> msp >> >> Anthony Minessale-2 wrote: >> > >> > if you are using unix you can use the supplied script >> > >> > scripts/freeswitch-gcore >> > >> > to capture a copy of the resident memory and I can have a look perhaps. >> > >> > Trunk is safe for production as we are in beta stage for a release of >> > 1.0.3 >> > at this time. >> > >> > >> > >> > On Fri, Jan 30, 2009 at 9:29 AM, shehzad p wrote: >> > >> >> >> >> When freeswitch freezes, we can't connect to it to check sps status, >> >> but once we were able to connect and at that time it was showing 0/0 >> sps. >> >> >> >> thanks... >> >> >> >> shehzad p wrote: >> >> > >> >> > Thanks, Anthony >> >> > >> >> > In my previous test sps did not changed, >> >> > but in recent test sps was dropped to 0 itself (as below). >> >> > =============================================================== >> >> > UP 0 years, 0 days, 5 hours, 1 minute, 53 seconds, 878 milliseconds, >> >> 190 >> >> > microseconds >> >> > 5474 session(s) since startup >> >> > 75 session(s) 0/0 >> >> > ============================================================= >> >> > >> >> > My system is 32 bit. >> >> > CPU is Intel(R) Xeon(R) CPU X3220 @ 2.40GHz >> >> > And RAM is 4GB >> >> > >> >> > Output of ulimit -a is: >> >> > ulimit -a: (set after first test) >> >> > core file size (blocks, -c) unlimited >> >> > data seg size (kbytes, -d) unlimited >> >> > max nice (-e) 20 >> >> > file size (blocks, -f) unlimited >> >> > pending signals (-i) unlimited >> >> > max locked memory (kbytes, -l) unlimited >> >> > max memory size (kbytes, -m) unlimited >> >> > open files (-n) 999999 >> >> > pipe size (512 bytes, -p) 8 >> >> > POSIX message queues (bytes, -q) unlimited >> >> > max rt priority (-r) unlimited >> >> > stack size (kbytes, -s) 244 >> >> > cpu time (seconds, -t) unlimited >> >> > max user processes (-u) unlimited >> >> > virtual memory (kbytes, -v) unlimited >> >> > file locks (-x) unlimited >> >> > =================================================== >> >> > >> >> > >> >> > BTW using trunk on production system is safe? >> >> > >> >> > Warm thanks for kind responses... >> >> > >> >> > >> >> > >> >> > Anthony Minessale-2 wrote: >> >> >> >> >> >> When you get it in that state what do you see when you execute >> >> >> >> >> >> fsctl sps >> >> >> >> >> >> is the sps a very low number? >> >> >> >> >> >> Did the sps drop by itself from the value you originally set it to? >> >> >> >> >> >> Are you using 32 bit? >> >> >> >> >> >> if so try all of these commands in your shell before starting FS >> >> >> >> >> >> ulimit -c unlimited >> >> >> ulimit -d unlimited >> >> >> ulimit -f unlimited >> >> >> ulimit -i unlimited >> >> >> ulimit -n 999999 >> >> >> ulimit -q unlimited >> >> >> ulimit -u unlimited >> >> >> ulimit -v unlimited >> >> >> ulimit -x unlimited >> >> >> ulimit -s 244 >> >> >> ulimit -l unlimited >> >> >> >> >> >> >> >> >> DO NOT put them in a script unless you source the script with . >> >> >> . myscript or they will be undone instantly when the script exits >> >> >> >> >> >> BTW, I said to try latest trunk not 1.0.2 We can only debug the >> >> >> development >> >> >> code at this point. >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> On Thu, Jan 29, 2009 at 10:06 AM, shehzad p >> wrote: >> >> >> >> >> >>> >> >> >>> Hi Anthony, >> >> >>> >> >> >>> I found interesting result while testing Freeswitch, and it might >> be >> >> >>> cause >> >> >>> of freezing out of freeswitch, >> >> >>> >> >> >>> I updated my system (as you told) with latest stable version >> >> Freeswitch >> >> >>> 1.0.2 >> >> >>> First of all I set sps to 100, >> >> >>> Then I sends call approximately 100 per seconds, Freeswitch works >> >> fine >> >> >>> and >> >> >>> handles all the calls very well. >> >> >>> >> >> >>> After that I send 130 calls per seconds, and magic happen now, >> >> >>> Freeswitch >> >> >>> handles first 100 calls only. >> >> >>> all the preceding calls were failed (even not appeared in >> freeswitch >> >> >>> console >> >> >>> why?) >> >> >>> >> >> >>> When I put ngrep trace, System responds with 503 Maximum Calls In >> >> >>> Progress. >> >> >>> (as below) >> >> >>> ########################################################### >> >> >>> # >> >> >>> U FSFSFSFSFS -> GWGWGWGWGW >> >> >>> SIP/2.0 503 Maximum Calls In Progress. >> >> >>> Via: SIP/2.0/UDP GWGWGWGWGW;branch=z9hG4bK53eafabb;rport=5060. >> >> >>> From: "99999" ;tag=as2e10c170. >> >> >>> To: ;tag=K3jSUFrDHpmmB. >> >> >>> Call-ID: 0feb229e58afe7be17110deb361dc234 at GWGWGWGWGW. >> >> >>> CSeq: 102 INVITE. >> >> >>> Retry-After: 300. >> >> >>> User-Agent: FreeSWITCH-mod_sofia/1.0.2-exported. >> >> >>> Accept: application/sdp. >> >> >>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, >> SUBSCRIBE, >> >> >>> NOTIFY, >> >> >>> REFER, UPDATE, REGISTER, INFO, PUBLISH. >> >> >>> Supported: timer, precondition, path, replaces. >> >> >>> Allow-Events: talk, presence, dialog, call-info, sla, >> >> >>> include-session-description, presence.winfo, message-summary, >> refer. >> >> >>> Content-Length: 0. >> >> >>> . >> >> >>> >> ##################################################################### >> >> >>> >> >> >>> >> >> >>> Now another issue to note down is that, >> >> >>> After all above happened and active calls comes to zero, >> >> >>> I just make a single call which also fails with response 503 - >> >> Maximum >> >> >>> Calls >> >> >>> In Progress. >> >> >>> >> >> >>> >> >> >>> Is this intended behaviour, should I increase SPS to overcome >> this. >> >> or >> >> >>> something like bug. >> >> >>> >> >> >>> Please let me know what should be the resolution for this. >> >> >>> >> >> >>> Thanks, >> >> >>> msp >> >> >>> >> >> >>> >> >> >>> >> >> >>> Anthony Minessale-2 wrote: >> >> >>> > >> >> >>> > Also remember, >> >> >>> > Actually completely uninstall and erase /usr/local/freeswitch >> and >> >> the >> >> >>> > 1.0.1 >> >> >>> > source tree and freshly install the new one. >> >> >>> > If you try to upgrade on top of a release with trunk it will >> cause >> >> >>> more >> >> >>> > problems for you. >> >> >>> > >> >> >>> > >> >> >>> > On Wed, Jan 28, 2009 at 3:11 AM, Ken Rice >> >> >>> wrote: >> >> >>> > >> >> >>> >> Upgrade to trunk... Many many issues have been resolved since >> >> 1.0.1 >> >> >>> was >> >> >>> >> the >> >> >>> >> current release >> >> >>> >> >> >> >>> >> >> >> >>> >> > From: shehzad p >> >> >>> >> > Reply-To: >> >> >>> >> > Date: Wed, 28 Jan 2009 00:54:13 -0800 (PST) >> >> >>> >> > To: >> >> >>> >> > Subject: [Freeswitch-users] Freeswitch freezes on increasing >> >> call >> >> >>> >> traffic >> >> >>> >> > >> >> >>> >> > >> >> >>> >> > Hi all, >> >> >>> >> > >> >> >>> >> > Yesterday my Freeswitch server faced a problem when call >> traffic >> >> >>> >> increased >> >> >>> >> > to more than 100. >> >> >>> >> > >> >> >>> >> > When I start Freeswitch, it works fine and then after some >> time >> >> >>> >> > (approximately 15 to 20 minutes) it stops functioning (means >> no >> >> >>> call >> >> >>> >> is >> >> >>> >> > being processed, no CLI command is working and it just >> freezes) >> >> >>> until >> >> >>> I >> >> >>> >> > restart the freeswitch. >> >> >>> >> > >> >> >>> >> > I am using Freeswitch 1.0.1. >> >> >>> >> > Debug (gdb) trace as on wiki page >> >> >>> >> > >> >> http://wiki.freeswitch.org/wiki/Debugging_Freeswitch#fs_debug.shis >> >> >>> >> attached >> >> >>> >> > http://www.nabble.com/file/p21701744/fs_debgu.txt >> fs_debgu.txt >> >> >>> >> > -- >> >> >>> >> > View this message in context: >> >> >>> >> > >> >> >>> >> >> >> >>> >> >> >> http://www.nabble.com/Freeswitch-freezes-on-increasing-call-traffic-tp21701744 >> >> >>> >> > p21701744.html >> >> >>> >> > Sent from the Freeswitch-users mailing list archive at >> >> Nabble.com. >> >> >>> >> > >> >> >>> >> > >> >> >>> >> > _______________________________________________ >> >> >>> >> > Freeswitch-users mailing list >> >> >>> >> > Freeswitch-users at lists.freeswitch.org >> >> >>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >>> >> > >> >> >>> >> UNSUBSCRIBE: >> >> >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> >>> >> > http://www.freeswitch.org >> >> >>> >> >> >> >>> >> >> >> >>> >> >> >> >>> >> _______________________________________________ >> >> >>> >> Freeswitch-users mailing list >> >> >>> >> Freeswitch-users at lists.freeswitch.org >> >> >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >>> >> UNSUBSCRIBE: >> >> >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> >>> >> http://www.freeswitch.org >> >> >>> >> >> >> >>> > >> >> >>> > >> >> >>> > >> >> >>> > -- >> >> >>> > Anthony Minessale II >> >> >>> > >> >> >>> > FreeSWITCH http://www.freeswitch.org/ >> >> >>> > ClueCon http://www.cluecon.com/ >> >> >>> > >> >> >>> > AIM: anthm >> >> >>> > >> >> MSN:anthony_minessale at hotmail.com >> >> >> > >> >> >>> >> >> >> >> >> > >> >> >< >> >> >>> >> >> >> MSN%3Aanthony_minessale at hotmail.com >> >> > >> >> >> >> >> > >> >> > >> >> >>> > >> >> >>> > >> >> >>> >> >> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> >> > >> >> >> >> >> > >> >> > >> >> >>> >> >> >> >> >> > >> >> >> >> >> > >> >> > >> >> >>> > >> >> >>> > IRC: irc.freenode.net #freeswitch >> >> >>> > >> >> >>> > FreeSWITCH Developer Conference >> >> >>> > >> >> sip:888 at conference.freeswitch.org >> >> >> > >> >> >>> >> >> >> >> >> > >> >> >< >> >> >>> >> >> >> sip%3A888 at conference.freeswitch.org >> >> > >> >> >> >> >> > >> >> > >> >> >>> > >> >> >>> > iax:guest at conference.freeswitch.org/888 >> >> >>> > >> >> >>> >> >> >> googletalk:conf+888 at conference.freeswitch.org >> >> > >> >> >> >> >> > >> >> > >> >> >>> >> >> >> >> >> > >> >> >> >> >> > >> >> > >> >> >>> > >> >> >>> > pstn:213-799-1400 >> >> >>> > >> >> >>> > _______________________________________________ >> >> >>> > Freeswitch-users mailing list >> >> >>> > Freeswitch-users at lists.freeswitch.org >> >> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >>> > >> >> >>> UNSUBSCRIBE: >> >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> >>> > http://www.freeswitch.org >> >> >>> > >> >> >>> > >> >> >>> >> >> >>> -- >> >> >>> View this message in context: >> >> >>> >> >> >> http://www.nabble.com/Freeswitch-freezes-on-increasing-call-traffic-tp21701744p21729863.html >> >> >>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >>> >> >> >>> >> >> >>> _______________________________________________ >> >> >>> Freeswitch-users mailing list >> >> >>> Freeswitch-users at lists.freeswitch.org >> >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >>> UNSUBSCRIBE: >> >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> >>> http://www.freeswitch.org >> >> >>> >> >> >> >> >> >> >> >> >> >> >> >> -- >> >> >> Anthony Minessale II >> >> >> >> >> >> FreeSWITCH http://www.freeswitch.org/ >> >> >> ClueCon http://www.cluecon.com/ >> >> >> >> >> >> AIM: anthm >> >> >> >> MSN:anthony_minessale at hotmail.com >> >> >> >> >< >> >> >> MSN%3Aanthony_minessale at hotmail.com >> >> > >> >> > >> >> >> >> >> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> >> > >> >> >> >> >> > >> >> > >> >> >> IRC: irc.freenode.net #freeswitch >> >> >> >> >> >> FreeSWITCH Developer Conference >> >> >> >> sip:888 at conference.freeswitch.org >> >> >> >> >< >> >> >> sip%3A888 at conference.freeswitch.org >> >> > >> >> > >> >> >> iax:guest at conference.freeswitch.org/888 >> >> >> >> >> >> googletalk:conf+888 at conference.freeswitch.org >> >> > >> >> >> >> >> > >> >> > >> >> >> pstn:213-799-1400 >> >> >> >> >> >> _______________________________________________ >> >> >> Freeswitch-users mailing list >> >> >> Freeswitch-users at lists.freeswitch.org >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> UNSUBSCRIBE: >> >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> >> http://www.freeswitch.org >> >> >> >> >> >> >> >> > >> >> > >> >> >> >> -- >> >> View this message in context: >> >> >> http://www.nabble.com/Freeswitch-freezes-on-increasing-call-traffic-tp21701744p21749375.html >> >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> >> >> >> _______________________________________________ >> >> Freeswitch-users mailing list >> >> Freeswitch-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> >> >> > >> > >> > >> > -- >> > Anthony Minessale II >> > >> > FreeSWITCH http://www.freeswitch.org/ >> > ClueCon http://www.cluecon.com/ >> > >> > AIM: anthm >> > MSN:anthony_minessale at hotmail.com >> < >> MSN%3Aanthony_minessale at hotmail.com >> > >> > >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> >> > >> > IRC: irc.freenode.net #freeswitch >> > >> > FreeSWITCH Developer Conference >> > sip:888 at conference.freeswitch.org >> < >> sip%3A888 at conference.freeswitch.org >> > >> > iax:guest at conference.freeswitch.org/888 >> > >> googletalk:conf+888 at conference.freeswitch.org >> >> > >> > pstn:213-799-1400 >> > >> > _______________________________________________ >> > Freeswitch-users mailing list >> > Freeswitch-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> >> -- >> View this message in context: >> http://www.nabble.com/Freeswitch-freezes-on-increasing-call-traffic-tp21701744p21761523.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/Freeswitch-freezes-on-increasing-call-traffic-tp21701744p21773300.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From blansky at interwise.com Sun Feb 1 02:37:26 2009 From: blansky at interwise.com (Boris Lansky) Date: Sun, 1 Feb 2009 12:37:26 +0200 Subject: [Freeswitch-users] Outbound Proxy configuration In-Reply-To: <4981D3F8.2030207@freeswitch.org> References: <7160CD9F-F06E-4964-943A-F4B0C12150CC@jerris.com> <4981D3F8.2030207@freeswitch.org> Message-ID: I have checked the "fs_path" usage ... again (I have done it before I have issued my question as well). And once again I can't understand why this thing is useful for me. I have found only two small examples refer the issue that use "fs_path" for a an API call. What I need is a real configuration example that shows configuration of a Proxy Server for all outbound calls going out from a Free Switch. I will real appreciate if I will get an exact answer and not just a general link to a FS doc. Regards, Boris Lansky Unified Communications Telephony Team AT&T Unified Communications Phone: +972.3.976.7604 Fax: +972.3.976.7712 blansky at interwise.com From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Raymond Chandler Sent: Thursday, January 29, 2009 6:06 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Outbound Proxy configuration Boris Lansky wrote: Sorry for the stupid question but in what configuration file should I add such line "sofia/foo/user at that.domain ;fs_path=sip:proxy.this.domain" ? appology accepted... look in the dialplan, there should be plenty of documentations on using the dialplan on our wiki... wiki.freeswitch.org... look for sofia syntax too on the mod_sofia page -Ray -------------- next part -------------- An HTML attachment was scrubbed... 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Name: not available Type: image/png Size: 16211 bytes Desc: image001.png Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090201/3f1f8beb/attachment-0002.png From pmhshz at gmail.com Sun Feb 1 05:35:07 2009 From: pmhshz at gmail.com (shehzad p) Date: Sun, 1 Feb 2009 05:35:07 -0800 (PST) Subject: [Freeswitch-users] Freeswitch freezes on increasing call traffic In-Reply-To: <21773300.post@talk.nabble.com> References: <21701744.post@talk.nabble.com> <191c3a030901280601t3e185ec2m76295110e4888837@mail.gmail.com> <21729863.post@talk.nabble.com> <191c3a030901290831y2ff329b3maee6c1aafa3c68e2@mail.gmail.com> <21745312.post@talk.nabble.com> <21749375.post@talk.nabble.com> <191c3a030901300818i5d85a7aawa0c53b4e54892f40@mail.gmail.com> <21761523.post@talk.nabble.com> <191c3a030901310850s553f8c11oc4e559a710def6d2@mail.gmail.com> <21773300.post@talk.nabble.com> Message-ID: <21775256.post@talk.nabble.com> Hi anthony, There are so many crash occured, on FS 1.0.1 Server, all of them were same (means BT was same) but this last one was looking totally different so I feel that it should be posted... http://www.nabble.com/file/p21775256/bt_full_new.txt bt_full_new.txt shehzad p wrote: > > Hi Anthony, > > How can I handle Garbage collector related things. Is there any way to > look out where js variables are causing some kind of fault related to > Garbage collector? > > There are some loops but they finish in hardly two or three iterations. > > I didn't check session.ready() before/after fetching values from database > (odbc), which might block until response from server comes back. might > this cause fault as you said? > > Thanks for help... > > > > > Anthony Minessale-2 wrote: >> >> Clearly you have an issue with your javascript code. >> >> You have the Garbage collector blocking in every thread. >> >> Are you doing any endless loops in your code where you do not check >> session.ready() as a condition for >> continuing the script? >> >> any time session.ready() fails you must immediately exit. >> >> Are you using session.execute to execute long blocking operations like >> bridging many calls or entering a conference? >> You should avoid doing this as all the collective scripts on the system >> share a common Garbage Collector provided by the >> JS engine and it can lead to the exact issues you describe if the code is >> not properly designed. >> >> What else does you script do that are things provided by FS such as >> playing >> files and executing applications. >> >> >> >> On Sat, Jan 31, 2009 at 3:44 AM, shehzad p wrote: >> >>> >>> Hi Anthony, >>> >>> Freeswitch 1.0.2 was crashed on last test again... >>> BT is on http://pastebin.freeswitch.org/6979. >>> >>> I tried to use scripts/freeswitch-gcore, to capture resident memory, but >>> before the command complete the process system hanged up so only half >>> output >>> was captured.< http://www.nabble.com/file/p21761523/gcore-fs.txt >>> gcore-fs.txt > >>> >>> >>> Now I have checkout from trunk and will post back if any new thing >>> found... >>> >>> Thanks, >>> msp >>> >>> Anthony Minessale-2 wrote: >>> > >>> > if you are using unix you can use the supplied script >>> > >>> > scripts/freeswitch-gcore >>> > >>> > to capture a copy of the resident memory and I can have a look >>> perhaps. >>> > >>> > Trunk is safe for production as we are in beta stage for a release of >>> > 1.0.3 >>> > at this time. >>> > >>> > >>> > >>> > On Fri, Jan 30, 2009 at 9:29 AM, shehzad p wrote: >>> > >>> >> >>> >> When freeswitch freezes, we can't connect to it to check sps status, >>> >> but once we were able to connect and at that time it was showing 0/0 >>> sps. >>> >> >>> >> thanks... >>> >> >>> >> shehzad p wrote: >>> >> > >>> >> > Thanks, Anthony >>> >> > >>> >> > In my previous test sps did not changed, >>> >> > but in recent test sps was dropped to 0 itself (as below). >>> >> > =============================================================== >>> >> > UP 0 years, 0 days, 5 hours, 1 minute, 53 seconds, 878 >>> milliseconds, >>> >> 190 >>> >> > microseconds >>> >> > 5474 session(s) since startup >>> >> > 75 session(s) 0/0 >>> >> > ============================================================= >>> >> > >>> >> > My system is 32 bit. >>> >> > CPU is Intel(R) Xeon(R) CPU X3220 @ 2.40GHz >>> >> > And RAM is 4GB >>> >> > >>> >> > Output of ulimit -a is: >>> >> > ulimit -a: (set after first test) >>> >> > core file size (blocks, -c) unlimited >>> >> > data seg size (kbytes, -d) unlimited >>> >> > max nice (-e) 20 >>> >> > file size (blocks, -f) unlimited >>> >> > pending signals (-i) unlimited >>> >> > max locked memory (kbytes, -l) unlimited >>> >> > max memory size (kbytes, -m) unlimited >>> >> > open files (-n) 999999 >>> >> > pipe size (512 bytes, -p) 8 >>> >> > POSIX message queues (bytes, -q) unlimited >>> >> > max rt priority (-r) unlimited >>> >> > stack size (kbytes, -s) 244 >>> >> > cpu time (seconds, -t) unlimited >>> >> > max user processes (-u) unlimited >>> >> > virtual memory (kbytes, -v) unlimited >>> >> > file locks (-x) unlimited >>> >> > =================================================== >>> >> > >>> >> > >>> >> > BTW using trunk on production system is safe? >>> >> > >>> >> > Warm thanks for kind responses... >>> >> > >>> >> > >>> >> > >>> >> > Anthony Minessale-2 wrote: >>> >> >> >>> >> >> When you get it in that state what do you see when you execute >>> >> >> >>> >> >> fsctl sps >>> >> >> >>> >> >> is the sps a very low number? >>> >> >> >>> >> >> Did the sps drop by itself from the value you originally set it >>> to? >>> >> >> >>> >> >> Are you using 32 bit? >>> >> >> >>> >> >> if so try all of these commands in your shell before starting FS >>> >> >> >>> >> >> ulimit -c unlimited >>> >> >> ulimit -d unlimited >>> >> >> ulimit -f unlimited >>> >> >> ulimit -i unlimited >>> >> >> ulimit -n 999999 >>> >> >> ulimit -q unlimited >>> >> >> ulimit -u unlimited >>> >> >> ulimit -v unlimited >>> >> >> ulimit -x unlimited >>> >> >> ulimit -s 244 >>> >> >> ulimit -l unlimited >>> >> >> >>> >> >> >>> >> >> DO NOT put them in a script unless you source the script with . >>> >> >> . myscript or they will be undone instantly when the script exits >>> >> >> >>> >> >> BTW, I said to try latest trunk not 1.0.2 We can only debug the >>> >> >> development >>> >> >> code at this point. >>> >> >> >>> >> >> >>> >> >> >>> >> >> >>> >> >> >>> >> >> On Thu, Jan 29, 2009 at 10:06 AM, shehzad p >>> wrote: >>> >> >> >>> >> >>> >>> >> >>> Hi Anthony, >>> >> >>> >>> >> >>> I found interesting result while testing Freeswitch, and it might >>> be >>> >> >>> cause >>> >> >>> of freezing out of freeswitch, >>> >> >>> >>> >> >>> I updated my system (as you told) with latest stable version >>> >> Freeswitch >>> >> >>> 1.0.2 >>> >> >>> First of all I set sps to 100, >>> >> >>> Then I sends call approximately 100 per seconds, Freeswitch works >>> >> fine >>> >> >>> and >>> >> >>> handles all the calls very well. >>> >> >>> >>> >> >>> After that I send 130 calls per seconds, and magic happen now, >>> >> >>> Freeswitch >>> >> >>> handles first 100 calls only. >>> >> >>> all the preceding calls were failed (even not appeared in >>> freeswitch >>> >> >>> console >>> >> >>> why?) >>> >> >>> >>> >> >>> When I put ngrep trace, System responds with 503 Maximum Calls In >>> >> >>> Progress. >>> >> >>> (as below) >>> >> >>> ########################################################### >>> >> >>> # >>> >> >>> U FSFSFSFSFS -> GWGWGWGWGW >>> >> >>> SIP/2.0 503 Maximum Calls In Progress. >>> >> >>> Via: SIP/2.0/UDP GWGWGWGWGW;branch=z9hG4bK53eafabb;rport=5060. >>> >> >>> From: "99999" ;tag=as2e10c170. >>> >> >>> To: ;tag=K3jSUFrDHpmmB. >>> >> >>> Call-ID: 0feb229e58afe7be17110deb361dc234 at GWGWGWGWGW. >>> >> >>> CSeq: 102 INVITE. >>> >> >>> Retry-After: 300. >>> >> >>> User-Agent: FreeSWITCH-mod_sofia/1.0.2-exported. >>> >> >>> Accept: application/sdp. >>> >> >>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, >>> SUBSCRIBE, >>> >> >>> NOTIFY, >>> >> >>> REFER, UPDATE, REGISTER, INFO, PUBLISH. >>> >> >>> Supported: timer, precondition, path, replaces. >>> >> >>> Allow-Events: talk, presence, dialog, call-info, sla, >>> >> >>> include-session-description, presence.winfo, message-summary, >>> refer. >>> >> >>> Content-Length: 0. >>> >> >>> . >>> >> >>> >>> ##################################################################### >>> >> >>> >>> >> >>> >>> >> >>> Now another issue to note down is that, >>> >> >>> After all above happened and active calls comes to zero, >>> >> >>> I just make a single call which also fails with response 503 - >>> >> Maximum >>> >> >>> Calls >>> >> >>> In Progress. >>> >> >>> >>> >> >>> >>> >> >>> Is this intended behaviour, should I increase SPS to overcome >>> this. >>> >> or >>> >> >>> something like bug. >>> >> >>> >>> >> >>> Please let me know what should be the resolution for this. >>> >> >>> >>> >> >>> Thanks, >>> >> >>> msp >>> >> >>> >>> >> >>> >>> >> >>> >>> >> >>> Anthony Minessale-2 wrote: >>> >> >>> > >>> >> >>> > Also remember, >>> >> >>> > Actually completely uninstall and erase /usr/local/freeswitch >>> and >>> >> the >>> >> >>> > 1.0.1 >>> >> >>> > source tree and freshly install the new one. >>> >> >>> > If you try to upgrade on top of a release with trunk it will >>> cause >>> >> >>> more >>> >> >>> > problems for you. >>> >> >>> > >>> >> >>> > >>> >> >>> > On Wed, Jan 28, 2009 at 3:11 AM, Ken Rice >>> >>> >> >>> wrote: >>> >> >>> > >>> >> >>> >> Upgrade to trunk... Many many issues have been resolved since >>> >> 1.0.1 >>> >> >>> was >>> >> >>> >> the >>> >> >>> >> current release >>> >> >>> >> >>> >> >>> >> >>> >> >>> >> > From: shehzad p >>> >> >>> >> > Reply-To: >>> >> >>> >> > Date: Wed, 28 Jan 2009 00:54:13 -0800 (PST) >>> >> >>> >> > To: >>> >> >>> >> > Subject: [Freeswitch-users] Freeswitch freezes on >>> increasing >>> >> call >>> >> >>> >> traffic >>> >> >>> >> > >>> >> >>> >> > >>> >> >>> >> > Hi all, >>> >> >>> >> > >>> >> >>> >> > Yesterday my Freeswitch server faced a problem when call >>> traffic >>> >> >>> >> increased >>> >> >>> >> > to more than 100. >>> >> >>> >> > >>> >> >>> >> > When I start Freeswitch, it works fine and then after some >>> time >>> >> >>> >> > (approximately 15 to 20 minutes) it stops functioning >>> (means >>> no >>> >> >>> call >>> >> >>> >> is >>> >> >>> >> > being processed, no CLI command is working and it just >>> freezes) >>> >> >>> until >>> >> >>> I >>> >> >>> >> > restart the freeswitch. >>> >> >>> >> > >>> >> >>> >> > I am using Freeswitch 1.0.1. >>> >> >>> >> > Debug (gdb) trace as on wiki page >>> >> >>> >> > >>> >> http://wiki.freeswitch.org/wiki/Debugging_Freeswitch#fs_debug.shis >>> >> >>> >> attached >>> >> >>> >> > http://www.nabble.com/file/p21701744/fs_debgu.txt >>> fs_debgu.txt >>> >> >>> >> > -- >>> >> >>> >> > View this message in context: >>> >> >>> >> > >>> >> >>> >> >>> >> >>> >>> >> >>> http://www.nabble.com/Freeswitch-freezes-on-increasing-call-traffic-tp21701744 >>> >> >>> >> > p21701744.html >>> >> >>> >> > Sent from the Freeswitch-users mailing list archive at >>> >> Nabble.com. >>> >> >>> >> > >>> >> >>> >> > >>> >> >>> >> > _______________________________________________ >>> >> >>> >> > Freeswitch-users mailing list >>> >> >>> >> > Freeswitch-users at lists.freeswitch.org >>> >> >>> >> > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >> >>> >> > >>> >> >>> >> UNSUBSCRIBE: >>> >> >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >> >>> >> > http://www.freeswitch.org >>> >> >>> >> >>> >> >>> >> >>> >> >>> >> >>> >> >>> >> _______________________________________________ >>> >> >>> >> Freeswitch-users mailing list >>> >> >>> >> Freeswitch-users at lists.freeswitch.org >>> >> >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >> >>> >> UNSUBSCRIBE: >>> >> >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >> >>> >> http://www.freeswitch.org >>> >> >>> >> >>> >> >>> > >>> >> >>> > >>> >> >>> > >>> >> >>> > -- >>> >> >>> > Anthony Minessale II >>> >> >>> > >>> >> >>> > FreeSWITCH http://www.freeswitch.org/ >>> >> >>> > ClueCon http://www.cluecon.com/ >>> >> >>> > >>> >> >>> > AIM: anthm >>> >> >>> > >>> >> MSN:anthony_minessale at hotmail.com >>> >>> >>> > >>> >> >>> >>> >> >>> >>> >>> > >>> >> >< >>> >> >>> >>> >> >>> MSN%3Aanthony_minessale at hotmail.com >>> >>> > >>> >> >>> >>> >>> > >>> >> > >>> >> >>> > >>> >> >>> > >>> >> >>> >>> >> >>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>> >>> > >>> >> >>> >>> >>> > >>> >> > >>> >> >>> >>> >> >>> >>> >>> > >>> >> >>> >>> >>> > >>> >> > >>> >> >>> > >>> >> >>> > IRC: irc.freenode.net #freeswitch >>> >> >>> > >>> >> >>> > FreeSWITCH Developer Conference >>> >> >>> > >>> >> sip:888 at conference.freeswitch.org >>> >>> >>> > >>> >> >>> >>> >> >>> >>> >>> > >>> >> >< >>> >> >>> >>> >> >>> sip%3A888 at conference.freeswitch.org >>> >>> > >>> >> >>> >>> >>> > >>> >> > >>> >> >>> > >>> >> >>> > iax:guest at conference.freeswitch.org/888 >>> >> >>> > >>> >> >>> >>> >> >>> googletalk:conf+888 at conference.freeswitch.org >>> >>> > >>> >> >>> >>> >>> > >>> >> > >>> >> >>> >>> >> >>> >>> >>> > >>> >> >>> >>> >>> > >>> >> > >>> >> >>> > >>> >> >>> > pstn:213-799-1400 >>> >> >>> > >>> >> >>> > _______________________________________________ >>> >> >>> > Freeswitch-users mailing list >>> >> >>> > Freeswitch-users at lists.freeswitch.org >>> >> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >> >>> > >>> >> >>> UNSUBSCRIBE: >>> >> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >> >>> > http://www.freeswitch.org >>> >> >>> > >>> >> >>> > >>> >> >>> >>> >> >>> -- >>> >> >>> View this message in context: >>> >> >>> >>> >> >>> http://www.nabble.com/Freeswitch-freezes-on-increasing-call-traffic-tp21701744p21729863.html >>> >> >>> Sent from the Freeswitch-users mailing list archive at >>> Nabble.com. >>> >> >>> >>> >> >>> >>> >> >>> _______________________________________________ >>> >> >>> Freeswitch-users mailing list >>> >> >>> Freeswitch-users at lists.freeswitch.org >>> >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >> >>> UNSUBSCRIBE: >>> >> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >> >>> http://www.freeswitch.org >>> >> >>> >>> >> >> >>> >> >> >>> >> >> >>> >> >> -- >>> >> >> Anthony Minessale II >>> >> >> >>> >> >> FreeSWITCH http://www.freeswitch.org/ >>> >> >> ClueCon http://www.cluecon.com/ >>> >> >> >>> >> >> AIM: anthm >>> >> >> >>> MSN:anthony_minessale at hotmail.com >>> >> >>> >>> >< >>> >> >>> MSN%3Aanthony_minessale at hotmail.com >>> >>> > >>> >> > >>> >> >> >>> >> >>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>> >>> > >>> >> >>> >>> >>> > >>> >> > >>> >> >> IRC: irc.freenode.net #freeswitch >>> >> >> >>> >> >> FreeSWITCH Developer Conference >>> >> >> >>> sip:888 at conference.freeswitch.org >>> >> >>> >>> >< >>> >> >>> sip%3A888 at conference.freeswitch.org >>> >>> > >>> >> > >>> >> >> iax:guest at conference.freeswitch.org/888 >>> >> >> >>> >> >>> googletalk:conf+888 at conference.freeswitch.org >>> >>> > >>> >> >>> >>> >>> > >>> >> > >>> >> >> pstn:213-799-1400 >>> >> >> >>> >> >> _______________________________________________ >>> >> >> Freeswitch-users mailing list >>> >> >> Freeswitch-users at lists.freeswitch.org >>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >> >> UNSUBSCRIBE: >>> >> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >> >> http://www.freeswitch.org >>> >> >> >>> >> >> >>> >> > >>> >> > >>> >> >>> >> -- >>> >> View this message in context: >>> >> >>> http://www.nabble.com/Freeswitch-freezes-on-increasing-call-traffic-tp21701744p21749375.html >>> >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>> >> >>> >> >>> >> _______________________________________________ >>> >> Freeswitch-users mailing list >>> >> Freeswitch-users at lists.freeswitch.org >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >> UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >> http://www.freeswitch.org >>> >> >>> > >>> > >>> > >>> > -- >>> > Anthony Minessale II >>> > >>> > FreeSWITCH http://www.freeswitch.org/ >>> > ClueCon http://www.cluecon.com/ >>> > >>> > AIM: anthm >>> > MSN:anthony_minessale at hotmail.com >>> < >>> MSN%3Aanthony_minessale at hotmail.com >>> > >>> > >>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>> >>> > >>> > IRC: irc.freenode.net #freeswitch >>> > >>> > FreeSWITCH Developer Conference >>> > sip:888 at conference.freeswitch.org >>> < >>> sip%3A888 at conference.freeswitch.org >>> > >>> > iax:guest at conference.freeswitch.org/888 >>> > >>> googletalk:conf+888 at conference.freeswitch.org >>> >>> > >>> > pstn:213-799-1400 >>> > >>> > _______________________________________________ >>> > Freeswitch-users mailing list >>> > Freeswitch-users at lists.freeswitch.org >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> > >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> > http://www.freeswitch.org >>> > >>> > >>> >>> -- >>> View this message in context: >>> http://www.nabble.com/Freeswitch-freezes-on-increasing-call-traffic-tp21701744p21761523.html >>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:213-799-1400 >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> > > -- View this message in context: http://www.nabble.com/Freeswitch-freezes-on-increasing-call-traffic-tp21701744p21775256.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From brian at freeswitch.org Sun Feb 1 08:43:50 2009 From: brian at freeswitch.org (Brian West) Date: Sun, 1 Feb 2009 10:43:50 -0600 Subject: [Freeswitch-users] Freeswitch freezes on increasing call traffic In-Reply-To: <21775256.post@talk.nabble.com> References: <21701744.post@talk.nabble.com> <191c3a030901280601t3e185ec2m76295110e4888837@mail.gmail.com> <21729863.post@talk.nabble.com> <191c3a030901290831y2ff329b3maee6c1aafa3c68e2@mail.gmail.com> <21745312.post@talk.nabble.com> <21749375.post@talk.nabble.com> <191c3a030901300818i5d85a7aawa0c53b4e54892f40@mail.gmail.com> <21761523.post@talk.nabble.com> <191c3a030901310850s553f8c11oc4e559a710def6d2@mail.gmail.com> <21773300.post@talk.nabble.com> <21775256.post@talk.nabble.com> Message-ID: <30BF0913-A90C-4472-9A70-6CD6AEE2AD08@freeswitch.org> I highly recommend you upgrade to 1.03RC1 or SVN Trunk. Chances are we have already fixed these issues. /b On Feb 1, 2009, at 7:35 AM, shehzad p wrote: > > Hi anthony, > > There are so many crash occured, on FS 1.0.1 Server, all of them > were same > (means BT was same) > but this last one was looking totally different so I feel that it > should be > posted... > http://www.nabble.com/file/p21775256/bt_full_new.txt bt_full_new.txt -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090201/3297f1f6/attachment-0002.html From peder at networkoblivion.com Sun Feb 1 10:03:11 2009 From: peder at networkoblivion.com (peder at networkoblivion.com) Date: Sun, 01 Feb 2009 12:03:11 -0600 Subject: [Freeswitch-users] Cisco 7975G and XML options for G722 In-Reply-To: <897947BF-9886-47BB-B8BB-8DE8849437F0@freeswitch.org> References: <0C8E2714-6D96-49DB-93CD-0D377E4DAE96@freeswitch.org> <3885f4fe0901312115k1309573lb160ecbc83f8d57@mail.gmail.com> <897947BF-9886-47BB-B8BB-8DE8849437F0@freeswitch.org> Message-ID: <4985E3DF.1000805@networkoblivion.com> I have an xml config file for the 79x1, but there is no mention of wideband and/or g722. I found a doc on CCO that says that the 7941 and 7961 support g722 for sccp and sip, so if you find some xml that mentions g722, I would appreciate seeing it as I have a couple of those phones. I know there is a "" entry, so maybe you just add it there like on the 79x0 versions 'preferred_codec: "g729"'. Peder Brian West wrote: > I found it finally. Now if I had the full XML I could pick apart for > all the options that would help too. > > /b > > On Jan 31, 2009, at 11:15 PM, Ron McCarthy wrote: > >> Wow, can't even find the tech doc via my CCO login even. Gotta love >> Cisco and the support for SIP... >> >> Probably have to find someone with CCM and look at a config that CCM >> made up, as that should have it if they enabled for the phone. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > ' From anthony.minessale at gmail.com Sun Feb 1 10:19:00 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sun, 1 Feb 2009 12:19:00 -0600 Subject: [Freeswitch-users] Outbound Proxy configuration In-Reply-To: References: <7160CD9F-F06E-4964-943A-F4B0C12150CC@jerris.com> <4981D3F8.2030207@freeswitch.org> Message-ID: <191c3a030902011019k94e5fa3i1a37c2c0b2e66e52@mail.gmail.com> They probably saw your signature that says you work for AT&T and assumed you could follow the documentation since AT&T is a telephone company. What they were trying to explain was if you want to send a call out of a sofia profile you can set the proxy on the fly with that extra parameter added to the dial string anywhere it's used. The explanation requires you understand the basic operation of freeswitch. the uri contained in fs_path indicates the sip address of a proxy server. On Sun, Feb 1, 2009 at 4:37 AM, Boris Lansky wrote: > I have checked the "fs_path" usage ? again (I have done it before I have > issued my question as well). And once again I can't understand why this > thing is useful for me. I have found only two small examples refer the issue > that use "fs_path" for a an API call. What I need is a real configuration > example that shows configuration of a Proxy Server for all outbound calls > going out from a Free Switch. I will real appreciate if I will get an exact > answer and not just a general link to a FS doc. > > > > Regards, > > > > *Boris Lansky* > *Unified Communications Telephony Team* > > *AT&T Unified Communications > *Phone: +972.3.976.7604 > > Fax: +972.3.976.7712 > > blansky at interwise.com > > [image: cid:image001.png at 01C8E00D.2EC2BC90] > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Raymond > Chandler > *Sent:* Thursday, January 29, 2009 6:06 PM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Outbound Proxy configuration > > > > Boris Lansky wrote: > > Sorry for the stupid question but in what configuration file should I add such line "sofia/foo/user at that.domain ;fs_path=sip:proxy.this.domain" ? > > > > appology accepted... look in the dialplan, there should be plenty of > documentations on using the dialplan on our wiki... wiki.freeswitch.org... > look for sofia syntax too on the mod_sofia page > > -Ray > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... 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Name: not available Type: image/png Size: 16211 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090201/0fdbe3e8/attachment-0002.png From brian at freeswitch.org Sun Feb 1 10:33:29 2009 From: brian at freeswitch.org (Brian West) Date: Sun, 1 Feb 2009 12:33:29 -0600 Subject: [Freeswitch-users] Cisco 7975G and XML options for G722 In-Reply-To: <4985E3DF.1000805@networkoblivion.com> References: <0C8E2714-6D96-49DB-93CD-0D377E4DAE96@freeswitch.org> <3885f4fe0901312115k1309573lb160ecbc83f8d57@mail.gmail.com> <897947BF-9886-47BB-B8BB-8DE8849437F0@freeswitch.org> <4985E3DF.1000805@networkoblivion.com> Message-ID: Cisco 7970 and the G.722 codecThe G.722 codec is disabled by default on the 7970G phone. - To enable the G722 codec, add the following line inside of the context: 2 - Add the following line within the context: 1 On a config generated by CUCM 6, the 'advertiseG722Codec' context usually appears on a new line following the context. - It is also useful to add the following lines inside of the context, but these are purely optional: 0 0 0 1 From voip-info. /b On Feb 1, 2009, at 12:03 PM, peder at networkoblivion.com wrote: > I have an xml config file for the 79x1, but there is no mention of > wideband and/or g722. I found a doc on CCO that says that the 7941 > and > 7961 support g722 for sccp and sip, so if you find some xml that > mentions g722, I would appreciate seeing it as I have a couple of > those > phones. I know there is a "" > entry, so > maybe you just add it there like on the 79x0 versions > 'preferred_codec: > "g729"'. > > Peder > > > Brian West wrote: >> I found it finally. Now if I had the full XML I could pick apart >> for >> all the options that would help too. >> >> /b >> >> On Jan 31, 2009, at 11:15 PM, Ron McCarthy wrote: >> >>> Wow, can't even find the tech doc via my CCO login even. Gotta love >>> Cisco and the support for SIP... >>> >>> Probably have to find someone with CCM and look at a config that CCM >>> made up, as that should have it if they enabled for the phone. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > ' > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From gkuri at ieee.org Sun Feb 1 12:55:07 2009 From: gkuri at ieee.org (Gabriel Kuri) Date: Sun, 01 Feb 2009 12:55:07 -0800 Subject: [Freeswitch-users] Cisco 7975G and XML options for G722 In-Reply-To: <897947BF-9886-47BB-B8BB-8DE8849437F0@freeswitch.org> References: <0C8E2714-6D96-49DB-93CD-0D377E4DAE96@freeswitch.org> <3885f4fe0901312115k1309573lb160ecbc83f8d57@mail.gmail.com> <897947BF-9886-47BB-B8BB-8DE8849437F0@freeswitch.org> Message-ID: <49860C2B.7020801@ieee.org> I don't know if this helps, but I attached a config file generated by CUCM for a 7975. I don't believe CUCM writes out all the possible config options into the XML file, although it does write out quite a bit. If you're looking for another option, let me know and I can see if I can enable it and write out the config again. Gabe Brian West wrote: > I found it finally. Now if I had the full XML I could pick apart for > all the options that would help too. > > /b > > On Jan 31, 2009, at 11:15 PM, Ron McCarthy wrote: > >> Wow, can't even find the tech doc via my CCO login even. Gotta love >> Cisco and the support for SIP... >> >> Probably have to find someone with CCM and look at a config that CCM >> made up, as that should have it if they enabled for the phone. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- A non-text attachment was scrubbed... Name: SEP000E03123456.cnf.xml Type: text/xml Size: 7980 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090201/bce27527/attachment-0002.xml From anthony.minessale at gmail.com Sun Feb 1 10:54:59 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sun, 1 Feb 2009 12:54:59 -0600 Subject: [Freeswitch-users] TDM400 FXO can dialout only once In-Reply-To: <7d0bfd8c0902010117h7e26a3a6x2112225cb5951cd8@mail.gmail.com> References: <7d0bfd8c0901250021q2969fe26u5ea79c3957989ffd@mail.gmail.com> <191c3a030901251818g284054ben60c0bfa61157824@mail.gmail.com> <7d0bfd8c0901252305v5a33fa02w4bc02deaa0c1b14b@mail.gmail.com> <87f2f3b90901260201r2e4d42a3x30367eee64b0e13b@mail.gmail.com> <7d0bfd8c0901260532g50deb359v3c92f5bb372f11a4@mail.gmail.com> <191c3a030901260833l2e07bfffkca9400d33a5483fe@mail.gmail.com> <7d0bfd8c0901261639x7c53f256yabf7347b5d22e7e4@mail.gmail.com> <7d0bfd8c0902010030n6b5d3c35k7539fac38e07300d@mail.gmail.com> <7d0bfd8c0902010117h7e26a3a6x2112225cb5951cd8@mail.gmail.com> Message-ID: <191c3a030902011054k1d3df502ydca8ae1b645abb10@mail.gmail.com> the value is dB already -7dB but for the detection section you do not specify anything but the list of frequencies needed to detect. detect-ring => 440,480 this means it needs to detect a 440+480 to know there is a dialtone, the generate value has nothing to do with it. openzap does not currently do busy detection to detect a hangup, but up in freeswitch you can use the tone_detect app to do this. On Sun, Feb 1, 2009 at 3:17 AM, Nandy Dagondon wrote: > hi, > > i found a major one. this time i deliberately set the dialtone freq to US > std on my PH definition. i expect FS wont dial at all. but to my surprise, > the problem is gone!! i checked the log. it indicates successful detection > of DIALTONE. > > going on further. i noticed FXO wont hangup on busy tone. > > one possibility is the volume settings. the default is -7. how many dBm is > this? and what is the dB equivalent per increment? > > tks n rgds, > nandy > > > On Sun, Feb 1, 2009 at 4:30 PM, Nandy Dagondon wrote: > >> hi everybody, >> >> i created [ph] tone definition per raul's suggestion and changed >> /etc/zaptel.conf entries to: >> tonezone=ph >> defaultzone=ph >> >> but it didn't solve the problem. >> i captured the console log during start-up and shutdown. i noticed openzap >> related errors during shutdown. here's the snippet of the log: >> >> STARTUP >> --------------- >> 2009-02-01 15:58:10 [NOTICE] zap_io.c:2517 zap_global_init() Modules >> configured: 1 >> 2009-02-01 15:58:10 [INFO] zap_io.c:2341 zap_load_module() Loading IO from >> /opt/freeswitch/mod/ozmod_zt.so >> 2009-02-01 15:58:10 [INFO] zap_io.c:2127 load_config() auto-loaded 'zt' >> 2009-02-01 15:58:10 [INFO] ozmod_zt.c:186 zt_open_range() configuring >> device /dev/zap/channel channel 1 as OpenZAP device 1:1 fd:39 >> 2009-02-01 15:58:10 [INFO] ozmod_zt.c:186 zt_open_range() configuring >> device /dev/zap/channel channel 2 as OpenZAP device 2:1 fd:40 >> 2009-02-01 15:58:10 [INFO] zap_io.c:2265 load_config() Configured 2 >> channel(s) >> 2009-02-01 15:58:10 [INFO] zap_io.c:2358 zap_load_module() Loading SIG >> from /opt/freeswitch/mod/ozmod_analog.so >> 2009-02-01 15:58:10 [INFO] zap_io.c:2474 zap_configure_span() auto-loaded >> 'analog' >> 2009-02-01 15:58:10 [CONSOLE] switch_loadable_module.c:857 >> switch_loadable_module_load_file() Successfully Loaded [mod_openzap] >> >> --- DIDN'T MAKE ANY CALL --- >> >> SHUTDOWN >> ------------------ >> 2009-02-01 15:59:07 [NOTICE] switch_loadable_module.c:536 >> switch_loadable_module_unprocess() Deleting API Function 'oz' >> 2009-02-01 15:59:07 [CONSOLE] switch_loadable_module.c:1231 do_shutdown() >> Stopping: mod_openzap >> 2009-02-01 15:59:07 [INFO] zap_io.c:256 zap_channel_destroy() Closing >> channel zt:1:1 fd:39 >> 2009-02-01 15:59:07 [INFO] zap_io.c:256 zap_channel_destroy() Closing >> channel zt:2:1 fd:40 >> 2009-02-01 15:59:08 [ERR] ozmod_analog.c:899 zap_analog_run() Failure >> Polling event! [no matching descriptor] >> 2009-02-01 15:59:08 [ERR] ozmod_analog.c:899 zap_analog_run() Failure >> Polling event! [no matching descriptor] >> 2009-02-01 15:59:08 [INFO] zap_io.c:2456 zap_unload_modules() Unloading >> /opt/freeswitch/mod/ozmod_analog.so >> 2009-02-01 15:59:08 [INFO] zap_io.c:2441 zap_unload_modules() Unloading IO >> zt >> 2009-02-01 15:59:08 [INFO] zap_io.c:2456 zap_unload_modules() Unloading >> /opt/freeswitch/mod/ozmod_zt.so >> >> i also notice the same ERR flag during shutdown after making test calls. >> >> any suggestion what to do next? >> >> tks for your assistance. >> >> rgds, >> -nandy >> >> >> On Tue, Jan 27, 2009 at 8:39 AM, Nandy Dagondon wrote: >> >>> i tested the SVN trunk version. still the same behaviour. >>> -nandy >>> >>> >>> On Tue, Jan 27, 2009 at 12:33 AM, Anthony Minessale < >>> anthony.minessale at gmail.com> wrote: >>> >>>> The unhanded type message just means that mod_openzap does not do >>>> anything with the TONE_DETECTED event that was passed >>>> up from the ozmod_analog. >>>> >>>> On Mon, Jan 26, 2009 at 7:32 AM, Nandy Dagondon wrote: >>>> >>>>> that's great. yes, i'm in the philippines. there's a difference in >>>>> dialtone - it's 425 Hz. >>>>> -nandy >>>>> >>>>> >>>>> >>>>> On Mon, Jan 26, 2009 at 6:01 PM, Michael Collins wrote: >>>>> >>>>>> I have a TDM400 clone and I will see if I can reproduce these >>>>>> symptoms. BTW, are you in the Philippines? Is there any difference in >>>>>> the dial tone there than in the US? >>>>>> -MC >>>>>> >>>>>> On Sun, Jan 25, 2009 at 11:05 PM, Nandy Dagondon wrote: >>>>>> > i monitored the line using another phone. there's indeed dialtone in >>>>>> all >>>>>> > attempts. >>>>>> > i see TONE_DETECTED in the first call but i wonder there's a WARNING >>>>>> message >>>>>> > immediately following [WARNING] mod_openzap.c:1196 on_fxo_signal() >>>>>> Unhandled >>>>>> > type for channel 2:1. >>>>>> > the dialtone freq should be okay since it's detected in the first >>>>>> call.could >>>>>> > the WARNING message gives us a hint of a possible problem other than >>>>>> the >>>>>> > dialtone freq? >>>>>> > >>>>>> > okay, i'll try the SVN version next. >>>>>> > >>>>>> > >>>>>> > On Mon, Jan 26, 2009 at 10:18 AM, Anthony Minessale >>>>>> > wrote: >>>>>> >> >>>>>> >> Its not detecting a dial tone on the failure case. >>>>>> >> Before dialing it waits until it picks up dialtone. >>>>>> >> Try the svn trunk version to see if it works any better or verify >>>>>> there is >>>>>> >> a dialtone on the line. >>>>>> >> >>>>>> >> On Jan 25, 2009 6:19 PM, "Nandy Dagondon" wrote: >>>>>> >> >>>>>> >> hi everybody, >>>>>> >> >>>>>> >> i installed FS 1.0.2 to an Atom mobo (Intel D945GCLF2). it's >>>>>> working using >>>>>> >> IP phones, softphones and digium FXS port. but there's a problem in >>>>>> dialing >>>>>> >> out to PSTN using digium tdm400 fxo - it works fine on the first >>>>>> attempt >>>>>> >> (after starting FS) but it fails on the subsequent attempts. i >>>>>> tested to >>>>>> >> call using the FXS port and IP phone. same problem. >>>>>> >> >>>>>> >> before i place any call, i checked >oz dump 2 1 (show current >>>>>> state = >>>>>> >> DOWN, last state = DOWN) >>>>>> >> >>>>>> >> in the first call, there's this message: >>>>>> >> [WARNING] mod_openzap.c:1196 on_fxo_signal() Unhandled type for >>>>>> channel >>>>>> >> 2:1 >>>>>> >> but >>>>>> >> >>>>>> >> then i hangup. checked >oz dump 21 (show current state=DOWN, last >>>>>> >> state=HANGUP) >>>>>> >> >>>>>> >> in the 2nd (and subsequent) attempts, the fxo just goes off-hook >>>>>> but >>>>>> >> doesn't send the dtmf tones. >>>>>> >> >oz dump 2 1 (shows current state = DIALING, last state = DOWN) >>>>>> >> >>>>>> >> has anyone encountered this problem before? i appreciate for any >>>>>> help to >>>>>> >> correct this problem. >>>>>> >> >>>>>> >> tks, >>>>>> >> nandy >>>>>> >> >>>>>> >> >>>>>> >> Environment: >>>>>> >> ================== >>>>>> >> kernel 2.6.18-92.1.22.el5 >>>>>> >> FS 1.0.2 >>>>>> >> zaptel 1.4.11 >>>>>> >> oslec >>>>>> >> digium TDM400P Rev. I (1-FXS, 2-FXO,3-4 vacant) >>>>>> >> >>>>>> >> zaptel.conf >>>>>> >> ======== >>>>>> >> loadzone = us >>>>>> >> defaultzone=us >>>>>> >> channels=1-2 >>>>>> >> alaw=1-4 >>>>>> >> fxsks=2 >>>>>> >> fxoks=1 >>>>>> >> >>>>>> >> >>>>>> >> openzap.conf.xml: >>>>>> >> =============== >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> >> openzap.conf >>>>>> >> ========== >>>>>> >> [span zt] >>>>>> >> name => OpenZAP FXS >>>>>> >> number => 1 >>>>>> >> fxs-channel => 1 >>>>>> >> >>>>>> >> [span zt] >>>>>> >> name => OpenZAP FXO >>>>>> >> number => 2 >>>>>> >> fxo-channel => 2 >>>>>> >> >>>>>> >> tones.conf (the dialtone and ring tone is set to Philipping tones) >>>>>> >> ======== >>>>>> >> [us] >>>>>> >> generate-dial => v=-7;%(1000,0,425) >>>>>> >> detect-dial => 425 >>>>>> >> >>>>>> >> generate-ring => v=-7;%(1000,4000,425,480) >>>>>> >> detect-ring => 425,480 >>>>>> >> >>>>>> >> generate-busy => v=-7;%(500,500,480,620) >>>>>> >> detect-busy => 480,620 >>>>>> >> >>>>>> >> generate-attn => v=0;%(200,300,1400,1800) >>>>>> >> detect-attn => 1400,1800 >>>>>> >> >>>>>> >> generate-callwaiting-sas => v=0;%(300,10000,440) >>>>>> >> detect-callwaiting-sas => 440 >>>>>> >> >>>>>> >> generate-callwaiting-cas => v=0;%(80,0,2750,2130) >>>>>> >> detect-callwaiting-cas => 2750,2130 >>>>>> >> >>>>>> >> detect-fail1 => 913.8 >>>>>> >> detect-fail2 => 1370.6 >>>>>> >> detect-fail3 => 776.7 >>>>>> >> >>>>>> >> LOG OF FIRST CALL (OK) >>>>>> >> ==================== >>>>>> >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:152 >>>>>> >> switch_core_standard_on_execute() OpenZAP/1:1/93400534 Execute >>>>>> >> bridge(openzap/2/1/3400534) >>>>>> >> 2009-01-25 10:35:58 [DEBUG] mod_openzap.c:340 tech_init() Set codec >>>>>> PCMU >>>>>> >> 20ms >>>>>> >> 2009-01-25 10:35:58 [DEBUG] mod_openzap.c:1017 >>>>>> channel_outgoing_channel() >>>>>> >> Connect outbound channel OpenZAP/2:1/3400534 >>>>>> >> 2009-01-25 10:35:58 [NOTICE] switch_channel.c:565 >>>>>> >> switch_channel_set_name() New Channel OpenZAP/2:1/3400534 >>>>>> >> [e5f12114-ea88-11dd-9f5c-290fb4a527a4] >>>>>> >> 2009-01-25 10:35:58 [DEBUG] mod_openzap.c:1026 >>>>>> channel_outgoing_channel() >>>>>> >> (OpenZAP/2:1/3400534) State Change CS_NEW -> CS_INIT >>>>>> >> 2009-01-25 10:35:58 [DEBUG] switch_core_session.c:807 >>>>>> >> switch_core_session_signal_state_change() Send signal >>>>>> OpenZAP/2:1/3400534 >>>>>> >> [BREAK] >>>>>> >> 2009-01-25 10:35:58 [DEBUG] ozmod_analog.c:52 >>>>>> analog_fxo_outgoing_call() >>>>>> >> Changing state on 2:1 from DOWN to DIALING >>>>>> >> 2009-01-25 10:35:58 [DEBUG] ozmod_analog.c:239 >>>>>> zap_analog_channel_run() >>>>>> >> ANALOG CHANNEL thread starting. >>>>>> >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:379 >>>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) Running State >>>>>> Change CS_INIT >>>>>> >> 2009-01-25 10:35:58 [DEBUG] ozmod_analog.c:410 >>>>>> zap_analog_channel_run() >>>>>> >> Executing state handler on 2:1 for DIALING >>>>>> >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:444 >>>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State INIT >>>>>> >> 2009-01-25 10:35:58 [DEBUG] mod_openzap.c:364 channel_on_init() >>>>>> >> (OpenZAP/2:1/3400534) State Change CS_INIT -> CS_ROUTING >>>>>> >> 2009-01-25 10:35:58 [DEBUG] switch_core_session.c:807 >>>>>> >> switch_core_session_signal_state_change() Send signal >>>>>> OpenZAP/2:1/3400534 >>>>>> >> [BREAK] >>>>>> >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:444 >>>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State INIT going to >>>>>> sleep >>>>>> >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:379 >>>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) Running State >>>>>> Change >>>>>> >> CS_ROUTING >>>>>> >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:447 >>>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State ROUTING >>>>>> >> 2009-01-25 10:35:58 [DEBUG] mod_openzap.c:387 channel_on_routing() >>>>>> >> OpenZAP/2:1/3400534 CHANNEL ROUTING >>>>>> >> 2009-01-25 10:35:58 [DEBUG] switch_ivr_originate.c:58 >>>>>> >> originate_on_routing() (OpenZAP/2:1/3400534) State Change >>>>>> CS_ROUTING -> >>>>>> >> CS_CONSUME_MEDIA >>>>>> >> 2009-01-25 10:35:58 [DEBUG] switch_core_session.c:807 >>>>>> >> switch_core_session_signal_state_change() Send signal >>>>>> OpenZAP/2:1/3400534 >>>>>> >> [BREAK] >>>>>> >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:447 >>>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State ROUTING going >>>>>> to sleep >>>>>> >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:379 >>>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) Running State >>>>>> Change >>>>>> >> CS_CONSUME_MEDIA >>>>>> >> 2009-01-25 10:35:58 [DEBUG] switch_core_state_machine.c:466 >>>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State CONSUME_MEDIA >>>>>> >> 2009-01-25 10:35:59 [DEBUG] ozmod_analog.c:615 >>>>>> zap_analog_channel_run() >>>>>> >> Detected tone DIAL on 2:1 >>>>>> >> 2009-01-25 10:35:59 [DEBUG] mod_openzap.c:1153 on_fxo_signal() got >>>>>> FXO sig >>>>>> >> 2:1 [TONE_DETECTED] >>>>>> >> 2009-01-25 10:35:59 [WARNING] mod_openzap.c:1196 on_fxo_signal() >>>>>> Unhandled >>>>>> >> type for channel 2:1 >>>>>> >> 2009-01-25 10:35:59 [DEBUG] zap_io.c:1179 >>>>>> zchan_activate_dtmf_buffer() >>>>>> >> Created DTMF Buffer! >>>>>> >> 2009-01-25 10:35:59 [DEBUG] zap_io.c:1715 handle_dtmf() 2:1 >>>>>> GENERATE DTMF >>>>>> >> [3400534] >>>>>> >> 2009-01-25 10:36:03 [DEBUG] ozmod_analog.c:302 >>>>>> zap_analog_channel_run() >>>>>> >> Changing state on 2:1 from DIALING to UP >>>>>> >> 2009-01-25 10:36:03 [DEBUG] ozmod_analog.c:410 >>>>>> zap_analog_channel_run() >>>>>> >> Executing state handler on 2:1 for UP >>>>>> >> 2009-01-25 10:36:03 [DEBUG] mod_openzap.c:1153 on_fxo_signal() got >>>>>> FXO sig >>>>>> >> 2:1 [UP] >>>>>> >> 2009-01-25 10:36:03 [DEBUG] switch_channel.c:1710 >>>>>> >> switch_channel_perform_mark_answered() Send signal >>>>>> OpenZAP/1:1/93400534 >>>>>> >> [BREAK] >>>>>> >> 2009-01-25 10:36:03 [NOTICE] mod_openzap.c:1180 on_fxo_signal() >>>>>> Channel >>>>>> >> [OpenZAP/2:1/3400534] has been answered >>>>>> >> 2009-01-25 10:36:03 [DEBUG] switch_channel.c:177 >>>>>> >> switch_channel_audio_sync() OpenZAP/2:1/3400534 receive message >>>>>> [AUDIO_SYNC] >>>>>> >> 2009-01-25 10:36:03 [DEBUG] switch_channel.c:1768 >>>>>> >> switch_channel_perform_answer() OpenZAP/1:1/93400534 receive >>>>>> message >>>>>> >> [ANSWER] >>>>>> >> 2009-01-25 10:36:03 [DEBUG] mod_openzap.c:813 >>>>>> >> channel_receive_message_fxs() Changing state on 1:1 from IDLE to UP >>>>>> >> 2009-01-25 10:36:03 [NOTICE] mod_openzap.c:814 >>>>>> >> channel_receive_message_fxs() Channel [OpenZAP/1:1/93400534] has >>>>>> been >>>>>> >> answered >>>>>> >> 2009-01-25 10:36:03 [DEBUG] switch_channel.c:177 >>>>>> >> switch_channel_audio_sync() OpenZAP/1:1/93400534 receive message >>>>>> >> [AUDIO_SYNC] >>>>>> >> 2009-01-25 10:36:03 [DEBUG] switch_core_session.c:511 >>>>>> >> switch_core_session_perform_receive_message() Send signal >>>>>> >> OpenZAP/1:1/93400534 [BREAK] >>>>>> >> 2009-01-25 10:36:03 [DEBUG] switch_ivr_originate.c:1627 >>>>>> >> switch_ivr_originate() Originate Resulted in Success: >>>>>> [OpenZAP/2:1/3400534] >>>>>> >> 2009-01-25 10:36:03 [DEBUG] switch_channel.c:177 >>>>>> >> switch_channel_audio_sync() OpenZAP/2:1/3400534 receive message >>>>>> [AUDIO_SYNC] >>>>>> >> 2009-01-25 10:36:03 [DEBUG] switch_channel.c:177 >>>>>> >> switch_channel_audio_sync() OpenZAP/1:1/93400534 receive message >>>>>> >> [AUDIO_SYNC] >>>>>> >> 2009-01-25 10:36:03 [DEBUG] switch_ivr_bridge.c:862 >>>>>> >> switch_ivr_multi_threaded_bridge() OpenZAP/2:1/3400534 receive >>>>>> message >>>>>> >> [BRIDGE] >>>>>> >> 2009-01-25 10:36:03 [DEBUG] switch_core_session.c:511 >>>>>> >> switch_core_session_perform_receive_message() Send signal >>>>>> >> OpenZAP/2:1/3400534 [BREAK] >>>>>> >> 2009-01-25 10:36:03 [DEBUG] switch_ivr_bridge.c:869 >>>>>> >> switch_ivr_multi_threaded_bridge() OpenZAP/1:1/93400534 receive >>>>>> message >>>>>> >> [BRIDGE] >>>>>> >> 2009-01-25 10:36:03 [DEBUG] switch_core_session.c:511 >>>>>> >> switch_core_session_perform_receive_message() Send signal >>>>>> >> OpenZAP/1:1/93400534 [BREAK] >>>>>> >> 2009-01-25 10:36:03 [DEBUG] switch_ivr_bridge.c:913 >>>>>> >> switch_ivr_multi_threaded_bridge() (OpenZAP/2:1/3400534) State >>>>>> Change >>>>>> >> CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA >>>>>> >> 2009-01-25 10:36:03 [DEBUG] switch_core_session.c:807 >>>>>> >> switch_core_session_signal_state_change() Send signal >>>>>> OpenZAP/2:1/3400534 >>>>>> >> [BREAK] >>>>>> >> 2009-01-25 10:36:03 [DEBUG] ozmod_analog.c:410 >>>>>> zap_analog_channel_run() >>>>>> >> Executing state handler on 1:1 for UP >>>>>> >> 2009-01-25 10:36:03 [DEBUG] mod_openzap.c:1212 on_fxs_signal() got >>>>>> FXS sig >>>>>> >> [UP] >>>>>> >> 2009-01-25 10:36:03 [DEBUG] switch_core_state_machine.c:466 >>>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State CONSUME_MEDIA >>>>>> going to >>>>>> >> sleep >>>>>> >> 2009-01-25 10:36:03 [DEBUG] switch_core_state_machine.c:379 >>>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) Running State >>>>>> Change >>>>>> >> CS_EXCHANGE_MEDIA >>>>>> >> 2009-01-25 10:36:03 [DEBUG] switch_core_state_machine.c:457 >>>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State >>>>>> EXCHANGE_MEDIA >>>>>> >> 2009-01-25 10:36:03 [DEBUG] mod_openzap.c:511 >>>>>> channel_on_exchange_media() >>>>>> >> CHANNEL EXCHANGE_MEDIA >>>>>> >> 2009-01-25 10:36:16 [DEBUG] ozmod_analog.c:744 process_event() >>>>>> EVENT >>>>>> >> [ONHOOK][1:1] STATE [UP] >>>>>> >> 2009-01-25 10:36:16 [DEBUG] ozmod_analog.c:780 process_event() >>>>>> Changing >>>>>> >> state on 1:1 from UP to DOWN >>>>>> >> 2009-01-25 10:36:16 [DEBUG] ozmod_analog.c:410 >>>>>> zap_analog_channel_run() >>>>>> >> Executing state handler on 1:1 for DOWN >>>>>> >> 2009-01-25 10:36:16 [DEBUG] mod_openzap.c:1212 on_fxs_signal() got >>>>>> FXS sig >>>>>> >> [STOP] >>>>>> >> 2009-01-25 10:36:16 [NOTICE] mod_openzap.c:1300 on_fxs_signal() >>>>>> Hangup >>>>>> >> OpenZAP/1:1/93400534 [CS_EXECUTE] [NORMAL_CLEARING] >>>>>> >> 2009-01-25 10:36:16 [DEBUG] switch_channel.c:1494 >>>>>> >> switch_channel_perform_hangup() Send signal OpenZAP/1:1/93400534 >>>>>> [KILL] >>>>>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_session.c:807 >>>>>> >> switch_core_session_signal_state_change() Send signal >>>>>> OpenZAP/1:1/93400534 >>>>>> >> [BREAK] >>>>>> >> 2009-01-25 10:36:16 [DEBUG] zap_io.c:1125 zap_channel_done() >>>>>> channel done >>>>>> >> 1:1 >>>>>> >> 2009-01-25 10:36:16 [DEBUG] ozmod_analog.c:726 >>>>>> zap_analog_channel_run() >>>>>> >> ANALOG CHANNEL 1:1 thread ended. >>>>>> >> 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:360 >>>>>> audio_bridge_thread() >>>>>> >> OpenZAP/1:1/93400534 ending bridge by request from read function >>>>>> >> 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:435 >>>>>> audio_bridge_thread() >>>>>> >> BRIDGE THREAD DONE [OpenZAP/1:1/93400534] >>>>>> >> 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:439 >>>>>> audio_bridge_thread() >>>>>> >> Send signal OpenZAP/2:1/3400534 [BREAK] >>>>>> >> 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:354 >>>>>> audio_bridge_thread() >>>>>> >> OpenZAP/1:1/93400534 ending bridge by request from write function >>>>>> >> 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:409 >>>>>> audio_bridge_thread() >>>>>> >> OpenZAP/2:1/3400534 receive message [UNBRIDGE] >>>>>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_session.c:511 >>>>>> >> switch_core_session_perform_receive_message() Send signal >>>>>> >> OpenZAP/2:1/3400534 [BREAK] >>>>>> >> 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:435 >>>>>> audio_bridge_thread() >>>>>> >> BRIDGE THREAD DONE [OpenZAP/2:1/3400534] >>>>>> >> 2009-01-25 10:36:16 [DEBUG] switch_ivr_bridge.c:439 >>>>>> audio_bridge_thread() >>>>>> >> Send signal OpenZAP/1:1/93400534 [BREAK] >>>>>> >> 2009-01-25 10:36:16 [NOTICE] switch_ivr_bridge.c:470 >>>>>> >> audio_bridge_on_exchange_media() Hangup OpenZAP/2:1/3400534 >>>>>> >> [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] >>>>>> >> 2009-01-25 10:36:16 [DEBUG] switch_channel.c:1494 >>>>>> >> switch_channel_perform_hangup() Send signal OpenZAP/2:1/3400534 >>>>>> [KILL] >>>>>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_session.c:807 >>>>>> >> switch_core_session_signal_state_change() Send signal >>>>>> OpenZAP/2:1/3400534 >>>>>> >> [BREAK] >>>>>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:457 >>>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State >>>>>> EXCHANGE_MEDIA going >>>>>> >> to sleep >>>>>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:379 >>>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) Running State >>>>>> Change >>>>>> >> CS_HANGUP >>>>>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:410 >>>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State HANGUP >>>>>> >> 2009-01-25 10:36:16 [DEBUG] mod_openzap.c:429 channel_on_hangup() >>>>>> Changing >>>>>> >> state on 2:1 from UP to HANGUP >>>>>> >> 2009-01-25 10:36:16 [DEBUG] mod_openzap.c:472 channel_on_hangup() >>>>>> >> OpenZAP/2:1/3400534 CHANNEL HANGUP >>>>>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:46 >>>>>> >> switch_core_standard_on_hangup() OpenZAP/2:1/3400534 Standard >>>>>> HANGUP, cause: >>>>>> >> NORMAL_CLEARING >>>>>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:410 >>>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State HANGUP going >>>>>> to sleep >>>>>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_session.c:939 >>>>>> >> switch_core_session_thread() Session 2 (OpenZAP/2:1/3400534) >>>>>> Locked, Waiting >>>>>> >> on external entities >>>>>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:454 >>>>>> >> switch_core_session_run() (OpenZAP/1:1/93400534) State EXECUTE >>>>>> going to >>>>>> >> sleep >>>>>> >> 2009-01-25 10:36:16 [NOTICE] switch_core_session.c:957 >>>>>> >> switch_core_session_thread() Session 2 (OpenZAP/2:1/3400534) Ended >>>>>> >> 2009-01-25 10:36:16 [NOTICE] switch_core_session.c:959 >>>>>> >> switch_core_session_thread() Close Channel OpenZAP/2:1/3400534 >>>>>> [CS_HANGUP] >>>>>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:379 >>>>>> >> switch_core_session_run() (OpenZAP/1:1/93400534) Running State >>>>>> Change >>>>>> >> CS_HANGUP >>>>>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:410 >>>>>> >> switch_core_session_run() (OpenZAP/1:1/93400534) State HANGUP >>>>>> >> 2009-01-25 10:36:16 [DEBUG] mod_openzap.c:472 channel_on_hangup() >>>>>> >> OpenZAP/1:1/93400534 CHANNEL HANGUP >>>>>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:46 >>>>>> >> switch_core_standard_on_hangup() OpenZAP/1:1/93400534 Standard >>>>>> HANGUP, >>>>>> >> cause: NORMAL_CLEARING >>>>>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_state_machine.c:410 >>>>>> >> switch_core_session_run() (OpenZAP/1:1/93400534) State HANGUP going >>>>>> to sleep >>>>>> >> 2009-01-25 10:36:16 [DEBUG] switch_core_session.c:939 >>>>>> >> switch_core_session_thread() Session 1 (OpenZAP/1:1/93400534) >>>>>> Locked, >>>>>> >> Waiting on external entities >>>>>> >> 2009-01-25 10:36:16 [NOTICE] switch_core_session.c:957 >>>>>> >> switch_core_session_thread() Session 1 (OpenZAP/1:1/93400534) Ended >>>>>> >> 2009-01-25 10:36:16 [NOTICE] switch_core_session.c:959 >>>>>> >> switch_core_session_thread() Close Channel OpenZAP/1:1/93400534 >>>>>> [CS_HANGUP] >>>>>> >> 2009-01-25 10:36:16 [DEBUG] ozmod_analog.c:410 >>>>>> zap_analog_channel_run() >>>>>> >> Executing state handler on 2:1 for HANGUP >>>>>> >> 2009-01-25 10:36:17 [DEBUG] ozmod_analog.c:351 >>>>>> zap_analog_channel_run() >>>>>> >> Changing state on 2:1 from HANGUP to DOWN >>>>>> >> 2009-01-25 10:36:17 [DEBUG] ozmod_analog.c:410 >>>>>> zap_analog_channel_run() >>>>>> >> Executing state handler on 2:1 for DOWN >>>>>> >> 2009-01-25 10:36:17 [DEBUG] mod_openzap.c:1153 on_fxo_signal() got >>>>>> FXO sig >>>>>> >> 2:1 [STOP] >>>>>> >> 2009-01-25 10:36:17 [DEBUG] zap_io.c:1125 zap_channel_done() >>>>>> channel done >>>>>> >> 2:1 >>>>>> >> 2009-01-25 10:36:17 [DEBUG] ozmod_analog.c:726 >>>>>> zap_analog_channel_run() >>>>>> >> ANALOG CHANNEL 2:1 thread ended. >>>>>> >> >>>>>> >> LOG OF FAILED CALLS >>>>>> >> ================== >>>>>> >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:152 >>>>>> >> switch_core_standard_on_execute() OpenZAP/1:1/93400534 Execute >>>>>> >> bridge(openzap/2/1/3400534) >>>>>> >> 2009-01-25 10:36:55 [DEBUG] mod_openzap.c:340 tech_init() Set codec >>>>>> PCMU >>>>>> >> 20ms >>>>>> >> 2009-01-25 10:36:55 [DEBUG] mod_openzap.c:1017 >>>>>> channel_outgoing_channel() >>>>>> >> Connect outbound channel OpenZAP/2:1/3400534 >>>>>> >> 2009-01-25 10:36:55 [NOTICE] switch_channel.c:565 >>>>>> >> switch_channel_set_name() New Channel OpenZAP/2:1/3400534 >>>>>> >> [079f5420-ea89-11dd-9f5c-290fb4a527a4] >>>>>> >> 2009-01-25 10:36:55 [DEBUG] mod_openzap.c:1026 >>>>>> channel_outgoing_channel() >>>>>> >> (OpenZAP/2:1/3400534) State Change CS_NEW -> CS_INIT >>>>>> >> 2009-01-25 10:36:55 [DEBUG] switch_core_session.c:807 >>>>>> >> switch_core_session_signal_state_change() Send signal >>>>>> OpenZAP/2:1/3400534 >>>>>> >> [BREAK] >>>>>> >> 2009-01-25 10:36:55 [DEBUG] ozmod_analog.c:52 >>>>>> analog_fxo_outgoing_call() >>>>>> >> Changing state on 2:1 from DOWN to DIALING >>>>>> >> 2009-01-25 10:36:55 [DEBUG] ozmod_analog.c:239 >>>>>> zap_analog_channel_run() >>>>>> >> ANALOG CHANNEL thread starting. >>>>>> >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:379 >>>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) Running State >>>>>> Change CS_INIT >>>>>> >> 2009-01-25 10:36:55 [DEBUG] ozmod_analog.c:410 >>>>>> zap_analog_channel_run() >>>>>> >> Executing state handler on 2:1 for DIALING >>>>>> >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:444 >>>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State INIT >>>>>> >> 2009-01-25 10:36:55 [DEBUG] mod_openzap.c:364 channel_on_init() >>>>>> >> (OpenZAP/2:1/3400534) State Change CS_INIT -> CS_ROUTING >>>>>> >> 2009-01-25 10:36:55 [DEBUG] switch_core_session.c:807 >>>>>> >> switch_core_session_signal_state_change() Send signal >>>>>> OpenZAP/2:1/3400534 >>>>>> >> [BREAK] >>>>>> >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:444 >>>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State INIT going to >>>>>> sleep >>>>>> >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:379 >>>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) Running State >>>>>> Change >>>>>> >> CS_ROUTING >>>>>> >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:447 >>>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State ROUTING >>>>>> >> 2009-01-25 10:36:55 [DEBUG] mod_openzap.c:387 channel_on_routing() >>>>>> >> OpenZAP/2:1/3400534 CHANNEL ROUTING >>>>>> >> 2009-01-25 10:36:55 [DEBUG] switch_ivr_originate.c:58 >>>>>> >> originate_on_routing() (OpenZAP/2:1/3400534) State Change >>>>>> CS_ROUTING -> >>>>>> >> CS_CONSUME_MEDIA >>>>>> >> 2009-01-25 10:36:55 [DEBUG] switch_core_session.c:807 >>>>>> >> switch_core_session_signal_state_change() Send signal >>>>>> OpenZAP/2:1/3400534 >>>>>> >> [BREAK] >>>>>> >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:447 >>>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State ROUTING going >>>>>> to sleep >>>>>> >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:379 >>>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) Running State >>>>>> Change >>>>>> >> CS_CONSUME_MEDIA >>>>>> >> 2009-01-25 10:36:55 [DEBUG] switch_core_state_machine.c:466 >>>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State CONSUME_MEDIA >>>>>> >> 2009-01-25 10:37:08 [DEBUG] ozmod_analog.c:744 process_event() >>>>>> EVENT >>>>>> >> [ONHOOK][1:1] STATE [IDLE] >>>>>> >> 2009-01-25 10:37:08 [DEBUG] ozmod_analog.c:780 process_event() >>>>>> Changing >>>>>> >> state on 1:1 from IDLE to DOWN >>>>>> >> 2009-01-25 10:37:08 [DEBUG] ozmod_analog.c:410 >>>>>> zap_analog_channel_run() >>>>>> >> Executing state handler on 1:1 for DOWN >>>>>> >> 2009-01-25 10:37:08 [DEBUG] mod_openzap.c:1212 on_fxs_signal() got >>>>>> FXS sig >>>>>> >> [STOP] >>>>>> >> 2009-01-25 10:37:08 [NOTICE] mod_openzap.c:1300 on_fxs_signal() >>>>>> Hangup >>>>>> >> OpenZAP/1:1/93400534 [CS_EXECUTE] [NORMAL_CLEARING] >>>>>> >> 2009-01-25 10:37:08 [DEBUG] switch_channel.c:1494 >>>>>> >> switch_channel_perform_hangup() Send signal OpenZAP/1:1/93400534 >>>>>> [KILL] >>>>>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_session.c:807 >>>>>> >> switch_core_session_signal_state_change() Send signal >>>>>> OpenZAP/1:1/93400534 >>>>>> >> [BREAK] >>>>>> >> 2009-01-25 10:37:08 [DEBUG] zap_io.c:1125 zap_channel_done() >>>>>> channel done >>>>>> >> 1:1 >>>>>> >> 2009-01-25 10:37:08 [DEBUG] ozmod_analog.c:726 >>>>>> zap_analog_channel_run() >>>>>> >> ANALOG CHANNEL 1:1 thread ended. >>>>>> >> 2009-01-25 10:37:08 [NOTICE] switch_ivr_originate.c:1566 >>>>>> >> switch_ivr_originate() Hangup OpenZAP/2:1/3400534 >>>>>> [CS_CONSUME_MEDIA] >>>>>> >> [ORIGINATOR_CANCEL] >>>>>> >> 2009-01-25 10:37:08 [DEBUG] switch_channel.c:1494 >>>>>> >> switch_channel_perform_hangup() Send signal OpenZAP/2:1/3400534 >>>>>> [KILL] >>>>>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_session.c:807 >>>>>> >> switch_core_session_signal_state_change() Send signal >>>>>> OpenZAP/2:1/3400534 >>>>>> >> [BREAK] >>>>>> >> 2009-01-25 10:37:08 [DEBUG] switch_ivr_originate.c:1691 >>>>>> >> switch_ivr_originate() Originate Cancelled by originator >>>>>> termination Cause: >>>>>> >> 487 [ORIGINATOR_CANCEL] >>>>>> >> 2009-01-25 10:37:08 [INFO] mod_dptools.c:1909 >>>>>> audio_bridge_function() >>>>>> >> Originate Failed. Cause: ORIGINATOR_CANCEL >>>>>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:454 >>>>>> >> switch_core_session_run() (OpenZAP/1:1/93400534) State EXECUTE >>>>>> going to >>>>>> >> sleep >>>>>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:379 >>>>>> >> switch_core_session_run() (OpenZAP/1:1/93400534) Running State >>>>>> Change >>>>>> >> CS_HANGUP >>>>>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:410 >>>>>> >> switch_core_session_run() (OpenZAP/1:1/93400534) State HANGUP >>>>>> >> 2009-01-25 10:37:08 [DEBUG] mod_openzap.c:472 channel_on_hangup() >>>>>> >> OpenZAP/1:1/93400534 CHANNEL HANGUP >>>>>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:46 >>>>>> >> switch_core_standard_on_hangup() OpenZAP/1:1/93400534 Standard >>>>>> HANGUP, >>>>>> >> cause: NORMAL_CLEARING >>>>>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:410 >>>>>> >> switch_core_session_run() (OpenZAP/1:1/93400534) State HANGUP going >>>>>> to sleep >>>>>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_session.c:939 >>>>>> >> switch_core_session_thread() Session 3 (OpenZAP/1:1/93400534) >>>>>> Locked, >>>>>> >> Waiting on external entities >>>>>> >> 2009-01-25 10:37:08 [NOTICE] switch_core_session.c:957 >>>>>> >> switch_core_session_thread() Session 3 (OpenZAP/1:1/93400534) Ended >>>>>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:466 >>>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State CONSUME_MEDIA >>>>>> going to >>>>>> >> sleep >>>>>> >> 2009-01-25 10:37:08 [NOTICE] switch_core_session.c:959 >>>>>> >> switch_core_session_thread() Close Channel OpenZAP/1:1/93400534 >>>>>> [CS_HANGUP] >>>>>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:379 >>>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) Running State >>>>>> Change >>>>>> >> CS_HANGUP >>>>>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:410 >>>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State HANGUP >>>>>> >> 2009-01-25 10:37:08 [DEBUG] mod_openzap.c:429 channel_on_hangup() >>>>>> Changing >>>>>> >> state on 2:1 from DIALING to HANGUP >>>>>> >> 2009-01-25 10:37:08 [DEBUG] mod_openzap.c:472 channel_on_hangup() >>>>>> >> OpenZAP/2:1/3400534 CHANNEL HANGUP >>>>>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:46 >>>>>> >> switch_core_standard_on_hangup() OpenZAP/2:1/3400534 Standard >>>>>> HANGUP, cause: >>>>>> >> ORIGINATOR_CANCEL >>>>>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_state_machine.c:410 >>>>>> >> switch_core_session_run() (OpenZAP/2:1/3400534) State HANGUP going >>>>>> to sleep >>>>>> >> 2009-01-25 10:37:08 [DEBUG] switch_core_session.c:939 >>>>>> >> switch_core_session_thread() Session 4 (OpenZAP/2:1/3400534) >>>>>> Locked, Waiting >>>>>> >> on external entities >>>>>> >> 2009-01-25 10:37:08 [NOTICE] switch_core_session.c:957 >>>>>> >> switch_core_session_thread() Session 4 (OpenZAP/2:1/3400534) Ended >>>>>> >> 2009-01-25 10:37:08 [NOTICE] switch_core_session.c:959 >>>>>> >> switch_core_session_thread() Close Channel OpenZAP/2:1/3400534 >>>>>> [CS_HANGUP] >>>>>> >> 2009-01-25 10:37:08 [DEBUG] ozmod_analog.c:410 >>>>>> zap_analog_channel_run() >>>>>> >> Executing state handler on 2:1 for HANGUP >>>>>> >> 2009-01-25 10:37:09 [DEBUG] ozmod_analog.c:351 >>>>>> zap_analog_channel_run() >>>>>> >> Changing state on 2:1 from HANGUP to DOWN >>>>>> >> 2009-01-25 10:37:09 [DEBUG] ozmod_analog.c:410 >>>>>> zap_analog_channel_run() >>>>>> >> Executing state handler on 2:1 for DOWN >>>>>> >> 2009-01-25 10:37:09 [DEBUG] mod_openzap.c:1153 on_fxo_signal() got >>>>>> FXO sig >>>>>> >> 2:1 [STOP] >>>>>> >> 2009-01-25 10:37:09 [DEBUG] zap_io.c:1125 zap_channel_done() >>>>>> channel done >>>>>> >> 2:1 >>>>>> >> 2009-01-25 10:37:09 [DEBUG] ozmod_analog.c:726 >>>>>> zap_analog_channel_run() >>>>>> >> ANALOG CHANNEL 2:1 thread ended. >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> >> _______________________________________________ >>>>>> >> Freeswitch-users mailing list >>>>>> >> Freeswitch-users at lists.freeswitch.org >>>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> >> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> >> http://www.freeswitch.org >>>>>> >> >>>>>> >> >>>>>> >> _______________________________________________ >>>>>> >> Freeswitch-users mailing list >>>>>> >> Freeswitch-users at lists.freeswitch.org >>>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> >> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> >> http://www.freeswitch.org >>>>>> >> >>>>>> > >>>>>> > >>>>>> > _______________________________________________ >>>>>> > Freeswitch-users mailing list >>>>>> > Freeswitch-users at lists.freeswitch.org >>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> > UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> > http://www.freeswitch.org >>>>>> > >>>>>> > >>>>>> >>>>>> _______________________________________________ >>>>>> Freeswitch-users mailing list >>>>>> Freeswitch-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>> >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> -- >>>> Anthony Minessale II >>>> >>>> FreeSWITCH http://www.freeswitch.org/ >>>> ClueCon http://www.cluecon.com/ >>>> >>>> AIM: anthm >>>> MSN:anthony_minessale at hotmail.com >>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>>> IRC: irc.freenode.net #freeswitch >>>> >>>> FreeSWITCH Developer Conference >>>> sip:888 at conference.freeswitch.org >>>> iax:guest at conference.freeswitch.org/888 >>>> googletalk:conf+888 at conference.freeswitch.org >>>> pstn:213-799-1400 >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090201/e94b380d/attachment-0002.html From brian at freeswitch.org Sun Feb 1 10:59:04 2009 From: brian at freeswitch.org (Brian West) Date: Sun, 1 Feb 2009 12:59:04 -0600 Subject: [Freeswitch-users] Cisco 7975G and XML options for G722 In-Reply-To: <49860C2B.7020801@ieee.org> References: <0C8E2714-6D96-49DB-93CD-0D377E4DAE96@freeswitch.org> <3885f4fe0901312115k1309573lb160ecbc83f8d57@mail.gmail.com> <897947BF-9886-47BB-B8BB-8DE8849437F0@freeswitch.org> <49860C2B.7020801@ieee.org> Message-ID: <1D9807B8-61ED-41B8-9AC3-C6C6BD306BE5@freeswitch.org> Why yes it does ;) /b On Feb 1, 2009, at 2:55 PM, Gabriel Kuri wrote: > 722 From pmhshz at gmail.com Sun Feb 1 21:26:56 2009 From: pmhshz at gmail.com (shehzad p) Date: Sun, 1 Feb 2009 21:26:56 -0800 (PST) Subject: [Freeswitch-users] Freeswitch freezes on increasing call traffic In-Reply-To: <30BF0913-A90C-4472-9A70-6CD6AEE2AD08@freeswitch.org> References: <21701744.post@talk.nabble.com> <191c3a030901280601t3e185ec2m76295110e4888837@mail.gmail.com> <21729863.post@talk.nabble.com> <191c3a030901290831y2ff329b3maee6c1aafa3c68e2@mail.gmail.com> <21745312.post@talk.nabble.com> <21749375.post@talk.nabble.com> <191c3a030901300818i5d85a7aawa0c53b4e54892f40@mail.gmail.com> <21761523.post@talk.nabble.com> <191c3a030901310850s553f8c11oc4e559a710def6d2@mail.gmail.com> <21773300.post@talk.nabble.com> <21775256.post@talk.nabble.com> <30BF0913-A90C-4472-9A70-6CD6AEE2AD08@freeswitch.org> Message-ID: <21784344.post@talk.nabble.com> Hi Brian, I am first looking to modify the script, as there are several things to modify (thanks to Anthony) then paralally I will test 1.0.3 RC1 also and any new thing come out then I will post update here... Thanks, msp Brian West-3 wrote: > > I highly recommend you upgrade to 1.03RC1 or SVN Trunk. Chances are > we have already fixed these issues. > > /b > > On Feb 1, 2009, at 7:35 AM, shehzad p wrote: > >> >> Hi anthony, >> >> There are so many crash occured, on FS 1.0.1 Server, all of them >> were same >> (means BT was same) >> but this last one was looking totally different so I feel that it >> should be >> posted... >> http://www.nabble.com/file/p21775256/bt_full_new.txt bt_full_new.txt > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/Freeswitch-freezes-on-increasing-call-traffic-tp21701744p21784344.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From brian at freeswitch.org Sun Feb 1 21:37:59 2009 From: brian at freeswitch.org (Brian West) Date: Sun, 1 Feb 2009 23:37:59 -0600 Subject: [Freeswitch-users] Freeswitch freezes on increasing call traffic In-Reply-To: <21784344.post@talk.nabble.com> References: <21701744.post@talk.nabble.com> <191c3a030901280601t3e185ec2m76295110e4888837@mail.gmail.com> <21729863.post@talk.nabble.com> <191c3a030901290831y2ff329b3maee6c1aafa3c68e2@mail.gmail.com> <21745312.post@talk.nabble.com> <21749375.post@talk.nabble.com> <191c3a030901300818i5d85a7aawa0c53b4e54892f40@mail.gmail.com> <21761523.post@talk.nabble.com> <191c3a030901310850s553f8c11oc4e559a710def6d2@mail.gmail.com> <21773300.post@talk.nabble.com> <21775256.post@talk.nabble.com> <30BF0913-A90C-4472-9A70-6CD6AEE2AD08@freeswitch.org> <21784344.post@talk.nabble.com> Message-ID: <54089955-A273-48CD-8657-D2A1CD4E0B89@freeswitch.org> The RC1 tarball has all the SVN dirs so you can "make current" on it. /b On Feb 1, 2009, at 11:26 PM, shehzad p wrote: > > Hi Brian, > > I am first looking to modify the script, as there are several things > to > modify (thanks to Anthony) then paralally I will test 1.0.3 RC1 also > and any > new thing come out then I will post update here... > > Thanks, > msp From kawarod at laposte.net Sun Feb 1 23:04:17 2009 From: kawarod at laposte.net (rod) Date: Mon, 02 Feb 2009 11:04:17 +0400 Subject: [Freeswitch-users] Strange Performance when using as SBC In-Reply-To: <191c3a030901300556s534823a1if4b79b4279962ed1@mail.gmail.com> References: <4982EAD5.1010200@laposte.net> <191c3a030901300556s534823a1if4b79b4279962ed1@mail.gmail.com> Message-ID: <49869AF1.1070903@laposte.net> Thanks Anthony, the setup is like this: sipp server ---- FS 1 ---- FS2 FS1 is the AMD CPU that has only one extension in dialplan that bridges 9999 to FS2. 9999 is the first extension in FS2 dialplan that plays moh, FS2 has no CPU pbm. FS1 is maxing out at 60 bridged calls without your option -hp. Using -hp, I'm now able to bridge 200 concurrent calls (a great improvement) and the system is still reactive. CPU load is high but not 100% and as the system responds well, I think that doesn't matter. The 2GB of memory are completely consumed (top command shows 700MB for FS process). I understand that FS1 server is not the best hardware platform, and I'm waiting for new 4 cores server for testing. I will update those numbers when testing with the new hardware. regards, rod. Anthony Minessale wrote: > Which of the 2 machines has the load issue? You said it was one box > calling the other. > > You have 2 major things against you, single CPU and AMD, but you > should at least be able to get in the vicinity of 800-1000 calls on a > box like that. > > Are you calling the default 9999? It's not really an appropriate > extension for load testing. > On the terminating box you should set up a manual extension that is > the first one in the dial plan > to play a wav file from preferably a ram disk or /tmp > > If you do plan on using this in production accept nothing less than a > multi-core intel machine with at least 4 cores, the more cores the > better because that parallel processing is where FS gets it's atvantage. > > > > On Fri, Jan 30, 2009 at 5:56 AM, rod > wrote: > > Dear list, > > I've been playing with freeswitch for some time (2 months) and the > fact > is that I'm very pleased with the functionnalities of this software. > > I'd like to use FS as a SBC handling media and I'm doing some > tests with > sipp to load the machine but I'm unable to bridge more than 60 calls > without seeing the CPU being loaded at 100%. I'm sure something is > going > wrong with my setup but I'm unable to see what. > > The test machine has the following specs: > Athlon XP 3500+ with 2GB of memory (I know this is not a high end > machine :p) > > Freeswitch:/opt/freeswitch/log# cat /proc/cpuinfo > processor : 0 > vendor_id : AuthenticAMD > cpu family : 15 > model : 95 > model name : AMD Athlon(tm) 64 Processor 3500+ > stepping : 2 > cpu MHz : 2199.973 > cache size : 512 KB > fpu : yes > fpu_exception : yes > cpuid level : 1 > wp : yes > flags : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge > mca cmov pat pse36 clflush mmx fxsr sse sse2 syscall nx mmxext > fxsr_opt > rdtscp lm 3dnowext 3dnow up rep_good pni cx16 lahf_lm svm extapic > cr8_legacy > bogomips : 4402.97 > TLB size : 1024 4K pages > clflush size : 64 > cache_alignment : 64 > address sizes : 40 bits physical, 48 bits virtual > power management: ts fid vid ttp tm stc > > I installed FS on a fresh debian 64: > Linux Freeswitch 2.6.26-1-amd64 #1 SMP Sat Jan 10 17:57:00 UTC 2009 > x86_64 GNU/Linux > > I set the ulimit parameters like those on the website: > freeswitch at internal> ... > Freeswitch:/opt/free-svn/bin# ulimit -a > core file size (blocks, -c) unlimited > data seg size (kbytes, -d) unlimited > scheduling priority (-e) 0 > file size (blocks, -f) unlimited > pending signals (-i) unlimited > max locked memory (kbytes, -l) unlimited > max memory size (kbytes, -m) unlimited > open files (-n) 999999 > pipe size (512 bytes, -p) 8 > POSIX message queues (bytes, -q) unlimited > real-time priority (-r) 0 > stack size (kbytes, -s) 244 > cpu time (seconds, -t) unlimited > max user processes (-u) unlimited > virtual memory (kbytes, -v) unlimited > file locks (-x) unlimited > > > My network setup is the following: > > SIPP machine (10.10.10.1/24)----------------vlan > 55 > ----------(10.10.10.254/24 ) FS > (10.10.20.254/24)-------------- > vlan56 > -------------------(10.10.20.100/24 ) > OTHER STOCK FS > > > I launched sipp with: > sipp -sn uac_pcap -s 9999 -r 10 -l 80 -d 60000 -mi 10.10.10.1 -i > 10.10.10.1 -mp 25000 10.10.10.254:5060 > > The dialplan on FS is very simple: > > > > > > > > > data="sofia/external/9999 at 10.10.20.100 "/> > > > > > > > FreeSWITCH Version 1.0.trunk (11560M) Started. > Crash Protection [Disabled] > Max Sessions[1000] > Session Rate[100] > SQL [Enabled] > > > The test is very simple: sipp dial 9999 that matches in my FS dialplan > and this is bridged to an other FS machine playing music on hold. > When I launch "top" I see after 30 to 40 s that FS consumes all > the CPU > ressources (with a mean of 50-60 % before), with 80 calls. > When I set 70 calls, I have to wait 70-80 s before seeing the same > issue. > > Presence is set to false on the 2 profile. > > I have the same issue with FS 1.0.2 that' s why I tried FS 11560. > > When I use the FS machine as a router to test the packet per second > performance, I'm reaching 100Mbps with 8000pps in each direction (from > vlan 55 to vlan56) with less than 12% CPU. So that I don't think > there's > an issue with the network. > > Here is an "mpstat -P ALL 1" to show you what's happening suddenly > with > 70 bridge calls: > 12:31:26 CPU %user %nice %sys %iowait %irq %soft > %steal %idle intr/s > 12:31:27 all 3,00 0,00 3,00 0,00 1,00 4,00 > 0,00 89,00 6241,00 > 12:31:27 0 3,00 0,00 3,00 0,00 1,00 4,00 > 0,00 89,00 6241,00 > > 12:31:27 CPU %user %nice %sys %iowait %irq %soft > %steal %idle intr/s > 12:31:28 all 14,14 0,00 56,57 0,00 2,02 5,05 > 0,00 22,22 6035,35 > 12:31:28 0 14,14 0,00 56,57 0,00 2,02 5,05 > 0,00 22,22 6035,35 > > 12:31:28 CPU %user %nice %sys %iowait %irq %soft > %steal %idle intr/s > 12:31:29 all 24,75 0,00 67,33 0,00 0,99 6,93 > 0,00 0,00 5483,17 > 12:31:29 0 24,75 0,00 67,33 0,00 0,99 6,93 > 0,00 0,00 5483,17 > > > The CPU is going from 89% idle to 0% in less than 2 seconds. > > I know that I don't have to expect too much from this kind of > hardware, > but it seems strange that the CPU power vanished so suddenly. > > Thanks a lot for the guys that have read this long mail :p > > kind regards, > rod > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From krice at freeswitch.org Sun Feb 1 23:17:50 2009 From: krice at freeswitch.org (Ken Rice) Date: Mon, 2 Feb 2009 01:17:50 -0600 Subject: [Freeswitch-users] Strange Performance when using as SBC In-Reply-To: <49869AF1.1070903@laposte.net> References: <4982EAD5.1010200@laposte.net> <191c3a030901300556s534823a1if4b79b4279962ed1@mail.gmail.com> <49869AF1.1070903@laposte.net> Message-ID: <94b568920902012317h8411fccg216435380f62d649@mail.gmail.com> Dont forget there are several things you can do to increase performance... 1) where possible use bypass media or media proxy modes 2) mount freeswitch/db as a ram drive (if you are using voicemail with the internal FS DBs you'll need a way to make this persistant across reboots) 3) see the wiki for setting reasonable ulimits 4) (this is my oppinion others may vary) dont use mod_cdr_csv 5) turn off (or reduce logging) in switch.conf.xml all of these thing can greatly improve performance. On Mon, Feb 2, 2009 at 1:04 AM, rod wrote: > Thanks Anthony, > > the setup is like this: > > sipp server ---- FS 1 ---- FS2 > > FS1 is the AMD CPU that has only one extension in dialplan that bridges > 9999 to FS2. 9999 is the first extension in FS2 dialplan that plays moh, > FS2 has no CPU pbm. > > FS1 is maxing out at 60 bridged calls without your option -hp. > > Using -hp, I'm now able to bridge 200 concurrent calls (a great > improvement) and the system is still reactive. CPU load is high but not > 100% and as the system responds well, I think that doesn't matter. The > 2GB of memory are completely consumed (top command shows 700MB for FS > process). > > I understand that FS1 server is not the best hardware platform, and I'm > waiting for new 4 cores server for testing. > I will update those numbers when testing with the new hardware. > > regards, > rod. > > Anthony Minessale wrote: > > Which of the 2 machines has the load issue? You said it was one box > > calling the other. > > > > You have 2 major things against you, single CPU and AMD, but you > > should at least be able to get in the vicinity of 800-1000 calls on a > > box like that. > > > > Are you calling the default 9999? It's not really an appropriate > > extension for load testing. > > On the terminating box you should set up a manual extension that is > > the first one in the dial plan > > to play a wav file from preferably a ram disk or /tmp > > > > If you do plan on using this in production accept nothing less than a > > multi-core intel machine with at least 4 cores, the more cores the > > better because that parallel processing is where FS gets it's atvantage. > > > > > > > > On Fri, Jan 30, 2009 at 5:56 AM, rod > > wrote: > > > > Dear list, > > > > I've been playing with freeswitch for some time (2 months) and the > > fact > > is that I'm very pleased with the functionnalities of this software. > > > > I'd like to use FS as a SBC handling media and I'm doing some > > tests with > > sipp to load the machine but I'm unable to bridge more than 60 calls > > without seeing the CPU being loaded at 100%. I'm sure something is > > going > > wrong with my setup but I'm unable to see what. > > > > The test machine has the following specs: > > Athlon XP 3500+ with 2GB of memory (I know this is not a high end > > machine :p) > > > > Freeswitch:/opt/freeswitch/log# cat /proc/cpuinfo > > processor : 0 > > vendor_id : AuthenticAMD > > cpu family : 15 > > model : 95 > > model name : AMD Athlon(tm) 64 Processor 3500+ > > stepping : 2 > > cpu MHz : 2199.973 > > cache size : 512 KB > > fpu : yes > > fpu_exception : yes > > cpuid level : 1 > > wp : yes > > flags : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr > pge > > mca cmov pat pse36 clflush mmx fxsr sse sse2 syscall nx mmxext > > fxsr_opt > > rdtscp lm 3dnowext 3dnow up rep_good pni cx16 lahf_lm svm extapic > > cr8_legacy > > bogomips : 4402.97 > > TLB size : 1024 4K pages > > clflush size : 64 > > cache_alignment : 64 > > address sizes : 40 bits physical, 48 bits virtual > > power management: ts fid vid ttp tm stc > > > > I installed FS on a fresh debian 64: > > Linux Freeswitch 2.6.26-1-amd64 #1 SMP Sat Jan 10 17:57:00 UTC 2009 > > x86_64 GNU/Linux > > > > I set the ulimit parameters like those on the website: > > freeswitch at internal> ... > > Freeswitch:/opt/free-svn/bin# ulimit -a > > core file size (blocks, -c) unlimited > > data seg size (kbytes, -d) unlimited > > scheduling priority (-e) 0 > > file size (blocks, -f) unlimited > > pending signals (-i) unlimited > > max locked memory (kbytes, -l) unlimited > > max memory size (kbytes, -m) unlimited > > open files (-n) 999999 > > pipe size (512 bytes, -p) 8 > > POSIX message queues (bytes, -q) unlimited > > real-time priority (-r) 0 > > stack size (kbytes, -s) 244 > > cpu time (seconds, -t) unlimited > > max user processes (-u) unlimited > > virtual memory (kbytes, -v) unlimited > > file locks (-x) unlimited > > > > > > My network setup is the following: > > > > SIPP machine (10.10.10.1/24)----------------vlan > > 55 > > ----------(10.10.10.254/24 ) FS > > (10.10.20.254/24)-------------- > > vlan56 > > -------------------(10.10.20.100/24 ) > > OTHER STOCK FS > > > > > > I launched sipp with: > > sipp -sn uac_pcap -s 9999 -r 10 -l 80 -d 60000 -mi 10.10.10.1 -i > > 10.10.10.1 -mp 25000 10.10.10.254:5060 > > > > The dialplan on FS is very simple: > > > > > > > > > > > > > > > > > > > data="sofia/external/9999 at 10.10.20.100 "/> > > > > > > > > > > > > > > FreeSWITCH Version 1.0.trunk (11560M) Started. > > Crash Protection [Disabled] > > Max Sessions[1000] > > Session Rate[100] > > SQL [Enabled] > > > > > > The test is very simple: sipp dial 9999 that matches in my FS > dialplan > > and this is bridged to an other FS machine playing music on hold. > > When I launch "top" I see after 30 to 40 s that FS consumes all > > the CPU > > ressources (with a mean of 50-60 % before), with 80 calls. > > When I set 70 calls, I have to wait 70-80 s before seeing the same > > issue. > > > > Presence is set to false on the 2 profile. > > > > I have the same issue with FS 1.0.2 that' s why I tried FS 11560. > > > > When I use the FS machine as a router to test the packet per second > > performance, I'm reaching 100Mbps with 8000pps in each direction > (from > > vlan 55 to vlan56) with less than 12% CPU. So that I don't think > > there's > > an issue with the network. > > > > Here is an "mpstat -P ALL 1" to show you what's happening suddenly > > with > > 70 bridge calls: > > 12:31:26 CPU %user %nice %sys %iowait %irq %soft > > %steal %idle intr/s > > 12:31:27 all 3,00 0,00 3,00 0,00 1,00 4,00 > > 0,00 89,00 6241,00 > > 12:31:27 0 3,00 0,00 3,00 0,00 1,00 4,00 > > 0,00 89,00 6241,00 > > > > 12:31:27 CPU %user %nice %sys %iowait %irq %soft > > %steal %idle intr/s > > 12:31:28 all 14,14 0,00 56,57 0,00 2,02 5,05 > > 0,00 22,22 6035,35 > > 12:31:28 0 14,14 0,00 56,57 0,00 2,02 5,05 > > 0,00 22,22 6035,35 > > > > 12:31:28 CPU %user %nice %sys %iowait %irq %soft > > %steal %idle intr/s > > 12:31:29 all 24,75 0,00 67,33 0,00 0,99 6,93 > > 0,00 0,00 5483,17 > > 12:31:29 0 24,75 0,00 67,33 0,00 0,99 6,93 > > 0,00 0,00 5483,17 > > > > > > The CPU is going from 89% idle to 0% in less than 2 seconds. > > > > I know that I don't have to expect too much from this kind of > > hardware, > > but it seems strange that the CPU power vanished so suddenly. > > > > Thanks a lot for the guys that have read this long mail :p > > > > kind regards, > > rod > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com > > > > > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > > > > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > > > > > iax:guest at conference.freeswitch.org/888 > > > > googletalk:conf+888 at conference.freeswitch.org > > > > > > pstn:213-799-1400 > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090202/e5cef9ec/attachment-0002.html From kawarod at laposte.net Sun Feb 1 23:36:35 2009 From: kawarod at laposte.net (rod) Date: Mon, 02 Feb 2009 11:36:35 +0400 Subject: [Freeswitch-users] Strange Performance when using as SBC In-Reply-To: <94b568920902012317h8411fccg216435380f62d649@mail.gmail.com> References: <4982EAD5.1010200@laposte.net> <191c3a030901300556s534823a1if4b79b4279962ed1@mail.gmail.com> <49869AF1.1070903@laposte.net> <94b568920902012317h8411fccg216435380f62d649@mail.gmail.com> Message-ID: <4986A283.3090707@laposte.net> Hi Ken, 1) I'd like to use FS to hide topology, so bypass media is not possible 2) done 3) done 4) not used 5) i'm using this ins switch.xml -> , if you think an other log level is more suitable. Regarding logging, I can see in console and in the freeswitch.log that there is still a lot of NOTICE logging, see below: 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:960 switch_core_session_thread() Session 8721 (sofia/internal/sipp at 10.10.10.1:5060) Ended 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:962 switch_core_session_thread() Close Channel sofia/internal/sipp at 10.10.10.1:5060 [CS_HANGUP] 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:960 switch_core_session_thread() Session 8722 (sofia/external/9998 at 10.10.20.100) Ended 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:962 switch_core_session_thread() Close Channel sofia/external/9998 at 10.10.20.100 [CS_HANGUP] 2009-02-02 08:33:56 [NOTICE] sofia.c:3164 sofia_handle_sip_i_state() Channel [sofia/external/9998 at 10.10.20.100] has been answered 2009-02-02 08:33:56 [WARNING] mod_sofia.c:740 sofia_read_frame() Changing codec ptime to 30. I bet you have a linksys/sipura =D Do you have any idea where I can switch off this kind of logging. I thought it should be in /dialplan/internal.xml, but I see that in internal.xml -> thanks a lot for your suggestion. regards, rod Ken Rice wrote: > Dont forget there are several things you can do to increase performance... > > 1) where possible use bypass media or media proxy modes > 2) mount freeswitch/db as a ram drive (if you are using voicemail with > the internal FS DBs you'll need a way to make this persistant across > reboots) > 3) see the wiki for setting reasonable ulimits > 4) (this is my oppinion others may vary) dont use mod_cdr_csv > 5) turn off (or reduce logging) in switch.conf.xml > > all of these thing can greatly improve performance. > > On Mon, Feb 2, 2009 at 1:04 AM, rod > wrote: > > Thanks Anthony, > > the setup is like this: > > sipp server ---- FS 1 ---- FS2 > > FS1 is the AMD CPU that has only one extension in dialplan that > bridges > 9999 to FS2. 9999 is the first extension in FS2 dialplan that > plays moh, > FS2 has no CPU pbm. > > FS1 is maxing out at 60 bridged calls without your option -hp. > > Using -hp, I'm now able to bridge 200 concurrent calls (a great > improvement) and the system is still reactive. CPU load is high > but not > 100% and as the system responds well, I think that doesn't matter. The > 2GB of memory are completely consumed (top command shows 700MB for FS > process). > > I understand that FS1 server is not the best hardware platform, > and I'm > waiting for new 4 cores server for testing. > I will update those numbers when testing with the new hardware. > > regards, > rod. > > Anthony Minessale wrote: > > Which of the 2 machines has the load issue? You said it was one box > > calling the other. > > > > You have 2 major things against you, single CPU and AMD, but you > > should at least be able to get in the vicinity of 800-1000 calls > on a > > box like that. > > > > Are you calling the default 9999? It's not really an appropriate > > extension for load testing. > > On the terminating box you should set up a manual extension that is > > the first one in the dial plan > > to play a wav file from preferably a ram disk or /tmp > > > > If you do plan on using this in production accept nothing less > than a > > multi-core intel machine with at least 4 cores, the more cores the > > better because that parallel processing is where FS gets it's > atvantage. > > > > > > > > On Fri, Jan 30, 2009 at 5:56 AM, rod > > >> wrote: > > > > Dear list, > > > > I've been playing with freeswitch for some time (2 months) > and the > > fact > > is that I'm very pleased with the functionnalities of this > software. > > > > I'd like to use FS as a SBC handling media and I'm doing some > > tests with > > sipp to load the machine but I'm unable to bridge more than > 60 calls > > without seeing the CPU being loaded at 100%. I'm sure > something is > > going > > wrong with my setup but I'm unable to see what. > > > > The test machine has the following specs: > > Athlon XP 3500+ with 2GB of memory (I know this is not a > high end > > machine :p) > > > > Freeswitch:/opt/freeswitch/log# cat /proc/cpuinfo > > processor : 0 > > vendor_id : AuthenticAMD > > cpu family : 15 > > model : 95 > > model name : AMD Athlon(tm) 64 Processor 3500+ > > stepping : 2 > > cpu MHz : 2199.973 > > cache size : 512 KB > > fpu : yes > > fpu_exception : yes > > cpuid level : 1 > > wp : yes > > flags : fpu vme de pse tsc msr pae mce cx8 apic > sep mtrr pge > > mca cmov pat pse36 clflush mmx fxsr sse sse2 syscall nx mmxext > > fxsr_opt > > rdtscp lm 3dnowext 3dnow up rep_good pni cx16 lahf_lm svm > extapic > > cr8_legacy > > bogomips : 4402.97 > > TLB size : 1024 4K pages > > clflush size : 64 > > cache_alignment : 64 > > address sizes : 40 bits physical, 48 bits virtual > > power management: ts fid vid ttp tm stc > > > > I installed FS on a fresh debian 64: > > Linux Freeswitch 2.6.26-1-amd64 #1 SMP Sat Jan 10 17:57:00 > UTC 2009 > > x86_64 GNU/Linux > > > > I set the ulimit parameters like those on the website: > > freeswitch at internal> ... > > Freeswitch:/opt/free-svn/bin# ulimit -a > > core file size (blocks, -c) unlimited > > data seg size (kbytes, -d) unlimited > > scheduling priority (-e) 0 > > file size (blocks, -f) unlimited > > pending signals (-i) unlimited > > max locked memory (kbytes, -l) unlimited > > max memory size (kbytes, -m) unlimited > > open files (-n) 999999 > > pipe size (512 bytes, -p) 8 > > POSIX message queues (bytes, -q) unlimited > > real-time priority (-r) 0 > > stack size (kbytes, -s) 244 > > cpu time (seconds, -t) unlimited > > max user processes (-u) unlimited > > virtual memory (kbytes, -v) unlimited > > file locks (-x) unlimited > > > > > > My network setup is the following: > > > > SIPP machine (10.10.10.1/24)----------------vlan > > > 55 > > ----------(10.10.10.254/24 > ) FS > > (10.10.20.254/24)-------------- > > > vlan56 > > -------------------(10.10.20.100/24 > ) > > OTHER STOCK FS > > > > > > I launched sipp with: > > sipp -sn uac_pcap -s 9999 -r 10 -l 80 -d 60000 -mi > 10.10.10.1 -i > > 10.10.10.1 -mp 25000 10.10.10.254:5060 > > > > > The dialplan on FS is very simple: > > > > > > > > > > > > > > > > > > > data="sofia/external/9999 at 10.10.20.100 > >"/> > > > > > > > > > > > > > > FreeSWITCH Version 1.0.trunk (11560M) Started. > > Crash Protection [Disabled] > > Max Sessions[1000] > > Session Rate[100] > > SQL [Enabled] > > > > > > The test is very simple: sipp dial 9999 that matches in my > FS dialplan > > and this is bridged to an other FS machine playing music on > hold. > > When I launch "top" I see after 30 to 40 s that FS consumes all > > the CPU > > ressources (with a mean of 50-60 % before), with 80 calls. > > When I set 70 calls, I have to wait 70-80 s before seeing > the same > > issue. > > > > Presence is set to false on the 2 profile. > > > > I have the same issue with FS 1.0.2 that' s why I tried FS > 11560. > > > > When I use the FS machine as a router to test the packet per > second > > performance, I'm reaching 100Mbps with 8000pps in each > direction (from > > vlan 55 to vlan56) with less than 12% CPU. So that I don't think > > there's > > an issue with the network. > > > > Here is an "mpstat -P ALL 1" to show you what's happening > suddenly > > with > > 70 bridge calls: > > 12:31:26 CPU %user %nice %sys %iowait %irq %soft > > %steal %idle intr/s > > 12:31:27 all 3,00 0,00 3,00 0,00 1,00 4,00 > > 0,00 89,00 6241,00 > > 12:31:27 0 3,00 0,00 3,00 0,00 1,00 4,00 > > 0,00 89,00 6241,00 > > > > 12:31:27 CPU %user %nice %sys %iowait %irq %soft > > %steal %idle intr/s > > 12:31:28 all 14,14 0,00 56,57 0,00 2,02 5,05 > > 0,00 22,22 6035,35 > > 12:31:28 0 14,14 0,00 56,57 0,00 2,02 5,05 > > 0,00 22,22 6035,35 > > > > 12:31:28 CPU %user %nice %sys %iowait %irq %soft > > %steal %idle intr/s > > 12:31:29 all 24,75 0,00 67,33 0,00 0,99 6,93 > > 0,00 0,00 5483,17 > > 12:31:29 0 24,75 0,00 67,33 0,00 0,99 6,93 > > 0,00 0,00 5483,17 > > > > > > The CPU is going from 89% idle to 0% in less than 2 seconds. > > > > I know that I don't have to expect too much from this kind of > > hardware, > > but it seems strange that the CPU power vanished so suddenly. > > > > Thanks a lot for the guys that have read this long mail :p > > > > kind regards, > > rod > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com > > > > > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > > > > > IRC: irc.freenode.net > #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > > > > > iax:guest at conference.freeswitch.org/888 > > > > > googletalk:conf+888 at conference.freeswitch.org > > > > > > pstn:213-799-1400 > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From blansky at interwise.com Mon Feb 2 00:33:19 2009 From: blansky at interwise.com (Boris Lansky) Date: Mon, 2 Feb 2009 10:33:19 +0200 Subject: [Freeswitch-users] Outbound Proxy configuration In-Reply-To: <191c3a030902011019k94e5fa3i1a37c2c0b2e66e52@mail.gmail.com> References: <7160CD9F-F06E-4964-943A-F4B0C12150CC@jerris.com><4981D3F8.2030207@freeswitch.org> <191c3a030902011019k94e5fa3i1a37c2c0b2e66e52@mail.gmail.com> Message-ID: Thanks Anthony, A little explanation of yours was very helpful. The directive works for me. Regards, Boris Lansky Unified Communications Telephony Team AT&T Unified Communications Phone: +972.3.976.7604 Fax: +972.3.976.7712 blansky at interwise.com From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Sunday, February 01, 2009 8:19 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Outbound Proxy configuration They probably saw your signature that says you work for AT&T and assumed you could follow the documentation since AT&T is a telephone company. What they were trying to explain was if you want to send a call out of a sofia profile you can set the proxy on the fly with that extra parameter added to the dial string anywhere it's used. The explanation requires you understand the basic operation of freeswitch. the uri contained in fs_path indicates the sip address of a proxy server. On Sun, Feb 1, 2009 at 4:37 AM, Boris Lansky wrote: I have checked the "fs_path" usage ... again (I have done it before I have issued my question as well). And once again I can't understand why this thing is useful for me. I have found only two small examples refer the issue that use "fs_path" for a an API call. What I need is a real configuration example that shows configuration of a Proxy Server for all outbound calls going out from a Free Switch. I will real appreciate if I will get an exact answer and not just a general link to a FS doc. Regards, Boris Lansky Unified Communications Telephony Team AT&T Unified Communications Phone: +972.3.976.7604 Fax: +972.3.976.7712 blansky at interwise.com From: freeswitch-users-bounces at lists.freeswitch.org [mailto: freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Raymond Chandler Sent: Thursday, January 29, 2009 6:06 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Outbound Proxy configuration Boris Lansky wrote: Sorry for the stupid question but in what configuration file should I add such line "sofia/foo/user at that.domain ;fs_path=sip:proxy.this.domain" ? appology accepted... look in the dialplan, there should be plenty of documentations on using the dialplan on our wiki... wiki.freeswitch.org... look for sofia syntax too on the mod_sofia page -Ray _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090202/3cea3f07/attachment-0002.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/png Size: 16211 bytes Desc: image001.png Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090202/3cea3f07/attachment-0002.png From sias at cpdata.co.za Mon Feb 2 00:51:57 2009 From: sias at cpdata.co.za (Sias Mey) Date: Mon, 2 Feb 2009 10:51:57 +0200 Subject: [Freeswitch-users] Conference dialing and uuid In-Reply-To: <191c3a030901300605r78dd8943w7fb1075ef851e1d@mail.gmail.com> References: <20090130103944.GA23536@cpdata.co.za> <20090130133315.GB23536@cpdata.co.za> <191c3a030901300605r78dd8943w7fb1075ef851e1d@mail.gmail.com> Message-ID: <20090202085157.GA3555@cpdata.co.za> Yes ... yes indeed I can. That works quite a bit better than generating 4 channels and getting massively confused with what uuid does what... but now im stuck without ringback again :-(. In my conference dial string I send: {ringback=\'%(400,200,400,450)\',transfer_ringback=\'%(400,200,400,450)\', .... }sofia/internal/1001 at xxx.xxx.xxx.xxx A dump of all the channel variables shows ringback is set to %25(400,200,400,450)%3B%25(400,2200,400,450) %25(400,200,400,450)%3B%25(400,2200,400,450) This seems ok to me but I still dont get any ringback. Thanks again for answering all the anoying questions from the same guy :-P, Sias On Fri, Jan 30, 2009 at 08:05:07AM -0600, Anthony Minessale wrote: > you should be able to use {} in the dial command > you also should be able to do > originate {...}sofia/profile/[1]user at domain.com > conference:@ inline > to the api interface > > On Fri, Jan 30, 2009 at 7:33 AM, Sias Mey <[2]sias at cpdata.co.za> wrote: > > Hi Brian, > Hmmm Ill do some more testing on it later. But I got a destination > out > of order when I tried. Right now Im busy implementing the string > checking. Which seems like it will work out ok, but is clearly not > ideal. > Thanks for the replay > > On Fri, Jan 30, 2009 at 04:45:54AM -0600, Brian West wrote: > > What wasn't working about this? The {} can be used everywhere > without > > a problem... Maybe you can provide more details on this. > > > > /b > > > > > > > > > > On Jan 30, 2009, at 4:39 AM, Sias Mey wrote: > > > > > > > > I couldent find a way of setting channel variables or executing > > > javascript directly on the conference dial since it expects and > > > endpoint > > > and the {} syntax produced an error. So now I am using the Loopback > > > inteface to register some values. > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > [3]Freeswitch-users at lists.freeswitch.org > > [4]http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:[5]http://lists.freeswitch.org/mailman/options/freeswitch-u > sers > > [6]http://www.freeswitch.org > _______________________________________________ > Freeswitch-users mailing list > [7]Freeswitch-users at lists.freeswitch.org > [8]http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:[9]http://lists.freeswitch.org/mailman/options/freeswitch-u > sers > [10]http://www.freeswitch.org > > -- > Anthony Minessale II > FreeSWITCH [11]http://www.freeswitch.org/ > ClueCon [12]http://www.cluecon.com/ > AIM: anthm > [13]MSN:anthony_minessale at hotmail.com > GTALK/JABBER/[14]PAYPAL:anthony.minessale at gmail.com > IRC: [15]irc.freenode.net #freeswitch > FreeSWITCH Developer Conference > [16]sip:888 at conference.freeswitch.org > [17]iax:guest at conference.freeswitch.org/888 > [18]googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > References > > 1. mailto:user at domain.com > 2. mailto:sias at cpdata.co.za > 3. mailto:Freeswitch-users at lists.freeswitch.org > 4. http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > 5. http://lists.freeswitch.org/mailman/options/freeswitch-users > 6. http://www.freeswitch.org/ > 7. mailto:Freeswitch-users at lists.freeswitch.org > 8. http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > 9. http://lists.freeswitch.org/mailman/options/freeswitch-users > 10. http://www.freeswitch.org/ > 11. http://www.freeswitch.org/ > 12. http://www.cluecon.com/ > 13. mailto:MSN%3Aanthony_minessale at hotmail.com > 14. mailto:PAYPAL%3Aanthony.minessale at gmail.com > 15. http://irc.freenode.net/ > 16. mailto:sip%3A888 at conference.freeswitch.org > 17. http://iax:guest at conference.freeswitch.org/888 > 18. mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Mon Feb 2 00:55:42 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 2 Feb 2009 02:55:42 -0600 Subject: [Freeswitch-users] Conference dialing and uuid In-Reply-To: <20090202085157.GA3555@cpdata.co.za> References: <20090130103944.GA23536@cpdata.co.za> <20090130133315.GB23536@cpdata.co.za> <191c3a030901300605r78dd8943w7fb1075ef851e1d@mail.gmail.com> <20090202085157.GA3555@cpdata.co.za> Message-ID: <0F863750-5C77-4CD9-ADC6-CFDDF2B149B4@freeswitch.org> You can't get ringback dialing out from a conference its not possible as it is now. /b On Feb 2, 2009, at 2:51 AM, Sias Mey wrote: > Yes ... yes indeed I can. > > That works quite a bit better than generating 4 channels and getting > massively confused with what uuid does what... but now im stuck > without > ringback again :-(. > > In my conference dial string I send: > {ringback=\'%(400,200,400,450)\',transfer_ringback= > \'%(400,200,400,450)\', > .... }sofia/internal/1001 at xxx.xxx.xxx.xxx > > A dump of all the channel variables shows ringback is set to > > %25(400,200,400,450)%3B%25(400,2200,400,450) > %25(400,200,400,450)%3B%25(400,2200,400,450) transfer_ringback> > > This seems ok to me but I still dont get any ringback. > > Thanks again for answering all the anoying questions from the same guy > :-P, > Sias -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090202/ede8482f/attachment-0002.html From jaybinks at gmail.com Mon Feb 2 01:09:10 2009 From: jaybinks at gmail.com (jay binks) Date: Mon, 2 Feb 2009 19:09:10 +1000 Subject: [Freeswitch-users] Strange Performance when using as SBC In-Reply-To: <4986A283.3090707@laposte.net> References: <4982EAD5.1010200@laposte.net> <191c3a030901300556s534823a1if4b79b4279962ed1@mail.gmail.com> <49869AF1.1070903@laposte.net> <94b568920902012317h8411fccg216435380f62d649@mail.gmail.com> <4986A283.3090707@laposte.net> Message-ID: for topology hiding, use proxy media. it means FS ignores the RTP stream totally, and just passes it through. On Mon, Feb 2, 2009 at 5:36 PM, rod wrote: > Hi Ken, > > 1) I'd like to use FS to hide topology, so bypass media is not possible > 2) done > 3) done > 4) not used > 5) i'm using this ins switch.xml -> value="info"/>, if you think an other log level is more suitable. > > Regarding logging, I can see in console and in the freeswitch.log that > there is still a lot of NOTICE logging, see below: > 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:960 > switch_core_session_thread() Session 8721 > (sofia/internal/sipp at 10.10.10.1:5060) Ended > 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:962 > switch_core_session_thread() Close Channel > sofia/internal/sipp at 10.10.10.1:5060 [CS_HANGUP] > 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:960 > switch_core_session_thread() Session 8722 > (sofia/external/9998 at 10.10.20.100) Ended > 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:962 > switch_core_session_thread() Close Channel > sofia/external/9998 at 10.10.20.100 [CS_HANGUP] > 2009-02-02 08:33:56 [NOTICE] sofia.c:3164 sofia_handle_sip_i_state() > Channel [sofia/external/9998 at 10.10.20.100] has been answered > 2009-02-02 08:33:56 [WARNING] mod_sofia.c:740 sofia_read_frame() > Changing codec ptime to 30. I bet you have a linksys/sipura =D > > Do you have any idea where I can switch off this kind of logging. I > thought it should be in /dialplan/internal.xml, but I see that in > internal.xml -> > > thanks a lot for your suggestion. > > regards, > rod > > Ken Rice wrote: > > Dont forget there are several things you can do to increase > performance... > > > > 1) where possible use bypass media or media proxy modes > > 2) mount freeswitch/db as a ram drive (if you are using voicemail with > > the internal FS DBs you'll need a way to make this persistant across > > reboots) > > 3) see the wiki for setting reasonable ulimits > > 4) (this is my oppinion others may vary) dont use mod_cdr_csv > > 5) turn off (or reduce logging) in switch.conf.xml > > > > all of these thing can greatly improve performance. > > > > On Mon, Feb 2, 2009 at 1:04 AM, rod > > wrote: > > > > Thanks Anthony, > > > > the setup is like this: > > > > sipp server ---- FS 1 ---- FS2 > > > > FS1 is the AMD CPU that has only one extension in dialplan that > > bridges > > 9999 to FS2. 9999 is the first extension in FS2 dialplan that > > plays moh, > > FS2 has no CPU pbm. > > > > FS1 is maxing out at 60 bridged calls without your option -hp. > > > > Using -hp, I'm now able to bridge 200 concurrent calls (a great > > improvement) and the system is still reactive. CPU load is high > > but not > > 100% and as the system responds well, I think that doesn't matter. > The > > 2GB of memory are completely consumed (top command shows 700MB for FS > > process). > > > > I understand that FS1 server is not the best hardware platform, > > and I'm > > waiting for new 4 cores server for testing. > > I will update those numbers when testing with the new hardware. > > > > regards, > > rod. > > > > Anthony Minessale wrote: > > > Which of the 2 machines has the load issue? You said it was one box > > > calling the other. > > > > > > You have 2 major things against you, single CPU and AMD, but you > > > should at least be able to get in the vicinity of 800-1000 calls > > on a > > > box like that. > > > > > > Are you calling the default 9999? It's not really an appropriate > > > extension for load testing. > > > On the terminating box you should set up a manual extension that is > > > the first one in the dial plan > > > to play a wav file from preferably a ram disk or /tmp > > > > > > If you do plan on using this in production accept nothing less > > than a > > > multi-core intel machine with at least 4 cores, the more cores the > > > better because that parallel processing is where FS gets it's > > atvantage. > > > > > > > > > > > > On Fri, Jan 30, 2009 at 5:56 AM, rod > > > > >> wrote: > > > > > > Dear list, > > > > > > I've been playing with freeswitch for some time (2 months) > > and the > > > fact > > > is that I'm very pleased with the functionnalities of this > > software. > > > > > > I'd like to use FS as a SBC handling media and I'm doing some > > > tests with > > > sipp to load the machine but I'm unable to bridge more than > > 60 calls > > > without seeing the CPU being loaded at 100%. I'm sure > > something is > > > going > > > wrong with my setup but I'm unable to see what. > > > > > > The test machine has the following specs: > > > Athlon XP 3500+ with 2GB of memory (I know this is not a > > high end > > > machine :p) > > > > > > Freeswitch:/opt/freeswitch/log# cat /proc/cpuinfo > > > processor : 0 > > > vendor_id : AuthenticAMD > > > cpu family : 15 > > > model : 95 > > > model name : AMD Athlon(tm) 64 Processor 3500+ > > > stepping : 2 > > > cpu MHz : 2199.973 > > > cache size : 512 KB > > > fpu : yes > > > fpu_exception : yes > > > cpuid level : 1 > > > wp : yes > > > flags : fpu vme de pse tsc msr pae mce cx8 apic > > sep mtrr pge > > > mca cmov pat pse36 clflush mmx fxsr sse sse2 syscall nx mmxext > > > fxsr_opt > > > rdtscp lm 3dnowext 3dnow up rep_good pni cx16 lahf_lm svm > > extapic > > > cr8_legacy > > > bogomips : 4402.97 > > > TLB size : 1024 4K pages > > > clflush size : 64 > > > cache_alignment : 64 > > > address sizes : 40 bits physical, 48 bits virtual > > > power management: ts fid vid ttp tm stc > > > > > > I installed FS on a fresh debian 64: > > > Linux Freeswitch 2.6.26-1-amd64 #1 SMP Sat Jan 10 17:57:00 > > UTC 2009 > > > x86_64 GNU/Linux > > > > > > I set the ulimit parameters like those on the website: > > > freeswitch at internal> ... > > > Freeswitch:/opt/free-svn/bin# ulimit -a > > > core file size (blocks, -c) unlimited > > > data seg size (kbytes, -d) unlimited > > > scheduling priority (-e) 0 > > > file size (blocks, -f) unlimited > > > pending signals (-i) unlimited > > > max locked memory (kbytes, -l) unlimited > > > max memory size (kbytes, -m) unlimited > > > open files (-n) 999999 > > > pipe size (512 bytes, -p) 8 > > > POSIX message queues (bytes, -q) unlimited > > > real-time priority (-r) 0 > > > stack size (kbytes, -s) 244 > > > cpu time (seconds, -t) unlimited > > > max user processes (-u) unlimited > > > virtual memory (kbytes, -v) unlimited > > > file locks (-x) unlimited > > > > > > > > > My network setup is the following: > > > > > > SIPP machine (10.10.10.1/24)----------------vlan > > > > > 55 > > > ----------(10.10.10.254/24 > > ) FS > > > (10.10.20.254/24)-------------- > > > > > vlan56 > > > -------------------(10.10.20.100/24 > > ) > > > OTHER STOCK FS > > > > > > > > > I launched sipp with: > > > sipp -sn uac_pcap -s 9999 -r 10 -l 80 -d 60000 -mi > > 10.10.10.1 -i > > > 10.10.10.1 -mp 25000 10.10.10.254:5060 > > > > > > > > The dialplan on FS is very simple: > > > > > > > > > > > > > > > > > > > > > > > > > > > > > data="sofia/external/9999 at 10.10.20.100 > > > >"/> > > > > > > > > > > > > > > > > > > > > > FreeSWITCH Version 1.0.trunk (11560M) Started. > > > Crash Protection [Disabled] > > > Max Sessions[1000] > > > Session Rate[100] > > > SQL [Enabled] > > > > > > > > > The test is very simple: sipp dial 9999 that matches in my > > FS dialplan > > > and this is bridged to an other FS machine playing music on > > hold. > > > When I launch "top" I see after 30 to 40 s that FS consumes > all > > > the CPU > > > ressources (with a mean of 50-60 % before), with 80 calls. > > > When I set 70 calls, I have to wait 70-80 s before seeing > > the same > > > issue. > > > > > > Presence is set to false on the 2 profile. > > > > > > I have the same issue with FS 1.0.2 that' s why I tried FS > > 11560. > > > > > > When I use the FS machine as a router to test the packet per > > second > > > performance, I'm reaching 100Mbps with 8000pps in each > > direction (from > > > vlan 55 to vlan56) with less than 12% CPU. So that I don't > think > > > there's > > > an issue with the network. > > > > > > Here is an "mpstat -P ALL 1" to show you what's happening > > suddenly > > > with > > > 70 bridge calls: > > > 12:31:26 CPU %user %nice %sys %iowait %irq > %soft > > > %steal %idle intr/s > > > 12:31:27 all 3,00 0,00 3,00 0,00 1,00 > 4,00 > > > 0,00 89,00 6241,00 > > > 12:31:27 0 3,00 0,00 3,00 0,00 1,00 > 4,00 > > > 0,00 89,00 6241,00 > > > > > > 12:31:27 CPU %user %nice %sys %iowait %irq > %soft > > > %steal %idle intr/s > > > 12:31:28 all 14,14 0,00 56,57 0,00 2,02 > 5,05 > > > 0,00 22,22 6035,35 > > > 12:31:28 0 14,14 0,00 56,57 0,00 2,02 > 5,05 > > > 0,00 22,22 6035,35 > > > > > > 12:31:28 CPU %user %nice %sys %iowait %irq > %soft > > > %steal %idle intr/s > > > 12:31:29 all 24,75 0,00 67,33 0,00 0,99 > 6,93 > > > 0,00 0,00 5483,17 > > > 12:31:29 0 24,75 0,00 67,33 0,00 0,99 > 6,93 > > > 0,00 0,00 5483,17 > > > > > > > > > The CPU is going from 89% idle to 0% in less than 2 seconds. > > > > > > I know that I don't have to expect too much from this kind of > > > hardware, > > > but it seems strange that the CPU power vanished so suddenly. > > > > > > Thanks a lot for the guys that have read this long mail :p > > > > > > kind regards, > > > rod > > > > > > > > > _______________________________________________ > > > Freeswitch-users mailing list > > > Freeswitch-users at lists.freeswitch.org > > > > > > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > > > > > > > > > > -- > > > Anthony Minessale II > > > > > > FreeSWITCH http://www.freeswitch.org/ > > > ClueCon http://www.cluecon.com/ > > > > > > AIM: anthm > > > MSN:anthony_minessale at hotmail.com > > > > > > > > > > >> > > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > > > > > > > > > >> > > > IRC: irc.freenode.net > > #freeswitch > > > > > > FreeSWITCH Developer Conference > > > sip:888 at conference.freeswitch.org > > > > > > > > > > >> > > > iax:guest at conference.freeswitch.org/888 > > > > > > > > googletalk:conf+888 at conference.freeswitch.org > > > > > > > > > > >> > > > pstn:213-799-1400 > > > > > > ------------------------------------------------------------------------ > > > > > > _______________________________________________ > > > Freeswitch-users mailing list > > > Freeswitch-users at lists.freeswitch.org > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Sincerely Jay -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090202/c0181163/attachment-0002.html From krice at freeswitch.org Mon Feb 2 01:09:46 2009 From: krice at freeswitch.org (Ken Rice) Date: Mon, 02 Feb 2009 03:09:46 -0600 Subject: [Freeswitch-users] Strange Performance when using as SBC In-Reply-To: <4986A283.3090707@laposte.net> Message-ID: If you don't have to transcode, using proxy media mode will still save you some CPU time. This is 1/2 way between bypass media and the default media interactive mode. The other draw back to this mode is if you are using FS to clean up RTP and DTMF you loose those functions but they are not needed in most use cases. As far as the log level goes, I found that once I had things stable setting the loglevel to helped a good deal... Info is probably a bit too high of a loglevel I would probably go for CRIT or ERR (2 or 1 respectively) if you insist on leaving logging turned on... On a busy system these can and will generate a good deal of activity (and disk IO if using mod_logfile) Ken > From: rod > Reply-To: > Date: Mon, 02 Feb 2009 11:36:35 +0400 > To: > Subject: Re: [Freeswitch-users] Strange Performance when using as SBC > > Hi Ken, > > 1) I'd like to use FS to hide topology, so bypass media is not possible > 2) done > 3) done > 4) not used > 5) i'm using this ins switch.xml -> value="info"/>, if you think an other log level is more suitable. > > Regarding logging, I can see in console and in the freeswitch.log that > there is still a lot of NOTICE logging, see below: > 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:960 > switch_core_session_thread() Session 8721 > (sofia/internal/sipp at 10.10.10.1:5060) Ended > 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:962 > switch_core_session_thread() Close Channel > sofia/internal/sipp at 10.10.10.1:5060 [CS_HANGUP] > 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:960 > switch_core_session_thread() Session 8722 > (sofia/external/9998 at 10.10.20.100) Ended > 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:962 > switch_core_session_thread() Close Channel > sofia/external/9998 at 10.10.20.100 [CS_HANGUP] > 2009-02-02 08:33:56 [NOTICE] sofia.c:3164 sofia_handle_sip_i_state() > Channel [sofia/external/9998 at 10.10.20.100] has been answered > 2009-02-02 08:33:56 [WARNING] mod_sofia.c:740 sofia_read_frame() > Changing codec ptime to 30. I bet you have a linksys/sipura =D > > Do you have any idea where I can switch off this kind of logging. I > thought it should be in /dialplan/internal.xml, but I see that in > internal.xml -> > > thanks a lot for your suggestion. > > regards, > rod > > Ken Rice wrote: >> Dont forget there are several things you can do to increase performance... >> >> 1) where possible use bypass media or media proxy modes >> 2) mount freeswitch/db as a ram drive (if you are using voicemail with >> the internal FS DBs you'll need a way to make this persistant across >> reboots) >> 3) see the wiki for setting reasonable ulimits >> 4) (this is my oppinion others may vary) dont use mod_cdr_csv >> 5) turn off (or reduce logging) in switch.conf.xml >> >> all of these thing can greatly improve performance. >> >> On Mon, Feb 2, 2009 at 1:04 AM, rod > > wrote: >> >> Thanks Anthony, >> >> the setup is like this: >> >> sipp server ---- FS 1 ---- FS2 >> >> FS1 is the AMD CPU that has only one extension in dialplan that >> bridges >> 9999 to FS2. 9999 is the first extension in FS2 dialplan that >> plays moh, >> FS2 has no CPU pbm. >> >> FS1 is maxing out at 60 bridged calls without your option -hp. >> >> Using -hp, I'm now able to bridge 200 concurrent calls (a great >> improvement) and the system is still reactive. CPU load is high >> but not >> 100% and as the system responds well, I think that doesn't matter. The >> 2GB of memory are completely consumed (top command shows 700MB for FS >> process). >> >> I understand that FS1 server is not the best hardware platform, >> and I'm >> waiting for new 4 cores server for testing. >> I will update those numbers when testing with the new hardware. >> >> regards, >> rod. >> >> Anthony Minessale wrote: >>> Which of the 2 machines has the load issue? You said it was one box >>> calling the other. >>> >>> You have 2 major things against you, single CPU and AMD, but you >>> should at least be able to get in the vicinity of 800-1000 calls >> on a >>> box like that. >>> >>> Are you calling the default 9999? It's not really an appropriate >>> extension for load testing. >>> On the terminating box you should set up a manual extension that is >>> the first one in the dial plan >>> to play a wav file from preferably a ram disk or /tmp >>> >>> If you do plan on using this in production accept nothing less >> than a >>> multi-core intel machine with at least 4 cores, the more cores the >>> better because that parallel processing is where FS gets it's >> atvantage. >>> >>> >>> >>> On Fri, Jan 30, 2009 at 5:56 AM, rod > >>> >> wrote: >>> >>> Dear list, >>> >>> I've been playing with freeswitch for some time (2 months) >> and the >>> fact >>> is that I'm very pleased with the functionnalities of this >> software. >>> >>> I'd like to use FS as a SBC handling media and I'm doing some >>> tests with >>> sipp to load the machine but I'm unable to bridge more than >> 60 calls >>> without seeing the CPU being loaded at 100%. I'm sure >> something is >>> going >>> wrong with my setup but I'm unable to see what. >>> >>> The test machine has the following specs: >>> Athlon XP 3500+ with 2GB of memory (I know this is not a >> high end >>> machine :p) >>> >>> Freeswitch:/opt/freeswitch/log# cat /proc/cpuinfo >>> processor : 0 >>> vendor_id : AuthenticAMD >>> cpu family : 15 >>> model : 95 >>> model name : AMD Athlon(tm) 64 Processor 3500+ >>> stepping : 2 >>> cpu MHz : 2199.973 >>> cache size : 512 KB >>> fpu : yes >>> fpu_exception : yes >>> cpuid level : 1 >>> wp : yes >>> flags : fpu vme de pse tsc msr pae mce cx8 apic >> sep mtrr pge >>> mca cmov pat pse36 clflush mmx fxsr sse sse2 syscall nx mmxext >>> fxsr_opt >>> rdtscp lm 3dnowext 3dnow up rep_good pni cx16 lahf_lm svm >> extapic >>> cr8_legacy >>> bogomips : 4402.97 >>> TLB size : 1024 4K pages >>> clflush size : 64 >>> cache_alignment : 64 >>> address sizes : 40 bits physical, 48 bits virtual >>> power management: ts fid vid ttp tm stc >>> >>> I installed FS on a fresh debian 64: >>> Linux Freeswitch 2.6.26-1-amd64 #1 SMP Sat Jan 10 17:57:00 >> UTC 2009 >>> x86_64 GNU/Linux >>> >>> I set the ulimit parameters like those on the website: >>> freeswitch at internal> ... >>> Freeswitch:/opt/free-svn/bin# ulimit -a >>> core file size (blocks, -c) unlimited >>> data seg size (kbytes, -d) unlimited >>> scheduling priority (-e) 0 >>> file size (blocks, -f) unlimited >>> pending signals (-i) unlimited >>> max locked memory (kbytes, -l) unlimited >>> max memory size (kbytes, -m) unlimited >>> open files (-n) 999999 >>> pipe size (512 bytes, -p) 8 >>> POSIX message queues (bytes, -q) unlimited >>> real-time priority (-r) 0 >>> stack size (kbytes, -s) 244 >>> cpu time (seconds, -t) unlimited >>> max user processes (-u) unlimited >>> virtual memory (kbytes, -v) unlimited >>> file locks (-x) unlimited >>> >>> >>> My network setup is the following: >>> >>> SIPP machine (10.10.10.1/24)----------------vlan >> >>> 55 >>> ----------(10.10.10.254/24 >> ) FS >>> (10.10.20.254/24)-------------- >> >>> vlan56 >>> -------------------(10.10.20.100/24 >> ) >>> OTHER STOCK FS >>> >>> >>> I launched sipp with: >>> sipp -sn uac_pcap -s 9999 -r 10 -l 80 -d 60000 -mi >> 10.10.10.1 -i >>> 10.10.10.1 -mp 25000 10.10.10.254:5060 >> >>> >>> The dialplan on FS is very simple: >>> >>> >>> >>> >>> >>> >>> >>> >>> >> data="sofia/external/9999 at 10.10.20.100 >> > >"/> >>> >>> >>> >>> >>> >>> >>> FreeSWITCH Version 1.0.trunk (11560M) Started. >>> Crash Protection [Disabled] >>> Max Sessions[1000] >>> Session Rate[100] >>> SQL [Enabled] >>> >>> >>> The test is very simple: sipp dial 9999 that matches in my >> FS dialplan >>> and this is bridged to an other FS machine playing music on >> hold. >>> When I launch "top" I see after 30 to 40 s that FS consumes all >>> the CPU >>> ressources (with a mean of 50-60 % before), with 80 calls. >>> When I set 70 calls, I have to wait 70-80 s before seeing >> the same >>> issue. >>> >>> Presence is set to false on the 2 profile. >>> >>> I have the same issue with FS 1.0.2 that' s why I tried FS >> 11560. >>> >>> When I use the FS machine as a router to test the packet per >> second >>> performance, I'm reaching 100Mbps with 8000pps in each >> direction (from >>> vlan 55 to vlan56) with less than 12% CPU. So that I don't think >>> there's >>> an issue with the network. >>> >>> Here is an "mpstat -P ALL 1" to show you what's happening >> suddenly >>> with >>> 70 bridge calls: >>> 12:31:26 CPU %user %nice %sys %iowait %irq %soft >>> %steal %idle intr/s >>> 12:31:27 all 3,00 0,00 3,00 0,00 1,00 4,00 >>> 0,00 89,00 6241,00 >>> 12:31:27 0 3,00 0,00 3,00 0,00 1,00 4,00 >>> 0,00 89,00 6241,00 >>> >>> 12:31:27 CPU %user %nice %sys %iowait %irq %soft >>> %steal %idle intr/s >>> 12:31:28 all 14,14 0,00 56,57 0,00 2,02 5,05 >>> 0,00 22,22 6035,35 >>> 12:31:28 0 14,14 0,00 56,57 0,00 2,02 5,05 >>> 0,00 22,22 6035,35 >>> >>> 12:31:28 CPU %user %nice %sys %iowait %irq %soft >>> %steal %idle intr/s >>> 12:31:29 all 24,75 0,00 67,33 0,00 0,99 6,93 >>> 0,00 0,00 5483,17 >>> 12:31:29 0 24,75 0,00 67,33 0,00 0,99 6,93 >>> 0,00 0,00 5483,17 >>> >>> >>> The CPU is going from 89% idle to 0% in less than 2 seconds. >>> >>> I know that I don't have to expect too much from this kind of >>> hardware, >>> but it seems strange that the CPU power vanished so suddenly. >>> >>> Thanks a lot for the guys that have read this long mail :p >>> >>> kind regards, >>> rod >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >> >>> > > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >>> >>> -- >>> Anthony Minessale II >>> >>> FreeSWITCH http://www.freeswitch.org/ >>> ClueCon http://www.cluecon.com/ >>> >>> AIM: anthm >>> MSN:anthony_minessale at hotmail.com >> >>> > > >>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> >>> > > >>> IRC: irc.freenode.net >> #freeswitch >>> >>> FreeSWITCH Developer Conference >>> sip:888 at conference.freeswitch.org >> >>> > > >>> iax:guest at conference.freeswitch.org/888 >> >>> >>> googletalk:conf+888 at conference.freeswitch.org >> >>> > > >>> pstn:213-799-1400 >>> >> ------------------------------------------------------------------------ >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From kawarod at laposte.net Mon Feb 2 02:00:12 2009 From: kawarod at laposte.net (rod) Date: Mon, 02 Feb 2009 14:00:12 +0400 Subject: [Freeswitch-users] Strange Performance when using as SBC In-Reply-To: References: Message-ID: <4986C42C.7030700@laposte.net> Hi Ken, Jay, thanks for pointing to proxy media, I will test. Ken, you are right, I was brain damaged (a stupid mistake) when setting INFO cause this kind of level could be very verbose. I'm switching to CRIT or ERR. Thanks guys, rod. thanks for Ken Rice wrote: > If you don't have to transcode, using proxy media mode will still save you > some CPU time. This is 1/2 way between bypass media and the default media > interactive mode. The other draw back to this mode is if you are using FS to > clean up RTP and DTMF you loose those functions but they are not needed in > most use cases. > > As far as the log level goes, I found that once I had things stable setting > the loglevel to helped a good deal... Info is probably a bit too high of a > loglevel I would probably go for CRIT or ERR (2 or 1 respectively) if you > insist on leaving logging turned on... On a busy system these can and will > generate a good deal of activity (and disk IO if using mod_logfile) > > Ken > > > >> From: rod >> Reply-To: >> Date: Mon, 02 Feb 2009 11:36:35 +0400 >> To: >> Subject: Re: [Freeswitch-users] Strange Performance when using as SBC >> >> Hi Ken, >> >> 1) I'd like to use FS to hide topology, so bypass media is not possible >> 2) done >> 3) done >> 4) not used >> 5) i'm using this ins switch.xml -> > value="info"/>, if you think an other log level is more suitable. >> >> Regarding logging, I can see in console and in the freeswitch.log that >> there is still a lot of NOTICE logging, see below: >> 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:960 >> switch_core_session_thread() Session 8721 >> (sofia/internal/sipp at 10.10.10.1:5060) Ended >> 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:962 >> switch_core_session_thread() Close Channel >> sofia/internal/sipp at 10.10.10.1:5060 [CS_HANGUP] >> 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:960 >> switch_core_session_thread() Session 8722 >> (sofia/external/9998 at 10.10.20.100) Ended >> 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:962 >> switch_core_session_thread() Close Channel >> sofia/external/9998 at 10.10.20.100 [CS_HANGUP] >> 2009-02-02 08:33:56 [NOTICE] sofia.c:3164 sofia_handle_sip_i_state() >> Channel [sofia/external/9998 at 10.10.20.100] has been answered >> 2009-02-02 08:33:56 [WARNING] mod_sofia.c:740 sofia_read_frame() >> Changing codec ptime to 30. I bet you have a linksys/sipura =D >> >> Do you have any idea where I can switch off this kind of logging. I >> thought it should be in /dialplan/internal.xml, but I see that in >> internal.xml -> >> >> thanks a lot for your suggestion. >> >> regards, >> rod >> >> Ken Rice wrote: >> >>> Dont forget there are several things you can do to increase performance... >>> >>> 1) where possible use bypass media or media proxy modes >>> 2) mount freeswitch/db as a ram drive (if you are using voicemail with >>> the internal FS DBs you'll need a way to make this persistant across >>> reboots) >>> 3) see the wiki for setting reasonable ulimits >>> 4) (this is my oppinion others may vary) dont use mod_cdr_csv >>> 5) turn off (or reduce logging) in switch.conf.xml >>> >>> all of these thing can greatly improve performance. >>> >>> On Mon, Feb 2, 2009 at 1:04 AM, rod >> > wrote: >>> >>> Thanks Anthony, >>> >>> the setup is like this: >>> >>> sipp server ---- FS 1 ---- FS2 >>> >>> FS1 is the AMD CPU that has only one extension in dialplan that >>> bridges >>> 9999 to FS2. 9999 is the first extension in FS2 dialplan that >>> plays moh, >>> FS2 has no CPU pbm. >>> >>> FS1 is maxing out at 60 bridged calls without your option -hp. >>> >>> Using -hp, I'm now able to bridge 200 concurrent calls (a great >>> improvement) and the system is still reactive. CPU load is high >>> but not >>> 100% and as the system responds well, I think that doesn't matter. The >>> 2GB of memory are completely consumed (top command shows 700MB for FS >>> process). >>> >>> I understand that FS1 server is not the best hardware platform, >>> and I'm >>> waiting for new 4 cores server for testing. >>> I will update those numbers when testing with the new hardware. >>> >>> regards, >>> rod. >>> >>> Anthony Minessale wrote: >>> >>>> Which of the 2 machines has the load issue? You said it was one box >>>> calling the other. >>>> >>>> You have 2 major things against you, single CPU and AMD, but you >>>> should at least be able to get in the vicinity of 800-1000 calls >>>> >>> on a >>> >>>> box like that. >>>> >>>> Are you calling the default 9999? It's not really an appropriate >>>> extension for load testing. >>>> On the terminating box you should set up a manual extension that is >>>> the first one in the dial plan >>>> to play a wav file from preferably a ram disk or /tmp >>>> >>>> If you do plan on using this in production accept nothing less >>>> >>> than a >>> >>>> multi-core intel machine with at least 4 cores, the more cores the >>>> better because that parallel processing is where FS gets it's >>>> >>> atvantage. >>> >>>> >>>> On Fri, Jan 30, 2009 at 5:56 AM, rod >>> >>> >>> >>>> >> wrote: >>>> >>>> Dear list, >>>> >>>> I've been playing with freeswitch for some time (2 months) >>>> >>> and the >>> >>>> fact >>>> is that I'm very pleased with the functionnalities of this >>>> >>> software. >>> >>>> I'd like to use FS as a SBC handling media and I'm doing some >>>> tests with >>>> sipp to load the machine but I'm unable to bridge more than >>>> >>> 60 calls >>> >>>> without seeing the CPU being loaded at 100%. I'm sure >>>> >>> something is >>> >>>> going >>>> wrong with my setup but I'm unable to see what. >>>> >>>> The test machine has the following specs: >>>> Athlon XP 3500+ with 2GB of memory (I know this is not a >>>> >>> high end >>> >>>> machine :p) >>>> >>>> Freeswitch:/opt/freeswitch/log# cat /proc/cpuinfo >>>> processor : 0 >>>> vendor_id : AuthenticAMD >>>> cpu family : 15 >>>> model : 95 >>>> model name : AMD Athlon(tm) 64 Processor 3500+ >>>> stepping : 2 >>>> cpu MHz : 2199.973 >>>> cache size : 512 KB >>>> fpu : yes >>>> fpu_exception : yes >>>> cpuid level : 1 >>>> wp : yes >>>> flags : fpu vme de pse tsc msr pae mce cx8 apic >>>> >>> sep mtrr pge >>> >>>> mca cmov pat pse36 clflush mmx fxsr sse sse2 syscall nx mmxext >>>> fxsr_opt >>>> rdtscp lm 3dnowext 3dnow up rep_good pni cx16 lahf_lm svm >>>> >>> extapic >>> >>>> cr8_legacy >>>> bogomips : 4402.97 >>>> TLB size : 1024 4K pages >>>> clflush size : 64 >>>> cache_alignment : 64 >>>> address sizes : 40 bits physical, 48 bits virtual >>>> power management: ts fid vid ttp tm stc >>>> >>>> I installed FS on a fresh debian 64: >>>> Linux Freeswitch 2.6.26-1-amd64 #1 SMP Sat Jan 10 17:57:00 >>>> >>> UTC 2009 >>> >>>> x86_64 GNU/Linux >>>> >>>> I set the ulimit parameters like those on the website: >>>> freeswitch at internal> ... >>>> Freeswitch:/opt/free-svn/bin# ulimit -a >>>> core file size (blocks, -c) unlimited >>>> data seg size (kbytes, -d) unlimited >>>> scheduling priority (-e) 0 >>>> file size (blocks, -f) unlimited >>>> pending signals (-i) unlimited >>>> max locked memory (kbytes, -l) unlimited >>>> max memory size (kbytes, -m) unlimited >>>> open files (-n) 999999 >>>> pipe size (512 bytes, -p) 8 >>>> POSIX message queues (bytes, -q) unlimited >>>> real-time priority (-r) 0 >>>> stack size (kbytes, -s) 244 >>>> cpu time (seconds, -t) unlimited >>>> max user processes (-u) unlimited >>>> virtual memory (kbytes, -v) unlimited >>>> file locks (-x) unlimited >>>> >>>> >>>> My network setup is the following: >>>> >>>> SIPP machine (10.10.10.1/24)----------------vlan >>>> >>> >>> >>>> 55 >>>> ----------(10.10.10.254/24 >>>> >>> ) FS >>> >>>> (10.10.20.254/24)-------------- >>>> >>> >>> >>>> vlan56 >>>> -------------------(10.10.20.100/24 >>>> >>> ) >>> >>>> OTHER STOCK FS >>>> >>>> >>>> I launched sipp with: >>>> sipp -sn uac_pcap -s 9999 -r 10 -l 80 -d 60000 -mi >>>> >>> 10.10.10.1 -i >>> >>>> 10.10.10.1 -mp 25000 10.10.10.254:5060 >>>> >>> >>> >>>> The dialplan on FS is very simple: >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>> data="sofia/external/9999 at 10.10.20.100 >>>> >>> >> >"/> >>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> FreeSWITCH Version 1.0.trunk (11560M) Started. >>>> Crash Protection [Disabled] >>>> Max Sessions[1000] >>>> Session Rate[100] >>>> SQL [Enabled] >>>> >>>> >>>> The test is very simple: sipp dial 9999 that matches in my >>>> >>> FS dialplan >>> >>>> and this is bridged to an other FS machine playing music on >>>> >>> hold. >>> >>>> When I launch "top" I see after 30 to 40 s that FS consumes all >>>> the CPU >>>> ressources (with a mean of 50-60 % before), with 80 calls. >>>> When I set 70 calls, I have to wait 70-80 s before seeing >>>> >>> the same >>> >>>> issue. >>>> >>>> Presence is set to false on the 2 profile. >>>> >>>> I have the same issue with FS 1.0.2 that' s why I tried FS >>>> >>> 11560. >>> >>>> When I use the FS machine as a router to test the packet per >>>> >>> second >>> >>>> performance, I'm reaching 100Mbps with 8000pps in each >>>> >>> direction (from >>> >>>> vlan 55 to vlan56) with less than 12% CPU. So that I don't think >>>> there's >>>> an issue with the network. >>>> >>>> Here is an "mpstat -P ALL 1" to show you what's happening >>>> >>> suddenly >>> >>>> with >>>> 70 bridge calls: >>>> 12:31:26 CPU %user %nice %sys %iowait %irq %soft >>>> %steal %idle intr/s >>>> 12:31:27 all 3,00 0,00 3,00 0,00 1,00 4,00 >>>> 0,00 89,00 6241,00 >>>> 12:31:27 0 3,00 0,00 3,00 0,00 1,00 4,00 >>>> 0,00 89,00 6241,00 >>>> >>>> 12:31:27 CPU %user %nice %sys %iowait %irq %soft >>>> %steal %idle intr/s >>>> 12:31:28 all 14,14 0,00 56,57 0,00 2,02 5,05 >>>> 0,00 22,22 6035,35 >>>> 12:31:28 0 14,14 0,00 56,57 0,00 2,02 5,05 >>>> 0,00 22,22 6035,35 >>>> >>>> 12:31:28 CPU %user %nice %sys %iowait %irq %soft >>>> %steal %idle intr/s >>>> 12:31:29 all 24,75 0,00 67,33 0,00 0,99 6,93 >>>> 0,00 0,00 5483,17 >>>> 12:31:29 0 24,75 0,00 67,33 0,00 0,99 6,93 >>>> 0,00 0,00 5483,17 >>>> >>>> >>>> The CPU is going from 89% idle to 0% in less than 2 seconds. >>>> >>>> I know that I don't have to expect too much from this kind of >>>> hardware, >>>> but it seems strange that the CPU power vanished so suddenly. >>>> >>>> Thanks a lot for the guys that have read this long mail :p >>>> >>>> kind regards, >>>> rod >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> >>> >>> >>>> >>> >>> > >>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> >>>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >>>> http://www.freeswitch.org >>>> >>>> >>>> >>>> >>>> -- >>>> Anthony Minessale II >>>> >>>> FreeSWITCH http://www.freeswitch.org/ >>>> ClueCon http://www.cluecon.com/ >>>> >>>> AIM: anthm >>>> MSN:anthony_minessale at hotmail.com >>>> >>> >>> >>>> >>> >>> > >>> >>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>>> >>> >>> >>>> >>> >>> > >>> >>>> IRC: irc.freenode.net >>>> >>> #freeswitch >>> >>>> FreeSWITCH Developer Conference >>>> sip:888 at conference.freeswitch.org >>>> >>> >>> >>>> >>> >>> > >>> >>>> iax:guest at conference.freeswitch.org/888 >>>> >>> >>> >>>> >>>> googletalk:conf+888 at conference.freeswitch.org >>>> >>> >>> >>>> >>> >>> > >>> >>>> pstn:213-799-1400 >>>> >>>> >>> ------------------------------------------------------------------------ >>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> >>> >>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> >>>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >>>> http://www.freeswitch.org >>>> >>>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> ------------------------------------------------------------------------ >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > From saeedahmad1981 at gmail.com Mon Feb 2 02:21:06 2009 From: saeedahmad1981 at gmail.com (Saeed Ahmed) Date: Mon, 2 Feb 2009 11:21:06 +0100 Subject: [Freeswitch-users] Strange Performance when using as SBC In-Reply-To: <4986C42C.7030700@laposte.net> References: <4986C42C.7030700@laposte.net> Message-ID: <1C31435E19AB4E57981CF2AAC2CA74AA@SaeedLaptop> Hi Rod, Could you please share how you configured Sipp & FS to create a test environment? Especially the dial plan, sofia settings etc..., actually I am a newbie. I want to test it on a single FS machine. Kind Regards Saeed -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of rod Sent: Monday, February 02, 2009 11:00 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Strange Performance when using as SBC Hi Ken, Jay, thanks for pointing to proxy media, I will test. Ken, you are right, I was brain damaged (a stupid mistake) when setting INFO cause this kind of level could be very verbose. I'm switching to CRIT or ERR. Thanks guys, rod. thanks for Ken Rice wrote: > If you don't have to transcode, using proxy media mode will still save you > some CPU time. This is 1/2 way between bypass media and the default media > interactive mode. The other draw back to this mode is if you are using FS to > clean up RTP and DTMF you loose those functions but they are not needed in > most use cases. > > As far as the log level goes, I found that once I had things stable setting > the loglevel to helped a good deal... Info is probably a bit too high of a > loglevel I would probably go for CRIT or ERR (2 or 1 respectively) if you > insist on leaving logging turned on... On a busy system these can and will > generate a good deal of activity (and disk IO if using mod_logfile) > > Ken > > > >> From: rod >> Reply-To: >> Date: Mon, 02 Feb 2009 11:36:35 +0400 >> To: >> Subject: Re: [Freeswitch-users] Strange Performance when using as SBC >> >> Hi Ken, >> >> 1) I'd like to use FS to hide topology, so bypass media is not possible >> 2) done >> 3) done >> 4) not used >> 5) i'm using this ins switch.xml -> > value="info"/>, if you think an other log level is more suitable. >> >> Regarding logging, I can see in console and in the freeswitch.log that >> there is still a lot of NOTICE logging, see below: >> 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:960 >> switch_core_session_thread() Session 8721 >> (sofia/internal/sipp at 10.10.10.1:5060) Ended >> 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:962 >> switch_core_session_thread() Close Channel >> sofia/internal/sipp at 10.10.10.1:5060 [CS_HANGUP] >> 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:960 >> switch_core_session_thread() Session 8722 >> (sofia/external/9998 at 10.10.20.100) Ended >> 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:962 >> switch_core_session_thread() Close Channel >> sofia/external/9998 at 10.10.20.100 [CS_HANGUP] >> 2009-02-02 08:33:56 [NOTICE] sofia.c:3164 sofia_handle_sip_i_state() >> Channel [sofia/external/9998 at 10.10.20.100] has been answered >> 2009-02-02 08:33:56 [WARNING] mod_sofia.c:740 sofia_read_frame() >> Changing codec ptime to 30. I bet you have a linksys/sipura =D >> >> Do you have any idea where I can switch off this kind of logging. I >> thought it should be in /dialplan/internal.xml, but I see that in >> internal.xml -> >> >> thanks a lot for your suggestion. >> >> regards, >> rod >> >> Ken Rice wrote: >> >>> Dont forget there are several things you can do to increase performance... >>> >>> 1) where possible use bypass media or media proxy modes >>> 2) mount freeswitch/db as a ram drive (if you are using voicemail with >>> the internal FS DBs you'll need a way to make this persistant across >>> reboots) >>> 3) see the wiki for setting reasonable ulimits >>> 4) (this is my oppinion others may vary) dont use mod_cdr_csv >>> 5) turn off (or reduce logging) in switch.conf.xml >>> >>> all of these thing can greatly improve performance. >>> >>> On Mon, Feb 2, 2009 at 1:04 AM, rod >> > wrote: >>> >>> Thanks Anthony, >>> >>> the setup is like this: >>> >>> sipp server ---- FS 1 ---- FS2 >>> >>> FS1 is the AMD CPU that has only one extension in dialplan that >>> bridges >>> 9999 to FS2. 9999 is the first extension in FS2 dialplan that >>> plays moh, >>> FS2 has no CPU pbm. >>> >>> FS1 is maxing out at 60 bridged calls without your option -hp. >>> >>> Using -hp, I'm now able to bridge 200 concurrent calls (a great >>> improvement) and the system is still reactive. CPU load is high >>> but not >>> 100% and as the system responds well, I think that doesn't matter. The >>> 2GB of memory are completely consumed (top command shows 700MB for FS >>> process). >>> >>> I understand that FS1 server is not the best hardware platform, >>> and I'm >>> waiting for new 4 cores server for testing. >>> I will update those numbers when testing with the new hardware. >>> >>> regards, >>> rod. >>> >>> Anthony Minessale wrote: >>> >>>> Which of the 2 machines has the load issue? You said it was one box >>>> calling the other. >>>> >>>> You have 2 major things against you, single CPU and AMD, but you >>>> should at least be able to get in the vicinity of 800-1000 calls >>>> >>> on a >>> >>>> box like that. >>>> >>>> Are you calling the default 9999? It's not really an appropriate >>>> extension for load testing. >>>> On the terminating box you should set up a manual extension that is >>>> the first one in the dial plan >>>> to play a wav file from preferably a ram disk or /tmp >>>> >>>> If you do plan on using this in production accept nothing less >>>> >>> than a >>> >>>> multi-core intel machine with at least 4 cores, the more cores the >>>> better because that parallel processing is where FS gets it's >>>> >>> atvantage. >>> >>>> >>>> On Fri, Jan 30, 2009 at 5:56 AM, rod >>> >>> >>> >>>> >> wrote: >>>> >>>> Dear list, >>>> >>>> I've been playing with freeswitch for some time (2 months) >>>> >>> and the >>> >>>> fact >>>> is that I'm very pleased with the functionnalities of this >>>> >>> software. >>> >>>> I'd like to use FS as a SBC handling media and I'm doing some >>>> tests with >>>> sipp to load the machine but I'm unable to bridge more than >>>> >>> 60 calls >>> >>>> without seeing the CPU being loaded at 100%. I'm sure >>>> >>> something is >>> >>>> going >>>> wrong with my setup but I'm unable to see what. >>>> >>>> The test machine has the following specs: >>>> Athlon XP 3500+ with 2GB of memory (I know this is not a >>>> >>> high end >>> >>>> machine :p) >>>> >>>> Freeswitch:/opt/freeswitch/log# cat /proc/cpuinfo >>>> processor : 0 >>>> vendor_id : AuthenticAMD >>>> cpu family : 15 >>>> model : 95 >>>> model name : AMD Athlon(tm) 64 Processor 3500+ >>>> stepping : 2 >>>> cpu MHz : 2199.973 >>>> cache size : 512 KB >>>> fpu : yes >>>> fpu_exception : yes >>>> cpuid level : 1 >>>> wp : yes >>>> flags : fpu vme de pse tsc msr pae mce cx8 apic >>>> >>> sep mtrr pge >>> >>>> mca cmov pat pse36 clflush mmx fxsr sse sse2 syscall nx mmxext >>>> fxsr_opt >>>> rdtscp lm 3dnowext 3dnow up rep_good pni cx16 lahf_lm svm >>>> >>> extapic >>> >>>> cr8_legacy >>>> bogomips : 4402.97 >>>> TLB size : 1024 4K pages >>>> clflush size : 64 >>>> cache_alignment : 64 >>>> address sizes : 40 bits physical, 48 bits virtual >>>> power management: ts fid vid ttp tm stc >>>> >>>> I installed FS on a fresh debian 64: >>>> Linux Freeswitch 2.6.26-1-amd64 #1 SMP Sat Jan 10 17:57:00 >>>> >>> UTC 2009 >>> >>>> x86_64 GNU/Linux >>>> >>>> I set the ulimit parameters like those on the website: >>>> freeswitch at internal> ... >>>> Freeswitch:/opt/free-svn/bin# ulimit -a >>>> core file size (blocks, -c) unlimited >>>> data seg size (kbytes, -d) unlimited >>>> scheduling priority (-e) 0 >>>> file size (blocks, -f) unlimited >>>> pending signals (-i) unlimited >>>> max locked memory (kbytes, -l) unlimited >>>> max memory size (kbytes, -m) unlimited >>>> open files (-n) 999999 >>>> pipe size (512 bytes, -p) 8 >>>> POSIX message queues (bytes, -q) unlimited >>>> real-time priority (-r) 0 >>>> stack size (kbytes, -s) 244 >>>> cpu time (seconds, -t) unlimited >>>> max user processes (-u) unlimited >>>> virtual memory (kbytes, -v) unlimited >>>> file locks (-x) unlimited >>>> >>>> >>>> My network setup is the following: >>>> >>>> SIPP machine (10.10.10.1/24)----------------vlan >>>> >>> >>> >>>> 55 >>>> ----------(10.10.10.254/24 >>>> >>> ) FS >>> >>>> (10.10.20.254/24)-------------- >>>> >>> >>> >>>> vlan56 >>>> -------------------(10.10.20.100/24 >>>> >>> ) >>> >>>> OTHER STOCK FS >>>> >>>> >>>> I launched sipp with: >>>> sipp -sn uac_pcap -s 9999 -r 10 -l 80 -d 60000 -mi >>>> >>> 10.10.10.1 -i >>> >>>> 10.10.10.1 -mp 25000 10.10.10.254:5060 >>>> >>> >>> >>>> The dialplan on FS is very simple: >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>> data="sofia/external/9999 at 10.10.20.100 >>>> >>> >> >"/> >>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> FreeSWITCH Version 1.0.trunk (11560M) Started. >>>> Crash Protection [Disabled] >>>> Max Sessions[1000] >>>> Session Rate[100] >>>> SQL [Enabled] >>>> >>>> >>>> The test is very simple: sipp dial 9999 that matches in my >>>> >>> FS dialplan >>> >>>> and this is bridged to an other FS machine playing music on >>>> >>> hold. >>> >>>> When I launch "top" I see after 30 to 40 s that FS consumes all >>>> the CPU >>>> ressources (with a mean of 50-60 % before), with 80 calls. >>>> When I set 70 calls, I have to wait 70-80 s before seeing >>>> >>> the same >>> >>>> issue. >>>> >>>> Presence is set to false on the 2 profile. >>>> >>>> I have the same issue with FS 1.0.2 that' s why I tried FS >>>> >>> 11560. >>> >>>> When I use the FS machine as a router to test the packet per >>>> >>> second >>> >>>> performance, I'm reaching 100Mbps with 8000pps in each >>>> >>> direction (from >>> >>>> vlan 55 to vlan56) with less than 12% CPU. So that I don't think >>>> there's >>>> an issue with the network. >>>> >>>> Here is an "mpstat -P ALL 1" to show you what's happening >>>> >>> suddenly >>> >>>> with >>>> 70 bridge calls: >>>> 12:31:26 CPU %user %nice %sys %iowait %irq %soft >>>> %steal %idle intr/s >>>> 12:31:27 all 3,00 0,00 3,00 0,00 1,00 4,00 >>>> 0,00 89,00 6241,00 >>>> 12:31:27 0 3,00 0,00 3,00 0,00 1,00 4,00 >>>> 0,00 89,00 6241,00 >>>> >>>> 12:31:27 CPU %user %nice %sys %iowait %irq %soft >>>> %steal %idle intr/s >>>> 12:31:28 all 14,14 0,00 56,57 0,00 2,02 5,05 >>>> 0,00 22,22 6035,35 >>>> 12:31:28 0 14,14 0,00 56,57 0,00 2,02 5,05 >>>> 0,00 22,22 6035,35 >>>> >>>> 12:31:28 CPU %user %nice %sys %iowait %irq %soft >>>> %steal %idle intr/s >>>> 12:31:29 all 24,75 0,00 67,33 0,00 0,99 6,93 >>>> 0,00 0,00 5483,17 >>>> 12:31:29 0 24,75 0,00 67,33 0,00 0,99 6,93 >>>> 0,00 0,00 5483,17 >>>> >>>> >>>> The CPU is going from 89% idle to 0% in less than 2 seconds. >>>> >>>> I know that I don't have to expect too much from this kind of >>>> hardware, >>>> but it seems strange that the CPU power vanished so suddenly. >>>> >>>> Thanks a lot for the guys that have read this long mail :p >>>> >>>> kind regards, >>>> rod >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> >>> >>> >>>> >>> >>> > >>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> >>>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >>>> http://www.freeswitch.org >>>> >>>> >>>> >>>> >>>> -- >>>> Anthony Minessale II >>>> >>>> FreeSWITCH http://www.freeswitch.org/ >>>> ClueCon http://www.cluecon.com/ >>>> >>>> AIM: anthm >>>> MSN:anthony_minessale at hotmail.com >>>> >>> >>> >>>> >>> >>> > >>> >>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>>> >>> >>> >>>> >>> >>> > >>> >>>> IRC: irc.freenode.net >>>> >>> #freeswitch >>> >>>> FreeSWITCH Developer Conference >>>> sip:888 at conference.freeswitch.org >>>> >>> >>> >>>> >>> >>> > >>> >>>> iax:guest at conference.freeswitch.org/888 >>>> >>> >>> >>>> >>>> googletalk:conf+888 at conference.freeswitch.org >>>> >>> >>> >>>> >>> >>> > >>> >>>> pstn:213-799-1400 >>>> >>>> >>> ------------------------------------------------------------------------ >>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> >>> >>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> >>>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >>>> http://www.freeswitch.org >>>> >>>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> ------------------------------------------------------------------------ >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From sias at cpdata.co.za Mon Feb 2 02:24:59 2009 From: sias at cpdata.co.za (Sias Mey) Date: Mon, 2 Feb 2009 12:24:59 +0200 Subject: [Freeswitch-users] Conference dialing and uuid In-Reply-To: <0F863750-5C77-4CD9-ADC6-CFDDF2B149B4@freeswitch.org> References: <20090130103944.GA23536@cpdata.co.za> <20090130133315.GB23536@cpdata.co.za> <191c3a030901300605r78dd8943w7fb1075ef851e1d@mail.gmail.com> <20090202085157.GA3555@cpdata.co.za> <0F863750-5C77-4CD9-ADC6-CFDDF2B149B4@freeswitch.org> Message-ID: <20090202102459.GA4179@cpdata.co.za> Aaah ok. Thanks for clearing that up. So using loopback is still the only real workable sollution for me, since that generates ringback from and alternative endpoint and plays it into the conference. I might play with some javascript that streams ring into the channel eventually but for now the string comparisons at least get me the right uuid. Thank you again, Sias On Mon, Feb 02, 2009 at 02:55:42AM -0600, Brian West wrote: > You can't get ringback dialing out from a conference its not possible > as it is now. > > /b > > On Feb 2, 2009, at 2:51 AM, Sias Mey wrote: > > Yes ... yes indeed I can. > That works quite a bit better than generating 4 channels and getting > massively confused with what uuid does what... but now im stuck > without > ringback again :-(. > In my conference dial string I send: > {ringback=\'%(400,200,400,450)\',transfer_ringback=\'%(400,200,400,4 > 50)\', > .... [1]}sofia/internal/1001 at xxx.xxx.xxx.xxx > A dump of all the channel variables shows ringback is set to > %25(400,200,400,450)%3B%25(400,2200,400,450) > %25(400,200,400,450)%3B%25(400,2200,400,450) transfer_ringback> > This seems ok to me but I still dont get any ringback. > Thanks again for answering all the anoying questions from the same > guy > :-P, > Sias > > References > > 1. mailto:}sofia/internal/1001 at xxx.xxx.xxx.xxx > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Mon Feb 2 02:29:25 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 2 Feb 2009 04:29:25 -0600 Subject: [Freeswitch-users] Conference dialing and uuid In-Reply-To: <20090202102459.GA4179@cpdata.co.za> References: <20090130103944.GA23536@cpdata.co.za> <20090130133315.GB23536@cpdata.co.za> <191c3a030901300605r78dd8943w7fb1075ef851e1d@mail.gmail.com> <20090202085157.GA3555@cpdata.co.za> <0F863750-5C77-4CD9-ADC6-CFDDF2B149B4@freeswitch.org> <20090202102459.GA4179@cpdata.co.za> Message-ID: <9FE1733C-A46A-4A17-8AEE-077DE8010235@freeswitch.org> Loopback will not work in that case either. If the far end plays ringback inband you should hear that if you use the conference dial api call. /b On Feb 2, 2009, at 4:24 AM, Sias Mey wrote: > Aaah ok. > > Thanks for clearing that up. > > So using loopback is still the only real workable sollution for me, > since that generates ringback from and alternative endpoint and > plays it > into the conference. > > I might play with some javascript that streams ring into the channel > eventually but for now the string comparisons at least get me the > right > uuid. > > Thank you again, > Sias From leon at scarlet-internet.nl Mon Feb 2 03:42:05 2009 From: leon at scarlet-internet.nl (Leon de Rooij) Date: Mon, 2 Feb 2009 12:42:05 +0100 Subject: [Freeswitch-users] LDAP Integration In-Reply-To: <4983C258.6080705@skopis.com> References: <49417123.10709@ydeasolutions.com.br> <49417538.9040203@ydeasolutions.com.br> <200812120842.00808.hads@nice.net.nz> <49418790.60001@ydeasolutions.com.br> <87f2f3b90812111241q3b16b307lbf4d1251c7d8aad7@mail.gmail.com> <494198F3.10806@ydeasolutions.com.br> <4947A61F.6060806@ydeasolutions.com.br> <49485AD1.5070708@skopis.com> <4949025F.9040008@ydeasolutions.com.br> <49621532.5080003@ydeasolutions.com.br> <4962D64D.3080809@skopis.com> <2DCF79D8-E2B5-43EB-93D1-EED92506E8DF@scarlet-internet.nl> <4983C258.6080705@skopis.com> Message-ID: <87556643-1B94-40AE-B84C-871AEA593593@scarlet-internet.nl> On Jan 31, 2009, at 4:15 AM, John Skopis (Lists) wrote: > Leon de Rooij wrote: >> Hi John, >> >> I've been trying to get your mod_xml_ldap module running, but didn't >> get very far yet.. >> >> What is the official way to get the module built ? >> > > The official way to build all fs modules is to uncomment the entry in > modules.conf. > > If you want to build a specific module there are targets > > make mod_name-clean > make mod_name-install Thanks, I'll try that. > as for mod_xml_ldap, I really do not feel that it is as quality as I > would expect a production quality module to be. I understand, it's just that I'm very interested in it as we're using ldap everywhere over here. >> I tried modifying trunk/freeswitch.spec so that >> >> XML_INT_MODULES contains xml_int/mod_xml_ldap >> >> There's also a directories/mod_ldap in DISABLED_MODULES in the same >> file, but I don't suppose it's necessary to enable it, or is it ? >> > > mod_ldap is a separate module, implementing the directory interface, > not > to be confused with the "directory", which is queried for user + > domain > configuration (e.g., conf/directory/default.xml). > > perhaps it should be renamed to mod_dbi? > >> The mod_xml_ldap doesn't get built by running make make or dpkg- >> buildpackage from trunk/ >> >> Also I tried building it from the module directory itself, but then I >> get the following error: >> >> fsbuilder at sv:~/trunk/src/mod/xml_int/mod_xml_ldap$ make >> Compiling mod_xml_ldap.c... >> cc1: warnings being treated as errors >> mod_xml_ldap.c: In function 'xml_ldap_search': >> mod_xml_ldap.c:356: warning: cast from pointer to integer of >> different >> size >> make[1]: *** [mod_xml_ldap.o] Error 1 >> make: *** [all] Error 1 >> > > > > I have been working on a new module called mod_entity that works off a > simple description of an xml entitiy (domain, user, extension, > condition, action, anti-action currently) querying a db backend via > the > directory interface for fields used to build the entity. It still > needs > a bit of work but I am hoping to get a patch together this weekend. I > will post it to the freeswitch-dev list asking for comments. > > Off the top of my head at least the wishlist TODO is: > > implement connection pooling for mod_directory > > implement a cache either as a module used by an xml_int mod or in > switch_xml to cache a switch_xml_t > > >> (Also I had to apt-get install libsasl2 libsasl2-dev, otherwise make >> from this dir errored with missing sasl/sasl.h) >> >> Can you see what I'm doing wrong ? >> >> (I'm using svn rev 11560) >> >> thanks & regards, >> >> Leon >> >> On Jan 6, 2009, at 4:55 AM, John Skopis (Lists) wrote: >> >>> Vinicius Kobashi wrote: >>>> hi ppl. >>>> >>>> i tried hard to make it work, but still i couldnt find a complete >>>> openldap scheme that provides these information, and i still >>>> could't >>>> find out where to put these configuration... >>>> >>>> can anyone help me? >>>> >>>> thankz! >>>> >>>> vinicius escreveu: >>>>> thankz! >>>>> >>>>> ill set my openldap to provide these information.. >>>>> >>>>> but these about these binding settings... where should i set them? >>>>> >>>>> best regards >>>>> >>>>> John Skopis (Lists) wrote: >>>>>> vinicius wrote: >>>>>> >>>>>>> hi ppl.. i tried to find something at google, but i couldnt >>>>>>> manage to find >>>>>>> anything. >>>>>>> i still dont know what to do to make the mod_xml_ldap work. >>>>>>> i couldnt find information about how to build a config file for >>>>>>> the >>>>>>> module, and where to store it... >>>>>>> >>>>>>> can anyone give me a help? >>>>>>> >>>>>>> >>>>>> Be advised mod_xml_ldap is probably not production quality and >>>>>> will >>>>>> undoubtedly change, eventually at least. >>>>>> >>>>>> Here is what I used once: >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>> bindings="configuration"/> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> which should/probably/might work with ldap objects like these: >>>>>> >>>>>> dn: cn=John Skopis,ou=people,dc=example >>>>>> objectClass: person >>>>>> objectClass: inetOrgPerson >>>>>> objectClass: organizationalPerson >>>>>> objectClass: FreeSWITCH-Exten-Object >>>>>> objectClass: top >>>>>> cn: John Skopis >>>>>> sn: Skopis >>>>>> givenName: John >>>>>> FSid: 1001 >>>>>> FSmailbox: 1001 >>>>>> FSpassword: 1234 >>>>>> FSvm-password: 1001 >>>>>> FSemail-addr: john+fs at skopis.com >>>>>> FSvm-email-all-messages: TRUE >>>>>> FSvm-delete-file: TRUE >>>>>> FSvm-attach-file: TRUE >>>>>> >>>>>> dn: SIPIdentityUserName=1001,ou=h350,dc=example >>>>>> objectClass: person >>>>>> objectClass: SIPIdentity >>>>>> objectClass: top >>>>>> cn: 1001 >>>>>> sn: 1001 >>>>>> SIPIdentitySIPURI: sip:1001 at 172.16.75.129 >>>>>> SIPIdentityRegistrarAddress: 172.16.75.128 >>>>>> SIPIdentityProxyAddress: 172.16.75.128 >>>>>> SIPIdentityPassword: 1234 >>>>>> SIPIdentityUserName: 1001 >>>>>> SIPIdentityServiceLevel: premium >>>>>> >>>>>> >>> Again, the module is not production quality. Hopefully I will >>> conjurer >>> the time and know-how to put something decent together eventually. >>> >>> To load configuration for any fs module you need to define the XML >>> configuration element under the section "configuration". >>> >>> A good starting point is the file >>> $PREFIX/conf/freeswitch.xml >>> >>> http://wiki.freeswitch.org/wiki/Freeswitch.xml >>> >>> Also take a look at $PREFIX/logs/freeswitch.xml.fsxml >>> >>> to load mod_xml_ldap you would need to add something like this to >>> modules.conf.xml >>> >>> >>> >>> and create an xml_ldap.conf.xml in >>> $PREFIX/autoload_configs/xml_ldap.conf.xml >>> >>> >>> ... >>> >>> >>> The ITU is doing some work called h.350: >>> http://www.itu.int/ITU-T/studygroups/com16/h350/index.html >>> >>> Here is what I was working with: >>> attributetype ( 1.3.6.1.4.1.65535.2.1.1 NAME 'FSid' >>> DESC 'FreeSWITCH Extension ID' >>> EQUALITY caseIgnoreIA5Match >>> SYNTAX 1.3.6.1.4.1.1466.115.121.1.26 ) >>> >>> attributetype ( 1.3.6.1.4.1.65535.2.1.2 NAME 'FSmailbox' >>> DESC 'FreeSWITCH Extension Mailbox' >>> EQUALITY caseIgnoreIA5Match >>> SYNTAX 1.3.6.1.4.1.1466.115.121.1.26 ) >>> >>> attributetype ( 1.3.6.1.4.1.65535.2.1.3 NAME 'FSpassword' >>> DESC 'FreeSWITCH Password' >>> EQUALITY caseExactIA5Match >>> SYNTAX 1.3.6.1.4.1.1466.115.121.1.26 >>> SINGLE-VALUE ) >>> >>> attributetype ( 1.3.6.1.4.1.65535.2.1.4 NAME 'FSa1hash' >>> DESC 'FreeSWITCH Crypted Password' >>> EQUALITY caseExactIA5Match >>> SYNTAX 1.3.6.1.4.1.1466.115.121.1.26 >>> SINGLE-VALUE ) >>> >>> attributetype ( 1.3.6.1.4.1.65535.2.1.5 NAME 'FSvm-password' >>> DESC 'FreeSWITCH VoiceMail Password' >>> EQUALITY integerMatch >>> SYNTAX 1.3.6.1.4.1.1466.115.121.1.27 >>> SINGLE-VALUE ) >>> >>> attributetype ( 1.3.6.1.4.1.65535.2.1.6 NAME 'FSemail-addr' >>> DESC 'E-mail address to send voicemail' >>> EQUALITY caseIgnoreIA5Match >>> SYNTAX 1.3.6.1.4.1.1466.115.121.1.26 ) >>> >>> attributetype ( 1.3.6.1.4.1.65535.2.1.7 NAME 'FSvm-email-all- >>> messages' >>> DESC 'FreeSWITCH Email All Mesages' >>> EQUALITY booleanMatch >>> SYNTAX 1.3.6.1.4.1.1466.115.121.1.7 >>> SINGLE-VALUE ) >>> >>> attributetype ( 1.3.6.1.4.1.65535.2.1.8 NAME 'FSvm-delete-file' >>> DESC 'FreeSWITCH VoiceMail Delete File' >>> EQUALITY booleanMatch >>> SYNTAX 1.3.6.1.4.1.1466.115.121.1.7 >>> SINGLE-VALUE ) >>> >>> attributetype ( 1.3.6.1.4.1.65535.2.1.9 NAME 'FSvm-attach-file' >>> DESC 'FreeSWITCH VoiceMail Attach file' >>> EQUALITY booleanMatch >>> SYNTAX 1.3.6.1.4.1.1466.115.121.1.7 >>> SINGLE-VALUE ) >>> >>> >>> >>> >>> >>> objectclass ( 1.3.6.1.4.1.65535.2.2.1 NAME 'FreeSWITCH-Exten-Object' >>> SUP top AUXILIARY >>> DESC '%obj_desc%' >>> MUST ( FSid $ FSpassword ) >>> MAY ( FSmailbox $ FSa1hash $ FSvm-password $ FSemail-addr $ >>> FSvm-email-all-messages $ FSvm-delete-file $ FSvm-attach-file ) ) >>> >>> hth >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From pmhshz at gmail.com Mon Feb 2 04:07:57 2009 From: pmhshz at gmail.com (shehzad p) Date: Mon, 2 Feb 2009 04:07:57 -0800 (PST) Subject: [Freeswitch-users] Call Variable not available when call hangup Message-ID: <21788550.post@talk.nabble.com> Hi all, I need to process some CDR variables in Dialplan, like call duration, Answered time etc. but when I place info application after bridge, it is not listing them properly as below: =========================================== Caller-Channel-Created-Time: [1233573341672157] Caller-Channel-Answered-Time: [1233573342712939] Caller-Channel-Hangup-Time: [0] ========================================== Here Hangup time is 0, So how can I find actual values? --I know that we can use xml_cdr or cdr_csv, but my current need is to get those values from dialplan itself so that can be passed to some script... thanks, msp -- View this message in context: http://www.nabble.com/Call-Variable-not-available-when-call-hangup-tp21788550p21788550.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From krice at freeswitch.org Mon Feb 2 04:11:28 2009 From: krice at freeswitch.org (Ken Rice) Date: Mon, 02 Feb 2009 06:11:28 -0600 Subject: [Freeswitch-users] Call Variable not available when call hangup In-Reply-To: <21788550.post@talk.nabble.com> Message-ID: As I told you on IRC, the call is not completed at that stage... So there is no hangup time... You must post process the call or figure out your own start answer and stop times.... > From: shehzad p > Reply-To: > Date: Mon, 2 Feb 2009 04:07:57 -0800 (PST) > To: > Subject: [Freeswitch-users] Call Variable not available when call hangup > > > Hi all, > > I need to process some CDR variables in Dialplan, like call duration, > Answered time etc. > but when I place info application after bridge, it is not listing them > properly as below: > =========================================== > Caller-Channel-Created-Time: [1233573341672157] > Caller-Channel-Answered-Time: [1233573342712939] > Caller-Channel-Hangup-Time: [0] > ========================================== > Here Hangup time is 0, So how can I find actual values? > > --I know that we can use xml_cdr or cdr_csv, but my current need is to get > those values from dialplan itself so that can be passed to some script... > > > thanks, > msp > -- > View this message in context: > http://www.nabble.com/Call-Variable-not-available-when-call-hangup-tp21788550p > 21788550.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From kawarod at laposte.net Mon Feb 2 04:36:44 2009 From: kawarod at laposte.net (rod) Date: Mon, 02 Feb 2009 16:36:44 +0400 Subject: [Freeswitch-users] Strange Performance when using as SBC In-Reply-To: <4986C42C.7030700@laposte.net> References: <4986C42C.7030700@laposte.net> Message-ID: <4986E8DC.20707@laposte.net> Some update: - I switched log level to "err" - I'm now using proxy-media - and I erased the directive answer in the dialplan (useless and seems that it consumes lots of CPU, don't know why) the dialplan now looks like this: instead of The box is now able to bridge 300 calls with 20-30% of free CPU. I will run a long term test. I see this error in the log and don't understand exactly if somebody could help (I'm running latest trunk 11592M): 2009-02-02 13:29:54 [ERR] switch_core_io.c:117 switch_core_session_read_frame() sofia/external/9998 at 10.10.20.100 has no read codec. regards, rodrigue rod wrote: > Hi Ken, Jay, > > thanks for pointing to proxy media, I will test. > > Ken, you are right, I was brain damaged (a stupid mistake) when setting > INFO cause this kind of level could be very verbose. I'm switching to > CRIT or ERR. > > Thanks guys, > rod. > > thanks for > > Ken Rice wrote: > >> If you don't have to transcode, using proxy media mode will still save you >> some CPU time. This is 1/2 way between bypass media and the default media >> interactive mode. The other draw back to this mode is if you are using FS to >> clean up RTP and DTMF you loose those functions but they are not needed in >> most use cases. >> >> As far as the log level goes, I found that once I had things stable setting >> the loglevel to helped a good deal... Info is probably a bit too high of a >> loglevel I would probably go for CRIT or ERR (2 or 1 respectively) if you >> insist on leaving logging turned on... On a busy system these can and will >> generate a good deal of activity (and disk IO if using mod_logfile) >> >> Ken >> >> >> >> >>> From: rod >>> Reply-To: >>> Date: Mon, 02 Feb 2009 11:36:35 +0400 >>> To: >>> Subject: Re: [Freeswitch-users] Strange Performance when using as SBC >>> >>> Hi Ken, >>> >>> 1) I'd like to use FS to hide topology, so bypass media is not possible >>> 2) done >>> 3) done >>> 4) not used >>> 5) i'm using this ins switch.xml -> >> value="info"/>, if you think an other log level is more suitable. >>> >>> Regarding logging, I can see in console and in the freeswitch.log that >>> there is still a lot of NOTICE logging, see below: >>> 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:960 >>> switch_core_session_thread() Session 8721 >>> (sofia/internal/sipp at 10.10.10.1:5060) Ended >>> 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:962 >>> switch_core_session_thread() Close Channel >>> sofia/internal/sipp at 10.10.10.1:5060 [CS_HANGUP] >>> 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:960 >>> switch_core_session_thread() Session 8722 >>> (sofia/external/9998 at 10.10.20.100) Ended >>> 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:962 >>> switch_core_session_thread() Close Channel >>> sofia/external/9998 at 10.10.20.100 [CS_HANGUP] >>> 2009-02-02 08:33:56 [NOTICE] sofia.c:3164 sofia_handle_sip_i_state() >>> Channel [sofia/external/9998 at 10.10.20.100] has been answered >>> 2009-02-02 08:33:56 [WARNING] mod_sofia.c:740 sofia_read_frame() >>> Changing codec ptime to 30. I bet you have a linksys/sipura =D >>> >>> Do you have any idea where I can switch off this kind of logging. I >>> thought it should be in /dialplan/internal.xml, but I see that in >>> internal.xml -> >>> >>> thanks a lot for your suggestion. >>> >>> regards, >>> rod >>> >>> Ken Rice wrote: >>> >>> >>>> Dont forget there are several things you can do to increase performance... >>>> >>>> 1) where possible use bypass media or media proxy modes >>>> 2) mount freeswitch/db as a ram drive (if you are using voicemail with >>>> the internal FS DBs you'll need a way to make this persistant across >>>> reboots) >>>> 3) see the wiki for setting reasonable ulimits >>>> 4) (this is my oppinion others may vary) dont use mod_cdr_csv >>>> 5) turn off (or reduce logging) in switch.conf.xml >>>> >>>> all of these thing can greatly improve performance. >>>> >>>> On Mon, Feb 2, 2009 at 1:04 AM, rod >>> > wrote: >>>> >>>> Thanks Anthony, >>>> >>>> the setup is like this: >>>> >>>> sipp server ---- FS 1 ---- FS2 >>>> >>>> FS1 is the AMD CPU that has only one extension in dialplan that >>>> bridges >>>> 9999 to FS2. 9999 is the first extension in FS2 dialplan that >>>> plays moh, >>>> FS2 has no CPU pbm. >>>> >>>> FS1 is maxing out at 60 bridged calls without your option -hp. >>>> >>>> Using -hp, I'm now able to bridge 200 concurrent calls (a great >>>> improvement) and the system is still reactive. CPU load is high >>>> but not >>>> 100% and as the system responds well, I think that doesn't matter. The >>>> 2GB of memory are completely consumed (top command shows 700MB for FS >>>> process). >>>> >>>> I understand that FS1 server is not the best hardware platform, >>>> and I'm >>>> waiting for new 4 cores server for testing. >>>> I will update those numbers when testing with the new hardware. >>>> >>>> regards, >>>> rod. >>>> >>>> Anthony Minessale wrote: >>>> >>>> >>>>> Which of the 2 machines has the load issue? You said it was one box >>>>> calling the other. >>>>> >>>>> You have 2 major things against you, single CPU and AMD, but you >>>>> should at least be able to get in the vicinity of 800-1000 calls >>>>> >>>>> >>>> on a >>>> >>>> >>>>> box like that. >>>>> >>>>> Are you calling the default 9999? It's not really an appropriate >>>>> extension for load testing. >>>>> On the terminating box you should set up a manual extension that is >>>>> the first one in the dial plan >>>>> to play a wav file from preferably a ram disk or /tmp >>>>> >>>>> If you do plan on using this in production accept nothing less >>>>> >>>>> >>>> than a >>>> >>>> >>>>> multi-core intel machine with at least 4 cores, the more cores the >>>>> better because that parallel processing is where FS gets it's >>>>> >>>>> >>>> atvantage. >>>> >>>> >>>>> On Fri, Jan 30, 2009 at 5:56 AM, rod >>>> >>>>> >>>> >>>> >>>> >>>>> >> wrote: >>>>> >>>>> Dear list, >>>>> >>>>> I've been playing with freeswitch for some time (2 months) >>>>> >>>>> >>>> and the >>>> >>>> >>>>> fact >>>>> is that I'm very pleased with the functionnalities of this >>>>> >>>>> >>>> software. >>>> >>>> >>>>> I'd like to use FS as a SBC handling media and I'm doing some >>>>> tests with >>>>> sipp to load the machine but I'm unable to bridge more than >>>>> >>>>> >>>> 60 calls >>>> >>>> >>>>> without seeing the CPU being loaded at 100%. I'm sure >>>>> >>>>> >>>> something is >>>> >>>> >>>>> going >>>>> wrong with my setup but I'm unable to see what. >>>>> >>>>> The test machine has the following specs: >>>>> Athlon XP 3500+ with 2GB of memory (I know this is not a >>>>> >>>>> >>>> high end >>>> >>>> >>>>> machine :p) >>>>> >>>>> Freeswitch:/opt/freeswitch/log# cat /proc/cpuinfo >>>>> processor : 0 >>>>> vendor_id : AuthenticAMD >>>>> cpu family : 15 >>>>> model : 95 >>>>> model name : AMD Athlon(tm) 64 Processor 3500+ >>>>> stepping : 2 >>>>> cpu MHz : 2199.973 >>>>> cache size : 512 KB >>>>> fpu : yes >>>>> fpu_exception : yes >>>>> cpuid level : 1 >>>>> wp : yes >>>>> flags : fpu vme de pse tsc msr pae mce cx8 apic >>>>> >>>>> >>>> sep mtrr pge >>>> >>>> >>>>> mca cmov pat pse36 clflush mmx fxsr sse sse2 syscall nx mmxext >>>>> fxsr_opt >>>>> rdtscp lm 3dnowext 3dnow up rep_good pni cx16 lahf_lm svm >>>>> >>>>> >>>> extapic >>>> >>>> >>>>> cr8_legacy >>>>> bogomips : 4402.97 >>>>> TLB size : 1024 4K pages >>>>> clflush size : 64 >>>>> cache_alignment : 64 >>>>> address sizes : 40 bits physical, 48 bits virtual >>>>> power management: ts fid vid ttp tm stc >>>>> >>>>> I installed FS on a fresh debian 64: >>>>> Linux Freeswitch 2.6.26-1-amd64 #1 SMP Sat Jan 10 17:57:00 >>>>> >>>>> >>>> UTC 2009 >>>> >>>> >>>>> x86_64 GNU/Linux >>>>> >>>>> I set the ulimit parameters like those on the website: >>>>> freeswitch at internal> ... >>>>> Freeswitch:/opt/free-svn/bin# ulimit -a >>>>> core file size (blocks, -c) unlimited >>>>> data seg size (kbytes, -d) unlimited >>>>> scheduling priority (-e) 0 >>>>> file size (blocks, -f) unlimited >>>>> pending signals (-i) unlimited >>>>> max locked memory (kbytes, -l) unlimited >>>>> max memory size (kbytes, -m) unlimited >>>>> open files (-n) 999999 >>>>> pipe size (512 bytes, -p) 8 >>>>> POSIX message queues (bytes, -q) unlimited >>>>> real-time priority (-r) 0 >>>>> stack size (kbytes, -s) 244 >>>>> cpu time (seconds, -t) unlimited >>>>> max user processes (-u) unlimited >>>>> virtual memory (kbytes, -v) unlimited >>>>> file locks (-x) unlimited >>>>> >>>>> >>>>> My network setup is the following: >>>>> >>>>> SIPP machine (10.10.10.1/24)----------------vlan >>>>> >>>>> >>>> >>>> >>>> >>>>> 55 >>>>> ----------(10.10.10.254/24 >>>>> >>>>> >>>> ) FS >>>> >>>> >>>>> (10.10.20.254/24)-------------- >>>>> >>>>> >>>> >>>> >>>> >>>>> vlan56 >>>>> -------------------(10.10.20.100/24 >>>>> >>>>> >>>> ) >>>> >>>> >>>>> OTHER STOCK FS >>>>> >>>>> >>>>> I launched sipp with: >>>>> sipp -sn uac_pcap -s 9999 -r 10 -l 80 -d 60000 -mi >>>>> >>>>> >>>> 10.10.10.1 -i >>>> >>>> >>>>> 10.10.10.1 -mp 25000 10.10.10.254:5060 >>>>> >>>>> >>>> >>>> >>>> >>>>> The dialplan on FS is very simple: >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>> data="sofia/external/9999 at 10.10.20.100 >>>>> >>>>> >>>> >>> >"/> >>>> >>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> FreeSWITCH Version 1.0.trunk (11560M) Started. >>>>> Crash Protection [Disabled] >>>>> Max Sessions[1000] >>>>> Session Rate[100] >>>>> SQL [Enabled] >>>>> >>>>> >>>>> The test is very simple: sipp dial 9999 that matches in my >>>>> >>>>> >>>> FS dialplan >>>> >>>> >>>>> and this is bridged to an other FS machine playing music on >>>>> >>>>> >>>> hold. >>>> >>>> >>>>> When I launch "top" I see after 30 to 40 s that FS consumes all >>>>> the CPU >>>>> ressources (with a mean of 50-60 % before), with 80 calls. >>>>> When I set 70 calls, I have to wait 70-80 s before seeing >>>>> >>>>> >>>> the same >>>> >>>> >>>>> issue. >>>>> >>>>> Presence is set to false on the 2 profile. >>>>> >>>>> I have the same issue with FS 1.0.2 that' s why I tried FS >>>>> >>>>> >>>> 11560. >>>> >>>> >>>>> When I use the FS machine as a router to test the packet per >>>>> >>>>> >>>> second >>>> >>>> >>>>> performance, I'm reaching 100Mbps with 8000pps in each >>>>> >>>>> >>>> direction (from >>>> >>>> >>>>> vlan 55 to vlan56) with less than 12% CPU. So that I don't think >>>>> there's >>>>> an issue with the network. >>>>> >>>>> Here is an "mpstat -P ALL 1" to show you what's happening >>>>> >>>>> >>>> suddenly >>>> >>>> >>>>> with >>>>> 70 bridge calls: >>>>> 12:31:26 CPU %user %nice %sys %iowait %irq %soft >>>>> %steal %idle intr/s >>>>> 12:31:27 all 3,00 0,00 3,00 0,00 1,00 4,00 >>>>> 0,00 89,00 6241,00 >>>>> 12:31:27 0 3,00 0,00 3,00 0,00 1,00 4,00 >>>>> 0,00 89,00 6241,00 >>>>> >>>>> 12:31:27 CPU %user %nice %sys %iowait %irq %soft >>>>> %steal %idle intr/s >>>>> 12:31:28 all 14,14 0,00 56,57 0,00 2,02 5,05 >>>>> 0,00 22,22 6035,35 >>>>> 12:31:28 0 14,14 0,00 56,57 0,00 2,02 5,05 >>>>> 0,00 22,22 6035,35 >>>>> >>>>> 12:31:28 CPU %user %nice %sys %iowait %irq %soft >>>>> %steal %idle intr/s >>>>> 12:31:29 all 24,75 0,00 67,33 0,00 0,99 6,93 >>>>> 0,00 0,00 5483,17 >>>>> 12:31:29 0 24,75 0,00 67,33 0,00 0,99 6,93 >>>>> 0,00 0,00 5483,17 >>>>> >>>>> >>>>> The CPU is going from 89% idle to 0% in less than 2 seconds. >>>>> >>>>> I know that I don't have to expect too much from this kind of >>>>> hardware, >>>>> but it seems strange that the CPU power vanished so suddenly. >>>>> >>>>> Thanks a lot for the guys that have read this long mail :p >>>>> >>>>> kind regards, >>>>> rod >>>>> >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> >>>>> >>>> >>>>> >>>> > >>>> >>>> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> >>>>> >>>>> >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> >>>> >>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> >>>>> >>>>> -- >>>>> Anthony Minessale II >>>>> >>>>> FreeSWITCH http://www.freeswitch.org/ >>>>> ClueCon http://www.cluecon.com/ >>>>> >>>>> AIM: anthm >>>>> MSN:anthony_minessale at hotmail.com >>>>> >>>>> >>>> >>>> >>>> >>>>> >>>> >>>>> >>>> > >>>> >>>> >>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>>>> >>>>> >>>> >>>> >>>> >>>>> >>>> >>>>> >>>> > >>>> >>>> >>>>> IRC: irc.freenode.net >>>>> >>>>> >>>> #freeswitch >>>> >>>> >>>>> FreeSWITCH Developer Conference >>>>> sip:888 at conference.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> >>>>> >>>> >>>>> >>>> > >>>> >>>> >>>>> iax:guest at conference.freeswitch.org/888 >>>>> >>>>> >>>> >>>> >>>> >>>>> >>>>> googletalk:conf+888 at conference.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> >>>>> >>>> >>>>> >>>> > >>>> >>>> >>>>> pstn:213-799-1400 >>>>> >>>>> >>>>> >>>> ------------------------------------------------------------------------ >>>> >>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> >>>>> >>>>> >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> >>>> >>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> ------------------------------------------------------------------------ >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > From pmhshz at gmail.com Mon Feb 2 04:53:16 2009 From: pmhshz at gmail.com (shehzad p) Date: Mon, 2 Feb 2009 04:53:16 -0800 (PST) Subject: [Freeswitch-users] Call Variable not available when call hangup In-Reply-To: <21788550.post@talk.nabble.com> References: <21788550.post@talk.nabble.com> Message-ID: <21789152.post@talk.nabble.com> Is there any settings that when call hangup control can be transferred to another context and these CDR values can be accessible there? (just like in Asterisk, h extension) shehzad p wrote: > > Hi all, > > I need to process some CDR variables in Dialplan, like call duration, > Answered time etc. > but when I place info application after bridge, it is not listing them > properly as below: > =========================================== > Caller-Channel-Created-Time: [1233573341672157] > Caller-Channel-Answered-Time: [1233573342712939] > Caller-Channel-Hangup-Time: [0] > ========================================== > Here Hangup time is 0, So how can I find actual values? > > --I know that we can use xml_cdr or cdr_csv, but my current need is to get > those values from dialplan itself so that can be passed to some script... > > > thanks, > msp > -- View this message in context: http://www.nabble.com/Call-Variable-not-available-when-call-hangup-tp21788550p21789152.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From kawarod at laposte.net Mon Feb 2 04:52:57 2009 From: kawarod at laposte.net (rod) Date: Mon, 02 Feb 2009 16:52:57 +0400 Subject: [Freeswitch-users] Strange Performance when using as SBC In-Reply-To: <1C31435E19AB4E57981CF2AAC2CA74AA@SaeedLaptop> References: <4986C42C.7030700@laposte.net> <1C31435E19AB4E57981CF2AAC2CA74AA@SaeedLaptop> Message-ID: <4986ECA9.3040707@laposte.net> Hi Saeed, I just created an account to share my setup on the wiki. I will detail all the steps for a clean install of a debian64 lenny with FS used as a SBC (next step is to try the new LCR module :) )and what I'm doing do stress the server. I wrote nothing at this time so please be patient, I'm waiting for my new hardware so that I will detail as much as possible what I'll do. For beginning I suggest you reading the start page on the wiki, especially these pages: -http://wiki.freeswitch.org/wiki/Getting_Started_Guide -http://wiki.freeswitch.org/wiki/Dialplan_XML maybe you could tell more about the linux distribution you're using so that I can give you some pointers for sipp... regards. rod. Saeed Ahmed wrote: > Hi Rod, > > Could you please share how you configured Sipp & FS to create a test > environment? Especially the dial plan, sofia settings etc..., actually I am > a newbie. I want to test it on a single FS machine. > > Kind Regards > Saeed > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of rod > Sent: Monday, February 02, 2009 11:00 AM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Strange Performance when using as SBC > > Hi Ken, Jay, > > thanks for pointing to proxy media, I will test. > > Ken, you are right, I was brain damaged (a stupid mistake) when setting > INFO cause this kind of level could be very verbose. I'm switching to > CRIT or ERR. > > Thanks guys, > rod. > > thanks for > > Ken Rice wrote: > >> If you don't have to transcode, using proxy media mode will still save you >> some CPU time. This is 1/2 way between bypass media and the default media >> interactive mode. The other draw back to this mode is if you are using FS >> > to > >> clean up RTP and DTMF you loose those functions but they are not needed in >> most use cases. >> >> As far as the log level goes, I found that once I had things stable >> > setting > >> the loglevel to helped a good deal... Info is probably a bit too high of a >> loglevel I would probably go for CRIT or ERR (2 or 1 respectively) if you >> insist on leaving logging turned on... On a busy system these can and will >> generate a good deal of activity (and disk IO if using mod_logfile) >> >> Ken >> >> >> >> >>> From: rod >>> Reply-To: >>> Date: Mon, 02 Feb 2009 11:36:35 +0400 >>> To: >>> Subject: Re: [Freeswitch-users] Strange Performance when using as SBC >>> >>> Hi Ken, >>> >>> 1) I'd like to use FS to hide topology, so bypass media is not possible >>> 2) done >>> 3) done >>> 4) not used >>> 5) i'm using this ins switch.xml -> >> value="info"/>, if you think an other log level is more suitable. >>> >>> Regarding logging, I can see in console and in the freeswitch.log that >>> there is still a lot of NOTICE logging, see below: >>> 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:960 >>> switch_core_session_thread() Session 8721 >>> (sofia/internal/sipp at 10.10.10.1:5060) Ended >>> 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:962 >>> switch_core_session_thread() Close Channel >>> sofia/internal/sipp at 10.10.10.1:5060 [CS_HANGUP] >>> 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:960 >>> switch_core_session_thread() Session 8722 >>> (sofia/external/9998 at 10.10.20.100) Ended >>> 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:962 >>> switch_core_session_thread() Close Channel >>> sofia/external/9998 at 10.10.20.100 [CS_HANGUP] >>> 2009-02-02 08:33:56 [NOTICE] sofia.c:3164 sofia_handle_sip_i_state() >>> Channel [sofia/external/9998 at 10.10.20.100] has been answered >>> 2009-02-02 08:33:56 [WARNING] mod_sofia.c:740 sofia_read_frame() >>> Changing codec ptime to 30. I bet you have a linksys/sipura =D >>> >>> Do you have any idea where I can switch off this kind of logging. I >>> thought it should be in /dialplan/internal.xml, but I see that in >>> internal.xml -> >>> >>> thanks a lot for your suggestion. >>> >>> regards, >>> rod >>> >>> Ken Rice wrote: >>> >>> >>>> Dont forget there are several things you can do to increase >>>> > performance... > >>>> 1) where possible use bypass media or media proxy modes >>>> 2) mount freeswitch/db as a ram drive (if you are using voicemail with >>>> the internal FS DBs you'll need a way to make this persistant across >>>> reboots) >>>> 3) see the wiki for setting reasonable ulimits >>>> 4) (this is my oppinion others may vary) dont use mod_cdr_csv >>>> 5) turn off (or reduce logging) in switch.conf.xml >>>> >>>> all of these thing can greatly improve performance. >>>> >>>> On Mon, Feb 2, 2009 at 1:04 AM, rod >>> > wrote: >>>> >>>> Thanks Anthony, >>>> >>>> the setup is like this: >>>> >>>> sipp server ---- FS 1 ---- FS2 >>>> >>>> FS1 is the AMD CPU that has only one extension in dialplan that >>>> bridges >>>> 9999 to FS2. 9999 is the first extension in FS2 dialplan that >>>> plays moh, >>>> FS2 has no CPU pbm. >>>> >>>> FS1 is maxing out at 60 bridged calls without your option -hp. >>>> >>>> Using -hp, I'm now able to bridge 200 concurrent calls (a great >>>> improvement) and the system is still reactive. CPU load is high >>>> but not >>>> 100% and as the system responds well, I think that doesn't matter. >>>> > The > >>>> 2GB of memory are completely consumed (top command shows 700MB for >>>> > FS > >>>> process). >>>> >>>> I understand that FS1 server is not the best hardware platform, >>>> and I'm >>>> waiting for new 4 cores server for testing. >>>> I will update those numbers when testing with the new hardware. >>>> >>>> regards, >>>> rod. >>>> >>>> Anthony Minessale wrote: >>>> >>>> >>>>> Which of the 2 machines has the load issue? You said it was one box >>>>> calling the other. >>>>> >>>>> You have 2 major things against you, single CPU and AMD, but you >>>>> should at least be able to get in the vicinity of 800-1000 calls >>>>> >>>>> >>>> on a >>>> >>>> >>>>> box like that. >>>>> >>>>> Are you calling the default 9999? It's not really an appropriate >>>>> extension for load testing. >>>>> On the terminating box you should set up a manual extension that is >>>>> the first one in the dial plan >>>>> to play a wav file from preferably a ram disk or /tmp >>>>> >>>>> If you do plan on using this in production accept nothing less >>>>> >>>>> >>>> than a >>>> >>>> >>>>> multi-core intel machine with at least 4 cores, the more cores the >>>>> better because that parallel processing is where FS gets it's >>>>> >>>>> >>>> atvantage. >>>> >>>> >>>>> On Fri, Jan 30, 2009 at 5:56 AM, rod >>>> >>>>> >>>> >>>> >>>> >>>>> >> wrote: >>>>> >>>>> Dear list, >>>>> >>>>> I've been playing with freeswitch for some time (2 months) >>>>> >>>>> >>>> and the >>>> >>>> >>>>> fact >>>>> is that I'm very pleased with the functionnalities of this >>>>> >>>>> >>>> software. >>>> >>>> >>>>> I'd like to use FS as a SBC handling media and I'm doing some >>>>> tests with >>>>> sipp to load the machine but I'm unable to bridge more than >>>>> >>>>> >>>> 60 calls >>>> >>>> >>>>> without seeing the CPU being loaded at 100%. I'm sure >>>>> >>>>> >>>> something is >>>> >>>> >>>>> going >>>>> wrong with my setup but I'm unable to see what. >>>>> >>>>> The test machine has the following specs: >>>>> Athlon XP 3500+ with 2GB of memory (I know this is not a >>>>> >>>>> >>>> high end >>>> >>>> >>>>> machine :p) >>>>> >>>>> Freeswitch:/opt/freeswitch/log# cat /proc/cpuinfo >>>>> processor : 0 >>>>> vendor_id : AuthenticAMD >>>>> cpu family : 15 >>>>> model : 95 >>>>> model name : AMD Athlon(tm) 64 Processor 3500+ >>>>> stepping : 2 >>>>> cpu MHz : 2199.973 >>>>> cache size : 512 KB >>>>> fpu : yes >>>>> fpu_exception : yes >>>>> cpuid level : 1 >>>>> wp : yes >>>>> flags : fpu vme de pse tsc msr pae mce cx8 apic >>>>> >>>>> >>>> sep mtrr pge >>>> >>>> >>>>> mca cmov pat pse36 clflush mmx fxsr sse sse2 syscall nx mmxext >>>>> fxsr_opt >>>>> rdtscp lm 3dnowext 3dnow up rep_good pni cx16 lahf_lm svm >>>>> >>>>> >>>> extapic >>>> >>>> >>>>> cr8_legacy >>>>> bogomips : 4402.97 >>>>> TLB size : 1024 4K pages >>>>> clflush size : 64 >>>>> cache_alignment : 64 >>>>> address sizes : 40 bits physical, 48 bits virtual >>>>> power management: ts fid vid ttp tm stc >>>>> >>>>> I installed FS on a fresh debian 64: >>>>> Linux Freeswitch 2.6.26-1-amd64 #1 SMP Sat Jan 10 17:57:00 >>>>> >>>>> >>>> UTC 2009 >>>> >>>> >>>>> x86_64 GNU/Linux >>>>> >>>>> I set the ulimit parameters like those on the website: >>>>> freeswitch at internal> ... >>>>> Freeswitch:/opt/free-svn/bin# ulimit -a >>>>> core file size (blocks, -c) unlimited >>>>> data seg size (kbytes, -d) unlimited >>>>> scheduling priority (-e) 0 >>>>> file size (blocks, -f) unlimited >>>>> pending signals (-i) unlimited >>>>> max locked memory (kbytes, -l) unlimited >>>>> max memory size (kbytes, -m) unlimited >>>>> open files (-n) 999999 >>>>> pipe size (512 bytes, -p) 8 >>>>> POSIX message queues (bytes, -q) unlimited >>>>> real-time priority (-r) 0 >>>>> stack size (kbytes, -s) 244 >>>>> cpu time (seconds, -t) unlimited >>>>> max user processes (-u) unlimited >>>>> virtual memory (kbytes, -v) unlimited >>>>> file locks (-x) unlimited >>>>> >>>>> >>>>> My network setup is the following: >>>>> >>>>> SIPP machine (10.10.10.1/24)----------------vlan >>>>> >>>>> >>>> >>>> >>>> >>>>> 55 >>>>> ----------(10.10.10.254/24 >>>>> >>>>> >>>> ) FS >>>> >>>> >>>>> (10.10.20.254/24)-------------- >>>>> >>>>> >>>> >>>> >>>> >>>>> vlan56 >>>>> -------------------(10.10.20.100/24 >>>>> >>>>> >>>> ) >>>> >>>> >>>>> OTHER STOCK FS >>>>> >>>>> >>>>> I launched sipp with: >>>>> sipp -sn uac_pcap -s 9999 -r 10 -l 80 -d 60000 -mi >>>>> >>>>> >>>> 10.10.10.1 -i >>>> >>>> >>>>> 10.10.10.1 -mp 25000 10.10.10.254:5060 >>>>> >>>>> >>>> >>>> >>>> >>>>> The dialplan on FS is very simple: >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>> data="sofia/external/9999 at 10.10.20.100 >>>>> >>>>> >>>> >>> >"/> >>>> >>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> FreeSWITCH Version 1.0.trunk (11560M) Started. >>>>> Crash Protection [Disabled] >>>>> Max Sessions[1000] >>>>> Session Rate[100] >>>>> SQL [Enabled] >>>>> >>>>> >>>>> The test is very simple: sipp dial 9999 that matches in my >>>>> >>>>> >>>> FS dialplan >>>> >>>> >>>>> and this is bridged to an other FS machine playing music on >>>>> >>>>> >>>> hold. >>>> >>>> >>>>> When I launch "top" I see after 30 to 40 s that FS consumes all >>>>> the CPU >>>>> ressources (with a mean of 50-60 % before), with 80 calls. >>>>> When I set 70 calls, I have to wait 70-80 s before seeing >>>>> >>>>> >>>> the same >>>> >>>> >>>>> issue. >>>>> >>>>> Presence is set to false on the 2 profile. >>>>> >>>>> I have the same issue with FS 1.0.2 that' s why I tried FS >>>>> >>>>> >>>> 11560. >>>> >>>> >>>>> When I use the FS machine as a router to test the packet per >>>>> >>>>> >>>> second >>>> >>>> >>>>> performance, I'm reaching 100Mbps with 8000pps in each >>>>> >>>>> >>>> direction (from >>>> >>>> >>>>> vlan 55 to vlan56) with less than 12% CPU. So that I don't think >>>>> there's >>>>> an issue with the network. >>>>> >>>>> Here is an "mpstat -P ALL 1" to show you what's happening >>>>> >>>>> >>>> suddenly >>>> >>>> >>>>> with >>>>> 70 bridge calls: >>>>> 12:31:26 CPU %user %nice %sys %iowait %irq %soft >>>>> %steal %idle intr/s >>>>> 12:31:27 all 3,00 0,00 3,00 0,00 1,00 4,00 >>>>> 0,00 89,00 6241,00 >>>>> 12:31:27 0 3,00 0,00 3,00 0,00 1,00 4,00 >>>>> 0,00 89,00 6241,00 >>>>> >>>>> 12:31:27 CPU %user %nice %sys %iowait %irq %soft >>>>> %steal %idle intr/s >>>>> 12:31:28 all 14,14 0,00 56,57 0,00 2,02 5,05 >>>>> 0,00 22,22 6035,35 >>>>> 12:31:28 0 14,14 0,00 56,57 0,00 2,02 5,05 >>>>> 0,00 22,22 6035,35 >>>>> >>>>> 12:31:28 CPU %user %nice %sys %iowait %irq %soft >>>>> %steal %idle intr/s >>>>> 12:31:29 all 24,75 0,00 67,33 0,00 0,99 6,93 >>>>> 0,00 0,00 5483,17 >>>>> 12:31:29 0 24,75 0,00 67,33 0,00 0,99 6,93 >>>>> 0,00 0,00 5483,17 >>>>> >>>>> >>>>> The CPU is going from 89% idle to 0% in less than 2 seconds. >>>>> >>>>> I know that I don't have to expect too much from this kind of >>>>> hardware, >>>>> but it seems strange that the CPU power vanished so suddenly. >>>>> >>>>> Thanks a lot for the guys that have read this long mail :p >>>>> >>>>> kind regards, >>>>> rod >>>>> >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> >>>>> >>>> >>>>> >>>> > >>>> >>>> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> >>>>> >>>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >>>> >>>> >>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> >>>>> >>>>> -- >>>>> Anthony Minessale II >>>>> >>>>> FreeSWITCH http://www.freeswitch.org/ >>>>> ClueCon http://www.cluecon.com/ >>>>> >>>>> AIM: anthm >>>>> MSN:anthony_minessale at hotmail.com >>>>> >>>>> >>>> >>>> >>>> >>>>> >>>> >>>>> >>>> > >>>> >>>> >>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>>>> >>>>> >>>> >>>> >>>> >>>>> >>>> >>>>> >>>> > >>>> >>>> >>>>> IRC: irc.freenode.net >>>>> >>>>> >>>> #freeswitch >>>> >>>> >>>>> FreeSWITCH Developer Conference >>>>> sip:888 at conference.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> >>>>> >>>> >>>>> >>>> > >>>> >>>> >>>>> iax:guest at conference.freeswitch.org/888 >>>>> >>>>> >>>> >>>> >>>> >>>>> >>>>> googletalk:conf+888 at conference.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> >>>>> >>>> >>>>> >>>> > >>>> >>>> >>>>> pstn:213-799-1400 >>>>> >>>>> >>>>> > ------------------------------------------------------------------------ > >>>> >>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> >>>>> >>>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >>>> >>>> >>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> >>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >>>> http://www.freeswitch.org >>>> >>>> >>>> ------------------------------------------------------------------------ >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > From palletboy at gmail.com Mon Feb 2 05:02:58 2009 From: palletboy at gmail.com (J. G.) Date: Mon, 2 Feb 2009 08:02:58 -0500 Subject: [Freeswitch-users] Phonebooth? Message-ID: <3093591d0902020502s6f726ba8h819402da0705a76a@mail.gmail.com> I got a group email from Anders Brownworth this weekend regarding him donating Phonebooth to the FreePBX project? Wonder what the impact will be.. -- ----- Jason Gehman General Manager North Voice Communications www.NorthVC.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090202/7cc6cf1f/attachment-0002.html From saeedahmad1981 at gmail.com Mon Feb 2 05:03:55 2009 From: saeedahmad1981 at gmail.com (Saeed Ahmed) Date: Mon, 2 Feb 2009 14:03:55 +0100 Subject: [Freeswitch-users] Strange Performance when using as SBC In-Reply-To: <4986ECA9.3040707@laposte.net> References: <4986C42C.7030700@laposte.net><1C31435E19AB4E57981CF2AAC2CA74AA@SaeedLaptop> <4986ECA9.3040707@laposte.net> Message-ID: <9232C06D5362494791AF713E1DF61343@SaeedLaptop> Thanks rod for a quick answer, FS is installed on Ubuntu Server. I am planning to replace Nextone SBC with FS, Later I'll also use openZAP to communicate with TDM but this all depends how much calls it can take, or maybe we can also do something in clustering environment ( I am not sure about it). But thanks again and any further help will be highly appreciated. Kind Regards Saeed Ahmed Tariq -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of rod Sent: Monday, February 02, 2009 1:53 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Strange Performance when using as SBC Hi Saeed, I just created an account to share my setup on the wiki. I will detail all the steps for a clean install of a debian64 lenny with FS used as a SBC (next step is to try the new LCR module :) )and what I'm doing do stress the server. I wrote nothing at this time so please be patient, I'm waiting for my new hardware so that I will detail as much as possible what I'll do. For beginning I suggest you reading the start page on the wiki, especially these pages: -http://wiki.freeswitch.org/wiki/Getting_Started_Guide -http://wiki.freeswitch.org/wiki/Dialplan_XML maybe you could tell more about the linux distribution you're using so that I can give you some pointers for sipp... regards. rod. Saeed Ahmed wrote: > Hi Rod, > > Could you please share how you configured Sipp & FS to create a test > environment? Especially the dial plan, sofia settings etc..., actually I am > a newbie. I want to test it on a single FS machine. > > Kind Regards > Saeed > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of rod > Sent: Monday, February 02, 2009 11:00 AM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Strange Performance when using as SBC > > Hi Ken, Jay, > > thanks for pointing to proxy media, I will test. > > Ken, you are right, I was brain damaged (a stupid mistake) when setting > INFO cause this kind of level could be very verbose. I'm switching to > CRIT or ERR. > > Thanks guys, > rod. > > thanks for > > Ken Rice wrote: > >> If you don't have to transcode, using proxy media mode will still save you >> some CPU time. This is 1/2 way between bypass media and the default media >> interactive mode. The other draw back to this mode is if you are using FS >> > to > >> clean up RTP and DTMF you loose those functions but they are not needed in >> most use cases. >> >> As far as the log level goes, I found that once I had things stable >> > setting > >> the loglevel to helped a good deal... Info is probably a bit too high of a >> loglevel I would probably go for CRIT or ERR (2 or 1 respectively) if you >> insist on leaving logging turned on... On a busy system these can and will >> generate a good deal of activity (and disk IO if using mod_logfile) >> >> Ken >> >> >> >> >>> From: rod >>> Reply-To: >>> Date: Mon, 02 Feb 2009 11:36:35 +0400 >>> To: >>> Subject: Re: [Freeswitch-users] Strange Performance when using as SBC >>> >>> Hi Ken, >>> >>> 1) I'd like to use FS to hide topology, so bypass media is not possible >>> 2) done >>> 3) done >>> 4) not used >>> 5) i'm using this ins switch.xml -> >> value="info"/>, if you think an other log level is more suitable. >>> >>> Regarding logging, I can see in console and in the freeswitch.log that >>> there is still a lot of NOTICE logging, see below: >>> 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:960 >>> switch_core_session_thread() Session 8721 >>> (sofia/internal/sipp at 10.10.10.1:5060) Ended >>> 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:962 >>> switch_core_session_thread() Close Channel >>> sofia/internal/sipp at 10.10.10.1:5060 [CS_HANGUP] >>> 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:960 >>> switch_core_session_thread() Session 8722 >>> (sofia/external/9998 at 10.10.20.100) Ended >>> 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:962 >>> switch_core_session_thread() Close Channel >>> sofia/external/9998 at 10.10.20.100 [CS_HANGUP] >>> 2009-02-02 08:33:56 [NOTICE] sofia.c:3164 sofia_handle_sip_i_state() >>> Channel [sofia/external/9998 at 10.10.20.100] has been answered >>> 2009-02-02 08:33:56 [WARNING] mod_sofia.c:740 sofia_read_frame() >>> Changing codec ptime to 30. I bet you have a linksys/sipura =D >>> >>> Do you have any idea where I can switch off this kind of logging. I >>> thought it should be in /dialplan/internal.xml, but I see that in >>> internal.xml -> >>> >>> thanks a lot for your suggestion. >>> >>> regards, >>> rod >>> >>> Ken Rice wrote: >>> >>> >>>> Dont forget there are several things you can do to increase >>>> > performance... > >>>> 1) where possible use bypass media or media proxy modes >>>> 2) mount freeswitch/db as a ram drive (if you are using voicemail with >>>> the internal FS DBs you'll need a way to make this persistant across >>>> reboots) >>>> 3) see the wiki for setting reasonable ulimits >>>> 4) (this is my oppinion others may vary) dont use mod_cdr_csv >>>> 5) turn off (or reduce logging) in switch.conf.xml >>>> >>>> all of these thing can greatly improve performance. >>>> >>>> On Mon, Feb 2, 2009 at 1:04 AM, rod >>> > wrote: >>>> >>>> Thanks Anthony, >>>> >>>> the setup is like this: >>>> >>>> sipp server ---- FS 1 ---- FS2 >>>> >>>> FS1 is the AMD CPU that has only one extension in dialplan that >>>> bridges >>>> 9999 to FS2. 9999 is the first extension in FS2 dialplan that >>>> plays moh, >>>> FS2 has no CPU pbm. >>>> >>>> FS1 is maxing out at 60 bridged calls without your option -hp. >>>> >>>> Using -hp, I'm now able to bridge 200 concurrent calls (a great >>>> improvement) and the system is still reactive. CPU load is high >>>> but not >>>> 100% and as the system responds well, I think that doesn't matter. >>>> > The > >>>> 2GB of memory are completely consumed (top command shows 700MB for >>>> > FS > >>>> process). >>>> >>>> I understand that FS1 server is not the best hardware platform, >>>> and I'm >>>> waiting for new 4 cores server for testing. >>>> I will update those numbers when testing with the new hardware. >>>> >>>> regards, >>>> rod. >>>> >>>> Anthony Minessale wrote: >>>> >>>> >>>>> Which of the 2 machines has the load issue? You said it was one box >>>>> calling the other. >>>>> >>>>> You have 2 major things against you, single CPU and AMD, but you >>>>> should at least be able to get in the vicinity of 800-1000 calls >>>>> >>>>> >>>> on a >>>> >>>> >>>>> box like that. >>>>> >>>>> Are you calling the default 9999? It's not really an appropriate >>>>> extension for load testing. >>>>> On the terminating box you should set up a manual extension that is >>>>> the first one in the dial plan >>>>> to play a wav file from preferably a ram disk or /tmp >>>>> >>>>> If you do plan on using this in production accept nothing less >>>>> >>>>> >>>> than a >>>> >>>> >>>>> multi-core intel machine with at least 4 cores, the more cores the >>>>> better because that parallel processing is where FS gets it's >>>>> >>>>> >>>> atvantage. >>>> >>>> >>>>> On Fri, Jan 30, 2009 at 5:56 AM, rod >>>> >>>>> >>>> >>>> >>>> >>>>> >> wrote: >>>>> >>>>> Dear list, >>>>> >>>>> I've been playing with freeswitch for some time (2 months) >>>>> >>>>> >>>> and the >>>> >>>> >>>>> fact >>>>> is that I'm very pleased with the functionnalities of this >>>>> >>>>> >>>> software. >>>> >>>> >>>>> I'd like to use FS as a SBC handling media and I'm doing some >>>>> tests with >>>>> sipp to load the machine but I'm unable to bridge more than >>>>> >>>>> >>>> 60 calls >>>> >>>> >>>>> without seeing the CPU being loaded at 100%. I'm sure >>>>> >>>>> >>>> something is >>>> >>>> >>>>> going >>>>> wrong with my setup but I'm unable to see what. >>>>> >>>>> The test machine has the following specs: >>>>> Athlon XP 3500+ with 2GB of memory (I know this is not a >>>>> >>>>> >>>> high end >>>> >>>> >>>>> machine :p) >>>>> >>>>> Freeswitch:/opt/freeswitch/log# cat /proc/cpuinfo >>>>> processor : 0 >>>>> vendor_id : AuthenticAMD >>>>> cpu family : 15 >>>>> model : 95 >>>>> model name : AMD Athlon(tm) 64 Processor 3500+ >>>>> stepping : 2 >>>>> cpu MHz : 2199.973 >>>>> cache size : 512 KB >>>>> fpu : yes >>>>> fpu_exception : yes >>>>> cpuid level : 1 >>>>> wp : yes >>>>> flags : fpu vme de pse tsc msr pae mce cx8 apic >>>>> >>>>> >>>> sep mtrr pge >>>> >>>> >>>>> mca cmov pat pse36 clflush mmx fxsr sse sse2 syscall nx mmxext >>>>> fxsr_opt >>>>> rdtscp lm 3dnowext 3dnow up rep_good pni cx16 lahf_lm svm >>>>> >>>>> >>>> extapic >>>> >>>> >>>>> cr8_legacy >>>>> bogomips : 4402.97 >>>>> TLB size : 1024 4K pages >>>>> clflush size : 64 >>>>> cache_alignment : 64 >>>>> address sizes : 40 bits physical, 48 bits virtual >>>>> power management: ts fid vid ttp tm stc >>>>> >>>>> I installed FS on a fresh debian 64: >>>>> Linux Freeswitch 2.6.26-1-amd64 #1 SMP Sat Jan 10 17:57:00 >>>>> >>>>> >>>> UTC 2009 >>>> >>>> >>>>> x86_64 GNU/Linux >>>>> >>>>> I set the ulimit parameters like those on the website: >>>>> freeswitch at internal> ... >>>>> Freeswitch:/opt/free-svn/bin# ulimit -a >>>>> core file size (blocks, -c) unlimited >>>>> data seg size (kbytes, -d) unlimited >>>>> scheduling priority (-e) 0 >>>>> file size (blocks, -f) unlimited >>>>> pending signals (-i) unlimited >>>>> max locked memory (kbytes, -l) unlimited >>>>> max memory size (kbytes, -m) unlimited >>>>> open files (-n) 999999 >>>>> pipe size (512 bytes, -p) 8 >>>>> POSIX message queues (bytes, -q) unlimited >>>>> real-time priority (-r) 0 >>>>> stack size (kbytes, -s) 244 >>>>> cpu time (seconds, -t) unlimited >>>>> max user processes (-u) unlimited >>>>> virtual memory (kbytes, -v) unlimited >>>>> file locks (-x) unlimited >>>>> >>>>> >>>>> My network setup is the following: >>>>> >>>>> SIPP machine (10.10.10.1/24)----------------vlan >>>>> >>>>> >>>> >>>> >>>> >>>>> 55 >>>>> ----------(10.10.10.254/24 >>>>> >>>>> >>>> ) FS >>>> >>>> >>>>> (10.10.20.254/24)-------------- >>>>> >>>>> >>>> >>>> >>>> >>>>> vlan56 >>>>> -------------------(10.10.20.100/24 >>>>> >>>>> >>>> ) >>>> >>>> >>>>> OTHER STOCK FS >>>>> >>>>> >>>>> I launched sipp with: >>>>> sipp -sn uac_pcap -s 9999 -r 10 -l 80 -d 60000 -mi >>>>> >>>>> >>>> 10.10.10.1 -i >>>> >>>> >>>>> 10.10.10.1 -mp 25000 10.10.10.254:5060 >>>>> >>>>> >>>> >>>> >>>> >>>>> The dialplan on FS is very simple: >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>> data="sofia/external/9999 at 10.10.20.100 >>>>> >>>>> >>>> >>> >"/> >>>> >>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> FreeSWITCH Version 1.0.trunk (11560M) Started. >>>>> Crash Protection [Disabled] >>>>> Max Sessions[1000] >>>>> Session Rate[100] >>>>> SQL [Enabled] >>>>> >>>>> >>>>> The test is very simple: sipp dial 9999 that matches in my >>>>> >>>>> >>>> FS dialplan >>>> >>>> >>>>> and this is bridged to an other FS machine playing music on >>>>> >>>>> >>>> hold. >>>> >>>> >>>>> When I launch "top" I see after 30 to 40 s that FS consumes all >>>>> the CPU >>>>> ressources (with a mean of 50-60 % before), with 80 calls. >>>>> When I set 70 calls, I have to wait 70-80 s before seeing >>>>> >>>>> >>>> the same >>>> >>>> >>>>> issue. >>>>> >>>>> Presence is set to false on the 2 profile. >>>>> >>>>> I have the same issue with FS 1.0.2 that' s why I tried FS >>>>> >>>>> >>>> 11560. >>>> >>>> >>>>> When I use the FS machine as a router to test the packet per >>>>> >>>>> >>>> second >>>> >>>> >>>>> performance, I'm reaching 100Mbps with 8000pps in each >>>>> >>>>> >>>> direction (from >>>> >>>> >>>>> vlan 55 to vlan56) with less than 12% CPU. So that I don't think >>>>> there's >>>>> an issue with the network. >>>>> >>>>> Here is an "mpstat -P ALL 1" to show you what's happening >>>>> >>>>> >>>> suddenly >>>> >>>> >>>>> with >>>>> 70 bridge calls: >>>>> 12:31:26 CPU %user %nice %sys %iowait %irq %soft >>>>> %steal %idle intr/s >>>>> 12:31:27 all 3,00 0,00 3,00 0,00 1,00 4,00 >>>>> 0,00 89,00 6241,00 >>>>> 12:31:27 0 3,00 0,00 3,00 0,00 1,00 4,00 >>>>> 0,00 89,00 6241,00 >>>>> >>>>> 12:31:27 CPU %user %nice %sys %iowait %irq %soft >>>>> %steal %idle intr/s >>>>> 12:31:28 all 14,14 0,00 56,57 0,00 2,02 5,05 >>>>> 0,00 22,22 6035,35 >>>>> 12:31:28 0 14,14 0,00 56,57 0,00 2,02 5,05 >>>>> 0,00 22,22 6035,35 >>>>> >>>>> 12:31:28 CPU %user %nice %sys %iowait %irq %soft >>>>> %steal %idle intr/s >>>>> 12:31:29 all 24,75 0,00 67,33 0,00 0,99 6,93 >>>>> 0,00 0,00 5483,17 >>>>> 12:31:29 0 24,75 0,00 67,33 0,00 0,99 6,93 >>>>> 0,00 0,00 5483,17 >>>>> >>>>> >>>>> The CPU is going from 89% idle to 0% in less than 2 seconds. >>>>> >>>>> I know that I don't have to expect too much from this kind of >>>>> hardware, >>>>> but it seems strange that the CPU power vanished so suddenly. >>>>> >>>>> Thanks a lot for the guys that have read this long mail :p >>>>> >>>>> kind regards, >>>>> rod >>>>> >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> >>>>> >>>> >>>>> >>>> > >>>> >>>> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> >>>>> >>>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >>>> >>>> >>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> >>>>> >>>>> -- >>>>> Anthony Minessale II >>>>> >>>>> FreeSWITCH http://www.freeswitch.org/ >>>>> ClueCon http://www.cluecon.com/ >>>>> >>>>> AIM: anthm >>>>> MSN:anthony_minessale at hotmail.com >>>>> >>>>> >>>> >>>> >>>> >>>>> >>>> >>>>> >>>> > >>>> >>>> >>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>>>> >>>>> >>>> >>>> >>>> >>>>> >>>> >>>>> >>>> > >>>> >>>> >>>>> IRC: irc.freenode.net >>>>> >>>>> >>>> #freeswitch >>>> >>>> >>>>> FreeSWITCH Developer Conference >>>>> sip:888 at conference.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> >>>>> >>>> >>>>> >>>> > >>>> >>>> >>>>> iax:guest at conference.freeswitch.org/888 >>>>> >>>>> >>>> >>>> >>>> >>>>> >>>>> googletalk:conf+888 at conference.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> >>>>> >>>> >>>>> >>>> > >>>> >>>> >>>>> pstn:213-799-1400 >>>>> >>>>> >>>>> > ------------------------------------------------------------------------ > >>>> >>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> >>>>> >>>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >>>> >>>> >>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> >>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >>>> http://www.freeswitch.org >>>> >>>> >>>> ------------------------------------------------------------------------ >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From anthony.minessale at gmail.com Mon Feb 2 05:57:43 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 2 Feb 2009 07:57:43 -0600 Subject: [Freeswitch-users] Strange Performance when using as SBC In-Reply-To: <9232C06D5362494791AF713E1DF61343@SaeedLaptop> References: <4986C42C.7030700@laposte.net> <1C31435E19AB4E57981CF2AAC2CA74AA@SaeedLaptop> <4986ECA9.3040707@laposte.net> <9232C06D5362494791AF713E1DF61343@SaeedLaptop> Message-ID: <191c3a030902020557v70d3777fqb83e591385a64a59@mail.gmail.com> if you want to use ubuntu, be sure to use hardy and not intrepid. On Mon, Feb 2, 2009 at 7:03 AM, Saeed Ahmed wrote: > Thanks rod for a quick answer, > > FS is installed on Ubuntu Server. > > I am planning to replace Nextone SBC with FS, Later I'll also use openZAP > to > communicate with TDM but this all depends how much calls it can take, or > maybe we can also do something in clustering environment ( I am not sure > about it). But thanks again and any further help will be highly > appreciated. > > > Kind Regards > Saeed Ahmed Tariq > > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of rod > Sent: Monday, February 02, 2009 1:53 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Strange Performance when using as SBC > > Hi Saeed, > > I just created an account to share my setup on the wiki. I will detail > all the steps for a clean install of a debian64 lenny with FS used as a > SBC (next step is to try the new LCR module :) )and what I'm doing do > stress the server. > > I wrote nothing at this time so please be patient, I'm waiting for my > new hardware so that I will detail as much as possible what I'll do. > > For beginning I suggest you reading the start page on the wiki, > especially these pages: > -http://wiki.freeswitch.org/wiki/Getting_Started_Guide > -http://wiki.freeswitch.org/wiki/Dialplan_XML > > maybe you could tell more about the linux distribution you're using so > that I can give you some pointers for sipp... > > regards. > rod. > > > Saeed Ahmed wrote: > > Hi Rod, > > > > Could you please share how you configured Sipp & FS to create a test > > environment? Especially the dial plan, sofia settings etc..., actually I > am > > a newbie. I want to test it on a single FS machine. > > > > Kind Regards > > Saeed > > -----Original Message----- > > From: freeswitch-users-bounces at lists.freeswitch.org > > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of rod > > Sent: Monday, February 02, 2009 11:00 AM > > To: freeswitch-users at lists.freeswitch.org > > Subject: Re: [Freeswitch-users] Strange Performance when using as SBC > > > > Hi Ken, Jay, > > > > thanks for pointing to proxy media, I will test. > > > > Ken, you are right, I was brain damaged (a stupid mistake) when setting > > INFO cause this kind of level could be very verbose. I'm switching to > > CRIT or ERR. > > > > Thanks guys, > > rod. > > > > thanks for > > > > Ken Rice wrote: > > > >> If you don't have to transcode, using proxy media mode will still save > you > >> some CPU time. This is 1/2 way between bypass media and the default > media > >> interactive mode. The other draw back to this mode is if you are using > FS > >> > > to > > > >> clean up RTP and DTMF you loose those functions but they are not needed > in > >> most use cases. > >> > >> As far as the log level goes, I found that once I had things stable > >> > > setting > > > >> the loglevel to helped a good deal... Info is probably a bit too high of > a > >> loglevel I would probably go for CRIT or ERR (2 or 1 respectively) if > you > >> insist on leaving logging turned on... On a busy system these can and > will > >> generate a good deal of activity (and disk IO if using mod_logfile) > >> > >> Ken > >> > >> > >> > >> > >>> From: rod > >>> Reply-To: > >>> Date: Mon, 02 Feb 2009 11:36:35 +0400 > >>> To: > >>> Subject: Re: [Freeswitch-users] Strange Performance when using as SBC > >>> > >>> Hi Ken, > >>> > >>> 1) I'd like to use FS to hide topology, so bypass media is not possible > >>> 2) done > >>> 3) done > >>> 4) not used > >>> 5) i'm using this ins switch.xml -> >>> value="info"/>, if you think an other log level is more suitable. > >>> > >>> Regarding logging, I can see in console and in the freeswitch.log that > >>> there is still a lot of NOTICE logging, see below: > >>> 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:960 > >>> switch_core_session_thread() Session 8721 > >>> (sofia/internal/sipp at 10.10.10.1:5060) Ended > >>> 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:962 > >>> switch_core_session_thread() Close Channel > >>> sofia/internal/sipp at 10.10.10.1:5060 [CS_HANGUP] > >>> 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:960 > >>> switch_core_session_thread() Session 8722 > >>> (sofia/external/9998 at 10.10.20.100) Ended > >>> 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:962 > >>> switch_core_session_thread() Close Channel > >>> sofia/external/9998 at 10.10.20.100 [CS_HANGUP] > >>> 2009-02-02 08:33:56 [NOTICE] sofia.c:3164 sofia_handle_sip_i_state() > >>> Channel [sofia/external/9998 at 10.10.20.100] has been answered > >>> 2009-02-02 08:33:56 [WARNING] mod_sofia.c:740 sofia_read_frame() > >>> Changing codec ptime to 30. I bet you have a linksys/sipura =D > >>> > >>> Do you have any idea where I can switch off this kind of logging. I > >>> thought it should be in /dialplan/internal.xml, but I see that in > >>> internal.xml -> > >>> > >>> thanks a lot for your suggestion. > >>> > >>> regards, > >>> rod > >>> > >>> Ken Rice wrote: > >>> > >>> > >>>> Dont forget there are several things you can do to increase > >>>> > > performance... > > > >>>> 1) where possible use bypass media or media proxy modes > >>>> 2) mount freeswitch/db as a ram drive (if you are using voicemail with > >>>> the internal FS DBs you'll need a way to make this persistant across > >>>> reboots) > >>>> 3) see the wiki for setting reasonable ulimits > >>>> 4) (this is my oppinion others may vary) dont use mod_cdr_csv > >>>> 5) turn off (or reduce logging) in switch.conf.xml > >>>> > >>>> all of these thing can greatly improve performance. > >>>> > >>>> On Mon, Feb 2, 2009 at 1:04 AM, rod >>>> > wrote: > >>>> > >>>> Thanks Anthony, > >>>> > >>>> the setup is like this: > >>>> > >>>> sipp server ---- FS 1 ---- FS2 > >>>> > >>>> FS1 is the AMD CPU that has only one extension in dialplan that > >>>> bridges > >>>> 9999 to FS2. 9999 is the first extension in FS2 dialplan that > >>>> plays moh, > >>>> FS2 has no CPU pbm. > >>>> > >>>> FS1 is maxing out at 60 bridged calls without your option -hp. > >>>> > >>>> Using -hp, I'm now able to bridge 200 concurrent calls (a great > >>>> improvement) and the system is still reactive. CPU load is high > >>>> but not > >>>> 100% and as the system responds well, I think that doesn't matter. > >>>> > > The > > > >>>> 2GB of memory are completely consumed (top command shows 700MB for > >>>> > > FS > > > >>>> process). > >>>> > >>>> I understand that FS1 server is not the best hardware platform, > >>>> and I'm > >>>> waiting for new 4 cores server for testing. > >>>> I will update those numbers when testing with the new hardware. > >>>> > >>>> regards, > >>>> rod. > >>>> > >>>> Anthony Minessale wrote: > >>>> > >>>> > >>>>> Which of the 2 machines has the load issue? You said it was one box > >>>>> calling the other. > >>>>> > >>>>> You have 2 major things against you, single CPU and AMD, but you > >>>>> should at least be able to get in the vicinity of 800-1000 calls > >>>>> > >>>>> > >>>> on a > >>>> > >>>> > >>>>> box like that. > >>>>> > >>>>> Are you calling the default 9999? It's not really an appropriate > >>>>> extension for load testing. > >>>>> On the terminating box you should set up a manual extension that is > >>>>> the first one in the dial plan > >>>>> to play a wav file from preferably a ram disk or /tmp > >>>>> > >>>>> If you do plan on using this in production accept nothing less > >>>>> > >>>>> > >>>> than a > >>>> > >>>> > >>>>> multi-core intel machine with at least 4 cores, the more cores the > >>>>> better because that parallel processing is where FS gets it's > >>>>> > >>>>> > >>>> atvantage. > >>>> > >>>> > >>>>> On Fri, Jan 30, 2009 at 5:56 AM, rod >>>>> > >>>>> > >>>> > >>>> > >>>> > >>>>> >> wrote: > >>>>> > >>>>> Dear list, > >>>>> > >>>>> I've been playing with freeswitch for some time (2 months) > >>>>> > >>>>> > >>>> and the > >>>> > >>>> > >>>>> fact > >>>>> is that I'm very pleased with the functionnalities of this > >>>>> > >>>>> > >>>> software. > >>>> > >>>> > >>>>> I'd like to use FS as a SBC handling media and I'm doing some > >>>>> tests with > >>>>> sipp to load the machine but I'm unable to bridge more than > >>>>> > >>>>> > >>>> 60 calls > >>>> > >>>> > >>>>> without seeing the CPU being loaded at 100%. I'm sure > >>>>> > >>>>> > >>>> something is > >>>> > >>>> > >>>>> going > >>>>> wrong with my setup but I'm unable to see what. > >>>>> > >>>>> The test machine has the following specs: > >>>>> Athlon XP 3500+ with 2GB of memory (I know this is not a > >>>>> > >>>>> > >>>> high end > >>>> > >>>> > >>>>> machine :p) > >>>>> > >>>>> Freeswitch:/opt/freeswitch/log# cat /proc/cpuinfo > >>>>> processor : 0 > >>>>> vendor_id : AuthenticAMD > >>>>> cpu family : 15 > >>>>> model : 95 > >>>>> model name : AMD Athlon(tm) 64 Processor 3500+ > >>>>> stepping : 2 > >>>>> cpu MHz : 2199.973 > >>>>> cache size : 512 KB > >>>>> fpu : yes > >>>>> fpu_exception : yes > >>>>> cpuid level : 1 > >>>>> wp : yes > >>>>> flags : fpu vme de pse tsc msr pae mce cx8 apic > >>>>> > >>>>> > >>>> sep mtrr pge > >>>> > >>>> > >>>>> mca cmov pat pse36 clflush mmx fxsr sse sse2 syscall nx mmxext > >>>>> fxsr_opt > >>>>> rdtscp lm 3dnowext 3dnow up rep_good pni cx16 lahf_lm svm > >>>>> > >>>>> > >>>> extapic > >>>> > >>>> > >>>>> cr8_legacy > >>>>> bogomips : 4402.97 > >>>>> TLB size : 1024 4K pages > >>>>> clflush size : 64 > >>>>> cache_alignment : 64 > >>>>> address sizes : 40 bits physical, 48 bits virtual > >>>>> power management: ts fid vid ttp tm stc > >>>>> > >>>>> I installed FS on a fresh debian 64: > >>>>> Linux Freeswitch 2.6.26-1-amd64 #1 SMP Sat Jan 10 17:57:00 > >>>>> > >>>>> > >>>> UTC 2009 > >>>> > >>>> > >>>>> x86_64 GNU/Linux > >>>>> > >>>>> I set the ulimit parameters like those on the website: > >>>>> freeswitch at internal> ... > >>>>> Freeswitch:/opt/free-svn/bin# ulimit -a > >>>>> core file size (blocks, -c) unlimited > >>>>> data seg size (kbytes, -d) unlimited > >>>>> scheduling priority (-e) 0 > >>>>> file size (blocks, -f) unlimited > >>>>> pending signals (-i) unlimited > >>>>> max locked memory (kbytes, -l) unlimited > >>>>> max memory size (kbytes, -m) unlimited > >>>>> open files (-n) 999999 > >>>>> pipe size (512 bytes, -p) 8 > >>>>> POSIX message queues (bytes, -q) unlimited > >>>>> real-time priority (-r) 0 > >>>>> stack size (kbytes, -s) 244 > >>>>> cpu time (seconds, -t) unlimited > >>>>> max user processes (-u) unlimited > >>>>> virtual memory (kbytes, -v) unlimited > >>>>> file locks (-x) unlimited > >>>>> > >>>>> > >>>>> My network setup is the following: > >>>>> > >>>>> SIPP machine (10.10.10.1/24)----------------vlan > >>>>> > >>>>> > >>>> > >>>> > >>>> > >>>>> 55 > >>>>> ----------(10.10.10.254/24 > >>>>> > >>>>> > >>>> ) FS > >>>> > >>>> > >>>>> (10.10.20.254/24)-------------- > >>>>> > >>>>> > >>>> > >>>> > >>>> > >>>>> vlan56 > >>>>> -------------------(10.10.20.100/24 > >>>>> > >>>>> > >>>> ) > >>>> > >>>> > >>>>> OTHER STOCK FS > >>>>> > >>>>> > >>>>> I launched sipp with: > >>>>> sipp -sn uac_pcap -s 9999 -r 10 -l 80 -d 60000 -mi > >>>>> > >>>>> > >>>> 10.10.10.1 -i > >>>> > >>>> > >>>>> 10.10.10.1 -mp 25000 10.10.10.254:5060 > >>>>> > >>>>> > >>>> > >>>> > >>>> > >>>>> The dialplan on FS is very simple: > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> >>>>> data="sofia/external/9999 at 10.10.20.100 > >>>>> > >>>>> > >>>> >>>> >"/> > >>>> > >>>> > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> FreeSWITCH Version 1.0.trunk (11560M) Started. > >>>>> Crash Protection [Disabled] > >>>>> Max Sessions[1000] > >>>>> Session Rate[100] > >>>>> SQL [Enabled] > >>>>> > >>>>> > >>>>> The test is very simple: sipp dial 9999 that matches in my > >>>>> > >>>>> > >>>> FS dialplan > >>>> > >>>> > >>>>> and this is bridged to an other FS machine playing music on > >>>>> > >>>>> > >>>> hold. > >>>> > >>>> > >>>>> When I launch "top" I see after 30 to 40 s that FS consumes all > >>>>> the CPU > >>>>> ressources (with a mean of 50-60 % before), with 80 calls. > >>>>> When I set 70 calls, I have to wait 70-80 s before seeing > >>>>> > >>>>> > >>>> the same > >>>> > >>>> > >>>>> issue. > >>>>> > >>>>> Presence is set to false on the 2 profile. > >>>>> > >>>>> I have the same issue with FS 1.0.2 that' s why I tried FS > >>>>> > >>>>> > >>>> 11560. > >>>> > >>>> > >>>>> When I use the FS machine as a router to test the packet per > >>>>> > >>>>> > >>>> second > >>>> > >>>> > >>>>> performance, I'm reaching 100Mbps with 8000pps in each > >>>>> > >>>>> > >>>> direction (from > >>>> > >>>> > >>>>> vlan 55 to vlan56) with less than 12% CPU. So that I don't think > >>>>> there's > >>>>> an issue with the network. > >>>>> > >>>>> Here is an "mpstat -P ALL 1" to show you what's happening > >>>>> > >>>>> > >>>> suddenly > >>>> > >>>> > >>>>> with > >>>>> 70 bridge calls: > >>>>> 12:31:26 CPU %user %nice %sys %iowait %irq %soft > >>>>> %steal %idle intr/s > >>>>> 12:31:27 all 3,00 0,00 3,00 0,00 1,00 4,00 > >>>>> 0,00 89,00 6241,00 > >>>>> 12:31:27 0 3,00 0,00 3,00 0,00 1,00 4,00 > >>>>> 0,00 89,00 6241,00 > >>>>> > >>>>> 12:31:27 CPU %user %nice %sys %iowait %irq %soft > >>>>> %steal %idle intr/s > >>>>> 12:31:28 all 14,14 0,00 56,57 0,00 2,02 5,05 > >>>>> 0,00 22,22 6035,35 > >>>>> 12:31:28 0 14,14 0,00 56,57 0,00 2,02 5,05 > >>>>> 0,00 22,22 6035,35 > >>>>> > >>>>> 12:31:28 CPU %user %nice %sys %iowait %irq %soft > >>>>> %steal %idle intr/s > >>>>> 12:31:29 all 24,75 0,00 67,33 0,00 0,99 6,93 > >>>>> 0,00 0,00 5483,17 > >>>>> 12:31:29 0 24,75 0,00 67,33 0,00 0,99 6,93 > >>>>> 0,00 0,00 5483,17 > >>>>> > >>>>> > >>>>> The CPU is going from 89% idle to 0% in less than 2 seconds. > >>>>> > >>>>> I know that I don't have to expect too much from this kind of > >>>>> hardware, > >>>>> but it seems strange that the CPU power vanished so suddenly. > >>>>> > >>>>> Thanks a lot for the guys that have read this long mail :p > >>>>> > >>>>> kind regards, > >>>>> rod > >>>>> > >>>>> > >>>>> _______________________________________________ > >>>>> Freeswitch-users mailing list > >>>>> Freeswitch-users at lists.freeswitch.org > >>>>> > >>>>> > >>>> > >>>> > >>>> > >>>>> >>>>> > >>>>> > >>>> > > >>>> > >>>> > >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>>>> > >>>>> > >>>>> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > >>>> > >>>> > >>>>> http://www.freeswitch.org > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> -- > >>>>> Anthony Minessale II > >>>>> > >>>>> FreeSWITCH http://www.freeswitch.org/ > >>>>> ClueCon http://www.cluecon.com/ > >>>>> > >>>>> AIM: anthm > >>>>> MSN:anthony_minessale at hotmail.com > >>>>> > >>>>> > >>>> > > > >>>> > >>>> > >>>>> > >>>>> > >>>>> > >>>> > >> > >>>> > >>>> > >>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > >>>>> > >>>>> > >>>> > > > >>>> > >>>> > >>>>> > >>>>> > >>>>> > >>>> > >> > >>>> > >>>> > >>>>> IRC: irc.freenode.net > >>>>> > >>>>> > >>>> #freeswitch > >>>> > >>>> > >>>>> FreeSWITCH Developer Conference > >>>>> sip:888 at conference.freeswitch.org > >>>>> > >>>>> > >>>> > > > >>>> > >>>> > >>>>> > >>>>> > >>>>> > >>>> > >> > >>>> > >>>> > >>>>> iax:guest at conference.freeswitch.org/888 > >>>>> > >>>>> > >>>> > >>>> > >>>> > >>>>> > >>>>> googletalk:conf+888 at conference.freeswitch.org > >>>>> > >>>>> > >>>> > > > >>>> > >>>> > >>>>> > >>>>> > >>>>> > >>>> > >> > >>>> > >>>> > >>>>> pstn:213-799-1400 > >>>>> > >>>>> > >>>>> > > ------------------------------------------------------------------------ > > > >>>> > >>>> > >>>>> _______________________________________________ > >>>>> Freeswitch-users mailing list > >>>>> Freeswitch-users at lists.freeswitch.org > >>>>> > >>>>> > >>>> > >>>> > >>>> > >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>>>> > >>>>> > >>>>> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > >>>> > >>>> > >>>>> http://www.freeswitch.org > >>>>> > >>>>> > >>>>> > >>>> _______________________________________________ > >>>> Freeswitch-users mailing list > >>>> Freeswitch-users at lists.freeswitch.org > >>>> > >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>>> > >>>> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > >>>> http://www.freeswitch.org > >>>> > >>>> > >>>> > ------------------------------------------------------------------------ > >>>> > >>>> _______________________________________________ > >>>> Freeswitch-users mailing list > >>>> Freeswitch-users at lists.freeswitch.org > >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >>>> http://www.freeswitch.org > >>>> > >>>> > >>>> > >>> _______________________________________________ > >>> Freeswitch-users mailing list > >>> Freeswitch-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> http://www.freeswitch.org > >>> > >>> > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > >> > >> > >> > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090202/e58251ab/attachment-0002.html From anthony.minessale at gmail.com Mon Feb 2 06:01:25 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 2 Feb 2009 08:01:25 -0600 Subject: [Freeswitch-users] Conference dialing and uuid In-Reply-To: <9FE1733C-A46A-4A17-8AEE-077DE8010235@freeswitch.org> References: <20090130103944.GA23536@cpdata.co.za> <20090130133315.GB23536@cpdata.co.za> <191c3a030901300605r78dd8943w7fb1075ef851e1d@mail.gmail.com> <20090202085157.GA3555@cpdata.co.za> <0F863750-5C77-4CD9-ADC6-CFDDF2B149B4@freeswitch.org> <20090202102459.GA4179@cpdata.co.za> <9FE1733C-A46A-4A17-8AEE-077DE8010235@freeswitch.org> Message-ID: <191c3a030902020601v1109b1e9he867e9f1e1d4a401@mail.gmail.com> you could set the conference moh sound to be tone_stream::// with the teletone spec for ring sound and it use ignore_early_media=true in your originates so the first caller would hear ringback until the 2nd one arrived. On Mon, Feb 2, 2009 at 4:29 AM, Brian West wrote: > Loopback will not work in that case either. If the far end plays > ringback inband you should hear that if you use the conference dial > api call. > > /b > > On Feb 2, 2009, at 4:24 AM, Sias Mey wrote: > > > Aaah ok. > > > > Thanks for clearing that up. > > > > So using loopback is still the only real workable sollution for me, > > since that generates ringback from and alternative endpoint and > > plays it > > into the conference. > > > > I might play with some javascript that streams ring into the channel > > eventually but for now the string comparisons at least get me the > > right > > uuid. > > > > Thank you again, > > Sias > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090202/1754895a/attachment-0002.html From anthony.minessale at gmail.com Mon Feb 2 06:06:14 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 2 Feb 2009 08:06:14 -0600 Subject: [Freeswitch-users] Call Variable not available when call hangup In-Reply-To: <21789152.post@talk.nabble.com> References: <21788550.post@talk.nabble.com> <21789152.post@talk.nabble.com> Message-ID: <191c3a030902020606r1a42ef44n7a73bd1e5157392e@mail.gmail.com> the leg you are running the script on is not hungup, the other leg of the call is. If it was hungup you would not be executing the script. Asterisk and the h ext and the whole dead-agi thing are all poor design showing it's teeth. We do not support anything like it. You can however try this: (see the link below) http://www.nabble.com/Re:-Conference-javascript-and-hanuphooks-giving-me-headaches-p21614840.html On Mon, Feb 2, 2009 at 6:53 AM, shehzad p wrote: > > Is there any settings that when call hangup control can be transferred to > another context and these CDR values can be accessible there? (just like in > Asterisk, h extension) > > shehzad p wrote: > > > > Hi all, > > > > I need to process some CDR variables in Dialplan, like call duration, > > Answered time etc. > > but when I place info application after bridge, it is not listing them > > properly as below: > > =========================================== > > Caller-Channel-Created-Time: [1233573341672157] > > Caller-Channel-Answered-Time: [1233573342712939] > > Caller-Channel-Hangup-Time: [0] > > ========================================== > > Here Hangup time is 0, So how can I find actual values? > > > > --I know that we can use xml_cdr or cdr_csv, but my current need is to > get > > those values from dialplan itself so that can be passed to some script... > > > > > > thanks, > > msp > > > > -- > View this message in context: > http://www.nabble.com/Call-Variable-not-available-when-call-hangup-tp21788550p21789152.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090202/6d471696/attachment-0002.html From saeedahmad1981 at gmail.com Mon Feb 2 06:29:49 2009 From: saeedahmad1981 at gmail.com (Saeed Ahmed) Date: Mon, 2 Feb 2009 15:29:49 +0100 Subject: [Freeswitch-users] Strange Performance when using as SBC In-Reply-To: <191c3a030902020557v70d3777fqb83e591385a64a59@mail.gmail.com> References: <4986C42C.7030700@laposte.net><1C31435E19AB4E57981CF2AAC2CA74AA@SaeedLaptop><4986ECA9.3040707@laposte.net><9232C06D5362494791AF713E1DF61343@SaeedLaptop> <191c3a030902020557v70d3777fqb83e591385a64a59@mail.gmail.com> Message-ID: <0AF29744A97A482CB167B5C1F185F8B6@SaeedLaptop> Its Ubuntu 8.04 Hardy, 2.6.24-16 kernel. I hope it will be OK Kind Regards Saeed Ahmed Tariq _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Monday, February 02, 2009 2:58 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Strange Performance when using as SBC if you want to use ubuntu, be sure to use hardy and not intrepid. On Mon, Feb 2, 2009 at 7:03 AM, Saeed Ahmed wrote: Thanks rod for a quick answer, FS is installed on Ubuntu Server. I am planning to replace Nextone SBC with FS, Later I'll also use openZAP to communicate with TDM but this all depends how much calls it can take, or maybe we can also do something in clustering environment ( I am not sure about it). But thanks again and any further help will be highly appreciated. Kind Regards Saeed Ahmed Tariq -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of rod Sent: Monday, February 02, 2009 1:53 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Strange Performance when using as SBC Hi Saeed, I just created an account to share my setup on the wiki. I will detail all the steps for a clean install of a debian64 lenny with FS used as a SBC (next step is to try the new LCR module :) )and what I'm doing do stress the server. I wrote nothing at this time so please be patient, I'm waiting for my new hardware so that I will detail as much as possible what I'll do. For beginning I suggest you reading the start page on the wiki, especially these pages: -http://wiki.freeswitch.org/wiki/Getting_Started_Guide -http://wiki.freeswitch.org/wiki/Dialplan_XML maybe you could tell more about the linux distribution you're using so that I can give you some pointers for sipp... regards. rod. Saeed Ahmed wrote: > Hi Rod, > > Could you please share how you configured Sipp & FS to create a test > environment? Especially the dial plan, sofia settings etc..., actually I am > a newbie. I want to test it on a single FS machine. > > Kind Regards > Saeed > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of rod > Sent: Monday, February 02, 2009 11:00 AM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Strange Performance when using as SBC > > Hi Ken, Jay, > > thanks for pointing to proxy media, I will test. > > Ken, you are right, I was brain damaged (a stupid mistake) when setting > INFO cause this kind of level could be very verbose. I'm switching to > CRIT or ERR. > > Thanks guys, > rod. > > thanks for > > Ken Rice wrote: > >> If you don't have to transcode, using proxy media mode will still save you >> some CPU time. This is 1/2 way between bypass media and the default media >> interactive mode. The other draw back to this mode is if you are using FS >> > to > >> clean up RTP and DTMF you loose those functions but they are not needed in >> most use cases. >> >> As far as the log level goes, I found that once I had things stable >> > setting > >> the loglevel to helped a good deal... Info is probably a bit too high of a >> loglevel I would probably go for CRIT or ERR (2 or 1 respectively) if you >> insist on leaving logging turned on... On a busy system these can and will >> generate a good deal of activity (and disk IO if using mod_logfile) >> >> Ken >> >> >> >> >>> From: rod >>> Reply-To: >>> Date: Mon, 02 Feb 2009 11:36:35 +0400 >>> To: >>> Subject: Re: [Freeswitch-users] Strange Performance when using as SBC >>> >>> Hi Ken, >>> >>> 1) I'd like to use FS to hide topology, so bypass media is not possible >>> 2) done >>> 3) done >>> 4) not used >>> 5) i'm using this ins switch.xml -> >> value="info"/>, if you think an other log level is more suitable. >>> >>> Regarding logging, I can see in console and in the freeswitch.log that >>> there is still a lot of NOTICE logging, see below: >>> 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:960 >>> switch_core_session_thread() Session 8721 >>> (sofia/internal/sipp at 10.10.10.1:5060) Ended >>> 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:962 >>> switch_core_session_thread() Close Channel >>> sofia/internal/sipp at 10.10.10.1:5060 [CS_HANGUP] >>> 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:960 >>> switch_core_session_thread() Session 8722 >>> (sofia/external/9998 at 10.10.20.100) Ended >>> 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:962 >>> switch_core_session_thread() Close Channel >>> sofia/external/9998 at 10.10.20.100 [CS_HANGUP] >>> 2009-02-02 08:33:56 [NOTICE] sofia.c:3164 sofia_handle_sip_i_state() >>> Channel [sofia/external/9998 at 10.10.20.100] has been answered >>> 2009-02-02 08:33:56 [WARNING] mod_sofia.c:740 sofia_read_frame() >>> Changing codec ptime to 30. I bet you have a linksys/sipura =D >>> >>> Do you have any idea where I can switch off this kind of logging. I >>> thought it should be in /dialplan/internal.xml, but I see that in >>> internal.xml -> >>> >>> thanks a lot for your suggestion. >>> >>> regards, >>> rod >>> >>> Ken Rice wrote: >>> >>> >>>> Dont forget there are several things you can do to increase >>>> > performance... > >>>> 1) where possible use bypass media or media proxy modes >>>> 2) mount freeswitch/db as a ram drive (if you are using voicemail with >>>> the internal FS DBs you'll need a way to make this persistant across >>>> reboots) >>>> 3) see the wiki for setting reasonable ulimits >>>> 4) (this is my oppinion others may vary) dont use mod_cdr_csv >>>> 5) turn off (or reduce logging) in switch.conf.xml >>>> >>>> all of these thing can greatly improve performance. >>>> >>>> On Mon, Feb 2, 2009 at 1:04 AM, rod >>> > wrote: >>>> >>>> Thanks Anthony, >>>> >>>> the setup is like this: >>>> >>>> sipp server ---- FS 1 ---- FS2 >>>> >>>> FS1 is the AMD CPU that has only one extension in dialplan that >>>> bridges >>>> 9999 to FS2. 9999 is the first extension in FS2 dialplan that >>>> plays moh, >>>> FS2 has no CPU pbm. >>>> >>>> FS1 is maxing out at 60 bridged calls without your option -hp. >>>> >>>> Using -hp, I'm now able to bridge 200 concurrent calls (a great >>>> improvement) and the system is still reactive. CPU load is high >>>> but not >>>> 100% and as the system responds well, I think that doesn't matter. >>>> > The > >>>> 2GB of memory are completely consumed (top command shows 700MB for >>>> > FS > >>>> process). >>>> >>>> I understand that FS1 server is not the best hardware platform, >>>> and I'm >>>> waiting for new 4 cores server for testing. >>>> I will update those numbers when testing with the new hardware. >>>> >>>> regards, >>>> rod. >>>> >>>> Anthony Minessale wrote: >>>> >>>> >>>>> Which of the 2 machines has the load issue? You said it was one box >>>>> calling the other. >>>>> >>>>> You have 2 major things against you, single CPU and AMD, but you >>>>> should at least be able to get in the vicinity of 800-1000 calls >>>>> >>>>> >>>> on a >>>> >>>> >>>>> box like that. >>>>> >>>>> Are you calling the default 9999? It's not really an appropriate >>>>> extension for load testing. >>>>> On the terminating box you should set up a manual extension that is >>>>> the first one in the dial plan >>>>> to play a wav file from preferably a ram disk or /tmp >>>>> >>>>> If you do plan on using this in production accept nothing less >>>>> >>>>> >>>> than a >>>> >>>> >>>>> multi-core intel machine with at least 4 cores, the more cores the >>>>> better because that parallel processing is where FS gets it's >>>>> >>>>> >>>> atvantage. >>>> >>>> >>>>> On Fri, Jan 30, 2009 at 5:56 AM, rod >>>> >>>>> >>>> >>>> >>>> >>>>> >> wrote: >>>>> >>>>> Dear list, >>>>> >>>>> I've been playing with freeswitch for some time (2 months) >>>>> >>>>> >>>> and the >>>> >>>> >>>>> fact >>>>> is that I'm very pleased with the functionnalities of this >>>>> >>>>> >>>> software. >>>> >>>> >>>>> I'd like to use FS as a SBC handling media and I'm doing some >>>>> tests with >>>>> sipp to load the machine but I'm unable to bridge more than >>>>> >>>>> >>>> 60 calls >>>> >>>> >>>>> without seeing the CPU being loaded at 100%. I'm sure >>>>> >>>>> >>>> something is >>>> >>>> >>>>> going >>>>> wrong with my setup but I'm unable to see what. >>>>> >>>>> The test machine has the following specs: >>>>> Athlon XP 3500+ with 2GB of memory (I know this is not a >>>>> >>>>> >>>> high end >>>> >>>> >>>>> machine :p) >>>>> >>>>> Freeswitch:/opt/freeswitch/log# cat /proc/cpuinfo >>>>> processor : 0 >>>>> vendor_id : AuthenticAMD >>>>> cpu family : 15 >>>>> model : 95 >>>>> model name : AMD Athlon(tm) 64 Processor 3500+ >>>>> stepping : 2 >>>>> cpu MHz : 2199.973 >>>>> cache size : 512 KB >>>>> fpu : yes >>>>> fpu_exception : yes >>>>> cpuid level : 1 >>>>> wp : yes >>>>> flags : fpu vme de pse tsc msr pae mce cx8 apic >>>>> >>>>> >>>> sep mtrr pge >>>> >>>> >>>>> mca cmov pat pse36 clflush mmx fxsr sse sse2 syscall nx mmxext >>>>> fxsr_opt >>>>> rdtscp lm 3dnowext 3dnow up rep_good pni cx16 lahf_lm svm >>>>> >>>>> >>>> extapic >>>> >>>> >>>>> cr8_legacy >>>>> bogomips : 4402.97 >>>>> TLB size : 1024 4K pages >>>>> clflush size : 64 >>>>> cache_alignment : 64 >>>>> address sizes : 40 bits physical, 48 bits virtual >>>>> power management: ts fid vid ttp tm stc >>>>> >>>>> I installed FS on a fresh debian 64: >>>>> Linux Freeswitch 2.6.26-1-amd64 #1 SMP Sat Jan 10 17:57:00 >>>>> >>>>> >>>> UTC 2009 >>>> >>>> >>>>> x86_64 GNU/Linux >>>>> >>>>> I set the ulimit parameters like those on the website: >>>>> freeswitch at internal> ... >>>>> Freeswitch:/opt/free-svn/bin# ulimit -a >>>>> core file size (blocks, -c) unlimited >>>>> data seg size (kbytes, -d) unlimited >>>>> scheduling priority (-e) 0 >>>>> file size (blocks, -f) unlimited >>>>> pending signals (-i) unlimited >>>>> max locked memory (kbytes, -l) unlimited >>>>> max memory size (kbytes, -m) unlimited >>>>> open files (-n) 999999 >>>>> pipe size (512 bytes, -p) 8 >>>>> POSIX message queues (bytes, -q) unlimited >>>>> real-time priority (-r) 0 >>>>> stack size (kbytes, -s) 244 >>>>> cpu time (seconds, -t) unlimited >>>>> max user processes (-u) unlimited >>>>> virtual memory (kbytes, -v) unlimited >>>>> file locks (-x) unlimited >>>>> >>>>> >>>>> My network setup is the following: >>>>> >>>>> SIPP machine (10.10.10.1/24)----------------vlan >>>>> >>>>> >>>> >>>> >>>> >>>>> 55 >>>>> ----------(10.10.10.254/24 >>>>> >>>>> >>>> ) FS >>>> >>>> >>>>> (10.10.20.254/24)-------------- >>>>> >>>>> >>>> >>>> >>>> >>>>> vlan56 >>>>> -------------------(10.10.20.100/24 >>>>> >>>>> >>>> ) >>>> >>>> >>>>> OTHER STOCK FS >>>>> >>>>> >>>>> I launched sipp with: >>>>> sipp -sn uac_pcap -s 9999 -r 10 -l 80 -d 60000 -mi >>>>> >>>>> >>>> 10.10.10.1 -i >>>> >>>> >>>>> 10.10.10.1 -mp 25000 10.10.10.254:5060 >>>>> >>>>> >>>> >>>> >>>> >>>>> The dialplan on FS is very simple: >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>> data="sofia/external/9999 at 10.10.20.100 >>>>> >>>>> >>>> >>> >"/> >>>> >>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> FreeSWITCH Version 1.0.trunk (11560M) Started. >>>>> Crash Protection [Disabled] >>>>> Max Sessions[1000] >>>>> Session Rate[100] >>>>> SQL [Enabled] >>>>> >>>>> >>>>> The test is very simple: sipp dial 9999 that matches in my >>>>> >>>>> >>>> FS dialplan >>>> >>>> >>>>> and this is bridged to an other FS machine playing music on >>>>> >>>>> >>>> hold. >>>> >>>> >>>>> When I launch "top" I see after 30 to 40 s that FS consumes all >>>>> the CPU >>>>> ressources (with a mean of 50-60 % before), with 80 calls. >>>>> When I set 70 calls, I have to wait 70-80 s before seeing >>>>> >>>>> >>>> the same >>>> >>>> >>>>> issue. >>>>> >>>>> Presence is set to false on the 2 profile. >>>>> >>>>> I have the same issue with FS 1.0.2 that' s why I tried FS >>>>> >>>>> >>>> 11560. >>>> >>>> >>>>> When I use the FS machine as a router to test the packet per >>>>> >>>>> >>>> second >>>> >>>> >>>>> performance, I'm reaching 100Mbps with 8000pps in each >>>>> >>>>> >>>> direction (from >>>> >>>> >>>>> vlan 55 to vlan56) with less than 12% CPU. So that I don't think >>>>> there's >>>>> an issue with the network. >>>>> >>>>> Here is an "mpstat -P ALL 1" to show you what's happening >>>>> >>>>> >>>> suddenly >>>> >>>> >>>>> with >>>>> 70 bridge calls: >>>>> 12:31:26 CPU %user %nice %sys %iowait %irq %soft >>>>> %steal %idle intr/s >>>>> 12:31:27 all 3,00 0,00 3,00 0,00 1,00 4,00 >>>>> 0,00 89,00 6241,00 >>>>> 12:31:27 0 3,00 0,00 3,00 0,00 1,00 4,00 >>>>> 0,00 89,00 6241,00 >>>>> >>>>> 12:31:27 CPU %user %nice %sys %iowait %irq %soft >>>>> %steal %idle intr/s >>>>> 12:31:28 all 14,14 0,00 56,57 0,00 2,02 5,05 >>>>> 0,00 22,22 6035,35 >>>>> 12:31:28 0 14,14 0,00 56,57 0,00 2,02 5,05 >>>>> 0,00 22,22 6035,35 >>>>> >>>>> 12:31:28 CPU %user %nice %sys %iowait %irq %soft >>>>> %steal %idle intr/s >>>>> 12:31:29 all 24,75 0,00 67,33 0,00 0,99 6,93 >>>>> 0,00 0,00 5483,17 >>>>> 12:31:29 0 24,75 0,00 67,33 0,00 0,99 6,93 >>>>> 0,00 0,00 5483,17 >>>>> >>>>> >>>>> The CPU is going from 89% idle to 0% in less than 2 seconds. >>>>> >>>>> I know that I don't have to expect too much from this kind of >>>>> hardware, >>>>> but it seems strange that the CPU power vanished so suddenly. >>>>> >>>>> Thanks a lot for the guys that have read this long mail :p >>>>> >>>>> kind regards, >>>>> rod >>>>> >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> >>>>> >>>> >>>>> >>>> > >>>> >>>> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> >>>>> >>>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >>>> >>>> >>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> >>>>> >>>>> -- >>>>> Anthony Minessale II >>>>> >>>>> FreeSWITCH http://www.freeswitch.org/ >>>>> ClueCon http://www.cluecon.com/ >>>>> >>>>> AIM: anthm >>>>> MSN:anthony_minessale at hotmail.com >>>>> >>>>> >>>> > >>>> >>>> >>>>> >>>>> >>>>> >>>> >> >>>> >>>> >>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>>>> >>>>> >>>> > >>>> >>>> >>>>> >>>>> >>>>> >>>> >> >>>> >>>> >>>>> IRC: irc.freenode.net >>>>> >>>>> >>>> #freeswitch >>>> >>>> >>>>> FreeSWITCH Developer Conference >>>>> sip:888 at conference.freeswitch.org >>>>> >>>>> >>>> > >>>> >>>> >>>>> >>>>> >>>>> >>>> >> >>>> >>>> >>>>> iax:guest at conference.freeswitch.org/888 >>>>> >>>>> >>>> >>>> >>>> >>>>> >>>>> googletalk:conf+888 at conference.freeswitch.org >>>>> >>>>> >>>> > >>>> >>>> >>>>> >>>>> >>>>> >>>> >> >>>> >>>> >>>>> pstn:213-799-1400 >>>>> >>>>> >>>>> > ------------------------------------------------------------------------ > >>>> >>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> >>>>> >>>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >>>> >>>> >>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> >>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >>>> http://www.freeswitch.org >>>> >>>> >>>> ------------------------------------------------------------------------ >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 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URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090202/72e54e09/attachment-0002.html From kawarod at laposte.net Mon Feb 2 06:33:22 2009 From: kawarod at laposte.net (rod) Date: Mon, 02 Feb 2009 18:33:22 +0400 Subject: [Freeswitch-users] Strange Performance when using as SBC In-Reply-To: <9232C06D5362494791AF713E1DF61343@SaeedLaptop> References: <4986C42C.7030700@laposte.net><1C31435E19AB4E57981CF2AAC2CA74AA@SaeedLaptop> <4986ECA9.3040707@laposte.net> <9232C06D5362494791AF713E1DF61343@SaeedLaptop> Message-ID: <49870432.5050301@laposte.net> Hi Saeed, Here is a first draft of what I did to install FS on my server. Configuration are not present, they'll be in a next release :p http://wiki.freeswitch.org/wiki/SBC_Setup My aim is to setup FS as a SBC, I hope this page could be a great startup point for others. I will update regularly based on what I did. Saeed, why are you replacing your Nextone, it's said to be one of the best commercial SBC on the market. regards. Saeed Ahmed wrote: > Thanks rod for a quick answer, > > FS is installed on Ubuntu Server. > > I am planning to replace Nextone SBC with FS, Later I'll also use openZAP to > communicate with TDM but this all depends how much calls it can take, or > maybe we can also do something in clustering environment ( I am not sure > about it). But thanks again and any further help will be highly appreciated. > > > Kind Regards > Saeed Ahmed Tariq > > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of rod > Sent: Monday, February 02, 2009 1:53 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Strange Performance when using as SBC > > Hi Saeed, > > I just created an account to share my setup on the wiki. I will detail > all the steps for a clean install of a debian64 lenny with FS used as a > SBC (next step is to try the new LCR module :) )and what I'm doing do > stress the server. > > I wrote nothing at this time so please be patient, I'm waiting for my > new hardware so that I will detail as much as possible what I'll do. > > For beginning I suggest you reading the start page on the wiki, > especially these pages: > -http://wiki.freeswitch.org/wiki/Getting_Started_Guide > -http://wiki.freeswitch.org/wiki/Dialplan_XML > > maybe you could tell more about the linux distribution you're using so > that I can give you some pointers for sipp... > > regards. > rod. > > > Saeed Ahmed wrote: > >> Hi Rod, >> >> Could you please share how you configured Sipp & FS to create a test >> environment? Especially the dial plan, sofia settings etc..., actually I >> > am > >> a newbie. I want to test it on a single FS machine. >> >> Kind Regards >> Saeed >> -----Original Message----- >> From: freeswitch-users-bounces at lists.freeswitch.org >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of rod >> Sent: Monday, February 02, 2009 11:00 AM >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] Strange Performance when using as SBC >> >> Hi Ken, Jay, >> >> thanks for pointing to proxy media, I will test. >> >> Ken, you are right, I was brain damaged (a stupid mistake) when setting >> INFO cause this kind of level could be very verbose. I'm switching to >> CRIT or ERR. >> >> Thanks guys, >> rod. >> >> thanks for >> >> Ken Rice wrote: >> >> >>> If you don't have to transcode, using proxy media mode will still save >>> > you > >>> some CPU time. This is 1/2 way between bypass media and the default media >>> interactive mode. The other draw back to this mode is if you are using FS >>> >>> >> to >> >> >>> clean up RTP and DTMF you loose those functions but they are not needed >>> > in > >>> most use cases. >>> >>> As far as the log level goes, I found that once I had things stable >>> >>> >> setting >> >> >>> the loglevel to helped a good deal... Info is probably a bit too high of >>> > a > >>> loglevel I would probably go for CRIT or ERR (2 or 1 respectively) if you >>> insist on leaving logging turned on... On a busy system these can and >>> > will > >>> generate a good deal of activity (and disk IO if using mod_logfile) >>> >>> Ken >>> >>> >>> >>> >>> >>>> From: rod >>>> Reply-To: >>>> Date: Mon, 02 Feb 2009 11:36:35 +0400 >>>> To: >>>> Subject: Re: [Freeswitch-users] Strange Performance when using as SBC >>>> >>>> Hi Ken, >>>> >>>> 1) I'd like to use FS to hide topology, so bypass media is not possible >>>> 2) done >>>> 3) done >>>> 4) not used >>>> 5) i'm using this ins switch.xml -> >>> value="info"/>, if you think an other log level is more suitable. >>>> >>>> Regarding logging, I can see in console and in the freeswitch.log that >>>> there is still a lot of NOTICE logging, see below: >>>> 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:960 >>>> switch_core_session_thread() Session 8721 >>>> (sofia/internal/sipp at 10.10.10.1:5060) Ended >>>> 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:962 >>>> switch_core_session_thread() Close Channel >>>> sofia/internal/sipp at 10.10.10.1:5060 [CS_HANGUP] >>>> 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:960 >>>> switch_core_session_thread() Session 8722 >>>> (sofia/external/9998 at 10.10.20.100) Ended >>>> 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:962 >>>> switch_core_session_thread() Close Channel >>>> sofia/external/9998 at 10.10.20.100 [CS_HANGUP] >>>> 2009-02-02 08:33:56 [NOTICE] sofia.c:3164 sofia_handle_sip_i_state() >>>> Channel [sofia/external/9998 at 10.10.20.100] has been answered >>>> 2009-02-02 08:33:56 [WARNING] mod_sofia.c:740 sofia_read_frame() >>>> Changing codec ptime to 30. I bet you have a linksys/sipura =D >>>> >>>> Do you have any idea where I can switch off this kind of logging. I >>>> thought it should be in /dialplan/internal.xml, but I see that in >>>> internal.xml -> >>>> >>>> thanks a lot for your suggestion. >>>> >>>> regards, >>>> rod >>>> >>>> Ken Rice wrote: >>>> >>>> >>>> >>>>> Dont forget there are several things you can do to increase >>>>> >>>>> >> performance... >> >> >>>>> 1) where possible use bypass media or media proxy modes >>>>> 2) mount freeswitch/db as a ram drive (if you are using voicemail with >>>>> the internal FS DBs you'll need a way to make this persistant across >>>>> reboots) >>>>> 3) see the wiki for setting reasonable ulimits >>>>> 4) (this is my oppinion others may vary) dont use mod_cdr_csv >>>>> 5) turn off (or reduce logging) in switch.conf.xml >>>>> >>>>> all of these thing can greatly improve performance. >>>>> >>>>> On Mon, Feb 2, 2009 at 1:04 AM, rod >>>> > wrote: >>>>> >>>>> Thanks Anthony, >>>>> >>>>> the setup is like this: >>>>> >>>>> sipp server ---- FS 1 ---- FS2 >>>>> >>>>> FS1 is the AMD CPU that has only one extension in dialplan that >>>>> bridges >>>>> 9999 to FS2. 9999 is the first extension in FS2 dialplan that >>>>> plays moh, >>>>> FS2 has no CPU pbm. >>>>> >>>>> FS1 is maxing out at 60 bridged calls without your option -hp. >>>>> >>>>> Using -hp, I'm now able to bridge 200 concurrent calls (a great >>>>> improvement) and the system is still reactive. CPU load is high >>>>> but not >>>>> 100% and as the system responds well, I think that doesn't matter. >>>>> >>>>> >> The >> >> >>>>> 2GB of memory are completely consumed (top command shows 700MB for >>>>> >>>>> >> FS >> >> >>>>> process). >>>>> >>>>> I understand that FS1 server is not the best hardware platform, >>>>> and I'm >>>>> waiting for new 4 cores server for testing. >>>>> I will update those numbers when testing with the new hardware. >>>>> >>>>> regards, >>>>> rod. >>>>> >>>>> Anthony Minessale wrote: >>>>> >>>>> >>>>> >>>>>> Which of the 2 machines has the load issue? You said it was one box >>>>>> calling the other. >>>>>> >>>>>> You have 2 major things against you, single CPU and AMD, but you >>>>>> should at least be able to get in the vicinity of 800-1000 calls >>>>>> >>>>>> >>>>>> >>>>> on a >>>>> >>>>> >>>>> >>>>>> box like that. >>>>>> >>>>>> Are you calling the default 9999? It's not really an appropriate >>>>>> extension for load testing. >>>>>> On the terminating box you should set up a manual extension that is >>>>>> the first one in the dial plan >>>>>> to play a wav file from preferably a ram disk or /tmp >>>>>> >>>>>> If you do plan on using this in production accept nothing less >>>>>> >>>>>> >>>>>> >>>>> than a >>>>> >>>>> >>>>> >>>>>> multi-core intel machine with at least 4 cores, the more cores the >>>>>> better because that parallel processing is where FS gets it's >>>>>> >>>>>> >>>>>> >>>>> atvantage. >>>>> >>>>> >>>>> >>>>>> On Fri, Jan 30, 2009 at 5:56 AM, rod >>>>> >>>>>> >>>>>> >>>>> >>>>> >>>>> >>>>> >>>>>> >> wrote: >>>>>> >>>>>> Dear list, >>>>>> >>>>>> I've been playing with freeswitch for some time (2 months) >>>>>> >>>>>> >>>>>> >>>>> and the >>>>> >>>>> >>>>> >>>>>> fact >>>>>> is that I'm very pleased with the functionnalities of this >>>>>> >>>>>> >>>>>> >>>>> software. >>>>> >>>>> >>>>> >>>>>> I'd like to use FS as a SBC handling media and I'm doing some >>>>>> tests with >>>>>> sipp to load the machine but I'm unable to bridge more than >>>>>> >>>>>> >>>>>> >>>>> 60 calls >>>>> >>>>> >>>>> >>>>>> without seeing the CPU being loaded at 100%. I'm sure >>>>>> >>>>>> >>>>>> >>>>> something is >>>>> >>>>> >>>>> >>>>>> going >>>>>> wrong with my setup but I'm unable to see what. >>>>>> >>>>>> The test machine has the following specs: >>>>>> Athlon XP 3500+ with 2GB of memory (I know this is not a >>>>>> >>>>>> >>>>>> >>>>> high end >>>>> >>>>> >>>>> >>>>>> machine :p) >>>>>> >>>>>> Freeswitch:/opt/freeswitch/log# cat /proc/cpuinfo >>>>>> processor : 0 >>>>>> vendor_id : AuthenticAMD >>>>>> cpu family : 15 >>>>>> model : 95 >>>>>> model name : AMD Athlon(tm) 64 Processor 3500+ >>>>>> stepping : 2 >>>>>> cpu MHz : 2199.973 >>>>>> cache size : 512 KB >>>>>> fpu : yes >>>>>> fpu_exception : yes >>>>>> cpuid level : 1 >>>>>> wp : yes >>>>>> flags : fpu vme de pse tsc msr pae mce cx8 apic >>>>>> >>>>>> >>>>>> >>>>> sep mtrr pge >>>>> >>>>> >>>>> >>>>>> mca cmov pat pse36 clflush mmx fxsr sse sse2 syscall nx mmxext >>>>>> fxsr_opt >>>>>> rdtscp lm 3dnowext 3dnow up rep_good pni cx16 lahf_lm svm >>>>>> >>>>>> >>>>>> >>>>> extapic >>>>> >>>>> >>>>> >>>>>> cr8_legacy >>>>>> bogomips : 4402.97 >>>>>> TLB size : 1024 4K pages >>>>>> clflush size : 64 >>>>>> cache_alignment : 64 >>>>>> address sizes : 40 bits physical, 48 bits virtual >>>>>> power management: ts fid vid ttp tm stc >>>>>> >>>>>> I installed FS on a fresh debian 64: >>>>>> Linux Freeswitch 2.6.26-1-amd64 #1 SMP Sat Jan 10 17:57:00 >>>>>> >>>>>> >>>>>> >>>>> UTC 2009 >>>>> >>>>> >>>>> >>>>>> x86_64 GNU/Linux >>>>>> >>>>>> I set the ulimit parameters like those on the website: >>>>>> freeswitch at internal> ... >>>>>> Freeswitch:/opt/free-svn/bin# ulimit -a >>>>>> core file size (blocks, -c) unlimited >>>>>> data seg size (kbytes, -d) unlimited >>>>>> scheduling priority (-e) 0 >>>>>> file size (blocks, -f) unlimited >>>>>> pending signals (-i) unlimited >>>>>> max locked memory (kbytes, -l) unlimited >>>>>> max memory size (kbytes, -m) unlimited >>>>>> open files (-n) 999999 >>>>>> pipe size (512 bytes, -p) 8 >>>>>> POSIX message queues (bytes, -q) unlimited >>>>>> real-time priority (-r) 0 >>>>>> stack size (kbytes, -s) 244 >>>>>> cpu time (seconds, -t) unlimited >>>>>> max user processes (-u) unlimited >>>>>> virtual memory (kbytes, -v) unlimited >>>>>> file locks (-x) unlimited >>>>>> >>>>>> >>>>>> My network setup is the following: >>>>>> >>>>>> SIPP machine (10.10.10.1/24)----------------vlan >>>>>> >>>>>> >>>>>> >>>>> >>>>> >>>>> >>>>> >>>>>> 55 >>>>>> ----------(10.10.10.254/24 >>>>>> >>>>>> >>>>>> >>>>> ) FS >>>>> >>>>> >>>>> >>>>>> (10.10.20.254/24)-------------- >>>>>> >>>>>> >>>>>> >>>>> >>>>> >>>>> >>>>> >>>>>> vlan56 >>>>>> -------------------(10.10.20.100/24 >>>>>> >>>>>> >>>>>> >>>>> ) >>>>> >>>>> >>>>> >>>>>> OTHER STOCK FS >>>>>> >>>>>> >>>>>> I launched sipp with: >>>>>> sipp -sn uac_pcap -s 9999 -r 10 -l 80 -d 60000 -mi >>>>>> >>>>>> >>>>>> >>>>> 10.10.10.1 -i >>>>> >>>>> >>>>> >>>>>> 10.10.10.1 -mp 25000 10.10.10.254:5060 >>>>>> >>>>>> >>>>>> >>>>> >>>>> >>>>> >>>>> >>>>>> The dialplan on FS is very simple: >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>> data="sofia/external/9999 at 10.10.20.100 >>>>>> >>>>>> >>>>>> >>>>> >>>> >"/> >>>>> >>>>> >>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> FreeSWITCH Version 1.0.trunk (11560M) Started. >>>>>> Crash Protection [Disabled] >>>>>> Max Sessions[1000] >>>>>> Session Rate[100] >>>>>> SQL [Enabled] >>>>>> >>>>>> >>>>>> The test is very simple: sipp dial 9999 that matches in my >>>>>> >>>>>> >>>>>> >>>>> FS dialplan >>>>> >>>>> >>>>> >>>>>> and this is bridged to an other FS machine playing music on >>>>>> >>>>>> >>>>>> >>>>> hold. >>>>> >>>>> >>>>> >>>>>> When I launch "top" I see after 30 to 40 s that FS consumes all >>>>>> the CPU >>>>>> ressources (with a mean of 50-60 % before), with 80 calls. >>>>>> When I set 70 calls, I have to wait 70-80 s before seeing >>>>>> >>>>>> >>>>>> >>>>> the same >>>>> >>>>> >>>>> >>>>>> issue. >>>>>> >>>>>> Presence is set to false on the 2 profile. >>>>>> >>>>>> I have the same issue with FS 1.0.2 that' s why I tried FS >>>>>> >>>>>> >>>>>> >>>>> 11560. >>>>> >>>>> >>>>> >>>>>> When I use the FS machine as a router to test the packet per >>>>>> >>>>>> >>>>>> >>>>> second >>>>> >>>>> >>>>> >>>>>> performance, I'm reaching 100Mbps with 8000pps in each >>>>>> >>>>>> >>>>>> >>>>> direction (from >>>>> >>>>> >>>>> >>>>>> vlan 55 to vlan56) with less than 12% CPU. So that I don't think >>>>>> there's >>>>>> an issue with the network. >>>>>> >>>>>> Here is an "mpstat -P ALL 1" to show you what's happening >>>>>> >>>>>> >>>>>> >>>>> suddenly >>>>> >>>>> >>>>> >>>>>> with >>>>>> 70 bridge calls: >>>>>> 12:31:26 CPU %user %nice %sys %iowait %irq %soft >>>>>> %steal %idle intr/s >>>>>> 12:31:27 all 3,00 0,00 3,00 0,00 1,00 4,00 >>>>>> 0,00 89,00 6241,00 >>>>>> 12:31:27 0 3,00 0,00 3,00 0,00 1,00 4,00 >>>>>> 0,00 89,00 6241,00 >>>>>> >>>>>> 12:31:27 CPU %user %nice %sys %iowait %irq %soft >>>>>> %steal %idle intr/s >>>>>> 12:31:28 all 14,14 0,00 56,57 0,00 2,02 5,05 >>>>>> 0,00 22,22 6035,35 >>>>>> 12:31:28 0 14,14 0,00 56,57 0,00 2,02 5,05 >>>>>> 0,00 22,22 6035,35 >>>>>> >>>>>> 12:31:28 CPU %user %nice %sys %iowait %irq %soft >>>>>> %steal %idle intr/s >>>>>> 12:31:29 all 24,75 0,00 67,33 0,00 0,99 6,93 >>>>>> 0,00 0,00 5483,17 >>>>>> 12:31:29 0 24,75 0,00 67,33 0,00 0,99 6,93 >>>>>> 0,00 0,00 5483,17 >>>>>> >>>>>> >>>>>> The CPU is going from 89% idle to 0% in less than 2 seconds. >>>>>> >>>>>> I know that I don't have to expect too much from this kind of >>>>>> hardware, >>>>>> but it seems strange that the CPU power vanished so suddenly. >>>>>> >>>>>> Thanks a lot for the guys that have read this long mail :p >>>>>> >>>>>> kind regards, >>>>>> rod >>>>>> >>>>>> >>>>>> _______________________________________________ >>>>>> Freeswitch-users mailing list >>>>>> Freeswitch-users at lists.freeswitch.org >>>>>> >>>>>> >>>>>> >>>>> >>>>> >>>>> >>>>> >>>>>> >>>>> >>>>>> >>>>>> >>>>> > >>>>> >>>>> >>>>> >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> >>>>>> >>>>>> >>>>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> >>>>> >>>>> >>>>> >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> -- >>>>>> Anthony Minessale II >>>>>> >>>>>> FreeSWITCH http://www.freeswitch.org/ >>>>>> ClueCon http://www.cluecon.com/ >>>>>> >>>>>> AIM: anthm >>>>>> MSN:anthony_minessale at hotmail.com >>>>>> >>>>>> >>>>>> >>>>> >>>>> >>>>> >>>>> >>>>>> >>>>> >>>>>> >>>>>> >>>>> > >>>>> >>>>> >>>>> >>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>>>>> >>>>>> >>>>>> >>>>> >>>>> >>>>> >>>>> >>>>>> >>>>> >>>>>> >>>>>> >>>>> > >>>>> >>>>> >>>>> >>>>>> IRC: irc.freenode.net >>>>>> >>>>>> >>>>>> >>>>> #freeswitch >>>>> >>>>> >>>>> >>>>>> FreeSWITCH Developer Conference >>>>>> sip:888 at conference.freeswitch.org >>>>>> >>>>>> >>>>>> >>>>> >>>>> >>>>> >>>>> >>>>>> >>>>> >>>>>> >>>>>> >>>>> > >>>>> >>>>> >>>>> >>>>>> iax:guest at conference.freeswitch.org/888 >>>>>> >>>>>> >>>>>> >>>>> >>>>> >>>>> >>>>> >>>>>> >>>>>> googletalk:conf+888 at conference.freeswitch.org >>>>>> >>>>>> >>>>>> >>>>> >>>>> >>>>> >>>>> >>>>>> >>>>> >>>>>> >>>>>> >>>>> > >>>>> >>>>> >>>>> >>>>>> pstn:213-799-1400 >>>>>> >>>>>> >>>>>> >>>>>> >> ------------------------------------------------------------------------ >> >> >>>>> >>>>> >>>>> >>>>>> _______________________________________________ >>>>>> Freeswitch-users mailing list >>>>>> Freeswitch-users at lists.freeswitch.org >>>>>> >>>>>> >>>>>> >>>>> >>>>> >>>>> >>>>> >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> >>>>>> >>>>>> >>>>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> >>>>> >>>>> >>>>> >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>>> >>>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> >>>>> >>>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> >>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> >>>>> > ------------------------------------------------------------------------ > >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> >>>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> >>>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >>> >>> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > From saeedahmad1981 at gmail.com Mon Feb 2 07:15:01 2009 From: saeedahmad1981 at gmail.com (Saeed Ahmed) Date: Mon, 2 Feb 2009 16:15:01 +0100 Subject: [Freeswitch-users] Strange Performance when using as SBC In-Reply-To: <49870432.5050301@laposte.net> References: <4986C42C.7030700@laposte.net><1C31435E19AB4E57981CF2AAC2CA74AA@SaeedLaptop> <4986ECA9.3040707@laposte.net><9232C06D5362494791AF713E1DF61343@SaeedLaptop> <49870432.5050301@laposte.net> Message-ID: <47EAE16A9F2547488578BCF6C0884BBD@SaeedLaptop> Thanks Rod, Its really helpful contribution. @Nextone: I don't want to say much about it, but simply I am not happy with it, have you heard someone satisfied with NX who also owns it? Kind Regards Saeed Ahmed Tariq -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of rod Sent: Monday, February 02, 2009 3:33 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Strange Performance when using as SBC Hi Saeed, Here is a first draft of what I did to install FS on my server. Configuration are not present, they'll be in a next release :p http://wiki.freeswitch.org/wiki/SBC_Setup My aim is to setup FS as a SBC, I hope this page could be a great startup point for others. I will update regularly based on what I did. Saeed, why are you replacing your Nextone, it's said to be one of the best commercial SBC on the market. regards. Saeed Ahmed wrote: > Thanks rod for a quick answer, > > FS is installed on Ubuntu Server. > > I am planning to replace Nextone SBC with FS, Later I'll also use openZAP to > communicate with TDM but this all depends how much calls it can take, or > maybe we can also do something in clustering environment ( I am not sure > about it). But thanks again and any further help will be highly appreciated. > > > Kind Regards > Saeed Ahmed Tariq > > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of rod > Sent: Monday, February 02, 2009 1:53 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Strange Performance when using as SBC > > Hi Saeed, > > I just created an account to share my setup on the wiki. I will detail > all the steps for a clean install of a debian64 lenny with FS used as a > SBC (next step is to try the new LCR module :) )and what I'm doing do > stress the server. > > I wrote nothing at this time so please be patient, I'm waiting for my > new hardware so that I will detail as much as possible what I'll do. > > For beginning I suggest you reading the start page on the wiki, > especially these pages: > -http://wiki.freeswitch.org/wiki/Getting_Started_Guide > -http://wiki.freeswitch.org/wiki/Dialplan_XML > > maybe you could tell more about the linux distribution you're using so > that I can give you some pointers for sipp... > > regards. > rod. > > > Saeed Ahmed wrote: > >> Hi Rod, >> >> Could you please share how you configured Sipp & FS to create a test >> environment? Especially the dial plan, sofia settings etc..., actually I >> > am > >> a newbie. I want to test it on a single FS machine. >> >> Kind Regards >> Saeed >> -----Original Message----- >> From: freeswitch-users-bounces at lists.freeswitch.org >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of rod >> Sent: Monday, February 02, 2009 11:00 AM >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] Strange Performance when using as SBC >> >> Hi Ken, Jay, >> >> thanks for pointing to proxy media, I will test. >> >> Ken, you are right, I was brain damaged (a stupid mistake) when setting >> INFO cause this kind of level could be very verbose. I'm switching to >> CRIT or ERR. >> >> Thanks guys, >> rod. >> >> thanks for >> >> Ken Rice wrote: >> >> >>> If you don't have to transcode, using proxy media mode will still save >>> > you > >>> some CPU time. This is 1/2 way between bypass media and the default media >>> interactive mode. The other draw back to this mode is if you are using FS >>> >>> >> to >> >> >>> clean up RTP and DTMF you loose those functions but they are not needed >>> > in > >>> most use cases. >>> >>> As far as the log level goes, I found that once I had things stable >>> >>> >> setting >> >> >>> the loglevel to helped a good deal... Info is probably a bit too high of >>> > a > >>> loglevel I would probably go for CRIT or ERR (2 or 1 respectively) if you >>> insist on leaving logging turned on... On a busy system these can and >>> > will > >>> generate a good deal of activity (and disk IO if using mod_logfile) >>> >>> Ken >>> >>> >>> >>> >>> >>>> From: rod >>>> Reply-To: >>>> Date: Mon, 02 Feb 2009 11:36:35 +0400 >>>> To: >>>> Subject: Re: [Freeswitch-users] Strange Performance when using as SBC >>>> >>>> Hi Ken, >>>> >>>> 1) I'd like to use FS to hide topology, so bypass media is not possible >>>> 2) done >>>> 3) done >>>> 4) not used >>>> 5) i'm using this ins switch.xml -> >>> value="info"/>, if you think an other log level is more suitable. >>>> >>>> Regarding logging, I can see in console and in the freeswitch.log that >>>> there is still a lot of NOTICE logging, see below: >>>> 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:960 >>>> switch_core_session_thread() Session 8721 >>>> (sofia/internal/sipp at 10.10.10.1:5060) Ended >>>> 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:962 >>>> switch_core_session_thread() Close Channel >>>> sofia/internal/sipp at 10.10.10.1:5060 [CS_HANGUP] >>>> 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:960 >>>> switch_core_session_thread() Session 8722 >>>> (sofia/external/9998 at 10.10.20.100) Ended >>>> 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:962 >>>> switch_core_session_thread() Close Channel >>>> sofia/external/9998 at 10.10.20.100 [CS_HANGUP] >>>> 2009-02-02 08:33:56 [NOTICE] sofia.c:3164 sofia_handle_sip_i_state() >>>> Channel [sofia/external/9998 at 10.10.20.100] has been answered >>>> 2009-02-02 08:33:56 [WARNING] mod_sofia.c:740 sofia_read_frame() >>>> Changing codec ptime to 30. I bet you have a linksys/sipura =D >>>> >>>> Do you have any idea where I can switch off this kind of logging. I >>>> thought it should be in /dialplan/internal.xml, but I see that in >>>> internal.xml -> >>>> >>>> thanks a lot for your suggestion. >>>> >>>> regards, >>>> rod >>>> >>>> Ken Rice wrote: >>>> >>>> >>>> >>>>> Dont forget there are several things you can do to increase >>>>> >>>>> >> performance... >> >> >>>>> 1) where possible use bypass media or media proxy modes >>>>> 2) mount freeswitch/db as a ram drive (if you are using voicemail with >>>>> the internal FS DBs you'll need a way to make this persistant across >>>>> reboots) >>>>> 3) see the wiki for setting reasonable ulimits >>>>> 4) (this is my oppinion others may vary) dont use mod_cdr_csv >>>>> 5) turn off (or reduce logging) in switch.conf.xml >>>>> >>>>> all of these thing can greatly improve performance. >>>>> >>>>> On Mon, Feb 2, 2009 at 1:04 AM, rod >>>> > wrote: >>>>> >>>>> Thanks Anthony, >>>>> >>>>> the setup is like this: >>>>> >>>>> sipp server ---- FS 1 ---- FS2 >>>>> >>>>> FS1 is the AMD CPU that has only one extension in dialplan that >>>>> bridges >>>>> 9999 to FS2. 9999 is the first extension in FS2 dialplan that >>>>> plays moh, >>>>> FS2 has no CPU pbm. >>>>> >>>>> FS1 is maxing out at 60 bridged calls without your option -hp. >>>>> >>>>> Using -hp, I'm now able to bridge 200 concurrent calls (a great >>>>> improvement) and the system is still reactive. CPU load is high >>>>> but not >>>>> 100% and as the system responds well, I think that doesn't matter. >>>>> >>>>> >> The >> >> >>>>> 2GB of memory are completely consumed (top command shows 700MB for >>>>> >>>>> >> FS >> >> >>>>> process). >>>>> >>>>> I understand that FS1 server is not the best hardware platform, >>>>> and I'm >>>>> waiting for new 4 cores server for testing. >>>>> I will update those numbers when testing with the new hardware. >>>>> >>>>> regards, >>>>> rod. >>>>> >>>>> Anthony Minessale wrote: >>>>> >>>>> >>>>> >>>>>> Which of the 2 machines has the load issue? You said it was one box >>>>>> calling the other. >>>>>> >>>>>> You have 2 major things against you, single CPU and AMD, but you >>>>>> should at least be able to get in the vicinity of 800-1000 calls >>>>>> >>>>>> >>>>>> >>>>> on a >>>>> >>>>> >>>>> >>>>>> box like that. >>>>>> >>>>>> Are you calling the default 9999? It's not really an appropriate >>>>>> extension for load testing. >>>>>> On the terminating box you should set up a manual extension that is >>>>>> the first one in the dial plan >>>>>> to play a wav file from preferably a ram disk or /tmp >>>>>> >>>>>> If you do plan on using this in production accept nothing less >>>>>> >>>>>> >>>>>> >>>>> than a >>>>> >>>>> >>>>> >>>>>> multi-core intel machine with at least 4 cores, the more cores the >>>>>> better because that parallel processing is where FS gets it's >>>>>> >>>>>> >>>>>> >>>>> atvantage. >>>>> >>>>> >>>>> >>>>>> On Fri, Jan 30, 2009 at 5:56 AM, rod >>>>> >>>>>> >>>>>> >>>>> >>>>> >>>>> >>>>> >>>>>> >> wrote: >>>>>> >>>>>> Dear list, >>>>>> >>>>>> I've been playing with freeswitch for some time (2 months) >>>>>> >>>>>> >>>>>> >>>>> and the >>>>> >>>>> >>>>> >>>>>> fact >>>>>> is that I'm very pleased with the functionnalities of this >>>>>> >>>>>> >>>>>> >>>>> software. >>>>> >>>>> >>>>> >>>>>> I'd like to use FS as a SBC handling media and I'm doing some >>>>>> tests with >>>>>> sipp to load the machine but I'm unable to bridge more than >>>>>> >>>>>> >>>>>> >>>>> 60 calls >>>>> >>>>> >>>>> >>>>>> without seeing the CPU being loaded at 100%. I'm sure >>>>>> >>>>>> >>>>>> >>>>> something is >>>>> >>>>> >>>>> >>>>>> going >>>>>> wrong with my setup but I'm unable to see what. >>>>>> >>>>>> The test machine has the following specs: >>>>>> Athlon XP 3500+ with 2GB of memory (I know this is not a >>>>>> >>>>>> >>>>>> >>>>> high end >>>>> >>>>> >>>>> >>>>>> machine :p) >>>>>> >>>>>> Freeswitch:/opt/freeswitch/log# cat /proc/cpuinfo >>>>>> processor : 0 >>>>>> vendor_id : AuthenticAMD >>>>>> cpu family : 15 >>>>>> model : 95 >>>>>> model name : AMD Athlon(tm) 64 Processor 3500+ >>>>>> stepping : 2 >>>>>> cpu MHz : 2199.973 >>>>>> cache size : 512 KB >>>>>> fpu : yes >>>>>> fpu_exception : yes >>>>>> cpuid level : 1 >>>>>> wp : yes >>>>>> flags : fpu vme de pse tsc msr pae mce cx8 apic >>>>>> >>>>>> >>>>>> >>>>> sep mtrr pge >>>>> >>>>> >>>>> >>>>>> mca cmov pat pse36 clflush mmx fxsr sse sse2 syscall nx mmxext >>>>>> fxsr_opt >>>>>> rdtscp lm 3dnowext 3dnow up rep_good pni cx16 lahf_lm svm >>>>>> >>>>>> >>>>>> >>>>> extapic >>>>> >>>>> >>>>> >>>>>> cr8_legacy >>>>>> bogomips : 4402.97 >>>>>> TLB size : 1024 4K pages >>>>>> clflush size : 64 >>>>>> cache_alignment : 64 >>>>>> address sizes : 40 bits physical, 48 bits virtual >>>>>> power management: ts fid vid ttp tm stc >>>>>> >>>>>> I installed FS on a fresh debian 64: >>>>>> Linux Freeswitch 2.6.26-1-amd64 #1 SMP Sat Jan 10 17:57:00 >>>>>> >>>>>> >>>>>> >>>>> UTC 2009 >>>>> >>>>> >>>>> >>>>>> x86_64 GNU/Linux >>>>>> >>>>>> I set the ulimit parameters like those on the website: >>>>>> freeswitch at internal> ... >>>>>> Freeswitch:/opt/free-svn/bin# ulimit -a >>>>>> core file size (blocks, -c) unlimited >>>>>> data seg size (kbytes, -d) unlimited >>>>>> scheduling priority (-e) 0 >>>>>> file size (blocks, -f) unlimited >>>>>> pending signals (-i) unlimited >>>>>> max locked memory (kbytes, -l) unlimited >>>>>> max memory size (kbytes, -m) unlimited >>>>>> open files (-n) 999999 >>>>>> pipe size (512 bytes, -p) 8 >>>>>> POSIX message queues (bytes, -q) unlimited >>>>>> real-time priority (-r) 0 >>>>>> stack size (kbytes, -s) 244 >>>>>> cpu time (seconds, -t) unlimited >>>>>> max user processes (-u) unlimited >>>>>> virtual memory (kbytes, -v) unlimited >>>>>> file locks (-x) unlimited >>>>>> >>>>>> >>>>>> My network setup is the following: >>>>>> >>>>>> SIPP machine (10.10.10.1/24)----------------vlan >>>>>> >>>>>> >>>>>> >>>>> >>>>> >>>>> >>>>> >>>>>> 55 >>>>>> ----------(10.10.10.254/24 >>>>>> >>>>>> >>>>>> >>>>> ) FS >>>>> >>>>> >>>>> >>>>>> (10.10.20.254/24)-------------- >>>>>> >>>>>> >>>>>> >>>>> >>>>> >>>>> >>>>> >>>>>> vlan56 >>>>>> -------------------(10.10.20.100/24 >>>>>> >>>>>> >>>>>> >>>>> ) >>>>> >>>>> >>>>> >>>>>> OTHER STOCK FS >>>>>> >>>>>> >>>>>> I launched sipp with: >>>>>> sipp -sn uac_pcap -s 9999 -r 10 -l 80 -d 60000 -mi >>>>>> >>>>>> >>>>>> >>>>> 10.10.10.1 -i >>>>> >>>>> >>>>> >>>>>> 10.10.10.1 -mp 25000 10.10.10.254:5060 >>>>>> >>>>>> >>>>>> >>>>> >>>>> >>>>> >>>>> >>>>>> The dialplan on FS is very simple: >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>> data="sofia/external/9999 at 10.10.20.100 >>>>>> >>>>>> >>>>>> >>>>> >>>> >"/> >>>>> >>>>> >>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> FreeSWITCH Version 1.0.trunk (11560M) Started. >>>>>> Crash Protection [Disabled] >>>>>> Max Sessions[1000] >>>>>> Session Rate[100] >>>>>> SQL [Enabled] >>>>>> >>>>>> >>>>>> The test is very simple: sipp dial 9999 that matches in my >>>>>> >>>>>> >>>>>> >>>>> FS dialplan >>>>> >>>>> >>>>> >>>>>> and this is bridged to an other FS machine playing music on >>>>>> >>>>>> >>>>>> >>>>> hold. >>>>> >>>>> >>>>> >>>>>> When I launch "top" I see after 30 to 40 s that FS consumes all >>>>>> the CPU >>>>>> ressources (with a mean of 50-60 % before), with 80 calls. >>>>>> When I set 70 calls, I have to wait 70-80 s before seeing >>>>>> >>>>>> >>>>>> >>>>> the same >>>>> >>>>> >>>>> >>>>>> issue. >>>>>> >>>>>> Presence is set to false on the 2 profile. >>>>>> >>>>>> I have the same issue with FS 1.0.2 that' s why I tried FS >>>>>> >>>>>> >>>>>> >>>>> 11560. >>>>> >>>>> >>>>> >>>>>> When I use the FS machine as a router to test the packet per >>>>>> >>>>>> >>>>>> >>>>> second >>>>> >>>>> >>>>> >>>>>> performance, I'm reaching 100Mbps with 8000pps in each >>>>>> >>>>>> >>>>>> >>>>> direction (from >>>>> >>>>> >>>>> >>>>>> vlan 55 to vlan56) with less than 12% CPU. So that I don't think >>>>>> there's >>>>>> an issue with the network. >>>>>> >>>>>> Here is an "mpstat -P ALL 1" to show you what's happening >>>>>> >>>>>> >>>>>> >>>>> suddenly >>>>> >>>>> >>>>> >>>>>> with >>>>>> 70 bridge calls: >>>>>> 12:31:26 CPU %user %nice %sys %iowait %irq %soft >>>>>> %steal %idle intr/s >>>>>> 12:31:27 all 3,00 0,00 3,00 0,00 1,00 4,00 >>>>>> 0,00 89,00 6241,00 >>>>>> 12:31:27 0 3,00 0,00 3,00 0,00 1,00 4,00 >>>>>> 0,00 89,00 6241,00 >>>>>> >>>>>> 12:31:27 CPU %user %nice %sys %iowait %irq %soft >>>>>> %steal %idle intr/s >>>>>> 12:31:28 all 14,14 0,00 56,57 0,00 2,02 5,05 >>>>>> 0,00 22,22 6035,35 >>>>>> 12:31:28 0 14,14 0,00 56,57 0,00 2,02 5,05 >>>>>> 0,00 22,22 6035,35 >>>>>> >>>>>> 12:31:28 CPU %user %nice %sys %iowait %irq %soft >>>>>> %steal %idle intr/s >>>>>> 12:31:29 all 24,75 0,00 67,33 0,00 0,99 6,93 >>>>>> 0,00 0,00 5483,17 >>>>>> 12:31:29 0 24,75 0,00 67,33 0,00 0,99 6,93 >>>>>> 0,00 0,00 5483,17 >>>>>> >>>>>> >>>>>> The CPU is going from 89% idle to 0% in less than 2 seconds. >>>>>> >>>>>> I know that I don't have to expect too much from this kind of >>>>>> hardware, >>>>>> but it seems strange that the CPU power vanished so suddenly. >>>>>> >>>>>> Thanks a lot for the guys that have read this long mail :p >>>>>> >>>>>> kind regards, >>>>>> rod >>>>>> >>>>>> >>>>>> _______________________________________________ >>>>>> Freeswitch-users mailing list >>>>>> Freeswitch-users at lists.freeswitch.org >>>>>> >>>>>> >>>>>> >>>>> >>>>> >>>>> >>>>> >>>>>> >>>>> >>>>>> >>>>>> >>>>> > >>>>> >>>>> >>>>> >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> >>>>>> >>>>>> >>>>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> >>>>> >>>>> >>>>> >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> -- >>>>>> Anthony Minessale II >>>>>> >>>>>> FreeSWITCH http://www.freeswitch.org/ >>>>>> ClueCon http://www.cluecon.com/ >>>>>> >>>>>> AIM: anthm >>>>>> MSN:anthony_minessale at hotmail.com >>>>>> >>>>>> >>>>>> >>>>> >>>>> >>>>> >>>>> >>>>>> >>>>> >>>>>> >>>>>> >>>>> > >>>>> >>>>> >>>>> >>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>>>>> >>>>>> >>>>>> >>>>> >>>>> >>>>> >>>>> >>>>>> >>>>> >>>>>> >>>>>> >>>>> > >>>>> >>>>> >>>>> >>>>>> IRC: irc.freenode.net >>>>>> >>>>>> >>>>>> >>>>> #freeswitch >>>>> >>>>> >>>>> >>>>>> FreeSWITCH Developer Conference >>>>>> sip:888 at conference.freeswitch.org >>>>>> >>>>>> >>>>>> >>>>> >>>>> >>>>> >>>>> >>>>>> >>>>> >>>>>> >>>>>> >>>>> > >>>>> >>>>> >>>>> >>>>>> iax:guest at conference.freeswitch.org/888 >>>>>> >>>>>> >>>>>> >>>>> >>>>> >>>>> >>>>> >>>>>> >>>>>> googletalk:conf+888 at conference.freeswitch.org >>>>>> >>>>>> >>>>>> >>>>> >>>>> >>>>> >>>>> >>>>>> >>>>> >>>>>> >>>>>> >>>>> > >>>>> >>>>> >>>>> >>>>>> pstn:213-799-1400 >>>>>> >>>>>> >>>>>> >>>>>> >> ------------------------------------------------------------------------ >> >> >>>>> >>>>> >>>>> >>>>>> _______________________________________________ >>>>>> Freeswitch-users mailing list >>>>>> Freeswitch-users at lists.freeswitch.org >>>>>> >>>>>> >>>>>> >>>>> >>>>> >>>>> >>>>> >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> >>>>>> >>>>>> >>>>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> >>>>> >>>>> >>>>> >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>>> >>>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> >>>>> >>>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> >>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> >>>>> > ------------------------------------------------------------------------ > >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> >>>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> >>>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >>> >>> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From pmhshz at gmail.com Mon Feb 2 07:21:32 2009 From: pmhshz at gmail.com (shehzad p) Date: Mon, 2 Feb 2009 07:21:32 -0800 (PST) Subject: [Freeswitch-users] Call Variable not available when call hangup In-Reply-To: <191c3a030902020606r1a42ef44n7a73bd1e5157392e@mail.gmail.com> References: <21788550.post@talk.nabble.com> <21789152.post@talk.nabble.com> <191c3a030902020606r1a42ef44n7a73bd1e5157392e@mail.gmail.com> Message-ID: <21791503.post@talk.nabble.com> one question is that when javascript is being called from dial plan, I get the session object already available, It is for A leg of channel, So when javascript is called after Bridge how can I get the session object for B leg also? Anthony Minessale-2 wrote: > > the leg you are running the script on is not hungup, the other leg of the > call is. > > If it was hungup you would not be executing the script. > > Asterisk and the h ext and the whole dead-agi thing are all poor design > showing it's teeth. > We do not support anything like it. > > > You can however try this: (see the link below) > > http://www.nabble.com/Re:-Conference-javascript-and-hanuphooks-giving-me-headaches-p21614840.html > > > > On Mon, Feb 2, 2009 at 6:53 AM, shehzad p wrote: > >> >> Is there any settings that when call hangup control can be transferred to >> another context and these CDR values can be accessible there? (just like >> in >> Asterisk, h extension) >> >> shehzad p wrote: >> > >> > Hi all, >> > >> > I need to process some CDR variables in Dialplan, like call duration, >> > Answered time etc. >> > but when I place info application after bridge, it is not listing them >> > properly as below: >> > =========================================== >> > Caller-Channel-Created-Time: [1233573341672157] >> > Caller-Channel-Answered-Time: [1233573342712939] >> > Caller-Channel-Hangup-Time: [0] >> > ========================================== >> > Here Hangup time is 0, So how can I find actual values? >> > >> > --I know that we can use xml_cdr or cdr_csv, but my current need is to >> get >> > those values from dialplan itself so that can be passed to some >> script... >> > >> > >> > thanks, >> > msp >> > >> >> -- >> View this message in context: >> http://www.nabble.com/Call-Variable-not-available-when-call-hangup-tp21788550p21789152.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/Call-Variable-not-available-when-call-hangup-tp21788550p21791503.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From clif at eugeneweb.com Sun Feb 1 13:24:20 2009 From: clif at eugeneweb.com (clif at eugeneweb.com) Date: Sun, 1 Feb 2009 13:24:20 -0800 (PST) Subject: [Freeswitch-users] How do I set my FS internal ip address to a "static" value. Message-ID: Hi Gang, I've been struggleing with this also. Actually I can get it to bind to my address, the problem is it randomly drops my calls. :-( I have a FS running on a box with a static IP and I can start a call between two extensions and it will go for hours. Then I add anther interface say eth0:0 with a new static IP and reconfigure my phones and FS to use that, and the calls drop after about 15-20 mins. Though it's pretty random. Here is my setup. I have Debian Linux 2.6.23.1 kernel, and freeswitch-1.0.1. Here is my /etc/network/interfaces: # /etc/network/interfaces -- configuration file for ifup(8), ifdown(8) # The loopback interface auto lo iface lo inet loopback # The first network card - this entry was created during the Debian installation auto eth0 eth0:0 iface eth0 inet dhcp iface eth0:0 inet static address 192.168.0.249 netmask 255.255.255.0 gateway 192.168.0.254 The only change I made to the FS config is in Vars.xml. I added this line close to the top: Here is the console log of the call being dropped: freeswitch at archive> sofia status API CALL [sofia(status)] output: Name Type Data State ================================================================================================= external profile sip:mod_sofia at 67.171.158.226:5080 RUNNING (0) internal profile sip:mod_sofia at 192.168.0.249:5060 RUNNING (2) nat profile sip:mod_sofia at 67.171.158.226:5070 RUNNING (0) default alias internal ALIASED outbound alias external ALIASED 192.168.0.249 alias internal ALIASED ================================================================================================= 3 profiles 3 aliases freeswitch at archive> 2009-02-01 13:23:19 [NOTICE] sofia_glue.c:2634 sofia_glue_restart_all_profiles() Reload XML [Success] 2009-02-01 13:23:19 [INFO] mod_enum.c:817 event_handler() ENUM Reloaded 2009-02-01 13:23:19 [NOTICE] mod_sofia.c:568 sofia_read_frame() Hangup sofia/internal/1003 at 192.168.0.53:5060;user=phone;transport=udp;fs_nat=yes [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 2009-02-01 13:23:19 [NOTICE] switch_ivr_bridge.c:820 switch_ivr_multi_threaded_bridge() Hangup sofia/internal/1001 at 192.168.0.249 [CS_EXECUTE] [NORMAL_CLEARING] 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:807 switch_core_session_thread() Session 6 (sofia/internal/1003 at 192.168.0.53:5060;user=phone;transport=udp;fs_nat=yes) Ended 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:809 switch_core_session_thread() Close Channel sofia/internal/1003 at 192.168.0.53:5060;user=phone;transport=udp;fs_nat=yes [CS_HANGUP] 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:807 switch_core_session_thread() Session 5 (sofia/internal/1001 at 192.168.0.249) Ended 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:809 switch_core_session_thread() Close Channel sofia/internal/1001 at 192.168.0.249 [CS_HANGUP] 2009-02-01 13:23:19 [NOTICE] sofia.c:645 sofia_profile_thread_run() waiting for worker thread 2009-02-01 13:23:19 [NOTICE] sofia.c:645 sofia_profile_thread_run() waiting for worker thread 2009-02-01 13:23:19 [NOTICE] sofia.c:1865 config_sofia() Adding Alias [192.168.0.249] for profile [internal] 2009-02-01 13:23:19 [NOTICE] sofia.c:1865 config_sofia() Adding Alias [default] for profile [internal] 2009-02-01 13:23:19 [NOTICE] sofia.c:1875 config_sofia() Started Profile internal [sofia_reg_internal] 2009-02-01 13:23:20 [NOTICE] sofia.c:1865 config_sofia() Adding Alias [outbound] for profile [external] 2009-02-01 13:23:20 [NOTICE] sofia.c:1875 config_sofia() Started Profile external [sofia_reg_external] 2009-02-01 13:23:20 [NOTICE] sofia.c:645 sofia_profile_thread_run() waiting for worker thread 2009-02-01 13:23:20 [NOTICE] sofia.c:1875 config_sofia() Started Profile nat [sofia_reg_nat] sofia status API CALL [sofia(status)] output: Name Type Data State ================================================================================================= external profile sip:mod_sofia at 67.171.158.226:5080 RUNNING (0) internal profile sip:mod_sofia at 192.168.0.249:5060 RUNNING (0) outbound alias external ALIASED 192.168.0.249 alias internal ALIASED nat profile sip:mod_sofia at 67.171.158.226:5070 RUNNING (0) default alias internal ALIASED ================================================================================================= 3 profiles 3 aliases There is an older thread that says one should set but in this (later) thread is says only Jingleling usese that variable. ie. see: http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg00695.html http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg07345.html So what do you think causes this? What is the correct way? ;-) Thanks, Clif From anthony.minessale at gmail.com Mon Feb 2 07:41:05 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 2 Feb 2009 09:41:05 -0600 Subject: [Freeswitch-users] Call Variable not available when call hangup In-Reply-To: <21791503.post@talk.nabble.com> References: <21788550.post@talk.nabble.com> <21789152.post@talk.nabble.com> <191c3a030902020606r1a42ef44n7a73bd1e5157392e@mail.gmail.com> <21791503.post@talk.nabble.com> Message-ID: <191c3a030902020741k779e2488o38ca578a3b40e9ad@mail.gmail.com> you can't that's why i said it was a horrible approach. That's also why i posted you the instructions on the only elegant solution to your problem. On Mon, Feb 2, 2009 at 9:21 AM, shehzad p wrote: > > > one question is that when javascript is being called from dial plan, I get > the session object already available, It is for A leg of channel, > So when javascript is called after Bridge how can I get the session object > for B leg also? > > > Anthony Minessale-2 wrote: > > > > the leg you are running the script on is not hungup, the other leg of the > > call is. > > > > If it was hungup you would not be executing the script. > > > > Asterisk and the h ext and the whole dead-agi thing are all poor design > > showing it's teeth. > > We do not support anything like it. > > > > > > You can however try this: (see the link below) > > > > > http://www.nabble.com/Re:-Conference-javascript-and-hanuphooks-giving-me-headaches-p21614840.html > > > > > > > > On Mon, Feb 2, 2009 at 6:53 AM, shehzad p wrote: > > > >> > >> Is there any settings that when call hangup control can be transferred > to > >> another context and these CDR values can be accessible there? (just like > >> in > >> Asterisk, h extension) > >> > >> shehzad p wrote: > >> > > >> > Hi all, > >> > > >> > I need to process some CDR variables in Dialplan, like call duration, > >> > Answered time etc. > >> > but when I place info application after bridge, it is not listing them > >> > properly as below: > >> > =========================================== > >> > Caller-Channel-Created-Time: [1233573341672157] > >> > Caller-Channel-Answered-Time: [1233573342712939] > >> > Caller-Channel-Hangup-Time: [0] > >> > ========================================== > >> > Here Hangup time is 0, So how can I find actual values? > >> > > >> > --I know that we can use xml_cdr or cdr_csv, but my current need is to > >> get > >> > those values from dialplan itself so that can be passed to some > >> script... > >> > > >> > > >> > thanks, > >> > msp > >> > > >> > >> -- > >> View this message in context: > >> > http://www.nabble.com/Call-Variable-not-available-when-call-hangup-tp21788550p21789152.html > >> Sent from the Freeswitch-users mailing list archive at Nabble.com. > >> > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com < > MSN%3Aanthony_minessale at hotmail.com > > > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > > > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org < > sip%3A888 at conference.freeswitch.org > > > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > > > > pstn:213-799-1400 > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > View this message in context: > http://www.nabble.com/Call-Variable-not-available-when-call-hangup-tp21788550p21791503.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090202/2d430e44/attachment-0002.html From brian at freeswitch.org Mon Feb 2 07:41:39 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 2 Feb 2009 09:41:39 -0600 Subject: [Freeswitch-users] How do I set my FS internal ip address to a "static" value. In-Reply-To: References: Message-ID: <05AC6B23-7F34-4C89-8BD0-1744BEF20B4C@freeswitch.org> you need to add this setting to sofia.conf.xml You'll also need to edit the sofia profiles and input the exact IP you wish it to bind to. The params are sip-ip and rtp-ip. /b On Feb 1, 2009, at 3:24 PM, clif at eugeneweb.com wrote: > Hi Gang, > > I've been struggleing with this also. Actually I can get it to bind > to my > address, the problem is it randomly drops my calls. :-( > > I have a FS running on a box with a static IP and I can start a call > between > two extensions and it will go for hours. Then I add anther interface > say eth0:0 > with a new static IP and reconfigure my phones and FS to use that, > and the > calls drop after about 15-20 mins. Though it's pretty random. > > Here is my setup. I have Debian Linux 2.6.23.1 kernel, and > freeswitch-1.0.1. > Here is my /etc/network/interfaces: > > # /etc/network/interfaces -- configuration file for ifup(8), ifdown(8) > > # The loopback interface > auto lo > iface lo inet loopback > > # The first network card - this entry was created during the Debian > installation > auto eth0 eth0:0 > iface eth0 inet dhcp > iface eth0:0 inet static > address 192.168.0.249 > netmask 255.255.255.0 > gateway 192.168.0.254 > > The only change I made to the FS config is in Vars.xml. I added this > line close > to the top: > > > > Here is the console log of the call being dropped: > > freeswitch at archive> sofia status > API CALL [sofia(status)] output: > Name Type Data > State > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > ====================================================================== > external profile sip:mod_sofia at 67.171.158.226:5080 > RUNNING (0) > internal profile sip:mod_sofia at 192.168.0.249:5060 > RUNNING (2) > nat profile sip:mod_sofia at 67.171.158.226:5070 > RUNNING (0) > default alias internal > ALIASED > outbound alias external > ALIASED > 192.168.0.249 alias internal > ALIASED > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > ====================================================================== > 3 profiles 3 aliases > > freeswitch at archive> 2009-02-01 13:23:19 [NOTICE] sofia_glue.c:2634 > sofia_glue_restart_all_profiles() Reload XML [Success] > 2009-02-01 13:23:19 [INFO] mod_enum.c:817 event_handler() ENUM > Reloaded > 2009-02-01 13:23:19 [NOTICE] mod_sofia.c:568 sofia_read_frame() Hangup > sofia/internal/ > 1003 at 192.168.0.53:5060;user=phone;transport=udp;fs_nat=yes > [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] > 2009-02-01 13:23:19 [NOTICE] switch_ivr_bridge.c:820 > switch_ivr_multi_threaded_bridge() Hangup sofia/internal/1001 at 192.168.0.249 > [CS_EXECUTE] [NORMAL_CLEARING] > 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:807 > switch_core_session_thread() Session 6 > (sofia/internal/ > 1003 at 192.168.0.53:5060;user=phone;transport=udp;fs_nat=yes) > Ended > 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:809 > switch_core_session_thread() Close Channel > sofia/internal/ > 1003 at 192.168.0.53:5060;user=phone;transport=udp;fs_nat=yes > [CS_HANGUP] > 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:807 > switch_core_session_thread() Session 5 (sofia/internal/1001 at 192.168.0.249 > ) > Ended > 2009-02-01 13:23:19 [NOTICE] switch_core_session.c:809 > switch_core_session_thread() Close Channel sofia/internal/1001 at 192.168.0.249 > [CS_HANGUP] > 2009-02-01 13:23:19 [NOTICE] sofia.c:645 sofia_profile_thread_run() > waiting for > worker thread > 2009-02-01 13:23:19 [NOTICE] sofia.c:645 sofia_profile_thread_run() > waiting for > worker thread > 2009-02-01 13:23:19 [NOTICE] sofia.c:1865 config_sofia() Adding Alias > [192.168.0.249] for profile [internal] > 2009-02-01 13:23:19 [NOTICE] sofia.c:1865 config_sofia() Adding > Alias [default] > for profile [internal] > 2009-02-01 13:23:19 [NOTICE] sofia.c:1875 config_sofia() Started > Profile > internal [sofia_reg_internal] > 2009-02-01 13:23:20 [NOTICE] sofia.c:1865 config_sofia() Adding Alias > [outbound] for profile [external] > 2009-02-01 13:23:20 [NOTICE] sofia.c:1875 config_sofia() Started > Profile > external [sofia_reg_external] > 2009-02-01 13:23:20 [NOTICE] sofia.c:645 sofia_profile_thread_run() > waiting for > worker thread > 2009-02-01 13:23:20 [NOTICE] sofia.c:1875 config_sofia() Started > Profile nat > [sofia_reg_nat] > sofia status > API CALL [sofia(status)] output: > Name Type Data > State > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > ====================================================================== > external profile sip:mod_sofia at 67.171.158.226:5080 > RUNNING (0) > internal profile sip:mod_sofia at 192.168.0.249:5060 > RUNNING (0) > outbound alias external > ALIASED > 192.168.0.249 alias internal > ALIASED > nat profile sip:mod_sofia at 67.171.158.226:5070 > RUNNING (0) > default alias internal > ALIASED > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > ====================================================================== > 3 profiles 3 aliases > > There is an older thread that says one should set > > but in this (later) thread is says only Jingleling usese that > variable. > ie. see: > http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg00695.html > http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg07345.html > > So what do you think causes this? What is the correct way? ;-) > > > Thanks, > Clif > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From helmut.kuper at ewetel.de Mon Feb 2 10:00:48 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Mon, 02 Feb 2009 19:00:48 +0100 Subject: [Freeswitch-users] mod_voicemail: Limiting the number how often a menu is repeated Message-ID: <498734D0.5060004@ewetel.de> Hello, today I searched for a way to limit the number of menu repeatings in mod_voicemail to let's say 3 times and when it reached the limit voicemail should abort. But I couldn't find a hint. Any ideas? regards helmut From helmut.kuper at ewetel.de Mon Feb 2 10:04:11 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Mon, 02 Feb 2009 19:04:11 +0100 Subject: [Freeswitch-users] Q931 decoding Update In-Reply-To: <49830576.6080907@ewetel.de> References: <4972046E.8020102@ewetel.de> <49802B7F.70100@ewetel.de> <49818AE9.7030605@ewetel.de> <200901291202.41144.stkn@freeswitch.org> <191c3a030901290549x29c87976k68b3f9b455f00d0f@mail.gmail.com> <4981B8E6.9070708@ewetel.de> <5A2ED995-CF05-4AEF-9103-43BC84465514@freeswitch.org> <4981D6F8.4060409@ewetel.de> <191c3a030901290832k27a18b30u7bc531157fd89b19@mail.gmail.com> <49830576.6080907@ewetel.de> Message-ID: <4987359B.2020402@ewetel.de> Hello, today I uploaded a little patch for openzap concerning missed linking of the pcap library. So loading ozmod_isdn failed with some kind of "unknown symbol pcap_flush_dump" error message. This keeps mod_openzap from loading at FS startup. regards helmut From peder at networkoblivion.com Mon Feb 2 12:01:36 2009 From: peder at networkoblivion.com (peder at networkoblivion.com) Date: Mon, 02 Feb 2009 14:01:36 -0600 Subject: [Freeswitch-users] Strange Performance when using as SBC In-Reply-To: <191c3a030902020557v70d3777fqb83e591385a64a59@mail.gmail.com> References: <4986C42C.7030700@laposte.net> <1C31435E19AB4E57981CF2AAC2CA74AA@SaeedLaptop> <4986ECA9.3040707@laposte.net> <9232C06D5362494791AF713E1DF61343@SaeedLaptop> <191c3a030902020557v70d3777fqb83e591385a64a59@mail.gmail.com> Message-ID: <49875120.1040502@networkoblivion.com> What is wrong with Intrepid? Anthony Minessale wrote: > if you want to use ubuntu, be sure to use hardy and not intrepid. > > On Mon, Feb 2, 2009 at 7:03 AM, Saeed Ahmed > wrote: > > Thanks rod for a quick answer, > > FS is installed on Ubuntu Server. > > I am planning to replace Nextone SBC with FS, Later I'll also use > openZAP to > communicate with TDM but this all depends how much calls it can take, or > maybe we can also do something in clustering environment ( I am not sure > about it). But thanks again and any further help will be highly > appreciated. > > > Kind Regards > Saeed Ahmed Tariq > > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > > [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of rod > Sent: Monday, February 02, 2009 1:53 PM > To: freeswitch-users at lists.freeswitch.org > > Subject: Re: [Freeswitch-users] Strange Performance when using as SBC > > Hi Saeed, > > I just created an account to share my setup on the wiki. I will detail > all the steps for a clean install of a debian64 lenny with FS used as a > SBC (next step is to try the new LCR module :) )and what I'm doing do > stress the server. > > I wrote nothing at this time so please be patient, I'm waiting for my > new hardware so that I will detail as much as possible what I'll do. > > For beginning I suggest you reading the start page on the wiki, > especially these pages: > -http://wiki.freeswitch.org/wiki/Getting_Started_Guide > -http://wiki.freeswitch.org/wiki/Dialplan_XML > > maybe you could tell more about the linux distribution you're using so > that I can give you some pointers for sipp... > > regards. > rod. > > > Saeed Ahmed wrote: > > Hi Rod, > > > > Could you please share how you configured Sipp & FS to create a test > > environment? Especially the dial plan, sofia settings etc..., > actually I > am > > a newbie. I want to test it on a single FS machine. > > > > Kind Regards > > Saeed > > -----Original Message----- > > From: freeswitch-users-bounces at lists.freeswitch.org > > > [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of rod > > Sent: Monday, February 02, 2009 11:00 AM > > To: freeswitch-users at lists.freeswitch.org > > > Subject: Re: [Freeswitch-users] Strange Performance when using as SBC > > > > Hi Ken, Jay, > > > > thanks for pointing to proxy media, I will test. > > > > Ken, you are right, I was brain damaged (a stupid mistake) when > setting > > INFO cause this kind of level could be very verbose. I'm switching to > > CRIT or ERR. > > > > Thanks guys, > > rod. > > > > thanks for > > > > Ken Rice wrote: > > > >> If you don't have to transcode, using proxy media mode will > still save > you > >> some CPU time. This is 1/2 way between bypass media and the > default media > >> interactive mode. The other draw back to this mode is if you are > using FS > >> > > to > > > >> clean up RTP and DTMF you loose those functions but they are not > needed > in > >> most use cases. > >> > >> As far as the log level goes, I found that once I had things stable > >> > > setting > > > >> the loglevel to helped a good deal... Info is probably a bit too > high of > a > >> loglevel I would probably go for CRIT or ERR (2 or 1 > respectively) if you > >> insist on leaving logging turned on... On a busy system these > can and > will > >> generate a good deal of activity (and disk IO if using mod_logfile) > >> > >> Ken > >> > >> > >> > >> > >>> From: rod > > >>> Reply-To: > > >>> Date: Mon, 02 Feb 2009 11:36:35 +0400 > >>> To: > > >>> Subject: Re: [Freeswitch-users] Strange Performance when using > as SBC > >>> > >>> Hi Ken, > >>> > >>> 1) I'd like to use FS to hide topology, so bypass media is not > possible > >>> 2) done > >>> 3) done > >>> 4) not used > >>> 5) i'm using this ins switch.xml -> >>> value="info"/>, if you think an other log level is more suitable. > >>> > >>> Regarding logging, I can see in console and in the > freeswitch.log that > >>> there is still a lot of NOTICE logging, see below: > >>> 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:960 > >>> switch_core_session_thread() Session 8721 > >>> (sofia/internal/sipp at 10.10.10.1:5060 > ) Ended > >>> 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:962 > >>> switch_core_session_thread() Close Channel > >>> sofia/internal/sipp at 10.10.10.1:5060 > [CS_HANGUP] > >>> 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:960 > >>> switch_core_session_thread() Session 8722 > >>> (sofia/external/9998 at 10.10.20.100 ) Ended > >>> 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:962 > >>> switch_core_session_thread() Close Channel > >>> sofia/external/9998 at 10.10.20.100 > [CS_HANGUP] > >>> 2009-02-02 08:33:56 [NOTICE] sofia.c:3164 > sofia_handle_sip_i_state() > >>> Channel [sofia/external/9998 at 10.10.20.100 > ] has been answered > >>> 2009-02-02 08:33:56 [WARNING] mod_sofia.c:740 sofia_read_frame() > >>> Changing codec ptime to 30. I bet you have a linksys/sipura =D > >>> > >>> Do you have any idea where I can switch off this kind of logging. I > >>> thought it should be in /dialplan/internal.xml, but I see that in > >>> internal.xml -> > >>> > >>> thanks a lot for your suggestion. > >>> > >>> regards, > >>> rod > >>> > >>> Ken Rice wrote: > >>> > >>> > >>>> Dont forget there are several things you can do to increase > >>>> > > performance... > > > >>>> 1) where possible use bypass media or media proxy modes > >>>> 2) mount freeswitch/db as a ram drive (if you are using > voicemail with > >>>> the internal FS DBs you'll need a way to make this persistant > across > >>>> reboots) > >>>> 3) see the wiki for setting reasonable ulimits > >>>> 4) (this is my oppinion others may vary) dont use mod_cdr_csv > >>>> 5) turn off (or reduce logging) in switch.conf.xml > >>>> > >>>> all of these thing can greatly improve performance. > >>>> > >>>> On Mon, Feb 2, 2009 at 1:04 AM, rod > >>>> >> wrote: > >>>> > >>>> Thanks Anthony, > >>>> > >>>> the setup is like this: > >>>> > >>>> sipp server ---- FS 1 ---- FS2 > >>>> > >>>> FS1 is the AMD CPU that has only one extension in dialplan > that > >>>> bridges > >>>> 9999 to FS2. 9999 is the first extension in FS2 dialplan that > >>>> plays moh, > >>>> FS2 has no CPU pbm. > >>>> > >>>> FS1 is maxing out at 60 bridged calls without your option -hp. > >>>> > >>>> Using -hp, I'm now able to bridge 200 concurrent calls (a > great > >>>> improvement) and the system is still reactive. CPU load is > high > >>>> but not > >>>> 100% and as the system responds well, I think that doesn't > matter. > >>>> > > The > > > >>>> 2GB of memory are completely consumed (top command shows > 700MB for > >>>> > > FS > > > >>>> process). > >>>> > >>>> I understand that FS1 server is not the best hardware > platform, > >>>> and I'm > >>>> waiting for new 4 cores server for testing. > >>>> I will update those numbers when testing with the new > hardware. > >>>> > >>>> regards, > >>>> rod. > >>>> > >>>> Anthony Minessale wrote: > >>>> > >>>> > >>>>> Which of the 2 machines has the load issue? You said it was > one box > >>>>> calling the other. > >>>>> > >>>>> You have 2 major things against you, single CPU and AMD, but you > >>>>> should at least be able to get in the vicinity of 800-1000 calls > >>>>> > >>>>> > >>>> on a > >>>> > >>>> > >>>>> box like that. > >>>>> > >>>>> Are you calling the default 9999? It's not really an appropriate > >>>>> extension for load testing. > >>>>> On the terminating box you should set up a manual extension > that is > >>>>> the first one in the dial plan > >>>>> to play a wav file from preferably a ram disk or /tmp > >>>>> > >>>>> If you do plan on using this in production accept nothing less > >>>>> > >>>>> > >>>> than a > >>>> > >>>> > >>>>> multi-core intel machine with at least 4 cores, the more > cores the > >>>>> better because that parallel processing is where FS gets it's > >>>>> > >>>>> > >>>> atvantage. > >>>> > >>>> > >>>>> On Fri, Jan 30, 2009 at 5:56 AM, rod > >>>>> > >>>>> > >>>> > > >>>> > >>>> > >>>>> > >>> wrote: > >>>>> > >>>>> Dear list, > >>>>> > >>>>> I've been playing with freeswitch for some time (2 months) > >>>>> > >>>>> > >>>> and the > >>>> > >>>> > >>>>> fact > >>>>> is that I'm very pleased with the functionnalities of this > >>>>> > >>>>> > >>>> software. > >>>> > >>>> > >>>>> I'd like to use FS as a SBC handling media and I'm doing some > >>>>> tests with > >>>>> sipp to load the machine but I'm unable to bridge more than > >>>>> > >>>>> > >>>> 60 calls > >>>> > >>>> > >>>>> without seeing the CPU being loaded at 100%. I'm sure > >>>>> > >>>>> > >>>> something is > >>>> > >>>> > >>>>> going > >>>>> wrong with my setup but I'm unable to see what. > >>>>> > >>>>> The test machine has the following specs: > >>>>> Athlon XP 3500+ with 2GB of memory (I know this is not a > >>>>> > >>>>> > >>>> high end > >>>> > >>>> > >>>>> machine :p) > >>>>> > >>>>> Freeswitch:/opt/freeswitch/log# cat /proc/cpuinfo > >>>>> processor : 0 > >>>>> vendor_id : AuthenticAMD > >>>>> cpu family : 15 > >>>>> model : 95 > >>>>> model name : AMD Athlon(tm) 64 Processor 3500+ > >>>>> stepping : 2 > >>>>> cpu MHz : 2199.973 > >>>>> cache size : 512 KB > >>>>> fpu : yes > >>>>> fpu_exception : yes > >>>>> cpuid level : 1 > >>>>> wp : yes > >>>>> flags : fpu vme de pse tsc msr pae mce cx8 apic > >>>>> > >>>>> > >>>> sep mtrr pge > >>>> > >>>> > >>>>> mca cmov pat pse36 clflush mmx fxsr sse sse2 syscall nx > mmxext > >>>>> fxsr_opt > >>>>> rdtscp lm 3dnowext 3dnow up rep_good pni cx16 lahf_lm svm > >>>>> > >>>>> > >>>> extapic > >>>> > >>>> > >>>>> cr8_legacy > >>>>> bogomips : 4402.97 > >>>>> TLB size : 1024 4K pages > >>>>> clflush size : 64 > >>>>> cache_alignment : 64 > >>>>> address sizes : 40 bits physical, 48 bits virtual > >>>>> power management: ts fid vid ttp tm stc > >>>>> > >>>>> I installed FS on a fresh debian 64: > >>>>> Linux Freeswitch 2.6.26-1-amd64 #1 SMP Sat Jan 10 17:57:00 > >>>>> > >>>>> > >>>> UTC 2009 > >>>> > >>>> > >>>>> x86_64 GNU/Linux > >>>>> > >>>>> I set the ulimit parameters like those on the website: > >>>>> freeswitch at internal> ... > >>>>> Freeswitch:/opt/free-svn/bin# ulimit -a > >>>>> core file size (blocks, -c) unlimited > >>>>> data seg size (kbytes, -d) unlimited > >>>>> scheduling priority (-e) 0 > >>>>> file size (blocks, -f) unlimited > >>>>> pending signals (-i) unlimited > >>>>> max locked memory (kbytes, -l) unlimited > >>>>> max memory size (kbytes, -m) unlimited > >>>>> open files (-n) 999999 > >>>>> pipe size (512 bytes, -p) 8 > >>>>> POSIX message queues (bytes, -q) unlimited > >>>>> real-time priority (-r) 0 > >>>>> stack size (kbytes, -s) 244 > >>>>> cpu time (seconds, -t) unlimited > >>>>> max user processes (-u) unlimited > >>>>> virtual memory (kbytes, -v) unlimited > >>>>> file locks (-x) unlimited > >>>>> > >>>>> > >>>>> My network setup is the following: > >>>>> > >>>>> SIPP machine (10.10.10.1/24)----------------vlan > > >>>>> > >>>>> > >>>> > >>>> > >>>> > >>>>> 55 > >>>>> ----------(10.10.10.254/24 > > >>>>> > >>>>> > >>>> ) FS > >>>> > >>>> > >>>>> (10.10.20.254/24)-------------- > > >>>>> > >>>>> > >>>> > >>>> > >>>> > >>>>> vlan56 > >>>>> -------------------(10.10.20.100/24 > > >>>>> > >>>>> > >>>> ) > >>>> > >>>> > >>>>> OTHER STOCK FS > >>>>> > >>>>> > >>>>> I launched sipp with: > >>>>> sipp -sn uac_pcap -s 9999 -r 10 -l 80 -d 60000 -mi > >>>>> > >>>>> > >>>> 10.10.10.1 -i > >>>> > >>>> > >>>>> 10.10.10.1 -mp 25000 10.10.10.254:5060 > > >>>>> > >>>>> > >>>> > >>>> > >>>> > >>>>> The dialplan on FS is very simple: > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> expression="^9999$"> > >>>>> > >>>>> >>>>> data="sofia/external/9999 at 10.10.20.100 > > >>>>> > >>>>> > >>>> > > > >>>> >>"/> > >>>> > >>>> > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> FreeSWITCH Version 1.0.trunk (11560M) Started. > >>>>> Crash Protection [Disabled] > >>>>> Max Sessions[1000] > >>>>> Session Rate[100] > >>>>> SQL [Enabled] > >>>>> > >>>>> > >>>>> The test is very simple: sipp dial 9999 that matches in my > >>>>> > >>>>> > >>>> FS dialplan > >>>> > >>>> > >>>>> and this is bridged to an other FS machine playing music on > >>>>> > >>>>> > >>>> hold. > >>>> > >>>> > >>>>> When I launch "top" I see after 30 to 40 s that FS > consumes all > >>>>> the CPU > >>>>> ressources (with a mean of 50-60 % before), with 80 calls. > >>>>> When I set 70 calls, I have to wait 70-80 s before seeing > >>>>> > >>>>> > >>>> the same > >>>> > >>>> > >>>>> issue. > >>>>> > >>>>> Presence is set to false on the 2 profile. > >>>>> > >>>>> I have the same issue with FS 1.0.2 that' s why I tried FS > >>>>> > >>>>> > >>>> 11560. > >>>> > >>>> > >>>>> When I use the FS machine as a router to test the packet per > >>>>> > >>>>> > >>>> second > >>>> > >>>> > >>>>> performance, I'm reaching 100Mbps with 8000pps in each > >>>>> > >>>>> > >>>> direction (from > >>>> > >>>> > >>>>> vlan 55 to vlan56) with less than 12% CPU. So that I > don't think > >>>>> there's > >>>>> an issue with the network. > >>>>> > >>>>> Here is an "mpstat -P ALL 1" to show you what's happening > >>>>> > >>>>> > >>>> suddenly > >>>> > >>>> > >>>>> with > >>>>> 70 bridge calls: > >>>>> 12:31:26 CPU %user %nice %sys %iowait %irq > %soft > >>>>> %steal %idle intr/s > >>>>> 12:31:27 all 3,00 0,00 3,00 0,00 1,00 > 4,00 > >>>>> 0,00 89,00 6241,00 > >>>>> 12:31:27 0 3,00 0,00 3,00 0,00 1,00 > 4,00 > >>>>> 0,00 89,00 6241,00 > >>>>> > >>>>> 12:31:27 CPU %user %nice %sys %iowait %irq > %soft > >>>>> %steal %idle intr/s > >>>>> 12:31:28 all 14,14 0,00 56,57 0,00 2,02 > 5,05 > >>>>> 0,00 22,22 6035,35 > >>>>> 12:31:28 0 14,14 0,00 56,57 0,00 2,02 > 5,05 > >>>>> 0,00 22,22 6035,35 > >>>>> > >>>>> 12:31:28 CPU %user %nice %sys %iowait %irq > %soft > >>>>> %steal %idle intr/s > >>>>> 12:31:29 all 24,75 0,00 67,33 0,00 0,99 > 6,93 > >>>>> 0,00 0,00 5483,17 > >>>>> 12:31:29 0 24,75 0,00 67,33 0,00 0,99 > 6,93 > >>>>> 0,00 0,00 5483,17 > >>>>> > >>>>> > >>>>> The CPU is going from 89% idle to 0% in less than 2 seconds. > >>>>> > >>>>> I know that I don't have to expect too much from this kind of > >>>>> hardware, > >>>>> but it seems strange that the CPU power vanished so suddenly. > >>>>> > >>>>> Thanks a lot for the guys that have read this long mail :p > >>>>> > >>>>> kind regards, > >>>>> rod > >>>>> > >>>>> > >>>>> _______________________________________________ > >>>>> Freeswitch-users mailing list > >>>>> Freeswitch-users at lists.freeswitch.org > > >>>>> > >>>>> > >>>> > > >>>> > >>>> > >>>>> > >>>>> > >>>>> > >>>> >> > >>>> > >>>> > >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>>>> > >>>>> > >>>>> > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > >>>> > >>>> > >>>>> http://www.freeswitch.org > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> -- > >>>>> Anthony Minessale II > >>>>> > >>>>> FreeSWITCH http://www.freeswitch.org/ > >>>>> ClueCon http://www.cluecon.com/ > >>>>> > >>>>> AIM: anthm > >>>>> MSN:anthony_minessale at hotmail.com > > >>>>> > >>>>> > >>>> > > >>>> > >>>> > >>>>> > >>>>> > >>>>> > >>>> >> > >>>> > >>>> > >>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > >>>>> > >>>>> > >>>> > > >>>> > >>>> > >>>>> > >>>>> > >>>>> > >>>> >> > >>>> > >>>> > >>>>> IRC: irc.freenode.net > > >>>>> > >>>>> > >>>> #freeswitch > >>>> > >>>> > >>>>> FreeSWITCH Developer Conference > >>>>> sip:888 at conference.freeswitch.org > > >>>>> > >>>>> > >>>> > > >>>> > >>>> > >>>>> > >>>>> > >>>>> > >>>> >> > >>>> > >>>> > >>>>> iax:guest at conference.freeswitch.org/888 > > >>>>> > >>>>> > >>>> > >>>> > >>>> > >>>>> > >>>>> googletalk:conf+888 at conference.freeswitch.org > > >>>>> > >>>>> > >>>> > > >>>> > >>>> > >>>>> > >>>>> > >>>>> > >>>> > >> > >>>> > >>>> > >>>>> pstn:213-799-1400 > >>>>> > >>>>> > >>>>> > > > ------------------------------------------------------------------------ > > > >>>> > >>>> > >>>>> _______________________________________________ > >>>>> Freeswitch-users mailing list > >>>>> Freeswitch-users at lists.freeswitch.org > > >>>>> > >>>>> > >>>> > > >>>> > >>>> > >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>>>> > >>>>> > >>>>> > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > >>>> > >>>> > >>>>> http://www.freeswitch.org > >>>>> > >>>>> > >>>>> > >>>> _______________________________________________ > >>>> Freeswitch-users mailing list > >>>> Freeswitch-users at lists.freeswitch.org > > >>>> > > >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>>> > >>>> > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > >>>> http://www.freeswitch.org > >>>> > >>>> > >>>> > ------------------------------------------------------------------------ > >>>> > >>>> _______________________________________________ > >>>> Freeswitch-users mailing list > >>>> Freeswitch-users at lists.freeswitch.org > > >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >>>> http://www.freeswitch.org > >>>> > >>>> > >>>> > >>> _______________________________________________ > >>> Freeswitch-users mailing list > >>> Freeswitch-users at lists.freeswitch.org > > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> http://www.freeswitch.org > >>> > >>> > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > >> > >> > >> > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Mon Feb 2 12:10:22 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 2 Feb 2009 14:10:22 -0600 Subject: [Freeswitch-users] Strange Performance when using as SBC In-Reply-To: <49875120.1040502@networkoblivion.com> References: <4986C42C.7030700@laposte.net> <1C31435E19AB4E57981CF2AAC2CA74AA@SaeedLaptop> <4986ECA9.3040707@laposte.net> <9232C06D5362494791AF713E1DF61343@SaeedLaptop> <191c3a030902020557v70d3777fqb83e591385a64a59@mail.gmail.com> <49875120.1040502@networkoblivion.com> Message-ID: Its too bleeding edge and you had better know what you're doing if you use it. It comes with libtool 2.2 which you can't use to build FreeSWITCH. /b On Feb 2, 2009, at 2:01 PM, peder at networkoblivion.com wrote: > What is wrong with Intrepid? From raul at etellicom.com Mon Feb 2 12:34:45 2009 From: raul at etellicom.com (Raul Fragoso) Date: Mon, 02 Feb 2009 18:34:45 -0200 Subject: [Freeswitch-users] Strange Performance when using as SBC In-Reply-To: References: <4986C42C.7030700@laposte.net> <1C31435E19AB4E57981CF2AAC2CA74AA@SaeedLaptop> <4986ECA9.3040707@laposte.net> <9232C06D5362494791AF713E1DF61343@SaeedLaptop> <191c3a030902020557v70d3777fqb83e591385a64a59@mail.gmail.com> <49875120.1040502@networkoblivion.com> Message-ID: <1233606885.28870.5.camel@stargate> Yes, that's exactly the issue: libtool. Provided that you use libtool 1.5.22-4 or some other 1.5.x version, FS seems to work fine with Intrepid. -- Raul On Mon, 2009-02-02 at 14:10 -0600, Brian West wrote: > Its too bleeding edge and you had better know what you're doing if you > use it. It comes with libtool 2.2 which you can't use to build > FreeSWITCH. > > /b > > On Feb 2, 2009, at 2:01 PM, peder at networkoblivion.com wrote: > > > What is wrong with Intrepid? > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From raul at etellicom.com Mon Feb 2 12:48:14 2009 From: raul at etellicom.com (Raul Fragoso) Date: Mon, 02 Feb 2009 18:48:14 -0200 Subject: [Freeswitch-users] Strange Performance when using as SBC In-Reply-To: References: <4986C42C.7030700@laposte.net> <1C31435E19AB4E57981CF2AAC2CA74AA@SaeedLaptop> <4986ECA9.3040707@laposte.net> <9232C06D5362494791AF713E1DF61343@SaeedLaptop> <191c3a030902020557v70d3777fqb83e591385a64a59@mail.gmail.com> <49875120.1040502@networkoblivion.com> Message-ID: <1233607694.28870.9.camel@stargate> In addition to libtool, you may have issues with the latest packages of gcc and some other tools that FS will need. In any case, it's better to not use Intrepid at all ;-) Use Hardy as suggested and you will be happy. On Mon, 2009-02-02 at 14:10 -0600, Brian West wrote: > Its too bleeding edge and you had better know what you're doing if you > use it. It comes with libtool 2.2 which you can't use to build > FreeSWITCH. > > /b > > On Feb 2, 2009, at 2:01 PM, peder at networkoblivion.com wrote: > > > What is wrong with Intrepid? > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From hads at nice.net.nz Mon Feb 2 13:02:34 2009 From: hads at nice.net.nz (Hadley Rich) Date: Tue, 3 Feb 2009 10:02:34 +1300 Subject: [Freeswitch-users] Strange Performance when using as SBC In-Reply-To: <1233607694.28870.9.camel@stargate> References: <1233607694.28870.9.camel@stargate> Message-ID: <200902031002.34785.hads@nice.net.nz> On Tue, 03 Feb 2009 09:48:14 Raul Fragoso wrote: > In addition to libtool, you may have issues with the latest packages of > gcc and some other tools that FS will need. In any case, it's better to > not use Intrepid at all ;-) Use Hardy as suggested and you will be > happy. You shouldn't have any issues. I've used Intrepid on a VM to compile and test FreeSWITCH quite a bit and haven't run across any issues at all after downgrading libtool. That said I would also recommend Hardy LTS for production servers. hads -- http://nicegear.co.nz VoIP, DVB and other Linux compatible hardware. From brian at freeswitch.org Mon Feb 2 13:05:40 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 2 Feb 2009 15:05:40 -0600 Subject: [Freeswitch-users] Strange Performance when using as SBC In-Reply-To: <200902031002.34785.hads@nice.net.nz> References: <1233607694.28870.9.camel@stargate> <200902031002.34785.hads@nice.net.nz> Message-ID: <46AB18C7-C28B-4DA7-BBAE-1BBAF8ECB430@freeswitch.org> gcc 4.3.2 caused a segfault to appear in openzap due to over optimization... so yes it can bite you. :) /b On Feb 2, 2009, at 3:02 PM, Hadley Rich wrote: > You shouldn't have any issues. I've used Intrepid on a VM to compile > and test > FreeSWITCH quite a bit and haven't run across any issues at all after > downgrading libtool. > > That said I would also recommend Hardy LTS for production servers. > > hads From mike at jerris.com Mon Feb 2 13:48:55 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 2 Feb 2009 16:48:55 -0500 Subject: [Freeswitch-users] Q931 decoding Update In-Reply-To: <4987359B.2020402@ewetel.de> References: <4972046E.8020102@ewetel.de> <49802B7F.70100@ewetel.de> <49818AE9.7030605@ewetel.de> <200901291202.41144.stkn@freeswitch.org> <191c3a030901290549x29c87976k68b3f9b455f00d0f@mail.gmail.com> <4981B8E6.9070708@ewetel.de> <5A2ED995-CF05-4AEF-9103-43BC84465514@freeswitch.org> <4981D6F8.4060409@ewetel.de> <191c3a030901290832k27a18b30u7bc531157fd89b19@mail.gmail.com> <49830576.6080907@ewetel.de> <4987359B.2020402@ewetel.de> Message-ID: <3D009334-DCFD-4AE5-A446-6C8165F2D77D@jerris.com> We need to add more than this including detection in openzap configure.in if libpcap is available (headers and lib) and if not, disabling the functionality. MIke On Feb 2, 2009, at 1:04 PM, Helmut Kuper wrote: > Hello, > > today I uploaded a little patch for openzap concerning missed > linking of > the pcap library. So loading ozmod_isdn failed with some kind of > "unknown symbol pcap_flush_dump" error message. This keeps mod_openzap > from loading at FS startup. > > regards > helmut > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From jaybinks at gmail.com Mon Feb 2 14:20:20 2009 From: jaybinks at gmail.com (jay binks) Date: Tue, 3 Feb 2009 08:20:20 +1000 Subject: [Freeswitch-users] Strange Performance when using as SBC In-Reply-To: <49870432.5050301@laposte.net> References: <4986C42C.7030700@laposte.net> <1C31435E19AB4E57981CF2AAC2CA74AA@SaeedLaptop> <4986ECA9.3040707@laposte.net> <9232C06D5362494791AF713E1DF61343@SaeedLaptop> <49870432.5050301@laposte.net> Message-ID: Rod, that wiki article is Awesome ! real good to see guides with start to finish steps. cant wait to see the next installment of your guide :) Jay On Tue, Feb 3, 2009 at 12:33 AM, rod wrote: > Hi Saeed, > > Here is a first draft of what I did to install FS on my server. > Configuration are not present, they'll be in a next release :p > http://wiki.freeswitch.org/wiki/SBC_Setup > > My aim is to setup FS as a SBC, I hope this page could be a great > startup point for others. I will update regularly based on what I did. > > Saeed, why are you replacing your Nextone, it's said to be one of the > best commercial SBC on the market. > > regards. > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090203/b84c521a/attachment-0002.html From nik.middleton at noblesolutions.co.uk Mon Feb 2 15:35:51 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Mon, 2 Feb 2009 23:35:51 -0000 Subject: [Freeswitch-users] Generating calls from external source Message-ID: Hi Guys, As a long time Asterisk user, I'm looking into freeswitch as an alternative mainly due to (list multiple reasons here) Can anyone give me a pointer as to how I would achieve the following? I need to replicate an emergency broadcast system currently running under Asterisk. At the moment, I run through a Mysql database and using the manager API, issues an Originate command to dial a number. When the call is answered, a message is played, and the recipient has the option of hitting a digit to confirm receipt. I then call an AGI script to update the database. Is this fairly easy to do in Freeswitch? Not looking for code, just some pointers as to what's available to do the above / Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090202/82eebf5b/attachment-0002.html From msc at freeswitch.org Mon Feb 2 17:16:43 2009 From: msc at freeswitch.org (Michael S Collins) Date: Mon, 2 Feb 2009 17:16:43 -0800 Subject: [Freeswitch-users] Generating calls from external source In-Reply-To: References: Message-ID: Nik, Welcome to FreeSWITCH! The short answer is "yes, FS can do that." The first thing that you should do is unlearn "the Asterisk way" of thinking. Usually there is an elegant way of doing things in FS that wasn't possible in Ast. I would recommend that you start by looking at the event socket, which is somewhat analogous to the AMI only cooler. :) I have personally done something similar to this using the event socket and a Perl script. The key is to learn the syntax of the originate command. (definitely hit the wiki and IRC channel) Are you using TDM cards for this? Just curious. -MC (IRC nick: mercutioviz) Sent from my iPhone On Feb 2, 2009, at 3:35 PM, "Nik Middleton" wrote: > Hi Guys, > > > > As a long time Asterisk user, I?m looking into freeswitch as an alte > rnative mainly due to (list multiple reasons here) > > > > Can anyone give me a pointer as to how I would achieve the following? > > > > I need to replicate an emergency broadcast system currently running > under Asterisk. > > > > At the moment, I run through a Mysql database and using the manager > API, issues an Originate command to dial a number. > > > > When the call is answered, a message is played, and the recipient > has the option of hitting a digit to confirm receipt. I then call > an AGI script to update the database. > > > > Is this fairly easy to do in Freeswitch? > > > > Not looking for code, just some pointers as to what?s available to d > o the above / > > > > Regards, > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090202/14b335d2/attachment-0002.html From ajlong at worldlink.net Mon Feb 2 19:05:17 2009 From: ajlong at worldlink.net (Adam Long) Date: Mon, 2 Feb 2009 22:05:17 -0500 Subject: [Freeswitch-users] FreeSwitch setup as a "Dumb" SBC Message-ID: <019501c985ac$4f00ee60$ed02cb20$@net> Hi Guys, I've been working at setting up a couple of FreeSwitch nodes as a topology hiding SBCs that handles both ingress traffic from my providers/peers and pass traffic up to an openser router that then routes call across the cluster of SBCs through which they reach the destination. I have OpenSIPS/SER setup doing DB route lookups and ENUM with LCR/Serial forking etc. My question is what would be the best way to send a call out to a destination choosen by the OpenSER router? For example: SIP Provider -- > SBC --- > OpenSER ---- ( route lookup returns 123.123.123.4 as dest ) -- > SBC --- > 123.123.123.4 I was thinking something along the lines of adding a "X-Route-To: +1NXXNXXXXXX at 123.123.123.4" with openser and then something like this in the SBC. Is this a wise approach, is there anything I could do to do this better? I'd like to keep the logic in the SBCs as simple as possible. I am pretty familiar with SIP but my knowledge fades when it gets into the nitty gritty of routing. ie the Contact: and Via: headers and all that good stuff. I should also state I have two profiles defined one for the internal/private "core" network and one for the outside "external" network. Any thoughts on this at all would be greatly appreciated. Am I missing something in the SIP spec that would allow for this is a standardized way? Regards, -Adam -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090202/1c9fdb9e/attachment-0002.html From jason at voicesession.com Mon Feb 2 18:21:25 2009 From: jason at voicesession.com (lee jason) Date: Tue, 3 Feb 2009 10:21:25 +0800 Subject: [Freeswitch-users] Application language to support C or C++? Message-ID: <2cbf225c0902021821u2bc8b622ka5d9f94f05a968ae@mail.gmail.com> Hi All, I saw the applications using FreeSwitch library can be written in JavaScript, Perl, Python and Lua but I need to use Linux C or C++ for applications, Is FreeSwitch can supported it? Where can I get the sample codes? My Linux platform is base on Fedora. Thanks a lot. Jason -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090203/cd4c9ffb/attachment-0002.html From brian at freeswitch.org Mon Feb 2 19:16:47 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 2 Feb 2009 21:16:47 -0600 Subject: [Freeswitch-users] Application language to support C or C++? In-Reply-To: <2cbf225c0902021821u2bc8b622ka5d9f94f05a968ae@mail.gmail.com> References: <2cbf225c0902021821u2bc8b622ka5d9f94f05a968ae@mail.gmail.com> Message-ID: <87A290D3-137D-4E54-A308-4A3D568104E2@freeswitch.org> Do you want to write modules in c++ for FreeSWITCH? If so then yes you can write modules in c++... If thats not what you mean please clarify. /b On Feb 2, 2009, at 8:21 PM, lee jason wrote: > Hi All, > > I saw the applications using FreeSwitch library can be written > in JavaScript, Perl, Python and Lua but I need to use Linux C or C+ > + for applications, Is FreeSwitch can supported it? Where can I get > the sample codes? My Linux platform is base on Fedora. > > > Thanks a lot. > > Jason From krice at freeswitch.org Mon Feb 2 20:24:44 2009 From: krice at freeswitch.org (Ken Rice) Date: Mon, 02 Feb 2009 22:24:44 -0600 Subject: [Freeswitch-users] Application language to support C or C++? In-Reply-To: <2cbf225c0902021821u2bc8b622ka5d9f94f05a968ae@mail.gmail.com> Message-ID: FreeSwitch is written in C mainly and some things like mod_opal are written in C++, you can create your own modules in C... Grab the source and look around its pretty straight forward From: lee jason Reply-To: Date: Tue, 3 Feb 2009 10:21:25 +0800 To: Subject: [Freeswitch-users] Application language to support C or C++? Hi All, I saw the applications using FreeSwitch library can be written in JavaScript, Perl, Python and Lua but I need to use Linux C or C++ for applications, Is FreeSwitch can supported it? Where can I get the sample codes? My Linux platform is base on Fedora. Thanks a lot. Jason _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090202/af71a030/attachment-0002.html From krice at freeswitch.org Mon Feb 2 20:28:17 2009 From: krice at freeswitch.org (Ken Rice) Date: Mon, 02 Feb 2009 22:28:17 -0600 Subject: [Freeswitch-users] FreeSwitch setup as a "Dumb" SBC In-Reply-To: <019501c985ac$4f00ee60$ed02cb20$@net> Message-ID: Yes you can do that, but there is nothing that says you cant have FreeSWITCH just do those lookups and ENUM (FS Supports ENUM out of the box) and do the exact same thing so it would work like Provider -> ingress SBC -> egress SBC/Registration Server -> customer Loosing a whole hop in the process From: Adam Long Reply-To: Date: Mon, 2 Feb 2009 22:05:17 -0500 To: Subject: [Freeswitch-users] FreeSwitch setup as a "Dumb" SBC Hi Guys, I?ve been working at setting up a couple of FreeSwitch nodes as a topology hiding SBCs that handles both ingress traffic from my providers/peers and pass traffic up to an openser router that then routes call across the cluster of SBCs through which they reach the destination. I have OpenSIPS/SER setup doing DB route lookups and ENUM with LCR/Serial forking etc. My question is what would be the best way to send a call out to a destination choosen by the OpenSER router? For example: SIP Provider -- > SBC --- > OpenSER ---- ( route lookup returns 123.123.123.4 as dest ) -- > SBC --- > 123.123.123.4 I was thinking something along the lines of adding a ?X-Route-To: +1NXXNXXXXXX at 123.123.123.4? with openser and then something like this in the SBC? Is this a wise approach, is there anything I could do to do this better? I?d like to keep the logic in the SBCs as simple as possible. I am pretty familiar with SIP but my knowledge fades when it gets into the nitty gritty of routing? ie the Contact: and Via: headers and all that good stuff. I should also state I have two profiles defined one for the internal/private ?core? network and one for the outside ?external? network. Any thoughts on this at all would be greatly appreciated. Am I missing something in the SIP spec that would allow for this is a standardized way? Regards, -Adam _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090202/be0fb682/attachment-0002.html From kawarod at laposte.net Mon Feb 2 22:33:11 2009 From: kawarod at laposte.net (rod) Date: Tue, 03 Feb 2009 10:33:11 +0400 Subject: [Freeswitch-users] FreeSwitch setup as a "Dumb" SBC In-Reply-To: <019501c985ac$4f00ee60$ed02cb20$@net> References: <019501c985ac$4f00ee60$ed02cb20$@net> Message-ID: <4987E527.1040909@laposte.net> Hi Adam, I'm in the process of using FS as a SBC. For the route lookup, I do it using OpenSER carrierroute, without having to flow through SBC---Openser---SBC. I'm using carrierroute at this time cause I need more than 200 000 routing entries and carrierroute has been tested with twice this number. Here is the setup: - install openser and carrierroute and make openser listening on 127.0.0.1:5062 (for example) on your SBC - populate carrierroute table What I do to use carrierroute module from FS is to use a specific X-header (X-LOOKUP). In the dialplan, in the default context, I have something like this: The process is simple: the export "sip_h_X-ROUTE=LOOKUP" had a sip header X-ROUTE=LOOKUP then I bridge the call to 127.0.0.1:5062 (openser process) In openser I have a route block that checks the presence of header LOOKUP and openser sends a "604: unable to route call" if the prefix is not found, or a "302: with the IP of the gateway found" In FS, you can get the IP using the variable "${sip_redirect_contact_host_0}". Then I transfer this to the context ROUTING, where the check condition is based on the LOOKUP header that has been rewritten with this variable. I will document all this setup (installation of openser/carrierroute and config file of FS and openser) on a wiki page I start writing yesterday, so please be indulgent and patient. The next step is to test the scalability of this. I'm a very bad programmer, so that's the only way for me to contribute to FS, and as I see many people interested for an SBC setup, I think it could be great if we share our work/knowlegde. The wiki page is there: http://wiki.freeswitch.org/wiki/SBC_Setup regards, rod. Adam Long wrote: > > Hi Guys, > > I?ve been working at setting up a couple of FreeSwitch nodes as a > topology hiding SBCs that handles both ingress traffic from my > > providers/peers and pass traffic up to an openser router that then > routes call across the cluster of SBCs through which they reach the > destination. > > I have OpenSIPS/SER setup doing DB route lookups and ENUM with > LCR/Serial forking etc. > > My question is what would be the best way to send a call out to a > destination choosen by the OpenSER router? > > For example: > > SIP Provider -- > SBC --- > OpenSER ---- ( route lookup returns > 123.123.123.4 as dest ) -- > SBC --- > 123.123.123.4 > > I was thinking something along the lines of adding a ?X-Route-To: > +1NXXNXXXXXX@ 123.123.123.4? with openser > > and then something like this in the SBC? > > > > > > > > > > > > Is this a wise approach, is there anything I could do to do this better? > > I?d like to keep the logic in the SBCs as simple as possible. > > I am pretty familiar with SIP but my knowledge fades when it gets into > the nitty gritty of routing? ie the Contact: and Via: headers > > and all that good stuff. > > I should also state I have two profiles defined one for the > internal/private ?core? network and one for the outside ?external? > network. > > Any thoughts on this at all would be greatly appreciated. > > Am I missing something in the SIP spec that would allow for this is a > standardized way? > > Regards, > > -Adam > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From sias at cpdata.co.za Mon Feb 2 23:22:21 2009 From: sias at cpdata.co.za (Sias Mey) Date: Tue, 3 Feb 2009 09:22:21 +0200 Subject: [Freeswitch-users] Conference dialing and uuid In-Reply-To: <191c3a030902020601v1109b1e9he867e9f1e1d4a401@mail.gmail.com> References: <20090130103944.GA23536@cpdata.co.za> <20090130133315.GB23536@cpdata.co.za> <191c3a030901300605r78dd8943w7fb1075ef851e1d@mail.gmail.com> <20090202085157.GA3555@cpdata.co.za> <0F863750-5C77-4CD9-ADC6-CFDDF2B149B4@freeswitch.org> <20090202102459.GA4179@cpdata.co.za> <9FE1733C-A46A-4A17-8AEE-077DE8010235@freeswitch.org> <191c3a030902020601v1109b1e9he867e9f1e1d4a401@mail.gmail.com> Message-ID: <20090203072221.GD16105@cpdata.co.za> Actually loopback does work. however as I said it generates a pair of extra channels. Hmmm I was trying to generate and extra call to a JS script that generated a teletone ring in an on_ring_execute for the second call however it seems to stop execution of the call itself. Event though I use api commands to originate and then transfer it into the conference so that I have direct access to its uuid. I think changeing the moh might be a bit simpler however and elimite some CoreDB stuff I was doing to keep track of the calls ring generating call (what a sentance). On Mon, Feb 02, 2009 at 08:01:25AM -0600, Anthony Minessale wrote: > you could set the conference moh sound to be tone_stream::// with the > teletone spec for ring sound and it use ignore_early_media=true in your > originates so the first caller would hear ringback until the 2nd one > arrived. > > On Mon, Feb 2, 2009 at 4:29 AM, Brian West <[1]brian at freeswitch.org> > wrote: > > Loopback will not work in that case either. If the far end plays > ringback inband you should hear that if you use the conference dial > api call. > /b > > On Feb 2, 2009, at 4:24 AM, Sias Mey wrote: > > Aaah ok. > > > > Thanks for clearing that up. > > > > So using loopback is still the only real workable sollution for me, > > since that generates ringback from and alternative endpoint and > > plays it > > into the conference. > > > > I might play with some javascript that streams ring into the channel > > eventually but for now the string comparisons at least get me the > > right > > uuid. > > > > Thank you again, > > Sias > > _______________________________________________ > Freeswitch-users mailing list > [2]Freeswitch-users at lists.freeswitch.org > [3]http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:[4]http://lists.freeswitch.org/mailman/options/freeswitch-u > sers > [5]http://www.freeswitch.org > > -- > Anthony Minessale II > FreeSWITCH [6]http://www.freeswitch.org/ > ClueCon [7]http://www.cluecon.com/ > AIM: anthm > [8]MSN:anthony_minessale at hotmail.com > GTALK/JABBER/[9]PAYPAL:anthony.minessale at gmail.com > IRC: [10]irc.freenode.net #freeswitch > FreeSWITCH Developer Conference > [11]sip:888 at conference.freeswitch.org > [12]iax:guest at conference.freeswitch.org/888 > [13]googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > References > > 1. mailto:brian at freeswitch.org > 2. mailto:Freeswitch-users at lists.freeswitch.org > 3. http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > 4. http://lists.freeswitch.org/mailman/options/freeswitch-users > 5. http://www.freeswitch.org/ > 6. http://www.freeswitch.org/ > 7. http://www.cluecon.com/ > 8. mailto:MSN%3Aanthony_minessale at hotmail.com > 9. mailto:PAYPAL%3Aanthony.minessale at gmail.com > 10. http://irc.freenode.net/ > 11. mailto:sip%3A888 at conference.freeswitch.org > 12. http://iax:guest at conference.freeswitch.org/888 > 13. mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From sias at cpdata.co.za Tue Feb 3 00:25:30 2009 From: sias at cpdata.co.za (Sias Mey) Date: Tue, 3 Feb 2009 10:25:30 +0200 Subject: [Freeswitch-users] Conference dialing and uuid In-Reply-To: <20090203072221.GD16105@cpdata.co.za> References: <20090130103944.GA23536@cpdata.co.za> <20090130133315.GB23536@cpdata.co.za> <191c3a030901300605r78dd8943w7fb1075ef851e1d@mail.gmail.com> <20090202085157.GA3555@cpdata.co.za> <0F863750-5C77-4CD9-ADC6-CFDDF2B149B4@freeswitch.org> <20090202102459.GA4179@cpdata.co.za> <9FE1733C-A46A-4A17-8AEE-077DE8010235@freeswitch.org> <191c3a030902020601v1109b1e9he867e9f1e1d4a401@mail.gmail.com> <20090203072221.GD16105@cpdata.co.za> Message-ID: <20090203082530.GA17166@cpdata.co.za> Hmmm no MOH wont work... since I am planning on pulling more than just 2 members into the conference and I still need ringback for the later members as well. Is there a direct way for me to use conference play to play teletone directly? or should I just records some ringing if I want to use that? And lastly for my own sanity ;-) why would the following in a on_ring_execute stop execution of the call at that point? call = argv[1]; conf = argv[2]; consoleLog("info","Making ringback channel for uuid : "+ session.uuid +"\n"); var ringuuid = apiExecute("originate","loopback/ringback-conf="+ conf +"-conf &park()") //I tried with and without a exit() at the end It seems to stop media detection??(not really sure about the term) for the call that executes this script. Freeswitch doesent recognize the pickup of that call and thus it doesent get bridged into the conference. when I uuid_kill the call that gets originated everything else starts happening again. Oh Im running this in FS ver. 1.0.trunk (11226:11561M) and that loopback points to and ringback.js is use("TeleTone"); session.answer(); var tts = new TeleTone(session); tts.addTone("u", 400.0, 450.0, 0.0); tts.addTone("r", 440.0, 480.0, 0.0); var RESET = "v=2000;>=0;+=0;"; var UK_RING = RESET + "L=2;u(400,200);u(400,2200)"; var US_RING = RESET + "r(2000,4000)"; while(session.ready()) { console_log("making UK ring\n"); for (x = 0 ; x < 2 ; x++) { tts.generate(UK_RING); } } A slight bastardisation of the teletone JS example. I would expected the new channel that is created via a api originate to be completely seperate from the JS I create it in. (thats why I use api instead of creating a new session, although I should probably try that as well). I use some CoreDB stuff to keep tabs on the uuid for the originated call so that I can uuid_kill it in the on_answer_script but as mentioned... the on_answer only executes after I uuid_kill the originated channel in the cli... Thanks again guys, Specially since it seems you two are always the ones that get back to me. On Tue, Feb 03, 2009 at 09:22:21AM +0200, Sias Mey wrote: > Actually loopback does work. > however as I said it generates a pair of extra channels. > > Hmmm I was trying to generate and extra call to a JS script that > generated a teletone ring in an on_ring_execute for the second call > however it seems to stop execution of the call itself. Event though I > use api commands to originate and then transfer it into the conference > so that I have direct access to its uuid. > > I think changeing the moh might be a bit simpler however and elimite > some CoreDB stuff I was doing to keep track of the calls ring generating > call (what a sentance). > > On Mon, Feb 02, 2009 at 08:01:25AM -0600, Anthony Minessale wrote: > > you could set the conference moh sound to be tone_stream::// with the > > teletone spec for ring sound and it use ignore_early_media=true in your > > originates so the first caller would hear ringback until the 2nd one > > arrived. > > > > On Mon, Feb 2, 2009 at 4:29 AM, Brian West <[1]brian at freeswitch.org> > > wrote: > > > > Loopback will not work in that case either. If the far end plays > > ringback inband you should hear that if you use the conference dial > > api call. > > /b > > > > On Feb 2, 2009, at 4:24 AM, Sias Mey wrote: > > > Aaah ok. > > > > > > Thanks for clearing that up. > > > > > > So using loopback is still the only real workable sollution for me, > > > since that generates ringback from and alternative endpoint and > > > plays it > > > into the conference. > > > > > > I might play with some javascript that streams ring into the channel > > > eventually but for now the string comparisons at least get me the > > > right > > > uuid. > > > > > > Thank you again, > > > Sias > > > > _______________________________________________ > > Freeswitch-users mailing list > > [2]Freeswitch-users at lists.freeswitch.org > > [3]http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:[4]http://lists.freeswitch.org/mailman/options/freeswitch-u > > sers > > [5]http://www.freeswitch.org > > > > -- > > Anthony Minessale II > > FreeSWITCH [6]http://www.freeswitch.org/ > > ClueCon [7]http://www.cluecon.com/ > > AIM: anthm > > [8]MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/[9]PAYPAL:anthony.minessale at gmail.com > > IRC: [10]irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > > [11]sip:888 at conference.freeswitch.org > > [12]iax:guest at conference.freeswitch.org/888 > > [13]googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > > > > References > > > > 1. mailto:brian at freeswitch.org > > 2. mailto:Freeswitch-users at lists.freeswitch.org > > 3. http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > 4. http://lists.freeswitch.org/mailman/options/freeswitch-users > > 5. http://www.freeswitch.org/ > > 6. http://www.freeswitch.org/ > > 7. http://www.cluecon.com/ > > 8. mailto:MSN%3Aanthony_minessale at hotmail.com > > 9. mailto:PAYPAL%3Aanthony.minessale at gmail.com > > 10. http://irc.freenode.net/ > > 11. mailto:sip%3A888 at conference.freeswitch.org > > 12. http://iax:guest at conference.freeswitch.org/888 > > 13. mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From kawarod at laposte.net Tue Feb 3 01:11:24 2009 From: kawarod at laposte.net (rod) Date: Tue, 03 Feb 2009 13:11:24 +0400 Subject: [Freeswitch-users] Strange Performance when using as SBC In-Reply-To: References: <4986C42C.7030700@laposte.net> <1C31435E19AB4E57981CF2AAC2CA74AA@SaeedLaptop> <4986ECA9.3040707@laposte.net> <9232C06D5362494791AF713E1DF61343@SaeedLaptop> <49870432.5050301@laposte.net> Message-ID: <49880A3C.3010506@laposte.net> Hi all, I completed the wiki page with the comments I made in the posts: Re: [Freeswitch-users] FreeSwitch setup as a "Dumb" SBC I detailed how to setup mysql/kamailio/carrierroute to use the carrierroute module of kamailio for LCR. I wrote this page using my memory and history of the linux commands. Maybe some things are missing and I will update as soon as I get my new servers for reinstallation. I have to cleanup the way it is displayed, cause it lacks some wiki rules. If some would like to contribute, they are welcome. http://wiki.freeswitch.org/wiki/SBC_Setup regards, rod jay binks wrote: > Rod, > that wiki article is Awesome ! > > real good to see guides with start to finish steps. > cant wait to see the next installment of your guide :) > > Jay > > On Tue, Feb 3, 2009 at 12:33 AM, rod > wrote: > > Hi Saeed, > > Here is a first draft of what I did to install FS on my server. > Configuration are not present, they'll be in a next release :p > http://wiki.freeswitch.org/wiki/SBC_Setup > > My aim is to setup FS as a SBC, I hope this page could be a great > startup point for others. I will update regularly based on what I did. > > Saeed, why are you replacing your Nextone, it's said to be one of the > best commercial SBC on the market. > > regards. > > From nik.middleton at noblesolutions.co.uk Tue Feb 3 01:28:34 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Tue, 3 Feb 2009 09:28:34 -0000 Subject: [Freeswitch-users] Generating calls from external source In-Reply-To: References: Message-ID: Thanks for that, coming from a C++ background it's a refreshing change to be looking at something that seems logical and efficient. I'd briefly looked at the event socket and wondered if that was the way to go. I presume that there's some sort of event generation that can trigger and external process as well somewhere, though all I need to do is update mysql (hopefully using some sort of pooled connection) I'm not using a TDM card, I have a direct interconnect with the PSTN breakout provider with 1,500 channels available to me. I'm finding Asterisk proving to be less than stable at high call volumes and load values spike at more than 100 calls with billing/accounting in place, hence my interest in FS. The only thing that's concerning me is XML at the moment. Lots of code and very wordy. I'm sure I'll appreciate why XML given time Regards, ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael S Collins Sent: 03 February 2009 01:17 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Generating calls from external source Nik, Welcome to FreeSWITCH! The short answer is "yes, FS can do that." The first thing that you should do is unlearn "the Asterisk way" of thinking. Usually there is an elegant way of doing things in FS that wasn't possible in Ast. I would recommend that you start by looking at the event socket, which is somewhat analogous to the AMI only cooler. :) I have personally done something similar to this using the event socket and a Perl script. The key is to learn the syntax of the originate command. (definitely hit the wiki and IRC channel) Are you using TDM cards for this? Just curious. -MC (IRC nick: mercutioviz) Sent from my iPhone On Feb 2, 2009, at 3:35 PM, "Nik Middleton" wrote: Hi Guys, As a long time Asterisk user, I'm looking into freeswitch as an alternative mainly due to (list multiple reasons here) Can anyone give me a pointer as to how I would achieve the following? I need to replicate an emergency broadcast system currently running under Asterisk. At the moment, I run through a Mysql database and using the manager API, issues an Originate command to dial a number. When the call is answered, a message is played, and the recipient has the option of hitting a digit to confirm receipt. I then call an AGI script to update the database. Is this fairly easy to do in Freeswitch? Not looking for code, just some pointers as to what's available to do the above / Regards, _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090203/fe01adb3/attachment-0002.html From dave at 3c.co.uk Tue Feb 3 01:46:00 2009 From: dave at 3c.co.uk (David Knell) Date: Tue, 3 Feb 2009 09:46:00 +0000 Subject: [Freeswitch-users] Generating calls from external source In-Reply-To: References: Message-ID: <40B58CBF-9D0F-43A3-B36F-BD05916756E4@3c.co.uk> Hi Nik, Here's a snipped in Perl that launches an outbound call: if (my $sock = IO::Socket::INET->new(Proto =>'tcp', PeerAddr => '127.0.0.1', PeerPort => 8021)) { print $sock "auth XXX\n\n"; print $sock "api originate {softivr_id=$siid,src_softivr_id=$siid,softivr_outdial=true}sofia/frombt/$ntd at 1.2.3.4 $service\n\n"; $sock->close(); } - it does no error checking or anything, but (line by line) it: - opens a socket to the event socket interface - authenticates - issues an originate which dials out to the number in $ntd. The bits in {} set a bunch of variables on the channel, which are used by the software which processes the call later on. The call is linked to the extension in $service - FS looks this up in the dialplan - which handles our end. - closes the socket Cheers -- Dave > Thanks for that, coming from a C++ background it?s a refreshing > change to be looking at something that seems logical and efficient. > > I?d briefly looked at the event socket and wondered if that was the > way to go. I presume that there?s some sort of event generation > that can trigger and external process as well somewhere, though all > I need to do is update mysql (hopefully using some sort of pooled > connection) > > I?m not using a TDM card, I have a direct interconnect with the PSTN > breakout provider with 1,500 channels available to me. I?m finding > Asterisk proving to be less than stable at high call volumes and > load values spike at more than 100 calls with billing/accounting in > place, hence my interest in FS. The only thing that?s concerning me > is XML at the moment. Lots of code and very wordy. I?m sure I?ll > appreciate why XML given time > > Regards, > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Michael S Collins > Sent: 03 February 2009 01:17 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Generating calls from external source > > Nik, > > Welcome to FreeSWITCH! The short answer is "yes, FS can do that." > The first thing that you should do is unlearn "the Asterisk way" of > thinking. Usually there is an elegant way of doing things in FS that > wasn't possible in Ast. > > I would recommend that you start by looking at the event socket, > which is somewhat analogous to the AMI only cooler. :) I have > personally done something similar to this using the event socket and > a Perl script. The key is to learn the syntax of the originate > command. (definitely hit the wiki and IRC channel) > Are you using TDM cards for this? Just curious. > > -MC (IRC nick: mercutioviz) > > Sent from my iPhone > > On Feb 2, 2009, at 3:35 PM, "Nik Middleton" > wrote: >> Hi Guys, >> >> As a long time Asterisk user, I?m looking into freeswitch as an >> alternative mainly due to (list multiple reasons here) >> >> Can anyone give me a pointer as to how I would achieve the following? >> >> I need to replicate an emergency broadcast system currently running >> under Asterisk. >> >> At the moment, I run through a Mysql database and using the manager >> API, issues an Originate command to dial a number. >> >> When the call is answered, a message is played, and the recipient >> has the option of hitting a digit to confirm receipt. I then call >> an AGI script to update the database. >> >> Is this fairly easy to do in Freeswitch? >> >> Not looking for code, just some pointers as to what?s available to >> do the above / >> >> Regards, >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090203/b539deb4/attachment-0002.html From leon at scarlet-internet.nl Tue Feb 3 03:55:41 2009 From: leon at scarlet-internet.nl (Leon de Rooij) Date: Tue, 3 Feb 2009 12:55:41 +0100 Subject: [Freeswitch-users] debuild breaks since the last few days Message-ID: <0B3BFADA-4589-48AD-9F18-EEBDC6BAB740@scarlet-internet.nl> Hi all, I've been trying to build new debs, but debuild seems to break.. I tried trunk rev 11608 and 1.0.3RC-1 and tried building the packages with: debuild -i -us -uc -b (which worked before) And now it breaks at openzap with: cc1: warnings being treated as errors src/ozmod/ozmod_isdn/ozmod_isdn.c: In function 'writeQ931PacketToPcap': src/ozmod/ozmod_isdn/ozmod_isdn.c:220: warning: implicit declaration of function 'pcap_dump_flush' make[7]: *** [src/ozmod/ozmod_isdn/ozmod_isdn.o] Error 1 make[7]: Leaving directory `/home/fsbuilder/freeswitch-1.0.3RC1/libs/ openzap' make[6]: *** [../libopenzap.so] Error 2 make[6]: Leaving directory `/home/fsbuilder/freeswitch-1.0.3RC1/libs/ openzap/mod_openzap' make[5]: *** [all] Error 1 make[5]: Leaving directory `/home/fsbuilder/freeswitch-1.0.3RC1/libs/ openzap/mod_openzap' make[4]: *** [../../libs/openzap/mod_openzap-all] Error 1 make[4]: Leaving directory `/home/fsbuilder/freeswitch-1.0.3RC1/src/mod' make[3]: *** [all-recursive] Error 1 make[3]: Leaving directory `/home/fsbuilder/freeswitch-1.0.3RC1/src' Making all in build make[3]: Entering directory `/home/fsbuilder/freeswitch-1.0.3RC1/build' +-------- FreeSWITCH Build Complete -----------+ + FreeSWITCH has been successfully built. + + Install by running: + + + + /usr/bin/make install + +----------------------------------------------+ make[3]: Leaving directory `/home/fsbuilder/freeswitch-1.0.3RC1/build' make[2]: *** [all-recursive] Error 1 make[2]: Leaving directory `/home/fsbuilder/freeswitch-1.0.3RC1' make[1]: *** [all] Error 2 make[1]: Leaving directory `/home/fsbuilder/freeswitch-1.0.3RC1' make: *** [build-stamp] Error 2 debuild: fatal error at line 1247: debian/rules build failed Does anyone know how to fix this ? thanks, Leon From saeedahmad1981 at gmail.com Tue Feb 3 04:41:03 2009 From: saeedahmad1981 at gmail.com (Saeed Ahmed) Date: Tue, 3 Feb 2009 13:41:03 +0100 Subject: [Freeswitch-users] Strange Performance when using as SBC In-Reply-To: <49870432.5050301@laposte.net> References: <4986C42C.7030700@laposte.net><1C31435E19AB4E57981CF2AAC2CA74AA@SaeedLaptop> <4986ECA9.3040707@laposte.net><9232C06D5362494791AF713E1DF61343@SaeedLaptop> <49870432.5050301@laposte.net> Message-ID: Hi rod, It's really amazing! Well described! Could you please explain a bit why we used Kamailio? Kind Regards Saeed Ahmed Tariq -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of rod Sent: Monday, February 02, 2009 3:33 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Strange Performance when using as SBC Hi Saeed, Here is a first draft of what I did to install FS on my server. Configuration are not present, they'll be in a next release :p http://wiki.freeswitch.org/wiki/SBC_Setup My aim is to setup FS as a SBC, I hope this page could be a great startup point for others. I will update regularly based on what I did. Saeed, why are you replacing your Nextone, it's said to be one of the best commercial SBC on the market. regards. Saeed Ahmed wrote: > Thanks rod for a quick answer, > > FS is installed on Ubuntu Server. > > I am planning to replace Nextone SBC with FS, Later I'll also use openZAP to > communicate with TDM but this all depends how much calls it can take, or > maybe we can also do something in clustering environment ( I am not sure > about it). But thanks again and any further help will be highly appreciated. > > > Kind Regards > Saeed Ahmed Tariq > > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of rod > Sent: Monday, February 02, 2009 1:53 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Strange Performance when using as SBC > > Hi Saeed, > > I just created an account to share my setup on the wiki. I will detail > all the steps for a clean install of a debian64 lenny with FS used as a > SBC (next step is to try the new LCR module :) )and what I'm doing do > stress the server. > > I wrote nothing at this time so please be patient, I'm waiting for my > new hardware so that I will detail as much as possible what I'll do. > > For beginning I suggest you reading the start page on the wiki, > especially these pages: > -http://wiki.freeswitch.org/wiki/Getting_Started_Guide > -http://wiki.freeswitch.org/wiki/Dialplan_XML > > maybe you could tell more about the linux distribution you're using so > that I can give you some pointers for sipp... > > regards. > rod. > > > Saeed Ahmed wrote: > >> Hi Rod, >> >> Could you please share how you configured Sipp & FS to create a test >> environment? Especially the dial plan, sofia settings etc..., actually I >> > am > >> a newbie. I want to test it on a single FS machine. >> >> Kind Regards >> Saeed >> -----Original Message----- >> From: freeswitch-users-bounces at lists.freeswitch.org >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of rod >> Sent: Monday, February 02, 2009 11:00 AM >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] Strange Performance when using as SBC >> >> Hi Ken, Jay, >> >> thanks for pointing to proxy media, I will test. >> >> Ken, you are right, I was brain damaged (a stupid mistake) when setting >> INFO cause this kind of level could be very verbose. I'm switching to >> CRIT or ERR. >> >> Thanks guys, >> rod. >> >> thanks for >> >> Ken Rice wrote: >> >> >>> If you don't have to transcode, using proxy media mode will still save >>> > you > >>> some CPU time. This is 1/2 way between bypass media and the default media >>> interactive mode. The other draw back to this mode is if you are using FS >>> >>> >> to >> >> >>> clean up RTP and DTMF you loose those functions but they are not needed >>> > in > >>> most use cases. >>> >>> As far as the log level goes, I found that once I had things stable >>> >>> >> setting >> >> >>> the loglevel to helped a good deal... Info is probably a bit too high of >>> > a > >>> loglevel I would probably go for CRIT or ERR (2 or 1 respectively) if you >>> insist on leaving logging turned on... On a busy system these can and >>> > will > >>> generate a good deal of activity (and disk IO if using mod_logfile) >>> >>> Ken >>> >>> >>> >>> >>> >>>> From: rod >>>> Reply-To: >>>> Date: Mon, 02 Feb 2009 11:36:35 +0400 >>>> To: >>>> Subject: Re: [Freeswitch-users] Strange Performance when using as SBC >>>> >>>> Hi Ken, >>>> >>>> 1) I'd like to use FS to hide topology, so bypass media is not possible >>>> 2) done >>>> 3) done >>>> 4) not used >>>> 5) i'm using this ins switch.xml -> >>> value="info"/>, if you think an other log level is more suitable. >>>> >>>> Regarding logging, I can see in console and in the freeswitch.log that >>>> there is still a lot of NOTICE logging, see below: >>>> 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:960 >>>> switch_core_session_thread() Session 8721 >>>> (sofia/internal/sipp at 10.10.10.1:5060) Ended >>>> 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:962 >>>> switch_core_session_thread() Close Channel >>>> sofia/internal/sipp at 10.10.10.1:5060 [CS_HANGUP] >>>> 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:960 >>>> switch_core_session_thread() Session 8722 >>>> (sofia/external/9998 at 10.10.20.100) Ended >>>> 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:962 >>>> switch_core_session_thread() Close Channel >>>> sofia/external/9998 at 10.10.20.100 [CS_HANGUP] >>>> 2009-02-02 08:33:56 [NOTICE] sofia.c:3164 sofia_handle_sip_i_state() >>>> Channel [sofia/external/9998 at 10.10.20.100] has been answered >>>> 2009-02-02 08:33:56 [WARNING] mod_sofia.c:740 sofia_read_frame() >>>> Changing codec ptime to 30. I bet you have a linksys/sipura =D >>>> >>>> Do you have any idea where I can switch off this kind of logging. I >>>> thought it should be in /dialplan/internal.xml, but I see that in >>>> internal.xml -> >>>> >>>> thanks a lot for your suggestion. >>>> >>>> regards, >>>> rod >>>> >>>> Ken Rice wrote: >>>> >>>> >>>> >>>>> Dont forget there are several things you can do to increase >>>>> >>>>> >> performance... >> >> >>>>> 1) where possible use bypass media or media proxy modes >>>>> 2) mount freeswitch/db as a ram drive (if you are using voicemail with >>>>> the internal FS DBs you'll need a way to make this persistant across >>>>> reboots) >>>>> 3) see the wiki for setting reasonable ulimits >>>>> 4) (this is my oppinion others may vary) dont use mod_cdr_csv >>>>> 5) turn off (or reduce logging) in switch.conf.xml >>>>> >>>>> all of these thing can greatly improve performance. >>>>> >>>>> On Mon, Feb 2, 2009 at 1:04 AM, rod >>>> > wrote: >>>>> >>>>> Thanks Anthony, >>>>> >>>>> the setup is like this: >>>>> >>>>> sipp server ---- FS 1 ---- FS2 >>>>> >>>>> FS1 is the AMD CPU that has only one extension in dialplan that >>>>> bridges >>>>> 9999 to FS2. 9999 is the first extension in FS2 dialplan that >>>>> plays moh, >>>>> FS2 has no CPU pbm. >>>>> >>>>> FS1 is maxing out at 60 bridged calls without your option -hp. >>>>> >>>>> Using -hp, I'm now able to bridge 200 concurrent calls (a great >>>>> improvement) and the system is still reactive. CPU load is high >>>>> but not >>>>> 100% and as the system responds well, I think that doesn't matter. >>>>> >>>>> >> The >> >> >>>>> 2GB of memory are completely consumed (top command shows 700MB for >>>>> >>>>> >> FS >> >> >>>>> process). >>>>> >>>>> I understand that FS1 server is not the best hardware platform, >>>>> and I'm >>>>> waiting for new 4 cores server for testing. >>>>> I will update those numbers when testing with the new hardware. >>>>> >>>>> regards, >>>>> rod. >>>>> >>>>> Anthony Minessale wrote: >>>>> >>>>> >>>>> >>>>>> Which of the 2 machines has the load issue? You said it was one box >>>>>> calling the other. >>>>>> >>>>>> You have 2 major things against you, single CPU and AMD, but you >>>>>> should at least be able to get in the vicinity of 800-1000 calls >>>>>> >>>>>> >>>>>> >>>>> on a >>>>> >>>>> >>>>> >>>>>> box like that. >>>>>> >>>>>> Are you calling the default 9999? It's not really an appropriate >>>>>> extension for load testing. >>>>>> On the terminating box you should set up a manual extension that is >>>>>> the first one in the dial plan >>>>>> to play a wav file from preferably a ram disk or /tmp >>>>>> >>>>>> If you do plan on using this in production accept nothing less >>>>>> >>>>>> >>>>>> >>>>> than a >>>>> >>>>> >>>>> >>>>>> multi-core intel machine with at least 4 cores, the more cores the >>>>>> better because that parallel processing is where FS gets it's >>>>>> >>>>>> >>>>>> >>>>> atvantage. >>>>> >>>>> >>>>> >>>>>> On Fri, Jan 30, 2009 at 5:56 AM, rod >>>>> >>>>>> >>>>>> >>>>> >>>>> >>>>> >>>>> >>>>>> >> wrote: >>>>>> >>>>>> Dear list, >>>>>> >>>>>> I've been playing with freeswitch for some time (2 months) >>>>>> >>>>>> >>>>>> >>>>> and the >>>>> >>>>> >>>>> >>>>>> fact >>>>>> is that I'm very pleased with the functionnalities of this >>>>>> >>>>>> >>>>>> >>>>> software. >>>>> >>>>> >>>>> >>>>>> I'd like to use FS as a SBC handling media and I'm doing some >>>>>> tests with >>>>>> sipp to load the machine but I'm unable to bridge more than >>>>>> >>>>>> >>>>>> >>>>> 60 calls >>>>> >>>>> >>>>> >>>>>> without seeing the CPU being loaded at 100%. I'm sure >>>>>> >>>>>> >>>>>> >>>>> something is >>>>> >>>>> >>>>> >>>>>> going >>>>>> wrong with my setup but I'm unable to see what. >>>>>> >>>>>> The test machine has the following specs: >>>>>> Athlon XP 3500+ with 2GB of memory (I know this is not a >>>>>> >>>>>> >>>>>> >>>>> high end >>>>> >>>>> >>>>> >>>>>> machine :p) >>>>>> >>>>>> Freeswitch:/opt/freeswitch/log# cat /proc/cpuinfo >>>>>> processor : 0 >>>>>> vendor_id : AuthenticAMD >>>>>> cpu family : 15 >>>>>> model : 95 >>>>>> model name : AMD Athlon(tm) 64 Processor 3500+ >>>>>> stepping : 2 >>>>>> cpu MHz : 2199.973 >>>>>> cache size : 512 KB >>>>>> fpu : yes >>>>>> fpu_exception : yes >>>>>> cpuid level : 1 >>>>>> wp : yes >>>>>> flags : fpu vme de pse tsc msr pae mce cx8 apic >>>>>> >>>>>> >>>>>> >>>>> sep mtrr pge >>>>> >>>>> >>>>> >>>>>> mca cmov pat pse36 clflush mmx fxsr sse sse2 syscall nx mmxext >>>>>> fxsr_opt >>>>>> rdtscp lm 3dnowext 3dnow up rep_good pni cx16 lahf_lm svm >>>>>> >>>>>> >>>>>> >>>>> extapic >>>>> >>>>> >>>>> >>>>>> cr8_legacy >>>>>> bogomips : 4402.97 >>>>>> TLB size : 1024 4K pages >>>>>> clflush size : 64 >>>>>> cache_alignment : 64 >>>>>> address sizes : 40 bits physical, 48 bits virtual >>>>>> power management: ts fid vid ttp tm stc >>>>>> >>>>>> I installed FS on a fresh debian 64: >>>>>> Linux Freeswitch 2.6.26-1-amd64 #1 SMP Sat Jan 10 17:57:00 >>>>>> >>>>>> >>>>>> >>>>> UTC 2009 >>>>> >>>>> >>>>> >>>>>> x86_64 GNU/Linux >>>>>> >>>>>> I set the ulimit parameters like those on the website: >>>>>> freeswitch at internal> ... >>>>>> Freeswitch:/opt/free-svn/bin# ulimit -a >>>>>> core file size (blocks, -c) unlimited >>>>>> data seg size (kbytes, -d) unlimited >>>>>> scheduling priority (-e) 0 >>>>>> file size (blocks, -f) unlimited >>>>>> pending signals (-i) unlimited >>>>>> max locked memory (kbytes, -l) unlimited >>>>>> max memory size (kbytes, -m) unlimited >>>>>> open files (-n) 999999 >>>>>> pipe size (512 bytes, -p) 8 >>>>>> POSIX message queues (bytes, -q) unlimited >>>>>> real-time priority (-r) 0 >>>>>> stack size (kbytes, -s) 244 >>>>>> cpu time (seconds, -t) unlimited >>>>>> max user processes (-u) unlimited >>>>>> virtual memory (kbytes, -v) unlimited >>>>>> file locks (-x) unlimited >>>>>> >>>>>> >>>>>> My network setup is the following: >>>>>> >>>>>> SIPP machine (10.10.10.1/24)----------------vlan >>>>>> >>>>>> >>>>>> >>>>> >>>>> >>>>> >>>>> >>>>>> 55 >>>>>> ----------(10.10.10.254/24 >>>>>> >>>>>> >>>>>> >>>>> ) FS >>>>> >>>>> >>>>> >>>>>> (10.10.20.254/24)-------------- >>>>>> >>>>>> >>>>>> >>>>> >>>>> >>>>> >>>>> >>>>>> vlan56 >>>>>> -------------------(10.10.20.100/24 >>>>>> >>>>>> >>>>>> >>>>> ) >>>>> >>>>> >>>>> >>>>>> OTHER STOCK FS >>>>>> >>>>>> >>>>>> I launched sipp with: >>>>>> sipp -sn uac_pcap -s 9999 -r 10 -l 80 -d 60000 -mi >>>>>> >>>>>> >>>>>> >>>>> 10.10.10.1 -i >>>>> >>>>> >>>>> >>>>>> 10.10.10.1 -mp 25000 10.10.10.254:5060 >>>>>> >>>>>> >>>>>> >>>>> >>>>> >>>>> >>>>> >>>>>> The dialplan on FS is very simple: >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>> data="sofia/external/9999 at 10.10.20.100 >>>>>> >>>>>> >>>>>> >>>>> >>>> >"/> >>>>> >>>>> >>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> FreeSWITCH Version 1.0.trunk (11560M) Started. >>>>>> Crash Protection [Disabled] >>>>>> Max Sessions[1000] >>>>>> Session Rate[100] >>>>>> SQL [Enabled] >>>>>> >>>>>> >>>>>> The test is very simple: sipp dial 9999 that matches in my >>>>>> >>>>>> >>>>>> >>>>> FS dialplan >>>>> >>>>> >>>>> >>>>>> and this is bridged to an other FS machine playing music on >>>>>> >>>>>> >>>>>> >>>>> hold. >>>>> >>>>> >>>>> >>>>>> When I launch "top" I see after 30 to 40 s that FS consumes all >>>>>> the CPU >>>>>> ressources (with a mean of 50-60 % before), with 80 calls. >>>>>> When I set 70 calls, I have to wait 70-80 s before seeing >>>>>> >>>>>> >>>>>> >>>>> the same >>>>> >>>>> >>>>> >>>>>> issue. >>>>>> >>>>>> Presence is set to false on the 2 profile. >>>>>> >>>>>> I have the same issue with FS 1.0.2 that' s why I tried FS >>>>>> >>>>>> >>>>>> >>>>> 11560. >>>>> >>>>> >>>>> >>>>>> When I use the FS machine as a router to test the packet per >>>>>> >>>>>> >>>>>> >>>>> second >>>>> >>>>> >>>>> >>>>>> performance, I'm reaching 100Mbps with 8000pps in each >>>>>> >>>>>> >>>>>> >>>>> direction (from >>>>> >>>>> >>>>> >>>>>> vlan 55 to vlan56) with less than 12% CPU. So that I don't think >>>>>> there's >>>>>> an issue with the network. >>>>>> >>>>>> Here is an "mpstat -P ALL 1" to show you what's happening >>>>>> >>>>>> >>>>>> >>>>> suddenly >>>>> >>>>> >>>>> >>>>>> with >>>>>> 70 bridge calls: >>>>>> 12:31:26 CPU %user %nice %sys %iowait %irq %soft >>>>>> %steal %idle intr/s >>>>>> 12:31:27 all 3,00 0,00 3,00 0,00 1,00 4,00 >>>>>> 0,00 89,00 6241,00 >>>>>> 12:31:27 0 3,00 0,00 3,00 0,00 1,00 4,00 >>>>>> 0,00 89,00 6241,00 >>>>>> >>>>>> 12:31:27 CPU %user %nice %sys %iowait %irq %soft >>>>>> %steal %idle intr/s >>>>>> 12:31:28 all 14,14 0,00 56,57 0,00 2,02 5,05 >>>>>> 0,00 22,22 6035,35 >>>>>> 12:31:28 0 14,14 0,00 56,57 0,00 2,02 5,05 >>>>>> 0,00 22,22 6035,35 >>>>>> >>>>>> 12:31:28 CPU %user %nice %sys %iowait %irq %soft >>>>>> %steal %idle intr/s >>>>>> 12:31:29 all 24,75 0,00 67,33 0,00 0,99 6,93 >>>>>> 0,00 0,00 5483,17 >>>>>> 12:31:29 0 24,75 0,00 67,33 0,00 0,99 6,93 >>>>>> 0,00 0,00 5483,17 >>>>>> >>>>>> >>>>>> The CPU is going from 89% idle to 0% in less than 2 seconds. >>>>>> >>>>>> I know that I don't have to expect too much from this kind of >>>>>> hardware, >>>>>> but it seems strange that the CPU power vanished so suddenly. >>>>>> >>>>>> Thanks a lot for the guys that have read this long mail :p >>>>>> >>>>>> kind regards, >>>>>> rod >>>>>> >>>>>> >>>>>> _______________________________________________ >>>>>> Freeswitch-users mailing list >>>>>> Freeswitch-users at lists.freeswitch.org >>>>>> >>>>>> >>>>>> >>>>> >>>>> >>>>> >>>>> >>>>>> >>>>> >>>>>> >>>>>> >>>>> > >>>>> >>>>> >>>>> >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> >>>>>> >>>>>> >>>>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> >>>>> >>>>> >>>>> >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> -- >>>>>> Anthony Minessale II >>>>>> >>>>>> FreeSWITCH http://www.freeswitch.org/ >>>>>> ClueCon http://www.cluecon.com/ >>>>>> >>>>>> AIM: anthm >>>>>> MSN:anthony_minessale at hotmail.com >>>>>> >>>>>> >>>>>> >>>>> >>>>> >>>>> >>>>> >>>>>> >>>>> >>>>>> >>>>>> >>>>> > >>>>> >>>>> >>>>> >>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>>>>> >>>>>> >>>>>> >>>>> >>>>> >>>>> >>>>> >>>>>> >>>>> >>>>>> >>>>>> >>>>> > >>>>> >>>>> >>>>> >>>>>> IRC: irc.freenode.net >>>>>> >>>>>> >>>>>> >>>>> #freeswitch >>>>> >>>>> >>>>> >>>>>> FreeSWITCH Developer Conference >>>>>> sip:888 at conference.freeswitch.org >>>>>> >>>>>> >>>>>> >>>>> >>>>> >>>>> >>>>> >>>>>> >>>>> >>>>>> >>>>>> >>>>> > >>>>> >>>>> >>>>> >>>>>> iax:guest at conference.freeswitch.org/888 >>>>>> >>>>>> >>>>>> >>>>> >>>>> >>>>> >>>>> >>>>>> >>>>>> googletalk:conf+888 at conference.freeswitch.org >>>>>> >>>>>> >>>>>> >>>>> >>>>> >>>>> >>>>> >>>>>> >>>>> >>>>>> >>>>>> >>>>> > >>>>> >>>>> >>>>> >>>>>> pstn:213-799-1400 >>>>>> >>>>>> >>>>>> >>>>>> >> ------------------------------------------------------------------------ >> >> >>>>> >>>>> >>>>> >>>>>> _______________________________________________ >>>>>> Freeswitch-users mailing list >>>>>> Freeswitch-users at lists.freeswitch.org >>>>>> >>>>>> >>>>>> >>>>> >>>>> >>>>> >>>>> >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> >>>>>> >>>>>> >>>>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> >>>>> >>>>> >>>>> >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>>> >>>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> >>>>> >>>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> >>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> >>>>> > ------------------------------------------------------------------------ > >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> >>>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> >>>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >>> >>> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From raul at etellicom.com Tue Feb 3 05:00:28 2009 From: raul at etellicom.com (Raul Fragoso) Date: Tue, 03 Feb 2009 11:00:28 -0200 Subject: [Freeswitch-users] debuild breaks since the last few days In-Reply-To: <0B3BFADA-4589-48AD-9F18-EEBDC6BAB740@scarlet-internet.nl> References: <0B3BFADA-4589-48AD-9F18-EEBDC6BAB740@scarlet-internet.nl> Message-ID: <1233666028.24619.0.camel@stargate> I believe that installing the libpcap and libpcap-dev packages may fix your problem. -- Raul On Tue, 2009-02-03 at 12:55 +0100, Leon de Rooij wrote: > Hi all, > > I've been trying to build new debs, but debuild seems to break.. > > I tried trunk rev 11608 and 1.0.3RC-1 and tried building the packages > with: > > debuild -i -us -uc -b > > (which worked before) > > And now it breaks at openzap with: > > cc1: warnings being treated as errors > src/ozmod/ozmod_isdn/ozmod_isdn.c: In function 'writeQ931PacketToPcap': > src/ozmod/ozmod_isdn/ozmod_isdn.c:220: warning: implicit declaration > of function 'pcap_dump_flush' > make[7]: *** [src/ozmod/ozmod_isdn/ozmod_isdn.o] Error 1 > make[7]: Leaving directory `/home/fsbuilder/freeswitch-1.0.3RC1/libs/ > openzap' > make[6]: *** [../libopenzap.so] Error 2 > make[6]: Leaving directory `/home/fsbuilder/freeswitch-1.0.3RC1/libs/ > openzap/mod_openzap' > make[5]: *** [all] Error 1 > make[5]: Leaving directory `/home/fsbuilder/freeswitch-1.0.3RC1/libs/ > openzap/mod_openzap' > make[4]: *** [../../libs/openzap/mod_openzap-all] Error 1 > make[4]: Leaving directory `/home/fsbuilder/freeswitch-1.0.3RC1/src/mod' > make[3]: *** [all-recursive] Error 1 > make[3]: Leaving directory `/home/fsbuilder/freeswitch-1.0.3RC1/src' > Making all in build > make[3]: Entering directory `/home/fsbuilder/freeswitch-1.0.3RC1/build' > +-------- FreeSWITCH Build Complete -----------+ > + FreeSWITCH has been successfully built. + > + Install by running: + > + + > + /usr/bin/make install + > +----------------------------------------------+ > make[3]: Leaving directory `/home/fsbuilder/freeswitch-1.0.3RC1/build' > make[2]: *** [all-recursive] Error 1 > make[2]: Leaving directory `/home/fsbuilder/freeswitch-1.0.3RC1' > make[1]: *** [all] Error 2 > make[1]: Leaving directory `/home/fsbuilder/freeswitch-1.0.3RC1' > make: *** [build-stamp] Error 2 > debuild: fatal error at line 1247: > debian/rules build failed > > Does anyone know how to fix this ? > > thanks, > > Leon > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From helmut.kuper at ewetel.de Tue Feb 3 05:03:31 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Tue, 03 Feb 2009 14:03:31 +0100 Subject: [Freeswitch-users] debuild breaks since the last few days In-Reply-To: <0B3BFADA-4589-48AD-9F18-EEBDC6BAB740@scarlet-internet.nl> References: <0B3BFADA-4589-48AD-9F18-EEBDC6BAB740@scarlet-internet.nl> Message-ID: <498840A3.30509@ewetel.de> Hello, yes, you have openzap upgraded to r632. Then recompile it. Make sure you have libpcap installed and pcap devel files regards helmut Am 03.02.2009 12:55, schrieb Leon de Rooij: > Hi all, > > I've been trying to build new debs, but debuild seems to break.. > > I tried trunk rev 11608 and 1.0.3RC-1 and tried building the packages > with: > > debuild -i -us -uc -b > > (which worked before) > > And now it breaks at openzap with: > > cc1: warnings being treated as errors > src/ozmod/ozmod_isdn/ozmod_isdn.c: In function 'writeQ931PacketToPcap': > src/ozmod/ozmod_isdn/ozmod_isdn.c:220: warning: implicit declaration > of function 'pcap_dump_flush' > make[7]: *** [src/ozmod/ozmod_isdn/ozmod_isdn.o] Error 1 > make[7]: Leaving directory `/home/fsbuilder/freeswitch-1.0.3RC1/libs/ > openzap' > make[6]: *** [../libopenzap.so] Error 2 > make[6]: Leaving directory `/home/fsbuilder/freeswitch-1.0.3RC1/libs/ > openzap/mod_openzap' > make[5]: *** [all] Error 1 > make[5]: Leaving directory `/home/fsbuilder/freeswitch-1.0.3RC1/libs/ > openzap/mod_openzap' > make[4]: *** [../../libs/openzap/mod_openzap-all] Error 1 > make[4]: Leaving directory `/home/fsbuilder/freeswitch-1.0.3RC1/src/mod' > make[3]: *** [all-recursive] Error 1 > make[3]: Leaving directory `/home/fsbuilder/freeswitch-1.0.3RC1/src' > Making all in build > make[3]: Entering directory `/home/fsbuilder/freeswitch-1.0.3RC1/build' > +-------- FreeSWITCH Build Complete -----------+ > + FreeSWITCH has been successfully built. + > + Install by running: + > + + > + /usr/bin/make install + > +----------------------------------------------+ > make[3]: Leaving directory `/home/fsbuilder/freeswitch-1.0.3RC1/build' > make[2]: *** [all-recursive] Error 1 > make[2]: Leaving directory `/home/fsbuilder/freeswitch-1.0.3RC1' > make[1]: *** [all] Error 2 > make[1]: Leaving directory `/home/fsbuilder/freeswitch-1.0.3RC1' > make: *** [build-stamp] Error 2 > debuild: fatal error at line 1247: > debian/rules build failed > > Does anyone know how to fix this ? > > thanks, > > Leon > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > -- Mit freundlichen Gr??en Helmut Kuper Finanzdienstleistungen und Entwicklung Telefax: (0441) 8000-2799 mailto:helmut.kuper at ewetel.de ___________________________________ EWE TEL GmbH Cloppenburger Stra?e 310 26133 Oldenburg EWE TEL GmbH Handelsregister Amtsgericht Oldenburg HRB 3723 Vorsitzender des Aufsichtsrates: Heiko Harms Gesch?ftsf?hrung: Hans-Joachim Iken (Vorsitzender), Dr. Norbert Schulz, Dirk Thole Homepage: http://www.ewetel.de ___________________________________ From raul at etellicom.com Tue Feb 3 05:12:28 2009 From: raul at etellicom.com (Raul Fragoso) Date: Tue, 03 Feb 2009 11:12:28 -0200 Subject: [Freeswitch-users] Generating calls from external source In-Reply-To: <40B58CBF-9D0F-43A3-B36F-BD05916756E4@3c.co.uk> References: <40B58CBF-9D0F-43A3-B36F-BD05916756E4@3c.co.uk> Message-ID: <1233666748.24619.8.camel@stargate> In addition do David's suggestion, you probably want to have your application to watch for some specific events after the call is originated and take action based on them. For example, you could watch for the CHANNEL_ANSWER event and play some audio file waiting for some digit, which is generated by the DTMF event. To watch only for those specific events, you should do the following just after authentication (still using Perl as an example, but the mod_event_socket is language agnostic), then you will receive those events from FreeSWITCH through the socket stream: ... print $sock "auth XXX\n\n"; print $sock "event plain CHANNEL_ANSWER DTMF\n\n"; ... To see a list of available events, please look at the following wiki pages: http://wiki.freeswitch.org/wiki/Mod_event_socket#event http://wiki.freeswitch.org/wiki/Event_list Regards, Raul On Tue, 2009-02-03 at 09:46 +0000, David Knell wrote: > Hi Nik, > > > Here's a snipped in Perl that launches an outbound call: > > > if (my $sock = IO::Socket::INET->new(Proto =>'tcp', PeerAddr => > '127.0.0.1', PeerPort => 8021)) { > print $sock "auth XXX\n\n"; > print $sock "api originate {softivr_id=$siid,src_softivr_id= > $siid,softivr_outdial=true}sofia/frombt/$ntd at 1.2.3.4 $service\n\n"; > $sock->close(); > } > > > - it does no error checking or anything, but (line by line) it: > - opens a socket to the event socket interface > - authenticates > - issues an originate which dials out to the number in $ntd. The > bits in {} set a bunch of variables on the channel, which are used by > the software which processes the call later on. The call is linked to > the extension in $service - FS looks this up in the dialplan - which > handles our end. > - closes the socket > > > Cheers -- > > > Dave > > > > > Thanks for that, coming from a C++ background it?s a refreshing > > change to be looking at something that seems logical and efficient. > > > > I?d briefly looked at the event socket and wondered if that was the > > way to go. I presume that there?s some sort of event generation > > that can trigger and external process as well somewhere, though all > > I need to do is update mysql (hopefully using some sort of pooled > > connection) > > > > I?m not using a TDM card, I have a direct interconnect with the PSTN > > breakout provider with 1,500 channels available to me. I?m finding > > Asterisk proving to be less than stable at high call volumes and > > load values spike at more than 100 calls with billing/accounting in > > place, hence my interest in FS. The only thing that?s concerning me > > is XML at the moment. Lots of code and very wordy. I?m sure I?ll > > appreciate why XML given time > > > > Regards, > > > > > > ____________________________________________________________________ > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael S Collins > > Sent: 03 February 2009 01:17 > > To: freeswitch-users at lists.freeswitch.org > > Subject: Re: [Freeswitch-users] Generating calls from external > > source > > > > Nik, > > > > Welcome to FreeSWITCH! The short answer is "yes, FS can do that." > > The first thing that you should do is unlearn "the Asterisk way" of > > thinking. Usually there is an elegant way of doing things in FS that > > wasn't possible in Ast. > > > > I would recommend that you start by looking at the event socket, > > which is somewhat analogous to the AMI only cooler. :) I have > > personally done something similar to this using the event socket and > > a Perl script. The key is to learn the syntax of the originate > > command. (definitely hit the wiki and IRC channel) > > Are you using TDM cards for this? Just curious. > > > > -MC (IRC nick: mercutioviz) > > > > Sent from my iPhone > > > > On Feb 2, 2009, at 3:35 PM, "Nik Middleton" > > wrote: > > > Hi Guys, > > > > > > As a long time Asterisk user, I?m looking into freeswitch as an > > > alternative mainly due to (list multiple reasons here) > > > > > > Can anyone give me a pointer as to how I would achieve the > > > following? > > > > > > I need to replicate an emergency broadcast system currently > > > running under Asterisk. > > > > > > At the moment, I run through a Mysql database and using the > > > manager API, issues an Originate command to dial a number. > > > > > > When the call is answered, a message is played, and the recipient > > > has the option of hitting a digit to confirm receipt. I then call > > > an AGI script to update the database. > > > > > > Is this fairly easy to do in Freeswitch? > > > > > > Not looking for code, just some pointers as to what?s available to > > > do the above / > > > > > > Regards, > > > _______________________________________________ > > > Freeswitch-users mailing list > > > Freeswitch-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From kawarod at laposte.net Tue Feb 3 05:27:35 2009 From: kawarod at laposte.net (rod) Date: Tue, 03 Feb 2009 17:27:35 +0400 Subject: [Freeswitch-users] Strange Performance when using as SBC In-Reply-To: References: <4986C42C.7030700@laposte.net><1C31435E19AB4E57981CF2AAC2CA74AA@SaeedLaptop> <4986ECA9.3040707@laposte.net><9232C06D5362494791AF713E1DF61343@SaeedLaptop> <49870432.5050301@laposte.net> Message-ID: <49884647.2080306@laposte.net> Hi Saaed, thanks for encouraging. I'm using Kamailio to get access to the carrierroute module. Carrierroute is a module that is able to handle very large routing table (excerpt from carrierroute page: "This modules scales up to more than a few million users, and is able to handle more than several hundred thousand routing table entries", Greatings to Henning Westerholt). When I did my first test with FS, LCR module was not available and as I'm not a programmer I had to deal with existing tools and being able to handle a route table with approx 160 000 entries. I'm not a programmer so I relies on SIP (which I understand better than C or C++ :p) and the possibility to define specific header to exchange message between FS and Kamailio at the cost of just an extra SIP invite parsing (maybe a bad thing for very very high call per second rate) So if you follow the setup on the wiki, FS will pass the number to examine, and Kamailio will send the best route to use depending on probability (for load sharing, eg: 10% on a gateway, 20% on an other and 70% on the last one) and matching longest prefix. Then FS uses those route. You could also update the kamailio database and then issue a "kamctl cr reload" to load the new routing table. Maybe this is not the best setup, but my aim is to share what I did so that we could converge to the best solution to use FS as a SBC, that's why I provided also some indications to optimize FS based on what I read on the list and the wiki. The next steps are scalability testing, maybe a php (or whatever else) frontend to populate carrierroute table depending on the cost of many carriers (any people willing to contribute, don't rely on me for this :o), FS redundancy (I'd like to use LVS and some tools like sipsack to check the SIP process, but I'm far from having done any interesting things on that) that is lacking against commercial SBC, some scripts to graph the number of calls... (please an SNMP module :p) An other way to achieve LCR could be to use the new LCR module, and I think that Ken Rice on this list can provide advices for a high performance LCR setup. I subscribed to this list a long time ago, and my feeling is that FS is a great piece of software with a great community, so that I decided that it could be great to contribute. regards, rod Saeed Ahmed wrote: > Hi rod, > > It's really amazing! Well described! > > Could you please explain a bit why we used Kamailio? > > Kind Regards > Saeed Ahmed Tariq > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of rod > Sent: Monday, February 02, 2009 3:33 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Strange Performance when using as SBC > > Hi Saeed, > > Here is a first draft of what I did to install FS on my server. > Configuration are not present, they'll be in a next release :p > http://wiki.freeswitch.org/wiki/SBC_Setup > > My aim is to setup FS as a SBC, I hope this page could be a great > startup point for others. I will update regularly based on what I did. > > Saeed, why are you replacing your Nextone, it's said to be one of the > best commercial SBC on the market. > > regards. > > Saeed Ahmed wrote: > >> Thanks rod for a quick answer, >> >> FS is installed on Ubuntu Server. >> >> I am planning to replace Nextone SBC with FS, Later I'll also use openZAP >> > to > >> communicate with TDM but this all depends how much calls it can take, or >> maybe we can also do something in clustering environment ( I am not sure >> about it). But thanks again and any further help will be highly >> > appreciated. > >> Kind Regards >> Saeed Ahmed Tariq >> >> >> >> -----Original Message----- >> From: freeswitch-users-bounces at lists.freeswitch.org >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of rod >> Sent: Monday, February 02, 2009 1:53 PM >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] Strange Performance when using as SBC >> >> Hi Saeed, >> >> I just created an account to share my setup on the wiki. I will detail >> all the steps for a clean install of a debian64 lenny with FS used as a >> SBC (next step is to try the new LCR module :) )and what I'm doing do >> stress the server. >> >> I wrote nothing at this time so please be patient, I'm waiting for my >> new hardware so that I will detail as much as possible what I'll do. >> >> For beginning I suggest you reading the start page on the wiki, >> especially these pages: >> -http://wiki.freeswitch.org/wiki/Getting_Started_Guide >> -http://wiki.freeswitch.org/wiki/Dialplan_XML >> >> maybe you could tell more about the linux distribution you're using so >> that I can give you some pointers for sipp... >> >> regards. >> rod. >> >> >> Saeed Ahmed wrote: >> >> >>> Hi Rod, >>> >>> Could you please share how you configured Sipp & FS to create a test >>> environment? Especially the dial plan, sofia settings etc..., actually I >>> >>> >> am >> >> >>> a newbie. I want to test it on a single FS machine. >>> >>> Kind Regards >>> Saeed >>> -----Original Message----- >>> From: freeswitch-users-bounces at lists.freeswitch.org >>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of rod >>> Sent: Monday, February 02, 2009 11:00 AM >>> To: freeswitch-users at lists.freeswitch.org >>> Subject: Re: [Freeswitch-users] Strange Performance when using as SBC >>> >>> Hi Ken, Jay, >>> >>> thanks for pointing to proxy media, I will test. >>> >>> Ken, you are right, I was brain damaged (a stupid mistake) when setting >>> INFO cause this kind of level could be very verbose. I'm switching to >>> CRIT or ERR. >>> >>> Thanks guys, >>> rod. >>> >>> thanks for >>> >>> Ken Rice wrote: >>> >>> >>> >>>> If you don't have to transcode, using proxy media mode will still save >>>> >>>> >> you >> >> >>>> some CPU time. This is 1/2 way between bypass media and the default >>>> > media > >>>> interactive mode. The other draw back to this mode is if you are using >>>> > FS > >>>> >>>> >>>> >>> to >>> >>> >>> >>>> clean up RTP and DTMF you loose those functions but they are not needed >>>> >>>> >> in >> >> >>>> most use cases. >>>> >>>> As far as the log level goes, I found that once I had things stable >>>> >>>> >>>> >>> setting >>> >>> >>> >>>> the loglevel to helped a good deal... Info is probably a bit too high of >>>> >>>> >> a >> >> >>>> loglevel I would probably go for CRIT or ERR (2 or 1 respectively) if >>>> > you > >>>> insist on leaving logging turned on... On a busy system these can and >>>> >>>> >> will >> >> >>>> generate a good deal of activity (and disk IO if using mod_logfile) >>>> >>>> Ken >>>> >>>> >>>> >>>> >>>> >>>> >>>>> From: rod >>>>> Reply-To: >>>>> Date: Mon, 02 Feb 2009 11:36:35 +0400 >>>>> To: >>>>> Subject: Re: [Freeswitch-users] Strange Performance when using as SBC >>>>> >>>>> Hi Ken, >>>>> >>>>> 1) I'd like to use FS to hide topology, so bypass media is not possible >>>>> 2) done >>>>> 3) done >>>>> 4) not used >>>>> 5) i'm using this ins switch.xml -> >>>> value="info"/>, if you think an other log level is more suitable. >>>>> >>>>> Regarding logging, I can see in console and in the freeswitch.log that >>>>> there is still a lot of NOTICE logging, see below: >>>>> 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:960 >>>>> switch_core_session_thread() Session 8721 >>>>> (sofia/internal/sipp at 10.10.10.1:5060) Ended >>>>> 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:962 >>>>> switch_core_session_thread() Close Channel >>>>> sofia/internal/sipp at 10.10.10.1:5060 [CS_HANGUP] >>>>> 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:960 >>>>> switch_core_session_thread() Session 8722 >>>>> (sofia/external/9998 at 10.10.20.100) Ended >>>>> 2009-02-02 08:33:56 [NOTICE] switch_core_session.c:962 >>>>> switch_core_session_thread() Close Channel >>>>> sofia/external/9998 at 10.10.20.100 [CS_HANGUP] >>>>> 2009-02-02 08:33:56 [NOTICE] sofia.c:3164 sofia_handle_sip_i_state() >>>>> Channel [sofia/external/9998 at 10.10.20.100] has been answered >>>>> 2009-02-02 08:33:56 [WARNING] mod_sofia.c:740 sofia_read_frame() >>>>> Changing codec ptime to 30. I bet you have a linksys/sipura =D >>>>> >>>>> Do you have any idea where I can switch off this kind of logging. I >>>>> thought it should be in /dialplan/internal.xml, but I see that in >>>>> internal.xml -> >>>>> >>>>> thanks a lot for your suggestion. >>>>> >>>>> regards, >>>>> rod >>>>> >>>>> Ken Rice wrote: >>>>> >>>>> >>>>> >>>>> >>>>>> Dont forget there are several things you can do to increase >>>>>> >>>>>> >>>>>> >>> performance... >>> >>> >>> >>>>>> 1) where possible use bypass media or media proxy modes >>>>>> 2) mount freeswitch/db as a ram drive (if you are using voicemail with >>>>>> the internal FS DBs you'll need a way to make this persistant across >>>>>> reboots) >>>>>> 3) see the wiki for setting reasonable ulimits >>>>>> 4) (this is my oppinion others may vary) dont use mod_cdr_csv >>>>>> 5) turn off (or reduce logging) in switch.conf.xml >>>>>> >>>>>> all of these thing can greatly improve performance. >>>>>> >>>>>> On Mon, Feb 2, 2009 at 1:04 AM, rod >>>>> > wrote: >>>>>> >>>>>> Thanks Anthony, >>>>>> >>>>>> the setup is like this: >>>>>> >>>>>> sipp server ---- FS 1 ---- FS2 >>>>>> >>>>>> FS1 is the AMD CPU that has only one extension in dialplan that >>>>>> bridges >>>>>> 9999 to FS2. 9999 is the first extension in FS2 dialplan that >>>>>> plays moh, >>>>>> FS2 has no CPU pbm. >>>>>> >>>>>> FS1 is maxing out at 60 bridged calls without your option -hp. >>>>>> >>>>>> Using -hp, I'm now able to bridge 200 concurrent calls (a great >>>>>> improvement) and the system is still reactive. CPU load is high >>>>>> but not >>>>>> 100% and as the system responds well, I think that doesn't matter. >>>>>> >>>>>> >>>>>> >>> The >>> >>> >>> >>>>>> 2GB of memory are completely consumed (top command shows 700MB for >>>>>> >>>>>> >>>>>> >>> FS >>> >>> >>> >>>>>> process). >>>>>> >>>>>> I understand that FS1 server is not the best hardware platform, >>>>>> and I'm >>>>>> waiting for new 4 cores server for testing. >>>>>> I will update those numbers when testing with the new hardware. >>>>>> >>>>>> regards, >>>>>> rod. >>>>>> >>>>>> Anthony Minessale wrote: >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> Which of the 2 machines has the load issue? You said it was one box >>>>>>> calling the other. >>>>>>> >>>>>>> You have 2 major things against you, single CPU and AMD, but you >>>>>>> should at least be able to get in the vicinity of 800-1000 calls >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> on a >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> box like that. >>>>>>> >>>>>>> Are you calling the default 9999? It's not really an appropriate >>>>>>> extension for load testing. >>>>>>> On the terminating box you should set up a manual extension that is >>>>>>> the first one in the dial plan >>>>>>> to play a wav file from preferably a ram disk or /tmp >>>>>>> >>>>>>> If you do plan on using this in production accept nothing less >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> than a >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> multi-core intel machine with at least 4 cores, the more cores the >>>>>>> better because that parallel processing is where FS gets it's >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> atvantage. >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> On Fri, Jan 30, 2009 at 5:56 AM, rod >>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> >> wrote: >>>>>>> >>>>>>> Dear list, >>>>>>> >>>>>>> I've been playing with freeswitch for some time (2 months) >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> and the >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> fact >>>>>>> is that I'm very pleased with the functionnalities of this >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> software. >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> I'd like to use FS as a SBC handling media and I'm doing some >>>>>>> tests with >>>>>>> sipp to load the machine but I'm unable to bridge more than >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> 60 calls >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> without seeing the CPU being loaded at 100%. I'm sure >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> something is >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> going >>>>>>> wrong with my setup but I'm unable to see what. >>>>>>> >>>>>>> The test machine has the following specs: >>>>>>> Athlon XP 3500+ with 2GB of memory (I know this is not a >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> high end >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> machine :p) >>>>>>> >>>>>>> Freeswitch:/opt/freeswitch/log# cat /proc/cpuinfo >>>>>>> processor : 0 >>>>>>> vendor_id : AuthenticAMD >>>>>>> cpu family : 15 >>>>>>> model : 95 >>>>>>> model name : AMD Athlon(tm) 64 Processor 3500+ >>>>>>> stepping : 2 >>>>>>> cpu MHz : 2199.973 >>>>>>> cache size : 512 KB >>>>>>> fpu : yes >>>>>>> fpu_exception : yes >>>>>>> cpuid level : 1 >>>>>>> wp : yes >>>>>>> flags : fpu vme de pse tsc msr pae mce cx8 apic >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> sep mtrr pge >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> mca cmov pat pse36 clflush mmx fxsr sse sse2 syscall nx mmxext >>>>>>> fxsr_opt >>>>>>> rdtscp lm 3dnowext 3dnow up rep_good pni cx16 lahf_lm svm >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> extapic >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> cr8_legacy >>>>>>> bogomips : 4402.97 >>>>>>> TLB size : 1024 4K pages >>>>>>> clflush size : 64 >>>>>>> cache_alignment : 64 >>>>>>> address sizes : 40 bits physical, 48 bits virtual >>>>>>> power management: ts fid vid ttp tm stc >>>>>>> >>>>>>> I installed FS on a fresh debian 64: >>>>>>> Linux Freeswitch 2.6.26-1-amd64 #1 SMP Sat Jan 10 17:57:00 >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> UTC 2009 >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> x86_64 GNU/Linux >>>>>>> >>>>>>> I set the ulimit parameters like those on the website: >>>>>>> freeswitch at internal> ... >>>>>>> Freeswitch:/opt/free-svn/bin# ulimit -a >>>>>>> core file size (blocks, -c) unlimited >>>>>>> data seg size (kbytes, -d) unlimited >>>>>>> scheduling priority (-e) 0 >>>>>>> file size (blocks, -f) unlimited >>>>>>> pending signals (-i) unlimited >>>>>>> max locked memory (kbytes, -l) unlimited >>>>>>> max memory size (kbytes, -m) unlimited >>>>>>> open files (-n) 999999 >>>>>>> pipe size (512 bytes, -p) 8 >>>>>>> POSIX message queues (bytes, -q) unlimited >>>>>>> real-time priority (-r) 0 >>>>>>> stack size (kbytes, -s) 244 >>>>>>> cpu time (seconds, -t) unlimited >>>>>>> max user processes (-u) unlimited >>>>>>> virtual memory (kbytes, -v) unlimited >>>>>>> file locks (-x) unlimited >>>>>>> >>>>>>> >>>>>>> My network setup is the following: >>>>>>> >>>>>>> SIPP machine (10.10.10.1/24)----------------vlan >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> 55 >>>>>>> ----------(10.10.10.254/24 >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> ) FS >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> (10.10.20.254/24)-------------- >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> vlan56 >>>>>>> -------------------(10.10.20.100/24 >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> ) >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> OTHER STOCK FS >>>>>>> >>>>>>> >>>>>>> I launched sipp with: >>>>>>> sipp -sn uac_pcap -s 9999 -r 10 -l 80 -d 60000 -mi >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> 10.10.10.1 -i >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> 10.10.10.1 -mp 25000 10.10.10.254:5060 >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> The dialplan on FS is very simple: >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> data="sofia/external/9999 at 10.10.20.100 >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> >>>>> >"/> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> FreeSWITCH Version 1.0.trunk (11560M) Started. >>>>>>> Crash Protection [Disabled] >>>>>>> Max Sessions[1000] >>>>>>> Session Rate[100] >>>>>>> SQL [Enabled] >>>>>>> >>>>>>> >>>>>>> The test is very simple: sipp dial 9999 that matches in my >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> FS dialplan >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> and this is bridged to an other FS machine playing music on >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> hold. >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> When I launch "top" I see after 30 to 40 s that FS consumes all >>>>>>> the CPU >>>>>>> ressources (with a mean of 50-60 % before), with 80 calls. >>>>>>> When I set 70 calls, I have to wait 70-80 s before seeing >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> the same >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> issue. >>>>>>> >>>>>>> Presence is set to false on the 2 profile. >>>>>>> >>>>>>> I have the same issue with FS 1.0.2 that' s why I tried FS >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> 11560. >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> When I use the FS machine as a router to test the packet per >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> second >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> performance, I'm reaching 100Mbps with 8000pps in each >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> direction (from >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> vlan 55 to vlan56) with less than 12% CPU. So that I don't think >>>>>>> there's >>>>>>> an issue with the network. >>>>>>> >>>>>>> Here is an "mpstat -P ALL 1" to show you what's happening >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> suddenly >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> with >>>>>>> 70 bridge calls: >>>>>>> 12:31:26 CPU %user %nice %sys %iowait %irq %soft >>>>>>> %steal %idle intr/s >>>>>>> 12:31:27 all 3,00 0,00 3,00 0,00 1,00 4,00 >>>>>>> 0,00 89,00 6241,00 >>>>>>> 12:31:27 0 3,00 0,00 3,00 0,00 1,00 4,00 >>>>>>> 0,00 89,00 6241,00 >>>>>>> >>>>>>> 12:31:27 CPU %user %nice %sys %iowait %irq %soft >>>>>>> %steal %idle intr/s >>>>>>> 12:31:28 all 14,14 0,00 56,57 0,00 2,02 5,05 >>>>>>> 0,00 22,22 6035,35 >>>>>>> 12:31:28 0 14,14 0,00 56,57 0,00 2,02 5,05 >>>>>>> 0,00 22,22 6035,35 >>>>>>> >>>>>>> 12:31:28 CPU %user %nice %sys %iowait %irq %soft >>>>>>> %steal %idle intr/s >>>>>>> 12:31:29 all 24,75 0,00 67,33 0,00 0,99 6,93 >>>>>>> 0,00 0,00 5483,17 >>>>>>> 12:31:29 0 24,75 0,00 67,33 0,00 0,99 6,93 >>>>>>> 0,00 0,00 5483,17 >>>>>>> >>>>>>> >>>>>>> The CPU is going from 89% idle to 0% in less than 2 seconds. >>>>>>> >>>>>>> I know that I don't have to expect too much from this kind of >>>>>>> hardware, >>>>>>> but it seems strange that the CPU power vanished so suddenly. >>>>>>> >>>>>>> Thanks a lot for the guys that have read this long mail :p >>>>>>> >>>>>>> kind regards, >>>>>>> rod >>>>>>> >>>>>>> >>>>>>> _______________________________________________ >>>>>>> Freeswitch-users mailing list >>>>>>> Freeswitch-users at lists.freeswitch.org >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> >>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> > >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >>> >>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> -- >>>>>>> Anthony Minessale II >>>>>>> >>>>>>> FreeSWITCH http://www.freeswitch.org/ >>>>>>> ClueCon http://www.cluecon.com/ >>>>>>> >>>>>>> AIM: anthm >>>>>>> MSN:anthony_minessale at hotmail.com >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> >>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> > >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> >>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> > >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> IRC: irc.freenode.net >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> #freeswitch >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> FreeSWITCH Developer Conference >>>>>>> sip:888 at conference.freeswitch.org >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> >>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> > >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> iax:guest at conference.freeswitch.org/888 >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> >>>>>>> googletalk:conf+888 at conference.freeswitch.org >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> >>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> > >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> pstn:213-799-1400 >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>> ------------------------------------------------------------------------ >>> >>> >>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> _______________________________________________ >>>>>>> Freeswitch-users mailing list >>>>>>> Freeswitch-users at lists.freeswitch.org >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >>> >>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> _______________________________________________ >>>>>> Freeswitch-users mailing list >>>>>> Freeswitch-users at lists.freeswitch.org >>>>>> >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> >>>>>> >>>>>> >>>>>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >>> >>> >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >> ------------------------------------------------------------------------ >> >> >>>>>> _______________________________________________ >>>>>> Freeswitch-users mailing list >>>>>> Freeswitch-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> >>>>>> >>>>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> >>>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> >>>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> >>>> >>>> >>>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >>> >>> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > From raul at etellicom.com Tue Feb 3 05:33:49 2009 From: raul at etellicom.com (Raul Fragoso) Date: Tue, 03 Feb 2009 11:33:49 -0200 Subject: [Freeswitch-users] Application language to support C or C++? In-Reply-To: <2cbf225c0902021821u2bc8b622ka5d9f94f05a968ae@mail.gmail.com> References: <2cbf225c0902021821u2bc8b622ka5d9f94f05a968ae@mail.gmail.com> Message-ID: <1233668029.24619.29.camel@stargate> Depending on what you want to do, I suggest having a look at mod_event_socket: http://wiki.freeswitch.org/wiki/Mod_event_socket That module is a socket based interface that provides a vast range of options to control FreeSWITCH and its applications. Just for the record, my application is entirely written in C++ and uses FreeSWITCH as a back-end for providing PBX functionality through a combination of mod_event_socket and mod_xml_curl. Regards, Raul On Tue, 2009-02-03 at 10:21 +0800, lee jason wrote: > Hi All, > > I saw the applications using FreeSwitch library can be written > in JavaScript, Perl, Python and Lua but I need to use Linux C or C++ > for applications, Is FreeSwitch can supported it? Where can I get the > sample codes? My Linux platform is base on Fedora. > > > > > Thanks a lot. > > > Jason > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From cstomi.levlist at gmail.com Tue Feb 3 05:51:29 2009 From: cstomi.levlist at gmail.com (Tamas Cseke) Date: Tue, 03 Feb 2009 14:51:29 +0100 Subject: [Freeswitch-users] fifo problem Message-ID: <49884BE1.2000603@gmail.com> Hello, We have a problem with mod_fifo. we monitor fifo push event on event socket, call consumer with originate & fifo out nowait Similar like fifo_outbound works, but we have an external strategy for consumer selection (eg.: skill-based routing) The problem is when a caller stops waiting in fifo, the originated calls kepp ringing the consumer, and when the consumer answer the call, he or she may grab somebody else from the fifo, which is a problem because the callers are identified and some data (eg name, phonenumber is shown for the consumer). so it can happen these data will be wrong. We tried to resolve this issue by a call tracking in the external script using event socket. we pushes a variable into the CHANNEL_ORIGINATE event calling the consumer containing the caller uuid. and if the caller aborts the fifo, we hangup the consumer call with (uuid_kill) But it's not prefect becasue it can happen that the consumer pop another caller from the fifo. and we hangup this call, so as a side-effect we loosing another caller. Could anybody advise a solution for this please? we thinking about to have a fifo_caller_uuid variable, that we set before calling fifo with the out method. and if this uuid is in the top of the fifo then pop it else don't pop anybody. it seems to be a hack anyway.... Thanks in advance, Tamas From anthony.minessale at gmail.com Tue Feb 3 05:54:29 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 3 Feb 2009 07:54:29 -0600 Subject: [Freeswitch-users] Conference dialing and uuid In-Reply-To: <20090203082530.GA17166@cpdata.co.za> References: <20090130103944.GA23536@cpdata.co.za> <20090130133315.GB23536@cpdata.co.za> <191c3a030901300605r78dd8943w7fb1075ef851e1d@mail.gmail.com> <20090202085157.GA3555@cpdata.co.za> <0F863750-5C77-4CD9-ADC6-CFDDF2B149B4@freeswitch.org> <20090202102459.GA4179@cpdata.co.za> <9FE1733C-A46A-4A17-8AEE-077DE8010235@freeswitch.org> <191c3a030902020601v1109b1e9he867e9f1e1d4a401@mail.gmail.com> <20090203072221.GD16105@cpdata.co.za> <20090203082530.GA17166@cpdata.co.za> Message-ID: <191c3a030902030554y13e3dd7aj2eeaa43b86b1ced4@mail.gmail.com> There is a file format called tone_stream that I was trying to explain yesterday. tone_stream:// or tone_stream://path=/path/to/text_file.ttml you can use this to play tones anywhere a filename is supposed to go. I guess loopback really is your only option if you must generate ringback. Typically, whatever gateway you are calling out over will go into early media and start playing the real ringback. You should not execute any apps during the on_ring_execute that block, (playing audio etc) Media has not even been established at that point and you have nobody to play the audio to anyway, But you will block from that point until the application you chose has ended so you should only execute small apps that return immediately such as setting a variable etc. As for ringback I think you have the whole thing reversed in your head. the ringback vars etc only apply to the origination (a) leg of a call. If you make an inbound call set the ringback variable and then call bridge, the ringback var is parsed on that inbound leg and the dialout process of the bridge app involves 2 channels the A leg and the B leg. When the B leg gets a ring indication and the A leg detects it, it will begin to play the ringback sound you chose back to the originator of that inbound leg. In the conference or using originate situation, you are doing an outbound call with no relevant inbound call, so there is nothing to generate ringback to. That's why loopback works because it cross connects an outbound call back to an inbound call which gives the bridge app everything it needs to be able to generate artificial ringback. On Tue, Feb 3, 2009 at 2:25 AM, Sias Mey wrote: > Hmmm no MOH wont work... since I am planning on pulling more than just 2 > members into the conference and I still need ringback for the later > members as well. > > Is there a direct way for me to use conference play > to play teletone directly? or should I just records some ringing if I > want to use that? > > And lastly for my own sanity ;-) why would the following in a > on_ring_execute stop execution of the call at that point? > > call = argv[1]; > conf = argv[2]; > > consoleLog("info","Making ringback channel for uuid : "+ session.uuid > +"\n"); > var ringuuid = apiExecute("originate","loopback/ringback-conf="+ conf > +"-conf &park()") > > //I tried with and without a exit() at the end > > It seems to stop media detection??(not really sure about the term) for the > call that executes this > script. > > Freeswitch doesent recognize the pickup of that call and thus it doesent > get bridged into the conference. when I uuid_kill the call that gets > originated everything else starts happening again. > > Oh Im running this in FS ver. 1.0.trunk (11226:11561M) > > and that loopback points to > > > > > > > and ringback.js is > > use("TeleTone"); > session.answer(); > var tts = new TeleTone(session); > > tts.addTone("u", 400.0, 450.0, 0.0); > tts.addTone("r", 440.0, 480.0, 0.0); > > var RESET = "v=2000;>=0;+=0;"; > var UK_RING = RESET + "L=2;u(400,200);u(400,2200)"; > var US_RING = RESET + "r(2000,4000)"; > > while(session.ready()) { > console_log("making UK ring\n"); > for (x = 0 ; x < 2 ; x++) { > tts.generate(UK_RING); > } > } > > A slight bastardisation of the teletone JS example. > > I would expected the new channel that is created via a api originate to > be completely seperate from the JS I create it in. (thats why I use api > instead of creating a new session, although I should probably try that > as well). > > I use some CoreDB stuff to keep tabs on the uuid for the originated call > so that I can uuid_kill it in the on_answer_script but as mentioned... > the on_answer only executes after I uuid_kill the originated channel in > the cli... > > Thanks again guys, > Specially since it seems you two are always the ones that get back to > me. > > On Tue, Feb 03, 2009 at 09:22:21AM +0200, Sias Mey wrote: > > Actually loopback does work. > > however as I said it generates a pair of extra channels. > > > > Hmmm I was trying to generate and extra call to a JS script that > > generated a teletone ring in an on_ring_execute for the second call > > however it seems to stop execution of the call itself. Event though I > > use api commands to originate and then transfer it into the conference > > so that I have direct access to its uuid. > > > > I think changeing the moh might be a bit simpler however and elimite > > some CoreDB stuff I was doing to keep track of the calls ring generating > > call (what a sentance). > > > > On Mon, Feb 02, 2009 at 08:01:25AM -0600, Anthony Minessale wrote: > > > you could set the conference moh sound to be tone_stream::// with > the > > > teletone spec for ring sound and it use ignore_early_media=true in > your > > > originates so the first caller would hear ringback until the 2nd one > > > arrived. > > > > > > On Mon, Feb 2, 2009 at 4:29 AM, Brian West <[1]brian at freeswitch.org > > > > > wrote: > > > > > > Loopback will not work in that case either. If the far end plays > > > ringback inband you should hear that if you use the conference > dial > > > api call. > > > /b > > > > > > On Feb 2, 2009, at 4:24 AM, Sias Mey wrote: > > > > Aaah ok. > > > > > > > > Thanks for clearing that up. > > > > > > > > So using loopback is still the only real workable sollution for > me, > > > > since that generates ringback from and alternative endpoint and > > > > plays it > > > > into the conference. > > > > > > > > I might play with some javascript that streams ring into the > channel > > > > eventually but for now the string comparisons at least get me the > > > > right > > > > uuid. > > > > > > > > Thank you again, > > > > Sias > > > > > > _______________________________________________ > > > Freeswitch-users mailing list > > > [2]Freeswitch-users at lists.freeswitch.org > > > [3]http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:[4] > http://lists.freeswitch.org/mailman/options/freeswitch-u > > > sers > > > [5]http://www.freeswitch.org > > > > > > -- > > > Anthony Minessale II > > > FreeSWITCH [6]http://www.freeswitch.org/ > > > ClueCon [7]http://www.cluecon.com/ > > > AIM: anthm > > > [8]MSN:anthony_minessale at hotmail.com > > > GTALK/JABBER/[9]PAYPAL:anthony.minessale at gmail.com > > > IRC: [10]irc.freenode.net #freeswitch > > > FreeSWITCH Developer Conference > > > [11]sip:888 at conference.freeswitch.org > > > [12]iax:guest at conference.freeswitch.org/888 > > > [13]googletalk:conf+888 at conference.freeswitch.org > > > pstn:213-799-1400 > > > > > > References > > > > > > 1. mailto:brian at freeswitch.org > > > 2. mailto:Freeswitch-users at lists.freeswitch.org > > > 3. http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > 4. http://lists.freeswitch.org/mailman/options/freeswitch-users > > > 5. http://www.freeswitch.org/ > > > 6. http://www.freeswitch.org/ > > > 7. http://www.cluecon.com/ > > > 8. mailto:MSN%3Aanthony_minessale at hotmail.com > > > 9. mailto:PAYPAL%3Aanthony.minessale at gmail.com > > > 10. http://irc.freenode.net/ > > > 11. mailto:sip%3A888 at conference.freeswitch.org > > > 12. http://iax:guest at conference.freeswitch.org/888 > > > 13. mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org > > > > > _______________________________________________ > > > Freeswitch-users mailing list > > > Freeswitch-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090203/0e6d61e3/attachment-0002.html From anthony.minessale at gmail.com Tue Feb 3 06:09:32 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 3 Feb 2009 08:09:32 -0600 Subject: [Freeswitch-users] Generating calls from external source In-Reply-To: <1233666748.24619.8.camel@stargate> References: <40B58CBF-9D0F-43A3-B36F-BD05916756E4@3c.co.uk> <1233666748.24619.8.camel@stargate> Message-ID: <191c3a030902030609i5c3f415fhed501aa492d29cd3@mail.gmail.com> There is also an event socket library written in C called esl that is in the fs tree in the libs directory. This has the ability to establish connections both inbound and outbound from FS. There is also a perl module FreeSWITCH::Client that mr collins may be interested in in the tree as well. On Tue, Feb 3, 2009 at 7:12 AM, Raul Fragoso wrote: > In addition do David's suggestion, you probably want to have your > application to watch for some specific events after the call is > originated and take action based on them. For example, you could watch > for the CHANNEL_ANSWER event and play some audio file waiting for some > digit, which is generated by the DTMF event. > To watch only for those specific events, you should do the following > just after authentication (still using Perl as an example, but the > mod_event_socket is language agnostic), then you will receive those > events from FreeSWITCH through the socket stream: > > ... > print $sock "auth XXX\n\n"; > print $sock "event plain CHANNEL_ANSWER DTMF\n\n"; > ... > > To see a list of available events, please look at the following wiki > pages: > http://wiki.freeswitch.org/wiki/Mod_event_socket#event > http://wiki.freeswitch.org/wiki/Event_list > > Regards, > > Raul > > On Tue, 2009-02-03 at 09:46 +0000, David Knell wrote: > > Hi Nik, > > > > > > Here's a snipped in Perl that launches an outbound call: > > > > > > if (my $sock = IO::Socket::INET->new(Proto =>'tcp', PeerAddr => > > '127.0.0.1', PeerPort => 8021)) { > > print $sock "auth XXX\n\n"; > > print $sock "api originate {softivr_id=$siid,src_softivr_id= > > $siid,softivr_outdial=true}sofia/frombt/$ntd at 1.2.3.4 $service\n\n"; > > $sock->close(); > > } > > > > > > - it does no error checking or anything, but (line by line) it: > > - opens a socket to the event socket interface > > - authenticates > > - issues an originate which dials out to the number in $ntd. The > > bits in {} set a bunch of variables on the channel, which are used by > > the software which processes the call later on. The call is linked to > > the extension in $service - FS looks this up in the dialplan - which > > handles our end. > > - closes the socket > > > > > > Cheers -- > > > > > > Dave > > > > > > > > > Thanks for that, coming from a C++ background it's a refreshing > > > change to be looking at something that seems logical and efficient. > > > > > > I'd briefly looked at the event socket and wondered if that was the > > > way to go. I presume that there's some sort of event generation > > > that can trigger and external process as well somewhere, though all > > > I need to do is update mysql (hopefully using some sort of pooled > > > connection) > > > > > > I'm not using a TDM card, I have a direct interconnect with the PSTN > > > breakout provider with 1,500 channels available to me. I'm finding > > > Asterisk proving to be less than stable at high call volumes and > > > load values spike at more than 100 calls with billing/accounting in > > > place, hence my interest in FS. The only thing that's concerning me > > > is XML at the moment. Lots of code and very wordy. I'm sure I'll > > > appreciate why XML given time > > > > > > Regards, > > > > > > > > > ____________________________________________________________________ > > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael S > Collins > > > Sent: 03 February 2009 01:17 > > > To: freeswitch-users at lists.freeswitch.org > > > Subject: Re: [Freeswitch-users] Generating calls from external > > > source > > > > > > Nik, > > > > > > Welcome to FreeSWITCH! The short answer is "yes, FS can do that." > > > The first thing that you should do is unlearn "the Asterisk way" of > > > thinking. Usually there is an elegant way of doing things in FS that > > > wasn't possible in Ast. > > > > > > I would recommend that you start by looking at the event socket, > > > which is somewhat analogous to the AMI only cooler. :) I have > > > personally done something similar to this using the event socket and > > > a Perl script. The key is to learn the syntax of the originate > > > command. (definitely hit the wiki and IRC channel) > > > Are you using TDM cards for this? Just curious. > > > > > > -MC (IRC nick: mercutioviz) > > > > > > Sent from my iPhone > > > > > > On Feb 2, 2009, at 3:35 PM, "Nik Middleton" > > > wrote: > > > > Hi Guys, > > > > > > > > As a long time Asterisk user, I'm looking into freeswitch as an > > > > alternative mainly due to (list multiple reasons here) > > > > > > > > Can anyone give me a pointer as to how I would achieve the > > > > following? > > > > > > > > I need to replicate an emergency broadcast system currently > > > > running under Asterisk. > > > > > > > > At the moment, I run through a Mysql database and using the > > > > manager API, issues an Originate command to dial a number. > > > > > > > > When the call is answered, a message is played, and the recipient > > > > has the option of hitting a digit to confirm receipt. I then call > > > > an AGI script to update the database. > > > > > > > > Is this fairly easy to do in Freeswitch? > > > > > > > > Not looking for code, just some pointers as to what's available to > > > > do the above / > > > > > > > > Regards, > > > > _______________________________________________ > > > > Freeswitch-users mailing list > > > > Freeswitch-users at lists.freeswitch.org > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > http://www.freeswitch.org > > > _______________________________________________ > > > Freeswitch-users mailing list > > > Freeswitch-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090203/586ea54a/attachment-0002.html From sias at cpdata.co.za Tue Feb 3 06:16:21 2009 From: sias at cpdata.co.za (Sias Mey) Date: Tue, 3 Feb 2009 16:16:21 +0200 Subject: [Freeswitch-users] Conference dialing and uuid In-Reply-To: <191c3a030902030554y13e3dd7aj2eeaa43b86b1ced4@mail.gmail.com> References: <20090130133315.GB23536@cpdata.co.za> <191c3a030901300605r78dd8943w7fb1075ef851e1d@mail.gmail.com> <20090202085157.GA3555@cpdata.co.za> <0F863750-5C77-4CD9-ADC6-CFDDF2B149B4@freeswitch.org> <20090202102459.GA4179@cpdata.co.za> <9FE1733C-A46A-4A17-8AEE-077DE8010235@freeswitch.org> <191c3a030902020601v1109b1e9he867e9f1e1d4a401@mail.gmail.com> <20090203072221.GD16105@cpdata.co.za> <20090203082530.GA17166@cpdata.co.za> <191c3a030902030554y13e3dd7aj2eeaa43b86b1ced4@mail.gmail.com> Message-ID: <20090203141621.GA8916@cpdata.co.za> Hmm ok ... Ill try that In my head though the api call to originate shouldent block? but I assume since it does my head is wrong. Thanks you for the explanation. I think you can put this one to bed now :-P On Tue, Feb 03, 2009 at 07:54:29AM -0600, Anthony Minessale wrote: > There is a file format called tone_stream that I was trying to explain > yesterday. > tone_stream:// > or > tone_stream://path=/path/to/text_file.ttml > you can use this to play tones anywhere a filename is supposed to go. > I guess loopback really is your only option if you must generate > ringback. > Typically, whatever gateway you are calling out over will go into early > media and start playing the real ringback. > You should not execute any apps during the on_ring_execute that block, > (playing audio etc) > Media has not even been established at that point and you have nobody > to play the audio to anyway, > But you will block from that point until the application you chose has > ended so you should only execute small apps that > return immediately such as setting a variable etc. > As for ringback I think you have the whole thing reversed in your > head. > the ringback vars etc only apply to the origination (a) leg of a call. > If you make an inbound call set the ringback variable and then call > bridge, the ringback var is parsed on that inbound leg > and the dialout process of the bridge app involves 2 channels the A leg > and the B leg. When the B leg gets a ring indication and the A leg > detects it, it will begin to play the ringback sound you chose back to > the originator of that inbound leg. > In the conference or using originate situation, you are doing an > outbound call with no relevant inbound call, so there is nothing > to generate ringback to. That's why loopback works because it cross > connects an outbound call back to an inbound call which gives the > bridge app everything it needs to be able to generate artificial > ringback. > > On Tue, Feb 3, 2009 at 2:25 AM, Sias Mey <[1]sias at cpdata.co.za> wrote: > > Hmmm no MOH wont work... since I am planning on pulling more than > just 2 > members into the conference and I still need ringback for the later > members as well. > Is there a direct way for me to use conference play > > to play teletone directly? or should I just records some ringing if > I > want to use that? > And lastly for my own sanity ;-) why would the following in a > on_ring_execute stop execution of the call at that point? > call = argv[1]; > conf = argv[2]; > consoleLog("info","Making ringback channel for uuid : "+ > session.uuid > +"\n"); > var ringuuid = apiExecute("originate","loopback/ringback-conf="+ > conf +"-conf &park()") > //I tried with and without a exit() at the end > It seems to stop media detection??(not really sure about the term) > for the call that executes this > script. > Freeswitch doesent recognize the pickup of that call and thus it > doesent > get bridged into the conference. when I uuid_kill the call that gets > originated everything else starts happening again. > Oh Im running this in FS ver. 1.0.trunk (11226:11561M) > and that loopback points to > > expression="^ringback-conf=(.*)$"> > > > > and ringback.js is > use("TeleTone"); > session.answer(); > var tts = new TeleTone(session); > tts.addTone("u", 400.0, 450.0, 0.0); > tts.addTone("r", 440.0, 480.0, 0.0); > var RESET = "v=2000;>=0;+=0;"; > var UK_RING = RESET + "L=2;u(400,200);u(400,2200)"; > var US_RING = RESET + "r(2000,4000)"; > while(session.ready()) { > console_log("making UK ring\n"); > for (x = 0 ; x < 2 ; x++) { > tts.generate(UK_RING); > } > } > A slight bastardisation of the teletone JS example. > I would expected the new channel that is created via a api originate > to > be completely seperate from the JS I create it in. (thats why I use > api > instead of creating a new session, although I should probably try > that > as well). > I use some CoreDB stuff to keep tabs on the uuid for the originated > call > so that I can uuid_kill it in the on_answer_script but as > mentioned... > the on_answer only executes after I uuid_kill the originated channel > in > the cli... > Thanks again guys, > Specially since it seems you two are always the ones that get back > to > me. > > On Tue, Feb 03, 2009 at 09:22:21AM +0200, Sias Mey wrote: > > Actually loopback does work. > > however as I said it generates a pair of extra channels. > > > > Hmmm I was trying to generate and extra call to a JS script that > > generated a teletone ring in an on_ring_execute for the second call > > however it seems to stop execution of the call itself. Event though I > > use api commands to originate and then transfer it into the > conference > > so that I have direct access to its uuid. > > > > I think changeing the moh might be a bit simpler however and elimite > > some CoreDB stuff I was doing to keep track of the calls ring > generating > > call (what a sentance). > > > > On Mon, Feb 02, 2009 at 08:01:25AM -0600, Anthony Minessale wrote: > > > you could set the conference moh sound to be tone_stream::// > with the > > > teletone spec for ring sound and it use ignore_early_media=true > in your > > > originates so the first caller would hear ringback until the 2nd > one > > > arrived. > > > > > > On Mon, Feb 2, 2009 at 4:29 AM, Brian West > <[1][2]brian at freeswitch.org> > > > wrote: > > > > > > Loopback will not work in that case either. If the far end > plays > > > ringback inband you should hear that if you use the conference > dial > > > api call. > > > /b > > > > > > On Feb 2, 2009, at 4:24 AM, Sias Mey wrote: > > > > Aaah ok. > > > > > > > > Thanks for clearing that up. > > > > > > > > So using loopback is still the only real workable sollution > for me, > > > > since that generates ringback from and alternative endpoint > and > > > > plays it > > > > into the conference. > > > > > > > > I might play with some javascript that streams ring into the > channel > > > > eventually but for now the string comparisons at least get me > the > > > > right > > > > uuid. > > > > > > > > Thank you again, > > > > Sias > > > > > > _______________________________________________ > > > Freeswitch-users mailing list > > > [2][3]Freeswitch-users at lists.freeswitch.org > > > > [3][4]http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > UNSUBSCRIBE:[4][5]http://lists.freeswitch.org/mailman/options/freeswitc > h-u > > > sers > > > [5][6]http://www.freeswitch.org > > > > > > -- > > > Anthony Minessale II > > > FreeSWITCH [6][7]http://www.freeswitch.org/ > > > ClueCon [7][8]http://www.cluecon.com/ > > > AIM: anthm > > > [8][9]MSN:anthony_minessale at hotmail.com > > > GTALK/JABBER/[9][10]PAYPAL:anthony.minessale at gmail.com > > > IRC: [10][11]irc.freenode.net #freeswitch > > > FreeSWITCH Developer Conference > > > [11][12]sip:888 at conference.freeswitch.org > > > [12][13]iax:guest at conference.freeswitch.org/888 > > > [13][14]googletalk:conf+888 at conference.freeswitch.org > > > pstn:213-799-1400 > > > > > > References > > > > > > 1. mailto:[15]brian at freeswitch.org > > > 2. mailto:[16]Freeswitch-users at lists.freeswitch.org > > > 3. > [17]http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > 4. > [18]http://lists.freeswitch.org/mailman/options/freeswitch-users > > > 5. [19]http://www.freeswitch.org/ > > > 6. [20]http://www.freeswitch.org/ > > > 7. [21]http://www.cluecon.com/ > > > 8. mailto:[22]MSN%3Aanthony_minessale at hotmail.com > > > 9. mailto:[23]PAYPAL%3Aanthony.minessale at gmail.com > > > 10. [24]http://irc.freenode.net/ > > > 11. mailto:[25]sip%3A888 at conference.freeswitch.org > > > 12. [26]http://iax:guest at conference.freeswitch.org/888 > > > 13. mailto:[27]googletalk%3Aconf%2B888 at conference.freeswitch.org > > > > > _______________________________________________ > > > Freeswitch-users mailing list > > > [28]Freeswitch-users at lists.freeswitch.org > > > [29]http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > UNSUBSCRIBE:[30]http://lists.freeswitch.org/mailman/options/freeswitch- > users > > > [31]http://www.freeswitch.org > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > [32]Freeswitch-users at lists.freeswitch.org > > [33]http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:[34]http://lists.freeswitch.org/mailman/options/freeswitch- > users > > [35]http://www.freeswitch.org > _______________________________________________ > Freeswitch-users mailing list > [36]Freeswitch-users at lists.freeswitch.org > [37]http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:[38]http://lists.freeswitch.org/mailman/options/freeswitch- > users > [39]http://www.freeswitch.org > > -- > Anthony Minessale II > FreeSWITCH [40]http://www.freeswitch.org/ > ClueCon [41]http://www.cluecon.com/ > AIM: anthm > [42]MSN:anthony_minessale at hotmail.com > GTALK/JABBER/[43]PAYPAL:anthony.minessale at gmail.com > IRC: [44]irc.freenode.net #freeswitch > FreeSWITCH Developer Conference > [45]sip:888 at conference.freeswitch.org > [46]iax:guest at conference.freeswitch.org/888 > [47]googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > References > > 1. mailto:sias at cpdata.co.za > 2. mailto:brian at freeswitch.org > 3. mailto:Freeswitch-users at lists.freeswitch.org > 4. http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > 5. http://lists.freeswitch.org/mailman/options/freeswitch-u > 6. http://www.freeswitch.org/ > 7. http://www.freeswitch.org/ > 8. http://www.cluecon.com/ > 9. mailto:MSN%3Aanthony_minessale at hotmail.com > 10. mailto:PAYPAL%3Aanthony.minessale at gmail.com > 11. http://irc.freenode.net/ > 12. mailto:sip%3A888 at conference.freeswitch.org > 13. http://iax:guest at conference.freeswitch.org/888 > 14. mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org > 15. mailto:brian at freeswitch.org > 16. mailto:Freeswitch-users at lists.freeswitch.org > 17. http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > 18. http://lists.freeswitch.org/mailman/options/freeswitch-users > 19. http://www.freeswitch.org/ > 20. http://www.freeswitch.org/ > 21. http://www.cluecon.com/ > 22. mailto:MSN%253Aanthony_minessale at hotmail.com > 23. mailto:PAYPAL%253Aanthony.minessale at gmail.com > 24. http://irc.freenode.net/ > 25. mailto:sip%253A888 at conference.freeswitch.org > 26. http://iax:guest at conference.freeswitch.org/888 > 27. mailto:googletalk%253Aconf%252B888 at conference.freeswitch.org > 28. mailto:Freeswitch-users at lists.freeswitch.org > 29. http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > 30. http://lists.freeswitch.org/mailman/options/freeswitch-users > 31. http://www.freeswitch.org/ > 32. mailto:Freeswitch-users at lists.freeswitch.org > 33. http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > 34. http://lists.freeswitch.org/mailman/options/freeswitch-users > 35. http://www.freeswitch.org/ > 36. mailto:Freeswitch-users at lists.freeswitch.org > 37. http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > 38. http://lists.freeswitch.org/mailman/options/freeswitch-users > 39. http://www.freeswitch.org/ > 40. http://www.freeswitch.org/ > 41. http://www.cluecon.com/ > 42. mailto:MSN%3Aanthony_minessale at hotmail.com > 43. mailto:PAYPAL%3Aanthony.minessale at gmail.com > 44. http://irc.freenode.net/ > 45. mailto:sip%3A888 at conference.freeswitch.org > 46. http://iax:guest at conference.freeswitch.org/888 > 47. mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From sias at cpdata.co.za Tue Feb 3 07:11:22 2009 From: sias at cpdata.co.za (Sias Mey) Date: Tue, 3 Feb 2009 17:11:22 +0200 Subject: [Freeswitch-users] Conference dialing and uuid In-Reply-To: <20090203141621.GA8916@cpdata.co.za> References: <191c3a030901300605r78dd8943w7fb1075ef851e1d@mail.gmail.com> <20090202085157.GA3555@cpdata.co.za> <0F863750-5C77-4CD9-ADC6-CFDDF2B149B4@freeswitch.org> <20090202102459.GA4179@cpdata.co.za> <9FE1733C-A46A-4A17-8AEE-077DE8010235@freeswitch.org> <191c3a030902020601v1109b1e9he867e9f1e1d4a401@mail.gmail.com> <20090203072221.GD16105@cpdata.co.za> <20090203082530.GA17166@cpdata.co.za> <191c3a030902030554y13e3dd7aj2eeaa43b86b1ced4@mail.gmail.com> <20090203141621.GA8916@cpdata.co.za> Message-ID: <20090203151122.GB8916@cpdata.co.za> hmmm ok indeed. small mods to js files to just play a loooong tone_stream full of ringy noises and then stop them in the on answer and I have what I wanted. Thank you very very much for all your help. On Tue, Feb 03, 2009 at 04:16:21PM +0200, Sias Mey wrote: > Hmm ok ... Ill try that In my head though the api call to originate > shouldent block? but I assume since it does my head is wrong. > > Thanks you for the explanation. I think you can put this one to bed now > :-P > > On Tue, Feb 03, 2009 at 07:54:29AM -0600, Anthony Minessale wrote: > > There is a file format called tone_stream that I was trying to explain > > yesterday. > > tone_stream:// > > or > > tone_stream://path=/path/to/text_file.ttml > > you can use this to play tones anywhere a filename is supposed to go. > > I guess loopback really is your only option if you must generate > > ringback. > > Typically, whatever gateway you are calling out over will go into early > > media and start playing the real ringback. > > You should not execute any apps during the on_ring_execute that block, > > (playing audio etc) > > Media has not even been established at that point and you have nobody > > to play the audio to anyway, > > But you will block from that point until the application you chose has > > ended so you should only execute small apps that > > return immediately such as setting a variable etc. > > As for ringback I think you have the whole thing reversed in your > > head. > > the ringback vars etc only apply to the origination (a) leg of a call. > > If you make an inbound call set the ringback variable and then call > > bridge, the ringback var is parsed on that inbound leg > > and the dialout process of the bridge app involves 2 channels the A leg > > and the B leg. When the B leg gets a ring indication and the A leg > > detects it, it will begin to play the ringback sound you chose back to > > the originator of that inbound leg. > > In the conference or using originate situation, you are doing an > > outbound call with no relevant inbound call, so there is nothing > > to generate ringback to. That's why loopback works because it cross > > connects an outbound call back to an inbound call which gives the > > bridge app everything it needs to be able to generate artificial > > ringback. > > > > On Tue, Feb 3, 2009 at 2:25 AM, Sias Mey <[1]sias at cpdata.co.za> wrote: > > > > Hmmm no MOH wont work... since I am planning on pulling more than > > just 2 > > members into the conference and I still need ringback for the later > > members as well. > > Is there a direct way for me to use conference play > > > > to play teletone directly? or should I just records some ringing if > > I > > want to use that? > > And lastly for my own sanity ;-) why would the following in a > > on_ring_execute stop execution of the call at that point? > > call = argv[1]; > > conf = argv[2]; > > consoleLog("info","Making ringback channel for uuid : "+ > > session.uuid > > +"\n"); > > var ringuuid = apiExecute("originate","loopback/ringback-conf="+ > > conf +"-conf &park()") > > //I tried with and without a exit() at the end > > It seems to stop media detection??(not really sure about the term) > > for the call that executes this > > script. > > Freeswitch doesent recognize the pickup of that call and thus it > > doesent > > get bridged into the conference. when I uuid_kill the call that gets > > originated everything else starts happening again. > > Oh Im running this in FS ver. 1.0.trunk (11226:11561M) > > and that loopback points to > > > > > expression="^ringback-conf=(.*)$"> > > > > > > > > and ringback.js is > > use("TeleTone"); > > session.answer(); > > var tts = new TeleTone(session); > > tts.addTone("u", 400.0, 450.0, 0.0); > > tts.addTone("r", 440.0, 480.0, 0.0); > > var RESET = "v=2000;>=0;+=0;"; > > var UK_RING = RESET + "L=2;u(400,200);u(400,2200)"; > > var US_RING = RESET + "r(2000,4000)"; > > while(session.ready()) { > > console_log("making UK ring\n"); > > for (x = 0 ; x < 2 ; x++) { > > tts.generate(UK_RING); > > } > > } > > A slight bastardisation of the teletone JS example. > > I would expected the new channel that is created via a api originate > > to > > be completely seperate from the JS I create it in. (thats why I use > > api > > instead of creating a new session, although I should probably try > > that > > as well). > > I use some CoreDB stuff to keep tabs on the uuid for the originated > > call > > so that I can uuid_kill it in the on_answer_script but as > > mentioned... > > the on_answer only executes after I uuid_kill the originated channel > > in > > the cli... > > Thanks again guys, > > Specially since it seems you two are always the ones that get back > > to > > me. > > > > On Tue, Feb 03, 2009 at 09:22:21AM +0200, Sias Mey wrote: > > > Actually loopback does work. > > > however as I said it generates a pair of extra channels. > > > > > > Hmmm I was trying to generate and extra call to a JS script that > > > generated a teletone ring in an on_ring_execute for the second call > > > however it seems to stop execution of the call itself. Event though I > > > use api commands to originate and then transfer it into the > > conference > > > so that I have direct access to its uuid. > > > > > > I think changeing the moh might be a bit simpler however and elimite > > > some CoreDB stuff I was doing to keep track of the calls ring > > generating > > > call (what a sentance). > > > > > > On Mon, Feb 02, 2009 at 08:01:25AM -0600, Anthony Minessale wrote: > > > > you could set the conference moh sound to be tone_stream::// > > with the > > > > teletone spec for ring sound and it use ignore_early_media=true > > in your > > > > originates so the first caller would hear ringback until the 2nd > > one > > > > arrived. > > > > > > > > On Mon, Feb 2, 2009 at 4:29 AM, Brian West > > <[1][2]brian at freeswitch.org> > > > > wrote: > > > > > > > > Loopback will not work in that case either. If the far end > > plays > > > > ringback inband you should hear that if you use the conference > > dial > > > > api call. > > > > /b > > > > > > > > On Feb 2, 2009, at 4:24 AM, Sias Mey wrote: > > > > > Aaah ok. > > > > > > > > > > Thanks for clearing that up. > > > > > > > > > > So using loopback is still the only real workable sollution > > for me, > > > > > since that generates ringback from and alternative endpoint > > and > > > > > plays it > > > > > into the conference. > > > > > > > > > > I might play with some javascript that streams ring into the > > channel > > > > > eventually but for now the string comparisons at least get me > > the > > > > > right > > > > > uuid. > > > > > > > > > > Thank you again, > > > > > Sias > > > > > > > > _______________________________________________ > > > > Freeswitch-users mailing list > > > > [2][3]Freeswitch-users at lists.freeswitch.org > > > > > > [3][4]http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > > UNSUBSCRIBE:[4][5]http://lists.freeswitch.org/mailman/options/freeswitc > > h-u > > > > sers > > > > [5][6]http://www.freeswitch.org > > > > > > > > -- > > > > Anthony Minessale II > > > > FreeSWITCH [6][7]http://www.freeswitch.org/ > > > > ClueCon [7][8]http://www.cluecon.com/ > > > > AIM: anthm > > > > [8][9]MSN:anthony_minessale at hotmail.com > > > > GTALK/JABBER/[9][10]PAYPAL:anthony.minessale at gmail.com > > > > IRC: [10][11]irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > > > [11][12]sip:888 at conference.freeswitch.org > > > > [12][13]iax:guest at conference.freeswitch.org/888 > > > > [13][14]googletalk:conf+888 at conference.freeswitch.org > > > > pstn:213-799-1400 > > > > > > > > References > > > > > > > > 1. mailto:[15]brian at freeswitch.org > > > > 2. mailto:[16]Freeswitch-users at lists.freeswitch.org > > > > 3. > > [17]http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > 4. > > [18]http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > 5. [19]http://www.freeswitch.org/ > > > > 6. [20]http://www.freeswitch.org/ > > > > 7. [21]http://www.cluecon.com/ > > > > 8. mailto:[22]MSN%3Aanthony_minessale at hotmail.com > > > > 9. mailto:[23]PAYPAL%3Aanthony.minessale at gmail.com > > > > 10. [24]http://irc.freenode.net/ > > > > 11. mailto:[25]sip%3A888 at conference.freeswitch.org > > > > 12. [26]http://iax:guest at conference.freeswitch.org/888 > > > > 13. mailto:[27]googletalk%3Aconf%2B888 at conference.freeswitch.org > > > > > > > _______________________________________________ > > > > Freeswitch-users mailing list > > > > [28]Freeswitch-users at lists.freeswitch.org > > > > [29]http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > > UNSUBSCRIBE:[30]http://lists.freeswitch.org/mailman/options/freeswitch- > > users > > > > [31]http://www.freeswitch.org > > > > > > > > > _______________________________________________ > > > Freeswitch-users mailing list > > > [32]Freeswitch-users at lists.freeswitch.org > > > [33]http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > UNSUBSCRIBE:[34]http://lists.freeswitch.org/mailman/options/freeswitch- > > users > > > [35]http://www.freeswitch.org > > _______________________________________________ > > Freeswitch-users mailing list > > [36]Freeswitch-users at lists.freeswitch.org > > [37]http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:[38]http://lists.freeswitch.org/mailman/options/freeswitch- > > users > > [39]http://www.freeswitch.org > > > > -- > > Anthony Minessale II > > FreeSWITCH [40]http://www.freeswitch.org/ > > ClueCon [41]http://www.cluecon.com/ > > AIM: anthm > > [42]MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/[43]PAYPAL:anthony.minessale at gmail.com > > IRC: [44]irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > > [45]sip:888 at conference.freeswitch.org > > [46]iax:guest at conference.freeswitch.org/888 > > [47]googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > > > > References > > > > 1. mailto:sias at cpdata.co.za > > 2. mailto:brian at freeswitch.org > > 3. mailto:Freeswitch-users at lists.freeswitch.org > > 4. http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > 5. http://lists.freeswitch.org/mailman/options/freeswitch-u > > 6. http://www.freeswitch.org/ > > 7. http://www.freeswitch.org/ > > 8. http://www.cluecon.com/ > > 9. mailto:MSN%3Aanthony_minessale at hotmail.com > > 10. mailto:PAYPAL%3Aanthony.minessale at gmail.com > > 11. http://irc.freenode.net/ > > 12. mailto:sip%3A888 at conference.freeswitch.org > > 13. http://iax:guest at conference.freeswitch.org/888 > > 14. mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org > > 15. mailto:brian at freeswitch.org > > 16. mailto:Freeswitch-users at lists.freeswitch.org > > 17. http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > 18. http://lists.freeswitch.org/mailman/options/freeswitch-users > > 19. http://www.freeswitch.org/ > > 20. http://www.freeswitch.org/ > > 21. http://www.cluecon.com/ > > 22. mailto:MSN%253Aanthony_minessale at hotmail.com > > 23. mailto:PAYPAL%253Aanthony.minessale at gmail.com > > 24. http://irc.freenode.net/ > > 25. mailto:sip%253A888 at conference.freeswitch.org > > 26. http://iax:guest at conference.freeswitch.org/888 > > 27. mailto:googletalk%253Aconf%252B888 at conference.freeswitch.org > > 28. mailto:Freeswitch-users at lists.freeswitch.org > > 29. http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > 30. http://lists.freeswitch.org/mailman/options/freeswitch-users > > 31. http://www.freeswitch.org/ > > 32. mailto:Freeswitch-users at lists.freeswitch.org > > 33. http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > 34. http://lists.freeswitch.org/mailman/options/freeswitch-users > > 35. http://www.freeswitch.org/ > > 36. mailto:Freeswitch-users at lists.freeswitch.org > > 37. http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > 38. http://lists.freeswitch.org/mailman/options/freeswitch-users > > 39. http://www.freeswitch.org/ > > 40. http://www.freeswitch.org/ > > 41. http://www.cluecon.com/ > > 42. mailto:MSN%3Aanthony_minessale at hotmail.com > > 43. mailto:PAYPAL%3Aanthony.minessale at gmail.com > > 44. http://irc.freenode.net/ > > 45. mailto:sip%3A888 at conference.freeswitch.org > > 46. http://iax:guest at conference.freeswitch.org/888 > > 47. mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From anthony.minessale at gmail.com Tue Feb 3 07:19:37 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 3 Feb 2009 09:19:37 -0600 Subject: [Freeswitch-users] fifo problem In-Reply-To: <49884BE1.2000603@gmail.com> References: <49884BE1.2000603@gmail.com> Message-ID: <191c3a030902030719l2d8eb824rc1472df6d0f8b779@mail.gmail.com> you could use the intercept app to unpark the caller without using fifo out, then it would only work if the caller existed. On Tue, Feb 3, 2009 at 7:51 AM, Tamas Cseke wrote: > Hello, > > We have a problem with mod_fifo. > > we monitor fifo push event on event socket, > call consumer with originate & fifo out nowait > Similar like fifo_outbound works, but we have an external strategy for > consumer selection (eg.: skill-based routing) > > The problem is when a caller stops waiting in fifo, the originated calls > kepp ringing the consumer, and when the consumer answer the call, > he or she may grab somebody else from the fifo, which is a problem > because the callers are identified and some data (eg name, phonenumber > is shown for the consumer). > so it can happen these data will be wrong. > > We tried to resolve this issue by a call tracking in the external script > using event socket. > we pushes a variable into the CHANNEL_ORIGINATE event calling the > consumer containing the caller uuid. > and if the caller aborts the fifo, we hangup the consumer call with > (uuid_kill) > But it's not prefect becasue it can happen that the consumer pop another > caller from the fifo. > and we hangup this call, so as a side-effect we loosing another caller. > > Could anybody advise a solution for this please? > we thinking about to have a fifo_caller_uuid variable, that we set > before calling fifo with the out method. > and if this uuid is in the top of the fifo then pop it else don't pop > anybody. > it seems to be a hack anyway.... > > Thanks in advance, > Tamas > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090203/a6993e45/attachment-0002.html From jsokulski at dotsystems.pl Tue Feb 3 05:36:19 2009 From: jsokulski at dotsystems.pl (Jacek Sokulski) Date: Tue, 03 Feb 2009 14:36:19 +0100 Subject: [Freeswitch-users] origainate through sofia gateway Message-ID: <1233668179.5354.15.camel@dotw1126.dotsystems.pl> Hello I am trying to initiate a call from javascript, it works fine for local numbers: > session1.originate(session1, "{ignore_early_media=true}user/1008 at 192.168.1.122"); but when I am trying to connect through sofia gateway, the connection is not being established: > session2.originate(session2, "sofia/gateway/halonet/0225490317"); although I can call to this number from softphone. I have also tried setting effective_caller_id_number: > session1.originate(session1, "{effective_caller_id_number=fixed0248b}sofia/gateway/halonet/0225490317"); with the same result. A configuration in the dialplan that works is: > > > > > > > > Would appreciate any help. Jacek From nik.middleton at noblesolutions.co.uk Tue Feb 3 08:20:30 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Tue, 3 Feb 2009 16:20:30 -0000 Subject: [Freeswitch-users] OPenser <-> FS Do I need this? In-Reply-To: <1233668029.24619.29.camel@stargate> References: <2cbf225c0902021821u2bc8b622ka5d9f94f05a968ae@mail.gmail.com> <1233668029.24619.29.camel@stargate> Message-ID: Newbie with FS, currently have Asterisk servers front ended by Openser Question: I have around 400 sip remote clients, if I were to deploy FS, do I need Openser? Is there any advantage in retaining Openser? Regards From msc at freeswitch.org Tue Feb 3 08:41:59 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 3 Feb 2009 08:41:59 -0800 Subject: [Freeswitch-users] Application language to support C or C++? In-Reply-To: <1233668029.24619.29.camel@stargate> References: <2cbf225c0902021821u2bc8b622ka5d9f94f05a968ae@mail.gmail.com> <1233668029.24619.29.camel@stargate> Message-ID: <87f2f3b90902030841i54f2064djd13c0fc82fc71ca1@mail.gmail.com> Lee, You also might want to take a look at some of the examples in the contrib folder in the source tree. There are several items there that use the event socket. The event socket is extremely powerful and is suitable for a wide range of applications. However, it isn't the only way to do things. You could also build an actual FreeSWITCH application like the ones found in the "mod" directory. That's a bit more involved and I don't recommend starting there unless you're C/C++ skills are well established. :) What is your application? Most likely others here have done something similar and can share with you their experiences, including what worked and what didn't work. -MC On Tue, Feb 3, 2009 at 5:33 AM, Raul Fragoso wrote: > Depending on what you want to do, I suggest having a look at > mod_event_socket: http://wiki.freeswitch.org/wiki/Mod_event_socket > That module is a socket based interface that provides a vast range of > options to control FreeSWITCH and its applications. > Just for the record, my application is entirely written in C++ and uses > FreeSWITCH as a back-end for providing PBX functionality through a > combination of mod_event_socket and mod_xml_curl. > > Regards, > > Raul > > On Tue, 2009-02-03 at 10:21 +0800, lee jason wrote: >> Hi All, >> >> I saw the applications using FreeSwitch library can be written >> in JavaScript, Perl, Python and Lua but I need to use Linux C or C++ >> for applications, Is FreeSwitch can supported it? Where can I get the >> sample codes? My Linux platform is base on Fedora. >> >> >> >> >> Thanks a lot. >> >> >> Jason >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From msc at freeswitch.org Tue Feb 3 08:52:05 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 3 Feb 2009 08:52:05 -0800 Subject: [Freeswitch-users] Generating calls from external source In-Reply-To: <191c3a030902030609i5c3f415fhed501aa492d29cd3@mail.gmail.com> References: <40B58CBF-9D0F-43A3-B36F-BD05916756E4@3c.co.uk> <1233666748.24619.8.camel@stargate> <191c3a030902030609i5c3f415fhed501aa492d29cd3@mail.gmail.com> Message-ID: <87f2f3b90902030852h6b6477e9u9025ca278ab9122f@mail.gmail.com> On Tue, Feb 3, 2009 at 6:09 AM, Anthony Minessale wrote: > There is also an event socket library written in C called esl that is in the > fs tree in the libs directory. > This has the ability to establish connections both inbound and outbound from > FS. > > There is also a perl module FreeSWITCH::Client that mr collins may be > interested in in the tree as well. As a matter of fact that is the module I used for my outbound IVR application. It simply handled the communications between my perl script and my FS instance. The script would read in pre-formatted originate strings from a text file that had been previously generated by another application. Then all I had to do was specify how many concurrent channels that I wanted - kind of like a throttle - and then I let the script go. I used the "bgapi originate" syntax so that I wouldn't have to wait to see what happened with each origination attempt. Then about every second or so I would issue an "oz dump 1" and parse the results to count how many b channels were in use. If the number of b channels in use was >= my throttle limit then I'd pause the script for 1000ms and then issue the oz dump again until the number of b channels in use had dropped down below my limit. Nothing too fancy. You're welcome to review my script, originate syntax, and dialplan entries if you are interested. -MC > > > On Tue, Feb 3, 2009 at 7:12 AM, Raul Fragoso wrote: >> >> In addition do David's suggestion, you probably want to have your >> application to watch for some specific events after the call is >> originated and take action based on them. For example, you could watch >> for the CHANNEL_ANSWER event and play some audio file waiting for some >> digit, which is generated by the DTMF event. >> To watch only for those specific events, you should do the following >> just after authentication (still using Perl as an example, but the >> mod_event_socket is language agnostic), then you will receive those >> events from FreeSWITCH through the socket stream: >> >> ... >> print $sock "auth XXX\n\n"; >> print $sock "event plain CHANNEL_ANSWER DTMF\n\n"; >> ... >> >> To see a list of available events, please look at the following wiki >> pages: >> http://wiki.freeswitch.org/wiki/Mod_event_socket#event >> http://wiki.freeswitch.org/wiki/Event_list >> >> Regards, >> >> Raul >> >> On Tue, 2009-02-03 at 09:46 +0000, David Knell wrote: >> > Hi Nik, >> > >> > >> > Here's a snipped in Perl that launches an outbound call: >> > >> > >> > if (my $sock = IO::Socket::INET->new(Proto =>'tcp', PeerAddr => >> > '127.0.0.1', PeerPort => 8021)) { >> > print $sock "auth XXX\n\n"; >> > print $sock "api originate {softivr_id=$siid,src_softivr_id= >> > $siid,softivr_outdial=true}sofia/frombt/$ntd at 1.2.3.4 $service\n\n"; >> > $sock->close(); >> > } >> > >> > >> > - it does no error checking or anything, but (line by line) it: >> > - opens a socket to the event socket interface >> > - authenticates >> > - issues an originate which dials out to the number in $ntd. The >> > bits in {} set a bunch of variables on the channel, which are used by >> > the software which processes the call later on. The call is linked to >> > the extension in $service - FS looks this up in the dialplan - which >> > handles our end. >> > - closes the socket >> > >> > >> > Cheers -- >> > >> > >> > Dave >> > >> > >> > >> > > Thanks for that, coming from a C++ background it's a refreshing >> > > change to be looking at something that seems logical and efficient. >> > > >> > > I'd briefly looked at the event socket and wondered if that was the >> > > way to go. I presume that there's some sort of event generation >> > > that can trigger and external process as well somewhere, though all >> > > I need to do is update mysql (hopefully using some sort of pooled >> > > connection) >> > > >> > > I'm not using a TDM card, I have a direct interconnect with the PSTN >> > > breakout provider with 1,500 channels available to me. I'm finding >> > > Asterisk proving to be less than stable at high call volumes and >> > > load values spike at more than 100 calls with billing/accounting in >> > > place, hence my interest in FS. The only thing that's concerning me >> > > is XML at the moment. Lots of code and very wordy. I'm sure I'll >> > > appreciate why XML given time >> > > >> > > Regards, >> > > >> > > >> > > ____________________________________________________________________ >> > > From: freeswitch-users-bounces at lists.freeswitch.org >> > > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael >> > > S Collins >> > > Sent: 03 February 2009 01:17 >> > > To: freeswitch-users at lists.freeswitch.org >> > > Subject: Re: [Freeswitch-users] Generating calls from external >> > > source >> > > >> > > Nik, >> > > >> > > Welcome to FreeSWITCH! The short answer is "yes, FS can do that." >> > > The first thing that you should do is unlearn "the Asterisk way" of >> > > thinking. Usually there is an elegant way of doing things in FS that >> > > wasn't possible in Ast. >> > > >> > > I would recommend that you start by looking at the event socket, >> > > which is somewhat analogous to the AMI only cooler. :) I have >> > > personally done something similar to this using the event socket and >> > > a Perl script. The key is to learn the syntax of the originate >> > > command. (definitely hit the wiki and IRC channel) >> > > Are you using TDM cards for this? Just curious. >> > > >> > > -MC (IRC nick: mercutioviz) >> > > >> > > Sent from my iPhone >> > > >> > > On Feb 2, 2009, at 3:35 PM, "Nik Middleton" >> > > wrote: >> > > > Hi Guys, >> > > > >> > > > As a long time Asterisk user, I'm looking into freeswitch as an >> > > > alternative mainly due to (list multiple reasons here) >> > > > >> > > > Can anyone give me a pointer as to how I would achieve the >> > > > following? >> > > > >> > > > I need to replicate an emergency broadcast system currently >> > > > running under Asterisk. >> > > > >> > > > At the moment, I run through a Mysql database and using the >> > > > manager API, issues an Originate command to dial a number. >> > > > >> > > > When the call is answered, a message is played, and the recipient >> > > > has the option of hitting a digit to confirm receipt. I then call >> > > > an AGI script to update the database. >> > > > >> > > > Is this fairly easy to do in Freeswitch? >> > > > >> > > > Not looking for code, just some pointers as to what's available to >> > > > do the above / >> > > > >> > > > Regards, >> > > > _______________________________________________ >> > > > Freeswitch-users mailing list >> > > > Freeswitch-users at lists.freeswitch.org >> > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > > > >> > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > > > http://www.freeswitch.org >> > > _______________________________________________ >> > > Freeswitch-users mailing list >> > > Freeswitch-users at lists.freeswitch.org >> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > > >> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > > http://www.freeswitch.org >> > > >> > >> > >> > _______________________________________________ >> > Freeswitch-users mailing list >> > Freeswitch-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From nik.middleton at noblesolutions.co.uk Tue Feb 3 08:53:20 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Tue, 3 Feb 2009 16:53:20 -0000 Subject: [Freeswitch-users] Generating calls from external source In-Reply-To: <1233666748.24619.8.camel@stargate> References: <40B58CBF-9D0F-43A3-B36F-BD05916756E4@3c.co.uk> <1233666748.24619.8.camel@stargate> Message-ID: Are you suggesting that I should process the call externally instead of using the dialplan? That would be neat as the audio file select could be driven from the db select for the number. I presume that I could also bridge the call to another number as well dependant on DTMF selection? Regards -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Raul Fragoso Sent: 03 February 2009 13:12 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Generating calls from external source In addition do David's suggestion, you probably want to have your application to watch for some specific events after the call is originated and take action based on them. For example, you could watch for the CHANNEL_ANSWER event and play some audio file waiting for some digit, which is generated by the DTMF event. To watch only for those specific events, you should do the following just after authentication (still using Perl as an example, but the mod_event_socket is language agnostic), then you will receive those events from FreeSWITCH through the socket stream: ... print $sock "auth XXX\n\n"; print $sock "event plain CHANNEL_ANSWER DTMF\n\n"; ... To see a list of available events, please look at the following wiki pages: http://wiki.freeswitch.org/wiki/Mod_event_socket#event http://wiki.freeswitch.org/wiki/Event_list Regards, Raul On Tue, 2009-02-03 at 09:46 +0000, David Knell wrote: > Hi Nik, > > > Here's a snipped in Perl that launches an outbound call: > > > if (my $sock = IO::Socket::INET->new(Proto =>'tcp', PeerAddr => > '127.0.0.1', PeerPort => 8021)) { > print $sock "auth XXX\n\n"; > print $sock "api originate {softivr_id=$siid,src_softivr_id= > $siid,softivr_outdial=true}sofia/frombt/$ntd at 1.2.3.4 $service\n\n"; > $sock->close(); > } > > > - it does no error checking or anything, but (line by line) it: > - opens a socket to the event socket interface > - authenticates > - issues an originate which dials out to the number in $ntd. The > bits in {} set a bunch of variables on the channel, which are used by > the software which processes the call later on. The call is linked to > the extension in $service - FS looks this up in the dialplan - which > handles our end. > - closes the socket > > > Cheers -- > > > Dave > > > > > Thanks for that, coming from a C++ background it's a refreshing > > change to be looking at something that seems logical and efficient. > > > > I'd briefly looked at the event socket and wondered if that was the > > way to go. I presume that there's some sort of event generation > > that can trigger and external process as well somewhere, though all > > I need to do is update mysql (hopefully using some sort of pooled > > connection) > > > > I'm not using a TDM card, I have a direct interconnect with the PSTN > > breakout provider with 1,500 channels available to me. I'm finding > > Asterisk proving to be less than stable at high call volumes and > > load values spike at more than 100 calls with billing/accounting in > > place, hence my interest in FS. The only thing that's concerning me > > is XML at the moment. Lots of code and very wordy. I'm sure I'll > > appreciate why XML given time > > > > Regards, > > > > > > ____________________________________________________________________ > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael S Collins > > Sent: 03 February 2009 01:17 > > To: freeswitch-users at lists.freeswitch.org > > Subject: Re: [Freeswitch-users] Generating calls from external > > source > > > > Nik, > > > > Welcome to FreeSWITCH! The short answer is "yes, FS can do that." > > The first thing that you should do is unlearn "the Asterisk way" of > > thinking. Usually there is an elegant way of doing things in FS that > > wasn't possible in Ast. > > > > I would recommend that you start by looking at the event socket, > > which is somewhat analogous to the AMI only cooler. :) I have > > personally done something similar to this using the event socket and > > a Perl script. The key is to learn the syntax of the originate > > command. (definitely hit the wiki and IRC channel) > > Are you using TDM cards for this? Just curious. > > > > -MC (IRC nick: mercutioviz) > > > > Sent from my iPhone > > > > On Feb 2, 2009, at 3:35 PM, "Nik Middleton" > > wrote: > > > Hi Guys, > > > > > > As a long time Asterisk user, I'm looking into freeswitch as an > > > alternative mainly due to (list multiple reasons here) > > > > > > Can anyone give me a pointer as to how I would achieve the > > > following? > > > > > > I need to replicate an emergency broadcast system currently > > > running under Asterisk. > > > > > > At the moment, I run through a Mysql database and using the > > > manager API, issues an Originate command to dial a number. > > > > > > When the call is answered, a message is played, and the recipient > > > has the option of hitting a digit to confirm receipt. I then call > > > an AGI script to update the database. > > > > > > Is this fairly easy to do in Freeswitch? > > > > > > Not looking for code, just some pointers as to what's available to > > > do the above / > > > > > > Regards, > > > _______________________________________________ > > > Freeswitch-users mailing list > > > Freeswitch-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From mike at jerris.com Tue Feb 3 09:02:38 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 3 Feb 2009 12:02:38 -0500 Subject: [Freeswitch-users] debuild breaks since the last few days In-Reply-To: <498840A3.30509@ewetel.de> References: <0B3BFADA-4589-48AD-9F18-EEBDC6BAB740@scarlet-internet.nl> <498840A3.30509@ewetel.de> Message-ID: <707A7340-2C7A-47E9-9F69-8CC514785D8D@jerris.com> I just finished adding full libpcap detection. Openzap will now build again with or without libpcap, of course the pcap features will not work without. Mike On Feb 3, 2009, at 8:03 AM, Helmut Kuper wrote: > Hello, > > yes, you have openzap upgraded to r632. Then recompile it. Make sure > you > have libpcap installed and pcap devel files > > regards > helmut > > > Am 03.02.2009 12:55, schrieb Leon de Rooij: >> Hi all, >> >> I've been trying to build new debs, but debuild seems to break.. >> From msc at freeswitch.org Tue Feb 3 09:05:57 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 3 Feb 2009 09:05:57 -0800 Subject: [Freeswitch-users] Conference dialing and uuid In-Reply-To: <20090203141621.GA8916@cpdata.co.za> References: <20090130133315.GB23536@cpdata.co.za> <20090202085157.GA3555@cpdata.co.za> <0F863750-5C77-4CD9-ADC6-CFDDF2B149B4@freeswitch.org> <20090202102459.GA4179@cpdata.co.za> <9FE1733C-A46A-4A17-8AEE-077DE8010235@freeswitch.org> <191c3a030902020601v1109b1e9he867e9f1e1d4a401@mail.gmail.com> <20090203072221.GD16105@cpdata.co.za> <20090203082530.GA17166@cpdata.co.za> <191c3a030902030554y13e3dd7aj2eeaa43b86b1ced4@mail.gmail.com> <20090203141621.GA8916@cpdata.co.za> Message-ID: <87f2f3b90902030905y348efa0dq90c0080e5792e63f@mail.gmail.com> On Tue, Feb 3, 2009 at 6:16 AM, Sias Mey wrote: > Hmm ok ... Ill try that In my head though the api call to originate > shouldent block? but I assume since it does my head is wrong. You can use "bgapi originate" to do it in a non-blocking way. See the very last example for the originate command: http://wiki.freeswitch.org/wiki/Mod_commands#originate -MC > > Thanks you for the explanation. I think you can put this one to bed now > :-P > > On Tue, Feb 03, 2009 at 07:54:29AM -0600, Anthony Minessale wrote: >> There is a file format called tone_stream that I was trying to explain >> yesterday. >> tone_stream:// >> or >> tone_stream://path=/path/to/text_file.ttml >> you can use this to play tones anywhere a filename is supposed to go. >> I guess loopback really is your only option if you must generate >> ringback. >> Typically, whatever gateway you are calling out over will go into early >> media and start playing the real ringback. >> You should not execute any apps during the on_ring_execute that block, >> (playing audio etc) >> Media has not even been established at that point and you have nobody >> to play the audio to anyway, >> But you will block from that point until the application you chose has >> ended so you should only execute small apps that >> return immediately such as setting a variable etc. >> As for ringback I think you have the whole thing reversed in your >> head. >> the ringback vars etc only apply to the origination (a) leg of a call. >> If you make an inbound call set the ringback variable and then call >> bridge, the ringback var is parsed on that inbound leg >> and the dialout process of the bridge app involves 2 channels the A leg >> and the B leg. When the B leg gets a ring indication and the A leg >> detects it, it will begin to play the ringback sound you chose back to >> the originator of that inbound leg. >> In the conference or using originate situation, you are doing an >> outbound call with no relevant inbound call, so there is nothing >> to generate ringback to. That's why loopback works because it cross >> connects an outbound call back to an inbound call which gives the >> bridge app everything it needs to be able to generate artificial >> ringback. >> >> On Tue, Feb 3, 2009 at 2:25 AM, Sias Mey <[1]sias at cpdata.co.za> wrote: >> >> Hmmm no MOH wont work... since I am planning on pulling more than >> just 2 >> members into the conference and I still need ringback for the later >> members as well. >> Is there a direct way for me to use conference play >> >> to play teletone directly? or should I just records some ringing if >> I >> want to use that? >> And lastly for my own sanity ;-) why would the following in a >> on_ring_execute stop execution of the call at that point? >> call = argv[1]; >> conf = argv[2]; >> consoleLog("info","Making ringback channel for uuid : "+ >> session.uuid >> +"\n"); >> var ringuuid = apiExecute("originate","loopback/ringback-conf="+ >> conf +"-conf &park()") >> //I tried with and without a exit() at the end >> It seems to stop media detection??(not really sure about the term) >> for the call that executes this >> script. >> Freeswitch doesent recognize the pickup of that call and thus it >> doesent >> get bridged into the conference. when I uuid_kill the call that gets >> originated everything else starts happening again. >> Oh Im running this in FS ver. 1.0.trunk (11226:11561M) >> and that loopback points to >> >> > expression="^ringback-conf=(.*)$"> >> >> >> >> and ringback.js is >> use("TeleTone"); >> session.answer(); >> var tts = new TeleTone(session); >> tts.addTone("u", 400.0, 450.0, 0.0); >> tts.addTone("r", 440.0, 480.0, 0.0); >> var RESET = "v=2000;>=0;+=0;"; >> var UK_RING = RESET + "L=2;u(400,200);u(400,2200)"; >> var US_RING = RESET + "r(2000,4000)"; >> while(session.ready()) { >> console_log("making UK ring\n"); >> for (x = 0 ; x < 2 ; x++) { >> tts.generate(UK_RING); >> } >> } >> A slight bastardisation of the teletone JS example. >> I would expected the new channel that is created via a api originate >> to >> be completely seperate from the JS I create it in. (thats why I use >> api >> instead of creating a new session, although I should probably try >> that >> as well). >> I use some CoreDB stuff to keep tabs on the uuid for the originated >> call >> so that I can uuid_kill it in the on_answer_script but as >> mentioned... >> the on_answer only executes after I uuid_kill the originated channel >> in >> the cli... >> Thanks again guys, >> Specially since it seems you two are always the ones that get back >> to >> me. >> >> On Tue, Feb 03, 2009 at 09:22:21AM +0200, Sias Mey wrote: >> > Actually loopback does work. >> > however as I said it generates a pair of extra channels. >> > >> > Hmmm I was trying to generate and extra call to a JS script that >> > generated a teletone ring in an on_ring_execute for the second call >> > however it seems to stop execution of the call itself. Event though I >> > use api commands to originate and then transfer it into the >> conference >> > so that I have direct access to its uuid. >> > >> > I think changeing the moh might be a bit simpler however and elimite >> > some CoreDB stuff I was doing to keep track of the calls ring >> generating >> > call (what a sentance). >> > >> > On Mon, Feb 02, 2009 at 08:01:25AM -0600, Anthony Minessale wrote: >> > > you could set the conference moh sound to be tone_stream::// >> with the >> > > teletone spec for ring sound and it use ignore_early_media=true >> in your >> > > originates so the first caller would hear ringback until the 2nd >> one >> > > arrived. >> > > >> > > On Mon, Feb 2, 2009 at 4:29 AM, Brian West >> <[1][2]brian at freeswitch.org> >> > > wrote: >> > > >> > > Loopback will not work in that case either. If the far end >> plays >> > > ringback inband you should hear that if you use the conference >> dial >> > > api call. >> > > /b >> > > >> > > On Feb 2, 2009, at 4:24 AM, Sias Mey wrote: >> > > > Aaah ok. >> > > > >> > > > Thanks for clearing that up. >> > > > >> > > > So using loopback is still the only real workable sollution >> for me, >> > > > since that generates ringback from and alternative endpoint >> and >> > > > plays it >> > > > into the conference. >> > > > >> > > > I might play with some javascript that streams ring into the >> channel >> > > > eventually but for now the string comparisons at least get me >> the >> > > > right >> > > > uuid. >> > > > >> > > > Thank you again, >> > > > Sias >> > > >> > > _______________________________________________ >> > > Freeswitch-users mailing list >> > > [2][3]Freeswitch-users at lists.freeswitch.org >> > > >> [3][4]http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > > >> UNSUBSCRIBE:[4][5]http://lists.freeswitch.org/mailman/options/freeswitc >> h-u >> > > sers >> > > [5][6]http://www.freeswitch.org >> > > >> > > -- >> > > Anthony Minessale II >> > > FreeSWITCH [6][7]http://www.freeswitch.org/ >> > > ClueCon [7][8]http://www.cluecon.com/ >> > > AIM: anthm >> > > [8][9]MSN:anthony_minessale at hotmail.com >> > > GTALK/JABBER/[9][10]PAYPAL:anthony.minessale at gmail.com >> > > IRC: [10][11]irc.freenode.net #freeswitch >> > > FreeSWITCH Developer Conference >> > > [11][12]sip:888 at conference.freeswitch.org >> > > [12][13]iax:guest at conference.freeswitch.org/888 >> > > [13][14]googletalk:conf+888 at conference.freeswitch.org >> > > pstn:213-799-1400 >> > > >> > > References >> > > >> > > 1. mailto:[15]brian at freeswitch.org >> > > 2. mailto:[16]Freeswitch-users at lists.freeswitch.org >> > > 3. >> [17]http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > > 4. >> [18]http://lists.freeswitch.org/mailman/options/freeswitch-users >> > > 5. [19]http://www.freeswitch.org/ >> > > 6. [20]http://www.freeswitch.org/ >> > > 7. [21]http://www.cluecon.com/ >> > > 8. mailto:[22]MSN%3Aanthony_minessale at hotmail.com >> > > 9. mailto:[23]PAYPAL%3Aanthony.minessale at gmail.com >> > > 10. [24]http://irc.freenode.net/ >> > > 11. mailto:[25]sip%3A888 at conference.freeswitch.org >> > > 12. [26]http://iax:guest at conference.freeswitch.org/888 >> > > 13. mailto:[27]googletalk%3Aconf%2B888 at conference.freeswitch.org >> > >> > > _______________________________________________ >> > > Freeswitch-users mailing list >> > > [28]Freeswitch-users at lists.freeswitch.org >> > > [29]http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > > >> UNSUBSCRIBE:[30]http://lists.freeswitch.org/mailman/options/freeswitch- >> users >> > > [31]http://www.freeswitch.org >> > >> > >> > _______________________________________________ >> > Freeswitch-users mailing list >> > [32]Freeswitch-users at lists.freeswitch.org >> > [33]http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > >> UNSUBSCRIBE:[34]http://lists.freeswitch.org/mailman/options/freeswitch- >> users >> > [35]http://www.freeswitch.org >> _______________________________________________ >> Freeswitch-users mailing list >> [36]Freeswitch-users at lists.freeswitch.org >> [37]http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:[38]http://lists.freeswitch.org/mailman/options/freeswitch- >> users >> [39]http://www.freeswitch.org >> >> -- >> Anthony Minessale II >> FreeSWITCH [40]http://www.freeswitch.org/ >> ClueCon [41]http://www.cluecon.com/ >> AIM: anthm >> [42]MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/[43]PAYPAL:anthony.minessale at gmail.com >> IRC: [44]irc.freenode.net #freeswitch >> FreeSWITCH Developer Conference >> [45]sip:888 at conference.freeswitch.org >> [46]iax:guest at conference.freeswitch.org/888 >> [47]googletalk:conf+888 at conference.freeswitch.org >> pstn:213-799-1400 >> >> References >> >> 1. mailto:sias at cpdata.co.za >> 2. mailto:brian at freeswitch.org >> 3. mailto:Freeswitch-users at lists.freeswitch.org >> 4. http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> 5. http://lists.freeswitch.org/mailman/options/freeswitch-u >> 6. http://www.freeswitch.org/ >> 7. http://www.freeswitch.org/ >> 8. http://www.cluecon.com/ >> 9. mailto:MSN%3Aanthony_minessale at hotmail.com >> 10. mailto:PAYPAL%3Aanthony.minessale at gmail.com >> 11. http://irc.freenode.net/ >> 12. mailto:sip%3A888 at conference.freeswitch.org >> 13. http://iax:guest at conference.freeswitch.org/888 >> 14. mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org >> 15. mailto:brian at freeswitch.org >> 16. mailto:Freeswitch-users at lists.freeswitch.org >> 17. http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> 18. http://lists.freeswitch.org/mailman/options/freeswitch-users >> 19. http://www.freeswitch.org/ >> 20. http://www.freeswitch.org/ >> 21. http://www.cluecon.com/ >> 22. mailto:MSN%253Aanthony_minessale at hotmail.com >> 23. mailto:PAYPAL%253Aanthony.minessale at gmail.com >> 24. http://irc.freenode.net/ >> 25. mailto:sip%253A888 at conference.freeswitch.org >> 26. http://iax:guest at conference.freeswitch.org/888 >> 27. mailto:googletalk%253Aconf%252B888 at conference.freeswitch.org >> 28. mailto:Freeswitch-users at lists.freeswitch.org >> 29. http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> 30. http://lists.freeswitch.org/mailman/options/freeswitch-users >> 31. http://www.freeswitch.org/ >> 32. mailto:Freeswitch-users at lists.freeswitch.org >> 33. http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> 34. http://lists.freeswitch.org/mailman/options/freeswitch-users >> 35. http://www.freeswitch.org/ >> 36. mailto:Freeswitch-users at lists.freeswitch.org >> 37. http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> 38. http://lists.freeswitch.org/mailman/options/freeswitch-users >> 39. http://www.freeswitch.org/ >> 40. http://www.freeswitch.org/ >> 41. http://www.cluecon.com/ >> 42. mailto:MSN%3Aanthony_minessale at hotmail.com >> 43. mailto:PAYPAL%3Aanthony.minessale at gmail.com >> 44. http://irc.freenode.net/ >> 45. mailto:sip%3A888 at conference.freeswitch.org >> 46. http://iax:guest at conference.freeswitch.org/888 >> 47. mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org > >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From msc at freeswitch.org Tue Feb 3 09:08:19 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 3 Feb 2009 09:08:19 -0800 Subject: [Freeswitch-users] OPenser <-> FS Do I need this? In-Reply-To: References: <2cbf225c0902021821u2bc8b622ka5d9f94f05a968ae@mail.gmail.com> <1233668029.24619.29.camel@stargate> Message-ID: <87f2f3b90902030908n629399bdmdd46f633b803b5f6@mail.gmail.com> On Tue, Feb 3, 2009 at 8:20 AM, Nik Middleton wrote: > Newbie with FS, currently have Asterisk servers front ended by Openser > > Question: I have around 400 sip remote clients, if I were to deploy FS, > do I need Openser? Is there any advantage in retaining Openser? If I may ask... why did you have OpenSER with your Asterisk deployment? Reason I ask is because some people do that "because Asterisk sucks" but others have a specific application or reason. What does OpenSER do for your Asterisk install? -MC > > Regards > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From helmut.kuper at ewetel.de Tue Feb 3 09:09:06 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Tue, 03 Feb 2009 18:09:06 +0100 Subject: [Freeswitch-users] mod_voicemail: Limiting the number how often a menu is repeated In-Reply-To: <498734D0.5060004@ewetel.de> References: <498734D0.5060004@ewetel.de> Message-ID: <49887A32.7060804@ewetel.de> Hi, has anybody an idea? regards Helmut Am 02.02.2009 19:00, schrieb Helmut Kuper: > Hello, > > today I searched for a way to limit the number of menu repeatings in > mod_voicemail to let's say 3 times and when it reached the limit > voicemail should abort. But I couldn't find a hint. Any ideas? > > > regards > helmut > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > -- Mit freundlichen Gr??en Helmut Kuper Finanzdienstleistungen und Entwicklung Telefax: (0441) 8000-2799 mailto:helmut.kuper at ewetel.de ___________________________________ EWE TEL GmbH Cloppenburger Stra?e 310 26133 Oldenburg EWE TEL GmbH Handelsregister Amtsgericht Oldenburg HRB 3723 Vorsitzender des Aufsichtsrates: Heiko Harms Gesch?ftsf?hrung: Hans-Joachim Iken (Vorsitzender), Dr. Norbert Schulz, Dirk Thole Homepage: http://www.ewetel.de ___________________________________ From sicfslist at gmail.com Tue Feb 3 09:13:19 2009 From: sicfslist at gmail.com (Shelby Ramsey) Date: Tue, 3 Feb 2009 11:13:19 -0600 Subject: [Freeswitch-users] Generating calls from external source In-Reply-To: References: <40B58CBF-9D0F-43A3-B36F-BD05916756E4@3c.co.uk> <1233666748.24619.8.camel@stargate> Message-ID: <35b355e90902030913s1cec764j46280ac2d9a162a7@mail.gmail.com> Nik, There are a lot of ways to make FS dial out and deliver messaging etc. We are going through the process of replacing * for this purpose. For us (getting started with the help of our friends here on the list) it has been pretty easy. With * we were using AMI to originate calls ... to migrate to FS we just changed that to use event_socket with bgapi to originate the call and connect the call to a context and extension. There are several ways to get the dialplan to FS after that ... a script, xml_curl, or statically configured in the conf directory. So as an example the application we have just logs into the FS socket (similar to * but much better) and then rips off calls like this: bgapi originate{$set_some_vars}sofia/external/$ANI@$IP:$PORT $EXTENSION xml $CONTEXT The beauty of it all is that: -- a lot of flexibility in what you can do (like drive the call through events) -- the CDR reporting is about 3 million times better than * -- obviously higher capacity I'd start playing with event_socket and some static dialplans to get the feel for it ... but if you have an application written already to work with * (i.e. the logic and backend) it will be very easy to migrate and you'll be glad you did it! Shelby On Tue, Feb 3, 2009 at 10:53 AM, Nik Middleton < nik.middleton at noblesolutions.co.uk> wrote: > Are you suggesting that I should process the call externally instead of > using the dialplan? That would be neat as the audio file select could > be driven from the db select for the number. I presume that I could > also bridge the call to another number as well dependant on DTMF > selection? > > Regards > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Raul > Fragoso > Sent: 03 February 2009 13:12 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Generating calls from external source > > In addition do David's suggestion, you probably want to have your > application to watch for some specific events after the call is > originated and take action based on them. For example, you could watch > for the CHANNEL_ANSWER event and play some audio file waiting for some > digit, which is generated by the DTMF event. > To watch only for those specific events, you should do the following > just after authentication (still using Perl as an example, but the > mod_event_socket is language agnostic), then you will receive those > events from FreeSWITCH through the socket stream: > > ... > print $sock "auth XXX\n\n"; > print $sock "event plain CHANNEL_ANSWER DTMF\n\n"; > ... > > To see a list of available events, please look at the following wiki > pages: > http://wiki.freeswitch.org/wiki/Mod_event_socket#event > http://wiki.freeswitch.org/wiki/Event_list > > Regards, > > Raul > > On Tue, 2009-02-03 at 09:46 +0000, David Knell wrote: > > Hi Nik, > > > > > > Here's a snipped in Perl that launches an outbound call: > > > > > > if (my $sock = IO::Socket::INET->new(Proto =>'tcp', PeerAddr => > > '127.0.0.1', PeerPort => 8021)) { > > print $sock "auth XXX\n\n"; > > print $sock "api originate {softivr_id=$siid,src_softivr_id= > > $siid,softivr_outdial=true}sofia/frombt/$ntd at 1.2.3.4 $service\n\n"; > > $sock->close(); > > } > > > > > > - it does no error checking or anything, but (line by line) it: > > - opens a socket to the event socket interface > > - authenticates > > - issues an originate which dials out to the number in $ntd. The > > bits in {} set a bunch of variables on the channel, which are used by > > the software which processes the call later on. The call is linked to > > the extension in $service - FS looks this up in the dialplan - which > > handles our end. > > - closes the socket > > > > > > Cheers -- > > > > > > Dave > > > > > > > > > Thanks for that, coming from a C++ background it's a refreshing > > > change to be looking at something that seems logical and efficient. > > > > > > I'd briefly looked at the event socket and wondered if that was the > > > way to go. I presume that there's some sort of event generation > > > that can trigger and external process as well somewhere, though all > > > I need to do is update mysql (hopefully using some sort of pooled > > > connection) > > > > > > I'm not using a TDM card, I have a direct interconnect with the PSTN > > > breakout provider with 1,500 channels available to me. I'm finding > > > Asterisk proving to be less than stable at high call volumes and > > > load values spike at more than 100 calls with billing/accounting in > > > place, hence my interest in FS. The only thing that's concerning me > > > is XML at the moment. Lots of code and very wordy. I'm sure I'll > > > appreciate why XML given time > > > > > > Regards, > > > > > > > > > ____________________________________________________________________ > > > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Michael S Collins > > > Sent: 03 February 2009 01:17 > > > To: freeswitch-users at lists.freeswitch.org > > > Subject: Re: [Freeswitch-users] Generating calls from external > > > source > > > > > > Nik, > > > > > > Welcome to FreeSWITCH! The short answer is "yes, FS can do that." > > > The first thing that you should do is unlearn "the Asterisk way" of > > > thinking. Usually there is an elegant way of doing things in FS that > > > wasn't possible in Ast. > > > > > > I would recommend that you start by looking at the event socket, > > > which is somewhat analogous to the AMI only cooler. :) I have > > > personally done something similar to this using the event socket and > > > a Perl script. The key is to learn the syntax of the originate > > > command. (definitely hit the wiki and IRC channel) > > > Are you using TDM cards for this? Just curious. > > > > > > -MC (IRC nick: mercutioviz) > > > > > > Sent from my iPhone > > > > > > On Feb 2, 2009, at 3:35 PM, "Nik Middleton" > > > wrote: > > > > Hi Guys, > > > > > > > > As a long time Asterisk user, I'm looking into freeswitch as an > > > > alternative mainly due to (list multiple reasons here) > > > > > > > > Can anyone give me a pointer as to how I would achieve the > > > > following? > > > > > > > > I need to replicate an emergency broadcast system currently > > > > running under Asterisk. > > > > > > > > At the moment, I run through a Mysql database and using the > > > > manager API, issues an Originate command to dial a number. > > > > > > > > When the call is answered, a message is played, and the recipient > > > > has the option of hitting a digit to confirm receipt. I then call > > > > an AGI script to update the database. > > > > > > > > Is this fairly easy to do in Freeswitch? > > > > > > > > Not looking for code, just some pointers as to what's available to > > > > do the above / > > > > > > > > Regards, > > > > _______________________________________________ > > > > Freeswitch-users mailing list > > > > Freeswitch-users at lists.freeswitch.org > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > http://www.freeswitch.org > > > _______________________________________________ > > > Freeswitch-users mailing list > > > Freeswitch-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090203/2dc96b6e/attachment-0002.html From e.schmidbauer at gmail.com Tue Feb 3 09:16:59 2009 From: e.schmidbauer at gmail.com (e schmidbauer) Date: Tue, 3 Feb 2009 12:16:59 -0500 Subject: [Freeswitch-users] shoutcast skips Message-ID: <2cef777b0902030916s77339375pabc3d3c6fd9aec1a@mail.gmail.com> hey everyone. just wondering if anyone has tested recording conferences at 48000h celt to a shoutcast stream or wav file. we are able to have cd quality conferences with 3 members each using the celt codec with little or no noise disturbances or skipping. but when we try to record the conference either to a wav file or to a shoutcast stream, the quality significantly decreases due to skipping or popping noises. im not sure but maybe we are having this problem because our server doesnt have the CPU power to handle reencoding on the fly like that. we are using a 2.8ghz amd64 dual core, 4gig ddr 800 as our freeswitch server. im thinking if there is a way to record the conference as a celt audio file (instead of reencoding to mp3) that may reduce the CPU power needed and therefore solve the problem or we just need a more powerful server. could anyone recommended what kind of server we would need to handle such instances as i described above? thank you. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090203/d9a2483d/attachment-0002.html From nicolas at medularis.com Tue Feb 3 09:20:56 2009 From: nicolas at medularis.com (Nicolas Brenner) Date: Tue, 3 Feb 2009 14:20:56 -0300 Subject: [Freeswitch-users] origainate through sofia gateway In-Reply-To: <1233668179.5354.15.camel@dotw1126.dotsystems.pl> References: <1233668179.5354.15.camel@dotw1126.dotsystems.pl> Message-ID: <1b46b4e80902030920p521d6079s4c9b78d8d39a924a@mail.gmail.com> Jacek, I had a similar problem once. It actually depends on your sip gateway, but I was able to solve the problem by setting the caller id, ie: session1 = new Session(); session1.setCallerData("caller_id_name", "8280052500"); session1.setCallerData("caller_id_number", "8280052500"); session1.originate(session1, "{ignore_early_media=true}sofia/gateway/sip.ipcorp.cl/0225490317", 60); In this case, the caller_id was the number assigned to me by the external gateway. Hope it helps. Nicolas On Tue, Feb 3, 2009 at 10:36 AM, Jacek Sokulski wrote: > Hello > I am trying to initiate a call from javascript, it works fine for local numbers: > >> session1.originate(session1, "{ignore_early_media=true}user/1008 at 192.168.1.122"); > > but when I am trying to connect through sofia gateway, the connection is not being established: > >> session2.originate(session2, "sofia/gateway/halonet/0225490317"); > > although I can call to this number from softphone. > I have also tried setting effective_caller_id_number: > >> session1.originate(session1, "{effective_caller_id_number=fixed0248b}sofia/gateway/halonet/0225490317"); > > with the same result. > > A configuration in the dialplan that works is: > >> >> >> >> >> >> >> >> > > > Would appreciate any help. > Jacek > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From brian at freeswitch.org Tue Feb 3 09:23:02 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 3 Feb 2009 11:23:02 -0600 Subject: [Freeswitch-users] shoutcast skips In-Reply-To: <2cef777b0902030916s77339375pabc3d3c6fd9aec1a@mail.gmail.com> References: <2cef777b0902030916s77339375pabc3d3c6fd9aec1a@mail.gmail.com> Message-ID: <898222CF-3335-4E2D-AE79-E57CFEE7EFB8@freeswitch.org> You forgot to tell us what revision of the code you're on? /b On Feb 3, 2009, at 11:16 AM, e schmidbauer wrote: > hey everyone. just wondering if anyone has tested recording > conferences at 48000h celt to a shoutcast stream or wav file. > we are able to have cd quality conferences with 3 members each using > the celt codec with little or no noise disturbances or skipping. > but when we try to record the conference either to a wav file or to > a shoutcast stream, the quality significantly decreases due to > skipping or popping noises. > im not sure but maybe we are having this problem because our server > doesnt have the CPU power to handle reencoding on the fly like that. > we are using a 2.8ghz amd64 dual core, 4gig ddr 800 as our > freeswitch server. > im thinking if there is a way to record the conference as a celt > audio file (instead of reencoding to mp3) that may reduce the CPU > power needed and therefore solve the problem or we just need a more > powerful server. > could anyone recommended what kind of server we would need to handle > such instances as i described above? thank you. > _______________________________________________ > Freeswitch-users mailing list From brian at freeswitch.org Tue Feb 3 09:25:33 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 3 Feb 2009 11:25:33 -0600 Subject: [Freeswitch-users] origainate through sofia gateway In-Reply-To: <1b46b4e80902030920p521d6079s4c9b78d8d39a924a@mail.gmail.com> References: <1233668179.5354.15.camel@dotw1126.dotsystems.pl> <1b46b4e80902030920p521d6079s4c9b78d8d39a924a@mail.gmail.com> Message-ID: <5E2B217D-8C85-48DB-800A-535A7FAAE24B@freeswitch.org> YOU should NEVER use this method or call setCallerData at all you should use the correct methods to override the callerid. If its a B-Leg born from an A-Leg you use these on the on the A-Leg: http://wiki.freeswitch.org/wiki/Channel_Variables#effective_caller_id_name http://wiki.freeswitch.org/wiki/Channel_Variables#effective_caller_id_number If you're originating you use this: http://wiki.freeswitch.org/wiki/Channel_Variables#origination_caller_id_name http://wiki.freeswitch.org/wiki/Channel_Variables#origination_caller_id_number /b On Feb 3, 2009, at 11:20 AM, Nicolas Brenner wrote: > Jacek, > > I had a similar problem once. It actually depends on your sip gateway, > but I was able to solve the problem by setting the caller id, ie: > > session1 = new Session(); > session1.setCallerData("caller_id_name", "8280052500"); > session1.setCallerData("caller_id_number", "8280052500"); > session1.originate(session1, > "{ignore_early_media=true}sofia/gateway/sip.ipcorp.cl/0225490317", > 60); > > In this case, the caller_id was the number assigned to me by the > external gateway. > > Hope it helps. > > Nicolas From msc at freeswitch.org Tue Feb 3 09:27:57 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 3 Feb 2009 09:27:57 -0800 Subject: [Freeswitch-users] Generating calls from external source In-Reply-To: References: <40B58CBF-9D0F-43A3-B36F-BD05916756E4@3c.co.uk> <1233666748.24619.8.camel@stargate> Message-ID: <87f2f3b90902030927t2566577cm68248a16eff0246b@mail.gmail.com> On Tue, Feb 3, 2009 at 8:53 AM, Nik Middleton wrote: > Are you suggesting that I should process the call externally instead of > using the dialplan? That would be neat as the audio file select could I'm not saying you should, merely that you could. What I did was create a bunch of extensions in my dialplan that handled various steps of the IVR outbound call: start, answered, busy, not answered, SIT tones, etc. So my originate command would originate the call (A leg) and drop the B leg into the dialplan at the "start" extension and then it goes from there. It listens for early media busy or SIT tones and also does an "execute_on_answer" to the extension that does the actual IVR. (Only need the IVR on an answered call.) If the call is not answered after 25 seconds then I run a Lua script that checks for the presence of certain channel variables that I set with the "tone_detect" application (busy and SIT). If none of those are present then I assume the call went unanswered and do the post-processing. > be driven from the db select for the number. I presume that I could > also bridge the call to another number as well dependant on DTMF > selection? Yes, you can do this as well. You can build an IVR in XML or you can build in a scripting language like Lua: demo IVR: http://svn.freeswitch.org/svn/freeswitch/trunk/conf/autoload_configs/ivr.conf.xml Lua IVR info: http://wiki.freeswitch.org/wiki/IVR#Lua_IVRs Sorry if this is all a bit overwhelming, but you'll be glad that you dove in to FS because it does soooo much and does it so well. Enjoy! -MC > > Regards > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Raul > Fragoso > Sent: 03 February 2009 13:12 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Generating calls from external source > > In addition do David's suggestion, you probably want to have your > application to watch for some specific events after the call is > originated and take action based on them. For example, you could watch > for the CHANNEL_ANSWER event and play some audio file waiting for some > digit, which is generated by the DTMF event. > To watch only for those specific events, you should do the following > just after authentication (still using Perl as an example, but the > mod_event_socket is language agnostic), then you will receive those > events from FreeSWITCH through the socket stream: > > ... > print $sock "auth XXX\n\n"; > print $sock "event plain CHANNEL_ANSWER DTMF\n\n"; > ... > > To see a list of available events, please look at the following wiki > pages: > http://wiki.freeswitch.org/wiki/Mod_event_socket#event > http://wiki.freeswitch.org/wiki/Event_list > > Regards, > > Raul > > On Tue, 2009-02-03 at 09:46 +0000, David Knell wrote: >> Hi Nik, >> >> >> Here's a snipped in Perl that launches an outbound call: >> >> >> if (my $sock = IO::Socket::INET->new(Proto =>'tcp', PeerAddr => >> '127.0.0.1', PeerPort => 8021)) { >> print $sock "auth XXX\n\n"; >> print $sock "api originate {softivr_id=$siid,src_softivr_id= >> $siid,softivr_outdial=true}sofia/frombt/$ntd at 1.2.3.4 $service\n\n"; >> $sock->close(); >> } >> >> >> - it does no error checking or anything, but (line by line) it: >> - opens a socket to the event socket interface >> - authenticates >> - issues an originate which dials out to the number in $ntd. The >> bits in {} set a bunch of variables on the channel, which are used by >> the software which processes the call later on. The call is linked to >> the extension in $service - FS looks this up in the dialplan - which >> handles our end. >> - closes the socket >> >> >> Cheers -- >> >> >> Dave >> >> >> >> > Thanks for that, coming from a C++ background it's a refreshing >> > change to be looking at something that seems logical and efficient. >> > >> > I'd briefly looked at the event socket and wondered if that was the >> > way to go. I presume that there's some sort of event generation >> > that can trigger and external process as well somewhere, though all >> > I need to do is update mysql (hopefully using some sort of pooled >> > connection) >> > >> > I'm not using a TDM card, I have a direct interconnect with the PSTN >> > breakout provider with 1,500 channels available to me. I'm finding >> > Asterisk proving to be less than stable at high call volumes and >> > load values spike at more than 100 calls with billing/accounting in >> > place, hence my interest in FS. The only thing that's concerning me >> > is XML at the moment. Lots of code and very wordy. I'm sure I'll >> > appreciate why XML given time >> > >> > Regards, >> > >> > >> > ____________________________________________________________________ >> > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Michael S Collins >> > Sent: 03 February 2009 01:17 >> > To: freeswitch-users at lists.freeswitch.org >> > Subject: Re: [Freeswitch-users] Generating calls from external >> > source >> > >> > Nik, >> > >> > Welcome to FreeSWITCH! The short answer is "yes, FS can do that." >> > The first thing that you should do is unlearn "the Asterisk way" of >> > thinking. Usually there is an elegant way of doing things in FS that >> > wasn't possible in Ast. >> > >> > I would recommend that you start by looking at the event socket, >> > which is somewhat analogous to the AMI only cooler. :) I have >> > personally done something similar to this using the event socket and >> > a Perl script. The key is to learn the syntax of the originate >> > command. (definitely hit the wiki and IRC channel) >> > Are you using TDM cards for this? Just curious. >> > >> > -MC (IRC nick: mercutioviz) >> > >> > Sent from my iPhone >> > >> > On Feb 2, 2009, at 3:35 PM, "Nik Middleton" >> > wrote: >> > > Hi Guys, >> > > >> > > As a long time Asterisk user, I'm looking into freeswitch as an >> > > alternative mainly due to (list multiple reasons here) >> > > >> > > Can anyone give me a pointer as to how I would achieve the >> > > following? >> > > >> > > I need to replicate an emergency broadcast system currently >> > > running under Asterisk. >> > > >> > > At the moment, I run through a Mysql database and using the >> > > manager API, issues an Originate command to dial a number. >> > > >> > > When the call is answered, a message is played, and the recipient >> > > has the option of hitting a digit to confirm receipt. I then call >> > > an AGI script to update the database. >> > > >> > > Is this fairly easy to do in Freeswitch? >> > > >> > > Not looking for code, just some pointers as to what's available to >> > > do the above / >> > > >> > > Regards, >> > > _______________________________________________ >> > > Freeswitch-users mailing list >> > > Freeswitch-users at lists.freeswitch.org >> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > > http://www.freeswitch.org >> > _______________________________________________ >> > Freeswitch-users mailing list >> > Freeswitch-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From nicolas at medularis.com Tue Feb 3 09:29:26 2009 From: nicolas at medularis.com (Nicolas Brenner) Date: Tue, 3 Feb 2009 14:29:26 -0300 Subject: [Freeswitch-users] Generating calls from external source In-Reply-To: <35b355e90902030913s1cec764j46280ac2d9a162a7@mail.gmail.com> References: <40B58CBF-9D0F-43A3-B36F-BD05916756E4@3c.co.uk> <1233666748.24619.8.camel@stargate> <35b355e90902030913s1cec764j46280ac2d9a162a7@mail.gmail.com> Message-ID: <1b46b4e80902030929h4d4b4b39x9884f856b6028ad@mail.gmail.com> Nik, There's also a PHP library fs_sock.php under contrib in the source code. I used it to create a simple app that originates calls and then run some other sutff when it detects the call has ended. The actual call originate command is executed inside a javascript file which is run using bgapi jsrun. The js script also makes a POST request to an external URL using CURL. There's plenty to play around with, Freeswitch is really great, and mostly easy, a world of difference with *. Good luck! Nicolas On Tue, Feb 3, 2009 at 2:13 PM, Shelby Ramsey wrote: > Nik, > There are a lot of ways to make FS dial out and deliver messaging etc. We > are going through the process of replacing * for this purpose. For us > (getting started with the help of our friends here on the list) it has been > pretty easy. > With * we were using AMI to originate calls ... to migrate to FS we just > changed that to use event_socket with bgapi to originate the call and > connect the call to a context and extension. There are several ways to get > the dialplan to FS after that ... a script, xml_curl, or statically > configured in the conf directory. > So as an example the application we have just logs into the FS socket > (similar to * but much better) and then rips off calls like this: > bgapi originate{$set_some_vars}sofia/external/$ANI@$IP:$PORT $EXTENSION xml > $CONTEXT > The beauty of it all is that: > -- a lot of flexibility in what you can do (like drive the call through > events) > -- the CDR reporting is about 3 million times better than * > -- obviously higher capacity > I'd start playing with event_socket and some static dialplans to get the > feel for it ... but if you have an application written already to work with > * (i.e. the logic and backend) it will be very easy to migrate and you'll be > glad you did it! > Shelby > > > On Tue, Feb 3, 2009 at 10:53 AM, Nik Middleton > wrote: >> >> Are you suggesting that I should process the call externally instead of >> using the dialplan? That would be neat as the audio file select could >> be driven from the db select for the number. I presume that I could >> also bridge the call to another number as well dependant on DTMF >> selection? >> >> Regards >> >> >> -----Original Message----- >> From: freeswitch-users-bounces at lists.freeswitch.org >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Raul >> Fragoso >> Sent: 03 February 2009 13:12 >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] Generating calls from external source >> >> In addition do David's suggestion, you probably want to have your >> application to watch for some specific events after the call is >> originated and take action based on them. For example, you could watch >> for the CHANNEL_ANSWER event and play some audio file waiting for some >> digit, which is generated by the DTMF event. >> To watch only for those specific events, you should do the following >> just after authentication (still using Perl as an example, but the >> mod_event_socket is language agnostic), then you will receive those >> events from FreeSWITCH through the socket stream: >> >> ... >> print $sock "auth XXX\n\n"; >> print $sock "event plain CHANNEL_ANSWER DTMF\n\n"; >> ... >> >> To see a list of available events, please look at the following wiki >> pages: >> http://wiki.freeswitch.org/wiki/Mod_event_socket#event >> http://wiki.freeswitch.org/wiki/Event_list >> >> Regards, >> >> Raul >> >> On Tue, 2009-02-03 at 09:46 +0000, David Knell wrote: >> > Hi Nik, >> > >> > >> > Here's a snipped in Perl that launches an outbound call: >> > >> > >> > if (my $sock = IO::Socket::INET->new(Proto =>'tcp', PeerAddr => >> > '127.0.0.1', PeerPort => 8021)) { >> > print $sock "auth XXX\n\n"; >> > print $sock "api originate {softivr_id=$siid,src_softivr_id= >> > $siid,softivr_outdial=true}sofia/frombt/$ntd at 1.2.3.4 $service\n\n"; >> > $sock->close(); >> > } >> > >> > >> > - it does no error checking or anything, but (line by line) it: >> > - opens a socket to the event socket interface >> > - authenticates >> > - issues an originate which dials out to the number in $ntd. The >> > bits in {} set a bunch of variables on the channel, which are used by >> > the software which processes the call later on. The call is linked to >> > the extension in $service - FS looks this up in the dialplan - which >> > handles our end. >> > - closes the socket >> > >> > >> > Cheers -- >> > >> > >> > Dave >> > >> > >> > >> > > Thanks for that, coming from a C++ background it's a refreshing >> > > change to be looking at something that seems logical and efficient. >> > > >> > > I'd briefly looked at the event socket and wondered if that was the >> > > way to go. I presume that there's some sort of event generation >> > > that can trigger and external process as well somewhere, though all >> > > I need to do is update mysql (hopefully using some sort of pooled >> > > connection) >> > > >> > > I'm not using a TDM card, I have a direct interconnect with the PSTN >> > > breakout provider with 1,500 channels available to me. I'm finding >> > > Asterisk proving to be less than stable at high call volumes and >> > > load values spike at more than 100 calls with billing/accounting in >> > > place, hence my interest in FS. The only thing that's concerning me >> > > is XML at the moment. Lots of code and very wordy. I'm sure I'll >> > > appreciate why XML given time >> > > >> > > Regards, >> > > >> > > >> > > ____________________________________________________________________ >> > > From: freeswitch-users-bounces at lists.freeswitch.org >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of >> Michael S Collins >> > > Sent: 03 February 2009 01:17 >> > > To: freeswitch-users at lists.freeswitch.org >> > > Subject: Re: [Freeswitch-users] Generating calls from external >> > > source >> > > >> > > Nik, >> > > >> > > Welcome to FreeSWITCH! The short answer is "yes, FS can do that." >> > > The first thing that you should do is unlearn "the Asterisk way" of >> > > thinking. Usually there is an elegant way of doing things in FS that >> > > wasn't possible in Ast. >> > > >> > > I would recommend that you start by looking at the event socket, >> > > which is somewhat analogous to the AMI only cooler. :) I have >> > > personally done something similar to this using the event socket and >> > > a Perl script. The key is to learn the syntax of the originate >> > > command. (definitely hit the wiki and IRC channel) >> > > Are you using TDM cards for this? Just curious. >> > > >> > > -MC (IRC nick: mercutioviz) >> > > >> > > Sent from my iPhone >> > > >> > > On Feb 2, 2009, at 3:35 PM, "Nik Middleton" >> > > wrote: >> > > > Hi Guys, >> > > > >> > > > As a long time Asterisk user, I'm looking into freeswitch as an >> > > > alternative mainly due to (list multiple reasons here) >> > > > >> > > > Can anyone give me a pointer as to how I would achieve the >> > > > following? >> > > > >> > > > I need to replicate an emergency broadcast system currently >> > > > running under Asterisk. >> > > > >> > > > At the moment, I run through a Mysql database and using the >> > > > manager API, issues an Originate command to dial a number. >> > > > >> > > > When the call is answered, a message is played, and the recipient >> > > > has the option of hitting a digit to confirm receipt. I then call >> > > > an AGI script to update the database. >> > > > >> > > > Is this fairly easy to do in Freeswitch? >> > > > >> > > > Not looking for code, just some pointers as to what's available to >> > > > do the above / >> > > > >> > > > Regards, >> > > > _______________________________________________ >> > > > Freeswitch-users mailing list >> > > > Freeswitch-users at lists.freeswitch.org >> > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > > > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > > > http://www.freeswitch.org >> > > _______________________________________________ >> > > Freeswitch-users mailing list >> > > Freeswitch-users at lists.freeswitch.org >> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > > http://www.freeswitch.org >> > > >> > >> > >> > _______________________________________________ >> > Freeswitch-users mailing list >> > Freeswitch-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From nicolas at medularis.com Tue Feb 3 09:31:32 2009 From: nicolas at medularis.com (Nicolas Brenner) Date: Tue, 3 Feb 2009 14:31:32 -0300 Subject: [Freeswitch-users] origainate through sofia gateway In-Reply-To: <5E2B217D-8C85-48DB-800A-535A7FAAE24B@freeswitch.org> References: <1233668179.5354.15.camel@dotw1126.dotsystems.pl> <1b46b4e80902030920p521d6079s4c9b78d8d39a924a@mail.gmail.com> <5E2B217D-8C85-48DB-800A-535A7FAAE24B@freeswitch.org> Message-ID: <1b46b4e80902030931y6a1679afw12ec37596ba9d77d@mail.gmail.com> Oops! Well, fortunately I don't use that voip provider anymore (nor the script). Thanks Brian. Nicolas On Tue, Feb 3, 2009 at 2:25 PM, Brian West wrote: > YOU should NEVER use this method or call setCallerData at all you > should use the correct methods to override the callerid. > > If its a B-Leg born from an A-Leg you use these on the on the A-Leg: > > http://wiki.freeswitch.org/wiki/Channel_Variables#effective_caller_id_name > http://wiki.freeswitch.org/wiki/Channel_Variables#effective_caller_id_number > > If you're originating you use this: > > http://wiki.freeswitch.org/wiki/Channel_Variables#origination_caller_id_name > http://wiki.freeswitch.org/wiki/Channel_Variables#origination_caller_id_number > > /b From raul at etellicom.com Tue Feb 3 09:36:06 2009 From: raul at etellicom.com (Raul Fragoso) Date: Tue, 03 Feb 2009 15:36:06 -0200 Subject: [Freeswitch-users] Generating calls from external source In-Reply-To: References: <40B58CBF-9D0F-43A3-B36F-BD05916756E4@3c.co.uk> <1233666748.24619.8.camel@stargate> Message-ID: <1233682566.24619.34.camel@stargate> Hi Nik, That's one possibility, yes. You could use mod_xml_curl to provide the dial-plan on the fly and then use mod_event_socket to send commands to FS and process events. That's exactly what I do actually, we have an IVR engine that is driven by mod_event_socket and another module that provides the XML dial-plan through mod_xml_curl. The beauty of FS is that you have many options to tack a problem, and all of those options are very elegant. I suggest looking at mod_event_socket first and then decide if you can live with the static dial-plan or go to a more dynamic dial-plan via mod_xml_curl. Regards, Raul On Tue, 2009-02-03 at 16:53 +0000, Nik Middleton wrote: > Are you suggesting that I should process the call externally instead of > using the dialplan? That would be neat as the audio file select could > be driven from the db select for the number. I presume that I could > also bridge the call to another number as well dependant on DTMF > selection? > > Regards > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Raul > Fragoso > Sent: 03 February 2009 13:12 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Generating calls from external source > > In addition do David's suggestion, you probably want to have your > application to watch for some specific events after the call is > originated and take action based on them. For example, you could watch > for the CHANNEL_ANSWER event and play some audio file waiting for some > digit, which is generated by the DTMF event. > To watch only for those specific events, you should do the following > just after authentication (still using Perl as an example, but the > mod_event_socket is language agnostic), then you will receive those > events from FreeSWITCH through the socket stream: > > ... > print $sock "auth XXX\n\n"; > print $sock "event plain CHANNEL_ANSWER DTMF\n\n"; > ... > > To see a list of available events, please look at the following wiki > pages: > http://wiki.freeswitch.org/wiki/Mod_event_socket#event > http://wiki.freeswitch.org/wiki/Event_list > > Regards, > > Raul > > On Tue, 2009-02-03 at 09:46 +0000, David Knell wrote: > > Hi Nik, > > > > > > Here's a snipped in Perl that launches an outbound call: > > > > > > if (my $sock = IO::Socket::INET->new(Proto =>'tcp', PeerAddr => > > '127.0.0.1', PeerPort => 8021)) { > > print $sock "auth XXX\n\n"; > > print $sock "api originate {softivr_id=$siid,src_softivr_id= > > $siid,softivr_outdial=true}sofia/frombt/$ntd at 1.2.3.4 $service\n\n"; > > $sock->close(); > > } > > > > > > - it does no error checking or anything, but (line by line) it: > > - opens a socket to the event socket interface > > - authenticates > > - issues an originate which dials out to the number in $ntd. The > > bits in {} set a bunch of variables on the channel, which are used by > > the software which processes the call later on. The call is linked to > > the extension in $service - FS looks this up in the dialplan - which > > handles our end. > > - closes the socket > > > > > > Cheers -- > > > > > > Dave > > > > > > > > > Thanks for that, coming from a C++ background it's a refreshing > > > change to be looking at something that seems logical and efficient. > > > > > > I'd briefly looked at the event socket and wondered if that was the > > > way to go. I presume that there's some sort of event generation > > > that can trigger and external process as well somewhere, though all > > > I need to do is update mysql (hopefully using some sort of pooled > > > connection) > > > > > > I'm not using a TDM card, I have a direct interconnect with the PSTN > > > breakout provider with 1,500 channels available to me. I'm finding > > > Asterisk proving to be less than stable at high call volumes and > > > load values spike at more than 100 calls with billing/accounting in > > > place, hence my interest in FS. The only thing that's concerning me > > > is XML at the moment. Lots of code and very wordy. I'm sure I'll > > > appreciate why XML given time > > > > > > Regards, > > > > > > > > > ____________________________________________________________________ > > > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Michael S Collins > > > Sent: 03 February 2009 01:17 > > > To: freeswitch-users at lists.freeswitch.org > > > Subject: Re: [Freeswitch-users] Generating calls from external > > > source > > > > > > Nik, > > > > > > Welcome to FreeSWITCH! The short answer is "yes, FS can do that." > > > The first thing that you should do is unlearn "the Asterisk way" of > > > thinking. Usually there is an elegant way of doing things in FS that > > > wasn't possible in Ast. > > > > > > I would recommend that you start by looking at the event socket, > > > which is somewhat analogous to the AMI only cooler. :) I have > > > personally done something similar to this using the event socket and > > > a Perl script. The key is to learn the syntax of the originate > > > command. (definitely hit the wiki and IRC channel) > > > Are you using TDM cards for this? Just curious. > > > > > > -MC (IRC nick: mercutioviz) > > > > > > Sent from my iPhone > > > > > > On Feb 2, 2009, at 3:35 PM, "Nik Middleton" > > > wrote: > > > > Hi Guys, > > > > > > > > As a long time Asterisk user, I'm looking into freeswitch as an > > > > alternative mainly due to (list multiple reasons here) > > > > > > > > Can anyone give me a pointer as to how I would achieve the > > > > following? > > > > > > > > I need to replicate an emergency broadcast system currently > > > > running under Asterisk. > > > > > > > > At the moment, I run through a Mysql database and using the > > > > manager API, issues an Originate command to dial a number. > > > > > > > > When the call is answered, a message is played, and the recipient > > > > has the option of hitting a digit to confirm receipt. I then call > > > > an AGI script to update the database. > > > > > > > > Is this fairly easy to do in Freeswitch? > > > > > > > > Not looking for code, just some pointers as to what's available to > > > > do the above / > > > > > > > > Regards, > > > > _______________________________________________ > > > > Freeswitch-users mailing list > > > > Freeswitch-users at lists.freeswitch.org > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > http://www.freeswitch.org > > > _______________________________________________ > > > Freeswitch-users mailing list > > > Freeswitch-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From nik.middleton at noblesolutions.co.uk Tue Feb 3 10:07:32 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Tue, 3 Feb 2009 18:07:32 -0000 Subject: [Freeswitch-users] OPenser <-> FS Do I need this? In-Reply-To: <87f2f3b90902030908n629399bdmdd46f633b803b5f6@mail.gmail.com> References: <2cbf225c0902021821u2bc8b622ka5d9f94f05a968ae@mail.gmail.com><1233668029.24619.29.camel@stargate> <87f2f3b90902030908n629399bdmdd46f633b803b5f6@mail.gmail.com> Message-ID: Well Openser has better NAT handling than Asterisk for a start. In addition it takes the load off of Asterisk with regards to registrations. Further, I'm able to have multiple asterisk servers fronted by Openser Finally, I've numerous posts that * chokes with sip clients > 200. I couldn't afford to take the risk. But the biggest issue is with load spikes and asterisk. I've never gotten to the bottom of it, and believe me a lot of people far smarter then me have tried to figure it out. So... The more I can keep asterisk out of the mundane stuff the better. It's been said to me many times, that the way Asterisk is put together is fundamentally flawed and this really shows it's self under load. Not knocking Asterisk, it's served me well for the last 4 years. Heck I've got a book being published on it in a couple of months, but for me, I need a scalable solution, hence my interest in FS. I also don't see * going beyond 1.4. 1.6 as far as I can tell has a very low take-up rate, why ? well because they've changed how everything works to the extent that hardly anything written for 1.4 can port to 1.6. The syntax changes don't appear to serve any real purpose. So to get back to my original question, if FS can handle a significantly higher number of call setups, then perhaps I don't need OpenSer, that was the thrust of my post. Regards -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 03 February 2009 17:08 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] OPenser <-> FS Do I need this? On Tue, Feb 3, 2009 at 8:20 AM, Nik Middleton wrote: > Newbie with FS, currently have Asterisk servers front ended by Openser > > Question: I have around 400 sip remote clients, if I were to deploy FS, > do I need Openser? Is there any advantage in retaining Openser? If I may ask... why did you have OpenSER with your Asterisk deployment? Reason I ask is because some people do that "because Asterisk sucks" but others have a specific application or reason. What does OpenSER do for your Asterisk install? -MC > > Regards > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From krice at freeswitch.org Tue Feb 3 10:16:08 2009 From: krice at freeswitch.org (Ken Rice) Date: Tue, 03 Feb 2009 12:16:08 -0600 Subject: [Freeswitch-users] OPenser <-> FS Do I need this? In-Reply-To: Message-ID: FreeSwitch is very capable of handling high call setup loads... The question is what do you consider high setup loads? Where it is true, OpenSER/SIP/whatever its called this week can handle a much higher packet per second load then freeswitch, freeswitch on the other hand is capable of handling much more call volume then asterisk... Certain people hate when I quote numbers but I have personally deployed FreeSwitch on projects that handle (per FS Box) > 500 calls/sec (that's 2 leg calls) and in excess of concurrent calls... The real question is not can FS hang, but what at what level do you call 'high volume'... What I call high volume is a telemarketer running at 2500 calls/sec and peak concurrent channel usage in the 10,000 to 15,000 channel range K > From: Nik Middleton > Subject: Re: [Freeswitch-users] OPenser <-> FS Do I need this? > > ----SNIP > So to get back to my original question, if FS can handle a significantly > higher number of call setups, then perhaps I don't need OpenSer, that > was the thrust of my post. > ----SNIP From nik.middleton at noblesolutions.co.uk Tue Feb 3 10:31:37 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Tue, 3 Feb 2009 18:31:37 -0000 Subject: [Freeswitch-users] OPenser <-> FS Do I need this? In-Reply-To: References: Message-ID: If you're telling me that FS can handle the figures quoted, that's plenty enough for me. I have 5,000 lines PSTN /channels, possibly double that shortly. I need to fill all of them as quickly as possible and maintain that level for a given period of time. So I guess I'm in the upper medium end of the scale. Regards, -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice Sent: 03 February 2009 18:16 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] OPenser <-> FS Do I need this? FreeSwitch is very capable of handling high call setup loads... The question is what do you consider high setup loads? Where it is true, OpenSER/SIP/whatever its called this week can handle a much higher packet per second load then freeswitch, freeswitch on the other hand is capable of handling much more call volume then asterisk... Certain people hate when I quote numbers but I have personally deployed FreeSwitch on projects that handle (per FS Box) > 500 calls/sec (that's 2 leg calls) and in excess of concurrent calls... The real question is not can FS hang, but what at what level do you call 'high volume'... What I call high volume is a telemarketer running at 2500 calls/sec and peak concurrent channel usage in the 10,000 to 15,000 channel range K > From: Nik Middleton > Subject: Re: [Freeswitch-users] OPenser <-> FS Do I need this? > > ----SNIP > So to get back to my original question, if FS can handle a significantly > higher number of call setups, then perhaps I don't need OpenSer, that > was the thrust of my post. > ----SNIP _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From ajlong at worldlink.net Tue Feb 3 10:47:59 2009 From: ajlong at worldlink.net (Adam Long) Date: Tue, 3 Feb 2009 13:47:59 -0500 Subject: [Freeswitch-users] FreeSwitch setup as a "Dumb" SBC In-Reply-To: <4987E527.1040909@laposte.net> References: <019501c985ac$4f00ee60$ed02cb20$@net> <4987E527.1040909@laposte.net> Message-ID: <022001c9862f$efd4b7d0$cf7e2770$@net> Hi Rod, Great info, Thanks! Glad to see others are interested in the same concept. My reasons for SER as routing core and implementation is slightly different yet similar. I like your Redirect model, with that you are truly using your Kamailio as route server only. I would imagine very scalable. - Are you able to do any round robin, serial or parallel forking with this? - I wonder if multiple Contacts in the 302 response maybe with some logic in FreeSwitch dialplan? If so I think your design is a bit more efficient than mine as it keeps SER out of the call path. My design is little different.. it is more of a "Stateful" setup. With SER staying in call path and FreeSwitch at Edge. I do this to enable Serial Forking to a series of SBCs (FreeSwitch) geo distributed, when one of the branches is congested it forks to the next SBC (route). The FreeSwitch guys are probably right tho... with mod_easyroute and mod_lcr we could probably implement all of this in FreeSwitch without SER. I would be curious to know if anyone is doing something similar at high volumes and what sort of concurrency and cps they are able to achieve. I am a Perl and C# guy, I thought about implementing a mod_manged_lcr with memcached support. Memcache support would prob boost the scalability by a factor of 10 at least. I will let you know if I end up developing a high performance FreeSwitch route module. Right now I use memcache in a OpenSIPS perl script for my route caching and its incredibly fast and clusters well. It actually might be easier to add memcached support to mod_lcr and mod_easyroute but im not real strong in C/C++ I'll jump on IRC later and chat with some of the experts on this as I know memcache has been discussed before. I'd be curious to know if any progress has been made there already. Regards, -Adam -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of rod Sent: Tuesday, February 03, 2009 1:33 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] FreeSwitch setup as a "Dumb" SBC Hi Adam, I'm in the process of using FS as a SBC. For the route lookup, I do it using OpenSER carrierroute, without having to flow through SBC---Openser---SBC. I'm using carrierroute at this time cause I need more than 200 000 routing entries and carrierroute has been tested with twice this number. Here is the setup: - install openser and carrierroute and make openser listening on 127.0.0.1:5062 (for example) on your SBC - populate carrierroute table What I do to use carrierroute module from FS is to use a specific X-header (X-LOOKUP). In the dialplan, in the default context, I have something like this: The process is simple: the export "sip_h_X-ROUTE=LOOKUP" had a sip header X-ROUTE=LOOKUP then I bridge the call to 127.0.0.1:5062 (openser process) In openser I have a route block that checks the presence of header LOOKUP and openser sends a "604: unable to route call" if the prefix is not found, or a "302: with the IP of the gateway found" In FS, you can get the IP using the variable "${sip_redirect_contact_host_0}". Then I transfer this to the context ROUTING, where the check condition is based on the LOOKUP header that has been rewritten with this variable. I will document all this setup (installation of openser/carrierroute and config file of FS and openser) on a wiki page I start writing yesterday, so please be indulgent and patient. The next step is to test the scalability of this. I'm a very bad programmer, so that's the only way for me to contribute to FS, and as I see many people interested for an SBC setup, I think it could be great if we share our work/knowlegde. The wiki page is there: http://wiki.freeswitch.org/wiki/SBC_Setup regards, rod. Adam Long wrote: > > Hi Guys, > > I've been working at setting up a couple of FreeSwitch nodes as a > topology hiding SBCs that handles both ingress traffic from my > > providers/peers and pass traffic up to an openser router that then > routes call across the cluster of SBCs through which they reach the > destination. > > I have OpenSIPS/SER setup doing DB route lookups and ENUM with > LCR/Serial forking etc. > > My question is what would be the best way to send a call out to a > destination choosen by the OpenSER router? > > For example: > > SIP Provider -- > SBC --- > OpenSER ---- ( route lookup returns > 123.123.123.4 as dest ) -- > SBC --- > 123.123.123.4 > > I was thinking something along the lines of adding a "X-Route-To: > +1NXXNXXXXXX@ 123.123.123.4" with openser > > and then something like this in the SBC. > > > > > > > > > > > > Is this a wise approach, is there anything I could do to do this better? > > I'd like to keep the logic in the SBCs as simple as possible. > > I am pretty familiar with SIP but my knowledge fades when it gets into > the nitty gritty of routing. ie the Contact: and Via: headers > > and all that good stuff. > > I should also state I have two profiles defined one for the > internal/private "core" network and one for the outside "external" > network. > > Any thoughts on this at all would be greatly appreciated. > > Am I missing something in the SIP spec that would allow for this is a > standardized way? > > Regards, > > -Adam > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From ajlong at worldlink.net Tue Feb 3 12:24:58 2009 From: ajlong at worldlink.net (Adam Long) Date: Tue, 3 Feb 2009 15:24:58 -0500 Subject: [Freeswitch-users] mod_sofia "ReINVITE" Message-ID: <002b01c9863d$7c107470$74315d50$@net> In every one of my SIP sessions FreeSwitch appears to be inserting .. Contact: sip:mod_sofia at XXX.XXX.XXX.XXX:5060 Is this normal? I only ask as it is causing some of my end points to RE-INVITE back to this after the initial ( INVITE <---- > 100 Trying < --- > 200 OK ) call setup. If this is not normal or by design I can provide more details on configuration and dialplan. Thanks! Regards, -Adam -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090203/dd3018e1/attachment-0002.html From e.schmidbauer at gmail.com Tue Feb 3 12:26:39 2009 From: e.schmidbauer at gmail.com (e schmidbauer) Date: Tue, 3 Feb 2009 15:26:39 -0500 Subject: [Freeswitch-users] shoutcast skips In-Reply-To: <898222CF-3335-4E2D-AE79-E57CFEE7EFB8@freeswitch.org> References: <2cef777b0902030916s77339375pabc3d3c6fd9aec1a@mail.gmail.com> <898222CF-3335-4E2D-AE79-E57CFEE7EFB8@freeswitch.org> Message-ID: <2cef777b0902031226l72f4a4ceia80ec00721456602@mail.gmail.com> im using the latest svn of freeswitch On Tue, Feb 3, 2009 at 12:23 PM, Brian West wrote: > You forgot to tell us what revision of the code you're on? > /b > > On Feb 3, 2009, at 11:16 AM, e schmidbauer wrote: > > > hey everyone. just wondering if anyone has tested recording > > conferences at 48000h celt to a shoutcast stream or wav file. > > we are able to have cd quality conferences with 3 members each using > > the celt codec with little or no noise disturbances or skipping. > > but when we try to record the conference either to a wav file or to > > a shoutcast stream, the quality significantly decreases due to > > skipping or popping noises. > > im not sure but maybe we are having this problem because our server > > doesnt have the CPU power to handle reencoding on the fly like that. > > we are using a 2.8ghz amd64 dual core, 4gig ddr 800 as our > > freeswitch server. > > im thinking if there is a way to record the conference as a celt > > audio file (instead of reencoding to mp3) that may reduce the CPU > > power needed and therefore solve the problem or we just need a more > > powerful server. > > could anyone recommended what kind of server we would need to handle > > such instances as i described above? thank you. > > _______________________________________________ > > Freeswitch-users mailing list > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090203/c90d68c5/attachment-0002.html From krice at freeswitch.org Tue Feb 3 11:26:37 2009 From: krice at freeswitch.org (Ken Rice) Date: Tue, 03 Feb 2009 13:26:37 -0600 Subject: [Freeswitch-users] FreeSwitch setup as a "Dumb" SBC In-Reply-To: <022001c9862f$efd4b7d0$cf7e2770$@net> Message-ID: Actually I currently deploy FreeSWITCH for high volume usage using FreeSWITCH + mod_easyroute (I'm the author) and an advanced LCR module that does things like load balancing across multiple media gateways, auto route advance, and a few other nifty things... (this LCR module uses a proprietary algorithm so its not open source but it is licensable) With these things we do run OpenSER but only as a proxy to aggregate traffic heading upstream toward certain carriers (like L3 who make any IP changes a royal pain) Now to get down to some hard numbers that we have experience Equipment: DB Servers: Dell 2650 RAID 3+1 or 0+1 depending on number of Spindles, Dual 3GHz XEON (single code old slow FSB ones), 4G RAM, running Centos 5.2 and PostgreSQL 8.3 SIP Servers: Dell 1950 Dual Quad Core 2Ghz (E5335 part), 4 to 8G of RAM, GIG-E ethernet, whatever hard drive was cheap at time of order. Nothing really lives on these boxes but FreeSWITCH with mod_easyroute, mod_lcr_adv, and some CDR processing stuff DB servers feed all the route information... (yes we do the route lookups from the DB in real-time, the problem with most LCRs in doing this is an algorithm Call Rates Sustained, 500 avg cps, > 2000 calls (that's 2 legs not 1), avg invite delay 115ms (INVITE in to INVITE out measured with 'ngrep -q -t INVITE' - Note this is not a true picture of PDD as a number of other factors affect that, this is a picture of how much time we are adding on box in delaying an INVITE message) On Registrations we have experienced Registration/second rates exceeding 150 registrations per second using mod_xml_curl to feed the users directory. I suspect, this number can be greatly increased if we were to feed directory with something that cut out the apache and php over head K > From: Adam Long > Reply-To: > Date: Tue, 3 Feb 2009 13:47:59 -0500 > To: > Subject: Re: [Freeswitch-users] FreeSwitch setup as a "Dumb" SBC > > The FreeSwitch guys are probably right tho... with mod_easyroute and mod_lcr > we could probably implement all of this in FreeSwitch without SER. > I would be curious to know if anyone is doing something similar at high > volumes and what sort of concurrency and cps they > are able to achieve. From brian at freeswitch.org Tue Feb 3 12:29:19 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 3 Feb 2009 14:29:19 -0600 Subject: [Freeswitch-users] mod_sofia "ReINVITE" In-Reply-To: <002b01c9863d$7c107470$74315d50$@net> References: <002b01c9863d$7c107470$74315d50$@net> Message-ID: Yes this is normal. Your contact is mod_sofia ... why would it change? Remember its a B2Bua. Now you can put in your sofia profile but be warned it will break some devices. /b On Feb 3, 2009, at 2:24 PM, Adam Long wrote: > In every one of my SIP sessions FreeSwitch appears to be inserting ?. > > Contact: sip:mod_sofia at XXX.XXX.XXX.XXX:5060 > > Is this normal? > > I only ask as it is causing some of my end points to RE-INVITE back > to this after the initial ( INVITE <---- > 100 Trying < --- > > 200 OK ) call setup. > > If this is not normal or by design I can provide more details on > configuration and dialplan. > Thanks! > > Regards, > -Adam -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090203/a393d45f/attachment-0002.html From brian at freeswitch.org Tue Feb 3 12:29:36 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 3 Feb 2009 14:29:36 -0600 Subject: [Freeswitch-users] shoutcast skips In-Reply-To: <2cef777b0902031226l72f4a4ceia80ec00721456602@mail.gmail.com> References: <2cef777b0902030916s77339375pabc3d3c6fd9aec1a@mail.gmail.com> <898222CF-3335-4E2D-AE79-E57CFEE7EFB8@freeswitch.org> <2cef777b0902031226l72f4a4ceia80ec00721456602@mail.gmail.com> Message-ID: "latest" isn't a number... Can you provide the exact SVN rev you're on? /b On Feb 3, 2009, at 2:26 PM, e schmidbauer wrote: > im using the latest svn of freeswitch From anthony.minessale at gmail.com Tue Feb 3 12:45:04 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 3 Feb 2009 14:45:04 -0600 Subject: [Freeswitch-users] mod_voicemail: Limiting the number how often a menu is repeated In-Reply-To: <49887A32.7060804@ewetel.de> References: <498734D0.5060004@ewetel.de> <49887A32.7060804@ewetel.de> Message-ID: <191c3a030902031245m76cca75dk48501976526423a3@mail.gmail.com> no, there is no way to do that. On Tue, Feb 3, 2009 at 11:09 AM, Helmut Kuper wrote: > Hi, > > has anybody an idea? > > regards > Helmut > > Am 02.02.2009 19:00, schrieb Helmut Kuper: > > Hello, > > > > today I searched for a way to limit the number of menu repeatings in > > mod_voicemail to let's say 3 times and when it reached the limit > > voicemail should abort. But I couldn't find a hint. Any ideas? > > > > > > regards > > helmut > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > -- > > Mit freundlichen Gr??en > Helmut Kuper > Finanzdienstleistungen und Entwicklung > Telefax: (0441) 8000-2799 > mailto:helmut.kuper at ewetel.de > ___________________________________ > EWE TEL GmbH > Cloppenburger Stra?e 310 > 26133 Oldenburg > EWE TEL GmbH > > Handelsregister Amtsgericht Oldenburg HRB 3723 > Vorsitzender des Aufsichtsrates: Heiko Harms > Gesch?ftsf?hrung: Hans-Joachim Iken (Vorsitzender), Dr. Norbert Schulz, > Dirk Thole > Homepage: http://www.ewetel.de > ___________________________________ > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090203/a228b229/attachment-0002.html From brian at freeswitch.org Tue Feb 3 12:51:49 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 3 Feb 2009 14:51:49 -0600 Subject: [Freeswitch-users] shoutcast skips In-Reply-To: <2cef777b0902031226l72f4a4ceia80ec00721456602@mail.gmail.com> References: <2cef777b0902030916s77339375pabc3d3c6fd9aec1a@mail.gmail.com> <898222CF-3335-4E2D-AE79-E57CFEE7EFB8@freeswitch.org> <2cef777b0902031226l72f4a4ceia80ec00721456602@mail.gmail.com> Message-ID: <574748E6-1EF0-4F5A-A1D6-338BAA711040@freeswitch.org> Can you get me a sample of the recording to listen to? /b On Feb 3, 2009, at 2:26 PM, e schmidbauer wrote: > im using the latest svn of freeswitch > > On Tue, Feb 3, 2009 at 12:23 PM, Brian West > wrote: > You forgot to tell us what revision of the code you're on? > /b > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090203/c47c0ff0/attachment-0002.html From e.schmidbauer at gmail.com Tue Feb 3 12:59:51 2009 From: e.schmidbauer at gmail.com (e schmidbauer) Date: Tue, 3 Feb 2009 15:59:51 -0500 Subject: [Freeswitch-users] shoutcast skips In-Reply-To: <574748E6-1EF0-4F5A-A1D6-338BAA711040@freeswitch.org> References: <2cef777b0902030916s77339375pabc3d3c6fd9aec1a@mail.gmail.com> <898222CF-3335-4E2D-AE79-E57CFEE7EFB8@freeswitch.org> <2cef777b0902031226l72f4a4ceia80ec00721456602@mail.gmail.com> <574748E6-1EF0-4F5A-A1D6-338BAA711040@freeswitch.org> Message-ID: <2cef777b0902031259t6210b7b2w9873ced0806b98b3@mail.gmail.com> FreeSWITCH Version 1.0.trunk (11567) check out these sample recordings http://bwrl.org/recordings/2009-01-31-12-07-49.mp3 http://bwrl.org/recordings/2009-01-31-12-07-49.wav http://bwrl.org/recordings/test2.mp3 http://bwrl.org/recordings/test2.wav the conferences were recorded as wav files, i then converted them to mp3, both sound the same to me On Tue, Feb 3, 2009 at 3:51 PM, Brian West wrote: > Can you get me a sample of the recording to listen to? > /b > > On Feb 3, 2009, at 2:26 PM, e schmidbauer wrote: > > im using the latest svn of freeswitch > > On Tue, Feb 3, 2009 at 12:23 PM, Brian West wrote: > >> You forgot to tell us what revision of the code you're on? >> /b >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090203/06621b25/attachment-0002.html From brian at freeswitch.org Tue Feb 3 13:21:05 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 3 Feb 2009 15:21:05 -0600 Subject: [Freeswitch-users] shoutcast skips In-Reply-To: <2cef777b0902031259t6210b7b2w9873ced0806b98b3@mail.gmail.com> References: <2cef777b0902030916s77339375pabc3d3c6fd9aec1a@mail.gmail.com> <898222CF-3335-4E2D-AE79-E57CFEE7EFB8@freeswitch.org> <2cef777b0902031226l72f4a4ceia80ec00721456602@mail.gmail.com> <574748E6-1EF0-4F5A-A1D6-338BAA711040@freeswitch.org> <2cef777b0902031259t6210b7b2w9873ced0806b98b3@mail.gmail.com> Message-ID: <81F41092-76A7-4DCD-8274-9584177F8752@freeswitch.org> You're doing distributed radio right? So callers are calling in with CELT from all over the place? Can you contact us on IRC because we are very interested in debugging this issue. You can get us on IRC #freeswitch on irc.freenode.net Thanks, /b On Feb 3, 2009, at 2:59 PM, e schmidbauer wrote: > FreeSWITCH Version 1.0.trunk (11567) > check out these sample recordings > http://bwrl.org/recordings/2009-01-31-12-07-49.mp3 > http://bwrl.org/recordings/2009-01-31-12-07-49.wav > http://bwrl.org/recordings/test2.mp3 > http://bwrl.org/recordings/test2.wav > > the conferences were recorded as wav files, i then converted them to > mp3, both sound the same to me -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090203/35c1f885/attachment-0002.html From e.schmidbauer at gmail.com Tue Feb 3 13:27:42 2009 From: e.schmidbauer at gmail.com (e schmidbauer) Date: Tue, 3 Feb 2009 16:27:42 -0500 Subject: [Freeswitch-users] shoutcast skips In-Reply-To: <81F41092-76A7-4DCD-8274-9584177F8752@freeswitch.org> References: <2cef777b0902030916s77339375pabc3d3c6fd9aec1a@mail.gmail.com> <898222CF-3335-4E2D-AE79-E57CFEE7EFB8@freeswitch.org> <2cef777b0902031226l72f4a4ceia80ec00721456602@mail.gmail.com> <574748E6-1EF0-4F5A-A1D6-338BAA711040@freeswitch.org> <2cef777b0902031259t6210b7b2w9873ced0806b98b3@mail.gmail.com> <81F41092-76A7-4DCD-8274-9584177F8752@freeswitch.org> Message-ID: <2cef777b0902031327v4725f174kcaf976c04926fb21@mail.gmail.com> We are attempting distributed radio. We plan on having the hosts of the shows join the conference using CELT. But callers to the show would be joining using regular phones therefore using lower end codecs. I will be in the IRC shortly. On Tue, Feb 3, 2009 at 4:21 PM, Brian West wrote: > You're doing distributed radio right? So callers are calling in with CELT > from all over the place? Can you contact us on IRC because we are very > interested in debugging this issue. > You can get us on IRC #freeswitch on irc.freenode.net > > Thanks, > > /b > > On Feb 3, 2009, at 2:59 PM, e schmidbauer wrote: > > FreeSWITCH Version 1.0.trunk (11567) > check out these sample recordings > http://bwrl.org/recordings/2009-01-31-12-07-49.mp3 > http://bwrl.org/recordings/2009-01-31-12-07-49.wav > http://bwrl.org/recordings/test2.mp3 > http://bwrl.org/recordings/test2.wav > > the conferences were recorded as wav files, i then converted them to mp3, > both sound the same to me > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090203/b0bea706/attachment-0002.html From kokoska.rokoska at post.cz Tue Feb 3 14:11:58 2009 From: kokoska.rokoska at post.cz (kokoska.rokoska) Date: Tue, 03 Feb 2009 23:11:58 +0100 Subject: [Freeswitch-users] FreeSwitch setup as a "Dumb" SBC In-Reply-To: References: Message-ID: <4988C12E.1090109@post.cz> Ken Rice napsal(a): ... > On Registrations we have experienced Registration/second rates exceeding 150 > registrations per second using mod_xml_curl to feed the users directory. I > suspect, this number can be greatly increased if we were to feed directory > with something that cut out the apache and php over head > If someone interested I have few numbers on Registrar performance: DB server: 2x Quad core E5345 @ 2.33GHz, 16 GiB RAM Centos 5 x86_64, MySQL 5.0 Registrar server: 2x Quad core E5345 @ 2.33GHz, 16 GiB RAM Centos 5 x86_64 Tested using sipp with 10.000 and 30.000 "users". FreeSWITCH as registrar - current trunk: 1. FreeSwitch si simply modified (code doing NAT-ping is commented out :-) 2. Directory is served through lighttpd and simple "C" binary doing one trivial select. Lighttpd runs on the same machine as FS. When I move lighhtpd to another machine, I cannot see any significat performance boost. Result: I can go up to the 470-500 reg/s. and FS is heavy overloaded and retransmissions occurs. Kamailio as registrar - 1.4.3. no TLS: 1. Kamailio runs with usrloc db_mode 3 (no caching) Result: I can go up to the 3500-3700 reg/s. and Kamailio server is at 0.3 load and all 8 cores are bellow 15 %. Without retransmissions. The limit is DB throughput. Just for "curiosity" I switched userloc to db_mode 2 (write back) and at 5000 regs/s I stopped the sipp test, because I saw the bottle neck becomes the server runnig sipp (very old P4 box). Conclusion: While I see amazing FreeSWITCH performance on INVITEs per seconds and concurrent calls (another galaxy from * point of view :-), if you have to handle lots of registrations per second, it is IMO better to use Kamailio/OpenSIPS/SER as separate registrar and "propagate" users to FS through SQL view. Hope this helps someone... Best regards, kokoska.rokoska From anthony.minessale at gmail.com Tue Feb 3 14:34:38 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 3 Feb 2009 16:34:38 -0600 Subject: [Freeswitch-users] FreeSwitch setup as a "Dumb" SBC In-Reply-To: <4988C12E.1090109@post.cz> References: <4988C12E.1090109@post.cz> Message-ID: <191c3a030902031434v1129d684j18b09a954d871f7b@mail.gmail.com> What does it look like if you serve the directory from the static xml file out of curiosity. On Tue, Feb 3, 2009 at 4:11 PM, kokoska.rokoska wrote: > Ken Rice napsal(a): > ... > > > On Registrations we have experienced Registration/second rates exceeding > 150 > > registrations per second using mod_xml_curl to feed the users directory. > I > > suspect, this number can be greatly increased if we were to feed > directory > > with something that cut out the apache and php over head > > > > If someone interested I have few numbers on Registrar performance: > > DB server: > 2x Quad core E5345 @ 2.33GHz, 16 GiB RAM > Centos 5 x86_64, MySQL 5.0 > > Registrar server: > 2x Quad core E5345 @ 2.33GHz, 16 GiB RAM > Centos 5 x86_64 > > Tested using sipp with 10.000 and 30.000 "users". > > > FreeSWITCH as registrar - current trunk: > 1. FreeSwitch si simply modified (code doing NAT-ping is commented out :-) > 2. Directory is served through lighttpd and simple "C" binary doing one > trivial select. Lighttpd runs on the same machine as FS. When I move > lighhtpd to another machine, I cannot see any significat performance boost. > > Result: I can go up to the 470-500 reg/s. and FS is heavy overloaded and > retransmissions occurs. > > > Kamailio as registrar - 1.4.3. no TLS: > 1. Kamailio runs with usrloc db_mode 3 (no caching) > > Result: I can go up to the 3500-3700 reg/s. and Kamailio server is at > 0.3 load and all 8 cores are bellow 15 %. Without retransmissions. The > limit is DB throughput. > Just for "curiosity" I switched userloc to db_mode 2 (write back) and at > 5000 regs/s I stopped the sipp test, because I saw the bottle neck > becomes the server runnig sipp (very old P4 box). > > > Conclusion: > While I see amazing FreeSWITCH performance on INVITEs per seconds and > concurrent calls (another galaxy from * point of view :-), if you have > to handle lots of registrations per second, it is IMO better to use > Kamailio/OpenSIPS/SER as separate registrar and "propagate" users to FS > through SQL view. > > Hope this helps someone... > > Best regards, > > kokoska.rokoska > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090203/3676ff23/attachment-0002.html From kokoska.rokoska at post.cz Tue Feb 3 14:51:00 2009 From: kokoska.rokoska at post.cz (kokoska.rokoska) Date: Tue, 03 Feb 2009 23:51:00 +0100 Subject: [Freeswitch-users] FreeSwitch setup as a "Dumb" SBC In-Reply-To: <191c3a030902031434v1129d684j18b09a954d871f7b@mail.gmail.com> References: <4988C12E.1090109@post.cz> <191c3a030902031434v1129d684j18b09a954d871f7b@mail.gmail.com> Message-ID: <4988CA54.4080003@post.cz> Anthony Minessale napsal(a): > What does it look like if you serve the directory from the static xml > file out of curiosity. > Good question :-) I have never thing about it, becasue I need "dynamic" users. But it should show up very impressive number :-) I'll try it tommorow (here is midnight) and let you know. BTW: I try to find some another server in colocation with higher performace. With mentioned P4 I'm affraid have no chance to stress FS with static xml directory... Thank you for your interest, Anthony! Best regards, kokoska.rokoska > On Tue, Feb 3, 2009 at 4:11 PM, kokoska.rokoska > wrote: > > Ken Rice napsal(a): > ... > > > On Registrations we have experienced Registration/second rates > exceeding 150 > > registrations per second using mod_xml_curl to feed the users > directory. I > > suspect, this number can be greatly increased if we were to feed > directory > > with something that cut out the apache and php over head > > > > If someone interested I have few numbers on Registrar performance: > > DB server: > 2x Quad core E5345 @ 2.33GHz, 16 GiB RAM > Centos 5 x86_64, MySQL 5.0 > > Registrar server: > 2x Quad core E5345 @ 2.33GHz, 16 GiB RAM > Centos 5 x86_64 > > Tested using sipp with 10.000 and 30.000 "users". > > > FreeSWITCH as registrar - current trunk: > 1. FreeSwitch si simply modified (code doing NAT-ping is commented > out :-) > 2. Directory is served through lighttpd and simple "C" binary doing one > trivial select. Lighttpd runs on the same machine as FS. When I move > lighhtpd to another machine, I cannot see any significat performance > boost. > > Result: I can go up to the 470-500 reg/s. and FS is heavy overloaded and > retransmissions occurs. > > > Kamailio as registrar - 1.4.3. no TLS: > 1. Kamailio runs with usrloc db_mode 3 (no caching) > > Result: I can go up to the 3500-3700 reg/s. and Kamailio server is at > 0.3 load and all 8 cores are bellow 15 %. Without retransmissions. The > limit is DB throughput. > Just for "curiosity" I switched userloc to db_mode 2 (write back) and at > 5000 regs/s I stopped the sipp test, because I saw the bottle neck > becomes the server runnig sipp (very old P4 box). > > > Conclusion: > While I see amazing FreeSWITCH performance on INVITEs per seconds and > concurrent calls (another galaxy from * point of view :-), if you have > to handle lots of registrations per second, it is IMO better to use > Kamailio/OpenSIPS/SER as separate registrar and "propagate" users to FS > through SQL view. > > Hope this helps someone... > > Best regards, > > kokoska.rokoska > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From krice at freeswitch.org Tue Feb 3 14:56:17 2009 From: krice at freeswitch.org (Ken Rice) Date: Tue, 03 Feb 2009 16:56:17 -0600 Subject: [Freeswitch-users] FreeSwitch setup as a "Dumb" SBC In-Reply-To: <191c3a030902031434v1129d684j18b09a954d871f7b@mail.gmail.com> Message-ID: Never tried hah... From: Anthony Minessale Reply-To: Date: Tue, 3 Feb 2009 16:34:38 -0600 To: Subject: Re: [Freeswitch-users] FreeSwitch setup as a "Dumb" SBC What does it look like if you serve the directory from the static xml file out of curiosity. On Tue, Feb 3, 2009 at 4:11 PM, kokoska.rokoska wrote: > Ken Rice napsal(a): > ... > >> > On Registrations we have experienced Registration/second rates exceeding >> 150 >> > registrations per second using mod_xml_curl to feed the users directory. I >> > suspect, this number can be greatly increased if we were to feed directory >> > with something that cut out the apache and php over head >> > > > If someone interested I have few numbers on Registrar performance: > > DB server: > 2x Quad core E5345 @ 2.33GHz, 16 GiB RAM > Centos 5 x86_64, MySQL 5.0 > > Registrar server: > 2x Quad core E5345 @ 2.33GHz, 16 GiB RAM > Centos 5 x86_64 > > Tested using sipp with 10.000 and 30.000 "users". > > > FreeSWITCH as registrar - current trunk: > 1. FreeSwitch si simply modified (code doing NAT-ping is commented out :-) > 2. Directory is served through lighttpd and simple "C" binary doing one > trivial select. Lighttpd runs on the same machine as FS. When I move > lighhtpd to another machine, I cannot see any significat performance boost. > > Result: I can go up to the 470-500 reg/s. and FS is heavy overloaded and > retransmissions occurs. > > > Kamailio as registrar - 1.4.3. no TLS: > 1. Kamailio runs with usrloc db_mode 3 (no caching) > > Result: I can go up to the 3500-3700 reg/s. and Kamailio server is at > 0.3 load and all 8 cores are bellow 15 %. Without retransmissions. The > limit is DB throughput. > Just for "curiosity" I switched userloc to db_mode 2 (write back) and at > 5000 regs/s I stopped the sipp test, because I saw the bottle neck > becomes the server runnig sipp (very old P4 box). > > > Conclusion: > While I see amazing FreeSWITCH performance on INVITEs per seconds and > concurrent calls (another galaxy from * point of view :-), if you have > to handle lots of registrations per second, it is IMO better to use > Kamailio/OpenSIPS/SER as separate registrar and "propagate" users to FS > through SQL view. > > Hope this helps someone... > > Best regards, > > kokoska.rokoska > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090203/79c8824b/attachment-0002.html From Daniell at airg.com Tue Feb 3 18:21:07 2009 From: Daniell at airg.com (Daniel Liang) Date: Tue, 3 Feb 2009 18:21:07 -0800 Subject: [Freeswitch-users] Recording background music and voice is out of sync In-Reply-To: <022001c9862f$efd4b7d0$cf7e2770$@net> References: <019501c985ac$4f00ee60$ed02cb20$@net><4987E527.1040909@laposte.net> <022001c9862f$efd4b7d0$cf7e2770$@net> Message-ID: <0B02E756F603CC409EB553879B090CC80A23EB2F@HPEXCHVS01.exchange.airg> Hi, I was trying to record a background music with a user's voice at the same time. I did a playback and started recording. But the recorded user's voice and the background music is about 0.5 second out of sync. I also tried to use uuid_displace instead of playback, but I got the same result. I guess it was the transfer delay between freeswitch and the end user. Is there a way to avoid that? One of the solution that I can think of is to route the background music to the end user and then route it back to freeswitch and let freeswitch recorded user's voice and the routed music together. But I don't know how I can do that in freeswitch. Any idea? Thanks. Daniel From brian at freeswitch.org Tue Feb 3 18:35:42 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 3 Feb 2009 20:35:42 -0600 Subject: [Freeswitch-users] Recording background music and voice is out of sync In-Reply-To: <0B02E756F603CC409EB553879B090CC80A23EB2F@HPEXCHVS01.exchange.airg> References: <019501c985ac$4f00ee60$ed02cb20$@net><4987E527.1040909@laposte.net> <022001c9862f$efd4b7d0$cf7e2770$@net> <0B02E756F603CC409EB553879B090CC80A23EB2F@HPEXCHVS01.exchange.airg> Message-ID: <341AE5F8-20B2-4CF3-92EE-7311B3E71C7E@freeswitch.org> Can you show us an example of how you're doing this? Playback and Record aren't async so you'll need to show us how you're doing this. Also don't hijack threads you hit replay on the one "Re: [Freeswitch- users] FreeSwitch setup as a "Dumb" SBC" as the subject, deleted the subject and started a new body. That hijacks the thread and that can cause your problem to go ignored in some cases if people aren't interested in the thread topic depending on how their reader threads the emails. Please click new message and type freeswitch- users at lists.freeswitch.org in and then input your subject and body to start a new thread. Thanks, Brian West FreeSWITCH.org On Feb 3, 2009, at 8:21 PM, Daniel Liang wrote: > Hi, > > I was trying to record a background music with a user's voice at the > same time. I did a playback and started recording. But the recorded > user's voice and the background music is about 0.5 second out of > sync. I > also tried to use uuid_displace instead of playback, but I got the > same > result. From kawarod at laposte.net Tue Feb 3 22:35:37 2009 From: kawarod at laposte.net (rod) Date: Wed, 04 Feb 2009 10:35:37 +0400 Subject: [Freeswitch-users] FreeSwitch setup as a "Dumb" SBC In-Reply-To: <191c3a030902031434v1129d684j18b09a954d871f7b@mail.gmail.com> References: <4988C12E.1090109@post.cz> <191c3a030902031434v1129d684j18b09a954d871f7b@mail.gmail.com> Message-ID: <49893739.1090400@laposte.net> I did the test when I start looking at FS. With 10 000 files in conf/directory/default mounted as a ramdisk (if not in Ramdisk, the IO are too high) and an intel quad core q9550 (2.83Ghz) with 4GB RAM and the db also in Ramdisk, I was stuck at approx 150cps with a very high CPU usage. The version I used was 1.0.1, but not sure. Anthony Minessale wrote: > What does it look like if you serve the directory from the static xml > file out of curiosity. > > > On Tue, Feb 3, 2009 at 4:11 PM, kokoska.rokoska > > wrote: > > Ken Rice napsal(a): > ... > > > On Registrations we have experienced Registration/second rates > exceeding 150 > > registrations per second using mod_xml_curl to feed the users > directory. I > > suspect, this number can be greatly increased if we were to feed > directory > > with something that cut out the apache and php over head > > > > If someone interested I have few numbers on Registrar performance: > > DB server: > 2x Quad core E5345 @ 2.33GHz, 16 GiB RAM > Centos 5 x86_64, MySQL 5.0 > > Registrar server: > 2x Quad core E5345 @ 2.33GHz, 16 GiB RAM > Centos 5 x86_64 > > Tested using sipp with 10.000 and 30.000 "users". > > > FreeSWITCH as registrar - current trunk: > 1. FreeSwitch si simply modified (code doing NAT-ping is commented > out :-) > 2. Directory is served through lighttpd and simple "C" binary > doing one > trivial select. Lighttpd runs on the same machine as FS. When I move > lighhtpd to another machine, I cannot see any significat > performance boost. > > Result: I can go up to the 470-500 reg/s. and FS is heavy > overloaded and > retransmissions occurs. > > > Kamailio as registrar - 1.4.3. no TLS: > 1. Kamailio runs with usrloc db_mode 3 (no caching) > > Result: I can go up to the 3500-3700 reg/s. and Kamailio server is at > 0.3 load and all 8 cores are bellow 15 %. Without retransmissions. The > limit is DB throughput. > Just for "curiosity" I switched userloc to db_mode 2 (write back) > and at > 5000 regs/s I stopped the sipp test, because I saw the bottle neck > becomes the server runnig sipp (very old P4 box). > > > Conclusion: > While I see amazing FreeSWITCH performance on INVITEs per seconds and > concurrent calls (another galaxy from * point of view :-), if you have > to handle lots of registrations per second, it is IMO better to use > Kamailio/OpenSIPS/SER as separate registrar and "propagate" users > to FS > through SQL view. > > Hope this helps someone... > > Best regards, > > kokoska.rokoska > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From pmhshz at gmail.com Tue Feb 3 22:38:24 2009 From: pmhshz at gmail.com (shehzad p) Date: Tue, 3 Feb 2009 22:38:24 -0800 (PST) Subject: [Freeswitch-users] Freeswitch freezes on increasing call traffic In-Reply-To: <191c3a030901310850s553f8c11oc4e559a710def6d2@mail.gmail.com> References: <21701744.post@talk.nabble.com> <191c3a030901280601t3e185ec2m76295110e4888837@mail.gmail.com> <21729863.post@talk.nabble.com> <191c3a030901290831y2ff329b3maee6c1aafa3c68e2@mail.gmail.com> <21745312.post@talk.nabble.com> <21749375.post@talk.nabble.com> <191c3a030901300818i5d85a7aawa0c53b4e54892f40@mail.gmail.com> <21761523.post@talk.nabble.com> <191c3a030901310850s553f8c11oc4e559a710def6d2@mail.gmail.com> Message-ID: <21825226.post@talk.nabble.com> Hi anthony, I Modified the whole architecture of call routing system, Now after getting required routes, script exit and, control comes back to Dialplan, and call is bridged there, And call hangup, CDR is posted to cdr.php file (using xml_cdr). So now there is no blocking statement (bridge or anything like that) in current javascript, It return back control instantly. So, setting up all above architecture... First I tested FS 1.0.1 , It get crashed two times, in interval of 3 to 5 hours and simultaneous call of about 100 to 150. BT is the same as before... http://www.nabble.com/file/p21825226/bt_new_arch.txt bt_new_arch.txt Now I am also testing 1.0.3RC1, and post it back if any found. Thanks msp Clearly you have an issue with your javascript code. You have the Garbage collector blocking in every thread. Are you doing any endless loops in your code where you do not check session.ready() as a condition for continuing the script? any time session.ready() fails you must immediately exit. Are you using session.execute to execute long blocking operations like bridging many calls or entering a conference? You should avoid doing this as all the collective scripts on the system share a common Garbage Collector provided by the JS engine and it can lead to the exact issues you describe if the code is not properly designed. What else does you script do that are things provided by FS such as playing files and executing applications. -- View this message in context: http://www.nabble.com/Freeswitch-freezes-on-increasing-call-traffic-tp21701744p21825226.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From kawarod at laposte.net Tue Feb 3 22:59:48 2009 From: kawarod at laposte.net (rod) Date: Wed, 04 Feb 2009 10:59:48 +0400 Subject: [Freeswitch-users] FreeSwitch setup as a "Dumb" SBC In-Reply-To: <022001c9862f$efd4b7d0$cf7e2770$@net> References: <019501c985ac$4f00ee60$ed02cb20$@net> <4987E527.1040909@laposte.net> <022001c9862f$efd4b7d0$cf7e2770$@net> Message-ID: <49893CE4.7080300@laposte.net> Hi Adam, I detailed a bit more my previous mail on this page: http://wiki.freeswitch.org/wiki/SBC_Setup Round robin is managed by the carrierroute module. Carrierroute will reply based on the probability you defined for a route, so if you define 0.3 and 0.7 for the same prefix, your traffic will point to 2 different gateways with a probability of 30% for one and 70% for the others (If I understood well the behaviour of carrierroute). For forking, what I do is that carrierroute replies with a code and not an IP address. This code, is then used as a condition in FS and the dialplan matched could then propose serial or parallel forking (in the wiki, I detailed serial forking). The idea is that you could define many combination of GWs, eg: - code01: try IP_A then IP_B (serial) - code02: try IP_B then IP_A (serial) - code03: try IP_A and IP_C (parallel) this setup is working for me as I do not have 1000 of GWs but I need a big routing table (approx 160000). I'm sure it could be possible to use the failure route functionnality of carrierroute to define a new route when the first one failed without having to define code. The drawback of this method is that you can't define metrics/properties for a route (quality, cost, fax compliance...) in realtime, and this is where using/enhancing the native FS module mod_lcr could be better (I have no idea on how mod_lcr performs, I will give it a try). rod Adam Long wrote: > Hi Rod, > > Great info, Thanks! > Glad to see others are interested in the same concept. > My reasons for SER as routing core and implementation is slightly different > yet similar. > > I like your Redirect model, with that you are truly using your Kamailio as > route server only. I would imagine very scalable. > - Are you able to do any round robin, serial or parallel forking > with this? > - I wonder if multiple Contacts in the 302 response maybe with some > logic in FreeSwitch dialplan? > If so I think your design is a bit more efficient than mine as it keeps SER > out of the call path. > > My design is little different.. it is more of a "Stateful" setup. With SER > staying in call path and FreeSwitch at Edge. > I do this to enable Serial Forking to a series of SBCs (FreeSwitch) geo > distributed, when one of the branches is congested it > forks to the next SBC (route). > > The FreeSwitch guys are probably right tho... with mod_easyroute and mod_lcr > we could probably implement all of this in FreeSwitch without SER. > I would be curious to know if anyone is doing something similar at high > volumes and what sort of concurrency and cps they > are able to achieve. > > I am a Perl and C# guy, I thought about implementing a mod_manged_lcr with > memcached support. > Memcache support would prob boost the scalability by a factor of 10 at > least. > > I will let you know if I end up developing a high performance FreeSwitch > route module. > Right now I use memcache in a OpenSIPS perl script for my route caching and > its incredibly fast > and clusters well. > > It actually might be easier to add memcached support to mod_lcr and > mod_easyroute but im not real strong in C/C++ > > I'll jump on IRC later and chat with some of the experts on this as I know > memcache has been discussed before. > I'd be curious to know if any progress has been made there already. > > > Regards, > -Adam > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of rod > Sent: Tuesday, February 03, 2009 1:33 AM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] FreeSwitch setup as a "Dumb" SBC > > Hi Adam, > > I'm in the process of using FS as a SBC. For the route lookup, I do it > using OpenSER carrierroute, without having to flow through > SBC---Openser---SBC. I'm using carrierroute at this time cause I need > more than 200 000 routing entries and carrierroute has been tested with > twice this number. > > Here is the setup: > > - install openser and carrierroute and make openser listening on > 127.0.0.1:5062 (for example) on your SBC > - populate carrierroute table > > What I do to use carrierroute module from FS is to use a specific > X-header (X-LOOKUP). > > In the dialplan, in the default context, I have something like this: > > > > > > data="sofia/internal/${sip_req_user}@127.0.0.1:5062"/> > data="sip_h_X-ROUTE=${sip_redirect_contact_host_0}"/> > > > > > The process is simple: > the export "sip_h_X-ROUTE=LOOKUP" had a sip header X-ROUTE=LOOKUP > then I bridge the call to 127.0.0.1:5062 (openser process) > > In openser I have a route block that checks the presence of header > LOOKUP and openser sends a "604: unable to route call" if the prefix is > not found, or a "302: with the IP of the gateway found" > > In FS, you can get the IP using the variable > "${sip_redirect_contact_host_0}". Then I transfer this to the context > ROUTING, where the check condition is based on the LOOKUP header that > has been rewritten with this variable. > > I will document all this setup (installation of openser/carrierroute and > config file of FS and openser) on a wiki page I start writing yesterday, > so please be indulgent and patient. > The next step is to test the scalability of this. > > I'm a very bad programmer, so that's the only way for me to contribute > to FS, and as I see many people interested for an SBC setup, I think it > could be great if we share our work/knowlegde. > > The wiki page is there: > http://wiki.freeswitch.org/wiki/SBC_Setup > > regards, > rod. > > > > > > Adam Long wrote: > >> Hi Guys, >> >> I've been working at setting up a couple of FreeSwitch nodes as a >> topology hiding SBCs that handles both ingress traffic from my >> >> providers/peers and pass traffic up to an openser router that then >> routes call across the cluster of SBCs through which they reach the >> destination. >> >> I have OpenSIPS/SER setup doing DB route lookups and ENUM with >> LCR/Serial forking etc. >> >> My question is what would be the best way to send a call out to a >> destination choosen by the OpenSER router? >> >> For example: >> >> SIP Provider -- > SBC --- > OpenSER ---- ( route lookup returns >> 123.123.123.4 as dest ) -- > SBC --- > 123.123.123.4 >> >> I was thinking something along the lines of adding a "X-Route-To: >> +1NXXNXXXXXX@ 123.123.123.4" with openser >> >> and then something like this in the SBC. >> >> >> >> >> >> >> >> >> >> >> >> Is this a wise approach, is there anything I could do to do this better? >> >> I'd like to keep the logic in the SBCs as simple as possible. >> >> I am pretty familiar with SIP but my knowledge fades when it gets into >> the nitty gritty of routing. ie the Contact: and Via: headers >> >> and all that good stuff. >> >> I should also state I have two profiles defined one for the >> internal/private "core" network and one for the outside "external" >> network. >> >> Any thoughts on this at all would be greatly appreciated. >> >> Am I missing something in the SIP spec that would allow for this is a >> standardized way? >> >> Regards, >> >> -Adam >> >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > From kawarod at laposte.net Tue Feb 3 23:09:52 2009 From: kawarod at laposte.net (rod) Date: Wed, 04 Feb 2009 11:09:52 +0400 Subject: [Freeswitch-users] FreeSwitch setup as a "Dumb" SBC In-Reply-To: <022001c9862f$efd4b7d0$cf7e2770$@net> References: <019501c985ac$4f00ee60$ed02cb20$@net> <4987E527.1040909@laposte.net> <022001c9862f$efd4b7d0$cf7e2770$@net> Message-ID: <49893F40.9040904@laposte.net> One more thing, I worked on a setup like yours: - Kamailio as a registrar that do the routing decision - FS as a SBC What you have to do is just append an header with Kamailio and send the invite to your FS server using something like that (use of pseudo variables in Kamailio): #------------------------------------------- # PREPARE ROUTING USING REWRITING OF DOMAIN #------------------------------------------- if (is_method("INVITE") && from_uri==myself && src_ip!=10.10.10.254){ if(!cr_route("default", "0", "$rU", "$rU", "call_id")){ xlog("$ci CALLEE ROUTING FAILED: no route found"); sl_send_reply("604", "Unable to route this call"); exit; } else { xlog("$ci Route found for $rU via $rd"); } } # ----------------------------------------------------------------- # Route to FREESWITCH using domain rewriting applied above for LCR # ----------------------------------------------------------------- xlog("$ci ROUTE: $rd"); append_hf("X-ROUTE: $rd\r\n"); rewritehostport("10.10.10.254:5062"); # there you have to distribute the invite to your FS servers, take a look at the dispatcher module Using that, the FS server receiving the Invite, just need to parse the X-ROUTE header and route the call, without having to resend the call to a Kamailio server. I think you can adapt this scenario to your perl script using variable exportation and append_hf function. rod. Adam Long wrote: > Hi Rod, > > Great info, Thanks! > Glad to see others are interested in the same concept. > My reasons for SER as routing core and implementation is slightly different > yet similar. > > I like your Redirect model, with that you are truly using your Kamailio as > route server only. I would imagine very scalable. > - Are you able to do any round robin, serial or parallel forking > with this? > - I wonder if multiple Contacts in the 302 response maybe with some > logic in FreeSwitch dialplan? > If so I think your design is a bit more efficient than mine as it keeps SER > out of the call path. > > My design is little different.. it is more of a "Stateful" setup. With SER > staying in call path and FreeSwitch at Edge. > I do this to enable Serial Forking to a series of SBCs (FreeSwitch) geo > distributed, when one of the branches is congested it > forks to the next SBC (route). > > The FreeSwitch guys are probably right tho... with mod_easyroute and mod_lcr > we could probably implement all of this in FreeSwitch without SER. > I would be curious to know if anyone is doing something similar at high > volumes and what sort of concurrency and cps they > are able to achieve. > > I am a Perl and C# guy, I thought about implementing a mod_manged_lcr with > memcached support. > Memcache support would prob boost the scalability by a factor of 10 at > least. > > I will let you know if I end up developing a high performance FreeSwitch > route module. > Right now I use memcache in a OpenSIPS perl script for my route caching and > its incredibly fast > and clusters well. > > It actually might be easier to add memcached support to mod_lcr and > mod_easyroute but im not real strong in C/C++ > > I'll jump on IRC later and chat with some of the experts on this as I know > memcache has been discussed before. > I'd be curious to know if any progress has been made there already. > > > Regards, > -Adam > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of rod > Sent: Tuesday, February 03, 2009 1:33 AM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] FreeSwitch setup as a "Dumb" SBC > > Hi Adam, > > I'm in the process of using FS as a SBC. For the route lookup, I do it > using OpenSER carrierroute, without having to flow through > SBC---Openser---SBC. I'm using carrierroute at this time cause I need > more than 200 000 routing entries and carrierroute has been tested with > twice this number. > > Here is the setup: > > - install openser and carrierroute and make openser listening on > 127.0.0.1:5062 (for example) on your SBC > - populate carrierroute table > > What I do to use carrierroute module from FS is to use a specific > X-header (X-LOOKUP). > > In the dialplan, in the default context, I have something like this: > > > > > > data="sofia/internal/${sip_req_user}@127.0.0.1:5062"/> > data="sip_h_X-ROUTE=${sip_redirect_contact_host_0}"/> > > > > > The process is simple: > the export "sip_h_X-ROUTE=LOOKUP" had a sip header X-ROUTE=LOOKUP > then I bridge the call to 127.0.0.1:5062 (openser process) > > In openser I have a route block that checks the presence of header > LOOKUP and openser sends a "604: unable to route call" if the prefix is > not found, or a "302: with the IP of the gateway found" > > In FS, you can get the IP using the variable > "${sip_redirect_contact_host_0}". Then I transfer this to the context > ROUTING, where the check condition is based on the LOOKUP header that > has been rewritten with this variable. > > I will document all this setup (installation of openser/carrierroute and > config file of FS and openser) on a wiki page I start writing yesterday, > so please be indulgent and patient. > The next step is to test the scalability of this. > > I'm a very bad programmer, so that's the only way for me to contribute > to FS, and as I see many people interested for an SBC setup, I think it > could be great if we share our work/knowlegde. > > The wiki page is there: > http://wiki.freeswitch.org/wiki/SBC_Setup > > regards, > rod. > > > > > > Adam Long wrote: > >> Hi Guys, >> >> I've been working at setting up a couple of FreeSwitch nodes as a >> topology hiding SBCs that handles both ingress traffic from my >> >> providers/peers and pass traffic up to an openser router that then >> routes call across the cluster of SBCs through which they reach the >> destination. >> >> I have OpenSIPS/SER setup doing DB route lookups and ENUM with >> LCR/Serial forking etc. >> >> My question is what would be the best way to send a call out to a >> destination choosen by the OpenSER router? >> >> For example: >> >> SIP Provider -- > SBC --- > OpenSER ---- ( route lookup returns >> 123.123.123.4 as dest ) -- > SBC --- > 123.123.123.4 >> >> I was thinking something along the lines of adding a "X-Route-To: >> +1NXXNXXXXXX@ 123.123.123.4" with openser >> >> and then something like this in the SBC. >> >> >> >> >> >> >> >> >> >> >> >> Is this a wise approach, is there anything I could do to do this better? >> >> I'd like to keep the logic in the SBCs as simple as possible. >> >> I am pretty familiar with SIP but my knowledge fades when it gets into >> the nitty gritty of routing. ie the Contact: and Via: headers >> >> and all that good stuff. >> >> I should also state I have two profiles defined one for the >> internal/private "core" network and one for the outside "external" >> network. >> >> Any thoughts on this at all would be greatly appreciated. >> >> Am I missing something in the SIP spec that would allow for this is a >> standardized way? >> >> Regards, >> >> -Adam >> >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > From jsokulski at dotsystems.pl Wed Feb 4 00:51:34 2009 From: jsokulski at dotsystems.pl (Jacek Sokulski) Date: Wed, 04 Feb 2009 09:51:34 +0100 Subject: [Freeswitch-users] origainate through sofia gateway In-Reply-To: <1b46b4e80902030931y6a1679afw12ec37596ba9d77d@mail.gmail.com> References: <1233668179.5354.15.camel@dotw1126.dotsystems.pl> <1b46b4e80902030920p521d6079s4c9b78d8d39a924a@mail.gmail.com> <5E2B217D-8C85-48DB-800A-535A7FAAE24B@freeswitch.org> <1b46b4e80902030931y6a1679afw12ec37596ba9d77d@mail.gmail.com> Message-ID: <1233737494.5405.12.camel@dotw1126.dotsystems.pl> We have tried setting both effective_caller_id_number and origination_caller_id_number: session1.originate(session1,"{origination_caller_id_number=fixed0248b}sofia/gateway/halonet/0225490317",15); but the problem still exists. The solution we have found for the case when we originate two calls, local and external, is as follow: session1 = new Session(); session1.originate(session1,"user/1003 at 192.168.1.122",15);//local if(session1.ready()) { session1.execute("execute_extension","00930691688627 XML default");//external } so the external call goes through the dialplan. It does not work if both calls are external. One possible solution could be to pass the originating call through dialplan (loopback?) but we have not managed to figure out how to do it. Thanks Jacek Dnia 03-02-2009, wto o godzinie 14:31 -0300, Nicolas Brenner pisze: > Oops! Well, fortunately I don't use that voip provider anymore (nor the script). > > Thanks Brian. > > Nicolas > > On Tue, Feb 3, 2009 at 2:25 PM, Brian West wrote: > > YOU should NEVER use this method or call setCallerData at all you > > should use the correct methods to override the callerid. > > > > If its a B-Leg born from an A-Leg you use these on the on the A-Leg: > > > > http://wiki.freeswitch.org/wiki/Channel_Variables#effective_caller_id_name > > http://wiki.freeswitch.org/wiki/Channel_Variables#effective_caller_id_number > > > > If you're originating you use this: > > > > http://wiki.freeswitch.org/wiki/Channel_Variables#origination_caller_id_name > > http://wiki.freeswitch.org/wiki/Channel_Variables#origination_caller_id_number > > > > /b > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From kokoska.rokoska at post.cz Wed Feb 4 02:31:41 2009 From: kokoska.rokoska at post.cz (kokoska rokoska) Date: Wed, 04 Feb 2009 11:31:41 +0100 Subject: [Freeswitch-users] FreeSwitch setup as a "Dumb" SBC In-Reply-To: <191c3a030902031434v1129d684j18b09a954d871f7b@mail.gmail.com> References: <4988C12E.1090109@post.cz> <191c3a030902031434v1129d684j18b09a954d871f7b@mail.gmail.com> Message-ID: <49896E8D.3010609@post.cz> Anthony Minessale napsal(a): > What does it look like if you serve the directory from the static xml > file out of curiosity. > Well, I write all user infos into static xml files loaded at startup :-) For the first try (without "tuning", see below) I can't go beyond 220 reg/s - it is just about 10-12 % higher rate then with "lighty" etc. BTW: Lighty alone can serve about 2600-2700 "dynamic directory xml files from DB" per second - tested with "ab". The only difference (but big one :-) is that CPU utilization is below 15% on all cores and load is about 0.2, so machine is idle :-) It tells me, that somthing is wron with my setup :-) The only optimizations done are: 1. No logging 2. FS in "high priority" mode 3. "ulimits" applied Next (today evenings - tommorow mornings, don't know) I try to "tune" FS for better preformance - as I did with previous test last month: 1. Move FS internal SQL light to ramdisk - I'm not sure if it helps, because OS caches all HDD reads/writes, but if SQL light forces sync after every DB update/insert it can make sense - I try it. May be I move all FS dir to ramdisk. 2. Slightly change mod_sofia to disable NAT ping loop (unnecessary DB operations) and, mainly, disable retrieving and sending of NOTIFY messages containing VM info. I'll look into my notes to see what I have done before and do the same... Be patient please, as soon as I have the results, I post them here :-) Best regards, kokoska.rokoska > On Tue, Feb 3, 2009 at 4:11 PM, kokoska.rokoska > wrote: > > Ken Rice napsal(a): > ... > > > On Registrations we have experienced Registration/second rates > exceeding 150 > > registrations per second using mod_xml_curl to feed the users > directory. I > > suspect, this number can be greatly increased if we were to feed > directory > > with something that cut out the apache and php over head > > > > If someone interested I have few numbers on Registrar performance: > > DB server: > 2x Quad core E5345 @ 2.33GHz, 16 GiB RAM > Centos 5 x86_64, MySQL 5.0 > > Registrar server: > 2x Quad core E5345 @ 2.33GHz, 16 GiB RAM > Centos 5 x86_64 > > Tested using sipp with 10.000 and 30.000 "users". > > > FreeSWITCH as registrar - current trunk: > 1. FreeSwitch si simply modified (code doing NAT-ping is commented > out :-) > 2. Directory is served through lighttpd and simple "C" binary doing one > trivial select. Lighttpd runs on the same machine as FS. When I move > lighhtpd to another machine, I cannot see any significat performance > boost. > > Result: I can go up to the 470-500 reg/s. and FS is heavy overloaded and > retransmissions occurs. > > > Kamailio as registrar - 1.4.3. no TLS: > 1. Kamailio runs with usrloc db_mode 3 (no caching) > > Result: I can go up to the 3500-3700 reg/s. and Kamailio server is at > 0.3 load and all 8 cores are bellow 15 %. Without retransmissions. The > limit is DB throughput. > Just for "curiosity" I switched userloc to db_mode 2 (write back) and at > 5000 regs/s I stopped the sipp test, because I saw the bottle neck > becomes the server runnig sipp (very old P4 box). > > > Conclusion: > While I see amazing FreeSWITCH performance on INVITEs per seconds and > concurrent calls (another galaxy from * point of view :-), if you have > to handle lots of registrations per second, it is IMO better to use > Kamailio/OpenSIPS/SER as separate registrar and "propagate" users to FS > through SQL view. > > Hope this helps someone... > > Best regards, > > kokoska.rokoska > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From Claudio.Cavalera at italtel.it Wed Feb 4 03:03:12 2009 From: Claudio.Cavalera at italtel.it (Cavalera Claudio Luigi) Date: Wed, 4 Feb 2009 12:03:12 +0100 Subject: [Freeswitch-users] 16 threads didn't exit Message-ID: Hello list, I'm trying to track down a seg fault issue with a fs Revision: 11489 Here is the backtrace pastebin: http://pastebin.freeswitch.org/7009 but before digging the dump I would like to understand: am I the only one having error like this in fs console: "Error in my_thread_global_end(): 16 threads didn't exit" I'm asking this because googling around did not take me to much relation between this error and fs. In fact as you can see the error does not have the usual fs logging format with date time and logging level, it's just a yellow line printed out in console. It seems to me related on php and mysql from what I've read here http://forums.mysql.com/read.php?10,153077,206930#msg-206930 It's possible that my fs segfaults not because of this error at all, but I just wanted to inform you asap in case someone could face similar issues. I've enabled crash-protection in switch.conf.xml although I'm not sure what it does. Best Regards, Claudio Internet Email Confidentiality Footer ----------------------------------------------------------------------------------------------------- La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. ----------------------------------------------------------------------------------------------------- From sias at cpdata.co.za Wed Feb 4 03:49:55 2009 From: sias at cpdata.co.za (Sias Mey) Date: Wed, 4 Feb 2009 13:49:55 +0200 Subject: [Freeswitch-users] 16 threads didn't exit In-Reply-To: References: Message-ID: <20090204114955.GD6752@cpdata.co.za> I have seen that error myself, however I assumed it was due te me hanging up other cals from the api_hangup_hook of a related call. I use this to set a "master" call in a conference so that if it hangs up all calls in the conference hangup. On Wed, Feb 04, 2009 at 12:03:12PM +0100, Cavalera Claudio Luigi wrote: > Hello list, > I'm trying to track down a seg fault issue with a fs Revision: 11489 > Here is the backtrace pastebin: > http://pastebin.freeswitch.org/7009 > > but before digging the dump I would like to understand: am I the only > one having error like this in fs console: > "Error in my_thread_global_end(): 16 threads didn't exit" > > I'm asking this because googling around did not take me to much relation > between this error and fs. > In fact as you can see the error does not have the usual fs logging > format with date time and logging level, it's just a yellow line printed > out in console. > It seems to me related on php and mysql from what I've read here > http://forums.mysql.com/read.php?10,153077,206930#msg-206930 > > It's possible that my fs segfaults not because of this error at all, but > I just wanted to inform you asap in case someone could face similar > issues. > > I've enabled crash-protection in switch.conf.xml although I'm not sure > what it does. > > Best Regards, > Claudio > > > Internet Email Confidentiality Footer > ----------------------------------------------------------------------------------------------------- > La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. > > This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. > ----------------------------------------------------------------------------------------------------- > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From jonas.gauffin at gmail.com Wed Feb 4 04:26:04 2009 From: jonas.gauffin at gmail.com (Jonas Gauffin) Date: Wed, 4 Feb 2009 13:26:04 +0100 Subject: [Freeswitch-users] gateway Message-ID: Hello I'm trying to make outbound calls through my gateway provider. My calls got rejected and I asked them why. Apparently I need to use 5060 as source port, since they validate both my IP and the port that the messages come from. Is this possible with freeswitch? If so, what config settings should I set? Regards, Jonas -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090204/68095d5b/attachment-0002.html From Claudio.Cavalera at italtel.it Wed Feb 4 05:28:48 2009 From: Claudio.Cavalera at italtel.it (Cavalera Claudio Luigi) Date: Wed, 4 Feb 2009 14:28:48 +0100 Subject: [Freeswitch-users] 16 threads didn't exit In-Reply-To: <20090204114955.GD6752@cpdata.co.za> Message-ID: freeswitch-users-bounces at lists.freeswitch.org wrote: > I have seen that error myself, however I assumed it was due te me > hanging up other cals from the api_hangup_hook of a related call. > > I use this to set a "master" call in a conference so that if > it hangs up > all calls in the conference hangup. > Thanks Sias for this info, if you are able to reproduce this message in a systematic way please let me know so that I can reproduce it myself. I've searched for the "threads didn't exit" message in fs code and I did not found it. I think it comes from mysql and I'm not sure it's even related to my segfault. Best regards, Claudio Internet Email Confidentiality Footer ----------------------------------------------------------------------------------------------------- La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. ----------------------------------------------------------------------------------------------------- From anthony.minessale at gmail.com Wed Feb 4 05:51:06 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 4 Feb 2009 07:51:06 -0600 Subject: [Freeswitch-users] Freeswitch freezes on increasing call traffic In-Reply-To: <21825226.post@talk.nabble.com> References: <21701744.post@talk.nabble.com> <191c3a030901280601t3e185ec2m76295110e4888837@mail.gmail.com> <21729863.post@talk.nabble.com> <191c3a030901290831y2ff329b3maee6c1aafa3c68e2@mail.gmail.com> <21745312.post@talk.nabble.com> <21749375.post@talk.nabble.com> <191c3a030901300818i5d85a7aawa0c53b4e54892f40@mail.gmail.com> <21761523.post@talk.nabble.com> <191c3a030901310850s553f8c11oc4e559a710def6d2@mail.gmail.com> <21825226.post@talk.nabble.com> Message-ID: <191c3a030902040551w5d867deta55c98bd3a3c03f8@mail.gmail.com> If you still get a crash on SVN trunk please post the bt even if you think it's the same, since it won't be exactly the same, the line numbers etc will be accurate with our development code making it easier to debug. On Wed, Feb 4, 2009 at 12:38 AM, shehzad p wrote: > > Hi anthony, > > I Modified the whole architecture of call routing system, > Now after getting required routes, script exit and, > control comes back to Dialplan, and call is bridged there, > And call hangup, CDR is posted to cdr.php file (using xml_cdr). > > So now there is no blocking statement (bridge or anything like that) in > current javascript, It return back control instantly. > > So, setting up all above architecture... > First I tested FS 1.0.1 , It get crashed two times, in interval of 3 to 5 > hours and simultaneous call of about 100 to 150. > BT is the same as before... > http://www.nabble.com/file/p21825226/bt_new_arch.txt bt_new_arch.txt > > Now I am also testing 1.0.3RC1, and post it back if any found. > > Thanks > msp > > > Clearly you have an issue with your javascript code. > > You have the Garbage collector blocking in every thread. > > Are you doing any endless loops in your code where you do not check > session.ready() as a condition for > continuing the script? > > any time session.ready() fails you must immediately exit. > > Are you using session.execute to execute long blocking operations like > bridging many calls or entering a conference? > You should avoid doing this as all the collective scripts on the system > share a common Garbage Collector provided by the > JS engine and it can lead to the exact issues you describe if the code is > not properly designed. > > What else does you script do that are things provided by FS such as playing > files and executing applications. > > > > > -- > View this message in context: > http://www.nabble.com/Freeswitch-freezes-on-increasing-call-traffic-tp21701744p21825226.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090204/deb1ce9a/attachment-0002.html From anthony.minessale at gmail.com Wed Feb 4 06:09:07 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 4 Feb 2009 08:09:07 -0600 Subject: [Freeswitch-users] origainate through sofia gateway In-Reply-To: <1233737494.5405.12.camel@dotw1126.dotsystems.pl> References: <1233668179.5354.15.camel@dotw1126.dotsystems.pl> <1b46b4e80902030920p521d6079s4c9b78d8d39a924a@mail.gmail.com> <5E2B217D-8C85-48DB-800A-535A7FAAE24B@freeswitch.org> <1b46b4e80902030931y6a1679afw12ec37596ba9d77d@mail.gmail.com> <1233737494.5405.12.camel@dotw1126.dotsystems.pl> Message-ID: <191c3a030902040609s600798c8t720950a2e13ee05a@mail.gmail.com> Where did you learn how to use js this way? session.originate is being misused here and is depricated and may be removed. the first arg to session.originate is either undefined or a *different* session (the a leg) session1 = new Session(); session1.originate(undefined, "{ignore_early_media=true}user/ 1008 at 192.168.1.122"); session1.setVariable("effective_caller_id_number","fixed0248b"); //once you have session1 when you originate session2 you pass session1 as the arg // the effective_caller_id is taken from session1 session2 = new Session(); session2.originate(session1, "sofia/gateway/halonet/0225490317"); Anyway this whole code is depricated in favor of this: session1 = new Session("{ignore_early_media=true}user/1008 at 192.168.1.122"); if (session1.ready()) { session1.setVariable("effective_caller_id_number","fixed0248b"); session2 = new Session("sofia/gateway/halonet/0225490317", session1); } and could be further refactored down to this: session1 = new Session("{ignore_early_media=true}user/1008 at 192.168.1.122"); if (session1.ready()) { session1.setVariable("effective_caller_id_number","fixed0248b"); session1.execute("bridge", "sofia/gateway/halonet/0225490317"); } or down to this one line of code that will setup the call detached from the script and exit. var result = apiExecute("originate", "{effective_caller_id_number=fixed0248b,origination_caller_id_number=1000,ignore_early_media=true}user/ 1008 at 192.168.1.122 bridge:sofia/gateway/halonet/0225490317 inline"); if you dont care about the result and want to exit even before the call is completed. var result = apiExecute("bgapi", "originate {effective_caller_id_number=fixed0248b,origination_caller_id_number=1000,ignore_early_media=true}user/ 1008 at 192.168.1.122 bridge:sofia/gateway/halonet/0225490317 inline"); On Wed, Feb 4, 2009 at 2:51 AM, Jacek Sokulski wrote: > > We have tried setting both effective_caller_id_number and > origination_caller_id_number: > > > session1.originate(session1,"{origination_caller_id_number=fixed0248b}sofia/gateway/halonet/0225490317",15); > but the problem still exists. The solution we have found for the case > when we originate two calls, local and external, is as follow: > > session1 = new Session(); > session1.originate(session1,"user/1003 at 192.168.1.122",15);//local > if(session1.ready()) { > session1.execute("execute_extension","00930691688627 XML > default");//external > } > > so the external call goes through the dialplan. > It does not work if both calls are external. One possible solution could be > to pass the originating call through dialplan (loopback?) but we have not > managed > to figure out how to do it. > > Thanks > Jacek > > Dnia 03-02-2009, wto o godzinie 14:31 -0300, Nicolas Brenner pisze: > > Oops! Well, fortunately I don't use that voip provider anymore (nor the > script). > > > > Thanks Brian. > > > > Nicolas > > > > On Tue, Feb 3, 2009 at 2:25 PM, Brian West wrote: > > > YOU should NEVER use this method or call setCallerData at all you > > > should use the correct methods to override the callerid. > > > > > > If its a B-Leg born from an A-Leg you use these on the on the A-Leg: > > > > > > > http://wiki.freeswitch.org/wiki/Channel_Variables#effective_caller_id_name > > > > http://wiki.freeswitch.org/wiki/Channel_Variables#effective_caller_id_number > > > > > > If you're originating you use this: > > > > > > > http://wiki.freeswitch.org/wiki/Channel_Variables#origination_caller_id_name > > > > http://wiki.freeswitch.org/wiki/Channel_Variables#origination_caller_id_number > > > > > > /b > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090204/d81ea72e/attachment-0002.html From anthony.minessale at gmail.com Wed Feb 4 06:14:52 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 4 Feb 2009 08:14:52 -0600 Subject: [Freeswitch-users] gateway In-Reply-To: References: Message-ID: <191c3a030902040614q32c05f02le7e8f41dd70d1fc3@mail.gmail.com> if you are not behind any nat then as long as you run your profile on 5060, the source port on every packet will be 5060. If you *are* behind nat the nat mapping will pick a random port unless you have a firewall that allows you to set specific mappings. On Wed, Feb 4, 2009 at 6:26 AM, Jonas Gauffin wrote: > Hello > I'm trying to make outbound calls through my gateway provider. > My calls got rejected and I asked them why. > > Apparently I need to use 5060 as source port, since they validate both my > IP and the port that the messages come from. > Is this possible with freeswitch? If so, what config settings should I set? > > Regards, > Jonas > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090204/84ddc63c/attachment-0002.html From Claudio.Cavalera at italtel.it Wed Feb 4 06:27:31 2009 From: Claudio.Cavalera at italtel.it (Cavalera Claudio Luigi) Date: Wed, 4 Feb 2009 15:27:31 +0100 Subject: [Freeswitch-users] Errors compiling trunk on a fresh system Message-ID: Hello, I'm trying to compile a brand new fs on a clean system. Revision: 11630 After the usual ./bootstrap.sh ./configure --enable-core-odbc-support I was getting this at make http://pastebin.freeswitch.org/7011 so i cd into utils under libs/sofia-sip/utils issued a doxygen -u but still getting this http://pastebin.freeswitch.org/7012 any hint on what could be? Best Regards, Claudio Internet Email Confidentiality Footer ----------------------------------------------------------------------------------------------------- La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. 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If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. ----------------------------------------------------------------------------------------------------- From jsokulski at dotsystems.pl Wed Feb 4 06:46:07 2009 From: jsokulski at dotsystems.pl (Jacek Sokulski) Date: Wed, 04 Feb 2009 15:46:07 +0100 Subject: [Freeswitch-users] origainate through sofia gateway In-Reply-To: <191c3a030902040609s600798c8t720950a2e13ee05a@mail.gmail.com> References: <1233668179.5354.15.camel@dotw1126.dotsystems.pl> <1b46b4e80902030920p521d6079s4c9b78d8d39a924a@mail.gmail.com> <5E2B217D-8C85-48DB-800A-535A7FAAE24B@freeswitch.org> <1b46b4e80902030931y6a1679afw12ec37596ba9d77d@mail.gmail.com> <1233737494.5405.12.camel@dotw1126.dotsystems.pl> <191c3a030902040609s600798c8t720950a2e13ee05a@mail.gmail.com> Message-ID: <1233758767.5405.22.camel@dotw1126.dotsystems.pl> Thanks Anthony, the js snippets are very instructive. A couple of points: 1. The code with apiExecute does not work (local phone is connected, but after picking up it hungs up immediately), other examples are working fine. 2. It does not show how initiate external call without existing session. 3. How can one pass the call through dialplan? Jacek PS. we got the code probable from wiki or from this mialing list. Dnia 04-02-2009, ?ro o godzinie 08:09 -0600, Anthony Minessale pisze: > Where did you learn how to use js this way? > session.originate is being misused here and is depricated and may be > removed. > > the first arg to session.originate is either undefined or a > *different* session (the a leg) > > session1 = new Session(); > session1.originate(undefined, > "{ignore_early_media=true}user/1008 at 192.168.1.122"); > > session1.setVariable("effective_caller_id_number","fixed0248b"); > > //once you have session1 when you originate session2 you pass session1 > as the arg > // the effective_caller_id is taken from session1 > > session2 = new Session(); > session2.originate(session1, "sofia/gateway/halonet/0225490317"); > > Anyway this whole code is depricated in favor of this: > > session1 = new > Session("{ignore_early_media=true}user/1008 at 192.168.1.122"); > > if (session1.ready()) { > session1.setVariable("effective_caller_id_number","fixed0248b"); > session2 = new Session("sofia/gateway/halonet/0225490317", > session1); > } > > and could be further refactored down to this: > > session1 = new > Session("{ignore_early_media=true}user/1008 at 192.168.1.122"); > if (session1.ready()) { > session1.setVariable("effective_caller_id_number","fixed0248b"); > session1.execute("bridge", "sofia/gateway/halonet/0225490317"); > } > > or down to this one line of code that will setup the call detached > from the script and exit. > > var result = apiExecute("originate", > "{effective_caller_id_number=fixed0248b,origination_caller_id_number=1000,ignore_early_media=true}user/1008 at 192.168.1.122 bridge:sofia/gateway/halonet/0225490317 inline"); > > if you dont care about the result and want to exit even before the > call is completed. > > var result = apiExecute("bgapi", "originate > {effective_caller_id_number=fixed0248b,origination_caller_id_number=1000,ignore_early_media=true}user/1008 at 192.168.1.122 bridge:sofia/gateway/halonet/0225490317 inline"); > > > > On Wed, Feb 4, 2009 at 2:51 AM, Jacek Sokulski > wrote: > > We have tried setting both effective_caller_id_number and > origination_caller_id_number: > > session1.originate(session1,"{origination_caller_id_number=fixed0248b}sofia/gateway/halonet/0225490317",15); > but the problem still exists. The solution we have found for > the case > when we originate two calls, local and external, is as follow: > > session1 = new Session(); > session1.originate(session1,"user/1003 at 192.168.1.122",15);//local > if(session1.ready()) { > session1.execute("execute_extension","00930691688627 XML > default");//external > } > > so the external call goes through the dialplan. > It does not work if both calls are external. One possible > solution could be > to pass the originating call through dialplan (loopback?) but > we have not managed > to figure out how to do it. > > Thanks > Jacek > > Dnia 03-02-2009, wto o godzinie 14:31 -0300, Nicolas Brenner > pisze: > > > Oops! Well, fortunately I don't use that voip provider > anymore (nor the script). > > > > Thanks Brian. > > > > Nicolas > > > > On Tue, Feb 3, 2009 at 2:25 PM, Brian West > wrote: > > > YOU should NEVER use this method or call setCallerData at > all you > > > should use the correct methods to override the callerid. > > > > > > If its a B-Leg born from an A-Leg you use these on the on > the A-Leg: > > > > > > > http://wiki.freeswitch.org/wiki/Channel_Variables#effective_caller_id_name > > > > http://wiki.freeswitch.org/wiki/Channel_Variables#effective_caller_id_number > > > > > > If you're originating you use this: > > > > > > > http://wiki.freeswitch.org/wiki/Channel_Variables#origination_caller_id_name > > > > http://wiki.freeswitch.org/wiki/Channel_Variables#origination_caller_id_number > > > > > > /b > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From jonas.gauffin at gmail.com Wed Feb 4 07:20:42 2009 From: jonas.gauffin at gmail.com (Jonas Gauffin) Date: Wed, 4 Feb 2009 16:20:42 +0100 Subject: [Freeswitch-users] gateway In-Reply-To: <191c3a030902040614q32c05f02le7e8f41dd70d1fc3@mail.gmail.com> References: <191c3a030902040614q32c05f02le7e8f41dd70d1fc3@mail.gmail.com> Message-ID: I'm behind NAT. Is it FS that picks the random port, or the FW? I've mapped port 5060 to the freeswitch ip in my FW. On Wed, Feb 4, 2009 at 3:14 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > if you are not behind any nat then as long as you run your profile on 5060, > the source port on every packet will be 5060. > If you *are* behind nat the nat mapping will pick a random port unless you > have a firewall that allows you to set specific mappings. > > > On Wed, Feb 4, 2009 at 6:26 AM, Jonas Gauffin wrote: > >> Hello >> I'm trying to make outbound calls through my gateway provider. >> My calls got rejected and I asked them why. >> >> Apparently I need to use 5060 as source port, since they validate both my >> IP and the port that the messages come from. >> Is this possible with freeswitch? If so, what config settings should I >> set? >> >> Regards, >> Jonas >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090204/393aa237/attachment-0002.html From wasim at convergence.pk Wed Feb 4 07:25:40 2009 From: wasim at convergence.pk (Wasim Baig) Date: Wed, 4 Feb 2009 20:25:40 +0500 Subject: [Freeswitch-users] gateway In-Reply-To: References: <191c3a030902040614q32c05f02le7e8f41dd70d1fc3@mail.gmail.com> Message-ID: On Wed, Feb 4, 2009 at 8:20 PM, Jonas Gauffin wrote: > I'm behind NAT. Is it FS that picks the random port, or the FW? > the FW > I've mapped port 5060 to the freeswitch ip in my FW. > thats inbound, now you need to tell your firewall to nail port 5060 on the outbound side when it comes from port 5060 of your freeswitch IP -- wasim h. baig | principal consultant | convergence pk | +92 300 8508070 | peace be upon you ... -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090204/4cde34eb/attachment-0002.html From intralanman at freeswitch.org Wed Feb 4 07:26:40 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Wed, 04 Feb 2009 10:26:40 -0500 Subject: [Freeswitch-users] gateway In-Reply-To: References: <191c3a030902040614q32c05f02le7e8f41dd70d1fc3@mail.gmail.com> Message-ID: <4989B3B0.3060407@freeswitch.org> Its the firewall... some fw's will give you port 5060 if there's nothing else in the network using 5060, but others will randomize it always. -Ray Jonas Gauffin wrote: > I'm behind NAT. > Is it FS that picks the random port, or the FW? > > I've mapped port 5060 to the freeswitch ip in my FW. > > On Wed, Feb 4, 2009 at 3:14 PM, Anthony Minessale > > wrote: > > if you are not behind any nat then as long as you run your profile > on 5060, the source port on every packet will be 5060. > If you *are* behind nat the nat mapping will pick a random port > unless you have a firewall that allows you to set specific mappings. > > > On Wed, Feb 4, 2009 at 6:26 AM, Jonas Gauffin > > wrote: > > Hello > > I'm trying to make outbound calls through my gateway provider. > My calls got rejected and I asked them why. > > Apparently I need to use 5060 as source port, since they > validate both my IP and the port that the messages come from. > Is this possible with freeswitch? If so, what config settings > should I set? > > Regards, > Jonas > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090204/eaf37c7a/attachment-0002.html From freeswitch-users at lists.rupa.com Wed Feb 4 07:27:52 2009 From: freeswitch-users at lists.rupa.com (Rupa Schomaker (lists)) Date: Wed, 04 Feb 2009 09:27:52 -0600 Subject: [Freeswitch-users] gateway In-Reply-To: References: <191c3a030902040614q32c05f02le7e8f41dd70d1fc3@mail.gmail.com> Message-ID: <4989B3F8.6000604@lists.rupa.com> It is the firewall. Most consumer firewalls allow mapping inbound ports (probably what you describe). I don't know of any that do outbound mapping. Linux or *bsd firewalls should be able to do what you want. I'm sure a cisco with IOS could but it has been ages since I've played with that. On 2/4/2009 9:20 AM, Jonas Gauffin wrote: > I'm behind NAT. > Is it FS that picks the random port, or the FW? > > I've mapped port 5060 to the freeswitch ip in my FW. > > On Wed, Feb 4, 2009 at 3:14 PM, Anthony Minessale > > wrote: > > if you are not behind any nat then as long as you run your profile > on 5060, the source port on every packet will be 5060. > If you *are* behind nat the nat mapping will pick a random port > unless you have a firewall that allows you to set specific mappings. > > > On Wed, Feb 4, 2009 at 6:26 AM, Jonas Gauffin > > wrote: > > Hello > > I'm trying to make outbound calls through my gateway provider. > My calls got rejected and I asked them why. > > Apparently I need to use 5060 as source port, since they > validate both my IP and the port that the messages come from. > Is this possible with freeswitch? If so, what config settings > should I set? > > Regards, > Jonas > > From anthony.minessale at gmail.com Wed Feb 4 08:18:15 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 4 Feb 2009 10:18:15 -0600 Subject: [Freeswitch-users] 16 threads didn't exit In-Reply-To: References: <20090204114955.GD6752@cpdata.co.za> Message-ID: <191c3a030902040818r102dce59qb021b93857307353@mail.gmail.com> can you "make current" to rule out any issues with outdated code and maybe describe what you are using in FS such as scripting langs or anything else that was not enabled by default. 2009/2/4 Cavalera Claudio Luigi > freeswitch-users-bounces at lists.freeswitch.org wrote: > > I have seen that error myself, however I assumed it was due te me > > hanging up other cals from the api_hangup_hook of a related call. > > > > I use this to set a "master" call in a conference so that if > > it hangs up > > all calls in the conference hangup. > > > > Thanks Sias for this info, > if you are able to reproduce this message in a systematic way please let > me know so that I can reproduce it myself. > > I've searched for the "threads didn't exit" message in fs code and I did > not found it. > I think it comes from mysql and I'm not sure it's even related to my > segfault. > Best regards, > Claudio > > > Internet Email Confidentiality Footer > > ----------------------------------------------------------------------------------------------------- > La presente comunicazione, con le informazioni in essa contenute e ogni > documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' > indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete > i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, > comunicazione, divulgazione o simili basate sul contenuto di tali > informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., > D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se > avete ricevuto questa comunicazione per errore, vi preghiamo di darne > immediata notizia al mittente e di distruggere il messaggio originale e ogni > file allegato senza farne copia alcuna o riprodurne in alcun modo il > contenuto. > > This e-mail and its attachments are intended for the addressee(s) only and > are confidential and/or may contain legally privileged information. If you > have received this message by mistake or are not one of the addressees > above, you may take no action based on it, and you may not copy or show it > to anyone; please reply to this e-mail and point out the error which has > occurred. > > ----------------------------------------------------------------------------------------------------- > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090204/5699d6b4/attachment-0002.html From anthony.minessale at gmail.com Wed Feb 4 08:22:05 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 4 Feb 2009 10:22:05 -0600 Subject: [Freeswitch-users] origainate through sofia gateway In-Reply-To: <1233758767.5405.22.camel@dotw1126.dotsystems.pl> References: <1233668179.5354.15.camel@dotw1126.dotsystems.pl> <1b46b4e80902030920p521d6079s4c9b78d8d39a924a@mail.gmail.com> <5E2B217D-8C85-48DB-800A-535A7FAAE24B@freeswitch.org> <1b46b4e80902030931y6a1679afw12ec37596ba9d77d@mail.gmail.com> <1233737494.5405.12.camel@dotw1126.dotsystems.pl> <191c3a030902040609s600798c8t720950a2e13ee05a@mail.gmail.com> <1233758767.5405.22.camel@dotw1126.dotsystems.pl> Message-ID: <191c3a030902040822o1ffededbwd592361aa09f6b46@mail.gmail.com> can you press f8 for debug and try that apiExecute and post the results? On Wed, Feb 4, 2009 at 8:46 AM, Jacek Sokulski wrote: > Thanks Anthony, > the js snippets are very instructive. > A couple of points: > 1. The code with apiExecute does not work (local phone is connected, but > after picking up it hungs up immediately), other examples are working > fine. > > 2. It does not show how initiate external call without existing session. > > 3. How can one pass the call through dialplan? > > Jacek > > PS. > we got the code probable from wiki or from this mialing list. > > Dnia 04-02-2009, ?ro o godzinie 08:09 -0600, Anthony Minessale pisze: > > Where did you learn how to use js this way? > > session.originate is being misused here and is depricated and may be > > removed. > > > > the first arg to session.originate is either undefined or a > > *different* session (the a leg) > > > > session1 = new Session(); > > session1.originate(undefined, > > "{ignore_early_media=true}user/1008 at 192.168.1.122"); > > > > session1.setVariable("effective_caller_id_number","fixed0248b"); > > > > //once you have session1 when you originate session2 you pass session1 > > as the arg > > // the effective_caller_id is taken from session1 > > > > session2 = new Session(); > > session2.originate(session1, "sofia/gateway/halonet/0225490317"); > > > > Anyway this whole code is depricated in favor of this: > > > > session1 = new > > Session("{ignore_early_media=true}user/1008 at 192.168.1.122"); > > > > if (session1.ready()) { > > session1.setVariable("effective_caller_id_number","fixed0248b"); > > session2 = new Session("sofia/gateway/halonet/0225490317", > > session1); > > } > > > > and could be further refactored down to this: > > > > session1 = new > > Session("{ignore_early_media=true}user/1008 at 192.168.1.122"); > > if (session1.ready()) { > > session1.setVariable("effective_caller_id_number","fixed0248b"); > > session1.execute("bridge", "sofia/gateway/halonet/0225490317"); > > } > > > > or down to this one line of code that will setup the call detached > > from the script and exit. > > > > var result = apiExecute("originate", > > > "{effective_caller_id_number=fixed0248b,origination_caller_id_number=1000,ignore_early_media=true}user/ > 1008 at 192.168.1.122 bridge:sofia/gateway/halonet/0225490317 inline"); > > > > if you dont care about the result and want to exit even before the > > call is completed. > > > > var result = apiExecute("bgapi", "originate > > > {effective_caller_id_number=fixed0248b,origination_caller_id_number=1000,ignore_early_media=true}user/ > 1008 at 192.168.1.122 bridge:sofia/gateway/halonet/0225490317 inline"); > > > > > > > > On Wed, Feb 4, 2009 at 2:51 AM, Jacek Sokulski > > wrote: > > > > We have tried setting both effective_caller_id_number and > > origination_caller_id_number: > > > > > session1.originate(session1,"{origination_caller_id_number=fixed0248b}sofia/gateway/halonet/0225490317",15); > > but the problem still exists. The solution we have found for > > the case > > when we originate two calls, local and external, is as follow: > > > > session1 = new Session(); > > session1.originate(session1,"user/1003 at 192.168.1.122 > ",15);//local > > if(session1.ready()) { > > session1.execute("execute_extension","00930691688627 XML > > default");//external > > } > > > > so the external call goes through the dialplan. > > It does not work if both calls are external. One possible > > solution could be > > to pass the originating call through dialplan (loopback?) but > > we have not managed > > to figure out how to do it. > > > > Thanks > > Jacek > > > > Dnia 03-02-2009, wto o godzinie 14:31 -0300, Nicolas Brenner > > pisze: > > > > > Oops! Well, fortunately I don't use that voip provider > > anymore (nor the script). > > > > > > Thanks Brian. > > > > > > Nicolas > > > > > > On Tue, Feb 3, 2009 at 2:25 PM, Brian West > > wrote: > > > > YOU should NEVER use this method or call setCallerData at > > all you > > > > should use the correct methods to override the callerid. > > > > > > > > If its a B-Leg born from an A-Leg you use these on the on > > the A-Leg: > > > > > > > > > > > http://wiki.freeswitch.org/wiki/Channel_Variables#effective_caller_id_name > > > > > > > http://wiki.freeswitch.org/wiki/Channel_Variables#effective_caller_id_number > > > > > > > > If you're originating you use this: > > > > > > > > > > > http://wiki.freeswitch.org/wiki/Channel_Variables#origination_caller_id_name > > > > > > > http://wiki.freeswitch.org/wiki/Channel_Variables#origination_caller_id_number > > > > > > > > /b > > > > > > _______________________________________________ > > > Freeswitch-users mailing list > > > Freeswitch-users at lists.freeswitch.org > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090204/6ede27db/attachment-0002.html From mike at jerris.com Wed Feb 4 08:30:38 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 04 Feb 2009 11:30:38 -0500 Subject: [Freeswitch-users] Errors compiling trunk on a fresh system In-Reply-To: Message-ID: This should now be fixed in trunk in revision 11632. Can you please test and confirm. Mike On 2/4/09 9:27 AM, "Cavalera Claudio Luigi" wrote: > Hello, > I'm trying to compile a brand new fs on a clean system. > Revision: 11630 > > After the usual ./bootstrap.sh > ./configure --enable-core-odbc-support > I was getting this at make > http://pastebin.freeswitch.org/7011 > > so > i cd into utils under libs/sofia-sip/utils > issued a doxygen -u > but still getting this > http://pastebin.freeswitch.org/7012 > > any hint on what could be? > > Best Regards, > Claudio > > From kerrada2003 at yahoo.com Wed Feb 4 08:17:12 2009 From: kerrada2003 at yahoo.com (Ali Al-Rubaie) Date: Wed, 4 Feb 2009 08:17:12 -0800 (PST) Subject: [Freeswitch-users] SIP Authentication Message-ID: <423759.89818.qm@web33708.mail.mud.yahoo.com> Hi, I have a problem in SIP registration (authentication) with FreeSWITCH server. The SIP messages are: recv 292 bytes from udp/[209.82.10.250]:3458 at 16:35:24.758862: ?? ------------------------------------------------------------------------ ?? REGISTER sip:209.82.10.235 SIP/2.0 ?? Via: SIP/2.0/UDP 209.82.10.250:1059 ?? From: sip:1001 at 209.82.10.235 ?? To: sip:1001 at 209.82.10.235 ?? Contact: sip:1001 at 209.82.10.250:1059 ?? Call-ID: fc383368-e3e0-4c32-b87b-c3127d0cc2d8 at 192.168.10.23 ?? CSeq: 597775306 REGISTER ?? Content-Length: 0 ?? Expires: 3600 ? ?? ------------------------------------------------------------------------ send 582 bytes to udp/[209.82.10.250]:1059 at 16:35:24.763948: ?? ------------------------------------------------------------------------ ?? SIP/2.0 401 Unauthorized ?? Via: SIP/2.0/UDP 209.82.10.250:1059 ?? From: sip:1001 at 209.82.10.235 ?? To: ;tag=7yam2F01ZH3vH ?? Call-ID: fc383368-e3e0-4c32-b87b-c3127d0cc2d8 at 192.168.10.23 ?? CSeq: 597775306 REGISTER ?? User-Agent: FreeSWITCH-mod_sofia/1.0.2-hacked ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH ?? Supported: timer, precondition, path, replaces ?? WWW-Authenticate: Digest realm="209.82.10.235", nonce="40b63193-85c2-4ed9-874e-c03f81be313d", algorithm=MD5, qop="auth" ?? Content-Length: 0 ? ?? ------------------------------------------------------------------------ recv 466 bytes from udp/[209.82.10.250]:3458 at 16:35:24.772834: ?? ------------------------------------------------------------------------ ?? REGISTER sip:209.82.10.235 SIP/2.0 ?? Via: SIP/2.0/UDP 209.82.10.250:1059 ?? From: sip:1001 at 209.82.10.235 ?? To: sip:1001 at 209.82.10.235 ?? Contact: sip:1001 at 209.82.10.250:1059 ?? Call-ID: fc383368-e3e0-4c32-b87b-c3127d0cc2d8 at 192.168.10.23 ?? CSeq: 597775307 REGISTER ?? Content-Length: 0 ?? Expires: 3600 ?? Authorization: Digest username="1001",realm="209.82.10.235",nonce="40b63193-85c2-4ed9-874e-c03f81be313d",response="eebe0ea43319e82cc5f6dba5877de706",uri="sip:209.82.10.235" ? ?? ------------------------------------------------------------------------ send 458 bytes to udp/[209.82.10.250]:1059 at 16:35:24.774354: ?? ------------------------------------------------------------------------ ?? SIP/2.0 403 Forbidden ?? Via: SIP/2.0/UDP 209.82.10.250:1059 ?? From: sip:1001 at 209.82.10.235 ?? To: ;tag=873c4aH5vtSFD ?? Call-ID: fc383368-e3e0-4c32-b87b-c3127d0cc2d8 at 192.168.10.23 ?? CSeq: 597775307 REGISTER ?? User-Agent: FreeSWITCH-mod_sofia/1.0.2-hacked ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH ?? Supported: timer, precondition, path, replaces ?? Content-Length: 0 ? ?? ------------------------------------------------------------------------ What I have noted is that the client does not send the values for "cnonce" and "nc" in the response. I'm not sure if this is the reason, however how this problem can be solved? Thanks, Ali -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090204/1b23e643/attachment-0002.html From anthony.minessale at gmail.com Wed Feb 4 08:52:57 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 4 Feb 2009 10:52:57 -0600 Subject: [Freeswitch-users] SIP Authentication In-Reply-To: <423759.89818.qm@web33708.mail.mud.yahoo.com> References: <423759.89818.qm@web33708.mail.mud.yahoo.com> Message-ID: <191c3a030902040852k2c54fceekd73bbd128eb36765@mail.gmail.com> That's the reason, the missing params. The client has a bug in it. On Wed, Feb 4, 2009 at 10:17 AM, Ali Al-Rubaie wrote: > Hi, > > I have a problem in SIP registration (authentication) with FreeSWITCH > server. The SIP messages are: > > recv 292 bytes from udp/[209.82.10.250]:3458 at 16:35:24.758862: > > > ------------------------------------------------------------------------ > > REGISTER sip:209.82.10.235 SIP/2.0 > > Via: SIP/2.0/UDP 209.82.10.250:1059 > > From: sip:1001 at 209.82.10.235 > > To: sip:1001 at 209.82.10.235 > > Contact: sip:1001 at 209.82.10.250:1059 > > Call-ID: fc383368-e3e0-4c32-b87b-c3127d0cc2d8 at 192.168.10.23 > > CSeq: 597775306 REGISTER > > Content-Length: 0 > > Expires: 3600 > > > > > ------------------------------------------------------------------------ > > send 582 bytes to udp/[209.82.10.250]:1059 at 16:35:24.763948: > > > ------------------------------------------------------------------------ > > SIP/2.0 401 Unauthorized > > Via: SIP/2.0/UDP 209.82.10.250:1059 > > From: sip:1001 at 209.82.10.235 > > To: > >;tag=7yam2F01ZH3vH > > Call-ID: fc383368-e3e0-4c32-b87b-c3127d0cc2d8 at 192.168.10.23 > > CSeq: 597775306 REGISTER > > User-Agent: FreeSWITCH-mod_sofia/1.0.2-hacked > > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH > > Supported: timer, precondition, path, replaces > > WWW-Authenticate: Digest realm="209.82.10.235", > nonce="40b63193-85c2-4ed9-874e-c03f81be313d", algorithm=MD5, qop="auth" > > Content-Length: 0 > > > > > ------------------------------------------------------------------------ > > recv 466 bytes from udp/[209.82.10.250]:3458 at 16:35:24.772834: > > > ------------------------------------------------------------------------ > > REGISTER sip:209.82.10.235 SIP/2.0 > > Via: SIP/2.0/UDP 209.82.10.250:1059 > > From: sip:1001 at 209.82.10.235 > > To: sip:1001 at 209.82.10.235 > > Contact: sip:1001 at 209.82.10.250:1059 > > Call-ID: fc383368-e3e0-4c32-b87b-c3127d0cc2d8 at 192.168.10.23 > > CSeq: 597775307 REGISTER > > Content-Length: 0 > > Expires: 3600 > > Authorization: Digest > username="1001",realm="209.82.10.235",nonce="40b63193-85c2-4ed9-874e-c03f81be313d",response="eebe0ea43319e82cc5f6dba5877de706",uri="sip:209.82.10.235" > > > > > ------------------------------------------------------------------------ > > send 458 bytes to udp/[209.82.10.250]:1059 at 16:35:24.774354: > > > ------------------------------------------------------------------------ > > SIP/2.0 403 Forbidden > > Via: SIP/2.0/UDP 209.82.10.250:1059 > > From: sip:1001 at 209.82.10.235 > > To: > >;tag=873c4aH5vtSFD > > Call-ID: fc383368-e3e0-4c32-b87b-c3127d0cc2d8 at 192.168.10.23 > > CSeq: 597775307 REGISTER > > User-Agent: FreeSWITCH-mod_sofia/1.0.2-hacked > > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH > > Supported: timer, precondition, path, replaces > > Content-Length: 0 > > > > > ------------------------------------------------------------------------ > What I have noted is that the client does not send the values for "cnonce" > and "nc" in the response. I'm not sure if this is the reason, however how > this problem can be solved? > > Thanks, > > Ali > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090204/8a749fcb/attachment-0002.html From brian at freeswitch.org Wed Feb 4 08:52:45 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 4 Feb 2009 10:52:45 -0600 Subject: [Freeswitch-users] SIP Authentication In-Reply-To: <423759.89818.qm@web33708.mail.mud.yahoo.com> References: <423759.89818.qm@web33708.mail.mud.yahoo.com> Message-ID: <7DAC91F6-2FD3-464D-AA81-321EBCADC8C0@freeswitch.org> What client is this? I also notice we receive port 3458 and reply to port 1059... /b On Feb 4, 2009, at 10:17 AM, Ali Al-Rubaie wrote: > What I have noted is that the client does not send the values for > "cnonce" and "nc" in the response. I'm not sure if this is the > reason, however how this problem can be solved? > > Thanks, > > Ali From msc at freeswitch.org Wed Feb 4 09:41:07 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 4 Feb 2009 09:41:07 -0800 Subject: [Freeswitch-users] origainate through sofia gateway In-Reply-To: <191c3a030902040609s600798c8t720950a2e13ee05a@mail.gmail.com> References: <1233668179.5354.15.camel@dotw1126.dotsystems.pl> <1b46b4e80902030920p521d6079s4c9b78d8d39a924a@mail.gmail.com> <5E2B217D-8C85-48DB-800A-535A7FAAE24B@freeswitch.org> <1b46b4e80902030931y6a1679afw12ec37596ba9d77d@mail.gmail.com> <1233737494.5405.12.camel@dotw1126.dotsystems.pl> <191c3a030902040609s600798c8t720950a2e13ee05a@mail.gmail.com> Message-ID: <87f2f3b90902040941r61d669aaie949aa7cc8578a9a@mail.gmail.com> I'll make sure the substance of this is in the wiki and I'll look for references to the deprecated way and remove those. -MC On Wed, Feb 4, 2009 at 6:09 AM, Anthony Minessale wrote: > Where did you learn how to use js this way? > session.originate is being misused here and is depricated and may be > removed. > > the first arg to session.originate is either undefined or a *different* > session (the a leg) > > session1 = new Session(); > session1.originate(undefined, > "{ignore_early_media=true}user/1008 at 192.168.1.122"); > > session1.setVariable("effective_caller_id_number","fixed0248b"); > > //once you have session1 when you originate session2 you pass session1 as > the arg > // the effective_caller_id is taken from session1 > > session2 = new Session(); > session2.originate(session1, "sofia/gateway/halonet/0225490317"); > > Anyway this whole code is depricated in favor of this: > > session1 = new Session("{ignore_early_media=true}user/1008 at 192.168.1.122"); > if (session1.ready()) { > session1.setVariable("effective_caller_id_number","fixed0248b"); > session2 = new Session("sofia/gateway/halonet/0225490317", session1); > } > > and could be further refactored down to this: > > session1 = new Session("{ignore_early_media=true}user/1008 at 192.168.1.122"); > if (session1.ready()) { > session1.setVariable("effective_caller_id_number","fixed0248b"); > session1.execute("bridge", "sofia/gateway/halonet/0225490317"); > } > > or down to this one line of code that will setup the call detached from the > script and exit. > > var result = apiExecute("originate", > "{effective_caller_id_number=fixed0248b,origination_caller_id_number=1000,ignore_early_media=true}user/1008 at 192.168.1.122 > bridge:sofia/gateway/halonet/0225490317 inline"); > > if you dont care about the result and want to exit even before the call is > completed. > > var result = apiExecute("bgapi", "originate > {effective_caller_id_number=fixed0248b,origination_caller_id_number=1000,ignore_early_media=true}user/1008 at 192.168.1.122 > bridge:sofia/gateway/halonet/0225490317 inline"); > > > > On Wed, Feb 4, 2009 at 2:51 AM, Jacek Sokulski > wrote: >> >> We have tried setting both effective_caller_id_number and >> origination_caller_id_number: >> >> >> session1.originate(session1,"{origination_caller_id_number=fixed0248b}sofia/gateway/halonet/0225490317",15); >> but the problem still exists. The solution we have found for the case >> when we originate two calls, local and external, is as follow: >> >> session1 = new Session(); >> session1.originate(session1,"user/1003 at 192.168.1.122",15);//local >> if(session1.ready()) { >> session1.execute("execute_extension","00930691688627 XML >> default");//external >> } >> >> so the external call goes through the dialplan. >> It does not work if both calls are external. One possible solution could >> be >> to pass the originating call through dialplan (loopback?) but we have not >> managed >> to figure out how to do it. >> >> Thanks >> Jacek >> >> Dnia 03-02-2009, wto o godzinie 14:31 -0300, Nicolas Brenner pisze: >> > Oops! Well, fortunately I don't use that voip provider anymore (nor the >> > script). >> > >> > Thanks Brian. >> > >> > Nicolas >> > >> > On Tue, Feb 3, 2009 at 2:25 PM, Brian West wrote: >> > > YOU should NEVER use this method or call setCallerData at all you >> > > should use the correct methods to override the callerid. >> > > >> > > If its a B-Leg born from an A-Leg you use these on the on the A-Leg: >> > > >> > > >> > > http://wiki.freeswitch.org/wiki/Channel_Variables#effective_caller_id_name >> > > >> > > http://wiki.freeswitch.org/wiki/Channel_Variables#effective_caller_id_number >> > > >> > > If you're originating you use this: >> > > >> > > >> > > http://wiki.freeswitch.org/wiki/Channel_Variables#origination_caller_id_name >> > > >> > > http://wiki.freeswitch.org/wiki/Channel_Variables#origination_caller_id_number >> > > >> > > /b >> > >> > _______________________________________________ >> > Freeswitch-users mailing list >> > Freeswitch-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From Daniell at airg.com Wed Feb 4 09:43:10 2009 From: Daniell at airg.com (Daniel Liang) Date: Wed, 4 Feb 2009 09:43:10 -0800 Subject: [Freeswitch-users] Recording background music and voice is out of sync Message-ID: <0B02E756F603CC409EB553879B090CC80A23EBB5@HPEXCHVS01.exchange.airg> What I did was the following: First, I sent the playback command: call-command: execute execute-app-name: playback execute-app-arg: Then I send uuid_record (Sorry, it was not Record command): api uuid_record start 120 I also tried replacing the playback command with: api uuid_displace start 0 mux But the end results are the same. The recorded user's voice is about 0.5 second behind the expected result. Thanks, Daniel -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: February 3, 2009 6:36 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Recording background music and voice is outof sync Can you show us an example of how you're doing this? Playback and Record aren't async so you'll need to show us how you're doing this. Also don't hijack threads you hit replay on the one "Re: [Freeswitch- users] FreeSwitch setup as a "Dumb" SBC" as the subject, deleted the subject and started a new body. That hijacks the thread and that can cause your problem to go ignored in some cases if people aren't interested in the thread topic depending on how their reader threads the emails. Please click new message and type freeswitch- users at lists.freeswitch.org in and then input your subject and body to start a new thread. Thanks, Brian West FreeSWITCH.org On Feb 3, 2009, at 8:21 PM, Daniel Liang wrote: > Hi, > > I was trying to record a background music with a user's voice at the > same time. I did a playback and started recording. But the recorded > user's voice and the background music is about 0.5 second out of sync. > I also tried to use uuid_displace instead of playback, but I got the > same result. _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090204/2d1124e2/attachment-0002.html From saigop at gmail.com Wed Feb 4 09:56:14 2009 From: saigop at gmail.com (Gopalakrishnan A.N) Date: Wed, 4 Feb 2009 23:26:14 +0530 Subject: [Freeswitch-users] Q931 decoding Update In-Reply-To: <3D009334-DCFD-4AE5-A446-6C8165F2D77D@jerris.com> References: <4972046E.8020102@ewetel.de> <200901291202.41144.stkn@freeswitch.org> <191c3a030901290549x29c87976k68b3f9b455f00d0f@mail.gmail.com> <4981B8E6.9070708@ewetel.de> <5A2ED995-CF05-4AEF-9103-43BC84465514@freeswitch.org> <4981D6F8.4060409@ewetel.de> <191c3a030901290832k27a18b30u7bc531157fd89b19@mail.gmail.com> <49830576.6080907@ewetel.de> <4987359B.2020402@ewetel.de> <3D009334-DCFD-4AE5-A446-6C8165F2D77D@jerris.com> Message-ID: <2ea4d47e0902040956v75c5472foa4649c50b7340484@mail.gmail.com> Hi, Its a awesome. Can the packet capturing be done with event socket? -- Thank you with regards, Gopal, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090204/74f6434a/attachment-0002.html From chavpaskov at shaw.ca Wed Feb 4 09:59:48 2009 From: chavpaskov at shaw.ca (Chav Paskov) Date: Wed, 04 Feb 2009 09:59:48 -0800 Subject: [Freeswitch-users] mod_limit Message-ID: <4989D794.1010805@shaw.ca> Hi , is it possible to use mod_limit in case if the end point is not registered / gateway for example/. Regards Chav From msc at freeswitch.org Wed Feb 4 10:06:52 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 4 Feb 2009 10:06:52 -0800 Subject: [Freeswitch-users] mod_limit In-Reply-To: <4989D794.1010805@shaw.ca> References: <4989D794.1010805@shaw.ca> Message-ID: <87f2f3b90902041006v7d7d7ff2t9961274905d9d5b0@mail.gmail.com> On Wed, Feb 4, 2009 at 9:59 AM, Chav Paskov wrote: > Hi , > is it possible to use mod_limit in case if the end point is not > registered / gateway for example/. Could you add some detail to this question? What are you trying to do? (mod_limit may or may not work, but there might be another solution which is why I am asking.) -MC > Regards > Chav > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From chavpaskov at shaw.ca Wed Feb 4 10:54:56 2009 From: chavpaskov at shaw.ca (Chav Paskov) Date: Wed, 04 Feb 2009 10:54:56 -0800 Subject: [Freeswitch-users] mod_limit In-Reply-To: <87f2f3b90902041006v7d7d7ff2t9961274905d9d5b0@mail.gmail.com> References: <4989D794.1010805@shaw.ca> <87f2f3b90902041006v7d7d7ff2t9961274905d9d5b0@mail.gmail.com> Message-ID: <4989E480.1080105@shaw.ca> Michael Collins wrote: > On Wed, Feb 4, 2009 at 9:59 AM, Chav Paskov wrote: > >> Hi , >> is it possible to use mod_limit in case if the end point is not >> registered / gateway for example/. >> > > Could you add some detail to this question? What are you trying to do? > (mod_limit may or may not work, but there might be another solution > which is why I am asking.) > > -MC > > >> Regards >> Chav >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > i have few gateways under my ACL that are allowed to send calls to FS, but i want to be able to enforce "capacity" policy on the traffic coming from any one of them depending on total termination capacity on my termination end. Let say GW 1 has to be limited to make 10 simultaneous calls while GW2 could make up to 30 and so on. Regards Chav From msc at freeswitch.org Wed Feb 4 11:05:09 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 4 Feb 2009 11:05:09 -0800 Subject: [Freeswitch-users] mod_limit In-Reply-To: <4989E480.1080105@shaw.ca> References: <4989D794.1010805@shaw.ca> <87f2f3b90902041006v7d7d7ff2t9961274905d9d5b0@mail.gmail.com> <4989E480.1080105@shaw.ca> Message-ID: <87f2f3b90902041105l50f51f08t230bab8d69eefb4e@mail.gmail.com> On Wed, Feb 4, 2009 at 10:54 AM, Chav Paskov wrote: > Michael Collins wrote: >> On Wed, Feb 4, 2009 at 9:59 AM, Chav Paskov wrote: >> >>> Hi , >>> is it possible to use mod_limit in case if the end point is not >>> registered / gateway for example/. >>> >> >> Could you add some detail to this question? What are you trying to do? >> (mod_limit may or may not work, but there might be another solution >> which is why I am asking.) >> >> -MC >> >> >>> Regards >>> Chav >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > i have few gateways under my ACL that are allowed to send calls to FS, > but i want to be able to enforce "capacity" policy on the traffic > coming from any one of them depending on total termination capacity on > my termination end. > Let say GW 1 has to be limited to make 10 simultaneous calls while GW2 > could make up to 30 and so on. I'm sure that this is possible. I don't personally have a way to test all of this but I know that a number of our users are doing things like this currently. Can you hop on to the IRC channel? #freeswitch on irc.freenode.net. A lot of people there can help with this one. -MC (IRC: mercutioviz) > Regards > Chav > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From brian at freeswitch.org Wed Feb 4 11:13:54 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 4 Feb 2009 13:13:54 -0600 Subject: [Freeswitch-users] mod_limit In-Reply-To: <87f2f3b90902041105l50f51f08t230bab8d69eefb4e@mail.gmail.com> References: <4989D794.1010805@shaw.ca> <87f2f3b90902041006v7d7d7ff2t9961274905d9d5b0@mail.gmail.com> <4989E480.1080105@shaw.ca> <87f2f3b90902041105l50f51f08t230bab8d69eefb4e@mail.gmail.com> Message-ID: <49E707E6-2126-45E5-9D5D-D44D477245C5@freeswitch.org> mod_limit is just another lego brick in the set... You can put it in the dialplan before you go in our out a gateway and limit based on the realm... its no different for registered users either... I get the feeling you're trying to make this more complicated then it really is. /b PS when you reply please try to shorten the resulting text of the reply no need to have unsubscribe info in the email 10 times. On Feb 4, 2009, at 1:05 PM, Michael Collins wrote: >>> >>> >> i have few gateways under my ACL that are allowed to send calls to >> FS, >> but i want to be able to enforce "capacity" policy on the traffic >> coming from any one of them depending on total termination capacity >> on >> my termination end. >> Let say GW 1 has to be limited to make 10 simultaneous calls >> while GW2 >> could make up to 30 and so on. > > I'm sure that this is possible. I don't personally have a way to test > all of this but I know that a number of our users are doing things > like this currently. Can you hop on to the IRC channel? #freeswitch on > irc.freenode.net. A lot of people there can help with this one. > > -MC (IRC: mercutioviz) > >> Regards >> Chav From msc at freeswitch.org Wed Feb 4 12:52:29 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 4 Feb 2009 12:52:29 -0800 Subject: [Freeswitch-users] Q931 decoding Update In-Reply-To: <2ea4d47e0902040956v75c5472foa4649c50b7340484@mail.gmail.com> References: <4972046E.8020102@ewetel.de> <191c3a030901290549x29c87976k68b3f9b455f00d0f@mail.gmail.com> <4981B8E6.9070708@ewetel.de> <5A2ED995-CF05-4AEF-9103-43BC84465514@freeswitch.org> <4981D6F8.4060409@ewetel.de> <191c3a030901290832k27a18b30u7bc531157fd89b19@mail.gmail.com> <49830576.6080907@ewetel.de> <4987359B.2020402@ewetel.de> <3D009334-DCFD-4AE5-A446-6C8165F2D77D@jerris.com> <2ea4d47e0902040956v75c5472foa4649c50b7340484@mail.gmail.com> Message-ID: <87f2f3b90902041252h5ede448bq720c15ea23dd517a@mail.gmail.com> On Wed, Feb 4, 2009 at 9:56 AM, Gopalakrishnan A.N wrote: > Hi, > Its a awesome. Can the packet capturing be done with event socket? Not at this time. Would require some additional programming. Are you up for the task? ;) -MC > > -- > Thank you with regards, > Gopal, > > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From sias at cpdata.co.za Wed Feb 4 13:14:46 2009 From: sias at cpdata.co.za (Sias Mey) Date: Wed, 4 Feb 2009 23:14:46 +0200 Subject: [Freeswitch-users] 16 threads didn't exit In-Reply-To: <191c3a030902040818r102dce59qb021b93857307353@mail.gmail.com> References: <20090204114955.GD6752@cpdata.co.za> <191c3a030902040818r102dce59qb021b93857307353@mail.gmail.com> Message-ID: <20090204211445.GA15501@cpdata.co.za> Hi Anthony, I have been seeing this message for the for a couple of weeks now. And as you know(you asked me to) I have been keeping up to date. So it is definately not only a recent change. I use JS with odbc and core support. And also xml_rpc. I will do some more testing tommorow to see if I can narrow down what script and what bit of code seems to cause this. On Wed, Feb 04, 2009 at 10:18:15AM -0600, Anthony Minessale wrote: > can you "make current" to rule out any issues with outdated code and > maybe describe what you are using in FS such as scripting langs or > anything else that was not enabled by default. > > 2009/2/4 Cavalera Claudio Luigi <[1]Claudio.Cavalera at italtel.it> > > [2]freeswitch-users-bounces at lists.freeswitch.org wrote: > > I have seen that error myself, however I assumed it was due te me > > hanging up other cals from the api_hangup_hook of a related call. > > > > I use this to set a "master" call in a conference so that if > > it hangs up > > all calls in the conference hangup. > > > > Thanks Sias for this info, > if you are able to reproduce this message in a systematic way please > let > me know so that I can reproduce it myself. > I've searched for the "threads didn't exit" message in fs code and I > did > not found it. > I think it comes from mysql and I'm not sure it's even related to my > segfault. > > Best regards, > Claudio > Internet Email Confidentiality Footer > ----------------------------------------------------------------------- > ------------------------------ > La presente comunicazione, con le informazioni in essa contenute e ogni > documento o file allegato, e' rivolta unicamente alla/e persona/e cui > e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se > non siete i destinatari/autorizzati siete avvisati che qualsiasi > azione, copia, comunicazione, divulgazione o simili basate sul > contenuto di tali informazioni e' vietata e potrebbe essere contro la > legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione > dei dati personali). Se avete ricevuto questa comunicazione per errore, > vi preghiamo di darne immediata notizia al mittente e di distruggere il > messaggio originale e ogni file allegato senza farne copia alcuna o > riprodurne in alcun modo il contenuto. > This e-mail and its attachments are intended for the addressee(s) only > and are confidential and/or may contain legally privileged information. > If you have received this message by mistake or are not one of the > addressees above, you may take no action based on it, and you may not > copy or show it to anyone; please reply to this e-mail and point out > the error which has occurred. > ----------------------------------------------------------------------- > ------------------------------ > _______________________________________________ > Freeswitch-users mailing list > [3]Freeswitch-users at lists.freeswitch.org > [4]http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:[5]http://lists.freeswitch.org/mailman/options/freeswitch-u > sers > [6]http://www.freeswitch.org > > -- > Anthony Minessale II > FreeSWITCH [7]http://www.freeswitch.org/ > ClueCon [8]http://www.cluecon.com/ > AIM: anthm > [9]MSN:anthony_minessale at hotmail.com > GTALK/JABBER/[10]PAYPAL:anthony.minessale at gmail.com > IRC: [11]irc.freenode.net #freeswitch > FreeSWITCH Developer Conference > [12]sip:888 at conference.freeswitch.org > [13]iax:guest at conference.freeswitch.org/888 > [14]googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > References > > 1. mailto:Claudio.Cavalera at italtel.it > 2. mailto:freeswitch-users-bounces at lists.freeswitch.org > 3. mailto:Freeswitch-users at lists.freeswitch.org > 4. http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > 5. http://lists.freeswitch.org/mailman/options/freeswitch-users > 6. http://www.freeswitch.org/ > 7. http://www.freeswitch.org/ > 8. http://www.cluecon.com/ > 9. mailto:MSN%3Aanthony_minessale at hotmail.com > 10. mailto:PAYPAL%3Aanthony.minessale at gmail.com > 11. http://irc.freenode.net/ > 12. mailto:sip%3A888 at conference.freeswitch.org > 13. http://iax:guest at conference.freeswitch.org/888 > 14. mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From nik.middleton at noblesolutions.co.uk Wed Feb 4 14:30:09 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Wed, 4 Feb 2009 22:30:09 -0000 Subject: [Freeswitch-users] FS in ISP Mode In-Reply-To: <2ea4d47e0902040956v75c5472foa4649c50b7340484@mail.gmail.com> References: <4972046E.8020102@ewetel.de><200901291202.41144.stkn@freeswitch.org><191c3a030901290549x29c87976k68b3f9b455f00d0f@mail.gmail.com><4981B8E6.9070708@ewetel.de><5A2ED995-CF05-4AEF-9103-43BC84465514@freeswitch.org><4981D6F8.4060409@ewetel.de><191c3a030901290832k27a18b30u7bc531157fd89b19@mail.gmail.com><49830576.6080907@ewetel.de> <4987359B.2020402@ewetel.de><3D009334-DCFD-4AE5-A446-6C8165F2D77D@jerris.com> <2ea4d47e0902040956v75c5472foa4649c50b7340484@mail.gmail.com> Message-ID: Hi Guys, Excuse my ignorance, but I'm just starting with FS. I've loaded FS onto one of our servers in a datacenter. I'm registering with our PSTN breakout provider just fine, but I'm a little confused about internal/external. Given that we have no internal clients, as they're all external, should I switch the ports over so that 5060 is the external port? I think I know the answer, but would like confirmation Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090204/2b5c0a83/attachment-0002.html From nik.middleton at noblesolutions.co.uk Wed Feb 4 14:40:11 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Wed, 4 Feb 2009 22:40:11 -0000 Subject: [Freeswitch-users] Gateway setting In-Reply-To: <2ea4d47e0902040956v75c5472foa4649c50b7340484@mail.gmail.com> References: <4972046E.8020102@ewetel.de><200901291202.41144.stkn@freeswitch.org><191c3a030901290549x29c87976k68b3f9b455f00d0f@mail.gmail.com><4981B8E6.9070708@ewetel.de><5A2ED995-CF05-4AEF-9103-43BC84465514@freeswitch.org><4981D6F8.4060409@ewetel.de><191c3a030901290832k27a18b30u7bc531157fd89b19@mail.gmail.com><49830576.6080907@ewetel.de> <4987359B.2020402@ewetel.de><3D009334-DCFD-4AE5-A446-6C8165F2D77D@jerris.com> <2ea4d47e0902040956v75c5472foa4649c50b7340484@mail.gmail.com> Message-ID: Hi Guys, Need a little help here; I connect to my PSTN provider via the LAN, Question: As the provider authenticates on IP, how do I not send a password? In the .xml file if I remove the password entry it complains Secondly, the contact should be my local address, not the public one. What do I need to do here? Finally what does FS do to determine if the status is up, is there an asterisk equivalent of qualify going on here? Regards Name My Provider Scheme Digest Realm 172.16.1.2 Username myname Password yes >From Contact To sip: myname @172.168.1.2 Proxy myname:172.168.1.2 Context default Expires 600 Freq 600 Ping 0 PingFreq 0 State NOREG Status UP -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090204/3705c4b8/attachment-0002.html From brian at freeswitch.org Wed Feb 4 14:42:05 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 4 Feb 2009 16:42:05 -0600 Subject: [Freeswitch-users] FS in ISP Mode In-Reply-To: References: <4972046E.8020102@ewetel.de><200901291202.41144.stkn@freeswitch.org><191c3a030901290549x29c87976k68b3f9b455f00d0f@mail.gmail.com><4981B8E6.9070708@ewetel.de><5A2ED995-CF05-4AEF-9103-43BC84465514@freeswitch.org><4981D6F8.4060409@ewetel.de><191c3a030901290832k27a18b30u7bc531157fd89b19@mail.gmail.com><49830576.6080907@ewetel.de> <4987359B.2020402@ewetel.de><3D009334-DCFD-4AE5-A446-6C8165F2D77D@jerris.com> <2ea4d47e0902040956v75c5472foa4649c50b7340484@mail.gmail.com> Message-ID: <513C18B7-631A-4250-8484-81DD7F75D3A1@freeswitch.org> Don't let the names of the profiles confuse you... they are just names. internal is on port 5060 has auth on... external is on 5080 and doesn't' have auth on and lets all calls into the public context without auth. Also when you post to the mailing list do not hijack a thread. Hijacking happens when you take an existing message, click reply, change the subject and start a new body. That will hijack the thread. In your case the thread was "Re: [Freeswitch-users] Q931 decoding Update". So next time please click new message and input the address freeswitch-users at lists.freeswitch.org . Thanks, Brian West FreeSWITCH.org On Feb 4, 2009, at 4:30 PM, Nik Middleton wrote: > Hi Guys, > > Excuse my ignorance, but I?m just starting with FS. > > I?ve loaded FS onto one of our servers in a datacenter. I?m > registering with our PSTN breakout provider just fine, but I?m a > little confused about internal/external. > > Given that we have no internal clients, as they?re all external, > should I switch the ports over so that 5060 is the external port? > > I think I know the answer, but would like confirmation > > Regards > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Wed Feb 4 14:48:49 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 4 Feb 2009 16:48:49 -0600 Subject: [Freeswitch-users] Gateway setting In-Reply-To: References: <4972046E.8020102@ewetel.de><200901291202.41144.stkn@freeswitch.org><191c3a030901290549x29c87976k68b3f9b455f00d0f@mail.gmail.com><4981B8E6.9070708@ewetel.de><5A2ED995-CF05-4AEF-9103-43BC84465514@freeswitch.org><4981D6F8.4060409@ewetel.de><191c3a030901290832k27a18b30u7bc531157fd89b19@mail.gmail.com><49830576.6080907@ewetel.de> <4987359B.2020402@ewetel.de><3D009334-DCFD-4AE5-A446-6C8165F2D77D@jerris.com> <2ea4d47e0902040956v75c5472foa4649c50b7340484@mail.gmail.com> Message-ID: <4C4AD6CE-35C7-4F13-80C6-6F86DA71A51E@freeswitch.org> You don't use gateways if they auth by IP... just dial sofia/profile/ number at remoteip Also please stop hijacking threads. /b On Feb 4, 2009, at 4:40 PM, Nik Middleton wrote: > Hi Guys, > > Need a little help here; I connect to my PSTN provider via the LAN, > > Question: As the provider authenticates on IP, how do I not send a > password? In the .xml file if I remove the password entry it > complains > > Secondly, the contact should be my local address, not the public > one. What do I need to do here? > > Finally what does FS do to determine if the status is up, is there > an asterisk equivalent of qualify going on here? > > Regards > > > Name My Provider > Scheme Digest > Realm 172.16.1.2 > Username myname > Password yes > From > Contact > To sip: myname @172.168.1.2 > Proxy myname:172.168.1.2 > Context default > Expires 600 > Freq 600 > Ping 0 > PingFreq 0 > State NOREG > Status UP > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090204/def37ca8/attachment-0002.html From nik.middleton at noblesolutions.co.uk Wed Feb 4 14:50:35 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Wed, 4 Feb 2009 22:50:35 -0000 Subject: [Freeswitch-users] FS in ISP Mode Message-ID: Do apologise about the hijacking, Question: My ISP sends inbound calls via 5060, so it seems I need to renumber the ports, but that leaves my SIP end points who authenticate also needing 5060, can they be combined? Regards, -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: 04 February 2009 22:42 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] FS in ISP Mode Don't let the names of the profiles confuse you... they are just names. internal is on port 5060 has auth on... external is on 5080 and doesn't' have auth on and lets all calls into the public context without auth. Also when you post to the mailing list do not hijack a thread. Hijacking happens when you take an existing message, click reply, change the subject and start a new body. That will hijack the thread. In your case the thread was "Re: [Freeswitch-users] Q931 decoding Update". So next time please click new message and input the address freeswitch-users at lists.freeswitch.org . Thanks, Brian West FreeSWITCH.org On Feb 4, 2009, at 4:30 PM, Nik Middleton wrote: > Hi Guys, > > Excuse my ignorance, but I'm just starting with FS. > > I've loaded FS onto one of our servers in a datacenter. I'm > registering with our PSTN breakout provider just fine, but I'm a > little confused about internal/external. > > Given that we have no internal clients, as they're all external, > should I switch the ports over so that 5060 is the external port? > > I think I know the answer, but would like confirmation > > Regards > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090204/f98b8f0d/attachment-0002.html From nik.middleton at noblesolutions.co.uk Wed Feb 4 14:52:49 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Wed, 4 Feb 2009 22:52:49 -0000 Subject: [Freeswitch-users] Gateway settings Message-ID: Hi Guys, Need a little help here; I connect to my PSTN provider via the LAN, Question: As the provider authenticates on IP, how do I not send a password? In the .xml file if I remove the password entry it complains Secondly, the contact should be my local address, not the public one. What do I need to do here? Finally what does FS do to determine if the status is up, is there an asterisk equivalent of qualify going on here? Regards Name My Provider Scheme Digest Realm 172.16.1.2 Username myname Password yes >From Contact To sip: myname @172.168.1.2 Proxy myname:172.168.1.2 Context default Expires 600 Freq 600 Ping 0 PingFreq 0 State NOREG Status UP -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090204/15a3fcd1/attachment-0002.html From nik.middleton at noblesolutions.co.uk Wed Feb 4 14:54:41 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Wed, 4 Feb 2009 22:54:41 -0000 Subject: [Freeswitch-users] Gateway setting In-Reply-To: <4C4AD6CE-35C7-4F13-80C6-6F86DA71A51E@freeswitch.org> References: <4972046E.8020102@ewetel.de><200901291202.41144.stkn@freeswitch.org><191c3a030901290549x29c87976k68b3f9b455f00d0f@mail.gmail.com><4981B8E6.9070708@ewetel.de><5A2ED995-CF05-4AEF-9103-43BC84465514@freeswitch.org><4981D6F8.4060409@ewetel.de><191c3a030901290832k27a18b30u7bc531157fd89b19@mail.gmail.com><49830576.6080907@ewetel.de><4987359B.2020402@ewetel.de><3D009334-DCFD-4AE5-A446-6C8165F2D77D@jerris.com><2ea4d47e0902040956v75c5472foa4649c50b7340484@mail.gmail.com> <4C4AD6CE-35C7-4F13-80C6-6F86DA71A51E@freeswitch.org> Message-ID: Sorry, 2 messages sent before I understood how this forum processes posts ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: 04 February 2009 22:49 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Gateway setting You don't use gateways if they auth by IP... just dial sofia/profile/number at remoteip Also please stop hijacking threads. /b On Feb 4, 2009, at 4:40 PM, Nik Middleton wrote: Hi Guys, Need a little help here; I connect to my PSTN provider via the LAN, Question: As the provider authenticates on IP, how do I not send a password? In the .xml file if I remove the password entry it complains Secondly, the contact should be my local address, not the public one. What do I need to do here? Finally what does FS do to determine if the status is up, is there an asterisk equivalent of qualify going on here? Regards Name My Provider Scheme Digest Realm 172.16.1.2 Username myname Password yes >From Contact To sip: myname @172.168.1.2 Proxy myname:172.168.1.2 Context default Expires 600 Freq 600 Ping 0 PingFreq 0 State NOREG Status UP _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090204/440f0b90/attachment-0002.html From brian at freeswitch.org Wed Feb 4 14:55:49 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 4 Feb 2009 16:55:49 -0600 Subject: [Freeswitch-users] FS in ISP Mode In-Reply-To: References: Message-ID: <05317B37-EDAF-4286-BE27-5DAB8C0A0289@freeswitch.org> If your ITSP requires you to come from 5060 on your request then they are seriously broken. But yes you can move the ports around on the profiles or turn auth to false on the internal profile if you don't require any digest auth or phones registering. /b On Feb 4, 2009, at 4:50 PM, Nik Middleton wrote: > Do apologise about the hijacking, > > Question: My ISP sends inbound calls via 5060, so it seems I need > to renumber the ports, but that leaves my SIP end points who > authenticate also needing 5060, can they be combined? > > Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090204/9f91c9aa/attachment-0002.html From brian at freeswitch.org Wed Feb 4 14:59:57 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 4 Feb 2009 16:59:57 -0600 Subject: [Freeswitch-users] Gateway setting In-Reply-To: References: <4972046E.8020102@ewetel.de><200901291202.41144.stkn@freeswitch.org><191c3a030901290549x29c87976k68b3f9b455f00d0f@mail.gmail.com><4981B8E6.9070708@ewetel.de><5A2ED995-CF05-4AEF-9103-43BC84465514@freeswitch.org><4981D6F8.4060409@ewetel.de><191c3a030901290832k27a18b30u7bc531157fd89b19@mail.gmail.com><49830576.6080907@ewetel.de><4987359B.2020402@ewetel.de><3D009334-DCFD-4AE5-A446-6C8165F2D77D@jerris.com><2ea4d47e0902040956v75c5472foa4649c50b7340484@mail.gmail.com> <4C4AD6CE-35C7-4F13-80C6-6F86DA71A51E@freeswitch.org> Message-ID: <43782D4C-3B15-427D-98A7-F72E3CE0F922@freeswitch.org> Its ok ;) We'll get you taken care of.. you should join us on IRC... #freenode its a faster way to get help. irc.freenode.net /b On Feb 4, 2009, at 4:54 PM, Nik Middleton wrote: > Sorry, 2 messages sent before I understood how this forum processes > posts -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090204/65288a5d/attachment-0002.html From nik.middleton at noblesolutions.co.uk Wed Feb 4 15:05:52 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Wed, 4 Feb 2009 23:05:52 -0000 Subject: [Freeswitch-users] Gateway setting In-Reply-To: <43782D4C-3B15-427D-98A7-F72E3CE0F922@freeswitch.org> References: <4972046E.8020102@ewetel.de><200901291202.41144.stkn@freeswitch.org><191c3a030901290549x29c87976k68b3f9b455f00d0f@mail.gmail.com><4981B8E6.9070708@ewetel.de><5A2ED995-CF05-4AEF-9103-43BC84465514@freeswitch.org><4981D6F8.4060409@ewetel.de><191c3a030901290832k27a18b30u7bc531157fd89b19@mail.gmail.com><49830576.6080907@ewetel.de><4987359B.2020402@ewetel.de><3D009334-DCFD-4AE5-A446-6C8165F2D77D@jerris.com><2ea4d47e0902040956v75c5472foa4649c50b7340484@mail.gmail.com><4C4AD6CE-35C7-4F13-80C6-6F86DA71A51E@freeswitch.org> <43782D4C-3B15-427D-98A7-F72E3CE0F922@freeswitch.org> Message-ID: Well try as I might, I can't connect to that server, others are fine, but I get DNS pool errors ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: 04 February 2009 23:00 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Gateway setting Its ok ;) We'll get you taken care of.. you should join us on IRC... #freenode its a faster way to get help. irc.freenode.net /b On Feb 4, 2009, at 4:54 PM, Nik Middleton wrote: Sorry, 2 messages sent before I understood how this forum processes posts -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090204/ec162a50/attachment-0002.html From nik.middleton at noblesolutions.co.uk Wed Feb 4 15:07:58 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Wed, 4 Feb 2009 23:07:58 -0000 Subject: [Freeswitch-users] FS in ISP Mode In-Reply-To: <05317B37-EDAF-4286-BE27-5DAB8C0A0289@freeswitch.org> References: <05317B37-EDAF-4286-BE27-5DAB8C0A0289@freeswitch.org> Message-ID: Sorry, not being clear. If external user dials a geo number, my pstn provider forwards call to 5060 at my server address. They expect me to be listening on 5060 Regards ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: 04 February 2009 22:56 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] FS in ISP Mode If your ITSP requires you to come from 5060 on your request then they are seriously broken. But yes you can move the ports around on the profiles or turn auth to false on the internal profile if you don't require any digest auth or phones registering. /b On Feb 4, 2009, at 4:50 PM, Nik Middleton wrote: Do apologise about the hijacking, Question: My ISP sends inbound calls via 5060, so it seems I need to renumber the ports, but that leaves my SIP end points who authenticate also needing 5060, can they be combined? Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090204/0eafa860/attachment-0002.html From krice at freeswitch.org Wed Feb 4 15:11:07 2009 From: krice at freeswitch.org (Ken Rice) Date: Wed, 04 Feb 2009 17:11:07 -0600 Subject: [Freeswitch-users] FS in ISP Mode In-Reply-To: Message-ID: You can reverse the ports or try to get your provider to send to 5060 or you can bind an additional IP and have the external profile listen there K From: Nik Middleton Reply-To: Date: Wed, 4 Feb 2009 23:07:58 -0000 To: Subject: Re: [Freeswitch-users] FS in ISP Mode Sorry, not being clear. If external user dials a geo number, my pstn provider forwards call to 5060 at my server address. They expect me to be listening on 5060 Regards From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: 04 February 2009 22:56 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] FS in ISP Mode If your ITSP requires you to come from 5060 on your request then they are seriously broken. But yes you can move the ports around on the profiles or turn auth to false on the internal profile if you don't require any digest auth or phones registering. /b On Feb 4, 2009, at 4:50 PM, Nik Middleton wrote: Do apologise about the hijacking, Question: My ISP sends inbound calls via 5060, so it seems I need to renumber the ports, but that leaves my SIP end points who authenticate also needing 5060, can they be combined? Regards, _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090204/8d522b32/attachment-0002.html From brian at freeswitch.org Wed Feb 4 15:11:21 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 4 Feb 2009 17:11:21 -0600 Subject: [Freeswitch-users] Gateway setting In-Reply-To: References: <4972046E.8020102@ewetel.de><200901291202.41144.stkn@freeswitch.org><191c3a030901290549x29c87976k68b3f9b455f00d0f@mail.gmail.com><4981B8E6.9070708@ewetel.de><5A2ED995-CF05-4AEF-9103-43BC84465514@freeswitch.org><4981D6F8.4060409@ewetel.de><191c3a030901290832k27a18b30u7bc531157fd89b19@mail.gmail.com><49830576.6080907@ewetel.de><4987359B.2020402@ewetel.de><3D009334-DCFD-4AE5-A446-6C8165F2D77D@jerris.com><2ea4d47e0902040956v75c5472foa4649c50b7340484@mail.gmail.com><4C4AD6CE-35C7-4F13-80C6-6F86DA71A51E@freeswitch.org> <43782D4C-3B15-427D-98A7-F72E3CE0F922@freeswitch.org> Message-ID: <217639A0-29CD-410E-B571-FAD71636D083@freeswitch.org> You'll need to get an IRC client or do it from the www.freeswitch.org site on the right. /b On Feb 4, 2009, at 5:05 PM, Nik Middleton wrote: > Well try as I might, I can?t connect to that server, others are > fine, but I get DNS pool errors > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090204/c3f69e1b/attachment-0002.html From nik.middleton at noblesolutions.co.uk Wed Feb 4 15:15:45 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Wed, 4 Feb 2009 23:15:45 -0000 Subject: [Freeswitch-users] FS in ISP Mode In-Reply-To: References: Message-ID: Ah Ha, that would work (second IP) Regards ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice Sent: 04 February 2009 23:11 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] FS in ISP Mode You can reverse the ports or try to get your provider to send to 5060 or you can bind an additional IP and have the external profile listen there K ________________________________ From: Nik Middleton Reply-To: Date: Wed, 4 Feb 2009 23:07:58 -0000 To: Subject: Re: [Freeswitch-users] FS in ISP Mode Sorry, not being clear. If external user dials a geo number, my pstn provider forwards call to 5060 at my server address. They expect me to be listening on 5060 Regards ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: 04 February 2009 22:56 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] FS in ISP Mode If your ITSP requires you to come from 5060 on your request then they are seriously broken. But yes you can move the ports around on the profiles or turn auth to false on the internal profile if you don't require any digest auth or phones registering. /b On Feb 4, 2009, at 4:50 PM, Nik Middleton wrote: Do apologise about the hijacking, Question: My ISP sends inbound calls via 5060, so it seems I need to renumber the ports, but that leaves my SIP end points who authenticate also needing 5060, can they be combined? Regards, ________________________________ _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090204/4d319639/attachment-0002.html From saigop at gmail.com Wed Feb 4 21:11:26 2009 From: saigop at gmail.com (Gopalakrishnan A.N) Date: Thu, 5 Feb 2009 10:41:26 +0530 Subject: [Freeswitch-users] Q931 decoding Update In-Reply-To: <87f2f3b90902041252h5ede448bq720c15ea23dd517a@mail.gmail.com> References: <4972046E.8020102@ewetel.de> <4981B8E6.9070708@ewetel.de> <5A2ED995-CF05-4AEF-9103-43BC84465514@freeswitch.org> <4981D6F8.4060409@ewetel.de> <191c3a030901290832k27a18b30u7bc531157fd89b19@mail.gmail.com> <49830576.6080907@ewetel.de> <4987359B.2020402@ewetel.de> <3D009334-DCFD-4AE5-A446-6C8165F2D77D@jerris.com> <2ea4d47e0902040956v75c5472foa4649c50b7340484@mail.gmail.com> <87f2f3b90902041252h5ede448bq720c15ea23dd517a@mail.gmail.com> Message-ID: <2ea4d47e0902042111x5aa08410oe99c6ae02df3d7de@mail.gmail.com> Yes I can do that with any integration On Thu, Feb 5, 2009 at 2:22 AM, Michael Collins wrote: > On Wed, Feb 4, 2009 at 9:56 AM, Gopalakrishnan A.N > wrote: > > Hi, > > Its a awesome. Can the packet capturing be done with event socket? > > Not at this time. Would require some additional programming. Are you > up for the task? ;) > -MC > > > > > -- > > Thank you with regards, > > Gopal, > > > > > > > > > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Thank you with regards, Gopal, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090205/8e3617a4/attachment-0002.html From sicfslist at gmail.com Wed Feb 4 23:33:07 2009 From: sicfslist at gmail.com (Shelby Ramsey) Date: Thu, 5 Feb 2009 01:33:07 -0600 Subject: [Freeswitch-users] XML CDR ERROR ... Message-ID: <35b355e90902042333p3009d6c1md2a76355a90509bd@mail.gmail.com> Hello, I'm having it on both Fedora and Ubuntu boxes: 2009-02-05 01:26:27 [ERR] mod_xml_cdr.c:290 mod_xml_cdr_load() Open of xml_cdr.conf failed 2009-02-05 01:26:27 [CRIT] switch_loadable_module.c:839 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_xml_cdr.so **Module load routine returned an error** Details: -- Ubuntu 6.04 LTS -- Fedora 8 Tried a couple of things: -- messing with libcurl -- ./configure --without-libcurl Thanks! SDR -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090205/1ced279a/attachment-0002.html From brian at freeswitch.org Wed Feb 4 23:42:59 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 5 Feb 2009 01:42:59 -0600 Subject: [Freeswitch-users] XML CDR ERROR ... In-Reply-To: <35b355e90902042333p3009d6c1md2a76355a90509bd@mail.gmail.com> References: <35b355e90902042333p3009d6c1md2a76355a90509bd@mail.gmail.com> Message-ID: <640834B6-0A23-4855-BB0F-F3622AA334FF@freeswitch.org> Make sure your config file is installed and issue a reloadxml then load mod_xml_cdr /b On Feb 5, 2009, at 1:33 AM, Shelby Ramsey wrote: > Hello, > > I'm having it on both Fedora and Ubuntu boxes: > > 2009-02-05 01:26:27 [ERR] mod_xml_cdr.c:290 mod_xml_cdr_load() Open > of xml_cdr.conf failed > 2009-02-05 01:26:27 [CRIT] switch_loadable_module.c:839 > switch_loadable_module_load_file() Error Loading module /usr/local/ > freeswitch/mod/mod_xml_cdr.so > **Module load routine returned an error** > > Details: > -- Ubuntu 6.04 LTS > -- Fedora 8 > > Tried a couple of things: > -- messing with libcurl > -- ./configure --without-libcurl > > Thanks! > > SDR From kawarod at laposte.net Thu Feb 5 00:25:09 2009 From: kawarod at laposte.net (rod) Date: Thu, 05 Feb 2009 12:25:09 +0400 Subject: [Freeswitch-users] FreeSwitch setup as a "Dumb" SBC In-Reply-To: <49896E8D.3010609@post.cz> References: <4988C12E.1090109@post.cz> <191c3a030902031434v1129d684j18b09a954d871f7b@mail.gmail.com> <49896E8D.3010609@post.cz> Message-ID: <498AA265.4060307@laposte.net> Hi, how many static xml files did you create for your test ? rod. kokoska rokoska wrote: > Anthony Minessale napsal(a): > >> What does it look like if you serve the directory from the static xml >> file out of curiosity. >> >> > > Well, I write all user infos into static xml files loaded at startup :-) > > For the first try (without "tuning", see below) I can't go beyond 220 > reg/s - it is just about 10-12 % higher rate then with "lighty" etc. > BTW: Lighty alone can serve about 2600-2700 "dynamic directory xml files > from DB" per second - tested with "ab". > > The only difference (but big one :-) is that CPU utilization is below > 15% on all cores and load is about 0.2, so machine is idle :-) > > It tells me, that somthing is wron with my setup :-) > > > The only optimizations done are: > > 1. No logging > 2. FS in "high priority" mode > 3. "ulimits" applied > > Next (today evenings - tommorow mornings, don't know) I try to "tune" FS > for better preformance - as I did with previous test last month: > > 1. Move FS internal SQL light to ramdisk - I'm not sure if it helps, > because OS caches all HDD reads/writes, but if SQL light forces sync > after every DB update/insert it can make sense - I try it. > May be I move all FS dir to ramdisk. > > 2. Slightly change mod_sofia to disable NAT ping loop (unnecessary DB > operations) and, mainly, disable retrieving and sending of NOTIFY > messages containing VM info. > I'll look into my notes to see what I have done before and do the same... > > > Be patient please, as soon as I have the results, I post them here :-) > > > Best regards, > > kokoska.rokoska > > > >> On Tue, Feb 3, 2009 at 4:11 PM, kokoska.rokoska > > wrote: >> >> Ken Rice napsal(a): >> ... >> >> > On Registrations we have experienced Registration/second rates >> exceeding 150 >> > registrations per second using mod_xml_curl to feed the users >> directory. I >> > suspect, this number can be greatly increased if we were to feed >> directory >> > with something that cut out the apache and php over head >> > >> >> If someone interested I have few numbers on Registrar performance: >> >> DB server: >> 2x Quad core E5345 @ 2.33GHz, 16 GiB RAM >> Centos 5 x86_64, MySQL 5.0 >> >> Registrar server: >> 2x Quad core E5345 @ 2.33GHz, 16 GiB RAM >> Centos 5 x86_64 >> >> Tested using sipp with 10.000 and 30.000 "users". >> >> >> FreeSWITCH as registrar - current trunk: >> 1. FreeSwitch si simply modified (code doing NAT-ping is commented >> out :-) >> 2. Directory is served through lighttpd and simple "C" binary doing one >> trivial select. Lighttpd runs on the same machine as FS. When I move >> lighhtpd to another machine, I cannot see any significat performance >> boost. >> >> Result: I can go up to the 470-500 reg/s. and FS is heavy overloaded and >> retransmissions occurs. >> >> >> Kamailio as registrar - 1.4.3. no TLS: >> 1. Kamailio runs with usrloc db_mode 3 (no caching) >> >> Result: I can go up to the 3500-3700 reg/s. and Kamailio server is at >> 0.3 load and all 8 cores are bellow 15 %. Without retransmissions. The >> limit is DB throughput. >> Just for "curiosity" I switched userloc to db_mode 2 (write back) and at >> 5000 regs/s I stopped the sipp test, because I saw the bottle neck >> becomes the server runnig sipp (very old P4 box). >> >> >> Conclusion: >> While I see amazing FreeSWITCH performance on INVITEs per seconds and >> concurrent calls (another galaxy from * point of view :-), if you have >> to handle lots of registrations per second, it is IMO better to use >> Kamailio/OpenSIPS/SER as separate registrar and "propagate" users to FS >> through SQL view. >> >> Hope this helps someone... >> >> Best regards, >> >> kokoska.rokoska >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> >> iax:guest at conference.freeswitch.org/888 >> >> googletalk:conf+888 at conference.freeswitch.org >> >> pstn:213-799-1400 >> >> >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > From pmhshz at gmail.com Thu Feb 5 00:42:15 2009 From: pmhshz at gmail.com (shehzad p) Date: Thu, 5 Feb 2009 00:42:15 -0800 (PST) Subject: [Freeswitch-users] Freeswitch freezes on increasing call traffic In-Reply-To: <191c3a030902040551w5d867deta55c98bd3a3c03f8@mail.gmail.com> References: <21701744.post@talk.nabble.com> <191c3a030901280601t3e185ec2m76295110e4888837@mail.gmail.com> <21729863.post@talk.nabble.com> <191c3a030901290831y2ff329b3maee6c1aafa3c68e2@mail.gmail.com> <21745312.post@talk.nabble.com> <21749375.post@talk.nabble.com> <191c3a030901300818i5d85a7aawa0c53b4e54892f40@mail.gmail.com> <21761523.post@talk.nabble.com> <191c3a030901310850s553f8c11oc4e559a710def6d2@mail.gmail.com> <21825226.post@talk.nabble.com> <191c3a030902040551w5d867deta55c98bd3a3c03f8@mail.gmail.com> Message-ID: <21847332.post@talk.nabble.com> Hi anthony, In my previous post I already attached the BT for the testing of FS 1.0.1 Posting it again, please find it on link==>http://www.nabble.com/file/p21825226/bt_new_arch.txt Now I got the same result while testing FS 1.0.3RC1, And its BT is also same... BT Link: http://www.nabble.com/file/p21847332/fs_1_0_3_bt_new_arch.log fs_1_0_3_bt_new_arch.log (Note: Same in the sens the functions listed in the sequence are almost same as before...) Anthony Minessale-2 wrote: > > If you still get a crash on SVN trunk please post the bt even if you think > it's the same, since it won't be > exactly the same, the line numbers etc will be accurate with our > development > code making it easier to debug. > > > On Wed, Feb 4, 2009 at 12:38 AM, shehzad p wrote: > >> >> Hi anthony, >> >> I Modified the whole architecture of call routing system, >> Now after getting required routes, script exit and, >> control comes back to Dialplan, and call is bridged there, >> And call hangup, CDR is posted to cdr.php file (using xml_cdr). >> >> So now there is no blocking statement (bridge or anything like that) in >> current javascript, It return back control instantly. >> >> So, setting up all above architecture... >> First I tested FS 1.0.1 , It get crashed two times, in interval of 3 to 5 >> hours and simultaneous call of about 100 to 150. >> BT is the same as before... >> http://www.nabble.com/file/p21825226/bt_new_arch.txt bt_new_arch.txt >> >> Now I am also testing 1.0.3RC1, and post it back if any found. >> >> Thanks >> msp >> >> >> Clearly you have an issue with your javascript code. >> >> You have the Garbage collector blocking in every thread. >> >> Are you doing any endless loops in your code where you do not check >> session.ready() as a condition for >> continuing the script? >> >> any time session.ready() fails you must immediately exit. >> >> Are you using session.execute to execute long blocking operations like >> bridging many calls or entering a conference? >> You should avoid doing this as all the collective scripts on the system >> share a common Garbage Collector provided by the >> JS engine and it can lead to the exact issues you describe if the code is >> not properly designed. >> >> What else does you script do that are things provided by FS such as >> playing >> files and executing applications. >> >> >> >> >> -- >> View this message in context: >> http://www.nabble.com/Freeswitch-freezes-on-increasing-call-traffic-tp21701744p21825226.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/Freeswitch-freezes-on-increasing-call-traffic-tp21701744p21847332.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From brian at freeswitch.org Thu Feb 5 00:46:55 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 5 Feb 2009 02:46:55 -0600 Subject: [Freeswitch-users] Freeswitch freezes on increasing call traffic In-Reply-To: <21847332.post@talk.nabble.com> References: <21701744.post@talk.nabble.com> <191c3a030901280601t3e185ec2m76295110e4888837@mail.gmail.com> <21729863.post@talk.nabble.com> <191c3a030901290831y2ff329b3maee6c1aafa3c68e2@mail.gmail.com> <21745312.post@talk.nabble.com> <21749375.post@talk.nabble.com> <191c3a030901300818i5d85a7aawa0c53b4e54892f40@mail.gmail.com> <21761523.post@talk.nabble.com> <191c3a030901310850s553f8c11oc4e559a710def6d2@mail.gmail.com> <21825226.post@talk.nabble.com> <191c3a030902040551w5d867deta55c98bd3a3c03f8@mail.gmail.com> <21847332.post@talk.nabble.com> Message-ID: <4C2A4389-BEF6-454B-8B5F-FDCCC76E174A@freeswitch.org> Can you give me the output of uname -a and the contents of /proc/ cpuinfo? Not sure I asked for this info already or not. Thanks, Brian On Feb 5, 2009, at 2:42 AM, shehzad p wrote: > > Hi anthony, > In my previous post I already attached the BT for the testing of FS > 1.0.1 > Posting it again, please find it on > link==>http://www.nabble.com/file/p21825226/bt_new_arch.txt > > Now I got the same result while testing FS 1.0.3RC1, And its BT is > also > same... BT Link: > http://www.nabble.com/file/p21847332/fs_1_0_3_bt_new_arch.log > fs_1_0_3_bt_new_arch.log > > (Note: Same in the sens the functions listed in the sequence are > almost same > as before...) > > > Anthony Minessale-2 wrote: >> >> If you still get a crash on SVN trunk please post the bt even if >> you think >> it's the same, since it won't be >> exactly the same, the line numbers etc will be accurate with our >> development >> code making it easier to debug. >> >> >> On Wed, Feb 4, 2009 at 12:38 AM, shehzad p wrote: >> >>> >>> Hi anthony, >>> >>> I Modified the whole architecture of call routing system, >>> Now after getting required routes, script exit and, >>> control comes back to Dialplan, and call is bridged there, >>> And call hangup, CDR is posted to cdr.php file (using xml_cdr). >>> >>> So now there is no blocking statement (bridge or anything like >>> that) in >>> current javascript, It return back control instantly. >>> >>> So, setting up all above architecture... >>> First I tested FS 1.0.1 , It get crashed two times, in interval of >>> 3 to 5 >>> hours and simultaneous call of about 100 to 150. >>> BT is the same as before... >>> http://www.nabble.com/file/p21825226/bt_new_arch.txt bt_new_arch.txt >>> >>> Now I am also testing 1.0.3RC1, and post it back if any found. >>> >>> Thanks >>> msp >>> >>> >>> Clearly you have an issue with your javascript code. >>> >>> You have the Garbage collector blocking in every thread. >>> >>> Are you doing any endless loops in your code where you do not check >>> session.ready() as a condition for >>> continuing the script? >>> >>> any time session.ready() fails you must immediately exit. >>> >>> Are you using session.execute to execute long blocking operations >>> like >>> bridging many calls or entering a conference? >>> You should avoid doing this as all the collective scripts on the >>> system >>> share a common Garbage Collector provided by the >>> JS engine and it can lead to the exact issues you describe if the >>> code is >>> not properly designed. >>> >>> What else does you script do that are things provided by FS such as >>> playing >>> files and executing applications. >>> >>> >>> >>> >>> -- >>> View this message in context: >>> http://www.nabble.com/Freeswitch-freezes-on-increasing-call-traffic-tp21701744p21825226.html >>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com > > >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com> > >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org > > >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org> > >> pstn:213-799-1400 >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > View this message in context: http://www.nabble.com/Freeswitch-freezes-on-increasing-call-traffic-tp21701744p21847332.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From kokoska.rokoska at post.cz Thu Feb 5 00:49:48 2009 From: kokoska.rokoska at post.cz (kokoska rokoska) Date: Thu, 05 Feb 2009 09:49:48 +0100 Subject: [Freeswitch-users] FreeSwitch setup as a "Dumb" SBC In-Reply-To: <498AA265.4060307@laposte.net> References: <4988C12E.1090109@post.cz> <191c3a030902031434v1129d684j18b09a954d871f7b@mail.gmail.com> <49896E8D.3010609@post.cz> <498AA265.4060307@laposte.net> Message-ID: <498AA82C.4050302@post.cz> rod napsal(a): > Hi, > > how many static xml files did you create for your test ? > > rod. > Hi rod, I created 10.000 files xml directory files, but all of them were "included" into main FreeSWITCH xml config (using preprocessor). So I hope => no disk reads, all in memory (freeswitch.xml.fsxml is memory mapped)... Best regards, kokoska.rokoska From krice at suspicious.org Thu Feb 5 00:55:08 2009 From: krice at suspicious.org (Ken Rice) Date: Thu, 05 Feb 2009 02:55:08 -0600 Subject: [Freeswitch-users] FreeSwitch setup as a "Dumb" SBC In-Reply-To: <498AA82C.4050302@post.cz> Message-ID: That file is memory mapped but it can still take an IO hit... Mounting fs/db in a ramdrive still helps out and it doesn't have to be that big of a ram drive for the testing you are doing > From: kokoska rokoska > Reply-To: > Date: Thu, 05 Feb 2009 09:49:48 +0100 > To: > Subject: Re: [Freeswitch-users] FreeSwitch setup as a "Dumb" SBC > > > > > rod napsal(a): >> Hi, >> >> how many static xml files did you create for your test ? >> >> rod. >> > > Hi rod, > > I created 10.000 files xml directory files, but all of them were > "included" into main FreeSWITCH xml config (using preprocessor). So I > hope => no disk reads, all in memory (freeswitch.xml.fsxml is memory > mapped)... > > Best regards, > > kokoska.rokoska > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From kokoska.rokoska at post.cz Thu Feb 5 01:07:37 2009 From: kokoska.rokoska at post.cz (kokoska rokoska) Date: Thu, 05 Feb 2009 10:07:37 +0100 Subject: [Freeswitch-users] FreeSwitch setup as a "Dumb" SBC In-Reply-To: References: Message-ID: <498AAC59.60407@post.cz> Ken Rice napsal(a): > That file is memory mapped but it can still take an IO hit... Mounting fs/db > in a ramdrive still helps out and it doesn't have to be that big of a ram > drive for the testing you are doing > Thank you very much, Ken, for that info! I do it like I wrote before, but I have to wait till the "blades" aren't busy. They are not only for my testing and the other users have "tasks" with higher priority than my sipp testing :-) As soon as I have the result, I'll post them. I'm pretty sure, that changes in FS code and "all in RAM" helps much :-) Best regards, kokoska.rokoska From Claudio.Cavalera at italtel.it Thu Feb 5 01:11:37 2009 From: Claudio.Cavalera at italtel.it (Cavalera Claudio Luigi) Date: Thu, 5 Feb 2009 10:11:37 +0100 Subject: [Freeswitch-users] Errors compiling trunk on a fresh system In-Reply-To: Message-ID: freeswitch-users-bounces at lists.freeswitch.org wrote: > This should now be fixed in trunk in revision 11632. Can you please > test and confirm. > > Mike I confirm it's fixed (revision 11649), thanks as always. 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If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. ----------------------------------------------------------------------------------------------------- From pmhshz at gmail.com Thu Feb 5 01:51:37 2009 From: pmhshz at gmail.com (shehzad p) Date: Thu, 5 Feb 2009 01:51:37 -0800 (PST) Subject: [Freeswitch-users] Freeswitch freezes on increasing call traffic In-Reply-To: <4C2A4389-BEF6-454B-8B5F-FDCCC76E174A@freeswitch.org> References: <21701744.post@talk.nabble.com> <191c3a030901280601t3e185ec2m76295110e4888837@mail.gmail.com> <21729863.post@talk.nabble.com> <191c3a030901290831y2ff329b3maee6c1aafa3c68e2@mail.gmail.com> <21745312.post@talk.nabble.com> <21749375.post@talk.nabble.com> <191c3a030901300818i5d85a7aawa0c53b4e54892f40@mail.gmail.com> <21761523.post@talk.nabble.com> <191c3a030901310850s553f8c11oc4e559a710def6d2@mail.gmail.com> <21825226.post@talk.nabble.com> <191c3a030902040551w5d867deta55c98bd3a3c03f8@mail.gmail.com> <21847332.post@talk.nabble.com> <4C2A4389-BEF6-454B-8B5F-FDCCC76E174A@freeswitch.org> Message-ID: <21848148.post@talk.nabble.com> HI Brian, Output of ulimit -a and /proc/cpuinfo is attached... http://www.nabble.com/file/p21848148/12_ulimit_and_cpuinfo.log 12_ulimit_and_cpuinfo.log BUT...................... I am running the freeswitch using below command (So ulimit set according to Anthony's previous post): =================================================================================== ulimit -d unlimited; ulimit -f unlimited; ulimit -i unlimited; ulimit -n 999999; ulimit -q unlimited ; ulimit -u unlimited; ulimit -v unlimited ; ulimit -x unlimited; ulimit -s 244; ulimit -l unlimited; freeswitch =================================================================================== Thanks msp Brian West-3 wrote: > > Can you give me the output of uname -a and the contents of /proc/ > cpuinfo? Not sure I asked for this info already or not. > > Thanks, > Brian > > On Feb 5, 2009, at 2:42 AM, shehzad p wrote: > >> >> Hi anthony, >> In my previous post I already attached the BT for the testing of FS >> 1.0.1 >> Posting it again, please find it on >> link==>http://www.nabble.com/file/p21825226/bt_new_arch.txt >> >> Now I got the same result while testing FS 1.0.3RC1, And its BT is >> also >> same... BT Link: >> http://www.nabble.com/file/p21847332/fs_1_0_3_bt_new_arch.log >> fs_1_0_3_bt_new_arch.log >> >> (Note: Same in the sens the functions listed in the sequence are >> almost same >> as before...) >> >> >> Anthony Minessale-2 wrote: >>> >>> If you still get a crash on SVN trunk please post the bt even if >>> you think >>> it's the same, since it won't be >>> exactly the same, the line numbers etc will be accurate with our >>> development >>> code making it easier to debug. >>> >>> >>> On Wed, Feb 4, 2009 at 12:38 AM, shehzad p wrote: >>> >>>> >>>> Hi anthony, >>>> >>>> I Modified the whole architecture of call routing system, >>>> Now after getting required routes, script exit and, >>>> control comes back to Dialplan, and call is bridged there, >>>> And call hangup, CDR is posted to cdr.php file (using xml_cdr). >>>> >>>> So now there is no blocking statement (bridge or anything like >>>> that) in >>>> current javascript, It return back control instantly. >>>> >>>> So, setting up all above architecture... >>>> First I tested FS 1.0.1 , It get crashed two times, in interval of >>>> 3 to 5 >>>> hours and simultaneous call of about 100 to 150. >>>> BT is the same as before... >>>> http://www.nabble.com/file/p21825226/bt_new_arch.txt bt_new_arch.txt >>>> >>>> Now I am also testing 1.0.3RC1, and post it back if any found. >>>> >>>> Thanks >>>> msp >>>> >>>> >>>> Clearly you have an issue with your javascript code. >>>> >>>> You have the Garbage collector blocking in every thread. >>>> >>>> Are you doing any endless loops in your code where you do not check >>>> session.ready() as a condition for >>>> continuing the script? >>>> >>>> any time session.ready() fails you must immediately exit. >>>> >>>> Are you using session.execute to execute long blocking operations >>>> like >>>> bridging many calls or entering a conference? >>>> You should avoid doing this as all the collective scripts on the >>>> system >>>> share a common Garbage Collector provided by the >>>> JS engine and it can lead to the exact issues you describe if the >>>> code is >>>> not properly designed. >>>> >>>> What else does you script do that are things provided by FS such as >>>> playing >>>> files and executing applications. >>>> >>>> >>>> >>>> >>>> -- >>>> View this message in context: >>>> http://www.nabble.com/Freeswitch-freezes-on-increasing-call-traffic-tp21701744p21825226.html >>>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> >>> -- >>> Anthony Minessale II >>> >>> FreeSWITCH http://www.freeswitch.org/ >>> ClueCon http://www.cluecon.com/ >>> >>> AIM: anthm >>> MSN:anthony_minessale at hotmail.com >> > >>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com>> > >>> IRC: irc.freenode.net #freeswitch >>> >>> FreeSWITCH Developer Conference >>> sip:888 at conference.freeswitch.org >> > >>> iax:guest at conference.freeswitch.org/888 >>> googletalk:conf+888 at conference.freeswitch.org>> > >>> pstn:213-799-1400 >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> -- >> View this message in context: >> http://www.nabble.com/Freeswitch-freezes-on-increasing-call-traffic-tp21701744p21847332.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/Freeswitch-freezes-on-increasing-call-traffic-tp21701744p21848148.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From helmut.kuper at ewetel.de Thu Feb 5 01:57:21 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Thu, 05 Feb 2009 10:57:21 +0100 Subject: [Freeswitch-users] Q931 decoding Update In-Reply-To: <2ea4d47e0902042111x5aa08410oe99c6ae02df3d7de@mail.gmail.com> References: <4972046E.8020102@ewetel.de> <4981B8E6.9070708@ewetel.de> <5A2ED995-CF05-4AEF-9103-43BC84465514@freeswitch.org> <4981D6F8.4060409@ewetel.de> <191c3a030901290832k27a18b30u7bc531157fd89b19@mail.gmail.com> <49830576.6080907@ewetel.de> <4987359B.2020402@ewetel.de> <3D009334-DCFD-4AE5-A446-6C8165F2D77D@jerris.com> <2ea4d47e0902040956v75c5472foa4649c50b7340484@mail.gmail.com> <87f2f3b90902041252h5ede448bq720c15ea23dd517a@mail.gmail.com> <2ea4d47e0902042111x5aa08410oe99c6ae02df3d7de@mail.gmail.com> Message-ID: <498AB801.7030007@ewetel.de> Hello, if you have more than one call in your q931.pcap file captured, you may like to seperate the call flows in wireshark's packet list. wireshark allows you to sort the packets by e.g. q931 call reference first. All you have to do is this: Open q931 pcap file in wireshark, goto edit->preferences...->Columns Enter a title of your new column e.g "Q931 Call Ref" in title-field. Select "Custom" from Format-field. Enter exactly "q931.call_ref" without quotes into the field next to Format-field. Then apply it and close the window. Now you have a "Q931 Call Ref" column in the packet list. Click on it and the Flows a sorted first by "q931 call reference" and second by time. regards Helmut From Laurent.Fabre at kirranet.com Thu Feb 5 03:29:10 2009 From: Laurent.Fabre at kirranet.com (Laurent Fabre) Date: Thu, 5 Feb 2009 12:29:10 +0100 Subject: [Freeswitch-users] Directory User Password Message-ID: Hi everyone, Any chance I could populate user accounts with hashed passwords instead of cleartext ? If not, which block of code should I look for to propose a patch ? Thanks in advance, Laurent -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090205/616db3b6/attachment-0002.html From hads at nice.net.nz Thu Feb 5 03:39:06 2009 From: hads at nice.net.nz (Hadley Rich) Date: Fri, 6 Feb 2009 00:39:06 +1300 Subject: [Freeswitch-users] Directory User Password In-Reply-To: References: Message-ID: <200902060039.07078.hads@nice.net.nz> On Friday 06 February 2009 00:29:10 Laurent Fabre wrote: > Hi everyone, > > Any chance I could populate user accounts with hashed passwords instead of > cleartext ? Sure can; http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide#Basic_User hads -- http://nicegear.co.nz New Zealands Open Source Hardware Supplier From jacredit at gmail.com Wed Feb 4 22:06:17 2009 From: jacredit at gmail.com (John Hyde) Date: Wed, 4 Feb 2009 22:06:17 -0800 Subject: [Freeswitch-users] does anyone have a working FS / aastra config Message-ID: <777d76f40902042206u7c448985i8690c6df4472f3b7@mail.gmail.com> I am having problems getting an Aastra 57i to make calls through FS. the phone registers fine, but all calls fail. If i use xlite or a nokia sip phone, i have no problems. Here is a packet capture of an attempted call: http://pastebin.freeswitch.org/7039 notice packet 9, it should have been a SIP INVITE, but it turned out to be a Fragmented IP protocol The phone and FS are both on the same lan subnet, and the phone connects fine with an asterisk server on the same subnet. Is there a known config for aastra phones that I can reference, or does anyone know why I am having this issue? -- john -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090204/39a5e4c6/attachment-0002.html From jsokulski at dotsystems.pl Thu Feb 5 05:09:15 2009 From: jsokulski at dotsystems.pl (Jacek Sokulski) Date: Thu, 05 Feb 2009 14:09:15 +0100 Subject: [Freeswitch-users] origainate through sofia gateway In-Reply-To: <191c3a030902040822o1ffededbwd592361aa09f6b46@mail.gmail.com> References: <1233668179.5354.15.camel@dotw1126.dotsystems.pl> <1b46b4e80902030920p521d6079s4c9b78d8d39a924a@mail.gmail.com> <5E2B217D-8C85-48DB-800A-535A7FAAE24B@freeswitch.org> <1b46b4e80902030931y6a1679afw12ec37596ba9d77d@mail.gmail.com> <1233737494.5405.12.camel@dotw1126.dotsystems.pl> <191c3a030902040609s600798c8t720950a2e13ee05a@mail.gmail.com> <1233758767.5405.22.camel@dotw1126.dotsystems.pl> <191c3a030902040822o1ffededbwd592361aa09f6b46@mail.gmail.com> Message-ID: <1233839355.5346.31.camel@dotw1126.dotsystems.pl> the result of apiExecute("bgapi", "originate {effective_caller_id_number=fixed0248b,origination_caller_id_number=1000,ignore_early_media=true}user/1008 at 192.168.1.122 bridge:sofia/gateway/halonet/0225490317 inline"); > /: > 2009-02-05 14:04:48 [DEBUG] switch_ivr_originate.c:777 switch_ivr_originate() variable string 0 = [effective_caller_id_number=fixed0248b] > 2009-02-05 14:04:48 [DEBUG] switch_ivr_originate.c:777 switch_ivr_originate() variable string 1 = [origination_caller_id_number=1000] > 2009-02-05 14:04:48 [DEBUG] switch_ivr_originate.c:777 switch_ivr_originate() variable string 2 = [ignore_early_media=true] > 2009-02-05 14:04:48 [DEBUG] switch_ivr_originate.c:777 switch_ivr_originate() variable string 0 = [presence_id=1008 at 192.168.1.122] > 2009-02-05 14:04:48 [NOTICE] switch_channel.c:565 switch_channel_set_name() New Channel sofia/internal/sip:1008 at 192.168.1.126:5070 [f9b18b2e-b0a9-4e24-8452-40e1cce047bd] > 2009-02-05 14:04:48 [DEBUG] mod_sofia.c:2518 sofia_outgoing_channel() (sofia/internal/sip:1008 at 192.168.1.126:5070) State Change CS_NEW -> CS_INIT > 2009-02-05 14:04:48 [DEBUG] switch_core_session.c:807 switch_core_session_signal_state_change() Send signal sofia/internal/sip:1008 at 192.168.1.126:5070 [BREAK] > 2009-02-05 14:04:48 [DEBUG] switch_core_state_machine.c:379 switch_core_session_run() (sofia/internal/sip:1008 at 192.168.1.126:5070) Running State Change CS_INIT > 2009-02-05 14:04:48 [DEBUG] switch_core_state_machine.c:444 switch_core_session_run() (sofia/internal/sip:1008 at 192.168.1.126:5070) State INIT > 2009-02-05 14:04:48 [DEBUG] mod_sofia.c:83 sofia_on_init() sofia/internal/sip:1008 at 192.168.1.126:5070 SOFIA INIT > send 1196 bytes to udp/[192.168.1.126]:5070 at 13:04:48.127004: > ------------------------------------------------------------------------ > INVITE sip:1008 at 192.168.1.126:5070 SIP/2.0 > Via: SIP/2.0/UDP 192.168.1.122;rport;branch=z9hG4bKFpQjUmp1vNHZB > Max-Forwards: 70 > From: "FreeSWITCH" ;tag=9vpFm3tBc8gSe > To: > Call-ID: 68736fc7-6e28-122c-8396-000c7680c408 > CSeq: 110801208 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.2-wyeksportowane > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH > Supported: timer, precondition, path, replaces > Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer > Min-SE: 120 > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 335 > Remote-Party-ID: "FreeSWITCH" ;screen=yes;privacy=off > > v=0 > o=FreeSWITCH 1927164999227225404 3276738624485570081 IN IP4 192.168.1.122 > s=FreeSWITCH > c=IN IP4 192.168.1.122 > t=0 0 > m=audio 27044 RTP/AVP 9 0 8 3 101 13 > a=rtpmap:9 G722/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=rtpmap:13 CN/8000 > a=ptime:20 > ------------------------------------------------------------------------ > 2009-02-05 14:04:48 [DEBUG] mod_sofia.c:111 sofia_on_init() (sofia/internal/sip:1008 at 192.168.1.126:5070) State Change CS_INIT -> CS_ROUTING > 2009-02-05 14:04:48 [DEBUG] switch_core_session.c:807 switch_core_session_signal_state_change() Send signal sofia/internal/sip:1008 at 192.168.1.126:5070 [BREAK] > 2009-02-05 14:04:48 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state() Channel sofia/internal/sip:1008 at 192.168.1.126:5070 entering state [calling] > 2009-02-05 14:04:48 [DEBUG] switch_core_state_machine.c:444 switch_core_session_run() (sofia/internal/sip:1008 at 192.168.1.126:5070) State INIT going to sleep > 2009-02-05 14:04:48 [DEBUG] switch_core_state_machine.c:379 switch_core_session_run() (sofia/internal/sip:1008 at 192.168.1.126:5070) Running State Change CS_ROUTING > 2009-02-05 14:04:48 [DEBUG] switch_core_state_machine.c:447 switch_core_session_run() (sofia/internal/sip:1008 at 192.168.1.126:5070) State ROUTING > 2009-02-05 14:04:48 [DEBUG] mod_sofia.c:130 sofia_on_routing() sofia/internal/sip:1008 at 192.168.1.126:5070 SOFIA ROUTING > 2009-02-05 14:04:48 [DEBUG] switch_ivr_originate.c:58 originate_on_routing() (sofia/internal/sip:1008 at 192.168.1.126:5070) State Change CS_ROUTING -> CS_CONSUME_MEDIA > 2009-02-05 14:04:48 [DEBUG] switch_core_session.c:807 switch_core_session_signal_state_change() Send signal sofia/internal/sip:1008 at 192.168.1.126:5070 [BREAK] > 2009-02-05 14:04:48 [DEBUG] switch_core_state_machine.c:447 switch_core_session_run() (sofia/internal/sip:1008 at 192.168.1.126:5070) State ROUTING going to sleep > 2009-02-05 14:04:48 [DEBUG] switch_core_state_machine.c:379 switch_core_session_run() (sofia/internal/sip:1008 at 192.168.1.126:5070) Running State Change CS_CONSUME_MEDIA > 2009-02-05 14:04:48 [DEBUG] switch_core_state_machine.c:466 switch_core_session_run() (sofia/internal/sip:1008 at 192.168.1.126:5070) State CONSUME_MEDIA > recv 361 bytes from udp/[192.168.1.126]:5070 at 13:04:48.130745: > ------------------------------------------------------------------------ > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 192.168.1.122;rport;branch=z9hG4bKFpQjUmp1vNHZB > From: "FreeSWITCH" ;tag=9vpFm3tBc8gSe > To: ;tag=2111843199 > Contact: > Call-ID: 68736fc7-6e28-122c-8396-000c7680c408 > CSeq: 110801208 INVITE > Server: X-Lite release 1105d > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 362 bytes from udp/[192.168.1.126]:5070 at 13:04:48.181571: > ------------------------------------------------------------------------ > SIP/2.0 180 Ringing > Via: SIP/2.0/UDP 192.168.1.122;rport;branch=z9hG4bKFpQjUmp1vNHZB > From: "FreeSWITCH" ;tag=9vpFm3tBc8gSe > To: ;tag=2111843199 > Contact: > Call-ID: 68736fc7-6e28-122c-8396-000c7680c408 > CSeq: 110801208 INVITE > Server: X-Lite release 1105d > Content-Length: 0 > > ------------------------------------------------------------------------ > 2009-02-05 14:04:48 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state() Channel sofia/internal/sip:1008 at 192.168.1.126:5070 entering state [proceeding] > 2009-02-05 14:04:48 [NOTICE] sofia.c:2596 sofia_handle_sip_i_state() Ring-Ready sofia/internal/sip:1008 at 192.168.1.126:5070! > recv 678 bytes from udp/[192.168.1.126]:5070 at 13:04:51.914714: > ------------------------------------------------------------------------ > SIP/2.0 200 Ok > Via: SIP/2.0/UDP 192.168.1.122;rport;branch=z9hG4bKFpQjUmp1vNHZB > From: "FreeSWITCH" ;tag=9vpFm3tBc8gSe > To: ;tag=2111843199 > Contact: > Call-ID: 68736fc7-6e28-122c-8396-000c7680c408 > CSeq: 110801208 INVITE > Content-Type: application/sdp > Server: X-Lite release 1105d > Content-Length: 288 > > v=0 > o=1008 1183460117 1183463903 IN IP4 192.168.1.126 > s=X-Lite > c=IN IP4 192.168.1.126 > t=0 0 > m=audio 9000 RTP/AVP 0 8 98 97 101 > a=rtpmap:0 pcmu/8000 > a=rtpmap:8 pcma/8000 > a=rtpmap:98 iLBC/8000 > a=rtpmap:97 speex/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=sendrecv > ------------------------------------------------------------------------ > send 372 bytes to udp/[192.168.1.126]:5070 at 13:04:51.915303: > ------------------------------------------------------------------------ > ACK sip:1008 at 192.168.1.126:5070 SIP/2.0 > Via: SIP/2.0/UDP 192.168.1.122;rport;branch=z9hG4bKgZgBXF74Sy7HQ > Max-Forwards: 70 > From: "FreeSWITCH" ;tag=9vpFm3tBc8gSe > To: ;tag=2111843199 > Call-ID: 68736fc7-6e28-122c-8396-000c7680c408 > CSeq: 110801208 ACK > Contact: > Content-Length: 0 > > ------------------------------------------------------------------------ > 2009-02-05 14:04:51 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state() Channel sofia/internal/sip:1008 at 192.168.1.126:5070 entering state [ready] > 2009-02-05 14:04:51 [DEBUG] sofia.c:2546 sofia_handle_sip_i_state() Remote SDP: > v=0 > o=1008 1183460117 1183463903 IN IP4 192.168.1.126 > s=X-Lite > c=IN IP4 192.168.1.126 > t=0 0 > m=audio 9000 RTP/AVP 0 8 98 97 101 > a=rtpmap:0 pcmu/8000 > a=rtpmap:8 pcma/8000 > a=rtpmap:98 iLBC/8000 > a=rtpmap:97 speex/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > > 2009-02-05 14:04:51 [DEBUG] sofia_glue.c:2509 sofia_glue_negotiate_sdp() Audio Codec Compare [pcmu:0:8000]/[G722:9:8000] > 2009-02-05 14:04:51 [DEBUG] sofia_glue.c:2509 sofia_glue_negotiate_sdp() Audio Codec Compare [pcmu:0:8000]/[PCMU:0:8000] > 2009-02-05 14:04:51 [DEBUG] sofia_glue.c:1670 sofia_glue_tech_set_codec() Set Codec sofia/internal/sip:1008 at 192.168.1.126:5070 PCMU/8000 20 ms 160 samples > 2009-02-05 14:04:51 [DEBUG] sofia_glue.c:2473 sofia_glue_negotiate_sdp() Set 2833 dtmf payload to 101 > 2009-02-05 14:04:51 [DEBUG] sofia_glue.c:1890 sofia_glue_activate_rtp() AUDIO RTP [sofia/internal/sip:1008 at 192.168.1.126:5070] 192.168.1.122 port 27044 -> 192.168.1.126 port 9000 codec: 0 ms: 20 > 2009-02-05 14:04:51 [DEBUG] switch_rtp.c:865 switch_rtp_create() Starting timer [soft] 160 bytes per 20000ms > 2009-02-05 14:04:51 [NOTICE] sofia.c:3031 sofia_handle_sip_i_state() Channel [sofia/internal/sip:1008 at 192.168.1.126:5070] has been answered > 2009-02-05 14:04:51 [DEBUG] switch_channel.c:177 switch_channel_audio_sync() sofia/internal/sip:1008 at 192.168.1.126:5070 receive message [AUDIO_SYNC] > 2009-02-05 14:04:51 [DEBUG] switch_ivr_originate.c:1627 switch_ivr_originate() Originate Resulted in Success: [sofia/internal/sip:1008 at 192.168.1.126:5070] > 2009-02-05 14:04:51 [DEBUG] switch_channel.c:177 switch_channel_audio_sync() sofia/internal/sip:1008 at 192.168.1.126:5070 receive message [AUDIO_SYNC] > 2009-02-05 14:04:51 [DEBUG] switch_ivr_originate.c:1627 switch_ivr_originate() Originate Resulted in Success: [sofia/internal/sip:1008 at 192.168.1.126:5070] > 2009-02-05 14:04:51 [DEBUG] switch_channel.c:177 switch_channel_audio_sync() sofia/internal/sip:1008 at 192.168.1.126:5070 receive message [AUDIO_SYNC] > 2009-02-05 14:04:51 [DEBUG] switch_ivr.c:1245 switch_ivr_session_transfer() (sofia/internal/sip:1008 at 192.168.1.126:5070) State Change CS_CONSUME_MEDIA -> CS_ROUTING > 2009-02-05 14:04:51 [DEBUG] switch_core_session.c:807 switch_core_session_signal_state_change() Send signal sofia/internal/sip:1008 at 192.168.1.126:5070 [BREAK] > 2009-02-05 14:04:51 [DEBUG] switch_ivr.c:1249 switch_ivr_session_transfer() sofia/internal/sip:1008 at 192.168.1.126:5070 receive message [TRANSFER] > 2009-02-05 14:04:51 [DEBUG] switch_core_session.c:511 switch_core_session_perform_receive_message() Send signal sofia/internal/sip:1008 at 192.168.1.126:5070 [BREAK] > 2009-02-05 14:04:51 [NOTICE] switch_ivr.c:1251 switch_ivr_session_transfer() Transfer sofia/internal/sip:1008 at 192.168.1.126:5070 to inline[bridge:sofia/gateway/halonet/0225490317 at default] > 2009-02-05 14:04:51 [DEBUG] switch_core_state_machine.c:466 switch_core_session_run() (sofia/internal/sip:1008 at 192.168.1.126:5070) State CONSUME_MEDIA going to sleep > 2009-02-05 14:04:51 [DEBUG] switch_core_state_machine.c:379 switch_core_session_run() (sofia/internal/sip:1008 at 192.168.1.126:5070) Running State Change CS_ROUTING > 2009-02-05 14:04:51 [DEBUG] switch_core_state_machine.c:447 switch_core_session_run() (sofia/internal/sip:1008 at 192.168.1.126:5070) State ROUTING > 2009-02-05 14:04:51 [DEBUG] mod_sofia.c:130 sofia_on_routing() sofia/internal/sip:1008 at 192.168.1.126:5070 SOFIA ROUTING > 2009-02-05 14:04:51 [DEBUG] switch_core_state_machine.c:64 switch_core_standard_on_routing() sofia/internal/sip:1008 at 192.168.1.126:5070 Standard ROUTING > 2009-02-05 14:04:51 [DEBUG] switch_core_state_machine.c:100 switch_core_standard_on_routing() (sofia/internal/sip:1008 at 192.168.1.126:5070) State Change CS_ROUTING -> CS_EXECUTE > 2009-02-05 14:04:51 [DEBUG] switch_core_session.c:807 switch_core_session_signal_state_change() Send signal sofia/internal/sip:1008 at 192.168.1.126:5070 [BREAK] > 2009-02-05 14:04:51 [DEBUG] switch_core_state_machine.c:447 switch_core_session_run() (sofia/internal/sip:1008 at 192.168.1.126:5070) State ROUTING going to sleep > 2009-02-05 14:04:51 [DEBUG] switch_core_state_machine.c:379 switch_core_session_run() (sofia/internal/sip:1008 at 192.168.1.126:5070) Running State Change CS_EXECUTE > 2009-02-05 14:04:51 [DEBUG] switch_core_state_machine.c:454 switch_core_session_run() (sofia/internal/sip:1008 at 192.168.1.126:5070) State EXECUTE > 2009-02-05 14:04:51 [DEBUG] mod_sofia.c:173 sofia_on_execute() sofia/internal/sip:1008 at 192.168.1.126:5070 SOFIA EXECUTE > 2009-02-05 14:04:51 [DEBUG] switch_core_state_machine.c:137 switch_core_standard_on_execute() sofia/internal/sip:1008 at 192.168.1.126:5070 Standard EXECUTE > 2009-02-05 14:04:51 [DEBUG] switch_core_state_machine.c:152 switch_core_standard_on_execute() sofia/internal/sip:1008 at 192.168.1.126:5070 Execute bridge(sofia/gateway/halonet/0225490317) > 2009-02-05 14:04:51 [NOTICE] switch_channel.c:565 switch_channel_set_name() New Channel sofia/external/0225490317 [bf4fece9-38fc-40fa-8e9e-91f836f05e55] > 2009-02-05 14:04:51 [DEBUG] mod_sofia.c:2518 sofia_outgoing_channel() (sofia/external/0225490317) State Change CS_NEW -> CS_INIT > 2009-02-05 14:04:51 [DEBUG] switch_core_session.c:807 switch_core_session_signal_state_change() Send signal sofia/external/0225490317 [BREAK] > 2009-02-05 14:04:51 [DEBUG] switch_core_state_machine.c:379 switch_core_session_run() (sofia/external/0225490317) Running State Change CS_INIT > 2009-02-05 14:04:51 [DEBUG] switch_core_state_machine.c:444 switch_core_session_run() (sofia/external/0225490317) State INIT > 2009-02-05 14:04:51 [DEBUG] mod_sofia.c:83 sofia_on_init() sofia/external/0225490317 SOFIA INIT > 2009-02-05 14:04:52 [DEBUG] sofia_glue.c:566 sofia_glue_ext_address_lookup() STUN Success [89.77.89.244]:[50863] > 2009-02-05 14:04:52 [DEBUG] mod_sofia.c:111 sofia_on_init() (sofia/external/0225490317) State Change CS_INIT -> CS_ROUTING > 2009-02-05 14:04:52 [DEBUG] switch_core_session.c:807 switch_core_session_signal_state_change() Send signal sofia/external/0225490317 [BREAK] > 2009-02-05 14:04:52 [DEBUG] switch_core_state_machine.c:444 switch_core_session_run() (sofia/external/0225490317) State INIT going to sleep > 2009-02-05 14:04:52 [DEBUG] switch_core_state_machine.c:379 switch_core_session_run() (sofia/external/0225490317) Running State Change CS_ROUTING > 2009-02-05 14:04:52 [DEBUG] switch_core_state_machine.c:447 switch_core_session_run() (sofia/external/0225490317) State ROUTING > 2009-02-05 14:04:52 [DEBUG] mod_sofia.c:130 sofia_on_routing() sofia/external/0225490317 SOFIA ROUTING > 2009-02-05 14:04:52 [DEBUG] switch_ivr_originate.c:58 originate_on_routing() (sofia/external/0225490317) State Change CS_ROUTING -> CS_CONSUME_MEDIA > 2009-02-05 14:04:52 [DEBUG] switch_core_session.c:807 switch_core_session_signal_state_change() Send signal sofia/external/0225490317 [BREAK] > 2009-02-05 14:04:52 [DEBUG] switch_core_state_machine.c:447 switch_core_session_run() (sofia/external/0225490317) State ROUTING going to sleep > 2009-02-05 14:04:52 [DEBUG] switch_core_state_machine.c:379 switch_core_session_run() (sofia/external/0225490317) Running State Change CS_CONSUME_MEDIA > 2009-02-05 14:04:52 [DEBUG] switch_core_state_machine.c:466 switch_core_session_run() (sofia/external/0225490317) State CONSUME_MEDIA > 2009-02-05 14:04:52 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state() Channel sofia/external/0225490317 entering state [calling] > send 1117 bytes to udp/[194.9.25.21]:5060 at 13:04:52.572008: > ------------------------------------------------------------------------ > INVITE sip:0225490317 at sip.halonet.pl SIP/2.0 > Via: SIP/2.0/UDP 83.13.98.165:5080;rport;branch=z9hG4bKyvKBDBpvBQe2p > Max-Forwards: 69 > From: "Extension 1008" ;tag=FZp353yettFNN > To: > Call-ID: 6b05cee4-6e28-122c-8396-000c7680c408 > CSeq: 110801210 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.2-wyeksportowane > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO > Supported: timer, precondition, path, replaces > Allow-Events: talk, refer > Min-SE: 120 > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 309 > Remote-Party-ID: "Extension 1008" ;screen=yes;privacy=off > > v=0 > o=FreeSWITCH 3996368035436745512 7588776177330183331 IN IP4 89.77.89.244 > s=FreeSWITCH > c=IN IP4 89.77.89.244 > t=0 0 > m=audio 50863 RTP/AVP 0 8 3 101 13 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=rtpmap:13 CN/8000 > a=ptime:20 > ------------------------------------------------------------------------ > recv 760 bytes from udp/[194.9.25.21]:5060 at 13:04:52.993246: > ------------------------------------------------------------------------ > SIP/2.0 407 Proxy Authentication Required > Via: SIP/2.0/UDP 83.13.98.165:5080;rport=5080;branch=z9hG4bKyvKBDBpvBQe2p;received=89.77.89.244 > From: "Extension 1008" ;tag=FZp353yettFNN > To: ;tag=03bd8e75ec97c9ee65c772e401792a5a.5fa3 > Call-ID: 6b05cee4-6e28-122c-8396-000c7680c408 > CSeq: 110801210 INVITE > Proxy-Authenticate: Digest realm="sip.halonet.pl", nonce="498ae51439454939d240f51ab9455e2c8505c7aa", qop="auth" > Server: Sip EXpress router (2.0.0-rc1 (i386/linux)) > Content-Length: 0 > Warning: 392 194.9.25.21:5060 "Noisy feedback tells: pid=22400 req_src_ip=89.77.89.244 req_src_port=50866 in_uri=sip:0225490317 at sip.halonet.pl out_uri=sip:0225490317 at sip.halonet.pl via_cnt==1" > > ------------------------------------------------------------------------ > send 387 bytes to udp/[194.9.25.21]:5060 at 13:04:52.993616: > ------------------------------------------------------------------------ > ACK sip:0225490317 at sip.halonet.pl SIP/2.0 > Via: SIP/2.0/UDP 83.13.98.165:5080;rport;branch=z9hG4bKyvKBDBpvBQe2p > Max-Forwards: 69 > From: "Extension 1008" ;tag=FZp353yettFNN > To: ;tag=03bd8e75ec97c9ee65c772e401792a5a.5fa3 > Call-ID: 6b05cee4-6e28-122c-8396-000c7680c408 > CSeq: 110801210 ACK > Content-Length: 0 > > ------------------------------------------------------------------------ > send 1409 bytes to udp/[194.9.25.21]:5060 at 13:04:52.994530: > ------------------------------------------------------------------------ > INVITE sip:0225490317 at sip.halonet.pl SIP/2.0 > Via: SIP/2.0/UDP 83.13.98.165:5080;rport;branch=z9hG4bKZ5c4e66Z8Z4mj > Max-Forwards: 69 > From: "Extension 1008" ;tag=FZp353yettFNN > To: > Call-ID: 6b05cee4-6e28-122c-8396-000c7680c408 > CSeq: 110801211 INVITE > Contact: > Expires: 600 > User-Agent: FreeSWITCH-mod_sofia/1.0.2-wyeksportowane > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO > Supported: timer, precondition, path, replaces > Allow-Events: talk, refer > Proxy-Authorization: Digest username="fixed0248b", realm="sip.halonet.pl", nonce="498ae51439454939d240f51ab9455e2c8505c7aa", cnonce="a1oY524oEiyWgwAMdoDECA", algorithm=MD5, uri="sip:0225490317 at sip.halonet.pl", response="f05fad749d4ff6d5a75d61e3283c3afa", qop=auth, nc=00000001 > Min-SE: 120 > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 309 > Remote-Party-ID: "Extension 1008" ;screen=yes;privacy=off > > v=0 > o=FreeSWITCH 3996368035436745512 7588776177330183331 IN IP4 89.77.89.244 > s=FreeSWITCH > c=IN IP4 89.77.89.244 > t=0 0 > m=audio 50863 RTP/AVP 0 8 3 101 13 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=rtpmap:13 CN/8000 > a=ptime:20 > ------------------------------------------------------------------------ > 2009-02-05 14:04:52 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state() Channel sofia/external/0225490317 entering state [calling] > recv 618 bytes from udp/[194.9.25.21]:5060 at 13:04:53.349700: > ------------------------------------------------------------------------ > SIP/2.0 100 trying -- your call is important to us > Via: SIP/2.0/UDP 83.13.98.165:5080;rport=5080;branch=z9hG4bKZ5c4e66Z8Z4mj;received=89.77.89.244 > From: "Extension 1008" ;tag=FZp353yettFNN > To: > Call-ID: 6b05cee4-6e28-122c-8396-000c7680c408 > CSeq: 110801211 INVITE > Server: Sip EXpress router (2.0.0-rc1 (i386/linux)) > Content-Length: 0 > Warning: 392 194.9.25.21:5060 "Noisy feedback tells: pid=22332 req_src_ip=89.77.89.244 req_src_port=50866 in_uri=sip:0225490317 at sip.halonet.pl out_uri=sip:0225490317 at 217.11.128.50:5060 via_cnt==1" > > ------------------------------------------------------------------------ > recv 469 bytes from udp/[194.9.25.21]:5060 at 13:04:53.861812: > ------------------------------------------------------------------------ > SIP/2.0 500 Internal Server Error > Via: SIP/2.0/UDP 83.13.98.165:5080;received=89.77.89.244;rport=5080;branch=z9hG4bKZ5c4e66Z8Z4mj > From: "Extension 1008" ;tag=FZp353yettFNN > To: ;tag=4961D2C4-DA7 > Date: Thu, 05 Feb 2009 13:04:39 GMT > Call-ID: 6b05cee4-6e28-122c-8396-000c7680c408 > Server: Cisco-SIPGateway/IOS-12.x > CSeq: 110801211 INVITE > Allow-Events: telephone-event > Content-Length: 0 > > ------------------------------------------------------------------------ > send 362 bytes to udp/[194.9.25.21]:5060 at 13:04:53.862199: > ------------------------------------------------------------------------ > ACK sip:0225490317 at sip.halonet.pl SIP/2.0 > Via: SIP/2.0/UDP 83.13.98.165:5080;rport;branch=z9hG4bKZ5c4e66Z8Z4mj > Max-Forwards: 69 > From: "Extension 1008" ;tag=FZp353yettFNN > To: ;tag=4961D2C4-DA7 > Call-ID: 6b05cee4-6e28-122c-8396-000c7680c408 > CSeq: 110801211 ACK > Content-Length: 0 > > ------------------------------------------------------------------------ > 2009-02-05 14:04:53 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state() Channel sofia/external/0225490317 entering state [terminated] > 2009-02-05 14:04:53 [NOTICE] sofia.c:3090 sofia_handle_sip_i_state() Hangup sofia/external/0225490317 [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE] > 2009-02-05 14:04:53 [DEBUG] switch_channel.c:1494 switch_channel_perform_hangup() Send signal sofia/external/0225490317 [KILL] > 2009-02-05 14:04:53 [DEBUG] switch_core_session.c:807 switch_core_session_signal_state_change() Send signal sofia/external/0225490317 [BREAK] > 2009-02-05 14:04:53 [DEBUG] switch_core_state_machine.c:466 switch_core_session_run() (sofia/external/0225490317) State CONSUME_MEDIA going to sleep > 2009-02-05 14:04:53 [DEBUG] switch_core_state_machine.c:379 switch_core_session_run() (sofia/external/0225490317) Running State Change CS_HANGUP > 2009-02-05 14:04:53 [DEBUG] switch_core_state_machine.c:410 switch_core_session_run() (sofia/external/0225490317) State HANGUP > 2009-02-05 14:04:53 [DEBUG] mod_sofia.c:253 sofia_on_hangup() sofia/external/0225490317 Overriding SIP cause 503 with 500 from the other leg > 2009-02-05 14:04:53 [DEBUG] mod_sofia.c:287 sofia_on_hangup() Channel sofia/external/0225490317 hanging up, cause: NORMAL_TEMPORARY_FAILURE > 2009-02-05 14:04:53 [DEBUG] switch_core_state_machine.c:46 switch_core_standard_on_hangup() sofia/external/0225490317 Standard HANGUP, cause: NORMAL_TEMPORARY_FAILURE > 2009-02-05 14:04:53 [DEBUG] switch_core_state_machine.c:410 switch_core_session_run() (sofia/external/0225490317) State HANGUP going to sleep > 2009-02-05 14:04:53 [DEBUG] switch_core_session.c:939 switch_core_session_thread() Session 16 (sofia/external/0225490317) Locked, Waiting on external entities > 2009-02-05 14:04:53 [DEBUG] switch_ivr_originate.c:1695 switch_ivr_originate() Originate Resulted in Error Cause: 41 [NORMAL_TEMPORARY_FAILURE] > 2009-02-05 14:04:53 [DEBUG] switch_channel.c:177 switch_channel_audio_sync() sofia/internal/sip:1008 at 192.168.1.126:5070 receive message [AUDIO_SYNC] > 2009-02-05 14:04:53 [NOTICE] switch_core_session.c:957 switch_core_session_thread() Session 16 (sofia/external/0225490317) Ended > 2009-02-05 14:04:53 [INFO] mod_dptools.c:1909 audio_bridge_function() Originate Failed. Cause: NORMAL_TEMPORARY_FAILURE > 2009-02-05 14:04:53 [NOTICE] mod_dptools.c:1936 audio_bridge_function() Hangup sofia/internal/sip:1008 at 192.168.1.126:5070 [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE] > 2009-02-05 14:04:53 [NOTICE] switch_core_session.c:959 switch_core_session_thread() Close Channel sofia/external/0225490317 [CS_HANGUP] > 2009-02-05 14:04:53 [DEBUG] switch_channel.c:1494 switch_channel_perform_hangup() Send signal sofia/internal/sip:1008 at 192.168.1.126:5070 [KILL] > 2009-02-05 14:04:53 [DEBUG] switch_core_session.c:807 switch_core_session_signal_state_change() Send signal sofia/internal/sip:1008 at 192.168.1.126:5070 [BREAK] > 2009-02-05 14:04:53 [DEBUG] switch_core_state_machine.c:454 switch_core_session_run() (sofia/internal/sip:1008 at 192.168.1.126:5070) State EXECUTE going to sleep > 2009-02-05 14:04:53 [DEBUG] switch_core_state_machine.c:379 switch_core_session_run() (sofia/internal/sip:1008 at 192.168.1.126:5070) Running State Change CS_HANGUP > 2009-02-05 14:04:53 [DEBUG] switch_core_state_machine.c:410 switch_core_session_run() (sofia/internal/sip:1008 at 192.168.1.126:5070) State HANGUP > 2009-02-05 14:04:53 [DEBUG] mod_sofia.c:253 sofia_on_hangup() sofia/internal/sip:1008 at 192.168.1.126:5070 Overriding SIP cause 503 with 500 from the other leg > 2009-02-05 14:04:53 [DEBUG] mod_sofia.c:287 sofia_on_hangup() Channel sofia/internal/sip:1008 at 192.168.1.126:5070 hanging up, cause: NORMAL_TEMPORARY_FAILURE > 2009-02-05 14:04:53 [DEBUG] mod_sofia.c:344 sofia_on_hangup() Sending BYE to sofia/internal/sip:1008 at 192.168.1.126:5070 > send 648 bytes to udp/[192.168.1.126]:5070 at 13:04:53.868506: > ------------------------------------------------------------------------ > BYE sip:1008 at 192.168.1.126:5070 SIP/2.0 > Via: SIP/2.0/UDP 192.168.1.122;rport;branch=z9hG4bKH893yar8p7X4j > Max-Forwards: 70 > From: "FreeSWITCH" ;tag=9vpFm3tBc8gSe > To: ;tag=2111843199 > Call-ID: 68736fc7-6e28-122c-8396-000c7680c408 > CSeq: 110801209 BYE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.2-wyeksportowane > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH > Supported: timer, precondition, path, replaces > Reason: Q.850;cause=41;text="NORMAL_TEMPORARY_FAILURE" > Content-Length: 0 > > ------------------------------------------------------------------------ > 2009-02-05 14:04:53 [DEBUG] switch_core_state_machine.c:46 switch_core_standard_on_hangup() sofia/internal/sip:1008 at 192.168.1.126:5070 Standard HANGUP, cause: NORMAL_TEMPORARY_FAILURE > 2009-02-05 14:04:53 [DEBUG] switch_core_state_machine.c:410 switch_core_session_run() (sofia/internal/sip:1008 at 192.168.1.126:5070) State HANGUP going to sleep > 2009-02-05 14:04:53 [DEBUG] switch_core_session.c:939 switch_core_session_thread() Session 15 (sofia/internal/sip:1008 at 192.168.1.126:5070) Locked, Waiting on external entities > 2009-02-05 14:04:53 [NOTICE] switch_core_session.c:957 switch_core_session_thread() Session 15 (sofia/internal/sip:1008 at 192.168.1.126:5070) Ended > 2009-02-05 14:04:53 [NOTICE] switch_core_session.c:959 switch_core_session_thread() Close Channel sofia/internal/sip:1008 at 192.168.1.126:5070 [CS_HANGUP] > recv 354 bytes from udp/[192.168.1.126]:5070 at 13:04:53.882344: > ------------------------------------------------------------------------ > SIP/2.0 200 Ok > Via: SIP/2.0/UDP 192.168.1.122;rport;branch=z9hG4bKH893yar8p7X4j > From: "FreeSWITCH" ;tag=9vpFm3tBc8gSe > To: ;tag=2111843199 > Contact: > Call-ID: 68736fc7-6e28-122c-8396-000c7680c408 > CSeq: 110801209 BYE > Server: X-Lite release 1105d > Content-Length: 0 > > ------------------------------------------------------------------------ Dnia 04-02-2009, ?ro o godzinie 10:22 -0600, Anthony Minessale pisze: > can you press f8 for debug and try that apiExecute and post the > results? > > On Wed, Feb 4, 2009 at 8:46 AM, Jacek Sokulski > wrote: > Thanks Anthony, > the js snippets are very instructive. > A couple of points: > 1. The code with apiExecute does not work (local phone is > connected, but > after picking up it hungs up immediately), other examples are > working > fine. > > 2. It does not show how initiate external call without > existing session. > > 3. How can one pass the call through dialplan? > > Jacek > > PS. > we got the code probable from wiki or from this mialing list. > > Dnia 04-02-2009, ?ro o godzinie 08:09 -0600, Anthony Minessale > pisze: > > > Where did you learn how to use js this way? > > session.originate is being misused here and is depricated > and may be > > removed. > > > > the first arg to session.originate is either undefined or a > > *different* session (the a leg) > > > > session1 = new Session(); > > session1.originate(undefined, > > "{ignore_early_media=true}user/1008 at 192.168.1.122"); > > > > > session1.setVariable("effective_caller_id_number","fixed0248b"); > > > > //once you have session1 when you originate session2 you > pass session1 > > as the arg > > // the effective_caller_id is taken from session1 > > > > session2 = new Session(); > > session2.originate(session1, > "sofia/gateway/halonet/0225490317"); > > > > Anyway this whole code is depricated in favor of this: > > > > session1 = new > > Session("{ignore_early_media=true}user/1008 at 192.168.1.122"); > > > > if (session1.ready()) { > > > session1.setVariable("effective_caller_id_number","fixed0248b"); > > session2 = new Session("sofia/gateway/halonet/0225490317", > > session1); > > } > > > > and could be further refactored down to this: > > > > session1 = new > > Session("{ignore_early_media=true}user/1008 at 192.168.1.122"); > > if (session1.ready()) { > > > session1.setVariable("effective_caller_id_number","fixed0248b"); > > session1.execute("bridge", > "sofia/gateway/halonet/0225490317"); > > } > > > > or down to this one line of code that will setup the call > detached > > from the script and exit. > > > > var result = apiExecute("originate", > > > "{effective_caller_id_number=fixed0248b,origination_caller_id_number=1000,ignore_early_media=true}user/1008 at 192.168.1.122 bridge:sofia/gateway/halonet/0225490317 inline"); > > > > if you dont care about the result and want to exit even > before the > > call is completed. > > > > var result = apiExecute("bgapi", "originate > > > {effective_caller_id_number=fixed0248b,origination_caller_id_number=1000,ignore_early_media=true}user/1008 at 192.168.1.122 bridge:sofia/gateway/halonet/0225490317 inline"); > > > > > > > > On Wed, Feb 4, 2009 at 2:51 AM, Jacek Sokulski > > wrote: > > > > We have tried setting both > effective_caller_id_number and > > origination_caller_id_number: > > > > > session1.originate(session1,"{origination_caller_id_number=fixed0248b}sofia/gateway/halonet/0225490317",15); > > but the problem still exists. The solution we have > found for > > the case > > when we originate two calls, local and external, is > as follow: > > > > session1 = new Session(); > > > session1.originate(session1,"user/1003 at 192.168.1.122",15);//local > > if(session1.ready()) { > > > session1.execute("execute_extension","00930691688627 XML > > default");//external > > } > > > > so the external call goes through the dialplan. > > It does not work if both calls are external. One > possible > > solution could be > > to pass the originating call through dialplan > (loopback?) but > > we have not managed > > to figure out how to do it. > > > > Thanks > > Jacek > > > > Dnia 03-02-2009, wto o godzinie 14:31 -0300, Nicolas > Brenner > > pisze: > > > > > Oops! Well, fortunately I don't use that voip > provider > > anymore (nor the script). > > > > > > Thanks Brian. > > > > > > Nicolas > > > > > > On Tue, Feb 3, 2009 at 2:25 PM, Brian West > > wrote: > > > > YOU should NEVER use this method or call > setCallerData at > > all you > > > > should use the correct methods to override the > callerid. > > > > > > > > If its a B-Leg born from an A-Leg you use these > on the on > > the A-Leg: > > > > > > > > > > > http://wiki.freeswitch.org/wiki/Channel_Variables#effective_caller_id_name > > > > > > > http://wiki.freeswitch.org/wiki/Channel_Variables#effective_caller_id_number > > > > > > > > If you're originating you use this: > > > > > > > > > > > http://wiki.freeswitch.org/wiki/Channel_Variables#origination_caller_id_name > > > > > > > http://wiki.freeswitch.org/wiki/Channel_Variables#origination_caller_id_number > > > > > > > > /b > > > > > > _______________________________________________ > > > Freeswitch-users mailing list > > > Freeswitch-users at lists.freeswitch.org > > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From anthony.minessale at gmail.com Thu Feb 5 06:21:32 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 5 Feb 2009 08:21:32 -0600 Subject: [Freeswitch-users] Freeswitch freezes on increasing call traffic In-Reply-To: <21848148.post@talk.nabble.com> References: <21701744.post@talk.nabble.com> <21749375.post@talk.nabble.com> <191c3a030901300818i5d85a7aawa0c53b4e54892f40@mail.gmail.com> <21761523.post@talk.nabble.com> <191c3a030901310850s553f8c11oc4e559a710def6d2@mail.gmail.com> <21825226.post@talk.nabble.com> <191c3a030902040551w5d867deta55c98bd3a3c03f8@mail.gmail.com> <21847332.post@talk.nabble.com> <4C2A4389-BEF6-454B-8B5F-FDCCC76E174A@freeswitch.org> <21848148.post@talk.nabble.com> Message-ID: <191c3a030902050621k7534e2d4n35eec6dc36fd439a@mail.gmail.com> First of all please stop using the mailing list as a bug tracker. All issues should be put into jira and managed with that. Secondly, Didn't I ask you multiple times to stop using release snapshots and please use the SVN trunk? I don't understand why you keep ignoring me and using everything but what I asked. I am not telling you to use SVN because I think it will be fixed it's so we are on the development copy of the code to get the proper line numbers etc. If you look at your 2 bt you posted, the line numbers are different on each one. What are you using on the other side of ODBC? as you can see in your bt, the call goes into ODBC then into several libs with no symbols and crashes on free. This can be a sign of corrupt memory, running out of memory or an issue in either ODBC or the database specific lib. What distro is it? What ODBC version? unixODBC? version xxx? What database driver version xxx? Is it mysl not using the proper reentrant version of the plugin? Sometimes packaged libs have bugs in them which fall out of our control. Can you build unixODBC and the plugins yourself with debug symbols so we can see if that is the cause or at the very least then we can see the debug info in the bt. please make sure you address *all* my questions in your jira report. Starting with using svn trunk, *hint* type "make current" from your rc1 distro. On Thu, Feb 5, 2009 at 3:51 AM, shehzad p wrote: > > HI Brian, > > Output of ulimit -a and /proc/cpuinfo is attached... > http://www.nabble.com/file/p21848148/12_ulimit_and_cpuinfo.log > 12_ulimit_and_cpuinfo.log > > BUT...................... > I am running the freeswitch using below command (So ulimit set according to > Anthony's previous post): > > =================================================================================== > ulimit -d unlimited; ulimit -f unlimited; ulimit -i unlimited; ulimit -n > 999999; ulimit -q unlimited ; ulimit -u unlimited; ulimit -v unlimited ; > ulimit -x unlimited; ulimit -s 244; ulimit -l unlimited; freeswitch > > =================================================================================== > > Thanks > msp > > > > Brian West-3 wrote: > > > > Can you give me the output of uname -a and the contents of /proc/ > > cpuinfo? Not sure I asked for this info already or not. > > > > Thanks, > > Brian > > > > On Feb 5, 2009, at 2:42 AM, shehzad p wrote: > > > >> > >> Hi anthony, > >> In my previous post I already attached the BT for the testing of FS > >> 1.0.1 > >> Posting it again, please find it on > >> link==>http://www.nabble.com/file/p21825226/bt_new_arch.txt > >> > >> Now I got the same result while testing FS 1.0.3RC1, And its BT is > >> also > >> same... BT Link: > >> http://www.nabble.com/file/p21847332/fs_1_0_3_bt_new_arch.log > >> fs_1_0_3_bt_new_arch.log > >> > >> (Note: Same in the sens the functions listed in the sequence are > >> almost same > >> as before...) > >> > >> > >> Anthony Minessale-2 wrote: > >>> > >>> If you still get a crash on SVN trunk please post the bt even if > >>> you think > >>> it's the same, since it won't be > >>> exactly the same, the line numbers etc will be accurate with our > >>> development > >>> code making it easier to debug. > >>> > >>> > >>> On Wed, Feb 4, 2009 at 12:38 AM, shehzad p wrote: > >>> > >>>> > >>>> Hi anthony, > >>>> > >>>> I Modified the whole architecture of call routing system, > >>>> Now after getting required routes, script exit and, > >>>> control comes back to Dialplan, and call is bridged there, > >>>> And call hangup, CDR is posted to cdr.php file (using xml_cdr). > >>>> > >>>> So now there is no blocking statement (bridge or anything like > >>>> that) in > >>>> current javascript, It return back control instantly. > >>>> > >>>> So, setting up all above architecture... > >>>> First I tested FS 1.0.1 , It get crashed two times, in interval of > >>>> 3 to 5 > >>>> hours and simultaneous call of about 100 to 150. > >>>> BT is the same as before... > >>>> http://www.nabble.com/file/p21825226/bt_new_arch.txt bt_new_arch.txt > >>>> > >>>> Now I am also testing 1.0.3RC1, and post it back if any found. > >>>> > >>>> Thanks > >>>> msp > >>>> > >>>> > >>>> Clearly you have an issue with your javascript code. > >>>> > >>>> You have the Garbage collector blocking in every thread. > >>>> > >>>> Are you doing any endless loops in your code where you do not check > >>>> session.ready() as a condition for > >>>> continuing the script? > >>>> > >>>> any time session.ready() fails you must immediately exit. > >>>> > >>>> Are you using session.execute to execute long blocking operations > >>>> like > >>>> bridging many calls or entering a conference? > >>>> You should avoid doing this as all the collective scripts on the > >>>> system > >>>> share a common Garbage Collector provided by the > >>>> JS engine and it can lead to the exact issues you describe if the > >>>> code is > >>>> not properly designed. > >>>> > >>>> What else does you script do that are things provided by FS such as > >>>> playing > >>>> files and executing applications. > >>>> > >>>> > >>>> > >>>> > >>>> -- > >>>> View this message in context: > >>>> > http://www.nabble.com/Freeswitch-freezes-on-increasing-call-traffic-tp21701744p21825226.html > >>>> Sent from the Freeswitch-users mailing list archive at Nabble.com. > >>>> > >>>> > >>>> _______________________________________________ > >>>> Freeswitch-users mailing list > >>>> Freeswitch-users at lists.freeswitch.org > >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>>> http://www.freeswitch.org > >>>> > >>> > >>> > >>> > >>> -- > >>> Anthony Minessale II > >>> > >>> FreeSWITCH http://www.freeswitch.org/ > >>> ClueCon http://www.cluecon.com/ > >>> > >>> AIM: anthm > >>> MSN:anthony_minessale at hotmail.com< > MSN%3Aanthony_minessale at hotmail.com > >>> > > >>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > >>> > > >>> IRC: irc.freenode.net #freeswitch > >>> > >>> FreeSWITCH Developer Conference > >>> sip:888 at conference.freeswitch.org< > sip%3A888 at conference.freeswitch.org > >>> > > >>> iax:guest at conference.freeswitch.org/888 > >>> googletalk:conf+888 at conference.freeswitch.org > > >>> > > >>> pstn:213-799-1400 > >>> > >>> _______________________________________________ > >>> Freeswitch-users mailing list > >>> Freeswitch-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> http://www.freeswitch.org > >>> > >>> > >> > >> -- > >> View this message in context: > >> > http://www.nabble.com/Freeswitch-freezes-on-increasing-call-traffic-tp21701744p21847332.html > >> Sent from the Freeswitch-users mailing list archive at Nabble.com. > >> > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > View this message in context: > http://www.nabble.com/Freeswitch-freezes-on-increasing-call-traffic-tp21701744p21848148.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090205/a427c469/attachment-0002.html From pmhshz at gmail.com Thu Feb 5 06:22:49 2009 From: pmhshz at gmail.com (shehzad p) Date: Thu, 5 Feb 2009 06:22:49 -0800 (PST) Subject: [Freeswitch-users] Freeswitch freezes on increasing call traffic In-Reply-To: <21848148.post@talk.nabble.com> References: <21701744.post@talk.nabble.com> <191c3a030901280601t3e185ec2m76295110e4888837@mail.gmail.com> <21729863.post@talk.nabble.com> <191c3a030901290831y2ff329b3maee6c1aafa3c68e2@mail.gmail.com> <21745312.post@talk.nabble.com> <21749375.post@talk.nabble.com> <191c3a030901300818i5d85a7aawa0c53b4e54892f40@mail.gmail.com> <21761523.post@talk.nabble.com> <191c3a030901310850s553f8c11oc4e559a710def6d2@mail.gmail.com> <21825226.post@talk.nabble.com> <191c3a030902040551w5d867deta55c98bd3a3c03f8@mail.gmail.com> <21847332.post@talk.nabble.com> <4C2A4389-BEF6-454B-8B5F-FDCCC76E174A@freeswitch.org> <21848148.post@talk.nabble.com> Message-ID: <21852304.post@talk.nabble.com> Hi Brian, As it can be seen from the system information, there require any change in system or any suggestion... out put of uname -a is : Linux localhost.localdomain 2.6.23.1-42.fc8 #1 SMP Tue Oct 30 13:55:12 EDT 2007 i686 i686 i386 GNU/Linux Thanks, msp shehzad p wrote: > > HI Brian, > > Output of ulimit -a and /proc/cpuinfo is attached... > http://www.nabble.com/file/p21848148/12_ulimit_and_cpuinfo.log > 12_ulimit_and_cpuinfo.log > > BUT...................... > I am running the freeswitch using below command (So ulimit set according > to Anthony's previous post): > =================================================================================== > ulimit -d unlimited; ulimit -f unlimited; ulimit -i unlimited; ulimit -n > 999999; ulimit -q unlimited ; ulimit -u unlimited; ulimit -v unlimited ; > ulimit -x unlimited; ulimit -s 244; ulimit -l unlimited; freeswitch > =================================================================================== > > Thanks > msp > > > > Brian West-3 wrote: >> >> Can you give me the output of uname -a and the contents of /proc/ >> cpuinfo? Not sure I asked for this info already or not. >> >> Thanks, >> Brian >> >> On Feb 5, 2009, at 2:42 AM, shehzad p wrote: >> >>> >>> Hi anthony, >>> In my previous post I already attached the BT for the testing of FS >>> 1.0.1 >>> Posting it again, please find it on >>> link==>http://www.nabble.com/file/p21825226/bt_new_arch.txt >>> >>> Now I got the same result while testing FS 1.0.3RC1, And its BT is >>> also >>> same... BT Link: >>> http://www.nabble.com/file/p21847332/fs_1_0_3_bt_new_arch.log >>> fs_1_0_3_bt_new_arch.log >>> >>> (Note: Same in the sens the functions listed in the sequence are >>> almost same >>> as before...) >>> >>> >>> Anthony Minessale-2 wrote: >>>> >>>> If you still get a crash on SVN trunk please post the bt even if >>>> you think >>>> it's the same, since it won't be >>>> exactly the same, the line numbers etc will be accurate with our >>>> development >>>> code making it easier to debug. >>>> >>>> >>>> On Wed, Feb 4, 2009 at 12:38 AM, shehzad p wrote: >>>> >>>>> >>>>> Hi anthony, >>>>> >>>>> I Modified the whole architecture of call routing system, >>>>> Now after getting required routes, script exit and, >>>>> control comes back to Dialplan, and call is bridged there, >>>>> And call hangup, CDR is posted to cdr.php file (using xml_cdr). >>>>> >>>>> So now there is no blocking statement (bridge or anything like >>>>> that) in >>>>> current javascript, It return back control instantly. >>>>> >>>>> So, setting up all above architecture... >>>>> First I tested FS 1.0.1 , It get crashed two times, in interval of >>>>> 3 to 5 >>>>> hours and simultaneous call of about 100 to 150. >>>>> BT is the same as before... >>>>> http://www.nabble.com/file/p21825226/bt_new_arch.txt bt_new_arch.txt >>>>> >>>>> Now I am also testing 1.0.3RC1, and post it back if any found. >>>>> >>>>> Thanks >>>>> msp >>>>> >>>>> >>>>> Clearly you have an issue with your javascript code. >>>>> >>>>> You have the Garbage collector blocking in every thread. >>>>> >>>>> Are you doing any endless loops in your code where you do not check >>>>> session.ready() as a condition for >>>>> continuing the script? >>>>> >>>>> any time session.ready() fails you must immediately exit. >>>>> >>>>> Are you using session.execute to execute long blocking operations >>>>> like >>>>> bridging many calls or entering a conference? >>>>> You should avoid doing this as all the collective scripts on the >>>>> system >>>>> share a common Garbage Collector provided by the >>>>> JS engine and it can lead to the exact issues you describe if the >>>>> code is >>>>> not properly designed. >>>>> >>>>> What else does you script do that are things provided by FS such as >>>>> playing >>>>> files and executing applications. >>>>> >>>>> >>>>> >>>>> >>>>> -- >>>>> View this message in context: >>>>> http://www.nabble.com/Freeswitch-freezes-on-increasing-call-traffic-tp21701744p21825226.html >>>>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>>>> >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> >>>> -- >>>> Anthony Minessale II >>>> >>>> FreeSWITCH http://www.freeswitch.org/ >>>> ClueCon http://www.cluecon.com/ >>>> >>>> AIM: anthm >>>> MSN:anthony_minessale at hotmail.com >>> > >>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com>>> > >>>> IRC: irc.freenode.net #freeswitch >>>> >>>> FreeSWITCH Developer Conference >>>> sip:888 at conference.freeswitch.org >>> > >>>> iax:guest at conference.freeswitch.org/888 >>>> googletalk:conf+888 at conference.freeswitch.org>>> > >>>> pstn:213-799-1400 >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> -- >>> View this message in context: >>> http://www.nabble.com/Freeswitch-freezes-on-increasing-call-traffic-tp21701744p21847332.html >>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- View this message in context: http://www.nabble.com/Freeswitch-freezes-on-increasing-call-traffic-tp21701744p21852304.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From paul.degt at gmail.com Thu Feb 5 07:12:15 2009 From: paul.degt at gmail.com (paul.degt) Date: Thu, 05 Feb 2009 10:12:15 -0500 Subject: [Freeswitch-users] Freeswitch freezes on increasing call traffic In-Reply-To: <21852304.post@talk.nabble.com> References: <21701744.post@talk.nabble.com> <191c3a030901280601t3e185ec2m76295110e4888837@mail.gmail.com> <21729863.post@talk.nabble.com> <191c3a030901290831y2ff329b3maee6c1aafa3c68e2@mail.gmail.com> <21745312.post@talk.nabble.com> <21749375.post@talk.nabble.com> <191c3a030901300818i5d85a7aawa0c53b4e54892f40@mail.gmail.com> <21761523.post@talk.nabble.com> <191c3a030901310850s553f8c11oc4e559a710def6d2@mail.gmail.com> <21825226.post@talk.nabble.com> <191c3a030902040551w5d867deta55c98bd3a3c03f8@mail.gmail.com> <21847332.post@talk.nabble.com> <4C2A4389-BEF6-454B-8B5F-FDCCC76E174A@freeswitch.org> <21848148.post@talk.nabble.com> <21852304.post@talk.nabble.com> Message-ID: <498B01CF.6080902@gmail.com> Look like you use Fedora. I had a lot of issues with using Fedora as production or load test system, in my opinion it's more like work in progress than a production ready stable linux. If you cannot buy RHEL or SLES use Centos. shehzad p wrote: > Hi Brian, > > As it can be seen from the system information, there require any change in > system or any suggestion... > > out put of uname -a is : > Linux localhost.localdomain 2.6.23.1-42.fc8 #1 SMP Tue Oct 30 13:55:12 EDT > 2007 i686 i686 i386 GNU/Linux > > > Thanks, > msp > > > shehzad p wrote: > >> HI Brian, >> >> Output of ulimit -a and /proc/cpuinfo is attached... >> http://www.nabble.com/file/p21848148/12_ulimit_and_cpuinfo.log >> 12_ulimit_and_cpuinfo.log >> >> BUT...................... >> I am running the freeswitch using below command (So ulimit set according >> to Anthony's previous post): >> =================================================================================== >> ulimit -d unlimited; ulimit -f unlimited; ulimit -i unlimited; ulimit -n >> 999999; ulimit -q unlimited ; ulimit -u unlimited; ulimit -v unlimited ; >> ulimit -x unlimited; ulimit -s 244; ulimit -l unlimited; freeswitch >> =================================================================================== >> >> Thanks >> msp >> >> >> >> Brian West-3 wrote: >> >>> Can you give me the output of uname -a and the contents of /proc/ >>> cpuinfo? Not sure I asked for this info already or not. >>> >>> Thanks, >>> Brian >>> >>> On Feb 5, 2009, at 2:42 AM, shehzad p wrote: >>> >>> > From sicfslist at gmail.com Thu Feb 5 07:28:37 2009 From: sicfslist at gmail.com (Shelby Ramsey) Date: Thu, 5 Feb 2009 09:28:37 -0600 Subject: [Freeswitch-users] XML CDR ERROR ... In-Reply-To: <640834B6-0A23-4855-BB0F-F3622AA334FF@freeswitch.org> References: <35b355e90902042333p3009d6c1md2a76355a90509bd@mail.gmail.com> <640834B6-0A23-4855-BB0F-F3622AA334FF@freeswitch.org> Message-ID: <35b355e90902050728u4f0e1dbk2386518ed3985b74@mail.gmail.com> Just to make sure I'm clear: -- start FS (without in modules.conf.xml) -- but have xml_cdr.conf.xml in autoload configs -- then reload xml -- then execute load mod_xml_cdr via the api That seems like a challenging way to start FS ... SDR On Thu, Feb 5, 2009 at 1:42 AM, Brian West wrote: > Make sure your config file is installed and issue a reloadxml then > load mod_xml_cdr > > /b > > On Feb 5, 2009, at 1:33 AM, Shelby Ramsey wrote: > > > Hello, > > > > I'm having it on both Fedora and Ubuntu boxes: > > > > 2009-02-05 01:26:27 [ERR] mod_xml_cdr.c:290 mod_xml_cdr_load() Open > > of xml_cdr.conf failed > > 2009-02-05 01:26:27 [CRIT] switch_loadable_module.c:839 > > switch_loadable_module_load_file() Error Loading module /usr/local/ > > freeswitch/mod/mod_xml_cdr.so > > **Module load routine returned an error** > > > > Details: > > -- Ubuntu 6.04 LTS > > -- Fedora 8 > > > > Tried a couple of things: > > -- messing with libcurl > > -- ./configure --without-libcurl > > > > Thanks! > > > > SDR > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090205/4e4915dd/attachment-0002.html From freeswitch-users at lists.rupa.com Thu Feb 5 07:38:54 2009 From: freeswitch-users at lists.rupa.com (Rupa Schomaker (lists)) Date: Thu, 05 Feb 2009 09:38:54 -0600 Subject: [Freeswitch-users] XML CDR ERROR ... In-Reply-To: <35b355e90902050728u4f0e1dbk2386518ed3985b74@mail.gmail.com> References: <35b355e90902042333p3009d6c1md2a76355a90509bd@mail.gmail.com> <640834B6-0A23-4855-BB0F-F3622AA334FF@freeswitch.org> <35b355e90902050728u4f0e1dbk2386518ed3985b74@mail.gmail.com> Message-ID: <498B080E.6080202@lists.rupa.com> The error is that the conf file couldn't be loaded. If you've verified teh file is actually in autoload configs dir, also ensure the file is readable by whatever user/group freeswitch is running as. Once you have your config squared away, just having hte in modules.conf.xml is sufficient. Brian was trying to assist you in debugging.... On 2/5/2009 9:28 AM, Shelby Ramsey wrote: > Just to make sure I'm clear: > -- start FS (without in modules.conf.xml) > -- but have xml_cdr.conf.xml in autoload configs > -- then reload xml > -- then execute load mod_xml_cdr via the api > > That seems like a challenging way to start FS ... > > SDR > > On Thu, Feb 5, 2009 at 1:42 AM, Brian West > wrote: > > Make sure your config file is installed and issue a reloadxml then > load mod_xml_cdr > > /b > > On Feb 5, 2009, at 1:33 AM, Shelby Ramsey wrote: > > > Hello, > > > > I'm having it on both Fedora and Ubuntu boxes: > > > > 2009-02-05 01:26:27 [ERR] mod_xml_cdr.c:290 mod_xml_cdr_load() Open > > of xml_cdr.conf failed > > 2009-02-05 01:26:27 [CRIT] switch_loadable_module.c:839 > > switch_loadable_module_load_file() Error Loading module /usr/local/ > > freeswitch/mod/mod_xml_cdr.so > > **Module load routine returned an error** > > > > Details: > > -- Ubuntu 6.04 LTS > > -- Fedora 8 > > > > Tried a couple of things: > > -- messing with libcurl > > -- ./configure --without-libcurl > > > > Thanks! > > > > SDR From anthony.minessale at gmail.com Thu Feb 5 07:40:20 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 5 Feb 2009 09:40:20 -0600 Subject: [Freeswitch-users] origainate through sofia gateway In-Reply-To: <1233839355.5346.31.camel@dotw1126.dotsystems.pl> References: <1233668179.5354.15.camel@dotw1126.dotsystems.pl> <1b46b4e80902030920p521d6079s4c9b78d8d39a924a@mail.gmail.com> <5E2B217D-8C85-48DB-800A-535A7FAAE24B@freeswitch.org> <1b46b4e80902030931y6a1679afw12ec37596ba9d77d@mail.gmail.com> <1233737494.5405.12.camel@dotw1126.dotsystems.pl> <191c3a030902040609s600798c8t720950a2e13ee05a@mail.gmail.com> <1233758767.5405.22.camel@dotw1126.dotsystems.pl> <191c3a030902040822o1ffededbwd592361aa09f6b46@mail.gmail.com> <1233839355.5346.31.camel@dotw1126.dotsystems.pl> Message-ID: <191c3a030902050740w7f7c4ef2hdb7a6a0e355c46f5@mail.gmail.com> Oddly the provider returns SIP/2.0 500 Internal Server Error to your invite. Maybe you can ask them why they do that in this particular case. On Thu, Feb 5, 2009 at 7:09 AM, Jacek Sokulski wrote: > the result of > apiExecute("bgapi", "originate > > {effective_caller_id_number=fixed0248b,origination_caller_id_number=1000,ignore_early_media=true}user/ > 1008 at 192.168.1.122 bridge:sofia/gateway/halonet/0225490317 inline"); > > /: > > 2009-02-05 14:04:48 [DEBUG] switch_ivr_originate.c:777 > switch_ivr_originate() variable string 0 = > [effective_caller_id_number=fixed0248b] > > 2009-02-05 14:04:48 [DEBUG] switch_ivr_originate.c:777 > switch_ivr_originate() variable string 1 = > [origination_caller_id_number=1000] > > 2009-02-05 14:04:48 [DEBUG] switch_ivr_originate.c:777 > switch_ivr_originate() variable string 2 = [ignore_early_media=true] > > 2009-02-05 14:04:48 [DEBUG] switch_ivr_originate.c:777 > switch_ivr_originate() variable string 0 = [presence_id=1008 at 192.168.1.122 > ] > > 2009-02-05 14:04:48 [NOTICE] switch_channel.c:565 > switch_channel_set_name() New Channel sofia/internal/ > sip:1008 at 192.168.1.126:5070 [f9b18b2e-b0a9-4e24-8452-40e1cce047bd] > > 2009-02-05 14:04:48 [DEBUG] mod_sofia.c:2518 sofia_outgoing_channel() > (sofia/internal/sip:1008 at 192.168.1.126:5070) State Change CS_NEW -> > CS_INIT > > 2009-02-05 14:04:48 [DEBUG] switch_core_session.c:807 > switch_core_session_signal_state_change() Send signal sofia/internal/ > sip:1008 at 192.168.1.126:5070 [BREAK] > > 2009-02-05 14:04:48 [DEBUG] switch_core_state_machine.c:379 > switch_core_session_run() (sofia/internal/sip:1008 at 192.168.1.126:5070) > Running State Change CS_INIT > > 2009-02-05 14:04:48 [DEBUG] switch_core_state_machine.c:444 > switch_core_session_run() (sofia/internal/sip:1008 at 192.168.1.126:5070) > State INIT > > 2009-02-05 14:04:48 [DEBUG] mod_sofia.c:83 sofia_on_init() > sofia/internal/sip:1008 at 192.168.1.126:5070 SOFIA INIT > > send 1196 bytes to udp/[192.168.1.126]:5070 at 13:04:48.127004: > > > ------------------------------------------------------------------------ > > INVITE sip:1008 at 192.168.1.126:5070 SIP/2.0 > > Via: SIP/2.0/UDP 192.168.1.122;rport;branch=z9hG4bKFpQjUmp1vNHZB > > Max-Forwards: 70 > > From: "FreeSWITCH" > >;tag=9vpFm3tBc8gSe > > To: > > Call-ID: 68736fc7-6e28-122c-8396-000c7680c408 > > CSeq: 110801208 INVITE > > Contact: > > User-Agent: FreeSWITCH-mod_sofia/1.0.2-wyeksportowane > > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH > > Supported: timer, precondition, path, replaces > > Allow-Events: talk, presence, dialog, call-info, sla, > include-session-description, presence.winfo, message-summary, refer > > Min-SE: 120 > > Content-Type: application/sdp > > Content-Disposition: session > > Content-Length: 335 > > Remote-Party-ID: "FreeSWITCH" > >;screen=yes;privacy=off > > > > v=0 > > o=FreeSWITCH 1927164999227225404 3276738624485570081 IN IP4 > 192.168.1.122 > > s=FreeSWITCH > > c=IN IP4 192.168.1.122 > > t=0 0 > > m=audio 27044 RTP/AVP 9 0 8 3 101 13 > > a=rtpmap:9 G722/8000 > > a=rtpmap:0 PCMU/8000 > > a=rtpmap:8 PCMA/8000 > > a=rtpmap:3 GSM/8000 > > a=rtpmap:101 telephone-event/8000 > > a=fmtp:101 0-16 > > a=rtpmap:13 CN/8000 > > a=ptime:20 > > > ------------------------------------------------------------------------ > > 2009-02-05 14:04:48 [DEBUG] mod_sofia.c:111 sofia_on_init() > (sofia/internal/sip:1008 at 192.168.1.126:5070) State Change CS_INIT -> > CS_ROUTING > > 2009-02-05 14:04:48 [DEBUG] switch_core_session.c:807 > switch_core_session_signal_state_change() Send signal sofia/internal/ > sip:1008 at 192.168.1.126:5070 [BREAK] > > 2009-02-05 14:04:48 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state() > Channel sofia/internal/sip:1008 at 192.168.1.126:5070 entering state > [calling] > > 2009-02-05 14:04:48 [DEBUG] switch_core_state_machine.c:444 > switch_core_session_run() (sofia/internal/sip:1008 at 192.168.1.126:5070) > State INIT going to sleep > > 2009-02-05 14:04:48 [DEBUG] switch_core_state_machine.c:379 > switch_core_session_run() (sofia/internal/sip:1008 at 192.168.1.126:5070) > Running State Change CS_ROUTING > > 2009-02-05 14:04:48 [DEBUG] switch_core_state_machine.c:447 > switch_core_session_run() (sofia/internal/sip:1008 at 192.168.1.126:5070) > State ROUTING > > 2009-02-05 14:04:48 [DEBUG] mod_sofia.c:130 sofia_on_routing() > sofia/internal/sip:1008 at 192.168.1.126:5070 SOFIA ROUTING > > 2009-02-05 14:04:48 [DEBUG] switch_ivr_originate.c:58 > originate_on_routing() (sofia/internal/sip:1008 at 192.168.1.126:5070) State > Change CS_ROUTING -> CS_CONSUME_MEDIA > > 2009-02-05 14:04:48 [DEBUG] switch_core_session.c:807 > switch_core_session_signal_state_change() Send signal sofia/internal/ > sip:1008 at 192.168.1.126:5070 [BREAK] > > 2009-02-05 14:04:48 [DEBUG] switch_core_state_machine.c:447 > switch_core_session_run() (sofia/internal/sip:1008 at 192.168.1.126:5070) > State ROUTING going to sleep > > 2009-02-05 14:04:48 [DEBUG] switch_core_state_machine.c:379 > switch_core_session_run() (sofia/internal/sip:1008 at 192.168.1.126:5070) > Running State Change CS_CONSUME_MEDIA > > 2009-02-05 14:04:48 [DEBUG] switch_core_state_machine.c:466 > switch_core_session_run() (sofia/internal/sip:1008 at 192.168.1.126:5070) > State CONSUME_MEDIA > > recv 361 bytes from udp/[192.168.1.126]:5070 at 13:04:48.130745: > > > ------------------------------------------------------------------------ > > SIP/2.0 100 Trying > > Via: SIP/2.0/UDP 192.168.1.122;rport;branch=z9hG4bKFpQjUmp1vNHZB > > From: "FreeSWITCH" > >;tag=9vpFm3tBc8gSe > > To: ;tag=2111843199 > > Contact: > > Call-ID: 68736fc7-6e28-122c-8396-000c7680c408 > > CSeq: 110801208 INVITE > > Server: X-Lite release 1105d > > Content-Length: 0 > > > > > ------------------------------------------------------------------------ > > recv 362 bytes from udp/[192.168.1.126]:5070 at 13:04:48.181571: > > > ------------------------------------------------------------------------ > > SIP/2.0 180 Ringing > > Via: SIP/2.0/UDP 192.168.1.122;rport;branch=z9hG4bKFpQjUmp1vNHZB > > From: "FreeSWITCH" > >;tag=9vpFm3tBc8gSe > > To: ;tag=2111843199 > > Contact: > > Call-ID: 68736fc7-6e28-122c-8396-000c7680c408 > > CSeq: 110801208 INVITE > > Server: X-Lite release 1105d > > Content-Length: 0 > > > > > ------------------------------------------------------------------------ > > 2009-02-05 14:04:48 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state() > Channel sofia/internal/sip:1008 at 192.168.1.126:5070 entering state > [proceeding] > > 2009-02-05 14:04:48 [NOTICE] sofia.c:2596 sofia_handle_sip_i_state() > Ring-Ready sofia/internal/sip:1008 at 192.168.1.126:5070! > > recv 678 bytes from udp/[192.168.1.126]:5070 at 13:04:51.914714: > > > ------------------------------------------------------------------------ > > SIP/2.0 200 Ok > > Via: SIP/2.0/UDP 192.168.1.122;rport;branch=z9hG4bKFpQjUmp1vNHZB > > From: "FreeSWITCH" > >;tag=9vpFm3tBc8gSe > > To: ;tag=2111843199 > > Contact: > > Call-ID: 68736fc7-6e28-122c-8396-000c7680c408 > > CSeq: 110801208 INVITE > > Content-Type: application/sdp > > Server: X-Lite release 1105d > > Content-Length: 288 > > > > v=0 > > o=1008 1183460117 1183463903 IN IP4 192.168.1.126 > > s=X-Lite > > c=IN IP4 192.168.1.126 > > t=0 0 > > m=audio 9000 RTP/AVP 0 8 98 97 101 > > a=rtpmap:0 pcmu/8000 > > a=rtpmap:8 pcma/8000 > > a=rtpmap:98 iLBC/8000 > > a=rtpmap:97 speex/8000 > > a=rtpmap:101 telephone-event/8000 > > a=fmtp:101 0-15 > > a=sendrecv > > > ------------------------------------------------------------------------ > > send 372 bytes to udp/[192.168.1.126]:5070 at 13:04:51.915303: > > > ------------------------------------------------------------------------ > > ACK sip:1008 at 192.168.1.126:5070 SIP/2.0 > > Via: SIP/2.0/UDP 192.168.1.122;rport;branch=z9hG4bKgZgBXF74Sy7HQ > > Max-Forwards: 70 > > From: "FreeSWITCH" > >;tag=9vpFm3tBc8gSe > > To: ;tag=2111843199 > > Call-ID: 68736fc7-6e28-122c-8396-000c7680c408 > > CSeq: 110801208 ACK > > Contact: > > Content-Length: 0 > > > > > ------------------------------------------------------------------------ > > 2009-02-05 14:04:51 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state() > Channel sofia/internal/sip:1008 at 192.168.1.126:5070 entering state [ready] > > 2009-02-05 14:04:51 [DEBUG] sofia.c:2546 sofia_handle_sip_i_state() > Remote SDP: > > v=0 > > o=1008 1183460117 1183463903 IN IP4 192.168.1.126 > > s=X-Lite > > c=IN IP4 192.168.1.126 > > t=0 0 > > m=audio 9000 RTP/AVP 0 8 98 97 101 > > a=rtpmap:0 pcmu/8000 > > a=rtpmap:8 pcma/8000 > > a=rtpmap:98 iLBC/8000 > > a=rtpmap:97 speex/8000 > > a=rtpmap:101 telephone-event/8000 > > a=fmtp:101 0-15 > > > > 2009-02-05 14:04:51 [DEBUG] sofia_glue.c:2509 sofia_glue_negotiate_sdp() > Audio Codec Compare [pcmu:0:8000]/[G722:9:8000] > > 2009-02-05 14:04:51 [DEBUG] sofia_glue.c:2509 sofia_glue_negotiate_sdp() > Audio Codec Compare [pcmu:0:8000]/[PCMU:0:8000] > > 2009-02-05 14:04:51 [DEBUG] sofia_glue.c:1670 sofia_glue_tech_set_codec() > Set Codec sofia/internal/sip:1008 at 192.168.1.126:5070 PCMU/8000 20 ms 160 > samples > > 2009-02-05 14:04:51 [DEBUG] sofia_glue.c:2473 sofia_glue_negotiate_sdp() > Set 2833 dtmf payload to 101 > > 2009-02-05 14:04:51 [DEBUG] sofia_glue.c:1890 sofia_glue_activate_rtp() > AUDIO RTP [sofia/internal/sip:1008 at 192.168.1.126:5070] 192.168.1.122 port > 27044 -> 192.168.1.126 port 9000 codec: 0 ms: 20 > > 2009-02-05 14:04:51 [DEBUG] switch_rtp.c:865 switch_rtp_create() Starting > timer [soft] 160 bytes per 20000ms > > 2009-02-05 14:04:51 [NOTICE] sofia.c:3031 sofia_handle_sip_i_state() > Channel [sofia/internal/sip:1008 at 192.168.1.126:5070] has been answered > > 2009-02-05 14:04:51 [DEBUG] switch_channel.c:177 > switch_channel_audio_sync() sofia/internal/sip:1008 at 192.168.1.126:5070receive message [AUDIO_SYNC] > > 2009-02-05 14:04:51 [DEBUG] switch_ivr_originate.c:1627 > switch_ivr_originate() Originate Resulted in Success: [sofia/internal/ > sip:1008 at 192.168.1.126:5070] > > 2009-02-05 14:04:51 [DEBUG] switch_channel.c:177 > switch_channel_audio_sync() sofia/internal/sip:1008 at 192.168.1.126:5070receive message [AUDIO_SYNC] > > 2009-02-05 14:04:51 [DEBUG] switch_ivr_originate.c:1627 > switch_ivr_originate() Originate Resulted in Success: [sofia/internal/ > sip:1008 at 192.168.1.126:5070] > > 2009-02-05 14:04:51 [DEBUG] switch_channel.c:177 > switch_channel_audio_sync() sofia/internal/sip:1008 at 192.168.1.126:5070receive message [AUDIO_SYNC] > > 2009-02-05 14:04:51 [DEBUG] switch_ivr.c:1245 > switch_ivr_session_transfer() (sofia/internal/sip:1008 at 192.168.1.126:5070) > State Change CS_CONSUME_MEDIA -> CS_ROUTING > > 2009-02-05 14:04:51 [DEBUG] switch_core_session.c:807 > switch_core_session_signal_state_change() Send signal sofia/internal/ > sip:1008 at 192.168.1.126:5070 [BREAK] > > 2009-02-05 14:04:51 [DEBUG] switch_ivr.c:1249 > switch_ivr_session_transfer() sofia/internal/sip:1008 at 192.168.1.126:5070receive message [TRANSFER] > > 2009-02-05 14:04:51 [DEBUG] switch_core_session.c:511 > switch_core_session_perform_receive_message() Send signal sofia/internal/ > sip:1008 at 192.168.1.126:5070 [BREAK] > > 2009-02-05 14:04:51 [NOTICE] switch_ivr.c:1251 > switch_ivr_session_transfer() Transfer sofia/internal/ > sip:1008 at 192.168.1.126:5070 to > inline[bridge:sofia/gateway/halonet/0225490317 at default] > > 2009-02-05 14:04:51 [DEBUG] switch_core_state_machine.c:466 > switch_core_session_run() (sofia/internal/sip:1008 at 192.168.1.126:5070) > State CONSUME_MEDIA going to sleep > > 2009-02-05 14:04:51 [DEBUG] switch_core_state_machine.c:379 > switch_core_session_run() (sofia/internal/sip:1008 at 192.168.1.126:5070) > Running State Change CS_ROUTING > > 2009-02-05 14:04:51 [DEBUG] switch_core_state_machine.c:447 > switch_core_session_run() (sofia/internal/sip:1008 at 192.168.1.126:5070) > State ROUTING > > 2009-02-05 14:04:51 [DEBUG] mod_sofia.c:130 sofia_on_routing() > sofia/internal/sip:1008 at 192.168.1.126:5070 SOFIA ROUTING > > 2009-02-05 14:04:51 [DEBUG] switch_core_state_machine.c:64 > switch_core_standard_on_routing() sofia/internal/ > sip:1008 at 192.168.1.126:5070 Standard ROUTING > > 2009-02-05 14:04:51 [DEBUG] switch_core_state_machine.c:100 > switch_core_standard_on_routing() (sofia/internal/ > sip:1008 at 192.168.1.126:5070) State Change CS_ROUTING -> CS_EXECUTE > > 2009-02-05 14:04:51 [DEBUG] switch_core_session.c:807 > switch_core_session_signal_state_change() Send signal sofia/internal/ > sip:1008 at 192.168.1.126:5070 [BREAK] > > 2009-02-05 14:04:51 [DEBUG] switch_core_state_machine.c:447 > switch_core_session_run() (sofia/internal/sip:1008 at 192.168.1.126:5070) > State ROUTING going to sleep > > 2009-02-05 14:04:51 [DEBUG] switch_core_state_machine.c:379 > switch_core_session_run() (sofia/internal/sip:1008 at 192.168.1.126:5070) > Running State Change CS_EXECUTE > > 2009-02-05 14:04:51 [DEBUG] switch_core_state_machine.c:454 > switch_core_session_run() (sofia/internal/sip:1008 at 192.168.1.126:5070) > State EXECUTE > > 2009-02-05 14:04:51 [DEBUG] mod_sofia.c:173 sofia_on_execute() > sofia/internal/sip:1008 at 192.168.1.126:5070 SOFIA EXECUTE > > 2009-02-05 14:04:51 [DEBUG] switch_core_state_machine.c:137 > switch_core_standard_on_execute() sofia/internal/ > sip:1008 at 192.168.1.126:5070 Standard EXECUTE > > 2009-02-05 14:04:51 [DEBUG] switch_core_state_machine.c:152 > switch_core_standard_on_execute() sofia/internal/ > sip:1008 at 192.168.1.126:5070 Execute > bridge(sofia/gateway/halonet/0225490317) > > 2009-02-05 14:04:51 [NOTICE] switch_channel.c:565 > switch_channel_set_name() New Channel sofia/external/0225490317 > [bf4fece9-38fc-40fa-8e9e-91f836f05e55] > > 2009-02-05 14:04:51 [DEBUG] mod_sofia.c:2518 sofia_outgoing_channel() > (sofia/external/0225490317) State Change CS_NEW -> CS_INIT > > 2009-02-05 14:04:51 [DEBUG] switch_core_session.c:807 > switch_core_session_signal_state_change() Send signal > sofia/external/0225490317 [BREAK] > > 2009-02-05 14:04:51 [DEBUG] switch_core_state_machine.c:379 > switch_core_session_run() (sofia/external/0225490317) Running State Change > CS_INIT > > 2009-02-05 14:04:51 [DEBUG] switch_core_state_machine.c:444 > switch_core_session_run() (sofia/external/0225490317) State INIT > > 2009-02-05 14:04:51 [DEBUG] mod_sofia.c:83 sofia_on_init() > sofia/external/0225490317 SOFIA INIT > > 2009-02-05 14:04:52 [DEBUG] sofia_glue.c:566 > sofia_glue_ext_address_lookup() STUN Success [89.77.89.244]:[50863] > > 2009-02-05 14:04:52 [DEBUG] mod_sofia.c:111 sofia_on_init() > (sofia/external/0225490317) State Change CS_INIT -> CS_ROUTING > > 2009-02-05 14:04:52 [DEBUG] switch_core_session.c:807 > switch_core_session_signal_state_change() Send signal > sofia/external/0225490317 [BREAK] > > 2009-02-05 14:04:52 [DEBUG] switch_core_state_machine.c:444 > switch_core_session_run() (sofia/external/0225490317) State INIT going to > sleep > > 2009-02-05 14:04:52 [DEBUG] switch_core_state_machine.c:379 > switch_core_session_run() (sofia/external/0225490317) Running State Change > CS_ROUTING > > 2009-02-05 14:04:52 [DEBUG] switch_core_state_machine.c:447 > switch_core_session_run() (sofia/external/0225490317) State ROUTING > > 2009-02-05 14:04:52 [DEBUG] mod_sofia.c:130 sofia_on_routing() > sofia/external/0225490317 SOFIA ROUTING > > 2009-02-05 14:04:52 [DEBUG] switch_ivr_originate.c:58 > originate_on_routing() (sofia/external/0225490317) State Change CS_ROUTING > -> CS_CONSUME_MEDIA > > 2009-02-05 14:04:52 [DEBUG] switch_core_session.c:807 > switch_core_session_signal_state_change() Send signal > sofia/external/0225490317 [BREAK] > > 2009-02-05 14:04:52 [DEBUG] switch_core_state_machine.c:447 > switch_core_session_run() (sofia/external/0225490317) State ROUTING going to > sleep > > 2009-02-05 14:04:52 [DEBUG] switch_core_state_machine.c:379 > switch_core_session_run() (sofia/external/0225490317) Running State Change > CS_CONSUME_MEDIA > > 2009-02-05 14:04:52 [DEBUG] switch_core_state_machine.c:466 > switch_core_session_run() (sofia/external/0225490317) State CONSUME_MEDIA > > 2009-02-05 14:04:52 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state() > Channel sofia/external/0225490317 entering state [calling] > > send 1117 bytes to udp/[194.9.25.21]:5060 at 13:04:52.572008: > > > ------------------------------------------------------------------------ > > INVITE sip:0225490317 at sip.halonet.plSIP/2.0 > > Via: SIP/2.0/UDP 83.13.98.165:5080;rport;branch=z9hG4bKyvKBDBpvBQe2p > > Max-Forwards: 69 > > From: "Extension 1008" > ;transport=udp>;tag=FZp353yettFNN > > To: > > > Call-ID: 6b05cee4-6e28-122c-8396-000c7680c408 > > CSeq: 110801210 INVITE > > Contact: > > User-Agent: FreeSWITCH-mod_sofia/1.0.2-wyeksportowane > > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO > > Supported: timer, precondition, path, replaces > > Allow-Events: talk, refer > > Min-SE: 120 > > Content-Type: application/sdp > > Content-Disposition: session > > Content-Length: 309 > > Remote-Party-ID: "Extension 1008" > >;screen=yes;privacy=off > > > > v=0 > > o=FreeSWITCH 3996368035436745512 7588776177330183331 IN IP4 > 89.77.89.244 > > s=FreeSWITCH > > c=IN IP4 89.77.89.244 > > t=0 0 > > m=audio 50863 RTP/AVP 0 8 3 101 13 > > a=rtpmap:0 PCMU/8000 > > a=rtpmap:8 PCMA/8000 > > a=rtpmap:3 GSM/8000 > > a=rtpmap:101 telephone-event/8000 > > a=fmtp:101 0-16 > > a=rtpmap:13 CN/8000 > > a=ptime:20 > > > ------------------------------------------------------------------------ > > recv 760 bytes from udp/[194.9.25.21]:5060 at 13:04:52.993246: > > > ------------------------------------------------------------------------ > > SIP/2.0 407 Proxy Authentication Required > > Via: SIP/2.0/UDP 83.13.98.165:5080 > ;rport=5080;branch=z9hG4bKyvKBDBpvBQe2p;received=89.77.89.244 > > From: "Extension 1008" > ;transport=udp>;tag=FZp353yettFNN > > To: > >;tag=03bd8e75ec97c9ee65c772e401792a5a.5fa3 > > Call-ID: 6b05cee4-6e28-122c-8396-000c7680c408 > > CSeq: 110801210 INVITE > > Proxy-Authenticate: Digest realm="sip.halonet.pl", > nonce="498ae51439454939d240f51ab9455e2c8505c7aa", qop="auth" > > Server: Sip EXpress router (2.0.0-rc1 (i386/linux)) > > Content-Length: 0 > > Warning: 392 194.9.25.21:5060 "Noisy feedback tells: pid=22400 > req_src_ip=89.77.89.244 req_src_port=50866 in_uri= > sip:0225490317 at sip.halonet.pl out_uri= > sip:0225490317 at sip.halonet.pl via_cnt==1" > > > > > ------------------------------------------------------------------------ > > send 387 bytes to udp/[194.9.25.21]:5060 at 13:04:52.993616: > > > ------------------------------------------------------------------------ > > ACK sip:0225490317 at sip.halonet.pl SIP/2.0 > > Via: SIP/2.0/UDP 83.13.98.165:5080;rport;branch=z9hG4bKyvKBDBpvBQe2p > > Max-Forwards: 69 > > From: "Extension 1008" > ;transport=udp>;tag=FZp353yettFNN > > To: > >;tag=03bd8e75ec97c9ee65c772e401792a5a.5fa3 > > Call-ID: 6b05cee4-6e28-122c-8396-000c7680c408 > > CSeq: 110801210 ACK > > Content-Length: 0 > > > > > ------------------------------------------------------------------------ > > send 1409 bytes to udp/[194.9.25.21]:5060 at 13:04:52.994530: > > > ------------------------------------------------------------------------ > > INVITE sip:0225490317 at sip.halonet.plSIP/2.0 > > Via: SIP/2.0/UDP 83.13.98.165:5080;rport;branch=z9hG4bKZ5c4e66Z8Z4mj > > Max-Forwards: 69 > > From: "Extension 1008" > ;transport=udp>;tag=FZp353yettFNN > > To: > > > Call-ID: 6b05cee4-6e28-122c-8396-000c7680c408 > > CSeq: 110801211 INVITE > > Contact: > > Expires: 600 > > User-Agent: FreeSWITCH-mod_sofia/1.0.2-wyeksportowane > > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO > > Supported: timer, precondition, path, replaces > > Allow-Events: talk, refer > > Proxy-Authorization: Digest username="fixed0248b", realm=" > sip.halonet.pl", nonce="498ae51439454939d240f51ab9455e2c8505c7aa", > cnonce="a1oY524oEiyWgwAMdoDECA", algorithm=MD5, uri=" > sip:0225490317 at sip.halonet.pl ", > response="f05fad749d4ff6d5a75d61e3283c3afa", qop=auth, nc=00000001 > > Min-SE: 120 > > Content-Type: application/sdp > > Content-Disposition: session > > Content-Length: 309 > > Remote-Party-ID: "Extension 1008" > >;screen=yes;privacy=off > > > > v=0 > > o=FreeSWITCH 3996368035436745512 7588776177330183331 IN IP4 > 89.77.89.244 > > s=FreeSWITCH > > c=IN IP4 89.77.89.244 > > t=0 0 > > m=audio 50863 RTP/AVP 0 8 3 101 13 > > a=rtpmap:0 PCMU/8000 > > a=rtpmap:8 PCMA/8000 > > a=rtpmap:3 GSM/8000 > > a=rtpmap:101 telephone-event/8000 > > a=fmtp:101 0-16 > > a=rtpmap:13 CN/8000 > > a=ptime:20 > > > ------------------------------------------------------------------------ > > 2009-02-05 14:04:52 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state() > Channel sofia/external/0225490317 entering state [calling] > > recv 618 bytes from udp/[194.9.25.21]:5060 at 13:04:53.349700: > > > ------------------------------------------------------------------------ > > SIP/2.0 100 trying -- your call is important to us > > Via: SIP/2.0/UDP 83.13.98.165:5080 > ;rport=5080;branch=z9hG4bKZ5c4e66Z8Z4mj;received=89.77.89.244 > > From: "Extension 1008" > ;transport=udp>;tag=FZp353yettFNN > > To: > > > Call-ID: 6b05cee4-6e28-122c-8396-000c7680c408 > > CSeq: 110801211 INVITE > > Server: Sip EXpress router (2.0.0-rc1 (i386/linux)) > > Content-Length: 0 > > Warning: 392 194.9.25.21:5060 "Noisy feedback tells: pid=22332 > req_src_ip=89.77.89.244 req_src_port=50866 in_uri= > sip:0225490317 at sip.halonet.pl out_uri= > sip:0225490317 at 217.11.128.50:5060 via_cnt==1" > > > > > ------------------------------------------------------------------------ > > recv 469 bytes from udp/[194.9.25.21]:5060 at 13:04:53.861812: > > > ------------------------------------------------------------------------ > > SIP/2.0 500 Internal Server Error > > Via: SIP/2.0/UDP 83.13.98.165:5080 > ;received=89.77.89.244;rport=5080;branch=z9hG4bKZ5c4e66Z8Z4mj > > From: "Extension 1008" > ;transport=udp>;tag=FZp353yettFNN > > To: > >;tag=4961D2C4-DA7 > > Date: Thu, 05 Feb 2009 13:04:39 GMT > > Call-ID: 6b05cee4-6e28-122c-8396-000c7680c408 > > Server: Cisco-SIPGateway/IOS-12.x > > CSeq: 110801211 INVITE > > Allow-Events: telephone-event > > Content-Length: 0 > > > > > ------------------------------------------------------------------------ > > send 362 bytes to udp/[194.9.25.21]:5060 at 13:04:53.862199: > > > ------------------------------------------------------------------------ > > ACK sip:0225490317 at sip.halonet.pl SIP/2.0 > > Via: SIP/2.0/UDP 83.13.98.165:5080;rport;branch=z9hG4bKZ5c4e66Z8Z4mj > > Max-Forwards: 69 > > From: "Extension 1008" > ;transport=udp>;tag=FZp353yettFNN > > To: > >;tag=4961D2C4-DA7 > > Call-ID: 6b05cee4-6e28-122c-8396-000c7680c408 > > CSeq: 110801211 ACK > > Content-Length: 0 > > > > > ------------------------------------------------------------------------ > > 2009-02-05 14:04:53 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state() > Channel sofia/external/0225490317 entering state [terminated] > > 2009-02-05 14:04:53 [NOTICE] sofia.c:3090 sofia_handle_sip_i_state() > Hangup sofia/external/0225490317 [CS_CONSUME_MEDIA] > [NORMAL_TEMPORARY_FAILURE] > > 2009-02-05 14:04:53 [DEBUG] switch_channel.c:1494 > switch_channel_perform_hangup() Send signal sofia/external/0225490317 [KILL] > > 2009-02-05 14:04:53 [DEBUG] switch_core_session.c:807 > switch_core_session_signal_state_change() Send signal > sofia/external/0225490317 [BREAK] > > 2009-02-05 14:04:53 [DEBUG] switch_core_state_machine.c:466 > switch_core_session_run() (sofia/external/0225490317) State CONSUME_MEDIA > going to sleep > > 2009-02-05 14:04:53 [DEBUG] switch_core_state_machine.c:379 > switch_core_session_run() (sofia/external/0225490317) Running State Change > CS_HANGUP > > 2009-02-05 14:04:53 [DEBUG] switch_core_state_machine.c:410 > switch_core_session_run() (sofia/external/0225490317) State HANGUP > > 2009-02-05 14:04:53 [DEBUG] mod_sofia.c:253 sofia_on_hangup() > sofia/external/0225490317 Overriding SIP cause 503 with 500 from the other > leg > > 2009-02-05 14:04:53 [DEBUG] mod_sofia.c:287 sofia_on_hangup() Channel > sofia/external/0225490317 hanging up, cause: NORMAL_TEMPORARY_FAILURE > > 2009-02-05 14:04:53 [DEBUG] switch_core_state_machine.c:46 > switch_core_standard_on_hangup() sofia/external/0225490317 Standard HANGUP, > cause: NORMAL_TEMPORARY_FAILURE > > 2009-02-05 14:04:53 [DEBUG] switch_core_state_machine.c:410 > switch_core_session_run() (sofia/external/0225490317) State HANGUP going to > sleep > > 2009-02-05 14:04:53 [DEBUG] switch_core_session.c:939 > switch_core_session_thread() Session 16 (sofia/external/0225490317) Locked, > Waiting on external entities > > 2009-02-05 14:04:53 [DEBUG] switch_ivr_originate.c:1695 > switch_ivr_originate() Originate Resulted in Error Cause: 41 > [NORMAL_TEMPORARY_FAILURE] > > 2009-02-05 14:04:53 [DEBUG] switch_channel.c:177 > switch_channel_audio_sync() sofia/internal/sip:1008 at 192.168.1.126:5070receive message [AUDIO_SYNC] > > 2009-02-05 14:04:53 [NOTICE] switch_core_session.c:957 > switch_core_session_thread() Session 16 (sofia/external/0225490317) Ended > > 2009-02-05 14:04:53 [INFO] mod_dptools.c:1909 audio_bridge_function() > Originate Failed. Cause: NORMAL_TEMPORARY_FAILURE > > 2009-02-05 14:04:53 [NOTICE] mod_dptools.c:1936 audio_bridge_function() > Hangup sofia/internal/sip:1008 at 192.168.1.126:5070 [CS_EXECUTE] > [NORMAL_TEMPORARY_FAILURE] > > 2009-02-05 14:04:53 [NOTICE] switch_core_session.c:959 > switch_core_session_thread() Close Channel sofia/external/0225490317 > [CS_HANGUP] > > 2009-02-05 14:04:53 [DEBUG] switch_channel.c:1494 > switch_channel_perform_hangup() Send signal sofia/internal/ > sip:1008 at 192.168.1.126:5070 [KILL] > > 2009-02-05 14:04:53 [DEBUG] switch_core_session.c:807 > switch_core_session_signal_state_change() Send signal sofia/internal/ > sip:1008 at 192.168.1.126:5070 [BREAK] > > 2009-02-05 14:04:53 [DEBUG] switch_core_state_machine.c:454 > switch_core_session_run() (sofia/internal/sip:1008 at 192.168.1.126:5070) > State EXECUTE going to sleep > > 2009-02-05 14:04:53 [DEBUG] switch_core_state_machine.c:379 > switch_core_session_run() (sofia/internal/sip:1008 at 192.168.1.126:5070) > Running State Change CS_HANGUP > > 2009-02-05 14:04:53 [DEBUG] switch_core_state_machine.c:410 > switch_core_session_run() (sofia/internal/sip:1008 at 192.168.1.126:5070) > State HANGUP > > 2009-02-05 14:04:53 [DEBUG] mod_sofia.c:253 sofia_on_hangup() > sofia/internal/sip:1008 at 192.168.1.126:5070 Overriding SIP cause 503 with > 500 from the other leg > > 2009-02-05 14:04:53 [DEBUG] mod_sofia.c:287 sofia_on_hangup() Channel > sofia/internal/sip:1008 at 192.168.1.126:5070 hanging up, cause: > NORMAL_TEMPORARY_FAILURE > > 2009-02-05 14:04:53 [DEBUG] mod_sofia.c:344 sofia_on_hangup() Sending BYE > to sofia/internal/sip:1008 at 192.168.1.126:5070 > > send 648 bytes to udp/[192.168.1.126]:5070 at 13:04:53.868506: > > > ------------------------------------------------------------------------ > > BYE sip:1008 at 192.168.1.126:5070 SIP/2.0 > > Via: SIP/2.0/UDP 192.168.1.122;rport;branch=z9hG4bKH893yar8p7X4j > > Max-Forwards: 70 > > From: "FreeSWITCH" > >;tag=9vpFm3tBc8gSe > > To: ;tag=2111843199 > > Call-ID: 68736fc7-6e28-122c-8396-000c7680c408 > > CSeq: 110801209 BYE > > Contact: > > User-Agent: FreeSWITCH-mod_sofia/1.0.2-wyeksportowane > > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH > > Supported: timer, precondition, path, replaces > > Reason: Q.850;cause=41;text="NORMAL_TEMPORARY_FAILURE" > > Content-Length: 0 > > > > > ------------------------------------------------------------------------ > > 2009-02-05 14:04:53 [DEBUG] switch_core_state_machine.c:46 > switch_core_standard_on_hangup() sofia/internal/ > sip:1008 at 192.168.1.126:5070 Standard HANGUP, cause: > NORMAL_TEMPORARY_FAILURE > > 2009-02-05 14:04:53 [DEBUG] switch_core_state_machine.c:410 > switch_core_session_run() (sofia/internal/sip:1008 at 192.168.1.126:5070) > State HANGUP going to sleep > > 2009-02-05 14:04:53 [DEBUG] switch_core_session.c:939 > switch_core_session_thread() Session 15 (sofia/internal/ > sip:1008 at 192.168.1.126:5070) Locked, Waiting on external entities > > 2009-02-05 14:04:53 [NOTICE] switch_core_session.c:957 > switch_core_session_thread() Session 15 (sofia/internal/ > sip:1008 at 192.168.1.126:5070) Ended > > 2009-02-05 14:04:53 [NOTICE] switch_core_session.c:959 > switch_core_session_thread() Close Channel sofia/internal/ > sip:1008 at 192.168.1.126:5070 [CS_HANGUP] > > recv 354 bytes from udp/[192.168.1.126]:5070 at 13:04:53.882344: > > > ------------------------------------------------------------------------ > > SIP/2.0 200 Ok > > Via: SIP/2.0/UDP 192.168.1.122;rport;branch=z9hG4bKH893yar8p7X4j > > From: "FreeSWITCH" > >;tag=9vpFm3tBc8gSe > > To: ;tag=2111843199 > > Contact: > > Call-ID: 68736fc7-6e28-122c-8396-000c7680c408 > > CSeq: 110801209 BYE > > Server: X-Lite release 1105d > > Content-Length: 0 > > > > > ------------------------------------------------------------------------ > Dnia 04-02-2009, ?ro o godzinie 10:22 -0600, Anthony Minessale pisze: > > can you press f8 for debug and try that apiExecute and post the > > results? > > > > On Wed, Feb 4, 2009 at 8:46 AM, Jacek Sokulski > > wrote: > > Thanks Anthony, > > the js snippets are very instructive. > > A couple of points: > > 1. The code with apiExecute does not work (local phone is > > connected, but > > after picking up it hungs up immediately), other examples are > > working > > fine. > > > > 2. It does not show how initiate external call without > > existing session. > > > > 3. How can one pass the call through dialplan? > > > > Jacek > > > > PS. > > we got the code probable from wiki or from this mialing list. > > > > Dnia 04-02-2009, ?ro o godzinie 08:09 -0600, Anthony Minessale > > pisze: > > > > > Where did you learn how to use js this way? > > > session.originate is being misused here and is depricated > > and may be > > > removed. > > > > > > the first arg to session.originate is either undefined or a > > > *different* session (the a leg) > > > > > > session1 = new Session(); > > > session1.originate(undefined, > > > "{ignore_early_media=true}user/1008 at 192.168.1.122"); > > > > > > > > session1.setVariable("effective_caller_id_number","fixed0248b"); > > > > > > //once you have session1 when you originate session2 you > > pass session1 > > > as the arg > > > // the effective_caller_id is taken from session1 > > > > > > session2 = new Session(); > > > session2.originate(session1, > > "sofia/gateway/halonet/0225490317"); > > > > > > Anyway this whole code is depricated in favor of this: > > > > > > session1 = new > > > Session("{ignore_early_media=true}user/1008 at 192.168.1.122"); > > > > > > if (session1.ready()) { > > > > > session1.setVariable("effective_caller_id_number","fixed0248b"); > > > session2 = new Session("sofia/gateway/halonet/0225490317", > > > session1); > > > } > > > > > > and could be further refactored down to this: > > > > > > session1 = new > > > Session("{ignore_early_media=true}user/1008 at 192.168.1.122"); > > > if (session1.ready()) { > > > > > session1.setVariable("effective_caller_id_number","fixed0248b"); > > > session1.execute("bridge", > > "sofia/gateway/halonet/0225490317"); > > > } > > > > > > or down to this one line of code that will setup the call > > detached > > > from the script and exit. > > > > > > var result = apiExecute("originate", > > > > > > "{effective_caller_id_number=fixed0248b,origination_caller_id_number=1000,ignore_early_media=true}user/ > 1008 at 192.168.1.122 bridge:sofia/gateway/halonet/0225490317 inline"); > > > > > > if you dont care about the result and want to exit even > > before the > > > call is completed. > > > > > > var result = apiExecute("bgapi", "originate > > > > > > {effective_caller_id_number=fixed0248b,origination_caller_id_number=1000,ignore_early_media=true}user/ > 1008 at 192.168.1.122 bridge:sofia/gateway/halonet/0225490317 inline"); > > > > > > > > > > > > On Wed, Feb 4, 2009 at 2:51 AM, Jacek Sokulski > > > wrote: > > > > > > We have tried setting both > > effective_caller_id_number and > > > origination_caller_id_number: > > > > > > > > > session1.originate(session1,"{origination_caller_id_number=fixed0248b}sofia/gateway/halonet/0225490317",15); > > > but the problem still exists. The solution we have > > found for > > > the case > > > when we originate two calls, local and external, is > > as follow: > > > > > > session1 = new Session(); > > > > > session1.originate(session1,"user/1003 at 192.168.1.122 > ",15);//local > > > if(session1.ready()) { > > > > > session1.execute("execute_extension","00930691688627 XML > > > default");//external > > > } > > > > > > so the external call goes through the dialplan. > > > It does not work if both calls are external. One > > possible > > > solution could be > > > to pass the originating call through dialplan > > (loopback?) but > > > we have not managed > > > to figure out how to do it. > > > > > > Thanks > > > Jacek > > > > > > Dnia 03-02-2009, wto o godzinie 14:31 -0300, Nicolas > > Brenner > > > pisze: > > > > > > > Oops! Well, fortunately I don't use that voip > > provider > > > anymore (nor the script). > > > > > > > > Thanks Brian. > > > > > > > > Nicolas > > > > > > > > On Tue, Feb 3, 2009 at 2:25 PM, Brian West > > > wrote: > > > > > YOU should NEVER use this method or call > > setCallerData at > > > all you > > > > > should use the correct methods to override the > > callerid. > > > > > > > > > > If its a B-Leg born from an A-Leg you use these > > on the on > > > the A-Leg: > > > > > > > > > > > > > > > > http://wiki.freeswitch.org/wiki/Channel_Variables#effective_caller_id_name > > > > > > > > > > > http://wiki.freeswitch.org/wiki/Channel_Variables#effective_caller_id_number > > > > > > > > > > If you're originating you use this: > > > > > > > > > > > > > > > > http://wiki.freeswitch.org/wiki/Channel_Variables#origination_caller_id_name > > > > > > > > > > > http://wiki.freeswitch.org/wiki/Channel_Variables#origination_caller_id_number > > > > > > > > > > /b > > > > > > > > _______________________________________________ > > > > Freeswitch-users mailing list > > > > Freeswitch-users at lists.freeswitch.org > > > > > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > > > > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > http://www.freeswitch.org > > > > > > > > > _______________________________________________ > > > Freeswitch-users mailing list > > > Freeswitch-users at lists.freeswitch.org > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > > > > > > > > > > > > > -- > > > Anthony Minessale II > > > > > > FreeSWITCH http://www.freeswitch.org/ > > > ClueCon http://www.cluecon.com/ > > > > > > AIM: anthm > > > MSN:anthony_minessale at hotmail.com > > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > > IRC: irc.freenode.net #freeswitch > > > > > > FreeSWITCH Developer Conference > > > sip:888 at conference.freeswitch.org > > > iax:guest at conference.freeswitch.org/888 > > > googletalk:conf+888 at conference.freeswitch.org > > > pstn:213-799-1400 > > > _______________________________________________ > > > Freeswitch-users mailing list > > > Freeswitch-users at lists.freeswitch.org > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090205/0f69fe0e/attachment-0002.html From anthony.minessale at gmail.com Thu Feb 5 07:48:12 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 5 Feb 2009 09:48:12 -0600 Subject: [Freeswitch-users] FreeSwitch setup as a "Dumb" SBC In-Reply-To: <498AAC59.60407@post.cz> References: <498AAC59.60407@post.cz> Message-ID: <191c3a030902050748p44a86740hd823aedee5d81f56@mail.gmail.com> fyi, I am pretty sure i disabled the mmap calls so it's not memory mapped it's all in ram the whole time. The fsxml file is just left behind to show you the last successful load On Thu, Feb 5, 2009 at 3:07 AM, kokoska rokoska wrote: > > > > Ken Rice napsal(a): > > That file is memory mapped but it can still take an IO hit... Mounting > fs/db > > in a ramdrive still helps out and it doesn't have to be that big of a ram > > drive for the testing you are doing > > > > Thank you very much, Ken, for that info! > > I do it like I wrote before, but I have to wait till the "blades" aren't > busy. They are not only for my testing and the other users have "tasks" > with higher priority than my sipp testing :-) > > As soon as I have the result, I'll post them. I'm pretty sure, that > changes in FS code and "all in RAM" helps much :-) > > Best regards, > > kokoska.rokoska > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090205/7563273b/attachment-0002.html From kerrada2003 at yahoo.com Thu Feb 5 08:34:08 2009 From: kerrada2003 at yahoo.com (Ali Al-Rubaie) Date: Thu, 5 Feb 2009 08:34:08 -0800 (PST) Subject: [Freeswitch-users] SIP Authentication In-Reply-To: Message-ID: <939444.77774.qm@web33707.mail.mud.yahoo.com> We're using HelpCaster softphone. The issue here is that in Digest Authentication, if the server sends the parameter "qop" in the challenge then the client should respond with the "cnonce" parameter. The parameter "qop" is optional in Digest Auth. So the question here is that, can we configure FreeSWITCH so that it will not send "qop" in the challenge? Thanks! --- On Wed, 2/4/09, freeswitch-users-request at lists.freeswitch.org wrote: From: freeswitch-users-request at lists.freeswitch.org Subject: Freeswitch-users Digest, Vol 32, Issue 39 To: freeswitch-users at lists.freeswitch.org Date: Wednesday, February 4, 2009, 2:05 PM Send Freeswitch-users mailing list submissions to freeswitch-users at lists.freeswitch.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.freeswitch.org/mailman/listinfo/freeswitch-users or, via email, send a message with subject or body 'help' to freeswitch-users-request at lists.freeswitch.org You can reach the person managing the list at freeswitch-users-owner at lists.freeswitch.org When replying, please edit your Subject line so it is more specific than "Re: Contents of Freeswitch-users digest..." Today's Topics: 1. Re: SIP Authentication (Brian West) 2. Re: origainate through sofia gateway (Michael Collins) 3. Recording background music and voice is out of sync (Daniel Liang) 4. Re: Q931 decoding Update (Gopalakrishnan A.N) 5. mod_limit (Chav Paskov) 6. Re: mod_limit (Michael Collins) 7. Re: mod_limit (Chav Paskov) 8. Re: mod_limit (Michael Collins) ---------------------------------------------------------------------- Message: 1 Date: Wed, 4 Feb 2009 10:52:45 -0600 From: Brian West Subject: Re: [Freeswitch-users] SIP Authentication To: freeswitch-users at lists.freeswitch.org Message-ID: <7DAC91F6-2FD3-464D-AA81-321EBCADC8C0 at freeswitch.org> Content-Type: text/plain; charset=US-ASCII; format=flowed; delsp=yes What client is this? I also notice we receive port 3458 and reply to port 1059... /b On Feb 4, 2009, at 10:17 AM, Ali Al-Rubaie wrote: > What I have noted is that the client does not send the values for > "cnonce" and "nc" in the response. I'm not sure if this is the > reason, however how this problem can be solved? > > Thanks, > > Ali ------------------------------ Message: 2 Date: Wed, 4 Feb 2009 09:41:07 -0800 From: Michael Collins Subject: Re: [Freeswitch-users] origainate through sofia gateway To: freeswitch-users at lists.freeswitch.org Message-ID: <87f2f3b90902040941r61d669aaie949aa7cc8578a9a at mail.gmail.com> Content-Type: text/plain; charset=ISO-8859-1 I'll make sure the substance of this is in the wiki and I'll look for references to the deprecated way and remove those. -MC On Wed, Feb 4, 2009 at 6:09 AM, Anthony Minessale wrote: > Where did you learn how to use js this way? > session.originate is being misused here and is depricated and may be > removed. > > the first arg to session.originate is either undefined or a *different* > session (the a leg) > > session1 = new Session(); > session1.originate(undefined, > "{ignore_early_media=true}user/1008 at 192.168.1.122"); > > session1.setVariable("effective_caller_id_number","fixed0248b"); > > //once you have session1 when you originate session2 you pass session1 as > the arg > // the effective_caller_id is taken from session1 > > session2 = new Session(); > session2.originate(session1, "sofia/gateway/halonet/0225490317"); > > Anyway this whole code is depricated in favor of this: > > session1 = new Session("{ignore_early_media=true}user/1008 at 192.168.1.122"); > if (session1.ready()) { > session1.setVariable("effective_caller_id_number","fixed0248b"); > session2 = new Session("sofia/gateway/halonet/0225490317", session1); > } > > and could be further refactored down to this: > > session1 = new Session("{ignore_early_media=true}user/1008 at 192.168.1.122"); > if (session1.ready()) { > session1.setVariable("effective_caller_id_number","fixed0248b"); > session1.execute("bridge", "sofia/gateway/halonet/0225490317"); > } > > or down to this one line of code that will setup the call detached from the > script and exit. > > var result = apiExecute("originate", > "{effective_caller_id_number=fixed0248b,origination_caller_id_number=1000,ignore_early_media=true}user/1008 at 192.168.1.122 > bridge:sofia/gateway/halonet/0225490317 inline"); > > if you dont care about the result and want to exit even before the call is > completed. > > var result = apiExecute("bgapi", "originate > {effective_caller_id_number=fixed0248b,origination_caller_id_number=1000,ignore_early_media=true}user/1008 at 192.168.1.122 > bridge:sofia/gateway/halonet/0225490317 inline"); > > > > On Wed, Feb 4, 2009 at 2:51 AM, Jacek Sokulski > wrote: >> >> We have tried setting both effective_caller_id_number and >> origination_caller_id_number: >> >> >> session1.originate(session1,"{origination_caller_id_number=fixed0248b}sofia/gateway/halonet/0225490317",15); >> but the problem still exists. The solution we have found for the case >> when we originate two calls, local and external, is as follow: >> >> session1 = new Session(); >> session1.originate(session1,"user/1003 at 192.168.1.122",15);//local >> if(session1.ready()) { >> session1.execute("execute_extension","00930691688627 XML >> default");//external >> } >> >> so the external call goes through the dialplan. >> It does not work if both calls are external. One possible solution could >> be >> to pass the originating call through dialplan (loopback?) but we have not >> managed >> to figure out how to do it. >> >> Thanks >> Jacek >> >> Dnia 03-02-2009, wto o godzinie 14:31 -0300, Nicolas Brenner pisze: >> > Oops! Well, fortunately I don't use that voip provider anymore (nor the >> > script). >> > >> > Thanks Brian. >> > >> > Nicolas >> > >> > On Tue, Feb 3, 2009 at 2:25 PM, Brian West wrote: >> > > YOU should NEVER use this method or call setCallerData at all you >> > > should use the correct methods to override the callerid. >> > > >> > > If its a B-Leg born from an A-Leg you use these on the on the A-Leg: >> > > >> > > >> > > http://wiki.freeswitch.org/wiki/Channel_Variables#effective_caller_id_name >> > > >> > > http://wiki.freeswitch.org/wiki/Channel_Variables#effective_caller_id_number >> > > >> > > If you're originating you use this: >> > > >> > > >> > > http://wiki.freeswitch.org/wiki/Channel_Variables#origination_caller_id_name >> > > >> > > http://wiki.freeswitch.org/wiki/Channel_Variables#origination_caller_id_number >> > > >> > > /b >> > >> > _______________________________________________ >> > Freeswitch-users mailing list >> > Freeswitch-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > ------------------------------ Message: 3 Date: Wed, 4 Feb 2009 09:43:10 -0800 From: "Daniel Liang" Subject: [Freeswitch-users] Recording background music and voice is out of sync To: Message-ID: <0B02E756F603CC409EB553879B090CC80A23EBB5 at HPEXCHVS01.exchange.airg> Content-Type: text/plain; charset="us-ascii" What I did was the following: First, I sent the playback command: call-command: execute execute-app-name: playback execute-app-arg: Then I send uuid_record (Sorry, it was not Record command): api uuid_record start 120 I also tried replacing the playback command with: api uuid_displace start 0 mux But the end results are the same. The recorded user's voice is about 0.5 second behind the expected result. Thanks, Daniel -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: February 3, 2009 6:36 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Recording background music and voice is outof sync Can you show us an example of how you're doing this? Playback and Record aren't async so you'll need to show us how you're doing this. Also don't hijack threads you hit replay on the one "Re: [Freeswitch- users] FreeSwitch setup as a "Dumb" SBC" as the subject, deleted the subject and started a new body. That hijacks the thread and that can cause your problem to go ignored in some cases if people aren't interested in the thread topic depending on how their reader threads the emails. Please click new message and type freeswitch- users at lists.freeswitch.org in and then input your subject and body to start a new thread. Thanks, Brian West FreeSWITCH.org On Feb 3, 2009, at 8:21 PM, Daniel Liang wrote: > Hi, > > I was trying to record a background music with a user's voice at the > same time. I did a playback and started recording. But the recorded > user's voice and the background music is about 0.5 second out of sync. > I also tried to use uuid_displace instead of playback, but I got the > same result. _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090204/2d1124e2/attachment-0001.html ------------------------------ Message: 4 Date: Wed, 4 Feb 2009 23:26:14 +0530 From: "Gopalakrishnan A.N" Subject: Re: [Freeswitch-users] Q931 decoding Update To: freeswitch-users at lists.freeswitch.org Message-ID: <2ea4d47e0902040956v75c5472foa4649c50b7340484 at mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" Hi, Its a awesome. Can the packet capturing be done with event socket? -- Thank you with regards, Gopal, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090204/74f6434a/attachment-0001.html ------------------------------ Message: 5 Date: Wed, 04 Feb 2009 09:59:48 -0800 From: Chav Paskov Subject: [Freeswitch-users] mod_limit To: freeswitch-users at lists.freeswitch.org Message-ID: <4989D794.1010805 at shaw.ca> Content-Type: text/plain; charset=ISO-8859-1; format=flowed Hi , is it possible to use mod_limit in case if the end point is not registered / gateway for example/. Regards Chav ------------------------------ Message: 6 Date: Wed, 4 Feb 2009 10:06:52 -0800 From: Michael Collins Subject: Re: [Freeswitch-users] mod_limit To: freeswitch-users at lists.freeswitch.org Message-ID: <87f2f3b90902041006v7d7d7ff2t9961274905d9d5b0 at mail.gmail.com> Content-Type: text/plain; charset=ISO-8859-1 On Wed, Feb 4, 2009 at 9:59 AM, Chav Paskov wrote: > Hi , > is it possible to use mod_limit in case if the end point is not > registered / gateway for example/. Could you add some detail to this question? What are you trying to do? (mod_limit may or may not work, but there might be another solution which is why I am asking.) -MC > Regards > Chav > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > ------------------------------ Message: 7 Date: Wed, 04 Feb 2009 10:54:56 -0800 From: Chav Paskov Subject: Re: [Freeswitch-users] mod_limit To: freeswitch-users at lists.freeswitch.org Message-ID: <4989E480.1080105 at shaw.ca> Content-Type: text/plain; charset=ISO-8859-1; format=flowed Michael Collins wrote: > On Wed, Feb 4, 2009 at 9:59 AM, Chav Paskov wrote: > >> Hi , >> is it possible to use mod_limit in case if the end point is not >> registered / gateway for example/. >> > > Could you add some detail to this question? What are you trying to do? > (mod_limit may or may not work, but there might be another solution > which is why I am asking.) > > -MC > > >> Regards >> Chav >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > i have few gateways under my ACL that are allowed to send calls to FS, but i want to be able to enforce "capacity" policy on the traffic coming from any one of them depending on total termination capacity on my termination end. Let say GW 1 has to be limited to make 10 simultaneous calls while GW2 could make up to 30 and so on. Regards Chav ------------------------------ Message: 8 Date: Wed, 4 Feb 2009 11:05:09 -0800 From: Michael Collins Subject: Re: [Freeswitch-users] mod_limit To: freeswitch-users at lists.freeswitch.org Message-ID: <87f2f3b90902041105l50f51f08t230bab8d69eefb4e at mail.gmail.com> Content-Type: text/plain; charset=ISO-8859-1 On Wed, Feb 4, 2009 at 10:54 AM, Chav Paskov wrote: > Michael Collins wrote: >> On Wed, Feb 4, 2009 at 9:59 AM, Chav Paskov wrote: >> >>> Hi , >>> is it possible to use mod_limit in case if the end point is not >>> registered / gateway for example/. >>> >> >> Could you add some detail to this question? What are you trying to do? >> (mod_limit may or may not work, but there might be another solution >> which is why I am asking.) >> >> -MC >> >> >>> Regards >>> Chav >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > i have few gateways under my ACL that are allowed to send calls to FS, > but i want to be able to enforce "capacity" policy on the traffic > coming from any one of them depending on total termination capacity on > my termination end. > Let say GW 1 has to be limited to make 10 simultaneous calls while GW2 > could make up to 30 and so on. I'm sure that this is possible. I don't personally have a way to test all of this but I know that a number of our users are doing things like this currently. Can you hop on to the IRC channel? #freeswitch on irc.freenode.net. A lot of people there can help with this one. -MC (IRC: mercutioviz) > Regards > Chav > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > ------------------------------ _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org End of Freeswitch-users Digest, Vol 32, Issue 39 ************************************************ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090205/15fa27e7/attachment-0002.html From kokoska.rokoska at post.cz Thu Feb 5 08:41:01 2009 From: kokoska.rokoska at post.cz (kokoska rokoska) Date: Thu, 05 Feb 2009 17:41:01 +0100 Subject: [Freeswitch-users] FreeSwitch setup as a "Dumb" SBC In-Reply-To: <191c3a030902050748p44a86740hd823aedee5d81f56@mail.gmail.com> References: <498AAC59.60407@post.cz> <191c3a030902050748p44a86740hd823aedee5d81f56@mail.gmail.com> Message-ID: <498B169D.8080105@post.cz> Anthony Minessale napsal(a): > fyi, > > I am pretty sure i disabled the mmap calls so it's not memory mapped > it's all in ram the whole time. > The fsxml file is just left behind to show you the last successful load > Thank you very much, Anthony, for very useful info! Best regards, kokoska.rokoska From anthony.minessale at gmail.com Thu Feb 5 08:46:54 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 5 Feb 2009 10:46:54 -0600 Subject: [Freeswitch-users] SIP Authentication In-Reply-To: <939444.77774.qm@web33707.mail.mud.yahoo.com> References: <939444.77774.qm@web33707.mail.mud.yahoo.com> Message-ID: <191c3a030902050846o60047c30pa2890707eae386d6@mail.gmail.com> It's optional for us but it's mandatory for the client if we exercise the option which we have opted to always do =D There is no way in the code to disable sending it because we prefer the more secure version of SIP auth. So again it's a bug in the client for not following the protocol. It would be considered a feature in FreeSWITCH to support limping for the sake of this broken client and we currently do not have any plans for implementing this feature. On Thu, Feb 5, 2009 at 10:34 AM, Ali Al-Rubaie wrote: > > We're using HelpCaster softphone. > > The issue here is that in Digest Authentication, if the server sends the > parameter "qop" in the challenge then the client should respond with the > "cnonce" parameter. The parameter "qop" is optional in Digest Auth. So the > question here is that, can we configure FreeSWITCH so that it will not send > "qop" in the challenge? > > Thanks! > > --- On *Wed, 2/4/09, freeswitch-users-request at lists.freeswitch.org < > freeswitch-users-request at lists.freeswitch.org>* wrote: > > From: freeswitch-users-request at lists.freeswitch.org < > freeswitch-users-request at lists.freeswitch.org> > Subject: Freeswitch-users Digest, Vol 32, Issue 39 > To: freeswitch-users at lists.freeswitch.org > Date: Wednesday, February 4, 2009, 2:05 PM > > Send Freeswitch-users mailing list submissions to > freeswitch-users at lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request at lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner at lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Freeswitch-users digest..." > > > Today's Topics: > > 1. Re: SIP Authentication (Brian West) > 2. Re: origainate through sofia gateway (Michael Collins) > 3. Recording background music and voice is out of sync (Daniel Liang) > 4. Re: Q931 decoding Update (Gopalakrishnan A.N) > 5. mod_limit (Chav Paskov) > 6. Re: mod_limit (Michael Collins) > 7. Re: mod_limit (Chav > Paskov) > 8. Re: mod_limit (Michael Collins) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Wed, 4 Feb 2009 10:52:45 -0600 > From: Brian West > Subject: Re: [Freeswitch-users] SIP Authentication > To: freeswitch-users at lists.freeswitch.org > Message-ID: <7DAC91F6-2FD3-464D-AA81-321EBCADC8C0 at freeswitch.org> > Content-Type: text/plain; charset=US-ASCII; format=flowed; delsp=yes > > What client is this? I also notice we receive port 3458 and reply to > port 1059... > > /b > > On Feb 4, 2009, at 10:17 AM, Ali Al-Rubaie wrote: > > > What I have noted is that the client does not send the values for > > "cnonce" and "nc" in the response. I'm not sure if > this is the > > reason, however how this problem can be solved? > > > > Thanks, > > > > Ali > > > > > ------------------------------ > > Message: > 2 > Date: Wed, 4 Feb 2009 09:41:07 -0800 > From: Michael Collins > Subject: Re: [Freeswitch-users] origainate through sofia gateway > To: freeswitch-users at lists.freeswitch.org > Message-ID: > <87f2f3b90902040941r61d669aaie949aa7cc8578a9a at mail.gmail.com> > Content-Type: text/plain; charset=ISO-8859-1 > > I'll make sure the substance of this is in the wiki and I'll look for > references to the deprecated way and remove those. > -MC > > On Wed, Feb 4, 2009 at 6:09 AM, Anthony Minessale > wrote: > > Where did you learn how to use js this way? > > session.originate is being misused here and is depricated and may be > > removed. > > > > the first arg to session.originate is either undefined or a *different* > > session (the a leg) > > > > session1 = new Session(); > > session1.originate(undefined, > > > "{ignore_early_media=true}user/1008 at 192.168.1.122"); > > > > > session1.setVariable("effective_caller_id_number","fixed0248b"); > > > > //once you have session1 when you originate session2 you pass session1 as > > the arg > > // the effective_caller_id is taken from session1 > > > > session2 = new Session(); > > session2.originate(session1, > "sofia/gateway/halonet/0225490317"); > > > > Anyway this whole code is depricated in favor of this: > > > > session1 = new > Session("{ignore_early_media=true}user/1008 at 192.168.1.122"); > > if (session1.ready()) { > > > session1.setVariable("effective_caller_id_number","fixed0248b"); > > session2 = new Session("sofia/gateway/halonet/0225490317", > session1); > > } > > > > and could be further refactored down to this: > > > > session1 = new > Session("{ignore_early_media=true}user/1008 at 192.168.1.122"); > > if > (session1.ready()) { > > > session1.setVariable("effective_caller_id_number","fixed0248b"); > > session1.execute("bridge", > "sofia/gateway/halonet/0225490317"); > > } > > > > or down to this one line of code that will setup the call detached from > the > > script and exit. > > > > var result = apiExecute("originate", > > > "{effective_caller_id_number=fixed0248b,origination_caller_id_number=1000,ignore_early_media=true}user/1008 at 192.168.1.122 > > bridge:sofia/gateway/halonet/0225490317 inline"); > > > > if you dont care about the result and want to exit even before the call is > > completed. > > > > var result = apiExecute("bgapi", "originate > > > {effective_caller_id_number=fixed0248b,origination_caller_id_number=1000,ignore_early_media=true}user/1008 at 192.168.1.122 > > bridge:sofia/gateway/halonet/0225490317 inline"); > > > > > > > > On Wed, Feb 4, 2009 at > 2:51 AM, Jacek Sokulski > > > wrote: > >> > >> We have tried setting both effective_caller_id_number and > >> origination_caller_id_number: > >> > >> > >> > session1.originate(session1,"{origination_caller_id_number=fixed0248b}sofia/gateway/halonet/0225490317",15); > >> but the problem still exists. The solution we have found for the case > >> when we originate two calls, local and external, is as follow: > >> > >> session1 = new Session(); > >> > session1.originate(session1,"user/1003 at 192.168.1.122",15);//local > >> if(session1.ready()) { > >> session1.execute("execute_extension","00930691688627 > XML > >> default");//external > >> } > >> > >> so the external call goes through the dialplan. > >> It does not work if both calls are external. One possible solution > could > >> > be > >> to pass the originating call through dialplan (loopback?) but we have > not > >> managed > >> to figure out how to do it. > >> > >> Thanks > >> Jacek > >> > >> Dnia 03-02-2009, wto o godzinie 14:31 -0300, Nicolas Brenner pisze: > >> > Oops! Well, fortunately I don't use that voip provider > anymore (nor the > >> > script). > >> > > >> > Thanks Brian. > >> > > >> > Nicolas > >> > > >> > On Tue, Feb 3, 2009 at 2:25 PM, Brian West > wrote: > >> > > YOU should NEVER use this method or call setCallerData at > all you > >> > > should use the correct methods to override the callerid. > >> > > > >> > > If its a B-Leg born from an A-Leg you use these on the on > the A-Leg: > >> > > > >> > > > >> > > > > http://wiki.freeswitch.org/wiki/Channel_Variables#effective_caller_id_name > >> > > > >> > > > http://wiki.freeswitch.org/wiki/Channel_Variables#effective_caller_id_number > >> > > > >> > > If you're originating you use this: > >> > > > >> > > > >> > > > http://wiki.freeswitch.org/wiki/Channel_Variables#origination_caller_id_name > >> > > > >> > > > http://wiki.freeswitch.org/wiki/Channel_Variables#origination_caller_id_number > >> > > > >> > > /b > >> > > >> > _______________________________________________ > >> > Freeswitch-users mailing list > >> > Freeswitch-users at lists.freeswitch.org > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> > http://www.freeswitch.org > >> > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > ------------------------------ > > Message: 3 > Date: Wed, 4 Feb 2009 09:43:10 -0800 > From: "Daniel Liang" > Subject: [Freeswitch-users] Recording background music and voice is > out of sync > To: > Message-ID: > <0B02E756F603CC409EB553879B090CC80A23EBB5 at HPEXCHVS01.exchange.airg> > Content-Type: text/plain; charset="us-ascii" > > What I did was the following: > > First, I sent the > playback command: > > call-command: execute > execute-app-name: playback > execute-app-arg: > > Then I send uuid_record (Sorry, it was not Record command): > > api uuid_record start 120 > > I also tried replacing the playback command with: > api uuid_displace start 0 mux > > But the end results are the same. The recorded user's voice is about 0.5 > second behind the expected result. > > Thanks, > Daniel > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Brian West > Sent: February 3, 2009 6:36 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Recording background music and voice is > outof sync > > Can you show us an example of how you're doing this? Playback and > Record aren't async so > you'll need to show us how you're doing > this. > > Also don't hijack threads you hit replay on the one "Re: [Freeswitch- > users] FreeSwitch setup as a "Dumb" SBC" as the subject, deleted > the > subject and started a new body. That hijacks the thread and that can > cause your problem to go ignored in some cases if people aren't > interested in the thread topic depending on how their reader threads the > emails. > > Please click new message and type freeswitch- users at lists.freeswitch.org > in and then input your subject and body to start a new thread. > > Thanks, > Brian West > FreeSWITCH.org > > > On Feb 3, 2009, at 8:21 PM, Daniel Liang wrote: > > > Hi, > > > > I was trying to record a background music with a user's voice at the > > same time. I did a playback and started recording. But the recorded > > user's voice and the background music is about 0.5 second out of sync. > > > I also tried > to use uuid_displace instead of playback, but I got the > > same result. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090204/2d1124e2/attachment-0001.html > > > ------------------------------ > > Message: 4 > Date: Wed, 4 Feb 2009 23:26:14 +0530 > From: "Gopalakrishnan A.N" > Subject: Re: [Freeswitch-users] Q931 decoding Update > To: freeswitch-users at lists.freeswitch.org > Message-ID: > > <2ea4d47e0902040956v75c5472foa4649c50b7340484 at mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > Hi, > Its a awesome. Can the packet capturing be done with event socket? > > -- > Thank you with regards, > Gopal, > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090204/74f6434a/attachment-0001.html > > > ------------------------------ > > Message: 5 > Date: Wed, 04 Feb 2009 09:59:48 -0800 > From: Chav Paskov > Subject: [Freeswitch-users] mod_limit > To: freeswitch-users at lists.freeswitch.org > Message-ID: <4989D794.1010805 at shaw.ca> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Hi , > is it possible to use mod_limit in case if the end point is not > registered / gateway for > example/. > Regards > Chav > > > > ------------------------------ > > Message: 6 > Date: Wed, 4 Feb 2009 10:06:52 -0800 > From: Michael Collins > Subject: Re: [Freeswitch-users] mod_limit > To: freeswitch-users at lists.freeswitch.org > Message-ID: > <87f2f3b90902041006v7d7d7ff2t9961274905d9d5b0 at mail.gmail.com> > Content-Type: text/plain; charset=ISO-8859-1 > > On Wed, Feb 4, 2009 at 9:59 AM, Chav Paskov wrote: > > Hi , > > is it possible to use mod_limit in case if the end point is not > > registered / gateway for example/. > > Could you add some detail to this question? What are you trying to do? > (mod_limit may or may not work, but there might be another solution > which is why I am asking.) > > -MC > > > Regards > > Chav > > > > _______________________________________________ > > Freeswitch-users mailing list > > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > ------------------------------ > > Message: 7 > Date: Wed, 04 Feb 2009 10:54:56 -0800 > From: Chav Paskov > Subject: Re: [Freeswitch-users] mod_limit > To: freeswitch-users at lists.freeswitch.org > Message-ID: <4989E480.1080105 at shaw.ca> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Michael Collins wrote: > > On Wed, Feb 4, 2009 at 9:59 AM, Chav Paskov > wrote: > > > >> Hi , > >> is it possible to use mod_limit in case if the end point is not > >> registered / gateway for example/. > >> > > > > Could you add some detail to this question? What are you trying to do? > > > (mod_limit may or may not work, but there might be another solution > > which is why I am asking.) > > > > -MC > > > > > >> Regards > >> Chav > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > >> > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > i have few gateways under my ACL that > are allowed to send calls to FS, > but i want to be able to enforce "capacity" policy on the traffic > coming from any one of them depending on total termination capacity on > my termination end. > Let say GW 1 has to be limited to make 10 simultaneous calls while GW2 > could make up to 30 and so on. > Regards > Chav > > > > ------------------------------ > > Message: 8 > Date: Wed, 4 Feb 2009 11:05:09 -0800 > From: Michael Collins > Subject: Re: [Freeswitch-users] mod_limit > To: freeswitch-users at lists.freeswitch.org > Message-ID: > <87f2f3b90902041105l50f51f08t230bab8d69eefb4e at mail.gmail.com> > Content-Type: text/plain; charset=ISO-8859-1 > > On Wed, Feb 4, 2009 at 10:54 AM, Chav Paskov wrote: > > Michael Collins wrote: > >> On Wed, Feb 4, 2009 at 9:59 AM, Chav Paskov > wrote: > >> > >>> Hi > , > >>> is it possible to use mod_limit in case if the end point is not > >>> registered / gateway for example/. > >>> > >> > >> Could you add some detail to this question? What are you trying to do? > >> (mod_limit may or may not work, but there might be another solution > >> which is why I am asking.) > >> > >> -MC > >> > >> > >>> Regards > >>> Chav > >>> > >>> _______________________________________________ > >>> Freeswitch-users mailing list > >>> Freeswitch-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> http://www.freeswitch.org > >>> > >>> > >> > >> _______________________________________________ > >> > Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > >> > > i have few gateways under my ACL that are allowed to send calls to FS, > > but i want to be able to enforce "capacity" policy on the > traffic > > coming from any one of them depending on total termination capacity on > > my termination end. > > Let say GW 1 has to be limited to make 10 simultaneous calls while GW2 > > could make up to 30 and so on. > > I'm sure that this is possible. I don't personally have a way to test > all of this but I know that a number of our users are doing things > like this currently. Can you hop on to the IRC channel? #freeswitch on > irc.freenode.net. A lot of people there can help with > this one. > > -MC (IRC: mercutioviz) > > > Regards > > Chav > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > ------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > End of Freeswitch-users Digest, Vol 32, Issue 39 > ************************************************ > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090205/61200d9e/attachment-0002.html From kerrada2003 at yahoo.com Thu Feb 5 11:50:30 2009 From: kerrada2003 at yahoo.com (Ali Al-Rubaie) Date: Thu, 5 Feb 2009 11:50:30 -0800 (PST) Subject: [Freeswitch-users] Transcoding G723 Message-ID: <592485.77688.qm@web33706.mail.mud.yahoo.com> Hi, I need FreeSWITCH to transcode from G711 to G723 but I couldn't do so because it supports G723 only in passthru mode. So, what's the solution? Thanks, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090205/0768e009/attachment-0002.html From kerrada2003 at yahoo.com Thu Feb 5 11:54:07 2009 From: kerrada2003 at yahoo.com (Ali Al-Rubaie) Date: Thu, 5 Feb 2009 11:54:07 -0800 (PST) Subject: [Freeswitch-users] FreeSWITCH calls database Message-ID: <254392.33195.qm@web33701.mail.mud.yahoo.com> Hi, Is there a way to find some calls' statistics in FreeSWITCH, like no. of calls, durations, calls' records..etc. Thanks, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090205/ab07641a/attachment-0002.html From intralanman at freeswitch.org Thu Feb 5 11:58:15 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Thu, 05 Feb 2009 14:58:15 -0500 Subject: [Freeswitch-users] Transcoding G723 In-Reply-To: <592485.77688.qm@web33706.mail.mud.yahoo.com> References: <592485.77688.qm@web33706.mail.mud.yahoo.com> Message-ID: <498B44D7.4050108@freeswitch.org> Ali Al-Rubaie wrote: > So, what's the solution? > don't try to transcode to patent encumbered codecs ;-) -Ray -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090205/ceac53ab/attachment-0002.html From krice at freeswitch.org Thu Feb 5 11:59:41 2009 From: krice at freeswitch.org (Ken Rice) Date: Thu, 05 Feb 2009 13:59:41 -0600 Subject: [Freeswitch-users] Transcoding G723 In-Reply-To: <592485.77688.qm@web33706.mail.mud.yahoo.com> Message-ID: G723 is a patent encumbered codec as is G729. A codec module for trans-coding either of these is not available at this time. G729 may become available at some point in the future, however I don?t see G723 coming available anytime soon do to facts that there is not much of a demand for it coupled with the high costs of licensing the required patents. From: Ali Al-Rubaie Reply-To: Date: Thu, 5 Feb 2009 11:50:30 -0800 (PST) To: Subject: [Freeswitch-users] Transcoding G723 Hi, I need FreeSWITCH to transcode from G711 to G723 but I couldn't do so because it supports G723 only in passthru mode. So, what's the solution? Thanks, _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090205/071b337b/attachment-0002.html From krice at freeswitch.org Thu Feb 5 12:00:37 2009 From: krice at freeswitch.org (Ken Rice) Date: Thu, 05 Feb 2009 14:00:37 -0600 Subject: [Freeswitch-users] FreeSWITCH calls database In-Reply-To: <254392.33195.qm@web33701.mail.mud.yahoo.com> Message-ID: There are several ways to do this... See mod_xml_cdr and mod_cdr_csv K From: Ali Al-Rubaie Reply-To: Date: Thu, 5 Feb 2009 11:54:07 -0800 (PST) To: Subject: [Freeswitch-users] FreeSWITCH calls database Hi, Is there a way to find some calls' statistics in FreeSWITCH, like no. of calls, durations, calls' records..etc. Thanks, _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090205/5827cdb6/attachment-0002.html From brian at freeswitch.org Thu Feb 5 12:03:28 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 5 Feb 2009 14:03:28 -0600 Subject: [Freeswitch-users] Transcoding G723 In-Reply-To: References: Message-ID: <2FEEE7F7-DCE7-4919-A0FD-A0042BBB7EEE@freeswitch.org> This codec patent expires in a few years anyway. /b On Feb 5, 2009, at 1:59 PM, Ken Rice wrote: > I don?t see G723 coming available anytime soon do to facts that > there is not much of a demand for it coupled with the high costs of > licensing the required patents. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090205/302077ad/attachment-0002.html From sicfslist at gmail.com Thu Feb 5 12:26:32 2009 From: sicfslist at gmail.com (Shelby Ramsey) Date: Thu, 5 Feb 2009 14:26:32 -0600 Subject: [Freeswitch-users] Transcoding G723 In-Reply-To: <2FEEE7F7-DCE7-4919-A0FD-A0042BBB7EEE@freeswitch.org> References: <2FEEE7F7-DCE7-4919-A0FD-A0042BBB7EEE@freeswitch.org> Message-ID: <35b355e90902051226g12756d2fsf85fb996ec1ed188@mail.gmail.com> This is a tough deal ... and something that I do think keeps FS out of the same boat as proprietary solutions like Nextone, etc. In the real world (from a service provider view) you have customers who want to send one thing (i.e. g729 and g723 a lot from international carriers) and your vendors who will only accept a limited set (specifically g729 (maybe) and certainly g711 ulaw). So you have to really restrict what people can send you and in some cases it can be a deal killer. I'm seeing more and more wholesale vendors (especially smaller niche guys) getting away from accepting anything other than g711. I would be interested in seeing if there would be a way to have the RTP transverse a media processing blade like the ones offered from Audiocodes etc. Most have some method to tell the device to set up ports and bridge without being involved in the signaling itself. There are a couple of major advantages: -- removing the transcoding from the host to risc based processors -- not worrying about the licensing because it comes with the card and would support all codecs SDR -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090205/da5661de/attachment-0002.html From nik.middleton at noblesolutions.co.uk Thu Feb 5 12:36:33 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Thu, 5 Feb 2009 20:36:33 -0000 Subject: [Freeswitch-users] Dialplan variables Message-ID: Hi Guys, Simple question, tried asking on IRC but no joy, they're too busy slating other systems. I'm trying to dial out via a remote sip gateway via the dial plan This works fine, but I'd like to wild card the extension so it matches on anything starting with a 0 the a number > 0 How do I pass the number dialed using a variable? In asterisk I would put ${EXTEN} Finally I also have the sip gateway registered Mag gateway sip:xxx at hostname.net REGED Is it possible to use the name of this gateway instead of the IP address as in 21X.XXX.XXX/XXX ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090205/997125ce/attachment-0002.html From brian at freeswitch.org Thu Feb 5 12:45:52 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 5 Feb 2009 14:45:52 -0600 Subject: [Freeswitch-users] Transcoding G723 In-Reply-To: <35b355e90902051226g12756d2fsf85fb996ec1ed188@mail.gmail.com> References: <2FEEE7F7-DCE7-4919-A0FD-A0042BBB7EEE@freeswitch.org> <35b355e90902051226g12756d2fsf85fb996ec1ed188@mail.gmail.com> Message-ID: <11C3C72A-247D-4F8C-B106-52D49949D026@freeswitch.org> If its that critical you can do it: http://jira.freeswitch.org/browse/MODCODEC-7 /b On Feb 5, 2009, at 2:26 PM, Shelby Ramsey wrote: > This is a tough deal ... and something that I do think keeps FS out > of the same boat as proprietary solutions like Nextone, etc. In the > real world (from a service provider view) you have customers who > want to send one thing (i.e. g729 and g723 a lot from international > carriers) and your vendors who will only accept a limited set > (specifically g729 (maybe) and certainly g711 ulaw). So you have to > really restrict what people can send you and in some cases it can be > a deal killer. I'm seeing more and more wholesale vendors > (especially smaller niche guys) getting away from accepting anything > other than g711. > > I would be interested in seeing if there would be a way to have the > RTP transverse a media processing blade like the ones offered from > Audiocodes etc. > Most have some method to tell the device to set up ports and bridge > without being involved in the signaling itself. > > There are a couple of major advantages: > -- removing the transcoding from the host to risc based processors > -- not worrying about the licensing because it comes with the card > and would support all codecs > > SDR > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Thu Feb 5 12:45:22 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 5 Feb 2009 14:45:22 -0600 Subject: [Freeswitch-users] Dialplan variables In-Reply-To: References: Message-ID: <4A7EAF44-59DE-4D1F-A52E-A5B37C488BBB@freeswitch.org> Nik, We did answer you twice. ${destination_number}, But you need to not approach this with an Asterisk mindset. Example: In this example the regular expression would match everything start with a 0 and capture the zero plus all digits and put it into $1, Then in the next line you use $1 to pass what the regular expression matched. This concept is a bit different vs Asterisk. /b On Feb 5, 2009, at 2:36 PM, Nik Middleton wrote: > Hi Guys, > > Simple question, tried asking on IRC but no joy, they?re too busy > slating other systems. > > I?m trying to dial out via a remote sip gateway via the dial plan > > > > > > > > This works fine, but I?d like to wild card the extension so it > matches on anything starting with a 0 the a number > 0 > > How do I pass the number dialed using a variable? In asterisk I > would put ${EXTEN} > > Finally I also have the sip gateway registered > > Mag gateway sip:xxx at hostname.net REGED > > Is it possible to use the name of this gateway instead of the IP > address as in 21X.XXX.XXX/XXX ? > > Regards > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090205/d4836e77/attachment-0002.html From brian at freeswitch.org Thu Feb 5 12:48:07 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 5 Feb 2009 14:48:07 -0600 Subject: [Freeswitch-users] Dialplan variables In-Reply-To: References: Message-ID: <98F86FCA-5D19-4F57-A77D-9A4F57632A64@freeswitch.org> Btw I just noticed you're using a gateway... in that case you use sofia/gateway/$gatewayname_here/$1 /b On Feb 5, 2009, at 2:36 PM, Nik Middleton wrote: > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090205/63fa0b0e/attachment-0002.html From nicolas at medularis.com Thu Feb 5 12:49:19 2009 From: nicolas at medularis.com (Nicolas Brenner) Date: Thu, 5 Feb 2009 17:49:19 -0300 Subject: [Freeswitch-users] Transcoding G723 In-Reply-To: <35b355e90902051226g12756d2fsf85fb996ec1ed188@mail.gmail.com> References: <2FEEE7F7-DCE7-4919-A0FD-A0042BBB7EEE@freeswitch.org> <35b355e90902051226g12756d2fsf85fb996ec1ed188@mail.gmail.com> Message-ID: <1b46b4e80902051249x35b6b398k2dccf78a8f3d24e1@mail.gmail.com> I had to go with Asterisk because my VoIP providers only accept G729. I love FS and I have it on standby until either my providers accept other codecs (I'm trying to convince them of using Speex), or FS can transcode G729. Anyway, congratulations to the whole development team, and everybody on this list who help other people get started with FS, this is a really great project/software/platform! On Thu, Feb 5, 2009 at 5:26 PM, Shelby Ramsey wrote: > This is a tough deal ... and something that I do think keeps FS out of the > same boat as proprietary solutions like Nextone, etc. In the real world > (from a service provider view) you have customers who want to send one thing > (i.e. g729 and g723 a lot from international carriers) and your vendors who > will only accept a limited set (specifically g729 (maybe) and certainly g711 > ulaw). So you have to really restrict what people can send you and in some > cases it can be a deal killer. I'm seeing more and more wholesale vendors > (especially smaller niche guys) getting away from accepting anything other > than g711. > I would be interested in seeing if there would be a way to have the RTP > transverse a media processing blade like the ones offered from Audiocodes > etc. > Most have some method to tell the device to set up ports and bridge without > being involved in the signaling itself. > > There are a couple of major advantages: > -- removing the transcoding from the host to risc based processors > -- not worrying about the licensing because it comes with the card and > would support all codecs > SDR > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Nicol?s Brenner From nicolas at medularis.com Thu Feb 5 12:52:03 2009 From: nicolas at medularis.com (Nicolas Brenner) Date: Thu, 5 Feb 2009 17:52:03 -0300 Subject: [Freeswitch-users] Transcoding G723 In-Reply-To: <11C3C72A-247D-4F8C-B106-52D49949D026@freeswitch.org> References: <2FEEE7F7-DCE7-4919-A0FD-A0042BBB7EEE@freeswitch.org> <35b355e90902051226g12756d2fsf85fb996ec1ed188@mail.gmail.com> <11C3C72A-247D-4F8C-B106-52D49949D026@freeswitch.org> Message-ID: <1b46b4e80902051252o1b5a5c75l9f8d242cbaeec552@mail.gmail.com> In my case, I can't use hardware transcoding since I don't have physical access to the servers, I rent them. Hence I need a pure software/IP solution. On Thu, Feb 5, 2009 at 5:45 PM, Brian West wrote: > If its that critical you can do it: http://jira.freeswitch.org/browse/MODCODEC-7 > > /b > > On Feb 5, 2009, at 2:26 PM, Shelby Ramsey wrote: > >> This is a tough deal ... and something that I do think keeps FS out >> of the same boat as proprietary solutions like Nextone, etc. In the >> real world (from a service provider view) you have customers who >> want to send one thing (i.e. g729 and g723 a lot from international >> carriers) and your vendors who will only accept a limited set >> (specifically g729 (maybe) and certainly g711 ulaw). So you have to >> really restrict what people can send you and in some cases it can be >> a deal killer. I'm seeing more and more wholesale vendors >> (especially smaller niche guys) getting away from accepting anything >> other than g711. >> >> I would be interested in seeing if there would be a way to have the >> RTP transverse a media processing blade like the ones offered from >> Audiocodes etc. >> Most have some method to tell the device to set up ports and bridge >> without being involved in the signaling itself. >> >> There are a couple of major advantages: >> -- removing the transcoding from the host to risc based processors >> -- not worrying about the licensing because it comes with the card >> and would support all codecs >> >> SDR >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Nicol?s Brenner From brian at freeswitch.org Thu Feb 5 12:54:24 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 5 Feb 2009 14:54:24 -0600 Subject: [Freeswitch-users] Transcoding G723 In-Reply-To: <1b46b4e80902051252o1b5a5c75l9f8d242cbaeec552@mail.gmail.com> References: <2FEEE7F7-DCE7-4919-A0FD-A0042BBB7EEE@freeswitch.org> <35b355e90902051226g12756d2fsf85fb996ec1ed188@mail.gmail.com> <11C3C72A-247D-4F8C-B106-52D49949D026@freeswitch.org> <1b46b4e80902051252o1b5a5c75l9f8d242cbaeec552@mail.gmail.com> Message-ID: well that'll not scale far :P /b On Feb 5, 2009, at 2:52 PM, Nicolas Brenner wrote: > In my case, I can't use hardware transcoding since I don't have > physical access to the servers, I rent them. Hence I need a pure > software/IP solution. From mike at jerris.com Thu Feb 5 12:54:33 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 5 Feb 2009 15:54:33 -0500 Subject: [Freeswitch-users] spidermonkey problems In-Reply-To: <07F43D7C-3CE2-4A55-819B-E8F1095C5C8B@jerris.com> References: <191c3a030901151548w7504e2a0j5650449e20eff557@mail.gmail.com> <07F43D7C-3CE2-4A55-819B-E8F1095C5C8B@jerris.com> Message-ID: <722A8256-ECD2-4BBA-AEA2-E0B832AB60EA@jerris.com> This should now be fixed in svn trunk. Please re-test this with trunk and confirm that all is working correctly now. Mike On Jan 16, 2009, at 12:03 PM, Michael Jerris wrote: > All long running non js code should be wrapped in the suspend/resume > gc stuff. For example: > > cb_state.ret = BOOLEAN_TO_JSVAL(JS_FALSE); > cb_state.saveDepth = JS_SuspendRequest(cx); > args.input_callback = dtmf_func; > args.buf = bp; > args.buflen = len; > switch_ivr_sleep(jss->session, ms, sync, &args); > JS_ResumeRequest(cx, cb_state.saveDepth); > > I think this is your issue. Can you please file a bug on jira for > this issue (even better with a patch) > > Mike > > > > On Jan 16, 2009, at 5:54 AM, Jonas Gauffin wrote: > >> I've found the problem. one js thread wait in socket.read >> (mod_spidermonkey_socket) on data. >> That caller have hangup, which means that the garbage collector >> waits on it to close. >> >> All new javascript sessions waits in JS_AWAIT_GC_DONE for the >> garbage collector to be done before proceeding (which means that >> all new javascript calls don't do anything after being launched). >> >> My server will not send anything until an agent gets free or the >> session hangs up (detects it through the event socket). And the >> event socket will not send that the session has been hangup until >> the socket have received anything (and the script can exit). So >> it's kind of deadlock between my server and the spidermonkey_socket. >> >> Is it possible to add an option to socket.read to make it abort if >> the session have been closed? I know that I wrote >> mod_spidermonkey_socket from the start, but I can't figure out how >> to do it. >> >> Will new sessions always wait on old ones to be garbage collected >> properly? For instance, what happens if a script have a lenghty >> post process after caller have hang up? >> >> On Fri, Jan 16, 2009 at 9:38 AM, Jonas Gauffin > > wrote: >> I've got a loop, but the first thing checked in each iteration is >> if session.ready() returns false (and in that case exit the loop). >> >> I do create sessions in the script: create, try to originate to a >> destination and then finally bridge together the caller and the new >> session. >> >> I'll try to give you more details during the day. >> >> On Fri, Jan 16, 2009 at 12:48 AM, Anthony Minessale > > wrote: >> do you have any loops in your code that might not check for >> session.ready() in a exit when its not true. >> >> The symptoms you posted would be consistent with held readlocks so >> if you got a gcore (or windows equiv) of the process you might be >> able to see what threads where doing what to hang on to the read >> lock. >> >> also are you creating sessions in the script then executing app >> with them, beware of this because the thread of the script is used >> to execute apps on a session created that way and not the session >> thread. >> >> >> >> >> On Thu, Jan 15, 2009 at 5:20 AM, Jonas Gauffin > > wrote: >> Hello >> >> I got problems with hanging spidermonkey sessions and need some >> advice on how to debug them. >> >> I've made a javascript queue application that uses >> mod_spidermonkey_socket. It works fine for a while, >> but after some calls I noticed that calls didnt get transferred to >> agents. The reason was that earlier >> calls had not been terminated properly. >> >> freeswitch at test1> hupall >> 2009-01-15 12:15:04 [CRIT] switch_core_session.c:147 >> switch_core_session_hupall() Giving up with 8 sessions remaining >> API CALL [hupall()] output: >> +OK hangup all channels with cause MANAGER_REQUEST >> >> >> freeswitch at test1> show calls >> API CALL [show(calls)] output: >> >> 0 total. >> >> >> As you can see, 8 sessions are alive, but none of them are listed >> as calls. What kind of logs should I turn on to see what is >> happening with those sessions? >> >> Thanks, >> Jonas >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:213-799-1400 >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090205/92b66141/attachment-0002.html From sicfslist at gmail.com Thu Feb 5 13:01:17 2009 From: sicfslist at gmail.com (Shelby Ramsey) Date: Thu, 5 Feb 2009 15:01:17 -0600 Subject: [Freeswitch-users] Transcoding G723 In-Reply-To: References: <2FEEE7F7-DCE7-4919-A0FD-A0042BBB7EEE@freeswitch.org> <35b355e90902051226g12756d2fsf85fb996ec1ed188@mail.gmail.com> <11C3C72A-247D-4F8C-B106-52D49949D026@freeswitch.org> <1b46b4e80902051252o1b5a5c75l9f8d242cbaeec552@mail.gmail.com> Message-ID: <35b355e90902051301r4c2c53b6tb958774f4754e60f@mail.gmail.com> > > Brian, > Thanks for the link. Is anyone using this in the real world? I did think it was interesting that the author was from Sangoma ... SDR -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090205/8b71cfee/attachment-0002.html From brian at freeswitch.org Thu Feb 5 13:09:00 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 5 Feb 2009 15:09:00 -0600 Subject: [Freeswitch-users] Transcoding G723 In-Reply-To: <35b355e90902051301r4c2c53b6tb958774f4754e60f@mail.gmail.com> References: <2FEEE7F7-DCE7-4919-A0FD-A0042BBB7EEE@freeswitch.org> <35b355e90902051226g12756d2fsf85fb996ec1ed188@mail.gmail.com> <11C3C72A-247D-4F8C-B106-52D49949D026@freeswitch.org> <1b46b4e80902051252o1b5a5c75l9f8d242cbaeec552@mail.gmail.com> <35b355e90902051301r4c2c53b6tb958774f4754e60f@mail.gmail.com> Message-ID: No clue.. it was just put on jira this past weekend ;) /b On Feb 5, 2009, at 3:01 PM, Shelby Ramsey wrote: > Brian, > > Thanks for the link. Is anyone using this in the real world? I did > think it was interesting that the author was from Sangoma ... > > SDR From e.schmidbauer at gmail.com Thu Feb 5 13:11:50 2009 From: e.schmidbauer at gmail.com (e schmidbauer) Date: Thu, 5 Feb 2009 16:11:50 -0500 Subject: [Freeswitch-users] shoutcast skips In-Reply-To: <2cef777b0902031327v4725f174kcaf976c04926fb21@mail.gmail.com> References: <2cef777b0902030916s77339375pabc3d3c6fd9aec1a@mail.gmail.com> <898222CF-3335-4E2D-AE79-E57CFEE7EFB8@freeswitch.org> <2cef777b0902031226l72f4a4ceia80ec00721456602@mail.gmail.com> <574748E6-1EF0-4F5A-A1D6-338BAA711040@freeswitch.org> <2cef777b0902031259t6210b7b2w9873ced0806b98b3@mail.gmail.com> <81F41092-76A7-4DCD-8274-9584177F8752@freeswitch.org> <2cef777b0902031327v4725f174kcaf976c04926fb21@mail.gmail.com> Message-ID: <2cef777b0902051311q3e404a43q2c59fe971169ac8c@mail.gmail.com> Hey just wanted to report back on this issue. I switched OS's from Ubuntu 8.10 to Centos5.2 x64.and issue seems to be resolved. Thanks for the help on this issue. Regards. Emmanuel Schmidbauer On Tue, Feb 3, 2009 at 4:27 PM, e schmidbauer wrote: > We are attempting distributed radio. We plan on having the hosts of the > shows join the conference using CELT. But callers to the show would be > joining using regular phones therefore using lower end codecs. I will be in > the IRC shortly. > > On Tue, Feb 3, 2009 at 4:21 PM, Brian West wrote: > >> You're doing distributed radio right? So callers are calling in with CELT >> from all over the place? Can you contact us on IRC because we are very >> interested in debugging this issue. >> You can get us on IRC #freeswitch on irc.freenode.net >> >> Thanks, >> >> /b >> >> On Feb 3, 2009, at 2:59 PM, e schmidbauer wrote: >> >> FreeSWITCH Version 1.0.trunk (11567) >> check out these sample recordings >> http://bwrl.org/recordings/2009-01-31-12-07-49.mp3 >> http://bwrl.org/recordings/2009-01-31-12-07-49.wav >> http://bwrl.org/recordings/test2.mp3 >> http://bwrl.org/recordings/test2.wav >> >> the conferences were recorded as wav files, i then converted them to mp3, >> both sound the same to me >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090205/55872948/attachment-0002.html From nicolas at medularis.com Thu Feb 5 13:14:40 2009 From: nicolas at medularis.com (Nicolas Brenner) Date: Thu, 5 Feb 2009 18:14:40 -0300 Subject: [Freeswitch-users] Transcoding G723 In-Reply-To: References: <2FEEE7F7-DCE7-4919-A0FD-A0042BBB7EEE@freeswitch.org> <35b355e90902051226g12756d2fsf85fb996ec1ed188@mail.gmail.com> <11C3C72A-247D-4F8C-B106-52D49949D026@freeswitch.org> <1b46b4e80902051252o1b5a5c75l9f8d242cbaeec552@mail.gmail.com> Message-ID: <1b46b4e80902051314k238d0fb3x6ee41fa8daab1d48@mail.gmail.com> Tell that to the Amazon S3 and Amazon EC2 people ;) On Thu, Feb 5, 2009 at 5:54 PM, Brian West wrote: > well that'll not scale far :P > > /b > > On Feb 5, 2009, at 2:52 PM, Nicolas Brenner wrote: > >> In my case, I can't use hardware transcoding since I don't have >> physical access to the servers, I rent them. Hence I need a pure >> software/IP solution. > > -- Nicol?s Brenner From nik.middleton at noblesolutions.co.uk Thu Feb 5 13:14:39 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Thu, 5 Feb 2009 21:14:39 -0000 Subject: [Freeswitch-users] Dialplan variables In-Reply-To: <98F86FCA-5D19-4F57-A77D-9A4F57632A64@freeswitch.org> References: <98F86FCA-5D19-4F57-A77D-9A4F57632A64@freeswitch.org> Message-ID: That didn't work, until I removed the $ in front of the gateway name as in sofia/gateway/gatewayname_here/$1 Why is that, surely it's a variable? Regards, ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: 05 February 2009 20:48 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Dialplan variables Btw I just noticed you're using a gateway... in that case you use sofia/gateway/$gatewayname_here/$1 /b On Feb 5, 2009, at 2:36 PM, Nik Middleton wrote: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090205/870bf357/attachment-0002.html From brian at freeswitch.org Thu Feb 5 13:20:27 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 5 Feb 2009 15:20:27 -0600 Subject: [Freeswitch-users] Dialplan variables In-Reply-To: References: <98F86FCA-5D19-4F57-A77D-9A4F57632A64@freeswitch.org> Message-ID: <081F27CA-12D7-4047-8011-AED40397500A@freeswitch.org> I mean for you to replace it with your gateway name. In your case its Mag I think? /b On Feb 5, 2009, at 3:14 PM, Nik Middleton wrote: > That didn?t work, until I removed the $ in front of the gateway name > as in > > sofia/gateway/gatewayname_here/$1 > > Why is that, surely it?s a variable? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090205/a81465d7/attachment-0002.html From brian at freeswitch.org Thu Feb 5 13:20:47 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 5 Feb 2009 15:20:47 -0600 Subject: [Freeswitch-users] shoutcast skips In-Reply-To: <2cef777b0902051311q3e404a43q2c59fe971169ac8c@mail.gmail.com> References: <2cef777b0902030916s77339375pabc3d3c6fd9aec1a@mail.gmail.com> <898222CF-3335-4E2D-AE79-E57CFEE7EFB8@freeswitch.org> <2cef777b0902031226l72f4a4ceia80ec00721456602@mail.gmail.com> <574748E6-1EF0-4F5A-A1D6-338BAA711040@freeswitch.org> <2cef777b0902031259t6210b7b2w9873ced0806b98b3@mail.gmail.com> <81F41092-76A7-4DCD-8274-9584177F8752@freeswitch.org> <2cef777b0902031327v4725f174kcaf976c04926fb21@mail.gmail.com> <2cef777b0902051311q3e404a43q2c59fe971169ac8c@mail.gmail.com> Message-ID: <06712653-71EF-4C87-AA2C-CE3825F00FA3@freeswitch.org> Anyway we can look closer at the ubuntu issue also? /b On Feb 5, 2009, at 3:11 PM, e schmidbauer wrote: > Hey just wanted to report back on this issue. I switched OS's from > Ubuntu 8.10 to Centos5.2 x64.and issue seems to be resolved. Thanks > for the help on this issue. Regards. > Emmanuel Schmidbauer From nik.middleton at noblesolutions.co.uk Thu Feb 5 14:19:19 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Thu, 5 Feb 2009 22:19:19 -0000 Subject: [Freeswitch-users] Caller ID not being passed Message-ID: Try as I might, I cannot seem to get caller ID passed to the external sip gateway This GW happily processes caller id from Asterisk If tried adding param name="caller-id-in-from" value="true" in gw definition, and even in the dial plan to no avail Can anyone shed some light on this? Regards, This is a sip debug sent 1382 bytes to udp/[XXX.XXX.XXX.XXX]:5060 at 21:45:17.404609: INVITE sip:07539000000 at mygw.net SIP/2.0 Via: SIP/2.0/UDP 87.238.75.206:5080;rport;branch=z9hG4bKm2UveeUvtNc5D Max-Forwards: 69 From: "Extension 1000" ;tag=j6m5U7g4F3XNa To: Call-ID: 1e8c5d1d-6e71-122c-fdba-001a4b0a67ca CSeq: 110816823 INVITE Contact: Expires: 600 User-Agent: FreeSWITCH-mod_sofia/1.0.2-exported Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: timer, precondition, path, replaces Allow-Events: talk, refer Proxy-Authorization: Digest username="me", realm="mygw.net", nonce="498b5f19b089db1bf9d13b2c83f45407048ede9b", algorithm=MD5, uri="sip:07539600000 at mygw.net", response="03dbbeef1387a11c7100965dcfd01052" Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 311 P-Key-Flags: keys="3" -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090205/0ff99ee7/attachment-0002.html From anthony.minessale at gmail.com Thu Feb 5 14:20:18 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 5 Feb 2009 16:20:18 -0600 Subject: [Freeswitch-users] shoutcast skips In-Reply-To: <06712653-71EF-4C87-AA2C-CE3825F00FA3@freeswitch.org> References: <2cef777b0902030916s77339375pabc3d3c6fd9aec1a@mail.gmail.com> <898222CF-3335-4E2D-AE79-E57CFEE7EFB8@freeswitch.org> <2cef777b0902031226l72f4a4ceia80ec00721456602@mail.gmail.com> <574748E6-1EF0-4F5A-A1D6-338BAA711040@freeswitch.org> <2cef777b0902031259t6210b7b2w9873ced0806b98b3@mail.gmail.com> <81F41092-76A7-4DCD-8274-9584177F8752@freeswitch.org> <2cef777b0902031327v4725f174kcaf976c04926fb21@mail.gmail.com> <2cef777b0902051311q3e404a43q2c59fe971169ac8c@mail.gmail.com> <06712653-71EF-4C87-AA2C-CE3825F00FA3@freeswitch.org> Message-ID: <191c3a030902051420k5606076fub1a2c313dcb84693@mail.gmail.com> I changed the code in tree that probably fixed it in both cases. It should be good now. On Thu, Feb 5, 2009 at 3:20 PM, Brian West wrote: > Anyway we can look closer at the ubuntu issue also? > > /b > > On Feb 5, 2009, at 3:11 PM, e schmidbauer wrote: > > > Hey just wanted to report back on this issue. I switched OS's from > > Ubuntu 8.10 to Centos5.2 x64.and issue seems to be resolved. Thanks > > for the help on this issue. Regards. > > Emmanuel Schmidbauer > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090205/673ad786/attachment-0002.html From brian at freeswitch.org Thu Feb 5 14:29:24 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 5 Feb 2009 16:29:24 -0600 Subject: [Freeswitch-users] Caller ID not being passed In-Reply-To: References: Message-ID: Try application export /b On Feb 5, 2009, at 4:19 PM, Nik Middleton wrote: > Try as I might, I cannot seem to get caller ID passed to the > external sip gateway > > This GW happily processes caller id from Asterisk > > If tried adding param name="caller-id-in-from" value="true" in gw > definition, and even > > data="effective_caller_id_number=07539600000"/> in the dial plan to > no avail > > Can anyone shed some light on this? > > Regards, > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090205/65778ccf/attachment-0002.html From nik.middleton at noblesolutions.co.uk Thu Feb 5 14:43:53 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Thu, 5 Feb 2009 22:43:53 -0000 Subject: [Freeswitch-users] Caller ID not being passed In-Reply-To: References: Message-ID: No good, I tried But surely, If I have the proper values in the sip phones xml files, these should be passed to the GW should they not? Regards ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: 05 February 2009 22:29 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Caller ID not being passed Try application export /b On Feb 5, 2009, at 4:19 PM, Nik Middleton wrote: Try as I might, I cannot seem to get caller ID passed to the external sip gateway This GW happily processes caller id from Asterisk If tried adding param name="caller-id-in-from" value="true" in gw definition, and even in the dial plan to no avail Can anyone shed some light on this? Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090205/076d9ec5/attachment-0002.html From brian at freeswitch.org Thu Feb 5 14:50:18 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 5 Feb 2009 16:50:18 -0600 Subject: [Freeswitch-users] Caller ID not being passed In-Reply-To: References: Message-ID: Nope still doing it wrong. Try this: use export instead of set. > data="effective_caller_id_number=07539600000"/> -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090205/8e373dd3/attachment-0002.html From brian at freeswitch.org Thu Feb 5 14:52:24 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 5 Feb 2009 16:52:24 -0600 Subject: [Freeswitch-users] Caller ID not being passed In-Reply-To: References: Message-ID: <356C95D2-B637-47F7-91AA-BD22C83BEC2F@freeswitch.org> Nik, Ignore me... set should have worked... You're using caller-id-in-from let me look closer at this. /b On Feb 5, 2009, at 4:43 PM, Nik Middleton wrote: > No good, I tried > > data="effective_caller_id_number=07539600000"/> > > > > But surely, If I have the proper values in the sip phones xml files, > these should be passed to the GW should they not? > > Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090205/d1ac086c/attachment-0002.html From brian at freeswitch.org Thu Feb 5 14:53:36 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 5 Feb 2009 16:53:36 -0600 Subject: [Freeswitch-users] Caller ID not being passed In-Reply-To: References: Message-ID: <2294453B-8059-4AB9-8566-44E22B859C9A@freeswitch.org> Nik, While I'm looking at this can you post your full gateway and dialplan for us to see? /b On Feb 5, 2009, at 4:43 PM, Nik Middleton wrote: > No good, I tried > > data="effective_caller_id_number=07539600000"/> > > > > But surely, If I have the proper values in the sip phones xml files, > these should be passed to the GW should they not? > > Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090205/f8822068/attachment-0002.html From nik.middleton at noblesolutions.co.uk Thu Feb 5 15:12:18 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Thu, 5 Feb 2009 23:12:18 -0000 Subject: [Freeswitch-users] Caller ID not being passed In-Reply-To: <2294453B-8059-4AB9-8566-44E22B859C9A@freeswitch.org> References: <2294453B-8059-4AB9-8566-44E22B859C9A@freeswitch.org> Message-ID: Dial plan is as per default setup with the addition of the following. To be honest, and I'm no SIP guru, I can't see the caller-id being set in the sip headers Mag.xml ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: 05 February 2009 22:54 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Caller ID not being passed Nik, While I'm looking at this can you post your full gateway and dialplan for us to see? /b On Feb 5, 2009, at 4:43 PM, Nik Middleton wrote: No good, I tried But surely, If I have the proper values in the sip phones xml files, these should be passed to the GW should they not? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090205/cc69fe92/attachment-0002.html From brian at freeswitch.org Thu Feb 5 15:20:35 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 5 Feb 2009 17:20:35 -0600 Subject: [Freeswitch-users] Caller ID not being passed In-Reply-To: References: <2294453B-8059-4AB9-8566-44E22B859C9A@freeswitch.org> Message-ID: <4726C6D0-897E-46D9-91F6-6D2EF646344F@freeswitch.org> I notice you're using 1.0.2 any way you can test this with 1.0.3 RC1 tarball? /b On Feb 5, 2009, at 5:12 PM, Nik Middleton wrote: > Dial plan is as per default setup with the addition of the > following. To be honest, and I?m no SIP guru, I can?t see the caller- > id being set in the sip headers > > > > > data="effective_caller_id_number=0------00000000006"/> > data="effective_caller_id_number"/> > > > > > > Mag.xml > > > > > > > > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090205/e1250d97/attachment-0002.html From nik.middleton at noblesolutions.co.uk Thu Feb 5 15:26:27 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Thu, 5 Feb 2009 23:26:27 -0000 Subject: [Freeswitch-users] Caller ID not being passed In-Reply-To: <4726C6D0-897E-46D9-91F6-6D2EF646344F@freeswitch.org> References: <2294453B-8059-4AB9-8566-44E22B859C9A@freeswitch.org> <4726C6D0-897E-46D9-91F6-6D2EF646344F@freeswitch.org> Message-ID: Yes, I'll report back tomorrow, Regards ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: 05 February 2009 23:21 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Caller ID not being passed I notice you're using 1.0.2 any way you can test this with 1.0.3 RC1 tarball? /b On Feb 5, 2009, at 5:12 PM, Nik Middleton wrote: Dial plan is as per default setup with the addition of the following. To be honest, and I'm no SIP guru, I can't see the caller-id being set in the sip headers Mag.xml -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090205/2b86ef00/attachment-0002.html From steveu at coppice.org Thu Feb 5 15:32:01 2009 From: steveu at coppice.org (Steve Underwood) Date: Fri, 06 Feb 2009 07:32:01 +0800 Subject: [Freeswitch-users] Transcoding G723 In-Reply-To: References: <2FEEE7F7-DCE7-4919-A0FD-A0042BBB7EEE@freeswitch.org> <35b355e90902051226g12756d2fsf85fb996ec1ed188@mail.gmail.com> <11C3C72A-247D-4F8C-B106-52D49949D026@freeswitch.org> <1b46b4e80902051252o1b5a5c75l9f8d242cbaeec552@mail.gmail.com> Message-ID: <498B76F1.5070406@coppice.org> Brian West wrote: > well that'll not scale far :P > That transcoding card does 120 channels. A modern quad core CPU with a well implemented codec can do several hundred. A dual quad core chassis can do twice as much. Which one has a scaling problem? Steve > /b > > On Feb 5, 2009, at 2:52 PM, Nicolas Brenner wrote: > > >> In my case, I can't use hardware transcoding since I don't have >> physical access to the servers, I rent them. Hence I need a pure >> software/IP solution. >> From brian at freeswitch.org Thu Feb 5 15:37:24 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 5 Feb 2009 17:37:24 -0600 Subject: [Freeswitch-users] Transcoding G723 In-Reply-To: <498B76F1.5070406@coppice.org> References: <2FEEE7F7-DCE7-4919-A0FD-A0042BBB7EEE@freeswitch.org> <35b355e90902051226g12756d2fsf85fb996ec1ed188@mail.gmail.com> <11C3C72A-247D-4F8C-B106-52D49949D026@freeswitch.org> <1b46b4e80902051252o1b5a5c75l9f8d242cbaeec552@mail.gmail.com> <498B76F1.5070406@coppice.org> Message-ID: <7A658893-9EA4-4CD9-9AB6-18067FF0E445@freeswitch.org> The hardware in this case... which is why I said it wouldn't scale far :P /b On Feb 5, 2009, at 5:32 PM, Steve Underwood wrote: > That transcoding card does 120 channels. A modern quad core CPU with a > well implemented codec can do several hundred. A dual quad core > chassis > can do twice as much. Which one has a scaling problem? > > Steve From moises.silva at gmail.com Thu Feb 5 21:17:33 2009 From: moises.silva at gmail.com (Moises Silva) Date: Thu, 5 Feb 2009 23:17:33 -0600 Subject: [Freeswitch-users] Transcoding G723 In-Reply-To: <35b355e90902051301r4c2c53b6tb958774f4754e60f@mail.gmail.com> References: <2FEEE7F7-DCE7-4919-A0FD-A0042BBB7EEE@freeswitch.org> <35b355e90902051226g12756d2fsf85fb996ec1ed188@mail.gmail.com> <11C3C72A-247D-4F8C-B106-52D49949D026@freeswitch.org> <1b46b4e80902051252o1b5a5c75l9f8d242cbaeec552@mail.gmail.com> <35b355e90902051301r4c2c53b6tb958774f4754e60f@mail.gmail.com> Message-ID: At least 1 company is using it in their FS gateway for a call center of around 125 positions with their this scenario: Asterisk servers <---- IAX G711 ----> FS Gateway <--- SIP G729 ---> SIP Provider The G723 has only been tested in my laptop with an IAX connection to the FS server though. Any testing is certainly appreciated to squeeze bugs out. Moy On Thu, Feb 5, 2009 at 3:01 PM, Shelby Ramsey wrote: >> Brian, > > Thanks for the link. Is anyone using this in the real world? I did think > it was interesting that the author was from Sangoma ... > SDR > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- "I do not agree with what you have to say, but I'll defend to the death your right to say it." Voltaire From pmhshz at gmail.com Fri Feb 6 00:30:56 2009 From: pmhshz at gmail.com (shehzad p) Date: Fri, 6 Feb 2009 00:30:56 -0800 (PST) Subject: [Freeswitch-users] Freeswitch freezes on increasing call traffic In-Reply-To: <498B01CF.6080902@gmail.com> References: <21701744.post@talk.nabble.com> <191c3a030901280601t3e185ec2m76295110e4888837@mail.gmail.com> <21729863.post@talk.nabble.com> <191c3a030901290831y2ff329b3maee6c1aafa3c68e2@mail.gmail.com> <21745312.post@talk.nabble.com> <21749375.post@talk.nabble.com> <191c3a030901300818i5d85a7aawa0c53b4e54892f40@mail.gmail.com> <21761523.post@talk.nabble.com> <191c3a030901310850s553f8c11oc4e559a710def6d2@mail.gmail.com> <21825226.post@talk.nabble.com> <191c3a030902040551w5d867deta55c98bd3a3c03f8@mail.gmail.com> <21847332.post@talk.nabble.com> <4C2A4389-BEF6-454B-8B5F-FDCCC76E174A@freeswitch.org> <21848148.post@talk.nabble.com> <21852304.post@talk.nabble.com> <498B01CF.6080902@gmail.com> Message-ID: <21868458.post@talk.nabble.com> Hi all, I have opened JIRA for the same. http://jira.freeswitch.org/browse/FSCORE-285 One system is Fedora, and another one is Ubuntu. Although fs 1.0.1 was also get crashed in Ubuntu many times. Now 1.0.3.RC1 is loaded on Ubuntu, so this happen again with my Ubuntu I will surely post it. And I am now going to test latest trunk version and post back if any thing found... thanks, msp paul.degt wrote: > > Look like you use Fedora. I had a lot of issues with using Fedora as > production or load test system, in my opinion it's more like work in > progress than a production ready stable linux. If you cannot buy RHEL or > SLES use Centos. > > shehzad p wrote: >> Hi Brian, >> >> As it can be seen from the system information, there require any change >> in >> system or any suggestion... >> >> out put of uname -a is : >> Linux localhost.localdomain 2.6.23.1-42.fc8 #1 SMP Tue Oct 30 13:55:12 >> EDT >> 2007 i686 i686 i386 GNU/Linux >> >> >> Thanks, >> msp >> >> >> shehzad p wrote: >> >>> HI Brian, >>> >>> Output of ulimit -a and /proc/cpuinfo is attached... >>> http://www.nabble.com/file/p21848148/12_ulimit_and_cpuinfo.log >>> 12_ulimit_and_cpuinfo.log >>> >>> BUT...................... >>> I am running the freeswitch using below command (So ulimit set according >>> to Anthony's previous post): >>> =================================================================================== >>> ulimit -d unlimited; ulimit -f unlimited; ulimit -i unlimited; ulimit -n >>> 999999; ulimit -q unlimited ; ulimit -u unlimited; ulimit -v unlimited ; >>> ulimit -x unlimited; ulimit -s 244; ulimit -l unlimited; freeswitch >>> =================================================================================== >>> >>> Thanks >>> msp >>> >>> >>> >>> Brian West-3 wrote: >>> >>>> Can you give me the output of uname -a and the contents of /proc/ >>>> cpuinfo? Not sure I asked for this info already or not. >>>> >>>> Thanks, >>>> Brian >>>> >>>> On Feb 5, 2009, at 2:42 AM, shehzad p wrote: >>>> >>>> >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/Freeswitch-freezes-on-increasing-call-traffic-tp21701744p21868458.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From ahgindia308 at gmail.com Fri Feb 6 00:57:07 2009 From: ahgindia308 at gmail.com (Ankit Gandhi) Date: Fri, 6 Feb 2009 00:57:07 -0800 (PST) Subject: [Freeswitch-users] Freeswitch hangup cause code issue Message-ID: <21868781.post@talk.nabble.com> Hello, I want my caller to detect hangup cause as 34 so that he can try next provider according to lcr. Here is my setup. Caller -> switch (fs) -> Terminator. Now when terminator sends "503 Service Unavailable", I want to override this cause, so that the caller gets hangup case 34. (According to the terminator he is sending hangup cause 34 from his side, but in freeswitch we are getting hangup cause 41 for that call and the same hangup cause on caller side). When I tried asterisk as caller, I get hangup cause 34 in that case. But when I tried freeswitch as caller, then we are getting hangup cause 41, the same as we are getting in switch (fs). >From the switch, I tried one of this condition through javascript to override the hangup cause before sending to caller: -> session.execute("respond","503"); -> session.execute("hangup","NORMAL_CIRCUIT_CONGESTION"); -> session.execute("hangup","34"); -> session.hangup(34); In all the above cases, asterisk properly detects the hangup cause 34, but freeswitch does not detect that. It detects the same hangup cause 41 for the call. Other callers also get the same hangup cause 41 for such calls. How can I override this cause, so that the caller gets hangup cause 34 in such cases? Here is the sip trace, on the caller side returned through switch (fs). ss = switch cc = caller -------------------------------------------------------------------- U ss.ss.ss.ss:5060 -> cc.cc.cc.cc:5080 SIP/2.0 503 Service Unavailable. Via: SIP/2.0/UDP cc.cc.cc.cc:5080;rport=5080;branch=z9hG4bK8HvDU4Z88KU5m;received=122.169.29.122. From: "654321" ;tag=8p7aF18aN0mSj. To: ;tag=Nj8ypjDyvKUXe. Call-ID: fd1ab774-6ec5-122c-9fac-001cc086141d. CSeq: 110835048 INVITE. User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-hacked. Accept: application/sdp. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. Supported: timer, precondition, path, replaces. Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Reason: Q.850;cause=34;text="NORMAL_CIRCUIT_CONGESTION". Content-Length: 0. ----------------------------------------------------------------------- -- View this message in context: http://www.nabble.com/Freeswitch-hangup-cause-code-issue-tp21868781p21868781.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From helmut.kuper at ewetel.de Fri Feb 6 04:32:03 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Fri, 06 Feb 2009 13:32:03 +0100 Subject: [Freeswitch-users] mod_openzap stops working after some calls Message-ID: <498C2DC3.40701@ewetel.de> Hello, since yesterday I do a real life test with FS in a 40 sip extension environment with TDM connection via wanpipe/sangoma A104d. I detected two times the problem, that openzap stops working, while SIP calls worked. Only restarting helped. Maybe reloading of mod_openzap helps as well, but I didn't tested that, yet. I filtered FS logfile for hints of that problem and I found a growing number of this line: 2009-02-06 11:09:51 [INFO] ozmod_isdn.c:706 zap_isdn_931_34() Duplicate SETUP message(?) for Channel 1:21 ~ 1:21 in state DOWN [ignoring] Befor restart I saw such a line for *each* TDM channel and no one was able to dial out. mod_openzap handled around 40 outgoing calls until last restart . Since last restart FS runs good for 2 hours now without any duplicate SETUP in log ... Has anybody similar problems? regards Helmut From pmhshz at gmail.com Fri Feb 6 06:51:31 2009 From: pmhshz at gmail.com (shehzad p) Date: Fri, 6 Feb 2009 06:51:31 -0800 (PST) Subject: [Freeswitch-users] streamFile and read doesn't get DTMF while used with originate Message-ID: <21873924.post@talk.nabble.com> Hi all My set up is : When i originate the call from CLI, using originate command, It comes in a dialplan and from there it goes to javascript for handling simple IVR. In IVR I tested both streamFile and dialplan read application to get the DTMF from user, but it was not working, it just play the file and does nothing (streamFile doesn't call the onInput function at all).. This setup was working before, when I call a javascript from CLI, and in JS I create a session to originate the call. Is there any settings missing for my first setup, so that I can get DTMF properly... Thank.. msp -- View this message in context: http://www.nabble.com/streamFile-and-read-doesn%27t-get-DTMF-while-used-with-originate-tp21873924p21873924.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From anthony.minessale at gmail.com Fri Feb 6 07:09:08 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 6 Feb 2009 09:09:08 -0600 Subject: [Freeswitch-users] Freeswitch hangup cause code issue In-Reply-To: <21868781.post@talk.nabble.com> References: <21868781.post@talk.nabble.com> Message-ID: <191c3a030902060709m786873f9i7c31f5c0d57dce03@mail.gmail.com> you would need to provide the console output of FreeSWITCH of the entire call. with TPORT_LOG=1 env var set and console loglevel debug (press f8) On Fri, Feb 6, 2009 at 2:57 AM, Ankit Gandhi wrote: > > Hello, > I want my caller to detect hangup cause as 34 so that he can try next > provider according to lcr. > Here is my setup. > Caller -> switch (fs) -> Terminator. > Now when terminator sends "503 Service Unavailable", I want to override > this > cause, so that the caller gets hangup case 34. (According to the terminator > he is sending hangup cause 34 from his side, but in freeswitch we are > getting hangup cause 41 for that call and the same hangup cause on caller > side). > When I tried asterisk as caller, I get hangup cause 34 in that case. But > when I tried freeswitch as caller, then we are getting hangup cause 41, the > same as we are getting in switch (fs). > >From the switch, I tried one of this condition through javascript to > override the hangup cause before sending to caller: > -> session.execute("respond","503"); > -> session.execute("hangup","NORMAL_CIRCUIT_CONGESTION"); > -> session.execute("hangup","34"); > -> session.hangup(34); > In all the above cases, asterisk properly detects the hangup cause 34, but > freeswitch does not detect that. It detects the same hangup cause 41 for > the > call. Other callers also get the same hangup cause 41 for such calls. > How can I override this cause, so that the caller gets hangup cause 34 in > such cases? > > Here is the sip trace, on the caller side returned through switch (fs). > ss = switch > cc = caller > -------------------------------------------------------------------- > U ss.ss.ss.ss:5060 -> cc.cc.cc.cc:5080 > SIP/2.0 503 Service Unavailable. > Via: SIP/2.0/UDP > cc.cc.cc.cc:5080 > ;rport=5080;branch=z9hG4bK8HvDU4Z88KU5m;received=122.169.29.122. > From: "654321" > >;tag=8p7aF18aN0mSj. > To: ;tag=Nj8ypjDyvKUXe. > Call-ID: fd1ab774-6ec5-122c-9fac-001cc086141d. > CSeq: 110835048 INVITE. > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-hacked. > Accept: application/sdp. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, > REFER, UPDATE, REGISTER, INFO, PUBLISH. > Supported: timer, precondition, path, replaces. > Allow-Events: talk, presence, dialog, call-info, sla, > include-session-description, presence.winfo, message-summary, refer. > Reason: Q.850;cause=34;text="NORMAL_CIRCUIT_CONGESTION". > Content-Length: 0. > ----------------------------------------------------------------------- > > > -- > View this message in context: > http://www.nabble.com/Freeswitch-hangup-cause-code-issue-tp21868781p21868781.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090206/7cf508d5/attachment-0002.html From sicfslist at gmail.com Fri Feb 6 07:16:11 2009 From: sicfslist at gmail.com (Shelby Ramsey) Date: Fri, 6 Feb 2009 09:16:11 -0600 Subject: [Freeswitch-users] Transcoding G723 In-Reply-To: References: <2FEEE7F7-DCE7-4919-A0FD-A0042BBB7EEE@freeswitch.org> <35b355e90902051226g12756d2fsf85fb996ec1ed188@mail.gmail.com> <11C3C72A-247D-4F8C-B106-52D49949D026@freeswitch.org> <1b46b4e80902051252o1b5a5c75l9f8d242cbaeec552@mail.gmail.com> <35b355e90902051301r4c2c53b6tb958774f4754e60f@mail.gmail.com> Message-ID: <35b355e90902060716v2dff2586t253a69c97ab1adac@mail.gmail.com> Thanks Moises. It looks like good work. When is Sangoma coming out with a similar product ... Doug told me it was in the works, then not in the works, then back in the works ... The problem is this particular card is PCI only and it will only do 120 channels .... Thanks! 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URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090206/bd1c27fc/attachment-0002.html From anthony.minessale at gmail.com Fri Feb 6 07:18:18 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 6 Feb 2009 09:18:18 -0600 Subject: [Freeswitch-users] Freeswitch freezes on increasing call traffic In-Reply-To: <21868458.post@talk.nabble.com> References: <21701744.post@talk.nabble.com> <191c3a030901310850s553f8c11oc4e559a710def6d2@mail.gmail.com> <21825226.post@talk.nabble.com> <191c3a030902040551w5d867deta55c98bd3a3c03f8@mail.gmail.com> <21847332.post@talk.nabble.com> <4C2A4389-BEF6-454B-8B5F-FDCCC76E174A@freeswitch.org> <21848148.post@talk.nabble.com> <21852304.post@talk.nabble.com> <498B01CF.6080902@gmail.com> <21868458.post@talk.nabble.com> Message-ID: <191c3a030902060718m2d58cab9p757c0e95d5f8c559@mail.gmail.com> I am providing you with free software and free support. I am willing to investigate you problem for you but you must be more cooperative. I took the time to ask you several questions yesterday and you have sent 4 emails since and not addressed it. Please move any further correspondence on this issue to your jira ticket. If you do not have the necessary skill to build your own ODBC and driver as the email requests, will you please try CentOS 64 bit linux with only SVN Trunk which is a platform we trust the unixODBC and its depends. Please E-mail me privately your latest JS code. Also please try to produce a minimal script that will reproduce your issue that we can reproduce. If you continue to ignore my requests I will have no choice but to close your issue. Transcript of ignored e-mail ---------------------------------- First of all please stop using the mailing list as a bug tracker. All issues should be put into jira and managed with that. Secondly, Didn't I ask you multiple times to stop using release snapshots and please use the SVN trunk? I don't understand why you keep ignoring me and using everything but what I asked. I am not telling you to use SVN because I think it will be fixed it's so we are on the development copy of the code to get the proper line numbers etc. If you look at your 2 bt you posted, the line numbers are different on each one. What are you using on the other side of ODBC? as you can see in your bt, the call goes into ODBC then into several libs with no symbols and crashes on free. This can be a sign of corrupt memory, running out of memory or an issue in either ODBC or the database specific lib. What distro is it? What ODBC version? unixODBC? version xxx? What database driver version xxx? Is it mysl not using the proper reentrant version of the plugin? Sometimes packaged libs have bugs in them which fall out of our control. Can you build unixODBC and the plugins yourself with debug symbols so we can see if that is the cause or at the very least then we can see the debug info in the bt. please make sure you address *all* my questions in your jira report. Starting with using svn trunk, *hint* type "make current" from your rc1 distro. On Fri, Feb 6, 2009 at 2:30 AM, shehzad p wrote: > > Hi all, > > I have opened JIRA for the same. > http://jira.freeswitch.org/browse/FSCORE-285 > > One system is Fedora, and another one is Ubuntu. Although fs 1.0.1 was also > get crashed in Ubuntu many times. > Now 1.0.3.RC1 is loaded on Ubuntu, so this happen again with my Ubuntu I > will surely post it. > > And I am now going to test latest trunk version and post back if any thing > found... > > thanks, > msp > > paul.degt wrote: > > > > Look like you use Fedora. I had a lot of issues with using Fedora as > > production or load test system, in my opinion it's more like work in > > progress than a production ready stable linux. If you cannot buy RHEL or > > SLES use Centos. > > > > shehzad p wrote: > >> Hi Brian, > >> > >> As it can be seen from the system information, there require any change > >> in > >> system or any suggestion... > >> > >> out put of uname -a is : > >> Linux localhost.localdomain 2.6.23.1-42.fc8 #1 SMP Tue Oct 30 13:55:12 > >> EDT > >> 2007 i686 i686 i386 GNU/Linux > >> > >> > >> Thanks, > >> msp > >> > >> > >> shehzad p wrote: > >> > >>> HI Brian, > >>> > >>> Output of ulimit -a and /proc/cpuinfo is attached... > >>> http://www.nabble.com/file/p21848148/12_ulimit_and_cpuinfo.log > >>> 12_ulimit_and_cpuinfo.log > >>> > >>> BUT...................... > >>> I am running the freeswitch using below command (So ulimit set > according > >>> to Anthony's previous post): > >>> > =================================================================================== > >>> ulimit -d unlimited; ulimit -f unlimited; ulimit -i unlimited; ulimit > -n > >>> 999999; ulimit -q unlimited ; ulimit -u unlimited; ulimit -v unlimited > ; > >>> ulimit -x unlimited; ulimit -s 244; ulimit -l unlimited; freeswitch > >>> > =================================================================================== > >>> > >>> Thanks > >>> msp > >>> > >>> > >>> > >>> Brian West-3 wrote: > >>> > >>>> Can you give me the output of uname -a and the contents of /proc/ > >>>> cpuinfo? Not sure I asked for this info already or not. > >>>> > >>>> Thanks, > >>>> Brian > >>>> > >>>> On Feb 5, 2009, at 2:42 AM, shehzad p wrote: > >>>> > >>>> > >> > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > View this message in context: > http://www.nabble.com/Freeswitch-freezes-on-increasing-call-traffic-tp21701744p21868458.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090206/75346abc/attachment-0002.html From nik.middleton at noblesolutions.co.uk Fri Feb 6 07:51:37 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Fri, 6 Feb 2009 15:51:37 -0000 Subject: [Freeswitch-users] Caller ID not being passed In-Reply-To: References: <2294453B-8059-4AB9-8566-44E22B859C9A@freeswitch.org><4726C6D0-897E-46D9-91F6-6D2EF646344F@freeswitch.org> Message-ID: Ok, It's now working as expected, looks like I had something odd set in the phone's peer definition. I'll try and back track to see what I was doing wrong, but what ever it was, was preventing the caller id from being sent in the INVITE. Regards ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Nik Middleton Sent: 05 February 2009 23:26 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Caller ID not being passed Yes, I'll report back tomorrow, Regards ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: 05 February 2009 23:21 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Caller ID not being passed I notice you're using 1.0.2 any way you can test this with 1.0.3 RC1 tarball? /b On Feb 5, 2009, at 5:12 PM, Nik Middleton wrote: Dial plan is as per default setup with the addition of the following. To be honest, and I'm no SIP guru, I can't see the caller-id being set in the sip headers Mag.xml -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090206/3eabbdc6/attachment-0002.html From steveu at coppice.org Fri Feb 6 08:02:21 2009 From: steveu at coppice.org (Steve Underwood) Date: Sat, 07 Feb 2009 00:02:21 +0800 Subject: [Freeswitch-users] Transcoding G723 In-Reply-To: <35b355e90902060716v2dff2586t253a69c97ab1adac@mail.gmail.com> References: <2FEEE7F7-DCE7-4919-A0FD-A0042BBB7EEE@freeswitch.org> <35b355e90902051226g12756d2fsf85fb996ec1ed188@mail.gmail.com> <11C3C72A-247D-4F8C-B106-52D49949D026@freeswitch.org> <1b46b4e80902051252o1b5a5c75l9f8d242cbaeec552@mail.gmail.com> <35b355e90902051301r4c2c53b6tb958774f4754e60f@mail.gmail.com> <35b355e90902060716v2dff2586t253a69c97ab1adac@mail.gmail.com> Message-ID: <498C5F0D.9040409@coppice.org> Shelby Ramsey wrote: > Thanks Moises. It looks like good work. When is Sangoma coming out > with a similar product ... Doug told me it was in the works, then not > in the works, then back in the works ... > > The problem is this particular card is PCI only and it will only do > 120 channels .... If I were them, I'd think long and hard about such a product, and in the end probably not do it. Its the only realistic way to do G.723.1, but the market for that is not so big. For the much larger number wanting G.729, the main CPUs keep getting faster. There are 6 core Xeons now, and there are supposed to be 8 cores by the end of 2009. You can have 4 of those in a 1U chassis. Their speed is just going to keep running ahead of a card. The only real DSP that needs to be offloaded is EC, because a large number of channels of that is probably going to challenge the main CPUs for some time to come. Steve From anthony.minessale at gmail.com Fri Feb 6 08:02:41 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 6 Feb 2009 10:02:41 -0600 Subject: [Freeswitch-users] mod_openzap stops working after some calls In-Reply-To: <498C2DC3.40701@ewetel.de> References: <498C2DC3.40701@ewetel.de> Message-ID: <191c3a030902060802r424f8a22tb4887dff7c6aead5@mail.gmail.com> I think we have some trouble surviving issues. So when everything is ok we do fine but if something goes wrong we don't recover. We are still missing state timers in the q931. maybe you can use your new pcap thing to see what goes wrong =D On Fri, Feb 6, 2009 at 6:32 AM, Helmut Kuper wrote: > Hello, > > since yesterday I do a real life test with FS in a 40 sip extension > environment with TDM connection via wanpipe/sangoma A104d. > > I detected two times the problem, that openzap stops working, while SIP > calls worked. Only restarting helped. Maybe reloading of mod_openzap > helps as well, but I didn't tested that, yet. I filtered FS logfile for > hints of that problem and I found a growing number of this line: > > 2009-02-06 11:09:51 [INFO] ozmod_isdn.c:706 zap_isdn_931_34() Duplicate > SETUP message(?) for Channel 1:21 ~ 1:21 in state DOWN [ignoring] > > Befor restart I saw such a line for *each* TDM channel and no one was > able to dial out. mod_openzap handled around 40 outgoing calls until > last restart . > > Since last restart FS runs good for 2 hours now without any duplicate > SETUP in log ... > > Has anybody similar problems? > > > regards > Helmut > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090206/e4d49369/attachment-0002.html From nik.middleton at noblesolutions.co.uk Fri Feb 6 08:09:45 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Fri, 6 Feb 2009 16:09:45 -0000 Subject: [Freeswitch-users] Call accounting - CDR's Message-ID: Hi Guys I'm looking for some pointers on how to collect CDR's and store in mysql. Is there anything built in yet? I can rate the calls as a batch process, I simply need the call data. Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090206/fd9f1ada/attachment-0002.html From freeswitch-users at lists.rupa.com Fri Feb 6 08:27:17 2009 From: freeswitch-users at lists.rupa.com (Rupa Schomaker (lists)) Date: Fri, 06 Feb 2009 10:27:17 -0600 Subject: [Freeswitch-users] Call accounting - CDR's In-Reply-To: References: Message-ID: <498C64E5.3040103@lists.rupa.com> Use the script in scripts/contrib/wasim/ as a starting point. Basically, you log to csv files, and then the script periodically picks them up and loads to your DB. This is using mysql as an example, but you can do the same with postgres as well. If you need realtime inserts, then use mod_cdr_xml and have those post to a script on a webserver that parses the xml and inserts into appropriate tables. This is what I use along with a rails app. Remember that if you do real time, you also need to periodically scrape the error directory and load those (mod_cdr_xml will save to error if it can't successfully post to your script). On 2/6/2009 10:09 AM, Nik Middleton wrote: > Hi Guys > > > > I?m looking for some pointers on how to collect CDR?s and store in > mysql. Is there anything built in yet? > > > > I can rate the calls as a batch process, I simply need the call data. > > > > Regards > From sicfslist at gmail.com Fri Feb 6 08:28:54 2009 From: sicfslist at gmail.com (Shelby Ramsey) Date: Fri, 6 Feb 2009 10:28:54 -0600 Subject: [Freeswitch-users] Transcoding G723 In-Reply-To: <498C5F0D.9040409@coppice.org> References: <2FEEE7F7-DCE7-4919-A0FD-A0042BBB7EEE@freeswitch.org> <35b355e90902051226g12756d2fsf85fb996ec1ed188@mail.gmail.com> <11C3C72A-247D-4F8C-B106-52D49949D026@freeswitch.org> <1b46b4e80902051252o1b5a5c75l9f8d242cbaeec552@mail.gmail.com> <35b355e90902051301r4c2c53b6tb958774f4754e60f@mail.gmail.com> <35b355e90902060716v2dff2586t253a69c97ab1adac@mail.gmail.com> <498C5F0D.9040409@coppice.org> Message-ID: <35b355e90902060828g2552e05i882f8a8f1a05d37c@mail.gmail.com> Steve, You definitely have a better grasp on this topic than me. But I think it's a tough sell on the host based processing ... when you look at products like what audio codes can do on a card (3 DS-3's worth of transcoding) ... but I have had a couple of soft switch vendors claim though that they could do 2,000 calls per host (but I seriously doubt it). I agree that producing a card that does 120 channels is pretty worthless ... but having something that could do say a 1000 would be very helpful. G723 is a pretty big deal internationally ... I'm even seeing crazy requests like AMR from folks trying to originate VoIP off of mobile devices in Europe. But I agree just being able to g729 --> ulaw or ulaw --> g729 would be a great first step (host based or otherwise). SDR -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090206/2ced76a5/attachment-0002.html From sicfslist at gmail.com Fri Feb 6 08:32:36 2009 From: sicfslist at gmail.com (Shelby Ramsey) Date: Fri, 6 Feb 2009 10:32:36 -0600 Subject: [Freeswitch-users] Call accounting - CDR's In-Reply-To: References: Message-ID: <35b355e90902060832r4bb1b70y9ca850726ae1510e@mail.gmail.com> Nik, There are a bunch of ways to do this ... mod_xml_cdr posts to a url then you can parse and dump ... or you can use mod_cdr_csv which allows you to dictate exactly what you want to collect and then parse the file and dump into mysql. There are also a couple of examples here --> http://wiki.freeswitch.org/wiki/Mod_cdr for hacking up the cdr_csv.conf and making it do what you want. SDR -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090206/85911ac0/attachment-0002.html From nik.middleton at noblesolutions.co.uk Fri Feb 6 08:59:09 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Fri, 6 Feb 2009 16:59:09 -0000 Subject: [Freeswitch-users] Call accounting - CDR's In-Reply-To: <498C64E5.3040103@lists.rupa.com> References: <498C64E5.3040103@lists.rupa.com> Message-ID: Thanks, I was confused because I saw that mod cdr had been dropped. Due anticipated call volumes, batch processing is ideal, it keeps any MySql load issues away from FS Regards, -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rupa Schomaker (lists) Sent: 06 February 2009 16:27 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Call accounting - CDR's Use the script in scripts/contrib/wasim/ as a starting point. Basically, you log to csv files, and then the script periodically picks them up and loads to your DB. This is using mysql as an example, but you can do the same with postgres as well. If you need realtime inserts, then use mod_cdr_xml and have those post to a script on a webserver that parses the xml and inserts into appropriate tables. This is what I use along with a rails app. Remember that if you do real time, you also need to periodically scrape the error directory and load those (mod_cdr_xml will save to error if it can't successfully post to your script). On 2/6/2009 10:09 AM, Nik Middleton wrote: > Hi Guys > > > > I'm looking for some pointers on how to collect CDR's and store in > mysql. Is there anything built in yet? > > > > I can rate the calls as a batch process, I simply need the call data. > > > > Regards > _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From ajlong at worldlink.net Fri Feb 6 09:30:34 2009 From: ajlong at worldlink.net (Adam Long) Date: Fri, 6 Feb 2009 12:30:34 -0500 Subject: [Freeswitch-users] Call accounting - CDR's In-Reply-To: <498C64E5.3040103@lists.rupa.com> References: <498C64E5.3040103@lists.rupa.com> Message-ID: <030001c98880$9e660f90$db322eb0$@net> Is mod_cdr_xml asynchronous ... by that I mean .. if there is a communication failure how does this effect load on the system or the call for that matter... it wouldn?t hold up the progress or delay the turn up of the call or anything would it? I understand it would write to the error directory (but my thoughts are what is the impact of this beyond the obvious IO hit) I guess a good question is what is more load intensive??? 1.) mod_cdr_csv (with batch script that loads into DB somewhere) 2.) mod_cdr_xml (posting to lighttpd on remote host inserting into DB) I'm thinking about this for a system that would be handling in excess of 200-300 call setups per second. -Adam -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rupa Schomaker (lists) Sent: Friday, February 06, 2009 11:27 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Call accounting - CDR's Use the script in scripts/contrib/wasim/ as a starting point. Basically, you log to csv files, and then the script periodically picks them up and loads to your DB. This is using mysql as an example, but you can do the same with postgres as well. If you need realtime inserts, then use mod_cdr_xml and have those post to a script on a webserver that parses the xml and inserts into appropriate tables. This is what I use along with a rails app. Remember that if you do real time, you also need to periodically scrape the error directory and load those (mod_cdr_xml will save to error if it can't successfully post to your script). On 2/6/2009 10:09 AM, Nik Middleton wrote: > Hi Guys > > > > I?m looking for some pointers on how to collect CDR?s and store in > mysql. Is there anything built in yet? > > > > I can rate the calls as a batch process, I simply need the call data. > > > > Regards > _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From krice at freeswitch.org Fri Feb 6 09:40:16 2009 From: krice at freeswitch.org (Ken Rice) Date: Fri, 06 Feb 2009 11:40:16 -0600 Subject: [Freeswitch-users] Call accounting - CDR's In-Reply-To: <030001c98880$9e660f90$db322eb0$@net> Message-ID: Mod_xml_cdr will drop a file to the file system on failure to post. You can also leverage this drop a file to the file system and run a CDR processor locally. We handle call rates in the 500+ range using the local file system as a caching mechanism and a simple PHP script to rate the CDRs and load them into a pgsql db. > From: Adam Long > Reply-To: > Date: Fri, 6 Feb 2009 12:30:34 -0500 > To: > Subject: Re: [Freeswitch-users] Call accounting - CDR's > > Is mod_cdr_xml asynchronous ... by that I mean .. if there is a communication > failure how does this effect load on the system or the call for that > matter... it wouldn?t hold up the progress or delay the turn up of the call or > anything would it? I understand it would write to the error directory (but my > thoughts are what is the impact of this beyond the obvious IO hit) I guess a > good question is what is more load intensive??? 1.) mod_cdr_csv (with batch > script that loads into DB somewhere) 2.) mod_cdr_xml (posting to lighttpd on > remote host inserting into DB) I'm thinking about this for a system that > would be handling in excess of 200-300 call setups per > second. -Adam -----Original Message----- From: > freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rupa > Schomaker (lists) Sent: Friday, February 06, 2009 11:27 AM To: > freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Call > accounting - CDR's Use the script in scripts/contrib/wasim/ as a starting > point. Basically, you log to csv files, and then the script periodically > picks them up and loads to your DB. This is using mysql as an example, > but you can do the same with postgres as well. If you need realtime inserts, > then use mod_cdr_xml and have those post to a script on a webserver that > parses the xml and inserts into appropriate tables. This is what I use along > with a rails app. Remember that if you do real time, you also need to > periodically scrape the error directory and load those (mod_cdr_xml will save > to error if it can't successfully post to your script). On 2/6/2009 10:09 AM, > Nik Middleton wrote: > Hi Guys > > > > I?m looking for some pointers on > how to collect CDR?s and store in > mysql. Is there anything built in yet? > > > > > I can rate the calls as a batch process, I simply need the call > data. > > > > Regards > > _______________________________________________ Freeswitch-users mailing > list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman > /listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/opt > ions/freeswitch-users http://www.freeswitch.org ____________________________ > ___________________ Freeswitch-users mailing > list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman > /listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/opt > ions/freeswitch-users http://www.freeswitch.org From freeswitch-users at digitaldan.com Fri Feb 6 10:59:12 2009 From: freeswitch-users at digitaldan.com (Dan) Date: Fri, 6 Feb 2009 11:59:12 -0700 (MST) Subject: [Freeswitch-users] mod_shout delay in trunk Message-ID: <11000168.01233946752072.JavaMail.root@zimbra> Hi, With the 1.0.2 release i was able to to stream a call using mod_shout to an icecast server with only a 1 or two second delay to clients. With the current trunk that delay is now 8 to 10 seconds. I thought it might have been a change to mod_shout.c. I tried tweaking a few outbound buffer sizes with no luck so I just copied the 1.0.2 version of mod_shout.c over, compiled and reinstalled the module, restarted fs and still the delay is 8 to 10 seconds. I'm a little stumped. I currently have both versions installed (trunk and 1.0.2) for testing. Both are streaming to the same icecast server. My current svn revision is 11669, the calls are coming in via sip using g.711 ulaw and it looks like lame/mod_shout is streaming it as a 16kbs, 8khz mono mp3 stream. I'm using a flash/flex applet I wrote to consume the icecast stream, although I have used totem to listen to the stream as well. Any thoughts -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090206/40be49c6/attachment-0002.html From brian at freeswitch.org Fri Feb 6 11:11:54 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 6 Feb 2009 13:11:54 -0600 Subject: [Freeswitch-users] mod_shout delay in trunk In-Reply-To: <11000168.01233946752072.JavaMail.root@zimbra> References: <11000168.01233946752072.JavaMail.root@zimbra> Message-ID: Yes this will be normal due to buffering. Have you tested svn trunk? /b On Feb 6, 2009, at 12:59 PM, Dan wrote: > Hi, > > With the 1.0.2 release i was able to to stream a call using > mod_shout to an icecast server with only a 1 or two second delay to > clients. With the current trunk that delay is now 8 to 10 seconds. > I thought it might have been a change to mod_shout.c. I tried > tweaking a few outbound buffer sizes with no luck so I just copied > the 1.0.2 version of mod_shout.c over, compiled and reinstalled the > module, restarted fs and still the delay is 8 to 10 seconds. I'm a > little stumped. I currently have both versions installed (trunk > and 1.0.2) for testing. Both are streaming to the same icecast > server. > > My current svn revision is 11669, the calls are coming in via sip > using g.711 ulaw and it looks like lame/mod_shout is streaming it as > a 16kbs, 8khz mono mp3 stream. I'm using a flash/flex applet I > wrote to consume the icecast stream, although I have used totem to > listen to the stream as well. > > Any thoughts -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090206/56093979/attachment-0002.html From mitul at enterux.com Fri Feb 6 11:27:39 2009 From: mitul at enterux.com (Mitul Limbani) Date: Sat, 7 Feb 2009 00:57:39 +0530 Subject: [Freeswitch-users] Transcoding G723 In-Reply-To: <35b355e90902060828g2552e05i882f8a8f1a05d37c@mail.gmail.com> References: <2FEEE7F7-DCE7-4919-A0FD-A0042BBB7EEE@freeswitch.org> <35b355e90902051226g12756d2fsf85fb996ec1ed188@mail.gmail.com> <11C3C72A-247D-4F8C-B106-52D49949D026@freeswitch.org> <1b46b4e80902051252o1b5a5c75l9f8d242cbaeec552@mail.gmail.com> <35b355e90902051301r4c2c53b6tb958774f4754e60f@mail.gmail.com> <35b355e90902060716v2dff2586t253a69c97ab1adac@mail.gmail.com> <498C5F0D.9040409@coppice.org> <35b355e90902060828g2552e05i882f8a8f1a05d37c@mail.gmail.com> Message-ID: <26923699-F29A-42AB-BACE-11921955F0C2@enterux.com> Hi, I came across a company who was selling hardware which could do like 5000 G729 conversion simultanoeusly, I was like this sounds cool, they have support for asterisk, I haven't enquirer yet how they do this, but anyone wishes to buy it cost ?15000/year for the hrdware + support Any one on the list requires such high throughput calls can connect with me so that we can really test if it can work with freeswitch. Regards, Mitul Limbani, Founder & CEO, Enterux Solutions Pvt Ltd, The Enterprise Linux Company(r), http://www.enterux.com/ On 06-Feb-09, at 21:58, Shelby Ramsey wrote: > Steve, > > You definitely have a better grasp on this topic than me. But I > think it's a tough sell on the host based processing ... when you > look at products like what audio codes can do on a card (3 DS-3's > worth of transcoding) ... but I have had a couple of soft switch > vendors claim though that they could do 2,000 calls per host (but I > seriously doubt it). > > I agree that producing a card that does 120 channels is pretty > worthless ... but having something that could do say a 1000 would be > very helpful. > > G723 is a pretty big deal internationally ... I'm even seeing crazy > requests like AMR from folks trying to originate VoIP off of mobile > devices in Europe. > > But I agree just being able to g729 --> ulaw or ulaw --> g729 would > be a great first step (host based or otherwise). > > SDR > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Fri Feb 6 11:40:53 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 6 Feb 2009 13:40:53 -0600 Subject: [Freeswitch-users] Transcoding G723 In-Reply-To: <26923699-F29A-42AB-BACE-11921955F0C2@enterux.com> References: <2FEEE7F7-DCE7-4919-A0FD-A0042BBB7EEE@freeswitch.org> <35b355e90902051226g12756d2fsf85fb996ec1ed188@mail.gmail.com> <11C3C72A-247D-4F8C-B106-52D49949D026@freeswitch.org> <1b46b4e80902051252o1b5a5c75l9f8d242cbaeec552@mail.gmail.com> <35b355e90902051301r4c2c53b6tb958774f4754e60f@mail.gmail.com> <35b355e90902060716v2dff2586t253a69c97ab1adac@mail.gmail.com> <498C5F0D.9040409@coppice.org> <35b355e90902060828g2552e05i882f8a8f1a05d37c@mail.gmail.com> <26923699-F29A-42AB-BACE-11921955F0C2@enterux.com> Message-ID: PER YEAR? ARE THEY DAFT? /b On Feb 6, 2009, at 1:27 PM, Mitul Limbani wrote: > it cost ?15000/year for the hrdware + support From freeswitch-users at digitaldan.com Fri Feb 6 11:43:17 2009 From: freeswitch-users at digitaldan.com (freeswitch-users at digitaldan.com) Date: Fri, 6 Feb 2009 12:43:17 -0700 (MST) Subject: [Freeswitch-users] mod_shout delay in trunk In-Reply-To: <1047580.31233949312392.JavaMail.root@zimbra> Message-ID: <17591403.51233949397584.JavaMail.root@zimbra> I have, do you know what would have changed between 1.0.2 and trunk that would cause the buffer to change? Also if its not in mod_shout.c (which I copied from 1.0.2 to trunk for testing with no luck), where else would fs be buffering? One thing I have noticed is that in 1.0.2 as soon as the dial plan hits my record statement I see mod_shout logging that it has connected to the icecast server, in trunk it takes about 5 seconds to see the same log mesage. Below is my current svn info Path: . URL: http://svn.freeswitch.org/svn/freeswitch/trunk Repository Root: http://svn.freeswitch.org/svn Repository UUID: d0543943-73ff-0310-b7d9-9358b9ac24b2 Revision: 11669 Node Kind: directory Schedule: normal Last Changed Author: brian Last Changed Rev: 11669 Last Changed Date: 2009-02-06 11:29:51 -0700 (Fri, 06 Feb 2009) ----- Original Message ----- From: "Brian West" To: freeswitch-users at lists.freeswitch.org Sent: Friday, February 6, 2009 12:11:54 PM GMT -07:00 US/Canada Mountain Subject: Re: [Freeswitch-users] mod_shout delay in trunk Yes this will be normal due to buffering. Have you tested svn trunk? /b On Feb 6, 2009, at 12:59 PM, Dan wrote: Hi, With the 1.0.2 release i was able to to stream a call using mod_shout to an icecast server with only a 1 or two second delay to clients. With the current trunk that delay is now 8 to 10 seconds. I thought it might have been a change to mod_shout.c. I tried tweaking a few outbound buffer sizes with no luck so I just copied the 1.0.2 version of mod_shout.c over, compiled and reinstalled the module, restarted fs and still the delay is 8 to 10 seconds. I'm a little stumped. I currently have both versions installed (trunk and 1.0.2) for testing. Both are streaming to the same icecast server. My current svn revision is 11669, the calls are coming in via sip using g.711 ulaw and it looks like lame/mod_shout is streaming it as a 16kbs, 8khz mono mp3 stream. I'm using a flash/flex applet I wrote to consume the icecast stream, although I have used totem to listen to the stream as well. Any thoughts _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090206/daf345e3/attachment-0002.html From brian at freeswitch.org Fri Feb 6 11:47:53 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 6 Feb 2009 13:47:53 -0600 Subject: [Freeswitch-users] mod_shout delay in trunk In-Reply-To: <17591403.51233949397584.JavaMail.root@zimbra> References: <17591403.51233949397584.JavaMail.root@zimbra> Message-ID: <824A1B7F-A0CE-4ED9-AEF2-97037E8F1E59@freeswitch.org> Let me clarify.. yes this is normal file buffering was added so we wouldn't thrash your hard drive with tiny bits of data when recording calls so now it buffers and writes larger chunks to disk. This is why you have this delay which is 100% normal.... is realtime a critical thing? It is shout cast so you know it doesn't have to be realtime.. in fact some clients will buffer a little bit anyway and add to it. /b On Feb 6, 2009, at 1:43 PM, freeswitch-users at digitaldan.com wrote: > I have, do you know what would have changed between 1.0.2 and trunk > that would cause the buffer to change? Also if its not in > mod_shout.c (which I copied from 1.0.2 to trunk for testing with no > luck), where else would fs be buffering? One thing I have noticed > is that in 1.0.2 as soon as the dial plan hits my record statement I > see mod_shout logging that it has connected to the icecast server, > in trunk it takes about 5 seconds to see the same log mesage. Below > is my current svn info > Path: . -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090206/eabd461d/attachment-0002.html From freeswitch-users at digitaldan.com Fri Feb 6 12:01:23 2009 From: freeswitch-users at digitaldan.com (freeswitch-users at digitaldan.com) Date: Fri, 6 Feb 2009 13:01:23 -0700 (MST) Subject: [Freeswitch-users] mod_shout delay in trunk In-Reply-To: <16242086.81233950401907.JavaMail.root@zimbra> Message-ID: <8732241.101233950483385.JavaMail.root@zimbra> For me it is. For what I'm using it for I can tolerate around a second or two delay. I have the icecast server setup to only buffer 1K for their on-connect burst as well as my flash/flex player to only buffer 1k (yes I might as well not buffer at all, which I may end up doing). In 1.0.2 this worked very well. Is this buffer configurable? If not, where is it being set? Thanks Dan- ----- Original Message ----- From: "Brian West" To: freeswitch-users at lists.freeswitch.org Sent: Friday, February 6, 2009 12:47:53 PM GMT -07:00 US/Canada Mountain Subject: Re: [Freeswitch-users] mod_shout delay in trunk Let me clarify.. yes this is normal file buffering was added so we wouldn't thrash your hard drive with tiny bits of data when recording calls so now it buffers and writes larger chunks to disk. This is why you have this delay which is 100% normal.... is realtime a critical thing? It is shout cast so you know it doesn't have to be realtime.. in fact some clients will buffer a little bit anyway and add to it. /b On Feb 6, 2009, at 1:43 PM, freeswitch-users at digitaldan.com wrote: I have, do you know what would have changed between 1.0.2 and trunk that would cause the buffer to change? Also if its not in mod_shout.c (which I copied from 1.0.2 to trunk for testing with no luck), where else would fs be buffering? One thing I have noticed is that in 1.0.2 as soon as the dial plan hits my record statement I see mod_shout logging that it has connected to the icecast server, in trunk it takes about 5 seconds to see the same log mesage. Below is my current svn info Path: . _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090206/217ca9cf/attachment-0002.html From anthony.minessale at gmail.com Fri Feb 6 12:07:44 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 6 Feb 2009 14:07:44 -0600 Subject: [Freeswitch-users] mod_shout delay in trunk In-Reply-To: <8732241.101233950483385.JavaMail.root@zimbra> References: <16242086.81233950401907.JavaMail.root@zimbra> <8732241.101233950483385.JavaMail.root@zimbra> Message-ID: <191c3a030902061207x24bb8f05h901f1f7ad208926c@mail.gmail.com> edit switch_ivr_play_say.c line 423 comment the line out and recompile. Tell me if it helps you and i will consider making it configurable. On Fri, Feb 6, 2009 at 2:01 PM, wrote: > For me it is. For what I'm using it for I can tolerate around a second or > two delay. I have the icecast server setup to only buffer 1K for their > on-connect burst as well as my flash/flex player to only buffer 1k (yes I > might as well not buffer at all, which I may end up doing). In 1.0.2 this > worked very well. Is this buffer configurable? If not, where is it being > set? > > Thanks > Dan- > ----- Original Message ----- > From: "Brian West" > To: freeswitch-users at lists.freeswitch.org > Sent: Friday, February 6, 2009 12:47:53 PM GMT -07:00 US/Canada Mountain > Subject: Re: [Freeswitch-users] mod_shout delay in trunk > > Let me clarify.. yes this is normal file buffering was added so we wouldn't > thrash your hard drive with tiny bits of data when recording calls so now it > buffers and writes larger chunks to disk. This is why you have this delay > which is 100% normal.... is realtime a critical thing? It is shout cast so > you know it doesn't have to be realtime.. in fact some clients will buffer a > little bit anyway and add to it. > /b > > On Feb 6, 2009, at 1:43 PM, freeswitch-users at digitaldan.com wrote: > > I have, do you know what would have changed between 1.0.2 and trunk that > would cause the buffer to change? Also if its not in mod_shout.c (which I > copied from 1.0.2 to trunk for testing with no luck), where else would fs be > buffering? One thing I have noticed is that in 1.0.2 as soon as the dial > plan hits my record statement I see mod_shout logging that it has connected > to the icecast server, in trunk it takes about 5 seconds to see the same log > mesage. Below is my current svn info > Path: . > > > > _______________________________________________ Freeswitch-users mailing > list Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090206/562d328c/attachment-0002.html From kristian.kielhofner at gmail.com Fri Feb 6 12:20:25 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Fri, 6 Feb 2009 15:20:25 -0500 Subject: [Freeswitch-users] Compiling FreeSWITCH for AstLinux Message-ID: <2d9149cd0902061220i11b87fd9se253109d7a39249a@mail.gmail.com> Hey guys, I've finally gotten around to trying to compile FreeSWITCH for AstLinux. Here is the branch I've created for it: http://astlinux.svn.sourceforge.net/viewvc/astlinux/branches/astlinux-freeswitch/package/freeswitch/ Right now it is bombing trying to run configure for libs/sqlite, as shown in the build log here: http://astbuild.star2star.com/astlinux-freeswitch-build.log Here is the config.log for sqlite: http://astbuild.star2star.com/sqlite-config.log I'll continue to dig into this but in the meantime I thought I'd get some extra eyeballs on it... Thanks! P.S. - Yes, yes I know "AstLinux" isn't the best name for a distro with FreeSWITCH. Depending on my success here I have some other ideas... -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From darren at aleph-com.net Fri Feb 6 12:29:59 2009 From: darren at aleph-com.net (Darren Wiebe) Date: Fri, 06 Feb 2009 13:29:59 -0700 Subject: [Freeswitch-users] Call accounting - CDR's In-Reply-To: References: Message-ID: <498C9DC7.6030104@aleph-com.net> Nik Middleton wrote: > > Hi Guys > > I?m looking for some pointers on how to collect CDR?s and store in > mysql. Is there anything built in yet? > > I can rate the calls as a batch process, I simply need the call data. > > Regards > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > I use mod_cdr_csv and then run a process every minute to import them into the mysql database. It seems to work well. -- Darren Wiebe darren at aleph-com.net Aleph Communications www.aleph-com.net From mgg at giagnocavo.net Fri Feb 6 12:30:38 2009 From: mgg at giagnocavo.net (Michael Giagnocavo) Date: Fri, 6 Feb 2009 15:30:38 -0500 Subject: [Freeswitch-users] Transcoding G723 In-Reply-To: <26923699-F29A-42AB-BACE-11921955F0C2@enterux.com> References: <2FEEE7F7-DCE7-4919-A0FD-A0042BBB7EEE@freeswitch.org> <35b355e90902051226g12756d2fsf85fb996ec1ed188@mail.gmail.com> <11C3C72A-247D-4F8C-B106-52D49949D026@freeswitch.org> <1b46b4e80902051252o1b5a5c75l9f8d242cbaeec552@mail.gmail.com> <35b355e90902051301r4c2c53b6tb958774f4754e60f@mail.gmail.com> <35b355e90902060716v2dff2586t253a69c97ab1adac@mail.gmail.com> <498C5F0D.9040409@coppice.org> <35b355e90902060828g2552e05i882f8a8f1a05d37c@mail.gmail.com> <26923699-F29A-42AB-BACE-11921955F0C2@enterux.com> Message-ID: <6E8D2069C08AA84A83D336E996AE4C6702362F6BCE@mse17be1.mse17.exchange.ms> $22K would buy quite a few machines with many core Xeons. I just don't see how it'd be effective at that price. Not to mention a yearly figure. The only thing to take into consideration would be the G729 licenses. But in bulk, the price should be pretty effective, even figuring in hardware. (Not to mention what things like Larrabee will mean for encoding -- 32 1.5GHz "Pentium 4 x64" cores, each with 4 threads?) And... if you're running Asterisk, um, isn't 5000 channels a wee bit over what you can handle anyways? ;) -Michael -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mitul Limbani Sent: Friday, February 06, 2009 12:28 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Transcoding G723 Hi, I came across a company who was selling hardware which could do like 5000 G729 conversion simultanoeusly, I was like this sounds cool, they have support for asterisk, I haven't enquirer yet how they do this, but anyone wishes to buy it cost ?15000/year for the hrdware + support Any one on the list requires such high throughput calls can connect with me so that we can really test if it can work with freeswitch. Regards, Mitul Limbani, Founder & CEO, Enterux Solutions Pvt Ltd, The Enterprise Linux Company(r), http://www.enterux.com/ On 06-Feb-09, at 21:58, Shelby Ramsey wrote: > Steve, > > You definitely have a better grasp on this topic than me. But I > think it's a tough sell on the host based processing ... when you > look at products like what audio codes can do on a card (3 DS-3's > worth of transcoding) ... but I have had a couple of soft switch > vendors claim though that they could do 2,000 calls per host (but I > seriously doubt it). > > I agree that producing a card that does 120 channels is pretty > worthless ... but having something that could do say a 1000 would be > very helpful. > > G723 is a pretty big deal internationally ... I'm even seeing crazy > requests like AMR from folks trying to originate VoIP off of mobile > devices in Europe. > > But I agree just being able to g729 --> ulaw or ulaw --> g729 would > be a great first step (host based or otherwise). > > SDR > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From freeswitch-users at digitaldan.com Fri Feb 6 12:36:14 2009 From: freeswitch-users at digitaldan.com (freeswitch-users at digitaldan.com) Date: Fri, 6 Feb 2009 13:36:14 -0700 (MST) Subject: [Freeswitch-users] mod_shout delay in trunk In-Reply-To: <15830211.131233952452225.JavaMail.root@zimbra> Message-ID: <28165623.151233952574612.JavaMail.root@zimbra> That worked great! I wanted to say just how awesome Freeswitch is, I have been doing voip related development with SIP since 2000 and this is by far the most well designed piece of voip software I have used or developed on. I currently have a homegrown sip server built on the NIST sip stack with Sun's JMF libraries for RTP processing. 95% of the code and complexity is handling the SIP and RTP sessions, the other 5% is the final application logic and what is most important to me . By letting freeswitch do whats its good at (call routing, sip and media handling) it allows me to focus on what I'm good at (what should we do with those streams, like record them). I have been bragging about this project to anybody who will listen! Dan- ----- Original Message ----- From: "Anthony Minessale" To: freeswitch-users at lists.freeswitch.org Sent: Friday, February 6, 2009 1:07:44 PM GMT -07:00 US/Canada Mountain Subject: Re: [Freeswitch-users] mod_shout delay in trunk edit switch_ivr_play_say.c line 423 comment the line out and recompile. Tell me if it helps you and i will consider making it configurable. On Fri, Feb 6, 2009 at 2:01 PM, < freeswitch-users at digitaldan.com > wrote: For me it is. For what I'm using it for I can tolerate around a second or two delay. I have the icecast server setup to only buffer 1K for their on-connect burst as well as my flash/flex player to only buffer 1k (yes I might as well not buffer at all, which I may end up doing). In 1.0.2 this worked very well. Is this buffer configurable? If not, where is it being set? Thanks Dan- ----- Original Message ----- From: "Brian West" < brian at freeswitch.org > To: freeswitch-users at lists.freeswitch.org Sent: Friday, February 6, 2009 12:47:53 PM GMT -07:00 US/Canada Mountain Subject: Re: [Freeswitch-users] mod_shout delay in trunk Let me clarify.. yes this is normal file buffering was added so we wouldn't thrash your hard drive with tiny bits of data when recording calls so now it buffers and writes larger chunks to disk. This is why you have this delay which is 100% normal.... is realtime a critical thing? It is shout cast so you know it doesn't have to be realtime.. in fact some clients will buffer a little bit anyway and add to it. /b On Feb 6, 2009, at 1:43 PM, freeswitch-users at digitaldan.com wrote: I have, do you know what would have changed between 1.0.2 and trunk that would cause the buffer to change? Also if its not in mod_shout.c (which I copied from 1.0.2 to trunk for testing with no luck), where else would fs be buffering? One thing I have noticed is that in 1.0.2 as soon as the dial plan hits my record statement I see mod_shout logging that it has connected to the icecast server, in trunk it takes about 5 seconds to see the same log mesage. Below is my current svn info Path: . _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/ PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090206/64231bc1/attachment-0002.html From wasim at convergence.pk Fri Feb 6 12:40:11 2009 From: wasim at convergence.pk (Wasim Baig) Date: Sat, 7 Feb 2009 01:40:11 +0500 Subject: [Freeswitch-users] Transcoding G723 In-Reply-To: <6E8D2069C08AA84A83D336E996AE4C6702362F6BCE@mse17be1.mse17.exchange.ms> References: <1b46b4e80902051252o1b5a5c75l9f8d242cbaeec552@mail.gmail.com> <35b355e90902051301r4c2c53b6tb958774f4754e60f@mail.gmail.com> <35b355e90902060716v2dff2586t253a69c97ab1adac@mail.gmail.com> <498C5F0D.9040409@coppice.org> <35b355e90902060828g2552e05i882f8a8f1a05d37c@mail.gmail.com> <26923699-F29A-42AB-BACE-11921955F0C2@enterux.com> <6E8D2069C08AA84A83D336E996AE4C6702362F6BCE@mse17be1.mse17.exchange.ms> Message-ID: On Sat, Feb 7, 2009 at 1:30 AM, Michael Giagnocavo wrote: $22K would buy quite a few machines with many core Xeons. I just don't see > how it'd be effective at that price. Not to mention a yearly figure. G729 is roughly 25 MIPS (encode+decode), coppice, please correct as necessary. A dual quad-core xeon 3 ghz should do roughly 1k calls, assuming nothing else is being run. > The only thing to take into consideration would be the G729 licenses. But > in bulk, the price should be pretty effective, even figuring in hardware. Or places where licensing isn't enforced, like offshore g729 farms :) > (Not to mention what things like Larrabee will mean for encoding -- 32 > 1.5GHz "Pentium 4 x64" cores, each with 4 threads?) 32 x 1.5 GHZ = 48 Ghz or roughly 2000 channels ... > And... if you're running Asterisk, um, isn't 5000 channels a wee bit over > what you can handle anyways? ;) Yeh, but not when you have FS :) which is why talks of this capacity on HMP are becoming more and more relevant ... -- wasim h. baig | principal consultant | convergence pk | +92 300 8508070 | peace be upon you ... -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090207/b9a20c5d/attachment-0002.html From nik.middleton at noblesolutions.co.uk Fri Feb 6 12:40:51 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Fri, 6 Feb 2009 20:40:51 -0000 Subject: [Freeswitch-users] Call accounting - CDR's In-Reply-To: <030001c98880$9e660f90$db322eb0$@net> References: <498C64E5.3040103@lists.rupa.com> <030001c98880$9e660f90$db322eb0$@net> Message-ID: What about using a radius server, would that be more resilient? Regards -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Adam Long Sent: 06 February 2009 17:31 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Call accounting - CDR's Is mod_cdr_xml asynchronous ... by that I mean .. if there is a communication failure how does this effect load on the system or the call for that matter... it wouldn't hold up the progress or delay the turn up of the call or anything would it? I understand it would write to the error directory (but my thoughts are what is the impact of this beyond the obvious IO hit) I guess a good question is what is more load intensive??? 1.) mod_cdr_csv (with batch script that loads into DB somewhere) 2.) mod_cdr_xml (posting to lighttpd on remote host inserting into DB) I'm thinking about this for a system that would be handling in excess of 200-300 call setups per second. -Adam -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rupa Schomaker (lists) Sent: Friday, February 06, 2009 11:27 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Call accounting - CDR's Use the script in scripts/contrib/wasim/ as a starting point. Basically, you log to csv files, and then the script periodically picks them up and loads to your DB. This is using mysql as an example, but you can do the same with postgres as well. If you need realtime inserts, then use mod_cdr_xml and have those post to a script on a webserver that parses the xml and inserts into appropriate tables. This is what I use along with a rails app. Remember that if you do real time, you also need to periodically scrape the error directory and load those (mod_cdr_xml will save to error if it can't successfully post to your script). On 2/6/2009 10:09 AM, Nik Middleton wrote: > Hi Guys > > > > I'm looking for some pointers on how to collect CDR's and store in > mysql. Is there anything built in yet? > > > > I can rate the calls as a batch process, I simply need the call data. > > > > Regards > _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From anthony.minessale at gmail.com Fri Feb 6 12:48:31 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 6 Feb 2009 14:48:31 -0600 Subject: [Freeswitch-users] mod_shout delay in trunk In-Reply-To: <28165623.151233952574612.JavaMail.root@zimbra> References: <15830211.131233952452225.JavaMail.root@zimbra> <28165623.151233952574612.JavaMail.root@zimbra> Message-ID: <191c3a030902061248r3d307647le5574c1004c6da24@mail.gmail.com> Thanks, We appreciate the positive feedback! if you revert the change I suggested and update i added a new variable enable_file_write_buffering=false set this variable on the channel before you start recording it with the set application or in the dialstring in {} on outbound calls and it should skip the buffering. Could you test it for me and confirm it works? Thank you On Fri, Feb 6, 2009 at 2:36 PM, wrote: > That worked great! > > I wanted to say just how awesome Freeswitch is, I have been doing voip > related development with SIP since 2000 and this is by far the most well > designed piece of voip software I have used or developed on. I currently > have a homegrown sip server built on the NIST sip stack with Sun's JMF > libraries for RTP processing. 95% of the code and complexity is handling > the SIP and RTP sessions, the other 5% is the final application logic and > what is most important to me. By letting freeswitch do whats its good at > (call routing, sip and media handling) it allows me to focus on what I'm > good at (what should we do with those streams, like record them). I have > been bragging about this project to anybody who will listen! > > Dan- > ----- Original Message ----- > From: "Anthony Minessale" > To: freeswitch-users at lists.freeswitch.org > Sent: Friday, February 6, 2009 1:07:44 PM GMT -07:00 US/Canada Mountain > Subject: Re: [Freeswitch-users] mod_shout delay in trunk > > edit switch_ivr_play_say.c line 423 > > comment the line out and recompile. > Tell me if it helps you and i will consider making it configurable. > > > On Fri, Feb 6, 2009 at 2:01 PM, wrote: > >> For me it is. For what I'm using it for I can tolerate around a second or >> two delay. I have the icecast server setup to only buffer 1K for their >> on-connect burst as well as my flash/flex player to only buffer 1k (yes I >> might as well not buffer at all, which I may end up doing). In 1.0.2 this >> worked very well. Is this buffer configurable? If not, where is it being >> set? >> >> Thanks >> Dan- >> ----- Original Message ----- >> From: "Brian West" >> To: freeswitch-users at lists.freeswitch.org >> Sent: Friday, February 6, 2009 12:47:53 PM GMT -07:00 US/Canada Mountain >> Subject: Re: [Freeswitch-users] mod_shout delay in trunk >> >> Let me clarify.. yes this is normal file buffering was added so we >> wouldn't thrash your hard drive with tiny bits of data when recording calls >> so now it buffers and writes larger chunks to disk. This is why you have >> this delay which is 100% normal.... is realtime a critical thing? It is >> shout cast so you know it doesn't have to be realtime.. in fact some clients >> will buffer a little bit anyway and add to it. >> /b >> >> On Feb 6, 2009, at 1:43 PM, freeswitch-users at digitaldan.com wrote: >> >> I have, do you know what would have changed between 1.0.2 and trunk that >> would cause the buffer to change? Also if its not in mod_shout.c (which I >> copied from 1.0.2 to trunk for testing with no luck), where else would fs be >> buffering? One thing I have noticed is that in 1.0.2 as soon as the dial >> plan hits my record statement I see mod_shout logging that it has connected >> to the icecast server, in trunk it takes about 5 seconds to see the same log >> mesage. Below is my current svn info >> Path: . >> >> >> >> _______________________________________________ Freeswitch-users mailing >> list Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ Freeswitch-users mailing > list Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090206/50050217/attachment-0002.html From nik.middleton at noblesolutions.co.uk Fri Feb 6 13:07:12 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Fri, 6 Feb 2009 21:07:12 -0000 Subject: [Freeswitch-users] Call accounting - CDR's In-Reply-To: References: <030001c98880$9e660f90$db322eb0$@net> Message-ID: So you're simply posting this file to a web server? How do you find the load on it at this rate of calls? BTW can anyone point me to resources discussing how to do this? (Using a web server to post data to a db) I've not this sort of thing before, and I'm not too sure what I should be goggling for Regards, -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice Sent: 06 February 2009 17:40 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Call accounting - CDR's Mod_xml_cdr will drop a file to the file system on failure to post. You can also leverage this drop a file to the file system and run a CDR processor locally. We handle call rates in the 500+ range using the local file system as a caching mechanism and a simple PHP script to rate the CDRs and load them into a pgsql db. > From: Adam Long > Reply-To: > Date: Fri, 6 Feb 2009 12:30:34 -0500 > To: > Subject: Re: [Freeswitch-users] Call accounting - CDR's > > Is mod_cdr_xml asynchronous ... by that I mean .. if there is a communication > failure how does this effect load on the system or the call for that > matter... it wouldn?t hold up the progress or delay the turn up of the call or > anything would it? I understand it would write to the error directory (but my > thoughts are what is the impact of this beyond the obvious IO hit) I guess a > good question is what is more load intensive??? 1.) mod_cdr_csv (with batch > script that loads into DB somewhere) 2.) mod_cdr_xml (posting to lighttpd on > remote host inserting into DB) I'm thinking about this for a system that > would be handling in excess of 200-300 call setups per > second. -Adam -----Original Message----- From: > freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rupa > Schomaker (lists) Sent: Friday, February 06, 2009 11:27 AM To: > freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Call > accounting - CDR's Use the script in scripts/contrib/wasim/ as a starting > point. Basically, you log to csv files, and then the script periodically > picks them up and loads to your DB. This is using mysql as an example, > but you can do the same with postgres as well. If you need realtime inserts, > then use mod_cdr_xml and have those post to a script on a webserver that > parses the xml and inserts into appropriate tables. This is what I use along > with a rails app. Remember that if you do real time, you also need to > periodically scrape the error directory and load those (mod_cdr_xml will save > to error if it can't successfully post to your script). On 2/6/2009 10:09 AM, > Nik Middleton wrote: > Hi Guys > > > > I?m looking for some pointers on > how to collect CDR?s and store in > mysql. Is there anything built in yet? > > > > > I can rate the calls as a batch process, I simply need the call > data. > > > > Regards > > _______________________________________________ Freeswitch-users mailing > list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman > /listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/opt > ions/freeswitch-users http://www.freeswitch.org ____________________________ > ___________________ Freeswitch-users mailing > list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman > /listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/opt > ions/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From mitul at enterux.com Fri Feb 6 15:02:13 2009 From: mitul at enterux.com (Mitul Limbani) Date: Fri, 6 Feb 2009 18:02:13 -0500 Subject: [Freeswitch-users] Transcoding G723 Message-ID: <49796.1233961333@enterux.com> BODY { font-family:Arial, Helvetica, sans-serif;font-size:12px; }Hello guys, Sorry it was my mistake, i re-read the entire proposal and it looks like their specialized 1U Hardware with custom CPU can handle close to 1500 simultaneous G729 encoding n transmission, also this hardware offers from any codec to any codec transcoding path. Also this box easily works over FreeSWITCH, Asterisk, RTPProxy, OpenSIPS are their claims. and their pricing are First year ?16,000 Second year ?13,000/annum recurring This is found slightly on higher side though, Thanks & Regards, Mitul Limbani, Founder & CEO, Enterux Solutions, The Enterprise Linux Company (TM), www.enterux.com +91-9820332422 On Sat 07/02/09 02:00 , Michael Giagnocavo mgg at giagnocavo.net sent: $22K would buy quite a few machines with many core Xeons. I just don't see how it'd be effective at that price. Not to mention a yearly figure. The only thing to take into consideration would be the G729 licenses. But in bulk, the price should be pretty effective, even figuring in hardware. (Not to mention what things like Larrabee will mean for encoding -- 32 1.5GHz "Pentium 4 x64" cores, each with 4 threads?) And... if you're running Asterisk, um, isn't 5000 channels a wee bit over what you can handle anyways? ;) -Michael -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org [1]] On Behalf Of Mitul Limbani Sent: Friday, February 06, 2009 12:28 PM To: freeswitch-users at lists.freeswitch.org [2] Subject: Re: [Freeswitch-users] Transcoding G723 Hi, I came across a company who was selling hardware which could do like 5000 G729 conversion simultanoeusly, I was like this sounds cool, they have support for asterisk, I haven't enquirer yet how they do this, but anyone wishes to buy it cost ?15000/year for the hrdware + support Any one on the list requires such high throughput calls can connect with me so that we can really test if it can work with freeswitch. Regards, Mitul Limbani, Founder & CEO, Enterux Solutions Pvt Ltd, The Enterprise Linux Company(r), http://www.enterux.com/ On 06-Feb-09, at 21:58, Shelby Ramsey wrote: > Steve, > > You definitely have a better grasp on this topic than me. But I > think it's a tough sell on the host based processing ... when you > look at products like what audio codes can do on a card (3 DS-3's > worth of transcoding) ... but I have had a couple of soft switch > vendors claim though that they could do 2,000 calls per host (but I > seriously doubt it). > > I agree that producing a card that does 120 channels is pretty > worthless ... but having something that could do say a 1000 would be > very helpful. > > G723 is a pretty big deal internationally ... I'm even seeing crazy > requests like AMR from folks trying to originate VoIP off of mobile > devices in Europe. > > But I agree just being able to g729 --> ulaw or ulaw --> g729 would > be a great first step (host based or otherwise). > > SDR > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org [4] > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org [5] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org [6] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ------------------------- Msg sent via Enterux Enterprise Email Server : http://www.enterux.com/ Links: ------ [1] mailto:freeswitch-users-bounces at lists.freeswitch.org [2] mailto:freeswitch-users at lists.freeswitch.org [3] mailto:sicfslist at gmail.com [4] mailto:Freeswitch-users at lists.freeswitch.org [5] mailto:Freeswitch-users at lists.freeswitch.org [6] mailto:Freeswitch-users at lists.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090206/cd736ff9/attachment-0002.html From freeswitch-users at digitaldan.com Fri Feb 6 13:13:06 2009 From: freeswitch-users at digitaldan.com (Dan) Date: Fri, 6 Feb 2009 14:13:06 -0700 (MST) Subject: [Freeswitch-users] mod_shout delay in trunk In-Reply-To: <191c3a030902061248r3d307647le5574c1004c6da24@mail.gmail.com> Message-ID: <15850979.181233954786201.JavaMail.root@zimbra> On line 424 I think it needs to be changed from if (!vval || !switch_true(vval)) { to if (!vval || switch_true(vval)) { Other wise it works, thanks! ----- Original Message ----- From: "Anthony Minessale" To: freeswitch-users at lists.freeswitch.org Sent: Friday, February 6, 2009 1:48:31 PM GMT -07:00 US/Canada Mountain Subject: Re: [Freeswitch-users] mod_shout delay in trunk Thanks, We appreciate the positive feedback! if you revert the change I suggested and update i added a new variable enable_file_write_buffering=false set this variable on the channel before you start recording it with the set application or in the dialstring in {} on outbound calls and it should skip the buffering. Could you test it for me and confirm it works? Thank you On Fri, Feb 6, 2009 at 2:36 PM, < freeswitch-users at digitaldan.com > wrote: That worked great! I wanted to say just how awesome Freeswitch is, I have been doing voip related development with SIP since 2000 and this is by far the most well designed piece of voip software I have used or developed on. I currently have a homegrown sip server built on the NIST sip stack with Sun's JMF libraries for RTP processing. 95% of the code and complexity is handling the SIP and RTP sessions, the other 5% is the final application logic and what is most important to me. By letting freeswitch do whats its good at (call routing, sip and media handling) it allows me to focus on what I'm good at (what should we do with those streams, like record them). I have been bragging about this project to anybody who will listen! Dan- ----- Original Message ----- From: "Anthony Minessale" < anthony.minessale at gmail.com > To: freeswitch-users at lists.freeswitch.org Sent: Friday, February 6, 2009 1:07:44 PM GMT -07:00 US/Canada Mountain Subject: Re: [Freeswitch-users] mod_shout delay in trunk edit switch_ivr_play_say.c line 423 comment the line out and recompile. Tell me if it helps you and i will consider making it configurable. On Fri, Feb 6, 2009 at 2:01 PM, < freeswitch-users at digitaldan.com > wrote: For me it is. For what I'm using it for I can tolerate around a second or two delay. I have the icecast server setup to only buffer 1K for their on-connect burst as well as my flash/flex player to only buffer 1k (yes I might as well not buffer at all, which I may end up doing). In 1.0.2 this worked very well. Is this buffer configurable? If not, where is it being set? Thanks Dan- ----- Original Message ----- From: "Brian West" < brian at freeswitch.org > To: freeswitch-users at lists.freeswitch.org Sent: Friday, February 6, 2009 12:47:53 PM GMT -07:00 US/Canada Mountain Subject: Re: [Freeswitch-users] mod_shout delay in trunk Let me clarify.. yes this is normal file buffering was added so we wouldn't thrash your hard drive with tiny bits of data when recording calls so now it buffers and writes larger chunks to disk. This is why you have this delay which is 100% normal.... is realtime a critical thing? It is shout cast so you know it doesn't have to be realtime.. in fact some clients will buffer a little bit anyway and add to it. /b On Feb 6, 2009, at 1:43 PM, freeswitch-users at digitaldan.com wrote: I have, do you know what would have changed between 1.0.2 and trunk that would cause the buffer to change? Also if its not in mod_shout.c (which I copied from 1.0.2 to trunk for testing with no luck), where else would fs be buffering? One thing I have noticed is that in 1.0.2 as soon as the dial plan hits my record statement I see mod_shout logging that it has connected to the icecast server, in trunk it takes about 5 seconds to see the same log mesage. Below is my current svn info Path: . _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/ PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/ PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090206/cf3646ae/attachment-0002.html From msc at freeswitch.org Fri Feb 6 13:16:28 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 6 Feb 2009 13:16:28 -0800 Subject: [Freeswitch-users] Compiling FreeSWITCH for AstLinux In-Reply-To: <2d9149cd0902061220i11b87fd9se253109d7a39249a@mail.gmail.com> References: <2d9149cd0902061220i11b87fd9se253109d7a39249a@mail.gmail.com> Message-ID: <87f2f3b90902061316h5fe6e8afw253c95a55ddf3aa0@mail.gmail.com> > P.S. - Yes, yes I know "AstLinux" isn't the best name for a distro > with FreeSWITCH. Depending on my success here I have some other > ideas... > How about KickAstLinux? ;) -MC From krice at suspicious.org Fri Feb 6 13:18:39 2009 From: krice at suspicious.org (Ken Rice) Date: Fri, 06 Feb 2009 15:18:39 -0600 Subject: [Freeswitch-users] Call accounting - CDR's In-Reply-To: Message-ID: This is built into mod_xml_cdr and should be covered on its wiki page... I personally don't post the records to a web server as I think that's too much over head at this load... I use a script to scrape the directory and process the CDRs... This gives me the FileSystem as a buffer, and allows me to get a ton of info out of the xml cdrs like custom channel variables etc K > From: Nik Middleton > Reply-To: > Date: Fri, 6 Feb 2009 21:07:12 -0000 > To: > Subject: Re: [Freeswitch-users] Call accounting - CDR's > > So you're simply posting this file to a web server? How do you find the load > on it at this rate of calls? > > BTW can anyone point me to resources discussing how to do this? (Using a web > server to post data to a db) I've not this sort of thing before, and I'm not > too sure what I should be goggling for > > > Regards, > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice > Sent: 06 February 2009 17:40 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Call accounting - CDR's > > Mod_xml_cdr will drop a file to the file system on failure to post. You can > also leverage this drop a file to the file system and run a CDR processor > locally. > > We handle call rates in the 500+ range using the local file system as a > caching mechanism and a simple PHP script to rate the CDRs and load them > into a pgsql db. > > > > >> From: Adam Long >> Reply-To: >> Date: Fri, 6 Feb 2009 12:30:34 -0500 >> To: >> Subject: Re: [Freeswitch-users] Call accounting - CDR's >> >> Is mod_cdr_xml asynchronous ... by that I mean .. if there is a communication >> failure how > does this effect load on the system or the call for that >> matter... > it wouldn?t hold up the progress or delay the turn up of the call or >> anything would it? > > I understand it would write to the error directory (but my >> thoughts are what is the impact of > this beyond the obvious IO hit) > > I guess a >> good question is what is more load intensive??? > > 1.) mod_cdr_csv (with batch >> script that loads into DB somewhere) > 2.) mod_cdr_xml (posting to lighttpd on >> remote host inserting into DB) > > I'm thinking about this for a system that >> would be handling in excess of 200-300 call setups > per >> second. > > -Adam > > -----Original Message----- > From: >> freeswitch-users-bounces at lists.freeswitch.org >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rupa >> Schomaker (lists) > Sent: Friday, February 06, 2009 11:27 AM > To: >> freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Call >> accounting - CDR's > > Use the script in scripts/contrib/wasim/ as a starting >> point. > > Basically, you log to csv files, and then the script periodically >> picks > them up and loads to your DB. This is using mysql as an example, >> but > you can do the same with postgres as well. > > If you need realtime inserts, >> then use mod_cdr_xml and have those post > to a script on a webserver that >> parses the xml and inserts into > appropriate tables. This is what I use along >> with a rails app. > > Remember that if you do real time, you also need to >> periodically scrape > the error directory and load those (mod_cdr_xml will save >> to error if it > can't successfully post to your script). > > On 2/6/2009 10:09 AM, >> Nik Middleton wrote: >> Hi Guys >> >> >> >> I?m looking for some pointers on >> how to collect CDR?s and store in >> mysql. Is there anything built in yet? >> >> >> >> >> I can rate the calls as a batch process, I simply need the call >> data. >> >> >> >> Regards >> >> > > > _______________________________________________ > Freeswitch-users mailing >> list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman >> /listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/opt >> ions/freeswitch-users > http://www.freeswitch.org > > > ____________________________ >> ___________________ > Freeswitch-users mailing >> list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman >> /listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/opt >> ions/freeswitch-users > http://www.freeswitch.org > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From freeswitch-users at lists.rupa.com Fri Feb 6 13:24:59 2009 From: freeswitch-users at lists.rupa.com (Rupa Schomaker (lists)) Date: Fri, 06 Feb 2009 15:24:59 -0600 Subject: [Freeswitch-users] Call accounting - CDR's In-Reply-To: References: <030001c98880$9e660f90$db322eb0$@net> Message-ID: <498CAAAB.2040405@lists.rupa.com> I'm doing this on low volume pbx setup, so posting to the web server is fine with my load. If you are doing high load, then definitely write to files and batch process them. On 2/6/2009 3:07 PM, Nik Middleton wrote: > So you're simply posting this file to a web server? How do you find > the load on it at this rate of calls? > > BTW can anyone point me to resources discussing how to do this? > (Using a web server to post data to a db) I've not this sort of thing > before, and I'm not too sure what I should be goggling for > > > Regards, > > > -----Original Message----- From: > freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Ken Rice Sent: 06 February 2009 17:40 To: > freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] > Call accounting - CDR's > > Mod_xml_cdr will drop a file to the file system on failure to post. > You can also leverage this drop a file to the file system and run a > CDR processor locally. > > We handle call rates in the 500+ range using the local file system as > a caching mechanism and a simple PHP script to rate the CDRs and load > them into a pgsql db. > > > > >> From: Adam Long Reply-To: >> Date: Fri, 6 Feb 2009 >> 12:30:34 -0500 To: Subject: >> Re: [Freeswitch-users] Call accounting - CDR's >> >> Is mod_cdr_xml asynchronous ... by that I mean .. if there is a >> communication failure how > does this effect load on the system or the call for that >> matter... > it wouldn?t hold up the progress or delay the turn up of the call or >> anything would it? > > I understand it would write to the error directory (but my >> thoughts are what is the impact of > this beyond the obvious IO hit) > > I guess a >> good question is what is more load intensive??? > > 1.) mod_cdr_csv (with batch >> script that loads into DB somewhere) > 2.) mod_cdr_xml (posting to lighttpd on >> remote host inserting into DB) > > I'm thinking about this for a system that >> would be handling in excess of 200-300 call setups > per >> second. > > -Adam > > -----Original Message----- From: >> freeswitch-users-bounces at lists.freeswitch.org >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of >> Rupa Schomaker (lists) > Sent: Friday, February 06, 2009 11:27 AM To: >> freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Call >> accounting - CDR's > > Use the script in scripts/contrib/wasim/ as a starting >> point. > > Basically, you log to csv files, and then the script periodically >> picks > them up and loads to your DB. This is using mysql as an example, >> but > you can do the same with postgres as well. > > If you need realtime inserts, >> then use mod_cdr_xml and have those post > to a script on a webserver that >> parses the xml and inserts into > appropriate tables. This is what I use along >> with a rails app. > > Remember that if you do real time, you also need to >> periodically scrape > the error directory and load those (mod_cdr_xml will save >> to error if it > can't successfully post to your script). > > On 2/6/2009 10:09 AM, >> Nik Middleton wrote: Hi Guys >> >> >> >> I?m looking for some pointers on how to collect CDR?s and store in >> mysql. Is there anything built in yet? >> >> >> >> >> I can rate the calls as a batch process, I simply need the call >> data. >> >> >> >> Regards >> From sicfslist at gmail.com Fri Feb 6 13:30:13 2009 From: sicfslist at gmail.com (Shelby Ramsey) Date: Fri, 6 Feb 2009 15:30:13 -0600 Subject: [Freeswitch-users] Call accounting - CDR's In-Reply-To: References: Message-ID: <35b355e90902061330t34c22f27x11013de60b101c76@mail.gmail.com> Just out of curiosity ... You actually set the values in xml_cdr_conf.xml to an invalid value ... and then FS tries it and then dumps it into the err_dir? Nik, Just configure the xml_conf_cdr and it will post all of the channel variables to your web server ... you can look at the variables and see what you want. Or I actually like Ken's suggestion ... that makes a lot of sense ... same benefit of having all of the channel variables ... no overhead. SDR -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090206/d1bc6461/attachment-0002.html From krice at freeswitch.org Fri Feb 6 13:33:13 2009 From: krice at freeswitch.org (Ken Rice) Date: Fri, 06 Feb 2009 15:33:13 -0600 Subject: [Freeswitch-users] Call accounting - CDR's In-Reply-To: <35b355e90902061330t34c22f27x11013de60b101c76@mail.gmail.com> Message-ID: Nope you comment out that line and it wont even attempt to post and will drop it into the log/xml_cdr directory From: Shelby Ramsey Reply-To: Date: Fri, 6 Feb 2009 15:30:13 -0600 To: Subject: Re: [Freeswitch-users] Call accounting - CDR's Just out of curiosity ... You actually set the values in xml_cdr_conf.xml to an invalid value ... and then FS tries it and then dumps it into the err_dir? Nik, Just configure the xml_conf_cdr and it will post all of the channel variables to your web server ... you can look at the variables and see what you want. Or I actually like Ken's suggestion ... that makes a lot of sense ... same benefit of having all of the channel variables ... no overhead. SDR _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090206/7cc7417b/attachment-0002.html From jalsot at gmail.com Fri Feb 6 13:36:51 2009 From: jalsot at gmail.com (Tamas) Date: Fri, 06 Feb 2009 22:36:51 +0100 Subject: [Freeswitch-users] mod_shout delay in trunk In-Reply-To: <191c3a030902061248r3d307647le5574c1004c6da24@mail.gmail.com> References: <15830211.131233952452225.JavaMail.root@zimbra> <28165623.151233952574612.JavaMail.root@zimbra> <191c3a030902061248r3d307647le5574c1004c6da24@mail.gmail.com> Message-ID: <498CAD73.2040709@gmail.com> Hello, could this option be used to lower I/O load - to rather write more bytes at once rather than one by one - on file recording (record_session)? Regards, Tamas Anthony Minessale ?rta: > Thanks, > > We appreciate the positive feedback! > > if you revert the change I suggested and update i added a new variable > > enable_file_write_buffering=false > > set this variable on the channel before you start recording it with > the set application or in the dialstring in {} > on outbound calls and it should skip the buffering. > > Could you test it for me and confirm it works? > > Thank you > > > On Fri, Feb 6, 2009 at 2:36 PM, > wrote: > > That worked great! > > I wanted to say just how awesome Freeswitch is, I have been doing > voip related development with SIP since 2000 and this is by far > the most well designed piece of voip software I have used or > developed on. I currently have a homegrown sip server built on > the NIST sip stack with Sun's JMF libraries for RTP processing. > 95% of the code and complexity is handling the SIP and RTP > sessions, the other 5% is the final application logic and what is > most important to me. By letting freeswitch do whats its good at > (call routing, sip and media handling) it allows me to focus on > what I'm good at (what should we do with those streams, like > record them). I have been bragging about this project to anybody > who will listen! > > Dan- > > ----- Original Message ----- > From: "Anthony Minessale" > > To: freeswitch-users at lists.freeswitch.org > > Sent: Friday, February 6, 2009 1:07:44 PM GMT -07:00 US/Canada > Mountain > Subject: Re: [Freeswitch-users] mod_shout delay in trunk > > edit switch_ivr_play_say.c line 423 > > comment the line out and recompile. > Tell me if it helps you and i will consider making it configurable. > > > On Fri, Feb 6, 2009 at 2:01 PM, > wrote: > > For me it is. For what I'm using it for I can tolerate around > a second or two delay. I have the icecast server setup to > only buffer 1K for their on-connect burst as well as my > flash/flex player to only buffer 1k (yes I might as well not > buffer at all, which I may end up doing). In 1.0.2 this > worked very well. Is this buffer configurable? If not, where > is it being set? > > Thanks > Dan- > > ----- Original Message ----- > From: "Brian West" > > To: freeswitch-users at lists.freeswitch.org > > Sent: Friday, February 6, 2009 12:47:53 PM GMT -07:00 > US/Canada Mountain > Subject: Re: [Freeswitch-users] mod_shout delay in trunk > > Let me clarify.. yes this is normal file buffering was added > so we wouldn't thrash your hard drive with tiny bits of data > when recording calls so now it buffers and writes larger > chunks to disk. This is why you have this delay which is 100% > normal.... is realtime a critical thing? It is shout cast so > you know it doesn't have to be realtime.. in fact some clients > will buffer a little bit anyway and add to it. > > /b > > On Feb 6, 2009, at 1:43 PM, freeswitch-users at digitaldan.com > wrote: > > I have, do you know what would have changed between 1.0.2 > and trunk that would cause the buffer to change? Also if > its not in mod_shout.c (which I copied from 1.0.2 to trunk > for testing with no luck), where else would fs be > buffering? One thing I have noticed is that in 1.0.2 as > soon as the dial plan hits my record statement I see > mod_shout logging that it has connected to the icecast > server, in trunk it takes about 5 seconds to see the same > log mesage. Below is my current svn info > Path: . > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > > _______________________________________________ Freeswitch-users > mailing list Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From msc at freeswitch.org Fri Feb 6 13:36:39 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 6 Feb 2009 13:36:39 -0800 Subject: [Freeswitch-users] mod_shout delay in trunk In-Reply-To: <15850979.181233954786201.JavaMail.root@zimbra> References: <191c3a030902061248r3d307647le5574c1004c6da24@mail.gmail.com> <15850979.181233954786201.JavaMail.root@zimbra> Message-ID: <87f2f3b90902061336s5bbbdce3o921023b83590df88@mail.gmail.com> > if you revert the change I suggested and update i added a new variable > > enable_file_write_buffering=false > > set this variable on the channel before you start recording it with the set > application or in the dialstring in {} > on outbound calls and it should skip the buffering. > > Could you test it for me and confirm it works? > > Thank you Also, as payment for services rendered could you please add this variable and description to the wiki? http://wiki.freeswitch.org/wiki/Channel_Variables Thanks! -MC From mgg at giagnocavo.net Fri Feb 6 13:38:17 2009 From: mgg at giagnocavo.net (Michael Giagnocavo) Date: Fri, 6 Feb 2009 16:38:17 -0500 Subject: [Freeswitch-users] Transcoding G723 In-Reply-To: <49796.1233961333@enterux.com> References: <49796.1233961333@enterux.com> Message-ID: <6E8D2069C08AA84A83D336E996AE4C6702362F6C4C@mse17be1.mse17.exchange.ms> Sure ?Custom CPU? isn?t just a 4 socket Intel setup? -Michael From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mitul Limbani Sent: Friday, February 06, 2009 4:02 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Transcoding G723 Hello guys, Sorry it was my mistake, i re-read the entire proposal and it looks like their specialized 1U Hardware with custom CPU can handle close to 1500 simultaneous G729 encoding n transmission, also this hardware offers from any codec to any codec transcoding path. Also this box easily works over FreeSWITCH, Asterisk, RTPProxy, OpenSIPS are their claims. and their pricing are First year ?16,000 Second year ?13,000/annum recurring This is found slightly on higher side though, Thanks & Regards, Mitul Limbani, Founder & CEO, Enterux Solutions, The Enterprise Linux Company (TM), www.enterux.com +91-9820332422 On Sat 07/02/09 02:00 , Michael Giagnocavo mgg at giagnocavo.net sent: $22K would buy quite a few machines with many core Xeons. I just don't see how it'd be effective at that price. Not to mention a yearly figure. The only thing to take into consideration would be the G729 licenses. But in bulk, the price should be pretty effective, even figuring in hardware. (Not to mention what things like Larrabee will mean for encoding -- 32 1.5GHz "Pentium 4 x64" cores, each with 4 threads?) And... if you're running Asterisk, um, isn't 5000 channels a wee bit over what you can handle anyways? ;) -Michael -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mitul Limbani Sent: Friday, February 06, 2009 12:28 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Transcoding G723 Hi, I came across a company who was selling hardware which could do like 5000 G729 conversion simultanoeusly, I was like this sounds cool, they have support for asterisk, I haven't enquirer yet how they do this, but anyone wishes to buy it cost ?15000/year for the hrdware + support Any one on the list requires such high throughput calls can connect with me so that we can really test if it can work with freeswitch. Regards, Mitul Limbani, Founder & CEO, Enterux Solutions Pvt Ltd, The Enterprise Linux Company(r), http://www.enterux.com/ On 06-Feb-09, at 21:58, Shelby Ramsey > wrote: > Steve, > > You definitely have a better grasp on this topic than me. But I > think it's a tough sell on the host based processing ... when you > look at products like what audio codes can do on a card (3 DS-3's > worth of transcoding) ... but I have had a couple of soft switch > vendors claim though that they could do 2,000 calls per host (but I > seriously doubt it). > > I agree that producing a card that does 120 channels is pretty > worthless ... but having something that could do say a 1000 would be > very helpful. > > G723 is a pretty big deal internationally ... I'm even seeing crazy > requests like AMR from folks trying to originate VoIP off of mobile > devices in Europe. > > But I agree just being able to g729 --> ulaw or ulaw --> g729 would > be a great first step (host based or otherwise). > > SDR > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ________________________________ Msg sent via Enterux Enterprise Email Server : http://www.enterux.com/ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090206/adc47ed6/attachment-0002.html From sicfslist at gmail.com Fri Feb 6 13:39:02 2009 From: sicfslist at gmail.com (Shelby Ramsey) Date: Fri, 6 Feb 2009 15:39:02 -0600 Subject: [Freeswitch-users] Call accounting - CDR's In-Reply-To: References: <35b355e90902061330t34c22f27x11013de60b101c76@mail.gmail.com> Message-ID: <35b355e90902061339w2084ddfbvbe7ed95d11b438ea@mail.gmail.com> Even better ... Thanks Ken! SDR -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090206/7291a097/attachment-0002.html From freeswitch-users at digitaldan.com Fri Feb 6 13:54:07 2009 From: freeswitch-users at digitaldan.com (Dan) Date: Fri, 6 Feb 2009 14:54:07 -0700 (MST) Subject: [Freeswitch-users] mod_shout delay in trunk In-Reply-To: <87f2f3b90902061336s5bbbdce3o921023b83590df88@mail.gmail.com> Message-ID: <9128420.211233957247696.JavaMail.root@zimbra> Done! ----- Original Message ----- From: "Michael Collins" To: freeswitch-users at lists.freeswitch.org Sent: Friday, February 6, 2009 2:36:39 PM GMT -07:00 US/Canada Mountain Subject: Re: [Freeswitch-users] mod_shout delay in trunk > if you revert the change I suggested and update i added a new variable > > enable_file_write_buffering=false > > set this variable on the channel before you start recording it with the set > application or in the dialstring in {} > on outbound calls and it should skip the buffering. > > Could you test it for me and confirm it works? > > Thank you Also, as payment for services rendered could you please add this variable and description to the wiki? http://wiki.freeswitch.org/wiki/Channel_Variables Thanks! -MC _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090206/50f4be53/attachment-0002.html From msc at freeswitch.org Fri Feb 6 14:04:36 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 6 Feb 2009 14:04:36 -0800 Subject: [Freeswitch-users] mod_shout delay in trunk In-Reply-To: <9128420.211233957247696.JavaMail.root@zimbra> References: <87f2f3b90902061336s5bbbdce3o921023b83590df88@mail.gmail.com> <9128420.211233957247696.JavaMail.root@zimbra> Message-ID: <87f2f3b90902061404v74ff9552sbb8a0464c18c65bc@mail.gmail.com> > Done! Many thanks! From nik.middleton at noblesolutions.co.uk Fri Feb 6 14:28:36 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Fri, 6 Feb 2009 22:28:36 -0000 Subject: [Freeswitch-users] Call accounting - CDR's In-Reply-To: <35b355e90902061339w2084ddfbvbe7ed95d11b438ea@mail.gmail.com> References: <35b355e90902061330t34c22f27x11013de60b101c76@mail.gmail.com> <35b355e90902061339w2084ddfbvbe7ed95d11b438ea@mail.gmail.com> Message-ID: Guy's, Thanks for all the responses; it's truly refreshing to get so much valuable input. I'm reading the docs furiously, but I still don't know what I don't know yet. But given time I will return the favor to those that come later. Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090206/37e6dfb6/attachment-0002.html From msc at freeswitch.org Fri Feb 6 14:42:25 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 6 Feb 2009 14:42:25 -0800 Subject: [Freeswitch-users] Call accounting - CDR's In-Reply-To: References: <35b355e90902061330t34c22f27x11013de60b101c76@mail.gmail.com> <35b355e90902061339w2084ddfbvbe7ed95d11b438ea@mail.gmail.com> Message-ID: <87f2f3b90902061442s2d1ef36fq5199a6766c083581@mail.gmail.com> > > Thanks for all the responses; it's truly refreshing to get so much valuable > input. I'm reading the docs furiously, but I still don't know what I don't > know yet. But given time I will return the favor to those that come later. Sounds good! If you feel up to doing any wiki documentation please let me know and I'll be happy to offer some pointers, etc. -MC From nik.middleton at noblesolutions.co.uk Fri Feb 6 15:20:07 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Fri, 6 Feb 2009 23:20:07 -0000 Subject: [Freeswitch-users] Call accounting - CDR's In-Reply-To: <87f2f3b90902061442s2d1ef36fq5199a6766c083581@mail.gmail.com> References: <35b355e90902061330t34c22f27x11013de60b101c76@mail.gmail.com><35b355e90902061339w2084ddfbvbe7ed95d11b438ea@mail.gmail.com> <87f2f3b90902061442s2d1ef36fq5199a6766c083581@mail.gmail.com> Message-ID: More than happy to add my 2 cents worth when I have something useful to say Question regarding the xml cdr's Let's say I have a cron job looking at these files and processing them. How does FS create them. Does a MV occur from some other DIR, as otherwise it's possible I might try and open an in progress record. Regards -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 06 February 2009 22:42 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Call accounting - CDR's > > Thanks for all the responses; it's truly refreshing to get so much valuable > input. I'm reading the docs furiously, but I still don't know what I don't > know yet. But given time I will return the favor to those that come later. Sounds good! If you feel up to doing any wiki documentation please let me know and I'll be happy to offer some pointers, etc. -MC _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From gkuri at ieee.org Fri Feb 6 15:19:49 2009 From: gkuri at ieee.org (Gabriel Kuri) Date: Fri, 06 Feb 2009 15:19:49 -0800 Subject: [Freeswitch-users] Separate NICs for Performance Message-ID: <498CC595.8070601@ieee.org> Hey Folks: For a FS box that's generally handling higher amounts of inbound/outbound call traffic (say 500 - 700 calls) and registrations (30 - 50 per second), is it recommended to split off the signaling and media traffic onto separate NICs for performance reasons? Or is it better to split all the traffic for the phones and outside traffic to carriers into separate profiles and split those profiles onto separate NICs? or perhaps none of the above and there's still a better solution I didn't mention ;) ? Thanks, Gabe From brian at freeswitch.org Fri Feb 6 15:28:36 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 6 Feb 2009 17:28:36 -0600 Subject: [Freeswitch-users] Separate NICs for Performance In-Reply-To: <498CC595.8070601@ieee.org> References: <498CC595.8070601@ieee.org> Message-ID: <70153680-883F-4A6D-9D78-83FFE65C239B@freeswitch.org> If the nic's have their own bus you could do that to improve network performance of sip signaling and media... now the neat thing would be to have an option to have say three network interfaces and have FreeSWITCH round robin them per call. I smell a bounty. /b On Feb 6, 2009, at 5:19 PM, Gabriel Kuri wrote: > Hey Folks: > > For a FS box that's generally handling higher amounts of > inbound/outbound call traffic (say 500 - 700 calls) and registrations > (30 - 50 per second), is it recommended to split off the signaling and > media traffic onto separate NICs for performance reasons? Or is it > better to split all the traffic for the phones and outside traffic to > carriers into separate profiles and split those profiles onto separate > NICs? or perhaps none of the above and there's still a better > solution I > didn't mention ;) ? > > Thanks, > > Gabe From msc at freeswitch.org Fri Feb 6 16:00:11 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 6 Feb 2009 16:00:11 -0800 Subject: [Freeswitch-users] Call accounting - CDR's In-Reply-To: References: <35b355e90902061330t34c22f27x11013de60b101c76@mail.gmail.com> <35b355e90902061339w2084ddfbvbe7ed95d11b438ea@mail.gmail.com> <87f2f3b90902061442s2d1ef36fq5199a6766c083581@mail.gmail.com> Message-ID: <87f2f3b90902061600i5e3ce93esfe883b36afa6cc1c@mail.gmail.com> > Question regarding the xml cdr's > > Let's say I have a cron job looking at these files and processing them. > How does FS create them. Does a MV occur from some other DIR, as > otherwise it's possible I might try and open an in progress record. No worries - the file isn't "available" until it's ready to go, just like being mv'd into the dir. -MC From kristian.kielhofner at gmail.com Fri Feb 6 16:00:50 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Fri, 6 Feb 2009 19:00:50 -0500 Subject: [Freeswitch-users] Separate NICs for Performance In-Reply-To: <70153680-883F-4A6D-9D78-83FFE65C239B@freeswitch.org> References: <498CC595.8070601@ieee.org> <70153680-883F-4A6D-9D78-83FFE65C239B@freeswitch.org> Message-ID: <2d9149cd0902061600k1515998t8adf3dcc84c1ce21@mail.gmail.com> Bonding? Intel ANS? On Fri, Feb 6, 2009 at 6:28 PM, Brian West wrote: > If the nic's have their own bus you could do that to improve network > performance of sip signaling and media... now the neat thing would be > to have an option to have say three network interfaces and have > FreeSWITCH round robin them per call. I smell a bounty. > > /b -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From brian at freeswitch.org Fri Feb 6 16:04:37 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 6 Feb 2009 18:04:37 -0600 Subject: [Freeswitch-users] Separate NICs for Performance In-Reply-To: <2d9149cd0902061600k1515998t8adf3dcc84c1ce21@mail.gmail.com> References: <498CC595.8070601@ieee.org> <70153680-883F-4A6D-9D78-83FFE65C239B@freeswitch.org> <2d9149cd0902061600k1515998t8adf3dcc84c1ce21@mail.gmail.com> Message-ID: <6BDE6A1E-673D-4B01-B8C0-99F2E7C771CC@freeswitch.org> That would work too I suspect. /b On Feb 6, 2009, at 6:00 PM, Kristian Kielhofner wrote: > Bonding? Intel ANS? > > On Fri, Feb 6, 2009 at 6:28 PM, Brian West > wrote: >> If the nic's have their own bus you could do that to improve network >> performance of sip signaling and media... now the neat thing would be >> to have an option to have say three network interfaces and have >> FreeSWITCH round robin them per call. I smell a bounty. >> >> /b > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090206/b8a9a8db/attachment-0002.html From kristian.kielhofner at gmail.com Fri Feb 6 16:06:56 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Fri, 6 Feb 2009 19:06:56 -0500 Subject: [Freeswitch-users] Separate NICs for Performance In-Reply-To: <6BDE6A1E-673D-4B01-B8C0-99F2E7C771CC@freeswitch.org> References: <498CC595.8070601@ieee.org> <70153680-883F-4A6D-9D78-83FFE65C239B@freeswitch.org> <2d9149cd0902061600k1515998t8adf3dcc84c1ce21@mail.gmail.com> <6BDE6A1E-673D-4B01-B8C0-99F2E7C771CC@freeswitch.org> Message-ID: <2d9149cd0902061606g61913108ree46ed95087c292e@mail.gmail.com> I think the big problem is still going to be interrupts and other networking stuff. On Fri, Feb 6, 2009 at 7:04 PM, Brian West wrote: > That would work too I suspect. > /b -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From brian at freeswitch.org Fri Feb 6 16:14:11 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 6 Feb 2009 18:14:11 -0600 Subject: [Freeswitch-users] Separate NICs for Performance In-Reply-To: <2d9149cd0902061606g61913108ree46ed95087c292e@mail.gmail.com> References: <498CC595.8070601@ieee.org> <70153680-883F-4A6D-9D78-83FFE65C239B@freeswitch.org> <2d9149cd0902061600k1515998t8adf3dcc84c1ce21@mail.gmail.com> <6BDE6A1E-673D-4B01-B8C0-99F2E7C771CC@freeswitch.org> <2d9149cd0902061606g61913108ree46ed95087c292e@mail.gmail.com> Message-ID: <483DA153-9427-4681-8872-770F4E0DF5BF@freeswitch.org> Wouldn't be a huge deal if each card has a dedicated bus then it wouldn't be fighting for bandwidth... its still going to be hitting a limit at some point but you might get more milage out of it. /b On Feb 6, 2009, at 6:06 PM, Kristian Kielhofner wrote: > I think the big problem is still going to be interrupts and other > networking stuff. From mkarp at securesilence.com Fri Feb 6 16:32:42 2009 From: mkarp at securesilence.com (Maxim Karp) Date: Fri, 6 Feb 2009 16:32:42 -0800 Subject: [Freeswitch-users] Voicemail prompts and playback speed Message-ID: <009301c988bb$99adb5d0$cd092170$@com> Hello all, Can anyone please let me know how I might be able to configure the voice mail prompts and their playback speed? Thanks, Maxim. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090206/d95402a0/attachment-0002.html From mike at jerris.com Fri Feb 6 18:27:48 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 6 Feb 2009 21:27:48 -0500 Subject: [Freeswitch-users] Voicemail prompts and playback speed In-Reply-To: <009301c988bb$99adb5d0$cd092170$@com> References: <009301c988bb$99adb5d0$cd092170$@com> Message-ID: Not sure what you mean by playback speed. All the prompts for voicemail are defined in the phrase macros in the configuration. Mike On Feb 6, 2009, at 7:32 PM, "Maxim Karp" wrote: > Hello all, > > > > Can anyone please let me know how I might be able to configure the > voice mail prompts and their playback speed? > > > > Thanks, > > > > Maxim. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From msc at freeswitch.org Fri Feb 6 20:26:33 2009 From: msc at freeswitch.org (Michael S Collins) Date: Fri, 6 Feb 2009 20:26:33 -0800 Subject: [Freeswitch-users] Voicemail prompts and playback speed Message-ID: On Feb 6, 2009, at 6:27 PM, Michael Jerris wrote: > Not sure what you mean by playback speed. All the prompts for > voicemail are defined in the phrase macros in the configuration. Check out conf/lang/en/vm/sounds.xml -MC From uv at yuvalhertzog.com Sat Feb 7 01:39:57 2009 From: uv at yuvalhertzog.com (UV) Date: Sat, 7 Feb 2009 20:39:57 +1100 Subject: [Freeswitch-users] Outgoing registration expiry Message-ID: I?m trying to set up a default provider via the example.com.xml using the variables set in vars.xml. The provider has a registration expiry of 120 seconds and I?m trying to set it up to register every 60 seconds but when I change the ?expire-seconds? variable (in directory/default/example.com.xml), it doesn?t seem to have any effect. Actually, doesn?t matter how long I wait, it doesn?t seem to re-register at all? Any idea what I?m missing? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090207/ecee7ffe/attachment-0002.html From uv at yuvalhertzog.com Sat Feb 7 01:40:09 2009 From: uv at yuvalhertzog.com (UV) Date: Sat, 7 Feb 2009 20:40:09 +1100 Subject: [Freeswitch-users] Global Variables forgotten through the public context? Message-ID: <0B4E2726927041D09D0425DA0242C805@UVix> Another question: When I try routing calls through the public context to the default context, global variables (set in vars.xml) seem to be ?forgotten? and appear blank. I?m trying a very simple scenario of an incoming call on the public context routed to a phone number which is rightfully captured at ?local.example.com? but all three variables ${outbound_caller_id_number}, ${outbound_caller_id_name} and ${default_gateway} appear to be blanked out (according to the console debug output). Again, any idea what am I doing wrong? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090207/3ca9c5a5/attachment-0002.html From brian at freeswitch.org Sat Feb 7 01:50:30 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 7 Feb 2009 03:50:30 -0600 Subject: [Freeswitch-users] Outgoing registration expiry In-Reply-To: References: Message-ID: <2EE57DC0-CF8B-4E56-B383-9FBFAA60CD7F@freeswitch.org> You would have to reload and restart the profile for that to take effect. You can't change the global and have it magically start using the new value. /b On Feb 7, 2009, at 3:39 AM, UV wrote: > I?m trying to set up a default provider via the example.com.xml > using the variables set in vars.xml. > The provider has a registration expiry of 120 seconds and I?m trying > to set it up to register every 60 seconds but when I change the > ?expire-seconds? variable (in directory/default/example.com.xml), it > doesn?t seem to have any effect. Actually, doesn?t matter how long I > wait, it doesn?t seem to re-register at all? > > Any idea what I?m missing? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090207/d21260cc/attachment-0002.html From brian at freeswitch.org Sat Feb 7 01:51:04 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 7 Feb 2009 03:51:04 -0600 Subject: [Freeswitch-users] Global Variables forgotten through the public context? In-Reply-To: <0B4E2726927041D09D0425DA0242C805@UVix> References: <0B4E2726927041D09D0425DA0242C805@UVix> Message-ID: <9D0CFA79-C120-4D66-B48C-7A5641B07AF4@freeswitch.org> No the vars are there can you provide more detail on what exactly you're doing? The default config uses the call_debug variable and its a global set in vars.xml. /b On Feb 7, 2009, at 3:40 AM, UV wrote: > Another question: > When I try routing calls through the public context to the default > context, global variables (set in vars.xml) seem to be ?forgotten? > and appear blank. > I?m trying a very simple scenario of an incoming call on the public > context routed to a phone number which is rightfully captured at > ?local.example.com? but all three variables $ > {outbound_caller_id_number}, ${outbound_caller_id_name} and $ > {default_gateway} appear to be blanked out (according to the console > debug output). > > Again, any idea what am I doing wrong? > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090207/d9014d47/attachment-0002.html From nik.middleton at noblesolutions.co.uk Sat Feb 7 02:07:34 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Sat, 7 Feb 2009 10:07:34 -0000 Subject: [Freeswitch-users] Call accounting - CDR's In-Reply-To: <87f2f3b90902061600i5e3ce93esfe883b36afa6cc1c@mail.gmail.com> References: <35b355e90902061330t34c22f27x11013de60b101c76@mail.gmail.com><35b355e90902061339w2084ddfbvbe7ed95d11b438ea@mail.gmail.com><87f2f3b90902061442s2d1ef36fq5199a6766c083581@mail.gmail.com> <87f2f3b90902061600i5e3ce93esfe883b36afa6cc1c@mail.gmail.com> Message-ID: Great, thanks for that. One of the big issues with Asterisk's way of billing is that if let's say a remote phone diverts a call to another number, say a mobile, because a local channel is created for the redirect, Asterisk loses critical information such as the account code and therefore cannot be billed for. I'm going to try this with FS today and see what happens. It would be awesome if it is accountable. That way I wouldn't have to force the user to do diverts from a web page Regards, -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 07 February 2009 00:00 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Call accounting - CDR's > Question regarding the xml cdr's > > Let's say I have a cron job looking at these files and processing them. > How does FS create them. Does a MV occur from some other DIR, as > otherwise it's possible I might try and open an in progress record. No worries - the file isn't "available" until it's ready to go, just like being mv'd into the dir. -MC _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From nik.middleton at noblesolutions.co.uk Sat Feb 7 03:38:15 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Sat, 7 Feb 2009 11:38:15 -0000 Subject: [Freeswitch-users] AMD Functionality Message-ID: Hi Guys, Is there any form of Answer phone detection in FS? A search hasn't really brought up anything Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090207/b52a1b5b/attachment-0002.html From uv at yuvalhertzog.com Sat Feb 7 05:14:34 2009 From: uv at yuvalhertzog.com (UV) Date: Sun, 8 Feb 2009 00:14:34 +1100 Subject: [Freeswitch-users] Outgoing registration expiry In-Reply-To: <2EE57DC0-CF8B-4E56-B383-9FBFAA60CD7F@freeswitch.org> References: <2EE57DC0-CF8B-4E56-B383-9FBFAA60CD7F@freeswitch.org> Message-ID: <057607FE113648A0B6D0DA614BDD3244@UVix> Obviously. XML was reloaded, sofia profiles were reloaded, freeswitch app was shutdown and restarted and last but not least, I?ve rebooted the computer several times just to make sure :-) No, seriously, I?ve done everything to verify the settings are there ? it?s just not re-registering. My question is does the ?expire-seconds? really work on per second interval? Value 5 equals to gateway being registered every 5 seconds? Because it doesn?t? Am I the only one to have this problem? Am I using it wrong? _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Saturday, February 07, 2009 8:51 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Outgoing registration expiry You would have to reload and restart the profile for that to take effect. You can't change the global and have it magically start using the new value. /b On Feb 7, 2009, at 3:39 AM, UV wrote: I?m trying to set up a default provider via the example.com.xml using the variables set in vars.xml. The provider has a registration expiry of 120 seconds and I?m trying to set it up to register every 60 seconds but when I change the ?expire-seconds? variable (in directory/default/example.com.xml), it doesn?t seem to have any effect. Actually, doesn?t matter how long I wait, it doesn?t seem to re-register at all? Any idea what I?m missing? No virus found in this incoming message. Checked by AVG i www.avg.com Version: 8.0.233 / Virus Database: 270.10.18/1937 i Release Date: 02/06/09 11:31:00 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090208/0ad85243/attachment-0002.html From uv at yuvalhertzog.com Sat Feb 7 05:30:46 2009 From: uv at yuvalhertzog.com (UV) Date: Sun, 8 Feb 2009 00:30:46 +1100 Subject: [Freeswitch-users] Global Variables forgotten through thepublic context? In-Reply-To: <9D0CFA79-C120-4D66-B48C-7A5641B07AF4@freeswitch.org> References: <0B4E2726927041D09D0425DA0242C805@UVix> <9D0CFA79-C120-4D66-B48C-7A5641B07AF4@freeswitch.org> Message-ID: <5B1ED1D4834B4719B309FAC03F1595C0@UVix> I took the out-of-the-box public context dialplan, added an entry to dial a 10 digit number through the default context and when it ran, I noticed in the console log that all the values below are either null or empty. To be more specific: In the public context I?ve added this extension in the beginning: If you want, I can pastebin the log for you to see. It would be easier for you to replicate it yourself and see. It?s quite easy to replicate. _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Saturday, February 07, 2009 8:51 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Global Variables forgotten through thepublic context? No the vars are there can you provide more detail on what exactly you're doing? The default config uses the call_debug variable and its a global set in vars.xml. /b On Feb 7, 2009, at 3:40 AM, UV wrote: Another question: When I try routing calls through the public context to the default context, global variables (set in vars.xml) seem to be ?forgotten? and appear blank. I?m trying a very simple scenario of an incoming call on the public context routed to a phone number which is rightfully captured at ?local.example.com? but all three variables ${outbound_caller_id_number}, ${outbound_caller_id_name} and ${default_gateway} appear to be blanked out (according to the console debug output). Again, any idea what am I doing wrong? No virus found in this incoming message. Checked by AVG i www.avg.com Version: 8.0.233 / Virus Database: 270.10.18/1937 i Release Date: 02/06/09 11:31:00 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090208/59771189/attachment-0002.html From brian at freeswitch.org Sat Feb 7 05:32:08 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 7 Feb 2009 07:32:08 -0600 Subject: [Freeswitch-users] Outgoing registration expiry In-Reply-To: <057607FE113648A0B6D0DA614BDD3244@UVix> References: <2EE57DC0-CF8B-4E56-B383-9FBFAA60CD7F@freeswitch.org> <057607FE113648A0B6D0DA614BDD3244@UVix> Message-ID: <465CCF29-33B5-4A93-A322-EE71270F8BE6@freeswitch.org> turn on the sofia debug. on the profile. I will suspect that the far end proxy is forcing your expire to a higher number. /b On Feb 7, 2009, at 7:14 AM, UV wrote: > Obviously. XML was reloaded, sofia profiles were reloaded, > freeswitch app was shutdown and restarted and last but not least, > I?ve rebooted the computer several times just to make sure J > No, seriously, I?ve done everything to verify the settings are there > ? it?s just not re-registering. My question is does the ?expire- > seconds? really work on per second interval? Value 5 equals to > gateway being registered every 5 seconds? > Because it doesn?t? Am I the only one to have this problem? Am I > using it wrong? > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090207/4a693178/attachment-0002.html From brian at freeswitch.org Sat Feb 7 05:32:43 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 7 Feb 2009 07:32:43 -0600 Subject: [Freeswitch-users] Global Variables forgotten through thepublic context? In-Reply-To: <5B1ED1D4834B4719B309FAC03F1595C0@UVix> References: <0B4E2726927041D09D0425DA0242C805@UVix> <9D0CFA79-C120-4D66-B48C-7A5641B07AF4@freeswitch.org> <5B1ED1D4834B4719B309FAC03F1595C0@UVix> Message-ID: Please do... also make sure you have a context param on your gateway. /b On Feb 7, 2009, at 7:30 AM, UV wrote: > I took the out-of-the-box public context dialplan, added an entry to > dial a 10 digit number through the default context and when it ran, > I noticed in the console log that all the values below are either > null or empty. > > To be more specific: > In the public context I?ve added this extension in the beginning: > > > > > > > If you want, I can pastebin the log for you to see. It would be > easier for you to replicate it yourself and see. It?s quite easy to > replicate. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090207/def372e5/attachment-0002.html From sicfslist at gmail.com Sat Feb 7 06:40:14 2009 From: sicfslist at gmail.com (Shelby Ramsey) Date: Sat, 7 Feb 2009 08:40:14 -0600 Subject: [Freeswitch-users] AMD Functionality In-Reply-To: References: Message-ID: <35b355e90902070640p4811b4fy88796dd62ee0c306@mail.gmail.com> Nik, Right now there is mod_vmd. It sets the channel variable vmd_detect if it detects a beep. If a beep is detected it will set vmd_detect=TRUE. If no beep is detected then it won't do anything. Example of usage as follows (with the outcome being hangup if answering machine is detected): And Ken Rice also has a mod that can be licensed that probably is a better solution and works much better. Hope this helps. SDR -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090207/5340510b/attachment-0002.html From uv at yuvalhertzog.com Sat Feb 7 07:24:44 2009 From: uv at yuvalhertzog.com (UV) Date: Sun, 8 Feb 2009 02:24:44 +1100 Subject: [Freeswitch-users] Outgoing registration expiry In-Reply-To: <465CCF29-33B5-4A93-A322-EE71270F8BE6@freeswitch.org> References: <2EE57DC0-CF8B-4E56-B383-9FBFAA60CD7F@freeswitch.org><057607FE113648A0B6D0DA614BDD3244@UVix> <465CCF29-33B5-4A93-A322-EE71270F8BE6@freeswitch.org> Message-ID: ?. And correct you are! Far end proxy does force to a higher number? Thanks! _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Sunday, February 08, 2009 12:32 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Outgoing registration expiry turn on the sofia debug. on the profile. I will suspect that the far end proxy is forcing your expire to a higher number. /b On Feb 7, 2009, at 7:14 AM, UV wrote: Obviously. XML was reloaded, sofia profiles were reloaded, freeswitch app was shutdown and restarted and last but not least, I?ve rebooted the computer several times just to make sure :-) No, seriously, I?ve done everything to verify the settings are there ? it?s just not re-registering. My question is does the ?expire-seconds? really work on per second interval? Value 5 equals to gateway being registered every 5 seconds? Because it doesn?t? Am I the only one to have this problem? Am I using it wrong? No virus found in this incoming message. Checked by AVG i www.avg.com Version: 8.0.233 / Virus Database: 270.10.18/1937 i Release Date: 02/06/09 11:31:00 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090208/0a4cad18/attachment-0002.html From uv at yuvalhertzog.com Sat Feb 7 07:26:47 2009 From: uv at yuvalhertzog.com (UV) Date: Sun, 8 Feb 2009 02:26:47 +1100 Subject: [Freeswitch-users] Global Variables forgotten through thepubliccontext? In-Reply-To: References: <0B4E2726927041D09D0425DA0242C805@UVix><9D0CFA79-C120-4D66-B48C-7A5641B07AF4@freeswitch.org><5B1ED1D4834B4719B309FAC03F1595C0@UVix> Message-ID: Will do. Now I have a little problem delaying me as the latest build changed something with the sound file playing and now the FS can?t find any local file to play (adds mysterious \16000\ to the file location?). I?ll try to isolate this first then get back to this issue. _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Sunday, February 08, 2009 12:33 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Global Variables forgotten through thepubliccontext? Please do... also make sure you have a context param on your gateway. /b On Feb 7, 2009, at 7:30 AM, UV wrote: I took the out-of-the-box public context dialplan, added an entry to dial a 10 digit number through the default context and when it ran, I noticed in the console log that all the values below are either null or empty. To be more specific: In the public context I?ve added this extension in the beginning: If you want, I can pastebin the log for you to see. It would be easier for you to replicate it yourself and see. It?s quite easy to replicate. No virus found in this incoming message. Checked by AVG i www.avg.com Version: 8.0.233 / Virus Database: 270.10.18/1937 i Release Date: 02/06/09 11:31:00 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090208/609c14c9/attachment-0002.html From juanbackson at gmail.com Sat Feb 7 08:44:16 2009 From: juanbackson at gmail.com (Juan Backson) Date: Sun, 8 Feb 2009 00:44:16 +0800 Subject: [Freeswitch-users] need suggestion on using mod_easyroute Message-ID: <27c25bc40902070844h5db40f22x26bcfce9ff21e852@mail.gmail.com> Hi, I notice that there is a newly-developed mod_easyroute model available. Has anyone used it with large amount of routes ( ex > 1M ) on a high traffic scenario? For that kind of scenario, would it be better to consider using out-going event socket to serve that purpose? I would greatly appreciate any recommendation. Thanks, JB -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090208/efd5c64b/attachment-0002.html From msc at freeswitch.org Sat Feb 7 10:14:57 2009 From: msc at freeswitch.org (Michael Collins) Date: Sat, 7 Feb 2009 10:14:57 -0800 Subject: [Freeswitch-users] Call accounting - CDR's In-Reply-To: References: <35b355e90902061330t34c22f27x11013de60b101c76@mail.gmail.com> <35b355e90902061339w2084ddfbvbe7ed95d11b438ea@mail.gmail.com> <87f2f3b90902061442s2d1ef36fq5199a6766c083581@mail.gmail.com> <87f2f3b90902061600i5e3ce93esfe883b36afa6cc1c@mail.gmail.com> Message-ID: <87f2f3b90902071014v2dd31baay43d8d6df57d10ed9@mail.gmail.com> On Sat, Feb 7, 2009 at 2:07 AM, Nik Middleton wrote: > Great, thanks for that. > > One of the big issues with Asterisk's way of billing is that if let's > say a remote phone diverts a call to another number, say a mobile, > because a local channel is created for the redirect, Asterisk loses > critical information such as the account code and therefore cannot be > billed for. I'm going to try this with FS today and see what happens. > It would be awesome if it is accountable. That way I wouldn't have to > force the user to do diverts from a web page Let us know how it goes and if you have any questions. I'm sure that we can help you nail this one down. -MC From krice at freeswitch.org Sat Feb 7 10:26:59 2009 From: krice at freeswitch.org (Ken Rice) Date: Sat, 07 Feb 2009 12:26:59 -0600 Subject: [Freeswitch-users] need suggestion on using mod_easyroute In-Reply-To: <27c25bc40902070844h5db40f22x26bcfce9ff21e852@mail.gmail.com> Message-ID: Mod_easyroute can handle millions of numbers... It is NOT however an LCR module... There are other things for that... Look at mod_lcr or if you need an extremely high performance LCR contact me off list From: Juan Backson Reply-To: Date: Sun, 8 Feb 2009 00:44:16 +0800 To: Subject: [Freeswitch-users] need suggestion on using mod_easyroute Hi, I notice that there is a newly-developed mod_easyroute model available. Has anyone used it with large amount of routes ( ex > 1M ) on a high traffic scenario? For that kind of scenario, would it be better to consider using out-going event socket to serve that purpose? I would greatly appreciate any recommendation. Thanks, JB _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090207/d8278e7c/attachment-0002.html From brian at freeswitch.org Sat Feb 7 10:30:31 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 7 Feb 2009 12:30:31 -0600 Subject: [Freeswitch-users] Global Variables forgotten through thepubliccontext? In-Reply-To: References: <0B4E2726927041D09D0425DA0242C805@UVix><9D0CFA79-C120-4D66-B48C-7A5641B07AF4@freeswitch.org><5B1ED1D4834B4719B309FAC03F1595C0@UVix> Message-ID: <03BD385F-E942-4B32-98B0-0A29E13BC5D5@freeswitch.org> Its trying to open the file to match the current channel rate... you can install the 16k files via "make hd-sounds-install hd-moh-install" /b On Feb 7, 2009, at 9:26 AM, UV wrote: > Will do. > Now I have a little problem delaying me as the latest build changed > something with the sound file playing and now the FS can?t find any > local file to play (adds mysterious \16000\ to the file location?). > I?ll try to isolate this first then get back to this issue. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090207/65fe3943/attachment-0002.html From anthony.minessale at gmail.com Sat Feb 7 13:23:54 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sat, 7 Feb 2009 15:23:54 -0600 Subject: [Freeswitch-users] mod_shout delay in trunk In-Reply-To: <498CAD73.2040709@gmail.com> References: <15830211.131233952452225.JavaMail.root@zimbra> <28165623.151233952574612.JavaMail.root@zimbra> <191c3a030902061248r3d307647le5574c1004c6da24@mail.gmail.com> <498CAD73.2040709@gmail.com> Message-ID: <191c3a030902071323w562c1f2cg9a7aeebb42781c8f@mail.gmail.com> tamas, the opposite. The default is to not do one by one and setting the var to false makes it more i/o intensive but it would provide more real time recording when recording to streams. BTW the reversed logic is fixed in tree On Fri, Feb 6, 2009 at 3:36 PM, Tamas wrote: > Hello, > > could this option be used to lower I/O load - to rather write more bytes > at once rather than one by one - on file recording (record_session)? > > Regards, > Tamas > > Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090207/b3c6be65/attachment-0002.html From uv at yuvalhertzog.com Sat Feb 7 16:46:51 2009 From: uv at yuvalhertzog.com (UV) Date: Sun, 8 Feb 2009 11:46:51 +1100 Subject: [Freeswitch-users] Global Variables forgotten throughthepubliccontext? In-Reply-To: <03BD385F-E942-4B32-98B0-0A29E13BC5D5@freeswitch.org> References: <0B4E2726927041D09D0425DA0242C805@UVix><9D0CFA79-C120-4D66-B48C-7A5641B07AF4@freeswitch.org><5B1ED1D4834B4719B309FAC03F1595C0@UVix> <03BD385F-E942-4B32-98B0-0A29E13BC5D5@freeswitch.org> Message-ID: <9F926F21CCB64A49A6EFF4425B111F67@UVix> Yeah, I have all the sounds installed. I don?t think it?s that. I?m getting error messages such as ?[ERR] mod_sndfile.c:185 sndfile_file_open() Error Opening File [E:\FS/sounds/en/us/callie\voicemail/8000\16000\vm-goodbye.w] [System error : The system cannot find the path specified.]? all across the board. The only thing still working is MoH? This started from one of yesterday?s builds (r11665 ? 11678). _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Sunday, February 08, 2009 5:31 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Global Variables forgotten throughthepubliccontext? Its trying to open the file to match the current channel rate... you can install the 16k files via "make hd-sounds-install hd-moh-install" /b On Feb 7, 2009, at 9:26 AM, UV wrote: Will do. Now I have a little problem delaying me as the latest build changed something with the sound file playing and now the FS can?t find any local file to play (adds mysterious \16000\ to the file location?). I?ll try to isolate this first then get back to this issue. No virus found in this incoming message. Checked by AVG i www.avg.com Version: 8.0.233 / Virus Database: 270.10.18/1937 i Release Date: 02/06/09 11:31:00 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090208/0cab0873/attachment-0002.html From brian at freeswitch.org Sat Feb 7 17:56:50 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 7 Feb 2009 19:56:50 -0600 Subject: [Freeswitch-users] Global Variables forgotten throughthepubliccontext? In-Reply-To: <9F926F21CCB64A49A6EFF4425B111F67@UVix> References: <0B4E2726927041D09D0425DA0242C805@UVix><9D0CFA79-C120-4D66-B48C-7A5641B07AF4@freeswitch.org><5B1ED1D4834B4719B309FAC03F1595C0@UVix> <03BD385F-E942-4B32-98B0-0A29E13BC5D5@freeswitch.org> <9F926F21CCB64A49A6EFF4425B111F67@UVix> Message-ID: <51C9B370-500D-4D95-A515-E9EEF1705014@freeswitch.org> You have a \ somewhere in your path... which doesn't make sense... you're on windows. Can you open a jira... I think this was the cause http://fisheye.freeswitch.org/browse/FreeSWITCH/src/mod/formats/mod_sndfile/mod_sndfile.c?r1=11090&r2=11601 /b On Feb 7, 2009, at 6:46 PM, UV wrote: > Yeah, I have all the sounds installed. I don?t think it?s that. > I?m getting error messages such as ?[ERR] mod_sndfile.c:185 > sndfile_file_open() Error Opening File [E:\FS/sounds/en/us/callie > \voicemail/8000\16000\vm-goodbye.w] [System error : The system > cannot find the path specified.]? all across the board. The only > thing still working is MoH? This started from one of yesterday?s > builds (r11665 ? 11678). -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090207/4553b646/attachment-0002.html From woodydickson at gmail.com Sat Feb 7 18:24:19 2009 From: woodydickson at gmail.com (Woody Dickson) Date: Sun, 8 Feb 2009 10:24:19 +0800 Subject: [Freeswitch-users] Seeking opinion on shared disk space Message-ID: Hi, In my deployment scenario, I plan to have two redundant freeswitch servers running on two different boxes. Two key features I am leveraging on freeswitch are voicemail and call recording and playback., and as a result of that, a shared storage for playback of the recorded wav files is needed. When the user traffic is high, I am affraid that NFS or even GFS can't scale well. On the other hand, a real SAN hardware with optical-fabric is too expensive for us. We are therefore considering using iSCSI SAN to build a cheap SAN for that purpose. Does anyone have experience setting up a shared storage between multiple freeswitch servers and can share some inputs with me? Thanks for all your help. Woody From krice at freeswitch.org Sat Feb 7 18:34:52 2009 From: krice at freeswitch.org (Ken Rice) Date: Sat, 07 Feb 2009 20:34:52 -0600 Subject: [Freeswitch-users] Seeking opinion on shared disk space In-Reply-To: Message-ID: Whats wrong with NFS? As long as you put a reasonable disk subsystem you'll be fine... GFS sucks for voice anyway... It can take several seconds to get a lock... No matter what you use, you have to remember that you *MUST* have a cluster aware file system, simply mounting the same iscsi or SAN LUN on 2 different boxes running ext3 won't work since things aren't guaranteed to be flushed until a sync is called > From: Woody Dickson > Reply-To: > Date: Sun, 8 Feb 2009 10:24:19 +0800 > To: > Subject: [Freeswitch-users] Seeking opinion on shared disk space > > Hi, > > In my deployment scenario, I plan to have two redundant freeswitch > servers running on two different boxes. Two key features I am > leveraging on freeswitch are voicemail and call recording and > playback., and as a result of that, a shared storage for playback of > the recorded wav files is needed. When the user traffic is high, I am > affraid that NFS or even GFS can't scale well. On the other hand, a > real SAN hardware with optical-fabric is too expensive for us. We are > therefore considering using iSCSI SAN to build a cheap SAN for that > purpose. > > Does anyone have experience setting up a shared storage between > multiple freeswitch servers and can share some inputs with me? > > Thanks for all your help. > > Woody > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From john at whitesmiths.com Sat Feb 7 13:25:32 2009 From: john at whitesmiths.com (John O'Brien) Date: Sun, 8 Feb 2009 08:25:32 +1100 Subject: [Freeswitch-users] Newbie - point me in the right direction Message-ID: <008F70F6-818D-476B-9DE9-33DA018D37D9@whitesmiths.com> Hi, I am a real newbie. I have been building Asterisk based applications for a couple of years now. I am looking at migrating these apps to FreeSwitch - eventually. I want to do this gradually - I need to keep things running in the meantime. I have two Asterisk boxes, A1 & A2, each running a separate telephony app. We have an external SIP service with DID's NNNNN200 -> NNNNN299. We want to direct the incoming SIP calls so that the DID's NNNNN200 -> NNNNN219 go to Asterisk server A1 and NNNNN220 -> NNNNN299 to Asterisk server A2. Yes we really just want the calls switched on the DID. I'm struggling to know where to start - can someone point me in the right direction? Regards, John From pauld at versafon.com Sat Feb 7 21:03:59 2009 From: pauld at versafon.com (Paul D.) Date: Sun, 08 Feb 2009 00:03:59 -0500 Subject: [Freeswitch-users] Cannot choose Cepstral voice from dialplan Message-ID: <498E67BF.3060207@versafon.com> Followed Wiki to install and configure mod_cepstral. The problem is FS always defaults to one voice, which I installed first, and ignores others. I did define SWIFT_HOME and added swift lib path to /etc/ld.so.conf. After I restart FS I see on FS console after dialing my test extension: Failed to load library libceplex_us.so due to: libceplex_us.so: cannot open shared object file: No such file or directory Failed to load language / lexical libraries for Callie-8kHz I do have this voice installed. This message does not appear after subsequent calls to the test extension, until next FS restart. I use this in my dialplan: This is all under centos 5.2 64 bit. Any suggestions would be greatly appreciated. From nik.middleton at noblesolutions.co.uk Sun Feb 8 08:21:22 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Sun, 8 Feb 2009 16:21:22 -0000 Subject: [Freeswitch-users] Struggling with Originate Message-ID: Hi Guys, I'm placing calls ok by using the event socket. However, I need to modify the To: Sip header prior to the call going out for routing purposes. Is it possible to do this in the Originate action? If not, can someone explain if it's possible to trigger part of the dial plan externally? I can then modify the headers and then place the call/ Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090208/b1b9cea1/attachment-0002.html From dave at 3c.co.uk Sun Feb 8 09:09:09 2009 From: dave at 3c.co.uk (David Knell) Date: Sun, 8 Feb 2009 17:09:09 +0000 Subject: [Freeswitch-users] Struggling with Originate In-Reply-To: References: Message-ID: <4BFB326A-23AE-4768-9FBA-9774EC0FC34A@3c.co.uk> Hi Nik, How do you need to modify it? Cheers -- Dave > Hi Guys, > > I?m placing calls ok by using the event socket. However, I need to > modify the To: Sip header prior to the call going out for routing > purposes. Is it possible to do this in the Originate action? > > If not, can someone explain if it?s possible to trigger part of the > dial plan externally? I can then modify the headers and then place > the call/ > > > Regards, > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090208/6f980746/attachment-0002.html From moises.silva at gmail.com Sun Feb 8 09:36:18 2009 From: moises.silva at gmail.com (Moises Silva) Date: Sun, 8 Feb 2009 11:36:18 -0600 Subject: [Freeswitch-users] Transcoding G723 In-Reply-To: <35b355e90902060716v2dff2586t253a69c97ab1adac@mail.gmail.com> References: <2FEEE7F7-DCE7-4919-A0FD-A0042BBB7EEE@freeswitch.org> <35b355e90902051226g12756d2fsf85fb996ec1ed188@mail.gmail.com> <11C3C72A-247D-4F8C-B106-52D49949D026@freeswitch.org> <1b46b4e80902051252o1b5a5c75l9f8d242cbaeec552@mail.gmail.com> <35b355e90902051301r4c2c53b6tb958774f4754e60f@mail.gmail.com> <35b355e90902060716v2dff2586t253a69c97ab1adac@mail.gmail.com> Message-ID: It seems I know the same you know, it was on the works then not, then back in the works. However I don't know the status on that. If you have contact with Doug he is a better person to ask to regarding new products coming out. Moy On Fri, Feb 6, 2009 at 9:16 AM, Shelby Ramsey wrote: > Thanks Moises. It looks like good work. When is Sangoma coming out with a > similar product ... Doug told me it was in the works, then not in the works, > then back in the works ... > The problem is this particular card is PCI only and it will only do 120 > channels .... > Thanks! > SDR > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- "I do not agree with what you have to say, but I'll defend to the death your right to say it." Voltaire From msc at freeswitch.org Sun Feb 8 14:11:00 2009 From: msc at freeswitch.org (Michael S Collins) Date: Sun, 8 Feb 2009 14:11:00 -0800 Subject: [Freeswitch-users] Cannot choose Cepstral voice from dialplan In-Reply-To: <498E67BF.3060207@versafon.com> References: <498E67BF.3060207@versafon.com> Message-ID: <8B964172-6EEB-4B5B-B4B1-D35BFEAEB460@freeswitch.org> Sent from my iPhone On Feb 7, 2009, at 9:03 PM, "Paul D." wrote: > Followed Wiki to install and configure mod_cepstral. The problem is FS > always defaults to one voice, which I installed first, and ignores > others. > I did define SWIFT_HOME and added swift lib path to /etc/ld.so.conf. > After I restart FS I see on FS console after dialing my test > extension: > > Failed to load library libceplex_us.so due to: libceplex_us.so: cannot > open shared object file: No such file or directory > Failed to load language / lexical libraries for Callie-8kHz > Look in /opt/swift for the dir that has the lib files. Most likely you'll find that they have the files are but have more specific names, perhaps with a version number. If you create symlinks to the files in question it should work. -MC > I do have this voice installed. This message does not appear after > subsequent calls to the test extension, until next FS restart. > I use this in my dialplan: > > > > > > > > > > This is all under centos 5.2 64 bit. Any suggestions would be greatly > appreciated. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From nik.middleton at noblesolutions.co.uk Sun Feb 8 14:31:35 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Sun, 8 Feb 2009 22:31:35 -0000 Subject: [Freeswitch-users] Problems passing arguments to lua Message-ID: Hi Guys, I'm having some issues passing an argument to an lua script. Basically in an originate command I pass the name of a .wav file If I hard code the file session:streamFile("myfile.wav"]); It works, However, using this: session:streamFile(argv[1]); causes this error 2009-02-08 22:09:07 [ERR] mod_lua.cpp:176 lua_parse_and_execute() cannot open /usr/local/freeswitch/scripts/helloworld.lua,myfile.wav: No such file or directory Any Ideas? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090208/c4f6b7b5/attachment-0002.html From nik.middleton at noblesolutions.co.uk Sun Feb 8 14:41:29 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Sun, 8 Feb 2009 22:41:29 -0000 Subject: [Freeswitch-users] connecting to mysql using lua Message-ID: Hi Guys I want to access Mysql 5 from lua. The wiki is not too clear on this. Do I need to install lua and lua mysql? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090208/250993c5/attachment-0002.html From pauld at versafon.com Sun Feb 8 15:14:25 2009 From: pauld at versafon.com (pauld) Date: Sun, 08 Feb 2009 18:14:25 -0500 Subject: [Freeswitch-users] Cannot choose Cepstral voice from dialplan In-Reply-To: <8B964172-6EEB-4B5B-B4B1-D35BFEAEB460@freeswitch.org> References: <498E67BF.3060207@versafon.com> <8B964172-6EEB-4B5B-B4B1-D35BFEAEB460@freeswitch.org> Message-ID: <498F6751.2000701@versafon.com> The libs are there with correct symlinks, see below. I tested both voices directly via swift command, works fine. Any other ideas? It's Cepstral 5.1, FS 1.0.2. ls -la /opt/swift/lib total 4352 drwxr-xr-x 2 root root 4096 Feb 7 22:59 . drwxr-xr-x 10 root root 4096 Feb 7 12:29 .. lrwxrwxrwx 1 999 20202 20 Feb 7 22:59 libceplang_en.so -> libceplang_en.so.5.1 lrwxrwxrwx 1 999 20202 20 Feb 7 22:59 libceplang_en.so.5 -> libceplang_en.so.5.1 -rwxrwxr-x 1 999 20202 412050 Jul 8 2008 libceplang_en.so.5.1 lrwxrwxrwx 1 999 20202 19 Feb 7 12:29 libceplex_uk.so -> libceplex_uk.so.5.1 lrwxrwxrwx 1 999 20202 19 Feb 7 12:29 libceplex_uk.so.5 -> libceplex_uk.so.5.1 -rwxrwxr-x 1 999 20202 904994 Jul 8 2008 libceplex_uk.so.5.1 lrwxrwxrwx 1 999 20202 19 Feb 7 22:59 libceplex_us.so -> libceplex_us.so.5.1 lrwxrwxrwx 1 999 20202 19 Feb 7 22:59 libceplex_us.so.5 -> libceplex_us.so.5.1 -rwxrwxr-x 1 999 20202 1009780 Jul 8 2008 libceplex_us.so.5.1 lrwxrwxrwx 1 999 20202 15 Feb 7 22:59 libswift.so -> libswift.so.5.1 lrwxrwxrwx 1 999 20202 15 Feb 7 22:59 libswift.so.5 -> libswift.so.5.1 -rwxrwxr-x 1 999 20202 2100418 Jul 8 2008 libswift.so.5.1 Michael S Collins wrote: > Sent from my iPhone > > On Feb 7, 2009, at 9:03 PM, "Paul D." wrote: > > >> Followed Wiki to install and configure mod_cepstral. The problem is FS >> always defaults to one voice, which I installed first, and ignores >> others. >> I did define SWIFT_HOME and added swift lib path to /etc/ld.so.conf. >> After I restart FS I see on FS console after dialing my test >> extension: >> >> Failed to load library libceplex_us.so due to: libceplex_us.so: cannot >> open shared object file: No such file or directory >> Failed to load language / lexical libraries for Callie-8kHz >> >> > > Look in /opt/swift for the dir that has the lib files. Most likely > you'll find that they have the files are but have more specific names, > perhaps with a version number. If you create symlinks to the files in > question it should work. > > -MC > > >> I do have this voice installed. This message does not appear after >> subsequent calls to the test extension, until next FS restart. >> I use this in my dialplan: >> >> >> >> >> >> >> >> >> >> This is all under centos 5.2 64 bit. Any suggestions would be greatly >> appreciated. >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From msc at freeswitch.org Sun Feb 8 15:20:50 2009 From: msc at freeswitch.org (Michael S Collins) Date: Sun, 8 Feb 2009 15:20:50 -0800 Subject: [Freeswitch-users] Problems passing arguments to lua In-Reply-To: References: Message-ID: <6E948AB3-1030-46C5-8E08-3AB74653F0AE@freeswitch.org> Print out the variable to make sure it is what you expect: io.write("argv is " .. argv[1] .. "\n"; Also, if you don't give the sound file an absolute path name then it will automatically use the sound dir path. -MC Sent from my iPhone On Feb 8, 2009, at 2:31 PM, "Nik Middleton" wrote: > Hi Guys, > > > > I?m having some issues passing an argument to an lua script. > > > > Basically in an originate command I pass the name of a .wav file > > > > If I hard code the file session:streamFile(?myfile.wav?]); > > > > It works, > > > > However, using this: > > > > session:streamFile(argv[1]); > > > > causes this error > > > > 2009-02-08 22:09:07 [ERR] mod_lua.cpp:176 lua_parse_and_execute() > cannot open /usr/local/freeswitch/scripts/helloworld.lua,myfile.wav: > No such file or directory > > > > Any Ideas? > > > > Regards > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090208/b7543fd6/attachment-0002.html From msc at freeswitch.org Sun Feb 8 15:27:19 2009 From: msc at freeswitch.org (Michael S Collins) Date: Sun, 8 Feb 2009 15:27:19 -0800 Subject: [Freeswitch-users] connecting to mysql using lua In-Reply-To: References: Message-ID: <78A54C9A-C807-445B-A3BA-0B09A3B27521@freeswitch.org> Nik, I see your point about the wiki entry regarding luasql. If someone could clarify then I will be happy to help get the wiki documentation updated appropriately. -MC Sent from my iPhone On Feb 8, 2009, at 2:41 PM, "Nik Middleton" wrote: > Hi Guys > > > > I want to access Mysql 5 from lua. The wiki is not too clear on > this. Do I need to install lua and lua mysql? > > > > Regards > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090208/9fa1eafc/attachment-0002.html From nik.middleton at noblesolutions.co.uk Sun Feb 8 16:21:28 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Mon, 9 Feb 2009 00:21:28 -0000 Subject: [Freeswitch-users] Problems passing arguments to lua In-Reply-To: <6E948AB3-1030-46C5-8E08-3AB74653F0AE@freeswitch.org> References: <6E948AB3-1030-46C5-8E08-3AB74653F0AE@freeswitch.org> Message-ID: Done that, still doesn't work. My guess is "related Regards, ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael S Collins Sent: 08 February 2009 23:21 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Problems passing arguments to lua Print out the variable to make sure it is what you expect: io.write("argv is " .. argv[1] .. "\n"; Also, if you don't give the sound file an absolute path name then it will automatically use the sound dir path. -MC Sent from my iPhone On Feb 8, 2009, at 2:31 PM, "Nik Middleton" wrote: Hi Guys, I'm having some issues passing an argument to an lua script. Basically in an originate command I pass the name of a .wav file If I hard code the file session:streamFile("myfile.wav"]); It works, However, using this: session:streamFile(argv[1]); causes this error 2009-02-08 22:09:07 [ERR] mod_lua.cpp:176 lua_parse_and_execute() cannot open /usr/local/freeswitch/scripts/helloworld.lua,myfile.wav: No such file or directory Any Ideas? Regards _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090209/3aec6f2d/attachment-0002.html From brian at freeswitch.org Sun Feb 8 16:54:46 2009 From: brian at freeswitch.org (Brian West) Date: Sun, 8 Feb 2009 18:54:46 -0600 Subject: [Freeswitch-users] Struggling with Originate In-Reply-To: <4BFB326A-23AE-4768-9FBA-9774EC0FC34A@3c.co.uk> References: <4BFB326A-23AE-4768-9FBA-9774EC0FC34A@3c.co.uk> Message-ID: If you're wanting to modify the invite domain you can do that via the sip_invite_domain variable either export it before the bridge or place it inside {} on the originate line... ie "{sip_invite_domain=example.com}sofia/blah/blah" /b On Feb 8, 2009, at 11:09 AM, David Knell wrote: > Hi Nik, > > How do you need to modify it? > > Cheers -- > > Dave From brian at freeswitch.org Sun Feb 8 16:55:40 2009 From: brian at freeswitch.org (Brian West) Date: Sun, 8 Feb 2009 18:55:40 -0600 Subject: [Freeswitch-users] Problems passing arguments to lua In-Reply-To: References: <6E948AB3-1030-46C5-8E08-3AB74653F0AE@freeswitch.org> Message-ID: <31920E97-233D-4331-9F07-B09E29B592CD@freeswitch.org> Looks like you put a , instead of a space when calling the script. /b On Feb 8, 2009, at 6:21 PM, Nik Middleton wrote: > cannot open /usr/local/freeswitch/scripts/helloworld.lua,myfile.wav -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090208/38935482/attachment-0002.html From dougblackstone at gmail.com Sun Feb 8 19:21:49 2009 From: dougblackstone at gmail.com (Doug Blacksone) Date: Mon, 9 Feb 2009 11:21:49 +0800 Subject: [Freeswitch-users] Dynamic Dialplan Message-ID: <743112140902081921n2bb1bf49mfe36e5e2193c6b04@mail.gmail.com> Hi, Right now, I am working on getting freeswitch configured for our call-center with more than 1000 agents. There are several areas where we need the dialplan to be configurable based on some user detail in the database. Therefore, the dialplan needs to be some-what dynamic based-on inputs from the database. I would like to know from other implementation as to the most scalable way of doing high performance dynamic dialplan that is super scalable. There are three ways I can think of: 1. Static dialplan using customized freeswitch mod to access postgres for data pros: best performance cons: harder to program 2. Static dialplan using lua to access postgres for data pros: easy to program, maybe-performance is better than curl cons: need to search through all the extensions to find one dialplan, performance is slower than the first one. 3. curl-based dialplan using Java Servlet and HTTP pros: easy to program, freeswitch only gets one extension and no extension search cons: performance is slow than the other two Is this a correct analysis? If from a pure performance's perspective, how much performance can a customized mod gains in comparison to lua? For a production system that needs to be highly scalable, what do you recommend? Thank you very much for any input to our critical design decision. Doug -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090209/ec2253c8/attachment-0002.html From krice at freeswitch.org Sun Feb 8 19:32:18 2009 From: krice at freeswitch.org (Ken Rice) Date: Sun, 08 Feb 2009 21:32:18 -0600 Subject: [Freeswitch-users] Dynamic Dialplan In-Reply-To: <743112140902081921n2bb1bf49mfe36e5e2193c6b04@mail.gmail.com> Message-ID: If you want an extremely high performance you write your own dialplan module... Its not that hard... Or option 1 is the more high performance way to fo... Curl with a serverlet will scale to a point but I doubt it will get to where you need in the long run Static and do what you need, but how scalable is lua is unknown at this point (someone should post some info on that) and its known that java script doesn?t scale that well for large installs K From: Doug Blacksone Reply-To: Date: Mon, 9 Feb 2009 11:21:49 +0800 To: Subject: [Freeswitch-users] Dynamic Dialplan Hi, Right now, I am working on getting freeswitch configured for our call-center with more than 1000 agents. There are several areas where we need the dialplan to be configurable based on some user detail in the database. Therefore, the dialplan needs to be some-what dynamic based-on inputs from the database. I would like to know from other implementation as to the most scalable way of doing high performance dynamic dialplan that is super scalable. There are three ways I can think of: 1. Static dialplan using customized freeswitch mod to access postgres for data pros: best performance cons: harder to program 2. Static dialplan using lua to access postgres for data pros: easy to program, maybe-performance is better than curl cons: need to search through all the extensions to find one dialplan, performance is slower than the first one. 3. curl-based dialplan using Java Servlet and HTTP pros: easy to program, freeswitch only gets one extension and no extension search cons: performance is slow than the other two Is this a correct analysis? If from a pure performance's perspective, how much performance can a customized mod gains in comparison to lua? For a production system that needs to be highly scalable, what do you recommend? Thank you very much for any input to our critical design decision. Doug _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090208/6ffdb348/attachment-0002.html From sicfslist at gmail.com Sun Feb 8 20:20:22 2009 From: sicfslist at gmail.com (Shelby Ramsey) Date: Sun, 8 Feb 2009 22:20:22 -0600 Subject: [Freeswitch-users] Dynamic Dialplan In-Reply-To: References: <743112140902081921n2bb1bf49mfe36e5e2193c6b04@mail.gmail.com> Message-ID: <35b355e90902082020s274fb805r7197a77e522b36f4@mail.gmail.com> Doug, Ken is right on this one. I know there are some guys on the list (like Ken) that could help you write a module. It's probably the best way to go (if you're going to have all agents running off of one or two boxes). If you're going to spread the agents / calls around on multiple boxes or use a combo of OpenSIPS / OpenSer / pick your flavor ... and FS then using xml_curl will work fine. We've got that working today and it's been acceptable. SDR -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090208/34593918/attachment-0002.html From krice at suspicious.org Sun Feb 8 20:50:31 2009 From: krice at suspicious.org (Ken Rice) Date: Sun, 08 Feb 2009 22:50:31 -0600 Subject: [Freeswitch-users] Dynamic Dialplan In-Reply-To: <35b355e90902082020s274fb805r7197a77e522b36f4@mail.gmail.com> Message-ID: Also depending on what your Timeframe is like there is a distributed queue mechanism with skills based routing on the way... From: Shelby Ramsey Reply-To: Date: Sun, 8 Feb 2009 22:20:22 -0600 To: Subject: Re: [Freeswitch-users] Dynamic Dialplan Doug, Ken is right on this one. I know there are some guys on the list (like Ken) that could help you write a module. It's probably the best way to go (if you're going to have all agents running off of one or two boxes). If you're going to spread the agents / calls around on multiple boxes or use a combo of OpenSIPS / OpenSer / pick your flavor ... and FS then using xml_curl will work fine. We've got that working today and it's been acceptable. SDR _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090208/d0d1e752/attachment-0002.html From andrew at hijacked.us Sun Feb 8 21:16:47 2009 From: andrew at hijacked.us (Andrew Thompson) Date: Mon, 9 Feb 2009 00:16:47 -0500 Subject: [Freeswitch-users] Dynamic Dialplan In-Reply-To: References: <35b355e90902082020s274fb805r7197a77e522b36f4@mail.gmail.com> Message-ID: <20090209051646.GD4963@hijacked.us> On Sun, Feb 08, 2009 at 10:50:31PM -0600, Ken Rice wrote: > Also depending on what your Timeframe is like there is a distributed queue > mechanism with skills based routing on the way... > It even managed to route 2 calls in a row this week ;) Still a ways off from anything production grade tho. Andrew From wasim at convergence.pk Sun Feb 8 21:14:41 2009 From: wasim at convergence.pk (Wasim Baig) Date: Mon, 9 Feb 2009 10:14:41 +0500 Subject: [Freeswitch-users] Dynamic Dialplan In-Reply-To: <20090209051646.GD4963@hijacked.us> References: <35b355e90902082020s274fb805r7197a77e522b36f4@mail.gmail.com> <20090209051646.GD4963@hijacked.us> Message-ID: On Mon, Feb 9, 2009 at 10:16 AM, Andrew Thompson wrote: > On Sun, Feb 08, 2009 at 10:50:31PM -0600, Ken Rice wrote: > > Also depending on what your Timeframe is like there is a distributed > queue > > mechanism with skills based routing on the way... > > > It even managed to route 2 calls in a row this week ;) Still a ways off > from anything production grade tho. Bravo, bravo! -- wasim h. baig | principal consultant | convergence pk | +92 300 8508070 | peace be upon you ... -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090209/2e57bf53/attachment-0002.html From kristian.kielhofner at gmail.com Sun Feb 8 23:15:07 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Mon, 9 Feb 2009 02:15:07 -0500 Subject: [Freeswitch-users] Mod_native_file FreeSWITCH files (and script) available Message-ID: <2d9149cd0902082315l73703eb2t79d0bbb96845ff49@mail.gmail.com> Hello everyone, Now that I've got FreeSWITCH compiling under AstLinux I'm starting to look at ways to optimize FreeSWITCH. First things first: minimize transcoding. I hate transcoding. I modified a concept I came up with back in the day for Asterisk. I've created a script to convert WAV files to FreeSWITCH mod_native_file formats for various codecs. Of course I also updated the wiki: http://wiki.freeswitch.org/wiki/Mod_native_file#Script_to_convert_a_sound_file_to_specific_formats_to_avoid_transcoding As you can see I've also converted the current FreeSWITCH prompts and MOH into various file formats (including G.729 and G.723). These files have undergone very little testing (basically none) so I would appreciate some feedback on them. I'm also working on G.729, iLBC, speex (VBR?!?), and G.722. What about Siren, CELT, and some of the others? I don't have much experience with these and I could certainly use some help. P.S. - I'm currently using sox and Asterisk res_convert to do the conversion. I don't know how else to legitimately (and consistently) convert to these various formats. Without bringing up the usual G.729/G.723 discussions, I'm in the US and this is how I could legally do all of these conversions. -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From helmut.kuper at ewetel.de Sun Feb 8 23:48:13 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Mon, 09 Feb 2009 08:48:13 +0100 Subject: [Freeswitch-users] Openzap r654 doesn't compile. Error with ozmod_libpri Message-ID: <498FDFBD.8050401@ewetel.de> Hello, during some imporvements on q931toPcap as well as debugging my TDM PRI problem with loosing state sync after some time I updated the code to latest trunk (r654). After doing a bootstrap in FS trunk directory I tried to compile openzap in libs/openzap. I got this error: /bin/sh ./libtool --tag=CC --mode=link gcc -g -O2 -o ozmod_libpri.la -rpath /usr/local/openzap/mod -lm -lpthread -ldl libtool: link: libtool library `ozmod_libpri.la' must begin with `lib' Try `libtool --help --mode=link' for more information. make: *** [ozmod_libpri.la] Error 1 Additionally I wonder what ozmod_libpri is. Do I have consider it for Q931 capturing? regards Helmut From helmut.kuper at ewetel.de Sun Feb 8 23:54:04 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Mon, 09 Feb 2009 08:54:04 +0100 Subject: [Freeswitch-users] Can I force the A-leg codec in FS? Message-ID: <498FE11C.2070105@ewetel.de> Hello, in my reallife setup of FS all internal extensions use G.722 as preferred codec. Unfortunately when there is an outgoing TDM call I found that FS starts transcoding instead of forcing G.711 for A leg. So is there a way to force the A codec? regards helmut From krice at suspicious.org Mon Feb 9 00:00:18 2009 From: krice at suspicious.org (Ken Rice) Date: Mon, 09 Feb 2009 02:00:18 -0600 Subject: [Freeswitch-users] Can I force the A-leg codec in FS? In-Reply-To: <498FE11C.2070105@ewetel.de> Message-ID: Set the codec negotiation to greedy > From: Helmut Kuper > Reply-To: > Date: Mon, 09 Feb 2009 08:54:04 +0100 > To: > Subject: [Freeswitch-users] Can I force the A-leg codec in FS? > > Hello, > > in my reallife setup of FS all internal extensions use G.722 as > preferred codec. Unfortunately when there is an outgoing TDM call I > found that FS starts transcoding instead of forcing G.711 for A leg. So > is there a way to force the A codec? > > regards > helmut > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From helmut.kuper at ewetel.de Mon Feb 9 00:26:27 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Mon, 09 Feb 2009 09:26:27 +0100 Subject: [Freeswitch-users] Can I force the A-leg codec in FS? In-Reply-To: References: Message-ID: <498FE8B3.8040904@ewetel.de> Hi Ken, thx for the hint. It looks quite static, so I guess each call (also internal) are then forced to g711. I'm looking for a dynamic way depending on destination number. regards helmut On 09.02.2009 09:00, Ken Rice wrote: > Set the codec negotiation to greedy > > > From helmut.kuper at ewetel.de Mon Feb 9 00:27:33 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Mon, 09 Feb 2009 09:27:33 +0100 Subject: [Freeswitch-users] Openzap r654 doesn't compile. Error with ozmod_libpri In-Reply-To: <498FDFBD.8050401@ewetel.de> References: <498FDFBD.8050401@ewetel.de> Message-ID: <498FE8F5.9060206@ewetel.de> Hello, update, when I remove all ozmod_ from ozmod_libpri lines in Makefile, it compiles without errors. regards helmut On 09.02.2009 08:48, Helmut Kuper wrote: > Hello, > > during some imporvements on q931toPcap as well as debugging my TDM PRI > problem with loosing state sync after some time I updated the code to > latest trunk (r654). After doing a bootstrap in FS trunk directory I > tried to compile openzap in libs/openzap. I got this error: > > /bin/sh ./libtool --tag=CC --mode=link gcc -g -O2 -o > ozmod_libpri.la -rpath /usr/local/openzap/mod -lm -lpthread -ldl > libtool: link: libtool library `ozmod_libpri.la' must begin with `lib' > Try `libtool --help --mode=link' for more information. > make: *** [ozmod_libpri.la] Error 1 > > Additionally I wonder what ozmod_libpri is. Do I have consider it for > Q931 capturing? > > regards > Helmut > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > -- Mit freundlichen Gr??en Helmut Kuper Finanzdienstleistungen und Entwicklung Telefax: (0441) 8000-2799 mailto:helmut.kuper at ewetel.de ___________________________________ EWE TEL GmbH Cloppenburger Stra?e 310 26133 Oldenburg EWE TEL GmbH Handelsregister Amtsgericht Oldenburg HRB 3723 Vorsitzender des Aufsichtsrates: Heiko Harms Gesch?ftsf?hrung: Hans-Joachim Iken (Vorsitzender), Dr. Norbert Schulz, Dirk Thole Homepage: http://www.ewetel.de ___________________________________ From nik.middleton at noblesolutions.co.uk Mon Feb 9 00:56:00 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Mon, 9 Feb 2009 08:56:00 -0000 Subject: [Freeswitch-users] Problems passing arguments to lua In-Reply-To: <31920E97-233D-4331-9F07-B09E29B592CD@freeswitch.org> References: <6E948AB3-1030-46C5-8E08-3AB74653F0AE@freeswitch.org> <31920E97-233D-4331-9F07-B09E29B592CD@freeswitch.org> Message-ID: That and not enclosing in single quotes, thanks Regards, ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: 09 February 2009 00:56 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Problems passing arguments to lua Looks like you put a , instead of a space when calling the script. /b On Feb 8, 2009, at 6:21 PM, Nik Middleton wrote: cannot open /usr/local/freeswitch/scripts/helloworld.lua,myfile.wav -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090209/35d592a8/attachment-0002.html From ahgindia308 at gmail.com Mon Feb 9 01:46:17 2009 From: ahgindia308 at gmail.com (Ankit Gandhi) Date: Mon, 9 Feb 2009 01:46:17 -0800 (PST) Subject: [Freeswitch-users] Freeswitch not processing calls Message-ID: <21909687.post@talk.nabble.com> Recently I noticed that fs (1.0.3 RC1) is not processing the calls. Previously about 50-60 calls were remaining active at any time. But then suddenly, only 5-6 active calls were there. So I checked sip INVITE's in ngrep, and I noticed that originator were sending CANCEL after INVITE. So I contacted them and they were telling that while sending calls, your gateways time outs after a call is sent, and so they send CANCEL after INVITE. My architecture is as follows : originator -> switch (our fs) -> terminator. CPU : Intel(R) Xeon(R) CPU X3220 @ 2.40GHz OS : Ubuntu 6.06.1 LTS 32-bit RAM : 4 GB We had also set ulimits on command prompt every time before starting fs with nc mode as follows : ulimit -c unlimited; ulimit -n 999999; ulimit -s 244; /usr/local/freeswitch/bin/freeswitch -nc In the ngrep trace, I noticed that fs was not sending INVITE to terminator in between the INVITE and CANCEL from originator. So what could be the reason for INVITE not being processed by fs and getting timeout from originator. Here is the sip trace for a sample call. Check the duration between INVITE and CANCEL from originator. -------------------------------------------------------------------- U 2009/02/07 13:16:44.354443 ori.ori.ori.ori:2000 -> fs.fs.fs.fs:5060 INVITE sip:yyyyyyyyyyyy at fs.fs.fs.fs:5060;user=phone SIP/2.0. Call-ID: 7008117790783130205-1234012604 at ori.ori.ori.ori. From: ;tag=13475. To: . Content-Type: application/sdp. CSeq: 1 INVITE. Via: SIP/2.0/UDP ori.ori.ori.ori:2000;branch=z9hG4bK-6141d60007b94a5d-4297d09d-1. Contact: sip:xxxxxxxxxxx at ori.ori.ori.ori:2000;user=phone. Allow: INVITE,CANCEL,BYE,ACK,REFER,UPDATE,INFO. Supported: timer,100rel. Max-Forwards: 70. Content-Length: 317. . v=0. o=MG4000|2.0 492612 492612 IN IP4 66.151.208.138. s=-. c=IN IP4 66.151.208.138. t=0 0. m=audio 38832 RTP/AVP 97 18 98 96 101 13. a=rtpmap:97 G.729b/8000. a=rtpmap:98 G.729a/8000. a=rtpmap:96 G.729ab/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=fmtp:18 annexb=yes. a=ptime:20. a=rtpmap:13 CN/8000. # U 2009/02/07 13:16:44.354633 fs.fs.fs.fs:5060 -> ori.ori.ori.ori:2000 SIP/2.0 100 Trying. Via: SIP/2.0/UDP ori.ori.ori.ori:2000;branch=z9hG4bK-6141d60007b94a5d-4297d09d-1. From: ;tag=13475. To: . Call-ID: 7008117790783130205-1234012604 at ori.ori.ori.ori. CSeq: 1 INVITE. User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-hacked. Content-Length: 0. U 2009/02/07 13:16:59.208008 ori.ori.ori.ori:2000 -> fs.fs.fs.fs:5060 CANCEL sip:yyyyyyyyyyyy at fs.fs.fs.fs:5060;user=phone SIP/2.0. Call-ID: 7008117790783130205-1234012604 at ori.ori.ori.ori. From: ;tag=13475. To: . CSeq: 1 CANCEL. Via: SIP/2.0/UDP ori.ori.ori.ori:2000;branch=z9hG4bK-6141d60007b94a5d-4297d09d-1. Supported: timer,100rel. Max-Forwards: 70. Content-Length: 0. . # U 2009/02/07 13:16:59.208095 fs.fs.fs.fs:5060 -> ori.ori.ori.ori:2000 SIP/2.0 200 OK. Via: SIP/2.0/UDP ori.ori.ori.ori:2000;branch=z9hG4bK-6141d60007b94a5d-4297d09d-1. From: ;tag=13475. To: ;tag=7B6p78S5SDF0j. Call-ID: 7008117790783130205-1234012604 at ori.ori.ori.ori. CSeq: 1 CANCEL. Content-Length: 0. . # U 2009/02/07 13:16:59.208134 fs.fs.fs.fs:5060 -> ori.ori.ori.ori:2000 SIP/2.0 487 Request Terminated. Via: SIP/2.0/UDP ori.ori.ori.ori:2000;branch=z9hG4bK-6141d60007b94a5d-4297d09d-1. From: ;tag=13475. To: ;tag=7B6p78S5SDF0j. Call-ID: 7008117790783130205-1234012604 at ori.ori.ori.ori. CSeq: 1 INVITE. User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-hacked. Accept: application/sdp. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. Supported: timer, precondition, path, replaces. Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Content-Length: 0. U 2009/02/07 13:16:59.253482 ori.ori.ori.ori:2000 -> fs.fs.fs.fs:5060 ACK sip:yyyyyyyyyyyy at fs.fs.fs.fs:5060;user=phone SIP/2.0. Call-ID: 7008117790783130205-1234012604 at ori.ori.ori.ori. From: ;tag=13475. To: ;tag=7B6p78S5SDF0j. CSeq: 1 ACK. Via: SIP/2.0/UDP ori.ori.ori.ori:2000;branch=z9hG4bK-6141d60007b94a5d-4297d09d-1. Max-Forwards: 70. Content-Length: 0. ------------------------------------------------------- -- View this message in context: http://www.nabble.com/Freeswitch-not-processing-calls-tp21909687p21909687.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From brian at freeswitch.org Mon Feb 9 02:14:44 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 9 Feb 2009 04:14:44 -0600 Subject: [Freeswitch-users] Freeswitch not processing calls In-Reply-To: <21909687.post@talk.nabble.com> References: <21909687.post@talk.nabble.com> Message-ID: I notice it offers 18 which is G729 but these listed below are 100% invalid. There is no such thing as G.729a, G.729b or G.729ab that are valid in the SDP. I suspect if you start FreeSWITCH and crank it up to debug level ("console loglevel debug") you'll clearly see why this is taking place. I think we talked to you on IRC about this and told this. If your termination gateway requires any of the above listed on 96,97 or 98 the call will fail. We can only do passthru on G729 which is the official name which is listed on payload 18. Your termination provider needs to be informed of this fact. The fmtp line is what controls annexb operation! /b On Feb 9, 2009, at 3:46 AM, Ankit Gandhi wrote: > a=rtpmap:97 G.729b/8000. > a=rtpmap:98 G.729a/8000. > a=rtpmap:96 G.729ab/8000. From ahgindia308 at gmail.com Mon Feb 9 03:48:12 2009 From: ahgindia308 at gmail.com (Ankit Gandhi) Date: Mon, 9 Feb 2009 03:48:12 -0800 (PST) Subject: [Freeswitch-users] Freeswitch not processing calls In-Reply-To: References: <21909687.post@talk.nabble.com> Message-ID: <21911300.post@talk.nabble.com> Hi Brian, But issue here is that, FS is not processing any such calls and not sending 488 to the caller. Also the sip trace I had provided was from the caller to fs. FS does not even bridge the call to terminator in between the INVITE and CANCEL from the caller. It just gives so many errors in log like following : 2009-02-09 11:31:04 [ERR] mod_g729.c:145 switch_g729_decode() This codec is only usable in passthrough mode! 2009-02-09 11:31:04 [ERR] mod_g729.c:145 switch_g729_decode() This codec is only usable in passthrough mode! 2009-02-09 11:31:04 [ERR] mod_g729.c:145 switch_g729_decode() This codec is only usable in passthrough mode! 2009-02-09 11:31:04 [ERR] mod_g729.c:145 switch_g729_decode() This codec is only usable in passthrough mode! 2009-02-09 11:31:04 [ERR] mod_g729.c:145 switch_g729_decode() This codec is only usable in passthrough mode! It seems odd that due to codec mismatch, fs does not process the call at all and just times out and the caller sends CANCEL to us. Waiting for your reply. Brian West-3 wrote: > > I notice it offers 18 which is G729 but these listed below are 100% > invalid. There is no such thing as G.729a, G.729b or G.729ab that are > valid in the SDP. I suspect if you start FreeSWITCH and crank it up > to debug level ("console loglevel debug") you'll clearly see why this > is taking place. I think we talked to you on IRC about this and told > this. If your termination gateway requires any of the above listed on > 96,97 or 98 the call will fail. We can only do passthru on G729 which > is the official name which is listed on payload 18. Your termination > provider needs to be informed of this fact. The fmtp line is what > controls annexb operation! > > /b > > On Feb 9, 2009, at 3:46 AM, Ankit Gandhi wrote: > >> a=rtpmap:97 G.729b/8000. >> a=rtpmap:98 G.729a/8000. >> a=rtpmap:96 G.729ab/8000. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/Freeswitch-not-processing-calls-tp21909687p21911300.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From ahgindia308 at gmail.com Mon Feb 9 04:06:24 2009 From: ahgindia308 at gmail.com (Ankit Gandhi) Date: Mon, 9 Feb 2009 04:06:24 -0800 (PST) Subject: [Freeswitch-users] Freeswitch not processing calls In-Reply-To: <21911300.post@talk.nabble.com> References: <21909687.post@talk.nabble.com> <21911300.post@talk.nabble.com> Message-ID: <21911561.post@talk.nabble.com> Here is the correct codec sent to fs, but it times out again. http://www.nabble.com/file/p21911561/correct_codec_with_cancel.txt correct_codec_with_cancel.txt Ankit Gandhi wrote: > > Hi Brian, > But issue here is that, FS is not processing any such calls and not > sending 488 to the caller. > Also the sip trace I had provided was from the caller to fs. FS does not > even bridge the call to terminator in between the INVITE and CANCEL from > the caller. > It just gives so many errors in log like following : > > 2009-02-09 11:31:04 [ERR] mod_g729.c:145 switch_g729_decode() This codec > is only usable in passthrough mode! > 2009-02-09 11:31:04 [ERR] mod_g729.c:145 switch_g729_decode() This codec > is only usable in passthrough mode! > 2009-02-09 11:31:04 [ERR] mod_g729.c:145 switch_g729_decode() This codec > is only usable in passthrough mode! > 2009-02-09 11:31:04 [ERR] mod_g729.c:145 switch_g729_decode() This codec > is only usable in passthrough mode! > 2009-02-09 11:31:04 [ERR] mod_g729.c:145 switch_g729_decode() This codec > is only usable in passthrough mode! > > It seems odd that due to codec mismatch, fs does not process the call at > all and just times out and the caller sends CANCEL to us. > Waiting for your reply. > > > Brian West-3 wrote: >> >> I notice it offers 18 which is G729 but these listed below are 100% >> invalid. There is no such thing as G.729a, G.729b or G.729ab that are >> valid in the SDP. I suspect if you start FreeSWITCH and crank it up >> to debug level ("console loglevel debug") you'll clearly see why this >> is taking place. I think we talked to you on IRC about this and told >> this. If your termination gateway requires any of the above listed on >> 96,97 or 98 the call will fail. We can only do passthru on G729 which >> is the official name which is listed on payload 18. Your termination >> provider needs to be informed of this fact. The fmtp line is what >> controls annexb operation! >> >> /b >> >> On Feb 9, 2009, at 3:46 AM, Ankit Gandhi wrote: >> >>> a=rtpmap:97 G.729b/8000. >>> a=rtpmap:98 G.729a/8000. >>> a=rtpmap:96 G.729ab/8000. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- View this message in context: http://www.nabble.com/Freeswitch-not-processing-calls-tp21909687p21911561.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From nik.middleton at noblesolutions.co.uk Mon Feb 9 05:30:35 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Mon, 9 Feb 2009 13:30:35 -0000 Subject: [Freeswitch-users] Making a system call with LUA Message-ID: In the absence of any directives on lua/mysql, is it possible to launch a PHP script from lua? All I need to do is pass some data to update a db record. I don't need to have any links to the call as I intend to call is in the hang-up callback Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090209/0a56c5c5/attachment-0002.html From anthony.minessale at gmail.com Mon Feb 9 05:51:44 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 9 Feb 2009 07:51:44 -0600 Subject: [Freeswitch-users] Freeswitch not processing calls In-Reply-To: <21911561.post@talk.nabble.com> References: <21909687.post@talk.nabble.com> <21911300.post@talk.nabble.com> <21911561.post@talk.nabble.com> Message-ID: <191c3a030902090551w75a6d5e5hc736ba7ab2784b73@mail.gmail.com> 1) please do not report bugs on the mailing list. 2) please report the bug on jira http://jira.freeswitch.org according to the rules: http://wiki.freeswitch.org/wiki/Reporting_Bugs If you have an issue that you want us to correct you will have to try the latest SVN trunk (not a snapshot) issue "make current" from your RC1 directory. Attach the entire console log output from start of call to finish with console loglevel debug. On Mon, Feb 9, 2009 at 6:06 AM, Ankit Gandhi wrote: > > Here is the correct codec sent to fs, but it times out again. > http://www.nabble.com/file/p21911561/correct_codec_with_cancel.txt > correct_codec_with_cancel.txt > > > Ankit Gandhi wrote: > > > > Hi Brian, > > But issue here is that, FS is not processing any such calls and not > > sending 488 to the caller. > > Also the sip trace I had provided was from the caller to fs. FS does not > > even bridge the call to terminator in between the INVITE and CANCEL from > > the caller. > > It just gives so many errors in log like following : > > > > 2009-02-09 11:31:04 [ERR] mod_g729.c:145 switch_g729_decode() This codec > > is only usable in passthrough mode! > > 2009-02-09 11:31:04 [ERR] mod_g729.c:145 switch_g729_decode() This codec > > is only usable in passthrough mode! > > 2009-02-09 11:31:04 [ERR] mod_g729.c:145 switch_g729_decode() This codec > > is only usable in passthrough mode! > > 2009-02-09 11:31:04 [ERR] mod_g729.c:145 switch_g729_decode() This codec > > is only usable in passthrough mode! > > 2009-02-09 11:31:04 [ERR] mod_g729.c:145 switch_g729_decode() This codec > > is only usable in passthrough mode! > > > > It seems odd that due to codec mismatch, fs does not process the call at > > all and just times out and the caller sends CANCEL to us. > > Waiting for your reply. > > > > > > Brian West-3 wrote: > >> > >> I notice it offers 18 which is G729 but these listed below are 100% > >> invalid. There is no such thing as G.729a, G.729b or G.729ab that are > >> valid in the SDP. I suspect if you start FreeSWITCH and crank it up > >> to debug level ("console loglevel debug") you'll clearly see why this > >> is taking place. I think we talked to you on IRC about this and told > >> this. If your termination gateway requires any of the above listed on > >> 96,97 or 98 the call will fail. We can only do passthru on G729 which > >> is the official name which is listed on payload 18. Your termination > >> provider needs to be informed of this fact. The fmtp line is what > >> controls annexb operation! > >> > >> /b > >> > >> On Feb 9, 2009, at 3:46 AM, Ankit Gandhi wrote: > >> > >>> a=rtpmap:97 G.729b/8000. > >>> a=rtpmap:98 G.729a/8000. > >>> a=rtpmap:96 G.729ab/8000. > >> > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > >> > > > > > > -- > View this message in context: > http://www.nabble.com/Freeswitch-not-processing-calls-tp21909687p21911561.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090209/5fe6dfae/attachment-0002.html From pauld at versafon.com Mon Feb 9 06:01:03 2009 From: pauld at versafon.com (pauld) Date: Mon, 09 Feb 2009 09:01:03 -0500 Subject: [Freeswitch-users] Dynamic Dialplan In-Reply-To: <743112140902081921n2bb1bf49mfe36e5e2193c6b04@mail.gmail.com> References: <743112140902081921n2bb1bf49mfe36e5e2193c6b04@mail.gmail.com> Message-ID: <4990371F.4050006@versafon.com> Option 3 does not have slow performance, Java apps can be highly scalable high performance when written right, this is a serious strong typed language unlike lua and javascript. I actually tested such solution against MySql cluster with 500 calls/m load script, scaled just fine. Contact me off list if you need professional help with that. Doug Blacksone wrote: > Hi, > > Right now, I am working on getting freeswitch configured for our > call-center with more than 1000 agents. There are several areas where > we need the dialplan to be configurable based on some user detail in > the database. Therefore, the dialplan needs to be some-what dynamic > based-on inputs from the database. > > I would like to know from other implementation as to the most scalable > way of doing high performance dynamic dialplan that is super scalable. > > There are three ways I can think of: > > 1. Static dialplan using customized freeswitch mod to access postgres > for data > pros: best performance > cons: harder to program > > 2. Static dialplan using lua to access postgres for data > pros: easy to program, maybe-performance is better than curl > cons: need to search through all the extensions to find one dialplan, > performance is slower than the first one. > > 3. curl-based dialplan using Java Servlet and HTTP > pros: easy to program, freeswitch only gets one extension and no > extension search > cons: performance is slow than the other two > > Is this a correct analysis? > If from a pure performance's perspective, how much performance can a > customized mod gains in comparison to lua? > > For a production system that needs to be highly scalable, what do you > recommend? > > > Thank you very much for any input to our critical design decision. > > Doug > > > > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From alex at sinapticode.ro Mon Feb 9 06:07:08 2009 From: alex at sinapticode.ro (Alexandru Nedelcu) Date: Mon, 09 Feb 2009 16:07:08 +0200 Subject: [Freeswitch-users] Making a system call with LUA In-Reply-To: References: Message-ID: <1234188428.6040.16.camel@gathern.lan> On Mon, 2009-02-09 at 13:30 +0000, Nik Middleton wrote: > In the absence of any directives on lua/mysql, is it possible to > launch a PHP script from lua? All I need to do is pass some data to > update a db record. I don?t need to have any links to the call as I > intend to call is in the hang-up callback I'm actually using Javascript, but os.execute should work: http://www.lua.org/pil/22.2.html From anthony.minessale at gmail.com Mon Feb 9 06:07:51 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 9 Feb 2009 08:07:51 -0600 Subject: [Freeswitch-users] Dynamic Dialplan In-Reply-To: <4990371F.4050006@versafon.com> References: <743112140902081921n2bb1bf49mfe36e5e2193c6b04@mail.gmail.com> <4990371F.4050006@versafon.com> Message-ID: <191c3a030902090607i3050187ay5c9047a36fb69a73@mail.gmail.com> hint: when you have "harder to program" under cons: that's usually how you find the right choice ;) On Mon, Feb 9, 2009 at 8:01 AM, pauld wrote: > Option 3 does not have slow performance, Java apps can be highly > scalable high performance when written right, this is a serious strong > typed language unlike lua and javascript. > I actually tested such solution against MySql cluster with 500 calls/m > load script, scaled just fine. > Contact me off list if you need professional help with that. > > Doug Blacksone wrote: > > Hi, > > > > Right now, I am working on getting freeswitch configured for our > > call-center with more than 1000 agents. There are several areas where > > we need the dialplan to be configurable based on some user detail in > > the database. Therefore, the dialplan needs to be some-what dynamic > > based-on inputs from the database. > > > > I would like to know from other implementation as to the most scalable > > way of doing high performance dynamic dialplan that is super scalable. > > > > There are three ways I can think of: > > > > 1. Static dialplan using customized freeswitch mod to access postgres > > for data > > pros: best performance > > cons: harder to program > > > > 2. Static dialplan using lua to access postgres for data > > pros: easy to program, maybe-performance is better than curl > > cons: need to search through all the extensions to find one dialplan, > > performance is slower than the first one. > > > > 3. curl-based dialplan using Java Servlet and HTTP > > pros: easy to program, freeswitch only gets one extension and no > > extension search > > cons: performance is slow than the other two > > > > Is this a correct analysis? > > If from a pure performance's perspective, how much performance can a > > customized mod gains in comparison to lua? > > > > For a production system that needs to be highly scalable, what do you > > recommend? > > > > > > Thank you very much for any input to our critical design decision. > > > > Doug > > > > > > > > > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090209/dee9c4d7/attachment-0002.html From jalsot at gmail.com Mon Feb 9 06:28:59 2009 From: jalsot at gmail.com (Tamas) Date: Mon, 09 Feb 2009 15:28:59 +0100 Subject: [Freeswitch-users] Questions about speex Message-ID: <49903DAB.3080706@gmail.com> Hello, I'm looking for the best codec/scenario for the last-mile and checked speex codec capabilities (http://www.speex.org/comparison/nb_codecs.png). This will be FS-FS interconnect (where one side uses portaudio, aka a simple softphone). As I see, it would be worth to use Speex in VBR mode and also to turn on DTX. Would it be possible with current mod_speex? As I saw, currently vad and vbr options are set to 0. Would setting these hardwired options to 1 make the trick? I was thinking about G.726-32 as it was suggested last week on irc, but it has still too high bandwidth requirements and Speex seems to have better MOS values over 10kbps (or 6kbps with VBR). Thanks in advance, Tamas From odermann at googlemail.com Mon Feb 9 06:59:53 2009 From: odermann at googlemail.com (Dennis) Date: Mon, 9 Feb 2009 15:59:53 +0100 Subject: [Freeswitch-users] DTMF: Mute sound for the other side? Message-ID: <5e414ed0902090659p18b970c9y8ba8f1bacda1f3ac@mail.gmail.com> hi, i am having a small problem with the dtmf-sounds... if i press a dtmf digit while i am bridged with another leg, the other side will hear the dtmf sound. this is very annoying and even worse in a conference, when multiple people can press dtmf digits (for (un-)muting themselves or using other functions). is there a way, to NOT let the other side hear the dtmf sound (but of course still make fs listening to it)? thanks for the help dennis From javieraristizabal at gmail.com Mon Feb 9 07:03:28 2009 From: javieraristizabal at gmail.com (=?ISO-8859-1?Q?Javier_Aristiz=E1bal?=) Date: Mon, 9 Feb 2009 10:03:28 -0500 Subject: [Freeswitch-users] connecting to mysql using lua In-Reply-To: <78A54C9A-C807-445B-A3BA-0B09A3B27521@freeswitch.org> References: <78A54C9A-C807-445B-A3BA-0B09A3B27521@freeswitch.org> Message-ID: Hi. Well you need to install luasql, and work only with lua 5.0 or major. You need a ODBC connection to MySQL. And there is an lua script example: ============================================================ #!/usr/local/bin/lua require "luasql.mysql" env = assert (luasql.mysql()) con = assert (env:connect("DB","user","password","localhost)) cur = assert (con:execute"SELECT id, name FROM test") row = cur:fetch ({}, "a") while row do print(string.format("Id: %s, Name: %s", row.id, row.name)) -- reusing the table of results row = cur:fetch (row, "a") end cur:close() con:close() env:close() ========================================================== regards javar -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090209/ab98f942/attachment-0002.html From kerrada2003 at yahoo.com Mon Feb 9 07:08:21 2009 From: kerrada2003 at yahoo.com (Ali Al-Rubaie) Date: Mon, 9 Feb 2009 07:08:21 -0800 (PST) Subject: [Freeswitch-users] SIP Authentication In-Reply-To: Message-ID: <367769.62870.qm@web33705.mail.mud.yahoo.com> Thanks so much Anthony but I have one more question: I was checking the source file sofia_reg.c and it seems that the code had been written iin such a way that FreeSWITCH can authenticate SIP agents based on RFC2069 and RFC2617. Is that conclusion correct? Thanks in advance, Message: 2 Date: Thu, 5 Feb 2009 10:46:54 -0600 From: Anthony Minessale Subject: Re: [Freeswitch-users] SIP Authentication To: freeswitch-users at lists.freeswitch.org Message-ID: <191c3a030902050846o60047c30pa2890707eae386d6 at mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" It's optional for us but it's mandatory for the client if we exercise the option which we have opted to always do =D There is no way in the code to disable sending it because we prefer the more secure version of SIP auth. So again it's a bug in the client for not following the protocol. It would be considered a feature in FreeSWITCH to support limping for the sake of this broken client and we currently do not have any plans for implementing this feature. On Thu, Feb 5, 2009 at 10:34 AM, Ali Al-Rubaie wrote: > > We're using HelpCaster softphone. > > The issue here is that in Digest Authentication, if the server sends the > parameter "qop" in the challenge then the client should respond with the > "cnonce" parameter. The parameter "qop" is optional in Digest Auth. So the > question here is that, can we configure FreeSWITCH so that it will not send > "qop" in the challenge? > > Thanks! > > --- On *Wed, 2/4/09, freeswitch-users-request at lists.freeswitch.org < > freeswitch-users-request at lists.freeswitch.org>* wrote: > > From: freeswitch-users-request at lists.freeswitch.org < > freeswitch-users-request at lists.freeswitch.org> > Subject: Freeswitch-users Digest, Vol 32, Issue 39 > To: freeswitch-users at lists.freeswitch.org > Date: Wednesday, February 4, 2009, 2:05 PM > > Send Freeswitch-users mailing list submissions to > freeswitch-users at lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request at lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner at lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Freeswitch-users digest..." > > > Today's Topics: > > 1. Re: SIP Authentication (Brian West) > 2. Re: origainate through sofia gateway (Michael Collins) > 3. Recording background music and voice is out of sync (Daniel Liang) > 4. Re: Q931 decoding Update (Gopalakrishnan A.N) > 5. mod_limit (Chav Paskov) > 6. Re: mod_limit (Michael Collins) > 7. Re: mod_limit (Chav > Paskov) > 8. Re: mod_limit (Michael Collins) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Wed, 4 Feb 2009 10:52:45 -0600 > From: Brian West > Subject: Re: [Freeswitch-users] SIP Authentication > To: freeswitch-users at lists.freeswitch.org > Message-ID: <7DAC91F6-2FD3-464D-AA81-321EBCADC8C0 at freeswitch.org> > Content-Type: text/plain; charset=US-ASCII; format=flowed; delsp=yes > > What client is this? I also notice we receive port 3458 and reply to > port 1059... > > /b > > On Feb 4, 2009, at 10:17 AM, Ali Al-Rubaie wrote: > > > What I have noted is that the client does not send the values for > > "cnonce" and "nc" in the response. I'm not sure if > this is the > > reason, however how this problem can be solved? > > > > Thanks, > > > > Ali > > > > > ------------------------------ > > Message: > 2 > Date: Wed, 4 Feb 2009 09:41:07 -0800 > From: Michael Collins > Subject: Re: [Freeswitch-users] origainate through sofia gateway > To: freeswitch-users at lists.freeswitch.org > Message-ID: > <87f2f3b90902040941r61d669aaie949aa7cc8578a9a at mail.gmail.com> > Content-Type: text/plain; charset=ISO-8859-1 > > I'll make sure the substance of this is in the wiki and I'll look for > references to the deprecated way and remove those. > -MC > > On Wed, Feb 4, 2009 at 6:09 AM, Anthony Minessale > wrote: > > Where did you learn how to use js this way? > > session.originate is being misused here and is depricated and may be > > removed. > > > > the first arg to session.originate is either undefined or a *different* > > session (the a leg) > > > > session1 = new Session(); > > session1.originate(undefined, > > > "{ignore_early_media=true}user/1008 at 192.168.1.122"); > > > > > session1.setVariable("effective_caller_id_number","fixed0248b"); > > > > //once you have session1 when you originate session2 you pass session1 as > > the arg > > // the effective_caller_id is taken from session1 > > > > session2 = new Session(); > > session2.originate(session1, > "sofia/gateway/halonet/0225490317"); > > > > Anyway this whole code is depricated in favor of this: > > > > session1 = new > Session("{ignore_early_media=true}user/1008 at 192.168.1.122"); > > if (session1.ready()) { > > > session1.setVariable("effective_caller_id_number","fixed0248b"); > > session2 = new Session("sofia/gateway/halonet/0225490317", > session1); > > } > > > > and could be further refactored down to this: > > > > session1 = new > Session("{ignore_early_media=true}user/1008 at 192.168.1.122"); > > if > (session1.ready()) { > > > session1.setVariable("effective_caller_id_number","fixed0248b"); > > session1.execute("bridge", > "sofia/gateway/halonet/0225490317"); > > } > > > > or down to this one line of code that will setup the call detached from > the > > script and exit. > > > > var result = apiExecute("originate", > > > "{effective_caller_id_number=fixed0248b,origination_caller_id_number=1000,ignore_early_media=true}user/1008 at 192.168.1.122 > > bridge:sofia/gateway/halonet/0225490317 inline"); > > > > if you dont care about the result and want to exit even before the call is > > completed. > > > > var result = apiExecute("bgapi", "originate > > > {effective_caller_id_number=fixed0248b,origination_caller_id_number=1000,ignore_early_media=true}user/1008 at 192.168.1.122 > > bridge:sofia/gateway/halonet/0225490317 inline"); > > > > > > > > On Wed, Feb 4, 2009 at > 2:51 AM, Jacek Sokulski > > > wrote: > >> > >> We have tried setting both effective_caller_id_number and > >> origination_caller_id_number: > >> > >> > >> > session1.originate(session1,"{origination_caller_id_number=fixed0248b}sofia/gateway/halonet/0225490317",15); > >> but the problem still exists. The solution we have found for the case > >> when we originate two calls, local and external, is as follow: > >> > >> session1 = new Session(); > >> > session1.originate(session1,"user/1003 at 192.168.1.122",15);//local > >> if(session1.ready()) { > >> session1.execute("execute_extension","00930691688627 > XML > >> default");//external > >> } > >> > >> so the external call goes through the dialplan. > >> It does not work if both calls are external. One possible solution > could > >> > be > >> to pass the originating call through dialplan (loopback?) but we have > not > >> managed > >> to figure out how to do it. > >> > >> Thanks > >> Jacek > >> > >> Dnia 03-02-2009, wto o godzinie 14:31 -0300, Nicolas Brenner pisze: > >> > Oops! Well, fortunately I don't use that voip provider > anymore (nor the > >> > script). > >> > > >> > Thanks Brian. > >> > > >> > Nicolas > >> > > >> > On Tue, Feb 3, 2009 at 2:25 PM, Brian West > wrote: > >> > > YOU should NEVER use this method or call setCallerData at > all you > >> > > should use the correct methods to override the callerid. > >> > > > >> > > If its a B-Leg born from an A-Leg you use these on the on > the A-Leg: > >> > > > >> > > > >> > > > > http://wiki.freeswitch.org/wiki/Channel_Variables#effective_caller_id_name > >> > > > >> > > > http://wiki.freeswitch.org/wiki/Channel_Variables#effective_caller_id_number > >> > > > >> > > If you're originating you use this: > >> > > > >> > > > >> > > > http://wiki.freeswitch.org/wiki/Channel_Variables#origination_caller_id_name > >> > > > >> > > > http://wiki.freeswitch.org/wiki/Channel_Variables#origination_caller_id_number > >> > > > >> > > /b > >> > > >> > _______________________________________________ > >> > Freeswitch-users mailing list > >> > Freeswitch-users at lists.freeswitch.org > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> > http://www.freeswitch.org > >> > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > ------------------------------ > > Message: 3 > Date: Wed, 4 Feb 2009 09:43:10 -0800 > From: "Daniel Liang" > Subject: [Freeswitch-users] Recording background music and voice is > out of sync > To: > Message-ID: > <0B02E756F603CC409EB553879B090CC80A23EBB5 at HPEXCHVS01.exchange.airg> > Content-Type: text/plain; charset="us-ascii" > > What I did was the following: > > First, I sent the > playback command: > > call-command: execute > execute-app-name: playback > execute-app-arg: > > Then I send uuid_record (Sorry, it was not Record command): > > api uuid_record start 120 > > I also tried replacing the playback command with: > api uuid_displace start 0 mux > > But the end results are the same. The recorded user's voice is about 0.5 > second behind the expected result. > > Thanks, > Daniel > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Brian West > Sent: February 3, 2009 6:36 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Recording background music and voice is > outof sync > > Can you show us an example of how you're doing this? Playback and > Record aren't async so > you'll need to show us how you're doing > this. > > Also don't hijack threads you hit replay on the one "Re: [Freeswitch- > users] FreeSwitch setup as a "Dumb" SBC" as the subject, deleted > the > subject and started a new body. That hijacks the thread and that can > cause your problem to go ignored in some cases if people aren't > interested in the thread topic depending on how their reader threads the > emails. > > Please click new message and type freeswitch- users at lists.freeswitch.org > in and then input your subject and body to start a new thread. > > Thanks, > Brian West > FreeSWITCH.org > > > On Feb 3, 2009, at 8:21 PM, Daniel Liang wrote: > > > Hi, > > > > I was trying to record a background music with a user's voice at the > > same time. I did a playback and started recording. But the recorded > > user's voice and the background music is about 0.5 second out of sync. > > > I also tried > to use uuid_displace instead of playback, but I got the > > same result. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090204/2d1124e2/attachment-0001.html > > > ------------------------------ > > Message: 4 > Date: Wed, 4 Feb 2009 23:26:14 +0530 > From: "Gopalakrishnan A.N" > Subject: Re: [Freeswitch-users] Q931 decoding Update > To: freeswitch-users at lists.freeswitch.org > Message-ID: > > <2ea4d47e0902040956v75c5472foa4649c50b7340484 at mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > Hi, > Its a awesome. Can the packet capturing be done with event socket? > > -- > Thank you with regards, > Gopal, > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090204/74f6434a/attachment-0001.html > > > ------------------------------ > > Message: 5 > Date: Wed, 04 Feb 2009 09:59:48 -0800 > From: Chav Paskov > Subject: [Freeswitch-users] mod_limit > To: freeswitch-users at lists.freeswitch.org > Message-ID: <4989D794.1010805 at shaw.ca> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Hi , > is it possible to use mod_limit in case if the end point is not > registered / gateway for > example/. > Regards > Chav > > > > ------------------------------ > > Message: 6 > Date: Wed, 4 Feb 2009 10:06:52 -0800 > From: Michael Collins > Subject: Re: [Freeswitch-users] mod_limit > To: freeswitch-users at lists.freeswitch.org > Message-ID: > <87f2f3b90902041006v7d7d7ff2t9961274905d9d5b0 at mail.gmail.com> > Content-Type: text/plain; charset=ISO-8859-1 > > On Wed, Feb 4, 2009 at 9:59 AM, Chav Paskov wrote: > > Hi , > > is it possible to use mod_limit in case if the end point is not > > registered / gateway for example/. > > Could you add some detail to this question? What are you trying to do? > (mod_limit may or may not work, but there might be another solution > which is why I am asking.) > > -MC > > > Regards > > Chav > > > > _______________________________________________ > > Freeswitch-users mailing list > > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > ------------------------------ > > Message: 7 > Date: Wed, 04 Feb 2009 10:54:56 -0800 > From: Chav Paskov > Subject: Re: [Freeswitch-users] mod_limit > To: freeswitch-users at lists.freeswitch.org > Message-ID: <4989E480.1080105 at shaw.ca> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Michael Collins wrote: > > On Wed, Feb 4, 2009 at 9:59 AM, Chav Paskov > wrote: > > > >> Hi , > >> is it possible to use mod_limit in case if the end point is not > >> registered / gateway for example/. > >> > > > > Could you add some detail to this question? What are you trying to do? > > > (mod_limit may or may not work, but there might be another solution > > which is why I am asking.) > > > > -MC > > > > > >> Regards > >> Chav > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > >> > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > i have few gateways under my ACL that > are allowed to send calls to FS, > but i want to be able to enforce "capacity" policy on the traffic > coming from any one of them depending on total termination capacity on > my termination end. > Let say GW 1 has to be limited to make 10 simultaneous calls while GW2 > could make up to 30 and so on. > Regards > Chav > > > > ------------------------------ > > Message: 8 > Date: Wed, 4 Feb 2009 11:05:09 -0800 > From: Michael Collins > Subject: Re: [Freeswitch-users] mod_limit > To: freeswitch-users at lists.freeswitch.org > Message-ID: > <87f2f3b90902041105l50f51f08t230bab8d69eefb4e at mail.gmail.com> > Content-Type: text/plain; charset=ISO-8859-1 > > On Wed, Feb 4, 2009 at 10:54 AM, Chav Paskov wrote: > > Michael Collins wrote: > >> On Wed, Feb 4, 2009 at 9:59 AM, Chav Paskov > wrote: > >> > >>> Hi > , > >>> is it possible to use mod_limit in case if the end point is not > >>> registered / gateway for example/. > >>> > >> > >> Could you add some detail to this question? What are you trying to do? > >> (mod_limit may or may not work, but there might be another solution > >> which is why I am asking.) > >> > >> -MC > >> > >> > >>> Regards > >>> Chav > >>> > >>> _______________________________________________ > >>> Freeswitch-users mailing list > >>> Freeswitch-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> http://www.freeswitch.org > >>> > >>> > >> > >> _______________________________________________ > >> > Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > >> > > i have few gateways under my ACL that are allowed to send calls to FS, > > but i want to be able to enforce "capacity" policy on the > traffic > > coming from any one of them depending on total termination capacity on > > my termination end. > > Let say GW 1 has to be limited to make 10 simultaneous calls while GW2 > > could make up to 30 and so on. > > I'm sure that this is possible. I don't personally have a way to test > all of this but I know that a number of our users are doing things > like this currently. Can you hop on to the IRC channel? #freeswitch on > irc.freenode.net. A lot of people there can help with > this one. > > -MC (IRC: mercutioviz) > > > Regards > > Chav > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > ------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > End of Freeswitch-users Digest, Vol 32, Issue 39 > ************************************************ > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090205/61200d9e/attachment.html ------------------------------ _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org End of Freeswitch-users Digest, Vol 32, Issue 50 ************************************************ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090209/70114813/attachment-0002.html From anthony.minessale at gmail.com Mon Feb 9 07:09:53 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 9 Feb 2009 09:09:53 -0600 Subject: [Freeswitch-users] Compiling FreeSWITCH for AstLinux In-Reply-To: <87f2f3b90902061316h5fe6e8afw253c95a55ddf3aa0@mail.gmail.com> References: <2d9149cd0902061220i11b87fd9se253109d7a39249a@mail.gmail.com> <87f2f3b90902061316h5fe6e8afw253c95a55ddf3aa0@mail.gmail.com> Message-ID: <191c3a030902090709j7a96e42csfbac48b9de1409f8@mail.gmail.com> How about PBlx I even have the domain name ;) On Fri, Feb 6, 2009 at 3:16 PM, Michael Collins wrote: > > P.S. - Yes, yes I know "AstLinux" isn't the best name for a distro > > with FreeSWITCH. Depending on my success here I have some other > > ideas... > > > > How about KickAstLinux? ;) > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090209/34791ee1/attachment-0002.html From odermann at googlemail.com Mon Feb 9 07:16:15 2009 From: odermann at googlemail.com (Dennis) Date: Mon, 9 Feb 2009 16:16:15 +0100 Subject: [Freeswitch-users] Socket-Event on "originate call"? Message-ID: <5e414ed0902090716y10f95f71o5760dfdf50d1c5cb@mail.gmail.com> hi, i am using socket outbound with fs. if i do an originate over the console, for starting an outbound call without having an inbound call, and send the originate directly to the socket, the socket is first started, if the call is in answer or ringing state. before this, i will not receive any event, because the socket was not started. therefore i will not know, if the target is "busy" (hangup, hangup cause: user busy). it would be very helpful, if the socket would start immediately on an event like "channel originate". thanks for the help dennis From kokoska.rokoska at post.cz Mon Feb 9 07:18:52 2009 From: kokoska.rokoska at post.cz (kokoska rokoska) Date: Mon, 09 Feb 2009 16:18:52 +0100 Subject: [Freeswitch-users] Dynamic Dialplan In-Reply-To: <743112140902081921n2bb1bf49mfe36e5e2193c6b04@mail.gmail.com> References: <743112140902081921n2bb1bf49mfe36e5e2193c6b04@mail.gmail.com> Message-ID: <4990495C.9010008@post.cz> Just my 2c: If you use curl with lighttpd and custom built fastcgi "C" responder (it is really simple with fcgi libs - even I can do it :-) performance could be not that bad. Like I wrote in the past it can handle about 2000 reguest per second (including SQL query wiht simple "postprocessing"). Best regards, kokoska.rokoska Doug Blacksone napsal(a): > Hi, > > Right now, I am working on getting freeswitch configured for our > call-center with more than 1000 agents. There are several areas where > we need the dialplan to be configurable based on some user detail in the > database. Therefore, the dialplan needs to be some-what dynamic > based-on inputs from the database. > > I would like to know from other implementation as to the most scalable > way of doing high performance dynamic dialplan that is super scalable. > > There are three ways I can think of: > > 1. Static dialplan using customized freeswitch mod to access postgres > for data > pros: best performance > cons: harder to program > > 2. Static dialplan using lua to access postgres for data > pros: easy to program, maybe-performance is better than curl > cons: need to search through all the extensions to find one dialplan, > performance is slower than the first one. > > 3. curl-based dialplan using Java Servlet and HTTP > pros: easy to program, freeswitch only gets one extension and no > extension search > cons: performance is slow than the other two > > Is this a correct analysis? > If from a pure performance's perspective, how much performance can a > customized mod gains in comparison to lua? > > For a production system that needs to be highly scalable, what do you > recommend? > > > Thank you very much for any input to our critical design decision. > > Doug > > > > > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From anthony.minessale at gmail.com Mon Feb 9 07:25:57 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 9 Feb 2009 09:25:57 -0600 Subject: [Freeswitch-users] DTMF: Mute sound for the other side? In-Reply-To: <5e414ed0902090659p18b970c9y8ba8f1bacda1f3ac@mail.gmail.com> References: <5e414ed0902090659p18b970c9y8ba8f1bacda1f3ac@mail.gmail.com> Message-ID: <191c3a030902090725q11c8b3eaka6ee5e8565b1a7ea@mail.gmail.com> 1) don't use inband tones for dtmf. 2) post a bounty to have FS clip the audio for x milliseconds when a tone is detected. (you will still hear faint clicks between the start of the tone and when the clipping activates) On Mon, Feb 9, 2009 at 8:59 AM, Dennis wrote: > hi, > > i am having a small problem with the dtmf-sounds... > > if i press a dtmf digit while i am bridged with another leg, the other > side will hear the dtmf sound. > this is very annoying and even worse in a conference, when multiple > people can press dtmf digits (for (un-)muting themselves or using > other functions). > > is there a way, to NOT let the other side hear the dtmf sound (but of > course still make fs listening to it)? > > > thanks for the help > dennis > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090209/bff5e75b/attachment-0002.html From anthony.minessale at gmail.com Mon Feb 9 07:27:38 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 9 Feb 2009 09:27:38 -0600 Subject: [Freeswitch-users] Socket-Event on "originate call"? In-Reply-To: <5e414ed0902090716y10f95f71o5760dfdf50d1c5cb@mail.gmail.com> References: <5e414ed0902090716y10f95f71o5760dfdf50d1c5cb@mail.gmail.com> Message-ID: <191c3a030902090727n44eef7d7x83704b66e628081@mail.gmail.com> when an originate is unsuccessful the failure and the cause code is returned as the reply to the originate request. On Mon, Feb 9, 2009 at 9:16 AM, Dennis wrote: > hi, > > i am using socket outbound with fs. > > if i do an originate over the console, for starting an outbound call > without having an inbound call, and send the originate directly to the > socket, the socket is first started, if the call is in answer or > ringing state. > before this, i will not receive any event, because the socket was not > started. therefore i will not know, if the target is "busy" (hangup, > hangup cause: user busy). > > it would be very helpful, if the socket would start immediately on an > event like "channel originate". > > > thanks for the help > dennis > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090209/05a2c7f0/attachment-0002.html From anthony.minessale at gmail.com Mon Feb 9 07:32:48 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 9 Feb 2009 09:32:48 -0600 Subject: [Freeswitch-users] Can I force the A-leg codec in FS? In-Reply-To: <498FE8B3.8040904@ewetel.de> References: <498FE8B3.8040904@ewetel.de> Message-ID: <191c3a030902090732v30f79c91qb06b3762e84b047@mail.gmail.com> 1) set late-negotation=true in the sofia profile 2) set absolute_codec_string channel variable to the exact codec you want as the first action in your dialplan. On Mon, Feb 9, 2009 at 2:26 AM, Helmut Kuper wrote: > Hi Ken, > > thx for the hint. It looks quite static, so I guess each call (also > internal) are then forced to g711. I'm looking for a dynamic way > depending on destination number. > > regards > helmut > > On 09.02.2009 09:00, Ken Rice wrote: > > Set the codec negotiation to greedy > > > > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090209/51042c84/attachment-0002.html From sprice at gmail.com Sun Feb 8 20:20:54 2009 From: sprice at gmail.com (SP) Date: Sun, 8 Feb 2009 22:20:54 -0600 Subject: [Freeswitch-users] Dynamic Dialplan In-Reply-To: <743112140902081921n2bb1bf49mfe36e5e2193c6b04@mail.gmail.com> References: <743112140902081921n2bb1bf49mfe36e5e2193c6b04@mail.gmail.com> Message-ID: <7e2ac3270902082020r3b8ec2f9of2721a6d1db0243c@mail.gmail.com> Everything you can do in a static dialplan you can do via curl as well. Multiple extensions, search/conditions are allowed. Don't sell the curl short, it's very powerful and can get the ball rolling. On Sun, Feb 8, 2009 at 21:21, Doug Blacksone wrote: > Hi, > > Right now, I am working on getting freeswitch configured for our call-center > with more than 1000 agents. There are several areas where we need the > dialplan to be configurable based on some user detail in the database. > Therefore, the dialplan needs to be some-what dynamic based-on inputs from > the database. > > I would like to know from other implementation as to the most scalable way > of doing high performance dynamic dialplan that is super scalable. > > There are three ways I can think of: > > 1. Static dialplan using customized freeswitch mod to access postgres for > data > pros: best performance > cons: harder to program > > 2. Static dialplan using lua to access postgres for data > pros: easy to program, maybe-performance is better than curl > cons: need to search through all the extensions to find one dialplan, > performance is slower than the first one. > > 3. curl-based dialplan using Java Servlet and HTTP > pros: easy to program, freeswitch only gets one extension and no extension > search > cons: performance is slow than the other two > > Is this a correct analysis? > If from a pure performance's perspective, how much performance can a > customized mod gains in comparison to lua? > > For a production system that needs to be highly scalable, what do you > recommend? > > > Thank you very much for any input to our critical design decision. > > Doug From anthony.minessale at gmail.com Mon Feb 9 07:37:44 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 9 Feb 2009 09:37:44 -0600 Subject: [Freeswitch-users] SIP Authentication In-Reply-To: <367769.62870.qm@web33705.mail.mud.yahoo.com> References: <367769.62870.qm@web33705.mail.mud.yahoo.com> Message-ID: <191c3a030902090737o61cbdf28i1e1ffb91198c504@mail.gmail.com> See this post: http://www.mail-archive.com/freeswitch-dev at lists.freeswitch.org/msg00926.html On Mon, Feb 9, 2009 at 9:08 AM, Ali Al-Rubaie wrote: > Thanks so much Anthony but I have one more question: > > I was checking the source file sofia_reg.c and it seems that the code had > been written iin such a way that FreeSWITCH can authenticate SIP agents > based on RFC2069 and RFC2617. Is that conclusion correct? > > Thanks in advance, > > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090209/ffef0342/attachment-0002.html From nik.middleton at noblesolutions.co.uk Mon Feb 9 07:47:17 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Mon, 9 Feb 2009 15:47:17 -0000 Subject: [Freeswitch-users] Making a system call with LUA In-Reply-To: <1234188428.6040.16.camel@gathern.lan> References: <1234188428.6040.16.camel@gathern.lan> Message-ID: Can I assume that info/functions in lua are all available in the embedded lua in FS? Regards -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Alexandru Nedelcu Sent: 09 February 2009 14:07 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Making a system call with LUA On Mon, 2009-02-09 at 13:30 +0000, Nik Middleton wrote: > In the absence of any directives on lua/mysql, is it possible to > launch a PHP script from lua? All I need to do is pass some data to > update a db record. I don't need to have any links to the call as I > intend to call is in the hang-up callback I'm actually using Javascript, but os.execute should work: http://www.lua.org/pil/22.2.html _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From mike at jerris.com Mon Feb 9 07:59:33 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 9 Feb 2009 10:59:33 -0500 Subject: [Freeswitch-users] Openzap r654 doesn't compile. Error with ozmod_libpri In-Reply-To: <498FE8F5.9060206@ewetel.de> References: <498FDFBD.8050401@ewetel.de> <498FE8F5.9060206@ewetel.de> Message-ID: <14E5C011-BCC8-4F97-99E7-471BD203C14F@jerris.com> It sounds like your automake got screwed up with some new changes. I tried and was unable to reproduce this issue, can you test a fresh checkout and see if you still see this issue? Mike On Feb 9, 2009, at 3:27 AM, Helmut Kuper wrote: > Hello, > > update, when I remove all ozmod_ from ozmod_libpri lines in > Makefile, it > compiles without errors. > > regards > helmut > > On 09.02.2009 08:48, Helmut Kuper wrote: >> Hello, >> >> during some imporvements on q931toPcap as well as debugging my TDM >> PRI >> problem with loosing state sync after some time I updated the code to >> latest trunk (r654). After doing a bootstrap in FS trunk directory I >> tried to compile openzap in libs/openzap. I got this error: >> >> /bin/sh ./libtool --tag=CC --mode=link gcc -g -O2 -o >> ozmod_libpri.la -rpath /usr/local/openzap/mod -lm -lpthread -ldl >> libtool: link: libtool library `ozmod_libpri.la' must begin with >> `lib' >> Try `libtool --help --mode=link' for more information. >> make: *** [ozmod_libpri.la] Error 1 >> >> Additionally I wonder what ozmod_libpri is. Do I have consider it for >> Q931 capturing? >> >> regards >> Helmut >> From helmut.kuper at ewetel.de Mon Feb 9 08:06:55 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Mon, 09 Feb 2009 17:06:55 +0100 Subject: [Freeswitch-users] Openzap r654 doesn't compile. Error with ozmod_libpri In-Reply-To: <14E5C011-BCC8-4F97-99E7-471BD203C14F@jerris.com> References: <498FDFBD.8050401@ewetel.de> <498FE8F5.9060206@ewetel.de> <14E5C011-BCC8-4F97-99E7-471BD203C14F@jerris.com> Message-ID: <4990549F.50207@ewetel.de> Hi Mike, of course I can ... will do it tomorrow. regards helmut On 09.02.2009 16:59, Michael Jerris wrote: > It sounds like your automake got screwed up with some new changes. I > tried and was unable to reproduce this issue, can you test a fresh > checkout and see if you still see this issue? > > Mike > > On Feb 9, 2009, at 3:27 AM, Helmut Kuper wrote: > From intralanman at freeswitch.org Mon Feb 9 08:15:21 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Mon, 09 Feb 2009 11:15:21 -0500 Subject: [Freeswitch-users] Cannot choose Cepstral voice from dialplan In-Reply-To: <498F6751.2000701@versafon.com> References: <498E67BF.3060207@versafon.com> <8B964172-6EEB-4B5B-B4B1-D35BFEAEB460@freeswitch.org> <498F6751.2000701@versafon.com> Message-ID: <49905699.40605@freeswitch.org> pauld wrote: > The libs are there with correct symlinks, see below. I tested both > voices directly via swift command, works fine. > Any other ideas? > It's Cepstral 5.1, FS 1.0.2. > Unpredictable issues have been reported using cepstral 5 with FreeSWITCH. I'd suggest using their 4.x release. If you have a really good reason to only use 5, then you might entice someone to work on reliable Cepstral 5 integration with a bounty... upgrade to FreeSWITCH trunk first though. -Ray From msc at freeswitch.org Mon Feb 9 09:04:57 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 9 Feb 2009 09:04:57 -0800 Subject: [Freeswitch-users] Cannot choose Cepstral voice from dialplan In-Reply-To: <498F6751.2000701@versafon.com> References: <498E67BF.3060207@versafon.com> <8B964172-6EEB-4B5B-B4B1-D35BFEAEB460@freeswitch.org> <498F6751.2000701@versafon.com> Message-ID: <87f2f3b90902090904i72e37241hb5d4808eb31fcc18@mail.gmail.com> On Sun, Feb 8, 2009 at 3:14 PM, pauld wrote: > The libs are there with correct symlinks, see below. I tested both > voices directly via swift command, works fine. > Any other ideas? > It's Cepstral 5.1, FS 1.0.2. > Well, first I recommend getting on latest trunk if that's at all possible for you. The devs have made a ton of improvements in the last five weeks. Second, this might actually be an issue with FS looking in its own lib directory for these .so files. Try a symlink from /usr/local/freeswitch/lib to your /opt/swift/lib (or whatever the name is) dir for each .so file. However, I think Raymond is correct - some weirdness has been reported by some Cepstral users on 5.1. We'd definitely like to hear about your experiences if and when you get it running. -MC From msc at freeswitch.org Mon Feb 9 09:22:20 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 9 Feb 2009 09:22:20 -0800 Subject: [Freeswitch-users] Making a system call with LUA In-Reply-To: References: <1234188428.6040.16.camel@gathern.lan> Message-ID: <87f2f3b90902090922x2950912xe44f4d750d57d4c2@mail.gmail.com> On Mon, Feb 9, 2009 at 7:47 AM, Nik Middleton wrote: > Can I assume that info/functions in lua are all available in the > embedded lua in FS? > > Regards > Generally speaking that is a safe assumption. -MC From msc at freeswitch.org Mon Feb 9 09:28:30 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 9 Feb 2009 09:28:30 -0800 Subject: [Freeswitch-users] Newbie - point me in the right direction In-Reply-To: <008F70F6-818D-476B-9DE9-33DA018D37D9@whitesmiths.com> References: <008F70F6-818D-476B-9DE9-33DA018D37D9@whitesmiths.com> Message-ID: <87f2f3b90902090928hd683219od44f1f6b075112c8@mail.gmail.com> > I have two Asterisk boxes, A1 & A2, each running a separate telephony > app. > We have an external SIP service with DID's NNNNN200 -> NNNNN299. > We want to direct the incoming SIP calls so that the DID's NNNNN200 -> > NNNNN219 go to Asterisk server A1 and NNNNN220 -> NNNNN299 to Asterisk > server A2. > Yes we really just want the calls switched on the DID. > Are you thinking about using FreeSWITCH to direct these calls? Something like this? SIP Provider <--> FS <--+--> A1 +--> A2 I just want to make sure that we understand what you are trying to accomplish and why you might need FS in this scenario... -MC From intralanman at freeswitch.org Mon Feb 9 09:31:53 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Mon, 09 Feb 2009 12:31:53 -0500 Subject: [Freeswitch-users] Dynamic Dialplan In-Reply-To: <4990495C.9010008@post.cz> References: <743112140902081921n2bb1bf49mfe36e5e2193c6b04@mail.gmail.com> <4990495C.9010008@post.cz> Message-ID: <49906889.9030801@freeswitch.org> kokoska rokoska wrote: > Just my 2c: > > If you use curl with lighttpd and custom built fastcgi "C" responder (it > is really simple with fcgi libs - even I can do it :-) performance could > be not that bad. hmmm, mod_xml_curl using C, interesting thought.. all of the complexities of writing your own module without the nice structured FS API... although, as a benefit, i guess you do get a little extra latency ;-) -Ray From Daniell at airg.com Mon Feb 9 09:40:16 2009 From: Daniell at airg.com (Daniel Liang) Date: Mon, 9 Feb 2009 09:40:16 -0800 Subject: [Freeswitch-users] Recording background music and voice is outof sync In-Reply-To: <341AE5F8-20B2-4CF3-92EE-7311B3E71C7E@freeswitch.org> References: <019501c985ac$4f00ee60$ed02cb20$@net><4987E527.1040909@laposte.net><022001c9862f$efd4b7d0$cf7e2770$@net><0B02E756F603CC409EB553879B090CC80A23EB2F@HPEXCHVS01.exchange.airg> <341AE5F8-20B2-4CF3-92EE-7311B3E71C7E@freeswitch.org> Message-ID: <0B02E756F603CC409EB553879B090CC80A23F578@HPEXCHVS01.exchange.airg> Hi Brian, I have created a new thread regarding this issue a few days ago, you may have missed it. So, I am reposting the same content there: What I did was the following: First, I sent the playback command: call-command: execute execute-app-name: playback execute-app-arg: Then I sent uuid_record (Sorry, it was not Record command): api uuid_record start 120 I also tried replacing the playback command with: api uuid_displace start 0 mux But the end results are the same. The recorded user's voice is about 0.5 second behind the expected result. Thanks, Daniel -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: February 3, 2009 6:36 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Recording background music and voice is outof sync Can you show us an example of how you're doing this? Playback and Record aren't async so you'll need to show us how you're doing this. Also don't hijack threads you hit replay on the one "Re: [Freeswitch- users] FreeSwitch setup as a "Dumb" SBC" as the subject, deleted the subject and started a new body. That hijacks the thread and that can cause your problem to go ignored in some cases if people aren't interested in the thread topic depending on how their reader threads the emails. Please click new message and type freeswitch- users at lists.freeswitch.org in and then input your subject and body to start a new thread. Thanks, Brian West FreeSWITCH.org On Feb 3, 2009, at 8:21 PM, Daniel Liang wrote: > Hi, > > I was trying to record a background music with a user's voice at the > same time. I did a playback and started recording. But the recorded > user's voice and the background music is about 0.5 second out of sync. > I also tried to use uuid_displace instead of playback, but I got the > same result. _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From helmut.kuper at ewetel.de Mon Feb 9 09:58:52 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Mon, 09 Feb 2009 18:58:52 +0100 Subject: [Freeswitch-users] Openzap r654 doesn't compile. Error with ozmod_libpri In-Reply-To: <4990549F.50207@ewetel.de> References: <498FDFBD.8050401@ewetel.de> <498FE8F5.9060206@ewetel.de> <14E5C011-BCC8-4F97-99E7-471BD203C14F@jerris.com> <4990549F.50207@ewetel.de> Message-ID: <49906EDC.4050408@ewetel.de> Hello, well, tomorrow is today ;) and so I compiled a fresh truch checout of FS and all went well ... Any idea to get my old trunk dir clean again without doing a sure or current? I don't want to clean up my binary directory due to a "make sure" ... thx for your help. And again: FS is a really nice piece of software. The more I crawl into it the more I'm impressed :) regards Helmut From kokoska.rokoska at post.cz Mon Feb 9 10:03:18 2009 From: kokoska.rokoska at post.cz (kokoska rokoska) Date: Mon, 09 Feb 2009 19:03:18 +0100 Subject: [Freeswitch-users] Dynamic Dialplan In-Reply-To: <49906889.9030801@freeswitch.org> References: <743112140902081921n2bb1bf49mfe36e5e2193c6b04@mail.gmail.com> <4990495C.9010008@post.cz> <49906889.9030801@freeswitch.org> Message-ID: <49906FE6.8000008@post.cz> Raymond Chandler napsal(a): > kokoska rokoska wrote: >> Just my 2c: >> >> If you use curl with lighttpd and custom built fastcgi "C" responder (it >> is really simple with fcgi libs - even I can do it :-) performance could >> be not that bad. > hmmm, mod_xml_curl using C, interesting thought.. May be not the best way, but very simple. Well, it depends on what you have to do, but "directory" serving based on DB queries (this what I'm using it for) is very simple - just few lines of code. > all of the > complexities of writing your own module without the nice structured FS > API... I should say I have no idea how hard is to write custom FreeSWITCH module (may be I should try it :-), but the FS code is very nice! > although, as a benefit, i guess you do get a little extra latency ;-) > :-) Yes, you are right. And as a bonus some CPU utilization... Like I wrote above, I didn't say it is faster, but IMO it is very simple and not as slow as it looks (when using apache + php + apc). Best regards, kokoska.rokoska From helmut.kuper at ewetel.de Mon Feb 9 10:17:11 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Mon, 09 Feb 2009 19:17:11 +0100 Subject: [Freeswitch-users] mod_openzap stops working after some calls In-Reply-To: <191c3a030902060802r424f8a22tb4887dff7c6aead5@mail.gmail.com> References: <498C2DC3.40701@ewetel.de> <191c3a030902060802r424f8a22tb4887dff7c6aead5@mail.gmail.com> Message-ID: <49907327.6010703@ewetel.de> Hello Anthony, :D yes that's what I'm doing ... beneath some code changes in openzap ... So I found a real timestamp in pcap is quite usefull if you have more than one call at a time ... I added that function today. It uses "libapr-1" functions. Unfortunately I introduced a dependency to libs/apr to openzap by that. If it delivers micro seconds, maybe it's better to use zap_time_now(). Have to check that tomorrow. I agree there are some problems in maintaining channel states correctly. Once a day I have to restart FS. I get "TOMANYCALLS" errors, no matching channels on RELEASE, SETUP duplicates and "oz dump 1" shows more and more channels with states other than DOWN, even, when no current calls are there. I did some timebased changes in ozmod_isdn SETUP handling and hope it helps out until state timers a available. If it works I would like to upload it to trunk, if you allow. regards helmut On 06.02.2009 17:02, Anthony Minessale wrote: > I think we have some trouble surviving issues. > So when everything is ok we do fine but if something goes wrong we > don't recover. > We are still missing state timers in the q931. > > maybe you can use your new pcap thing to see what goes wrong =D From mike at jerris.com Mon Feb 9 10:28:33 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 9 Feb 2009 13:28:33 -0500 Subject: [Freeswitch-users] Openzap r654 doesn't compile. Error with ozmod_libpri In-Reply-To: <49906EDC.4050408@ewetel.de> References: <498FDFBD.8050401@ewetel.de> <498FE8F5.9060206@ewetel.de> <14E5C011-BCC8-4F97-99E7-471BD203C14F@jerris.com> <4990549F.50207@ewetel.de> <49906EDC.4050408@ewetel.de> Message-ID: a fresh bootstrap and configure should fix it ... it may do it for you now if you just update. Mike On Feb 9, 2009, at 12:58 PM, Helmut Kuper wrote: > Hello, > > well, tomorrow is today ;) and so I compiled a fresh truch checout > of FS > and all went well ... Any idea to get my old trunk dir clean again > without doing a sure or current? I don't want to clean up my binary > directory due to a "make sure" ... > > thx for your help. And again: FS is a really nice piece of software. > The > more I crawl into it the more I'm impressed :) From mike at jerris.com Mon Feb 9 10:30:35 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 9 Feb 2009 13:30:35 -0500 Subject: [Freeswitch-users] mod_openzap stops working after some calls In-Reply-To: <49907327.6010703@ewetel.de> References: <498C2DC3.40701@ewetel.de> <191c3a030902060802r424f8a22tb4887dff7c6aead5@mail.gmail.com> <49907327.6010703@ewetel.de> Message-ID: <7CEFE4F8-7461-4DF3-ABE0-C0B818466AD0@jerris.com> We can not add apr dependency in openzap, we should use the native openzap calls instead. If there is anything you NEED that you don't have, please let me know and we will try to add replacement functions. Mike On Feb 9, 2009, at 1:17 PM, Helmut Kuper wrote: > Hello Anthony, > > > :D yes that's what I'm doing ... beneath some code changes in openzap > ... So I found a real timestamp in pcap is quite usefull if you have > more than one call at a time ... I added that function today. It uses > "libapr-1" functions. Unfortunately I introduced a dependency to > libs/apr to openzap by that. If it delivers micro seconds, maybe it's > better to use zap_time_now(). Have to check that tomorrow. > > I agree there are some problems in maintaining channel states > correctly. > Once a day I have to restart FS. I get "TOMANYCALLS" errors, no > matching > channels on RELEASE, SETUP duplicates and "oz dump 1" shows more and > more channels with states other than DOWN, even, when no current calls > are there. I did some timebased changes in ozmod_isdn SETUP handling > and hope it helps out until state timers a available. If it works I > would like to upload it to trunk, if you allow. > > regards > helmut > > > On 06.02.2009 17:02, Anthony Minessale wrote: >> I think we have some trouble surviving issues. >> So when everything is ok we do fine but if something goes wrong we >> don't recover. >> We are still missing state timers in the q931. >> >> maybe you can use your new pcap thing to see what goes wrong =D From helmut.kuper at ewetel.de Mon Feb 9 10:40:27 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Mon, 09 Feb 2009 19:40:27 +0100 Subject: [Freeswitch-users] mod_openzap stops working after some calls In-Reply-To: <7CEFE4F8-7461-4DF3-ABE0-C0B818466AD0@jerris.com> References: <498C2DC3.40701@ewetel.de> <191c3a030902060802r424f8a22tb4887dff7c6aead5@mail.gmail.com> <49907327.6010703@ewetel.de> <7CEFE4F8-7461-4DF3-ABE0-C0B818466AD0@jerris.com> Message-ID: <4990789B.40405@ewetel.de> Hi Mike, I would like to have a function which gives current time in sec, usec since unix epoch. It's only for pcap timestamp. I found a zap_time_now() somewhere in openzap maybe it helps ... regards helmut From anthony.minessale at gmail.com Mon Feb 9 11:45:38 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 9 Feb 2009 13:45:38 -0600 Subject: [Freeswitch-users] Dynamic Dialplan In-Reply-To: <49906FE6.8000008@post.cz> References: <743112140902081921n2bb1bf49mfe36e5e2193c6b04@mail.gmail.com> <4990495C.9010008@post.cz> <49906889.9030801@freeswitch.org> <49906FE6.8000008@post.cz> Message-ID: <191c3a030902091145i1fffdbdcoe49e16366dc84acf@mail.gmail.com> That's why I chose mod_xml_curl as a demo for the xml_hook api. It's not only a demo, it's rather functional =D You have 2 choices other than using the stuff we already have in tree. 1) write a custom dialplan module, this module gets a single callback function a dialplan_hunt function that has the session and the caller profile. you can see from mod_enum or mod_dialplan_xml how this can be used to make your own module that looks in a db and returns instructions to FS on the fly. 2) write a custom xml_hook and use it with mod_dialplan_xml, this type of module embeds itself into the xml lookups so when something tries to find something in the xml registry, your function is called and you can do your db lookups and generate the xml returned as binary xml obj built from a result of the query. This is more powerfule because it allows you to pre-empt any xml lookups so you can deliver directory, config, dialplan, phrase macros, etc mod_xml_curl is an example of #2, it turns the xml_req into a url req and feeds the xml returned over the http socket into an xml object and returns it as the result in place of the static contents of the xml file. On Mon, Feb 9, 2009 at 12:03 PM, kokoska rokoska wrote: > > > > Raymond Chandler napsal(a): > > kokoska rokoska wrote: > >> Just my 2c: > >> > >> If you use curl with lighttpd and custom built fastcgi "C" responder (it > >> is really simple with fcgi libs - even I can do it :-) performance could > >> be not that bad. > > hmmm, mod_xml_curl using C, interesting thought.. > > May be not the best way, but very simple. > Well, it depends on what you have to do, but "directory" serving based > on DB queries (this what I'm using it for) is very simple - just few > lines of code. > > > all of the > > complexities of writing your own module without the nice structured FS > > API... > > I should say I have no idea how hard is to write custom FreeSWITCH > module (may be I should try it :-), but the FS code is very nice! > > > although, as a benefit, i guess you do get a little extra latency ;-) > > > > :-) Yes, you are right. And as a bonus some CPU utilization... > > Like I wrote above, I didn't say it is faster, but IMO it is very simple > and not as slow as it looks (when using apache + php + apc). > > Best regards, > > kokoska.rokoska > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090209/e232d7c1/attachment-0002.html From anthony.minessale at gmail.com Mon Feb 9 12:03:34 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 9 Feb 2009 14:03:34 -0600 Subject: [Freeswitch-users] Global Variables forgotten throughthepubliccontext? In-Reply-To: <51C9B370-500D-4D95-A515-E9EEF1705014@freeswitch.org> References: <0B4E2726927041D09D0425DA0242C805@UVix> <9D0CFA79-C120-4D66-B48C-7A5641B07AF4@freeswitch.org> <5B1ED1D4834B4719B309FAC03F1595C0@UVix> <03BD385F-E942-4B32-98B0-0A29E13BC5D5@freeswitch.org> <9F926F21CCB64A49A6EFF4425B111F67@UVix> <51C9B370-500D-4D95-A515-E9EEF1705014@freeswitch.org> Message-ID: <191c3a030902091203j3accbda1o1be699d0025cf2da@mail.gmail.com> should be fixed in latest trunk On Sat, Feb 7, 2009 at 7:56 PM, Brian West wrote: > You have a \ somewhere in your path... which doesn't make sense... you're > on windows. > > Can you open a jira... I think this was the cause > http://fisheye.freeswitch.org/browse/FreeSWITCH/src/mod/formats/mod_sndfile/mod_sndfile.c?r1=11090&r2=11601 > > /b > > > On Feb 7, 2009, at 6:46 PM, UV wrote: > > Yeah, I have all the sounds installed. I don't think it's that. > I'm getting error messages such as "[ERR] mod_sndfile.c:185 > sndfile_file_open() Error Opening File [E:\FS/sounds/en/us/callie\voicemail/ > *8000\16000*\vm-goodbye.w] [System error : The system cannot find the path > specified.]" all across the board. The only thing still working is MoH? This > started from one of yesterday's builds (r11665 ? 11678). > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090209/c267bc68/attachment-0002.html From nik.middleton at noblesolutions.co.uk Mon Feb 9 12:10:15 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Mon, 9 Feb 2009 20:10:15 -0000 Subject: [Freeswitch-users] DTMF not being recognised Message-ID: Hi Guys, I have an IVR that's working fine on internal extensions, but when a call is via my sip GW, they're not being trapped. I have tried the following in the gw profile References: Message-ID: Further to this message, DTMF works with PMCU but not with PMCA which is the native format for this sip provider. Regards ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Nik Middleton Sent: 09 February 2009 20:10 To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] DTMF not being recognised Hi Guys, I have an IVR that's working fine on internal extensions, but when a call is via my sip GW, they're not being trapped. I have tried the following in the gw profile References: <0B4E2726927041D09D0425DA0242C805@UVix> <9D0CFA79-C120-4D66-B48C-7A5641B07AF4@freeswitch.org> <5B1ED1D4834B4719B309FAC03F1595C0@UVix> <03BD385F-E942-4B32-98B0-0A29E13BC5D5@freeswitch.org> <9F926F21CCB64A49A6EFF4425B111F67@UVix> <51C9B370-500D-4D95-A515-E9EEF1705014@freeswitch.org> <191c3a030902091203j3accbda1o1be699d0025cf2da@mail.gmail.com> Message-ID: <191c3a030902091318wf64698n77bb95c75338f059@mail.gmail.com> this was the wrong thread, i have no idea if this is fixed or is even a real issue. On Mon, Feb 9, 2009 at 2:03 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > should be fixed in latest trunk > > On Sat, Feb 7, 2009 at 7:56 PM, Brian West wrote: > >> You have a \ somewhere in your path... which doesn't make sense... you're >> on windows. >> >> Can you open a jira... I think this was the cause >> http://fisheye.freeswitch.org/browse/FreeSWITCH/src/mod/formats/mod_sndfile/mod_sndfile.c?r1=11090&r2=11601 >> >> /b >> >> >> On Feb 7, 2009, at 6:46 PM, UV wrote: >> >> Yeah, I have all the sounds installed. I don't think it's that. >> I'm getting error messages such as "[ERR] mod_sndfile.c:185 >> sndfile_file_open() Error Opening File [E:\FS/sounds/en/us/callie\voicemail/ >> *8000\16000*\vm-goodbye.w] [System error : The system cannot find the >> path specified.]" all across the board. The only thing still working is MoH? >> This started from one of yesterday's builds (r11665 ? 11678). >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090209/7c706b15/attachment-0002.html From msc at freeswitch.org Mon Feb 9 13:26:31 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 9 Feb 2009 13:26:31 -0800 Subject: [Freeswitch-users] DTMF not being recognised In-Reply-To: References: Message-ID: <87f2f3b90902091326t34847d95qd763ce55efcbe9b3@mail.gmail.com> On Mon, Feb 9, 2009 at 12:21 PM, Nik Middleton wrote: > Further to this message, DTMF works with PMCU but not with PMCA which is the > native format for this sip provider. > Any chance you could get some debug information? I'm wondering what is actually being sent vs. what is actually being received. A pcap at the far end to compare with a pcap at the near end would be quite enlightening. -MC From nik.middleton at noblesolutions.co.uk Mon Feb 9 13:34:13 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Mon, 9 Feb 2009 21:34:13 -0000 Subject: [Freeswitch-users] DTMF not being recognized In-Reply-To: <87f2f3b90902091326t34847d95qd763ce55efcbe9b3@mail.gmail.com> References: <87f2f3b90902091326t34847d95qd763ce55efcbe9b3@mail.gmail.com> Message-ID: Forgive me, I'm not sure how I get that info with FS, can you enlighten me? DTMF also works with GSM and others, but not Alaw Regards, -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 09 February 2009 21:27 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] DTMF not being recognised On Mon, Feb 9, 2009 at 12:21 PM, Nik Middleton wrote: > Further to this message, DTMF works with PMCU but not with PMCA which is the > native format for this sip provider. > Any chance you could get some debug information? I'm wondering what is actually being sent vs. what is actually being received. A pcap at the far end to compare with a pcap at the near end would be quite enlightening. -MC _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From msc at freeswitch.org Mon Feb 9 13:38:23 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 9 Feb 2009 13:38:23 -0800 Subject: [Freeswitch-users] DTMF not being recognized In-Reply-To: References: <87f2f3b90902091326t34847d95qd763ce55efcbe9b3@mail.gmail.com> Message-ID: <87f2f3b90902091338k126a839dya747b34fe5806aad@mail.gmail.com> On Mon, Feb 9, 2009 at 1:34 PM, Nik Middleton wrote: > Forgive me, I'm not sure how I get that info with FS, can you enlighten > me? > I was thinking of something like Wireshark. You can also check out this: http://wiki.freeswitch.org/wiki/Reporting_Bugs#Capturing_RTP_With_tshark_.28Advanced.29 Being able to see what *actually* is going out over the wire (or coming in on the wire) can take much of the guess work out of debugging. -MC From blake.france at gmail.com Mon Feb 9 10:04:28 2009 From: blake.france at gmail.com (Blake France) Date: Mon, 09 Feb 2009 12:04:28 -0600 Subject: [Freeswitch-users] Recording play end of sound file again Message-ID: <4990702C.9090305@gmail.com> Whenever I try to record and IVR or Voicemail Greeting, it will record and playback, but playback does something like this. "Please leave a message" ... "Message" It plays the end of the sound file AGAIN after playing the sound file. I've tried leaving extra time before and after speaking, but it still does this. Anyone ran into this issue? From john at argv.co.uk Mon Feb 9 14:50:52 2009 From: john at argv.co.uk (John Daragon) Date: Mon, 09 Feb 2009 22:50:52 +0000 Subject: [Freeswitch-users] BT IPExchange Interoperability Testing In-Reply-To: References: Message-ID: <4990B34C.2070505@argv.co.uk> Hi; We're looking to set up a CP which will interact with BT's 21CN network using the IPX gateway. We're running through the test scenarios (which, unfortunately, we have under NDA) now. Just wondering if anyone out there has already passed the test suite with Freeswitch ? jd -- John Daragon argv[0] limited Lambs Lawn Cottage, Staple Fitzpaine, Taunton, TA3 5SL, UK Registered in England Company Number 02947782 v +44 (0) 1460 234068 f +44 (0) 1460 234069 m +44 (0) 7836 576127 From brian at freeswitch.org Mon Feb 9 15:52:10 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 9 Feb 2009 17:52:10 -0600 Subject: [Freeswitch-users] Recording play end of sound file again In-Reply-To: <4990702C.9090305@gmail.com> References: <4990702C.9090305@gmail.com> Message-ID: Can you tell me what SVN rev you're on? /b On Feb 9, 2009, at 12:04 PM, Blake France wrote: > Whenever I try to record and IVR or Voicemail Greeting, it will record > and playback, but playback does something like this. > > "Please leave a message" ... "Message" > > It plays the end of the sound file AGAIN after playing the sound file. > I've tried leaving extra time before and after speaking, but it still > does this. Anyone ran into this issue? From brian at freeswitch.org Mon Feb 9 15:54:48 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 9 Feb 2009 17:54:48 -0600 Subject: [Freeswitch-users] BT IPExchange Interoperability Testing In-Reply-To: <4990B34C.2070505@argv.co.uk> References: <4990B34C.2070505@argv.co.uk> Message-ID: Yes search the mailing list people have interoped with BT in record time. On another note you hijacked the "DTMF not being recognized" by clicking reply, deleting the text and changing the subject. Please try not to do that in the future, click "new message" input freeswitch-users at lists.freeswitch.org then type your subject and message then click send. Your email client echo's back the headers that causes the mailing list server and many email clients to thread the message properly. /b On Feb 9, 2009, at 4:50 PM, John Daragon wrote: > Hi; > > We're looking to set up a CP which will interact with BT's 21CN > network > using the IPX gateway. > > We're running through the test scenarios (which, unfortunately, we > have > under NDA) now. > > Just wondering if anyone out there has already passed the test suite > with Freeswitch ? > > jd > > -- > John Daragon argv[0] limited > Lambs Lawn Cottage, Staple Fitzpaine, Taunton, TA3 5SL, UK > Registered in England Company Number 02947782 > v +44 (0) 1460 234068 f +44 (0) 1460 234069 m +44 (0) 7836 576127 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From shannon at sacredhearts.us Mon Feb 9 16:04:10 2009 From: shannon at sacredhearts.us (Shannon) Date: Mon, 9 Feb 2009 18:04:10 -0600 Subject: [Freeswitch-users] BT IPExchange Interoperability Testing In-Reply-To: References: <4990B34C.2070505@argv.co.uk> Message-ID: <7e2ac3270902091604y5e765b04l4934aa9088516abe@mail.gmail.com> Test A - proper user list manners -FAIL :) On 2/9/09, Brian West wrote: > Yes search the mailing list people have interoped with BT in record > time. On another note you hijacked the "DTMF not being recognized" by > clicking reply, deleting the text and changing the subject. Please > try not to do that in the future, click "new message" input > freeswitch-users at lists.freeswitch.org > then type your subject and message then click send. Your email > client echo's back the headers that causes the mailing list server and > many email clients to thread the message properly. > > /b > > On Feb 9, 2009, at 4:50 PM, John Daragon wrote: > >> Hi; >> >> We're looking to set up a CP which will interact with BT's 21CN >> network >> using the IPX gateway. >> >> We're running through the test scenarios (which, unfortunately, we >> have >> under NDA) now. >> >> Just wondering if anyone out there has already passed the test suite >> with Freeswitch ? >> >> jd >> >> -- >> John Daragon argv[0] limited >> Lambs Lawn Cottage, Staple Fitzpaine, Taunton, TA3 5SL, UK >> Registered in England Company Number 02947782 >> v +44 (0) 1460 234068 f +44 (0) 1460 234069 m +44 (0) 7836 576127 >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Shannon From brian at freeswitch.org Mon Feb 9 16:13:51 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 9 Feb 2009 18:13:51 -0600 Subject: [Freeswitch-users] BT IPExchange Interoperability Testing In-Reply-To: <7e2ac3270902091604y5e765b04l4934aa9088516abe@mail.gmail.com> References: <4990B34C.2070505@argv.co.uk> <7e2ac3270902091604y5e765b04l4934aa9088516abe@mail.gmail.com> Message-ID: <7C6E76E6-4CCB-45DC-8A91-561F3D575C9E@freeswitch.org> John, Here is the post http://lists.freeswitch.org/pipermail/freeswitch-users/2007-December/001825.html Shannon, I want to make sure everyone knows that list etiquette is critical to keep the SNR low. ;) Anyway welcome to FreeSWITCH, sit back, relax and enjoy the ride... ;) /b On Feb 9, 2009, at 6:04 PM, Shannon wrote: > Test A - proper user list manners -FAIL :) > > > On 2/9/09, Brian West wrote: >> Yes search the mailing list people have interoped with BT in record >> time. On another note you hijacked the "DTMF not being recognized" >> by >> clicking reply, deleting the text and changing the subject. Please >> try not to do that in the future, click "new message" input >> freeswitch-users at lists.freeswitch.org >> then type your subject and message then click send. Your email >> client echo's back the headers that causes the mailing list server >> and >> many email clients to thread the message properly. >> >> /b From brian at freeswitch.org Mon Feb 9 16:15:47 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 9 Feb 2009 18:15:47 -0600 Subject: [Freeswitch-users] BT IPExchange Interoperability Testing In-Reply-To: <7e2ac3270902091604y5e765b04l4934aa9088516abe@mail.gmail.com> References: <4990B34C.2070505@argv.co.uk> <7e2ac3270902091604y5e765b04l4934aa9088516abe@mail.gmail.com> Message-ID: <691796D7-3778-411A-BCE1-6493EDA6D6FE@freeswitch.org> What is funny the post about this from David Knell was also a thread hijack :P /b On Feb 9, 2009, at 6:04 PM, Shannon wrote: > Test A - proper user list manners -FAIL :) > > > On 2/9/09, Brian West wrote: >> Yes search the mailing list people have interoped with BT in record >> time. On another note you hijacked the "DTMF not being recognized" >> by >> clicking reply, deleting the text and changing the subject. Please >> try not to do that in the future, click "new message" input >> freeswitch-users at lists.freeswitch.org >> then type your subject and message then click send. Your email >> client echo's back the headers that causes the mailing list server >> and >> many email clients to thread the message properly. >> >> /b -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090209/d8cac34e/attachment-0002.html From dave at 3c.co.uk Mon Feb 9 23:04:57 2009 From: dave at 3c.co.uk (David Knell) Date: Tue, 10 Feb 2009 07:04:57 +0000 Subject: [Freeswitch-users] Thread hijacking and BT interop In-Reply-To: <691796D7-3778-411A-BCE1-6493EDA6D6FE@freeswitch.org> References: <4990B34C.2070505@argv.co.uk> <7e2ac3270902091604y5e765b04l4934aa9088516abe@mail.gmail.com> <691796D7-3778-411A-BCE1-6493EDA6D6FE@freeswitch.org> Message-ID: <9E97ECB0-825E-4622-85CE-D1961DE2A019@3c.co.uk> Oops - I did it again ;-) --Dave From dave at 3c.co.uk Mon Feb 9 23:19:00 2009 From: dave at 3c.co.uk (David Knell) Date: Tue, 10 Feb 2009 07:19:00 +0000 Subject: [Freeswitch-users] BT IPExchange Interoperability Testing In-Reply-To: <4990B34C.2070505@argv.co.uk> References: <4990B34C.2070505@argv.co.uk> Message-ID: Hi John, I think we had a chat at a show at Olympia(?) a couple of years back. We did an IPX interconnect some 12 months ago - it all went pretty well. I'll give you a call later on: depending on where you are in the process, we might be able to save you a pound or two. Cheers -- Dave > Hi; > > We're looking to set up a CP which will interact with BT's 21CN > network > using the IPX gateway. > > We're running through the test scenarios (which, unfortunately, we > have > under NDA) now. > > Just wondering if anyone out there has already passed the test suite > with Freeswitch ? > > jd > > -- > John Daragon argv[0] limited > Lambs Lawn Cottage, Staple Fitzpaine, Taunton, TA3 5SL, UK > Registered in England Company Number 02947782 > v +44 (0) 1460 234068 f +44 (0) 1460 234069 m +44 (0) 7836 576127 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From kristian.kielhofner at gmail.com Mon Feb 9 23:43:49 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Tue, 10 Feb 2009 02:43:49 -0500 Subject: [Freeswitch-users] FreeSWITCH uclibc segfault Message-ID: <2d9149cd0902092343k4109d132v9d974c77e7ac8de0@mail.gmail.com> I don't think this is worth filing a bug for (yet)... FS rev 11655 segfaults with AstLinux (uClibc). Backtrace: http://astbuild.star2star.com/astlinux-freeswitch-segfault.txt I'm sorry it doesn't have all the symbols... Everything except FS is stripped. Configure options: http://astlinux.svn.sourceforge.net/viewvc/astlinux/branches/astlinux-freeswitch/package/freeswitch/freeswitch.mk FreeSWITCH compiles cleanly. Are there any known issues with uclibc (couldn't find anything on Jira) or did I do something stupid? Thanks! -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From krice at suspicious.org Mon Feb 9 23:47:16 2009 From: krice at suspicious.org (Ken Rice) Date: Tue, 10 Feb 2009 01:47:16 -0600 Subject: [Freeswitch-users] FreeSWITCH uclibc segfault In-Reply-To: <2d9149cd0902092343k4109d132v9d974c77e7ac8de0@mail.gmail.com> Message-ID: There arent known issues cause I don't think anyone else has tried it hah > From: Kristian Kielhofner > Reply-To: > Date: Tue, 10 Feb 2009 02:43:49 -0500 > To: > Subject: [Freeswitch-users] FreeSWITCH uclibc segfault > > I don't think this is worth filing a bug for (yet)... > > FS rev 11655 segfaults with AstLinux (uClibc). > > Backtrace: > http://astbuild.star2star.com/astlinux-freeswitch-segfault.txt > > I'm sorry it doesn't have all the symbols... Everything except FS is > stripped. > > Configure options: > http://astlinux.svn.sourceforge.net/viewvc/astlinux/branches/astlinux-freeswit > ch/package/freeswitch/freeswitch.mk > > FreeSWITCH compiles cleanly. > > Are there any known issues with uclibc (couldn't find anything on > Jira) or did I do something stupid? > > Thanks! > > -- > Kristian Kielhofner > http://blog.krisk.org > http://www.submityoursip.com > http://www.astlinux.org > http://www.star2star.com > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From saigop at gmail.com Tue Feb 10 00:30:11 2009 From: saigop at gmail.com (Gopalakrishnan A.N) Date: Tue, 10 Feb 2009 14:00:11 +0530 Subject: [Freeswitch-users] javascript to get the status Message-ID: <2ea4d47e0902100030r7ca8fa94k32fb2004944ad939@mail.gmail.com> Hi, I am trying to execute the following script, its working fine for call origination, but cant able to get the status for dialed numbers, able to get only the last dialed number not for both the numbers. The script as follows, Javascript var array = [2]; array[0]="39841799874"; array[1]="39894929942"; for(var i=0;i References: <498FE8B3.8040904@ewetel.de> <191c3a030902090732v30f79c91qb06b3762e84b047@mail.gmail.com> Message-ID: <49914C75.7060104@ewetel.de> Hi Anthony, thanks, works perfectly :) regards helmut On 09.02.2009 16:32, Anthony Minessale wrote: > 1) set late-negotation=true in the sofia profile > 2) set absolute_codec_string channel variable to the exact codec you > want as the first action in your dialplan. From helmut.kuper at ewetel.de Tue Feb 10 02:12:14 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Tue, 10 Feb 2009 11:12:14 +0100 Subject: [Freeswitch-users] little problem with gateway registration Message-ID: <499152FE.7090909@ewetel.de> Hello, today I connected FS to my SIP-DDI SoftSwitch. It registered successfully, but not as expected at the register-proxy (sip2.ewetel.net), but at the proxy (proxy2.ewetel.net). My gateway config is this: The corresponding register request generated by FS is this: REGISTER sip:proxy2.ewetel.net;transport=udp SIP/2.0 Via: SIP/2.0/UDP 85.16.246.22;branch=z9hG4bK2FNpmSQHpZtye Via: SIP/2.0/UDP 85.16.246.6:5070;received=85.16.246.6;rport=5070;branch=z9hG4bK2FNpmSQHpZtye Max-Forwards: 69 From: ;tag=QrFtyDUrpg94N To: Call-ID: 8bd8f32c-f759-11dd-bb38-778d64a5265f CSeq: 111011668 REGISTER Contact: Expires: 60 User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-11698M Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: timer, precondition, path, replaces Authorization: Digest username="ippbx", realm="sip2.ewetel.net", nonce="4991527400000e269de527c4c699b9466bbf1e015cd0532c", cnonce="HE1PznH9Eiyb4QAUT+bjMA", algorithm=MD5, uri="sip:proxy2.ewetel.net;transport=udp", response="b5af17310919ec6b8f82acd4c89ffb9e", qop=auth, nc=00000001 Content-Length: 0 Shouldn't it register to register-proxy when it is given? regards Helmut From john at argv.co.uk Tue Feb 10 02:15:25 2009 From: john at argv.co.uk (John Daragon) Date: Tue, 10 Feb 2009 10:15:25 +0000 Subject: [Freeswitch-users] BT IPExchange Interoperability Testing In-Reply-To: References: <4990B34C.2070505@argv.co.uk> Message-ID: <499153BD.8010807@argv.co.uk> Brian West wrote: > Yes search the mailing list people have interoped with BT in record > time. On another note you hijacked the "DTMF not being recognized" by > clicking reply, deleting the text and changing the subject. Please > try not to do that in the future, click "new message" input freeswitch-users at lists.freeswitch.org > then type your subject and message then click send. Your email > client echo's back the headers that causes the mailing list server and > many email clients to thread the message properly. > Whoops, sorry! User IQ Error. jd -- John Daragon argv[0] limited Lambs Lawn Cottage, Staple Fitzpaine, Taunton, TA3 5SL, UK Registered in England Company Number 02947782 v +44 (0) 1460 234068 f +44 (0) 1460 234069 m +44 (0) 7836 576127 From blake.france at gmail.com Tue Feb 10 05:05:17 2009 From: blake.france at gmail.com (Blake France) Date: Tue, 10 Feb 2009 07:05:17 -0600 Subject: [Freeswitch-users] Recording play end of sound file again In-Reply-To: References: <4990702C.9090305@gmail.com> Message-ID: <49917B8D.3040409@gmail.com> Brian West wrote: > Can you tell me what SVN rev you're on? > /b > > On Feb 9, 2009, at 12:04 PM, Blake France wrote: > > >> Whenever I try to record and IVR or Voicemail Greeting, it will record >> and playback, but playback does something like this. >> >> "Please leave a message" ... "Message" >> >> It plays the end of the sound file AGAIN after playing the sound file. >> I've tried leaving extra time before and after speaking, but it still >> does this. Anyone ran into this issue? >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > Honestly I'm a noob and can't. I'm running the lastest release for PFSense. From anthony.minessale at gmail.com Tue Feb 10 06:13:17 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 10 Feb 2009 08:13:17 -0600 Subject: [Freeswitch-users] little problem with gateway registration In-Reply-To: <499152FE.7090909@ewetel.de> References: <499152FE.7090909@ewetel.de> Message-ID: <191c3a030902100613m1b9a8fc1w787b3e177a373557@mail.gmail.com> register-proxy is for where it actually sends the packet but it will still say the name of proxy in the packet. Did you check the destination address of the packet it should end up as the ip:port of that proxy. On Tue, Feb 10, 2009 at 4:12 AM, Helmut Kuper wrote: > Hello, > > today I connected FS to my SIP-DDI SoftSwitch. It registered > successfully, but not as expected at the register-proxy > (sip2.ewetel.net), but at the proxy (proxy2.ewetel.net). > > My gateway config is this: > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > The corresponding register request generated by FS is this: > > REGISTER sip:proxy2.ewetel.net;transport=udp SIP/2.0 > Via: SIP/2.0/UDP 85.16.246.22;branch=z9hG4bK2FNpmSQHpZtye > Via: SIP/2.0/UDP 85.16.246.6:5070 > ;received=85.16.246.6;rport=5070;branch=z9hG4bK2FNpmSQHpZtye > Max-Forwards: 69 > From: > ;transport=udp>;tag=QrFtyDUrpg94N > To: > ;transport=udp> > Call-ID: 8bd8f32c-f759-11dd-bb38-778d64a5265f > CSeq: 111011668 REGISTER > Contact: > Expires: 60 > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-11698M > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO > Supported: timer, precondition, path, replaces > Authorization: Digest username="ippbx", realm="sip2.ewetel.net", > nonce="4991527400000e269de527c4c699b9466bbf1e015cd0532c", > cnonce="HE1PznH9Eiyb4QAUT+bjMA", algorithm=MD5, uri="sip:proxy2.ewetel.net;transport=udp", > response="b5af17310919ec6b8f82acd4c89ffb9e", qop=auth, nc=00000001 > Content-Length: 0 > > > Shouldn't it register to register-proxy when it is given? > > regards > Helmut > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090210/a09f3d7d/attachment-0002.html From anthony.minessale at gmail.com Tue Feb 10 06:14:46 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 10 Feb 2009 08:14:46 -0600 Subject: [Freeswitch-users] Recording play end of sound file again In-Reply-To: <49917B8D.3040409@gmail.com> References: <4990702C.9090305@gmail.com> <49917B8D.3040409@gmail.com> Message-ID: <191c3a030902100614l51900772u2f88862159565aae@mail.gmail.com> your issue has already been fixed in FS. You will have to wait for pfsense to upgrade to get the fix. On Tue, Feb 10, 2009 at 7:05 AM, Blake France wrote: > Brian West wrote: > > Can you tell me what SVN rev you're on? > > /b > > > > On Feb 9, 2009, at 12:04 PM, Blake France wrote: > > > > > >> Whenever I try to record and IVR or Voicemail Greeting, it will record > >> and playback, but playback does something like this. > >> > >> "Please leave a message" ... "Message" > >> > >> It plays the end of the sound file AGAIN after playing the sound file. > >> I've tried leaving extra time before and after speaking, but it still > >> does this. Anyone ran into this issue? > >> > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > Honestly I'm a noob and can't. I'm running the lastest release for > PFSense. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090210/952b970d/attachment-0002.html From blake.france at gmail.com Tue Feb 10 06:18:50 2009 From: blake.france at gmail.com (Blake France) Date: Tue, 10 Feb 2009 08:18:50 -0600 Subject: [Freeswitch-users] Recording play end of sound file again In-Reply-To: <191c3a030902100614l51900772u2f88862159565aae@mail.gmail.com> References: <4990702C.9090305@gmail.com> <49917B8D.3040409@gmail.com> <191c3a030902100614l51900772u2f88862159565aae@mail.gmail.com> Message-ID: <49918CCA.8020701@gmail.com> Anthony Minessale wrote: > your issue has already been fixed in FS. You will have to wait for > pfsense to upgrade to get the fix. > > > On Tue, Feb 10, 2009 at 7:05 AM, Blake France > wrote: > > Brian West wrote: > > Can you tell me what SVN rev you're on? > > /b > > > > On Feb 9, 2009, at 12:04 PM, Blake France wrote: > > > > > >> Whenever I try to record and IVR or Voicemail Greeting, it will > record > >> and playback, but playback does something like this. > >> > >> "Please leave a message" ... "Message" > >> > >> It plays the end of the sound file AGAIN after playing the > sound file. > >> I've tried leaving extra time before and after speaking, but it > still > >> does this. Anyone ran into this issue? > >> > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > Honestly I'm a noob and can't. I'm running the lastest release > for PFSense. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > Thank you. From odermann at googlemail.com Tue Feb 10 06:58:32 2009 From: odermann at googlemail.com (Dennis) Date: Tue, 10 Feb 2009 15:58:32 +0100 Subject: [Freeswitch-users] DTMF: Mute sound for the other side? In-Reply-To: <191c3a030902090725q11c8b3eaka6ee5e8565b1a7ea@mail.gmail.com> References: <5e414ed0902090659p18b970c9y8ba8f1bacda1f3ac@mail.gmail.com> <191c3a030902090725q11c8b3eaka6ee5e8565b1a7ea@mail.gmail.com> Message-ID: <5e414ed0902100658j2656c453mb74d4840bc256063@mail.gmail.com> we are not using inband tones. we are using rfc2833. is it still neccessary, to do some extra programming? if yes: isn't there a way for fs to recognize, that there is a rfc2833 and simply does not play it back for the others? 2009/2/9 Anthony Minessale : > 1) don't use inband tones for dtmf. > 2) post a bounty to have FS clip the audio for x milliseconds when a tone is > detected. (you will still hear faint clicks between the start of the tone > and when the clipping activates) > > > > On Mon, Feb 9, 2009 at 8:59 AM, Dennis wrote: >> >> hi, >> >> i am having a small problem with the dtmf-sounds... >> >> if i press a dtmf digit while i am bridged with another leg, the other >> side will hear the dtmf sound. >> this is very annoying and even worse in a conference, when multiple >> people can press dtmf digits (for (un-)muting themselves or using >> other functions). >> >> is there a way, to NOT let the other side hear the dtmf sound (but of >> course still make fs listening to it)? >> >> >> thanks for the help >> dennis >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From msc at freeswitch.org Tue Feb 10 06:59:42 2009 From: msc at freeswitch.org (Michael S Collins) Date: Tue, 10 Feb 2009 06:59:42 -0800 Subject: [Freeswitch-users] Thread hijacking and BT interop In-Reply-To: <9E97ECB0-825E-4622-85CE-D1961DE2A019@3c.co.uk> References: <4990B34C.2070505@argv.co.uk> <7e2ac3270902091604y5e765b04l4934aa9088516abe@mail.gmail.com> <691796D7-3778-411A-BCE1-6493EDA6D6FE@freeswitch.org> <9E97ECB0-825E-4622-85CE-D1961DE2A019@3c.co.uk> Message-ID: On Feb 9, 2009, at 11:04 PM, David Knell wrote: > Oops - I did it again ;-) > You Britney Spears wannabe!! :p -MC > --Dave > From kawarod at laposte.net Tue Feb 10 07:07:14 2009 From: kawarod at laposte.net (rod) Date: Tue, 10 Feb 2009 19:07:14 +0400 Subject: [Freeswitch-users] mod_fax and sending a fax Message-ID: <49919822.3030101@laposte.net> Hi all, I don't understand how to use the fax commands for sending a fax. In the wiki I saw this: my question is how to specify the gateway/profile that will handle the call. For a call I can use the bridge application like this, but for the txfax ?? regards, rod From odermann at googlemail.com Tue Feb 10 07:16:28 2009 From: odermann at googlemail.com (Dennis) Date: Tue, 10 Feb 2009 16:16:28 +0100 Subject: [Freeswitch-users] Socket-Event on "originate call"? In-Reply-To: <191c3a030902090727n44eef7d7x83704b66e628081@mail.gmail.com> References: <5e414ed0902090716y10f95f71o5760dfdf50d1c5cb@mail.gmail.com> <191c3a030902090727n44eef7d7x83704b66e628081@mail.gmail.com> Message-ID: <5e414ed0902100716s2c053a48w2174f7587dd0641e@mail.gmail.com> yes, you are right. we are receiving the reply. but, we are using socket outbound and manage all calls over this socket. we also measure the durations (like variable_duration and variable_billsec) and count all outgoing calls over the socket. but, if the originate (without an inbound call) will not start the socket, we can not count up, how many calls failed because of "user busy" or how long the platform was in use. a possible workarround: is it possible to trigger a dialplan over the cli (like our default dialplan, which starts the socket), so that the dialplan starts the originates? the basic problem for us, that, if we just want to make dialouts, we are missing the inbound call to start the socket. kind regards dennis 2009/2/9 Anthony Minessale : > when an originate is unsuccessful the failure and the cause code is returned > as the reply to the originate request. > > > On Mon, Feb 9, 2009 at 9:16 AM, Dennis wrote: >> >> hi, >> >> i am using socket outbound with fs. >> >> if i do an originate over the console, for starting an outbound call >> without having an inbound call, and send the originate directly to the >> socket, the socket is first started, if the call is in answer or >> ringing state. >> before this, i will not receive any event, because the socket was not >> started. therefore i will not know, if the target is "busy" (hangup, >> hangup cause: user busy). >> >> it would be very helpful, if the socket would start immediately on an >> event like "channel originate". >> >> >> thanks for the help >> dennis From kristian.kielhofner at gmail.com Tue Feb 10 07:34:32 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Tue, 10 Feb 2009 10:34:32 -0500 Subject: [Freeswitch-users] FreeSWITCH uclibc segfault In-Reply-To: References: <2d9149cd0902092343k4109d132v9d974c77e7ac8de0@mail.gmail.com> Message-ID: <2d9149cd0902100734w4a0483a2p57f9548d9eeec1fc@mail.gmail.com> Uh oh, I was afraid of that. I haven't had to work around uClibc issues in a while. Hopefully I still remember some of that stuff. ;) On Tue, Feb 10, 2009 at 2:47 AM, Ken Rice wrote: > There arent known issues cause I don't think anyone else has tried it hah > > -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From f.koliqi at gmail.com Tue Feb 10 07:21:58 2009 From: f.koliqi at gmail.com (Fadil Berisha) Date: Tue, 10 Feb 2009 10:21:58 -0500 Subject: [Freeswitch-users] FreeSWITCH uclibc segfault In-Reply-To: <2d9149cd0902092343k4109d132v9d974c77e7ac8de0@mail.gmail.com> References: <2d9149cd0902092343k4109d132v9d974c77e7ac8de0@mail.gmail.com> Message-ID: <5c7d82f20902100721y65b60f26x79af9c770918bad6@mail.gmail.com> In my uClibc system ( busybox + uClibc), FreeSWITCH compiles cleanly with: ./bootstrap.sh make make install but after starting, getting segfaults in mod_lua and and spidermonkey. Without those modules, FreeSWITCH running. Need more tests in order to confirm OK koliqi On Tue, Feb 10, 2009 at 2:43 AM, Kristian Kielhofner < kristian.kielhofner at gmail.com> wrote: > I don't think this is worth filing a bug for (yet)... > > FS rev 11655 segfaults with AstLinux (uClibc). > > Backtrace: > http://astbuild.star2star.com/astlinux-freeswitch-segfault.txt > > I'm sorry it doesn't have all the symbols... Everything except FS is > stripped. > > Configure options: > > http://astlinux.svn.sourceforge.net/viewvc/astlinux/branches/astlinux-freeswitch/package/freeswitch/freeswitch.mk > > FreeSWITCH compiles cleanly. > > Are there any known issues with uclibc (couldn't find anything on > Jira) or did I do something stupid? > > Thanks! > > -- > Kristian Kielhofner > http://blog.krisk.org > http://www.submityoursip.com > http://www.astlinux.org > http://www.star2star.com > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090210/f97b496f/attachment-0002.html From anthony.minessale at gmail.com Tue Feb 10 08:15:34 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 10 Feb 2009 10:15:34 -0600 Subject: [Freeswitch-users] FreeSWITCH uclibc segfault In-Reply-To: <2d9149cd0902100734w4a0483a2p57f9548d9eeec1fc@mail.gmail.com> References: <2d9149cd0902092343k4109d132v9d974c77e7ac8de0@mail.gmail.com> <2d9149cd0902100734w4a0483a2p57f9548d9eeec1fc@mail.gmail.com> Message-ID: <191c3a030902100815p33a6c3b7pd5f23c35a4644703@mail.gmail.com> hmm crashing in mutex lock, maybe the pthread lib is messed up. how did you trick it into compiling? Maybe some of the answers are wrong and apr is using the wrong thread abstraction? Some guy made this wiki page regarding cross compiling, did you see it ? http://wiki.freeswitch.org/wiki/Download_%26_Installation_Guide#Cross_Compiling_for_ARM_on_Linux I was able to build FS in scratchbox for arm11 before. I do remember one time when trying to get asterisk to work on wrt in the old days that the pthread lib was bad and I had to use a different version of uClibc runtime to get around it. On Tue, Feb 10, 2009 at 9:34 AM, Kristian Kielhofner < kristian.kielhofner at gmail.com> wrote: > Uh oh, I was afraid of that. > > I haven't had to work around uClibc issues in a while. Hopefully I > still remember some of that stuff. ;) > > On Tue, Feb 10, 2009 at 2:47 AM, Ken Rice wrote: > > There arent known issues cause I don't think anyone else has tried it hah > > > > > > -- > Kristian Kielhofner > http://blog.krisk.org > http://www.submityoursip.com > http://www.astlinux.org > http://www.star2star.com > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090210/f4bb36c4/attachment-0002.html From mike at jerris.com Tue Feb 10 08:27:32 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 10 Feb 2009 11:27:32 -0500 Subject: [Freeswitch-users] DTMF: Mute sound for the other side? In-Reply-To: <5e414ed0902100658j2656c453mb74d4840bc256063@mail.gmail.com> References: <5e414ed0902090659p18b970c9y8ba8f1bacda1f3ac@mail.gmail.com> <191c3a030902090725q11c8b3eaka6ee5e8565b1a7ea@mail.gmail.com> <5e414ed0902100658j2656c453mb74d4840bc256063@mail.gmail.com> Message-ID: <174C0607-ADBE-4AB2-92C3-588A86B74EAE@jerris.com> If your in a conference and your hearing other people hitting dtmf digits that IS inband, it means that the place upstream that is doing inband to 2833 conversion is not properly clipping the dtmf, this probably needs to be fixed on that device. Mike On Feb 10, 2009, at 9:58 AM, Dennis wrote: > we are not using inband tones. we are using rfc2833. > > is it still neccessary, to do some extra programming? if yes: isn't > there a way for fs to recognize, that there is a rfc2833 and simply > does not play it back for the others? > > > 2009/2/9 Anthony Minessale : >> 1) don't use inband tones for dtmf. >> 2) post a bounty to have FS clip the audio for x milliseconds when >> a tone is >> detected. (you will still hear faint clicks between the start of >> the tone >> and when the clipping activates) >> >> >> >> On Mon, Feb 9, 2009 at 8:59 AM, Dennis >> wrote: >>> >>> hi, >>> >>> i am having a small problem with the dtmf-sounds... >>> >>> if i press a dtmf digit while i am bridged with another leg, the >>> other >>> side will hear the dtmf sound. >>> this is very annoying and even worse in a conference, when multiple >>> people can press dtmf digits (for (un-)muting themselves or using >>> other functions). >>> >>> is there a way, to NOT let the other side hear the dtmf sound (but >>> of >>> course still make fs listening to it)? >>> >>> >>> thanks for the help >>> dennis >>> From mike at jerris.com Tue Feb 10 08:34:13 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 10 Feb 2009 11:34:13 -0500 Subject: [Freeswitch-users] FreeSWITCH uclibc segfault In-Reply-To: <191c3a030902100815p33a6c3b7pd5f23c35a4644703@mail.gmail.com> References: <2d9149cd0902092343k4109d132v9d974c77e7ac8de0@mail.gmail.com> <2d9149cd0902100734w4a0483a2p57f9548d9eeec1fc@mail.gmail.com> <191c3a030902100815p33a6c3b7pd5f23c35a4644703@mail.gmail.com> Message-ID: <1AACA9A6-8228-4947-8B7D-75C599311EA0@jerris.com> From that link, these are the ones that are most likey causing the issue. I think the first one. Can you check your config.log and see what results it has for those checks? If you can compile native on the device I would suggest to on the device do a ./configure -C and then to use that config.cache file generated to get all your answers to feed your cross toolchain. You can probably just slim down that exact file and have everything you need. export apr_cv_mutex_recursive=yes; \ export ac_cv_func_pthread_rwlock_init=yes; \ export apr_cv_type_rwlock_t=yes; \ Mike On Feb 10, 2009, at 11:15 AM, Anthony Minessale wrote: > hmm crashing in mutex lock, maybe the pthread lib is messed up. > how did you trick it into compiling? Maybe some of the answers are > wrong and apr is using the wrong thread abstraction? > > Some guy made this wiki page regarding cross compiling, did you see > it ? > http://wiki.freeswitch.org/wiki/Download_%26_Installation_Guide#Cross_Compiling_for_ARM_on_Linux > > I was able to build FS in scratchbox for arm11 before. > > I do remember one time when trying to get asterisk to work on wrt in > the old days that the pthread lib > was bad and I had to use a different version of uClibc runtime to > get around it. > > > On Tue, Feb 10, 2009 at 9:34 AM, Kristian Kielhofner > wrote: > Uh oh, I was afraid of that. > > I haven't had to work around uClibc issues in a while. Hopefully I > still remember some of that stuff. ;) > > On Tue, Feb 10, 2009 at 2:47 AM, Ken Rice > wrote: > > There arent known issues cause I don't think anyone else has tried > it hah > > > > > > -- > Kristian Kielhofner > http://blog.krisk.org > http://www.submityoursip.com > http://www.astlinux.org > http://www.star2star.com > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090210/7a6639e4/attachment-0002.html From kristian.kielhofner at gmail.com Tue Feb 10 09:25:05 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Tue, 10 Feb 2009 12:25:05 -0500 Subject: [Freeswitch-users] FreeSWITCH uclibc segfault In-Reply-To: <191c3a030902100815p33a6c3b7pd5f23c35a4644703@mail.gmail.com> References: <2d9149cd0902092343k4109d132v9d974c77e7ac8de0@mail.gmail.com> <2d9149cd0902100734w4a0483a2p57f9548d9eeec1fc@mail.gmail.com> <191c3a030902100815p33a6c3b7pd5f23c35a4644703@mail.gmail.com> Message-ID: <2d9149cd0902100925n1ea6c99bn6b0ada20edab0be9@mail.gmail.com> Tony, Thanks for looking at this. That just goes to show you how useless gdb is to me. Now that you say mutex lock I can think of some configure variables to try... :) I had read that section of the wiki but there is no mention he is using uclibc. Chances are he probably is but there's no way to be sure. I'll make sure to update the wiki with whatever I find. I only have a cross compiler. Some tests cannot be run at all and other guesses are incorrect. That's the fun part. I think pthread is pretty decent in this version of uClibc but I could be wrong. On Tue, Feb 10, 2009 at 11:15 AM, Anthony Minessale wrote: > hmm crashing in mutex lock, maybe the pthread lib is messed up. > how did you trick it into compiling? Maybe some of the answers are wrong > and apr is using the wrong thread abstraction? > > Some guy made this wiki page regarding cross compiling, did you see it ? > http://wiki.freeswitch.org/wiki/Download_%26_Installation_Guide#Cross_Compiling_for_ARM_on_Linux > > I was able to build FS in scratchbox for arm11 before. > > I do remember one time when trying to get asterisk to work on wrt in the old > days that the pthread lib > was bad and I had to use a different version of uClibc runtime to get around > it. > -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From kristian.kielhofner at gmail.com Tue Feb 10 09:26:09 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Tue, 10 Feb 2009 12:26:09 -0500 Subject: [Freeswitch-users] FreeSWITCH uclibc segfault In-Reply-To: <5c7d82f20902100721y65b60f26x79af9c770918bad6@mail.gmail.com> References: <2d9149cd0902092343k4109d132v9d974c77e7ac8de0@mail.gmail.com> <5c7d82f20902100721y65b60f26x79af9c770918bad6@mail.gmail.com> Message-ID: <2d9149cd0902100926h33858165vfde06a2d6e175686@mail.gmail.com> Thanks, it's good to know it's possible. Lua and spidermonkey wouldn't even compile for me; I'm going to look into that once FS starts. On Tue, Feb 10, 2009 at 10:21 AM, Fadil Berisha wrote: > In my uClibc system ( busybox + uClibc), FreeSWITCH compiles cleanly with: > > ./bootstrap.sh > make > make install > > but after starting, getting segfaults in mod_lua and and spidermonkey. > Without those modules, FreeSWITCH running. Need more tests in order to > confirm OK > > koliqi -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From kristian.kielhofner at gmail.com Tue Feb 10 09:28:11 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Tue, 10 Feb 2009 12:28:11 -0500 Subject: [Freeswitch-users] FreeSWITCH uclibc segfault In-Reply-To: <1AACA9A6-8228-4947-8B7D-75C599311EA0@jerris.com> References: <2d9149cd0902092343k4109d132v9d974c77e7ac8de0@mail.gmail.com> <2d9149cd0902100734w4a0483a2p57f9548d9eeec1fc@mail.gmail.com> <191c3a030902100815p33a6c3b7pd5f23c35a4644703@mail.gmail.com> <1AACA9A6-8228-4947-8B7D-75C599311EA0@jerris.com> Message-ID: <2d9149cd0902100928ocfbc858x48a3f77527c1b87@mail.gmail.com> Yep, now that Tony boiled down that gdb output to a mutex lock I agree. I'm trying a compile now with these values (cross compiling, no native compiler). Cross compiling is so much fun! On Tue, Feb 10, 2009 at 11:34 AM, Michael Jerris wrote: > From that link, these are the ones that are most likey causing the issue. I > think the first one. Can you check your config.log and see what results it > has for those checks? If you can compile native on the device I would > suggest to on the device do a ./configure -C and then to use that > config.cache file generated to get all your answers to feed your cross > toolchain. You can probably just slim down that exact file and have > everything you need. > export apr_cv_mutex_recursive=yes; \ > export ac_cv_func_pthread_rwlock_init=yes; \ > export apr_cv_type_rwlock_t=yes; \ > > Mike -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From kristian.kielhofner at gmail.com Tue Feb 10 10:02:22 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Tue, 10 Feb 2009 13:02:22 -0500 Subject: [Freeswitch-users] FreeSWITCH uclibc segfault In-Reply-To: <2d9149cd0902100928ocfbc858x48a3f77527c1b87@mail.gmail.com> References: <2d9149cd0902092343k4109d132v9d974c77e7ac8de0@mail.gmail.com> <2d9149cd0902100734w4a0483a2p57f9548d9eeec1fc@mail.gmail.com> <191c3a030902100815p33a6c3b7pd5f23c35a4644703@mail.gmail.com> <1AACA9A6-8228-4947-8B7D-75C599311EA0@jerris.com> <2d9149cd0902100928ocfbc858x48a3f77527c1b87@mail.gmail.com> Message-ID: <2d9149cd0902101002i4d992397uca1f6c471f66c80@mail.gmail.com> Replying to myself: That was it! Time to do some testing... Thanks!!! On Tue, Feb 10, 2009 at 12:28 PM, Kristian Kielhofner wrote: > Yep, now that Tony boiled down that gdb output to a mutex lock I agree. > > I'm trying a compile now with these values (cross compiling, no native > compiler). > > Cross compiling is so much fun! > > On Tue, Feb 10, 2009 at 11:34 AM, Michael Jerris wrote: >> From that link, these are the ones that are most likey causing the issue. I >> think the first one. Can you check your config.log and see what results it >> has for those checks? If you can compile native on the device I would >> suggest to on the device do a ./configure -C and then to use that >> config.cache file generated to get all your answers to feed your cross >> toolchain. You can probably just slim down that exact file and have >> everything you need. >> export apr_cv_mutex_recursive=yes; \ >> export ac_cv_func_pthread_rwlock_init=yes; \ >> export apr_cv_type_rwlock_t=yes; \ >> >> Mike > > -- > Kristian Kielhofner > http://blog.krisk.org > http://www.submityoursip.com > http://www.astlinux.org > http://www.star2star.com > -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From ajlong at worldlink.net Tue Feb 10 11:01:38 2009 From: ajlong at worldlink.net (Adam Long) Date: Tue, 10 Feb 2009 14:01:38 -0500 Subject: [Freeswitch-users] Dynamic Dialplan In-Reply-To: <49906FE6.8000008@post.cz> References: <743112140902081921n2bb1bf49mfe36e5e2193c6b04@mail.gmail.com> <4990495C.9010008@post.cz> <49906889.9030801@freeswitch.org> <49906FE6.8000008@post.cz> Message-ID: <000301c98bb2$00d66560$02833020$@net> What about a mod_perl XML binding like the example here? http://wiki.freeswitch.org/wiki/Mod_perl and http://wiki.freeswitch.org/wiki/Mod_perl_and_Generating_XML Would this be faster than a setup like mod_curl_xml -> lighttpd -> FastCGI/Perl ? I guess it would depend on if mod_perl in FreeSWITCH spawns new interpreter per request or if it uses one interpreter instance with multiple threads to execute pre-loaded perl. Anyone know if this is the case? Regards, -Adam -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of kokoska rokoska Sent: Monday, February 09, 2009 1:03 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Dynamic Dialplan Raymond Chandler napsal(a): > kokoska rokoska wrote: >> Just my 2c: >> >> If you use curl with lighttpd and custom built fastcgi "C" responder (it >> is really simple with fcgi libs - even I can do it :-) performance could >> be not that bad. > hmmm, mod_xml_curl using C, interesting thought.. May be not the best way, but very simple. Well, it depends on what you have to do, but "directory" serving based on DB queries (this what I'm using it for) is very simple - just few lines of code. > all of the > complexities of writing your own module without the nice structured FS > API... I should say I have no idea how hard is to write custom FreeSWITCH module (may be I should try it :-), but the FS code is very nice! > although, as a benefit, i guess you do get a little extra latency ;-) > :-) Yes, you are right. And as a bonus some CPU utilization... Like I wrote above, I didn't say it is faster, but IMO it is very simple and not as slow as it looks (when using apache + php + apc). Best regards, kokoska.rokoska _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From pbd at suspiria.net Tue Feb 10 10:41:37 2009 From: pbd at suspiria.net (Public Dump) Date: Tue, 10 Feb 2009 19:41:37 +0100 Subject: [Freeswitch-users] High CPU load after starting Message-ID: <13C421883438EB42B9E2C30069FD4AB766F9B1F840@crushinator.central.local> After starting FreeSwitch (1.0.2) on a 4 core server running Windows Server 2008, the CPU load (privileged time/kernel) for one of the cores goes to 50% and stays there. Stoping FreeSwitch stops the load. I have tried to disable all modules but the problem persists. Has anybody seen this problem, can it be fixed ? regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090210/f57d864c/attachment-0002.html From brian at freeswitch.org Tue Feb 10 12:44:03 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 10 Feb 2009 14:44:03 -0600 Subject: [Freeswitch-users] High CPU load after starting In-Reply-To: <13C421883438EB42B9E2C30069FD4AB766F9B1F840@crushinator.central.local> References: <13C421883438EB42B9E2C30069FD4AB766F9B1F840@crushinator.central.local> Message-ID: <7A22A338-495E-4D95-8D0F-FD196838B0CD@freeswitch.org> Please update to SVN trunk or the latest Windows Build and this problem should go away. /b On Feb 10, 2009, at 12:41 PM, Public Dump wrote: > After starting FreeSwitch (1.0.2) on a 4 core server running > Windows Server 2008, the CPU load (privileged time/kernel) for one > of the cores goes to 50% and stays there. > Stoping FreeSwitch stops the load. I have tried to disable all > modules but the problem persists. > > Has anybody seen this problem, can it be fixed ? > > regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090210/2b4107ed/attachment-0002.html From gmaruzz at celliax.org Tue Feb 10 13:03:52 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Tue, 10 Feb 2009 22:03:52 +0100 Subject: [Freeswitch-users] High CPU load after starting In-Reply-To: <7A22A338-495E-4D95-8D0F-FD196838B0CD@freeswitch.org> References: <13C421883438EB42B9E2C30069FD4AB766F9B1F840@crushinator.central.local> <7A22A338-495E-4D95-8D0F-FD196838B0CD@freeswitch.org> Message-ID: <7b197bef0902101303r664fc8f0h280ce6f31517e9a1@mail.gmail.com> I use often the Windows svn on Vista, no problem seen Sincerely, Giovanni Maruzzelli ========================================= www.celliax.org via Pierlombardo 9, 20135 Milano Italy gmaruzz at celliax dot org Cell : +39-347-2665618 Fax : +39-02-87390039 On Tue, Feb 10, 2009 at 9:44 PM, Brian West wrote: > Please update to SVN trunk or the latest Windows Build and this problem > should go away. > /b > > On Feb 10, 2009, at 12:41 PM, Public Dump wrote: > > After starting FreeSwitch (1.0.2) on a 4 core server running Windows > Server 2008, the CPU load (privileged time/kernel) for one of the cores goes > to 50% and stays there. > Stoping FreeSwitch stops the load. I have tried to disable all modules but > the problem persists. > > Has anybody seen this problem, can it be fixed ? > > regards > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090210/81ca2b0c/attachment-0002.html From nik.middleton at noblesolutions.co.uk Tue Feb 10 13:04:21 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Tue, 10 Feb 2009 21:04:21 -0000 Subject: [Freeswitch-users] Strange error message Message-ID: Hi Guys, I'm baffled by this error. I'm updating the db on call hang-up If I comment out curs:close() no error, but I'm concerned about memory leaks. Can anyone tell me what FS is complaining about? The db gets updated in both cases Regards require "luasql.mysql" function myHangupHook(s, status, arg) freeswitch.consoleLog("info", " : They hung up on US!!!\n"); env = assert (luasql.mysql()) con = assert (env:connect("xxxxl","xxxxxxxxx","pass","192.168.3.205")) curs = assert (con:execute"UPDATE callers SET lastcall = 'BOB' WHERE id = 33292") curs:close() con:close() env:close() freeswitch.consoleLog("NOTICE", "myHangupHook: " .. status .. "\n"); --error() end 2009-02-10 20:53:20 [INFO] switch_cpp.cpp:1086 console_log() : They hung up on US!!! 2009-02-10 20:53:20 [ERR] mod_lua.cpp:176 lua_parse_and_execute() /usr/local/freeswitch/scripts/helloworld.lua:50: attempt to index global 'curs' (a number value) stack traceback: /usr/local/freeswitch/scripts/helloworld.lua:50: in function [C]: in function 'hangup' /usr/local/freeswitch/scripts/helloworld.lua:70: in main chunk 2009-02-10 20:53:20 [NOTICE] switch_core_session.c:957 switch_core_session_thread() Session 63 (sofia/internal/1001 at 192.168.3.206) Ended -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090210/82e04d3a/attachment-0002.html From jesse.peterson at exbiblio.com Tue Feb 10 12:49:37 2009 From: jesse.peterson at exbiblio.com (Jesse Peterson) Date: Tue, 10 Feb 2009 12:49:37 -0800 Subject: [Freeswitch-users] SIP registration/retry/authorization problem Message-ID: <86FB611F-47CE-4472-8CE7-E52F6F1AADF5@exbiblio.com> Hello, I seem to be experiencing the exact same issue as is documented here: http://lists.freeswitch.org/pipermail/freeswitch-users/2008-September/006272.html Like above a "sofia profile external restart" immediately resumes operation. Does anyone have an idea what this may be? Is there a debugging step I can take? If it were a predictable outage I could monitor the registration attempts and find out why there are [401][Unauthorized] errors suddenly (again after a 'sofia restart' all is well). This happens multiple times a day for us. Thanks, - Jesse From msc at freeswitch.org Tue Feb 10 13:18:02 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 10 Feb 2009 13:18:02 -0800 Subject: [Freeswitch-users] SIP registration/retry/authorization problem In-Reply-To: <86FB611F-47CE-4472-8CE7-E52F6F1AADF5@exbiblio.com> References: <86FB611F-47CE-4472-8CE7-E52F6F1AADF5@exbiblio.com> Message-ID: <87f2f3b90902101318x37bf045of2ec3bec5890d279@mail.gmail.com> On Tue, Feb 10, 2009 at 12:49 PM, Jesse Peterson wrote: > Hello, > > I seem to be experiencing the exact same issue as is documented here: > http://lists.freeswitch.org/pipermail/freeswitch-users/2008-September/006272.html > > Like above a "sofia profile external restart" immediately resumes > operation. Does anyone have an idea what this may be? > > Is there a debugging step I can take? If it were a predictable outage > I could monitor the registration attempts and find out why there are > [401][Unauthorized] errors suddenly (again after a 'sofia restart' > all is well). > > This happens multiple times a day for us. Which revision of FS are you running? If you can update to the latest trunk and reproduce the symptoms that would be helpful. -MC From c_cav_01 at yahoo.com Tue Feb 10 13:19:28 2009 From: c_cav_01 at yahoo.com (Chris) Date: Tue, 10 Feb 2009 13:19:28 -0800 (PST) Subject: [Freeswitch-users] Strange error message In-Reply-To: Message-ID: <77751.47799.qm@web55105.mail.re4.yahoo.com> Closing the connection will force the server to close any open transactions, as well as release recordsets in local memory that reference the connection. ? However curs is not a recordset.? An SQL update is going to return an integer (rows affected) or boolean depending on the which server you use since no recordset is actually requested. --- On Tue, 2/10/09, Nik Middleton wrote: From: Nik Middleton Subject: [Freeswitch-users] Strange error message To: freeswitch-users at lists.freeswitch.org Date: Tuesday, February 10, 2009, 2:04 PM Hi Guys, ? I?m baffled by this error.? I?m updating the db on call hang-up If I comment out curs:close() no error, but I?m concerned about memory leaks.? Can anyone tell me what FS is complaining about? ? The db gets updated in both cases ? Regards ? ? ? require "luasql.mysql" ? function myHangupHook(s, status, arg) ??????????? freeswitch.consoleLog("info", " : They hung up on US!!!\n"); ??? ??????? env = assert (luasql.mysql()) ??????????? con = assert (env:connect("xxxxl","xxxxxxxxx","pass","192.168.3.205")) ??????????? curs = assert (con:execute"UPDATE callers SET lastcall = 'BOB' WHERE id = 33292") ??????????? curs:close() ??????????? con:close() ??????????? env:close() ??????????? freeswitch.consoleLog("NOTICE", "myHangupHook: " .. status .. "\n"); ??? --error() end ? ? ? ? 2009-02-10 20:53:20 [INFO] switch_cpp.cpp:1086 console_log()? : They hung up on US!!! 2009-02-10 20:53:20 [ERR] mod_lua.cpp:176 lua_parse_and_execute() /usr/local/freeswitch/scripts/helloworld.lua:50: attempt to index global 'curs' (a number value) stack traceback: ??????? /usr/local/freeswitch/scripts/helloworld.lua:50: in function ??????? [C]: in function 'hangup' ??????? /usr/local/freeswitch/scripts/helloworld.lua:70: in main chunk 2009-02-10 20:53:20 [NOTICE] switch_core_session.c:957 switch_core_session_thread() Session 63 (sofia/internal/1001 at 192.168.3.206) Ended_______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090210/58f2bc72/attachment-0002.html From brian at freeswitch.org Tue Feb 10 13:25:43 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 10 Feb 2009 15:25:43 -0600 Subject: [Freeswitch-users] High CPU load after starting In-Reply-To: <7b197bef0902101303r664fc8f0h280ce6f31517e9a1@mail.gmail.com> References: <13C421883438EB42B9E2C30069FD4AB766F9B1F840@crushinator.central.local> <7A22A338-495E-4D95-8D0F-FD196838B0CD@freeswitch.org> <7b197bef0902101303r664fc8f0h280ce6f31517e9a1@mail.gmail.com> Message-ID: <5E26356E-2E69-4828-84B3-A537436FE2AB@freeswitch.org> Yes but this was a problem that was confirmed and fixed long ago... if you recall we have 1.0.3RC1 out and svn trunk. /b On Feb 10, 2009, at 3:03 PM, Giovanni Maruzzelli wrote: > I use often the Windows svn on Vista, no problem seen > > Sincerely, From brian at freeswitch.org Tue Feb 10 13:27:35 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 10 Feb 2009 15:27:35 -0600 Subject: [Freeswitch-users] SIP registration/retry/authorization problem In-Reply-To: <86FB611F-47CE-4472-8CE7-E52F6F1AADF5@exbiblio.com> References: <86FB611F-47CE-4472-8CE7-E52F6F1AADF5@exbiblio.com> Message-ID: <286A20C5-4088-4CD9-8BAA-D294777AF931@freeswitch.org> Try this: In sofia.conf.xml /b On Feb 10, 2009, at 2:49 PM, Jesse Peterson wrote: > Hello, > > I seem to be experiencing the exact same issue as is documented here: > http://lists.freeswitch.org/pipermail/freeswitch-users/2008-September/006272.html > > Like above a "sofia profile external restart" immediately resumes > operation. Does anyone have an idea what this may be? > > Is there a debugging step I can take? If it were a predictable outage > I could monitor the registration attempts and find out why there are > [401][Unauthorized] errors suddenly (again after a 'sofia restart' > all is well). > > This happens multiple times a day for us. > > Thanks, > - Jesse -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090210/794cd661/attachment-0002.html From kokoska.rokoska at post.cz Tue Feb 10 13:34:30 2009 From: kokoska.rokoska at post.cz (kokoska rokoska) Date: Tue, 10 Feb 2009 22:34:30 +0100 Subject: [Freeswitch-users] Dynamic Dialplan In-Reply-To: <191c3a030902091145i1fffdbdcoe49e16366dc84acf@mail.gmail.com> References: <743112140902081921n2bb1bf49mfe36e5e2193c6b04@mail.gmail.com> <4990495C.9010008@post.cz> <49906889.9030801@freeswitch.org> <49906FE6.8000008@post.cz> <191c3a030902091145i1fffdbdcoe49e16366dc84acf@mail.gmail.com> Message-ID: <4991F2E6.1020102@post.cz> Anthony Minessale napsal(a): > That's why I chose mod_xml_curl as a demo for the xml_hook api. It's > not only a demo, it's rather functional =D > :-)) > You have 2 choices other than using the stuff we already have in tree. > > 1) write a custom dialplan module, this module gets a single callback > function a dialplan_hunt function that has the session and the caller > profile. you can see from mod_enum or mod_dialplan_xml how this can be > used to make your own module that looks in a db and returns instructions > to FS on the fly. > I try to look at it. > 2) write a custom xml_hook and use it with mod_dialplan_xml, this type > of module embeds itself into the xml lookups so when something tries to > find something in the xml registry, your function is called and you can > do your db lookups and generate the xml returned as binary xml obj built > from a result of the query. This is more powerfule because it allows > you to pre-empt any xml lookups so you can deliver directory, config, > dialplan, phrase macros, etc > Super tip, Anthony! Thank you very much. In case of directory it can dramatically increase maximu of registers per second. > > mod_xml_curl is an example of #2, it turns the xml_req into a url req > and feeds the xml returned over the http socket into an xml object and > returns it as the result in place of the static contents of the xml file. > Well I will look deep into mod_xml_curl to see how it is done and than (may be :-) will introduce new module mod_db_directory :-) Thanks once more, Anthony, for very valuable info! Best regards, kokoska.rokoska From anthony.minessale at gmail.com Tue Feb 10 14:32:40 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 10 Feb 2009 16:32:40 -0600 Subject: [Freeswitch-users] Dynamic Dialplan In-Reply-To: <000301c98bb2$00d66560$02833020$@net> References: <743112140902081921n2bb1bf49mfe36e5e2193c6b04@mail.gmail.com> <4990495C.9010008@post.cz> <49906889.9030801@freeswitch.org> <49906FE6.8000008@post.cz> <000301c98bb2$00d66560$02833020$@net> Message-ID: <191c3a030902101432y5954389dje53f6dc05ce654cc@mail.gmail.com> oh yeah, i forgot about those too. python,perl and lua can all do that. On Tue, Feb 10, 2009 at 1:01 PM, Adam Long wrote: > What about a mod_perl XML binding like the example here? > http://wiki.freeswitch.org/wiki/Mod_perl and > http://wiki.freeswitch.org/wiki/Mod_perl_and_Generating_XML > > Would this be faster than a setup like > > mod_curl_xml -> lighttpd -> FastCGI/Perl ? > > I guess it would depend on if mod_perl in FreeSWITCH spawns new interpreter > per request > or if it uses one interpreter instance with multiple threads to execute > pre-loaded perl. > > Anyone know if this is the case? > > Regards, > -Adam > > > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of kokoska > rokoska > Sent: Monday, February 09, 2009 1:03 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Dynamic Dialplan > > > > > Raymond Chandler napsal(a): > > kokoska rokoska wrote: > >> Just my 2c: > >> > >> If you use curl with lighttpd and custom built fastcgi "C" responder (it > >> is really simple with fcgi libs - even I can do it :-) performance could > >> be not that bad. > > hmmm, mod_xml_curl using C, interesting thought.. > > May be not the best way, but very simple. > Well, it depends on what you have to do, but "directory" serving based > on DB queries (this what I'm using it for) is very simple - just few > lines of code. > > > all of the > > complexities of writing your own module without the nice structured FS > > API... > > I should say I have no idea how hard is to write custom FreeSWITCH > module (may be I should try it :-), but the FS code is very nice! > > > although, as a benefit, i guess you do get a little extra latency ;-) > > > > :-) Yes, you are right. And as a bonus some CPU utilization... > > Like I wrote above, I didn't say it is faster, but IMO it is very simple > and not as slow as it looks (when using apache + php + apc). > > Best regards, > > kokoska.rokoska > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090210/cac42683/attachment-0002.html From nik.middleton at noblesolutions.co.uk Tue Feb 10 15:22:00 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Tue, 10 Feb 2009 23:22:00 -0000 Subject: [Freeswitch-users] Strange error message In-Reply-To: <77751.47799.qm@web55105.mail.re4.yahoo.com> References: <77751.47799.qm@web55105.mail.re4.yahoo.com> Message-ID: So what you're saying is that I can comment out curs:close() as it's not needed? Regards, ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Chris Sent: 10 February 2009 21:19 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Strange error message Closing the connection will force the server to close any open transactions, as well as release recordsets in local memory that reference the connection. However curs is not a recordset. An SQL update is going to return an integer (rows affected) or boolean depending on the which server you use since no recordset is actually requested. --- On Tue, 2/10/09, Nik Middleton wrote: From: Nik Middleton Subject: [Freeswitch-users] Strange error message To: freeswitch-users at lists.freeswitch.org Date: Tuesday, February 10, 2009, 2:04 PM Hi Guys, I'm baffled by this error. I'm updating the db on call hang-up If I comment out curs:close() no error, but I'm concerned about memory leaks. Can anyone tell me what FS is complaining about? The db gets updated in both cases Regards require "luasql.mysql" function myHangupHook(s, status, arg) freeswitch.consoleLog("info", " : They hung up on US!!!\n"); env = assert (luasql.mysql()) con = assert (env:connect("xxxxl","xxxxxxxxx","pass","192.168.3.205")) curs = assert (con:execute"UPDATE callers SET lastcall = 'BOB' WHERE id = 33292") curs:close() con:close() env:close() freeswitch.consoleLog("NOTICE", "myHangupHook: " .. status .. "\n"); --error() end 2009-02-10 20:53:20 [INFO] switch_cpp.cpp:1086 console_log() : They hung up on US!!! 2009-02-10 20:53:20 [ERR] mod_lua.cpp:176 lua_parse_and_execute() /usr/local/freeswitch/scripts/helloworld.lua:50: attempt to index global 'curs' (a number value) stack traceback: /usr/local/freeswitch/scripts/helloworld.lua:50: in function [C]: in function 'hangup' /usr/local/freeswitch/scripts/helloworld.lua:70: in main chunk 2009-02-10 20:53:20 [NOTICE] switch_core_session.c:957 switch_core_session_thread() Session 63 (sofia/internal/1001 at 192.168.3.206) Ended _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090210/cb925ccd/attachment-0002.html From jonathan at myogre.com Tue Feb 10 13:22:58 2009 From: jonathan at myogre.com (Jonathan Creasy) Date: Tue, 10 Feb 2009 15:22:58 -0600 Subject: [Freeswitch-users] dialplan question Message-ID: <18cd54110902101322g76cc3aedj6dd38a52ecdada81@mail.gmail.com> I'm trying to setup a dialstring to send a call to a user as a DID number not their user. I want to be able to do one of two things: One, I want the Request URI to match the contact from their REGISTER and To header to be "+1NXXNXXXXXX at contactdomain" Two, I want to make both the Request URI and the To header be "+1NXXNXXXXXX at contactdomain". I looked at "force-user" but that gives me the ability to do one DID to a user (by forcing the user to the DID). So, how do I need to setup the call for these scenarios? From jonathan at myogre.com Tue Feb 10 17:00:23 2009 From: jonathan at myogre.com (Jonathan Creasy) Date: Tue, 10 Feb 2009 19:00:23 -0600 Subject: [Freeswitch-users] dialplan question Message-ID: <18cd54110902101700r3f039a2ex19c1b0818c1a3af4@mail.gmail.com> I'm trying to setup an extension to send a call to a user as a DID number not their user. I want to be able to do one of two things: One, I want the Request URI to match the contact from their REGISTER and To header to be "+1NXXNXXXXXX at contactdomain" Two, I want to make both the Request URI and the To header be "+1NXXNXXXXXX at contactdomain". I looked at "force-user" but that gives me the ability to do one DID to a user (by forcing the user to the DID). So, how do I need to setup the call for these scenarios? From jesse.peterson at exbiblio.com Tue Feb 10 17:39:18 2009 From: jesse.peterson at exbiblio.com (Jesse Peterson) Date: Tue, 10 Feb 2009 17:39:18 -0800 Subject: [Freeswitch-users] SIP registration/retry/authorization problem In-Reply-To: <87f2f3b90902101318x37bf045of2ec3bec5890d279@mail.gmail.com> References: <86FB611F-47CE-4472-8CE7-E52F6F1AADF5@exbiblio.com> <87f2f3b90902101318x37bf045of2ec3bec5890d279@mail.gmail.com> Message-ID: On Feb 10, 2009, at 1:18 PM, Michael Collins wrote: > Which revision of FS are you running? If you can update to the latest > trunk and reproduce the symptoms that would be helpful. > -MC This is FreeSwitch 1.0 running from a twixswitch 0.4 installation. The cited user was using FreeSwitch 1.0.1 with his symptoms. I unfortunately do not currently have the ability to try a different version. I have seen a similar issue using r10558M in the past. Thanks, - Jesse From jesse.peterson at exbiblio.com Tue Feb 10 17:43:33 2009 From: jesse.peterson at exbiblio.com (Jesse Peterson) Date: Tue, 10 Feb 2009 17:43:33 -0800 Subject: [Freeswitch-users] SIP registration/retry/authorization problem In-Reply-To: <286A20C5-4088-4CD9-8BAA-D294777AF931@freeswitch.org> References: <86FB611F-47CE-4472-8CE7-E52F6F1AADF5@exbiblio.com> <286A20C5-4088-4CD9-8BAA-D294777AF931@freeswitch.org> Message-ID: <24DD6D23-C3EB-48A4-AC4F-587F7DD16361@exbiblio.com> On Feb 10, 2009, at 1:27 PM, Brian West wrote: > Try this: > > > > In sofia.conf.xml > > /b I'm not able to find any documentation on this setting. I think it may be newer than my version of FreeSwitch (1.0). What does it do? Thanks, - Jesse From brian at freeswitch.org Tue Feb 10 17:48:13 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 10 Feb 2009 19:48:13 -0600 Subject: [Freeswitch-users] SIP registration/retry/authorization problem In-Reply-To: <24DD6D23-C3EB-48A4-AC4F-587F7DD16361@exbiblio.com> References: <86FB611F-47CE-4472-8CE7-E52F6F1AADF5@exbiblio.com> <286A20C5-4088-4CD9-8BAA-D294777AF931@freeswitch.org> <24DD6D23-C3EB-48A4-AC4F-587F7DD16361@exbiblio.com> Message-ID: <980C8919-ECFF-47F4-824E-842EBB0293BF@freeswitch.org> I highly recommend you wipe the box/install and install from Scratch using SVN trunk /b On Feb 10, 2009, at 7:43 PM, Jesse Peterson wrote: > I'm not able to find any documentation on this setting. I think it may > be newer than my version of FreeSwitch (1.0). What does it do? > > Thanks, > - Jesse From ajlong at worldlink.net Tue Feb 10 18:22:31 2009 From: ajlong at worldlink.net (Adam Long) Date: Tue, 10 Feb 2009 21:22:31 -0500 Subject: [Freeswitch-users] Dynamic Dialplan In-Reply-To: <191c3a030902101432y5954389dje53f6dc05ce654cc@mail.gmail.com> References: <743112140902081921n2bb1bf49mfe36e5e2193c6b04@mail.gmail.com> <4990495C.9010008@post.cz> <49906889.9030801@freeswitch.org> <49906FE6.8000008@post.cz> <000301c98bb2$00d66560$02833020$@net> <191c3a030902101432y5954389dje53f6dc05ce654cc@mail.gmail.com> Message-ID: <006401c98bef$980e04f0$c82a0ed0$@net> There sure are lots of options J Does the perl implementation use a single persistent embedded interpreter or does it spawn a new interpreter per request? Things like persistent database handles come to mind. Performance would be drastically impacted if it were to spawn new interpreter per request. Any ideas? Regards, -Adam From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Tuesday, February 10, 2009 5:33 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Dynamic Dialplan oh yeah, i forgot about those too. python,perl and lua can all do that. On Tue, Feb 10, 2009 at 1:01 PM, Adam Long wrote: What about a mod_perl XML binding like the example here? http://wiki.freeswitch.org/wiki/Mod_perl and http://wiki.freeswitch.org/wiki/Mod_perl_and_Generating_XML Would this be faster than a setup like mod_curl_xml -> lighttpd -> FastCGI/Perl ? I guess it would depend on if mod_perl in FreeSWITCH spawns new interpreter per request or if it uses one interpreter instance with multiple threads to execute pre-loaded perl. Anyone know if this is the case? Regards, -Adam -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of kokoska rokoska Sent: Monday, February 09, 2009 1:03 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Dynamic Dialplan Raymond Chandler napsal(a): > kokoska rokoska wrote: >> Just my 2c: >> >> If you use curl with lighttpd and custom built fastcgi "C" responder (it >> is really simple with fcgi libs - even I can do it :-) performance could >> be not that bad. > hmmm, mod_xml_curl using C, interesting thought.. May be not the best way, but very simple. Well, it depends on what you have to do, but "directory" serving based on DB queries (this what I'm using it for) is very simple - just few lines of code. > all of the > complexities of writing your own module without the nice structured FS > API... I should say I have no idea how hard is to write custom FreeSWITCH module (may be I should try it :-), but the FS code is very nice! > although, as a benefit, i guess you do get a little extra latency ;-) > :-) Yes, you are right. And as a bonus some CPU utilization... Like I wrote above, I didn't say it is faster, but IMO it is very simple and not as slow as it looks (when using apache + php + apc). Best regards, kokoska.rokoska _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090210/024df86e/attachment-0002.html From chavpaskov at shaw.ca Tue Feb 10 18:39:17 2009 From: chavpaskov at shaw.ca (Chav Paskov) Date: Tue, 10 Feb 2009 18:39:17 -0800 Subject: [Freeswitch-users] Dynamic Dialplan In-Reply-To: <000301c98bb2$00d66560$02833020$@net> References: <743112140902081921n2bb1bf49mfe36e5e2193c6b04@mail.gmail.com> <4990495C.9010008@post.cz> <49906889.9030801@freeswitch.org> <49906FE6.8000008@post.cz> <000301c98bb2$00d66560$02833020$@net> Message-ID: <49923A55.1070009@shaw.ca> Adam Long wrote: > What about a mod_perl XML binding like the example here? > http://wiki.freeswitch.org/wiki/Mod_perl and > http://wiki.freeswitch.org/wiki/Mod_perl_and_Generating_XML > > Would this be faster than a setup like > > mod_curl_xml -> lighttpd -> FastCGI/Perl ? > > I guess it would depend on if mod_perl in FreeSWITCH spawns new interpreter per request > or if it uses one interpreter instance with multiple threads to execute pre-loaded perl. > > Anyone know if this is the case? > > Regards, > -Adam > > > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of kokoska rokoska > Sent: Monday, February 09, 2009 1:03 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Dynamic Dialplan > > > > > Raymond Chandler napsal(a): > >> kokoska rokoska wrote: >> >>> Just my 2c: >>> >>> If you use curl with lighttpd and custom built fastcgi "C" responder (it >>> is really simple with fcgi libs - even I can do it :-) performance could >>> be not that bad. >>> >> hmmm, mod_xml_curl using C, interesting thought.. >> > > May be not the best way, but very simple. > Well, it depends on what you have to do, but "directory" serving based > on DB queries (this what I'm using it for) is very simple - just few > lines of code. > > >> all of the >> complexities of writing your own module without the nice structured FS >> API... >> > > I should say I have no idea how hard is to write custom FreeSWITCH > module (may be I should try it :-), but the FS code is very nice! > > >> although, as a benefit, i guess you do get a little extra latency ;-) >> >> > > :-) Yes, you are right. And as a bonus some CPU utilization... > > Like I wrote above, I didn't say it is faster, but IMO it is very simple > and not as slow as it looks (when using apache + php + apc). > > Best regards, > > kokoska.rokoska > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > I'm developing something similar and now on testing mode. xml works just fine. The key is FCGI and web server. i think this week i made a progress in the right direction. will report the progress promptly when done. If you need hand pls drop an e-mail i do not mind sharing. and if it works for you i'll come with kind of how to instructions if anybody else is interested. Regards Chav From chavpaskov at shaw.ca Tue Feb 10 18:41:36 2009 From: chavpaskov at shaw.ca (Chav Paskov) Date: Tue, 10 Feb 2009 18:41:36 -0800 Subject: [Freeswitch-users] Dynamic Dialplan In-Reply-To: <006401c98bef$980e04f0$c82a0ed0$@net> References: <743112140902081921n2bb1bf49mfe36e5e2193c6b04@mail.gmail.com> <4990495C.9010008@post.cz> <49906889.9030801@freeswitch.org> <49906FE6.8000008@post.cz> <000301c98bb2$00d66560$02833020$@net> <191c3a030902101432y5954389dje53f6dc05ce654cc@mail.gmail.com> <006401c98bef$980e04f0$c82a0ed0$@net> Message-ID: <49923AE0.1030802@shaw.ca> Adam Long wrote: > > There sure are lots of options J > > > > Does the perl implementation use a single persistent embedded > interpreter or does it spawn a new interpreter per request? > > > > Things like persistent database handles come to mind. > > Performance would be drastically impacted if it were to spawn new > interpreter per request. > > > > Any ideas? > > > > Regards, > > -Adam > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of > *Anthony Minessale > *Sent:* Tuesday, February 10, 2009 5:33 PM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Dynamic Dialplan > > > > oh yeah, i forgot about those too. > python,perl and lua can all do that. > > On Tue, Feb 10, 2009 at 1:01 PM, Adam Long > wrote: > > What about a mod_perl XML binding like the example here? > http://wiki.freeswitch.org/wiki/Mod_perl and > http://wiki.freeswitch.org/wiki/Mod_perl_and_Generating_XML > > Would this be faster than a setup like > > mod_curl_xml -> lighttpd -> FastCGI/Perl ? > > I guess it would depend on if mod_perl in FreeSWITCH spawns new > interpreter per request > or if it uses one interpreter instance with multiple threads to > execute pre-loaded perl. > > Anyone know if this is the case? > > Regards, > -Adam > > > > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > > [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of > kokoska rokoska > Sent: Monday, February 09, 2009 1:03 PM > To: freeswitch-users at lists.freeswitch.org > > Subject: Re: [Freeswitch-users] Dynamic Dialplan > > > > Raymond Chandler napsal(a): > > kokoska rokoska wrote: > >> Just my 2c: > >> > >> If you use curl with lighttpd and custom built fastcgi "C" > responder (it > >> is really simple with fcgi libs - even I can do it :-) performance > could > >> be not that bad. > > hmmm, mod_xml_curl using C, interesting thought.. > > May be not the best way, but very simple. > Well, it depends on what you have to do, but "directory" serving based > on DB queries (this what I'm using it for) is very simple - just few > lines of code. > > > all of the > > complexities of writing your own module without the nice structured FS > > API... > > I should say I have no idea how hard is to write custom FreeSWITCH > module (may be I should try it :-), but the FS code is very nice! > > > although, as a benefit, i guess you do get a little extra latency ;-) > > > > :-) Yes, you are right. And as a bonus some CPU utilization... > > Like I wrote above, I didn't say it is faster, but IMO it is very simple > and not as slow as it looks (when using apache + php + apc). > > Best regards, > > kokoska.rokoska > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > Perl is the language i'm using. and is working exactly as i expect. Chav From helmut.kuper at ewetel.de Wed Feb 11 02:19:17 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Wed, 11 Feb 2009 11:19:17 +0100 Subject: [Freeswitch-users] little problem with gateway registration In-Reply-To: <191c3a030902100613m1b9a8fc1w787b3e177a373557@mail.gmail.com> References: <499152FE.7090909@ewetel.de> <191c3a030902100613m1b9a8fc1w787b3e177a373557@mail.gmail.com> Message-ID: <4992A625.5080902@ewetel.de> Hi Anthony, well currently both ip addresses and ports (of proxy and registrar) are the same. And it works good as it is now. :) regards Helmut On 10.02.2009 15:13, Anthony Minessale wrote: > register-proxy is for where it actually sends the packet but it will > still say the name of proxy in the packet. > Did you check the destination address of the packet it should end up > as the ip:port of that proxy. > From helmut.kuper at ewetel.de Wed Feb 11 04:09:32 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Wed, 11 Feb 2009 13:09:32 +0100 Subject: [Freeswitch-users] segfault when shoutcast plays mp3 and extension hangs up Message-ID: <4992BFFC.50006@ewetel.de> Hello, today I tried to play a mp3. It works fine until extension hangs up. Then FS (FreeSWITCH Version 1.0.trunk (11698M)) crashed with segfault. The mp3 file was generated by MP3Splitter (http://www.codevisions.de/hp/upload/_files/mp3splitter20.zip) as a piece out of a complete mp3 song. There is a good chance that it generates corrupt mp3s. At least those mp3s are playable in winamp and media player. My dialplan: FS console output shows problems in mp3 file: freeswitch at ippbx-prod-node0> 2009-02-11 11:45:43 [DEBUG] Span:1 Q.931() Timer 0 of call 0 (CRV: 2, State: 0) timed out Note: Illegal Audio-MPEG-Header 0x00000000 at offset 0x10ec15. Note: Trying to resync... Note: Hit end of (available) data during resync. 2009-02-11 11:45:44 [DEBUG] switch_ivr_play_say.c:1261 switch_ivr_play_file() done playing file ./start_fs.sh: line 6: 27201 Segmentation fault (core dumped) bin/freeswitch $1 Here are the backtraces: (gdb) bt #0 0xabe0e408 in mpg123_delete at plt () from /opt/app/voip/ippbx.prod/mod/mod_shout.so #1 0xabe13325 in ?? () from /opt/app/voip/ippbx.prod/mod/mod_shout.so #2 0xb7dc04b9 in switch_core_file_seek (fh=0xa76fef28, cur_pos=0xa76fef88, samples=0, whence=1) at src/switch_core_file.c:345 #3 0xb7dfab82 in switch_ivr_play_file (session=0xa7700030, fh=0xa76fef28, file=0xa7711ed8 "/opt/app/voip/ippbx.prod/sounds/de/callie/qet.mp3", args=0xa76ff0d8) at src/switch_ivr_play_say.c:1262 #4 0xad71da25 in ?? () from /opt/app/voip/ippbx.prod/mod/mod_dptools.so #5 0xb7dc6c54 in switch_core_session_exec (session=0xa7700030, application_interface=0xb346b388, arg=0xb34fd790 "qet.mp3") at src/switch_core_session.c:1332 #6 0xb7dc70fe in switch_core_session_execute_application (session=0xa7700030, app=0xb34fd780 "playback", arg=0xb34fd790 "qet.mp3") at src/switch_core_session.c:1254 #7 0xb7dc93a4 in switch_core_session_run (session=0xa7700030) at src/switch_core_state_machine.c:155 #8 0xb7dc6725 in switch_core_session_thread (thread=0xb34fd3d0, obj=0xa7700030) at src/switch_core_session.c:940 #9 0xb7e30bf6 in dummy_worker (opaque=0xb34fd3d0) at threadproc/unix/thread.c:138 #10 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #11 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 Current language: auto; currently asm (gdb) bt full #0 0xabe0e408 in mpg123_delete at plt () from /opt/app/voip/ippbx.prod/mod/mod_shout.so No locals. #1 0xabe13325 in ?? () from /opt/app/voip/ippbx.prod/mod/mod_shout.so No locals. #2 0xb7dc04b9 in switch_core_file_seek (fh=0xa76fef28, cur_pos=0xa76fef88, samples=0, whence=1) at src/switch_core_file.c:345 status = 4294967295 __PRETTY_FUNCTION__ = "switch_core_file_seek" #3 0xb7dfab82 in switch_ivr_play_file (session=0xa7700030, fh=0xa76fef28, file=0xa7711ed8 "/opt/app/voip/ippbx.prod/sounds/de/callie/qet.mp3", args=0xa76ff0d8) at src/switch_ivr_play_say.c:1262 channel = (switch_channel_t *) 0xa7704598 dtmf = {digit = 0 '\0', duration = 0} interval = 2909916864 samples = 320 framelen = 640 sample_start = olen = 320 llen = 320 write_frame = {codec = 0xa76fefb8, source = 0x0, packet = 0x0, packetlen = 0, data = 0xa7743f30, datalen = 640, buflen = 32768, samples = 320, rate = 16000, payload = 0 '\0', timestamp = 0, seq = 0, ssrc = 0, m = SWITCH_FALSE, flags = 0} timer = {interval = 0, flags = 0, samples = 0, samplecount = 0, timer_interface = 0x0, memory_pool = 0x0, private_info = 0x0, diff = 0, tick = 0} codec = {codec_interface = 0x80cd8f8, implementation = 0x80cdf20, fmtp_in = 0x0, fmtp_out = 0x0, codec_settings = {quality = 0, complexity = 0, enhancement = 0, vad = 0, vbr = 0, vbr_quality = 0, abr = 0, dtx = 0, preproc = 0, pp_vad = 0, pp_agc = 0, pp_agc_level = 0, pp_denoise = 0, pp_dereverb = 0, pp_dereverb_decay = 0, pp_dereverb_level = 0}, flags = 3, memory_pool = 0xb34fbb38, private_info = 0x0, agreed_pt = 0 '\0', mutex = 0xa7711f10} pool = (switch_memory_pool_t *) 0xb34fbb38 status = SWITCH_STATUS_SUCCESS lfh = {file_interface = 0x8136258, flags = 3085, fd = 0x0, samples = 0, samplerate = 16000, native_rate = 16000, channels = 1 '\001', format = 0, sections = 0, seekable = 0, sample_count = 729088, speed = 0, memory_pool = 0xa7712040, prebuf = 0, interval = 0, private_info = 0xa7714048, handler = 0x0, pos = 0, audio_buffer = 0xb3476f88, sp_audio_buffer = 0x0, thresh = 0, silence_hits = 0, offset_pos = 364640, last_pos = 0, vol = 0, resampler = 0x0, buffer = 0x0, dbuf = 0x0, dbuflen = 0, pre_buffer = 0x0, pre_buffer_data = 0x0, pre_buffer_datalen = 0, file = 0xb7eb71f0 "src/switch_ivr_play_say.c", func = 0xb7eb794b "switch_ivr_play_file", line = 894} read_codec = (switch_codec_t *) 0xb34fcb90 p = 0xb7eaf8d4 "current_application" asis = 0 '\0' prefix = timer_name = 0x0 prebuf = eof = 1 bread = __func__ = "switch_ivr_play_file" __PRETTY_FUNCTION__ = "switch_ivr_play_file" #4 0xad71da25 in ?? () from /opt/app/voip/ippbx.prod/mod/mod_dptools.so No locals. #5 0xb7dc6c54 in switch_core_session_exec (session=0xa7700030, application_interface=0xb346b388, arg=0xb34fd790 "qet.mp3") at src/switch_core_session.c:1332 log = lp = event = (switch_event_t *) 0x0 var = channel = (switch_channel_t *) 0xa7704598 expanded = 0xb34fd790 "qet.mp3" app = 0xad725133 "playback" __PRETTY_FUNCTION__ = "switch_core_session_exec" __func__ = "switch_core_session_exec" #6 0xb7dc70fe in switch_core_session_execute_application (session=0xa7700030, app=0xb34fd780 "playback", arg=0xb34fd790 "qet.mp3") at src/switch_core_session.c:1254 application_interface = (switch_application_interface_t *) 0xb346b388 __func__ = "switch_core_session_execute_application" #7 0xb7dc93a4 in switch_core_session_run (session=0xa7700030) at src/switch_core_state_machine.c:155 proceed = global_proceed = do_extra_handlers = state = endstate = endpoint_interface = driver_state_handler = (const switch_state_handler_table_t *) 0xb331f860 application_state_handler = thread_id = 3084680427 env = {{__jmpbuf = {0, 0, 0, 0, 0, 0}, __mask_was_saved = 0, __saved_mask = {__val = {0 }}}} sig = __func__ = "switch_core_session_run" __PRETTY_FUNCTION__ = "switch_core_session_run" #8 0xb7dc6725 in switch_core_session_thread (thread=0xb34fd3d0, obj=0xa7700030) at src/switch_core_session.c:940 session = (switch_core_session_t *) 0xa7700030 event = event_str = 0x0 val = __func__ = "switch_core_session_thread" __PRETTY_FUNCTION__ = "switch_core_session_thread" #9 0xb7e30bf6 in dummy_worker (opaque=0xb34fd3d0) at threadproc/unix/thread.c:138 No locals. #10 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 No symbol table info available. #11 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. (gdb) thread apply all bt Thread 31 (process 27201): #0 0xb7f14410 in __kernel_vsyscall () #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 #2 0xb7e32799 in apr_sleep (t=1000000) at time/unix/time.c:246 #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 #4 0xb7db9154 in switch_console_loop () at src/switch_console.c:792 #5 0xb7dcedf0 in switch_core_runtime_loop (bg=0) at src/switch_core.c:659 #6 0x0804a36a in main (argc=1, argv=0xbffe65c4) at src/switch.c:666 Thread 30 (process 27202): #0 0xb7f14410 in __kernel_vsyscall () #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 #2 0xb7e32799 in apr_sleep (t=100000) at time/unix/time.c:246 #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 #4 0xb7dbe2a4 in pool_thread (thread=0xb7a07da8, obj=0x0) at src/switch_core_memory.c:421 #5 0xb7e30bf6 in dummy_worker (opaque=0xb7a07da8) at threadproc/unix/thread.c:138 #6 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #7 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 Thread 29 (process 27203): #0 0xb7f14410 in __kernel_vsyscall () #1 0xb7d2eaa5 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/tls/i686/cmov/libpthread.so.0 #2 0xb7e2acaa in apr_thread_cond_wait (cond=0xb6fb53b8, mutex=0xb6fb5388) at locks/unix/thread_cond.c:68 #3 0xb7e21a3c in apr_queue_pop (queue=0xb6fb5358, data=0xb6ec03a8) at misc/apr_queue.c:276 #4 0xb7dadd44 in switch_queue_pop (queue=0xb6fb5358, data=0xb6ec03a8) at src/switch_apr.c:879 #5 0xb7ddf879 in switch_event_dispatch_thread (thread=0x8068140, obj=0xb6fb5358) at src/switch_event.c:230 #6 0xb7e30bf6 in dummy_worker (opaque=0x8068140) at threadproc/unix/thread.c:138 #7 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #8 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 Thread 28 (process 27204): #0 0xb7f14410 in __kernel_vsyscall () #1 0xb7d2eaa5 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/tls/i686/cmov/libpthread.so.0 #2 0xb7e2acaa in apr_thread_cond_wait (cond=0x805e380, mutex=0x805e350) at locks/unix/thread_cond.c:68 #3 0xb7e21a3c in apr_queue_pop (queue=0x805e320, data=0xb66bf3a8) at misc/apr_queue.c:276 #4 0xb7dadd44 in switch_queue_pop (queue=0x805e320, data=0xb66bf3a8) at src/switch_apr.c:879 #5 0xb7ddec2d in switch_event_thread (thread=0x8068160, obj=0x805e320) at src/switch_event.c:273 #6 0xb7e30bf6 in dummy_worker (opaque=0x8068160) at threadproc/unix/thread.c:138 #7 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #8 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 Thread 27 (process 27205): #0 0xb7f14410 in __kernel_vsyscall () #1 0xb7d2eaa5 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/tls/i686/cmov/libpthread.so.0 #2 0xb7e2acaa in apr_thread_cond_wait (cond=0xb71d4b38, mutex=0xb71d4b08) at locks/unix/thread_cond.c:68 #3 0xb7e21a3c in apr_queue_pop (queue=0xb71d4ad8, data=0xb5ebe3a8) at misc/apr_queue.c:276 #4 0xb7dadd44 in switch_queue_pop (queue=0xb71d4ad8, data=0xb5ebe3a8) at src/switch_apr.c:879 #5 0xb7ddec2d in switch_event_thread (thread=0x8068180, obj=0xb71d4ad8) at src/switch_event.c:273 #6 0xb7e30bf6 in dummy_worker (opaque=0x8068180) at threadproc/unix/thread.c:138 #7 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 ---Type to continue, or q to quit--- #8 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 Thread 26 (process 27206): #0 0xb7f14410 in __kernel_vsyscall () #1 0xb7d2eaa5 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/tls/i686/cmov/libpthread.so.0 #2 0xb7e2acaa in apr_thread_cond_wait (cond=0xb7171b38, mutex=0xb7171b08) at locks/unix/thread_cond.c:68 #3 0xb7e21a3c in apr_queue_pop (queue=0xb7171ad8, data=0xb56bd3a8) at misc/apr_queue.c:276 #4 0xb7dadd44 in switch_queue_pop (queue=0xb7171ad8, data=0xb56bd3a8) at src/switch_apr.c:879 #5 0xb7ddec2d in switch_event_thread (thread=0x80681a0, obj=0xb7171ad8) at src/switch_event.c:273 #6 0xb7e30bf6 in dummy_worker (opaque=0x80681a0) at threadproc/unix/thread.c:138 #7 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #8 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 Thread 25 (process 27207): #0 0xb7f14410 in __kernel_vsyscall () #1 0xb7d2eaa5 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/tls/i686/cmov/libpthread.so.0 #2 0xb7e2acaa in apr_thread_cond_wait (cond=0xb6fb54a8, mutex=0xb6fb5478) at locks/unix/thread_cond.c:68 #3 0xb7e21a3c in apr_queue_pop (queue=0xb6fb5448, data=0xb4dce3a8) at misc/apr_queue.c:276 #4 0xb7dadd44 in switch_queue_pop (queue=0xb6fb5448, data=0xb4dce3a8) at src/switch_apr.c:879 #5 0xb7e082fd in log_thread (thread=0xb4e30ae0, obj=0x0) at src/switch_log.c:209 #6 0xb7e30bf6 in dummy_worker (opaque=0xb4e30ae0) at threadproc/unix/thread.c:138 #7 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #8 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 Thread 24 (process 27210): #0 0xb7f14410 in __kernel_vsyscall () #1 0xb7d2eaa5 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/tls/i686/cmov/libpthread.so.0 #2 0xb7e2acaa in apr_thread_cond_wait (cond=0x80c0c58, mutex=0x80c0c28) at locks/unix/thread_cond.c:68 #3 0xb7daed54 in switch_thread_cond_wait (cond=0x80c0c58, mutex=0x80c0c28) at src/switch_apr.c:359 #4 0xb7e11266 in switch_cond_next () at src/switch_time.c:203 #5 0xb7dc27a5 in switch_core_sql_thread (thread=0xb3567ae8, obj=0x0) at src/switch_core_sqldb.c:220 #6 0xb7e30bf6 in dummy_worker (opaque=0xb3567ae8) at threadproc/unix/thread.c:138 #7 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #8 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 Thread 23 (process 27211): #0 0xb7f14410 in __kernel_vsyscall () #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 #2 0xb7e32799 in apr_sleep (t=500000) at time/unix/time.c:246 #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 #4 0xb7dd2af4 in switch_scheduler_task_thread (thread=0x80baa90, obj=0x0) at src/switch_scheduler.c:171 #5 0xb7e30bf6 in dummy_worker (opaque=0x80baa90) at threadproc/unix/thread.c:138 #6 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #7 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 Thread 22 (process 27212): #0 0xb7f14410 in __kernel_vsyscall () #1 0xb7c7d676 in epoll_wait () from /lib/tls/i686/cmov/libc.so.6 #2 0xb32b7126 in su_epoll_port_wait_events (self=0x80c6340, tout=1000) at su_epoll_port.c:491 #3 0xb32bd9b8 in su_base_port_step (self=0x80c6340, tout=1000) at su_base_port.c:442 #4 0xb32b8551 in su_port_step (self=0x80c6340, tout=1000) at su_port.h:326 ---Type to continue, or q to quit--- #5 0xb32b8521 in su_root_step (self=0x80c68f0, tout=1000) at su_root.c:730 #6 0xb31e8f29 in sofia_profile_thread_run (thread=0x80d2aa8, obj=0x80d1e10) at sofia.c:831 #7 0xb7e30bf6 in dummy_worker (opaque=0x80d2aa8) at threadproc/unix/thread.c:138 #8 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #9 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 Thread 21 (process 27213): #0 0xb7f14410 in __kernel_vsyscall () #1 0xb7c7d676 in epoll_wait () from /lib/tls/i686/cmov/libc.so.6 #2 0xb32b7126 in su_epoll_port_wait_events (self=0x80dac30, tout=1000) at su_epoll_port.c:491 #3 0xb32bd89b in su_base_port_run (self=0x80dac30) at su_base_port.c:342 #4 0xb32b842b in su_port_run (self=0x80dac30) at su_port.h:312 #5 0xb32b8408 in su_root_run (self=0x80dacb0) at su_root.c:691 #6 0xb32a8b31 in su_pthread_port_clone_main (varg=0xb311c0a8) at su_pthread_port.c:321 #7 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #8 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 Thread 20 (process 27214): #0 0xb7f14410 in __kernel_vsyscall () #1 0xb7c7d676 in epoll_wait () from /lib/tls/i686/cmov/libc.so.6 #2 0xb32b7126 in su_epoll_port_wait_events (self=0x80da208, tout=1000) at su_epoll_port.c:491 #3 0xb32bd9b8 in su_base_port_step (self=0x80da208, tout=1000) at su_base_port.c:442 #4 0xb32b8551 in su_port_step (self=0x80da208, tout=1000) at su_port.h:326 #5 0xb32b8521 in su_root_step (self=0x80d7558, tout=1000) at su_root.c:730 #6 0xb31e8f29 in sofia_profile_thread_run (thread=0x80dde88, obj=0x80dd650) at sofia.c:831 #7 0xb7e30bf6 in dummy_worker (opaque=0x80dde88) at threadproc/unix/thread.c:138 #8 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #9 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 Thread 19 (process 27215): #0 0xb7f14410 in __kernel_vsyscall () #1 0xb7c7d676 in epoll_wait () from /lib/tls/i686/cmov/libc.so.6 #2 0xb32b7126 in su_epoll_port_wait_events (self=0x80e5228, tout=1000) at su_epoll_port.c:491 #3 0xb32bd89b in su_base_port_run (self=0x80e5228) at su_base_port.c:342 #4 0xb32b842b in su_port_run (self=0x80e5228) at su_port.h:312 #5 0xb32b8408 in su_root_run (self=0x80e3b40) at su_root.c:691 #6 0xb32a8b31 in su_pthread_port_clone_main (varg=0xb211a0a8) at su_pthread_port.c:321 #7 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #8 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 Thread 18 (process 27216): #0 0xb7f14410 in __kernel_vsyscall () #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 #2 0xb7e32799 in apr_sleep (t=10000) at time/unix/time.c:246 #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 #4 0xb31d4b37 in sofia_profile_worker_thread_run (thread=0x80d2b88, obj=0x80d1e10) at sofia.c:656 #5 0xb7e30bf6 in dummy_worker (opaque=0x80d2b88) at threadproc/unix/thread.c:138 #6 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #7 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 Thread 17 (process 27217): ---Type to continue, or q to quit--- #0 0xb7f14410 in __kernel_vsyscall () #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 #2 0xb7e32799 in apr_sleep (t=10000) at time/unix/time.c:246 #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 #4 0xb31d4b37 in sofia_profile_worker_thread_run (thread=0x80ddf68, obj=0x80dd650) at sofia.c:656 #5 0xb7e30bf6 in dummy_worker (opaque=0x80ddf68) at threadproc/unix/thread.c:138 #6 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #7 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 Thread 16 (process 27218): #0 0xb7f14410 in __kernel_vsyscall () #1 0xb7c7d676 in epoll_wait () from /lib/tls/i686/cmov/libc.so.6 #2 0xb32b7126 in su_epoll_port_wait_events (self=0x80e66d8, tout=1000) at su_epoll_port.c:491 #3 0xb32bd9b8 in su_base_port_step (self=0x80e66d8, tout=1000) at su_base_port.c:442 #4 0xb32b8551 in su_port_step (self=0x80e66d8, tout=1000) at su_port.h:326 #5 0xb32b8521 in su_root_step (self=0x80e49f0, tout=1000) at su_root.c:730 #6 0xb31e8f29 in sofia_profile_thread_run (thread=0x80e8710, obj=0x80e7e90) at sofia.c:831 #7 0xb7e30bf6 in dummy_worker (opaque=0x80e8710) at threadproc/unix/thread.c:138 #8 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #9 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 Thread 15 (process 27219): #0 0xb7f14410 in __kernel_vsyscall () #1 0xb7c7d676 in epoll_wait () from /lib/tls/i686/cmov/libc.so.6 #2 0xb32b7126 in su_epoll_port_wait_events (self=0x80ee648, tout=1000) at su_epoll_port.c:491 #3 0xb32bd89b in su_base_port_run (self=0x80ee648) at su_base_port.c:342 #4 0xb32b842b in su_port_run (self=0x80ee648) at su_port.h:312 #5 0xb32b8408 in su_root_run (self=0x80f0bd0) at su_root.c:691 #6 0xb32a8b31 in su_pthread_port_clone_main (varg=0xb00b20a8) at su_pthread_port.c:321 #7 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #8 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 Thread 14 (process 27220): #0 0xb7f14410 in __kernel_vsyscall () #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 #2 0xb7e32799 in apr_sleep (t=10000) at time/unix/time.c:246 #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 #4 0xb31d4b37 in sofia_profile_worker_thread_run (thread=0x80e87f0, obj=0x80e7e90) at sofia.c:656 #5 0xb7e30bf6 in dummy_worker (opaque=0x80e87f0) at threadproc/unix/thread.c:138 #6 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #7 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 Thread 13 (process 27221): #0 0xb7f14410 in __kernel_vsyscall () #1 0xb7c72c07 in poll () from /lib/tls/i686/cmov/libc.so.6 #2 0xb337af00 in wanpipe_wait (zchan=0xb3446128, flags=0xae751f80, to=100) at src/ozmod/ozmod_wanpipe/ozmod_wanpipe.c:255 #3 0xae837ae7 in zap_channel_wait (zchan=0x1, flags=0xae751f80, to=100) at src/zap_io.c:1479 #4 0xae80aee8 in zap_isdn_run (me=0xb3416528, obj=0xb341dbc8) at src/ozmod/ozmod_isdn/ozmod_isdn.c:1725 #5 0xae8421ba in thread_launch (args=0xb3416528) at src/zap_threadmutex.c:74 #6 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #7 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 ---Type to continue, or q to quit--- Thread 12 (process 27222): #0 0xb7f14410 in __kernel_vsyscall () #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 #2 0xb7e32799 in apr_sleep (t=100000) at time/unix/time.c:246 #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 #4 0xb31f929e in sofia_presence_event_thread_run (thread=0x80cf958, obj=0x0) at sofia_presence.c:664 #5 0xb7e30bf6 in dummy_worker (opaque=0x80cf958) at threadproc/unix/thread.c:138 #6 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #7 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 Thread 11 (process 27223): #0 0xb7f14410 in __kernel_vsyscall () #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 #2 0xb7e32799 in apr_sleep (t=1000000) at time/unix/time.c:246 #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 #4 0xad706ea6 in ?? () from /opt/app/voip/ippbx.prod/mod/mod_fifo.so #5 0xb7e30bf6 in dummy_worker (opaque=0xad386b90) at threadproc/unix/thread.c:138 #6 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #7 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 Thread 10 (process 27225): #0 0xb7f14410 in __kernel_vsyscall () #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 #2 0xb7e32799 in apr_sleep (t=1000000) at time/unix/time.c:246 #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 #4 0xabdf94ab in ?? () from /opt/app/voip/ippbx.prod/mod/mod_local_stream.so #5 0xb7e30bf6 in dummy_worker (opaque=0x81386e0) at threadproc/unix/thread.c:138 #6 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #7 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 Thread 9 (process 27226): #0 0xb7f14410 in __kernel_vsyscall () #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 #2 0xb7e32799 in apr_sleep (t=1000000) at time/unix/time.c:246 #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 #4 0xabdf94ab in ?? () from /opt/app/voip/ippbx.prod/mod/mod_local_stream.so #5 0xb7e30bf6 in dummy_worker (opaque=0x815e728) at threadproc/unix/thread.c:138 #6 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #7 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 Thread 8 (process 27227): #0 0xb7f14410 in __kernel_vsyscall () #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 #2 0xb7e32799 in apr_sleep (t=1000000) at time/unix/time.c:246 #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 #4 0xabdf94ab in ?? () from /opt/app/voip/ippbx.prod/mod/mod_local_stream.so #5 0xb7e30bf6 in dummy_worker (opaque=0x8184770) at threadproc/unix/thread.c:138 #6 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #7 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 ---Type to continue, or q to quit--- Thread 7 (process 27228): #0 0xb7f14410 in __kernel_vsyscall () #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 #2 0xb7e32799 in apr_sleep (t=1000) at time/unix/time.c:246 #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 #4 0xb7e12a59 in softtimer_runtime () at src/switch_time.c:459 #5 0xb7dd4fb3 in switch_loadable_module_exec (thread=0x80bf8d8, obj=0x80bf6c8) at src/switch_loadable_module.c:93 #6 0xb7e30bf6 in dummy_worker (opaque=0x80bf8d8) at threadproc/unix/thread.c:138 #7 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #8 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 Thread 6 (process 27229): #0 0xb7f14410 in __kernel_vsyscall () #1 0xb7d31bb8 in accept () from /lib/tls/i686/cmov/libpthread.so.0 #2 0xb7e2f90d in apr_socket_accept (new=0xaa2d434c, sock=0x81bbbb0, connection_context=0x81bdaa8) at network_io/unix/sockets.c:187 #3 0xb7dae3fb in switch_socket_accept (new_sock=0xaa2d434c, sock=0x81bbbb0, pool=0x81bdaa8) at src/switch_apr.c:664 #4 0xb33262f2 in mod_event_socket_runtime () at mod_event_socket.c:2134 #5 0xb7dd4fb3 in switch_loadable_module_exec (thread=0x80bfb40, obj=0x80bf930) at src/switch_loadable_module.c:93 #6 0xb7e30bf6 in dummy_worker (opaque=0x80bfb40) at threadproc/unix/thread.c:138 #7 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #8 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 Thread 5 (process 27230): #0 0xb7f14410 in __kernel_vsyscall () #1 0xb7c72c07 in poll () from /lib/tls/i686/cmov/libc.so.6 #2 0xb33ce86e in ?? () from /opt/app/voip/ippbx.prod/mod/mod_xml_rpc.so #3 0xb33c1464 in ChanSwitchAccept (chanSwitchP=0x81f5030, channelPP=0xa9ad30e0, channelInfoPP=0xa9ad30dc, errorP=0xa9ad30e4) at ../../../../libs/xmlrpc-c/lib/abyss/src/chanswitch.c:149 #4 0xb33cd37e in ServerRun (serverP=0xb33ffe4c) at ../../../../libs/xmlrpc-c/lib/abyss/src/server.c:908 #5 0xb33be832 in mod_xml_rpc_runtime () at mod_xml_rpc.c:837 #6 0xb7dd4fb3 in switch_loadable_module_exec (thread=0x80bfda8, obj=0x80bfb98) at src/switch_loadable_module.c:93 #7 0xb7e30bf6 in dummy_worker (opaque=0x80bfda8) at threadproc/unix/thread.c:138 #8 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #9 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 Thread 4 (process 27231): #0 0xb7f14410 in __kernel_vsyscall () #1 0xb7d3199b in read () from /lib/tls/i686/cmov/libpthread.so.0 #2 0xb7e9dfb3 in read_char (el=0x81c5040, cp=0xa92d235b "?|???`!\b\200a\"\b?#-?;\237?@P\034\b?#-?\a") at read.c:294 #3 0xb7e9da9c in el_getc (el=0x81c5040, cp=0xa92d235b "?|???`!\b\200a\"\b?#-?;\237?@P\034\b?#-?\a") at read.c:362 #4 0xb7e9dbdf in el_gets (el=0x81c5040, nread=0xa92d23a8) at read.c:241 #5 0xb7db9f3b in console_thread (thread=0x82102d0, obj=0x8210248) at src/switch_console.c:441 #6 0xb7e30bf6 in dummy_worker (opaque=0x82102d0) at threadproc/unix/thread.c:138 #7 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #8 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 Thread 3 (process 27235): #0 0xb7f14410 in __kernel_vsyscall () #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 #2 0xb7e32799 in apr_sleep (t=100000) at time/unix/time.c:246 #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 ---Type to continue, or q to quit--- #4 0xb7df019e in switch_ivr_originate (session=0xb34dd080, bleg=0xa8ad10c0, cause=0xa8ad10bc, bridgeto=0x823a528 "openzap/1/a/04855711", timelimit_sec=60, table=0xb7ecda20, cid_name_override=0x0, cid_num_override=0x0, caller_profile_override=0x0, ovars=0x0, flags=) at src/switch_ivr_originate.c:1793 #5 0xad7231af in ?? () from /opt/app/voip/ippbx.prod/mod/mod_dptools.so #6 0xb7dc6c54 in switch_core_session_exec (session=0xb34dd080, application_interface=0xb346ba10, arg=0x823a528 "openzap/1/a/04855711") at src/switch_core_session.c:1332 #7 0xb7dc70fe in switch_core_session_execute_application (session=0xb34dd080, app=0x823a520 "bridge", arg=0x823a528 "openzap/1/a/04855711") at src/switch_core_session.c:1254 #8 0xb7dc93a4 in switch_core_session_run (session=0xb34dd080) at src/switch_core_state_machine.c:155 #9 0xb7dc6725 in switch_core_session_thread (thread=0x823a050, obj=0xb34dd080) at src/switch_core_session.c:940 #10 0xb7e30bf6 in dummy_worker (opaque=0x823a050) at threadproc/unix/thread.c:138 #11 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #12 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 Thread 2 (process 27236): #0 0xb7f14410 in __kernel_vsyscall () #1 0xb7c72c07 in poll () from /lib/tls/i686/cmov/libc.so.6 #2 0xb337af00 in wanpipe_wait (zchan=0xb341f9d8, flags=0xa80fee60, to=40) at src/ozmod/ozmod_wanpipe/ozmod_wanpipe.c:255 #3 0xae837ae7 in zap_channel_wait (zchan=0x1, flags=0xa80fee60, to=40) at src/zap_io.c:1479 #4 0xae849059 in channel_read_frame (session=0xb34e5400, frame=0xa80ff170, flags=0, stream_id=0) at mod_openzap.c:593 #5 0xb7dcbe6f in switch_core_session_read_frame (session=0xb34e5400, frame=0xa80ff170, flags=0, stream_id=0) at src/switch_core_io.c:161 #6 0xb7e054ec in switch_ivr_sleep (session=0xb34e5400, ms=10, sync=SWITCH_FALSE, args=0x0) at src/switch_ivr.c:262 #7 0xb7deaf94 in originate_on_consume_media_transmit (session=0xb34e5400) at src/switch_ivr_originate.c:47 #8 0xb7dc8b74 in switch_core_session_run (session=0xb34e5400) at src/switch_core_state_machine.c:476 #9 0xb7dc6725 in switch_core_session_thread (thread=0xb34f1060, obj=0xb34e5400) at src/switch_core_session.c:940 #10 0xb7e30bf6 in dummy_worker (opaque=0xb34f1060) at threadproc/unix/thread.c:138 #11 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #12 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 Thread 1 (process 27234): #0 0xabe0e408 in mpg123_delete at plt () from /opt/app/voip/ippbx.prod/mod/mod_shout.so #1 0xabe13325 in ?? () from /opt/app/voip/ippbx.prod/mod/mod_shout.so #2 0xb7dc04b9 in switch_core_file_seek (fh=0xa76fef28, cur_pos=0xa76fef88, samples=0, whence=1) at src/switch_core_file.c:345 #3 0xb7dfab82 in switch_ivr_play_file (session=0xa7700030, fh=0xa76fef28, file=0xa7711ed8 "/opt/app/voip/ippbx.prod/sounds/de/callie/qet.mp3", args=0xa76ff0d8) at src/switch_ivr_play_say.c:1262 #4 0xad71da25 in ?? () from /opt/app/voip/ippbx.prod/mod/mod_dptools.so #5 0xb7dc6c54 in switch_core_session_exec (session=0xa7700030, application_interface=0xb346b388, arg=0xb34fd790 "qet.mp3") at src/switch_core_session.c:1332 #6 0xb7dc70fe in switch_core_session_execute_application (session=0xa7700030, app=0xb34fd780 "playback", arg=0xb34fd790 "qet.mp3") at src/switch_core_session.c:1254 #7 0xb7dc93a4 in switch_core_session_run (session=0xa7700030) at src/switch_core_state_machine.c:155 #8 0xb7dc6725 in switch_core_session_thread (thread=0xb34fd3d0, obj=0xa7700030) at src/switch_core_session.c:940 #9 0xb7e30bf6 in dummy_worker (opaque=0xb34fd3d0) at threadproc/unix/thread.c:138 #10 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 #11 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 (gdb) thread apply all bt full Thread 31 (process 27201): #0 0xb7f14410 in __kernel_vsyscall () No symbol table info available. #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. #2 0xb7e32799 in apr_sleep (t=1000000) at time/unix/time.c:246 tv = {tv_sec = 0, tv_usec = 940000} #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 No locals. #4 0xb7db9154 in switch_console_loop () at src/switch_console.c:792 arg = 1 thread = (switch_thread_t *) 0x82102d0 thd_attr = (switch_threadattr_t *) 0x8210298 pool = (switch_memory_pool_t *) 0x8210248 __func__ = "switch_console_loop" __PRETTY_FUNCTION__ = "switch_console_loop" #5 0xb7dcedf0 in switch_core_runtime_loop (bg=0) at src/switch_core.c:659 No locals. #6 0x0804a36a in main (argc=1, argv=0xbffe65c4) at src/switch.c:666 pid_path = "/opt/app/voip/ippbx.prod/log/freeswitch.pid", '\0' pid_buffer = "27201", '\0' old_pid_buffer = "27150", '\0' pid_len = 5 old_pid_len = 5 err = 0x0 nf = 0 runas_user = 0x0 runas_group = 0x0 nc = 0 pid = 27201 x = 1111804576 die = 0 alt_dirs = 0 known_opt = -1208927888 high_prio = 0 flags = 1 ret = destroy_status = fd = (switch_file_t *) 0x80529b0 pool = (switch_memory_pool_t *) 0x80528f0 __PRETTY_FUNCTION__ = "main" Thread 30 (process 27202): #0 0xb7f14410 in __kernel_vsyscall () No symbol table info available. #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. #2 0xb7e32799 in apr_sleep (t=100000) at time/unix/time.c:246 tv = {tv_sec = 0, tv_usec = 100000} #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 ---Type to continue, or q to quit--- No locals. #4 0xb7dbe2a4 in pool_thread (thread=0xb7a07da8, obj=0x0) at src/switch_core_memory.c:421 No locals. #5 0xb7e30bf6 in dummy_worker (opaque=0xb7a07da8) at threadproc/unix/thread.c:138 No locals. #6 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 No symbol table info available. #7 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. Thread 29 (process 27203): #0 0xb7f14410 in __kernel_vsyscall () No symbol table info available. #1 0xb7d2eaa5 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/tls/i686/cmov/libpthread.so.0 No symbol table info available. #2 0xb7e2acaa in apr_thread_cond_wait (cond=0xb6fb53b8, mutex=0xb6fb5388) at locks/unix/thread_cond.c:68 rv = -512 #3 0xb7e21a3c in apr_queue_pop (queue=0xb6fb5358, data=0xb6ec03a8) at misc/apr_queue.c:276 rv = 0 #4 0xb7dadd44 in switch_queue_pop (queue=0xb6fb5358, data=0xb6ec03a8) at src/switch_apr.c:879 No locals. #5 0xb7ddf879 in switch_event_dispatch_thread (thread=0x8068140, obj=0xb6fb5358) at src/switch_event.c:230 pop = (void *) 0x0 event = (switch_event_t *) 0x0 my_id = 0 __func__ = "switch_event_dispatch_thread" #6 0xb7e30bf6 in dummy_worker (opaque=0x8068140) at threadproc/unix/thread.c:138 No locals. #7 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 No symbol table info available. #8 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. Thread 28 (process 27204): #0 0xb7f14410 in __kernel_vsyscall () No symbol table info available. #1 0xb7d2eaa5 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/tls/i686/cmov/libpthread.so.0 No symbol table info available. #2 0xb7e2acaa in apr_thread_cond_wait (cond=0x805e380, mutex=0x805e350) at locks/unix/thread_cond.c:68 rv = -512 #3 0xb7e21a3c in apr_queue_pop (queue=0x805e320, data=0xb66bf3a8) at misc/apr_queue.c:276 rv = 0 #4 0xb7dadd44 in switch_queue_pop (queue=0x805e320, data=0xb66bf3a8) at src/switch_apr.c:879 No locals. #5 0xb7ddec2d in switch_event_thread (thread=0x8068160, obj=0x805e320) at src/switch_event.c:273 pop = (void *) 0x0 index = 0 my_id = 0 __func__ = "switch_event_thread" #6 0xb7e30bf6 in dummy_worker (opaque=0x8068160) at threadproc/unix/thread.c:138 No locals. ---Type to continue, or q to quit--- #7 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 No symbol table info available. #8 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. Thread 27 (process 27205): #0 0xb7f14410 in __kernel_vsyscall () No symbol table info available. #1 0xb7d2eaa5 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/tls/i686/cmov/libpthread.so.0 No symbol table info available. #2 0xb7e2acaa in apr_thread_cond_wait (cond=0xb71d4b38, mutex=0xb71d4b08) at locks/unix/thread_cond.c:68 rv = -512 #3 0xb7e21a3c in apr_queue_pop (queue=0xb71d4ad8, data=0xb5ebe3a8) at misc/apr_queue.c:276 rv = 0 #4 0xb7dadd44 in switch_queue_pop (queue=0xb71d4ad8, data=0xb5ebe3a8) at src/switch_apr.c:879 No locals. #5 0xb7ddec2d in switch_event_thread (thread=0x8068180, obj=0xb71d4ad8) at src/switch_event.c:273 pop = (void *) 0x0 index = 0 my_id = 1 __func__ = "switch_event_thread" #6 0xb7e30bf6 in dummy_worker (opaque=0x8068180) at threadproc/unix/thread.c:138 No locals. #7 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 No symbol table info available. #8 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. Thread 26 (process 27206): #0 0xb7f14410 in __kernel_vsyscall () No symbol table info available. #1 0xb7d2eaa5 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/tls/i686/cmov/libpthread.so.0 No symbol table info available. #2 0xb7e2acaa in apr_thread_cond_wait (cond=0xb7171b38, mutex=0xb7171b08) at locks/unix/thread_cond.c:68 rv = -512 #3 0xb7e21a3c in apr_queue_pop (queue=0xb7171ad8, data=0xb56bd3a8) at misc/apr_queue.c:276 rv = 0 #4 0xb7dadd44 in switch_queue_pop (queue=0xb7171ad8, data=0xb56bd3a8) at src/switch_apr.c:879 No locals. #5 0xb7ddec2d in switch_event_thread (thread=0x80681a0, obj=0xb7171ad8) at src/switch_event.c:273 pop = (void *) 0x0 index = 0 my_id = 2 __func__ = "switch_event_thread" #6 0xb7e30bf6 in dummy_worker (opaque=0x80681a0) at threadproc/unix/thread.c:138 No locals. #7 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 No symbol table info available. #8 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. ---Type to continue, or q to quit--- Thread 25 (process 27207): #0 0xb7f14410 in __kernel_vsyscall () No symbol table info available. #1 0xb7d2eaa5 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/tls/i686/cmov/libpthread.so.0 No symbol table info available. #2 0xb7e2acaa in apr_thread_cond_wait (cond=0xb6fb54a8, mutex=0xb6fb5478) at locks/unix/thread_cond.c:68 rv = -512 #3 0xb7e21a3c in apr_queue_pop (queue=0xb6fb5448, data=0xb4dce3a8) at misc/apr_queue.c:276 rv = 0 #4 0xb7dadd44 in switch_queue_pop (queue=0xb6fb5448, data=0xb4dce3a8) at src/switch_apr.c:879 No locals. #5 0xb7e082fd in log_thread (thread=0xb4e30ae0, obj=0x0) at src/switch_log.c:209 pop = (void *) 0x0 binding = (switch_log_binding_t *) 0x0 __func__ = "log_thread" #6 0xb7e30bf6 in dummy_worker (opaque=0xb4e30ae0) at threadproc/unix/thread.c:138 No locals. #7 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 No symbol table info available. #8 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. Thread 24 (process 27210): #0 0xb7f14410 in __kernel_vsyscall () No symbol table info available. #1 0xb7d2eaa5 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/tls/i686/cmov/libpthread.so.0 No symbol table info available. #2 0xb7e2acaa in apr_thread_cond_wait (cond=0x80c0c58, mutex=0x80c0c28) at locks/unix/thread_cond.c:68 rv = -512 #3 0xb7daed54 in switch_thread_cond_wait (cond=0x80c0c58, mutex=0x80c0c28) at src/switch_apr.c:359 No locals. #4 0xb7e11266 in switch_cond_next () at src/switch_time.c:203 No locals. #5 0xb7dc27a5 in switch_core_sql_thread (thread=0xb3567ae8, obj=0x0) at src/switch_core_sqldb.c:220 pop = (void *) 0x811bf00 itterations = 0 trans = 0 '\0' nothing_in_queue = 1 '\001' len = 100 sql_len = 65536 sqlbuf = 0x80aa9a8 "" newlen = lc = 0 __PRETTY_FUNCTION__ = "switch_core_sql_thread" __func__ = "switch_core_sql_thread" #6 0xb7e30bf6 in dummy_worker (opaque=0xb3567ae8) at threadproc/unix/thread.c:138 No locals. #7 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 No symbol table info available. #8 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. ---Type to continue, or q to quit--- Thread 23 (process 27211): #0 0xb7f14410 in __kernel_vsyscall () No symbol table info available. #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. #2 0xb7e32799 in apr_sleep (t=500000) at time/unix/time.c:246 tv = {tv_sec = 0, tv_usec = 340000} #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 No locals. #4 0xb7dd2af4 in switch_scheduler_task_thread (thread=0x80baa90, obj=0x0) at src/switch_scheduler.c:171 __func__ = "switch_scheduler_task_thread" #5 0xb7e30bf6 in dummy_worker (opaque=0x80baa90) at threadproc/unix/thread.c:138 No locals. #6 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 No symbol table info available. #7 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. Thread 22 (process 27212): #0 0xb7f14410 in __kernel_vsyscall () No symbol table info available. #1 0xb7c7d676 in epoll_wait () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. #2 0xb32b7126 in su_epoll_port_wait_events (self=0x80c6340, tout=1000) at su_epoll_port.c:491 j = -1290679408 n = -1288576964 events = 0 index = -1288973717 version = 1 M = 4 ev = 0xb311c0e0 __PRETTY_FUNCTION__ = "su_epoll_port_wait_events" #3 0xb32bd9b8 in su_base_port_step (self=0x80c6340, tout=1000) at su_base_port.c:442 now = {tv_sec = 3443337943, tv_usec = 249063} __PRETTY_FUNCTION__ = "su_base_port_step" #4 0xb32b8551 in su_port_step (self=0x80c6340, tout=1000) at su_port.h:326 base = (su_virtual_port_t *) 0x80c6340 #5 0xb32b8521 in su_root_step (self=0x80c68f0, tout=1000) at su_root.c:730 __PRETTY_FUNCTION__ = "su_root_step" #6 0xb31e8f29 in sofia_profile_thread_run (thread=0x80d2aa8, obj=0x80d1e10) at sofia.c:831 pool = node = (sip_alias_node_t *) 0xb32f45dc s_event = (switch_event_t *) 0x0 sanity = __func__ = "sofia_profile_thread_run" __PRETTY_FUNCTION__ = "sofia_profile_thread_run" #7 0xb7e30bf6 in dummy_worker (opaque=0x80d2aa8) at threadproc/unix/thread.c:138 No locals. #8 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 No symbol table info available. ---Type to continue, or q to quit--- #9 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. Thread 21 (process 27213): #0 0xb7f14410 in __kernel_vsyscall () No symbol table info available. #1 0xb7c7d676 in epoll_wait () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. #2 0xb32b7126 in su_epoll_port_wait_events (self=0x80dac30, tout=1000) at su_epoll_port.c:491 j = -1299074328 n = 1 events = 0 index = 2 version = 4 M = 4 ev = 0xb291b260 __PRETTY_FUNCTION__ = "su_epoll_port_wait_events" #3 0xb32bd89b in su_base_port_run (self=0x80dac30) at su_base_port.c:342 tout = 1000 __PRETTY_FUNCTION__ = "su_base_port_run" #4 0xb32b842b in su_port_run (self=0x80dac30) at su_port.h:312 base = (su_virtual_port_t *) 0x80dac30 #5 0xb32b8408 in su_root_run (self=0x80dacb0) at su_root.c:691 __PRETTY_FUNCTION__ = "su_root_run" #6 0xb32a8b31 in su_pthread_port_clone_main (varg=0xb311c0a8) at su_pthread_port.c:321 arg = (struct clone_args *) 0x0 task = {{sut_port = 0x80dac30, sut_root = 0x80dacb0}} zap = 1 #7 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 No symbol table info available. #8 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. Thread 20 (process 27214): #0 0xb7f14410 in __kernel_vsyscall () No symbol table info available. #1 0xb7c7d676 in epoll_wait () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. #2 0xb32b7126 in su_epoll_port_wait_events (self=0x80da208, tout=1000) at su_epoll_port.c:491 j = -1307464816 n = -1288576964 events = 0 index = -1288973717 version = 1 M = 4 ev = 0xb211a0e0 __PRETTY_FUNCTION__ = "su_epoll_port_wait_events" #3 0xb32bd9b8 in su_base_port_step (self=0x80da208, tout=1000) at su_base_port.c:442 now = {tv_sec = 3443337943, tv_usec = 148997} __PRETTY_FUNCTION__ = "su_base_port_step" #4 0xb32b8551 in su_port_step (self=0x80da208, tout=1000) at su_port.h:326 ---Type to continue, or q to quit--- base = (su_virtual_port_t *) 0x80da208 #5 0xb32b8521 in su_root_step (self=0x80d7558, tout=1000) at su_root.c:730 __PRETTY_FUNCTION__ = "su_root_step" #6 0xb31e8f29 in sofia_profile_thread_run (thread=0x80dde88, obj=0x80dd650) at sofia.c:831 pool = node = (sip_alias_node_t *) 0xb32f45dc s_event = (switch_event_t *) 0x0 sanity = __func__ = "sofia_profile_thread_run" __PRETTY_FUNCTION__ = "sofia_profile_thread_run" #7 0xb7e30bf6 in dummy_worker (opaque=0x80dde88) at threadproc/unix/thread.c:138 No locals. #8 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 No symbol table info available. #9 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. Thread 19 (process 27215): #0 0xb7f14410 in __kernel_vsyscall () No symbol table info available. #1 0xb7c7d676 in epoll_wait () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. #2 0xb32b7126 in su_epoll_port_wait_events (self=0x80e5228, tout=1000) at su_epoll_port.c:491 j = -1315859736 n = 1 events = 0 index = 1 version = 3 M = 4 ev = 0xb1919260 __PRETTY_FUNCTION__ = "su_epoll_port_wait_events" #3 0xb32bd89b in su_base_port_run (self=0x80e5228) at su_base_port.c:342 tout = 1000 __PRETTY_FUNCTION__ = "su_base_port_run" #4 0xb32b842b in su_port_run (self=0x80e5228) at su_port.h:312 base = (su_virtual_port_t *) 0x80e5228 #5 0xb32b8408 in su_root_run (self=0x80e3b40) at su_root.c:691 __PRETTY_FUNCTION__ = "su_root_run" #6 0xb32a8b31 in su_pthread_port_clone_main (varg=0xb211a0a8) at su_pthread_port.c:321 arg = (struct clone_args *) 0x0 task = {{sut_port = 0x80e5228, sut_root = 0x80e3b40}} zap = 1 #7 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 No symbol table info available. #8 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. Thread 18 (process 27216): #0 0xb7f14410 in __kernel_vsyscall () No symbol table info available. #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 ---Type to continue, or q to quit--- No symbol table info available. #2 0xb7e32799 in apr_sleep (t=10000) at time/unix/time.c:246 tv = {tv_sec = 0, tv_usec = 10000} #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 No locals. #4 0xb31d4b37 in sofia_profile_worker_thread_run (thread=0x80d2b88, obj=0x80d1e10) at sofia.c:656 ireg_loops = 1 gateway_loops = 0 loops = 95 qsize = 4294966782 __PRETTY_FUNCTION__ = "sofia_profile_worker_thread_run" #5 0xb7e30bf6 in dummy_worker (opaque=0x80d2b88) at threadproc/unix/thread.c:138 No locals. #6 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 No symbol table info available. #7 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. Thread 17 (process 27217): #0 0xb7f14410 in __kernel_vsyscall () No symbol table info available. #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. #2 0xb7e32799 in apr_sleep (t=10000) at time/unix/time.c:246 tv = {tv_sec = 0, tv_usec = 10000} #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 No locals. #4 0xb31d4b37 in sofia_profile_worker_thread_run (thread=0x80ddf68, obj=0x80dd650) at sofia.c:656 ireg_loops = 1 gateway_loops = 0 loops = 95 qsize = 4294966782 __PRETTY_FUNCTION__ = "sofia_profile_worker_thread_run" #5 0xb7e30bf6 in dummy_worker (opaque=0x80ddf68) at threadproc/unix/thread.c:138 No locals. #6 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 No symbol table info available. #7 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. Thread 16 (process 27218): #0 0xb7f14410 in __kernel_vsyscall () No symbol table info available. #1 0xb7c7d676 in epoll_wait () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. #2 0xb32b7126 in su_epoll_port_wait_events (self=0x80e66d8, tout=1000) at su_epoll_port.c:491 j = -1341445232 n = -1288576964 events = 0 index = -1288973717 version = 1 ---Type to continue, or q to quit--- M = 4 ev = 0xb00b20e0 __PRETTY_FUNCTION__ = "su_epoll_port_wait_events" #3 0xb32bd9b8 in su_base_port_step (self=0x80e66d8, tout=1000) at su_base_port.c:442 now = {tv_sec = 3443337943, tv_usec = 560073} __PRETTY_FUNCTION__ = "su_base_port_step" #4 0xb32b8551 in su_port_step (self=0x80e66d8, tout=1000) at su_port.h:326 base = (su_virtual_port_t *) 0x80e66d8 #5 0xb32b8521 in su_root_step (self=0x80e49f0, tout=1000) at su_root.c:730 __PRETTY_FUNCTION__ = "su_root_step" #6 0xb31e8f29 in sofia_profile_thread_run (thread=0x80e8710, obj=0x80e7e90) at sofia.c:831 pool = node = (sip_alias_node_t *) 0xb32f45dc s_event = (switch_event_t *) 0x0 sanity = __func__ = "sofia_profile_thread_run" __PRETTY_FUNCTION__ = "sofia_profile_thread_run" #7 0xb7e30bf6 in dummy_worker (opaque=0x80e8710) at threadproc/unix/thread.c:138 No locals. #8 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 No symbol table info available. #9 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. Thread 15 (process 27219): #0 0xb7f14410 in __kernel_vsyscall () No symbol table info available. #1 0xb7c7d676 in epoll_wait () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. #2 0xb32b7126 in su_epoll_port_wait_events (self=0x80ee648, tout=1000) at su_epoll_port.c:491 j = -1349840152 n = 1 events = 0 index = 1 version = 3 M = 4 ev = 0xaf8b1260 __PRETTY_FUNCTION__ = "su_epoll_port_wait_events" #3 0xb32bd89b in su_base_port_run (self=0x80ee648) at su_base_port.c:342 tout = 1000 __PRETTY_FUNCTION__ = "su_base_port_run" #4 0xb32b842b in su_port_run (self=0x80ee648) at su_port.h:312 base = (su_virtual_port_t *) 0x80ee648 #5 0xb32b8408 in su_root_run (self=0x80f0bd0) at su_root.c:691 __PRETTY_FUNCTION__ = "su_root_run" #6 0xb32a8b31 in su_pthread_port_clone_main (varg=0xb00b20a8) at su_pthread_port.c:321 arg = (struct clone_args *) 0x0 task = {{sut_port = 0x80ee648, sut_root = 0x80f0bd0}} zap = 1 #7 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 No symbol table info available. ---Type to continue, or q to quit--- #8 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. Thread 14 (process 27220): #0 0xb7f14410 in __kernel_vsyscall () No symbol table info available. #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. #2 0xb7e32799 in apr_sleep (t=10000) at time/unix/time.c:246 tv = {tv_sec = 0, tv_usec = 0} #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 No locals. #4 0xb31d4b37 in sofia_profile_worker_thread_run (thread=0x80e87f0, obj=0x80e7e90) at sofia.c:656 ireg_loops = 0 gateway_loops = 0 loops = 85 qsize = 4294966782 __PRETTY_FUNCTION__ = "sofia_profile_worker_thread_run" #5 0xb7e30bf6 in dummy_worker (opaque=0x80e87f0) at threadproc/unix/thread.c:138 No locals. #6 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 No symbol table info available. #7 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. Thread 13 (process 27221): #0 0xb7f14410 in __kernel_vsyscall () No symbol table info available. #1 0xb7c72c07 in poll () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. #2 0xb337af00 in wanpipe_wait (zchan=0xb3446128, flags=0xae751f80, to=100) at src/ozmod/ozmod_wanpipe/ozmod_wanpipe.c:255 inflags = 1 result = #3 0xae837ae7 in zap_channel_wait (zchan=0x1, flags=0xae751f80, to=100) at src/zap_io.c:1479 __PRETTY_FUNCTION__ = "zap_channel_wait" #4 0xae80aee8 in zap_isdn_run (me=0xb3416528, obj=0xb341dbc8) at src/ozmod/ozmod_isdn/ozmod_isdn.c:1725 flags = ZAP_READ status = ZAP_TIMEOUT span = isdn_data = (zap_isdn_data_t *) 0xae753008 frame = "\002\001\002\002\b\002\200\002\001\036\002\201\210", '\0' len = 13 errs = 0 __func__ = "zap_isdn_run" #5 0xae8421ba in thread_launch (args=0xb3416528) at src/zap_threadmutex.c:74 exit_val = #6 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 No symbol table info available. #7 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. ---Type to continue, or q to quit--- Thread 12 (process 27222): #0 0xb7f14410 in __kernel_vsyscall () No symbol table info available. #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. #2 0xb7e32799 in apr_sleep (t=100000) at time/unix/time.c:246 tv = {tv_sec = 0, tv_usec = 100000} #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 No locals. #4 0xb31f929e in sofia_presence_event_thread_run (thread=0x80cf958, obj=0x0) at sofia_presence.c:664 count = 0 pop = (void *) 0xb3462cb8 __func__ = "sofia_presence_event_thread_run" #5 0xb7e30bf6 in dummy_worker (opaque=0x80cf958) at threadproc/unix/thread.c:138 No locals. #6 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 No symbol table info available. #7 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. Thread 11 (process 27223): #0 0xb7f14410 in __kernel_vsyscall () No symbol table info available. #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. #2 0xb7e32799 in apr_sleep (t=1000000) at time/unix/time.c:246 tv = {tv_sec = 0, tv_usec = 320000} #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 No locals. #4 0xad706ea6 in ?? () from /opt/app/voip/ippbx.prod/mod/mod_fifo.so No locals. #5 0xb7e30bf6 in dummy_worker (opaque=0xad386b90) at threadproc/unix/thread.c:138 No locals. #6 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 No symbol table info available. #7 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. Thread 10 (process 27225): #0 0xb7f14410 in __kernel_vsyscall () No symbol table info available. #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. #2 0xb7e32799 in apr_sleep (t=1000000) at time/unix/time.c:246 tv = {tv_sec = 0, tv_usec = 840000} #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 No locals. #4 0xabdf94ab in ?? () from /opt/app/voip/ippbx.prod/mod/mod_local_stream.so No locals. #5 0xb7e30bf6 in dummy_worker (opaque=0x81386e0) at threadproc/unix/thread.c:138 No locals. ---Type to continue, or q to quit--- #6 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 No symbol table info available. #7 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. Thread 9 (process 27226): #0 0xb7f14410 in __kernel_vsyscall () No symbol table info available. #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. #2 0xb7e32799 in apr_sleep (t=1000000) at time/unix/time.c:246 tv = {tv_sec = 0, tv_usec = 840000} #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 No locals. #4 0xabdf94ab in ?? () from /opt/app/voip/ippbx.prod/mod/mod_local_stream.so No locals. #5 0xb7e30bf6 in dummy_worker (opaque=0x815e728) at threadproc/unix/thread.c:138 No locals. #6 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 No symbol table info available. #7 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. Thread 8 (process 27227): #0 0xb7f14410 in __kernel_vsyscall () No symbol table info available. #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. #2 0xb7e32799 in apr_sleep (t=1000000) at time/unix/time.c:246 tv = {tv_sec = 0, tv_usec = 840000} #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 No locals. #4 0xabdf94ab in ?? () from /opt/app/voip/ippbx.prod/mod/mod_local_stream.so No locals. #5 0xb7e30bf6 in dummy_worker (opaque=0x8184770) at threadproc/unix/thread.c:138 No locals. #6 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 No symbol table info available. #7 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. Thread 7 (process 27228): #0 0xb7f14410 in __kernel_vsyscall () No symbol table info available. #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. #2 0xb7e32799 in apr_sleep (t=1000) at time/unix/time.c:246 tv = {tv_sec = 0, tv_usec = 1000} #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 No locals. #4 0xb7e12a59 in softtimer_runtime () at src/switch_time.c:459 ---Type to continue, or q to quit--- current_ms = 1544 x = tick = 144 ts = last = 1234349144129112 fwd_errs = 0 rev_errs = 0 __func__ = "softtimer_runtime" #5 0xb7dd4fb3 in switch_loadable_module_exec (thread=0x80bf8d8, obj=0x80bf6c8) at src/switch_loadable_module.c:93 status = module = (switch_loadable_module_t *) 0x80c0bc8 __PRETTY_FUNCTION__ = "switch_loadable_module_exec" __func__ = "switch_loadable_module_exec" #6 0xb7e30bf6 in dummy_worker (opaque=0x80bf8d8) at threadproc/unix/thread.c:138 No locals. #7 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 No symbol table info available. #8 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. Thread 6 (process 27229): #0 0xb7f14410 in __kernel_vsyscall () No symbol table info available. #1 0xb7d31bb8 in accept () from /lib/tls/i686/cmov/libpthread.so.0 No symbol table info available. #2 0xb7e2f90d in apr_socket_accept (new=0xaa2d434c, sock=0x81bbbb0, connection_context=0x81bdaa8) at network_io/unix/sockets.c:187 No locals. #3 0xb7dae3fb in switch_socket_accept (new_sock=0xaa2d434c, sock=0x81bbbb0, pool=0x81bdaa8) at src/switch_apr.c:664 No locals. #4 0xb33262f2 in mod_event_socket_runtime () at mod_event_socket.c:2134 pool = (switch_memory_pool_t *) 0x81bbaa0 listener_pool = (switch_memory_pool_t *) 0x81bdaa8 rv = sa = (switch_sockaddr_t *) 0x81bbaf8 inbound_socket = (switch_socket_t *) 0x81bdb00 listener = x = __func__ = "mod_event_socket_runtime" #5 0xb7dd4fb3 in switch_loadable_module_exec (thread=0x80bfb40, obj=0x80bf930) at src/switch_loadable_module.c:93 status = module = (switch_loadable_module_t *) 0xb340ec28 __PRETTY_FUNCTION__ = "switch_loadable_module_exec" __func__ = "switch_loadable_module_exec" #6 0xb7e30bf6 in dummy_worker (opaque=0x80bfb40) at threadproc/unix/thread.c:138 No locals. #7 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 No symbol table info available. #8 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. Thread 5 (process 27230): ---Type to continue, or q to quit--- #0 0xb7f14410 in __kernel_vsyscall () No symbol table info available. #1 0xb7c72c07 in poll () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. #2 0xb33ce86e in ?? () from /opt/app/voip/ippbx.prod/mod/mod_xml_rpc.so No locals. #3 0xb33c1464 in ChanSwitchAccept (chanSwitchP=0x81f5030, channelPP=0xa9ad30e0, channelInfoPP=0xa9ad30dc, errorP=0xa9ad30e4) at ../../../../libs/xmlrpc-c/lib/abyss/src/chanswitch.c:149 No locals. #4 0xb33cd37e in ServerRun (serverP=0xb33ffe4c) at ../../../../libs/xmlrpc-c/lib/abyss/src/server.c:908 srvP = (struct _TServer * const) 0x81f4fc8 #5 0xb33be832 in mod_xml_rpc_runtime () at mod_xml_rpc.c:837 registryP = (xmlrpc_registry *) 0x81bb960 env = {fault_occurred = 0, fault_code = 0, fault_string = 0x0} logfile = "/opt/app/voip/ippbx.prod/log/freeswitch_http.log", '\0' hi = var = (const void *) 0x80609b8 val = (void *) 0x805dc78 __func__ = "mod_xml_rpc_runtime" #6 0xb7dd4fb3 in switch_loadable_module_exec (thread=0x80bfda8, obj=0x80bfb98) at src/switch_loadable_module.c:93 status = module = (switch_loadable_module_t *) 0xb3405178 __PRETTY_FUNCTION__ = "switch_loadable_module_exec" __func__ = "switch_loadable_module_exec" #7 0xb7e30bf6 in dummy_worker (opaque=0x80bfda8) at threadproc/unix/thread.c:138 No locals. #8 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 No symbol table info available. #9 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. Thread 4 (process 27231): #0 0xb7f14410 in __kernel_vsyscall () No symbol table info available. #1 0xb7d3199b in read () from /lib/tls/i686/cmov/libpthread.so.0 No symbol table info available. #2 0xb7e9dfb3 in read_char (el=0x81c5040, cp=0xa92d235b "?|???`!\b\200a\"\b?#-?;\237?@P\034\b?#-?\a") at read.c:294 num_read = 136073656 tried = 0 #3 0xb7e9da9c in el_getc (el=0x81c5040, cp=0xa92d235b "?|???`!\b\200a\"\b?#-?;\237?@P\034\b?#-?\a") at read.c:362 num_read = -1456659621 ma = (c_macro_t *) 0x81c52e0 #4 0xb7e9dbdf in el_gets (el=0x81c5040, nread=0xa92d23a8) at read.c:241 cmdnum = 232 '?' num = 136405224 ch = -1 '?' #5 0xb7db9f3b in console_thread (thread=0x82102d0, obj=0x8210248) at src/switch_console.c:441 arg = 1 count = 7 line = 0x82160e8 "" pool = (switch_memory_pool_t *) 0x8210248 ---Type to continue, or q to quit--- __func__ = "console_thread" __PRETTY_FUNCTION__ = "console_thread" #6 0xb7e30bf6 in dummy_worker (opaque=0x82102d0) at threadproc/unix/thread.c:138 No locals. #7 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 No symbol table info available. #8 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. Thread 3 (process 27235): #0 0xb7f14410 in __kernel_vsyscall () No symbol table info available. #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. #2 0xb7e32799 in apr_sleep (t=100000) at time/unix/time.c:246 tv = {tv_sec = 0, tv_usec = 20000} #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 No locals. #4 0xb7df019e in switch_ivr_originate (session=0xb34dd080, bleg=0xa8ad10c0, cause=0xa8ad10bc, bridgeto=0x823a528 "openzap/1/a/04855711", timelimit_sec=60, table=0xb7ecda20, cid_name_override=0x0, cid_num_override=0x0, caller_profile_override=0x0, ovars=0x0, flags=) at src/switch_ivr_originate.c:1793 end = var_begin = originate_status = {{peer_session = 0xb34e5400, peer_channel = 0xb34e9968, caller_profile = 0xb34e1dc8, ring_ready = 0 '\0', early_media = 1 '\001', answered = 0 '\0', per_channel_timelimit_sec = 0, per_channel_progress_timelimit_sec = 0}, {peer_session = 0x0, peer_channel = 0x0, caller_profile = 0x0, ring_ready = 0 '\0', early_media = 0 '\0', answered = 0 '\0', per_channel_timelimit_sec = 0, per_channel_progress_timelimit_sec = 0} } dftflags = 0 myflags = 0 pipe_names = {0xa811de20 "openzap", 0x0 } data = status = SWITCH_STATUS_SUCCESS caller_channel = (switch_channel_t *) 0xb34e15e8 peer_names = {0xa811de20 "openzap", 0x0 } new_session = (switch_core_session_t *) 0xb34e5400 peer_session = new_profile = (switch_caller_profile_t *) 0xb34e1dc8 caller_caller_profile = chan_type = chan_data = peer_channel = ringback = {audio_buffer = 0x0, ts = {TONES = {{freqs = {0, 0, 0, 0, 0, 0}} }, channels = 0, rate = 0, duration = 0, wait = 0, tmp_duration = 0, tmp_wait = 0, loops = 0, LOOPS = 0, decay_factor = 0, decay_direction = 0, decay_step = 0, volume = 0, debug = 0, debug_stream = 0x0, user_data = 0x0, buffer = 0x0, datalen = 0, samples = 0, dynamic = 0, handler = 0}, fhb = { file_interface = 0x0, flags = 0, fd = 0x0, samples = 0, samplerate = 0, native_rate = 0, channels = 0 '\0', format = 0, sections = 0, seekable = 0, sample_count = 0, speed = 0, memory_pool = 0x0, prebuf = 0, interval = 0, private_info = 0x0, handler = 0x0, pos = 0, audio_buffer = 0x0, sp_audio_buffer = 0x0, thresh = 0, silence_hits = 0, offset_pos = 0, last_pos = 0, vol = 0, resampler = 0x0, buffer = 0x0, dbuf = 0x0, dbuflen = 0, pre_buffer = 0x0, pre_buffer_data = 0x0, pre_buffer_datalen = 0, file = 0x0, func = 0x0, line = 0}, fh = 0x0, silence = 0, asis = 0 '\0'} start = 1234349134 read_frame = (switch_frame_t *) 0x0 ---Type to continue, or q to quit--- pool = (switch_memory_pool_t *) 0x0 r = 0 i = -1286728216 and_argc = 1 or_argc = 1 sleep_ms = 1000 try = 0 retries = 1 write_codec = {codec_interface = 0x0, implementation = 0x0, fmtp_in = 0x0, fmtp_out = 0x0, codec_settings = {quality = 0, complexity = 0, enhancement = 0, vad = 0, vbr = 0, vbr_quality = 0, abr = 0, dtx = 0, preproc = 0, pp_vad = 0, pp_agc = 0, pp_agc_level = 0, pp_denoise = 0, pp_dereverb = 0, pp_dereverb_decay = 0, pp_dereverb_level = 0}, flags = 0, memory_pool = 0x0, private_info = 0x0, agreed_pt = 0 '\0', mutex = 0x0} write_frame = {codec = 0x0, source = 0x0, packet = 0x0, packetlen = 0, data = 0x821ec80, datalen = 0, buflen = 4096, samples = 0, rate = 0, payload = 0 '\0', timestamp = 0, seq = 0, ssrc = 0, m = SWITCH_FALSE, flags = 0} pass = 0 '\0' odata = 0xa811e160 "openzap/1/a/04855711" var = reason = SWITCH_CAUSE_SUCCESS to = 0 '\0' var_val = vars = 0x0 var_block_count = 0 e = ringback_data = 0x0 read_codec = message = (switch_core_session_message_t *) 0x0 var_event = (switch_event_t *) 0xb346cea8 fail_on_single_reject = 0 '\0' fail_on_single_reject_var = 0x0 loop_data = 0xa811de20 "openzap" progress_timelimit_sec = 60 oglobals = {session = 0xb34dd080, idx = -1, hups = 0, file = '\0' , key = '\0' , early_ok = 0 '\0', ring_ready = 1 '\001', sent_ring = 1 '\001', progress = 1 '\001', return_ring_ready = 0 '\0', monitor_early_media_ring = 0 '\0', monitor_early_media_fail = 0 '\0', gen_ringback = 0 '\0', ignore_early_media = 1 '\001', ignore_ring_ready = 0 '\0'} __PRETTY_FUNCTION__ = "switch_ivr_originate" __func__ = "switch_ivr_originate" #5 0xad7231af in ?? () from /opt/app/voip/ippbx.prod/mod/mod_dptools.so No locals. #6 0xb7dc6c54 in switch_core_session_exec (session=0xb34dd080, application_interface=0xb346ba10, arg=0x823a528 "openzap/1/a/04855711") at src/switch_core_session.c:1332 log = lp = event = (switch_event_t *) 0x0 var = channel = (switch_channel_t *) 0xb34e15e8 expanded = 0x823a528 "openzap/1/a/04855711" app = 0xad7253b0 "bridge" __PRETTY_FUNCTION__ = "switch_core_session_exec" __func__ = "switch_core_session_exec" #7 0xb7dc70fe in switch_core_session_execute_application (session=0xb34dd080, app=0x823a520 "bridge", arg=0x823a528 "openzap/1/a/04855711") ---Type to continue, or q to quit--- at src/switch_core_session.c:1254 application_interface = (switch_application_interface_t *) 0xb346ba10 __func__ = "switch_core_session_execute_application" #8 0xb7dc93a4 in switch_core_session_run (session=0xb34dd080) at src/switch_core_state_machine.c:155 proceed = global_proceed = do_extra_handlers = state = endstate = endpoint_interface = driver_state_handler = (const switch_state_handler_table_t *) 0xb331f860 application_state_handler = thread_id = 3084680427 env = {{__jmpbuf = {134603552, -1287115232, -1209317480, 27232, -1287651304, 24}, __mask_was_saved = -1210923727, __saved_mask = { __val = {2819623592, 2829914760, 3084048908, 2819620880, 27232, 1, 3008239368, 2829914744, 3085760892, 250000, 3084048908, 2829914760, 3085085008, 1, 3085760892, 2829914792, 3085760892, 0, 134564192, 3084043569, 3085085008, 134564244, 3085760892, 2829914824, 27232, 134564240, 3007790064, 3084573819, 3085760892, 3008213112, 3084048908, 2829914856}}}} sig = __func__ = "switch_core_session_run" __PRETTY_FUNCTION__ = "switch_core_session_run" #9 0xb7dc6725 in switch_core_session_thread (thread=0x823a050, obj=0xb34dd080) at src/switch_core_session.c:940 session = (switch_core_session_t *) 0xb34dd080 event = event_str = 0x0 val = __func__ = "switch_core_session_thread" __PRETTY_FUNCTION__ = "switch_core_session_thread" #10 0xb7e30bf6 in dummy_worker (opaque=0x823a050) at threadproc/unix/thread.c:138 No locals. #11 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 No symbol table info available. #12 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. Thread 2 (process 27236): #0 0xb7f14410 in __kernel_vsyscall () No symbol table info available. #1 0xb7c72c07 in poll () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. #2 0xb337af00 in wanpipe_wait (zchan=0xb341f9d8, flags=0xa80fee60, to=40) at src/ozmod/ozmod_wanpipe/ozmod_wanpipe.c:255 inflags = 1 result = #3 0xae837ae7 in zap_channel_wait (zchan=0x1, flags=0xa80fee60, to=40) at src/zap_io.c:1479 __PRETTY_FUNCTION__ = "zap_channel_wait" #4 0xae849059 in channel_read_frame (session=0xb34e5400, frame=0xa80ff170, flags=0, stream_id=0) at mod_openzap.c:593 channel = (switch_channel_t *) 0xb34e9968 len = wflags = ZAP_READ dtmf = '\0' status = total_to = 80 ---Type to continue, or q to quit--- chunk = 40 do_break = 0 __PRETTY_FUNCTION__ = "channel_read_frame" __func__ = "channel_read_frame" #5 0xb7dcbe6f in switch_core_session_read_frame (session=0xb34e5400, frame=0xa80ff170, flags=0, stream_id=0) at src/switch_core_io.c:161 ptr = status = SWITCH_STATUS_SUCCESS need_codec = perfect = do_bugs = -1212118646 do_resample = 0 is_cng = 0 flag = 0 __PRETTY_FUNCTION__ = "switch_core_session_read_frame" __func__ = "switch_core_session_read_frame" #6 0xb7e054ec in switch_ivr_sleep (session=0xb34e5400, ms=10, sync=SWITCH_FALSE, args=0x0) at src/switch_ivr.c:262 channel = (switch_channel_t *) 0xb34e9968 status = SWITCH_STATUS_SUCCESS start = 1234349144119246 now = 0 done = 1234349144129246 read_frame = (switch_frame_t *) 0x0 cng_frame = {codec = 0x0, source = 0x0, packet = 0x0, packetlen = 0, data = 0xa80ff176, datalen = 2, buflen = 2, samples = 0, rate = 0, payload = 0 '\0', timestamp = 0, seq = 0, ssrc = 0, m = SWITCH_FALSE, flags = 1} data = "\000" write_frame = {codec = 0x0, source = 0x0, packet = 0x0, packetlen = 0, data = 0x0, datalen = 0, buflen = 0, samples = 0, rate = 0, payload = 0 '\0', timestamp = 0, seq = 0, ssrc = 0, m = SWITCH_FALSE, flags = 0} abuf = (unsigned char *) 0x0 imp = {codec_type = SWITCH_CODEC_TYPE_AUDIO, ianacode = 0 '\0', iananame = 0x0, fmtp = 0x0, samples_per_second = 0, actual_samples_per_second = 0, bits_per_second = 0, microseconds_per_packet = 0, samples_per_packet = 0, decoded_bytes_per_packet = 0, encoded_bytes_per_packet = 0, number_of_channels = 0 '\0', codec_frames_per_packet = 0, init = 0, encode = 0, decode = 0, destroy = 0, codec_id = 0, next = 0x0} codec = {codec_interface = 0x0, implementation = 0x0, fmtp_in = 0x0, fmtp_out = 0x0, codec_settings = {quality = 0, complexity = 0, enhancement = 0, vad = 0, vbr = 0, vbr_quality = 0, abr = 0, dtx = 0, preproc = 0, pp_vad = 0, pp_agc = 0, pp_agc_level = 0, pp_denoise = 0, pp_dereverb = 0, pp_dereverb_decay = 0, pp_dereverb_level = 0}, flags = 0, memory_pool = 0x0, private_info = 0x0, agreed_pt = 0 '\0', mutex = 0x0} sval = 0 var = __func__ = "switch_ivr_sleep" __PRETTY_FUNCTION__ = "switch_ivr_sleep" #7 0xb7deaf94 in originate_on_consume_media_transmit (session=0xb34e5400) at src/switch_ivr_originate.c:47 channel = (switch_channel_t *) 0xb34e9968 #8 0xb7dc8b74 in switch_core_session_run (session=0xb34e5400) at src/switch_core_state_machine.c:476 proceed = 1 global_proceed = 0 do_extra_handlers = state = endstate = endpoint_interface = driver_state_handler = (const switch_state_handler_table_t *) 0xae8523c0 application_state_handler = ---Type to continue, or q to quit--- thread_id = 3084680427 env = {{__jmpbuf = {0, -1287122688, -1475349896, 27233, -1211162264, -1210902852}, __mask_was_saved = -1210923727, __saved_mask = { __val = {0, 3084099572, 3084048908, 3085631456, 27233, 1, 3008273032, 2819617400, 3085760892, 250000, 3084048908, 2819617416, 3085085008, 1, 3085760892, 2819617448, 3085760892, 0, 134564192, 3084043569, 3085085008, 134564244, 3085760892, 2819617480, 27233, 134564240, 135380168, 3084573819, 3085760892, 3008246776, 3084048908, 2819617512}}}} sig = __func__ = "switch_core_session_run" __PRETTY_FUNCTION__ = "switch_core_session_run" #9 0xb7dc6725 in switch_core_session_thread (thread=0xb34f1060, obj=0xb34e5400) at src/switch_core_session.c:940 session = (switch_core_session_t *) 0xb34e5400 event = event_str = 0x0 val = __func__ = "switch_core_session_thread" __PRETTY_FUNCTION__ = "switch_core_session_thread" #10 0xb7e30bf6 in dummy_worker (opaque=0xb34f1060) at threadproc/unix/thread.c:138 No locals. #11 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 No symbol table info available. #12 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. Thread 1 (process 27234): #0 0xabe0e408 in mpg123_delete at plt () from /opt/app/voip/ippbx.prod/mod/mod_shout.so No locals. #1 0xabe13325 in ?? () from /opt/app/voip/ippbx.prod/mod/mod_shout.so No locals. #2 0xb7dc04b9 in switch_core_file_seek (fh=0xa76fef28, cur_pos=0xa76fef88, samples=0, whence=1) at src/switch_core_file.c:345 status = 4294967295 __PRETTY_FUNCTION__ = "switch_core_file_seek" #3 0xb7dfab82 in switch_ivr_play_file (session=0xa7700030, fh=0xa76fef28, file=0xa7711ed8 "/opt/app/voip/ippbx.prod/sounds/de/callie/qet.mp3", args=0xa76ff0d8) at src/switch_ivr_play_say.c:1262 channel = (switch_channel_t *) 0xa7704598 dtmf = {digit = 0 '\0', duration = 0} interval = 2909916864 samples = 320 framelen = 640 sample_start = olen = 320 llen = 320 write_frame = {codec = 0xa76fefb8, source = 0x0, packet = 0x0, packetlen = 0, data = 0xa7743f30, datalen = 640, buflen = 32768, samples = 320, rate = 16000, payload = 0 '\0', timestamp = 0, seq = 0, ssrc = 0, m = SWITCH_FALSE, flags = 0} timer = {interval = 0, flags = 0, samples = 0, samplecount = 0, timer_interface = 0x0, memory_pool = 0x0, private_info = 0x0, diff = 0, tick = 0} codec = {codec_interface = 0x80cd8f8, implementation = 0x80cdf20, fmtp_in = 0x0, fmtp_out = 0x0, codec_settings = {quality = 0, complexity = 0, enhancement = 0, vad = 0, vbr = 0, vbr_quality = 0, abr = 0, dtx = 0, preproc = 0, pp_vad = 0, pp_agc = 0, pp_agc_level = 0, pp_denoise = 0, pp_dereverb = 0, pp_dereverb_decay = 0, pp_dereverb_level = 0}, flags = 3, memory_pool = 0xb34fbb38, private_info = 0x0, agreed_pt = 0 '\0', mutex = 0xa7711f10} pool = (switch_memory_pool_t *) 0xb34fbb38 status = SWITCH_STATUS_SUCCESS lfh = {file_interface = 0x8136258, flags = 3085, fd = 0x0, samples = 0, samplerate = 16000, native_rate = 16000, channels = 1 '\001', ---Type to continue, or q to quit--- format = 0, sections = 0, seekable = 0, sample_count = 729088, speed = 0, memory_pool = 0xa7712040, prebuf = 0, interval = 0, private_info = 0xa7714048, handler = 0x0, pos = 0, audio_buffer = 0xb3476f88, sp_audio_buffer = 0x0, thresh = 0, silence_hits = 0, offset_pos = 364640, last_pos = 0, vol = 0, resampler = 0x0, buffer = 0x0, dbuf = 0x0, dbuflen = 0, pre_buffer = 0x0, pre_buffer_data = 0x0, pre_buffer_datalen = 0, file = 0xb7eb71f0 "src/switch_ivr_play_say.c", func = 0xb7eb794b "switch_ivr_play_file", line = 894} read_codec = (switch_codec_t *) 0xb34fcb90 p = 0xb7eaf8d4 "current_application" asis = 0 '\0' prefix = timer_name = 0x0 prebuf = eof = 1 bread = __func__ = "switch_ivr_play_file" __PRETTY_FUNCTION__ = "switch_ivr_play_file" #4 0xad71da25 in ?? () from /opt/app/voip/ippbx.prod/mod/mod_dptools.so No locals. #5 0xb7dc6c54 in switch_core_session_exec (session=0xa7700030, application_interface=0xb346b388, arg=0xb34fd790 "qet.mp3") at src/switch_core_session.c:1332 log = lp = event = (switch_event_t *) 0x0 var = channel = (switch_channel_t *) 0xa7704598 expanded = 0xb34fd790 "qet.mp3" app = 0xad725133 "playback" __PRETTY_FUNCTION__ = "switch_core_session_exec" __func__ = "switch_core_session_exec" #6 0xb7dc70fe in switch_core_session_execute_application (session=0xa7700030, app=0xb34fd780 "playback", arg=0xb34fd790 "qet.mp3") at src/switch_core_session.c:1254 application_interface = (switch_application_interface_t *) 0xb346b388 __func__ = "switch_core_session_execute_application" #7 0xb7dc93a4 in switch_core_session_run (session=0xa7700030) at src/switch_core_state_machine.c:155 proceed = global_proceed = do_extra_handlers = state = endstate = endpoint_interface = driver_state_handler = (const switch_state_handler_table_t *) 0xb331f860 application_state_handler = thread_id = 3084680427 env = {{__jmpbuf = {0, 0, 0, 0, 0, 0}, __mask_was_saved = 0, __saved_mask = {__val = {0 }}}} sig = __func__ = "switch_core_session_run" __PRETTY_FUNCTION__ = "switch_core_session_run" #8 0xb7dc6725 in switch_core_session_thread (thread=0xb34fd3d0, obj=0xa7700030) at src/switch_core_session.c:940 session = (switch_core_session_t *) 0xa7700030 event = event_str = 0x0 val = ---Type to continue, or q to quit--- __func__ = "switch_core_session_thread" __PRETTY_FUNCTION__ = "switch_core_session_thread" #9 0xb7e30bf6 in dummy_worker (opaque=0xb34fd3d0) at threadproc/unix/thread.c:138 No locals. #10 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/libpthread.so.0 No symbol table info available. #11 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 No symbol table info available. regards Helmut From brian at freeswitch.org Wed Feb 11 04:16:42 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 11 Feb 2009 06:16:42 -0600 Subject: [Freeswitch-users] segfault when shoutcast plays mp3 and extension hangs up In-Reply-To: <4992BFFC.50006@ewetel.de> References: <4992BFFC.50006@ewetel.de> Message-ID: The proper location to post this is on Jira... please in the future report ALL issues via Jira. They'll get lost if not done this way. Thanks, Brian On Feb 11, 2009, at 6:09 AM, Helmut Kuper wrote: > Hello, > > today I tried to play a mp3. It works fine until extension hangs up. > Then FS (FreeSWITCH Version 1.0.trunk (11698M)) crashed with segfault. > The mp3 file was generated by MP3Splitter > (http://www.codevisions.de/hp/upload/_files/mp3splitter20.zip) as a > piece out of a complete mp3 song. There is a good chance that it > generates corrupt mp3s. At least those mp3s are playable in winamp and > media player. > > My dialplan: > > expression="^9123$"> > data="absolute_codec_string=G722"/> > > > > > > > > FS console output shows problems in mp3 file: > > freeswitch at ippbx-prod-node0> 2009-02-11 11:45:43 [DEBUG] Span:1 Q. > 931() > Timer 0 of call 0 (CRV: 2, State: 0) timed out > Note: Illegal Audio-MPEG-Header 0x00000000 at offset 0x10ec15. > Note: Trying to resync... > Note: Hit end of (available) data during resync. > 2009-02-11 11:45:44 [DEBUG] switch_ivr_play_say.c:1261 > switch_ivr_play_file() done playing file > ./start_fs.sh: line 6: 27201 Segmentation fault (core dumped) > bin/freeswitch $1 > > > Here are the backtraces: > > (gdb) bt > #0 0xabe0e408 in mpg123_delete at plt () from > /opt/app/voip/ippbx.prod/mod/mod_shout.so > #1 0xabe13325 in ?? () from /opt/app/voip/ippbx.prod/mod/mod_shout.so > #2 0xb7dc04b9 in switch_core_file_seek (fh=0xa76fef28, > cur_pos=0xa76fef88, samples=0, whence=1) at src/switch_core_file.c:345 > #3 0xb7dfab82 in switch_ivr_play_file (session=0xa7700030, > fh=0xa76fef28, > file=0xa7711ed8 "/opt/app/voip/ippbx.prod/sounds/de/callie/ > qet.mp3", > args=0xa76ff0d8) at src/switch_ivr_play_say.c:1262 > #4 0xad71da25 in ?? () from /opt/app/voip/ippbx.prod/mod/ > mod_dptools.so > #5 0xb7dc6c54 in switch_core_session_exec (session=0xa7700030, > application_interface=0xb346b388, arg=0xb34fd790 "qet.mp3") > at src/switch_core_session.c:1332 > #6 0xb7dc70fe in switch_core_session_execute_application > (session=0xa7700030, app=0xb34fd780 "playback", arg=0xb34fd790 > "qet.mp3") > at src/switch_core_session.c:1254 > #7 0xb7dc93a4 in switch_core_session_run (session=0xa7700030) at > src/switch_core_state_machine.c:155 > #8 0xb7dc6725 in switch_core_session_thread (thread=0xb34fd3d0, > obj=0xa7700030) at src/switch_core_session.c:940 > #9 0xb7e30bf6 in dummy_worker (opaque=0xb34fd3d0) at > threadproc/unix/thread.c:138 > #10 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > #11 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > Current language: auto; currently asm > > > (gdb) bt full > #0 0xabe0e408 in mpg123_delete at plt () from > /opt/app/voip/ippbx.prod/mod/mod_shout.so > No locals. > #1 0xabe13325 in ?? () from /opt/app/voip/ippbx.prod/mod/mod_shout.so > No locals. > #2 0xb7dc04b9 in switch_core_file_seek (fh=0xa76fef28, > cur_pos=0xa76fef88, samples=0, whence=1) at src/switch_core_file.c:345 > status = 4294967295 > __PRETTY_FUNCTION__ = "switch_core_file_seek" > #3 0xb7dfab82 in switch_ivr_play_file (session=0xa7700030, > fh=0xa76fef28, > file=0xa7711ed8 "/opt/app/voip/ippbx.prod/sounds/de/callie/ > qet.mp3", > args=0xa76ff0d8) at src/switch_ivr_play_say.c:1262 > channel = (switch_channel_t *) 0xa7704598 > dtmf = {digit = 0 '\0', duration = 0} > interval = 2909916864 > samples = 320 > framelen = 640 > sample_start = > olen = 320 > llen = 320 > write_frame = {codec = 0xa76fefb8, source = 0x0, packet = 0x0, > packetlen = 0, data = 0xa7743f30, datalen = 640, buflen = 32768, > samples = 320, rate = 16000, payload = 0 '\0', timestamp = 0, seq = > 0, > ssrc = 0, m = SWITCH_FALSE, flags = 0} > timer = {interval = 0, flags = 0, samples = 0, samplecount = 0, > timer_interface = 0x0, memory_pool = 0x0, private_info = 0x0, > diff = 0, tick = 0} > codec = {codec_interface = 0x80cd8f8, implementation = > 0x80cdf20, fmtp_in = 0x0, fmtp_out = 0x0, codec_settings = {quality > = 0, > complexity = 0, enhancement = 0, vad = 0, vbr = 0, vbr_quality = 0, > abr = 0, dtx = 0, preproc = 0, pp_vad = 0, pp_agc = 0, > pp_agc_level = 0, pp_denoise = 0, pp_dereverb = 0, > pp_dereverb_decay > = 0, pp_dereverb_level = 0}, flags = 3, memory_pool = 0xb34fbb38, > private_info = 0x0, agreed_pt = 0 '\0', mutex = 0xa7711f10} > pool = (switch_memory_pool_t *) 0xb34fbb38 > status = SWITCH_STATUS_SUCCESS > lfh = {file_interface = 0x8136258, flags = 3085, fd = 0x0, > samples = 0, samplerate = 16000, native_rate = 16000, channels = 1 > '\001', > format = 0, sections = 0, seekable = 0, sample_count = 729088, > speed = > 0, memory_pool = 0xa7712040, prebuf = 0, interval = 0, > private_info = 0xa7714048, handler = 0x0, pos = 0, audio_buffer = > 0xb3476f88, sp_audio_buffer = 0x0, thresh = 0, silence_hits = 0, > offset_pos = 364640, last_pos = 0, vol = 0, resampler = 0x0, buffer = > 0x0, dbuf = 0x0, dbuflen = 0, pre_buffer = 0x0, > pre_buffer_data = 0x0, pre_buffer_datalen = 0, file = 0xb7eb71f0 > "src/switch_ivr_play_say.c", func = 0xb7eb794b "switch_ivr_play_file", > line = 894} > read_codec = (switch_codec_t *) 0xb34fcb90 > p = 0xb7eaf8d4 "current_application" > asis = 0 '\0' > prefix = > timer_name = 0x0 > prebuf = > eof = 1 > bread = > __func__ = "switch_ivr_play_file" > __PRETTY_FUNCTION__ = "switch_ivr_play_file" > #4 0xad71da25 in ?? () from /opt/app/voip/ippbx.prod/mod/ > mod_dptools.so > No locals. > #5 0xb7dc6c54 in switch_core_session_exec (session=0xa7700030, > application_interface=0xb346b388, arg=0xb34fd790 "qet.mp3") > at src/switch_core_session.c:1332 > log = > lp = > event = (switch_event_t *) 0x0 > var = > channel = (switch_channel_t *) 0xa7704598 > expanded = 0xb34fd790 "qet.mp3" > app = 0xad725133 "playback" > __PRETTY_FUNCTION__ = "switch_core_session_exec" > __func__ = "switch_core_session_exec" > #6 0xb7dc70fe in switch_core_session_execute_application > (session=0xa7700030, app=0xb34fd780 "playback", arg=0xb34fd790 > "qet.mp3") > at src/switch_core_session.c:1254 > application_interface = (switch_application_interface_t *) > 0xb346b388 > __func__ = "switch_core_session_execute_application" > #7 0xb7dc93a4 in switch_core_session_run (session=0xa7700030) at > src/switch_core_state_machine.c:155 > proceed = > global_proceed = > do_extra_handlers = > state = > endstate = > endpoint_interface = > driver_state_handler = (const switch_state_handler_table_t *) > 0xb331f860 > application_state_handler = > thread_id = 3084680427 > env = {{__jmpbuf = {0, 0, 0, 0, 0, 0}, __mask_was_saved = 0, > __saved_mask = {__val = {0 }}}} > sig = > __func__ = "switch_core_session_run" > __PRETTY_FUNCTION__ = "switch_core_session_run" > #8 0xb7dc6725 in switch_core_session_thread (thread=0xb34fd3d0, > obj=0xa7700030) at src/switch_core_session.c:940 > session = (switch_core_session_t *) 0xa7700030 > event = > event_str = 0x0 > val = > __func__ = "switch_core_session_thread" > __PRETTY_FUNCTION__ = "switch_core_session_thread" > #9 0xb7e30bf6 in dummy_worker (opaque=0xb34fd3d0) at > threadproc/unix/thread.c:138 > No locals. > #10 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > No symbol table info available. > #11 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > > > > > (gdb) thread apply all bt > > Thread 31 (process 27201): > #0 0xb7f14410 in __kernel_vsyscall () > #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 > #2 0xb7e32799 in apr_sleep (t=1000000) at time/unix/time.c:246 > #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 > #4 0xb7db9154 in switch_console_loop () at src/switch_console.c:792 > #5 0xb7dcedf0 in switch_core_runtime_loop (bg=0) at src/ > switch_core.c:659 > #6 0x0804a36a in main (argc=1, argv=0xbffe65c4) at src/switch.c:666 > > Thread 30 (process 27202): > #0 0xb7f14410 in __kernel_vsyscall () > #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 > #2 0xb7e32799 in apr_sleep (t=100000) at time/unix/time.c:246 > #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 > #4 0xb7dbe2a4 in pool_thread (thread=0xb7a07da8, obj=0x0) at > src/switch_core_memory.c:421 > #5 0xb7e30bf6 in dummy_worker (opaque=0xb7a07da8) at > threadproc/unix/thread.c:138 > #6 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > #7 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > > Thread 29 (process 27203): > #0 0xb7f14410 in __kernel_vsyscall () > #1 0xb7d2eaa5 in pthread_cond_wait@@GLIBC_2.3.2 () from > /lib/tls/i686/cmov/libpthread.so.0 > #2 0xb7e2acaa in apr_thread_cond_wait (cond=0xb6fb53b8, > mutex=0xb6fb5388) at locks/unix/thread_cond.c:68 > #3 0xb7e21a3c in apr_queue_pop (queue=0xb6fb5358, data=0xb6ec03a8) at > misc/apr_queue.c:276 > #4 0xb7dadd44 in switch_queue_pop (queue=0xb6fb5358, data=0xb6ec03a8) > at src/switch_apr.c:879 > #5 0xb7ddf879 in switch_event_dispatch_thread (thread=0x8068140, > obj=0xb6fb5358) at src/switch_event.c:230 > #6 0xb7e30bf6 in dummy_worker (opaque=0x8068140) at > threadproc/unix/thread.c:138 > #7 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > #8 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > > Thread 28 (process 27204): > #0 0xb7f14410 in __kernel_vsyscall () > #1 0xb7d2eaa5 in pthread_cond_wait@@GLIBC_2.3.2 () from > /lib/tls/i686/cmov/libpthread.so.0 > #2 0xb7e2acaa in apr_thread_cond_wait (cond=0x805e380, > mutex=0x805e350) > at locks/unix/thread_cond.c:68 > #3 0xb7e21a3c in apr_queue_pop (queue=0x805e320, data=0xb66bf3a8) at > misc/apr_queue.c:276 > #4 0xb7dadd44 in switch_queue_pop (queue=0x805e320, > data=0xb66bf3a8) at > src/switch_apr.c:879 > #5 0xb7ddec2d in switch_event_thread (thread=0x8068160, > obj=0x805e320) > at src/switch_event.c:273 > #6 0xb7e30bf6 in dummy_worker (opaque=0x8068160) at > threadproc/unix/thread.c:138 > #7 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > #8 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > > Thread 27 (process 27205): > #0 0xb7f14410 in __kernel_vsyscall () > #1 0xb7d2eaa5 in pthread_cond_wait@@GLIBC_2.3.2 () from > /lib/tls/i686/cmov/libpthread.so.0 > #2 0xb7e2acaa in apr_thread_cond_wait (cond=0xb71d4b38, > mutex=0xb71d4b08) at locks/unix/thread_cond.c:68 > #3 0xb7e21a3c in apr_queue_pop (queue=0xb71d4ad8, data=0xb5ebe3a8) at > misc/apr_queue.c:276 > #4 0xb7dadd44 in switch_queue_pop (queue=0xb71d4ad8, data=0xb5ebe3a8) > at src/switch_apr.c:879 > #5 0xb7ddec2d in switch_event_thread (thread=0x8068180, > obj=0xb71d4ad8) > at src/switch_event.c:273 > #6 0xb7e30bf6 in dummy_worker (opaque=0x8068180) at > threadproc/unix/thread.c:138 > #7 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > ---Type to continue, or q to quit--- > #8 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > > Thread 26 (process 27206): > #0 0xb7f14410 in __kernel_vsyscall () > #1 0xb7d2eaa5 in pthread_cond_wait@@GLIBC_2.3.2 () from > /lib/tls/i686/cmov/libpthread.so.0 > #2 0xb7e2acaa in apr_thread_cond_wait (cond=0xb7171b38, > mutex=0xb7171b08) at locks/unix/thread_cond.c:68 > #3 0xb7e21a3c in apr_queue_pop (queue=0xb7171ad8, data=0xb56bd3a8) at > misc/apr_queue.c:276 > #4 0xb7dadd44 in switch_queue_pop (queue=0xb7171ad8, data=0xb56bd3a8) > at src/switch_apr.c:879 > #5 0xb7ddec2d in switch_event_thread (thread=0x80681a0, > obj=0xb7171ad8) > at src/switch_event.c:273 > #6 0xb7e30bf6 in dummy_worker (opaque=0x80681a0) at > threadproc/unix/thread.c:138 > #7 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > #8 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > > Thread 25 (process 27207): > #0 0xb7f14410 in __kernel_vsyscall () > #1 0xb7d2eaa5 in pthread_cond_wait@@GLIBC_2.3.2 () from > /lib/tls/i686/cmov/libpthread.so.0 > #2 0xb7e2acaa in apr_thread_cond_wait (cond=0xb6fb54a8, > mutex=0xb6fb5478) at locks/unix/thread_cond.c:68 > #3 0xb7e21a3c in apr_queue_pop (queue=0xb6fb5448, data=0xb4dce3a8) at > misc/apr_queue.c:276 > #4 0xb7dadd44 in switch_queue_pop (queue=0xb6fb5448, data=0xb4dce3a8) > at src/switch_apr.c:879 > #5 0xb7e082fd in log_thread (thread=0xb4e30ae0, obj=0x0) at > src/switch_log.c:209 > #6 0xb7e30bf6 in dummy_worker (opaque=0xb4e30ae0) at > threadproc/unix/thread.c:138 > #7 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > #8 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > > Thread 24 (process 27210): > #0 0xb7f14410 in __kernel_vsyscall () > #1 0xb7d2eaa5 in pthread_cond_wait@@GLIBC_2.3.2 () from > /lib/tls/i686/cmov/libpthread.so.0 > #2 0xb7e2acaa in apr_thread_cond_wait (cond=0x80c0c58, > mutex=0x80c0c28) > at locks/unix/thread_cond.c:68 > #3 0xb7daed54 in switch_thread_cond_wait (cond=0x80c0c58, > mutex=0x80c0c28) at src/switch_apr.c:359 > #4 0xb7e11266 in switch_cond_next () at src/switch_time.c:203 > #5 0xb7dc27a5 in switch_core_sql_thread (thread=0xb3567ae8, > obj=0x0) at > src/switch_core_sqldb.c:220 > #6 0xb7e30bf6 in dummy_worker (opaque=0xb3567ae8) at > threadproc/unix/thread.c:138 > #7 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > #8 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > > Thread 23 (process 27211): > #0 0xb7f14410 in __kernel_vsyscall () > #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 > #2 0xb7e32799 in apr_sleep (t=500000) at time/unix/time.c:246 > #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 > #4 0xb7dd2af4 in switch_scheduler_task_thread (thread=0x80baa90, > obj=0x0) at src/switch_scheduler.c:171 > #5 0xb7e30bf6 in dummy_worker (opaque=0x80baa90) at > threadproc/unix/thread.c:138 > #6 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > #7 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > > Thread 22 (process 27212): > #0 0xb7f14410 in __kernel_vsyscall () > #1 0xb7c7d676 in epoll_wait () from /lib/tls/i686/cmov/libc.so.6 > #2 0xb32b7126 in su_epoll_port_wait_events (self=0x80c6340, > tout=1000) > at su_epoll_port.c:491 > #3 0xb32bd9b8 in su_base_port_step (self=0x80c6340, tout=1000) at > su_base_port.c:442 > #4 0xb32b8551 in su_port_step (self=0x80c6340, tout=1000) at > su_port.h:326 > ---Type to continue, or q to quit--- > #5 0xb32b8521 in su_root_step (self=0x80c68f0, tout=1000) at > su_root.c:730 > #6 0xb31e8f29 in sofia_profile_thread_run (thread=0x80d2aa8, > obj=0x80d1e10) at sofia.c:831 > #7 0xb7e30bf6 in dummy_worker (opaque=0x80d2aa8) at > threadproc/unix/thread.c:138 > #8 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > #9 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > > Thread 21 (process 27213): > #0 0xb7f14410 in __kernel_vsyscall () > #1 0xb7c7d676 in epoll_wait () from /lib/tls/i686/cmov/libc.so.6 > #2 0xb32b7126 in su_epoll_port_wait_events (self=0x80dac30, > tout=1000) > at su_epoll_port.c:491 > #3 0xb32bd89b in su_base_port_run (self=0x80dac30) at > su_base_port.c:342 > #4 0xb32b842b in su_port_run (self=0x80dac30) at su_port.h:312 > #5 0xb32b8408 in su_root_run (self=0x80dacb0) at su_root.c:691 > #6 0xb32a8b31 in su_pthread_port_clone_main (varg=0xb311c0a8) at > su_pthread_port.c:321 > #7 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > #8 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > > Thread 20 (process 27214): > #0 0xb7f14410 in __kernel_vsyscall () > #1 0xb7c7d676 in epoll_wait () from /lib/tls/i686/cmov/libc.so.6 > #2 0xb32b7126 in su_epoll_port_wait_events (self=0x80da208, > tout=1000) > at su_epoll_port.c:491 > #3 0xb32bd9b8 in su_base_port_step (self=0x80da208, tout=1000) at > su_base_port.c:442 > #4 0xb32b8551 in su_port_step (self=0x80da208, tout=1000) at > su_port.h:326 > #5 0xb32b8521 in su_root_step (self=0x80d7558, tout=1000) at > su_root.c:730 > #6 0xb31e8f29 in sofia_profile_thread_run (thread=0x80dde88, > obj=0x80dd650) at sofia.c:831 > #7 0xb7e30bf6 in dummy_worker (opaque=0x80dde88) at > threadproc/unix/thread.c:138 > #8 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > #9 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > > Thread 19 (process 27215): > #0 0xb7f14410 in __kernel_vsyscall () > #1 0xb7c7d676 in epoll_wait () from /lib/tls/i686/cmov/libc.so.6 > #2 0xb32b7126 in su_epoll_port_wait_events (self=0x80e5228, > tout=1000) > at su_epoll_port.c:491 > #3 0xb32bd89b in su_base_port_run (self=0x80e5228) at > su_base_port.c:342 > #4 0xb32b842b in su_port_run (self=0x80e5228) at su_port.h:312 > #5 0xb32b8408 in su_root_run (self=0x80e3b40) at su_root.c:691 > #6 0xb32a8b31 in su_pthread_port_clone_main (varg=0xb211a0a8) at > su_pthread_port.c:321 > #7 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > #8 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > > Thread 18 (process 27216): > #0 0xb7f14410 in __kernel_vsyscall () > #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 > #2 0xb7e32799 in apr_sleep (t=10000) at time/unix/time.c:246 > #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 > #4 0xb31d4b37 in sofia_profile_worker_thread_run (thread=0x80d2b88, > obj=0x80d1e10) at sofia.c:656 > #5 0xb7e30bf6 in dummy_worker (opaque=0x80d2b88) at > threadproc/unix/thread.c:138 > #6 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > #7 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > > Thread 17 (process 27217): > ---Type to continue, or q to quit--- > #0 0xb7f14410 in __kernel_vsyscall () > #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 > #2 0xb7e32799 in apr_sleep (t=10000) at time/unix/time.c:246 > #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 > #4 0xb31d4b37 in sofia_profile_worker_thread_run (thread=0x80ddf68, > obj=0x80dd650) at sofia.c:656 > #5 0xb7e30bf6 in dummy_worker (opaque=0x80ddf68) at > threadproc/unix/thread.c:138 > #6 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > #7 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > > Thread 16 (process 27218): > #0 0xb7f14410 in __kernel_vsyscall () > #1 0xb7c7d676 in epoll_wait () from /lib/tls/i686/cmov/libc.so.6 > #2 0xb32b7126 in su_epoll_port_wait_events (self=0x80e66d8, > tout=1000) > at su_epoll_port.c:491 > #3 0xb32bd9b8 in su_base_port_step (self=0x80e66d8, tout=1000) at > su_base_port.c:442 > #4 0xb32b8551 in su_port_step (self=0x80e66d8, tout=1000) at > su_port.h:326 > #5 0xb32b8521 in su_root_step (self=0x80e49f0, tout=1000) at > su_root.c:730 > #6 0xb31e8f29 in sofia_profile_thread_run (thread=0x80e8710, > obj=0x80e7e90) at sofia.c:831 > #7 0xb7e30bf6 in dummy_worker (opaque=0x80e8710) at > threadproc/unix/thread.c:138 > #8 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > #9 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > > Thread 15 (process 27219): > #0 0xb7f14410 in __kernel_vsyscall () > #1 0xb7c7d676 in epoll_wait () from /lib/tls/i686/cmov/libc.so.6 > #2 0xb32b7126 in su_epoll_port_wait_events (self=0x80ee648, > tout=1000) > at su_epoll_port.c:491 > #3 0xb32bd89b in su_base_port_run (self=0x80ee648) at > su_base_port.c:342 > #4 0xb32b842b in su_port_run (self=0x80ee648) at su_port.h:312 > #5 0xb32b8408 in su_root_run (self=0x80f0bd0) at su_root.c:691 > #6 0xb32a8b31 in su_pthread_port_clone_main (varg=0xb00b20a8) at > su_pthread_port.c:321 > #7 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > #8 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > > Thread 14 (process 27220): > #0 0xb7f14410 in __kernel_vsyscall () > #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 > #2 0xb7e32799 in apr_sleep (t=10000) at time/unix/time.c:246 > #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 > #4 0xb31d4b37 in sofia_profile_worker_thread_run (thread=0x80e87f0, > obj=0x80e7e90) at sofia.c:656 > #5 0xb7e30bf6 in dummy_worker (opaque=0x80e87f0) at > threadproc/unix/thread.c:138 > #6 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > #7 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > > Thread 13 (process 27221): > #0 0xb7f14410 in __kernel_vsyscall () > #1 0xb7c72c07 in poll () from /lib/tls/i686/cmov/libc.so.6 > #2 0xb337af00 in wanpipe_wait (zchan=0xb3446128, flags=0xae751f80, > to=100) at src/ozmod/ozmod_wanpipe/ozmod_wanpipe.c:255 > #3 0xae837ae7 in zap_channel_wait (zchan=0x1, flags=0xae751f80, > to=100) > at src/zap_io.c:1479 > #4 0xae80aee8 in zap_isdn_run (me=0xb3416528, obj=0xb341dbc8) at > src/ozmod/ozmod_isdn/ozmod_isdn.c:1725 > #5 0xae8421ba in thread_launch (args=0xb3416528) at > src/zap_threadmutex.c:74 > #6 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > #7 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > ---Type to continue, or q to quit--- > > Thread 12 (process 27222): > #0 0xb7f14410 in __kernel_vsyscall () > #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 > #2 0xb7e32799 in apr_sleep (t=100000) at time/unix/time.c:246 > #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 > #4 0xb31f929e in sofia_presence_event_thread_run (thread=0x80cf958, > obj=0x0) at sofia_presence.c:664 > #5 0xb7e30bf6 in dummy_worker (opaque=0x80cf958) at > threadproc/unix/thread.c:138 > #6 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > #7 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > > Thread 11 (process 27223): > #0 0xb7f14410 in __kernel_vsyscall () > #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 > #2 0xb7e32799 in apr_sleep (t=1000000) at time/unix/time.c:246 > #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 > #4 0xad706ea6 in ?? () from /opt/app/voip/ippbx.prod/mod/mod_fifo.so > #5 0xb7e30bf6 in dummy_worker (opaque=0xad386b90) at > threadproc/unix/thread.c:138 > #6 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > #7 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > > Thread 10 (process 27225): > #0 0xb7f14410 in __kernel_vsyscall () > #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 > #2 0xb7e32799 in apr_sleep (t=1000000) at time/unix/time.c:246 > #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 > #4 0xabdf94ab in ?? () from > /opt/app/voip/ippbx.prod/mod/mod_local_stream.so > #5 0xb7e30bf6 in dummy_worker (opaque=0x81386e0) at > threadproc/unix/thread.c:138 > #6 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > #7 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > > Thread 9 (process 27226): > #0 0xb7f14410 in __kernel_vsyscall () > #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 > #2 0xb7e32799 in apr_sleep (t=1000000) at time/unix/time.c:246 > #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 > #4 0xabdf94ab in ?? () from > /opt/app/voip/ippbx.prod/mod/mod_local_stream.so > #5 0xb7e30bf6 in dummy_worker (opaque=0x815e728) at > threadproc/unix/thread.c:138 > #6 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > #7 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > > Thread 8 (process 27227): > #0 0xb7f14410 in __kernel_vsyscall () > #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 > #2 0xb7e32799 in apr_sleep (t=1000000) at time/unix/time.c:246 > #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 > #4 0xabdf94ab in ?? () from > /opt/app/voip/ippbx.prod/mod/mod_local_stream.so > #5 0xb7e30bf6 in dummy_worker (opaque=0x8184770) at > threadproc/unix/thread.c:138 > #6 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > #7 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > > ---Type to continue, or q to quit--- > Thread 7 (process 27228): > #0 0xb7f14410 in __kernel_vsyscall () > #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 > #2 0xb7e32799 in apr_sleep (t=1000) at time/unix/time.c:246 > #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 > #4 0xb7e12a59 in softtimer_runtime () at src/switch_time.c:459 > #5 0xb7dd4fb3 in switch_loadable_module_exec (thread=0x80bf8d8, > obj=0x80bf6c8) at src/switch_loadable_module.c:93 > #6 0xb7e30bf6 in dummy_worker (opaque=0x80bf8d8) at > threadproc/unix/thread.c:138 > #7 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > #8 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > > Thread 6 (process 27229): > #0 0xb7f14410 in __kernel_vsyscall () > #1 0xb7d31bb8 in accept () from /lib/tls/i686/cmov/libpthread.so.0 > #2 0xb7e2f90d in apr_socket_accept (new=0xaa2d434c, sock=0x81bbbb0, > connection_context=0x81bdaa8) at network_io/unix/sockets.c:187 > #3 0xb7dae3fb in switch_socket_accept (new_sock=0xaa2d434c, > sock=0x81bbbb0, pool=0x81bdaa8) at src/switch_apr.c:664 > #4 0xb33262f2 in mod_event_socket_runtime () at mod_event_socket.c: > 2134 > #5 0xb7dd4fb3 in switch_loadable_module_exec (thread=0x80bfb40, > obj=0x80bf930) at src/switch_loadable_module.c:93 > #6 0xb7e30bf6 in dummy_worker (opaque=0x80bfb40) at > threadproc/unix/thread.c:138 > #7 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > #8 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > > Thread 5 (process 27230): > #0 0xb7f14410 in __kernel_vsyscall () > #1 0xb7c72c07 in poll () from /lib/tls/i686/cmov/libc.so.6 > #2 0xb33ce86e in ?? () from /opt/app/voip/ippbx.prod/mod/ > mod_xml_rpc.so > #3 0xb33c1464 in ChanSwitchAccept (chanSwitchP=0x81f5030, > channelPP=0xa9ad30e0, channelInfoPP=0xa9ad30dc, errorP=0xa9ad30e4) > at ../../../../libs/xmlrpc-c/lib/abyss/src/chanswitch.c:149 > #4 0xb33cd37e in ServerRun (serverP=0xb33ffe4c) at > ../../../../libs/xmlrpc-c/lib/abyss/src/server.c:908 > #5 0xb33be832 in mod_xml_rpc_runtime () at mod_xml_rpc.c:837 > #6 0xb7dd4fb3 in switch_loadable_module_exec (thread=0x80bfda8, > obj=0x80bfb98) at src/switch_loadable_module.c:93 > #7 0xb7e30bf6 in dummy_worker (opaque=0x80bfda8) at > threadproc/unix/thread.c:138 > #8 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > #9 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > > Thread 4 (process 27231): > #0 0xb7f14410 in __kernel_vsyscall () > #1 0xb7d3199b in read () from /lib/tls/i686/cmov/libpthread.so.0 > #2 0xb7e9dfb3 in read_char (el=0x81c5040, cp=0xa92d235b > "?|???`!\b\200a\"\b?#-?;\237?@P\034\b?#-?\a") at > read.c:294 > #3 0xb7e9da9c in el_getc (el=0x81c5040, cp=0xa92d235b > "?|???`!\b\200a\"\b?#-?;\237?@P\034\b?#-?\a") at > read.c:362 > #4 0xb7e9dbdf in el_gets (el=0x81c5040, nread=0xa92d23a8) at read.c: > 241 > #5 0xb7db9f3b in console_thread (thread=0x82102d0, obj=0x8210248) at > src/switch_console.c:441 > #6 0xb7e30bf6 in dummy_worker (opaque=0x82102d0) at > threadproc/unix/thread.c:138 > #7 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > #8 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > > Thread 3 (process 27235): > #0 0xb7f14410 in __kernel_vsyscall () > #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 > #2 0xb7e32799 in apr_sleep (t=100000) at time/unix/time.c:246 > #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 > ---Type to continue, or q to quit--- > #4 0xb7df019e in switch_ivr_originate (session=0xb34dd080, > bleg=0xa8ad10c0, cause=0xa8ad10bc, bridgeto=0x823a528 > "openzap/1/a/04855711", > timelimit_sec=60, table=0xb7ecda20, cid_name_override=0x0, > cid_num_override=0x0, caller_profile_override=0x0, ovars=0x0, > flags=) at src/switch_ivr_originate.c:1793 > #5 0xad7231af in ?? () from /opt/app/voip/ippbx.prod/mod/ > mod_dptools.so > #6 0xb7dc6c54 in switch_core_session_exec (session=0xb34dd080, > application_interface=0xb346ba10, arg=0x823a528 "openzap/1/a/ > 04855711") > at src/switch_core_session.c:1332 > #7 0xb7dc70fe in switch_core_session_execute_application > (session=0xb34dd080, app=0x823a520 "bridge", arg=0x823a528 > "openzap/1/a/04855711") > at src/switch_core_session.c:1254 > #8 0xb7dc93a4 in switch_core_session_run (session=0xb34dd080) at > src/switch_core_state_machine.c:155 > #9 0xb7dc6725 in switch_core_session_thread (thread=0x823a050, > obj=0xb34dd080) at src/switch_core_session.c:940 > #10 0xb7e30bf6 in dummy_worker (opaque=0x823a050) at > threadproc/unix/thread.c:138 > #11 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > #12 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > > Thread 2 (process 27236): > #0 0xb7f14410 in __kernel_vsyscall () > #1 0xb7c72c07 in poll () from /lib/tls/i686/cmov/libc.so.6 > #2 0xb337af00 in wanpipe_wait (zchan=0xb341f9d8, flags=0xa80fee60, > to=40) at src/ozmod/ozmod_wanpipe/ozmod_wanpipe.c:255 > #3 0xae837ae7 in zap_channel_wait (zchan=0x1, flags=0xa80fee60, > to=40) > at src/zap_io.c:1479 > #4 0xae849059 in channel_read_frame (session=0xb34e5400, > frame=0xa80ff170, flags=0, stream_id=0) at mod_openzap.c:593 > #5 0xb7dcbe6f in switch_core_session_read_frame (session=0xb34e5400, > frame=0xa80ff170, flags=0, stream_id=0) at src/switch_core_io.c:161 > #6 0xb7e054ec in switch_ivr_sleep (session=0xb34e5400, ms=10, > sync=SWITCH_FALSE, args=0x0) at src/switch_ivr.c:262 > #7 0xb7deaf94 in originate_on_consume_media_transmit > (session=0xb34e5400) at src/switch_ivr_originate.c:47 > #8 0xb7dc8b74 in switch_core_session_run (session=0xb34e5400) at > src/switch_core_state_machine.c:476 > #9 0xb7dc6725 in switch_core_session_thread (thread=0xb34f1060, > obj=0xb34e5400) at src/switch_core_session.c:940 > #10 0xb7e30bf6 in dummy_worker (opaque=0xb34f1060) at > threadproc/unix/thread.c:138 > #11 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > #12 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > > Thread 1 (process 27234): > #0 0xabe0e408 in mpg123_delete at plt () from > /opt/app/voip/ippbx.prod/mod/mod_shout.so > #1 0xabe13325 in ?? () from /opt/app/voip/ippbx.prod/mod/mod_shout.so > #2 0xb7dc04b9 in switch_core_file_seek (fh=0xa76fef28, > cur_pos=0xa76fef88, samples=0, whence=1) at src/switch_core_file.c:345 > #3 0xb7dfab82 in switch_ivr_play_file (session=0xa7700030, > fh=0xa76fef28, > file=0xa7711ed8 "/opt/app/voip/ippbx.prod/sounds/de/callie/ > qet.mp3", > args=0xa76ff0d8) at src/switch_ivr_play_say.c:1262 > #4 0xad71da25 in ?? () from /opt/app/voip/ippbx.prod/mod/ > mod_dptools.so > #5 0xb7dc6c54 in switch_core_session_exec (session=0xa7700030, > application_interface=0xb346b388, arg=0xb34fd790 "qet.mp3") > at src/switch_core_session.c:1332 > #6 0xb7dc70fe in switch_core_session_execute_application > (session=0xa7700030, app=0xb34fd780 "playback", arg=0xb34fd790 > "qet.mp3") > at src/switch_core_session.c:1254 > #7 0xb7dc93a4 in switch_core_session_run (session=0xa7700030) at > src/switch_core_state_machine.c:155 > #8 0xb7dc6725 in switch_core_session_thread (thread=0xb34fd3d0, > obj=0xa7700030) at src/switch_core_session.c:940 > #9 0xb7e30bf6 in dummy_worker (opaque=0xb34fd3d0) at > threadproc/unix/thread.c:138 > #10 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > #11 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > > > (gdb) thread apply all bt full > > Thread 31 (process 27201): > #0 0xb7f14410 in __kernel_vsyscall () > No symbol table info available. > #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > #2 0xb7e32799 in apr_sleep (t=1000000) at time/unix/time.c:246 > tv = {tv_sec = 0, tv_usec = 940000} > #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 > No locals. > #4 0xb7db9154 in switch_console_loop () at src/switch_console.c:792 > arg = 1 > thread = (switch_thread_t *) 0x82102d0 > thd_attr = (switch_threadattr_t *) 0x8210298 > pool = (switch_memory_pool_t *) 0x8210248 > __func__ = "switch_console_loop" > __PRETTY_FUNCTION__ = "switch_console_loop" > #5 0xb7dcedf0 in switch_core_runtime_loop (bg=0) at src/ > switch_core.c:659 > No locals. > #6 0x0804a36a in main (argc=1, argv=0xbffe65c4) at src/switch.c:666 > pid_path = "/opt/app/voip/ippbx.prod/log/freeswitch.pid", '\0' > > pid_buffer = "27201", '\0' > old_pid_buffer = "27150", '\0' > pid_len = 5 > old_pid_len = 5 > err = 0x0 > nf = 0 > runas_user = 0x0 > runas_group = 0x0 > nc = 0 > pid = 27201 > x = 1111804576 > die = 0 > alt_dirs = 0 > known_opt = -1208927888 > high_prio = 0 > flags = 1 > ret = > destroy_status = > fd = (switch_file_t *) 0x80529b0 > pool = (switch_memory_pool_t *) 0x80528f0 > __PRETTY_FUNCTION__ = "main" > > Thread 30 (process 27202): > #0 0xb7f14410 in __kernel_vsyscall () > No symbol table info available. > #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > #2 0xb7e32799 in apr_sleep (t=100000) at time/unix/time.c:246 > tv = {tv_sec = 0, tv_usec = 100000} > #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 > ---Type to continue, or q to quit--- > No locals. > #4 0xb7dbe2a4 in pool_thread (thread=0xb7a07da8, obj=0x0) at > src/switch_core_memory.c:421 > No locals. > #5 0xb7e30bf6 in dummy_worker (opaque=0xb7a07da8) at > threadproc/unix/thread.c:138 > No locals. > #6 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > No symbol table info available. > #7 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > > Thread 29 (process 27203): > #0 0xb7f14410 in __kernel_vsyscall () > No symbol table info available. > #1 0xb7d2eaa5 in pthread_cond_wait@@GLIBC_2.3.2 () from > /lib/tls/i686/cmov/libpthread.so.0 > No symbol table info available. > #2 0xb7e2acaa in apr_thread_cond_wait (cond=0xb6fb53b8, > mutex=0xb6fb5388) at locks/unix/thread_cond.c:68 > rv = -512 > #3 0xb7e21a3c in apr_queue_pop (queue=0xb6fb5358, data=0xb6ec03a8) at > misc/apr_queue.c:276 > rv = 0 > #4 0xb7dadd44 in switch_queue_pop (queue=0xb6fb5358, data=0xb6ec03a8) > at src/switch_apr.c:879 > No locals. > #5 0xb7ddf879 in switch_event_dispatch_thread (thread=0x8068140, > obj=0xb6fb5358) at src/switch_event.c:230 > pop = (void *) 0x0 > event = (switch_event_t *) 0x0 > my_id = 0 > __func__ = "switch_event_dispatch_thread" > #6 0xb7e30bf6 in dummy_worker (opaque=0x8068140) at > threadproc/unix/thread.c:138 > No locals. > #7 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > No symbol table info available. > #8 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > > Thread 28 (process 27204): > #0 0xb7f14410 in __kernel_vsyscall () > No symbol table info available. > #1 0xb7d2eaa5 in pthread_cond_wait@@GLIBC_2.3.2 () from > /lib/tls/i686/cmov/libpthread.so.0 > No symbol table info available. > #2 0xb7e2acaa in apr_thread_cond_wait (cond=0x805e380, > mutex=0x805e350) > at locks/unix/thread_cond.c:68 > rv = -512 > #3 0xb7e21a3c in apr_queue_pop (queue=0x805e320, data=0xb66bf3a8) at > misc/apr_queue.c:276 > rv = 0 > #4 0xb7dadd44 in switch_queue_pop (queue=0x805e320, > data=0xb66bf3a8) at > src/switch_apr.c:879 > No locals. > #5 0xb7ddec2d in switch_event_thread (thread=0x8068160, > obj=0x805e320) > at src/switch_event.c:273 > pop = (void *) 0x0 > index = 0 > my_id = 0 > __func__ = "switch_event_thread" > #6 0xb7e30bf6 in dummy_worker (opaque=0x8068160) at > threadproc/unix/thread.c:138 > No locals. > ---Type to continue, or q to quit--- > #7 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > No symbol table info available. > #8 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > > Thread 27 (process 27205): > #0 0xb7f14410 in __kernel_vsyscall () > No symbol table info available. > #1 0xb7d2eaa5 in pthread_cond_wait@@GLIBC_2.3.2 () from > /lib/tls/i686/cmov/libpthread.so.0 > No symbol table info available. > #2 0xb7e2acaa in apr_thread_cond_wait (cond=0xb71d4b38, > mutex=0xb71d4b08) at locks/unix/thread_cond.c:68 > rv = -512 > #3 0xb7e21a3c in apr_queue_pop (queue=0xb71d4ad8, data=0xb5ebe3a8) at > misc/apr_queue.c:276 > rv = 0 > #4 0xb7dadd44 in switch_queue_pop (queue=0xb71d4ad8, data=0xb5ebe3a8) > at src/switch_apr.c:879 > No locals. > #5 0xb7ddec2d in switch_event_thread (thread=0x8068180, > obj=0xb71d4ad8) > at src/switch_event.c:273 > pop = (void *) 0x0 > index = 0 > my_id = 1 > __func__ = "switch_event_thread" > #6 0xb7e30bf6 in dummy_worker (opaque=0x8068180) at > threadproc/unix/thread.c:138 > No locals. > #7 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > No symbol table info available. > #8 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > > Thread 26 (process 27206): > #0 0xb7f14410 in __kernel_vsyscall () > No symbol table info available. > #1 0xb7d2eaa5 in pthread_cond_wait@@GLIBC_2.3.2 () from > /lib/tls/i686/cmov/libpthread.so.0 > No symbol table info available. > #2 0xb7e2acaa in apr_thread_cond_wait (cond=0xb7171b38, > mutex=0xb7171b08) at locks/unix/thread_cond.c:68 > rv = -512 > #3 0xb7e21a3c in apr_queue_pop (queue=0xb7171ad8, data=0xb56bd3a8) at > misc/apr_queue.c:276 > rv = 0 > #4 0xb7dadd44 in switch_queue_pop (queue=0xb7171ad8, data=0xb56bd3a8) > at src/switch_apr.c:879 > No locals. > #5 0xb7ddec2d in switch_event_thread (thread=0x80681a0, > obj=0xb7171ad8) > at src/switch_event.c:273 > pop = (void *) 0x0 > index = 0 > my_id = 2 > __func__ = "switch_event_thread" > #6 0xb7e30bf6 in dummy_worker (opaque=0x80681a0) at > threadproc/unix/thread.c:138 > No locals. > #7 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > No symbol table info available. > #8 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > > ---Type to continue, or q to quit--- > Thread 25 (process 27207): > #0 0xb7f14410 in __kernel_vsyscall () > No symbol table info available. > #1 0xb7d2eaa5 in pthread_cond_wait@@GLIBC_2.3.2 () from > /lib/tls/i686/cmov/libpthread.so.0 > No symbol table info available. > #2 0xb7e2acaa in apr_thread_cond_wait (cond=0xb6fb54a8, > mutex=0xb6fb5478) at locks/unix/thread_cond.c:68 > rv = -512 > #3 0xb7e21a3c in apr_queue_pop (queue=0xb6fb5448, data=0xb4dce3a8) at > misc/apr_queue.c:276 > rv = 0 > #4 0xb7dadd44 in switch_queue_pop (queue=0xb6fb5448, data=0xb4dce3a8) > at src/switch_apr.c:879 > No locals. > #5 0xb7e082fd in log_thread (thread=0xb4e30ae0, obj=0x0) at > src/switch_log.c:209 > pop = (void *) 0x0 > binding = (switch_log_binding_t *) 0x0 > __func__ = "log_thread" > #6 0xb7e30bf6 in dummy_worker (opaque=0xb4e30ae0) at > threadproc/unix/thread.c:138 > No locals. > #7 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > No symbol table info available. > #8 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > > Thread 24 (process 27210): > #0 0xb7f14410 in __kernel_vsyscall () > No symbol table info available. > #1 0xb7d2eaa5 in pthread_cond_wait@@GLIBC_2.3.2 () from > /lib/tls/i686/cmov/libpthread.so.0 > No symbol table info available. > #2 0xb7e2acaa in apr_thread_cond_wait (cond=0x80c0c58, > mutex=0x80c0c28) > at locks/unix/thread_cond.c:68 > rv = -512 > #3 0xb7daed54 in switch_thread_cond_wait (cond=0x80c0c58, > mutex=0x80c0c28) at src/switch_apr.c:359 > No locals. > #4 0xb7e11266 in switch_cond_next () at src/switch_time.c:203 > No locals. > #5 0xb7dc27a5 in switch_core_sql_thread (thread=0xb3567ae8, > obj=0x0) at > src/switch_core_sqldb.c:220 > pop = (void *) 0x811bf00 > itterations = 0 > trans = 0 '\0' > nothing_in_queue = 1 '\001' > len = 100 > sql_len = 65536 > sqlbuf = 0x80aa9a8 "" > newlen = > lc = 0 > __PRETTY_FUNCTION__ = "switch_core_sql_thread" > __func__ = "switch_core_sql_thread" > #6 0xb7e30bf6 in dummy_worker (opaque=0xb3567ae8) at > threadproc/unix/thread.c:138 > No locals. > #7 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > No symbol table info available. > #8 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > ---Type to continue, or q to quit--- > > Thread 23 (process 27211): > #0 0xb7f14410 in __kernel_vsyscall () > No symbol table info available. > #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > #2 0xb7e32799 in apr_sleep (t=500000) at time/unix/time.c:246 > tv = {tv_sec = 0, tv_usec = 340000} > #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 > No locals. > #4 0xb7dd2af4 in switch_scheduler_task_thread (thread=0x80baa90, > obj=0x0) at src/switch_scheduler.c:171 > __func__ = "switch_scheduler_task_thread" > #5 0xb7e30bf6 in dummy_worker (opaque=0x80baa90) at > threadproc/unix/thread.c:138 > No locals. > #6 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > No symbol table info available. > #7 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > > Thread 22 (process 27212): > #0 0xb7f14410 in __kernel_vsyscall () > No symbol table info available. > #1 0xb7c7d676 in epoll_wait () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > #2 0xb32b7126 in su_epoll_port_wait_events (self=0x80c6340, > tout=1000) > at su_epoll_port.c:491 > j = -1290679408 > n = -1288576964 > events = 0 > index = -1288973717 > version = 1 > M = 4 > ev = 0xb311c0e0 > __PRETTY_FUNCTION__ = "su_epoll_port_wait_events" > #3 0xb32bd9b8 in su_base_port_step (self=0x80c6340, tout=1000) at > su_base_port.c:442 > now = {tv_sec = 3443337943, tv_usec = 249063} > __PRETTY_FUNCTION__ = "su_base_port_step" > #4 0xb32b8551 in su_port_step (self=0x80c6340, tout=1000) at > su_port.h:326 > base = (su_virtual_port_t *) 0x80c6340 > #5 0xb32b8521 in su_root_step (self=0x80c68f0, tout=1000) at > su_root.c:730 > __PRETTY_FUNCTION__ = "su_root_step" > #6 0xb31e8f29 in sofia_profile_thread_run (thread=0x80d2aa8, > obj=0x80d1e10) at sofia.c:831 > pool = > node = (sip_alias_node_t *) 0xb32f45dc > s_event = (switch_event_t *) 0x0 > sanity = > __func__ = "sofia_profile_thread_run" > __PRETTY_FUNCTION__ = "sofia_profile_thread_run" > #7 0xb7e30bf6 in dummy_worker (opaque=0x80d2aa8) at > threadproc/unix/thread.c:138 > No locals. > #8 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > No symbol table info available. > ---Type to continue, or q to quit--- > #9 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > > Thread 21 (process 27213): > #0 0xb7f14410 in __kernel_vsyscall () > No symbol table info available. > #1 0xb7c7d676 in epoll_wait () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > #2 0xb32b7126 in su_epoll_port_wait_events (self=0x80dac30, > tout=1000) > at su_epoll_port.c:491 > j = -1299074328 > n = 1 > events = 0 > index = 2 > version = 4 > M = 4 > ev = 0xb291b260 > __PRETTY_FUNCTION__ = "su_epoll_port_wait_events" > #3 0xb32bd89b in su_base_port_run (self=0x80dac30) at > su_base_port.c:342 > tout = 1000 > __PRETTY_FUNCTION__ = "su_base_port_run" > #4 0xb32b842b in su_port_run (self=0x80dac30) at su_port.h:312 > base = (su_virtual_port_t *) 0x80dac30 > #5 0xb32b8408 in su_root_run (self=0x80dacb0) at su_root.c:691 > __PRETTY_FUNCTION__ = "su_root_run" > #6 0xb32a8b31 in su_pthread_port_clone_main (varg=0xb311c0a8) at > su_pthread_port.c:321 > arg = (struct clone_args *) 0x0 > task = {{sut_port = 0x80dac30, sut_root = 0x80dacb0}} > zap = 1 > #7 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > No symbol table info available. > #8 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > > Thread 20 (process 27214): > #0 0xb7f14410 in __kernel_vsyscall () > No symbol table info available. > #1 0xb7c7d676 in epoll_wait () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > #2 0xb32b7126 in su_epoll_port_wait_events (self=0x80da208, > tout=1000) > at su_epoll_port.c:491 > j = -1307464816 > n = -1288576964 > events = 0 > index = -1288973717 > version = 1 > M = 4 > ev = 0xb211a0e0 > __PRETTY_FUNCTION__ = "su_epoll_port_wait_events" > #3 0xb32bd9b8 in su_base_port_step (self=0x80da208, tout=1000) at > su_base_port.c:442 > now = {tv_sec = 3443337943, tv_usec = 148997} > __PRETTY_FUNCTION__ = "su_base_port_step" > #4 0xb32b8551 in su_port_step (self=0x80da208, tout=1000) at > su_port.h:326 > ---Type to continue, or q to quit--- > base = (su_virtual_port_t *) 0x80da208 > #5 0xb32b8521 in su_root_step (self=0x80d7558, tout=1000) at > su_root.c:730 > __PRETTY_FUNCTION__ = "su_root_step" > #6 0xb31e8f29 in sofia_profile_thread_run (thread=0x80dde88, > obj=0x80dd650) at sofia.c:831 > pool = > node = (sip_alias_node_t *) 0xb32f45dc > s_event = (switch_event_t *) 0x0 > sanity = > __func__ = "sofia_profile_thread_run" > __PRETTY_FUNCTION__ = "sofia_profile_thread_run" > #7 0xb7e30bf6 in dummy_worker (opaque=0x80dde88) at > threadproc/unix/thread.c:138 > No locals. > #8 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > No symbol table info available. > #9 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > > Thread 19 (process 27215): > #0 0xb7f14410 in __kernel_vsyscall () > No symbol table info available. > #1 0xb7c7d676 in epoll_wait () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > #2 0xb32b7126 in su_epoll_port_wait_events (self=0x80e5228, > tout=1000) > at su_epoll_port.c:491 > j = -1315859736 > n = 1 > events = 0 > index = 1 > version = 3 > M = 4 > ev = 0xb1919260 > __PRETTY_FUNCTION__ = "su_epoll_port_wait_events" > #3 0xb32bd89b in su_base_port_run (self=0x80e5228) at > su_base_port.c:342 > tout = 1000 > __PRETTY_FUNCTION__ = "su_base_port_run" > #4 0xb32b842b in su_port_run (self=0x80e5228) at su_port.h:312 > base = (su_virtual_port_t *) 0x80e5228 > #5 0xb32b8408 in su_root_run (self=0x80e3b40) at su_root.c:691 > __PRETTY_FUNCTION__ = "su_root_run" > #6 0xb32a8b31 in su_pthread_port_clone_main (varg=0xb211a0a8) at > su_pthread_port.c:321 > arg = (struct clone_args *) 0x0 > task = {{sut_port = 0x80e5228, sut_root = 0x80e3b40}} > zap = 1 > #7 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > No symbol table info available. > #8 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > > Thread 18 (process 27216): > #0 0xb7f14410 in __kernel_vsyscall () > No symbol table info available. > #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 > ---Type to continue, or q to quit--- > No symbol table info available. > #2 0xb7e32799 in apr_sleep (t=10000) at time/unix/time.c:246 > tv = {tv_sec = 0, tv_usec = 10000} > #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 > No locals. > #4 0xb31d4b37 in sofia_profile_worker_thread_run (thread=0x80d2b88, > obj=0x80d1e10) at sofia.c:656 > ireg_loops = 1 > gateway_loops = 0 > loops = 95 > qsize = 4294966782 > __PRETTY_FUNCTION__ = "sofia_profile_worker_thread_run" > #5 0xb7e30bf6 in dummy_worker (opaque=0x80d2b88) at > threadproc/unix/thread.c:138 > No locals. > #6 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > No symbol table info available. > #7 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > > Thread 17 (process 27217): > #0 0xb7f14410 in __kernel_vsyscall () > No symbol table info available. > #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > #2 0xb7e32799 in apr_sleep (t=10000) at time/unix/time.c:246 > tv = {tv_sec = 0, tv_usec = 10000} > #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 > No locals. > #4 0xb31d4b37 in sofia_profile_worker_thread_run (thread=0x80ddf68, > obj=0x80dd650) at sofia.c:656 > ireg_loops = 1 > gateway_loops = 0 > loops = 95 > qsize = 4294966782 > __PRETTY_FUNCTION__ = "sofia_profile_worker_thread_run" > #5 0xb7e30bf6 in dummy_worker (opaque=0x80ddf68) at > threadproc/unix/thread.c:138 > No locals. > #6 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > No symbol table info available. > #7 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > > Thread 16 (process 27218): > #0 0xb7f14410 in __kernel_vsyscall () > No symbol table info available. > #1 0xb7c7d676 in epoll_wait () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > #2 0xb32b7126 in su_epoll_port_wait_events (self=0x80e66d8, > tout=1000) > at su_epoll_port.c:491 > j = -1341445232 > n = -1288576964 > events = 0 > index = -1288973717 > version = 1 > ---Type to continue, or q to quit--- > M = 4 > ev = 0xb00b20e0 > __PRETTY_FUNCTION__ = "su_epoll_port_wait_events" > #3 0xb32bd9b8 in su_base_port_step (self=0x80e66d8, tout=1000) at > su_base_port.c:442 > now = {tv_sec = 3443337943, tv_usec = 560073} > __PRETTY_FUNCTION__ = "su_base_port_step" > #4 0xb32b8551 in su_port_step (self=0x80e66d8, tout=1000) at > su_port.h:326 > base = (su_virtual_port_t *) 0x80e66d8 > #5 0xb32b8521 in su_root_step (self=0x80e49f0, tout=1000) at > su_root.c:730 > __PRETTY_FUNCTION__ = "su_root_step" > #6 0xb31e8f29 in sofia_profile_thread_run (thread=0x80e8710, > obj=0x80e7e90) at sofia.c:831 > pool = > node = (sip_alias_node_t *) 0xb32f45dc > s_event = (switch_event_t *) 0x0 > sanity = > __func__ = "sofia_profile_thread_run" > __PRETTY_FUNCTION__ = "sofia_profile_thread_run" > #7 0xb7e30bf6 in dummy_worker (opaque=0x80e8710) at > threadproc/unix/thread.c:138 > No locals. > #8 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > No symbol table info available. > #9 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > > Thread 15 (process 27219): > #0 0xb7f14410 in __kernel_vsyscall () > No symbol table info available. > #1 0xb7c7d676 in epoll_wait () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > #2 0xb32b7126 in su_epoll_port_wait_events (self=0x80ee648, > tout=1000) > at su_epoll_port.c:491 > j = -1349840152 > n = 1 > events = 0 > index = 1 > version = 3 > M = 4 > ev = 0xaf8b1260 > __PRETTY_FUNCTION__ = "su_epoll_port_wait_events" > #3 0xb32bd89b in su_base_port_run (self=0x80ee648) at > su_base_port.c:342 > tout = 1000 > __PRETTY_FUNCTION__ = "su_base_port_run" > #4 0xb32b842b in su_port_run (self=0x80ee648) at su_port.h:312 > base = (su_virtual_port_t *) 0x80ee648 > #5 0xb32b8408 in su_root_run (self=0x80f0bd0) at su_root.c:691 > __PRETTY_FUNCTION__ = "su_root_run" > #6 0xb32a8b31 in su_pthread_port_clone_main (varg=0xb00b20a8) at > su_pthread_port.c:321 > arg = (struct clone_args *) 0x0 > task = {{sut_port = 0x80ee648, sut_root = 0x80f0bd0}} > zap = 1 > #7 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > No symbol table info available. > ---Type to continue, or q to quit--- > #8 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > > Thread 14 (process 27220): > #0 0xb7f14410 in __kernel_vsyscall () > No symbol table info available. > #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > #2 0xb7e32799 in apr_sleep (t=10000) at time/unix/time.c:246 > tv = {tv_sec = 0, tv_usec = 0} > #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 > No locals. > #4 0xb31d4b37 in sofia_profile_worker_thread_run (thread=0x80e87f0, > obj=0x80e7e90) at sofia.c:656 > ireg_loops = 0 > gateway_loops = 0 > loops = 85 > qsize = 4294966782 > __PRETTY_FUNCTION__ = "sofia_profile_worker_thread_run" > #5 0xb7e30bf6 in dummy_worker (opaque=0x80e87f0) at > threadproc/unix/thread.c:138 > No locals. > #6 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > No symbol table info available. > #7 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > > Thread 13 (process 27221): > #0 0xb7f14410 in __kernel_vsyscall () > No symbol table info available. > #1 0xb7c72c07 in poll () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > #2 0xb337af00 in wanpipe_wait (zchan=0xb3446128, flags=0xae751f80, > to=100) at src/ozmod/ozmod_wanpipe/ozmod_wanpipe.c:255 > inflags = 1 > result = > #3 0xae837ae7 in zap_channel_wait (zchan=0x1, flags=0xae751f80, > to=100) > at src/zap_io.c:1479 > __PRETTY_FUNCTION__ = "zap_channel_wait" > #4 0xae80aee8 in zap_isdn_run (me=0xb3416528, obj=0xb341dbc8) at > src/ozmod/ozmod_isdn/ozmod_isdn.c:1725 > flags = ZAP_READ > status = ZAP_TIMEOUT > span = > isdn_data = (zap_isdn_data_t *) 0xae753008 > frame = "\002\001\002\002\b\002\200\002\001\036\002\201\210", > '\0' > len = 13 > errs = 0 > __func__ = "zap_isdn_run" > #5 0xae8421ba in thread_launch (args=0xb3416528) at > src/zap_threadmutex.c:74 > exit_val = > #6 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > No symbol table info available. > #7 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > > ---Type to continue, or q to quit--- > Thread 12 (process 27222): > #0 0xb7f14410 in __kernel_vsyscall () > No symbol table info available. > #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > #2 0xb7e32799 in apr_sleep (t=100000) at time/unix/time.c:246 > tv = {tv_sec = 0, tv_usec = 100000} > #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 > No locals. > #4 0xb31f929e in sofia_presence_event_thread_run (thread=0x80cf958, > obj=0x0) at sofia_presence.c:664 > count = 0 > pop = (void *) 0xb3462cb8 > __func__ = "sofia_presence_event_thread_run" > #5 0xb7e30bf6 in dummy_worker (opaque=0x80cf958) at > threadproc/unix/thread.c:138 > No locals. > #6 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > No symbol table info available. > #7 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > > Thread 11 (process 27223): > #0 0xb7f14410 in __kernel_vsyscall () > No symbol table info available. > #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > #2 0xb7e32799 in apr_sleep (t=1000000) at time/unix/time.c:246 > tv = {tv_sec = 0, tv_usec = 320000} > #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 > No locals. > #4 0xad706ea6 in ?? () from /opt/app/voip/ippbx.prod/mod/mod_fifo.so > No locals. > #5 0xb7e30bf6 in dummy_worker (opaque=0xad386b90) at > threadproc/unix/thread.c:138 > No locals. > #6 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > No symbol table info available. > #7 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > > Thread 10 (process 27225): > #0 0xb7f14410 in __kernel_vsyscall () > No symbol table info available. > #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > #2 0xb7e32799 in apr_sleep (t=1000000) at time/unix/time.c:246 > tv = {tv_sec = 0, tv_usec = 840000} > #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 > No locals. > #4 0xabdf94ab in ?? () from > /opt/app/voip/ippbx.prod/mod/mod_local_stream.so > No locals. > #5 0xb7e30bf6 in dummy_worker (opaque=0x81386e0) at > threadproc/unix/thread.c:138 > No locals. > ---Type to continue, or q to quit--- > #6 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > No symbol table info available. > #7 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > > Thread 9 (process 27226): > #0 0xb7f14410 in __kernel_vsyscall () > No symbol table info available. > #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > #2 0xb7e32799 in apr_sleep (t=1000000) at time/unix/time.c:246 > tv = {tv_sec = 0, tv_usec = 840000} > #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 > No locals. > #4 0xabdf94ab in ?? () from > /opt/app/voip/ippbx.prod/mod/mod_local_stream.so > No locals. > #5 0xb7e30bf6 in dummy_worker (opaque=0x815e728) at > threadproc/unix/thread.c:138 > No locals. > #6 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > No symbol table info available. > #7 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > > Thread 8 (process 27227): > #0 0xb7f14410 in __kernel_vsyscall () > No symbol table info available. > #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > #2 0xb7e32799 in apr_sleep (t=1000000) at time/unix/time.c:246 > tv = {tv_sec = 0, tv_usec = 840000} > #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 > No locals. > #4 0xabdf94ab in ?? () from > /opt/app/voip/ippbx.prod/mod/mod_local_stream.so > No locals. > #5 0xb7e30bf6 in dummy_worker (opaque=0x8184770) at > threadproc/unix/thread.c:138 > No locals. > #6 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > No symbol table info available. > #7 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > > Thread 7 (process 27228): > #0 0xb7f14410 in __kernel_vsyscall () > No symbol table info available. > #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > #2 0xb7e32799 in apr_sleep (t=1000) at time/unix/time.c:246 > tv = {tv_sec = 0, tv_usec = 1000} > #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 > No locals. > #4 0xb7e12a59 in softtimer_runtime () at src/switch_time.c:459 > ---Type to continue, or q to quit--- > current_ms = 1544 > x = > tick = 144 > ts = > last = 1234349144129112 > fwd_errs = 0 > rev_errs = 0 > __func__ = "softtimer_runtime" > #5 0xb7dd4fb3 in switch_loadable_module_exec (thread=0x80bf8d8, > obj=0x80bf6c8) at src/switch_loadable_module.c:93 > status = > module = (switch_loadable_module_t *) 0x80c0bc8 > __PRETTY_FUNCTION__ = "switch_loadable_module_exec" > __func__ = "switch_loadable_module_exec" > #6 0xb7e30bf6 in dummy_worker (opaque=0x80bf8d8) at > threadproc/unix/thread.c:138 > No locals. > #7 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > No symbol table info available. > #8 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > > Thread 6 (process 27229): > #0 0xb7f14410 in __kernel_vsyscall () > No symbol table info available. > #1 0xb7d31bb8 in accept () from /lib/tls/i686/cmov/libpthread.so.0 > No symbol table info available. > #2 0xb7e2f90d in apr_socket_accept (new=0xaa2d434c, sock=0x81bbbb0, > connection_context=0x81bdaa8) at network_io/unix/sockets.c:187 > No locals. > #3 0xb7dae3fb in switch_socket_accept (new_sock=0xaa2d434c, > sock=0x81bbbb0, pool=0x81bdaa8) at src/switch_apr.c:664 > No locals. > #4 0xb33262f2 in mod_event_socket_runtime () at mod_event_socket.c: > 2134 > pool = (switch_memory_pool_t *) 0x81bbaa0 > listener_pool = (switch_memory_pool_t *) 0x81bdaa8 > rv = > sa = (switch_sockaddr_t *) 0x81bbaf8 > inbound_socket = (switch_socket_t *) 0x81bdb00 > listener = > x = > __func__ = "mod_event_socket_runtime" > #5 0xb7dd4fb3 in switch_loadable_module_exec (thread=0x80bfb40, > obj=0x80bf930) at src/switch_loadable_module.c:93 > status = > module = (switch_loadable_module_t *) 0xb340ec28 > __PRETTY_FUNCTION__ = "switch_loadable_module_exec" > __func__ = "switch_loadable_module_exec" > #6 0xb7e30bf6 in dummy_worker (opaque=0x80bfb40) at > threadproc/unix/thread.c:138 > No locals. > #7 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > No symbol table info available. > #8 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > > Thread 5 (process 27230): > ---Type to continue, or q to quit--- > #0 0xb7f14410 in __kernel_vsyscall () > No symbol table info available. > #1 0xb7c72c07 in poll () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > #2 0xb33ce86e in ?? () from /opt/app/voip/ippbx.prod/mod/ > mod_xml_rpc.so > No locals. > #3 0xb33c1464 in ChanSwitchAccept (chanSwitchP=0x81f5030, > channelPP=0xa9ad30e0, channelInfoPP=0xa9ad30dc, errorP=0xa9ad30e4) > at ../../../../libs/xmlrpc-c/lib/abyss/src/chanswitch.c:149 > No locals. > #4 0xb33cd37e in ServerRun (serverP=0xb33ffe4c) at > ../../../../libs/xmlrpc-c/lib/abyss/src/server.c:908 > srvP = (struct _TServer * const) 0x81f4fc8 > #5 0xb33be832 in mod_xml_rpc_runtime () at mod_xml_rpc.c:837 > registryP = (xmlrpc_registry *) 0x81bb960 > env = {fault_occurred = 0, fault_code = 0, fault_string = 0x0} > logfile = "/opt/app/voip/ippbx.prod/log/freeswitch_http.log", > '\0' > hi = > var = (const void *) 0x80609b8 > val = (void *) 0x805dc78 > __func__ = "mod_xml_rpc_runtime" > #6 0xb7dd4fb3 in switch_loadable_module_exec (thread=0x80bfda8, > obj=0x80bfb98) at src/switch_loadable_module.c:93 > status = > module = (switch_loadable_module_t *) 0xb3405178 > __PRETTY_FUNCTION__ = "switch_loadable_module_exec" > __func__ = "switch_loadable_module_exec" > #7 0xb7e30bf6 in dummy_worker (opaque=0x80bfda8) at > threadproc/unix/thread.c:138 > No locals. > #8 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > No symbol table info available. > #9 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > > Thread 4 (process 27231): > #0 0xb7f14410 in __kernel_vsyscall () > No symbol table info available. > #1 0xb7d3199b in read () from /lib/tls/i686/cmov/libpthread.so.0 > No symbol table info available. > #2 0xb7e9dfb3 in read_char (el=0x81c5040, cp=0xa92d235b > "?|???`!\b\200a\"\b?#-?;\237?@P\034\b?#-?\a") at > read.c:294 > num_read = 136073656 > tried = 0 > #3 0xb7e9da9c in el_getc (el=0x81c5040, cp=0xa92d235b > "?|???`!\b\200a\"\b?#-?;\237?@P\034\b?#-?\a") at > read.c:362 > num_read = -1456659621 > ma = (c_macro_t *) 0x81c52e0 > #4 0xb7e9dbdf in el_gets (el=0x81c5040, nread=0xa92d23a8) at read.c: > 241 > cmdnum = 232 '?' > num = 136405224 > ch = -1 '?' > #5 0xb7db9f3b in console_thread (thread=0x82102d0, obj=0x8210248) at > src/switch_console.c:441 > arg = 1 > count = 7 > line = 0x82160e8 "" > pool = (switch_memory_pool_t *) 0x8210248 > ---Type to continue, or q to quit--- > __func__ = "console_thread" > __PRETTY_FUNCTION__ = "console_thread" > #6 0xb7e30bf6 in dummy_worker (opaque=0x82102d0) at > threadproc/unix/thread.c:138 > No locals. > #7 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > No symbol table info available. > #8 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > > Thread 3 (process 27235): > #0 0xb7f14410 in __kernel_vsyscall () > No symbol table info available. > #1 0xb7c75881 in select () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > #2 0xb7e32799 in apr_sleep (t=100000) at time/unix/time.c:246 > tv = {tv_sec = 0, tv_usec = 20000} > #3 0xb7e10e1e in do_sleep (t=4294966782) at src/switch_time.c:109 > No locals. > #4 0xb7df019e in switch_ivr_originate (session=0xb34dd080, > bleg=0xa8ad10c0, cause=0xa8ad10bc, bridgeto=0x823a528 > "openzap/1/a/04855711", > timelimit_sec=60, table=0xb7ecda20, cid_name_override=0x0, > cid_num_override=0x0, caller_profile_override=0x0, ovars=0x0, > flags=) at src/switch_ivr_originate.c:1793 > end = > var_begin = > originate_status = {{peer_session = 0xb34e5400, peer_channel = > 0xb34e9968, caller_profile = 0xb34e1dc8, ring_ready = 0 '\0', > early_media = 1 '\001', answered = 0 '\0', > per_channel_timelimit_sec > = 0, per_channel_progress_timelimit_sec = 0}, {peer_session = 0x0, > peer_channel = 0x0, caller_profile = 0x0, ring_ready = 0 '\0', > early_media = 0 '\0', answered = 0 '\0', per_channel_timelimit_sec = > 0, > per_channel_progress_timelimit_sec = 0} } > dftflags = 0 > myflags = 0 > pipe_names = {0xa811de20 "openzap", 0x0 } > data = > status = SWITCH_STATUS_SUCCESS > caller_channel = (switch_channel_t *) 0xb34e15e8 > peer_names = {0xa811de20 "openzap", 0x0 } > new_session = (switch_core_session_t *) 0xb34e5400 > peer_session = > new_profile = (switch_caller_profile_t *) 0xb34e1dc8 > caller_caller_profile = > chan_type = > chan_data = > peer_channel = > ringback = {audio_buffer = 0x0, ts = {TONES = {{freqs = {0, 0, > 0, 0, 0, 0}} }, channels = 0, rate = 0, > duration = 0, wait = 0, tmp_duration = 0, tmp_wait = 0, loops = 0, > LOOPS = 0, decay_factor = 0, decay_direction = 0, decay_step = 0, > volume = 0, debug = 0, debug_stream = 0x0, user_data = 0x0, > buffer = > 0x0, datalen = 0, samples = 0, dynamic = 0, handler = 0}, fhb = { > file_interface = 0x0, flags = 0, fd = 0x0, samples = 0, > samplerate = > 0, native_rate = 0, channels = 0 '\0', format = 0, sections = 0, > seekable = 0, sample_count = 0, speed = 0, memory_pool = 0x0, > prebuf > = 0, interval = 0, private_info = 0x0, handler = 0x0, pos = 0, > audio_buffer = 0x0, sp_audio_buffer = 0x0, thresh = 0, silence_hits > = 0, offset_pos = 0, last_pos = 0, vol = 0, resampler = 0x0, > buffer = 0x0, dbuf = 0x0, dbuflen = 0, pre_buffer = 0x0, > pre_buffer_data = 0x0, pre_buffer_datalen = 0, file = 0x0, func = 0x0, > line = 0}, fh = 0x0, silence = 0, asis = 0 '\0'} > start = 1234349134 > read_frame = (switch_frame_t *) 0x0 > ---Type to continue, or q to quit--- > pool = (switch_memory_pool_t *) 0x0 > r = 0 > i = -1286728216 > and_argc = 1 > or_argc = 1 > sleep_ms = 1000 > try = 0 > retries = 1 > write_codec = {codec_interface = 0x0, implementation = 0x0, > fmtp_in = 0x0, fmtp_out = 0x0, codec_settings = {quality = 0, > complexity = 0, enhancement = 0, vad = 0, vbr = 0, vbr_quality = 0, > abr = 0, dtx = 0, preproc = 0, pp_vad = 0, pp_agc = 0, > pp_agc_level = 0, pp_denoise = 0, pp_dereverb = 0, > pp_dereverb_decay > = 0, pp_dereverb_level = 0}, flags = 0, memory_pool = 0x0, > private_info = 0x0, agreed_pt = 0 '\0', mutex = 0x0} > write_frame = {codec = 0x0, source = 0x0, packet = 0x0, > packetlen = 0, data = 0x821ec80, datalen = 0, buflen = 4096, samples > = 0, > rate = 0, payload = 0 '\0', timestamp = 0, seq = 0, ssrc = 0, m = > SWITCH_FALSE, flags = 0} > pass = 0 '\0' > odata = 0xa811e160 "openzap/1/a/04855711" > var = > reason = SWITCH_CAUSE_SUCCESS > to = 0 '\0' > var_val = > vars = 0x0 > var_block_count = 0 > e = > ringback_data = 0x0 > read_codec = > message = (switch_core_session_message_t *) 0x0 > var_event = (switch_event_t *) 0xb346cea8 > fail_on_single_reject = 0 '\0' > fail_on_single_reject_var = 0x0 > loop_data = 0xa811de20 "openzap" > progress_timelimit_sec = 60 > oglobals = {session = 0xb34dd080, idx = -1, hups = 0, file = > '\0' , key = '\0' , > early_ok = 0 '\0', ring_ready = 1 '\001', sent_ring = 1 '\001', > progress = 1 '\001', return_ring_ready = 0 '\0', > monitor_early_media_ring = 0 '\0', monitor_early_media_fail = 0 '\0', > gen_ringback = 0 '\0', ignore_early_media = 1 '\001', > ignore_ring_ready = 0 '\0'} > __PRETTY_FUNCTION__ = "switch_ivr_originate" > __func__ = "switch_ivr_originate" > #5 0xad7231af in ?? () from /opt/app/voip/ippbx.prod/mod/ > mod_dptools.so > No locals. > #6 0xb7dc6c54 in switch_core_session_exec (session=0xb34dd080, > application_interface=0xb346ba10, arg=0x823a528 "openzap/1/a/ > 04855711") > at src/switch_core_session.c:1332 > log = > lp = > event = (switch_event_t *) 0x0 > var = > channel = (switch_channel_t *) 0xb34e15e8 > expanded = 0x823a528 "openzap/1/a/04855711" > app = 0xad7253b0 "bridge" > __PRETTY_FUNCTION__ = "switch_core_session_exec" > __func__ = "switch_core_session_exec" > #7 0xb7dc70fe in switch_core_session_execute_application > (session=0xb34dd080, app=0x823a520 "bridge", arg=0x823a528 > "openzap/1/a/04855711") > ---Type to continue, or q to quit--- > at src/switch_core_session.c:1254 > application_interface = (switch_application_interface_t *) > 0xb346ba10 > __func__ = "switch_core_session_execute_application" > #8 0xb7dc93a4 in switch_core_session_run (session=0xb34dd080) at > src/switch_core_state_machine.c:155 > proceed = > global_proceed = > do_extra_handlers = > state = > endstate = > endpoint_interface = > driver_state_handler = (const switch_state_handler_table_t *) > 0xb331f860 > application_state_handler = > thread_id = 3084680427 > env = {{__jmpbuf = {134603552, -1287115232, -1209317480, 27232, > -1287651304, 24}, __mask_was_saved = -1210923727, __saved_mask = { > __val = {2819623592, 2829914760, 3084048908, 2819620880, 27232, > 1, > 3008239368, 2829914744, 3085760892, 250000, 3084048908, 2829914760, > 3085085008, 1, 3085760892, 2829914792, 3085760892, 0, > 134564192, > 3084043569, 3085085008, 134564244, 3085760892, 2829914824, 27232, > 134564240, 3007790064, 3084573819, 3085760892, 3008213112, > 3084048908, 2829914856}}}} > sig = > __func__ = "switch_core_session_run" > __PRETTY_FUNCTION__ = "switch_core_session_run" > #9 0xb7dc6725 in switch_core_session_thread (thread=0x823a050, > obj=0xb34dd080) at src/switch_core_session.c:940 > session = (switch_core_session_t *) 0xb34dd080 > event = > event_str = 0x0 > val = > __func__ = "switch_core_session_thread" > __PRETTY_FUNCTION__ = "switch_core_session_thread" > #10 0xb7e30bf6 in dummy_worker (opaque=0x823a050) at > threadproc/unix/thread.c:138 > No locals. > #11 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > No symbol table info available. > #12 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > > Thread 2 (process 27236): > #0 0xb7f14410 in __kernel_vsyscall () > No symbol table info available. > #1 0xb7c72c07 in poll () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > #2 0xb337af00 in wanpipe_wait (zchan=0xb341f9d8, flags=0xa80fee60, > to=40) at src/ozmod/ozmod_wanpipe/ozmod_wanpipe.c:255 > inflags = 1 > result = > #3 0xae837ae7 in zap_channel_wait (zchan=0x1, flags=0xa80fee60, > to=40) > at src/zap_io.c:1479 > __PRETTY_FUNCTION__ = "zap_channel_wait" > #4 0xae849059 in channel_read_frame (session=0xb34e5400, > frame=0xa80ff170, flags=0, stream_id=0) at mod_openzap.c:593 > channel = (switch_channel_t *) 0xb34e9968 > len = > wflags = ZAP_READ > dtmf = '\0' > status = > total_to = 80 > ---Type to continue, or q to quit--- > chunk = 40 > do_break = 0 > __PRETTY_FUNCTION__ = "channel_read_frame" > __func__ = "channel_read_frame" > #5 0xb7dcbe6f in switch_core_session_read_frame (session=0xb34e5400, > frame=0xa80ff170, flags=0, stream_id=0) at src/switch_core_io.c:161 > ptr = > status = SWITCH_STATUS_SUCCESS > need_codec = > perfect = > do_bugs = -1212118646 > do_resample = 0 > is_cng = 0 > flag = 0 > __PRETTY_FUNCTION__ = "switch_core_session_read_frame" > __func__ = "switch_core_session_read_frame" > #6 0xb7e054ec in switch_ivr_sleep (session=0xb34e5400, ms=10, > sync=SWITCH_FALSE, args=0x0) at src/switch_ivr.c:262 > channel = (switch_channel_t *) 0xb34e9968 > status = SWITCH_STATUS_SUCCESS > start = 1234349144119246 > now = 0 > done = 1234349144129246 > read_frame = (switch_frame_t *) 0x0 > cng_frame = {codec = 0x0, source = 0x0, packet = 0x0, packetlen > = 0, data = 0xa80ff176, datalen = 2, buflen = 2, samples = 0, > rate = 0, payload = 0 '\0', timestamp = 0, seq = 0, ssrc = 0, m = > SWITCH_FALSE, flags = 1} > data = "\000" > write_frame = {codec = 0x0, source = 0x0, packet = 0x0, > packetlen = 0, data = 0x0, datalen = 0, buflen = 0, samples = 0, > rate = 0, > payload = 0 '\0', timestamp = 0, seq = 0, ssrc = 0, m = SWITCH_FALSE, > flags = 0} > abuf = (unsigned char *) 0x0 > imp = {codec_type = SWITCH_CODEC_TYPE_AUDIO, ianacode = 0 '\0', > iananame = 0x0, fmtp = 0x0, samples_per_second = 0, > actual_samples_per_second = 0, bits_per_second = 0, > microseconds_per_packet = 0, samples_per_packet = 0, > decoded_bytes_per_packet = 0, > encoded_bytes_per_packet = 0, number_of_channels = 0 '\0', > codec_frames_per_packet = 0, init = 0, encode = 0, decode = 0, > destroy = 0, > codec_id = 0, next = 0x0} > codec = {codec_interface = 0x0, implementation = 0x0, fmtp_in = > 0x0, fmtp_out = 0x0, codec_settings = {quality = 0, complexity = 0, > enhancement = 0, vad = 0, vbr = 0, vbr_quality = 0, abr = 0, dtx = > 0, preproc = 0, pp_vad = 0, pp_agc = 0, pp_agc_level = 0, > pp_denoise = 0, pp_dereverb = 0, pp_dereverb_decay = 0, > pp_dereverb_level = 0}, flags = 0, memory_pool = 0x0, private_info = > 0x0, > agreed_pt = 0 '\0', mutex = 0x0} > sval = 0 > var = > __func__ = "switch_ivr_sleep" > __PRETTY_FUNCTION__ = "switch_ivr_sleep" > #7 0xb7deaf94 in originate_on_consume_media_transmit > (session=0xb34e5400) at src/switch_ivr_originate.c:47 > channel = (switch_channel_t *) 0xb34e9968 > #8 0xb7dc8b74 in switch_core_session_run (session=0xb34e5400) at > src/switch_core_state_machine.c:476 > proceed = 1 > global_proceed = 0 > do_extra_handlers = > state = > endstate = > endpoint_interface = > driver_state_handler = (const switch_state_handler_table_t *) > 0xae8523c0 > application_state_handler = > ---Type to continue, or q to quit--- > thread_id = 3084680427 > env = {{__jmpbuf = {0, -1287122688, -1475349896, 27233, > -1211162264, -1210902852}, __mask_was_saved = -1210923727, > __saved_mask = { > __val = {0, 3084099572, 3084048908, 3085631456, 27233, 1, > 3008273032, 2819617400, 3085760892, 250000, 3084048908, 2819617416, > 3085085008, 1, 3085760892, 2819617448, 3085760892, 0, > 134564192, > 3084043569, 3085085008, 134564244, 3085760892, 2819617480, 27233, > 134564240, 135380168, 3084573819, 3085760892, 3008246776, > 3084048908, 2819617512}}}} > sig = > __func__ = "switch_core_session_run" > __PRETTY_FUNCTION__ = "switch_core_session_run" > #9 0xb7dc6725 in switch_core_session_thread (thread=0xb34f1060, > obj=0xb34e5400) at src/switch_core_session.c:940 > session = (switch_core_session_t *) 0xb34e5400 > event = > event_str = 0x0 > val = > __func__ = "switch_core_session_thread" > __PRETTY_FUNCTION__ = "switch_core_session_thread" > #10 0xb7e30bf6 in dummy_worker (opaque=0xb34f1060) at > threadproc/unix/thread.c:138 > No locals. > #11 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > No symbol table info available. > #12 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > > Thread 1 (process 27234): > #0 0xabe0e408 in mpg123_delete at plt () from > /opt/app/voip/ippbx.prod/mod/mod_shout.so > No locals. > #1 0xabe13325 in ?? () from /opt/app/voip/ippbx.prod/mod/mod_shout.so > No locals. > #2 0xb7dc04b9 in switch_core_file_seek (fh=0xa76fef28, > cur_pos=0xa76fef88, samples=0, whence=1) at src/switch_core_file.c:345 > status = 4294967295 > __PRETTY_FUNCTION__ = "switch_core_file_seek" > #3 0xb7dfab82 in switch_ivr_play_file (session=0xa7700030, > fh=0xa76fef28, > file=0xa7711ed8 "/opt/app/voip/ippbx.prod/sounds/de/callie/ > qet.mp3", > args=0xa76ff0d8) at src/switch_ivr_play_say.c:1262 > channel = (switch_channel_t *) 0xa7704598 > dtmf = {digit = 0 '\0', duration = 0} > interval = 2909916864 > samples = 320 > framelen = 640 > sample_start = > olen = 320 > llen = 320 > write_frame = {codec = 0xa76fefb8, source = 0x0, packet = 0x0, > packetlen = 0, data = 0xa7743f30, datalen = 640, buflen = 32768, > samples = 320, rate = 16000, payload = 0 '\0', timestamp = 0, seq = > 0, > ssrc = 0, m = SWITCH_FALSE, flags = 0} > timer = {interval = 0, flags = 0, samples = 0, samplecount = 0, > timer_interface = 0x0, memory_pool = 0x0, private_info = 0x0, > diff = 0, tick = 0} > codec = {codec_interface = 0x80cd8f8, implementation = > 0x80cdf20, fmtp_in = 0x0, fmtp_out = 0x0, codec_settings = {quality > = 0, > complexity = 0, enhancement = 0, vad = 0, vbr = 0, vbr_quality = 0, > abr = 0, dtx = 0, preproc = 0, pp_vad = 0, pp_agc = 0, > pp_agc_level = 0, pp_denoise = 0, pp_dereverb = 0, > pp_dereverb_decay > = 0, pp_dereverb_level = 0}, flags = 3, memory_pool = 0xb34fbb38, > private_info = 0x0, agreed_pt = 0 '\0', mutex = 0xa7711f10} > pool = (switch_memory_pool_t *) 0xb34fbb38 > status = SWITCH_STATUS_SUCCESS > lfh = {file_interface = 0x8136258, flags = 3085, fd = 0x0, > samples = 0, samplerate = 16000, native_rate = 16000, channels = 1 > '\001', > ---Type to continue, or q to quit--- > format = 0, sections = 0, seekable = 0, sample_count = 729088, > speed = > 0, memory_pool = 0xa7712040, prebuf = 0, interval = 0, > private_info = 0xa7714048, handler = 0x0, pos = 0, audio_buffer = > 0xb3476f88, sp_audio_buffer = 0x0, thresh = 0, silence_hits = 0, > offset_pos = 364640, last_pos = 0, vol = 0, resampler = 0x0, buffer = > 0x0, dbuf = 0x0, dbuflen = 0, pre_buffer = 0x0, > pre_buffer_data = 0x0, pre_buffer_datalen = 0, file = 0xb7eb71f0 > "src/switch_ivr_play_say.c", func = 0xb7eb794b "switch_ivr_play_file", > line = 894} > read_codec = (switch_codec_t *) 0xb34fcb90 > p = 0xb7eaf8d4 "current_application" > asis = 0 '\0' > prefix = > timer_name = 0x0 > prebuf = > eof = 1 > bread = > __func__ = "switch_ivr_play_file" > __PRETTY_FUNCTION__ = "switch_ivr_play_file" > #4 0xad71da25 in ?? () from /opt/app/voip/ippbx.prod/mod/ > mod_dptools.so > No locals. > #5 0xb7dc6c54 in switch_core_session_exec (session=0xa7700030, > application_interface=0xb346b388, arg=0xb34fd790 "qet.mp3") > at src/switch_core_session.c:1332 > log = > lp = > event = (switch_event_t *) 0x0 > var = > channel = (switch_channel_t *) 0xa7704598 > expanded = 0xb34fd790 "qet.mp3" > app = 0xad725133 "playback" > __PRETTY_FUNCTION__ = "switch_core_session_exec" > __func__ = "switch_core_session_exec" > #6 0xb7dc70fe in switch_core_session_execute_application > (session=0xa7700030, app=0xb34fd780 "playback", arg=0xb34fd790 > "qet.mp3") > at src/switch_core_session.c:1254 > application_interface = (switch_application_interface_t *) > 0xb346b388 > __func__ = "switch_core_session_execute_application" > #7 0xb7dc93a4 in switch_core_session_run (session=0xa7700030) at > src/switch_core_state_machine.c:155 > proceed = > global_proceed = > do_extra_handlers = > state = > endstate = > endpoint_interface = > driver_state_handler = (const switch_state_handler_table_t *) > 0xb331f860 > application_state_handler = > thread_id = 3084680427 > env = {{__jmpbuf = {0, 0, 0, 0, 0, 0}, __mask_was_saved = 0, > __saved_mask = {__val = {0 }}}} > sig = > __func__ = "switch_core_session_run" > __PRETTY_FUNCTION__ = "switch_core_session_run" > #8 0xb7dc6725 in switch_core_session_thread (thread=0xb34fd3d0, > obj=0xa7700030) at src/switch_core_session.c:940 > session = (switch_core_session_t *) 0xa7700030 > event = > event_str = 0x0 > val = > ---Type to continue, or q to quit--- > __func__ = "switch_core_session_thread" > __PRETTY_FUNCTION__ = "switch_core_session_thread" > #9 0xb7e30bf6 in dummy_worker (opaque=0xb34fd3d0) at > threadproc/unix/thread.c:138 > No locals. > #10 0xb7d2a4fb in start_thread () from /lib/tls/i686/cmov/ > libpthread.so.0 > No symbol table info available. > #11 0xb7c7ce5e in clone () from /lib/tls/i686/cmov/libc.so.6 > No symbol table info available. > > > regards > Helmut > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From helmut.kuper at ewetel.de Wed Feb 11 04:41:22 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Wed, 11 Feb 2009 13:41:22 +0100 Subject: [Freeswitch-users] Question: SIP BYE authentication Message-ID: <4992C772.4090906@ewetel.de> Hello, my FS is connected to my SIP-DDI softswitch, which requires all SIP requests sent by a registered SIP account to be authenticated. I found that when FS sends a BYE FreeSWITCH ignores the authentication challenge (SIP/2.0 407) received from proxy and simply terminates the session. Is there a way to configure FS in that way that it react on auth challenges for BYEs ? regards Helmut From helmut.kuper at ewetel.de Wed Feb 11 04:43:45 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Wed, 11 Feb 2009 13:43:45 +0100 Subject: [Freeswitch-users] segfault when shoutcast plays mp3 and extension hangs up In-Reply-To: References: <4992BFFC.50006@ewetel.de> Message-ID: <4992C801.4020906@ewetel.de> Hi Brian, of course. Will do it as soon as I have a second FS plattform for testing SVN trunks. thx and regards Helmut On 11.02.2009 13:16, Brian West wrote: > The proper location to post this is on Jira... please in the future > report ALL issues via Jira. They'll get lost if not done this way. > > Thanks, > Brian > From edpimentl at gmail.com Wed Feb 11 04:58:47 2009 From: edpimentl at gmail.com (EdPimentl) Date: Wed, 11 Feb 2009 07:58:47 -0500 Subject: [Freeswitch-users] TODAY/URGENT: Stop IETF Enactment of Patented Standard for TLS In-Reply-To: <4992C6B7.A0699D98@RealMeasures.dyndns.org> References: <48804AEE.2231AA61@RealMeasures.dyndns.org> <49524E6F.BB09B1C4@RealMeasures.dyndns.org> <4992C6B7.A0699D98@RealMeasures.dyndns.org> Message-ID: <9dc4a1670902110458w7858da12v449052271521d8af@mail.gmail.com> Have you seen this? and how will impact FS in the future? -------------------------------------------------------------------------------------------- >From Seth: (Urgent. Send your note TODAY and CONFIRM the automatic reply from IETF. You can cc campaigns at fsf.org . Three links below, FSF's action page, Glyn Moody's blog, and the list announcement for TLS-AUTHZ at IETF. -- Seth) > http://www.fsf.org/news/reoppose-tls-authz-standard Send comments opposing TLS-authz standard by February 11 Last January, the Free Software Foundation issued an alert to efforts at the Internet Engineering Task Force (IETF) to sneak a patent-encumbered standard for "TLS authorization" through a back-door approval process that was referenced as "experimental" or "informational" (http://www.fsf.org/news/reoppose-tls-authz-standard/newsitem_view). The many comments sent to IETF at that time alerted committee members to this attempt and successfully prevented the standard gaining approval. Unfortunately, attempts to push through this standard have been renewed and become more of a threat. The proposal now at the IETF has a changed status from "experimental" to "proposed standard". The FSF is again issuing an alert and request for comments to be sent urgently and prior to the February 11 deadline to ietf at ietf.org. Please include us in your message by a CC to campaigns at fsf.org. You should also expect an automated reply from ietf at ietf.org, which you will need to answer to confirm your original message. That patent in question is claimed by RedPhone Security (https://datatracker.ietf.org/ipr/1026/). RedPhone has given a license to anyone who implements the protocol, but they still threaten to sue anyone that uses it. If our voice is strong enough, the IETF will not approve this standard on any level unless the patent threat is removed entirely with a royalty-free license for all users. Further background for your comment See the IETF summary: > http://www.ietf.org/mail-archive/web/ietf-announce/current/msg05617.html Much of the communication on the Internet happens between computers according to standards that define common languages. If we are going to live in a free world using free software, our software must be allowed to speak these languages. Unfortunately, discussions about possible new standards are tempting opportunities for people who would prefer to profit by extending proprietary control over our communities. If someone holds a software patent on a technique that a programmer or user has to use in order to make use of a standard, then no one is free without getting permission from and paying the patent holder (http://www.gnu.org/philosophy/fighting-software-patents.html). If we are not careful, standards can become major barriers to computer users having and exercising their freedom. We depend on organizations like the Internet Engineering Task Force (IETF) and the Internet Engineering Steering Group (IESG) to evaluate new proposals for standards and make sure that they are not encumbered by patents or any other sort of restriction that would prevent free software users and programmers from participating in the world they define. In February 2006, a standard for "TLS authorization" was introduced in the IETF for consideration (http://tools.ietf.org/wg/tls/draft-housley-tls-authz-extns-07.txt). Very late in the discussion, a company called RedPhone Security disclosed (this disclosure has subsequently been unpublished from the IETF website) that they applied for a patent which would need to be licensed to anyone wanting to practice the standard (https://datatracker.ietf.org/ipr/833/). After this disclosure, the proposal was rejected. Despite claims that RedPhone have offered a license for implementation of this protocol, users of this protocol would still be threatened by the patent. The IETF should continue to oppose this standard until RedPhone provide a royalty-free license for all users. Media Contacts Peter T. Brown Executive Director Free Software Foundation (617)542-5942 campaigns at fsf.org --- > http://www.computerworlduk.com/community/blogs/index.cfm?blogid=14&entryid=1845 Help Fight This Patent-Encumbered IETF Standard February 10, 2009 Posted by: Glyn Moody I've written numerous times about the importance of writing to governments about their hare-brained schemes, but this one is rather different. In this case, it's the normally sane Internet Engineering Task Force that wants to do something really daft. The FSF explains: Last January, the Free Software Foundation issued an alert to efforts at the Internet Engineering Task Force (IETF) to sneak a patent-encumbered standard for "TLS authorization" through a back-door approval process that was referenced as "experimental" or "informational". The many comments sent to IETF at that time alerted committee members to this attempt and successfully prevented the standard gaining approval. Unfortunately, attempts to push through this standard have been renewed and become more of a threat. The proposal now at the IETF has a changed status from "experimental" to "proposed standard". This is a throwback to the bad old days of sneaking patents into nominal standards. It is yet another reason why such patents should not be given in the first place. But until such time as the patent offices around the world come to their senses, the only option is to fight patent-encumbered standards on an individual basis. Here are the details for doing so: The FSF is again issuing an alert and request for comments to be sent urgently and prior to the February 11 deadline to ietf at ietf.org. Please include us in your message by a CC to campaigns at fsf.org. You should also expect an automated reply from ietf at ietf.org, which you will need to answer to confirm your original message. Here's what I've sent: I am writing to ask you not to approve the proposed patent-encumbered standard for TLS authorisation. To do so would fly in the face of the IETF's fundamental commitment to openness. It would weaken not just the standard itself, but the IETF's authority in this sphere. --- > http://www.ietf.org/mail-archive/web/ietf-announce/current/msg05617.html Fourth Last Call: draft-housley-tls-authz-extns * To: IETF-Announce * Subject: Fourth Last Call: draft-housley-tls-authz-extns * From: The IESG * Date: Wed, 14 Jan 2009 08:18:20 -0800 (PST) * List-archive: * Reply-to: ietf at ietf.org On June 27, 2006, the IESG approved "Transport Layer Security (TLS) Authorization Extensions," (draft-housley-tls-authz-extns) as a proposed standard. On November 29, 2006, Redphone Security (with whom Mark Brown, a co-author of the draft is affiliated) filed IETF IPR disclosure 767. Because of the timing of the IPR Disclosure, the IESG withdrew its approval of draft-housley-tls-authz-extns. A second IETF Last Call was initiated to determine whether the IETF community still had consensus to publish draft-housley-tls-authz-extns as a proposed standard given the IPR claimed. Consensus to publish as a standards track document was not demonstrated, and the document was withdrawn from IESG consideration. A third IETF Last Call was initiated to determine whether the IETF community had consensus to publish draft-housley-tls-authz-extns as an experimental track RFC with knowledge of the IPR disclosure from Redphone Security. Consensus to publish as experimental was not demonstrated; a substantial segment of the community objected to publication on any track in light of the IPR terms. Since the third Last Call, RedPhone Security filed IETF IPR disclosure 1026. This disclosure statement asserts in part that "the techniques for sending and receiving authorizations defined in TLS Authorizations Extensions (version draft-housley-tls-authz-extns-07.txt) do not infringe upon RedPhone Security's intellectual property rights". The full text of IPR disclosure 1026 is available at: https://datatracker.ietf.org/ipr/1026/ This Last Call is intended to determine whether the IETF community had consensus to publish draft-housley-tls-authz-extns as a proposed standard given IPR Disclosure 1026. The IESG is considering approving this draft as a standards track RFC. The IESG solicits final comments on whether the IETF community has consensus to publish draft-housley-tls-authz-extns as a proposed standard. Comments can be sent to ietf at ietf.org or exceptionally to iesg at ietf.org. Comments should be sent by 2009-02-11. A URL of this Internet-Draft is: http://www.ietf.org/internet-drafts/draft-housley-tls-authz-extns-07.txt _______________________________________________ IETF-Announce mailing list IETF-Announce at ietf.org https://www.ietf.org/mailman/listinfo/ietf-announce -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090211/23c6d0ab/attachment-0002.html From anthony.minessale at gmail.com Wed Feb 11 05:58:56 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 11 Feb 2009 07:58:56 -0600 Subject: [Freeswitch-users] segfault when shoutcast plays mp3 and extension hangs up In-Reply-To: <4992C801.4020906@ewetel.de> References: <4992BFFC.50006@ewetel.de> <4992C801.4020906@ewetel.de> Message-ID: <191c3a030902110558i9702139g6b12aaf15f6d3aac@mail.gmail.com> Why do you need to wait? jira is just a website, just go there and file the bug and attach the file that causes the issue. On Wed, Feb 11, 2009 at 6:43 AM, Helmut Kuper wrote: > Hi Brian, > > of course. Will do it as soon as I have a second FS plattform for > testing SVN trunks. > > thx and regards > Helmut > > > On 11.02.2009 13:16, Brian West wrote: > > The proper location to post this is on Jira... please in the future > > report ALL issues via Jira. They'll get lost if not done this way. > > > > Thanks, > > Brian > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090211/2054327f/attachment-0002.html From helmut.kuper at ewetel.de Wed Feb 11 06:12:18 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Wed, 11 Feb 2009 15:12:18 +0100 Subject: [Freeswitch-users] segfault when shoutcast plays mp3 and extension hangs up In-Reply-To: <191c3a030902110558i9702139g6b12aaf15f6d3aac@mail.gmail.com> References: <4992BFFC.50006@ewetel.de> <4992C801.4020906@ewetel.de> <191c3a030902110558i9702139g6b12aaf15f6d3aac@mail.gmail.com> Message-ID: <4992DCC2.3020702@ewetel.de> Hi Anthony, On 11.02.2009 14:58, Anthony Minessale wrote: > Why do you need to wait? > jira is just a website, just go there and file the bug and attach the > file that causes the issue. Well, there is a question on jira, which makes sure that I have reproduced the bug on SVN trunk ... but I'm not on latest trunk currently. For Jira it's a mandatory field which has to be anwered with yes. Reproduced with SVN Trunk? is required. ^* Reproduced with SVN Trunk?: Have you tried to reproduce this with SVN Trunk? If not STOP, make current and try... if the problem still persists verify you're on the latest SVN rev. as of RIGHT NOW please continue with your issue report. regards Helmut -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090211/57c75022/attachment-0002.html From anthony.minessale at gmail.com Wed Feb 11 06:17:39 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 11 Feb 2009 08:17:39 -0600 Subject: [Freeswitch-users] segfault when shoutcast plays mp3 and extension hangs up In-Reply-To: <4992DCC2.3020702@ewetel.de> References: <4992BFFC.50006@ewetel.de> <4992C801.4020906@ewetel.de> <191c3a030902110558i9702139g6b12aaf15f6d3aac@mail.gmail.com> <4992DCC2.3020702@ewetel.de> Message-ID: <191c3a030902110617n3dcd19cfr204c00792e1e87ed@mail.gmail.com> if the alternative is to post it to the mailing list, you have our permission this one time to answer "not yet" so you have somewhere to attach the bad file so we can reproduce it. On Wed, Feb 11, 2009 at 8:12 AM, Helmut Kuper wrote: > Hi Anthony, > > On 11.02.2009 14:58, Anthony Minessale wrote: > > Why do you need to wait? > jira is just a website, just go there and file the bug and attach the file > that causes the issue. > > Well, there is a question on jira, which makes sure that I have reproduced > the bug on SVN trunk ... but I'm not on latest trunk currently. For Jira > it's a mandatory field which has to be anwered with yes. > > > > Reproduced with SVN Trunk? is required. * Reproduced with SVN Trunk?: Have > you tried to reproduce this with SVN Trunk? If not STOP, make current and > try... if the problem still persists verify you're on the latest SVN rev. as > of RIGHT NOW please continue with your issue report. > > > regards > Helmut > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090211/73631d18/attachment-0002.html From helmut.kuper at ewetel.de Wed Feb 11 06:48:57 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Wed, 11 Feb 2009 15:48:57 +0100 Subject: [Freeswitch-users] segfault when shoutcast plays mp3 and extension hangs up In-Reply-To: <191c3a030902110617n3dcd19cfr204c00792e1e87ed@mail.gmail.com> References: <4992BFFC.50006@ewetel.de> <4992C801.4020906@ewetel.de> <191c3a030902110558i9702139g6b12aaf15f6d3aac@mail.gmail.com> <4992DCC2.3020702@ewetel.de> <191c3a030902110617n3dcd19cfr204c00792e1e87ed@mail.gmail.com> Message-ID: <4992E559.6060506@ewetel.de> Hi Anthony, I quickly have setup a test server with current trunk. So I can now enter there a "YES" into that field. Current trunk crashed as well. But thx for stretching the jira rules a bit :) I attached the file on jira in http://jira.freeswitch.org/browse/MODFORM-24 Can you delete it asap because of copyright reasons, please? regards helmut On 11.02.2009 15:17, Anthony Minessale wrote: > if the alternative is to post it to the mailing list, you have our > permission this one time to answer "not yet" so you have somewhere to > attach the bad file so we can reproduce it. From c_cav_01 at yahoo.com Wed Feb 11 07:23:01 2009 From: c_cav_01 at yahoo.com (Chris) Date: Wed, 11 Feb 2009 07:23:01 -0800 (PST) Subject: [Freeswitch-users] Strange error message In-Reply-To: Message-ID: <226281.90672.qm@web55102.mail.re4.yahoo.com> In that particular instance, no, it's not needed.? It's assigned a numeric value and is just a variable that will disappear when it goes out of scope.? You could deallocate or release the variable if you wished to be tidy. ? In the case of a recordset being returned by the server (an SQL "SELECT") then it will be a recordset object and would need to be closed. ? If you look at your error, your trying to index a number.? e.g curs assigned numeric by the UPDATE returning the number of rows affected, then your trying to look for a :close method or property on a numeric var which isn't going to work and the compiler is interpreting that as an attempt to index it. ? In general, on most SQL servers, SQL insert, update and delete calls return numerics, only select returns recordsets. --- On Tue, 2/10/09, Nik Middleton wrote: From: Nik Middleton Subject: Re: [Freeswitch-users] Strange error message To: freeswitch-users at lists.freeswitch.org Date: Tuesday, February 10, 2009, 4:22 PM So what you?re saying is that I can comment out curs:close()? as it?s not needed? ? Regards, ? ? ? From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Chris Sent: 10 February 2009 21:19 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Strange error message ? Closing the connection will force the server to close any open transactions, as well as release recordsets in local memory that reference the connection. ? However curs is not a recordset.? An SQL update is going to return an integer (rows affected) or boolean depending on the which server you use since no recordset is actually requested. --- On Tue, 2/10/09, Nik Middleton wrote: From: Nik Middleton Subject: [Freeswitch-users] Strange error message To: freeswitch-users at lists.freeswitch.org Date: Tuesday, February 10, 2009, 2:04 PM Hi Guys, ? I?m baffled by this error.? I?m updating the db on call hang-up If I comment out curs:close() no error, but I?m concerned about memory leaks.? Can anyone tell me what FS is complaining about? ? The db gets updated in both cases ? Regards ? ? ? require "luasql.mysql" ? function myHangupHook(s, status, arg) ??????????? freeswitch.consoleLog("info", " : They hung up on US!!!\n"); ??? ??????? env = assert (luasql.mysql()) ??????????? con = assert (env:connect("xxxxl","xxxxxxxxx","pass","192.168.3.205")) ??????????? curs = assert (con:execute"UPDATE callers SET lastcall = 'BOB' WHERE id = 33292") ??????????? curs:close() ??????????? con:close() ??????????? env:close() ??????????? freeswitch.consoleLog("NOTICE", "myHangupHook: " .. status .. "\n"); ??? --error() end ? ? ? ? 2009-02-10 20:53:20 [INFO] switch_cpp.cpp:1086 console_log()? : They hung up on US!!! 2009-02-10 20:53:20 [ERR] mod_lua.cpp:176 lua_parse_and_execute() /usr/local/freeswitch/scripts/helloworld.lua:50: attempt to index global 'curs' (a number value) stack traceback: ??????? /usr/local/freeswitch/scripts/helloworld.lua:50: in function ??????? [C]: in function 'hangup' ??????? /usr/local/freeswitch/scripts/helloworld.lua:70: in main chunk 2009-02-10 20:53:20 [NOTICE] switch_core_session.c:957 switch_core_session_thread() Session 63 (sofia/internal/1001 at 192.168.3.206) Ended_______________________________________________Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org ?_______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090211/5773e5fb/attachment-0002.html From anthony.minessale at gmail.com Wed Feb 11 07:29:52 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 11 Feb 2009 09:29:52 -0600 Subject: [Freeswitch-users] segfault when shoutcast plays mp3 and extension hangs up In-Reply-To: <4992E559.6060506@ewetel.de> References: <4992BFFC.50006@ewetel.de> <4992C801.4020906@ewetel.de> <191c3a030902110558i9702139g6b12aaf15f6d3aac@mail.gmail.com> <4992DCC2.3020702@ewetel.de> <191c3a030902110617n3dcd19cfr204c00792e1e87ed@mail.gmail.com> <4992E559.6060506@ewetel.de> Message-ID: <191c3a030902110729g30b30d46gac0f56205f8d48cf@mail.gmail.com> I am highly suspicious of the ubuntu. you are using a prerelease of gcc that we have already found at least 1 bug. we tried the file on our box and it doesn't even say anything about the file being bad etc...... it plays and hangs up fine even 4 times at once. It would be a big help if you could try to reproduce it on CentOS 5 as a comparison. We have had 3 cases this week where doing so has fixed problems and i don't want to believe it so I would appropriate it if you could test it. On Wed, Feb 11, 2009 at 8:48 AM, Helmut Kuper wrote: > Hi Anthony, > > I quickly have setup a test server with current trunk. So I can now > enter there a "YES" into that field. Current trunk crashed as well. But > thx for stretching the jira rules a bit :) > > I attached the file on jira in > http://jira.freeswitch.org/browse/MODFORM-24 > > Can you delete it asap because of copyright reasons, please? > > regards > helmut > > > > On 11.02.2009 15:17, Anthony Minessale wrote: > > if the alternative is to post it to the mailing list, you have our > > permission this one time to answer "not yet" so you have somewhere to > > attach the bad file so we can reproduce it. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090211/278ae62f/attachment-0002.html From helmut.kuper at ewetel.de Wed Feb 11 07:45:25 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Wed, 11 Feb 2009 16:45:25 +0100 Subject: [Freeswitch-users] mod_openzap stops working after some calls Update In-Reply-To: <4990789B.40405@ewetel.de> References: <498C2DC3.40701@ewetel.de> <191c3a030902060802r424f8a22tb4887dff7c6aead5@mail.gmail.com> <49907327.6010703@ewetel.de> <7CEFE4F8-7461-4DF3-ABE0-C0B818466AD0@jerris.com> <4990789B.40405@ewetel.de> Message-ID: <4992F295.4070809@ewetel.de> Hi Mike, I removed the apr dependency for timestamp and use the function openzap delivers. Works good so far. For unstabilities in openzap and Q931: On my side main problem seems to be, that channels for inbound traffic sometimes not be freed during runtime. Maybe our remote TDM end (AVAYA) simply doesn't release calls as it should, maybe openzap doesn't catch all q931 messages. I added a hack, which forces channels which are in TERMINATE or beyond *AND* this state is older than e.g. 500ms for inbound SETUP to DOWN. Openzap uses those down forced channels. I have the patch in production and I saw the first aid hack serveral times "InUse" channels freeing. "oz dump 1" shows channel states which hung in TERMINATE or above are down serveral minutes later :) This is just a quite brutal hack because it assumes that the remote TDM end is able to recover channels as well. regards helmut From helmut.kuper at ewetel.de Wed Feb 11 07:57:44 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Wed, 11 Feb 2009 16:57:44 +0100 Subject: [Freeswitch-users] segfault when shoutcast plays mp3 and extension hangs up In-Reply-To: <191c3a030902110729g30b30d46gac0f56205f8d48cf@mail.gmail.com> References: <4992BFFC.50006@ewetel.de> <4992C801.4020906@ewetel.de> <191c3a030902110558i9702139g6b12aaf15f6d3aac@mail.gmail.com> <4992DCC2.3020702@ewetel.de> <191c3a030902110617n3dcd19cfr204c00792e1e87ed@mail.gmail.com> <4992E559.6060506@ewetel.de> <191c3a030902110729g30b30d46gac0f56205f8d48cf@mail.gmail.com> Message-ID: <4992F578.6000401@ewetel.de> Hi Anthony, hm ... have to dig for a centos5 machine here ... I have one somewhere ... I will test it there as well. Concerning the prerelease of gcc ... My svn trunk FS was compiled by "gcc version 4.1.2 (Ubuntu 4.1.2-0ubuntu4)". In Jira I entered the gcc version of FS in production. regards Helmut On 11.02.2009 16:29, Anthony Minessale wrote: > I am highly suspicious of the ubuntu. > you are using a prerelease of gcc that we have already found at least > 1 bug. > > we tried the file on our box and it doesn't even say anything about > the file being bad etc...... it plays and hangs up fine even 4 times > at once. > It would be a big help if you could try to reproduce it on CentOS 5 as > a comparison. We have had 3 cases this week where doing so has fixed > problems and i don't want to believe it so I would appropriate it if > you could test it. > From saeedahmad1981 at gmail.com Wed Feb 11 08:31:37 2009 From: saeedahmad1981 at gmail.com (Saeed Ahmed) Date: Wed, 11 Feb 2009 17:31:37 +0100 Subject: [Freeswitch-users] FS + Call Center Solution Message-ID: <6309E7515E4F43159B9564800564B562@SaeedLaptop> Hi List, Is there any open source call center tool available which works with FS? Kind Regards Saeed Ahmed Tariq -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090211/579e5bf4/attachment-0002.html From odermann at googlemail.com Wed Feb 11 08:42:51 2009 From: odermann at googlemail.com (Dennis) Date: Wed, 11 Feb 2009 17:42:51 +0100 Subject: [Freeswitch-users] DTMF: Mute sound for the other side? In-Reply-To: <174C0607-ADBE-4AB2-92C3-588A86B74EAE@jerris.com> References: <5e414ed0902090659p18b970c9y8ba8f1bacda1f3ac@mail.gmail.com> <191c3a030902090725q11c8b3eaka6ee5e8565b1a7ea@mail.gmail.com> <5e414ed0902100658j2656c453mb74d4840bc256063@mail.gmail.com> <174C0607-ADBE-4AB2-92C3-588A86B74EAE@jerris.com> Message-ID: <5e414ed0902110842q49e0e454v6b9401b759fd04f0@mail.gmail.com> that is interesting. we are receiving the dtmf digits over 2833. might it be possible, that we receive 2833 AND inband (we asked our carrier for 2833, because we had problems with inband and fs - and we got it)? is there something we can setup in fs or is it a problem wich only our carrier can solve? 2009/2/10 Michael Jerris : > If your in a conference and your hearing other people hitting dtmf > digits that IS inband, it means that the place upstream that is doing > inband to 2833 conversion is not properly clipping the dtmf, this > probably needs to be fixed on that device. From brian at freeswitch.org Wed Feb 11 08:56:58 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 11 Feb 2009 10:56:58 -0600 Subject: [Freeswitch-users] DTMF: Mute sound for the other side? In-Reply-To: <5e414ed0902110842q49e0e454v6b9401b759fd04f0@mail.gmail.com> References: <5e414ed0902090659p18b970c9y8ba8f1bacda1f3ac@mail.gmail.com> <191c3a030902090725q11c8b3eaka6ee5e8565b1a7ea@mail.gmail.com> <5e414ed0902100658j2656c453mb74d4840bc256063@mail.gmail.com> <174C0607-ADBE-4AB2-92C3-588A86B74EAE@jerris.com> <5e414ed0902110842q49e0e454v6b9401b759fd04f0@mail.gmail.com> Message-ID: <26CE501F-8F06-4E7D-A9E7-24BD34E22DFE@freeswitch.org> Well if they are sending both they are broken. I would call and yell at them and beat them with a cluebat. /b On Feb 11, 2009, at 10:42 AM, Dennis wrote: > that is interesting. we are receiving the dtmf digits over 2833. might > it be possible, that we receive 2833 AND inband (we asked our carrier > for 2833, because we had problems with inband and fs - and we got it)? > > is there something we can setup in fs or is it a problem wich only our > carrier can solve? From odermann at googlemail.com Wed Feb 11 09:14:32 2009 From: odermann at googlemail.com (Dennis) Date: Wed, 11 Feb 2009 18:14:32 +0100 Subject: [Freeswitch-users] DTMF: Mute sound for the other side? In-Reply-To: <26CE501F-8F06-4E7D-A9E7-24BD34E22DFE@freeswitch.org> References: <5e414ed0902090659p18b970c9y8ba8f1bacda1f3ac@mail.gmail.com> <191c3a030902090725q11c8b3eaka6ee5e8565b1a7ea@mail.gmail.com> <5e414ed0902100658j2656c453mb74d4840bc256063@mail.gmail.com> <174C0607-ADBE-4AB2-92C3-588A86B74EAE@jerris.com> <5e414ed0902110842q49e0e454v6b9401b759fd04f0@mail.gmail.com> <26CE501F-8F06-4E7D-A9E7-24BD34E22DFE@freeswitch.org> Message-ID: <5e414ed0902110914u78e13547pbb52bfea588bbc29@mail.gmail.com> i can't tell, if they are sending both, but it seems so. we get 2833 for sure. they were kind enough to give it to us, because inband seems to be quite unreliable over sip. how can in find out, if both are coming and is there a way to "block" inband to test? perhaps we need both: if we bridge an inbound with another ivr on the outbound side, which is not sip and does not understand 2833, we need to pass inband through or something like this. or am i wrong with this? 2009/2/11 Brian West : > Well if they are sending both they are broken. I would call and yell > at them and beat them with a cluebat. > > /b > > On Feb 11, 2009, at 10:42 AM, Dennis wrote: > >> that is interesting. we are receiving the dtmf digits over 2833. might >> it be possible, that we receive 2833 AND inband (we asked our carrier >> for 2833, because we had problems with inband and fs - and we got it)? >> >> is there something we can setup in fs or is it a problem wich only our >> carrier can solve? > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From brian at freeswitch.org Wed Feb 11 09:23:35 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 11 Feb 2009 11:23:35 -0600 Subject: [Freeswitch-users] DTMF: Mute sound for the other side? In-Reply-To: <5e414ed0902110914u78e13547pbb52bfea588bbc29@mail.gmail.com> References: <5e414ed0902090659p18b970c9y8ba8f1bacda1f3ac@mail.gmail.com> <191c3a030902090725q11c8b3eaka6ee5e8565b1a7ea@mail.gmail.com> <5e414ed0902100658j2656c453mb74d4840bc256063@mail.gmail.com> <174C0607-ADBE-4AB2-92C3-588A86B74EAE@jerris.com> <5e414ed0902110842q49e0e454v6b9401b759fd04f0@mail.gmail.com> <26CE501F-8F06-4E7D-A9E7-24BD34E22DFE@freeswitch.org> <5e414ed0902110914u78e13547pbb52bfea588bbc29@mail.gmail.com> Message-ID: <32609C6A-EF32-48BE-9FA1-188BF4E05282@freeswitch.org> turn on the start_dtmf app and dial digits from the outside.. if you get duplicate digits then they are sending both. /b On Feb 11, 2009, at 11:14 AM, Dennis wrote: > i can't tell, if they are sending both, but it seems so. we get 2833 > for sure. they were kind enough to give it to us, because inband seems > to be quite unreliable over sip. > > how can in find out, if both are coming and is there a way to "block" > inband to test? > > perhaps we need both: if we bridge an inbound with another ivr on the > outbound side, which is not sip and does not understand 2833, we need > to pass inband through or something like this. or am i wrong with > this? From msc at freeswitch.org Wed Feb 11 09:43:24 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 11 Feb 2009 09:43:24 -0800 Subject: [Freeswitch-users] mod_openzap stops working after some calls Update In-Reply-To: <4992F295.4070809@ewetel.de> References: <498C2DC3.40701@ewetel.de> <191c3a030902060802r424f8a22tb4887dff7c6aead5@mail.gmail.com> <49907327.6010703@ewetel.de> <7CEFE4F8-7461-4DF3-ABE0-C0B818466AD0@jerris.com> <4990789B.40405@ewetel.de> <4992F295.4070809@ewetel.de> Message-ID: <87f2f3b90902110943q12c550e4qf8e6ecae1ecc6bec@mail.gmail.com> > This is just a quite brutal hack because it assumes that the remote TDM > end is able to recover channels as well. Could you possibly modify your hack to be less brutal? For example, could it send a STATUS ENQ to the far end for the channel in question? Just curious. -MC From anthony.minessale at gmail.com Wed Feb 11 10:05:31 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 11 Feb 2009 12:05:31 -0600 Subject: [Freeswitch-users] mod_openzap stops working after some calls Update In-Reply-To: <87f2f3b90902110943q12c550e4qf8e6ecae1ecc6bec@mail.gmail.com> References: <498C2DC3.40701@ewetel.de> <191c3a030902060802r424f8a22tb4887dff7c6aead5@mail.gmail.com> <49907327.6010703@ewetel.de> <7CEFE4F8-7461-4DF3-ABE0-C0B818466AD0@jerris.com> <4990789B.40405@ewetel.de> <4992F295.4070809@ewetel.de> <87f2f3b90902110943q12c550e4qf8e6ecae1ecc6bec@mail.gmail.com> Message-ID: <191c3a030902111005q661aa154tbcf1e33fb6082055@mail.gmail.com> you should come into irc on irc.freenode.net and join #openzap Stefan Knoblich (stkn) in the channel is doing some work on implementation actual q931 timers which would solve the problem the real way. Maybe you could collaberate with him. On Wed, Feb 11, 2009 at 11:43 AM, Michael Collins wrote: > > This is just a quite brutal hack because it assumes that the remote TDM > > end is able to recover channels as well. > > Could you possibly modify your hack to be less brutal? For example, > could it send a STATUS ENQ to the far end for the channel in question? > Just curious. > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090211/ca25e702/attachment-0002.html From odermann at googlemail.com Wed Feb 11 10:23:00 2009 From: odermann at googlemail.com (Dennis) Date: Wed, 11 Feb 2009 19:23:00 +0100 Subject: [Freeswitch-users] DTMF: Mute sound for the other side? In-Reply-To: <32609C6A-EF32-48BE-9FA1-188BF4E05282@freeswitch.org> References: <5e414ed0902090659p18b970c9y8ba8f1bacda1f3ac@mail.gmail.com> <191c3a030902090725q11c8b3eaka6ee5e8565b1a7ea@mail.gmail.com> <5e414ed0902100658j2656c453mb74d4840bc256063@mail.gmail.com> <174C0607-ADBE-4AB2-92C3-588A86B74EAE@jerris.com> <5e414ed0902110842q49e0e454v6b9401b759fd04f0@mail.gmail.com> <26CE501F-8F06-4E7D-A9E7-24BD34E22DFE@freeswitch.org> <5e414ed0902110914u78e13547pbb52bfea588bbc29@mail.gmail.com> <32609C6A-EF32-48BE-9FA1-188BF4E05282@freeswitch.org> Message-ID: <5e414ed0902111023q6353787ckf9a5af746b97a433@mail.gmail.com> ok, i will try this, but how can it be possible, that inband tones are audible in conference, when we do not even have start_dtmf activated? i just don't understand, why it must be dtmf inband, if the tones are audible and how they can be audible, if start_dtmf is not set. is it, because the carrier just sends them as normal sound, which is played as a tone, without beeing used for dtmf? 2009/2/11 Brian West : > turn on the start_dtmf app and dial digits from the outside.. if you > get duplicate digits then they are sending both. > > /b > > On Feb 11, 2009, at 11:14 AM, Dennis wrote: > >> i can't tell, if they are sending both, but it seems so. we get 2833 >> for sure. they were kind enough to give it to us, because inband seems >> to be quite unreliable over sip. >> >> how can in find out, if both are coming and is there a way to "block" >> inband to test? >> >> perhaps we need both: if we bridge an inbound with another ivr on the >> outbound side, which is not sip and does not understand 2833, we need >> to pass inband through or something like this. or am i wrong with >> this? > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From brian at freeswitch.org Wed Feb 11 10:36:11 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 11 Feb 2009 12:36:11 -0600 Subject: [Freeswitch-users] DTMF: Mute sound for the other side? In-Reply-To: <5e414ed0902111023q6353787ckf9a5af746b97a433@mail.gmail.com> References: <5e414ed0902090659p18b970c9y8ba8f1bacda1f3ac@mail.gmail.com> <191c3a030902090725q11c8b3eaka6ee5e8565b1a7ea@mail.gmail.com> <5e414ed0902100658j2656c453mb74d4840bc256063@mail.gmail.com> <174C0607-ADBE-4AB2-92C3-588A86B74EAE@jerris.com> <5e414ed0902110842q49e0e454v6b9401b759fd04f0@mail.gmail.com> <26CE501F-8F06-4E7D-A9E7-24BD34E22DFE@freeswitch.org> <5e414ed0902110914u78e13547pbb52bfea588bbc29@mail.gmail.com> <32609C6A-EF32-48BE-9FA1-188BF4E05282@freeswitch.org> <5e414ed0902111023q6353787ckf9a5af746b97a433@mail.gmail.com> Message-ID: <73433A3E-7641-4966-8FF1-73E6C62C5D60@freeswitch.org> On Feb 11, 2009, at 12:23 PM, Dennis wrote: > ok, i will try this, but how can it be possible, that inband tones are > audible in conference, when we do not even have start_dtmf activated? They aren't really sending 2833. > > > i just don't understand, why it must be dtmf inband, if the tones are > audible and how they can be audible, if start_dtmf is not set. > is it, because the carrier just sends them as normal sound, which is > played as a tone, without beeing used for dtmf? I bet they don't know how to config their switch to do 2833. /b From odermann at googlemail.com Wed Feb 11 10:37:51 2009 From: odermann at googlemail.com (Dennis) Date: Wed, 11 Feb 2009 19:37:51 +0100 Subject: [Freeswitch-users] Socket-Event on "originate call"? In-Reply-To: <5e414ed0902100716s2c053a48w2174f7587dd0641e@mail.gmail.com> References: <5e414ed0902090716y10f95f71o5760dfdf50d1c5cb@mail.gmail.com> <191c3a030902090727n44eef7d7x83704b66e628081@mail.gmail.com> <5e414ed0902100716s2c053a48w2174f7587dd0641e@mail.gmail.com> Message-ID: <5e414ed0902111037q2a6f4097sebd08cb712f042bf@mail.gmail.com> anthony, did you make the changes with "add {instant_ringback=true} to make ringback not wait for indication to generate ringback" for the described problem? we read something like this out of it, but we can not test it, because we get errors with the latest fs version (switch_ivr.c:674 switch_ivr_park() Cannot park channels that have no read codec.). 2009/2/10 Dennis : > yes, you are right. we are receiving the reply. > > but, we are using socket outbound and manage all calls over this > socket. we also measure the durations (like variable_duration and > variable_billsec) and count all outgoing calls over the socket. > but, if the originate (without an inbound call) will not start the > socket, we can not count up, how many calls failed because of "user > busy" or how long the platform was in use. > > a possible workarround: is it possible to trigger a dialplan over the > cli (like our default dialplan, which starts the socket), so that the > dialplan starts the originates? > > the basic problem for us, that, if we just want to make dialouts, we > are missing the inbound call to start the socket. > > > kind regards > dennis > > > > 2009/2/9 Anthony Minessale : >> when an originate is unsuccessful the failure and the cause code is returned >> as the reply to the originate request. >> >> >> On Mon, Feb 9, 2009 at 9:16 AM, Dennis wrote: >>> >>> hi, >>> >>> i am using socket outbound with fs. >>> >>> if i do an originate over the console, for starting an outbound call >>> without having an inbound call, and send the originate directly to the >>> socket, the socket is first started, if the call is in answer or >>> ringing state. >>> before this, i will not receive any event, because the socket was not >>> started. therefore i will not know, if the target is "busy" (hangup, >>> hangup cause: user busy). >>> >>> it would be very helpful, if the socket would start immediately on an >>> event like "channel originate". >>> >>> >>> thanks for the help >>> dennis > From brian at freeswitch.org Wed Feb 11 11:13:36 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 11 Feb 2009 13:13:36 -0600 Subject: [Freeswitch-users] Socket-Event on "originate call"? In-Reply-To: <5e414ed0902111037q2a6f4097sebd08cb712f042bf@mail.gmail.com> References: <5e414ed0902090716y10f95f71o5760dfdf50d1c5cb@mail.gmail.com> <191c3a030902090727n44eef7d7x83704b66e628081@mail.gmail.com> <5e414ed0902100716s2c053a48w2174f7587dd0641e@mail.gmail.com> <5e414ed0902111037q2a6f4097sebd08cb712f042bf@mail.gmail.com> Message-ID: <397EFED1-22E6-4B78-BDE4-3AE5F4204E5B@freeswitch.org> Try answer or pre_answer before park. /b On Feb 11, 2009, at 12:37 PM, Dennis wrote: > anthony, did you make the changes with "add {instant_ringback=true} to > make ringback not wait for indication to generate ringback" for the > described problem? > > we read something like this out of it, but we can not test it, because > we get errors with the latest fs version (switch_ivr.c:674 > switch_ivr_park() Cannot park channels that have no read codec.). From nik.middleton at noblesolutions.co.uk Wed Feb 11 11:15:34 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Wed, 11 Feb 2009 19:15:34 -0000 Subject: [Freeswitch-users] FS 1.0.2 Crash and burn Message-ID: I have a situation where FS aborts I'm running an lua script with mysql statements First time it runs, on hangup I get [CONSOLE] switch_core_memory.c:374 switch_core_memory_reclaim() Returning 4 recycled memory pool(s) If I run it again, FS exits. Should there be an error log somewhere that explains why FS dies? Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090211/8a185422/attachment-0002.html From brian at freeswitch.org Wed Feb 11 11:17:40 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 11 Feb 2009 13:17:40 -0600 Subject: [Freeswitch-users] FS 1.0.2 Crash and burn In-Reply-To: References: Message-ID: <7EA294D9-389F-4543-8F24-ED4C08600BE7@freeswitch.org> Can you show us what you're doing? /b On Feb 11, 2009, at 1:15 PM, Nik Middleton wrote: > I have a situation where FS aborts > > I?m running an lua script with mysql statements > > First time it runs, on hangup I get > > [CONSOLE] switch_core_memory.c:374 switch_core_memory_reclaim() > Returning 4 recycled memory pool(s) > > If I run it again, FS exits. > > Should there be an error log somewhere that explains why FS dies? > > Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090211/5b672908/attachment-0002.html From nik.middleton at noblesolutions.co.uk Wed Feb 11 11:35:25 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Wed, 11 Feb 2009 19:35:25 -0000 Subject: [Freeswitch-users] FS 1.0.2 Crash and burn In-Reply-To: <7EA294D9-389F-4543-8F24-ED4C08600BE7@freeswitch.org> References: <7EA294D9-389F-4543-8F24-ED4C08600BE7@freeswitch.org> Message-ID: I was running in a screen session, so going back to the console it shows it's a seg fault 2009-02-11 19:27:53 [NOTICE] sofia.c:3090 sofia_handle_sip_i_state() Hangup sofia/internal/1001 at 192.168.3.206 [CS_EXECUTE] [NORMAL_CLEARING] Segmentation fault (core dumped) Seg fault occurs on hangup What seems to be causing the problem is an insert statement. Note I'm using the protected call function to trap on any sql error (script will abort on error otherwise) but even calling it unprotected, the result is the same. function updatecall() query = "INSERT INTO CONTACT phonenum, group values 0771111111111, " .. CALLER ; freeswitch.consoleLog("info", query.."\n"); res = assert (con:execute(query)); if unexpected_condition then error() end end if type == "dtmf" and obj['digit'] == '9' then CALL_STATUS = "ORDER"; pcall(updateDNC); session:streamFile("wait48.wav"); return "break"; end function myHangupHook(s, status, arg) freeswitch.consoleLog("info", " : They hung up on US!!!\n"); con:close() env:close() end ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: 11 February 2009 19:18 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] FS 1.0.2 Crash and burn Can you show us what you're doing? /b On Feb 11, 2009, at 1:15 PM, Nik Middleton wrote: I have a situation where FS aborts I'm running an lua script with mysql statements First time it runs, on hangup I get [CONSOLE] switch_core_memory.c:374 switch_core_memory_reclaim() Returning 4 recycled memory pool(s) If I run it again, FS exits. Should there be an error log somewhere that explains why FS dies? Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090211/ceb3bc31/attachment-0002.html From pbd at suspiria.net Wed Feb 11 11:49:29 2009 From: pbd at suspiria.net (Public Dump) Date: Wed, 11 Feb 2009 20:49:29 +0100 Subject: [Freeswitch-users] Compile Freeswitch 64bit for Windows Message-ID: <13C421883438EB42B9E2C30069FD4AB767BF49AF0A@crushinator.central.local> ... did anybody succeed with this ? The solution for VS2008 does not seem to have a valid 64bit configuration. Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090211/45a1310e/attachment-0002.html From brian at freeswitch.org Wed Feb 11 11:38:14 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 11 Feb 2009 13:38:14 -0600 Subject: [Freeswitch-users] FS 1.0.2 Crash and burn In-Reply-To: References: <7EA294D9-389F-4543-8F24-ED4C08600BE7@freeswitch.org> Message-ID: <75A42DD8-9BA0-46CC-8121-04CE9DB27A27@freeswitch.org> How about getting a backtrace of the core dump and opening a jira? http://wiki.freeswitch.org/wiki/Reporting_Bugs /b On Feb 11, 2009, at 1:35 PM, Nik Middleton wrote: > I was running in a screen session, so going back to the console it > shows it?s a seg fault > > 2009-02-11 19:27:53 [NOTICE] sofia.c:3090 sofia_handle_sip_i_state() > Hangup sofia/internal/1001 at 192.168.3.206 [CS_EXECUTE] > [NORMAL_CLEARING] > Segmentation fault (core dumped) > > Seg fault occurs on hangup > > What seems to be causing the problem is an insert statement. > > Note I?m using the protected call function to trap on any sql error > (script will abort on error otherwise) but even calling it > unprotected, the result is the same. > > function updatecall() > query = "INSERT INTO CONTACT phonenum, group values > 0771111111111, " .. CALLER ; > freeswitch.consoleLog("info", query.."\n"); > res = assert (con:execute(query)); > if unexpected_condition then error() end > > end > > > if type == "dtmf" and obj['digit'] == '9' then > CALL_STATUS = "ORDER"; > pcall(updateDNC); > session:streamFile("wait48.wav"); > return "break"; > end > > > function myHangupHook(s, status, arg) > freeswitch.consoleLog("info", " : They hung up on US!!! > \n"); > con:close() > env:close() > > end -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090211/8450db85/attachment-0002.html From mike at jerris.com Wed Feb 11 12:07:55 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 11 Feb 2009 15:07:55 -0500 Subject: [Freeswitch-users] Compile Freeswitch 64bit for Windows In-Reply-To: <13C421883438EB42B9E2C30069FD4AB767BF49AF0A@crushinator.central.local> References: <13C421883438EB42B9E2C30069FD4AB767BF49AF0A@crushinator.central.local> Message-ID: I don't have a 64 bit windows box/os to get this working. Someone with access to such a box would have to set this up and submit a patch. Mike On Feb 11, 2009, at 2:49 PM, Public Dump wrote: > ? did anybody succeed with this ? The solution for VS2008 does not > seem to have a valid 64bit configuration. > > Regards > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090211/54deb044/attachment-0002.html From odermann at googlemail.com Wed Feb 11 12:11:11 2009 From: odermann at googlemail.com (Dennis) Date: Wed, 11 Feb 2009 21:11:11 +0100 Subject: [Freeswitch-users] Socket-Event on "originate call"? In-Reply-To: <397EFED1-22E6-4B78-BDE4-3AE5F4204E5B@freeswitch.org> References: <5e414ed0902090716y10f95f71o5760dfdf50d1c5cb@mail.gmail.com> <191c3a030902090727n44eef7d7x83704b66e628081@mail.gmail.com> <5e414ed0902100716s2c053a48w2174f7587dd0641e@mail.gmail.com> <5e414ed0902111037q2a6f4097sebd08cb712f042bf@mail.gmail.com> <397EFED1-22E6-4B78-BDE4-3AE5F4204E5B@freeswitch.org> Message-ID: <5e414ed0902111211y3456d5eaqc99baa126cb9dcad@mail.gmail.com> this does not help. we are using socket outbound and everything worked before the changes yesterday. we have the same error with other dialplans. > 2009/2/11 Brian West : > Try answer or pre_answer before park. > > /b > > On Feb 11, 2009, at 12:37 PM, Dennis wrote: > >> anthony, did you make the changes with "add {instant_ringback=true} to >> make ringback not wait for indication to generate ringback" for the >> described problem? >> >> we read something like this out of it, but we can not test it, because >> we get errors with the latest fs version (switch_ivr.c:674 >> switch_ivr_park() Cannot park channels that have no read codec.). From brian at freeswitch.org Wed Feb 11 12:17:30 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 11 Feb 2009 14:17:30 -0600 Subject: [Freeswitch-users] Socket-Event on "originate call"? In-Reply-To: <5e414ed0902111211y3456d5eaqc99baa126cb9dcad@mail.gmail.com> References: <5e414ed0902090716y10f95f71o5760dfdf50d1c5cb@mail.gmail.com> <191c3a030902090727n44eef7d7x83704b66e628081@mail.gmail.com> <5e414ed0902100716s2c053a48w2174f7587dd0641e@mail.gmail.com> <5e414ed0902111037q2a6f4097sebd08cb712f042bf@mail.gmail.com> <397EFED1-22E6-4B78-BDE4-3AE5F4204E5B@freeswitch.org> <5e414ed0902111211y3456d5eaqc99baa126cb9dcad@mail.gmail.com> Message-ID: Please collect the backtrace and report it on Jira. /b On Feb 11, 2009, at 2:11 PM, Dennis wrote: > this does not help. we are using socket outbound and everything worked > before the changes yesterday. > > we have the same error with other dialplans. From nik.middleton at noblesolutions.co.uk Wed Feb 11 12:20:08 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Wed, 11 Feb 2009 20:20:08 -0000 Subject: [Freeswitch-users] FS 1.0.2 Crash and burn In-Reply-To: <75A42DD8-9BA0-46CC-8121-04CE9DB27A27@freeswitch.org> References: <7EA294D9-389F-4543-8F24-ED4C08600BE7@freeswitch.org> <75A42DD8-9BA0-46CC-8121-04CE9DB27A27@freeswitch.org> Message-ID: Where is the core dump written? ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: 11 February 2009 19:38 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] FS 1.0.2 Crash and burn How about getting a backtrace of the core dump and opening a jira? http://wiki.freeswitch.org/wiki/Reporting_Bugs /b On Feb 11, 2009, at 1:35 PM, Nik Middleton wrote: I was running in a screen session, so going back to the console it shows it's a seg fault 2009-02-11 19:27:53 [NOTICE] sofia.c:3090 sofia_handle_sip_i_state() Hangup sofia/internal/1001 at 192.168.3.206 [CS_EXECUTE] [NORMAL_CLEARING] Segmentation fault (core dumped) Seg fault occurs on hangup What seems to be causing the problem is an insert statement. Note I'm using the protected call function to trap on any sql error (script will abort on error otherwise) but even calling it unprotected, the result is the same. function updatecall() query = "INSERT INTO CONTACT phonenum, group values 0771111111111, " .. CALLER ; freeswitch.consoleLog("info", query.."\n"); res = assert (con:execute(query)); if unexpected_condition then error() end end if type == "dtmf" and obj['digit'] == '9' then CALL_STATUS = "ORDER"; pcall(updateDNC); session:streamFile("wait48.wav"); return "break"; end function myHangupHook(s, status, arg) freeswitch.consoleLog("info", " : They hung up on US!!!\n"); con:close() env:close() end -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090211/a52bc528/attachment-0002.html From nik.middleton at noblesolutions.co.uk Wed Feb 11 12:22:01 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Wed, 11 Feb 2009 20:22:01 -0000 Subject: [Freeswitch-users] FS 1.0.2 Crash and burn In-Reply-To: <75A42DD8-9BA0-46CC-8121-04CE9DB27A27@freeswitch.org> References: <7EA294D9-389F-4543-8F24-ED4C08600BE7@freeswitch.org> <75A42DD8-9BA0-46CC-8121-04CE9DB27A27@freeswitch.org> Message-ID: Forget my last, followed the link Regards, ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: 11 February 2009 19:38 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] FS 1.0.2 Crash and burn How about getting a backtrace of the core dump and opening a jira? http://wiki.freeswitch.org/wiki/Reporting_Bugs /b On Feb 11, 2009, at 1:35 PM, Nik Middleton wrote: I was running in a screen session, so going back to the console it shows it's a seg fault 2009-02-11 19:27:53 [NOTICE] sofia.c:3090 sofia_handle_sip_i_state() Hangup sofia/internal/1001 at 192.168.3.206 [CS_EXECUTE] [NORMAL_CLEARING] Segmentation fault (core dumped) Seg fault occurs on hangup What seems to be causing the problem is an insert statement. Note I'm using the protected call function to trap on any sql error (script will abort on error otherwise) but even calling it unprotected, the result is the same. function updatecall() query = "INSERT INTO CONTACT phonenum, group values 0771111111111, " .. CALLER ; freeswitch.consoleLog("info", query.."\n"); res = assert (con:execute(query)); if unexpected_condition then error() end end if type == "dtmf" and obj['digit'] == '9' then CALL_STATUS = "ORDER"; pcall(updateDNC); session:streamFile("wait48.wav"); return "break"; end function myHangupHook(s, status, arg) freeswitch.consoleLog("info", " : They hung up on US!!!\n"); con:close() env:close() end -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090211/6678be19/attachment-0002.html From brian at freeswitch.org Wed Feb 11 12:23:06 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 11 Feb 2009 14:23:06 -0600 Subject: [Freeswitch-users] FS 1.0.2 Crash and burn In-Reply-To: References: <7EA294D9-389F-4543-8F24-ED4C08600BE7@freeswitch.org> <75A42DD8-9BA0-46CC-8121-04CE9DB27A27@freeswitch.org> Message-ID: Try starting it from your /usr/local/freeswitch/bin... ./freeswitch it'll dump in the same folder. /b On Feb 11, 2009, at 2:20 PM, Nik Middleton wrote: > Where is the core dump written? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090211/52f62146/attachment-0002.html From celam at arn.com Wed Feb 11 12:32:08 2009 From: celam at arn.com (Chris Elam) Date: Wed, 11 Feb 2009 15:32:08 -0500 Subject: [Freeswitch-users] Originate call from one ext to another from php? Message-ID: Hi all, I?m just starting playing around with FS and I?ve searched for the answer to what I think is an easy question but I can?t find it. I have FS running, 2 X-lite clients on 2 different computers connected using the preconfigured 1000 and 1001 extenstions. Both can call each other and everything is fine. I?m trying to figure out though how to originate the call from 1000 to 1001 via php. I?m using this script: http://wiki.freeswitch.org/wiki/PHP_Event_Socket Except that I?ve changed this line: $cmd = "api help"; To: $cmd = "api originate sofia/mydomain.com/1000 at 192.168.15.50 &bridge(sofia/mydomain.com/1001 at 192.168.15.50)"; The result I get is : -ERR DESTINATION_OUT_OF_ORDER PS, I literally have ?mydomain.com? in there as it looks like from the wiki this is the default. Any help is much appreciated, thanks all. This email may contain confidential information and is solely for the use of the intended recipient. Any review, distribution, disclosure or other use of this information by anyone other than the intended recipient is prohibited. If you have received this communication in error, please notify the sender immediately and delete this message from your system. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090211/72aae965/attachment-0002.html From brian at freeswitch.org Wed Feb 11 12:48:25 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 11 Feb 2009 14:48:25 -0600 Subject: [Freeswitch-users] Originate call from one ext to another from php? In-Reply-To: References: Message-ID: show me "sofia status", Try changing the @ to a % but I really need to see the sofia status output. /b On Feb 11, 2009, at 2:32 PM, Chris Elam wrote: > $cmd = "api originate sofia/mydomain.com/1000 at 192.168.15.50 &bridge(sofia/mydomain.com/1001 at 192.168.15.50 > )"; > > The result I get is : -ERR DESTINATION_OUT_OF_ORDER -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090211/680faaee/attachment-0002.html From celam at arn.com Wed Feb 11 12:54:15 2009 From: celam at arn.com (Chris Elam) Date: Wed, 11 Feb 2009 15:54:15 -0500 Subject: [Freeswitch-users] Originate call from one ext to another from php? In-Reply-To: Message-ID: The % gives the same error. Here is the sofia status output: API CALL [sofia(status)] output: Name Type Data State ============================================================================ ===================== external profile sip:mod_sofia at myoutsideip:5080 RUNNING (0) example.com gateway sip:joeuser at example.com NOREG internal profile sip:mod_sofia at myinsideip:5060 RUNNING (0) myinsideip alias internal ALIASED internal-ipv6 profile sip:mod_sofia@[::1]:5060 RUNNING (0) default alias internal ALIASED nat alias external ALIASED outbound alias external ALIASED ============================================================================ ===================== 3 profiles 4 aliases On 2/11/09 3:48 PM, "Brian West" wrote: > show me "sofia status", Try changing the @ to a % but I really need to see > the sofia status output. > > /b > > On Feb 11, 2009, at 2:32 PM, Chris Elam wrote: > >> $cmd = "api originate sofia/mydomain.com/1000 at 192.168.15.50 >> &bridge(sofia/mydomain.com/1001 at 192.168.15.50)"; >> >> The result I get is : -ERR DESTINATION_OUT_OF_ORDER >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org This email may contain confidential information and is solely for the use of the intended recipient. Any review, distribution, disclosure or other use of this information by anyone other than the intended recipient is prohibited. If you have received this communication in error, please notify the sender immediately and delete this message from your system. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090211/9d4af77f/attachment-0002.html From brian at freeswitch.org Wed Feb 11 12:59:17 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 11 Feb 2009 14:59:17 -0600 Subject: [Freeswitch-users] Originate call from one ext to another from php? In-Reply-To: References: Message-ID: try sofia/myinsideip/1000 and sofia/myinsideip/1001 I sure hope it doesn't say myinsideip on there and you only tried to hide your IP. /b On Feb 11, 2009, at 2:54 PM, Chris Elam wrote: > The % gives the same error. Here is the sofia status output: > > API CALL [sofia(status)] output: > Name > Type Data State > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > ====================================================================== > external profile sip:mod_sofia at myoutsideip:5080 > RUNNING (0) > example.com gateway sip:joeuser at example.com > NOREG > internal profile sip:mod_sofia at myinsideip:5060 > RUNNING (0) > myinsideip alias > internal ALIASED > internal-ipv6 profile sip:mod_sofia@[:: > 1]:5060 RUNNING (0) > default alias > internal ALIASED > nat alias > external ALIASED > outbound alias > external ALIASED > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > = > ====================================================================== > 3 profiles 4 aliases From celam at arn.com Wed Feb 11 13:06:04 2009 From: celam at arn.com (Chris Elam) Date: Wed, 11 Feb 2009 16:06:04 -0500 Subject: [Freeswitch-users] Originate call from one ext to another from php? In-Reply-To: Message-ID: That's it, worked perfectly, thanks a bunch! On 2/11/09 3:59 PM, "Brian West" wrote: > try sofia/myinsideip/1000 and sofia/myinsideip/1001 > > I sure hope it doesn't say myinsideip on there and you only tried to > hide your IP. > > /b > > > On Feb 11, 2009, at 2:54 PM, Chris Elam wrote: > >> The % gives the same error. Here is the sofia status output: >> >> API CALL [sofia(status)] output: >> Name >> Type Data State >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> ====================================================================== >> external profile sip:mod_sofia at myoutsideip:5080 >> RUNNING (0) >> example.com gateway sip:joeuser at example.com >> NOREG >> internal profile sip:mod_sofia at myinsideip:5060 >> RUNNING (0) >> myinsideip alias >> internal ALIASED >> internal-ipv6 profile sip:mod_sofia@[:: >> 1]:5060 RUNNING (0) >> default alias >> internal ALIASED >> nat alias >> external ALIASED >> outbound alias >> external ALIASED >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> ====================================================================== >> 3 profiles 4 aliases > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org This email may contain confidential information and is solely for the use of the intended recipient. Any review, distribution, disclosure or other use of this information by anyone other than the intended recipient is prohibited. If you have received this communication in error, please notify the sender immediately and delete this message from your system. From pbd at suspiria.net Wed Feb 11 13:11:11 2009 From: pbd at suspiria.net (Public Dump) Date: Wed, 11 Feb 2009 22:11:11 +0100 Subject: [Freeswitch-users] High CPU load after starting Message-ID: <13C421883438EB42B9E2C30069FD4AB767BF49AF0C@crushinator.central.local> After reading you suggestions I deployed the version from SVN today, the problem persists. Regards Von: Public Dump Gesendet: Dienstag, 10. Februar 2009 19:42 An: 'freeswitch-users at lists.freeswitch.org' Betreff: High CPU load after starting After starting FreeSwitch (1.0.2) on a 4 core server running Windows Server 2008, the CPU load (privileged time/kernel) for one of the cores goes to 50% and stays there. Stoping FreeSwitch stops the load. I have tried to disable all modules but the problem persists. Has anybody seen this problem, can it be fixed ? regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090211/7ef613df/attachment-0002.html From brian at freeswitch.org Wed Feb 11 13:13:32 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 11 Feb 2009 15:13:32 -0600 Subject: [Freeswitch-users] Originate call from one ext to another from php? In-Reply-To: References: Message-ID: remember its sofia/profilename/user%domain or sofia/domain/user the latter requires an alias on the profile for the domain the user registers with. /b On Feb 11, 2009, at 3:06 PM, Chris Elam wrote: > That's it, worked perfectly, thanks a bunch! > > > On 2/11/09 3:59 PM, "Brian West" wrote: > >> try sofia/myinsideip/1000 and sofia/myinsideip/1001 >> >> I sure hope it doesn't say myinsideip on there and you only tried to >> hide your IP. >> >> /b > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090211/97380167/attachment-0002.html From brian at freeswitch.org Wed Feb 11 13:16:16 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 11 Feb 2009 15:16:16 -0600 Subject: [Freeswitch-users] High CPU load after starting In-Reply-To: <13C421883438EB42B9E2C30069FD4AB767BF49AF0C@crushinator.central.local> References: <13C421883438EB42B9E2C30069FD4AB767BF49AF0C@crushinator.central.local> Message-ID: <1A18CBF4-81F9-4E57-AF9D-54B6103B292B@freeswitch.org> Are you sure you rebuilt it clean? Are you doing anything special? Changing any configs? /b On Feb 11, 2009, at 3:11 PM, Public Dump wrote: > After reading you suggestions I deployed the version from SVN today, > the problem persists. > > Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090211/744b3a0a/attachment-0002.html From pbd at suspiria.net Wed Feb 11 13:32:10 2009 From: pbd at suspiria.net (Public Dump) Date: Wed, 11 Feb 2009 22:32:10 +0100 Subject: [Freeswitch-users] Compile Freeswitch 64bit for Windows Message-ID: <13C421883438EB42B9E2C30069FD4AB767BF49AF0D@crushinator.central.local> For compiling x64 code you shouldn't need a 64bit system, but you couldn't run it of course. I don't have a 64 bit windows box/os to get this working. Someone with access to such a box would have to set this up and submit a patch. Mike -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090211/83da0d55/attachment-0002.html From anthony.minessale at gmail.com Wed Feb 11 13:40:07 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 11 Feb 2009 15:40:07 -0600 Subject: [Freeswitch-users] FS 1.0.2 Crash and burn In-Reply-To: References: <7EA294D9-389F-4543-8F24-ED4C08600BE7@freeswitch.org> <75A42DD8-9BA0-46CC-8121-04CE9DB27A27@freeswitch.org> Message-ID: <191c3a030902111340s2a9de96cg2a1cfa8cfddad6b0@mail.gmail.com> and make sure it's svn trunk or at least a daily snapshot and not 1.0.2 On Wed, Feb 11, 2009 at 2:23 PM, Brian West wrote: > Try starting it from your /usr/local/freeswitch/bin... ./freeswitch it'll > dump in the same folder. > /b > > On Feb 11, 2009, at 2:20 PM, Nik Middleton wrote: > > Where is the core dump written? > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090211/e5efa5a8/attachment-0002.html From jacredit at gmail.com Wed Feb 11 14:12:24 2009 From: jacredit at gmail.com (John Hyde) Date: Wed, 11 Feb 2009 14:12:24 -0800 Subject: [Freeswitch-users] does anyone have a working FS / aastra config In-Reply-To: <777d76f40902042206u7c448985i8690c6df4472f3b7@mail.gmail.com> References: <777d76f40902042206u7c448985i8690c6df4472f3b7@mail.gmail.com> Message-ID: <777d76f40902111412t38280ecdjf86f8a761d381414@mail.gmail.com> Figured out the phone was sending packets that were too large, and the receiving system was not reassembling the fragmented packet. This can be fixed on the Aastra by enabling basic codecs: Go to the phone web-UI -- global SIP -- Codec Preference List -- Codec 1 -- change all to basic, save settings and restart the phone. Or in cfg files for aastra set: sip use basic codecs: 1 regards- John On Wed, Feb 4, 2009 at 10:06 PM, John Hyde wrote: > I am having problems getting an Aastra 57i to make calls through FS. the > phone registers fine, but all calls fail. If i use xlite or a nokia sip > phone, i have no problems. > > Here is a packet capture of an attempted call: > > http://pastebin.freeswitch.org/7039 > > notice packet 9, it should have been a SIP INVITE, but it turned out to be > a Fragmented IP protocol > > The phone and FS are both on the same lan subnet, and the phone connects > fine with an asterisk server on the same subnet. > > Is there a known config for aastra phones that I can reference, or does > anyone know why I am having this issue? > > -- john > -- - j -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090211/b760d9d0/attachment-0002.html From pbd at suspiria.net Wed Feb 11 14:21:23 2009 From: pbd at suspiria.net (Public Dump) Date: Wed, 11 Feb 2009 23:21:23 +0100 Subject: [Freeswitch-users] Freeswitch-users Digest, Vol 32, Issue 98 In-Reply-To: References: Message-ID: <13C421883438EB42B9E2C30069FD4AB767BF49AF0E@crushinator.central.local> > Message: 3 > Date: Wed, 11 Feb 2009 15:16:16 -0600 > From: Brian West > Subject: Re: [Freeswitch-users] High CPU load after starting > To: freeswitch-users at lists.freeswitch.org > Message-ID: <1A18CBF4-81F9-4E57-AF9D-54B6103B292B at freeswitch.org> > Content-Type: text/plain; charset="us-ascii" Downloaded tarball, performed SVN update. Nothing special, just building it right out of the box. Already tired disabling all modules and loading all cores. Problem persists. What is really striking, is that the load is all privileged time (kernel), consumes exactly 50% of one core and the core in question is always the same one. The Server is running Hyper-V (Host OS) but this also means it is running under the control of the hypervisor. > Are you sure you rebuilt it clean? Are you doing anything special? > Changing any configs? > > /b > From anthony.minessale at gmail.com Wed Feb 11 14:25:37 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 11 Feb 2009 16:25:37 -0600 Subject: [Freeswitch-users] Freeswitch-users Digest, Vol 32, Issue 98 In-Reply-To: <13C421883438EB42B9E2C30069FD4AB767BF49AF0E@crushinator.central.local> References: <13C421883438EB42B9E2C30069FD4AB767BF49AF0E@crushinator.central.local> Message-ID: <191c3a030902111425p158d29b5gf68d2993d39d5e40@mail.gmail.com> it was a completely clean build? as in compleletely new and/or clean solution? Which tarball was it ? On Wed, Feb 11, 2009 at 4:21 PM, Public Dump wrote: > > Message: 3 > > Date: Wed, 11 Feb 2009 15:16:16 -0600 > > From: Brian West > > Subject: Re: [Freeswitch-users] High CPU load after starting > > To: freeswitch-users at lists.freeswitch.org > > Message-ID: <1A18CBF4-81F9-4E57-AF9D-54B6103B292B at freeswitch.org> > > Content-Type: text/plain; charset="us-ascii" > > Downloaded tarball, performed SVN update. > Nothing special, just building it right out of the box. Already tired > disabling all modules and loading all cores. > Problem persists. > > What is really striking, is that the load is all privileged time (kernel), > consumes exactly 50% of one core and the core in question is always the same > one. > > The Server is running Hyper-V (Host OS) but this also means it is running > under the control of the hypervisor. > > > Are you sure you rebuilt it clean? Are you doing anything special? > > Changing any configs? > > > > /b > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090211/8df90bb6/attachment-0002.html From freeswitch at servercorps.com Wed Feb 11 14:34:47 2009 From: freeswitch at servercorps.com (Nik Martin) Date: Wed, 11 Feb 2009 16:34:47 -0600 Subject: [Freeswitch-users] FreeSWITCH VPSs Message-ID: <92e7d2090902111434l60afd3a8i634b4df688ed4011@mail.gmail.com> If any one needs a FreeSWITCH box with a public, static IP, I can provide them for you at a reasonable cost. I'm building a Virtualization platform for FreeSWITCH hosting, and have the first node complete. These are OpenVZ Virtual Engines with Centos 5.2, a full build environment, and the latest FreeSWITCH trunk. You get 1 static IP, no NAT, 256 mb ram, and 8 gb disk space, with 1 Megabit of bandwidth. Great for VOIP service providers, backup switch, testing, etc. You can contact me directly if you are interested. Nik From brian at freeswitch.org Wed Feb 11 14:38:31 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 11 Feb 2009 16:38:31 -0600 Subject: [Freeswitch-users] FreeSWITCH VPSs In-Reply-To: <92e7d2090902111434l60afd3a8i634b4df688ed4011@mail.gmail.com> References: <92e7d2090902111434l60afd3a8i634b4df688ed4011@mail.gmail.com> Message-ID: <40CC1F17-A4C3-4CA6-8DA0-C8B2A5346CCD@freeswitch.org> Quick note make sure you're 100% 64 bit.. if you need help with that I can show you how on CentOS 5.2 /b On Feb 11, 2009, at 4:34 PM, Nik Martin wrote: > If any one needs a FreeSWITCH box with a public, static IP, I can > provide them for you at a reasonable cost. I'm building a > Virtualization platform for FreeSWITCH hosting, and have the first > node complete. These are OpenVZ Virtual Engines with Centos 5.2, a > full build environment, and the latest FreeSWITCH trunk. You get 1 > static IP, no NAT, 256 mb ram, and 8 gb disk space, with 1 Megabit of > bandwidth. Great for VOIP service providers, backup switch, testing, > etc. You can contact me directly if you are interested. > > Nik From freeswitch at servercorps.com Wed Feb 11 14:47:23 2009 From: freeswitch at servercorps.com (Nik Martin) Date: Wed, 11 Feb 2009 16:47:23 -0600 Subject: [Freeswitch-users] FreeSWITCH VPSs In-Reply-To: <40CC1F17-A4C3-4CA6-8DA0-C8B2A5346CCD@freeswitch.org> References: <92e7d2090902111434l60afd3a8i634b4df688ed4011@mail.gmail.com> <40CC1F17-A4C3-4CA6-8DA0-C8B2A5346CCD@freeswitch.org> Message-ID: <92e7d2090902111447v1cbb00c0l8fc9f1c8b0c1a514@mail.gmail.com> On Wed, Feb 11, 2009 at 4:38 PM, Brian West wrote: > Quick note make sure you're 100% 64 bit.. if you need help with that I > can show you how on CentOS 5.2 > My hardware Node is running 64 bit Centos 5.2, with OpenVZ's kernel: 2.6.18-92.1.18.el5.028stab060.2 #1 SMP Tue Jan 13 11:38:36 MSK 2009 x86_64 x86_64 x86_64 GNU/Linux I think the VE I've built is too, but uname is a bit cryptic: 2.6.18-92.1.18.el5.028stab060.2 #1 SMP Tue Jan 13 11:38:36 MSK 2009 i686 i686 i386 GNU/Linux I can easily change it if FS will run better. Nik > /b > > On Feb 11, 2009, at 4:34 PM, Nik Martin wrote: > >> If any one needs a FreeSWITCH box with a public, static IP, I can >> provide them for you at a reasonable cost. I'm building a >> Virtualization platform for FreeSWITCH hosting, and have the first >> node complete. These are OpenVZ Virtual Engines with Centos 5.2, a >> full build environment, and the latest FreeSWITCH trunk. You get 1 >> static IP, no NAT, 256 mb ram, and 8 gb disk space, with 1 Megabit of >> bandwidth. Great for VOIP service providers, backup switch, testing, >> etc. You can contact me directly if you are interested. >> >> Nik > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From krice at freeswitch.org Wed Feb 11 14:50:23 2009 From: krice at freeswitch.org (Ken Rice) Date: Wed, 11 Feb 2009 16:50:23 -0600 Subject: [Freeswitch-users] FreeSWITCH VPSs In-Reply-To: <92e7d2090902111447v1cbb00c0l8fc9f1c8b0c1a514@mail.gmail.com> Message-ID: Be sure to make the Virt nodes 64bit too... FS works 100% better w/ 64bit! > From: Nik Martin > Reply-To: > Date: Wed, 11 Feb 2009 16:47:23 -0600 > To: > Subject: Re: [Freeswitch-users] FreeSWITCH VPSs > > On Wed, Feb 11, 2009 at 4:38 PM, Brian West wrote: >> Quick note make sure you're 100% 64 bit.. if you need help with that I >> can show you how on CentOS 5.2 >> > > My hardware Node is running 64 bit Centos 5.2, with OpenVZ's kernel: > 2.6.18-92.1.18.el5.028stab060.2 #1 SMP Tue Jan 13 11:38:36 MSK 2009 > x86_64 x86_64 x86_64 GNU/Linux > > I think the VE I've built is too, but uname is a bit cryptic: > 2.6.18-92.1.18.el5.028stab060.2 #1 SMP Tue Jan 13 11:38:36 MSK 2009 > i686 i686 i386 GNU/Linux > > I can easily change it if FS will run better. > > Nik > > > > >> /b >> >> On Feb 11, 2009, at 4:34 PM, Nik Martin wrote: >> >>> If any one needs a FreeSWITCH box with a public, static IP, I can >>> provide them for you at a reasonable cost. I'm building a >>> Virtualization platform for FreeSWITCH hosting, and have the first >>> node complete. These are OpenVZ Virtual Engines with Centos 5.2, a >>> full build environment, and the latest FreeSWITCH trunk. You get 1 >>> static IP, no NAT, 256 mb ram, and 8 gb disk space, with 1 Megabit of >>> bandwidth. Great for VOIP service providers, backup switch, testing, >>> etc. You can contact me directly if you are interested. >>> >>> Nik >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Wed Feb 11 14:54:50 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 11 Feb 2009 16:54:50 -0600 Subject: [Freeswitch-users] FreeSWITCH VPSs In-Reply-To: <92e7d2090902111447v1cbb00c0l8fc9f1c8b0c1a514@mail.gmail.com> References: <92e7d2090902111434l60afd3a8i634b4df688ed4011@mail.gmail.com> <40CC1F17-A4C3-4CA6-8DA0-C8B2A5346CCD@freeswitch.org> <92e7d2090902111447v1cbb00c0l8fc9f1c8b0c1a514@mail.gmail.com> Message-ID: <4285D10F-3B87-47B4-979A-14C3066B14E0@freeswitch.org> Your VE must be 64bit also. http://wiki.openvz.org/Install_OpenVZ_on_a_x86_64_system_Centos-Fedora If you need the set util listed on that page let me know I have a copy of it. /b On Feb 11, 2009, at 4:47 PM, Nik Martin wrote: > I think the VE I've built is too, but uname is a bit cryptic: > 2.6.18-92.1.18.el5.028stab060.2 #1 SMP Tue Jan 13 11:38:36 MSK 2009 > i686 i686 i386 GNU/Linux > > I can easily change it if FS will run better. From nicolas at medularis.com Wed Feb 11 14:55:11 2009 From: nicolas at medularis.com (Nicolas Brenner) Date: Wed, 11 Feb 2009 19:55:11 -0300 Subject: [Freeswitch-users] FreeSWITCH VPSs In-Reply-To: References: <92e7d2090902111447v1cbb00c0l8fc9f1c8b0c1a514@mail.gmail.com> Message-ID: <1b46b4e80902111455x5b2f87f6p109af34fa306fefc@mail.gmail.com> Also be sure to test it right. I had a mediatemple VPS (they use Virtuozzo I think, the paid version of OpenVZ) and FS would not work right, I had multiple problems, then I switched to a real server and all of that went away. FS would compile and run ok, but then calls wouldn't work or sound wouldn't go through... I never investigated what was the real problem, but switching made the difference. Best regards and good luck! Nicolas On Wed, Feb 11, 2009 at 7:50 PM, Ken Rice wrote: > Be sure to make the Virt nodes 64bit too... FS works 100% better w/ 64bit! > > >> From: Nik Martin >> Reply-To: >> Date: Wed, 11 Feb 2009 16:47:23 -0600 >> To: >> Subject: Re: [Freeswitch-users] FreeSWITCH VPSs >> >> On Wed, Feb 11, 2009 at 4:38 PM, Brian West wrote: >>> Quick note make sure you're 100% 64 bit.. if you need help with that I >>> can show you how on CentOS 5.2 >>> >> >> My hardware Node is running 64 bit Centos 5.2, with OpenVZ's kernel: >> 2.6.18-92.1.18.el5.028stab060.2 #1 SMP Tue Jan 13 11:38:36 MSK 2009 >> x86_64 x86_64 x86_64 GNU/Linux >> >> I think the VE I've built is too, but uname is a bit cryptic: >> 2.6.18-92.1.18.el5.028stab060.2 #1 SMP Tue Jan 13 11:38:36 MSK 2009 >> i686 i686 i386 GNU/Linux >> >> I can easily change it if FS will run better. >> >> Nik >> >> >> >> >>> /b >>> >>> On Feb 11, 2009, at 4:34 PM, Nik Martin wrote: >>> >>>> If any one needs a FreeSWITCH box with a public, static IP, I can >>>> provide them for you at a reasonable cost. I'm building a >>>> Virtualization platform for FreeSWITCH hosting, and have the first >>>> node complete. These are OpenVZ Virtual Engines with Centos 5.2, a >>>> full build environment, and the latest FreeSWITCH trunk. You get 1 >>>> static IP, no NAT, 256 mb ram, and 8 gb disk space, with 1 Megabit of >>>> bandwidth. Great for VOIP service providers, backup switch, testing, >>>> etc. You can contact me directly if you are interested. >>>> >>>> Nik >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From brian at freeswitch.org Wed Feb 11 14:57:35 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 11 Feb 2009 16:57:35 -0600 Subject: [Freeswitch-users] FreeSWITCH VPSs In-Reply-To: <1b46b4e80902111455x5b2f87f6p109af34fa306fefc@mail.gmail.com> References: <92e7d2090902111447v1cbb00c0l8fc9f1c8b0c1a514@mail.gmail.com> <1b46b4e80902111455x5b2f87f6p109af34fa306fefc@mail.gmail.com> Message-ID: <01AFA6BD-7B77-43CC-A22B-64E448E82F04@freeswitch.org> It runs fine under OpenVZ pure 64bit... /b On Feb 11, 2009, at 4:55 PM, Nicolas Brenner wrote: > Also be sure to test it right. I had a mediatemple VPS (they use > Virtuozzo I think, the paid version of OpenVZ) and FS would not work > right, I had multiple problems, then I switched to a real server and > all of that went away. FS would compile and run ok, but then calls > wouldn't work or sound wouldn't go through... I never investigated > what was the real problem, but switching made the difference. > > Best regards and good luck! > > Nicolas From freeswitch at servercorps.com Wed Feb 11 15:03:04 2009 From: freeswitch at servercorps.com (Nik Martin) Date: Wed, 11 Feb 2009 17:03:04 -0600 Subject: [Freeswitch-users] FreeSWITCH VPSs In-Reply-To: <1b46b4e80902111455x5b2f87f6p109af34fa306fefc@mail.gmail.com> References: <92e7d2090902111447v1cbb00c0l8fc9f1c8b0c1a514@mail.gmail.com> <1b46b4e80902111455x5b2f87f6p109af34fa306fefc@mail.gmail.com> Message-ID: <92e7d2090902111503s48b82b2apb2050dc8e33b7caa@mail.gmail.com> I've had a VE in (light) production for about 2 weeks, with no issues so far. I'm going to build a pure 64 bit VE container though, and will run in that for a while too. Brian, you sid you have a readme on that? On Wed, Feb 11, 2009 at 4:55 PM, Nicolas Brenner wrote: > Also be sure to test it right. I had a mediatemple VPS (they use > Virtuozzo I think, the paid version of OpenVZ) and FS would not work > right, I had multiple problems, then I switched to a real server and > all of that went away. FS would compile and run ok, but then calls > wouldn't work or sound wouldn't go through... I never investigated > what was the real problem, but switching made the difference. > > Best regards and good luck! > > Nicolas > > On Wed, Feb 11, 2009 at 7:50 PM, Ken Rice wrote: >> Be sure to make the Virt nodes 64bit too... FS works 100% better w/ 64bit! >> >> >>> From: Nik Martin >>> Reply-To: >>> Date: Wed, 11 Feb 2009 16:47:23 -0600 >>> To: >>> Subject: Re: [Freeswitch-users] FreeSWITCH VPSs >>> >>> On Wed, Feb 11, 2009 at 4:38 PM, Brian West wrote: >>>> Quick note make sure you're 100% 64 bit.. if you need help with that I >>>> can show you how on CentOS 5.2 >>>> >>> >>> My hardware Node is running 64 bit Centos 5.2, with OpenVZ's kernel: >>> 2.6.18-92.1.18.el5.028stab060.2 #1 SMP Tue Jan 13 11:38:36 MSK 2009 >>> x86_64 x86_64 x86_64 GNU/Linux >>> >>> I think the VE I've built is too, but uname is a bit cryptic: >>> 2.6.18-92.1.18.el5.028stab060.2 #1 SMP Tue Jan 13 11:38:36 MSK 2009 >>> i686 i686 i386 GNU/Linux >>> >>> I can easily change it if FS will run better. >>> >>> Nik >>> >>> >>> >>> >>>> /b >>>> >>>> On Feb 11, 2009, at 4:34 PM, Nik Martin wrote: >>>> >>>>> If any one needs a FreeSWITCH box with a public, static IP, I can >>>>> provide them for you at a reasonable cost. I'm building a >>>>> Virtualization platform for FreeSWITCH hosting, and have the first >>>>> node complete. These are OpenVZ Virtual Engines with Centos 5.2, a >>>>> full build environment, and the latest FreeSWITCH trunk. You get 1 >>>>> static IP, no NAT, 256 mb ram, and 8 gb disk space, with 1 Megabit of >>>>> bandwidth. Great for VOIP service providers, backup switch, testing, >>>>> etc. You can contact me directly if you are interested. >>>>> >>>>> Nik >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From brian at freeswitch.org Wed Feb 11 15:08:35 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 11 Feb 2009 17:08:35 -0600 Subject: [Freeswitch-users] FreeSWITCH VPSs In-Reply-To: <92e7d2090902111503s48b82b2apb2050dc8e33b7caa@mail.gmail.com> References: <92e7d2090902111447v1cbb00c0l8fc9f1c8b0c1a514@mail.gmail.com> <1b46b4e80902111455x5b2f87f6p109af34fa306fefc@mail.gmail.com> <92e7d2090902111503s48b82b2apb2050dc8e33b7caa@mail.gmail.com> Message-ID: http://wiki.openvz.org/Install_OpenVZ_on_a_x86_64_system_Centos-Fedora and http://linux.carreira.com.pt/ovzutils/setx86_64-0.3.tar.gz Will set it up for 64bit containers and patch everything to work correctly... /b On Feb 11, 2009, at 5:03 PM, Nik Martin wrote: > I've had a VE in (light) production for about 2 weeks, with no issues > so far. I'm going to build a pure 64 bit VE container though, and > will run in that for a while too. Brian, you sid you have a readme on > that? > From steveu at coppice.org Wed Feb 11 15:15:52 2009 From: steveu at coppice.org (Steve Underwood) Date: Thu, 12 Feb 2009 07:15:52 +0800 Subject: [Freeswitch-users] Skype as a path to wideband adoption Message-ID: <49935C28.8090008@coppice.org> Hi all, Consider the following: - FS has recently greatly enhanced its support for wide, wider and widest band telephony - That advantage is of no benefit when interworking with the PSTN - Skype has had wideband since day one, and just got super-ultra-duper-wideband. - FS is acquiring skype connectivity options (2, which hopefully will converge to one best in class option). One obvious conclusion is Skype offers the best possibility for having broad coverage with wideband voice. However, codec issues are certain to degrade quality. Lossy compressed codecs don't transcode well, and the codecs Skype uses are their own. FS is gaining a range of open and royalty free licence wideband codecs, but not the ones Skype uses. Skype choose widely supported narrowband codecs - G.711 and G.729 - but have not used a wideband codec with broad support. So.... the point of this note is "what can we do to optimise things?" Steve From nik.middleton at noblesolutions.co.uk Wed Feb 11 15:36:04 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Wed, 11 Feb 2009 23:36:04 -0000 Subject: [Freeswitch-users] FS 1.0.2 Crash and burn In-Reply-To: References: <7EA294D9-389F-4543-8F24-ED4C08600BE7@freeswitch.org><75A42DD8-9BA0-46CC-8121-04CE9DB27A27@freeswitch.org> Message-ID: I've abandoned LUA. All sorts of problems (DTMF etc). Also reports of memory leaks when using MYSQL driver. Looking on the WIKI, JavaScript seems very well supported; PLUS DTMF works just fine (pulling my hair out on LUA) Guess I'm going to follow the path of least resistance on this one and use JS and ODBC Regards, ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: 11 February 2009 20:23 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] FS 1.0.2 Crash and burn Try starting it from your /usr/local/freeswitch/bin... ./freeswitch it'll dump in the same folder. /b On Feb 11, 2009, at 2:20 PM, Nik Middleton wrote: Where is the core dump written? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090211/b1471e63/attachment-0002.html From nik.middleton at noblesolutions.co.uk Wed Feb 11 15:41:48 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Wed, 11 Feb 2009 23:41:48 -0000 Subject: [Freeswitch-users] Setting outbound callerid using js Message-ID: Hi Guys I'm trying to set the outbound caller-id in js. The params seem to be acceptable, except I'm getting the default +000000000 caller-ID sent. Should the below work with js? session.originate(session,'{accountcode=54321,ignore_early_media=true,or igination_caller_id_number=07630600000,originate_timeout=25}sofia/gatewa y/mygw/01XXXXXXXXXXX'); (this works using lua BTW) regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090211/41695d60/attachment-0002.html From brian at freeswitch.org Wed Feb 11 15:50:07 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 11 Feb 2009 17:50:07 -0600 Subject: [Freeswitch-users] FS 1.0.2 Crash and burn In-Reply-To: References: <7EA294D9-389F-4543-8F24-ED4C08600BE7@freeswitch.org><75A42DD8-9BA0-46CC-8121-04CE9DB27A27@freeswitch.org> Message-ID: <6C1B5368-931F-4FA0-87F1-F0EE011C35DA@freeswitch.org> Lua has known issues with MySQL you must use latest SVN builds of the luasql driver for that to avoid it.. and still its not stellar.. the unixODBC one on the other hand works fine. /b On Feb 11, 2009, at 5:36 PM, Nik Middleton wrote: > I?ve abandoned LUA. > > All sorts of problems (DTMF etc). Also reports of memory leaks when > using MYSQL driver. > > Looking on the WIKI, JavaScript seems very well supported; PLUS DTMF > works just fine (pulling my hair out on LUA) > > Guess I?m going to follow the path of least resistance on this one > and use JS and ODBC > > Regards, > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090211/0e081cdf/attachment-0002.html From freeswitch at servercorps.com Wed Feb 11 16:04:20 2009 From: freeswitch at servercorps.com (Nik Martin) Date: Wed, 11 Feb 2009 18:04:20 -0600 Subject: [Freeswitch-users] FreeSWITCH VPSs In-Reply-To: References: <92e7d2090902111447v1cbb00c0l8fc9f1c8b0c1a514@mail.gmail.com> <1b46b4e80902111455x5b2f87f6p109af34fa306fefc@mail.gmail.com> <92e7d2090902111503s48b82b2apb2050dc8e33b7caa@mail.gmail.com> Message-ID: <92e7d2090902111604y516b54c6ya8832cb1944a8a3a@mail.gmail.com> Great, thanks! Nik On Wed, Feb 11, 2009 at 5:08 PM, Brian West wrote: > http://wiki.openvz.org/Install_OpenVZ_on_a_x86_64_system_Centos-Fedora > and > http://linux.carreira.com.pt/ovzutils/setx86_64-0.3.tar.gz > > Will set it up for 64bit containers and patch everything to work > correctly... > > /b > > On Feb 11, 2009, at 5:03 PM, Nik Martin wrote: > >> I've had a VE in (light) production for about 2 weeks, with no issues >> so far. I'm going to build a pure 64 bit VE container though, and >> will run in that for a while too. Brian, you sid you have a readme on >> that? >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From edpimentl at gmail.com Wed Feb 11 16:08:14 2009 From: edpimentl at gmail.com (EdPimentl) Date: Wed, 11 Feb 2009 19:08:14 -0500 Subject: [Freeswitch-users] FreeSWITCH VPSs In-Reply-To: <92e7d2090902111434l60afd3a8i634b4df688ed4011@mail.gmail.com> References: <92e7d2090902111434l60afd3a8i634b4df688ed4011@mail.gmail.com> Message-ID: <9dc4a1670902111608h4d8ca43eq85800ee902113a1a@mail.gmail.com> 256 MB Ram ..... is this correct?... Does any VoIP provider to use this? -E -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090211/3afe87ef/attachment-0002.html From nik.middleton at noblesolutions.co.uk Wed Feb 11 16:09:53 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Thu, 12 Feb 2009 00:09:53 -0000 Subject: [Freeswitch-users] Call accounting not working as expected Message-ID: I'm having an issue with call accounting If I initiate a call, and it is then transferred to an IVR menu. Person selects 1 to talk to someone. In js else if (data.digit == "5") { if (session.ready()) { var new_session = new Session(); new_session.originate(..... This Second call leg is not accounted for in either CSV or xml logs Am I doing something wrong? In the XML record is shows that I've diverted to the new number, but the time is all bundled with the initial call. This is exactly the same issue in Asterisk, which I was hoping to avoid. In Other words, why isn't a new call record created for the second leg? Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090212/262d461a/attachment-0002.html From chavpaskov at shaw.ca Wed Feb 11 16:12:09 2009 From: chavpaskov at shaw.ca (Tchavdar Paskov) Date: Wed, 11 Feb 2009 16:12:09 -0800 Subject: [Freeswitch-users] How i can trigger action or application in case of sip 302 received Message-ID: Hi, Everybody i was wondering if anybody can give me a hint on how i can set a condition/action in? dial plan in case of SIP 302 being received. Regards Chav? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090211/4105d3cc/attachment-0002.html From brian at freeswitch.org Wed Feb 11 16:12:52 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 11 Feb 2009 18:12:52 -0600 Subject: [Freeswitch-users] FreeSWITCH VPSs In-Reply-To: <9dc4a1670902111608h4d8ca43eq85800ee902113a1a@mail.gmail.com> References: <92e7d2090902111434l60afd3a8i634b4df688ed4011@mail.gmail.com> <9dc4a1670902111608h4d8ca43eq85800ee902113a1a@mail.gmail.com> Message-ID: <28F5713C-838B-409A-897B-F9E40D39B928@freeswitch.org> You can run a small SOHO operation on 256 megs /b On Feb 11, 2009, at 6:08 PM, EdPimentl wrote: > 256 MB Ram ..... is this correct?... Does any VoIP provider to use > this? > -E From brian at freeswitch.org Wed Feb 11 16:14:17 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 11 Feb 2009 18:14:17 -0600 Subject: [Freeswitch-users] Call accounting not working as expected In-Reply-To: References: Message-ID: first off don't use the session.originate var new_session = new Session({var=val}sofia/blah/blah); will do it all for you in one step. Also can you point me to where on the wiki that keeps talking about session.originate? I need to clean them off there. /b On Feb 11, 2009, at 6:09 PM, Nik Middleton wrote: > else if (data.digit == "5") { > if (session.ready()) { > var new_session = new Session(); > new_session.originate(?.. > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090211/c9627e52/attachment-0002.html From brian at freeswitch.org Wed Feb 11 16:16:39 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 11 Feb 2009 18:16:39 -0600 Subject: [Freeswitch-users] How i can trigger action or application in case of sip 302 received In-Reply-To: References: Message-ID: <28FEEE0A-D0EA-4D78-BCB8-DDA0F47A1F1D@freeswitch.org> Please refer to the extension in public.xml and default.xml both will cause a deflect to be done so the 3 leg call gets turned back into a 2 leg call. In some cases it might be desired to do a 3 leg call so you can bill the party that caused the 302 and the original party also. /b On Feb 11, 2009, at 6:12 PM, Tchavdar Paskov wrote: > Hi, Everybody > > i was wondering if anybody can give me a hint on how i can set a > condition/action in dial plan in case of SIP 302 being received. > > Regards > > Chav From chavpaskov at shaw.ca Wed Feb 11 16:21:57 2009 From: chavpaskov at shaw.ca (Tchavdar Paskov) Date: Wed, 11 Feb 2009 16:21:57 -0800 Subject: [Freeswitch-users] How i can trigger action or application in case of sip 302 received In-Reply-To: <28FEEE0A-D0EA-4D78-BCB8-DDA0F47A1F1D@freeswitch.org> References: <28FEEE0A-D0EA-4D78-BCB8-DDA0F47A1F1D@freeswitch.org> Message-ID: Thank you Brian, is there any way to inspect? what exactly is sent in 302 message and if possible? to replace it? or remove it. Regards Chav ----- Original Message ----- From: Brian West Date: Wednesday, February 11, 2009 4:17 pm Subject: Re: [Freeswitch-users] How i can trigger action or application in case of sip 302 received To: freeswitch-users at lists.freeswitch.org > Please refer to the extension in > public.xml? > and default.xml? both will cause a deflect to be done so > the 3 leg? > call gets turned back into a 2 leg call.? In some cases it > might be? > desired to do a 3 leg call so you can bill the party that caused > the? > 302 and the original party also. > > /b > > On Feb 11, 2009, at 6:12 PM, Tchavdar Paskov wrote: > > > Hi, Everybody > > > > i was wondering if anybody can give me a hint on how i can set > a? > > condition/action in? dial plan in case of SIP 302 being > received.> > > Regards > > > > Chav > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090211/7380a6ea/attachment-0002.html From edpimentl at gmail.com Wed Feb 11 16:24:46 2009 From: edpimentl at gmail.com (EdPimentl) Date: Wed, 11 Feb 2009 19:24:46 -0500 Subject: [Freeswitch-users] FreeSWITCH VPSs In-Reply-To: <28F5713C-838B-409A-897B-F9E40D39B928@freeswitch.org> References: <92e7d2090902111434l60afd3a8i634b4df688ed4011@mail.gmail.com> <9dc4a1670902111608h4d8ca43eq85800ee902113a1a@mail.gmail.com> <28F5713C-838B-409A-897B-F9E40D39B928@freeswitch.org> Message-ID: <9dc4a1670902111624s456b7fbeg99c1e078fe8def3d@mail.gmail.com> Soho,,, yes of course... Voip (soho)Service Provider.... not convinced is possible to provide reliable QoS. My .02 cents -E -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090211/d6b45c24/attachment-0002.html From msc at freeswitch.org Wed Feb 11 16:26:26 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 11 Feb 2009 16:26:26 -0800 Subject: [Freeswitch-users] Setting outbound callerid using js In-Reply-To: References: Message-ID: <87f2f3b90902111626x6db5dc61t68a0170a3460b42d@mail.gmail.com> session.originate(session,'{accountcode=54321,ignore_early_media=true,origination_caller_id_number=07630600000,originate_timeout=25}sofia/gateway/mygw/01XXXXXXXXXXX'); > > > > (this works using lua BTW) > hmmmm... how about using "effective_caller_id_number" instead? I think the JavaScript paradigm is a bit different than the Lua/Perl one. Let us know if that works or not. -MC From brian at freeswitch.org Wed Feb 11 16:28:03 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 11 Feb 2009 18:28:03 -0600 Subject: [Freeswitch-users] How i can trigger action or application in case of sip 302 received In-Reply-To: References: <28FEEE0A-D0EA-4D78-BCB8-DDA0F47A1F1D@freeswitch.org> Message-ID: <97D545F6-8972-41CB-8D1B-0865E62B680C@freeswitch.org> Nope its on auto pilot... we don't get passed the 302 from sofia. So what you have there is all you can get at. /b On Feb 11, 2009, at 6:21 PM, Tchavdar Paskov wrote: > Thank you Brian, > is there any way to inspect what exactly is sent in 302 message and > if possible to replace it or remove it. > > Regards > Chav From brian at freeswitch.org Wed Feb 11 16:28:56 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 11 Feb 2009 18:28:56 -0600 Subject: [Freeswitch-users] FreeSWITCH VPSs In-Reply-To: <9dc4a1670902111624s456b7fbeg99c1e078fe8def3d@mail.gmail.com> References: <92e7d2090902111434l60afd3a8i634b4df688ed4011@mail.gmail.com> <9dc4a1670902111608h4d8ca43eq85800ee902113a1a@mail.gmail.com> <28F5713C-838B-409A-897B-F9E40D39B928@freeswitch.org> <9dc4a1670902111624s456b7fbeg99c1e078fe8def3d@mail.gmail.com> Message-ID: <76543A10-18D7-450D-ACA1-E4CAE64F471F@freeswitch.org> Actually you can if you don't overload the machine like most VPS providers do... The advantage with OpenVZ in this case is that you can migrate the running FreeSWITCH instance between hardware nodes and not drop calls at this size. /b On Feb 11, 2009, at 6:24 PM, EdPimentl wrote: > Soho,,, yes of course... > Voip (soho)Service Provider.... not convinced is possible to provide > reliable QoS. > My .02 cents > -E From msc at freeswitch.org Wed Feb 11 16:31:05 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 11 Feb 2009 16:31:05 -0800 Subject: [Freeswitch-users] Call accounting not working as expected In-Reply-To: References: Message-ID: <87f2f3b90902111631q717facfctd4b1771d62f31207@mail.gmail.com> > > > This Second call leg is not accounted for in either CSV or xml logs > > > > Am I doing something wrong? In the XML record is shows that I've diverted > to the new number, but the time is all bundled with the initial call. > > > > This is exactly the same issue in Asterisk, which I was hoping to avoid. In > Other words, why isn't a new call record created for the second leg? Could you pastebin the xml cdr? I'm curious to see if it's anything like the ones I have. -MC From anthony.minessale at gmail.com Wed Feb 11 16:31:57 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 11 Feb 2009 18:31:57 -0600 Subject: [Freeswitch-users] Skype as a path to wideband adoption In-Reply-To: <49935C28.8090008@coppice.org> References: <49935C28.8090008@coppice.org> Message-ID: <191c3a030902111631r143ea868y8ee6164c86f3f071@mail.gmail.com> Good question, Does anybody have any contacts at Skype to open a discussion with them? Should we just call them anyway? They have chosen to interop directly with asterisk which has not completed it's attempt at wideband support. Maybe they are more interested in connecting to the PSTN but it's worth a try to ask them. On Wed, Feb 11, 2009 at 5:15 PM, Steve Underwood wrote: > Hi all, > > Consider the following: > > - FS has recently greatly enhanced its support for wide, wider and > widest band telephony > - That advantage is of no benefit when interworking with the PSTN > - Skype has had wideband since day one, and just got > super-ultra-duper-wideband. > - FS is acquiring skype connectivity options (2, which hopefully > will converge to one best in class option). > > One obvious conclusion is Skype offers the best possibility for having > broad coverage with wideband voice. > However, codec issues are certain to degrade quality. Lossy compressed > codecs don't transcode well, and the > codecs Skype uses are their own. FS is gaining a range of open and > royalty free licence wideband codecs, but > not the ones Skype uses. Skype choose widely supported narrowband codecs > - G.711 and G.729 - but have not used a wideband codec with broad support. > > So.... the point of this note is "what can we do to optimise things?" > > Steve > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090211/1b5b4a5d/attachment-0002.html From nik.middleton at noblesolutions.co.uk Wed Feb 11 16:47:32 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Thu, 12 Feb 2009 00:47:32 -0000 Subject: [Freeswitch-users] Call accounting not working as expected In-Reply-To: References: Message-ID: Thanks, that cured the call accounting However, in the original originate, any ideas why {var=val} is not being processed? Regards, ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: 12 February 2009 00:14 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Call accounting not working as expected first off don't use the session.originate var new_session = new Session({var=val}sofia/blah/blah); will do it all for you in one step. Also can you point me to where on the wiki that keeps talking about session.originate? I need to clean them off there. /b On Feb 11, 2009, at 6:09 PM, Nik Middleton wrote: else if (data.digit == "5") { if (session.ready()) { var new_session = new Session(); new_session.originate(..... -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090212/2648c12d/attachment-0002.html From msc at freeswitch.org Wed Feb 11 16:59:10 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 11 Feb 2009 16:59:10 -0800 Subject: [Freeswitch-users] Skype as a path to wideband adoption In-Reply-To: <191c3a030902111631r143ea868y8ee6164c86f3f071@mail.gmail.com> References: <49935C28.8090008@coppice.org> <191c3a030902111631r143ea868y8ee6164c86f3f071@mail.gmail.com> Message-ID: <87f2f3b90902111659u71330393waf48c93d8564bfc9@mail.gmail.com> On Wed, Feb 11, 2009 at 4:31 PM, Anthony Minessale wrote: > Good question, > > Does anybody have any contacts at Skype to open a discussion with them? > Should we just call them anyway? > > They have chosen to interop directly with asterisk which has not completed > it's attempt at wideband support. > Maybe they are more interested in connecting to the PSTN but it's worth a > try to ask them. > If someone with some clout could call them that would be ideal. If no one cares to then I would be happy to contact them. Volunteers? :) -MC From nik.middleton at noblesolutions.co.uk Wed Feb 11 16:59:42 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Thu, 12 Feb 2009 00:59:42 -0000 Subject: [Freeswitch-users] Call accounting not working as expected In-Reply-To: References: Message-ID: Further to my last. It is kind of being processed, the account code is being set, XML cdr's are created and are correct, but csv cdr's for the account code are not Caller ID is not being set in the A leg but is in the B Leg ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Nik Middleton Sent: 12 February 2009 00:48 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Call accounting not working as expected Thanks, that cured the call accounting However, in the original originate, any ideas why {var=val} is not being processed? Regards, ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: 12 February 2009 00:14 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Call accounting not working as expected first off don't use the session.originate var new_session = new Session({var=val}sofia/blah/blah); will do it all for you in one step. Also can you point me to where on the wiki that keeps talking about session.originate? I need to clean them off there. /b On Feb 11, 2009, at 6:09 PM, Nik Middleton wrote: else if (data.digit == "5") { if (session.ready()) { var new_session = new Session(); new_session.originate(..... -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090212/71b67346/attachment-0002.html From msc at freeswitch.org Wed Feb 11 17:01:16 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 11 Feb 2009 17:01:16 -0800 Subject: [Freeswitch-users] Call accounting not working as expected In-Reply-To: References: Message-ID: <87f2f3b90902111701s3f2f4801jc80f1a192264f203@mail.gmail.com> > However, in the original originate, any ideas why {var=val} is not being > processed? I think Brian's suggestion is the way to go: > first off don't use the session.originate > > > > var new_session = new Session({var=val}sofia/blah/blah); The above syntax is the clean way to do it. > > > > will do it all for you in one step. Also can you point me to where on the > wiki that keeps talking about session.originate? I need to clean them off > there. From msc at freeswitch.org Wed Feb 11 17:09:59 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 11 Feb 2009 17:09:59 -0800 Subject: [Freeswitch-users] Call accounting not working as expected In-Reply-To: References: Message-ID: <87f2f3b90902111709t2c6db8e2jf190b799eeca6ee8@mail.gmail.com> > It is kind of being processed, the account code is being set, XML cdr's are > created and are correct, but csv cdr's for the account code are not > > > > Caller ID is not being set in the A leg but is in the B Leg DING DING DING!!! We have a weener! Okay, that was the key piece of info. Most likely you are logging only the A leg in the CSV CDRs. Go to conf/autoload_configs/cdr_csv.conf.xml and look for these two lines: Most likely you need to use "b" or "ab" depending on your scenario. Try it each way and see how you like the results, then please report back. Thanks! -MC From edpimentl at gmail.com Wed Feb 11 17:22:37 2009 From: edpimentl at gmail.com (EdPimentl) Date: Wed, 11 Feb 2009 20:22:37 -0500 Subject: [Freeswitch-users] FreeSWITCH VPSs In-Reply-To: <76543A10-18D7-450D-ACA1-E4CAE64F471F@freeswitch.org> References: <92e7d2090902111434l60afd3a8i634b4df688ed4011@mail.gmail.com> <9dc4a1670902111608h4d8ca43eq85800ee902113a1a@mail.gmail.com> <28F5713C-838B-409A-897B-F9E40D39B928@freeswitch.org> <9dc4a1670902111624s456b7fbeg99c1e078fe8def3d@mail.gmail.com> <76543A10-18D7-450D-ACA1-E4CAE64F471F@freeswitch.org> Message-ID: <9dc4a1670902111722k6dfba174m302681f902e7ac4c@mail.gmail.com> Thanks and agree 100% and appreciated the added insight. My thinking of service provide grade deployment something along the line of Ken Rice or Michal B. Or a FS / TelcoBridges External service deployment.... Best regards, E -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090211/b551a9db/attachment-0002.html From brian at freeswitch.org Wed Feb 11 17:29:21 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 11 Feb 2009 19:29:21 -0600 Subject: [Freeswitch-users] gateways not hitting right context now? Message-ID: <62B6AA3B-86E2-408D-8D5D-742E3B93A355@freeswitch.org> If you have outbound gateways registering make sure you set the context and extension param on the gateway so it'll go to the right spot. Recent changes made it work much smoother. /b From freeswitch at servercorps.com Wed Feb 11 17:59:50 2009 From: freeswitch at servercorps.com (Nik Martin) Date: Wed, 11 Feb 2009 19:59:50 -0600 Subject: [Freeswitch-users] FreeSWITCH VPSs In-Reply-To: <76543A10-18D7-450D-ACA1-E4CAE64F471F@freeswitch.org> References: <92e7d2090902111434l60afd3a8i634b4df688ed4011@mail.gmail.com> <9dc4a1670902111608h4d8ca43eq85800ee902113a1a@mail.gmail.com> <28F5713C-838B-409A-897B-F9E40D39B928@freeswitch.org> <9dc4a1670902111624s456b7fbeg99c1e078fe8def3d@mail.gmail.com> <76543A10-18D7-450D-ACA1-E4CAE64F471F@freeswitch.org> Message-ID: <92e7d2090902111759g629ca736k879103144d012155@mail.gmail.com> My goal is obviously not to provide carrier grade VOIP switch service, but a platform for sandbox testing of configurations, SOHO VOIP Switches, development of FS addons, backup switch capability, etc. Doing this stuff at home behind NAT and a consumer grade router is one reason Brian, Anthony, Mike, et al. are half crazy. I run my company's phone switch in a 32 bit OpenVZ VE with 256 Mb ram, and have no issues. When I goof around trying to transcode between 8 and 16 bit codecs and whatnot, sure it gets tight, but FS on an idle system keeps 23 mb of of ram resident, and rarely if ever hits a 256 mb bean counter (limit). Also, these limits are not hard, I just know what my hardware has, and am trying to offer as much value as I can for what I have in the systems. 128 Gb of ECC ram and Quad/Quad Core zeons are still pretty pricey! Nik On Wed, Feb 11, 2009 at 6:28 PM, Brian West wrote: > Actually you can if you don't overload the machine like most VPS > providers do... The advantage with OpenVZ in this case is that you can > migrate the running FreeSWITCH instance between hardware nodes and not > drop calls at this size. > > /b > > On Feb 11, 2009, at 6:24 PM, EdPimentl wrote: > >> Soho,,, yes of course... >> Voip (soho)Service Provider.... not convinced is possible to provide >> reliable QoS. >> My .02 cents >> -E > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From pbd at suspiria.net Wed Feb 11 17:59:56 2009 From: pbd at suspiria.net (Public Dump) Date: Thu, 12 Feb 2009 02:59:56 +0100 Subject: [Freeswitch-users] High CPU load after starting In-Reply-To: References: Message-ID: <13C421883438EB42B9E2C30069FD4AB767BF49AF13@crushinator.central.local> http://files.freeswitch.org/freeswitch-snapshot.tar.gz Extracted into empty directory, SVN update, compile. > it was a completely clean build? > as in compleletely new and/or clean solution? > > Which tarball was it ? > From brian at freeswitch.org Wed Feb 11 18:10:58 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 11 Feb 2009 20:10:58 -0600 Subject: [Freeswitch-users] High CPU load after starting In-Reply-To: <13C421883438EB42B9E2C30069FD4AB767BF49AF13@crushinator.central.local> References: <13C421883438EB42B9E2C30069FD4AB767BF49AF13@crushinator.central.local> Message-ID: <2D54170F-8539-4EAB-9297-8331DAA5E512@freeswitch.org> OK does it work now? We have tested this on various windows installs among the team here and not seeing this issue... it was a known issue back in Nov. or Dec. but thats long been fixed. /b On Feb 11, 2009, at 7:59 PM, Public Dump wrote: > http://files.freeswitch.org/freeswitch-snapshot.tar.gz > > Extracted into empty directory, SVN update, compile. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090211/a9abd53a/attachment-0002.html From red.rain.seven at gmail.com Wed Feb 11 16:37:03 2009 From: red.rain.seven at gmail.com (Henry Huang) Date: Wed, 11 Feb 2009 16:37:03 -0800 Subject: [Freeswitch-users] FreeSWITCH VPSs In-Reply-To: <92e7d2090902111434l60afd3a8i634b4df688ed4011@mail.gmail.com> References: <92e7d2090902111434l60afd3a8i634b4df688ed4011@mail.gmail.com> Message-ID: <59ad9ca10902111637n1459d3cep5bc73dffe6bb474f@mail.gmail.com> Brian: I am also running my freeswitch on my own openVZ containers. Just how do you verify if the freeswitch is compiled as 64bit? I would assume if I compile it under a 64bit container, I would automatically get a 64bit freeswitch right? On Wed, Feb 11, 2009 at 2:34 PM, Nik Martin wrote: > If any one needs a FreeSWITCH box with a public, static IP, I can > provide them for you at a reasonable cost. I'm building a > Virtualization platform for FreeSWITCH hosting, and have the first > node complete. These are OpenVZ Virtual Engines with Centos 5.2, a > full build environment, and the latest FreeSWITCH trunk. You get 1 > static IP, no NAT, 256 mb ram, and 8 gb disk space, with 1 Megabit of > bandwidth. Great for VOIP service providers, backup switch, testing, > etc. You can contact me directly if you are interested. > > Nik > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Henry Huang UniC Solution - Communication Unified -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090211/8e3d8474/attachment-0002.html From brian at freeswitch.org Wed Feb 11 18:55:36 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 11 Feb 2009 20:55:36 -0600 Subject: [Freeswitch-users] FreeSWITCH VPSs In-Reply-To: <59ad9ca10902111637n1459d3cep5bc73dffe6bb474f@mail.gmail.com> References: <92e7d2090902111434l60afd3a8i634b4df688ed4011@mail.gmail.com> <59ad9ca10902111637n1459d3cep5bc73dffe6bb474f@mail.gmail.com> Message-ID: ding ding ding .. yep! "file /usr/local/freeswitch/bin/freeswitch" will also confirm /b On Feb 11, 2009, at 6:37 PM, Henry Huang wrote: > Brian: > > I am also running my freeswitch on my own openVZ containers. Just > how do you verify if the freeswitch is compiled as 64bit? I would > assume if I compile it under a 64bit container, I would > automatically get a 64bit freeswitch right? From anthony.minessale at gmail.com Wed Feb 11 20:25:55 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 11 Feb 2009 22:25:55 -0600 Subject: [Freeswitch-users] FS 1.0.2 Crash and burn In-Reply-To: <6C1B5368-931F-4FA0-87F1-F0EE011C35DA@freeswitch.org> References: <7EA294D9-389F-4543-8F24-ED4C08600BE7@freeswitch.org> <75A42DD8-9BA0-46CC-8121-04CE9DB27A27@freeswitch.org> <6C1B5368-931F-4FA0-87F1-F0EE011C35DA@freeswitch.org> Message-ID: <191c3a030902112025l401c759dife416cf34d51d134@mail.gmail.com> There is always C, it's actually considered a high level language by many ;) On Wed, Feb 11, 2009 at 5:50 PM, Brian West wrote: > Lua has known issues with MySQL you must use latest SVN builds of the > luasql driver for that to avoid it.. and still its not stellar.. the > unixODBC one on the other hand works fine. > /b > > On Feb 11, 2009, at 5:36 PM, Nik Middleton wrote: > > I've abandoned LUA. > > All sorts of problems (DTMF etc). Also reports of memory leaks when using > MYSQL driver. > > Looking on the WIKI, JavaScript seems very well supported; PLUS DTMF works > just fine (pulling my hair out on LUA) > > Guess I'm going to follow the path of least resistance on this one and use > JS and ODBC > > Regards, > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090211/d102432c/attachment-0002.html From pauld at versafon.com Wed Feb 11 20:38:48 2009 From: pauld at versafon.com (pauld) Date: Wed, 11 Feb 2009 23:38:48 -0500 Subject: [Freeswitch-users] Cannot choose Cepstral voice from dialplan In-Reply-To: <87f2f3b90902090904i72e37241hb5d4808eb31fcc18@mail.gmail.com> References: <498E67BF.3060207@versafon.com> <8B964172-6EEB-4B5B-B4B1-D35BFEAEB460@freeswitch.org> <498F6751.2000701@versafon.com> <87f2f3b90902090904i72e37241hb5d4808eb31fcc18@mail.gmail.com> Message-ID: <4993A7D8.1090004@versafon.com> The issue was resolved by creating symlinks to cepstral libs in FS lib directory. I tried that on 1.0.3, but most probably it would work on 1.0.2 as well. Thanks for help. BTW, without that FS would do a core dump (seg fault) on shutdown after TTS was invoked at least once. Looking at FS logs I see "TRANSCODING_NECESSARY" when executing dynamic text even with 8 kHz voice. Why would that be? Looks like it's PCMU/8000 what it's transcoding to what? Michael Collins wrote: > On Sun, Feb 8, 2009 at 3:14 PM, pauld wrote: > >> The libs are there with correct symlinks, see below. I tested both >> voices directly via swift command, works fine. >> Any other ideas? >> It's Cepstral 5.1, FS 1.0.2. >> >> > > Well, first I recommend getting on latest trunk if that's at all > possible for you. The devs have made a ton of improvements in the last > five weeks. Second, this might actually be an issue with FS looking in > its own lib directory for these .so files. Try a symlink from > /usr/local/freeswitch/lib to your /opt/swift/lib (or whatever the name > is) dir for each .so file. However, I think Raymond is correct - some > weirdness has been reported by some Cepstral users on 5.1. We'd > definitely like to hear about your experiences if and when you get it > running. > > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From brian at freeswitch.org Wed Feb 11 20:43:53 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 11 Feb 2009 22:43:53 -0600 Subject: [Freeswitch-users] Cannot choose Cepstral voice from dialplan In-Reply-To: <4993A7D8.1090004@versafon.com> References: <498E67BF.3060207@versafon.com> <8B964172-6EEB-4B5B-B4B1-D35BFEAEB460@freeswitch.org> <498F6751.2000701@versafon.com> <87f2f3b90902090904i72e37241hb5d4808eb31fcc18@mail.gmail.com> <4993A7D8.1090004@versafon.com> Message-ID: <10703C0D-0BA2-4FB3-B669-B87BF93912E0@freeswitch.org> This is normal. Are you using 5.0? can you include examples of how you're doing this? /b On Feb 11, 2009, at 10:38 PM, pauld wrote: > "TRANSCODING_NECESSARY" From anthony.minessale at gmail.com Wed Feb 11 20:48:26 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 11 Feb 2009 22:48:26 -0600 Subject: [Freeswitch-users] FreeSWITCH VPSs In-Reply-To: <92e7d2090902111759g629ca736k879103144d012155@mail.gmail.com> References: <92e7d2090902111434l60afd3a8i634b4df688ed4011@mail.gmail.com> <9dc4a1670902111608h4d8ca43eq85800ee902113a1a@mail.gmail.com> <28F5713C-838B-409A-897B-F9E40D39B928@freeswitch.org> <9dc4a1670902111624s456b7fbeg99c1e078fe8def3d@mail.gmail.com> <76543A10-18D7-450D-ACA1-E4CAE64F471F@freeswitch.org> <92e7d2090902111759g629ca736k879103144d012155@mail.gmail.com> Message-ID: <191c3a030902112048l1d362f79nf2d0fbb65f2c4984@mail.gmail.com> cool, However.... you say... "but a platform for sandbox testing of configurations, SOHO VOIP Switches, development of FS addons, backup switch capability, etc. Doing this stuff at home behind NAT and a consumer grade router is one reason Brian, Anthony, Mike, et al. are half crazy. " First of all, you have no idea on what platform and where we do our development. If you only saw how many terminals to random servers spanning the globe we have open...... You have the NAT part right, if you knew the hours it took from behind NAT to get all the code right for SIP interop or even had to deal with half the bullshit it takes to get SIP working, you'd be in the madhouse so don't you dare try using us as an ad slogan. Secondly, we are not half crazy, we are completely crazy and most of it comes from spending all day on this list tending to your never-ending threads while trying to help the other people in the community who actually give something back. If you want to use our list to advertise this service maybe you should find a way to contribute to the project rather than constantly asking for help in 5 separate emails in one day followed by another thread trying to sell something. A reasonable cost would be FREE just like everything else around here. On Wed, Feb 11, 2009 at 7:59 PM, Nik Martin wrote: > My goal is obviously not to provide carrier grade VOIP switch service, > but a platform for sandbox testing of configurations, SOHO VOIP > Switches, development of FS addons, backup switch capability, etc. > Doing this stuff at home behind NAT and a consumer grade router is one > reason Brian, Anthony, Mike, et al. are half crazy. I run my > company's phone switch in a 32 bit OpenVZ VE with 256 Mb ram, and have > no issues. When I goof around trying to transcode between 8 and 16 > bit codecs and whatnot, sure it gets tight, but FS on an idle system > keeps 23 mb of of ram resident, and rarely if ever hits a 256 mb bean > counter (limit). > > Also, these limits are not hard, I just know what my hardware has, and > am trying to offer as much value as I can for what I have in the > systems. 128 Gb of ECC ram and Quad/Quad Core zeons are still pretty > pricey! > > Nik > > > > > On Wed, Feb 11, 2009 at 6:28 PM, Brian West wrote: > > Actually you can if you don't overload the machine like most VPS > > providers do... The advantage with OpenVZ in this case is that you can > > migrate the running FreeSWITCH instance between hardware nodes and not > > drop calls at this size. > > > > /b > > > > On Feb 11, 2009, at 6:24 PM, EdPimentl wrote: > > > >> Soho,,, yes of course... > >> Voip (soho)Service Provider.... not convinced is possible to provide > >> reliable QoS. > >> My .02 cents > >> -E > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090211/123eae22/attachment-0002.html From odermann at googlemail.com Wed Feb 11 23:39:30 2009 From: odermann at googlemail.com (Dennis) Date: Thu, 12 Feb 2009 08:39:30 +0100 Subject: [Freeswitch-users] Socket-Event on "originate call"? In-Reply-To: References: <5e414ed0902090716y10f95f71o5760dfdf50d1c5cb@mail.gmail.com> <191c3a030902090727n44eef7d7x83704b66e628081@mail.gmail.com> <5e414ed0902100716s2c053a48w2174f7587dd0641e@mail.gmail.com> <5e414ed0902111037q2a6f4097sebd08cb712f042bf@mail.gmail.com> <397EFED1-22E6-4B78-BDE4-3AE5F4204E5B@freeswitch.org> <5e414ed0902111211y3456d5eaqc99baa126cb9dcad@mail.gmail.com> Message-ID: <5e414ed0902112339o6c1c499cxe30d0e255d6756f0@mail.gmail.com> the problem is fixed in the latest version of fs - at least it is working as before without any errors. but there is still the question, if the changes where made because of our problem with the not starting socket!? we can see in the cli, that the var is set, but it does not change anything regarding our problem. 2009/2/11 Brian West : > Please collect the backtrace and report it on Jira. > > /b > > On Feb 11, 2009, at 2:11 PM, Dennis wrote: > >> this does not help. we are using socket outbound and everything worked >> before the changes yesterday. >> >> we have the same error with other dialplans. From helmut.kuper at ewetel.de Thu Feb 12 00:07:38 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Thu, 12 Feb 2009 09:07:38 +0100 Subject: [Freeswitch-users] segfault when shoutcast plays mp3 and extension hangs up In-Reply-To: <191c3a030902110729g30b30d46gac0f56205f8d48cf@mail.gmail.com> References: <4992BFFC.50006@ewetel.de> <4992C801.4020906@ewetel.de> <191c3a030902110558i9702139g6b12aaf15f6d3aac@mail.gmail.com> <4992DCC2.3020702@ewetel.de> <191c3a030902110617n3dcd19cfr204c00792e1e87ed@mail.gmail.com> <4992E559.6060506@ewetel.de> <191c3a030902110729g30b30d46gac0f56205f8d48cf@mail.gmail.com> Message-ID: <4993D8CA.1010602@ewetel.de> Hi Anthony, hm... on centos5 it works fine. No problems, no warning, no crash. regards Helmut On 11.02.2009 16:29, Anthony Minessale wrote: > I am highly suspicious of the ubuntu. > you are using a prerelease of gcc that we have already found at least > 1 bug. > > we tried the file on our box and it doesn't even say anything about > the file being bad etc...... it plays and hangs up fine even 4 times > at once. > It would be a big help if you could try to reproduce it on CentOS 5 as > a comparison. We have had 3 cases this week where doing so has fixed > problems and i don't want to believe it so I would appropriate it if > you could test it. From helmut.kuper at ewetel.de Thu Feb 12 00:34:21 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Thu, 12 Feb 2009 09:34:21 +0100 Subject: [Freeswitch-users] Question: SIP BYE authentication In-Reply-To: <4992C772.4090906@ewetel.de> References: <4992C772.4090906@ewetel.de> Message-ID: <4993DF0D.9000403@ewetel.de> Hi, any ideas how to get FS's BYEs authenticated ? On 11.02.2009 13:41, Helmut Kuper wrote: > Hello, > > my FS is connected to my SIP-DDI softswitch, which requires all SIP > requests sent by a registered SIP account to be authenticated. I found > that when FS sends a BYE FreeSWITCH ignores the authentication > challenge (SIP/2.0 407) received from proxy and simply terminates the > session. > > Is there a way to configure FS in that way that it react on auth > challenges for BYEs ? > > regards > Helmut > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > From thomas.mangin at exa-networks.co.uk Thu Feb 12 00:59:33 2009 From: thomas.mangin at exa-networks.co.uk (Thomas Mangin) Date: Thu, 12 Feb 2009 08:59:33 +0000 Subject: [Freeswitch-users] FreeSWITCH VPSs In-Reply-To: <4285D10F-3B87-47B4-979A-14C3066B14E0@freeswitch.org> References: <92e7d2090902111434l60afd3a8i634b4df688ed4011@mail.gmail.com> <40CC1F17-A4C3-4CA6-8DA0-C8B2A5346CCD@freeswitch.org> <92e7d2090902111447v1cbb00c0l8fc9f1c8b0c1a514@mail.gmail.com> <4285D10F-3B87-47B4-979A-14C3066B14E0@freeswitch.org> Message-ID: <9C725DB2-CD87-49A3-998F-74B1A9030708@exa-networks.co.uk> Hello Brian / Everyone, Like Nik and Nicolas, I created a openvz box to test things in a 'near production' environment. The box does only take 'test' calls, ie it never saw more that a few calls at a time. The design was 100% openser/opensips/kamailio but I since replaced the pstn gateways with FS and it has been working perfectly and I am not looking back. I am now planning to use freeswitch as the registrar/voicemail/media servers to only keep openser as a proxy inbound proxy (as it is possible to program it to assign a RTP proxies topologically near the caller and you can use it to fix some really broken sip packets - like LLU operators cheap DSL routers badly NAT fixing the contact header). Following this thread I am wondering if I should/could expect some issues with my setup which is 32 bits or if your comments are only related to the performance/behaviour of FS once under load (in which case I need not to worry). voip-master:~# uname -a Linux voip-master 2.6.18-ovz-028stab053.5-smp #1 SMP Sat Mar 1 12:19:31 CET 2008 i686 GNU/Linux voip-master:~# vzlist VEID NPROC STATUS IP_ADDR HOSTNAME 1001 24 running A.B.C.A proxy1.sip (openser phone outbound proxy - accept REGISTER - range locked) 1002 24 running A.B.C.B in1.sip (openser incoming calls from the net - enum, no REGISTER - open) 1003 19 running A.B.C.C out1.sip (openser outgoing calls to the net - enum - to be FS) 1004 4 running A.B.C.D rtp1.nat (rtpproxy nat) 1005 3 running A.B.C.E media1 (was sems for voicemail/media) 1006 29 running A.B.C.F database1 1107 23 running A.B.C.G registrar1.sip (openser) 1108 26 running A.B.C.H registrar2.sip (FS) 1109 8 running A.B.C.I ns1(auth DNS for the tested zone with ENUM info) 1110 8 running A.B.C.J internal1.cache (cache with internal ENUM routing) 1111 8 running A.B.C.K external1.cache (normal DNS cache) 1112 21 running A.B.C.L pstn-out-1 (FS gateway out to pstn) 1113 21 running A.B.C.M pstn-in-1 (FS gateway in from pstn) voip-master:~# cat /proc/cpuinfo | grep model model : 4 model name : Intel(R) Xeon(TM) CPU 3.00GHz model : 4 model name : Intel(R) Xeon(TM) CPU 3.00GHz voip-master:~# free -m total used free shared buffers cached Mem: 2023 1902 120 0 431 1009 -/+ buffers/cache: 460 1562 Swap: 2588 3 2585 yep, memory is short :p) Regards, Thomas On 11 Feb 2009, at 22:54, Brian West wrote: > Your VE must be 64bit also. > > http://wiki.openvz.org/Install_OpenVZ_on_a_x86_64_system_Centos-Fedora > > If you need the set util listed on that page let me know I have a > copy of it. > > /b > > On Feb 11, 2009, at 4:47 PM, Nik Martin wrote: > >> I think the VE I've built is too, but uname is a bit cryptic: >> 2.6.18-92.1.18.el5.028stab060.2 #1 SMP Tue Jan 13 11:38:36 MSK 2009 >> i686 i686 i386 GNU/Linux >> >> I can easily change it if FS will run better. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From pbd at suspiria.net Thu Feb 12 01:23:18 2009 From: pbd at suspiria.net (Public Dump) Date: Thu, 12 Feb 2009 10:23:18 +0100 Subject: [Freeswitch-users] High CPU load after starting (Brian West) In-Reply-To: References: Message-ID: <13C421883438EB42B9E2C30069FD4AB767BF49AF15@crushinator.central.local> > OK does it work now? We have tested this on various windows installs > among the team here and not seeing this issue... it was a known issue > back in Nov. or Dec. but thats long been fixed. No, the problem is still there. I have tested it on a Core AMD 32bit AMD machine = everything is fine. On a 64bit 4 Core Intel Xeon machine = Problem is there. From r.pankratz at fh-wolfenbuettel.de Thu Feb 12 03:05:46 2009 From: r.pankratz at fh-wolfenbuettel.de (Rene Pankratz) Date: Thu, 12 Feb 2009 12:05:46 +0100 Subject: [Freeswitch-users] Change registration information for SIP-Registrar via XML/RPC? Message-ID: <4994028A.9090005@fh-wolfenbuettel.de> Hello, we want to use mod_pa as a softphone, that registers to a SIPregistrar. But the username and password need to be changed over time without restarting freeswitch. Currently we are using XML/RPC to control the call functions. So it would be best (if possible) to use it also for changing registration information. Is there any way to do this? Thanks in advance Ren? From helmut.kuper at ewetel.de Thu Feb 12 05:03:32 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Thu, 12 Feb 2009 14:03:32 +0100 Subject: [Freeswitch-users] mod_openzap stops working after some calls Update In-Reply-To: <87f2f3b90902110943q12c550e4qf8e6ecae1ecc6bec@mail.gmail.com> References: <498C2DC3.40701@ewetel.de> <191c3a030902060802r424f8a22tb4887dff7c6aead5@mail.gmail.com> <49907327.6010703@ewetel.de> <7CEFE4F8-7461-4DF3-ABE0-C0B818466AD0@jerris.com> <4990789B.40405@ewetel.de> <4992F295.4070809@ewetel.de> <87f2f3b90902110943q12c550e4qf8e6ecae1ecc6bec@mail.gmail.com> Message-ID: <49941E24.2070002@ewetel.de> Hi Mike, at least for incoming calls this shouldn't be too brutal, cause far end seems to know that the channel should be free otherwise it never would allocate it. By now the hack works at least for me quite good. Nobody from AVAYA side moaned about it, yet. But I have to wait one or two further days to be sure ... I guess I have to talk to stkn in irc to get an idea how long I have to use it. regards helmut On 11.02.2009 18:43, Michael Collins wrote: >> This is just a quite brutal hack because it assumes that the remote TDM >> end is able to recover channels as well. >> > > Could you possibly modify your hack to be less brutal? For example, > could it send a STATUS ENQ to the far end for the channel in question? > Just curious. > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090212/02632a8c/attachment-0002.html From Thomas.Sluschny at siemens.com Thu Feb 12 06:00:51 2009 From: Thomas.Sluschny at siemens.com (Sluschny, Thomas) Date: Thu, 12 Feb 2009 15:00:51 +0100 Subject: [Freeswitch-users] stream a file multicast with mod_esf Message-ID: Hi, i want to stream a file per IP multicast with mod_esf. I can stream IP multicast with: pa call stream XML and in XML dialplan: and i can also play files with 'playback' app, BUT: how can put these 2 things together? May be its trivial, but i cant get it make working with 'originate' or 'uuid_broadcast' or 'bridge', 'transfer' and so on ... Thanks in advance, Thomas -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090212/8a8320b8/attachment-0002.html From pauld at versafon.com Thu Feb 12 06:08:58 2009 From: pauld at versafon.com (pauld) Date: Thu, 12 Feb 2009 09:08:58 -0500 Subject: [Freeswitch-users] Cannot choose Cepstral voice from dialplan In-Reply-To: <10703C0D-0BA2-4FB3-B669-B87BF93912E0@freeswitch.org> References: <498E67BF.3060207@versafon.com> <8B964172-6EEB-4B5B-B4B1-D35BFEAEB460@freeswitch.org> <498F6751.2000701@versafon.com> <87f2f3b90902090904i72e37241hb5d4808eb31fcc18@mail.gmail.com> <4993A7D8.1090004@versafon.com> <10703C0D-0BA2-4FB3-B669-B87BF93912E0@freeswitch.org> Message-ID: <49942D7A.2090503@versafon.com> Yes I am using 5.1, I haven't done anything special other than followed wiki and then the advice given here to create symlinks in FS lib dir to all cepstral libs. I have cepstral libs in a standard location /opt/swift/lib. I have given an example extension I used for testing earlier in this thread. Brian West wrote: > This is normal. Are you using 5.0? can you include examples of how > you're doing this? > > /b > > On Feb 11, 2009, at 10:38 PM, pauld wrote: > > >> "TRANSCODING_NECESSARY" >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From freeswitch at servercorps.com Thu Feb 12 07:01:40 2009 From: freeswitch at servercorps.com (Nik Martin) Date: Thu, 12 Feb 2009 09:01:40 -0600 Subject: [Freeswitch-users] FreeSWITCH VPSs In-Reply-To: <191c3a030902112048l1d362f79nf2d0fbb65f2c4984@mail.gmail.com> References: <92e7d2090902111434l60afd3a8i634b4df688ed4011@mail.gmail.com> <9dc4a1670902111608h4d8ca43eq85800ee902113a1a@mail.gmail.com> <28F5713C-838B-409A-897B-F9E40D39B928@freeswitch.org> <9dc4a1670902111624s456b7fbeg99c1e078fe8def3d@mail.gmail.com> <76543A10-18D7-450D-ACA1-E4CAE64F471F@freeswitch.org> <92e7d2090902111759g629ca736k879103144d012155@mail.gmail.com> <191c3a030902112048l1d362f79nf2d0fbb65f2c4984@mail.gmail.com> Message-ID: <92e7d2090902120701y15b0fffej34bb8c46c7aca0b1@mail.gmail.com> I didn't mean to touch a nerve there, I think you mis-interpreted 100% of my original post. You may also have me confused with some other Nik, as I contribute as much or more than I request . There is a nother Nik on here that are probably referring to. I am nikko from #freeswitch. I have made many edits and contributions to the wiki, patches and verified bug reports submitted to Jira. See my other responses inline. On Wed, Feb 11, 2009 at 10:48 PM, Anthony Minessale wrote: > cool, > > However.... you say... > > "but a platform for sandbox testing of configurations, SOHO VOIP > Switches, development of FS addons, backup switch capability, etc. > Doing this stuff at home behind NAT and a consumer grade router is one > reason Brian, Anthony, Mike, et al. are half crazy. " > > First of all, you have no idea on what platform and where we do our > development. > If you only saw how many terminals to random servers spanning the globe we > have open...... > You have the NAT part right, if you knew the hours it took from behind NAT > to get all the code right for SIP interop or even had to deal with half the > bullshit it takes to get SIP working, you'd be in the madhouse so don't you > dare try using us as an ad slogan. I was adressing all the support hours you and others waste trying to help people get FS running on their home servers behind NAT, when they should be testing in an environment that more closely matches what their production one will be. > > Secondly, we are not half crazy, we are completely crazy and most of it > comes from spending all day > on this list tending to your never-ending threads while trying to help the > other people in the community who actually give something back. > > If you want to use our list to advertise this service maybe you should find > a way to contribute to the project rather than constantly asking for help in > 5 separate emails in one day followed by another thread trying to sell > something. > Sorry, you are mistaking me with another Nik. I'm nikko from #freeswitch, and contribute PLENTY. > A reasonable cost would be FREE just like everything else around here. I'm just trying to cover costs, and suport a single NOC engineer, and get more people to adopt FreeSWITCH. If people like Paige and others that come and go had a reasonable environment to test and configure in, they would not be saying crap like "this works in asterisk, blah blah blah". Again, sorry to have upset you, but we had an email conversation an few days ago, and you and Brian were cool with my plans. > > > On Wed, Feb 11, 2009 at 7:59 PM, Nik Martin > wrote: >> >> My goal is obviously not to provide carrier grade VOIP switch service, >> but a platform for sandbox testing of configurations, SOHO VOIP >> Switches, development of FS addons, backup switch capability, etc. >> Doing this stuff at home behind NAT and a consumer grade router is one >> reason Brian, Anthony, Mike, et al. are half crazy. I run my >> company's phone switch in a 32 bit OpenVZ VE with 256 Mb ram, and have >> no issues. When I goof around trying to transcode between 8 and 16 >> bit codecs and whatnot, sure it gets tight, but FS on an idle system >> keeps 23 mb of of ram resident, and rarely if ever hits a 256 mb bean >> counter (limit). >> >> Also, these limits are not hard, I just know what my hardware has, and >> am trying to offer as much value as I can for what I have in the >> systems. 128 Gb of ECC ram and Quad/Quad Core zeons are still pretty >> pricey! >> >> Nik >> >> >> >> >> On Wed, Feb 11, 2009 at 6:28 PM, Brian West wrote: >> > Actually you can if you don't overload the machine like most VPS >> > providers do... The advantage with OpenVZ in this case is that you can >> > migrate the running FreeSWITCH instance between hardware nodes and not >> > drop calls at this size. >> > >> > /b >> > >> > On Feb 11, 2009, at 6:24 PM, EdPimentl wrote: >> > >> >> Soho,,, yes of course... >> >> Voip (soho)Service Provider.... not convinced is possible to provide >> >> reliable QoS. >> >> My .02 cents >> >> -E >> > >> > >> > _______________________________________________ >> > Freeswitch-users mailing list >> > Freeswitch-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From brian at freeswitch.org Thu Feb 12 07:07:36 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 12 Feb 2009 09:07:36 -0600 Subject: [Freeswitch-users] stream a file multicast with mod_esf In-Reply-To: References: Message-ID: <6F68FEDB-8976-44D4-9A67-F871EC7D5120@freeswitch.org> esf is for multi cast paging... it currently won't let you play files... we would have to create a multicast playback application. /b On Feb 12, 2009, at 8:00 AM, Sluschny, Thomas wrote: > Hi, > > i want to stream a file per IP multicast with mod_esf. > > I can stream IP multicast with: > pa call stream XML > and in XML dialplan: > > > > > > > > and i can also play files with 'playback' app, > > BUT: how can put these 2 things together? > > May be its trivial, but i cant get it make working with 'originate' > or 'uuid_broadcast' or 'bridge', 'transfer' and so on ... > > Thanks in advance, > Thomas -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090212/dd1bef26/attachment-0002.html From brian at freeswitch.org Thu Feb 12 07:08:04 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 12 Feb 2009 09:08:04 -0600 Subject: [Freeswitch-users] Cannot choose Cepstral voice from dialplan In-Reply-To: <49942D7A.2090503@versafon.com> References: <498E67BF.3060207@versafon.com> <8B964172-6EEB-4B5B-B4B1-D35BFEAEB460@freeswitch.org> <498F6751.2000701@versafon.com> <87f2f3b90902090904i72e37241hb5d4808eb31fcc18@mail.gmail.com> <4993A7D8.1090004@versafon.com> <10703C0D-0BA2-4FB3-B669-B87BF93912E0@freeswitch.org> <49942D7A.2090503@versafon.com> Message-ID: <9268AE39-6841-4819-9A61-37806B48BEFF@freeswitch.org> You still didn't answer my question. How are you trying to do this from the dialplan. /b On Feb 12, 2009, at 8:08 AM, pauld wrote: > Yes I am using 5.1, I haven't done anything special other than > followed > wiki and then the advice given here to create symlinks in FS lib dir > to all > cepstral libs. I have cepstral libs in a standard location /opt/ > swift/lib. > I have given an example extension I used for testing earlier in this > thread From brian at freeswitch.org Thu Feb 12 07:10:51 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 12 Feb 2009 09:10:51 -0600 Subject: [Freeswitch-users] Change registration information for SIP-Registrar via XML/RPC? In-Reply-To: <4994028A.9090005@fh-wolfenbuettel.de> References: <4994028A.9090005@fh-wolfenbuettel.de> Message-ID: <2BCCBF51-82DF-4F88-AABB-8C4A44480D3B@freeswitch.org> You could store the data in globals and then restart the profiles via XML PRC. ie global_setvar, reloadxml, sofia profile blah restart. /b On Feb 12, 2009, at 5:05 AM, Rene Pankratz wrote: > Hello, > we want to use mod_pa as a softphone, that registers to a > SIPregistrar. > But the username and password need to be changed over time without > restarting freeswitch. > Currently we are using XML/RPC to control the call functions. So it > would be best (if possible) to use it also for changing registration > information. Is there any way to do this? > > Thanks in advance > Ren? From brian at freeswitch.org Thu Feb 12 07:11:12 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 12 Feb 2009 09:11:12 -0600 Subject: [Freeswitch-users] Question: SIP BYE authentication In-Reply-To: <4993DF0D.9000403@ewetel.de> References: <4992C772.4090906@ewetel.de> <4993DF0D.9000403@ewetel.de> Message-ID: <3154F1F1-9286-42BC-9922-1675E004E4A1@freeswitch.org> Are you calling via a gateway? /b On Feb 12, 2009, at 2:34 AM, Helmut Kuper wrote: > Hi, > > any ideas how to get FS's BYEs authenticated ? From anthony.minessale at gmail.com Thu Feb 12 07:29:03 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 12 Feb 2009 09:29:03 -0600 Subject: [Freeswitch-users] Socket-Event on "originate call"? In-Reply-To: <5e414ed0902112339o6c1c499cxe30d0e255d6756f0@mail.gmail.com> References: <5e414ed0902090716y10f95f71o5760dfdf50d1c5cb@mail.gmail.com> <191c3a030902090727n44eef7d7x83704b66e628081@mail.gmail.com> <5e414ed0902100716s2c053a48w2174f7587dd0641e@mail.gmail.com> <5e414ed0902111037q2a6f4097sebd08cb712f042bf@mail.gmail.com> <397EFED1-22E6-4B78-BDE4-3AE5F4204E5B@freeswitch.org> <5e414ed0902111211y3456d5eaqc99baa126cb9dcad@mail.gmail.com> <5e414ed0902112339o6c1c499cxe30d0e255d6756f0@mail.gmail.com> Message-ID: <191c3a030902120729p6da3868eo99a05a5152fd5fce@mail.gmail.com> No, I have not made any changes to reflect anything you asked about. instant_ringback=true is designed to send artificial ringback to the a leg while it's executing the bridge app. it will be meaningless to you if you do not use it with the bridge application On Thu, Feb 12, 2009 at 1:39 AM, Dennis wrote: > the problem is fixed in the latest version of fs - at least it is > working as before without any errors. > > but there is still the question, if the changes where made because of > our problem with the not starting socket!? > we can see in the cli, that the var is set, but it does not change > anything regarding our problem. > > > > 2009/2/11 Brian West : > > Please collect the backtrace and report it on Jira. > > > > /b > > > > On Feb 11, 2009, at 2:11 PM, Dennis wrote: > > > >> this does not help. we are using socket outbound and everything worked > >> before the changes yesterday. > >> > >> we have the same error with other dialplans. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090212/933b9df0/attachment-0002.html From ivdreg at gmail.com Thu Feb 12 06:06:16 2009 From: ivdreg at gmail.com (ivdreg ivdreg) Date: Thu, 12 Feb 2009 16:06:16 +0200 Subject: [Freeswitch-users] Codec negotiation questions Message-ID: Hi all, Can I ask 2 questions about codec negotiation: 1. Is it possible Freeswitch to work negotiate codecs between two phones as it is described below. INVITE from A with some codecs in SDP ---> Freeswitch rewrites codec preference according absolute_codec_string but exclude all codecs not offered by A ----> INVITE to B with rewrited SDP. example: from A SDP:PCMA,PCMU,SPEEX ----> absolute_codec_string=G722,PCMU,PCMA,GSM ----> to B SDP: PCMU,PCMA 2. Can I get codec list in INVITE with mod_perl for example or via xml_curl without processing SDP variable (switch_r_sdp). It will be useful to be in format that absolute_codec_string variable takes. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090212/a04dd7f3/attachment-0002.html From codecomplete at free.fr Thu Feb 12 06:11:40 2009 From: codecomplete at free.fr (Fred) Date: Thu, 12 Feb 2009 15:11:40 +0100 Subject: [Freeswitch-users] Switching from Asterisk to Freeswitch? Message-ID: <7.0.1.0.2.20090212094510.02496ca0@fredshack.com> Hello I successfully used Asterisk to build a voice server for our SOHO business. I did read the article comparing Asterisk to Freeswitch, but I have a couple of questions: 1. What are the decisive reasons that would justify taking a look at Freeswitch? What makes it a better option? 2. I'd like to build an affordable solution based on Asus' EeeBox and (because it's too small to add a PCI card) Sangoma's USB device to connect the host to POTS. Has someone successfully used Freeswitch to work on this hardware? www.asus.com/products.aspx?l1=24&l2=165 http://wiki.sangoma.com/sangoma-wanpipe-usbfxo Thank you for your feedback. From Thomas.Sluschny at siemens.com Thu Feb 12 08:12:26 2009 From: Thomas.Sluschny at siemens.com (Sluschny, Thomas) Date: Thu, 12 Feb 2009 17:12:26 +0100 Subject: [Freeswitch-users] stream a file multicast with mod_esf In-Reply-To: References: Message-ID: Hi Brian, i thought if i can stream from portaudio it is almost the same with streaming from file, so it should working already now. Is this not the design idea of channels and media to do so? regards, thomas PS: sry for improper formatted mail, i cant reply at the moment and have to copy mail from archive :( ________________________________ >From brian at freeswitch.org Thu Feb 12 07:07:36 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 12 Feb 2009 09:07:36 -0600 Subject: [Freeswitch-users] stream a file multicast with mod_esf Message-ID: <6F68FEDB-8976-44D4-9A67-F871EC7D5120 at freeswitch.org > esf is for multi cast paging... it currently won't let you play files... we would have to create a multicast playback application. /b On Feb 12, 2009, at 8:00 AM, Sluschny, Thomas wrote: > Hi, > > i want to stream a file per IP multicast with mod_esf. > > I can stream IP multicast with: > pa call stream XML > and in XML dialplan: > > > > > > > > and i can also play files with 'playback' app, > > BUT: how can put these 2 things together? > > May be its trivial, but i cant get it make working with 'originate' > or 'uuid_broadcast' or 'bridge', 'transfer' and so on ... > > Thanks in advance, > Thomas -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090212/7538c00b/attachment-0002.html From brian at freeswitch.org Thu Feb 12 08:25:15 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 12 Feb 2009 10:25:15 -0600 Subject: [Freeswitch-users] stream a file multicast with mod_esf In-Reply-To: References: Message-ID: <7C996136-963A-42D2-B0FE-D2729F968E52@freeswitch.org> You could but I think you want to stream RTP to a multicast it would be better off building an rtp format mod so you can record rtp:// x.x.x.x:5000 and play from rtp://y.y.y.y:5000 /b On Feb 12, 2009, at 10:12 AM, Sluschny, Thomas wrote: > Hi Brian, > > i thought if i can stream from portaudio it is almost the same with > streaming from file, > so it should working already now. > Is this not the design idea of channels and media to do so? > > regards, > thomas > > PS: sry for improper formatted mail, i cant reply at the moment and > have to copy mail from archive :( -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090212/4c3f4a37/attachment-0002.html From anthony.minessale at gmail.com Thu Feb 12 08:52:10 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 12 Feb 2009 10:52:10 -0600 Subject: [Freeswitch-users] Switching from Asterisk to Freeswitch? In-Reply-To: <7.0.1.0.2.20090212094510.02496ca0@fredshack.com> References: <7.0.1.0.2.20090212094510.02496ca0@fredshack.com> Message-ID: <191c3a030902120852mfef78b1sef49eaec55277e94@mail.gmail.com> On Thu, Feb 12, 2009 at 8:11 AM, Fred wrote: > Hello > > I successfully used Asterisk to build a voice server for our SOHO > business. I did read the article comparing Asterisk to Freeswitch, > but I have a couple of questions: > > 1. What are the decisive reasons that would justify taking a look at > Freeswitch? What makes it a better option? > It's sort of a loaded question because we are likely to prefer FS having worked on it for some years. But anyway, since you asked, I prefer FS because it actually lets you do thing things you dreamed of doing when you first try Asterisk. Asterisk is somewhat like a mirage where you tend to see a pool of water and end up jumping into a sand pit. Asterisk doesn't lack at all in inspiration and possibilities but every time I tried to make something from Asterisk I ended up with a mouth full of sand. Keep in mind I spent 3 years as a core developer in Asterisk doing my best to contribute to its success so it was a pretty big challenge to have to start over with all that functionality right at my fingertips and put up with the claims it was "vaporware" but in 3 short years we have all the functionality back and it's more scalable and is reaching towards the future by supporting things like wideband and ultra wide band audio, resampling and im integration. The short answer to the question is because we are all perfectionists. > > 2. I'd like to build an affordable solution based on Asus' EeeBox and > (because it's too small to add a PCI card) Sangoma's USB device to > connect the host to POTS. Has someone successfully used Freeswitch to > work on this hardware? > www.asus.com/products.aspx?l1=24&l2=165 > http://wiki.sangoma.com/sangoma-wanpipe-usbfxo > > Thank you for your feedback. > We support Sangoma hardware so i am sure if it is not currently supported it will be in the near future. Sangoma has been a big proponent to FreeSWITCH and we have worked very closely over the years. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090212/3dbf297b/attachment-0002.html From mike at jerris.com Thu Feb 12 09:01:24 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 12 Feb 2009 12:01:24 -0500 Subject: [Freeswitch-users] Question: SIP BYE authentication In-Reply-To: <4993DF0D.9000403@ewetel.de> References: <4992C772.4090906@ewetel.de> <4993DF0D.9000403@ewetel.de> Message-ID: If using gayeway it should already do this. On Feb 12, 2009, at 3:34 AM, Helmut Kuper wrote: > Hi, > > any ideas how to get FS's BYEs authenticated ? > > On 11.02.2009 13:41, Helmut Kuper wrote: >> Hello, >> >> my FS is connected to my SIP-DDI softswitch, which requires all SIP >> requests sent by a registered SIP account to be authenticated. I >> found >> that when FS sends a BYE FreeSWITCH ignores the authentication >> challenge (SIP/2.0 407) received from proxy and simply terminates the >> session. >> >> Is there a way to configure FS in that way that it react on auth >> challenges for BYEs ? >> >> regards >> Helmut >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mike at jerris.com Thu Feb 12 09:02:39 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 12 Feb 2009 12:02:39 -0500 Subject: [Freeswitch-users] High CPU load after starting (Brian West) In-Reply-To: <13C421883438EB42B9E2C30069FD4AB767BF49AF15@crushinator.central.local> References: <13C421883438EB42B9E2C30069FD4AB767BF49AF15@crushinator.central.local> Message-ID: Is this running on 64 bit os or 32? On Feb 12, 2009, at 4:23 AM, Public Dump wrote: >> OK does it work now? We have tested this on various windows installs >> among the team here and not seeing this issue... it was a known issue >> back in Nov. or Dec. but thats long been fixed. > > No, the problem is still there. > > I have tested it on a Core AMD 32bit AMD machine = everything is fine. > On a 64bit 4 Core Intel Xeon machine = Problem is there. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mike at jerris.com Thu Feb 12 09:04:31 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 12 Feb 2009 12:04:31 -0500 Subject: [Freeswitch-users] Change registration information for SIP-Registrar via XML/RPC? In-Reply-To: <4994028A.9090005@fh-wolfenbuettel.de> References: <4994028A.9090005@fh-wolfenbuettel.de> Message-ID: You can change the config files on disk and then issue reloadxml or use mod_XML_curl Mike On Feb 12, 2009, at 6:05 AM, Rene Pankratz wrote: > Hello, > we want to use mod_pa as a softphone, that registers to a > SIPregistrar. > But the username and password need to be changed over time without > restarting freeswitch. > Currently we are using XML/RPC to control the call functions. So it > would be best (if possible) to use it also for changing registration > information. Is there any way to do this? > > Thanks in advance > Ren? > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From msc at freeswitch.org Thu Feb 12 09:03:23 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 12 Feb 2009 09:03:23 -0800 Subject: [Freeswitch-users] Switching from Asterisk to Freeswitch? In-Reply-To: <7.0.1.0.2.20090212094510.02496ca0@fredshack.com> References: <7.0.1.0.2.20090212094510.02496ca0@fredshack.com> Message-ID: <87f2f3b90902120903r6d71a685i89a1ea81a9203960@mail.gmail.com> On Thu, Feb 12, 2009 at 6:11 AM, Fred wrote: > Hello > > I successfully used Asterisk to build a voice server for our SOHO > business. I did read the article comparing Asterisk to Freeswitch, > but I have a couple of questions: > > 1. What are the decisive reasons that would justify taking a look at > Freeswitch? What makes it a better option? The answer is, of course, "It depends." If all you need is a simple PBX for your small office then Asterisk actually isn't necessarily a bad choice. My personal experience from talking to people is that when it works, it works well. As to the question of what makes FS and better option than Asterisk it goes back to what you want to use it for. But in a nutshell FreeSWITCH does a lot of things better: FS is more scalable FS is more modular FS is much better written in terms of code quality - no voodoo > > 2. I'd like to build an affordable solution based on Asus' EeeBox and > (because it's too small to add a PCI card) Sangoma's USB device to > connect the host to POTS. Has someone successfully used Freeswitch to > work on this hardware? > www.asus.com/products.aspx?l1=24&l2=165 > http://wiki.sangoma.com/sangoma-wanpipe-usbfxo I personally have not but keep asking around. -MC From nik.middleton at noblesolutions.co.uk Thu Feb 12 09:30:39 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Thu, 12 Feb 2009 17:30:39 -0000 Subject: [Freeswitch-users] Call accounting not working as expected In-Reply-To: <87f2f3b90902111709t2c6db8e2jf190b799eeca6ee8@mail.gmail.com> References: <87f2f3b90902111709t2c6db8e2jf190b799eeca6ee8@mail.gmail.com> Message-ID: Bang on, Thanks -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 12 February 2009 01:10 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Call accounting not working as expected > It is kind of being processed, the account code is being set, XML cdr's are > created and are correct, but csv cdr's for the account code are not > > > > Caller ID is not being set in the A leg but is in the B Leg DING DING DING!!! We have a weener! Okay, that was the key piece of info. Most likely you are logging only the A leg in the CSV CDRs. Go to conf/autoload_configs/cdr_csv.conf.xml and look for these two lines: Most likely you need to use "b" or "ab" depending on your scenario. Try it each way and see how you like the results, then please report back. Thanks! -MC _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From ajlong at worldlink.net Thu Feb 12 09:46:57 2009 From: ajlong at worldlink.net (Adam Long) Date: Thu, 12 Feb 2009 12:46:57 -0500 Subject: [Freeswitch-users] RFC 4497 Originate Timeout / Progress Timeout .. No 100 Trying ... triggering 480 Response Code??? Message-ID: <018901c98d39$e6bd7cc0$b4387640$@net> Hi Guys, I've been experimenting with originate_timeout and progress_timeout as follows. However, shouldn't the timeout trigger a 408 Request Timeout instead of 480 Temporary Failure if no Provisional response received? Just curious, it seems to make sense to me.. but maybe SIP gods see differently. I have also tried using ${originate_disposition} after both bridge attempts to fetch the timeout disposition but instead this is set to NO_ANSWER (which would be correct for first attempt) As I understand it originate_disposition is reset for each bridge completed either successfully or unsuccessfully. Shouldn't the second attempt with no 100 Trying ever received trigger a NO_USER_REPONSE on timeout? According to RFC 4497 that would map to 408 Request Timeout For this test (please note progress_timeout set to low "2" value to test timeout) Node 10.200.1.11 is setup in such a way it responds with 100 Trying but never reaches 180 or 183 before 2 sec timer expires (as desired for this test) Node 10.200.1.12 (is disconnected and never even sends a provisional response, as desired) I have tried.
As well as .
Any thoughts, am I completely nuts and missing something in the spec? Regards, -Adam -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090212/b286db1e/attachment-0002.html From nik.middleton at noblesolutions.co.uk Thu Feb 12 09:51:14 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Thu, 12 Feb 2009 17:51:14 -0000 Subject: [Freeswitch-users] FS equiv for waitforextension Message-ID: HI, Is there an equivalent function in FS to waitforexten ? Closest I've seen is collectInput? Right now I'm using stream file, which is ok if they hit a digit before stream ends, but I want them to have a certain period after the file is played to hit a button. Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090212/1a32b4e9/attachment-0002.html From brian at freeswitch.org Thu Feb 12 09:55:10 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 12 Feb 2009 11:55:10 -0600 Subject: [Freeswitch-users] FS equiv for waitforextension In-Reply-To: References: Message-ID: <65691A20-8D66-43A0-A859-69AB7DE3056A@freeswitch.org> Dialplan or language method...btw if you're on IRC its better to ask there.. faster response... ;) /b On Feb 12, 2009, at 11:51 AM, Nik Middleton wrote: > HI, > > Is there an equivalent function in FS to waitforexten ? Closest > I?ve seen is collectInput? > > Right now I?m using stream file, which is ok if they hit a digit > before stream ends, but I want them to have a certain period after > the file is played to hit a button. > > Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090212/ef1d97a6/attachment-0002.html From sicfslist at gmail.com Thu Feb 12 10:01:12 2009 From: sicfslist at gmail.com (Shelby Ramsey) Date: Thu, 12 Feb 2009 12:01:12 -0600 Subject: [Freeswitch-users] FS equiv for waitforextension In-Reply-To: References: Message-ID: <35b355e90902121001i3a049252x852336b18846ad21@mail.gmail.com> Nik, I'm not sure if this is the right way ... but I use application="read" data="0 1 /path/silence.wav var 1000 # I'm sure there is a better way ... but this seems to work for me. SDR On Thu, Feb 12, 2009 at 11:51 AM, Nik Middleton < nik.middleton at noblesolutions.co.uk> wrote: > HI, > > > > Is there an equivalent function in FS to waitforexten ? Closest I've seen > is collectInput? > > > > Right now I'm using stream file, which is ok if they hit a digit before > stream ends, but I want them to have a certain period after the file is > played to hit a button. > > > > Regards, > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090212/92f65b4a/attachment-0002.html From brian at freeswitch.org Thu Feb 12 10:07:40 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 12 Feb 2009 12:07:40 -0600 Subject: [Freeswitch-users] FS equiv for waitforextension In-Reply-To: <35b355e90902121001i3a049252x852336b18846ad21@mail.gmail.com> References: <35b355e90902121001i3a049252x852336b18846ad21@mail.gmail.com> Message-ID: <76D5D669-B367-4639-925F-DA213FC93539@freeswitch.org> Dialplan isn't for writing IVR's... doing so is against the design of FreeSWITCH.. you can do simple things in dialplan but more complex stuff needs to be in a language. /b On Feb 12, 2009, at 12:01 PM, Shelby Ramsey wrote: > Nik, > > I'm not sure if this is the right way ... but I use > application="read" data="0 1 /path/silence.wav var 1000 # > > I'm sure there is a better way ... but this seems to work for me. > > SDR From msc at freeswitch.org Thu Feb 12 10:14:56 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 12 Feb 2009 10:14:56 -0800 Subject: [Freeswitch-users] FS equiv for waitforextension In-Reply-To: <76D5D669-B367-4639-925F-DA213FC93539@freeswitch.org> References: <35b355e90902121001i3a049252x852336b18846ad21@mail.gmail.com> <76D5D669-B367-4639-925F-DA213FC93539@freeswitch.org> Message-ID: <87f2f3b90902121014t29ab0e25u129c4c824d63e11e@mail.gmail.com> On Thu, Feb 12, 2009 at 10:07 AM, Brian West wrote: > Dialplan isn't for writing IVR's... doing so is against the design of > FreeSWITCH.. you can do simple things in dialplan but more complex > stuff needs to be in a language. Or create an IVR and send the call there from the dialplan. You can do IVRs in Lua/JS/Perl or in XML. -MC From nik.middleton at noblesolutions.co.uk Thu Feb 12 10:15:11 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Thu, 12 Feb 2009 18:15:11 -0000 Subject: [Freeswitch-users] FS equiv for waitforextension In-Reply-To: <76D5D669-B367-4639-925F-DA213FC93539@freeswitch.org> References: <35b355e90902121001i3a049252x852336b18846ad21@mail.gmail.com> <76D5D669-B367-4639-925F-DA213FC93539@freeswitch.org> Message-ID: Sorry, should have said this was in js Regards, -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: 12 February 2009 18:08 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] FS equiv for waitforextension Dialplan isn't for writing IVR's... doing so is against the design of FreeSWITCH.. you can do simple things in dialplan but more complex stuff needs to be in a language. /b On Feb 12, 2009, at 12:01 PM, Shelby Ramsey wrote: > Nik, > > I'm not sure if this is the right way ... but I use > application="read" data="0 1 /path/silence.wav var 1000 # > > I'm sure there is a better way ... but this seems to work for me. > > SDR _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From pbd at suspiria.net Thu Feb 12 10:49:44 2009 From: pbd at suspiria.net (Public Dump) Date: Thu, 12 Feb 2009 19:49:44 +0100 Subject: [Freeswitch-users] Freeswitch-users Digest, Vol 32, Issue 108 In-Reply-To: References: Message-ID: <13C421883438EB42B9E2C30069FD4AB76AEA2B386F@crushinator.central.local> > > Is this running on 64 bit os or 32? A 64bit , Windows 2008 Server. From nik.middleton at noblesolutions.co.uk Thu Feb 12 11:36:52 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Thu, 12 Feb 2009 19:36:52 -0000 Subject: [Freeswitch-users] js and VMD Message-ID: Hi Guys, I'm trying to get VMD running in js, does anyone have an example of how it's called? If I try session:execute("vmd"); I get an error Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090212/0ebc0032/attachment-0002.html From msc at freeswitch.org Thu Feb 12 12:02:24 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 12 Feb 2009 12:02:24 -0800 Subject: [Freeswitch-users] FS + Call Center Solution In-Reply-To: <6309E7515E4F43159B9564800564B562@SaeedLaptop> References: <6309E7515E4F43159B9564800564B562@SaeedLaptop> Message-ID: <87f2f3b90902121202m5652f0e3x42eabb6ced2fd646@mail.gmail.com> On Wed, Feb 11, 2009 at 8:31 AM, Saeed Ahmed wrote: > Hi List, > > Is there any open source call center tool available which works with FS? Check this out: http://opencsm.org/wiki/index.php/Spice_Telephony -MC From msc at freeswitch.org Thu Feb 12 11:59:57 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 12 Feb 2009 11:59:57 -0800 Subject: [Freeswitch-users] js and VMD In-Reply-To: References: Message-ID: <87f2f3b90902121159k570e0f4waa3afc2629ac5920@mail.gmail.com> > I'm trying to get VMD running in js, does anyone have an example of how it's > called? http://wiki.freeswitch.org/wiki/Mod_vmd You need to use the event socket because that is the way VMD is designed. If called from the dialplan it will set a channel variable but that isn't of much use in a real-time application. Using it as an API (or bgapi) will yield an event when VMD is detected. This makes sense because you don't know when (or even if) VMD will be detected, so using the event system is the best choice. -MC From red.rain.seven at gmail.com Thu Feb 12 12:13:47 2009 From: red.rain.seven at gmail.com (Henry Huang) Date: Thu, 12 Feb 2009 12:13:47 -0800 Subject: [Freeswitch-users] FreeSWITCH VPSs In-Reply-To: References: <92e7d2090902111434l60afd3a8i634b4df688ed4011@mail.gmail.com> <59ad9ca10902111637n1459d3cep5bc73dffe6bb474f@mail.gmail.com> Message-ID: <59ad9ca10902121213v24100f30qfcbb4e2fdb350806@mail.gmail.com> I run /usr/local/freeswitch/bin/freeswitch but I don't see a place where it says it's 32bit or 64bit. at the end of the initial script, I do see a version statement though. FreeSWITCH Version 1.0.trunk (exported) Started. Is there other ways to check if it's 32bit or 64bit? On Wed, Feb 11, 2009 at 6:55 PM, Brian West wrote: > ding ding ding .. yep! > > "file /usr/local/freeswitch/bin/freeswitch" will also confirm > > /b > > On Feb 11, 2009, at 6:37 PM, Henry Huang wrote: > > > Brian: > > > > I am also running my freeswitch on my own openVZ containers. Just > > how do you verify if the freeswitch is compiled as 64bit? I would > > assume if I compile it under a 64bit container, I would > > automatically get a 64bit freeswitch right? > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Henry Huang UniC Solution - Communication Unified -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090212/bc8d2934/attachment-0002.html From william.suffill at gmail.com Thu Feb 12 12:18:45 2009 From: william.suffill at gmail.com (William Suffill) Date: Thu, 12 Feb 2009 15:18:45 -0500 Subject: [Freeswitch-users] FreeSWITCH VPSs In-Reply-To: <59ad9ca10902121213v24100f30qfcbb4e2fdb350806@mail.gmail.com> References: <92e7d2090902111434l60afd3a8i634b4df688ed4011@mail.gmail.com> <59ad9ca10902111637n1459d3cep5bc73dffe6bb474f@mail.gmail.com> <59ad9ca10902121213v24100f30qfcbb4e2fdb350806@mail.gmail.com> Message-ID: <6b65470d0902121218j247dc688gbc9e615f6c8af281@mail.gmail.com> If you run in your shell: file /usr/local/freeswitch/bin/freeswitch as Brian suggested it will return something like what I got below: /usr/local/freeswitch/bin/freeswitch: ELF 64-bit LSB executable, x86-64, version 1 (SYSV), for GNU/Linux 2.6.8, dynamically linked (uses shared libs), not stripped He didn't want you to start freeswitch but instead pass the path of the binary to the file command which will tell information about most files. -- W -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090212/aa1ddbf2/attachment-0002.html From brian at freeswitch.org Thu Feb 12 12:31:16 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 12 Feb 2009 14:31:16 -0600 Subject: [Freeswitch-users] FreeSWITCH VPSs In-Reply-To: <59ad9ca10902121213v24100f30qfcbb4e2fdb350806@mail.gmail.com> References: <92e7d2090902111434l60afd3a8i634b4df688ed4011@mail.gmail.com> <59ad9ca10902111637n1459d3cep5bc73dffe6bb474f@mail.gmail.com> <59ad9ca10902121213v24100f30qfcbb4e2fdb350806@mail.gmail.com> Message-ID: <21D9E0A3-29A7-442F-AB2B-B26FF2568BFD@freeswitch.org> Well when I do this: root at taz [Thu Feb 12 02:20 PM] /usr/src/freeswitch.trunk <13>:file /usr/local/freeswitch/bin/freeswitch /usr/local/freeswitch/bin/freeswitch: ELF 64-bit LSB executable, AMD x86-64, version 1 (SYSV), for GNU/Linux 2.6.9, dynamically linked (uses shared libs), for GNU/Linux 2.6.9, not stripped It should clearly tell you. run the "file" command on it. /b On Feb 12, 2009, at 2:13 PM, Henry Huang wrote: > I run /usr/local/freeswitch/bin/freeswitch > but I don't see a place where it says it's 32bit or 64bit. > at the end of the initial script, I do see a version statement though. > FreeSWITCH Version 1.0.trunk (exported) Started. > Is there other ways to check if it's 32bit or 64bit? From red.rain.seven at gmail.com Thu Feb 12 12:40:50 2009 From: red.rain.seven at gmail.com (Henry Huang) Date: Thu, 12 Feb 2009 12:40:50 -0800 Subject: [Freeswitch-users] FreeSWITCH VPSs In-Reply-To: <21D9E0A3-29A7-442F-AB2B-B26FF2568BFD@freeswitch.org> References: <92e7d2090902111434l60afd3a8i634b4df688ed4011@mail.gmail.com> <59ad9ca10902111637n1459d3cep5bc73dffe6bb474f@mail.gmail.com> <59ad9ca10902121213v24100f30qfcbb4e2fdb350806@mail.gmail.com> <21D9E0A3-29A7-442F-AB2B-B26FF2568BFD@freeswitch.org> Message-ID: <59ad9ca10902121240t3eec6913x2d83a03facab7387@mail.gmail.com> Thinak you, William and Brian I got it now, I didn't know file was a command before because it didn't come with my CentOS installation. Now I have installed the file package and able to see the file info. Thanks again On Thu, Feb 12, 2009 at 12:31 PM, Brian West wrote: > Well when I do this: > > > root at taz [Thu Feb 12 02:20 PM] /usr/src/freeswitch.trunk > <13>:file /usr/local/freeswitch/bin/freeswitch > /usr/local/freeswitch/bin/freeswitch: ELF 64-bit LSB executable, AMD > x86-64, version 1 (SYSV), for GNU/Linux 2.6.9, dynamically linked > (uses shared libs), for GNU/Linux 2.6.9, not stripped > > > It should clearly tell you. run the "file" command on it. > > /b > > > > On Feb 12, 2009, at 2:13 PM, Henry Huang wrote: > > > I run /usr/local/freeswitch/bin/freeswitch > > but I don't see a place where it says it's 32bit or 64bit. > > at the end of the initial script, I do see a version statement though. > > FreeSWITCH Version 1.0.trunk (exported) Started. > > Is there other ways to check if it's 32bit or 64bit? > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Henry Huang UniC Solution - Communication Unified -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090212/2d2ba4dc/attachment-0002.html From jaugenstine at gmail.com Thu Feb 12 12:44:54 2009 From: jaugenstine at gmail.com (jonathan augenstine) Date: Thu, 12 Feb 2009 12:44:54 -0800 Subject: [Freeswitch-users] deflect issue Message-ID: <207e7a5e0902121244t157960ar7d775430c60baa66@mail.gmail.com> I am trying to use the deflect command to transfer an inbound call. The call is established and the command seems to complete successfully. If I bump up the sofia logging, I see the command executed in the LUA script and I see output from the console from sofia that seems to indicate the deflect refer has been initiated, but there are never any SIP messages sent to the gateway. See the console output below. The next SIP message I see is a BYE from the gateway when I hang up. Do I have something configured incorrectly? Jonathan 2009-02-12 12:36:52 [INFO] switch_cpp.cpp:1086 console_log() Awake Lua execute(deflect, sofia/external/6265551212 at aristotle.mn.maestroconference.com:5080:5080) nua: nua_refer: entering nua(0x937f0f0): sent signal r_refer nua: nua_stack_set_params: entering soa_set_params(static::0x93e65c0, ...) called nua: nua_application_event: entering nua: nua_handle_magic: entering -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090212/f0cff6fc/attachment-0002.html From nik.middleton at noblesolutions.co.uk Thu Feb 12 12:49:45 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Thu, 12 Feb 2009 20:49:45 -0000 Subject: [Freeswitch-users] js and VMD In-Reply-To: <87f2f3b90902121159k570e0f4waa3afc2629ac5920@mail.gmail.com> References: <87f2f3b90902121159k570e0f4waa3afc2629ac5920@mail.gmail.com> Message-ID: That makes sense, though could it not have a call back mechanism similar to DTMF detect? I'm still not sure how I could use it even in an event socket. I plan to call my js IVR script using a socket, but that has the originate call in it which is nice and simple, but I'm unsure how I could abort it (js IVR. The functionality I'm looking for is really simple. I simply don't want to leave a voicemail message. So VMD looks just the ticket. Regards -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 12 February 2009 20:00 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] js and VMD > I'm trying to get VMD running in js, does anyone have an example of how it's > called? http://wiki.freeswitch.org/wiki/Mod_vmd You need to use the event socket because that is the way VMD is designed. If called from the dialplan it will set a channel variable but that isn't of much use in a real-time application. Using it as an API (or bgapi) will yield an event when VMD is detected. This makes sense because you don't know when (or even if) VMD will be detected, so using the event system is the best choice. -MC _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From brian at freeswitch.org Thu Feb 12 12:47:15 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 12 Feb 2009 14:47:15 -0600 Subject: [Freeswitch-users] deflect issue In-Reply-To: <207e7a5e0902121244t157960ar7d775430c60baa66@mail.gmail.com> References: <207e7a5e0902121244t157960ar7d775430c60baa66@mail.gmail.com> Message-ID: deflect takes one arg. and that isn't one. Try a SIP uri... not a sofia/ string. ie sip:blah at host:5080 /b On Feb 12, 2009, at 2:44 PM, jonathan augenstine wrote: > I am trying to use the deflect command to transfer an inbound call. > The call is established and the command seems to complete > successfully. If I bump up the sofia logging, I see the command > executed in the LUA script and I see output from the console from > sofia that seems to indicate the deflect refer has been initiated, > but there are never any SIP messages sent to the gateway. See the > console output below. The next SIP message I see is a BYE from the > gateway when I hang up. Do I have something configured incorrectly? > > Jonathan > > 2009-02-12 12:36:52 [INFO] switch_cpp.cpp:1086 console_log() Awake > Lua execute(deflect, sofia/external/6265551212 at aristotle.mn.maestroconference.com > :5080:5080) > nua: nua_refer: entering > nua(0x937f0f0): sent signal r_refer > nua: nua_stack_set_params: entering > soa_set_params(static::0x93e65c0, ...) called > nua: nua_application_event: entering > nua: nua_handle_magic: entering -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090212/91091bc8/attachment-0002.html From anthony.minessale at gmail.com Thu Feb 12 12:59:57 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 12 Feb 2009 14:59:57 -0600 Subject: [Freeswitch-users] Cannot choose Cepstral voice from dialplan In-Reply-To: <4993A7D8.1090004@versafon.com> References: <498E67BF.3060207@versafon.com> <8B964172-6EEB-4B5B-B4B1-D35BFEAEB460@freeswitch.org> <498F6751.2000701@versafon.com> <87f2f3b90902090904i72e37241hb5d4808eb31fcc18@mail.gmail.com> <4993A7D8.1090004@versafon.com> Message-ID: <191c3a030902121259x41e03e50yc5c7c5dc00f509a7@mail.gmail.com> transcoding from PCMU (g711) to PCM (raw signed linear) the format that cepstral speaks. On Wed, Feb 11, 2009 at 10:38 PM, pauld wrote: > The issue was resolved by creating symlinks to cepstral libs in FS lib > directory. I tried that on 1.0.3, but most probably it would work on > 1.0.2 as well. Thanks for help. > BTW, without that FS would do a core dump (seg fault) on shutdown after > TTS was invoked at least once. > Looking at FS logs I see "TRANSCODING_NECESSARY" when executing dynamic > text even with 8 kHz voice. Why would that be? Looks like it's PCMU/8000 > what it's transcoding to what? > > > Michael Collins wrote: > > On Sun, Feb 8, 2009 at 3:14 PM, pauld wrote: > > > >> The libs are there with correct symlinks, see below. I tested both > >> voices directly via swift command, works fine. > >> Any other ideas? > >> It's Cepstral 5.1, FS 1.0.2. > >> > >> > > > > Well, first I recommend getting on latest trunk if that's at all > > possible for you. The devs have made a ton of improvements in the last > > five weeks. Second, this might actually be an issue with FS looking in > > its own lib directory for these .so files. Try a symlink from > > /usr/local/freeswitch/lib to your /opt/swift/lib (or whatever the name > > is) dir for each .so file. However, I think Raymond is correct - some > > weirdness has been reported by some Cepstral users on 5.1. We'd > > definitely like to hear about your experiences if and when you get it > > running. > > > > -MC > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090212/839f9df5/attachment-0002.html From anthony.minessale at gmail.com Thu Feb 12 13:06:15 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 12 Feb 2009 15:06:15 -0600 Subject: [Freeswitch-users] Codec negotiation questions In-Reply-To: References: Message-ID: <191c3a030902121306l1dae21a3m614f400703d15bd2@mail.gmail.com> the entire sdp is available as a variable (route the call to the info app to see the variables) so if you have inbound-late-negotiation set to true on the sip profile then you can use a regex or a script to set absolute_codec string before you answer. On Thu, Feb 12, 2009 at 8:06 AM, ivdreg ivdreg wrote: > Hi all, > > Can I ask 2 questions about codec negotiation: > > 1. Is it possible Freeswitch to work negotiate codecs between two phones as > it is described below. > INVITE from A with some codecs in SDP ---> Freeswitch rewrites codec > preference according absolute_codec_string but exclude all codecs not > offered by A ----> INVITE to B with rewrited SDP. > > example: > from A SDP:PCMA,PCMU,SPEEX ----> absolute_codec_string=G722,PCMU,PCMA,GSM > ----> to B SDP: PCMU,PCMA > > 2. Can I get codec list in INVITE with mod_perl for example or via xml_curl > without processing SDP variable (switch_r_sdp). It will be useful to be in > format that absolute_codec_string variable takes. > > Thanks > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090212/4f0f84cf/attachment-0002.html From kerrada2003 at yahoo.com Thu Feb 12 13:41:29 2009 From: kerrada2003 at yahoo.com (Ali Al-Rubaie) Date: Thu, 12 Feb 2009 13:41:29 -0800 (PST) Subject: [Freeswitch-users] Realm value Message-ID: <268387.58846.qm@web33701.mail.mud.yahoo.com> Hi, ? How can the default value of "realm" be changed? I had changed the command: ? ? in the file internal.xml but FS still uses the server IP address as the challenge realm. ? Thanks in advance! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090212/9fecc8d4/attachment-0002.html From msc at freeswitch.org Thu Feb 12 13:44:45 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 12 Feb 2009 13:44:45 -0800 Subject: [Freeswitch-users] js and VMD In-Reply-To: References: <87f2f3b90902121159k570e0f4waa3afc2629ac5920@mail.gmail.com> Message-ID: <87f2f3b90902121344h487cafeai724c8a56b9195201@mail.gmail.com> On Thu, Feb 12, 2009 at 12:49 PM, Nik Middleton wrote: > That makes sense, though could it not have a call back mechanism similar > to DTMF detect? > It probably could but the mod's author was using it exclusively from event socket. I personally added the channel variable code for the sake of testing. I'm sure this could be added but it's beyond my skills presently. I would recommend opening up a JIRA and requesting this functionality as an improvement. Perhaps the author, Eric Des Courtis, could add it or perhaps another skilled programmer could add this functionality. In the grand scheme of things it probably isn't that difficult and with a little time even I could figure it out. > I'm still not sure how I could use it even in an event socket. I plan > to call my js IVR script using a socket, but that has the originate call > in it which is nice and simple, but I'm unsure how I could abort it (js > IVR. As a proof of concept you could have your script loop and check the value of ${vmd_status} every 1000ms or so, and if it ever has the value "TRUE" then you know VMD was positive and you could hangup and do whatever other cleanup is necessary. That solution would be a temp fix even though it wouldn't actually scale very well. How are you handling answered calls now? Do you just start playing a message? I'm wondering how this would work even if there was a callback. Would you mind doing a pastebin of your script? I'd like to see the big picture. -MC > > The functionality I'm looking for is really simple. I simply don't want > to leave a voicemail message. So VMD looks just the ticket. > > Regards From lfurrea at gmail.com Thu Feb 12 13:46:23 2009 From: lfurrea at gmail.com (Luis F Urrea) Date: Thu, 12 Feb 2009 15:46:23 -0600 Subject: [Freeswitch-users] xml_cdr call flow Message-ID: Hi all, We are writing a xml_cdr parser to load CDRs in SQLite. We are interested in logging times for both A leg and B leg so that transfers are reported as individual calls with accurate timing. eg Inboud call to AA lasted 14 seconds then call to operator 20s and then call to actual extension 5min As of now we are using the tag with the "number" attribute to find out who did the A leg talk to, then we open the B leg files and get the times from each jump from the tag within the tag on the B leg file. Is this right or maybe someone could suggest a better way to do it. TIA Luis -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090212/4422ed95/attachment-0002.html From brian at freeswitch.org Thu Feb 12 13:48:00 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 12 Feb 2009 15:48:00 -0600 Subject: [Freeswitch-users] Realm value In-Reply-To: <268387.58846.qm@web33701.mail.mud.yahoo.com> References: <268387.58846.qm@web33701.mail.mud.yahoo.com> Message-ID: What SVN rev? /b On Feb 12, 2009, at 3:41 PM, Ali Al-Rubaie wrote: > Hi, > > How can the default value of "realm" be changed? I had changed the > command: > > > > in the file internal.xml but FS still uses the server IP address as > the challenge realm. > > Thanks in advance! > From anthony.minessale at gmail.com Thu Feb 12 14:20:18 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 12 Feb 2009 16:20:18 -0600 Subject: [Freeswitch-users] segfault when shoutcast plays mp3 and extension hangs up In-Reply-To: <4993D8CA.1010602@ewetel.de> References: <4992BFFC.50006@ewetel.de> <4992C801.4020906@ewetel.de> <191c3a030902110558i9702139g6b12aaf15f6d3aac@mail.gmail.com> <4992DCC2.3020702@ewetel.de> <191c3a030902110617n3dcd19cfr204c00792e1e87ed@mail.gmail.com> <4992E559.6060506@ewetel.de> <191c3a030902110729g30b30d46gac0f56205f8d48cf@mail.gmail.com> <4993D8CA.1010602@ewetel.de> Message-ID: <191c3a030902121420o3b4ec7a8j1fa0059b849bd866@mail.gmail.com> That's scary.... So I wonder what about the distro you are using that makes the same exact code not work? maybe the GCC ? On Thu, Feb 12, 2009 at 2:07 AM, Helmut Kuper wrote: > Hi Anthony, > > hm... on centos5 it works fine. No problems, no warning, no crash. > > regards > Helmut > > On 11.02.2009 16:29, Anthony Minessale wrote: > > I am highly suspicious of the ubuntu. > > you are using a prerelease of gcc that we have already found at least > > 1 bug. > > > > we tried the file on our box and it doesn't even say anything about > > the file being bad etc...... it plays and hangs up fine even 4 times > > at once. > > It would be a big help if you could try to reproduce it on CentOS 5 as > > a comparison. We have had 3 cases this week where doing so has fixed > > problems and i don't want to believe it so I would appropriate it if > > you could test it. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090212/f8c65917/attachment-0002.html From nik.middleton at noblesolutions.co.uk Thu Feb 12 14:26:17 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Thu, 12 Feb 2009 22:26:17 -0000 Subject: [Freeswitch-users] js and VMD In-Reply-To: <87f2f3b90902121344h487cafeai724c8a56b9195201@mail.gmail.com> References: <87f2f3b90902121159k570e0f4waa3afc2629ac5920@mail.gmail.com> <87f2f3b90902121344h487cafeai724c8a56b9195201@mail.gmail.com> Message-ID: Just been chatting to Ken Rice, his view (and he may be mistaken) is that it should fire the call back event in much the same way as DTMF does, however, it's not working. I used to develop with C/C++ for about 10 years, but that was 12 years ago. Very rusty. However, I'm going to look at the start_dtmf code and try to replicate the functionality in mod_vmd. Regarding your suggestion, that wouldn't really work as I'm streaming a file. However, if memory serves me well, there is a timer function in C that you can set to run that can call a function. There is a function in js called setTimeout(time_func, 500) but sadly it's not available in spidermonkey. BTW this function would resolve a bounty on call duration timeouts Regards -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 12 February 2009 21:45 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] js and VMD On Thu, Feb 12, 2009 at 12:49 PM, Nik Middleton wrote: > That makes sense, though could it not have a call back mechanism similar > to DTMF detect? > It probably could but the mod's author was using it exclusively from event socket. I personally added the channel variable code for the sake of testing. I'm sure this could be added but it's beyond my skills presently. I would recommend opening up a JIRA and requesting this functionality as an improvement. Perhaps the author, Eric Des Courtis, could add it or perhaps another skilled programmer could add this functionality. In the grand scheme of things it probably isn't that difficult and with a little time even I could figure it out. > I'm still not sure how I could use it even in an event socket. I plan > to call my js IVR script using a socket, but that has the originate call > in it which is nice and simple, but I'm unsure how I could abort it (js > IVR. As a proof of concept you could have your script loop and check the value of ${vmd_status} every 1000ms or so, and if it ever has the value "TRUE" then you know VMD was positive and you could hangup and do whatever other cleanup is necessary. That solution would be a temp fix even though it wouldn't actually scale very well. How are you handling answered calls now? Do you just start playing a message? I'm wondering how this would work even if there was a callback. Would you mind doing a pastebin of your script? I'd like to see the big picture. -MC > > The functionality I'm looking for is really simple. I simply don't want > to leave a voicemail message. So VMD looks just the ticket. > > Regards _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From woof at nortel.com Thu Feb 12 14:34:58 2009 From: woof at nortel.com (Andy Spitzer) Date: Thu, 12 Feb 2009 17:34:58 -0500 Subject: [Freeswitch-users] segfault when shoutcast plays mp3 and extension hangs up In-Reply-To: <191c3a030902121420o3b4ec7a8j1fa0059b849bd866@mail.gmail.com> References: <4992BFFC.50006@ewetel.de> <4992C801.4020906@ewetel.de> <191c3a030902110558i9702139g6b12aaf15f6d3aac@mail.gmail.com> <4992DCC2.3020702@ewetel.de> <191c3a030902110617n3dcd19cfr204c00792e1e87ed@mail.gmail.com> <4992E559.6060506@ewetel.de> <191c3a030902110729g30b30d46gac0f56205f8d48cf@mail.gmail.com> <4993D8CA.1010602@ewetel.de> <191c3a030902121420o3b4ec7a8j1fa0059b849bd866@mail.gmail.com> Message-ID: Woof! On Thu, 12 Feb 2009 17:20:18 -0500, Anthony Minessale wrote: > So I wonder what about the distro you are using that makes the same exact code not work? > maybe the GCC ? Possibly. A recent (last year?) GCC change caused some order of operations to change, and so code that inadvertently relied on the previous behavior doesn't work any more. The "c standard" doesn't define many of the orders, and some code may have side effects that depend on one way or the other without realizing it. I discovered this the hard way last year some time. --Woof! From brian at freeswitch.org Thu Feb 12 14:38:06 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 12 Feb 2009 16:38:06 -0600 Subject: [Freeswitch-users] segfault when shoutcast plays mp3 and extension hangs up In-Reply-To: References: <4992BFFC.50006@ewetel.de> <4992C801.4020906@ewetel.de> <191c3a030902110558i9702139g6b12aaf15f6d3aac@mail.gmail.com> <4992DCC2.3020702@ewetel.de> <191c3a030902110617n3dcd19cfr204c00792e1e87ed@mail.gmail.com> <4992E559.6060506@ewetel.de> <191c3a030902110729g30b30d46gac0f56205f8d48cf@mail.gmail.com> <4993D8CA.1010602@ewetel.de> <191c3a030902121420o3b4ec7a8j1fa0059b849bd866@mail.gmail.com> Message-ID: <9CF2F413-1DE1-49A7-A550-6423C3444D14@freeswitch.org> This is prob. why we don't see this crazy stuff on CentOS since the compiler is 4.1.2 /b On Feb 12, 2009, at 4:34 PM, Andy Spitzer wrote: > Possibly. A recent (last year?) GCC change caused some order of > operations to change, and so code that inadvertently relied on the > previous behavior doesn't work any more. The "c standard" doesn't > define many of the orders, and some code may have side effects that > depend on one way or the other without realizing it. > > I discovered this the hard way last year some time. > > --Woof! From msc at freeswitch.org Thu Feb 12 14:42:55 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 12 Feb 2009 14:42:55 -0800 Subject: [Freeswitch-users] js and VMD In-Reply-To: References: <87f2f3b90902121159k570e0f4waa3afc2629ac5920@mail.gmail.com> <87f2f3b90902121344h487cafeai724c8a56b9195201@mail.gmail.com> Message-ID: <87f2f3b90902121442t22d9e5b7l3d2b7711646d8ca2@mail.gmail.com> On Thu, Feb 12, 2009 at 2:26 PM, Nik Middleton wrote: > Just been chatting to Ken Rice, his view (and he may be mistaken) is > that it should fire the call back event in much the same way as DTMF > does, however, it's not working. I used to develop with C/C++ for about > 10 years, but that was 12 years ago. Very rusty. However, I'm going to > look at the start_dtmf code and try to replicate the functionality in > mod_vmd. That would be awesome. I think once you get into the code you'll realize that it isn't like walking through a warzone or tapdancing in a minefield like some other codebases. :) The code is really well-ordered so you can frequently copy and paste from other files and functions and create new functionality. Feel free to get in there and start mixing it up! It's kinda fun. > > Regarding your suggestion, that wouldn't really work as I'm streaming a > file. However, if memory serves me well, there is a timer function in C > that you can set to run that can call a function. There is a function > in js called setTimeout(time_func, 500) but sadly it's not available in > spidermonkey. BTW this function would resolve a bounty on call duration > timeouts Understood. The other thing you could do, at least to test the events for VMD, is to create a little daemon kind of program that sits there and listens for VMD hits and have it uuid_kill those channels for you. Crude, to be sure, but it would definitely let you confirm that the system is working. -MC From lfurrea at gmail.com Thu Feb 12 14:50:47 2009 From: lfurrea at gmail.com (Luis F Urrea) Date: Thu, 12 Feb 2009 16:50:47 -0600 Subject: [Freeswitch-users] xml_cdr call flow In-Reply-To: References: Message-ID: On our test calls we haven't been able to correlate times from the A leg with times from the B leg. I would expect something as A-leg(duration)= B-leg1(duration)+B-leg2(duration) Also the tag within tag does not seem to be in epoch microseconds. so it does not seem that's where i should be looking for that info. Here's an example of the tag for a test call on the A-Leg: 1. 2. 1233942283835696 3. 1233942283835696 4. 1233942283999716 5. 1233942283999716 6. 1233942287291931 7. 0 8. 1233942303240916 9. any hint is appreciated On Thu, Feb 12, 2009 at 3:46 PM, Luis F Urrea wrote: > Hi all, > > We are writing a xml_cdr parser to load CDRs in SQLite. We are interested > in logging times for both A leg and B leg so that transfers are reported as > individual calls with accurate timing. eg Inboud call to AA lasted 14 > seconds then call to operator 20s and then call to actual extension 5min > > As of now we are using the tag with the "number" attribute to > find out who did the A leg talk to, then we open the B leg files and get the > times from each jump from the tag within the tag on the B > leg file. > > Is this right or maybe someone could suggest a better way to do it. > > TIA > > Luis > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090212/2077fea2/attachment-0002.html From nik.middleton at noblesolutions.co.uk Thu Feb 12 14:58:02 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Thu, 12 Feb 2009 22:58:02 -0000 Subject: [Freeswitch-users] FS equiv for waitforextension In-Reply-To: <87f2f3b90902121014t29ab0e25u129c4c824d63e11e@mail.gmail.com> References: <35b355e90902121001i3a049252x852336b18846ad21@mail.gmail.com><76D5D669-B367-4639-925F-DA213FC93539@freeswitch.org> <87f2f3b90902121014t29ab0e25u129c4c824d63e11e@mail.gmail.com> Message-ID: Hi, Not sure who updates the WIKI, but it's wrong on collectinput for the example. In the call, dtmf needs quotes, ie "dtmf" Correction is session.collectInput( mycb, "dtmf", 8000 ); Without it you get [ERR] voice.js:70 mod_spidermonkey() ReferenceError: dtmf is not defined if ( session.ready( ) ) { session.answer( ); session.streamFile( "sounds/typeSomeDigits.wav" ); session.collectInput( mycb, dtmf, 8000 ); console_log( "info", "Got " + dtmf.digits + "\n" ); session.streamFile( "sounds/thanksBye.wav" ); session.hangup( ); } -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 12 February 2009 18:15 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] FS equiv for waitforextension On Thu, Feb 12, 2009 at 10:07 AM, Brian West wrote: > Dialplan isn't for writing IVR's... doing so is against the design of > FreeSWITCH.. you can do simple things in dialplan but more complex > stuff needs to be in a language. Or create an IVR and send the call there from the dialplan. You can do IVRs in Lua/JS/Perl or in XML. -MC _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From msc at freeswitch.org Thu Feb 12 15:00:32 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 12 Feb 2009 15:00:32 -0800 Subject: [Freeswitch-users] xml_cdr call flow In-Reply-To: References: Message-ID: <87f2f3b90902121500m2b86efc2q722314688d532648@mail.gmail.com> Pastebin the whole file so that we can see it in context... -MC On Thu, Feb 12, 2009 at 2:50 PM, Luis F Urrea wrote: > On our test calls we haven't been able to correlate times from the A leg > with times from the B leg. > > I would expect something as A-leg(duration)= > B-leg1(duration)+B-leg2(duration) > > Also the tag within tag does not seem to be in epoch > microseconds. so it does not seem that's where i should be looking for that > info. > > Here's an example of the tag for a test call on the A-Leg: > > > 1233942283835696 > 1233942283835696 > 1233942283999716 > 1233942283999716 > 1233942287291931 > 0 > 1233942303240916 > > > any hint is appreciated > > > On Thu, Feb 12, 2009 at 3:46 PM, Luis F Urrea wrote: >> >> Hi all, >> >> We are writing a xml_cdr parser to load CDRs in SQLite. We are interested >> in logging times for both A leg and B leg so that transfers are reported as >> individual calls with accurate timing. eg Inboud call to AA lasted 14 >> seconds then call to operator 20s and then call to actual extension 5min >> >> As of now we are using the tag with the "number" attribute to >> find out who did the A leg talk to, then we open the B leg files and get the >> times from each jump from the tag within the tag on the B >> leg file. >> >> Is this right or maybe someone could suggest a better way to do it. >> >> TIA >> >> Luis > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From msc at freeswitch.org Thu Feb 12 15:01:43 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 12 Feb 2009 15:01:43 -0800 Subject: [Freeswitch-users] FS equiv for waitforextension In-Reply-To: References: <35b355e90902121001i3a049252x852336b18846ad21@mail.gmail.com> <76D5D669-B367-4639-925F-DA213FC93539@freeswitch.org> <87f2f3b90902121014t29ab0e25u129c4c824d63e11e@mail.gmail.com> Message-ID: <87f2f3b90902121501g2d0c69f9p52a7003880c42304@mail.gmail.com> On Thu, Feb 12, 2009 at 2:58 PM, Nik Middleton wrote: > Hi, > > Not sure who updates the WIKI, but it's wrong on collectinput for the > example. In the call, dtmf needs quotes, ie "dtmf" Thanks for the heads up. Actually, YOU can update the wiki. If you want me to do so I will be happy to. -MC From brian at freeswitch.org Thu Feb 12 15:01:08 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 12 Feb 2009 17:01:08 -0600 Subject: [Freeswitch-users] FS equiv for waitforextension In-Reply-To: References: <35b355e90902121001i3a049252x852336b18846ad21@mail.gmail.com><76D5D669-B367-4639-925F-DA213FC93539@freeswitch.org> <87f2f3b90902121014t29ab0e25u129c4c824d63e11e@mail.gmail.com> Message-ID: <7A3C8938-9231-4A61-93EB-7503D4778C85@freeswitch.org> YOU DO! ;) Its a user edited content portal. /b On Feb 12, 2009, at 4:58 PM, Nik Middleton wrote: > > Not sure who updates the WIKI, but it's wrong on collectinput for the > example. In the call, dtmf needs quotes, ie "dtmf" From nik.middleton at noblesolutions.co.uk Thu Feb 12 15:17:53 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Thu, 12 Feb 2009 23:17:53 -0000 Subject: [Freeswitch-users] FS equiv for waitforextension In-Reply-To: <7A3C8938-9231-4A61-93EB-7503D4778C85@freeswitch.org> References: <35b355e90902121001i3a049252x852336b18846ad21@mail.gmail.com><76D5D669-B367-4639-925F-DA213FC93539@freeswitch.org><87f2f3b90902121014t29ab0e25u129c4c824d63e11e@mail.gmail.com> <7A3C8938-9231-4A61-93EB-7503D4778C85@freeswitch.org> Message-ID: Done, that was easy, unlike FS :) -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: 12 February 2009 23:01 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] FS equiv for waitforextension YOU DO! ;) Its a user edited content portal. /b On Feb 12, 2009, at 4:58 PM, Nik Middleton wrote: > > Not sure who updates the WIKI, but it's wrong on collectinput for the > example. In the call, dtmf needs quotes, ie "dtmf" _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From lfurrea at gmail.com Thu Feb 12 15:31:16 2009 From: lfurrea at gmail.com (Luis F Urrea) Date: Thu, 12 Feb 2009 17:31:16 -0600 Subject: [Freeswitch-users] xml_cdr call flow In-Reply-To: <87f2f3b90902121500m2b86efc2q722314688d532648@mail.gmail.com> References: <87f2f3b90902121500m2b86efc2q722314688d532648@mail.gmail.com> Message-ID: Heres pastebin of the A-leg http://pastebin.com/m6731913d On Thu, Feb 12, 2009 at 5:00 PM, Michael Collins wrote: > Pastebin the whole file so that we can see it in context... > -MC > > On Thu, Feb 12, 2009 at 2:50 PM, Luis F Urrea wrote: > > On our test calls we haven't been able to correlate times from the A leg > > with times from the B leg. > > > > I would expect something as A-leg(duration)= > > B-leg1(duration)+B-leg2(duration) > > > > Also the tag within tag does not seem to be in epoch > > microseconds. so it does not seem that's where i should be looking for > that > > info. > > > > Here's an example of the tag for a test call on the A-Leg: > > > > > > 1233942283835696 > > 1233942283835696 > > 1233942283999716 > > 1233942283999716 > > 1233942287291931 > > 0 > > 1233942303240916 > > > > > > any hint is appreciated > > > > > > On Thu, Feb 12, 2009 at 3:46 PM, Luis F Urrea wrote: > >> > >> Hi all, > >> > >> We are writing a xml_cdr parser to load CDRs in SQLite. We are > interested > >> in logging times for both A leg and B leg so that transfers are reported > as > >> individual calls with accurate timing. eg Inboud call to AA lasted 14 > >> seconds then call to operator 20s and then call to actual extension 5min > >> > >> As of now we are using the tag with the "number" attribute to > >> find out who did the A leg talk to, then we open the B leg files and get > the > >> times from each jump from the tag within the tag on > the B > >> leg file. > >> > >> Is this right or maybe someone could suggest a better way to do it. > >> > >> TIA > >> > >> Luis > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090212/f15e763a/attachment-0002.html From msc at freeswitch.org Thu Feb 12 16:08:16 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 12 Feb 2009 16:08:16 -0800 Subject: [Freeswitch-users] xml_cdr call flow In-Reply-To: References: <87f2f3b90902121500m2b86efc2q722314688d532648@mail.gmail.com> Message-ID: <87f2f3b90902121608x217f682v289eae73012e27a5@mail.gmail.com> On Thu, Feb 12, 2009 at 3:31 PM, Luis F Urrea wrote: > Heres pastebin of the A-leg > > http://pastebin.com/m6731913d > > > On Thu, Feb 12, 2009 at 5:00 PM, Michael Collins wrote: >> >> Pastebin the whole file so that we can see it in context... >> -MC >> >> On Thu, Feb 12, 2009 at 2:50 PM, Luis F Urrea wrote: >> > On our test calls we haven't been able to correlate times from the A leg >> > with times from the B leg. >> > >> > I would expect something as A-leg(duration)= >> > B-leg1(duration)+B-leg2(duration) I don't know that this is necessarily true. Can you pastebin your dialplan entry (or whatever generated this call) so we can take a look? Also, please use our pastebin so that it's easier for us to find stuff: http://pastebin.freeswitch.org >> > >> > Also the tag within tag does not seem to be in epoch >> > microseconds. so it does not seem that's where i should be looking for >> > that >> > info. >> > >> > Here's an example of the tag for a test call on the A-Leg: >> > >> > >> > 1233942283835696 >> > 1233942283835696 >> > 1233942283999716 >> > 1233942283999716 >> > 1233942287291931 >> > 0 >> > 1233942303240916 >> > >> > >> > any hint is appreciated >> > Perhaps I'm missing something but they sure look like epoch microseconds to me. -MC From pauld at versafon.com Thu Feb 12 16:35:02 2009 From: pauld at versafon.com (pauld) Date: Thu, 12 Feb 2009 19:35:02 -0500 Subject: [Freeswitch-users] Cannot choose Cepstral voice from dialplan In-Reply-To: <9268AE39-6841-4819-9A61-37806B48BEFF@freeswitch.org> References: <498E67BF.3060207@versafon.com> <8B964172-6EEB-4B5B-B4B1-D35BFEAEB460@freeswitch.org> <498F6751.2000701@versafon.com> <87f2f3b90902090904i72e37241hb5d4808eb31fcc18@mail.gmail.com> <4993A7D8.1090004@versafon.com> <10703C0D-0BA2-4FB3-B669-B87BF93912E0@freeswitch.org> <49942D7A.2090503@versafon.com> <9268AE39-6841-4819-9A61-37806B48BEFF@freeswitch.org> Message-ID: <4994C036.30608@versafon.com> Brian West wrote: > You still didn't answer my question. How are you trying to do this > from the dialplan. > > /b > > On Feb 12, 2009, at 8:08 AM, pauld wrote: > > >> Yes I am using 5.1, I haven't done anything special other than >> followed >> wiki and then the advice given here to create symlinks in FS lib dir >> to all >> cepstral libs. I have cepstral libs in a standard location /opt/ >> swift/lib. >> I have given an example extension I used for testing earlier in this >> thread >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From helmut.kuper at ewetel.de Fri Feb 13 01:13:21 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Fri, 13 Feb 2009 10:13:21 +0100 Subject: [Freeswitch-users] segfault when shoutcast plays mp3 and extension hangs up In-Reply-To: <9CF2F413-1DE1-49A7-A550-6423C3444D14@freeswitch.org> References: <4992BFFC.50006@ewetel.de> <4992C801.4020906@ewetel.de> <191c3a030902110558i9702139g6b12aaf15f6d3aac@mail.gmail.com> <4992DCC2.3020702@ewetel.de> <191c3a030902110617n3dcd19cfr204c00792e1e87ed@mail.gmail.com> <4992E559.6060506@ewetel.de> <191c3a030902110729g30b30d46gac0f56205f8d48cf@mail.gmail.com> <4993D8CA.1010602@ewetel.de> <191c3a030902121420o3b4ec7a8j1fa0059b849bd866@mail.gmail.com> <9CF2F413-1DE1-49A7-A550-6423C3444D14@freeswitch.org> Message-ID: <499539B1.1000507@ewetel.de> Hello, it works now. I'm not really sure what it was, but I know what I did in what order: 1. update gcc to "gcc version 4.2.4 (Ubuntu 4.2.4-1ubuntu3)" 2. configure 3. make sure 4. make install 5 Tested it: FS still crash 6 did a gdb backtrace, last function call was mpg123_delete ... 7. delete lame directory an archive 8. delete mpg123 directory and archive 9. bootstrap.sh 10. configure ... 11. make sure 12. chmod 755 ./libs/libsndfile/src/create_symbols_file.py (execution permissions were missed) 13. autogen was missed, so I installed it 14. make all 15. make install 16. Tested it: SUCCESS - no crash anymore For me it seems to be caused by mpg123 and/or lame ... Maybe a simple recompile and reinstall helps as well. regards Helmut On 12.02.2009 23:38, Brian West wrote: > This is prob. why we don't see this crazy stuff on CentOS since the > compiler is 4.1.2 > > /b > From alex at sinapticode.ro Fri Feb 13 03:33:14 2009 From: alex at sinapticode.ro (Alexandru Nedelcu) Date: Fri, 13 Feb 2009 13:33:14 +0200 Subject: [Freeswitch-users] Problems with Originate Message-ID: <1234524794.4431.56.camel@gathern.lan> Hi all, I'm using "originate" to initiate calls, and streamFile to play audio files on the answered sessions. All the logic was encapsulated in a Javascript file. The problem with this setup is that origination_caller_id_number doesn't work from inside the JS file (when calling session.originate). My setup only works when doing a direct originate command, with the JS script attached as an application to it, i.e... originate {ignore_early_media=true}sofia/gateway/myprovider/873040711222222 '&javascript(dialer.js )' Now, my next problem ... this doesn't work properly because with "ignore_early_media=true" then "dialer.js" isn't executed on FAIL. And I need that. If "ignore_early_media" is not specified, then "dialer.js" executes, but the recording starts before the phone is answered. Can you give me any tips? Thanks, -- Alexandru Nedelcu Software Developer, Sinapticode From alex at sinapticode.ro Fri Feb 13 03:48:44 2009 From: alex at sinapticode.ro (Alexandru Nedelcu) Date: Fri, 13 Feb 2009 13:48:44 +0200 Subject: [Freeswitch-users] Problems with Originate In-Reply-To: <1234524794.4431.56.camel@gathern.lan> References: <1234524794.4431.56.camel@gathern.lan> Message-ID: <1234525724.4431.59.camel@gathern.lan> On Fri, 2009-02-13 at 13:33 +0200, Alexandru Nedelcu wrote: > The problem with this setup is that origination_caller_id_number doesn't > work from inside the JS file (when calling session.originate). I just discovered something interesting. When originating the call like this ... session = new Session("") instead of this ... session = new Session(); session.originate("") ... then it works. Is this some kind of bug, or what's the difference here? Thanks, -- Alexandru Nedelcu Software Developer, Sinapticode From nik.middleton at noblesolutions.co.uk Fri Feb 13 03:48:39 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Fri, 13 Feb 2009 11:48:39 -0000 Subject: [Freeswitch-users] Problems with Originate In-Reply-To: <1234524794.4431.56.camel@gathern.lan> References: <1234524794.4431.56.camel@gathern.lan> Message-ID: Use this method in js var session = new Session('{absolute_codec_string=PCMA,accountcode=54321,ignore_early_medi a=true,origination_caller_id_number=40711222222,originate_timeout=25}sof ia/gateway/myprovider/873040711222222); -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Alexandru Nedelcu Sent: 13 February 2009 11:33 To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Problems with Originate Hi all, I'm using "originate" to initiate calls, and streamFile to play audio files on the answered sessions. All the logic was encapsulated in a Javascript file. The problem with this setup is that origination_caller_id_number doesn't work from inside the JS file (when calling session.originate). My setup only works when doing a direct originate command, with the JS script attached as an application to it, i.e... originate {ignore_early_media=true}sofia/gateway/myprovider/873040711222222 '&javascript(dialer.js )' Now, my next problem ... this doesn't work properly because with "ignore_early_media=true" then "dialer.js" isn't executed on FAIL. And I need that. If "ignore_early_media" is not specified, then "dialer.js" executes, but the recording starts before the phone is answered. Can you give me any tips? Thanks, -- Alexandru Nedelcu Software Developer, Sinapticode _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From zolotov at altron.ua Fri Feb 13 04:10:01 2009 From: zolotov at altron.ua (Evgeniy Zolotov) Date: Fri, 13 Feb 2009 14:10:01 +0200 Subject: [Freeswitch-users] Sending channel variables Message-ID: <1234527001.5507.14.camel@opos20.altron.lan> Hello! I'm trying to make such scheme: ---> FS_A --> FS_B --> record Incoming calls to FS_A are redirected to FS_B with the help of this context: FS_B records them to the file: This works good. But I have a question - in what manner I can send back (from FS_B to FS_A) some channel variables? Thanks, Evgeniy. From anthony.minessale at gmail.com Fri Feb 13 05:54:59 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 13 Feb 2009 07:54:59 -0600 Subject: [Freeswitch-users] Problems with Originate In-Reply-To: <1234525724.4431.59.camel@gathern.lan> References: <1234524794.4431.56.camel@gathern.lan> <1234525724.4431.59.camel@gathern.lan> Message-ID: <191c3a030902130554w1dfe6b1dw9e5525cf6a210dc4@mail.gmail.com> The first way is deprecated and will be removed. The 2nd way is the correct way. On Fri, Feb 13, 2009 at 5:48 AM, Alexandru Nedelcu wrote: > On Fri, 2009-02-13 at 13:33 +0200, Alexandru Nedelcu wrote: > > The problem with this setup is that origination_caller_id_number doesn't > > work from inside the JS file (when calling session.originate). > > I just discovered something interesting. > > When originating the call like this ... > session = new Session("") > instead of this ... > session = new Session(); session.originate("") > > ... then it works. Is this some kind of bug, or what's the difference > here? > > Thanks, > > -- > Alexandru Nedelcu > Software Developer, Sinapticode > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090213/27f5aeaf/attachment-0002.html From anthony.minessale at gmail.com Fri Feb 13 06:05:47 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 13 Feb 2009 08:05:47 -0600 Subject: [Freeswitch-users] Setting outbound callerid using js In-Reply-To: <87f2f3b90902111626x6db5dc61t68a0170a3460b42d@mail.gmail.com> References: <87f2f3b90902111626x6db5dc61t68a0170a3460b42d@mail.gmail.com> Message-ID: <191c3a030902130605m5e834e3dh27cc88387408b8b5@mail.gmail.com> 1) session.originate is depricated. 2) the first arg to session.originate is *another* session (not the same one) *or* undefined..... session.originate(undefined, ""); session.originate(a_leg_session, ""); session.originate(session, "") is asking the session to use itself as it's own a leg which makes no sense. This is perhaps the 4th time i have seen someone do this, can you point out where this is incorrectly documented? BTW effective_caller_id_name/number are variables you set on the A leg so when it's used to generate b legs that var is copied instead. a_leg_session.setVariable("effective_caller_id_number=1234"); b_leg_session = new Session(a_leg_session, ""); which is of course pointless because you never need to create the session if you just use the bridge application. session.execute("bridge", ""); even better just set the dest to a var and exit the script and use that var in your dialplan. --- contents of get_dest.js --- session.setVariable("dial_string", ""); -- dialplan -- the JavaScript paradigm is a bit different than the Lua/Perl one. Let > us know if that works or not. > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090213/d378128c/attachment-0002.html From nik.middleton at noblesolutions.co.uk Fri Feb 13 06:12:37 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Fri, 13 Feb 2009 14:12:37 -0000 Subject: [Freeswitch-users] Setting outbound callerid using js In-Reply-To: <191c3a030902130605m5e834e3dh27cc88387408b8b5@mail.gmail.com> References: <87f2f3b90902111626x6db5dc61t68a0170a3460b42d@mail.gmail.com> <191c3a030902130605m5e834e3dh27cc88387408b8b5@mail.gmail.com> Message-ID: I think this page (external) is the source http://alexn.org/docs/dialer.html Regards, ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: 13 February 2009 14:06 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Setting outbound callerid using js 1) session.originate is depricated. 2) the first arg to session.originate is *another* session (not the same one) *or* undefined..... session.originate(undefined, ""); session.originate(a_leg_session, ""); session.originate(session, "") is asking the session to use itself as it's own a leg which makes no sense. This is perhaps the 4th time i have seen someone do this, can you point out where this is incorrectly documented? BTW effective_caller_id_name/number are variables you set on the A leg so when it's used to generate b legs that var is copied instead. a_leg_session.setVariable("effective_caller_id_number=1234"); b_leg_session = new Session(a_leg_session, ""); which is of course pointless because you never need to create the session if you just use the bridge application. session.execute("bridge", ""); even better just set the dest to a var and exit the script and use that var in your dialplan. --- contents of get_dest.js --- session.setVariable("dial_string", ""); -- dialplan -- GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090213/a1900005/attachment-0002.html From alex at sinapticode.ro Fri Feb 13 06:25:36 2009 From: alex at sinapticode.ro (Alexandru Nedelcu) Date: Fri, 13 Feb 2009 16:25:36 +0200 Subject: [Freeswitch-users] Setting outbound callerid using js In-Reply-To: References: <87f2f3b90902111626x6db5dc61t68a0170a3460b42d@mail.gmail.com> <191c3a030902130605m5e834e3dh27cc88387408b8b5@mail.gmail.com> Message-ID: <1234535136.4431.70.camel@gathern.lan> I wrote that document ... I can't remember from where I got the idea that you should specify the a-leg as being the same session. That document is a draft, but it got indexed by Google unfortunately :( On Fri, 2009-02-13 at 14:12 +0000, Nik Middleton wrote: > I think this page (external) is the source > > > > http://alexn.org/docs/dialer.html > > > > Regards, > > > > > ______________________________________________________________________ > From:freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Anthony Minessale > Sent: 13 February 2009 14:06 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Setting outbound callerid using js > > > > > 1) session.originate is depricated. > 2) the first arg to session.originate is *another* session (not the > same one) *or* undefined..... > session.originate(undefined, ""); > session.originate(a_leg_session, ""); > > session.originate(session, "") is asking the session to > use itself as it's own a leg which makes no sense. > > This is perhaps the 4th time i have seen someone do this, can you > point out where this is incorrectly documented? > > BTW > > effective_caller_id_name/number are variables you set on the A leg so > when it's used to generate b legs that var is copied instead. > > a_leg_session.setVariable("effective_caller_id_number=1234"); > b_leg_session = new Session(a_leg_session, ""); > > which is of course pointless because you never need to create the > session if you just use the bridge application. > > > session.execute("bridge", ""); > > even better just set the dest to a var and exit the script and use > that var in your dialplan. > > > --- contents of get_dest.js --- > session.setVariable("dial_string", ""); > > -- dialplan -- > > the JavaScript paradigm is a bit different than the Lua/Perl one. Let > us know if that works or not. > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From jan.kubr at gmail.com Fri Feb 13 06:27:17 2009 From: jan.kubr at gmail.com (Jan Kubr) Date: Fri, 13 Feb 2009 15:27:17 +0100 Subject: [Freeswitch-users] Ruby framework for event socket Message-ID: <698401620902130627l5e8dec56g5fd43bbeb8b9ac0f@mail.gmail.com> Hi all, I've created a simple framework in Ruby that you can use to talk to Freeswitch via even socket outbound. It won't suite your needs perfectly if you are doing anything non-trivial, but it might be a nice starting point. Check it out at http://github.com/jankubr/freec Cheers, Jan Kubr From lfurrea at gmail.com Fri Feb 13 06:42:11 2009 From: lfurrea at gmail.com (Luis F Urrea) Date: Fri, 13 Feb 2009 08:42:11 -0600 Subject: [Freeswitch-users] xml_cdr call flow In-Reply-To: <87f2f3b90902121608x217f682v289eae73012e27a5@mail.gmail.com> References: <87f2f3b90902121500m2b86efc2q722314688d532648@mail.gmail.com> <87f2f3b90902121608x217f682v289eae73012e27a5@mail.gmail.com> Message-ID: My mistake, they do seem to be microsecs. But still I cannot correlate times from the A-leg with the B-legs. I have included below the xml_cdr files generated for the test call. The test call was made using three registered extensions. Basically, Ext 201 calls ext 203 and they talk, then 203 blindly transfers to 202, 202 does not answer and call rolls to voicemail. A-leg: http://pastebin.freeswitch.org/7206 B-leg: http://pastebin.freeswitch.org/7204 B-leg: http://pastebin.freeswitch.org/7205 Thanks for your help On Thu, Feb 12, 2009 at 6:08 PM, Michael Collins wrote: > On Thu, Feb 12, 2009 at 3:31 PM, Luis F Urrea wrote: > > Heres pastebin of the A-leg > > > > http://pastebin.com/m6731913d > > > > > > On Thu, Feb 12, 2009 at 5:00 PM, Michael Collins > wrote: > >> > >> Pastebin the whole file so that we can see it in context... > >> -MC > >> > >> On Thu, Feb 12, 2009 at 2:50 PM, Luis F Urrea > wrote: > >> > On our test calls we haven't been able to correlate times from the A > leg > >> > with times from the B leg. > >> > > >> > I would expect something as A-leg(duration)= > >> > B-leg1(duration)+B-leg2(duration) > I don't know that this is necessarily true. Can you pastebin your > dialplan entry (or whatever generated this call) so we can take a > look? Also, please use our pastebin so that it's easier for us to find > stuff: > http://pastebin.freeswitch.org > > >> > > >> > Also the tag within tag does not seem to be in > epoch > >> > microseconds. so it does not seem that's where i should be looking for > >> > that > >> > info. > >> > > >> > Here's an example of the tag for a test call on the A-Leg: > >> > > >> > > >> > 1233942283835696 > >> > 1233942283835696 > >> > 1233942283999716 > >> > 1233942283999716 > >> > 1233942287291931 > >> > 0 > >> > 1233942303240916 > >> > > >> > > >> > any hint is appreciated > >> > > Perhaps I'm missing something but they sure look like epoch microseconds to > me. > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090213/51a2ae45/attachment-0002.html From sicfslist at gmail.com Fri Feb 13 06:47:36 2009 From: sicfslist at gmail.com (Shelby Ramsey) Date: Fri, 13 Feb 2009 08:47:36 -0600 Subject: [Freeswitch-users] Sending channel variables In-Reply-To: <1234527001.5507.14.camel@opos20.altron.lan> References: <1234527001.5507.14.camel@opos20.altron.lan> Message-ID: <35b355e90902130647k1726383co4ea360f2fb3db628@mail.gmail.com> I'm assuming that you are saying these are 2 boxes .... if the protocol is a sip you can append a sip header ... _sip_h_X- .... This should be available as a channel variable on FS A. SDR On Fri, Feb 13, 2009 at 6:10 AM, Evgeniy Zolotov wrote: > Hello! > > I'm trying to make such scheme: > > ---> FS_A --> FS_B --> record > > Incoming calls to FS_A are redirected to FS_B with the help of this > context: > > > > > > data="sofia/outbound/$1 at 1.2.3.4:5080" /> > > > > > FS_B records them to the file: > > > > > > data="$${base_dir}/recordings/test/testrec.wav" /> > > > > This works good. But I have a question - in what manner I can send back > (from FS_B to FS_A) some channel variables? > > Thanks, Evgeniy. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090213/70717dde/attachment-0002.html From alex at sinapticode.ro Fri Feb 13 06:53:38 2009 From: alex at sinapticode.ro (Alexandru Nedelcu) Date: Fri, 13 Feb 2009 16:53:38 +0200 Subject: [Freeswitch-users] http://alexn.org/docs/dialer.html (was: Setting outbound callerid using js) In-Reply-To: <1234535136.4431.70.camel@gathern.lan> References: <87f2f3b90902111626x6db5dc61t68a0170a3460b42d@mail.gmail.com> <191c3a030902130605m5e834e3dh27cc88387408b8b5@mail.gmail.com> <1234535136.4431.70.camel@gathern.lan> Message-ID: <1234536818.4431.81.camel@gathern.lan> Btw ... I fixed the document. Sorry about that guys, I'm a rookie and I thought other people would find my setup useful. Can you guys read it and tell me if it contains other mistakes? My intention was to publish it on the wiki once it was ready, but I temporarily moved on to another project. http://alexn.org/docs/dialer.html Thanks, On Fri, 2009-02-13 at 16:25 +0200, Alexandru Nedelcu wrote: > I wrote that document ... I can't remember from where I got the idea > that you should specify the a-leg as being the same session. > > That document is a draft, but it got indexed by Google unfortunately :( > > > On Fri, 2009-02-13 at 14:12 +0000, Nik Middleton wrote: > > I think this page (external) is the source > > > > http://alexn.org/docs/dialer.html From ivdreg at gmail.com Fri Feb 13 06:57:48 2009 From: ivdreg at gmail.com (ivdreg ivdreg) Date: Fri, 13 Feb 2009 16:57:48 +0200 Subject: [Freeswitch-users] Codec negotiation questions In-Reply-To: <191c3a030902121306l1dae21a3m614f400703d15bd2@mail.gmail.com> References: <191c3a030902121306l1dae21a3m614f400703d15bd2@mail.gmail.com> Message-ID: Hi Anthony, Excuse me if I'm wrong but inbound-late-negotiation must be used proxy_media as I see in documentation. I don't want to proxy media because of some issues with MOH or 3-way conferencing. Also I want to exclude media codecs that are supported only in pass-trough mode. Let mi give you an example: SDP from caller v=0 o=- 1 2 IN IP4 192.168.20.193 s=CounterPath eyeBeam 1.5 c=IN IP4 192.168.40.81 t=0 0 m=audio 56888 RTP/AVP 100 106 97 105 98 3 101 a=fmtp:101 0-15 a=rtpmap:100 SPEEX/16000 a=rtpmap:106 SPEEX-FEC/16000 a=rtpmap:97 SPEEX/8000 a=rtpmap:105 SPEEX-FEC/8000 a=rtpmap:98 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv a=x-rtp-session-id:61940567309B49E8909127E1393A966E m=video 46378 RTP/AVP 125 115 34 a=fmtp:125 profile-level-id=42e015; max-br=4000; max-mbps=19800 a=fmtp:115 QCIF=1 MAXBR=4520 a=fmtp:34 QCIF=1 MAXBR=4520 a=rtpmap:125 H264/90000 a=rtpmap:115 H263-1998/90000 a=rtpmap:34 H263/90000 a=sendrecv a=x-rtp-session-id:E8244E608F65445BA183BBE641C5DF3C a=nortpproxy:yes SDP from Freeswitch to called v=0 o=FreeSWITCH 6228154995644782318 4633980766357417433 IN IP4 10.10.10.10 s=FreeSWITCH c=IN IP4 10.10.10.10 t=0 0 m=audio 26920 RTP/AVP 3 101 13 * a=rtpmap:3 GSM/8000* a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 So we offer *only first* in codec preference list from called wich is normal receives SIP/2.0 488 Not Acceptable Here *called suports - PCMA,PCMU,iLBC * Codec preference to this vars.xml we have witch is used in provile: also we have in profile: In dialplan I've set: About my second question: Why I should parse variable_switch_r_sdp: [v=0 o=- 6 2 IN IP4 192.168.20.193 s=CounterPath eyeBeam 1.5 c=IN IP4 192.168.40.81 t=0 0 m=audio 60642 RTP/AVP 100 106 97 105 98 3 101 a=rtpmap:100 SPEEX/16000 a=rtpmap:106 SPEEX-FEC/16000 a=rtpmap:97 SPEEX/8000 a=rtpmap:105 SPEEX-FEC/8000 a=rtpmap:98 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=x-rtp-session-id:0674790128EB43ACB8C7F55829BCFF14 m=video 44938 RTP/AVP 125 115 34 a=rtpmap:125 H264/90000 a=fmtp:125 profile-level-id=42e015; max-br=4000; max-mbps=19800 a=rtpmap:115 H263-1998/90000 a=fmtp:115 QCIF=1 MAXBR=4520 a=rtpmap:34 H263/90000 a=fmtp:34 QCIF=1 MAXBR=4520 a=x-rtp-session-id:D1EAE543055746AC9C03006B91ADE6DF a=nortpproxy:yes ] In FS core this parse is already done I'm sure in much more intelligent way. It can be exported as a variable like a absolute codec string I think. Thanks again. 2009/2/12 Anthony Minessale > the entire sdp is available as a variable (route the call to the info app > to see the variables) > so if you have inbound-late-negotiation set to true on the sip profile > then you can use a regex or a script to set absolute_codec string before > you answer. > > > On Thu, Feb 12, 2009 at 8:06 AM, ivdreg ivdreg wrote: > >> Hi all, >> >> Can I ask 2 questions about codec negotiation: >> >> 1. Is it possible Freeswitch to work negotiate codecs between two phones >> as it is described below. >> INVITE from A with some codecs in SDP ---> Freeswitch rewrites codec >> preference according absolute_codec_string but exclude all codecs not >> offered by A ----> INVITE to B with rewrited SDP. >> >> example: >> from A SDP:PCMA,PCMU,SPEEX ----> absolute_codec_string=G722,PCMU,PCMA,GSM >> ----> to B SDP: PCMU,PCMA >> >> 2. Can I get codec list in INVITE with mod_perl for example or via >> xml_curl without processing SDP variable (switch_r_sdp). It will be useful >> to be in format that absolute_codec_string variable takes. >> >> Thanks >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090213/c27eeee3/attachment-0002.html From helmut.kuper at ewetel.de Fri Feb 13 07:00:02 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Fri, 13 Feb 2009 16:00:02 +0100 Subject: [Freeswitch-users] Question: SIP BYE authentication In-Reply-To: References: <4992C772.4090906@ewetel.de> <4993DF0D.9000403@ewetel.de> Message-ID: <49958AF2.4010407@ewetel.de> Hi, yes, I'm using gateway, but it ignores softswitch side challenges for BYE messages coming from FS. My dialplan: regards helmut On 12.02.2009 18:01, Michael Jerris wrote: > If using gayeway it should already do this. > > On Feb 12, 2009, at 3:34 AM, Helmut Kuper > wrote: > From anthony.minessale at gmail.com Fri Feb 13 07:27:29 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 13 Feb 2009 09:27:29 -0600 Subject: [Freeswitch-users] Setting outbound callerid using js In-Reply-To: <1234535136.4431.70.camel@gathern.lan> References: <87f2f3b90902111626x6db5dc61t68a0170a3460b42d@mail.gmail.com> <191c3a030902130605m5e834e3dh27cc88387408b8b5@mail.gmail.com> <1234535136.4431.70.camel@gathern.lan> Message-ID: <191c3a030902130727y132a55c2v8fb27e87576154da@mail.gmail.com> I made it an error now to do it this way which should clear things up. Doing it that way probably led to instability in js. On Fri, Feb 13, 2009 at 8:25 AM, Alexandru Nedelcu wrote: > I wrote that document ... I can't remember from where I got the idea > that you should specify the a-leg as being the same session. > > That document is a draft, but it got indexed by Google unfortunately :( > > > On Fri, 2009-02-13 at 14:12 +0000, Nik Middleton wrote: > > I think this page (external) is the source > > > > > > > > http://alexn.org/docs/dialer.html > > > > > > > > Regards, > > > > > > > > > > ______________________________________________________________________ > > From:freeswitch-users-bounces at lists.freeswitch.org > > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > > Anthony Minessale > > Sent: 13 February 2009 14:06 > > To: freeswitch-users at lists.freeswitch.org > > Subject: Re: [Freeswitch-users] Setting outbound callerid using js > > > > > > > > > > 1) session.originate is depricated. > > 2) the first arg to session.originate is *another* session (not the > > same one) *or* undefined..... > > session.originate(undefined, ""); > > session.originate(a_leg_session, ""); > > > > session.originate(session, "") is asking the session to > > use itself as it's own a leg which makes no sense. > > > > This is perhaps the 4th time i have seen someone do this, can you > > point out where this is incorrectly documented? > > > > BTW > > > > effective_caller_id_name/number are variables you set on the A leg so > > when it's used to generate b legs that var is copied instead. > > > > a_leg_session.setVariable("effective_caller_id_number=1234"); > > b_leg_session = new Session(a_leg_session, ""); > > > > which is of course pointless because you never need to create the > > session if you just use the bridge application. > > > > > > session.execute("bridge", ""); > > > > even better just set the dest to a var and exit the script and use > > that var in your dialplan. > > > > > > --- contents of get_dest.js --- > > session.setVariable("dial_string", ""); > > > > -- dialplan -- > > > > > the JavaScript paradigm is a bit different than the Lua/Perl one. Let > > us know if that works or not. > > -MC > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090213/7c2ec5e7/attachment-0002.html From saeedahmad1981 at gmail.com Fri Feb 13 07:36:50 2009 From: saeedahmad1981 at gmail.com (Saeed Ahmed) Date: Fri, 13 Feb 2009 16:36:50 +0100 Subject: [Freeswitch-users] FS + Call Center Solution In-Reply-To: <87f2f3b90902121202m5652f0e3x42eabb6ced2fd646@mail.gmail.com> References: <6309E7515E4F43159B9564800564B562@SaeedLaptop> <87f2f3b90902121202m5652f0e3x42eabb6ced2fd646@mail.gmail.com> Message-ID: <3BF8F4AE71434041994251CFC355785C@SaeedLaptop> Thanks -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Thursday, February 12, 2009 9:02 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] FS + Call Center Solution On Wed, Feb 11, 2009 at 8:31 AM, Saeed Ahmed wrote: > Hi List, > > Is there any open source call center tool available which works with FS? Check this out: http://opencsm.org/wiki/index.php/Spice_Telephony -MC _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From anthony.minessale at gmail.com Fri Feb 13 07:40:35 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 13 Feb 2009 09:40:35 -0600 Subject: [Freeswitch-users] Question: SIP BYE authentication In-Reply-To: <49958AF2.4010407@ewetel.de> References: <4992C772.4090906@ewetel.de> <4993DF0D.9000403@ewetel.de> <49958AF2.4010407@ewetel.de> Message-ID: <191c3a030902130740y201f1a5dr573e382f80e49111@mail.gmail.com> Turn up debug and look harder are you sure it does not say "no matching gateway" when it gets the challenge to bye? On Fri, Feb 13, 2009 at 9:00 AM, Helmut Kuper wrote: > Hi, > > yes, I'm using gateway, but it ignores softswitch side challenges for > BYE messages coming from FS. > > My dialplan: > > expression="^0([0-9]+|^940[0-9]+)" break="on-false"> > > > expression="^(4918|4919)$"> > > data="ignore_early_media=true"/> > data="effective_caller_id_name="/> > data="effective_caller_id_number=1234$1"/> > data="sofia/gateway/SIPDDI/${dialed}@sip2.ewetel.net > "/> > > > > regards > helmut > > > On 12.02.2009 18:01, Michael Jerris wrote: > > If using gayeway it should already do this. > > > > On Feb 12, 2009, at 3:34 AM, Helmut Kuper > > wrote: > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090213/075dc2db/attachment-0002.html From anthony.minessale at gmail.com Fri Feb 13 07:46:15 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 13 Feb 2009 09:46:15 -0600 Subject: [Freeswitch-users] Codec negotiation questions In-Reply-To: References: <191c3a030902121306l1dae21a3m614f400703d15bd2@mail.gmail.com> Message-ID: <191c3a030902130746h23018e17n5e5c6c4900233097@mail.gmail.com> yes you are wrong. inbound-late-negotiation setting delays the codec negotiation until the instant audio is needed. It is not tied to inbound-proxy-media. This allows the call to come into the dialplan before any codec negotiation is done giving you a chance to look at the SDP before the negotiation takes place and insert an absolute_codec string essentially letting you chose unique codec preferences per inbound call. >>> Why I should parse variable_switch_r_sdp Well....you must parse it because it's you who cares about what it says, as described above it lets you peek at the sdp and enforce a unique set of codec prefs per call. On Fri, Feb 13, 2009 at 8:57 AM, ivdreg ivdreg wrote: > Hi Anthony, > > Excuse me if I'm wrong but inbound-late-negotiation must be used > proxy_media as I see in documentation. I don't want to proxy media because > of some issues with MOH or 3-way conferencing. Also I want to exclude media > codecs that are supported only in pass-trough mode. Let mi give you an > example: > > SDP from caller > > v=0 > o=- 1 2 IN IP4 192.168.20.193 > s=CounterPath eyeBeam 1.5 > c=IN IP4 192.168.40.81 > t=0 0 > m=audio 56888 RTP/AVP 100 106 97 105 98 3 101 > a=fmtp:101 0-15 > a=rtpmap:100 SPEEX/16000 > a=rtpmap:106 SPEEX-FEC/16000 > a=rtpmap:97 SPEEX/8000 > a=rtpmap:105 SPEEX-FEC/8000 > a=rtpmap:98 iLBC/8000 > a=rtpmap:101 telephone-event/8000 > a=sendrecv > a=x-rtp-session-id:61940567309B49E8909127E1393A966E > m=video 46378 RTP/AVP 125 115 34 > a=fmtp:125 profile-level-id=42e015; max-br=4000; max-mbps=19800 > a=fmtp:115 QCIF=1 MAXBR=4520 > a=fmtp:34 QCIF=1 MAXBR=4520 > a=rtpmap:125 H264/90000 > a=rtpmap:115 H263-1998/90000 > a=rtpmap:34 H263/90000 > a=sendrecv > a=x-rtp-session-id:E8244E608F65445BA183BBE641C5DF3C > a=nortpproxy:yes > > SDP from Freeswitch to called > > v=0 > o=FreeSWITCH 6228154995644782318 4633980766357417433 IN IP4 10.10.10.10 > s=FreeSWITCH > c=IN IP4 10.10.10.10 > t=0 0 > m=audio 26920 RTP/AVP 3 101 13 > * a=rtpmap:3 GSM/8000* > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=rtpmap:13 CN/8000 > a=ptime:20 > > So we offer *only first* in codec preference list > > from called wich is normal receives > SIP/2.0 488 Not Acceptable Here > *called suports - PCMA,PCMU,iLBC > * > Codec preference to this vars.xml we have witch is used in provile: > > also we have in profile: > > > In dialplan I've set: > > > > About my second question: > Why I should parse variable_switch_r_sdp: [v=0 > o=- 6 2 IN IP4 192.168.20.193 > s=CounterPath eyeBeam 1.5 > c=IN IP4 192.168.40.81 > t=0 0 > m=audio 60642 RTP/AVP 100 106 97 105 98 3 101 > a=rtpmap:100 SPEEX/16000 > a=rtpmap:106 SPEEX-FEC/16000 > a=rtpmap:97 SPEEX/8000 > a=rtpmap:105 SPEEX-FEC/8000 > a=rtpmap:98 iLBC/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=x-rtp-session-id:0674790128EB43ACB8C7F55829BCFF14 > m=video 44938 RTP/AVP 125 115 34 > a=rtpmap:125 H264/90000 > a=fmtp:125 profile-level-id=42e015; max-br=4000; max-mbps=19800 > a=rtpmap:115 H263-1998/90000 > a=fmtp:115 QCIF=1 MAXBR=4520 > a=rtpmap:34 H263/90000 > a=fmtp:34 QCIF=1 MAXBR=4520 > a=x-rtp-session-id:D1EAE543055746AC9C03006B91ADE6DF > a=nortpproxy:yes > ] > In FS core this parse is already done I'm sure in much more intelligent > way. It can be exported as a variable like a absolute codec string I think. > > Thanks again. > > 2009/2/12 Anthony Minessale > > the entire sdp is available as a variable (route the call to the info app >> to see the variables) >> so if you have inbound-late-negotiation set to true on the sip profile >> then you can use a regex or a script to set absolute_codec string before >> you answer. >> >> >> On Thu, Feb 12, 2009 at 8:06 AM, ivdreg ivdreg wrote: >> >>> Hi all, >>> >>> Can I ask 2 questions about codec negotiation: >>> >>> 1. Is it possible Freeswitch to work negotiate codecs between two phones >>> as it is described below. >>> INVITE from A with some codecs in SDP ---> Freeswitch rewrites codec >>> preference according absolute_codec_string but exclude all codecs not >>> offered by A ----> INVITE to B with rewrited SDP. >>> >>> example: >>> from A SDP:PCMA,PCMU,SPEEX ----> absolute_codec_string=G722,PCMU,PCMA,GSM >>> ----> to B SDP: PCMU,PCMA >>> >>> 2. Can I get codec list in INVITE with mod_perl for example or via >>> xml_curl without processing SDP variable (switch_r_sdp). It will be useful >>> to be in format that absolute_codec_string variable takes. >>> >>> Thanks >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:213-799-1400 >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090213/b4d736f6/attachment-0002.html From anthony.minessale at gmail.com Fri Feb 13 07:57:38 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 13 Feb 2009 09:57:38 -0600 Subject: [Freeswitch-users] xml_cdr call flow In-Reply-To: References: <87f2f3b90902121500m2b86efc2q722314688d532648@mail.gmail.com> <87f2f3b90902121608x217f682v289eae73012e27a5@mail.gmail.com> Message-ID: <191c3a030902130757v757dfa3foaaf35e64fbfb5c82@mail.gmail.com> each b leg call on the a leg shows up in a tag At the bottom of http://pastebin.freeswitch.org/7206 1233942283835696 1233942283835696 1233942283999716 1233942283999716 1233942287291931 0 1233942303240916 note transfer time, this is the time that the call was transferred to another extension. 1233942303240916 divide this number by one million to get epoch time 1233942303240916 / 1000000 == 1233942303 now look at http://pastebin.freeswitch.org/7204 at the bottom 1233942303325062 1233942303325062 1233942303368916 0 0 1233942333010768 0 This is the b leg, see the created_time: 1233942303325062 1233942303325062 / 1000000 == 1233942303 so as you can see the epoch time of your b leg cdr has a created_time that is the same one second window that corresponds to the transfer_time in the callflow tag in your a leg cdr On Fri, Feb 13, 2009 at 8:42 AM, Luis F Urrea wrote: > My mistake, they do seem to be microsecs. > > > But still I cannot correlate times from the A-leg with the B-legs. > > I have included below the xml_cdr files generated for the test call. > > The test call was made using three registered extensions. Basically, Ext > 201 calls ext 203 and they talk, then 203 blindly transfers to 202, 202 does > not answer and call rolls to voicemail. > > A-leg: > http://pastebin.freeswitch.org/7206 > > B-leg: > http://pastebin.freeswitch.org/7204 > > B-leg: > http://pastebin.freeswitch.org/7205 > > > Thanks for your help > > > On Thu, Feb 12, 2009 at 6:08 PM, Michael Collins wrote: > >> On Thu, Feb 12, 2009 at 3:31 PM, Luis F Urrea wrote: >> > Heres pastebin of the A-leg >> > >> > http://pastebin.com/m6731913d >> > >> > >> > On Thu, Feb 12, 2009 at 5:00 PM, Michael Collins >> wrote: >> >> >> >> Pastebin the whole file so that we can see it in context... >> >> -MC >> >> >> >> On Thu, Feb 12, 2009 at 2:50 PM, Luis F Urrea >> wrote: >> >> > On our test calls we haven't been able to correlate times from the A >> leg >> >> > with times from the B leg. >> >> > >> >> > I would expect something as A-leg(duration)= >> >> > B-leg1(duration)+B-leg2(duration) >> I don't know that this is necessarily true. Can you pastebin your >> dialplan entry (or whatever generated this call) so we can take a >> look? Also, please use our pastebin so that it's easier for us to find >> stuff: >> http://pastebin.freeswitch.org >> >> >> > >> >> > Also the tag within tag does not seem to be in >> epoch >> >> > microseconds. so it does not seem that's where i should be looking >> for >> >> > that >> >> > info. >> >> > >> >> > Here's an example of the tag for a test call on the A-Leg: >> >> > >> >> > >> >> > 1233942283835696 >> >> > 1233942283835696 >> >> > 1233942283999716 >> >> > 1233942283999716 >> >> > 1233942287291931 >> >> > 0 >> >> > 1233942303240916 >> >> > >> >> > >> >> > any hint is appreciated >> >> > >> Perhaps I'm missing something but they sure look like epoch microseconds >> to me. >> -MC >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090213/83c53fb0/attachment-0002.html From helmut.kuper at ewetel.de Fri Feb 13 08:03:02 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Fri, 13 Feb 2009 17:03:02 +0100 Subject: [Freeswitch-users] Question: SIP BYE authentication In-Reply-To: <191c3a030902130740y201f1a5dr573e382f80e49111@mail.gmail.com> References: <4992C772.4090906@ewetel.de> <4993DF0D.9000403@ewetel.de> <49958AF2.4010407@ewetel.de> <191c3a030902130740y201f1a5dr573e382f80e49111@mail.gmail.com> Message-ID: <499599B6.2030106@ewetel.de> Hi Anthony, yes you are right, sorry. 2009-02-13 16:56:20 [ERR] sofia_reg.c:1358 sofia_reg_handle_sip_r_challenge() No Matching gateway found works now :) thx regards helmut On 13.02.2009 16:40, Anthony Minessale wrote: > Turn up debug and look harder are you sure it does not say "no > matching gateway" when it gets the challenge to bye? From ivdreg at gmail.com Fri Feb 13 08:50:00 2009 From: ivdreg at gmail.com (ivdreg ivdreg) Date: Fri, 13 Feb 2009 18:50:00 +0200 Subject: [Freeswitch-users] Codec negotiation questions In-Reply-To: <191c3a030902130746h23018e17n5e5c6c4900233097@mail.gmail.com> References: <191c3a030902121306l1dae21a3m614f400703d15bd2@mail.gmail.com> <191c3a030902130746h23018e17n5e5c6c4900233097@mail.gmail.com> Message-ID: Hi Anthony, I'm not sure that you understood the problem. As it shown bellow the offered codec in leg B contains only one codec (first matched in codec preference list for this profile). Is there way to offer in leg B not only first codec but all codecs that exists in INVITE in leg A that matches codec preference list. If not is the only way is to parse SDP and set absolute_codec_string manualy? Regards 2009/2/13 Anthony Minessale > yes you are wrong. > > inbound-late-negotiation setting delays the codec negotiation until the > instant audio is needed. > It is not tied to inbound-proxy-media. > > > This allows the call to come into the dialplan before any codec negotiation > is done giving you a chance to look at the SDP before the negotiation takes > place and insert an absolute_codec string essentially letting you chose > unique codec preferences per inbound call. > > >>> Why I should parse variable_switch_r_sdp > > Well....you must parse it because it's you who cares about what it says, as > described above it lets you peek at the sdp and enforce a unique set of > codec prefs per call. > > > > > On Fri, Feb 13, 2009 at 8:57 AM, ivdreg ivdreg wrote: > >> Hi Anthony, >> >> Excuse me if I'm wrong but inbound-late-negotiation must be used >> proxy_media as I see in documentation. I don't want to proxy media because >> of some issues with MOH or 3-way conferencing. Also I want to exclude media >> codecs that are supported only in pass-trough mode. Let mi give you an >> example: >> >> SDP from caller >> >> v=0 >> o=- 1 2 IN IP4 192.168.20.193 >> s=CounterPath eyeBeam 1.5 >> c=IN IP4 192.168.40.81 >> t=0 0 >> m=audio 56888 RTP/AVP 100 106 97 105 98 3 101 >> a=fmtp:101 0-15 >> a=rtpmap:100 SPEEX/16000 >> a=rtpmap:106 SPEEX-FEC/16000 >> a=rtpmap:97 SPEEX/8000 >> a=rtpmap:105 SPEEX-FEC/8000 >> a=rtpmap:98 iLBC/8000 >> a=rtpmap:101 telephone-event/8000 >> a=sendrecv >> a=x-rtp-session-id:61940567309B49E8909127E1393A966E >> m=video 46378 RTP/AVP 125 115 34 >> a=fmtp:125 profile-level-id=42e015; max-br=4000; max-mbps=19800 >> a=fmtp:115 QCIF=1 MAXBR=4520 >> a=fmtp:34 QCIF=1 MAXBR=4520 >> a=rtpmap:125 H264/90000 >> a=rtpmap:115 H263-1998/90000 >> a=rtpmap:34 H263/90000 >> a=sendrecv >> a=x-rtp-session-id:E8244E608F65445BA183BBE641C5DF3C >> a=nortpproxy:yes >> >> SDP from Freeswitch to called >> >> v=0 >> o=FreeSWITCH 6228154995644782318 4633980766357417433 IN IP4 10.10.10.10 >> s=FreeSWITCH >> c=IN IP4 10.10.10.10 >> t=0 0 >> m=audio 26920 RTP/AVP 3 101 13 >> * a=rtpmap:3 GSM/8000* >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> a=rtpmap:13 CN/8000 >> a=ptime:20 >> >> So we offer *only first* in codec preference list >> >> from called wich is normal receives >> SIP/2.0 488 Not Acceptable Here >> *called suports - PCMA,PCMU,iLBC >> * >> Codec preference to this vars.xml we have witch is used in provile: >> >> also we have in profile: >> >> >> In dialplan I've set: >> >> >> >> About my second question: >> Why I should parse variable_switch_r_sdp: [v=0 >> o=- 6 2 IN IP4 192.168.20.193 >> s=CounterPath eyeBeam 1.5 >> c=IN IP4 192.168.40.81 >> t=0 0 >> m=audio 60642 RTP/AVP 100 106 97 105 98 3 101 >> a=rtpmap:100 SPEEX/16000 >> a=rtpmap:106 SPEEX-FEC/16000 >> a=rtpmap:97 SPEEX/8000 >> a=rtpmap:105 SPEEX-FEC/8000 >> a=rtpmap:98 iLBC/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-15 >> a=x-rtp-session-id:0674790128EB43ACB8C7F55829BCFF14 >> m=video 44938 RTP/AVP 125 115 34 >> a=rtpmap:125 H264/90000 >> a=fmtp:125 profile-level-id=42e015; max-br=4000; max-mbps=19800 >> a=rtpmap:115 H263-1998/90000 >> a=fmtp:115 QCIF=1 MAXBR=4520 >> a=rtpmap:34 H263/90000 >> a=fmtp:34 QCIF=1 MAXBR=4520 >> a=x-rtp-session-id:D1EAE543055746AC9C03006B91ADE6DF >> a=nortpproxy:yes >> ] >> In FS core this parse is already done I'm sure in much more intelligent >> way. It can be exported as a variable like a absolute codec string I think. >> >> Thanks again. >> >> 2009/2/12 Anthony Minessale >> >> the entire sdp is available as a variable (route the call to the info app >>> to see the variables) >>> so if you have inbound-late-negotiation set to true on the sip profile >>> then you can use a regex or a script to set absolute_codec string before >>> you answer. >>> >>> >>> On Thu, Feb 12, 2009 at 8:06 AM, ivdreg ivdreg wrote: >>> >>>> Hi all, >>>> >>>> Can I ask 2 questions about codec negotiation: >>>> >>>> 1. Is it possible Freeswitch to work negotiate codecs between two phones >>>> as it is described below. >>>> INVITE from A with some codecs in SDP ---> Freeswitch rewrites codec >>>> preference according absolute_codec_string but exclude all codecs not >>>> offered by A ----> INVITE to B with rewrited SDP. >>>> >>>> example: >>>> from A SDP:PCMA,PCMU,SPEEX ----> absolute_codec_string=G722,PCMU,PCMA,GSM >>>> ----> to B SDP: PCMU,PCMA >>>> >>>> 2. Can I get codec list in INVITE with mod_perl for example or via >>>> xml_curl without processing SDP variable (switch_r_sdp). It will be useful >>>> to be in format that absolute_codec_string variable takes. >>>> >>>> Thanks >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> Anthony Minessale II >>> >>> FreeSWITCH http://www.freeswitch.org/ >>> ClueCon http://www.cluecon.com/ >>> >>> AIM: anthm >>> MSN:anthony_minessale at hotmail.com >>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>> IRC: irc.freenode.net #freeswitch >>> >>> FreeSWITCH Developer Conference >>> sip:888 at conference.freeswitch.org >>> iax:guest at conference.freeswitch.org/888 >>> googletalk:conf+888 at conference.freeswitch.org >>> pstn:213-799-1400 >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090213/edfd036b/attachment-0002.html From msc at freeswitch.org Fri Feb 13 08:57:41 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 13 Feb 2009 08:57:41 -0800 Subject: [Freeswitch-users] Ruby framework for event socket In-Reply-To: <698401620902130627l5e8dec56g5fd43bbeb8b9ac0f@mail.gmail.com> References: <698401620902130627l5e8dec56g5fd43bbeb8b9ac0f@mail.gmail.com> Message-ID: <87f2f3b90902130857g635bd8bfp37e67b96425e4bb@mail.gmail.com> On Fri, Feb 13, 2009 at 6:27 AM, Jan Kubr wrote: > Hi all, > I've created a simple framework in Ruby that you can use to talk to > Freeswitch via even socket outbound. It won't suite your needs > perfectly if you are doing anything non-trivial, but it might be a > nice starting point. > Check it out at http://github.com/jankubr/freec > Thanks for sharing your work with the community! It is appreciated. -MC From msc at freeswitch.org Fri Feb 13 09:00:08 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 13 Feb 2009 09:00:08 -0800 Subject: [Freeswitch-users] http://alexn.org/docs/dialer.html (was: Setting outbound callerid using js) In-Reply-To: <1234536818.4431.81.camel@gathern.lan> References: <87f2f3b90902111626x6db5dc61t68a0170a3460b42d@mail.gmail.com> <191c3a030902130605m5e834e3dh27cc88387408b8b5@mail.gmail.com> <1234535136.4431.70.camel@gathern.lan> <1234536818.4431.81.camel@gathern.lan> Message-ID: <87f2f3b90902130900v3f363055qafa5e28af19f71f5@mail.gmail.com> On Fri, Feb 13, 2009 at 6:53 AM, Alexandru Nedelcu wrote: > Btw ... I fixed the document. > Sorry about that guys, I'm a rookie and I thought other people would > find my setup useful. > > Can you guys read it and tell me if it contains other mistakes? My > intention was to publish it on the wiki once it was ready, but I > temporarily moved on to another project. I'll check it out. FYI, I recommend s/FreeSwitch/FreeSWITCH/g. :) -MC From anthony.minessale at gmail.com Fri Feb 13 09:21:25 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 13 Feb 2009 11:21:25 -0600 Subject: [Freeswitch-users] Codec negotiation questions In-Reply-To: References: <191c3a030902121306l1dae21a3m614f400703d15bd2@mail.gmail.com> <191c3a030902130746h23018e17n5e5c6c4900233097@mail.gmail.com> Message-ID: <191c3a030902130921o53e6752bxf90b5a3ebe65578f@mail.gmail.com> As i have already answered, no, it does not do what you want automaticly, the only way to influence codec negotiation is the way i have described. parsing the sdp string allows you to set absolute_codec_string going both ways. if you set it before you answer the channel with late negotiation enabled it will influence the codecs accepted on the inbound call. it you set it on the b leg either by using export instead of set on the a leg or putting it in {} in the dial string it controlls what codecs are offered in the outbound invite. On Fri, Feb 13, 2009 at 10:50 AM, ivdreg ivdreg wrote: > Hi Anthony, > > I'm not sure that you understood the problem. As it shown bellow the > offered codec in leg B contains only one codec (first matched in codec > preference list for this profile). Is there way to offer in leg B not only > first codec but all codecs that exists in INVITE in leg A that matches codec > preference list. If not is the only way is to parse SDP and set > absolute_codec_string manualy? > > Regards > > 2009/2/13 Anthony Minessale > > yes you are wrong. >> >> inbound-late-negotiation setting delays the codec negotiation until the >> instant audio is needed. >> It is not tied to inbound-proxy-media. >> >> >> This allows the call to come into the dialplan before any codec >> negotiation is done giving you a chance to look at the SDP before the >> negotiation takes place and insert an absolute_codec string essentially >> letting you chose unique codec preferences per inbound call. >> >> >>> Why I should parse variable_switch_r_sdp >> >> Well....you must parse it because it's you who cares about what it says, >> as described above it lets you peek at the sdp and enforce a unique set of >> codec prefs per call. >> >> >> >> >> On Fri, Feb 13, 2009 at 8:57 AM, ivdreg ivdreg wrote: >> >>> Hi Anthony, >>> >>> Excuse me if I'm wrong but inbound-late-negotiation must be used >>> proxy_media as I see in documentation. I don't want to proxy media because >>> of some issues with MOH or 3-way conferencing. Also I want to exclude media >>> codecs that are supported only in pass-trough mode. Let mi give you an >>> example: >>> >>> SDP from caller >>> >>> v=0 >>> o=- 1 2 IN IP4 192.168.20.193 >>> s=CounterPath eyeBeam 1.5 >>> c=IN IP4 192.168.40.81 >>> t=0 0 >>> m=audio 56888 RTP/AVP 100 106 97 105 98 3 101 >>> a=fmtp:101 0-15 >>> a=rtpmap:100 SPEEX/16000 >>> a=rtpmap:106 SPEEX-FEC/16000 >>> a=rtpmap:97 SPEEX/8000 >>> a=rtpmap:105 SPEEX-FEC/8000 >>> a=rtpmap:98 iLBC/8000 >>> a=rtpmap:101 telephone-event/8000 >>> a=sendrecv >>> a=x-rtp-session-id:61940567309B49E8909127E1393A966E >>> m=video 46378 RTP/AVP 125 115 34 >>> a=fmtp:125 profile-level-id=42e015; max-br=4000; max-mbps=19800 >>> a=fmtp:115 QCIF=1 MAXBR=4520 >>> a=fmtp:34 QCIF=1 MAXBR=4520 >>> a=rtpmap:125 H264/90000 >>> a=rtpmap:115 H263-1998/90000 >>> a=rtpmap:34 H263/90000 >>> a=sendrecv >>> a=x-rtp-session-id:E8244E608F65445BA183BBE641C5DF3C >>> a=nortpproxy:yes >>> >>> SDP from Freeswitch to called >>> >>> v=0 >>> o=FreeSWITCH 6228154995644782318 4633980766357417433 IN IP4 >>> 10.10.10.10 >>> s=FreeSWITCH >>> c=IN IP4 10.10.10.10 >>> t=0 0 >>> m=audio 26920 RTP/AVP 3 101 13 >>> * a=rtpmap:3 GSM/8000* >>> a=rtpmap:101 telephone-event/8000 >>> a=fmtp:101 0-16 >>> a=rtpmap:13 CN/8000 >>> a=ptime:20 >>> >>> So we offer *only first* in codec preference list >>> >>> from called wich is normal receives >>> SIP/2.0 488 Not Acceptable Here >>> *called suports - PCMA,PCMU,iLBC >>> * >>> Codec preference to this vars.xml we have witch is used in provile: >>> >>> also we have in profile: >>> >>> >>> In dialplan I've set: >>> >>> >>> >>> About my second question: >>> Why I should parse variable_switch_r_sdp: [v=0 >>> o=- 6 2 IN IP4 192.168.20.193 >>> s=CounterPath eyeBeam 1.5 >>> c=IN IP4 192.168.40.81 >>> t=0 0 >>> m=audio 60642 RTP/AVP 100 106 97 105 98 3 101 >>> a=rtpmap:100 SPEEX/16000 >>> a=rtpmap:106 SPEEX-FEC/16000 >>> a=rtpmap:97 SPEEX/8000 >>> a=rtpmap:105 SPEEX-FEC/8000 >>> a=rtpmap:98 iLBC/8000 >>> a=rtpmap:101 telephone-event/8000 >>> a=fmtp:101 0-15 >>> a=x-rtp-session-id:0674790128EB43ACB8C7F55829BCFF14 >>> m=video 44938 RTP/AVP 125 115 34 >>> a=rtpmap:125 H264/90000 >>> a=fmtp:125 profile-level-id=42e015; max-br=4000; max-mbps=19800 >>> a=rtpmap:115 H263-1998/90000 >>> a=fmtp:115 QCIF=1 MAXBR=4520 >>> a=rtpmap:34 H263/90000 >>> a=fmtp:34 QCIF=1 MAXBR=4520 >>> a=x-rtp-session-id:D1EAE543055746AC9C03006B91ADE6DF >>> a=nortpproxy:yes >>> ] >>> In FS core this parse is already done I'm sure in much more intelligent >>> way. It can be exported as a variable like a absolute codec string I think. >>> >>> Thanks again. >>> >>> 2009/2/12 Anthony Minessale >>> >>> the entire sdp is available as a variable (route the call to the info app >>>> to see the variables) >>>> so if you have inbound-late-negotiation set to true on the sip profile >>>> then you can use a regex or a script to set absolute_codec string before >>>> you answer. >>>> >>>> >>>> On Thu, Feb 12, 2009 at 8:06 AM, ivdreg ivdreg wrote: >>>> >>>>> Hi all, >>>>> >>>>> Can I ask 2 questions about codec negotiation: >>>>> >>>>> 1. Is it possible Freeswitch to work negotiate codecs between two >>>>> phones as it is described below. >>>>> INVITE from A with some codecs in SDP ---> Freeswitch rewrites codec >>>>> preference according absolute_codec_string but exclude all codecs not >>>>> offered by A ----> INVITE to B with rewrited SDP. >>>>> >>>>> example: >>>>> from A SDP:PCMA,PCMU,SPEEX ----> absolute_codec_string=G722,PCMU,PCMA,GSM >>>>> ----> to B SDP: PCMU,PCMA >>>>> >>>>> 2. Can I get codec list in INVITE with mod_perl for example or via >>>>> xml_curl without processing SDP variable (switch_r_sdp). It will be useful >>>>> to be in format that absolute_codec_string variable takes. >>>>> >>>>> Thanks >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> -- >>>> Anthony Minessale II >>>> >>>> FreeSWITCH http://www.freeswitch.org/ >>>> ClueCon http://www.cluecon.com/ >>>> >>>> AIM: anthm >>>> MSN:anthony_minessale at hotmail.com >>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>>> IRC: irc.freenode.net #freeswitch >>>> >>>> FreeSWITCH Developer Conference >>>> sip:888 at conference.freeswitch.org >>>> iax:guest at conference.freeswitch.org/888 >>>> googletalk:conf+888 at conference.freeswitch.org >>>> pstn:213-799-1400 >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:213-799-1400 >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090213/23d65dca/attachment-0002.html From msc at freeswitch.org Fri Feb 13 09:34:05 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 13 Feb 2009 09:34:05 -0800 Subject: [Freeswitch-users] Setting outbound callerid using js In-Reply-To: <191c3a030902130727y132a55c2v8fb27e87576154da@mail.gmail.com> References: <87f2f3b90902111626x6db5dc61t68a0170a3460b42d@mail.gmail.com> <191c3a030902130605m5e834e3dh27cc88387408b8b5@mail.gmail.com> <1234535136.4431.70.camel@gathern.lan> <191c3a030902130727y132a55c2v8fb27e87576154da@mail.gmail.com> Message-ID: <87f2f3b90902130934i25e52f07keff6d2f48067443b@mail.gmail.com> >> > This is perhaps the 4th time i have seen someone do this, can you >> > point out where this is incorrectly documented? FYI, I've updated this html file for the orig author and sent it to him for review. -MC From ivdreg at gmail.com Fri Feb 13 09:50:54 2009 From: ivdreg at gmail.com (ivdreg ivdreg) Date: Fri, 13 Feb 2009 19:50:54 +0200 Subject: [Freeswitch-users] Codec negotiation questions In-Reply-To: <191c3a030902130921o53e6752bxf90b5a3ebe65578f@mail.gmail.com> References: <191c3a030902121306l1dae21a3m614f400703d15bd2@mail.gmail.com> <191c3a030902130746h23018e17n5e5c6c4900233097@mail.gmail.com> <191c3a030902130921o53e6752bxf90b5a3ebe65578f@mail.gmail.com> Message-ID: Hi Antony, Can you tell me why you do codec negotiation like that. I'm just curious. If you do not have time do not reply me. Thanks a lot for your help. 2009/2/13 Anthony Minessale > As i have already answered, no, it does not do what you want automaticly, > the only way to influence codec negotiation is the way i have described. > > parsing the sdp string allows you to set absolute_codec_string going both > ways. > if you set it before you answer the channel with late negotiation enabled > it will influence the codecs accepted on the inbound call. > it you set it on the b leg either by using export instead of set on the a > leg or putting it in {} in the dial string it controlls what codecs are > offered in the outbound invite. > > > > > On Fri, Feb 13, 2009 at 10:50 AM, ivdreg ivdreg wrote: > >> Hi Anthony, >> >> I'm not sure that you understood the problem. As it shown bellow the >> offered codec in leg B contains only one codec (first matched in codec >> preference list for this profile). Is there way to offer in leg B not only >> first codec but all codecs that exists in INVITE in leg A that matches codec >> preference list. If not is the only way is to parse SDP and set >> absolute_codec_string manualy? >> >> Regards >> >> 2009/2/13 Anthony Minessale >> >> yes you are wrong. >>> >>> inbound-late-negotiation setting delays the codec negotiation until the >>> instant audio is needed. >>> It is not tied to inbound-proxy-media. >>> >>> >>> This allows the call to come into the dialplan before any codec >>> negotiation is done giving you a chance to look at the SDP before the >>> negotiation takes place and insert an absolute_codec string essentially >>> letting you chose unique codec preferences per inbound call. >>> >>> >>> Why I should parse variable_switch_r_sdp >>> >>> Well....you must parse it because it's you who cares about what it says, >>> as described above it lets you peek at the sdp and enforce a unique set of >>> codec prefs per call. >>> >>> >>> >>> >>> On Fri, Feb 13, 2009 at 8:57 AM, ivdreg ivdreg wrote: >>> >>>> Hi Anthony, >>>> >>>> Excuse me if I'm wrong but inbound-late-negotiation must be used >>>> proxy_media as I see in documentation. I don't want to proxy media because >>>> of some issues with MOH or 3-way conferencing. Also I want to exclude media >>>> codecs that are supported only in pass-trough mode. Let mi give you an >>>> example: >>>> >>>> SDP from caller >>>> >>>> v=0 >>>> o=- 1 2 IN IP4 192.168.20.193 >>>> s=CounterPath eyeBeam 1.5 >>>> c=IN IP4 192.168.40.81 >>>> t=0 0 >>>> m=audio 56888 RTP/AVP 100 106 97 105 98 3 101 >>>> a=fmtp:101 0-15 >>>> a=rtpmap:100 SPEEX/16000 >>>> a=rtpmap:106 SPEEX-FEC/16000 >>>> a=rtpmap:97 SPEEX/8000 >>>> a=rtpmap:105 SPEEX-FEC/8000 >>>> a=rtpmap:98 iLBC/8000 >>>> a=rtpmap:101 telephone-event/8000 >>>> a=sendrecv >>>> a=x-rtp-session-id:61940567309B49E8909127E1393A966E >>>> m=video 46378 RTP/AVP 125 115 34 >>>> a=fmtp:125 profile-level-id=42e015; max-br=4000; max-mbps=19800 >>>> a=fmtp:115 QCIF=1 MAXBR=4520 >>>> a=fmtp:34 QCIF=1 MAXBR=4520 >>>> a=rtpmap:125 H264/90000 >>>> a=rtpmap:115 H263-1998/90000 >>>> a=rtpmap:34 H263/90000 >>>> a=sendrecv >>>> a=x-rtp-session-id:E8244E608F65445BA183BBE641C5DF3C >>>> a=nortpproxy:yes >>>> >>>> SDP from Freeswitch to called >>>> >>>> v=0 >>>> o=FreeSWITCH 6228154995644782318 4633980766357417433 IN IP4 >>>> 10.10.10.10 >>>> s=FreeSWITCH >>>> c=IN IP4 10.10.10.10 >>>> t=0 0 >>>> m=audio 26920 RTP/AVP 3 101 13 >>>> * a=rtpmap:3 GSM/8000* >>>> a=rtpmap:101 telephone-event/8000 >>>> a=fmtp:101 0-16 >>>> a=rtpmap:13 CN/8000 >>>> a=ptime:20 >>>> >>>> So we offer *only first* in codec preference list >>>> >>>> from called wich is normal receives >>>> SIP/2.0 488 Not Acceptable Here >>>> *called suports - PCMA,PCMU,iLBC >>>> * >>>> Codec preference to this vars.xml we have witch is used in provile: >>>> >>>> also we have in profile: >>>> >>>> >>>> In dialplan I've set: >>>> >>>> >>>> >>>> About my second question: >>>> Why I should parse variable_switch_r_sdp: [v=0 >>>> o=- 6 2 IN IP4 192.168.20.193 >>>> s=CounterPath eyeBeam 1.5 >>>> c=IN IP4 192.168.40.81 >>>> t=0 0 >>>> m=audio 60642 RTP/AVP 100 106 97 105 98 3 101 >>>> a=rtpmap:100 SPEEX/16000 >>>> a=rtpmap:106 SPEEX-FEC/16000 >>>> a=rtpmap:97 SPEEX/8000 >>>> a=rtpmap:105 SPEEX-FEC/8000 >>>> a=rtpmap:98 iLBC/8000 >>>> a=rtpmap:101 telephone-event/8000 >>>> a=fmtp:101 0-15 >>>> a=x-rtp-session-id:0674790128EB43ACB8C7F55829BCFF14 >>>> m=video 44938 RTP/AVP 125 115 34 >>>> a=rtpmap:125 H264/90000 >>>> a=fmtp:125 profile-level-id=42e015; max-br=4000; max-mbps=19800 >>>> a=rtpmap:115 H263-1998/90000 >>>> a=fmtp:115 QCIF=1 MAXBR=4520 >>>> a=rtpmap:34 H263/90000 >>>> a=fmtp:34 QCIF=1 MAXBR=4520 >>>> a=x-rtp-session-id:D1EAE543055746AC9C03006B91ADE6DF >>>> a=nortpproxy:yes >>>> ] >>>> In FS core this parse is already done I'm sure in much more intelligent >>>> way. It can be exported as a variable like a absolute codec string I think. >>>> >>>> Thanks again. >>>> >>>> 2009/2/12 Anthony Minessale >>>> >>>> the entire sdp is available as a variable (route the call to the info >>>>> app to see the variables) >>>>> so if you have inbound-late-negotiation set to true on the sip profile >>>>> then you can use a regex or a script to set absolute_codec string >>>>> before you answer. >>>>> >>>>> >>>>> On Thu, Feb 12, 2009 at 8:06 AM, ivdreg ivdreg wrote: >>>>> >>>>>> Hi all, >>>>>> >>>>>> Can I ask 2 questions about codec negotiation: >>>>>> >>>>>> 1. Is it possible Freeswitch to work negotiate codecs between two >>>>>> phones as it is described below. >>>>>> INVITE from A with some codecs in SDP ---> Freeswitch rewrites codec >>>>>> preference according absolute_codec_string but exclude all codecs not >>>>>> offered by A ----> INVITE to B with rewrited SDP. >>>>>> >>>>>> example: >>>>>> from A SDP:PCMA,PCMU,SPEEX ----> absolute_codec_string=G722,PCMU,PCMA,GSM >>>>>> ----> to B SDP: PCMU,PCMA >>>>>> >>>>>> 2. Can I get codec list in INVITE with mod_perl for example or via >>>>>> xml_curl without processing SDP variable (switch_r_sdp). It will be useful >>>>>> to be in format that absolute_codec_string variable takes. >>>>>> >>>>>> Thanks >>>>>> >>>>>> _______________________________________________ >>>>>> Freeswitch-users mailing list >>>>>> Freeswitch-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>> >>>>> >>>>> -- >>>>> Anthony Minessale II >>>>> >>>>> FreeSWITCH http://www.freeswitch.org/ >>>>> ClueCon http://www.cluecon.com/ >>>>> >>>>> AIM: anthm >>>>> MSN:anthony_minessale at hotmail.com >>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>>>> IRC: irc.freenode.net #freeswitch >>>>> >>>>> FreeSWITCH Developer Conference >>>>> sip:888 at conference.freeswitch.org >>>>> iax:guest at conference.freeswitch.org/888 >>>>> googletalk:conf+888 at conference.freeswitch.org >>>>> pstn:213-799-1400 >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> Anthony Minessale II >>> >>> FreeSWITCH http://www.freeswitch.org/ >>> ClueCon http://www.cluecon.com/ >>> >>> AIM: anthm >>> MSN:anthony_minessale at hotmail.com >>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>> IRC: irc.freenode.net #freeswitch >>> >>> FreeSWITCH Developer Conference >>> sip:888 at conference.freeswitch.org >>> iax:guest at conference.freeswitch.org/888 >>> googletalk:conf+888 at conference.freeswitch.org >>> pstn:213-799-1400 >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090213/a251917a/attachment-0002.html From mike at jerris.com Fri Feb 13 10:11:21 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 13 Feb 2009 13:11:21 -0500 Subject: [Freeswitch-users] speex build issues in svn trunk. Message-ID: <0F5A9820-2EE1-4A72-BD29-D12C6B45C25C@jerris.com> I updated the version of the speex library we use in tree last night and it may cause some build issues for those with current working copies. To fix this issue you can type "make speex-reconf" MIke From gmaruzz at celliax.org Fri Feb 13 10:15:08 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Fri, 13 Feb 2009 19:15:08 +0100 Subject: [Freeswitch-users] Skypiax (Skype Network endpoint) for FreeSWITCH Message-ID: <7b197bef0902131015k456dee87x8f0f1b7059f1b49c@mail.gmail.com> Hello FreeSWITCHers, mod_skypiax is available for testing, feature requests, bug hunting. I would like to ask the help of you all to make Skypiax robust and feature full on FreeSWITCH, and particularly of Massimo Cetra (CtRiX on IRC), that has developed mod_airpe (another Skype endpoint). I've written a first documentation on Skypiax installation and usage at: http://wiki.freeswitch.org/wiki/Skypiax and there is a Jira module at: http://jira.freeswitch.org/browse/MODSKYPIAX So, please, test the software, edit the wiki page both for style and content, file bug reports and feature requests. FreeSWITCH is now the platform of first development for me, so the FreeSWITCH part of Skypiax is more tested (if any) and the code is more readable compared to the Asterisk part where lot of legacy from my other projects clutter the code. But Skypiax strive to be available as a Skype compatible endpoint for all the opensource telephony community, and in the near time the Asterisk part will be cleaned much more, and documented. As you will see, the code is made by skypiax_protocol.c (the interaction with Skype client), mod_skypiax.c (the interaction with FreeSWITCH), chan_skypiax.c (the interaction with Asterisk). Please consider me available for all infos, clarifications, discussions, etc. I would like to thanks all the peoples that helped me via mail and IRC (so bad to have different timezones, isn't?), the *very early adopters*, the testers, the patchers, and you all. Particularly Anthony Minessale, Michael Jerris, Brian West, Michael Collins, Ken Rice, Seven Du, Clif Cox, Hristo Trendev, Rehan Allah Wala, Jason Garland and Antonio Gallo. >From the wiki page (http://wiki.freeswitch.org/wiki/Skypiax): WHAT IS SKYPIAX This software (Skypiax) uses the Skype API but is not endorsed, certified or otherwise approved in any way by Skype. Skypiax is an endpoint (channel driver) that uses the Skype client as an interface to the Skype network, and allows incoming and outgoing Skype calls to/from FreeSWITCH (that can be bridged, originated, answered, etc. as in all other endpoints, e.g. sofia/SIP). Skypiax works in FreeSWITCH (FS) on both Linux and Windows, at both 8khz and 16khz (Skype client has 16khz audio I/O). Skypiax works on Asterisk too, at 8khz, on Linux and Windows (through CygWin). Think of Skypiax as similar to OpenZAP for analog lines. For each channel you need an interface (a Skype client). So, for example, two concurrent calls would need two channels, and therefor two Skype clients running on your FreeSWITCH server. If your Skype client(s) have Skype credits, then Skypiax works for SkypeOut calls as well. Sincerely, Giovanni Maruzzelli ========================================= www.celliax.org via Pierlombardo 9, 20135 Milano Italy gmaruzz at celliax dot org Cell : +39-347-2665618 Fax : +39-02-87390039 On Thu, Jan 15, 2009 at 6:39 PM, Giovanni Maruzzelli wrote: > Hi FreeSWITCH developers > > I would like to propose to the community my plans, so we can discuss > and coordinate efforts. > > I developed a couple of channel drivers for Asterisk in the past > (works on both Linux and Windows), and I would like to port them to FS > and further enhance them. > > The two endpoints are: > - Skypiax, Skype compatible, makes and receives calls to/from Skype > network and Skypeout service, using the Skype client as interface. > - Celliax, GSM and SMS endpoint, makes and receives voice calls and > SMSs to/from the GSM/CDMA network, using second hand cellphones and/or > embedded professional devices as interfaces > > My aims are: > > a) port both endpoints from Asterisk to FreeSWITCH > b) have both endpoints continue to support at least Linux and Windows on FS > c) I would like better having most of the endpoints code working for > both FreeSWITCH and Asterisk, maintaining separated the code that > interface with the GSM and Skype network, from the code that interface > with the core. > > Skypiax, the skype compatible endpoint, is a fork of celliax, the GSM > endpoint, and they share the same skeleton and logic, so porting > celliax after having ported skypiax will be easier and faster :-). > > Current situation and next steps: > 1) skypiax (http://wiki.freeswitch.org/wiki/Skypiax) is now available > for testing and debugging, needs to be polished and cleaned > 2) starting mid next week (I'll be back in office), I want to > integrate into skypiax the code and ideas from mod_airpe of Massimo > (ctrix), that has developed an alternative Skype compatible module, > and coordinate any future development with him and any other > interested developer > 3) begin the porting of celliax to FS, aiming at a pre-beta release > for Linux and Windows during February. > 4) coordinate further development of celliax with any other developer > interested in GSM, SMSs, CDMA, IDEN, AT commands, FBUS commands, > embedded devices, audio sampling > > I am gmaruzz on #freeswitch and #freeswitch-dev, you can find more > info at www.celliax.org. > > Sincerely, > > Giovanni Maruzzelli > ========================================= > www.celliax.org > via Pierlombardo 9, 20135 Milano > Italy > gmaruzz at celliax dot org > Cell : +39-347-2665618 > Fax : +39-02-87390039 > From anthony.minessale at gmail.com Fri Feb 13 10:29:31 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 13 Feb 2009 12:29:31 -0600 Subject: [Freeswitch-users] Codec negotiation questions In-Reply-To: References: <191c3a030902121306l1dae21a3m614f400703d15bd2@mail.gmail.com> <191c3a030902130746h23018e17n5e5c6c4900233097@mail.gmail.com> <191c3a030902130921o53e6752bxf90b5a3ebe65578f@mail.gmail.com> Message-ID: <191c3a030902131029g204582adxad6dd0f53be71f2c@mail.gmail.com> Do it like what? you are using FS as a b2bua, the default behavior is to make a list of codecs that are parsed on every inbound call as soon as the invite is received. if you bridge that inbound leg to an outbound leg it will again use that same list with the one used by the inbound leg as the first choice (this is a favor we do by passing the chosen codec on the a leg over to the outbound call on the b leg) if this is not satisfactory i have left you with the option to control the behaviour yourself in a script. On Fri, Feb 13, 2009 at 11:50 AM, ivdreg ivdreg wrote: > Hi Antony, > > Can you tell me why you do codec negotiation like that. I'm just curious. > If you do not have time do not reply me. > > Thanks a lot for your help. > > > 2009/2/13 Anthony Minessale > >> As i have already answered, no, it does not do what you want automaticly, >> the only way to influence codec negotiation is the way i have described. >> >> parsing the sdp string allows you to set absolute_codec_string going both >> ways. >> if you set it before you answer the channel with late negotiation enabled >> it will influence the codecs accepted on the inbound call. >> it you set it on the b leg either by using export instead of set on the a >> leg or putting it in {} in the dial string it controlls what codecs are >> offered in the outbound invite. >> >> >> >> >> On Fri, Feb 13, 2009 at 10:50 AM, ivdreg ivdreg wrote: >> >>> Hi Anthony, >>> >>> I'm not sure that you understood the problem. As it shown bellow the >>> offered codec in leg B contains only one codec (first matched in codec >>> preference list for this profile). Is there way to offer in leg B not only >>> first codec but all codecs that exists in INVITE in leg A that matches codec >>> preference list. If not is the only way is to parse SDP and set >>> absolute_codec_string manualy? >>> >>> Regards >>> >>> 2009/2/13 Anthony Minessale >>> >>> yes you are wrong. >>>> >>>> inbound-late-negotiation setting delays the codec negotiation until the >>>> instant audio is needed. >>>> It is not tied to inbound-proxy-media. >>>> >>>> >>>> This allows the call to come into the dialplan before any codec >>>> negotiation is done giving you a chance to look at the SDP before the >>>> negotiation takes place and insert an absolute_codec string essentially >>>> letting you chose unique codec preferences per inbound call. >>>> >>>> >>> Why I should parse variable_switch_r_sdp >>>> >>>> Well....you must parse it because it's you who cares about what it says, >>>> as described above it lets you peek at the sdp and enforce a unique set of >>>> codec prefs per call. >>>> >>>> >>>> >>>> >>>> On Fri, Feb 13, 2009 at 8:57 AM, ivdreg ivdreg wrote: >>>> >>>>> Hi Anthony, >>>>> >>>>> Excuse me if I'm wrong but inbound-late-negotiation must be used >>>>> proxy_media as I see in documentation. I don't want to proxy media because >>>>> of some issues with MOH or 3-way conferencing. Also I want to exclude media >>>>> codecs that are supported only in pass-trough mode. Let mi give you an >>>>> example: >>>>> >>>>> SDP from caller >>>>> >>>>> v=0 >>>>> o=- 1 2 IN IP4 192.168.20.193 >>>>> s=CounterPath eyeBeam 1.5 >>>>> c=IN IP4 192.168.40.81 >>>>> t=0 0 >>>>> m=audio 56888 RTP/AVP 100 106 97 105 98 3 101 >>>>> a=fmtp:101 0-15 >>>>> a=rtpmap:100 SPEEX/16000 >>>>> a=rtpmap:106 SPEEX-FEC/16000 >>>>> a=rtpmap:97 SPEEX/8000 >>>>> a=rtpmap:105 SPEEX-FEC/8000 >>>>> a=rtpmap:98 iLBC/8000 >>>>> a=rtpmap:101 telephone-event/8000 >>>>> a=sendrecv >>>>> a=x-rtp-session-id:61940567309B49E8909127E1393A966E >>>>> m=video 46378 RTP/AVP 125 115 34 >>>>> a=fmtp:125 profile-level-id=42e015; max-br=4000; max-mbps=19800 >>>>> a=fmtp:115 QCIF=1 MAXBR=4520 >>>>> a=fmtp:34 QCIF=1 MAXBR=4520 >>>>> a=rtpmap:125 H264/90000 >>>>> a=rtpmap:115 H263-1998/90000 >>>>> a=rtpmap:34 H263/90000 >>>>> a=sendrecv >>>>> a=x-rtp-session-id:E8244E608F65445BA183BBE641C5DF3C >>>>> a=nortpproxy:yes >>>>> >>>>> SDP from Freeswitch to called >>>>> >>>>> v=0 >>>>> o=FreeSWITCH 6228154995644782318 4633980766357417433 IN IP4 >>>>> 10.10.10.10 >>>>> s=FreeSWITCH >>>>> c=IN IP4 10.10.10.10 >>>>> t=0 0 >>>>> m=audio 26920 RTP/AVP 3 101 13 >>>>> * a=rtpmap:3 GSM/8000* >>>>> a=rtpmap:101 telephone-event/8000 >>>>> a=fmtp:101 0-16 >>>>> a=rtpmap:13 CN/8000 >>>>> a=ptime:20 >>>>> >>>>> So we offer *only first* in codec preference list >>>>> >>>>> from called wich is normal receives >>>>> SIP/2.0 488 Not Acceptable Here >>>>> *called suports - PCMA,PCMU,iLBC >>>>> * >>>>> Codec preference to this vars.xml we have witch is used in provile: >>>>> >>>>> also we have in profile: >>>>> >>>>> >>>>> In dialplan I've set: >>>>> >>>>> >>>>> >>>>> About my second question: >>>>> Why I should parse variable_switch_r_sdp: [v=0 >>>>> o=- 6 2 IN IP4 192.168.20.193 >>>>> s=CounterPath eyeBeam 1.5 >>>>> c=IN IP4 192.168.40.81 >>>>> t=0 0 >>>>> m=audio 60642 RTP/AVP 100 106 97 105 98 3 101 >>>>> a=rtpmap:100 SPEEX/16000 >>>>> a=rtpmap:106 SPEEX-FEC/16000 >>>>> a=rtpmap:97 SPEEX/8000 >>>>> a=rtpmap:105 SPEEX-FEC/8000 >>>>> a=rtpmap:98 iLBC/8000 >>>>> a=rtpmap:101 telephone-event/8000 >>>>> a=fmtp:101 0-15 >>>>> a=x-rtp-session-id:0674790128EB43ACB8C7F55829BCFF14 >>>>> m=video 44938 RTP/AVP 125 115 34 >>>>> a=rtpmap:125 H264/90000 >>>>> a=fmtp:125 profile-level-id=42e015; max-br=4000; max-mbps=19800 >>>>> a=rtpmap:115 H263-1998/90000 >>>>> a=fmtp:115 QCIF=1 MAXBR=4520 >>>>> a=rtpmap:34 H263/90000 >>>>> a=fmtp:34 QCIF=1 MAXBR=4520 >>>>> a=x-rtp-session-id:D1EAE543055746AC9C03006B91ADE6DF >>>>> a=nortpproxy:yes >>>>> ] >>>>> In FS core this parse is already done I'm sure in much more intelligent >>>>> way. It can be exported as a variable like a absolute codec string I think. >>>>> >>>>> Thanks again. >>>>> >>>>> 2009/2/12 Anthony Minessale >>>>> >>>>> the entire sdp is available as a variable (route the call to the info >>>>>> app to see the variables) >>>>>> so if you have inbound-late-negotiation set to true on the sip profile >>>>>> then you can use a regex or a script to set absolute_codec string >>>>>> before you answer. >>>>>> >>>>>> >>>>>> On Thu, Feb 12, 2009 at 8:06 AM, ivdreg ivdreg wrote: >>>>>> >>>>>>> Hi all, >>>>>>> >>>>>>> Can I ask 2 questions about codec negotiation: >>>>>>> >>>>>>> 1. Is it possible Freeswitch to work negotiate codecs between two >>>>>>> phones as it is described below. >>>>>>> INVITE from A with some codecs in SDP ---> Freeswitch rewrites codec >>>>>>> preference according absolute_codec_string but exclude all codecs not >>>>>>> offered by A ----> INVITE to B with rewrited SDP. >>>>>>> >>>>>>> example: >>>>>>> from A SDP:PCMA,PCMU,SPEEX ----> absolute_codec_string=G722,PCMU,PCMA,GSM >>>>>>> ----> to B SDP: PCMU,PCMA >>>>>>> >>>>>>> 2. Can I get codec list in INVITE with mod_perl for example or via >>>>>>> xml_curl without processing SDP variable (switch_r_sdp). It will be useful >>>>>>> to be in format that absolute_codec_string variable takes. >>>>>>> >>>>>>> Thanks >>>>>>> >>>>>>> _______________________________________________ >>>>>>> Freeswitch-users mailing list >>>>>>> Freeswitch-users at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE: >>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> -- >>>>>> Anthony Minessale II >>>>>> >>>>>> FreeSWITCH http://www.freeswitch.org/ >>>>>> ClueCon http://www.cluecon.com/ >>>>>> >>>>>> AIM: anthm >>>>>> MSN:anthony_minessale at hotmail.com >>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>>>>> IRC: irc.freenode.net #freeswitch >>>>>> >>>>>> FreeSWITCH Developer Conference >>>>>> sip:888 at conference.freeswitch.org >>>>>> iax:guest at conference.freeswitch.org/888 >>>>>> googletalk:conf+888 at conference.freeswitch.org >>>>>> pstn:213-799-1400 >>>>>> >>>>>> _______________________________________________ >>>>>> Freeswitch-users mailing list >>>>>> Freeswitch-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> -- >>>> Anthony Minessale II >>>> >>>> FreeSWITCH http://www.freeswitch.org/ >>>> ClueCon http://www.cluecon.com/ >>>> >>>> AIM: anthm >>>> MSN:anthony_minessale at hotmail.com >>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>>> IRC: irc.freenode.net #freeswitch >>>> >>>> FreeSWITCH Developer Conference >>>> sip:888 at conference.freeswitch.org >>>> iax:guest at conference.freeswitch.org/888 >>>> googletalk:conf+888 at conference.freeswitch.org >>>> pstn:213-799-1400 >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:213-799-1400 >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090213/9e43d6e6/attachment-0002.html From gmaruzz at celliax.org Fri Feb 13 11:55:35 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Fri, 13 Feb 2009 20:55:35 +0100 Subject: [Freeswitch-users] speex build issues in svn trunk. In-Reply-To: <0F5A9820-2EE1-4A72-BD29-D12C6B45C25C@jerris.com> References: <0F5A9820-2EE1-4A72-BD29-D12C6B45C25C@jerris.com> Message-ID: <7b197bef0902131155l13a598b4s7c4f39a1983964c6@mail.gmail.com> Yay for the new speex with good Acoustic Echo Cancellation. I'll put it to work when I'll port Celliax, the GSM endpoint, for cancelling the sidetone that certain interfaces give back. :-) Thanks MikeJ ! Sincerely, Giovanni Maruzzelli ========================================= www.celliax.org via Pierlombardo 9, 20135 Milano Italy gmaruzz at celliax dot org Cell : +39-347-2665618 Fax : +39-02-87390039 On Fri, Feb 13, 2009 at 7:11 PM, Michael Jerris wrote: > I updated the version of the speex library we use in tree last night > and it may cause some build issues for those with current working > copies. To fix this issue you can type "make speex-reconf" > > MIke > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From msc at freeswitch.org Fri Feb 13 13:38:20 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 13 Feb 2009 13:38:20 -0800 Subject: [Freeswitch-users] INFO: Some new content on main page Message-ID: <87f2f3b90902131338n671fd82fod862314960fa8aa8@mail.gmail.com> FYI, There are a few new items on the main page: www.freeswitch.org, just in case you haven't been there lately. :) -MC From nik.middleton at noblesolutions.co.uk Fri Feb 13 14:30:28 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Fri, 13 Feb 2009 22:30:28 -0000 Subject: [Freeswitch-users] Hangup hook in js is never called Message-ID: Can't figure this one out. I've enabled a hang-up hook in js to do some cleanup. I've followed the example on the wiki, but it would appear it's never called. http://wiki.freeswitch.org/wiki/Example_Hangup_hook Is the code in error? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090213/821459cd/attachment-0002.html From msc at freeswitch.org Fri Feb 13 14:51:15 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 13 Feb 2009 14:51:15 -0800 Subject: [Freeswitch-users] Hangup hook in js is never called In-Reply-To: References: Message-ID: <87f2f3b90902131451t56f84286p13d7a0b815db0068@mail.gmail.com> > http://wiki.freeswitch.org/wiki/Example_Hangup_hook > > > > Is the code in error? > It might just be. I think you are better off using api_hangup_hook. What are you trying to do on hangup? The api_hangup_hook lets you call any API, including running a script. Here's an example that we played with today that I haven't even put on the wiki yet. I renames the wav file after the call is hung up. -MC From nik.middleton at noblesolutions.co.uk Fri Feb 13 15:04:16 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Fri, 13 Feb 2009 23:04:16 -0000 Subject: [Freeswitch-users] FS 1.0.2 Crash and burn In-Reply-To: <191c3a030902112025l401c759dife416cf34d51d134@mail.gmail.com> References: <7EA294D9-389F-4543-8F24-ED4C08600BE7@freeswitch.org><75A42DD8-9BA0-46CC-8121-04CE9DB27A27@freeswitch.org><6C1B5368-931F-4FA0-87F1-F0EE011C35DA@freeswitch.org> <191c3a030902112025l401c759dife416cf34d51d134@mail.gmail.com> Message-ID: Guess I'll have to dust off K&R. Having made the mindset leap to c++ I find C very procedural. Still would be like old times. Code I've looked at so far is very neat, but boy is there a lack of in-line comments. Haven't looked at the main source yet though. I always used to work on 3 lines of comments to 1 major line of code. Call me pedantic, but it aids maintenance. I've always thought of C as a low level language, just up from assembler, and nearly as efficient. An for me, low level is good not bad. Regards ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: 12 February 2009 04:26 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] FS 1.0.2 Crash and burn There is always C, it's actually considered a high level language by many ;) On Wed, Feb 11, 2009 at 5:50 PM, Brian West wrote: Lua has known issues with MySQL you must use latest SVN builds of the luasql driver for that to avoid it.. and still its not stellar.. the unixODBC one on the other hand works fine. /b On Feb 11, 2009, at 5:36 PM, Nik Middleton wrote: I've abandoned LUA. All sorts of problems (DTMF etc). Also reports of memory leaks when using MYSQL driver. Looking on the WIKI, JavaScript seems very well supported; PLUS DTMF works just fine (pulling my hair out on LUA) Guess I'm going to follow the path of least resistance on this one and use JS and ODBC Regards, _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090213/a98d82cb/attachment-0002.html From nik.middleton at noblesolutions.co.uk Fri Feb 13 15:07:20 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Fri, 13 Feb 2009 23:07:20 -0000 Subject: [Freeswitch-users] Hangup hook in js is never called In-Reply-To: <87f2f3b90902131451t56f84286p13d7a0b815db0068@mail.gmail.com> References: <87f2f3b90902131451t56f84286p13d7a0b815db0068@mail.gmail.com> Message-ID: I'm trying to capture the hang-up reason and write it to the db (Was it busy etc). I also close the db in that function. That way I know I don't have any open connections. This is in JavaScript BTW -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 13 February 2009 22:51 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Hangup hook in js is never called > http://wiki.freeswitch.org/wiki/Example_Hangup_hook > > > > Is the code in error? > It might just be. I think you are better off using api_hangup_hook. What are you trying to do on hangup? The api_hangup_hook lets you call any API, including running a script. Here's an example that we played with today that I haven't even put on the wiki yet. I renames the wav file after the call is hung up. -MC _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From anthony.minessale at gmail.com Fri Feb 13 15:19:59 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 13 Feb 2009 17:19:59 -0600 Subject: [Freeswitch-users] FS 1.0.2 Crash and burn In-Reply-To: References: <7EA294D9-389F-4543-8F24-ED4C08600BE7@freeswitch.org> <75A42DD8-9BA0-46CC-8121-04CE9DB27A27@freeswitch.org> <6C1B5368-931F-4FA0-87F1-F0EE011C35DA@freeswitch.org> <191c3a030902112025l401c759dife416cf34d51d134@mail.gmail.com> Message-ID: <191c3a030902131519m808654eo268f56c4728adf12@mail.gmail.com> modules can be c++ too. See mod_opal , mod_python, mod_java, mod_soundtouch, mod_managed and mod_perl all use switch_cpp.cpp a wrapper used to bridge into scripting langs. On Fri, Feb 13, 2009 at 5:04 PM, Nik Middleton < nik.middleton at noblesolutions.co.uk> wrote: > Guess I'll have to dust off K&R. Having made the mindset leap to > c++ I find C very procedural. Still would be like old times. Code I've > looked at so far is very neat, but boy is there a lack of in-line comments. > Haven't looked at the main source yet though. I always used to work on 3 > lines of comments to 1 major line of code. Call me pedantic, but it aids > maintenance. > > > > I've always thought of C as a low level language, just up from assembler, > and nearly as efficient. An for me, low level is good not bad. > > > > Regards > > > > > ------------------------------ > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony > Minessale > *Sent:* 12 February 2009 04:26 > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] FS 1.0.2 Crash and burn > > > > There is always C, it's actually considered a high level language by many > ;) > > On Wed, Feb 11, 2009 at 5:50 PM, Brian West wrote: > > Lua has known issues with MySQL you must use latest SVN builds of the > luasql driver for that to avoid it.. and still its not stellar.. the > unixODBC one on the other hand works fine. > > > > /b > > > > On Feb 11, 2009, at 5:36 PM, Nik Middleton wrote: > > > > I've abandoned LUA. > > > > All sorts of problems (DTMF etc). Also reports of memory leaks when using > MYSQL driver. > > > > Looking on the WIKI, JavaScript seems very well supported; PLUS DTMF works > just fine (pulling my hair out on LUA) > > > > Guess I'm going to follow the path of least resistance on this one and use > JS and ODBC > > > > Regards, > > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090213/1cc7ead7/attachment-0002.html From jason at jasonjgw.net Fri Feb 13 15:24:26 2009 From: jason at jasonjgw.net (Jason White) Date: Sat, 14 Feb 2009 10:24:26 +1100 Subject: [Freeswitch-users] FS 1.0.2 Crash and burn In-Reply-To: References: <191c3a030902112025l401c759dife416cf34d51d134@mail.gmail.com> Message-ID: <20090213232426.GA5549@jdc.jasonjgw.net> Nik Middleton wrote: > Code > I've looked at so far is very neat, but boy is there a lack of in-line > comments. Haven't looked at the main source yet though. I always used > to work on 3 lines of comments to 1 major line of code. Call me > pedantic, but it aids maintenance. I find the FreeSWITCH code quite readable. Public API functions have comments, which are all that we need, I think. From nik.middleton at noblesolutions.co.uk Fri Feb 13 15:40:32 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Fri, 13 Feb 2009 23:40:32 -0000 Subject: [Freeswitch-users] FS 1.0.2 Crash and burn In-Reply-To: <20090213232426.GA5549@jdc.jasonjgw.net> References: <191c3a030902112025l401c759dife416cf34d51d134@mail.gmail.com> <20090213232426.GA5549@jdc.jasonjgw.net> Message-ID: That would assume that the underlying code is perfect, which it probably isn't. Not knocking the efforts, but in my view, you can't have too much in line documentation. I hope to make a contribution shortly. Right now I'm updating the WIKI where appropriate. Top level examples should work with a cut and paste, if they don't you're going to alienate new entrants. Regards -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jason White Sent: 13 February 2009 23:24 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] FS 1.0.2 Crash and burn Nik Middleton wrote: > Code > I've looked at so far is very neat, but boy is there a lack of in-line > comments. Haven't looked at the main source yet though. I always used > to work on 3 lines of comments to 1 major line of code. Call me > pedantic, but it aids maintenance. I find the FreeSWITCH code quite readable. Public API functions have comments, which are all that we need, I think. _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From msc at freeswitch.org Fri Feb 13 16:08:44 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 13 Feb 2009 16:08:44 -0800 Subject: [Freeswitch-users] FS 1.0.2 Crash and burn In-Reply-To: References: <191c3a030902112025l401c759dife416cf34d51d134@mail.gmail.com> <20090213232426.GA5549@jdc.jasonjgw.net> Message-ID: <87f2f3b90902131608p5f8ead6fxecebac1b3baf8f52@mail.gmail.com> > Right now I'm updating the WIKI where appropriate. Top level examples > should work with a cut and paste, if they don't you're going to alienate > new entrants. This is a valid point. I will be happy to help with the wiki since documentation is kind of my bailiwick. My challenge is just being in a position to test everything. I have only so much equipment (and time) so it isn't always easy for me to set things up and do testing. But I will definitely do my best. -MC From mkarp at securesilence.com Fri Feb 13 18:17:25 2009 From: mkarp at securesilence.com (Maxim Karp) Date: Fri, 13 Feb 2009 18:17:25 -0800 Subject: [Freeswitch-users] Not getting a ring back for local extensions on a specific device Message-ID: <002901c98e4a$62ae3d40$280ab7c0$@com> Hello, I am using a SNOM 320 and Windows Mobile 6 VoIP softphone on two separate extensions. When dialing from the SNOM to the WM6 device I get ringback on the SNOM but when calling the SNOM from the WM6 device I don't get ringback though the call does complete and I get voice after the connection. Any ideas? Maxim. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090213/da97b2dc/attachment-0002.html From brian at freeswitch.org Fri Feb 13 19:21:09 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 13 Feb 2009 21:21:09 -0600 Subject: [Freeswitch-users] Not getting a ring back for local extensions on a specific device In-Reply-To: <002901c98e4a$62ae3d40$280ab7c0$@com> References: <002901c98e4a$62ae3d40$280ab7c0$@com> Message-ID: Would need a sip trace to know. TPORT_LOG=1 ./freeswitch /b On Feb 13, 2009, at 8:17 PM, Maxim Karp wrote: > Hello, > > I am using a SNOM 320 and Windows Mobile 6 VoIP softphone on two > separate extensions. When dialing from the SNOM to the WM6 device I > get ringback on the SNOM but when calling the SNOM from the WM6 > device I don?t get ringback though the call does complete and I get > voice after the connection. > > Any ideas? > > Maxim. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090213/5c30238f/attachment-0002.html From woodydickson at gmail.com Fri Feb 13 19:41:28 2009 From: woodydickson at gmail.com (Woody Dickson) Date: Sat, 14 Feb 2009 11:41:28 +0800 Subject: [Freeswitch-users] No-media problem with opensips-freeswitch setup Message-ID: Hi, I tried to configure opensips as sip proxy and sip registrars and freeswitch as B2BUA. Everything works until I start to connect sip clients that are behind ADSL. Both freeswitch and opensips are on public IP and I am using external profile as well. Does anyone have experience in setting up opensips and freeswitch together and can share the configuration with me? Thank you very much in advance for any help. Regards, Woody From brian at freeswitch.org Fri Feb 13 19:55:28 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 13 Feb 2009 21:55:28 -0600 Subject: [Freeswitch-users] No-media problem with opensips-freeswitch setup In-Reply-To: References: Message-ID: You have let the names of the profiles confuse you. Chances are you're trying to hair pin the calls out and back into the same nat. That usually doesn't work. You will need to give me more details about your setup. /b On Feb 13, 2009, at 9:41 PM, Woody Dickson wrote: > Hi, > > I tried to configure opensips as sip proxy and sip registrars and > freeswitch as B2BUA. Everything works until I start to connect sip > clients that are behind ADSL. > > Both freeswitch and opensips are on public IP and I am using external > profile as well. > > Does anyone have experience in setting up opensips and freeswitch > together and can share the configuration with me? > > Thank you very much in advance for any help. > > Regards, > Woody From wiltingtree at gmail.com Fri Feb 13 21:45:30 2009 From: wiltingtree at gmail.com (Adam Wilt) Date: Sat, 14 Feb 2009 00:45:30 -0500 Subject: [Freeswitch-users] monitoring events in Python Message-ID: I'm trying to use custom events for a conference call in a Python script. I set-up the events in the conference.conf.xml file, and I send "bgapi event plain CUSTOM conference::maintenance" to enable them. But I don't know how to look for these events in my script. Does anybody have some example code, or maybe just point me in the right direction? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090214/c4b833cc/attachment-0002.html From egghunt at gmail.com Sat Feb 14 02:27:17 2009 From: egghunt at gmail.com (Arnaldo de Moraes Pereira) Date: Sat, 14 Feb 2009 08:27:17 -0200 Subject: [Freeswitch-users] monitoring events in Python In-Reply-To: References: Message-ID: You might use freepy, which can be found in freeswitch tree, on dir scripts/socket/freepy. fseventlistener.py is a good example to follow. You'll want to pass your custom event to sniff_custom_events(). On Sat, Feb 14, 2009 at 3:45 AM, Adam Wilt wrote: > I'm trying to use custom events for a conference call in a Python script. > I set-up the events in the conference.conf.xml file, and I send "bgapi event > plain CUSTOM conference::maintenance" to enable them. But I don't know how > to look for these events in my script. Does anybody have some example code, > or maybe just point me in the right direction? Thanks. > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Arnaldo M Pereira ap at arnaldopereira.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090214/3bef0643/attachment-0002.html From nik.middleton at noblesolutions.co.uk Sat Feb 14 02:26:54 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Sat, 14 Feb 2009 10:26:54 -0000 Subject: [Freeswitch-users] Hangup hook in js is never called [RESOLVED In-Reply-To: <87f2f3b90902131451t56f84286p13d7a0b815db0068@mail.gmail.com> References: <87f2f3b90902131451t56f84286p13d7a0b815db0068@mail.gmail.com> Message-ID: The JS hook does indeed work. New to js, I hadn't declared the function prior calling it. I can only guess that java scripts are processed sequentially and do not throw up errors if a call is made to a function that hasn't been processed yet Regards, -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 13 February 2009 22:51 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Hangup hook in js is never called > http://wiki.freeswitch.org/wiki/Example_Hangup_hook > > > > Is the code in error? > It might just be. I think you are better off using api_hangup_hook. What are you trying to do on hangup? The api_hangup_hook lets you call any API, including running a script. Here's an example that we played with today that I haven't even put on the wiki yet. I renames the wav file after the call is hung up. From nik.middleton at noblesolutions.co.uk Sat Feb 14 05:17:50 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Sat, 14 Feb 2009 13:17:50 -0000 Subject: [Freeswitch-users] Problems with Originate In-Reply-To: <191c3a030902130554w1dfe6b1dw9e5525cf6a210dc4@mail.gmail.com> References: <1234524794.4431.56.camel@gathern.lan><1234525724.4431.59.camel@gathern.lan> <191c3a030902130554w1dfe6b1dw9e5525cf6a210dc4@mail.gmail.com> Message-ID: Understood. However, using the second method, how can I trap on call failure? If I originate a call and the user is busy, the console reports this fact, but then the script continues to execute if (session.ready()) { console_log("notice","Session result=[" + session.cause + "] \n"); if (session.cause == "USER_BUSY") { Disposition = "BUSY"; session.Hangup(); } In this case session.cause reports 'NONE' and what's surprising is that even though the call failed (busy) session.ready returns a true value. ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: 13 February 2009 13:55 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Problems with Originate The first way is deprecated and will be removed. The 2nd way is the correct way. On Fri, Feb 13, 2009 at 5:48 AM, Alexandru Nedelcu wrote: On Fri, 2009-02-13 at 13:33 +0200, Alexandru Nedelcu wrote: > The problem with this setup is that origination_caller_id_number doesn't > work from inside the JS file (when calling session.originate). I just discovered something interesting. When originating the call like this ... session = new Session("") instead of this ... session = new Session(); session.originate("") ... then it works. Is this some kind of bug, or what's the difference here? Thanks, -- Alexandru Nedelcu Software Developer, Sinapticode _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090214/3674f880/attachment-0002.html From woodydickson at gmail.com Sat Feb 14 06:18:58 2009 From: woodydickson at gmail.com (Woody Dickson) Date: Sat, 14 Feb 2009 22:18:58 +0800 Subject: [Freeswitch-users] No-media problem with opensips-freeswitch setup In-Reply-To: References: Message-ID: Hi My external.xml is just the default configuration: In my opensips.cfg, all the nated traffic is sent to the external_sip_ip and external_rtp_port. Is there anything I should add or change to enable media for device behind nat? Regards, Woody From anthony.minessale at gmail.com Sat Feb 14 07:49:04 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sat, 14 Feb 2009 09:49:04 -0600 Subject: [Freeswitch-users] Not getting a ring back for local extensions on a specific device In-Reply-To: <002901c98e4a$62ae3d40$280ab7c0$@com> References: <002901c98e4a$62ae3d40$280ab7c0$@com> Message-ID: <191c3a030902140749o57172558xd0833bb0a0add7a5@mail.gmail.com> If the device doesn't support 183 early media you may not hear the ringback. The example dialplan uses generated ringback tones. you could edit the default extension and comment out the lines that set the ringback variable and put in one that says "ring_ready" that would send a 180. most devices who implement sip have no idea what they have gotten themselves into and only end up implementing their one test case and not all the possible nightmares that result from trying to "interop" On Fri, Feb 13, 2009 at 8:17 PM, Maxim Karp wrote: > Hello, > > > > I am using a SNOM 320 and Windows Mobile 6 VoIP softphone on two separate > extensions. When dialing from the SNOM to the WM6 device I get ringback on > the SNOM but when calling the SNOM from the WM6 device I don't get ringback > though the call does complete and I get voice after the connection. > > > > Any ideas? > > > > Maxim. > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090214/c08965c0/attachment-0002.html From mike at yes.net.ua Sat Feb 14 02:16:28 2009 From: mike at yes.net.ua (Mike Tkachuk) Date: Sat, 14 Feb 2009 12:16:28 +0200 Subject: [Freeswitch-users] Transcoding G723 Message-ID: <102435003.20090214121628@yes.net.ua> Hello Freeswitch-users, Check this one: http://freehg.org/u/deepwalker/fs_g729/ G.729 is not G.723 but may be interesting. IPP have also g.723 implementation, not too hard to port. That code is working OK for me on development servers. -- Mike From mike at jerris.com Sat Feb 14 09:07:21 2009 From: mike at jerris.com (Michael Jerris) Date: Sat, 14 Feb 2009 12:07:21 -0500 Subject: [Freeswitch-users] monitoring events in Python In-Reply-To: References: Message-ID: <37F78651-0690-4599-B08C-BDBC24562D17@jerris.com> its just "event plain CUSTOM conference::maintenance" no "bgapi" see: http://wiki.freeswitch.org/wiki/Event_Socket for more info. MIke On Feb 14, 2009, at 12:45 AM, Adam Wilt wrote: > I'm trying to use custom events for a conference call in a Python > script. I set-up the events in the conference.conf.xml file, and I > send "bgapi event plain CUSTOM conference::maintenance" to enable > them. But I don't know how to look for these events in my script. > Does anybody have some example code, or maybe just point me in the > right direction? Thanks. From mkarp at securesilence.com Sat Feb 14 09:16:25 2009 From: mkarp at securesilence.com (Maxim Karp) Date: Sat, 14 Feb 2009 09:16:25 -0800 Subject: [Freeswitch-users] Not getting a ring back for local extensions on a specific device In-Reply-To: <191c3a030902140749o57172558xd0833bb0a0add7a5@mail.gmail.com> References: <002901c98e4a$62ae3d40$280ab7c0$@com> <191c3a030902140749o57172558xd0833bb0a0add7a5@mail.gmail.com> Message-ID: <004c01c98ec7$f9163cb0$eb42b610$@com> Hi Anthony, Thanks for the suggestion. In which config file do I change the ringback variable and what should the exact syntax be? Maxim. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Saturday, February 14, 2009 7:49 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Not getting a ring back for local extensions on a specific device If the device doesn't support 183 early media you may not hear the ringback. The example dialplan uses generated ringback tones. you could edit the default extension and comment out the lines that set the ringback variable and put in one that says "ring_ready" that would send a 180. most devices who implement sip have no idea what they have gotten themselves into and only end up implementing their one test case and not all the possible nightmares that result from trying to "interop" On Fri, Feb 13, 2009 at 8:17 PM, Maxim Karp wrote: Hello, I am using a SNOM 320 and Windows Mobile 6 VoIP softphone on two separate extensions. When dialing from the SNOM to the WM6 device I get ringback on the SNOM but when calling the SNOM from the WM6 device I don't get ringback though the call does complete and I get voice after the connection. Any ideas? Maxim. _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090214/67e9a64f/attachment-0002.html From anthony.minessale at gmail.com Sat Feb 14 09:19:50 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sat, 14 Feb 2009 11:19:50 -0600 Subject: [Freeswitch-users] Problems with Originate In-Reply-To: References: <1234524794.4431.56.camel@gathern.lan> <1234525724.4431.59.camel@gathern.lan> <191c3a030902130554w1dfe6b1dw9e5525cf6a210dc4@mail.gmail.com> Message-ID: <191c3a030902140919w806923ex2dc44c2734010251@mail.gmail.com> are you running this as a dialplan app? session is a reserved variable name for the session you executed the app on. are you using an alternate name for your new session like my_session etc....? this works for me, try it yourself. var my_session = new Session("sofia/external/7003 at conference.freeswitch.org "); consoleLog("err", "ready: " + my_session.ready() + "\n"); On Sat, Feb 14, 2009 at 7:17 AM, Nik Middleton < nik.middleton at noblesolutions.co.uk> wrote: > Understood. > > > > However, using the second method, how can I trap on call failure? > > > > If I originate a call and the user is busy, the console reports this fact, > but then the script continues to execute > > > > if (session.ready()) { > > console_log("notice","Session result=[" + > session.cause + "] \n"); > > if (session.cause == "USER_BUSY") { > > Disposition = > "BUSY"; > > session.Hangup(); > > } > > In this case session.cause reports 'NONE' and what's surprising is that > even though the call failed (busy) session.ready returns a true value. > ------------------------------ > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony > Minessale > *Sent:* 13 February 2009 13:55 > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Problems with Originate > > > > The first way is deprecated and will be removed. > The 2nd way is the correct way. > > On Fri, Feb 13, 2009 at 5:48 AM, Alexandru Nedelcu > wrote: > > On Fri, 2009-02-13 at 13:33 +0200, Alexandru Nedelcu wrote: > > The problem with this setup is that origination_caller_id_number doesn't > > work from inside the JS file (when calling session.originate). > > I just discovered something interesting. > > When originating the call like this ... > session = new Session("") > instead of this ... > session = new Session(); session.originate("") > > ... then it works. Is this some kind of bug, or what's the difference > here? > > > Thanks, > > -- > Alexandru Nedelcu > Software Developer, Sinapticode > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090214/4592692a/attachment-0002.html From anthony.minessale at gmail.com Sat Feb 14 09:20:23 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sat, 14 Feb 2009 11:20:23 -0600 Subject: [Freeswitch-users] Not getting a ring back for local extensions on a specific device In-Reply-To: <004c01c98ec7$f9163cb0$eb42b610$@com> References: <002901c98e4a$62ae3d40$280ab7c0$@com> <191c3a030902140749o57172558xd0833bb0a0add7a5@mail.gmail.com> <004c01c98ec7$f9163cb0$eb42b610$@com> Message-ID: <191c3a030902140920t6e808796ue0c0419855fea326@mail.gmail.com> dialplan/defualt.xml in the conf directory look for the word ringback On Sat, Feb 14, 2009 at 11:16 AM, Maxim Karp wrote: > Hi Anthony, > > > > Thanks for the suggestion. In which config file do I change the ringback > variable and what should the exact syntax be? > > > > Maxim. > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony > Minessale > *Sent:* Saturday, February 14, 2009 7:49 AM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Not getting a ring back for local > extensions on a specific device > > > > If the device doesn't support 183 early media you may not hear the > ringback. > The example dialplan uses generated ringback tones. > > you could edit the default extension and comment out the lines that set the > ringback variable > and put in one that says "ring_ready" that would send a 180. > > most devices who implement sip have no idea what they have gotten > themselves into and only end up > implementing their one test case and not all the possible nightmares that > result from trying to "interop" > > On Fri, Feb 13, 2009 at 8:17 PM, Maxim Karp > wrote: > > Hello, > > > > I am using a SNOM 320 and Windows Mobile 6 VoIP softphone on two separate > extensions. When dialing from the SNOM to the WM6 device I get ringback on > the SNOM but when calling the SNOM from the WM6 device I don't get ringback > though the call does complete and I get voice after the connection. > > > > Any ideas? > > > > Maxim. > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090214/c9b3793e/attachment-0002.html From anthony.minessale at gmail.com Sat Feb 14 09:28:11 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sat, 14 Feb 2009 11:28:11 -0600 Subject: [Freeswitch-users] FS 1.0.2 Crash and burn In-Reply-To: <87f2f3b90902131608p5f8ead6fxecebac1b3baf8f52@mail.gmail.com> References: <191c3a030902112025l401c759dife416cf34d51d134@mail.gmail.com> <20090213232426.GA5549@jdc.jasonjgw.net> <87f2f3b90902131608p5f8ead6fxecebac1b3baf8f52@mail.gmail.com> Message-ID: <191c3a030902140928p4f6dada0y3c7f7b3dfb9b8fde@mail.gmail.com> Maybe you can start a new thread then. On Fri, Feb 13, 2009 at 6:08 PM, Michael Collins wrote: > > Right now I'm updating the WIKI where appropriate. Top level examples > > should work with a cut and paste, if they don't you're going to alienate > > new entrants. > > This is a valid point. I will be happy to help with the wiki since > documentation is kind of my bailiwick. My challenge is just being in a > position to test everything. I have only so much equipment (and time) > so it isn't always easy for me to set things up and do testing. But I > will definitely do my best. > > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090214/32a8b7af/attachment-0002.html From nik.middleton at noblesolutions.co.uk Sat Feb 14 11:47:20 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Sat, 14 Feb 2009 19:47:20 -0000 Subject: [Freeswitch-users] Problems with Originate In-Reply-To: <191c3a030902140919w806923ex2dc44c2734010251@mail.gmail.com> References: <1234524794.4431.56.camel@gathern.lan><1234525724.4431.59.camel@gathern.lan><191c3a030902130554w1dfe6b1dw9e5525cf6a210dc4@mail.gmail.com> <191c3a030902140919w806923ex2dc44c2734010251@mail.gmail.com> Message-ID: Nope, Still not working. Here's my little test javascript var new_session = new Session('{ignore_early_media=true,}sofia/internal/1001 at 192.168.3.206'); //set the on_hangup function to be called when this session is hungup new_session.setHangupHook(on_hangup,"hup"); var on_hangup = function(hup_session, how) { console_log("err","In hangup section\n"); //exit here would end the script so you could cleanup and just be done exit(); } if (new_session.ready()) { new_session.answer( ); new_session.sleep(1500); new_session.streamFile("female2.wav"); } And this is the output [NOTICE] sofia.c:3090 sofia_handle_sip_i_state() Hangup sofia/internal/sip:1001 at 192.168.0.29:5060 [CS_CONSUME_MEDIA] [USER_BUSY] [ERR] switch_ivr_originate.c:1116 switch_ivr_originate() Cannot create outgoing channel of type [user] cause: [USER_BUSY] [INFO] mod_dptools.c:1909 audio_bridge_function() Originate Failed. Cause: USER_BUSY [NOTICE] switch_core_session.c:957 switch_core_session_thread() Session 95 (sofia/internal/sip:1001 at 192.168.0.29:5060) Ended [NOTICE] switch_core_session.c:959 switch_core_session_thread() Close Channel sofia/internal/sip:1001 at 192.168.0.29:5060 [CS_HANGUP] [NOTICE] mod_dptools.c:600 answer_function() Channel [sofia/internal/0000000000 at 192.168.3.206] has been answered ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: 14 February 2009 17:20 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Problems with Originate are you running this as a dialplan app? session is a reserved variable name for the session you executed the app on. are you using an alternate name for your new session like my_session etc....? this works for me, try it yourself. var my_session = new Session("sofia/external/7003 at conference.freeswitch.org"); consoleLog("err", "ready: " + my_session.ready() + "\n"); On Sat, Feb 14, 2009 at 7:17 AM, Nik Middleton wrote: Understood. However, using the second method, how can I trap on call failure? If I originate a call and the user is busy, the console reports this fact, but then the script continues to execute if (session.ready()) { console_log("notice","Session result=[" + session.cause + "] \n"); if (session.cause == "USER_BUSY") { Disposition = "BUSY"; session.Hangup(); } In this case session.cause reports 'NONE' and what's surprising is that even though the call failed (busy) session.ready returns a true value. ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: 13 February 2009 13:55 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Problems with Originate The first way is deprecated and will be removed. The 2nd way is the correct way. On Fri, Feb 13, 2009 at 5:48 AM, Alexandru Nedelcu wrote: On Fri, 2009-02-13 at 13:33 +0200, Alexandru Nedelcu wrote: > The problem with this setup is that origination_caller_id_number doesn't > work from inside the JS file (when calling session.originate). I just discovered something interesting. When originating the call like this ... session = new Session("") instead of this ... session = new Session(); session.originate("") ... then it works. Is this some kind of bug, or what's the difference here? Thanks, -- Alexandru Nedelcu Software Developer, Sinapticode _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090214/939795c9/attachment-0002.html From anthony.minessale at gmail.com Sat Feb 14 14:45:13 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sat, 14 Feb 2009 16:45:13 -0600 Subject: [Freeswitch-users] Problems with Originate In-Reply-To: References: <1234524794.4431.56.camel@gathern.lan> <1234525724.4431.59.camel@gathern.lan> <191c3a030902130554w1dfe6b1dw9e5525cf6a210dc4@mail.gmail.com> <191c3a030902140919w806923ex2dc44c2734010251@mail.gmail.com> Message-ID: <191c3a030902141445q7ffb1febm242c6785a9afef7c@mail.gmail.com> What do you suggest is not working? the call failed and it *did not* run the code inside if (new_session.ready()) The call was never established therefore it would not run the hangup hook either. In order to trigger the hangup hook the session would need to exist. If the session could not originate the new_session obj is an empty shell with no actual session inside. var new_session = new Session(, ); if (!new_session.ready()) { // the call never was established. } I gave you real code to try in my last email that you completely ignored..... and finally you are declaring the function wrong declare it at the top. function on_hangup(hup_session, how) { ..... } I sense you seem to think things are going to magically transform into however you are thinking they should work instead of you perhaps learning how they actually work. On Sat, Feb 14, 2009 at 1:47 PM, Nik Middleton < nik.middleton at noblesolutions.co.uk> wrote: > Nope, > > > > Still not working. Here's my little test javascript > > > > var new_session = new Session('{ignore_early_media=true,}sofia/internal/ > 1001 at 192.168.3.206'); > > > > //set the on_hangup function to be called when this session is > hungup > > > new_session.setHangupHook(on_hangup,"hup"); > > > > var on_hangup = function(hup_session, how) > { > > > console_log("err","In hangup > section\n"); > > //exit here would end the script so you > could cleanup and just be > done > > > > exit(); > > > } > > > > > > if (new_session.ready()) { > > new_session.answer( ); > > new_session.sleep(1500); > > new_session.streamFile("female2.wav"); > > } > > > > And this is the output > > > > [NOTICE] sofia.c:3090 sofia_handle_sip_i_state() Hangup sofia/internal/ > sip:1001 at 192.168.0.29:5060 [CS_CONSUME_MEDIA] [USER_BUSY] > > [ERR] switch_ivr_originate.c:1116 switch_ivr_originate() Cannot create > outgoing channel of type [user] cause: [USER_BUSY] > > [INFO] mod_dptools.c:1909 audio_bridge_function() Originate Failed. Cause: > USER_BUSY > > [NOTICE] switch_core_session.c:957 switch_core_session_thread() Session 95 > (sofia/internal/sip:1001 at 192.168.0.29:5060) Ended > > [NOTICE] switch_core_session.c:959 switch_core_session_thread() Close > Channel sofia/internal/sip:1001 at 192.168.0.29:5060 [CS_HANGUP] > > [NOTICE] mod_dptools.c:600 answer_function() Channel [sofia/internal/ > 0000000000 at 192.168.3.206] has been answered > > > ------------------------------ > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony > Minessale > *Sent:* 14 February 2009 17:20 > > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Problems with Originate > > > > are you running this as a dialplan app? > session is a reserved variable name for the session you executed the app > on. > > are you using an alternate name for your new session like my_session > etc....? > > this works for me, try it yourself. > > var my_session = new Session("sofia/external/ > 7003 at conference.freeswitch.org"); > consoleLog("err", "ready: " + my_session.ready() + "\n"); > > > > On Sat, Feb 14, 2009 at 7:17 AM, Nik Middleton < > nik.middleton at noblesolutions.co.uk> wrote: > > Understood. > > > > However, using the second method, how can I trap on call failure? > > > > If I originate a call and the user is busy, the console reports this fact, > but then the script continues to execute > > > > if (session.ready()) { > > console_log("notice","Session result=[" + > session.cause + "] \n"); > > if (session.cause == "USER_BUSY") { > > Disposition = > "BUSY"; > > session.Hangup(); > > } > > In this case session.cause reports 'NONE' and what's surprising is that > even though the call failed (busy) session.ready returns a true value. > ------------------------------ > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony > Minessale > *Sent:* 13 February 2009 13:55 > > > *To:* freeswitch-users at lists.freeswitch.org > > *Subject:* Re: [Freeswitch-users] Problems with Originate > > > > The first way is deprecated and will be removed. > The 2nd way is the correct way. > > On Fri, Feb 13, 2009 at 5:48 AM, Alexandru Nedelcu > wrote: > > On Fri, 2009-02-13 at 13:33 +0200, Alexandru Nedelcu wrote: > > The problem with this setup is that origination_caller_id_number doesn't > > work from inside the JS file (when calling session.originate). > > I just discovered something interesting. > > When originating the call like this ... > session = new Session("") > instead of this ... > session = new Session(); session.originate("") > > ... then it works. Is this some kind of bug, or what's the difference > here? > > > Thanks, > > -- > Alexandru Nedelcu > Software Developer, Sinapticode > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090214/acf28ded/attachment-0002.html From rinorhoxha at hotmail.com Sat Feb 14 15:04:01 2009 From: rinorhoxha at hotmail.com (JCATS) Date: Sat, 14 Feb 2009 15:04:01 -0800 (PST) Subject: [Freeswitch-users] [ANN] Spice Telephony - an open source FreeSWITCH/Erlang callcenter platform Message-ID: <22015518.post@talk.nabble.com> Have you planned any predictive dialer features ( like VICIDIAL )? -- View this message in context: http://www.nabble.com/-ANN--Spice-Telephony---an-open-source-FreeSWITCH-Erlang-callcenter-platform-tp21384907p22015518.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From xspambox at gmail.com Sat Feb 14 16:02:16 2009 From: xspambox at gmail.com (xs) Date: Sun, 15 Feb 2009 01:02:16 +0100 Subject: [Freeswitch-users] [newbie] Clean start with a simple configuration Message-ID: <70ad8fdb0902141602w5829a059p3fe9b4159cba499a@mail.gmail.com> Hi, I installed Freeswitch with this http://wiki.freeswitch.org/wiki/Quick_Startmanual and it works! But it has al sorts of options enabled: 5000,9995, 9996, 9999, 80+[group], 30+[conf], voicemail, something about pizza (00_pizza_demo.xml) and a lot of other xml files and directories (default.xml and default/) etc. etc. That is not what i want. When i install Asterisk (the only PBX that i have experience with) and i clean out sip.conf and extensions.conf and replace them with this: ---------------------------------------- sip.conf [general] [1001] username=1001 secret=password1001 type=friend context=phones host=dynamic qualify=yes disallow=all allow=g726 [1002] username=1002 secret=password1002 type=friend context=phones host=dynamic qualify=yes disallow=all allow=ilbc ---------------------------------------- extensions.conf ---------------------------------------- [phones] exten => _[1001-1002],1,Dial(SIP/${EXTEN},60) exten => _[1001-1002],n,Hangup() ---------------------------------------- then i can simply call from one (local) sip phone to another and force transcoding between them. Nothing else. I want that also with Freeswitch. It is a good starting point but i am fiddeling with it for a couple of days now, read the docs but i can't get Freeswitch to do just this. So just calling between a few local sip phones with transcoding and _everything_ else disabled. Can anyone help me a little? Thanks in advance! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090215/80256b70/attachment-0002.html From brian at freeswitch.org Sat Feb 14 18:10:30 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 14 Feb 2009 20:10:30 -0600 Subject: [Freeswitch-users] [newbie] Clean start with a simple configuration In-Reply-To: <70ad8fdb0902141602w5829a059p3fe9b4159cba499a@mail.gmail.com> References: <70ad8fdb0902141602w5829a059p3fe9b4159cba499a@mail.gmail.com> Message-ID: <13479F65-7CE3-4C41-9845-6D92A647C946@freeswitch.org> FreeSWITCH default config already has this feature. Register two phones... 1000 and 1001 both with password of 1234 then you can call between them. That will work exactly as you want out of the box. Expect more simplified configs to show up after 1.0.3. /b On Feb 14, 2009, at 6:02 PM, xs wrote: > I want that also with Freeswitch. It is a good starting point but i > am fiddeling with it for a couple of days now, read the docs but i > can't get Freeswitch to do just this. So just calling between a few > local sip phones with transcoding and _everything_ else disabled. From brian at freeswitch.org Sat Feb 14 18:12:02 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 14 Feb 2009 20:12:02 -0600 Subject: [Freeswitch-users] [ANN] Spice Telephony - an open source FreeSWITCH/Erlang callcenter platform In-Reply-To: <22015518.post@talk.nabble.com> References: <22015518.post@talk.nabble.com> Message-ID: <1F887BD4-3B6B-4A87-89C9-6E6A823996FE@freeswitch.org> And are you planning on contributing anything useful back to the community? Or just take take take? /b On Feb 14, 2009, at 5:04 PM, JCATS wrote: > > Have you planned any predictive dialer features ( like VICIDIAL )? From krice at freeswitch.org Sat Feb 14 18:16:45 2009 From: krice at freeswitch.org (Ken Rice) Date: Sat, 14 Feb 2009 20:16:45 -0600 Subject: [Freeswitch-users] [ANN] Spice Telephony - an open source FreeSWITCH/Erlang callcenter platform In-Reply-To: <22015518.post@talk.nabble.com> Message-ID: Vicidial is not a predictive dialer... At the best its a power dialer... And spice is not an outbound thing its an inbound queue manager... If you need a predictive dialer contact me offlist or contact one of the other freeswitch developers directly we're be happy to consult for you on building a dialer. > From: JCATS > Reply-To: > Date: Sat, 14 Feb 2009 15:04:01 -0800 (PST) > To: > Subject: Re: [Freeswitch-users] [ANN] Spice Telephony - an open source > FreeSWITCH/Erlang callcenter platform > > > Have you planned any predictive dialer features ( like VICIDIAL )? > > -- > View this message in context: > http://www.nabble.com/-ANN--Spice-Telephony---an-open-source-FreeSWITCH-Erlang > -callcenter-platform-tp21384907p22015518.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From tim at marocon.com Sat Feb 14 18:20:13 2009 From: tim at marocon.com (Tim Mattison) Date: Sat, 14 Feb 2009 21:20:13 -0500 Subject: [Freeswitch-users] [newbie] Clean start with a simple configuration In-Reply-To: <13479F65-7CE3-4C41-9845-6D92A647C946@freeswitch.org> References: <70ad8fdb0902141602w5829a059p3fe9b4159cba499a@mail.gmail.com> <13479F65-7CE3-4C41-9845-6D92A647C946@freeswitch.org> Message-ID: <2FA0A303-9D53-453A-89A3-517D4B7FBC19@marocon.com> I think the crux of the matter is "I can't get Freeswitch to do _just_ this" (emphasis added). Users that are relatively new to FreeSWITCH but Asterisk veterans want to know how to build a simple, lean configuration. I'm trying to figure it out myself right now. If simplified configs are only going to be posted after 1.0.3 what's the best place to post my findings and configs? Is there an explicit, sanctioned place for this kind of thing on the Wiki already? Tim On Feb 14, 2009, at 9:10 PM, Brian West wrote: > FreeSWITCH default config already has this feature. Register two > phones... 1000 and 1001 both with password of 1234 then you can call > between them. That will work exactly as you want out of the box. > Expect more simplified configs to show up after 1.0.3. > > /b > > On Feb 14, 2009, at 6:02 PM, xs wrote: > >> I want that also with Freeswitch. It is a good starting point but i >> am fiddeling with it for a couple of days now, read the docs but i >> can't get Freeswitch to do just this. So just calling between a few >> local sip phones with transcoding and _everything_ else disabled. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090214/3dc7dd89/attachment-0002.html From brian at freeswitch.org Sat Feb 14 18:28:45 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 14 Feb 2009 20:28:45 -0600 Subject: [Freeswitch-users] [newbie] Clean start with a simple configuration In-Reply-To: <2FA0A303-9D53-453A-89A3-517D4B7FBC19@marocon.com> References: <70ad8fdb0902141602w5829a059p3fe9b4159cba499a@mail.gmail.com> <13479F65-7CE3-4C41-9845-6D92A647C946@freeswitch.org> <2FA0A303-9D53-453A-89A3-517D4B7FBC19@marocon.com> Message-ID: <2598EA3B-E07B-448A-8915-F974A7F1594F@freeswitch.org> Well the config itself is easy to follow. The problem is coming from Asterisk you have the wrong mindset to approach most things in FreeSWITCH. http://svn.freeswitch.org/svn/configs/ (softphone is the best small config to look at) The default config is easy to strip down once you take a few moments to look at it and maybe hop on IRC and ask questions. I wrote the default and the softphone config. So if you have questions I'm just the person to ask. More config sets will appear as I have time to write them. /b On Feb 14, 2009, at 8:20 PM, Tim Mattison wrote: > I think the crux of the matter is "I can't get Freeswitch to do > _just_ this" (emphasis added). Users that are relatively new to > FreeSWITCH but Asterisk veterans want to know how to build a simple, > lean configuration. I'm trying to figure it out myself right now. > > If simplified configs are only going to be posted after 1.0.3 what's > the best place to post my findings and configs? Is there an > explicit, sanctioned place for this kind of thing on the Wiki already? > > Tim From pauld at versafon.com Sat Feb 14 18:37:49 2009 From: pauld at versafon.com (Paul D.) Date: Sat, 14 Feb 2009 21:37:49 -0500 Subject: [Freeswitch-users] FS SIP audio quality? Message-ID: <49977FFD.1020002@versafon.com> Comparing FS 1.0.3 audio quality vs * 1.4.2, simple Sip-to-Sip call, or call to VM prompt, or call via gateway to PSTN - FS audio volume level (should I say gain?) seems noticeably lower than on *, this may be a reason that FS audio seems to be subpar, more noise less clear. Test calls made using PCMU codec from X-Lite and Linksys 2002. Is there anything can be tweaked in FS to correct that? Same issue was with 1.0.2. From brian at freeswitch.org Sat Feb 14 18:43:40 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 14 Feb 2009 20:43:40 -0600 Subject: [Freeswitch-users] FS SIP audio quality? In-Reply-To: <49977FFD.1020002@versafon.com> References: <49977FFD.1020002@versafon.com> Message-ID: I haven't ever experienced this issue can you maybe elaborate on the issue a little more? We usually hear that the audio quality is much better... have you tried latest SVN trunk? If resampling was involved it might cause some audio issues but those were usually gain issue and that has since been fixed in SVN trunk as of yesterday. /b On Feb 14, 2009, at 8:37 PM, Paul D. wrote: > Comparing FS 1.0.3 audio quality vs * 1.4.2, simple Sip-to-Sip call, > or > call to VM prompt, or call via gateway to PSTN - FS audio volume > level > (should I say gain?) seems noticeably lower than on *, this may be a > reason that FS audio seems to be subpar, more noise less clear. Test > calls made using PCMU codec from X-Lite and Linksys 2002. > Is there anything can be tweaked in FS to correct that? Same issue was > with 1.0.2. From pauld at versafon.com Sat Feb 14 19:02:54 2009 From: pauld at versafon.com (Paul D.) Date: Sat, 14 Feb 2009 22:02:54 -0500 Subject: [Freeswitch-users] FS SIP audio quality? In-Reply-To: References: <49977FFD.1020002@versafon.com> Message-ID: <499785DE.3010805@versafon.com> I am not sure what else I can add to that, I would love to elaborate more if you ask anything specific. I haven't tried the latest trunk, but since there's no difference between 1.0.2 and 1.0.3RC1 in audio quality I don't think it make sense trying. From what I see in FS logs there's no resampling involved, and that looks like true since I specifically restricted codecs in my test SIP equipment. But the fact is I tried different boxes, same OS centos 5.2 x64, and I had to bring audio volume and mic level all the way up in X-Lite to compensate for the difference to * audio, and in * such volume level sounds like way too high. FS installed cleanly from scratch, mostly default settings, except some dialplan/directory additions. Brian West wrote: > I haven't ever experienced this issue can you maybe elaborate on the > issue a little more? We usually hear that the audio quality is much > better... have you tried latest SVN trunk? If resampling was involved > it might cause some audio issues but those were usually gain issue and > that has since been fixed in SVN trunk as of yesterday. > > /b > > On Feb 14, 2009, at 8:37 PM, Paul D. wrote: > > >> Comparing FS 1.0.3 audio quality vs * 1.4.2, simple Sip-to-Sip call, >> or >> call to VM prompt, or call via gateway to PSTN - FS audio volume >> level >> (should I say gain?) seems noticeably lower than on *, this may be a >> reason that FS audio seems to be subpar, more noise less clear. Test >> calls made using PCMU codec from X-Lite and Linksys 2002. >> Is there anything can be tweaked in FS to correct that? Same issue was >> with 1.0.2. >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From brian at freeswitch.org Sat Feb 14 19:10:41 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 14 Feb 2009 21:10:41 -0600 Subject: [Freeswitch-users] FS SIP audio quality? In-Reply-To: <499785DE.3010805@versafon.com> References: <49977FFD.1020002@versafon.com> <499785DE.3010805@versafon.com> Message-ID: <034A0C6C-0C28-4C9C-8ECA-379E090387F6@freeswitch.org> What is odd some people have reported the same issue with Asterisk. I would like to get to the bottom of it but nobody can provide any more detail on what might be going on and I haven't experienced this issue with the 30 or so phones I have on my desk .... I highly recommend you try SVN trunk. Let me know how that goes. ;) /b On Feb 14, 2009, at 9:02 PM, Paul D. wrote: > I am not sure what else I can add to that, I would love to elaborate > more if you ask anything specific. > I haven't tried the latest trunk, but since there's no difference > between 1.0.2 and 1.0.3RC1 in audio quality I don't think > it make sense trying. From what I see in FS logs there's no resampling > involved, and that looks like true since I specifically restricted > codecs in my test SIP equipment. > But the fact is I tried different boxes, same OS centos 5.2 x64, and I > had to bring audio volume and mic level all the way up in X-Lite to > compensate for the difference to * audio, > and in * such volume level sounds like way too high. > FS installed cleanly from scratch, mostly default settings, except > some > dialplan/directory additions. From jason at jasonjgw.net Sat Feb 14 19:18:53 2009 From: jason at jasonjgw.net (Jason White) Date: Sun, 15 Feb 2009 14:18:53 +1100 Subject: [Freeswitch-users] FS SIP audio quality? In-Reply-To: <034A0C6C-0C28-4C9C-8ECA-379E090387F6@freeswitch.org> References: <49977FFD.1020002@versafon.com> <499785DE.3010805@versafon.com> <034A0C6C-0C28-4C9C-8ECA-379E090387F6@freeswitch.org> Message-ID: <20090215031853.GA4488@jdc.jasonjgw.net> Brian West wrote: > What is odd some people have reported the same issue with Asterisk. I > would like to get to the bottom of it but nobody can provide any more > detail on what might be going on and I haven't experienced this issue > with the 30 or so phones I have on my desk .... I highly recommend you A data point that may or may not be helpful: if I set up PortAudio on FreeSWITCH and call an Asterisk conference from there, the audio is significantly louder than a comparable SIP call with another FreeSWITCH box at the other end. > try SVN trunk. Let me know how that goes. ;) I'll recreate the above scenario with SVN trunk (I've just built rev. 12018), and report if there is still a problem. I sometimes get audio distortion in the above situation if anyone speaks too loudly. I suspect clipping somewhere in the audio processing. From brian at freeswitch.org Sat Feb 14 19:27:06 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 14 Feb 2009 21:27:06 -0600 Subject: [Freeswitch-users] FS SIP audio quality? In-Reply-To: <20090215031853.GA4488@jdc.jasonjgw.net> References: <49977FFD.1020002@versafon.com> <499785DE.3010805@versafon.com> <034A0C6C-0C28-4C9C-8ECA-379E090387F6@freeswitch.org> <20090215031853.GA4488@jdc.jasonjgw.net> Message-ID: <0DF0A788-0AA7-4FA6-A0FC-DABC19E0A148@freeswitch.org> This was a problem with the resampler which was replaced... we use the resampler in Speex now which will not exhibit the problem. /b On Feb 14, 2009, at 9:18 PM, Jason White wrote: > I sometimes get audio distortion in the above situation if anyone > speaks too > loudly. I suspect clipping somewhere in the audio processing. From jason at jasonjgw.net Sat Feb 14 23:02:16 2009 From: jason at jasonjgw.net (Jason White) Date: Sun, 15 Feb 2009 18:02:16 +1100 Subject: [Freeswitch-users] Build and Sofia issues with recent svn trunk revisions Message-ID: <20090215070216.GA20246@jdc.jasonjgw.net> Following the resampling discussion, I tried upgrading to revision 12018. which compiled cleanly, but then failed to load my internal SIP profile: [ERR] sofia.c:739 sofia_profile_thread_run() Error Creating SIP UA for profile: internal The same configuration works under revision 11488, to which I've temporarily downgraded. Either something has changed in FreeSWITCH that requires modifications to my SIP configuration, or this is a regression. I decided to try rev. 12027, which, on the same machine (Debian Sid) fails to build with the following error: x86_64-linux-gnu-gcc -c -ggdb -I. ./dftables.c In file included from ./dftables.c:50: ./pcre_internal.h:239:2: error: #error LINK_SIZE must be either 2, 3, or 4 make[2]: *** [dftables.o] Error 1 I suspect that recent changes to the build system are responsible for the latter. I'm sure these are minor matters that will be sorted out soon. Thanks once again to the developers for a great project! From alex at sinapticode.ro Sun Feb 15 00:47:50 2009 From: alex at sinapticode.ro (Alexandru Nedelcu) Date: Sun, 15 Feb 2009 10:47:50 +0200 Subject: [Freeswitch-users] Hangup hook in js is never called In-Reply-To: References: <87f2f3b90902131451t56f84286p13d7a0b815db0068@mail.gmail.com> Message-ID: <1234687670.4604.5.camel@gathern.lan> I'm using the CDR logs for that, because I need other info as well. To make the connection between a log-file and a DB record, I'm passing a custom channel variable on originate. I've written a short article about it, but as the other one, it's a draft: http://alexn.org/docs/dialer_part_2.html On Fri, 2009-02-13 at 23:07 +0000, Nik Middleton wrote: > I'm trying to capture the hang-up reason and write it to the db (Was it > busy etc). I also close the db in that function. That way I know I > don't have any open connections. This is in JavaScript BTW > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Michael Collins > Sent: 13 February 2009 22:51 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Hangup hook in js is never called > > > http://wiki.freeswitch.org/wiki/Example_Hangup_hook > > > > > > > > Is the code in error? > > > > It might just be. I think you are better off using api_hangup_hook. > What are you trying to do on hangup? The api_hangup_hook lets you call > any API, including running a script. Here's an example that we played > with today that I haven't even put on the wiki yet. I renames the wav > file after the call is hung up. > > > > > > > > > > > > > > > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From nik.middleton at noblesolutions.co.uk Sun Feb 15 03:39:12 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Sun, 15 Feb 2009 11:39:12 -0000 Subject: [Freeswitch-users] Problems with Originate In-Reply-To: <191c3a030902141445q7ffb1febm242c6785a9afef7c@mail.gmail.com> References: <1234524794.4431.56.camel@gathern.lan><1234525724.4431.59.camel@gathern.lan><191c3a030902130554w1dfe6b1dw9e5525cf6a210dc4@mail.gmail.com><191c3a030902140919w806923ex2dc44c2734010251@mail.gmail.com> <191c3a030902141445q7ffb1febm242c6785a9afef7c@mail.gmail.com> Message-ID: You are indeed correct. I still had my asterisk hat on, and was expecting a hang-up event to be fired with the call outcome. I was explicitly testing for call failure. I've not modified the code to test the result of the originate and it works as expected. I will add some words to the wiki explaining this for those converting from Asterisk Regards, ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: 14 February 2009 22:45 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Problems with Originate What do you suggest is not working? the call failed and it *did not* run the code inside if (new_session.ready()) The call was never established therefore it would not run the hangup hook either. In order to trigger the hangup hook the session would need to exist. If the session could not originate the new_session obj is an empty shell with no actual session inside. var new_session = new Session(, ); if (!new_session.ready()) { // the call never was established. } I gave you real code to try in my last email that you completely ignored..... and finally you are declaring the function wrong declare it at the top. function on_hangup(hup_session, how) { ..... } I sense you seem to think things are going to magically transform into however you are thinking they should work instead of you perhaps learning how they actually work. On Sat, Feb 14, 2009 at 1:47 PM, Nik Middleton wrote: Nope, Still not working. Here's my little test javascript var new_session = new Session('{ignore_early_media=true,}sofia/internal/1001 at 192.168.3.206'); //set the on_hangup function to be called when this session is hungup new_session.setHangupHook(on_hangup,"hup"); var on_hangup = function(hup_session, how) { console_log("err","In hangup section\n"); //exit here would end the script so you could cleanup and just be done exit(); } if (new_session.ready()) { new_session.answer( ); new_session.sleep(1500); new_session.streamFile("female2.wav"); } And this is the output [NOTICE] sofia.c:3090 sofia_handle_sip_i_state() Hangup sofia/internal/sip:1001 at 192.168.0.29:5060 [CS_CONSUME_MEDIA] [USER_BUSY] [ERR] switch_ivr_originate.c:1116 switch_ivr_originate() Cannot create outgoing channel of type [user] cause: [USER_BUSY] [INFO] mod_dptools.c:1909 audio_bridge_function() Originate Failed. Cause: USER_BUSY [NOTICE] switch_core_session.c:957 switch_core_session_thread() Session 95 (sofia/internal/sip:1001 at 192.168.0.29:5060) Ended [NOTICE] switch_core_session.c:959 switch_core_session_thread() Close Channel sofia/internal/sip:1001 at 192.168.0.29:5060 [CS_HANGUP] [NOTICE] mod_dptools.c:600 answer_function() Channel [sofia/internal/0000000000 at 192.168.3.206] has been answered ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: 14 February 2009 17:20 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Problems with Originate are you running this as a dialplan app? session is a reserved variable name for the session you executed the app on. are you using an alternate name for your new session like my_session etc....? this works for me, try it yourself. var my_session = new Session("sofia/external/7003 at conference.freeswitch.org"); consoleLog("err", "ready: " + my_session.ready() + "\n"); On Sat, Feb 14, 2009 at 7:17 AM, Nik Middleton wrote: Understood. However, using the second method, how can I trap on call failure? If I originate a call and the user is busy, the console reports this fact, but then the script continues to execute if (session.ready()) { console_log("notice","Session result=[" + session.cause + "] \n"); if (session.cause == "USER_BUSY") { Disposition = "BUSY"; session.Hangup(); } In this case session.cause reports 'NONE' and what's surprising is that even though the call failed (busy) session.ready returns a true value. ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: 13 February 2009 13:55 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Problems with Originate The first way is deprecated and will be removed. The 2nd way is the correct way. On Fri, Feb 13, 2009 at 5:48 AM, Alexandru Nedelcu wrote: On Fri, 2009-02-13 at 13:33 +0200, Alexandru Nedelcu wrote: > The problem with this setup is that origination_caller_id_number doesn't > work from inside the JS file (when calling session.originate). I just discovered something interesting. When originating the call like this ... session = new Session("") instead of this ... session = new Session(); session.originate("") ... then it works. Is this some kind of bug, or what's the difference here? Thanks, -- Alexandru Nedelcu Software Developer, Sinapticode _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090215/590620a3/attachment-0002.html From daletrub at gmail.com Sat Feb 14 21:15:04 2009 From: daletrub at gmail.com (Dale Trub) Date: Sat, 14 Feb 2009 21:15:04 -0800 Subject: [Freeswitch-users] help! DTMFs disappearing in mod_conference Message-ID: Hey folks, I'm having a very odd issue and I'm wondering if anyone else has seen this, or if there's a setting to change etc. I should mention that if anyone by chance helps THIS WEEKEND, it could SAVE my butt. We are doing an important demo monday morning and honestly this stops us in our tracks. We are listening for DTMFs from mod_conference and passing that via the socket on to a separate display layer (in development). It works perfectly, but at a certain point in a conference, it seems the switch stops sensing the DTMFs on most (but not all) lines. FYI, we saw this before with FS 1.0 running on a VPS slice and thought maybe it was somehow related to that box, or that DID provider. We've now switched to a full server and a different DID provider, and are getting the exact same behavior. Today, here was the deal: - 10 people called in (practice walkthrough of our demo this monday) - all lines: DTMFs displayed - tried them several times - 6= mute/unmute also works (doesn't go through our display layer) - about 30 minutes in, again asked everyone to hit 1 (which again we pass to display layer) - and now most lines do not pass DTMFs - a couple lines still do pass them - (the "6" which we trap within FS as "mute/unmute" also stops working on those lines that stopped passing others) - the FS logs STOP reflecting DTMFs from the lines where we don't see them - so, we know it's FS and not our application - some time passes - keep trying the working ones -- eventually they stop working - one caller (with DTMFs non functional) hangs up and calls back - that caller now does have DTMFs working - we hung up and called back in - this time DTMFs worked ~100 times, and then again stopped - switched logs from INFO to DEBUG - below are some log file entries We're on CENT-OS and FS 1.0.2 Besides the obvious question ("how do I fix this") Non-obvious Questions: - Is there any way to tell if the DID provider is trapping the DTMFs and sending them out of band, or is sending them in-band? - Is there any reasonably easy way to get in and see/sniff/visualize/measure the SIP packets to see what is coming in? - Could this be related to this? http://wiki.freeswitch.org/wiki/RTP_Issues - Any other thoughts on how to debug? Thanks!! -Dale Here's the last working DTMF, and then some events I don't know ... through a place where this definitely wasn't working. 2009-02-14 22:26:03 [DEBUG] switch_rtp.c:1701 switch_rtp_dequeue_dtmf() RTP RECV DTMF 5:2000 2009-02-14 22:37:06 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state() Channel sofi a/external/xxphonenumxx at 172.16.250.4 entering state [received] 2009-02-14 22:37:06 [DEBUG] sofia.c:2546 sofia_handle_sip_i_state() Remote SDP: v=0 o=FreeSWITCH 8044373728746667485 7321340529655007764 IN IP4 172.16.250.4 s=FreeSWITCH c=IN IP4 172.16.3.13 t=0 0 m=audio 33440 RTP/AVP 0 101 a=rtpmap:101 telephone-event/8000 a=ptime:20 2009-02-14 22:37:06 [DEBUG] sofia_glue.c:2447 sofia_glue_negotiate_sdp() Our exi sting sdp is still good [PCMU 172.16.3.13:33440], let's keep it. 2009-02-14 22:37:06 [DEBUG] sofia_glue.c:2473 sofia_glue_negotiate_sdp() Set 283 3 dtmf payload to 101 2009-02-14 22:37:06 [DEBUG] sofia_glue.c:1880 sofia_glue_activate_rtp() Audio pa rams are unchanged for sofia/external/xxphonenumxx at 172.16.250.4. 2009-02-14 22:37:06 [DEBUG] sofia.c:2896 sofia_handle_sip_i_state() Processing R einvite 2009-02-14 22:37:06 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state() Channel sofi a/external/xxphonenumxx at 172.16.250.4 entering state [completed] 2009-02-14 22:37:06 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state() Channel sofi a/external/xxphonenumxx at 172.16.250.4 entering state [ready] 2009-02-14 22:38:34 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state() Channel sofi a/external/xxphonenumxx2 at 172.16.250.4 entering state [received] 2009-02-14 22:38:34 [DEBUG] sofia.c:2546 sofia_handle_sip_i_state() Remote SDP: v=0 o=FreeSWITCH 934104982290142318 4836750446264379897 IN IP4 172.16.250.4 s=FreeSWITCH c=IN IP4 172.16.1.21 t=0 0 m=audio 35356 RTP/AVP 0 101 a=rtpmap:101 telephone-event/8000 a=ptime:20 2009-02-14 22:38:34 [DEBUG] sofia_glue.c:2447 sofia_glue_negotiate_sdp() Our exi sting sdp is still good [PCMU 172.16.1.21:35356], let's keep it. 2009-02-14 22:38:34 [DEBUG] sofia_glue.c:2473 sofia_glue_negotiate_sdp() Set 283 3 dtmf payload to 101 2009-02-14 22:38:34 [DEBUG] sofia_glue.c:1880 sofia_glue_activate_rtp() Audio pa rams are unchanged for sofia/external/xxphonenumxx2 at 172.16.250.4. 2009-02-14 22:38:34 [DEBUG] sofia_glue.c:1880 sofia_glue_activate_rtp() Audio pa rams are unchanged for sofia/external/xxphonenumxx2 at 172.16.250.4. 2009-02-14 22:38:34 [DEBUG] sofia.c:2896 sofia_handle_sip_i_state() Processing R einvite 2009-02-14 22:38:34 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state() Channel sofi a/external/xxphonenumxx2 at 172.16.250.4 entering state [completed] 2009-02-14 22:38:34 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state() Channel sofi a/external/xxphonenumxx2 at 172.16.250.4 entering state [ready] -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090214/5051ba09/attachment-0002.html From shahal at jajah.com Sat Feb 14 23:47:10 2009 From: shahal at jajah.com (Shahal Hazan) Date: Sun, 15 Feb 2009 09:47:10 +0200 Subject: [Freeswitch-users] Adding a third person to an ongoing conversation by dialing DTMF Message-ID: I would to add a third person to an ongoing conversation (between two SIP callers for example) by dialing a DTMF. The DTMF can be as simple as 1 or 2 or 3 to add a predefined person: person1 or person2 or person3 respectively. What is the best way to accomplish that? 1) Receiving the DTMF: After I added: I wasn't able to receive the DTMF on a non IVR call (I only got the IVR examples to work) Can I capture the DTMF in JS? 2) Adding the third person: a. Creating a conference on the fly and adding the third person? b. Bridging the third person? c. Using an API to "originate" a new call added to the current call? d. Creating a predefined group and adding members from that group per caller's choice? Thanks, Shahal Hazan This mail was sent via Mail-SeCure System. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090215/e93e747d/attachment-0002.html From daletrub at gmail.com Sun Feb 15 08:29:59 2009 From: daletrub at gmail.com (Dale Trub) Date: Sun, 15 Feb 2009 08:29:59 -0800 Subject: [Freeswitch-users] help! DTMFs disappearing in mod_conference In-Reply-To: References: Message-ID: The bug I describe sure looks a lot like: http://jira.freeswitch.org/browse/FSCORE-266 We have a direct Metaswitch-> FS connection, and both machines in the same LAN/location. It's 64-bit CentOS btw. Bug also occurred on a 32-bit CentOS dev machine. On Sat, Feb 14, 2009 at 9:15 PM, Dale Trub wrote: > Hey folks, > I'm having a very odd issue and I'm wondering if anyone else has seen this, > or if there's a setting to change etc. > > I should mention that if anyone by chance helps THIS WEEKEND, it could SAVE > my butt. We are doing an important demo monday morning and honestly this > stops us in our tracks. > > We are listening for DTMFs from mod_conference and passing that via the > socket on to a separate display layer (in development). > > It works perfectly, but at a certain point in a conference, it seems the > switch stops sensing the DTMFs on most (but not all) lines. > > FYI, we saw this before with FS 1.0 running on a VPS slice and thought > maybe it was somehow related to that box, or that DID provider. We've now > switched to a full server and a different DID provider, and are getting the > exact same behavior. > > Today, here was the deal: > > - 10 people called in (practice walkthrough of our demo this monday) > - all lines: DTMFs displayed - tried them several times > - 6= mute/unmute also works (doesn't go through our display layer) > - about 30 minutes in, again asked everyone to hit 1 (which again we > pass to display layer) > - and now most lines do not pass DTMFs > - a couple lines still do pass them > - (the "6" which we trap within FS as "mute/unmute" also stops > working on those lines that stopped passing others) > - the FS logs STOP reflecting DTMFs from the lines where we don't > see them > - so, we know it's FS and not our application > - some time passes > - keep trying the working ones -- eventually they stop working > - one caller (with DTMFs non functional) hangs up and calls back > - that caller now does have DTMFs working > - we hung up and called back in > - this time DTMFs worked ~100 times, and then again stopped > - switched logs from INFO to DEBUG > - below are some log file entries > > > We're on CENT-OS and FS 1.0.2 > > Besides the obvious question ("how do I fix this") > > Non-obvious Questions: > > - Is there any way to tell if the DID provider is trapping the DTMFs > and sending them out of band, or is sending them in-band? > - Is there any reasonably easy way to get in and > see/sniff/visualize/measure the SIP packets to see what is coming in? > - Could this be related to this? > http://wiki.freeswitch.org/wiki/RTP_Issues > - Any other thoughts on how to debug? > > Thanks!! > > -Dale > > Here's the last working DTMF, and then some events I don't know ... through > a place where this definitely wasn't working. > > > 2009-02-14 22:26:03 [DEBUG] switch_rtp.c:1701 switch_rtp_dequeue_dtmf() RTP > RECV > DTMF 5:2000 > 2009-02-14 22:37:06 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state() Channel > sofi > a/external/xxphonenumxx at 172.16.250.4 entering state [received] > 2009-02-14 22:37:06 [DEBUG] sofia.c:2546 sofia_handle_sip_i_state() Remote > SDP: > v=0 > o=FreeSWITCH 8044373728746667485 7321340529655007764 IN IP4 172.16.250.4 > s=FreeSWITCH > c=IN IP4 172.16.3.13 > t=0 0 > m=audio 33440 RTP/AVP 0 101 > a=rtpmap:101 telephone-event/8000 > a=ptime:20 > > 2009-02-14 22:37:06 [DEBUG] sofia_glue.c:2447 sofia_glue_negotiate_sdp() > Our exi > sting sdp is still good [PCMU 172.16.3.13:33440], let's keep it. > 2009-02-14 22:37:06 [DEBUG] sofia_glue.c:2473 sofia_glue_negotiate_sdp() > Set 283 > 3 dtmf payload to 101 > 2009-02-14 22:37:06 [DEBUG] sofia_glue.c:1880 sofia_glue_activate_rtp() > Audio pa > rams are unchanged for sofia/external/xxphonenumxx at 172.16.250.4. > 2009-02-14 22:37:06 [DEBUG] sofia.c:2896 sofia_handle_sip_i_state() > Processing R > einvite > 2009-02-14 22:37:06 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state() Channel > sofi > a/external/xxphonenumxx at 172.16.250.4 entering state [completed] > 2009-02-14 22:37:06 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state() Channel > sofi > a/external/xxphonenumxx at 172.16.250.4 entering state [ready] > 2009-02-14 22:38:34 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state() Channel > sofi > a/external/xxphonenumxx2 at 172.16.250.4 entering state [received] > 2009-02-14 22:38:34 [DEBUG] sofia.c:2546 sofia_handle_sip_i_state() Remote > SDP: > v=0 > o=FreeSWITCH 934104982290142318 4836750446264379897 IN IP4 172.16.250.4 > s=FreeSWITCH > c=IN IP4 172.16.1.21 > t=0 0 > m=audio 35356 RTP/AVP 0 101 > a=rtpmap:101 telephone-event/8000 > a=ptime:20 > > 2009-02-14 22:38:34 [DEBUG] sofia_glue.c:2447 sofia_glue_negotiate_sdp() > Our exi > sting sdp is still good [PCMU 172.16.1.21:35356], let's keep it. > 2009-02-14 22:38:34 [DEBUG] sofia_glue.c:2473 sofia_glue_negotiate_sdp() > Set 283 > 3 dtmf payload to 101 > 2009-02-14 22:38:34 [DEBUG] sofia_glue.c:1880 sofia_glue_activate_rtp() > Audio pa > rams are unchanged for sofia/external/xxphonenumxx2 at 172.16.250.4. > > 2009-02-14 22:38:34 [DEBUG] sofia_glue.c:1880 sofia_glue_activate_rtp() > Audio pa > rams are unchanged for sofia/external/xxphonenumxx2 at 172.16.250.4. > 2009-02-14 22:38:34 [DEBUG] sofia.c:2896 sofia_handle_sip_i_state() > Processing R > einvite > 2009-02-14 22:38:34 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state() Channel > sofi > a/external/xxphonenumxx2 at 172.16.250.4 entering state [completed] > 2009-02-14 22:38:34 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state() Channel > sofi > a/external/xxphonenumxx2 at 172.16.250.4 entering state [ready] > > > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090215/05fe60da/attachment-0002.html From pbd at suspiria.net Sun Feb 15 11:16:25 2009 From: pbd at suspiria.net (Public Dump) Date: Sun, 15 Feb 2009 20:16:25 +0100 Subject: [Freeswitch-users] High CPU load after starting Message-ID: <13C421883438EB42B9E2C30069FD4AB76AEA2B38A2@crushinator.central.local> So, no ideas left how to fix this problem ? Von: Public Dump Gesendet: Dienstag, 10. Februar 2009 19:42 An: 'freeswitch-users at lists.freeswitch.org' Betreff: High CPU load after starting After starting FreeSwitch (1.0.2) on a 4 core server running Windows Server 2008, the CPU load (privileged time/kernel) for one of the cores goes to 50% and stays there. Stoping FreeSwitch stops the load. I have tried to disable all modules but the problem persists. Has anybody seen this problem, can it be fixed ? regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090215/9615846e/attachment-0002.html From brian at freeswitch.org Sun Feb 15 12:09:50 2009 From: brian at freeswitch.org (Brian West) Date: Sun, 15 Feb 2009 14:09:50 -0600 Subject: [Freeswitch-users] help! DTMFs disappearing in mod_conference In-Reply-To: References: Message-ID: Yes this issue has already been fixed in SVN Trunk. I recommend you update. /b On Feb 15, 2009, at 10:29 AM, Dale Trub wrote: > The bug I describe sure looks a lot like: > http://jira.freeswitch.org/browse/FSCORE-266 > > We have a direct Metaswitch-> FS connection, and both machines in > the same LAN/location. > > It's 64-bit CentOS btw. Bug also occurred on a 32-bit CentOS dev > machine. > > On Sat, Feb 14, 2009 at 9:15 PM, Dale Trub wrote: > Hey folks, > > I'm having a very odd issue and I'm wondering if anyone else has > seen this, or if there's a setting to change etc. > > I should mention that if anyone by chance helps THIS WEEKEND, it > could SAVE my butt. We are doing an important demo monday morning > and honestly this stops us in our tracks. > > We are listening for DTMFs from mod_conference and passing that via > the socket on to a separate display layer (in development). > > It works perfectly, but at a certain point in a conference, it seems > the switch stops sensing the DTMFs on most (but not all) lines. > > FYI, we saw this before with FS 1.0 running on a VPS slice and > thought maybe it was somehow related to that box, or that DID > provider. We've now switched to a full server and a different DID > provider, and are getting the exact same behavior. > > Today, here was the deal: > 10 people called in (practice walkthrough of our demo this monday) > all lines: DTMFs displayed - tried them several times > 6= mute/unmute also works (doesn't go through our display layer) > about 30 minutes in, again asked everyone to hit 1 (which again we > pass to display layer) > and now most lines do not pass DTMFs > a couple lines still do pass them > (the "6" which we trap within FS as "mute/unmute" also stops working > on those lines that stopped passing others) > the FS logs STOP reflecting DTMFs from the lines where we don't see > them > so, we know it's FS and not our application > some time passes > keep trying the working ones -- eventually they stop working > one caller (with DTMFs non functional) hangs up and calls back > that caller now does have DTMFs working > we hung up and called back in > this time DTMFs worked ~100 times, and then again stopped > switched logs from INFO to DEBUG > below are some log file entries > > We're on CENT-OS and FS 1.0.2 > > Besides the obvious question ("how do I fix this") > > Non-obvious Questions: > Is there any way to tell if the DID provider is trapping the DTMFs > and sending them out of band, or is sending them in-band? > Is there any reasonably easy way to get in and see/sniff/visualize/ > measure the SIP packets to see what is coming in? > Could this be related to this? http://wiki.freeswitch.org/wiki/RTP_Issues > Any other thoughts on how to debug? > Thanks!! > > -Dale > > Here's the last working DTMF, and then some events I don't know ... > through a place where this definitely wasn't working. > > > 2009-02-14 22:26:03 [DEBUG] switch_rtp.c:1701 > switch_rtp_dequeue_dtmf() RTP RECV > DTMF 5:2000 > 2009-02-14 22:37:06 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state() > Channel sofi > a/external/xxphonenumxx at 172.16.250.4 entering state [received] > 2009-02-14 22:37:06 [DEBUG] sofia.c:2546 sofia_handle_sip_i_state() > Remote SDP: > v=0 > o=FreeSWITCH 8044373728746667485 7321340529655007764 IN IP4 > 172.16.250.4 > s=FreeSWITCH > c=IN IP4 172.16.3.13 > t=0 0 > m=audio 33440 RTP/AVP 0 101 > a=rtpmap:101 telephone-event/8000 > a=ptime:20 > > 2009-02-14 22:37:06 [DEBUG] sofia_glue.c:2447 > sofia_glue_negotiate_sdp() Our exi > sting sdp is still good [PCMU 172.16.3.13:33440], let's keep it. > 2009-02-14 22:37:06 [DEBUG] sofia_glue.c:2473 > sofia_glue_negotiate_sdp() Set 283 > 3 dtmf payload to 101 > 2009-02-14 22:37:06 [DEBUG] sofia_glue.c:1880 > sofia_glue_activate_rtp() Audio pa > rams are unchanged for sofia/external/xxphonenumxx at 172.16.250.4. > 2009-02-14 22:37:06 [DEBUG] sofia.c:2896 sofia_handle_sip_i_state() > Processing R > einvite > 2009-02-14 22:37:06 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state() > Channel sofi > a/external/xxphonenumxx at 172.16.250.4 entering state [completed] > 2009-02-14 22:37:06 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state() > Channel sofi > a/external/xxphonenumxx at 172.16.250.4 entering state [ready] > 2009-02-14 22:38:34 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state() > Channel sofi > a/external/xxphonenumxx2 at 172.16.250.4 entering state [received] > 2009-02-14 22:38:34 [DEBUG] sofia.c:2546 sofia_handle_sip_i_state() > Remote SDP: > v=0 > o=FreeSWITCH 934104982290142318 4836750446264379897 IN IP4 > 172.16.250.4 > s=FreeSWITCH > c=IN IP4 172.16.1.21 > t=0 0 > m=audio 35356 RTP/AVP 0 101 > a=rtpmap:101 telephone-event/8000 > a=ptime:20 > > 2009-02-14 22:38:34 [DEBUG] sofia_glue.c:2447 > sofia_glue_negotiate_sdp() Our exi > sting sdp is still good [PCMU 172.16.1.21:35356], let's keep it. > 2009-02-14 22:38:34 [DEBUG] sofia_glue.c:2473 > sofia_glue_negotiate_sdp() Set 283 > 3 dtmf payload to 101 > 2009-02-14 22:38:34 [DEBUG] sofia_glue.c:1880 > sofia_glue_activate_rtp() Audio pa > rams are unchanged for sofia/external/xxphonenumxx2 at 172.16.250.4. > > 2009-02-14 22:38:34 [DEBUG] sofia_glue.c:1880 > sofia_glue_activate_rtp() Audio pa > rams are unchanged for sofia/external/xxphonenumxx2 at 172.16.250.4. > 2009-02-14 22:38:34 [DEBUG] sofia.c:2896 sofia_handle_sip_i_state() > Processing R > einvite > 2009-02-14 22:38:34 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state() > Channel sofi > a/external/xxphonenumxx2 at 172.16.250.4 entering state [completed] > 2009-02-14 22:38:34 [DEBUG] sofia.c:2542 sofia_handle_sip_i_state() > Channel sofi > a/external/xxphonenumxx2 at 172.16.250.4 entering state [ready] > > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090215/4dafd255/attachment-0002.html From brian at freeswitch.org Sun Feb 15 12:10:23 2009 From: brian at freeswitch.org (Brian West) Date: Sun, 15 Feb 2009 14:10:23 -0600 Subject: [Freeswitch-users] High CPU load after starting In-Reply-To: <13C421883438EB42B9E2C30069FD4AB76AEA2B38A2@crushinator.central.local> References: <13C421883438EB42B9E2C30069FD4AB76AEA2B38A2@crushinator.central.local> Message-ID: <16A96D11-7880-4B03-A729-D50C0538C0EB@freeswitch.org> Since nobody can reproduce it... not sure how we can proceed... have you done a fresh checkout from SVN trunk and tried again? /b On Feb 15, 2009, at 1:16 PM, Public Dump wrote: > So, no ideas left how to fix this problem ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090215/520708ba/attachment-0002.html From nik.middleton at noblesolutions.co.uk Sun Feb 15 14:18:56 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Sun, 15 Feb 2009 22:18:56 -0000 Subject: [Freeswitch-users] Getting current call count Message-ID: Hi guys, I'd like to get the number of calls on the system so that I can manage the load. >From the cli, I've tried the following: Show channels This along with the call detail shows me the correct number of calls Show calls count This delivers a value of zero. I should add that I'm placing an outbound call from a JavaScript. If I originate another call within the script and bridge it with the first, it then shows 1 call It's as if it's only counting calls between two end points Status Shows correct number of sessions, BUT... shows 2/200 (200 is the value set in call setups/sec. I've set maximum calls to 1000, so I'm hoping that this is a typo) Am I missing something here? Is there another way of doing this? Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090215/ddd70464/attachment-0002.html From nik.middleton at noblesolutions.co.uk Sun Feb 15 15:21:17 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Sun, 15 Feb 2009 23:21:17 -0000 Subject: [Freeswitch-users] [newbie] Clean start with asimple configuration In-Reply-To: <2FA0A303-9D53-453A-89A3-517D4B7FBC19@marocon.com> References: <70ad8fdb0902141602w5829a059p3fe9b4159cba499a@mail.gmail.com><13479F65-7CE3-4C41-9845-6D92A647C946@freeswitch.org> <2FA0A303-9D53-453A-89A3-517D4B7FBC19@marocon.com> Message-ID: I'm in the same boat, finding the transition from Asterisk to FS very frustrating. Something I can do in Asterisk in 10 minutes is taking me a day with FS. Do I think it's worth it? Absolutely, but it's incredibly painful at times. What I've done is to create some WIKI pages to help those familiar with Asterisk to understand the nuances of FS. I posted them in the user pages. Hopefully when there are enough contributions we can have a section on the main WIKI entitled 'Asterisk conversion' or something. Asterisk is very forgiving and takes a lot of the pain away from doing simple tasks. FS on the other hand is less forgiving, but you have more control. Being a control freak I like that. I kind of liken Asterisk to the early versions of basic. Each command had a line number. You could be up and running basic apps in a few hours. Then jump to C, and you'll spend ages doing a simple task, but once you've mastered it, you'll never go back. (7 Day FS veteran :-)) Regards, ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Tim Mattison Sent: 15 February 2009 02:20 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] [newbie] Clean start with asimple configuration I think the crux of the matter is "I can't get Freeswitch to do _just_ this" (emphasis added). Users that are relatively new to FreeSWITCH but Asterisk veterans want to know how to build a simple, lean configuration. I'm trying to figure it out myself right now. If simplified configs are only going to be posted after 1.0.3 what's the best place to post my findings and configs? Is there an explicit, sanctioned place for this kind of thing on the Wiki already? Tim On Feb 14, 2009, at 9:10 PM, Brian West wrote: FreeSWITCH default config already has this feature. Register two phones... 1000 and 1001 both with password of 1234 then you can call between them. That will work exactly as you want out of the box. Expect more simplified configs to show up after 1.0.3. /b On Feb 14, 2009, at 6:02 PM, xs wrote: I want that also with Freeswitch. It is a good starting point but i am fiddeling with it for a couple of days now, read the docs but i can't get Freeswitch to do just this. So just calling between a few local sip phones with transcoding and _everything_ else disabled. _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090215/0b3f19cb/attachment-0002.html From krice at freeswitch.org Sun Feb 15 15:40:19 2009 From: krice at freeswitch.org (Ken Rice) Date: Sun, 15 Feb 2009 17:40:19 -0600 Subject: [Freeswitch-users] Getting current call count In-Reply-To: Message-ID: 2/200 is number of sessions per second/MAX number of sessions per second... If you do a ?fsctl max_sessions? that will show you max number of sessions Status outputs something like this API CALL [status()] output: UP 0 years, 0 days, 17 hours, 18 minutes, 36 seconds, 913 milliseconds, 565 microseconds 1574 session(s) since startup 4 session(s) 0/30 Uptime Total Number of session(s) since startup Active session(s) sessions per sec/max sessions per sec From: Nik Middleton Reply-To: Date: Sun, 15 Feb 2009 22:18:56 -0000 To: Subject: [Freeswitch-users] Getting current call count Hi guys, I?d like to get the number of calls on the system so that I can manage the load. >From the cli, I?ve tried the following: Show channels This along with the call detail shows me the correct number of calls Show calls count This delivers a value of zero. I should add that I?m placing an outbound call from a JavaScript. If I originate another call within the script and bridge it with the first, it then shows 1 call It?s as if it?s only counting calls between two end points Status Shows correct number of sessions, BUT? shows 2/200 (200 is the value set in call setups/sec. I?ve set maximum calls to 1000, so I?m hoping that this is a typo) Am I missing something here? Is there another way of doing this? Regards, _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090215/969f8e75/attachment-0002.html From anthony.minessale at gmail.com Sun Feb 15 16:43:31 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sun, 15 Feb 2009 18:43:31 -0600 Subject: [Freeswitch-users] FS SIP audio quality? In-Reply-To: <49977FFD.1020002@versafon.com> References: <49977FFD.1020002@versafon.com> Message-ID: <191c3a030902151643n62c2b1a0k2c341eb79c1393ea@mail.gmail.com> it's digital audio. The only thing doing sampling and reconstruction of the signal are the phones. The audio files have been captured long ago from the microphone in the studio. We do nothing to alter the volume of the audio signal or manipulate it in any way unless you are transcoding between sample rates or codecs which you are not because you mentioned it was PCMU. If you are making a call from x-lite to a linksys using just PCMU there is no transcoding going on at all and it would not be any more or less loud than if the devices were exchanging media directly because all we would be doing is passing the digital packets across. I believe you are somehow mistaken in your explanation. There is a good chance that your x-lite has the gain set lower when you are testing FS since that's the only device in your whole scenario that is capable of adjusting the gain. If you wish, please get a complete packet capture of a completed call in both situations. On Sat, Feb 14, 2009 at 8:37 PM, Paul D. wrote: > Comparing FS 1.0.3 audio quality vs * 1.4.2, simple Sip-to-Sip call, or > call to VM prompt, or call via gateway to PSTN - FS audio volume level > (should I say gain?) seems noticeably lower than on *, this may be a > reason that FS audio seems to be subpar, more noise less clear. Test > calls made using PCMU codec from X-Lite and Linksys 2002. > Is there anything can be tweaked in FS to correct that? Same issue was > with 1.0.2. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090215/413477e6/attachment-0002.html From anthony.minessale at gmail.com Sun Feb 15 16:51:59 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sun, 15 Feb 2009 18:51:59 -0600 Subject: [Freeswitch-users] [newbie] Clean start with asimple configuration In-Reply-To: References: <70ad8fdb0902141602w5829a059p3fe9b4159cba499a@mail.gmail.com> <13479F65-7CE3-4C41-9845-6D92A647C946@freeswitch.org> <2FA0A303-9D53-453A-89A3-517D4B7FBC19@marocon.com> Message-ID: <191c3a030902151651k4108d9a0he12fa555ac6ceab1@mail.gmail.com> We do have mod_dialplan_asterisk if you miss that. You do realize you did not do it in 10 minutes with asterisk when you first started using it only once you learned how to work it. Your problem is more with the paradigm shift than the complexity. This is common so it's good that you are adding some wiki pages from you perspective. I deputize you in charge of training all new users with an asterisk background. It's harder for us to explain it, however, We did all come from asterisk too ;) There are 2 distinct camps of new users. *) Those who try to make it work like asterisk and take month to learn how the rain in Spain falls gently on the plain. *) Those who never heard of asterisk before and understand everything instantly. We need more people to step up and contribute more to the documentation from each perspective , keep up the good work! On Sun, Feb 15, 2009 at 5:21 PM, Nik Middleton < nik.middleton at noblesolutions.co.uk> wrote: > I'm in the same boat, finding the transition from Asterisk to FS very > frustrating. Something I can do in Asterisk in 10 minutes is taking me a > day with FS. > > > > Do I think it's worth it? Absolutely, but it's incredibly painful at > times. > > > > What I've done is to create some WIKI pages to help those familiar with > Asterisk to understand the nuances of FS. I posted them in the user pages. > Hopefully when there are enough contributions we can have a section on the > main WIKI entitled 'Asterisk conversion' or something. > > > > Asterisk is very forgiving and takes a lot of the pain away from doing > simple tasks. FS on the other hand is less forgiving, but you have more > control. Being a control freak I like that. > > > > I kind of liken Asterisk to the early versions of basic. Each command had > a line number. You could be up and running basic apps in a few hours. Then > jump to C, and you'll spend ages doing a simple task, but once you've > mastered it, you'll never go back. > > > > (7 Day FS veteran J) > > > > Regards, > > > ------------------------------ > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Tim Mattison > *Sent:* 15 February 2009 02:20 > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] [newbie] Clean start with asimple > configuration > > > > I think the crux of the matter is "I can't get Freeswitch to do _just_ > this" (emphasis added). Users that are relatively new to FreeSWITCH but > Asterisk veterans want to know how to build a simple, lean configuration. > I'm trying to figure it out myself right now. > > > > If simplified configs are only going to be posted after 1.0.3 what's the > best place to post my findings and configs? Is there an explicit, > sanctioned place for this kind of thing on the Wiki already? > > > > Tim > > > > On Feb 14, 2009, at 9:10 PM, Brian West wrote: > > > > FreeSWITCH default config already has this feature. Register two > phones... 1000 and 1001 both with password of 1234 then you can call > between them. That will work exactly as you want out of the box. > Expect more simplified configs to show up after 1.0.3. > > /b > > On Feb 14, 2009, at 6:02 PM, xs wrote: > > > I want that also with Freeswitch. It is a good starting point but i > > am fiddeling with it for a couple of days now, read the docs but i > > can't get Freeswitch to do just this. So just calling between a few > > local sip phones with transcoding and _everything_ else disabled. > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090215/642e0e5d/attachment-0002.html From anthony.minessale at gmail.com Sun Feb 15 17:14:49 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sun, 15 Feb 2009 19:14:49 -0600 Subject: [Freeswitch-users] Problems with Originate In-Reply-To: References: <1234524794.4431.56.camel@gathern.lan> <1234525724.4431.59.camel@gathern.lan> <191c3a030902130554w1dfe6b1dw9e5525cf6a210dc4@mail.gmail.com> <191c3a030902140919w806923ex2dc44c2734010251@mail.gmail.com> <191c3a030902141445q7ffb1febm242c6785a9afef7c@mail.gmail.com> Message-ID: <191c3a030902151714u4be8e9dakd01547b5e49b9dec@mail.gmail.com> btw on the failed session you can still access the attribute session.cause and session.causecode to see why it failed to setup On Sun, Feb 15, 2009 at 5:39 AM, Nik Middleton < nik.middleton at noblesolutions.co.uk> wrote: > You are indeed correct. I still had my asterisk hat on, and was > expecting a hang-up event to be fired with the call outcome. I was > explicitly testing for call failure. I've not modified the code to test the > result of the originate and it works as expected. > > > > I will add some words to the wiki explaining this for those converting from > Asterisk > > > > Regards, > > > ------------------------------ > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony > Minessale > *Sent:* 14 February 2009 22:45 > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Problems with Originate > > > > > What do you suggest is not working? > > the call failed and it *did not* run the code inside if > (new_session.ready()) > The call was never established therefore it would not run the hangup hook > either. > In order to trigger the hangup hook the session would need to exist. If > the session could not originate the new_session obj > is an empty shell with no actual session inside. > > var new_session = new Session(, ); > if (!new_session.ready()) { > // the call never was established. > } > > I gave you real code to try in my last email that you completely > ignored..... > > > and finally you are declaring the function wrong > > declare it at the top. > > function on_hangup(hup_session, how) > { > ..... > } > > I sense you seem to think things are going to magically transform into > however you are thinking they should work > instead of you perhaps learning how they actually work. > > > On Sat, Feb 14, 2009 at 1:47 PM, Nik Middleton < > nik.middleton at noblesolutions.co.uk> wrote: > > Nope, > > > > Still not working. Here's my little test javascript > > > > var new_session = new Session('{ignore_early_media=true,}sofia/internal/ > 1001 at 192.168.3.206'); > > > > //set the on_hangup function to be called when this session is > hungup > > > new_session.setHangupHook(on_hangup,"hup"); > > > > var on_hangup = function(hup_session, how) > { > > > console_log("err","In hangup > section\n"); > > //exit here would end the script so you > could cleanup and just be > done > > > > exit(); > > > } > > > > > > if (new_session.ready()) { > > new_session.answer( ); > > new_session.sleep(1500); > > new_session.streamFile("female2.wav"); > > } > > > > And this is the output > > > > [NOTICE] sofia.c:3090 sofia_handle_sip_i_state() Hangup sofia/internal/ > sip:1001 at 192.168.0.29:5060 [CS_CONSUME_MEDIA] [USER_BUSY] > > [ERR] switch_ivr_originate.c:1116 switch_ivr_originate() Cannot create > outgoing channel of type [user] cause: [USER_BUSY] > > [INFO] mod_dptools.c:1909 audio_bridge_function() Originate Failed. Cause: > USER_BUSY > > [NOTICE] switch_core_session.c:957 switch_core_session_thread() Session 95 > (sofia/internal/sip:1001 at 192.168.0.29:5060) Ended > > [NOTICE] switch_core_session.c:959 switch_core_session_thread() Close > Channel sofia/internal/sip:1001 at 192.168.0.29:5060 [CS_HANGUP] > > [NOTICE] mod_dptools.c:600 answer_function() Channel [sofia/internal/ > 0000000000 at 192.168.3.206] has been answered > > > ------------------------------ > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony > Minessale > *Sent:* 14 February 2009 17:20 > > > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Problems with Originate > > > > are you running this as a dialplan app? > session is a reserved variable name for the session you executed the app > on. > > are you using an alternate name for your new session like my_session > etc....? > > this works for me, try it yourself. > > var my_session = new Session("sofia/external/ > 7003 at conference.freeswitch.org"); > consoleLog("err", "ready: " + my_session.ready() + "\n"); > > > On Sat, Feb 14, 2009 at 7:17 AM, Nik Middleton < > nik.middleton at noblesolutions.co.uk> wrote: > > Understood. > > > > However, using the second method, how can I trap on call failure? > > > > If I originate a call and the user is busy, the console reports this fact, > but then the script continues to execute > > > > if (session.ready()) { > > console_log("notice","Session result=[" + > session.cause + "] \n"); > > if (session.cause == "USER_BUSY") { > > Disposition = > "BUSY"; > > session.Hangup(); > > } > > In this case session.cause reports 'NONE' and what's surprising is that > even though the call failed (busy) session.ready returns a true value. > ------------------------------ > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony > Minessale > *Sent:* 13 February 2009 13:55 > > > *To:* freeswitch-users at lists.freeswitch.org > > *Subject:* Re: [Freeswitch-users] Problems with Originate > > > > The first way is deprecated and will be removed. > The 2nd way is the correct way. > > On Fri, Feb 13, 2009 at 5:48 AM, Alexandru Nedelcu > wrote: > > On Fri, 2009-02-13 at 13:33 +0200, Alexandru Nedelcu wrote: > > The problem with this setup is that origination_caller_id_number doesn't > > work from inside the JS file (when calling session.originate). > > I just discovered something interesting. > > When originating the call like this ... > session = new Session("") > instead of this ... > session = new Session(); session.originate("") > > ... then it works. Is this some kind of bug, or what's the difference > here? > > > Thanks, > > -- > Alexandru Nedelcu > Software Developer, Sinapticode > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090215/837fa4a5/attachment-0002.html From anthony.minessale at gmail.com Sun Feb 15 17:20:31 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sun, 15 Feb 2009 19:20:31 -0600 Subject: [Freeswitch-users] Adding a third person to an ongoing conversation by dialing DTMF In-Reply-To: References: Message-ID: <191c3a030902151720u79c06acw671d3e9d86ab5954@mail.gmail.com> bind-meta-app + (three_way app or transfer app with -both to a conference) On Sun, Feb 15, 2009 at 1:47 AM, Shahal Hazan wrote: > I would to add a third person to an ongoing conversation (between two SIP > callers for example) by dialing a DTMF. > > The DTMF can be as simple as 1 or 2 or 3 to add a predefined person: > person1 or person2 or person3 respectively. > > What is the best way to accomplish that? > > 1) Receiving the DTMF: > > After I added: > > ** > > I wasn't able to receive the DTMF on a non IVR call (I only got the IVR > examples to work) > > Can I capture the DTMF in JS? > > > > 2) Adding the third person: > > a. Creating a conference on the fly and adding the third person? > > b. Bridging the third person? > > c. Using an API to "originate" a new call added to the current call? > > d. Creating a predefined group and adding members from that group per > caller's choice? > > > > Thanks, > > Shahal Hazan > > > This mail was sent via Mail-SeCure System. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090215/989d4127/attachment-0002.html From anthony.minessale at gmail.com Sun Feb 15 17:21:19 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sun, 15 Feb 2009 19:21:19 -0600 Subject: [Freeswitch-users] Build and Sofia issues with recent svn trunk revisions In-Reply-To: <20090215070216.GA20246@jdc.jasonjgw.net> References: <20090215070216.GA20246@jdc.jasonjgw.net> Message-ID: <191c3a030902151721pae57190g1ede358ee717962@mail.gmail.com> [ERR] sofia.c:739 sofia_profile_thread_run() Error Creating SIP UA for profile: internal 9/10 times means something is already running and listening to the sip port. On Sun, Feb 15, 2009 at 1:02 AM, Jason White wrote: > Following the resampling discussion, I tried upgrading to revision 12018. > which compiled cleanly, but then failed to load my internal SIP profile: > > [ERR] sofia.c:739 sofia_profile_thread_run() Error Creating SIP UA for > profile: internal > > The same configuration works under revision 11488, to which I've > temporarily > downgraded. Either something has changed in FreeSWITCH that requires > modifications to my SIP configuration, or this is a regression. > > I decided to try rev. 12027, which, on the same machine (Debian Sid) fails > to > build with the following error: > x86_64-linux-gnu-gcc -c -ggdb -I. ./dftables.c > In file included from ./dftables.c:50: > ./pcre_internal.h:239:2: error: #error LINK_SIZE must be either 2, 3, or 4 > make[2]: *** [dftables.o] Error 1 > > I suspect that recent changes to the build system are responsible for the > latter. > > I'm sure these are minor matters that will be sorted out soon. Thanks once > again to the developers for a great project! > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090215/784db508/attachment-0002.html From jason at jasonjgw.net Sun Feb 15 17:44:38 2009 From: jason at jasonjgw.net (Jason White) Date: Mon, 16 Feb 2009 12:44:38 +1100 Subject: [Freeswitch-users] [newbie] Clean start with asimple configuration In-Reply-To: <191c3a030902151651k4108d9a0he12fa555ac6ceab1@mail.gmail.com> References: <70ad8fdb0902141602w5829a059p3fe9b4159cba499a@mail.gmail.com> <13479F65-7CE3-4C41-9845-6D92A647C946@freeswitch.org> <2FA0A303-9D53-453A-89A3-517D4B7FBC19@marocon.com> <191c3a030902151651k4108d9a0he12fa555ac6ceab1@mail.gmail.com> Message-ID: <20090216014438.GA4590@jdc.jasonjgw.net> Anthony Minessale wrote: > > There are 2 distinct camps of new users. > > *) Those who try to make it work like asterisk and take month to learn how > the rain in Spain falls gently on the plain. > *) Those who never heard of asterisk before and understand everything > instantly. I must be one of those rare users who stand in the middle: I had used Asterisk before, but I didn't try to apply my Asterisk knowledge to learning FreeSWITCH, other than to make sure that all of the desirable features of my Asterisk configuration eventually had counterparts in my FreeSWITCH configuration. From pauld at versafon.com Sun Feb 15 18:04:14 2009 From: pauld at versafon.com (Paul D.) Date: Sun, 15 Feb 2009 21:04:14 -0500 Subject: [Freeswitch-users] FS SIP audio quality? In-Reply-To: <191c3a030902151643n62c2b1a0k2c341eb79c1393ea@mail.gmail.com> References: <49977FFD.1020002@versafon.com> <191c3a030902151643n62c2b1a0k2c341eb79c1393ea@mail.gmail.com> Message-ID: <4998C99E.9060706@versafon.com> Well, I tried several call scenarios: 1. Call from X-Lite or Linksys to VM. 2. Call from X-Lite or Linksys to a conference. 3. Call from X-Lite or Linksys to a PSTN number via Gafachi and CallWithUs. I have now * 1.6.5 and FS 1.0.3RC1 installed on the same enterprise grade Intel server. So just comparing audio in the call scenarios above * somehow does noticeably better job, sounds clearer and volume is at the right level. I am not changing any phone settings of course when switching between * and FS. I am not biased towards FS or * at the moment, though FS seems to have a better designed configuration options and community. Just wanted to share my experience, and hear some opinions. Unfortunately I cannot spend whole amount of time investigating this case now, capturing packets etc., but I will try to do that once I have time. Meanwhile I will have to stick to * for prod. Anthony Minessale wrote: > it's digital audio. The only thing doing sampling and reconstruction > of the signal are the phones. The audio files have been captured long > ago from the microphone in the studio. > We do nothing to alter the volume of the audio signal or manipulate it > in any way unless you are transcoding between sample rates or codecs > which you are not because you mentioned it was PCMU. > > If you are making a call from x-lite to a linksys using just PCMU > there is no transcoding going on at all and it would not be any more > or less loud than if the > devices were exchanging media directly because all we would be doing > is passing the digital packets across. > > I believe you are somehow mistaken in your explanation. There is a > good chance that your x-lite has the gain set lower when you are > testing FS since that's the only device > in your whole scenario that is capable of adjusting the gain. > > If you wish, please get a complete packet capture of a completed call > in both situations. > > > On Sat, Feb 14, 2009 at 8:37 PM, Paul D. > wrote: > > Comparing FS 1.0.3 audio quality vs * 1.4.2, simple Sip-to-Sip > call, or > call to VM prompt, or call via gateway to PSTN - FS audio volume > level > (should I say gain?) seems noticeably lower than on *, this may be a > reason that FS audio seems to be subpar, more noise less clear. Test > calls made using PCMU codec from X-Lite and Linksys 2002. > Is there anything can be tweaked in FS to correct that? Same issue was > with 1.0.2. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From jason at jasonjgw.net Sun Feb 15 18:09:09 2009 From: jason at jasonjgw.net (Jason White) Date: Mon, 16 Feb 2009 13:09:09 +1100 Subject: [Freeswitch-users] Location of config files - Debian packaging issue? Message-ID: <20090216020909.GA5212@jdc.jasonjgw.net> I've found the cause of my problem: As of the 12018 build, FreeSWITCH is searching for its configuration files in /etc/freeswitch rather than /opt/freeswitch/conf. I am using Debian packages built from a copy of the repository. If this is a deliberate change, it's fine, but if it isn't deliberate then something is amiss with the packaging. From brian at freeswitch.org Sun Feb 15 18:11:05 2009 From: brian at freeswitch.org (Brian West) Date: Sun, 15 Feb 2009 20:11:05 -0600 Subject: [Freeswitch-users] FS SIP audio quality? In-Reply-To: <4998C99E.9060706@versafon.com> References: <49977FFD.1020002@versafon.com> <191c3a030902151643n62c2b1a0k2c341eb79c1393ea@mail.gmail.com> <4998C99E.9060706@versafon.com> Message-ID: I'm not able to reproduce this issue.. can you verify the codecs are what you think they are on both Asterisk and FreeSWITCH. /b On Feb 15, 2009, at 8:04 PM, Paul D. wrote: > Well, I tried several call scenarios: > 1. Call from X-Lite or Linksys to VM. > 2. Call from X-Lite or Linksys to a conference. > 3. Call from X-Lite or Linksys to a PSTN number via Gafachi and > CallWithUs. > > I have now * 1.6.5 and FS 1.0.3RC1 installed on the same enterprise > grade Intel server. So just comparing audio in the call scenarios > above > * somehow does noticeably better job, sounds clearer and volume is at > the right level. I am not changing any phone settings of course when > switching between * and FS. > I am not biased towards FS or * at the moment, though FS seems to > have a > better designed configuration options and community. > Just wanted to share my experience, and hear some opinions. > Unfortunately I cannot spend whole amount of time investigating this > case now, capturing packets etc., but I will try to do that once I > have > time. Meanwhile I will have to stick to * for prod. From brian at freeswitch.org Sun Feb 15 18:12:21 2009 From: brian at freeswitch.org (Brian West) Date: Sun, 15 Feb 2009 20:12:21 -0600 Subject: [Freeswitch-users] FS SIP audio quality? In-Reply-To: <4998C99E.9060706@versafon.com> References: <49977FFD.1020002@versafon.com> <191c3a030902151643n62c2b1a0k2c341eb79c1393ea@mail.gmail.com> <4998C99E.9060706@versafon.com> Message-ID: <4DA657EE-2150-435D-BD15-3A5605A0A10F@freeswitch.org> Also you didn't try SVN Trunk? /b On Feb 15, 2009, at 8:04 PM, Paul D. wrote: > I have now * 1.6.5 and FS 1.0.3RC1 From brian at freeswitch.org Sun Feb 15 18:13:57 2009 From: brian at freeswitch.org (Brian West) Date: Sun, 15 Feb 2009 20:13:57 -0600 Subject: [Freeswitch-users] Location of config files - Debian packaging issue? In-Reply-To: <20090216020909.GA5212@jdc.jasonjgw.net> References: <20090216020909.GA5212@jdc.jasonjgw.net> Message-ID: <083B5AE6-D74E-416F-930E-20CEC06DAB0F@freeswitch.org> I think this is in the process of getting corrected to beh the "debian" way. Please join on IRC and interact with everyone related to this. /b On Feb 15, 2009, at 8:09 PM, Jason White wrote: > I've found the cause of my problem: > As of the 12018 build, FreeSWITCH is searching for its configuration > files in > /etc/freeswitch rather than /opt/freeswitch/conf. I am using Debian > packages > built from a copy of the repository. > > If this is a deliberate change, it's fine, but if it isn't > deliberate then > something is amiss with the packaging. From krice at freeswitch.org Sun Feb 15 18:25:10 2009 From: krice at freeswitch.org (Ken Rice) Date: Sun, 15 Feb 2009 20:25:10 -0600 Subject: [Freeswitch-users] FS SIP audio quality? In-Reply-To: <4998C99E.9060706@versafon.com> Message-ID: Paul, If you are truly having a problem please do get us a full packet trace including the RTP... As one of the largest FS users, I can tell you we have not seen this issue and we interconnect with dozens of different endpoint manufacturers using FreeSWITCH. (I run tollfreegateway.com an Open SIP to North American Tollfree TDM termination gateway) If this problem was wide spread I would suspect that users of several ITSPs would be complaining and their ITSPs would be be complaining to me. Now that being said, you're post really smells of a troll. If it is meant as an honest problem please do get us the trace and we'll be more than happy to look at it. Also, as was stated earlier if you are running 1.0.3RC1 then you might see a re-sampling problem in a trans-coding scenario, this has been resolved and you were advised to run trunk to get this fix. As far as your comment on spending too much time to investigate this, all we have asked for is a simple packet trace... This is something that can be done in 5 minutes K > From: "Paul D." > Reply-To: > Date: Sun, 15 Feb 2009 21:04:14 -0500 > To: > Subject: Re: [Freeswitch-users] FS SIP audio quality? > > Well, I tried several call scenarios: > 1. Call from X-Lite or Linksys to VM. > 2. Call from X-Lite or Linksys to a conference. > 3. Call from X-Lite or Linksys to a PSTN number via Gafachi and CallWithUs. > > I have now * 1.6.5 and FS 1.0.3RC1 installed on the same enterprise > grade Intel server. So just comparing audio in the call scenarios above > * somehow does noticeably better job, sounds clearer and volume is at > the right level. I am not changing any phone settings of course when > switching between * and FS. > I am not biased towards FS or * at the moment, though FS seems to have a > better designed configuration options and community. > Just wanted to share my experience, and hear some opinions. > Unfortunately I cannot spend whole amount of time investigating this > case now, capturing packets etc., but I will try to do that once I have > time. Meanwhile I will have to stick to * for prod. > > > Anthony Minessale wrote: >> it's digital audio. The only thing doing sampling and reconstruction >> of the signal are the phones. The audio files have been captured long >> ago from the microphone in the studio. >> We do nothing to alter the volume of the audio signal or manipulate it >> in any way unless you are transcoding between sample rates or codecs >> which you are not because you mentioned it was PCMU. >> >> If you are making a call from x-lite to a linksys using just PCMU >> there is no transcoding going on at all and it would not be any more >> or less loud than if the >> devices were exchanging media directly because all we would be doing >> is passing the digital packets across. >> >> I believe you are somehow mistaken in your explanation. There is a >> good chance that your x-lite has the gain set lower when you are >> testing FS since that's the only device >> in your whole scenario that is capable of adjusting the gain. >> >> If you wish, please get a complete packet capture of a completed call >> in both situations. >> >> >> On Sat, Feb 14, 2009 at 8:37 PM, Paul D. > > wrote: >> >> Comparing FS 1.0.3 audio quality vs * 1.4.2, simple Sip-to-Sip >> call, or >> call to VM prompt, or call via gateway to PSTN - FS audio volume >> level >> (should I say gain?) seems noticeably lower than on *, this may be a >> reason that FS audio seems to be subpar, more noise less clear. Test >> calls made using PCMU codec from X-Lite and Linksys 2002. >> Is there anything can be tweaked in FS to correct that? Same issue was >> with 1.0.2. >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> >> iax:guest at conference.freeswitch.org/888 >> >> googletalk:conf+888 at conference.freeswitch.org >> >> pstn:213-799-1400 >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From anthony.minessale at gmail.com Sun Feb 15 18:43:25 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sun, 15 Feb 2009 20:43:25 -0600 Subject: [Freeswitch-users] FS SIP audio quality? In-Reply-To: <191c3a030902151842n30590accrd6c7df69e546fb49@mail.gmail.com> References: <49977FFD.1020002@versafon.com> <191c3a030902151643n62c2b1a0k2c341eb79c1393ea@mail.gmail.com> <4998C99E.9060706@versafon.com> <191c3a030902151842n30590accrd6c7df69e546fb49@mail.gmail.com> Message-ID: <191c3a030902151843p3fdd1903x990ae6cc991724b2@mail.gmail.com> The typing it takes to start a pcap of each call and email them is less than you have typed thusfar. Please just take the captures and send them to us to examine. That's all. If you have a real issue we would like to address it. On Feb 15, 2009 8:06 PM, "Paul D." wrote: Well, I tried several call scenarios: 1. Call from X-Lite or Linksys to VM. 2. Call from X-Lite or Linksys to a conference. 3. Call from X-Lite or Linksys to a PSTN number via Gafachi and CallWithUs. I have now * 1.6.5 and FS 1.0.3RC1 installed on the same enterprise grade Intel server. So just comparing audio in the call scenarios above * somehow does noticeably better job, sounds clearer and volume is at the right level. I am not changing any phone settings of course when switching between * and FS. I am not biased towards FS or * at the moment, though FS seems to have a better designed configuration options and community. Just wanted to share my experience, and hear some opinions. Unfortunately I cannot spend whole amount of time investigating this case now, capturing packets etc., but I will try to do that once I have time. Meanwhile I will have to stick to * for prod. Anthony Minessale wrote: > it's digital audio. The only thing doing sampling and reconstruction ... > > wrote: > > Comparing FS 1.0.3 audio quality vs * 1.4.2, simple Si... > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.f... > > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > > pstn:213-799-1400 > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-use... -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090215/81a80269/attachment-0002.html From jaybinks at gmail.com Sun Feb 15 18:51:09 2009 From: jaybinks at gmail.com (jay binks) Date: Mon, 16 Feb 2009 12:51:09 +1000 Subject: [Freeswitch-users] FS SIP audio quality? In-Reply-To: <4998C99E.9060706@versafon.com> References: <49977FFD.1020002@versafon.com> <191c3a030902151643n62c2b1a0k2c341eb79c1393ea@mail.gmail.com> <4998C99E.9060706@versafon.com> Message-ID: another thing to try here... is to put FS in RTP proxy and bypass mode. http://wiki.freeswitch.org/wiki/Bypass_Media it would be interesting to see if your still experiencing this problem in either of those 2 modes. Jay On Mon, Feb 16, 2009 at 12:04 PM, Paul D. wrote: > Well, I tried several call scenarios: > 1. Call from X-Lite or Linksys to VM. > 2. Call from X-Lite or Linksys to a conference. > 3. Call from X-Lite or Linksys to a PSTN number via Gafachi and CallWithUs. > > I have now * 1.6.5 and FS 1.0.3RC1 installed on the same enterprise > grade Intel server. So just comparing audio in the call scenarios above > * somehow does noticeably better job, sounds clearer and volume is at > the right level. I am not changing any phone settings of course when > switching between * and FS. > I am not biased towards FS or * at the moment, though FS seems to have a > better designed configuration options and community. > Just wanted to share my experience, and hear some opinions. > Unfortunately I cannot spend whole amount of time investigating this > case now, capturing packets etc., but I will try to do that once I have > time. Meanwhile I will have to stick to * for prod. > > > Anthony Minessale wrote: > > it's digital audio. The only thing doing sampling and reconstruction > > of the signal are the phones. The audio files have been captured long > > ago from the microphone in the studio. > > We do nothing to alter the volume of the audio signal or manipulate it > > in any way unless you are transcoding between sample rates or codecs > > which you are not because you mentioned it was PCMU. > > > > If you are making a call from x-lite to a linksys using just PCMU > > there is no transcoding going on at all and it would not be any more > > or less loud than if the > > devices were exchanging media directly because all we would be doing > > is passing the digital packets across. > > > > I believe you are somehow mistaken in your explanation. There is a > > good chance that your x-lite has the gain set lower when you are > > testing FS since that's the only device > > in your whole scenario that is capable of adjusting the gain. > > > > If you wish, please get a complete packet capture of a completed call > > in both situations. > > > > > > On Sat, Feb 14, 2009 at 8:37 PM, Paul D. > > wrote: > > > > Comparing FS 1.0.3 audio quality vs * 1.4.2, simple Sip-to-Sip > > call, or > > call to VM prompt, or call via gateway to PSTN - FS audio volume > > level > > (should I say gain?) seems noticeably lower than on *, this may be a > > reason that FS audio seems to be subpar, more noise less clear. Test > > calls made using PCMU codec from X-Lite and Linksys 2002. > > Is there anything can be tweaked in FS to correct that? Same issue > was > > with 1.0.2. > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com > > > > > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > > > > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > > > > > iax:guest at conference.freeswitch.org/888 > > > > googletalk:conf+888 at conference.freeswitch.org > > > > > > pstn:213-799-1400 > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Sincerely Jay -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090216/dd111844/attachment-0002.html From jason at jasonjgw.net Sun Feb 15 19:02:52 2009 From: jason at jasonjgw.net (Jason White) Date: Mon, 16 Feb 2009 14:02:52 +1100 Subject: [Freeswitch-users] Build and Sofia issues with recent svn trunk revisions In-Reply-To: <20090215070216.GA20246@jdc.jasonjgw.net> References: <20090215070216.GA20246@jdc.jasonjgw.net> Message-ID: <20090216030252.GA2229@jdc.jasonjgw.net> Jason White wrote: > I decided to try rev. 12027, which, on the same machine (Debian Sid) fails to > build with the following error: > x86_64-linux-gnu-gcc -c -ggdb -I. ./dftables.c > In file included from ./dftables.c:50: > ./pcre_internal.h:239:2: error: #error LINK_SIZE must be either 2, 3, or 4 > make[2]: *** [dftables.o] Error 1 This error occurred with a freshly exported copy of the sources; there were no pre-existing makefiles or autoconf-generated scripts lying around. From brian at freeswitch.org Sun Feb 15 19:14:30 2009 From: brian at freeswitch.org (Brian West) Date: Sun, 15 Feb 2009 21:14:30 -0600 Subject: [Freeswitch-users] Build and Sofia issues with recent svn trunk revisions In-Reply-To: <20090216030252.GA2229@jdc.jasonjgw.net> References: <20090215070216.GA20246@jdc.jasonjgw.net> <20090216030252.GA2229@jdc.jasonjgw.net> Message-ID: Please open a jira http://jira.freeswitch.org /b On Feb 15, 2009, at 9:02 PM, Jason White wrote: > Jason White wrote: > >> I decided to try rev. 12027, which, on the same machine (Debian >> Sid) fails to >> build with the following error: >> x86_64-linux-gnu-gcc -c -ggdb -I. ./dftables.c >> In file included from ./dftables.c:50: >> ./pcre_internal.h:239:2: error: #error LINK_SIZE must be either 2, >> 3, or 4 >> make[2]: *** [dftables.o] Error 1 > > This error occurred with a freshly exported copy of the sources; > there were no > pre-existing makefiles or autoconf-generated scripts lying around. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090215/35d58aa1/attachment-0002.html From jason at jasonjgw.net Sun Feb 15 20:15:25 2009 From: jason at jasonjgw.net (Jason White) Date: Mon, 16 Feb 2009 15:15:25 +1100 Subject: [Freeswitch-users] Build and Sofia issues with recent svn trunk revisions In-Reply-To: References: <20090215070216.GA20246@jdc.jasonjgw.net> <20090216030252.GA2229@jdc.jasonjgw.net> Message-ID: <20090216041525.GA18456@jdc.jasonjgw.net> Brian West wrote: > Please open a jira http://jira.freeswitch.org Has anyone succeeded in doing this with a text-based Web browser such as Lynx or Elinks? Jira keeps complaining that I haven't selected a valid project. There are reasons why I would rather avoid a graphical browser under X at the moment, including X bugs that cause the X server to crash on this machine. From codecomplete at free.fr Sun Feb 15 20:29:45 2009 From: codecomplete at free.fr (Fred) Date: Mon, 16 Feb 2009 05:29:45 +0100 Subject: [Freeswitch-users] Switching from Asterisk to Freeswitch? In-Reply-To: References: Message-ID: <7.0.1.0.2.20090216052901.06a91d28@free.fr> Thanks guys for the input. I'll download FS and give it a shot. From mike at jerris.com Sun Feb 15 22:06:17 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 16 Feb 2009 01:06:17 -0500 Subject: [Freeswitch-users] Location of config files - Debian packaging issue? In-Reply-To: <083B5AE6-D74E-416F-930E-20CEC06DAB0F@freeswitch.org> References: <20090216020909.GA5212@jdc.jasonjgw.net> <083B5AE6-D74E-416F-930E-20CEC06DAB0F@freeswitch.org> Message-ID: <3B7025A0-DAB9-498C-AEF1-2587845C7DE1@jerris.com> This patch was incorrect and was supposed to be reverted. I will correct this error. Mike On Feb 15, 2009, at 9:13 PM, Brian West wrote: > I think this is in the process of getting corrected to beh the > "debian" way. Please join on IRC and interact with everyone related > to this. > > /b > > On Feb 15, 2009, at 8:09 PM, Jason White wrote: > >> I've found the cause of my problem: >> As of the 12018 build, FreeSWITCH is searching for its configuration >> files in >> /etc/freeswitch rather than /opt/freeswitch/conf. I am using Debian >> packages >> built from a copy of the repository. >> >> If this is a deliberate change, it's fine, but if it isn't >> deliberate then >> something is amiss with the packaging. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From woodydickson at gmail.com Mon Feb 16 04:14:43 2009 From: woodydickson at gmail.com (Woody Dickson) Date: Mon, 16 Feb 2009 20:14:43 +0800 Subject: [Freeswitch-users] dynamically add ip to an ACL Message-ID: Hi, Is it possible to dynamically add entries to an ACL without having to go through the xml file? Can it be done via command line or api? Thanks, Woody From leon at scarlet-internet.nl Mon Feb 16 04:49:46 2009 From: leon at scarlet-internet.nl (Leon de Rooij) Date: Mon, 16 Feb 2009 13:49:46 +0100 Subject: [Freeswitch-users] ipauth - directory In-Reply-To: <496E068B.6050404@kinetix.gr> References: <496E0187.8090805@kinetix.gr> <23DEEC73-4DC7-4056-902F-D74DD6644385@freeswitch.org> <496E068B.6050404@kinetix.gr> Message-ID: Hi all, I'd really like to know more about this too. Currently, I have two sip_profiles: - residential (where users can do authenticated registers and invites) - transit (where other users can do un-authenticated invites) Right now, FS is not aware of *who* is accessing the transit profile except for an acl that is set on this profile so unauthorized use is not possible. But what should I do when I want to allow multiple parties (from different IP addresses) to send their invites to the transit profile, and still be able to differentiate between them ? I'd like to set some variables, like an accountcode for example, on the basis of what IP address the INVITE originates from. So, is it possible to not use digest authentication, but still use a dialplan-directory user with IP= field or some such ? thanks a lot & kind regards, Leon de Rooij On Jan 14, 2009, at 4:36 PM, Apostolos Pantsiopoulos wrote: > Yes I know that. But what does the "ip=" setting do? > > Brian West wrote: >> >> cidr= and the domains acl in acl.conf.xml then apply that ACL to the >> sofia profile. >> >> /b >> >> On Jan 14, 2009, at 9:15 AM, Apostolos Pantsiopoulos wrote: >> >> >>> I noticed an "ip=" setting in the brian.xml sample file. >>> The comments state that this is used for ipauth (IP based >>> authentication?) >>> >>> What exactly is this setting. I cannot find anything in the wiki >>> about it. >>> Does it replace the use of the >>> >>> + ACL >>> >>> mechanism for IP authentication? >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > -- > ------------------------------------------- > Apostolos Pantsiopoulos > Kinetix Tele.com R & D > email: regs at kinetix.gr > ------------------------------------------- > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090216/5823e3be/attachment-0002.html From anthony.minessale at gmail.com Mon Feb 16 05:49:12 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 16 Feb 2009 07:49:12 -0600 Subject: [Freeswitch-users] dynamically add ip to an ACL In-Reply-To: References: Message-ID: <191c3a030902160549k4081d4c2s29a3a86bd267febf@mail.gmail.com> no, it's not possible. On Mon, Feb 16, 2009 at 6:14 AM, Woody Dickson wrote: > Hi, > > Is it possible to dynamically add entries to an ACL without having to > go through the xml file? Can it be done via command line or api? > > Thanks, > Woody > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090216/26304763/attachment-0002.html From anthony.minessale at gmail.com Mon Feb 16 06:04:06 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 16 Feb 2009 08:04:06 -0600 Subject: [Freeswitch-users] ipauth - directory In-Reply-To: References: <496E0187.8090805@kinetix.gr> <23DEEC73-4DC7-4056-902F-D74DD6644385@freeswitch.org> <496E068B.6050404@kinetix.gr> Message-ID: <191c3a030902160604r77089a6s5b3b9f3d07914218@mail.gmail.com> you have 3 options. on authenticated users, every tag in his account will be set on each call from that authenticated user. 1) make them register, this sets the variables automatically 2) use the ACL list with cidr= this has the same effect with no auth needed. 3) use some other way to differentiate the user and use the set_user application in the dialplan to inherit that user's variables. On Mon, Feb 16, 2009 at 6:49 AM, Leon de Rooij wrote: > Hi all, > > I'd really like to know more about this too. > > Currently, I have two sip_profiles: > > - residential (where users can do authenticated registers and invites) > - transit (where other users can do un-authenticated invites) > > Right now, FS is not aware of *who* is accessing the transit profile except > for an acl that is set on this profile so unauthorized use is not possible. > > But what should I do when I want to allow multiple parties (from different > IP addresses) to send their invites to the transit profile, and still be > able to differentiate between them ? > > I'd like to set some variables, like an accountcode for example, on the > basis of what IP address the INVITE originates from. > > So, is it possible to not use digest authentication, but still use a > dialplan-directory user with IP= field or some such ? > > thanks a lot & kind regards, > > Leon de Rooij > > > > On Jan 14, 2009, at 4:36 PM, Apostolos Pantsiopoulos wrote: > > Yes I know that. But what does the "ip=" setting do? > > Brian West wrote: > > cidr= and the domains acl in acl.conf.xml then apply that ACL to the > sofia profile. > > /b > > On Jan 14, 2009, at 9:15 AM, Apostolos Pantsiopoulos wrote: > > > > I noticed an "ip=" setting in the brian.xml sample file. > The comments state that this is used for ipauth (IP based > authentication?) > > What exactly is this setting. I cannot find anything in the wiki > about it. > Does it replace the use of the > > + ACL > > mechanism for IP authentication? > > > _______________________________________________ > Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > -- > ------------------------------------------- > Apostolos Pantsiopoulos > Kinetix Tele.com R & D > email: regs at kinetix.gr > ------------------------------------------- > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090216/9bdb02e6/attachment-0002.html From fanatikneo at gmx.de Sun Feb 15 23:59:47 2009 From: fanatikneo at gmx.de (Jan Fricke) Date: Mon, 16 Feb 2009 08:59:47 +0100 Subject: [Freeswitch-users] mod_pa - pa list - call states Message-ID: <49991CF3.5060906@gmx.de> Hello, I'm using Freeswitch (1.0.trunk) as a softphone with mod_pa. My GUI communicates with freeswitch via xml-rpc and fetches calls with "pa list". If somebody is calling, the state of the call is hold. When the call is answered with "pa answer" it is active. If someone calls me while I'm in call the state of the second call is hold. So far so good. But if the first call ends, the second is marked as active although it is still ringing and should be "hold". Is this the intended behavior of "pa list" or did I missunderstand the command? Best regards Jan From anthony.minessale at gmail.com Mon Feb 16 07:10:53 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 16 Feb 2009 09:10:53 -0600 Subject: [Freeswitch-users] mod_pa - pa list - call states In-Reply-To: <49991CF3.5060906@gmx.de> References: <49991CF3.5060906@gmx.de> Message-ID: <191c3a030902160710g464d2210w40fe46e4e126374b@mail.gmail.com> I recommend you use events instead of polling On Mon, Feb 16, 2009 at 1:59 AM, Jan Fricke wrote: > Hello, > I'm using Freeswitch (1.0.trunk) as a softphone with mod_pa. My GUI > communicates with freeswitch via xml-rpc and fetches calls with "pa list". > If somebody is calling, the state of the call is hold. When the call is > answered with "pa answer" it is active. If someone calls me while I'm in > call the state of the second call is hold. So far so good. > But if the first call ends, the second is marked as active although it > is still ringing and should be "hold". > Is this the intended behavior of "pa list" or did I missunderstand the > command? > > Best regards > > Jan > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090216/b09f52fb/attachment-0002.html From brian at freeswitch.org Mon Feb 16 07:05:52 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 16 Feb 2009 09:05:52 -0600 Subject: [Freeswitch-users] mod_pa - pa list - call states In-Reply-To: <49991CF3.5060906@gmx.de> References: <49991CF3.5060906@gmx.de> Message-ID: <366DB45A-6DD3-4309-97A3-498907B5FADE@freeswitch.org> Try "pa switch x", x being the call number. /b On Feb 16, 2009, at 1:59 AM, Jan Fricke wrote: > Hello, > I'm using Freeswitch (1.0.trunk) as a softphone with mod_pa. My GUI > communicates with freeswitch via xml-rpc and fetches calls with "pa > list". > If somebody is calling, the state of the call is hold. When the call > is > answered with "pa answer" it is active. If someone calls me while > I'm in > call the state of the second call is hold. So far so good. > But if the first call ends, the second is marked as active although it > is still ringing and should be "hold". > Is this the intended behavior of "pa list" or did I missunderstand the > command? > > Best regards > > Jan From ajlong at worldlink.net Mon Feb 16 07:18:23 2009 From: ajlong at worldlink.net (Adam Long) Date: Mon, 16 Feb 2009 10:18:23 -0500 Subject: [Freeswitch-users] dynamically add ip to an ACL In-Reply-To: <191c3a030902160549k4081d4c2s29a3a86bd267febf@mail.gmail.com> References: <191c3a030902160549k4081d4c2s29a3a86bd267febf@mail.gmail.com> Message-ID: <014001c99049$cf0edd40$6d2c97c0$@net> Hi Anthony, could he use mod_xml_curl for this to serve up a dynamic acl.conf.xml? Or would reloadacl have to be called somehow? Regards, -Adam From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Monday, February 16, 2009 8:49 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] dynamically add ip to an ACL no, it's not possible. On Mon, Feb 16, 2009 at 6:14 AM, Woody Dickson wrote: Hi, Is it possible to dynamically add entries to an ACL without having to go through the xml file? Can it be done via command line or api? Thanks, Woody _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090216/3786e362/attachment-0002.html From brian at freeswitch.org Mon Feb 16 07:21:20 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 16 Feb 2009 09:21:20 -0600 Subject: [Freeswitch-users] dynamically add ip to an ACL In-Reply-To: <014001c99049$cf0edd40$6d2c97c0$@net> References: <191c3a030902160549k4081d4c2s29a3a86bd267febf@mail.gmail.com> <014001c99049$cf0edd40$6d2c97c0$@net> Message-ID: <6950B56E-25ED-4E99-8495-7966116B2C28@freeswitch.org> Reloadacl would have to be called in either case. /b On Feb 16, 2009, at 9:18 AM, Adam Long wrote: > Hi Anthony, could he use mod_xml_curl for this to serve up a dynamic > acl.conf.xml? > > Or would reloadacl have to be called somehow? > > Regards, > -Adam > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090216/0e5b3402/attachment-0002.html From asannucci at gmail.com Mon Feb 16 07:24:30 2009 From: asannucci at gmail.com (Andrea) Date: Mon, 16 Feb 2009 10:24:30 -0500 Subject: [Freeswitch-users] Perl error when compiling Message-ID: <39B5704908BC4C5BAF066DE0AC8AD32C@quos> Hi, when y try to compiling perl module on freeswitch (from tarball 1.0.3RC1) I have this error: making all mod_perl Creating mod_perl.so... /usr/bin/ld: cannot find -ldb collect2: ld returned 1 exit status Any idea? Thank you - Andrea - From brian at freeswitch.org Mon Feb 16 07:28:19 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 16 Feb 2009 09:28:19 -0600 Subject: [Freeswitch-users] Perl error when compiling In-Reply-To: <39B5704908BC4C5BAF066DE0AC8AD32C@quos> References: <39B5704908BC4C5BAF066DE0AC8AD32C@quos> Message-ID: <8728518C-0AB3-4DE4-BBBB-C85150ACBD31@freeswitch.org> install gdbm-devel and db4-devel. /b On Feb 16, 2009, at 9:24 AM, Andrea wrote: > > making all mod_perl > Creating mod_perl.so... > /usr/bin/ld: cannot find -ldb > collect2: ld returned 1 exit status From hochlehnert at hotmail.com Mon Feb 16 07:28:27 2009 From: hochlehnert at hotmail.com (Klaus Hochlehnert) Date: Mon, 16 Feb 2009 16:28:27 +0100 Subject: [Freeswitch-users] Perl error when compiling In-Reply-To: <39B5704908BC4C5BAF066DE0AC8AD32C@quos> References: <39B5704908BC4C5BAF066DE0AC8AD32C@quos> Message-ID: Hi, install libdb development files, e.g. for Ubuntu/Debian: aptitude install libdb-dev Regards, Klaus> From: asannucci at gmail.com> To: freeswitch-users at lists.freeswitch.org> Date: Mon, 16 Feb 2009 10:24:30 -0500> Subject: [Freeswitch-users] Perl error when compiling> > Hi,> > when y try to compiling perl module on freeswitch (from tarball 1.0.3RC1) I> have this error:> > making all mod_perl> Creating mod_perl.so...> /usr/bin/ld: cannot find -ldb> collect2: ld returned 1 exit status> > Any idea?> > Thank you> > - Andrea -> > > _______________________________________________> Freeswitch-users mailing list> Freeswitch-users at 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://redirect.gimas.net/?n=M0902xHMMobile Nie wieder eine Mail verpassen mit Hotmail f?rs Handy! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090216/eec0c086/attachment-0002.html From mike at jerris.com Mon Feb 16 08:04:59 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 16 Feb 2009 11:04:59 -0500 Subject: [Freeswitch-users] Location of config files - Debian packaging issue? In-Reply-To: <3B7025A0-DAB9-498C-AEF1-2587845C7DE1@jerris.com> References: <20090216020909.GA5212@jdc.jasonjgw.net> <083B5AE6-D74E-416F-930E-20CEC06DAB0F@freeswitch.org> <3B7025A0-DAB9-498C-AEF1-2587845C7DE1@jerris.com> Message-ID: I have reverted this patch, it should be in /opt/freeswitch/conf in trunk now properly. Mike On Feb 16, 2009, at 1:06 AM, Michael Jerris wrote: > This patch was incorrect and was supposed to be reverted. I will > correct this error. > > Mike > > On Feb 15, 2009, at 9:13 PM, Brian West wrote: > >> I think this is in the process of getting corrected to beh the >> "debian" way. Please join on IRC and interact with everyone related >> to this. >> >> /b >> >> On Feb 15, 2009, at 8:09 PM, Jason White wrote: >> >>> I've found the cause of my problem: >>> As of the 12018 build, FreeSWITCH is searching for its configuration >>> files in >>> /etc/freeswitch rather than /opt/freeswitch/conf. I am using Debian >>> packages >>> built from a copy of the repository. >>> >>> If this is a deliberate change, it's fine, but if it isn't >>> deliberate then >>> something is amiss with the packaging. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > From asannucci at gmail.com Mon Feb 16 08:57:27 2009 From: asannucci at gmail.com (Andrea) Date: Mon, 16 Feb 2009 11:57:27 -0500 Subject: [Freeswitch-users] Perl error when compiling References: <39B5704908BC4C5BAF066DE0AC8AD32C@quos> <8728518C-0AB3-4DE4-BBBB-C85150ACBD31@freeswitch.org> Message-ID: <243DB2D99A1149CB8022A4B43C5B1358@quos> Thank you. Now work fine Now when i unload and load mod_java i receive this error: -freeswitch@ unload mod_java 2009-02-16 11:51:36 [NOTICE] switch_loadable_module.c:510 switch_loadable_module_unprocess() Deleting Application 'java' 2009-02-16 11:51:36 [CONSOLE] switch_loadable_module.c:1231 do_shutdown() Stopping: mod_java API CALL [unload(mod_java)] output: +OK 2009-02-16 11:51:37 [CONSOLE] switch_loadable_module.c:1244 do_shutdown() mod_java unloaded. freeswitch@ load mod_java API CALL [load(mod_java)] output: -ERR [module load file routine returned an error] freeswitch@ 2009-02-16 11:51:41 [NOTICE] modjava.c:244 mod_java_load() Java Framework Loading... 2009-02-16 11:51:41 [ERR] modjava.c:222 create_java_vm() Error creating Java VM! 2009-02-16 11:51:41 [CRIT] switch_loadable_module.c:839 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_java.so **Module load routine returned an error** This is my java.conf.xml Y have configured the PATH for jdk to point to right directory and i run ./configure with the java path options I hope you understand. I'm new on this :) Regards - Andrea - From msc at freeswitch.org Mon Feb 16 13:14:37 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 16 Feb 2009 13:14:37 -0800 Subject: [Freeswitch-users] Getting current call count In-Reply-To: References: Message-ID: <87f2f3b90902161314y3b8387cayfb7eb3c5c883c736@mail.gmail.com> > Show calls count > > > > This delivers a value of zero. Remember that a "call" is defined as "two or more bridged channels" so you could have lots of channels that aren't bridged and therefore many channels but zero calls. -MC From simon at airg.com Mon Feb 16 14:26:23 2009 From: simon at airg.com (Simon Tang) Date: Mon, 16 Feb 2009 14:26:23 -0800 Subject: [Freeswitch-users] How to keep leg A from hanging up from a bridge when Leg B hangs up (and vice versa)? Message-ID: <872970CF4A55BF42A5337D570860209F018189D8@HPEXCHVS01.exchange.airg> Hello, I'm using event socket outbound and have a framework that does stuff based on the events that come back (this includes my own IVR). What I have now is an IVR system that allows 2 users to bridge to one another at will, and to 'unbridge' at will by catching DTMF events. I have 2 requirements: 1. When one leg hangs up during a bridge, the other leg is presented with the IVR 2. After a bridge, when one leg sends a DTMF tone, both legs will be presented with the IVR and no longer be bridged (they can bridge with other sessions again after this point if they desire) I have done multiple experiments by using netcat and 2 sessions. Here is what I have found: * The hangup_after_bridge variable does nothing for me. I've set it on both legs, but whenever one leg hangs up after a uuid_bridge, the other leg will automatically hang up * I've tried setting "park_after_bridge=true" on both legs, and this works to a certain extent. If one leg hangs up, the other leg will be parked, and I can present that user with my IVR. This meets requirement #1. However, requirement #2 won't be met because: o If I set "park_after_bridge=true" and one leg sends a DTMF tone to signal an unbridge, I will "unbridge" the legs by "parking" both legs and I am able to present them both with an IVR. If they decide to bridge with each other again (by selecting an option in the IVR), I will attempt to do a uuid_bridge and this will FAIL! (both parties do not hear each other.) In the simplest terms, I can't do "uuid_bridge uuidA uuidB", "park", "uuid_bridge uuidA uuidB". * With "park_after_bridge=false" (default), I can do "uuid_bridge uuidA uuidB", "park", "uuid_bridge uuidA uuidB" with no issues, meeting requirement #2. However, this will not meet requirement #1, because when one leg hangs up, it will trigger a hangup on the other. Please help. How can I meet both of my requirements? Thanks. Simon Tang Lead, Server Team Suite 706, 1155 Robson Street Vancouver, B.C. Canada V6E 1B5 T: +1.604.408.2228 Ext. 116 F: +1.866.874.8136 E: simon at airg.com W: www.airg.com The information transmitted is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material communicated under NDA. Any review, retransmission, dissemination or other use of, or taking of any action in reliance upon, this information by persons or entities other than the intended recipient is prohibited. If you received this in error, please contact the sender and delete the material from any computer. -------------- next part -------------- An HTML attachment was scrubbed... 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Name: not available Type: image/jpeg Size: 1010 bytes Desc: image001.jpg Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090216/392db3da/attachment-0002.jpe From msc at freeswitch.org Mon Feb 16 14:32:51 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 16 Feb 2009 14:32:51 -0800 Subject: [Freeswitch-users] [newbie] Clean start with asimple configuration In-Reply-To: <20090216014438.GA4590@jdc.jasonjgw.net> References: <70ad8fdb0902141602w5829a059p3fe9b4159cba499a@mail.gmail.com> <13479F65-7CE3-4C41-9845-6D92A647C946@freeswitch.org> <2FA0A303-9D53-453A-89A3-517D4B7FBC19@marocon.com> <191c3a030902151651k4108d9a0he12fa555ac6ceab1@mail.gmail.com> <20090216014438.GA4590@jdc.jasonjgw.net> Message-ID: <87f2f3b90902161432o66faf2c0pc32d1d906db4c22@mail.gmail.com> > I must be one of those rare users who stand in the middle: I had used Asterisk > before, but I didn't try to apply my Asterisk knowledge to learning > FreeSWITCH, other than to make sure that all of the desirable features of my > Asterisk configuration eventually had counterparts in my FreeSWITCH > configuration. > The key for people coming from Asterisk is to learn the difference between the WHAT and the HOW. It's admittedly difficult, so any who've overcome the challenges and can help other new ones to make the transition are welcome to document their steps. My advice to Asterisk users who want their FS box to do something that their Asterisk box can do is this: separate the WHAT from the HOW. In other words, start at the top (WHAT) and work your way down, not at the bottom (WHAT). The top-down view is like this: Asterisk allows to SIP phones to talk to each other. The bottom-up viewpoint is this: I edited sip.conf, created these PEER entries, then I edited extensions.conf and added exten => foo... See the difference? Start at the top: ask yourself, WHAT does my Ast box do? Now, ask yourself, HOW does FS implement that feature? As much as I like the Rosetta Stone page (because I started it) I don't think that it is the first place to go when you are migrating from Asterisk. (Go there *after* you've played around with FS for a bit.) -MC From jason at jasonjgw.net Mon Feb 16 14:44:52 2009 From: jason at jasonjgw.net (Jason White) Date: Tue, 17 Feb 2009 09:44:52 +1100 Subject: [Freeswitch-users] Location of config files - Debian packaging issue? In-Reply-To: References: <20090216020909.GA5212@jdc.jasonjgw.net> <083B5AE6-D74E-416F-930E-20CEC06DAB0F@freeswitch.org> <3B7025A0-DAB9-498C-AEF1-2587845C7DE1@jerris.com> Message-ID: <20090216224452.GA6177@jdc.jasonjgw.net> Michael Jerris wrote: > I have reverted this patch, it should be in /opt/freeswitch/conf in > trunk now properly. Thank you for the excellent work. I will upgrade to it as soon as this pcre build problem is fixed - it's still failing with latest trunk, by the way. From msc at freeswitch.org Mon Feb 16 14:49:31 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 16 Feb 2009 14:49:31 -0800 Subject: [Freeswitch-users] How to keep leg A from hanging up from a bridge when Leg B hangs up (and vice versa)? In-Reply-To: <872970CF4A55BF42A5337D570860209F018189D8@HPEXCHVS01.exchange.airg> References: <872970CF4A55BF42A5337D570860209F018189D8@HPEXCHVS01.exchange.airg> Message-ID: <87f2f3b90902161449s26a2f63u55454bd4ac9b338b@mail.gmail.com> On Mon, Feb 16, 2009 at 2:26 PM, Simon Tang wrote: > Hello, > > > > I'm using event socket outbound and have a framework that does stuff based > on the events that come back (this includes my own IVR). What I have now is > an IVR system that allows 2 users to bridge to one another at will, and to > 'unbridge' at will by catching DTMF events. Two questions: What version of FS? Preferably latest SVN Are you using the default config, the one created with "make samples"? -MC From simon at airg.com Mon Feb 16 15:04:05 2009 From: simon at airg.com (Simon Tang) Date: Mon, 16 Feb 2009 15:04:05 -0800 Subject: [Freeswitch-users] How to keep leg A from hanging up from abridge when Leg B hangs up (and vice versa)? In-Reply-To: <87f2f3b90902161449s26a2f63u55454bd4ac9b338b@mail.gmail.com> References: <872970CF4A55BF42A5337D570860209F018189D8@HPEXCHVS01.exchange.airg> <87f2f3b90902161449s26a2f63u55454bd4ac9b338b@mail.gmail.com> Message-ID: <872970CF4A55BF42A5337D570860209F018189E9@HPEXCHVS01.exchange.airg> Revision 10626. Default config. I haven't tried latest svn yet, because my framework breaks with it (probably due to some event formatting changes). Can you explain what you mean by "make samples" and what default config? I don't recall ever doing a "make samples". -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: February 16, 2009 2:50 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] How to keep leg A from hanging up from abridge when Leg B hangs up (and vice versa)? On Mon, Feb 16, 2009 at 2:26 PM, Simon Tang wrote: > Hello, > > > > I'm using event socket outbound and have a framework that does stuff based > on the events that come back (this includes my own IVR). What I have now is > an IVR system that allows 2 users to bridge to one another at will, and to > 'unbridge' at will by catching DTMF events. Two questions: What version of FS? Preferably latest SVN Are you using the default config, the one created with "make samples"? -MC _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From msc at freeswitch.org Mon Feb 16 15:30:22 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 16 Feb 2009 15:30:22 -0800 Subject: [Freeswitch-users] How to keep leg A from hanging up from abridge when Leg B hangs up (and vice versa)? In-Reply-To: <872970CF4A55BF42A5337D570860209F018189E9@HPEXCHVS01.exchange.airg> References: <872970CF4A55BF42A5337D570860209F018189D8@HPEXCHVS01.exchange.airg> <87f2f3b90902161449s26a2f63u55454bd4ac9b338b@mail.gmail.com> <872970CF4A55BF42A5337D570860209F018189E9@HPEXCHVS01.exchange.airg> Message-ID: <87f2f3b90902161530u32ab7035v615d4171df4e6407@mail.gmail.com> On Mon, Feb 16, 2009 at 3:04 PM, Simon Tang wrote: > Revision 10626. Default config. I haven't tried latest svn yet, > because my framework breaks with it (probably due to some event > formatting changes). > > Can you explain what you mean by "make samples" and what default config? > I don't recall ever doing a "make samples". > I suppose I should have asked what your platform is! :) Is this Windows, Linux, Unix, or Mac ? -MC From jason at jasonjgw.net Mon Feb 16 15:46:05 2009 From: jason at jasonjgw.net (Jason White) Date: Tue, 17 Feb 2009 10:46:05 +1100 Subject: [Freeswitch-users] Build and Sofia issues with recent svn trunk revisions In-Reply-To: <20090216030252.GA2229@jdc.jasonjgw.net> References: <20090215070216.GA20246@jdc.jasonjgw.net> <20090216030252.GA2229@jdc.jasonjgw.net> Message-ID: <20090216234605.GA26164@jdc.jasonjgw.net> A fresh checkout from svn trunk fixed my problem, giving me a working build. Interestingly, running svn export and building from a separate directly wasn't enough; a full svn checkout proved necessary to fix this. From lfurrea at gmail.com Mon Feb 16 15:52:47 2009 From: lfurrea at gmail.com (Luis F Urrea) Date: Mon, 16 Feb 2009 17:52:47 -0600 Subject: [Freeswitch-users] xml_cdr Call Flow for attended transfer Message-ID: Hi all, I am trying to understand xml_cdr for an attended (consultative) transfer, I was thinking that the A-leg that initially originated the call would remain untouched but I see that it's global tags get replaced. I have a test call that goes as follows: 201 originates a call and talks to 203 -----> A-leg(1) and B-leg(1) 203 puts 201 on hold and calls 202 (attended) ------> A-leg(2) and B-leg(2) 203 transfers the call 201 and 202 are talking ------> A-leg(1) w/ B-leg(2) ??? Here are the relevant captures: A-leg(1) http://pastebin.freeswitch.org/7253 B-leg(1) http://pastebin.freeswitch.org/7254 A-leg(2) http://pastebin.freeswitch.org/7252 B-leg(2) http://pastebin.freeswitch.org/7255 I was expecting A-leg(1) to have corresponding to 201 which is the original A-leg but it seems that on the transfer, it reverts and 202 appears as the A-leg and 201 as the B-leg. Can someone shed some light on how that transfer gets logged in terms of A-leg and B-leg? TIA -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090216/ced47935/attachment-0002.html From simon at airg.com Mon Feb 16 15:58:03 2009 From: simon at airg.com (Simon Tang) Date: Mon, 16 Feb 2009 15:58:03 -0800 Subject: [Freeswitch-users] How to keep leg A from hanging up fromabridge when Leg B hangs up (and vice versa)? In-Reply-To: <87f2f3b90902161530u32ab7035v615d4171df4e6407@mail.gmail.com> References: <872970CF4A55BF42A5337D570860209F018189D8@HPEXCHVS01.exchange.airg><87f2f3b90902161449s26a2f63u55454bd4ac9b338b@mail.gmail.com><872970CF4A55BF42A5337D570860209F018189E9@HPEXCHVS01.exchange.airg> <87f2f3b90902161530u32ab7035v615d4171df4e6407@mail.gmail.com> Message-ID: <872970CF4A55BF42A5337D570860209F018189FE@HPEXCHVS01.exchange.airg> Linux :). I've mucked around in the conf files, sip_profiles and dialplans to customize it for my use, but that's as far as I've done in playing with configs. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: February 16, 2009 3:30 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] How to keep leg A from hanging up fromabridge when Leg B hangs up (and vice versa)? On Mon, Feb 16, 2009 at 3:04 PM, Simon Tang wrote: > Revision 10626. Default config. I haven't tried latest svn yet, > because my framework breaks with it (probably due to some event > formatting changes). > > Can you explain what you mean by "make samples" and what default config? > I don't recall ever doing a "make samples". > I suppose I should have asked what your platform is! :) Is this Windows, Linux, Unix, or Mac ? -MC _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From pauld at versafon.com Mon Feb 16 16:06:18 2009 From: pauld at versafon.com (Paul D.) Date: Mon, 16 Feb 2009 19:06:18 -0500 Subject: [Freeswitch-users] FS SIP audio quality? In-Reply-To: References: Message-ID: <4999FF7A.6080704@versafon.com> Eh?? Ken Rice wrote: > Paul, > > > > Now that being said, you're post really smells of a troll. > > > From pauld at versafon.com Mon Feb 16 16:08:28 2009 From: pauld at versafon.com (Paul D.) Date: Mon, 16 Feb 2009 19:08:28 -0500 Subject: [Freeswitch-users] FS SIP audio quality? In-Reply-To: <191c3a030902151843p3fdd1903x990ae6cc991724b2@mail.gmail.com> References: <49977FFD.1020002@versafon.com> <191c3a030902151643n62c2b1a0k2c341eb79c1393ea@mail.gmail.com> <4998C99E.9060706@versafon.com> <191c3a030902151842n30590accrd6c7df69e546fb49@mail.gmail.com> <191c3a030902151843p3fdd1903x990ae6cc991724b2@mail.gmail.com> Message-ID: <4999FFFC.10504@versafon.com> I was trying to send tcp dumps today, but the message was rejected because of its size (zipped). How do I send them? Anthony Minessale wrote: > > The typing it takes to start a pcap of each call and email them is > less than you have typed thusfar. > Please just take the captures and send them to us to examine. That's > all. If you have a real issue we would like to address it. > From brian at freeswitch.org Mon Feb 16 16:15:57 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 16 Feb 2009 18:15:57 -0600 Subject: [Freeswitch-users] FS SIP audio quality? In-Reply-To: <4999FFFC.10504@versafon.com> References: <49977FFD.1020002@versafon.com> <191c3a030902151643n62c2b1a0k2c341eb79c1393ea@mail.gmail.com> <4998C99E.9060706@versafon.com> <191c3a030902151842n30590accrd6c7df69e546fb49@mail.gmail.com> <191c3a030902151843p3fdd1903x990ae6cc991724b2@mail.gmail.com> <4999FFFC.10504@versafon.com> Message-ID: <92BA7612-7F23-4079-BE13-755823F6EB49@freeswitch.org> You can send them directly to me brian at freeswitch.org Thanks, /b On Feb 16, 2009, at 6:08 PM, Paul D. wrote: > I was trying to send tcp dumps today, but the message was rejected > because of its size (zipped). How do I send them? > > > Anthony Minessale wrote: >> >> The typing it takes to start a pcap of each call and email them is >> less than you have typed thusfar. >> Please just take the captures and send them to us to examine. That's >> all. If you have a real issue we would like to address it. >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From msc at freeswitch.org Mon Feb 16 16:18:25 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 16 Feb 2009 16:18:25 -0800 Subject: [Freeswitch-users] FS SIP audio quality? In-Reply-To: <4999FFFC.10504@versafon.com> References: <49977FFD.1020002@versafon.com> <191c3a030902151643n62c2b1a0k2c341eb79c1393ea@mail.gmail.com> <4998C99E.9060706@versafon.com> <191c3a030902151842n30590accrd6c7df69e546fb49@mail.gmail.com> <191c3a030902151843p3fdd1903x990ae6cc991724b2@mail.gmail.com> <4999FFFC.10504@versafon.com> Message-ID: <87f2f3b90902161618r19413bb9p238ef00f0f7f10f2@mail.gmail.com> On Mon, Feb 16, 2009 at 4:08 PM, Paul D. wrote: > I was trying to send tcp dumps today, but the message was rejected > because of its size (zipped). How do I send them? Can you put them on a server where the devs can use wget or a browser to download them? -MC From chavpaskov at shaw.ca Mon Feb 16 17:40:04 2009 From: chavpaskov at shaw.ca (Tchavdar Paskov) Date: Mon, 16 Feb 2009 17:40:04 -0800 Subject: [Freeswitch-users] Passing Caller_ID to MOD_LCR Message-ID: Hi, Is there a way to pass Caller Id to mod lcr and somehow to include it in a custom? sql. currently? my dialplan looks like this: ?? ????? ???????? ?????? ????? ????? ??? i guess that Caller_ID is already passed but i was thinking? about making some LCR decisions based on Destination number and Caller_ID /Interstate,Intrastate for example/ Thanks for your time Regards Chav -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090216/a4ee4ffd/attachment-0002.html From krice at suspicious.org Mon Feb 16 18:21:13 2009 From: krice at suspicious.org (Ken Rice) Date: Mon, 16 Feb 2009 20:21:13 -0600 Subject: [Freeswitch-users] FS SIP audio quality? In-Reply-To: <4999FF7A.6080704@versafon.com> Message-ID: If it wasn't meant as a troll I personally, publically, and directly apologize to you... What I see all the time is people want everything for free and then think its the developers responsibility to give away free tech support on this software which is free in the first place. Tony and his crew work on FreeSWITCH as much to feed their families as they do to have an open platform that anyone can use K > From: "Paul D." > Reply-To: > Date: Mon, 16 Feb 2009 19:06:18 -0500 > To: > Subject: Re: [Freeswitch-users] FS SIP audio quality? > > Eh?? > > Ken Rice wrote: >> Paul, >> >> >> >> Now that being said, you're post really smells of a troll. >> >> >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From pauld at versafon.com Mon Feb 16 19:33:26 2009 From: pauld at versafon.com (Paul D.) Date: Mon, 16 Feb 2009 22:33:26 -0500 Subject: [Freeswitch-users] FS SIP audio quality? In-Reply-To: References: <49977FFD.1020002@versafon.com> <191c3a030902151643n62c2b1a0k2c341eb79c1393ea@mail.gmail.com> <4998C99E.9060706@versafon.com> Message-ID: <499A3006.6070208@versafon.com> I re-tested calls to VM replacing some of FS prompts with * ones, and it appears that * sounds were recorded with a better quality/higher volume, so FS itself has nothing to do with that. That's solved. :-) I am going to double check all the equipment we used for tests, like headphones, telephone sets, cables since I am almost convinced that there's nothing in FS which can produce effects I observe. I will post back if I find anything wrong, appreciate everybody's help with this. Brian West wrote: > I'm not able to reproduce this issue.. can you verify the codecs are > what you think they are on both Asterisk and FreeSWITCH. > > /b > > On Feb 15, 2009, at 8:04 PM, Paul D. wrote: > > >> Well, I tried several call scenarios: >> 1. Call from X-Lite or Linksys to VM. >> 2. Call from X-Lite or Linksys to a conference. >> 3. Call from X-Lite or Linksys to a PSTN number via Gafachi and >> CallWithUs. >> >> I have now * 1.6.5 and FS 1.0.3RC1 installed on the same enterprise >> grade Intel server. So just comparing audio in the call scenarios >> above >> * somehow does noticeably better job, sounds clearer and volume is at >> the right level. I am not changing any phone settings of course when >> switching between * and FS. >> I am not biased towards FS or * at the moment, though FS seems to >> have a >> better designed configuration options and community. >> Just wanted to share my experience, and hear some opinions. >> Unfortunately I cannot spend whole amount of time investigating this >> case now, capturing packets etc., but I will try to do that once I >> have >> time. Meanwhile I will have to stick to * for prod. >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From freeswitch-users at lists.rupa.com Mon Feb 16 21:01:11 2009 From: freeswitch-users at lists.rupa.com (Rupa Schomaker (lists)) Date: Mon, 16 Feb 2009 23:01:11 -0600 Subject: [Freeswitch-users] Passing Caller_ID to MOD_LCR In-Reply-To: References: Message-ID: <499A4497.9080001@lists.rupa.com> On 2/16/2009 7:40 PM, Tchavdar Paskov wrote: > > Hi, > > Is there a way to pass Caller Id to mod lcr and somehow to include it in > > a custom sql. > > currently my dialplan looks like this: > > > > > > > break="never"> > > > > > > > > > > > > i guess that Caller_ID is already passed but i was thinking about > > making some LCR decisions based on Destination number and Caller_ID > > /Interstate,Intrastate for example/ > > > > Thanks for your time mod_lcr doesn't make any decisions based on caller id. Probably the best way to handle this would be to use profiles. You could extract the areacode and use that to determine which profile to use. It would be awkward if you want to handle all area codes -- but for a smaller set of area codes it might be sufficient. > > Regards > > Chav > > From cesar at auronix.com Mon Feb 16 21:33:19 2009 From: cesar at auronix.com (Cesar Cepeda) Date: Mon, 16 Feb 2009 23:33:19 -0600 Subject: [Freeswitch-users] Playing a G729 file as ringback Message-ID: <088801c990c1$3e139b50$ba3ad1f0$@com> Hi, I'm using FS with g279 on passthrough mode and I'm trying to play a g729 file as ringback to the A-leg while bridging a call. As far as I understand it should go something like this: . originate {channel_vars}dialstring . set some combination of values on 'ringback', 'transfer_ringback', 'instant_ringback' . bridge The bridge works correctly, but no matter what combination of values I try for the 'xxxringback' vars I never hear the ringback on the A-leg. Can you tell me what I'm missing? Thanks. Cesar Cepeda. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090216/4b920469/attachment-0002.html From chavpaskov at shaw.ca Mon Feb 16 22:44:01 2009 From: chavpaskov at shaw.ca (Tchavdar Paskov) Date: Mon, 16 Feb 2009 22:44:01 -0800 Subject: [Freeswitch-users] Passing Caller_ID to MOD_LCR In-Reply-To: <499A4497.9080001@lists.rupa.com> References: <499A4497.9080001@lists.rupa.com> Message-ID: i was thinking more in direction of building custom sql that deals with both Caller_id and destination number. I'm aware that the current? off the box mod_lcr? has no such ability? and that's why i asked if there is a way from? dial plan to pass the caller_id_number? channel variable to? mod_lcr. Thanks? ----- Original Message ----- From: "Rupa Schomaker (lists)" Date: Monday, February 16, 2009 9:01 pm Subject: Re: [Freeswitch-users] Passing Caller_ID to MOD_LCR To: freeswitch-users at lists.freeswitch.org > On 2/16/2009 7:40 PM, Tchavdar Paskov wrote: > > > Hi, > > > Is there a way to pass Caller Id to mod lcr and somehow to > include it in > > > a custom? sql. > > > currently? my dialplan looks like this: > > > > > >??? > > >?????? field="destination_number" expression="^1(\d+)$" > > > break="never"> > > >????????? > > > >??????? application="lcr" data="$1"/> > > >?????? application="bridge" data="${lcr_auto_route}"/> > > >?????? > > >???? > > > i guess that Caller_ID is already passed but i was > thinking? about > > > making some LCR decisions based on Destination number and > Caller_ID> > /Interstate,Intrastate for example/ > > > > > > Thanks for your time > > mod_lcr doesn't make any decisions based on caller id.? > Probably the > best way to handle this would be to use profiles.? You > could extract the > areacode and use that to determine which profile to use.? > It would be > awkward if you want to handle all area codes -- but for a > smaller set of > area codes it might be sufficient. > > > > > Regards > > > Chav > > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090216/2ac5de86/attachment-0002.html From chavpaskov at shaw.ca Mon Feb 16 23:18:47 2009 From: chavpaskov at shaw.ca (Tchavdar Paskov) Date: Mon, 16 Feb 2009 23:18:47 -0800 Subject: [Freeswitch-users] Passing Caller_ID to MOD_LCR In-Reply-To: References: <499A4497.9080001@lists.rupa.com> Message-ID: I also have? another question. When i insert my custom query? it looks like the profile is loaded successfully? but then when i place a call or use ? lcr? ##########? default? /which is where i defined the sql query/? and check the console output? turns out that the switch is using? the default? sql guery . How i can make sure that? the custom sql is the only one that is to be executed. /All required fields in the custom sql are in accordance with the requirements - it returns the exact required names and returned field number is also correct/. Thank you agai for your time Chav ----- Original Message ----- From: Tchavdar Paskov Date: Monday, February 16, 2009 10:44 pm Subject: Re: [Freeswitch-users] Passing Caller_ID to MOD_LCR To: freeswitch-users at lists.freeswitch.org > i was thinking more in direction of building custom sql that > deals with both Caller_id and destination number. I'm aware that > the current? off the box mod_lcr? has no such ability? and > that's why i asked if there is a way from? dial plan to pass the > caller_id_number? channel variable to? mod_lcr. > Thanks? > > ----- Original Message ----- > From: "Rupa Schomaker (lists)" > Date: Monday, February 16, 2009 9:01 pm > Subject: Re: [Freeswitch-users] Passing Caller_ID to MOD_LCR > To: freeswitch-users at lists.freeswitch.org > > > On 2/16/2009 7:40 PM, Tchavdar Paskov wrote: > > > > Hi, > > > > Is there a way to pass Caller Id to mod lcr and somehow to > > include it in > > > > a custom? sql. > > > > currently? my dialplan looks like this: > > > > > > > >??? > > > >?????? > field="destination_number" expression="^1(\d+)$" > > > > break="never"> > > > >????????? > > > > > >??????? > application="lcr" data="$1"/> > > > >?????? > application="bridge" data="${lcr_auto_route}"/> > > > >?????? > > > >???? > > > > i guess that Caller_ID is already passed but i was > > thinking? about > > > > making some LCR decisions based on Destination number and > > Caller_ID> > /Interstate,Intrastate for example/ > > > > > > > > Thanks for your time > > > > mod_lcr doesn't make any decisions based on caller id.? > > Probably the > > best way to handle this would be to use profiles.? You > > could extract the > > areacode and use that to determine which profile to use.? > > It would be > > awkward if you want to handle all area codes -- but for a > > smaller set of > > area codes it might be sufficient. > > > > > > > > Regards > > > > Chav > > > > > > > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090216/20a911a0/attachment-0002.html From nik.middleton at noblesolutions.co.uk Tue Feb 17 02:05:12 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Tue, 17 Feb 2009 10:05:12 -0000 Subject: [Freeswitch-users] FS SIP audio quality? In-Reply-To: <499A3006.6070208@versafon.com> References: <49977FFD.1020002@versafon.com> <191c3a030902151643n62c2b1a0k2c341eb79c1393ea@mail.gmail.com> <4998C99E.9060706@versafon.com> <499A3006.6070208@versafon.com> Message-ID: For what it's worth, using Asterisk recordings, I found FS to be better than when played on an Asterisk system. I came to the same conclusion early on that the included prompts with FS were of a relatively poor nature. Not volunteering to record new ones, but they do let the product down, as they lead to discussions as below. Regards, -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Paul D. Sent: 17 February 2009 03:33 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] FS SIP audio quality? I re-tested calls to VM replacing some of FS prompts with * ones, and it appears that * sounds were recorded with a better quality/higher volume, so FS itself has nothing to do with that. That's solved. :-) I am going to double check all the equipment we used for tests, like headphones, telephone sets, cables since I am almost convinced that there's nothing in FS which can produce effects I observe. I will post back if I find anything wrong, appreciate everybody's help with this. Brian West wrote: > I'm not able to reproduce this issue.. can you verify the codecs are > what you think they are on both Asterisk and FreeSWITCH. > > /b > > On Feb 15, 2009, at 8:04 PM, Paul D. wrote: > > >> Well, I tried several call scenarios: >> 1. Call from X-Lite or Linksys to VM. >> 2. Call from X-Lite or Linksys to a conference. >> 3. Call from X-Lite or Linksys to a PSTN number via Gafachi and >> CallWithUs. >> >> I have now * 1.6.5 and FS 1.0.3RC1 installed on the same enterprise >> grade Intel server. So just comparing audio in the call scenarios >> above >> * somehow does noticeably better job, sounds clearer and volume is at >> the right level. I am not changing any phone settings of course when >> switching between * and FS. >> I am not biased towards FS or * at the moment, though FS seems to >> have a >> better designed configuration options and community. >> Just wanted to share my experience, and hear some opinions. >> Unfortunately I cannot spend whole amount of time investigating this >> case now, capturing packets etc., but I will try to do that once I >> have >> time. Meanwhile I will have to stick to * for prod. >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From gmaruzz at celliax.org Tue Feb 17 02:19:30 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Tue, 17 Feb 2009 11:19:30 +0100 Subject: [Freeswitch-users] FS SIP audio quality? In-Reply-To: References: <49977FFD.1020002@versafon.com> <191c3a030902151643n62c2b1a0k2c341eb79c1393ea@mail.gmail.com> <4998C99E.9060706@versafon.com> <499A3006.6070208@versafon.com> Message-ID: <7b197bef0902170219l217522aal4fff04b5e0e51d06@mail.gmail.com> There is also another side to make mimd to: the Asterisk sounds you hear more often (the demo ones) are very long ones. The ones of the FS demo are very very short (many times just one word) and concatenated with the insertion of sleeps. That is probably someway altering the equation between user experiences Sincerely, Giovanni Maruzzelli ========================================= www.celliax.org via Pierlombardo 9, 20135 Milano Italy gmaruzz at celliax dot org Cell : +39-347-2665618 Fax : +39-02-87390039 On Tue, Feb 17, 2009 at 11:05 AM, Nik Middleton wrote: > For what it's worth, using Asterisk recordings, I found FS to be better > than when played on an Asterisk system. > > I came to the same conclusion early on that the included prompts with FS > were of a relatively poor nature. Not volunteering to record new ones, > but they do let the product down, as they lead to discussions as below. > > > Regards, > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Paul > D. > Sent: 17 February 2009 03:33 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] FS SIP audio quality? > > I re-tested calls to VM replacing some of FS prompts with * ones, and it > > appears that * sounds were recorded with a better quality/higher volume, > > so FS itself has nothing to do with that. That's solved. :-) > I am going to double check all the equipment we used for tests, like > headphones, telephone sets, cables since I am almost convinced that > there's nothing in FS which can produce effects I observe. > I will post back if I find anything wrong, appreciate everybody's help > with this. > > Brian West wrote: >> I'm not able to reproduce this issue.. can you verify the codecs are >> what you think they are on both Asterisk and FreeSWITCH. >> >> /b >> >> On Feb 15, 2009, at 8:04 PM, Paul D. wrote: >> >> >>> Well, I tried several call scenarios: >>> 1. Call from X-Lite or Linksys to VM. >>> 2. Call from X-Lite or Linksys to a conference. >>> 3. Call from X-Lite or Linksys to a PSTN number via Gafachi and >>> CallWithUs. >>> >>> I have now * 1.6.5 and FS 1.0.3RC1 installed on the same enterprise >>> grade Intel server. So just comparing audio in the call scenarios >>> above >>> * somehow does noticeably better job, sounds clearer and volume is at >>> the right level. I am not changing any phone settings of course when >>> switching between * and FS. >>> I am not biased towards FS or * at the moment, though FS seems to >>> have a >>> better designed configuration options and community. >>> Just wanted to share my experience, and hear some opinions. >>> Unfortunately I cannot spend whole amount of time investigating this >>> case now, capturing packets etc., but I will try to do that once I >>> have >>> time. Meanwhile I will have to stick to * for prod. >>> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From jaybinks at gmail.com Tue Feb 17 02:27:42 2009 From: jaybinks at gmail.com (jay binks) Date: Tue, 17 Feb 2009 20:27:42 +1000 Subject: [Freeswitch-users] FS SIP audio quality? In-Reply-To: <7b197bef0902170219l217522aal4fff04b5e0e51d06@mail.gmail.com> References: <49977FFD.1020002@versafon.com> <191c3a030902151643n62c2b1a0k2c341eb79c1393ea@mail.gmail.com> <4998C99E.9060706@versafon.com> <499A3006.6070208@versafon.com> <7b197bef0902170219l217522aal4fff04b5e0e51d06@mail.gmail.com> Message-ID: Back in November, Brian ( BKW ) was raising money to get new sounds recorded ... intending to have them for the 1.0.2 release.. I wonder if they made it in, or if they are still coming ... Jay On Tue, Feb 17, 2009 at 8:19 PM, Giovanni Maruzzelli wrote: > There is also another side to make mimd to: the Asterisk sounds you > hear more often (the demo ones) are very long ones. > > The ones of the FS demo are very very short (many times just one word) > and concatenated with the insertion of sleeps. > > That is probably someway altering the equation between user experiences > > > Sincerely, > > Giovanni Maruzzelli > ========================================= > www.celliax.org > via Pierlombardo 9, 20135 Milano > Italy > gmaruzz at celliax dot org > Cell : +39-347-2665618 > Fax : +39-02-87390039 > > > > > On Tue, Feb 17, 2009 at 11:05 AM, Nik Middleton > wrote: > > For what it's worth, using Asterisk recordings, I found FS to be better > > than when played on an Asterisk system. > > > > I came to the same conclusion early on that the included prompts with FS > > were of a relatively poor nature. Not volunteering to record new ones, > > but they do let the product down, as they lead to discussions as below. > > > > > > Regards, > > > > -----Original Message----- > > From: freeswitch-users-bounces at lists.freeswitch.org > > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Paul > > D. > > Sent: 17 February 2009 03:33 > > To: freeswitch-users at lists.freeswitch.org > > Subject: Re: [Freeswitch-users] FS SIP audio quality? > > > > I re-tested calls to VM replacing some of FS prompts with * ones, and it > > > > appears that * sounds were recorded with a better quality/higher volume, > > > > so FS itself has nothing to do with that. That's solved. :-) > > I am going to double check all the equipment we used for tests, like > > headphones, telephone sets, cables since I am almost convinced that > > there's nothing in FS which can produce effects I observe. > > I will post back if I find anything wrong, appreciate everybody's help > > with this. > > > > Brian West wrote: > >> I'm not able to reproduce this issue.. can you verify the codecs are > >> what you think they are on both Asterisk and FreeSWITCH. > >> > >> /b > >> > >> On Feb 15, 2009, at 8:04 PM, Paul D. wrote: > >> > >> > >>> Well, I tried several call scenarios: > >>> 1. Call from X-Lite or Linksys to VM. > >>> 2. Call from X-Lite or Linksys to a conference. > >>> 3. Call from X-Lite or Linksys to a PSTN number via Gafachi and > >>> CallWithUs. > >>> > >>> I have now * 1.6.5 and FS 1.0.3RC1 installed on the same enterprise > >>> grade Intel server. So just comparing audio in the call scenarios > >>> above > >>> * somehow does noticeably better job, sounds clearer and volume is at > >>> the right level. I am not changing any phone settings of course when > >>> switching between * and FS. > >>> I am not biased towards FS or * at the moment, though FS seems to > >>> have a > >>> better designed configuration options and community. > >>> Just wanted to share my experience, and hear some opinions. > >>> Unfortunately I cannot spend whole amount of time investigating this > >>> case now, capturing packets etc., but I will try to do that once I > >>> have > >>> time. Meanwhile I will have to stick to * for prod. > >>> > >> > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > >> > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Sincerely Jay -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090217/9c1d8430/attachment-0002.html From jason at jasonjgw.net Tue Feb 17 02:35:10 2009 From: jason at jasonjgw.net (Jason White) Date: Tue, 17 Feb 2009 21:35:10 +1100 Subject: [Freeswitch-users] FS SIP audio quality? In-Reply-To: References: <49977FFD.1020002@versafon.com> <191c3a030902151643n62c2b1a0k2c341eb79c1393ea@mail.gmail.com> <4998C99E.9060706@versafon.com> <499A3006.6070208@versafon.com> <7b197bef0902170219l217522aal4fff04b5e0e51d06@mail.gmail.com> Message-ID: <20090217103510.GA29891@jdc.jasonjgw.net> jay binks wrote: > Back in November, Brian ( BKW ) was raising money to get new sounds recorded > ... > intending to have them for the 1.0.2 release.. > > I wonder if they made it in, or if they are still coming ... Release 1.0.7 of the sound files was made available soon thereafter, which I understand includes the new material that was recorded. I might be wrong, though. From dave at 3c.co.uk Tue Feb 17 04:35:09 2009 From: dave at 3c.co.uk (David Knell) Date: Tue, 17 Feb 2009 12:35:09 +0000 Subject: [Freeswitch-users] FS SIP audio quality? In-Reply-To: <499A3006.6070208@versafon.com> References: <49977FFD.1020002@versafon.com> <191c3a030902151643n62c2b1a0k2c341eb79c1393ea@mail.gmail.com> <4998C99E.9060706@versafon.com> <499A3006.6070208@versafon.com> Message-ID: <499AAEFD.3050307@3c.co.uk> Paul D. wrote: > I re-tested calls to VM replacing some of FS prompts with * ones, and it > appears that * sounds were recorded with a better quality/higher volume, > so FS itself has nothing to do with that. That's solved. :-) > There's a long history of people in A/B listening tests reporting louder as sounding better on the same source material - even if the additional volume isn't detectable as such. Which, I guess, explains my 25 years of going to Motorhead gigs. --Dave -- David Knell, Director, 3C Limited T: 020 8114 5002 F: 020 3002 7257 M: 07773 800623 http://www.3c.co.uk From pablosaro at gmail.com Tue Feb 17 05:29:00 2009 From: pablosaro at gmail.com (Pablo Hernan Saro) Date: Tue, 17 Feb 2009 11:29:00 -0200 Subject: [Freeswitch-users] Options for configure script Message-ID: <247f8100902170529v79ed65a4p7df7241b4c11cb2d@mail.gmail.com> Hi there, Anyone knows how to configure FS in order to have log folder in /var/log/freeswitch ? I did ./configure --prefix=/opt/freeswitch and after install I got logs, pid file and cdrs under /opt/freeswitch/log Thanks in advance. Pablo From kokoska.rokoska at post.cz Tue Feb 17 05:42:31 2009 From: kokoska.rokoska at post.cz (kokoska rokoska) Date: Tue, 17 Feb 2009 14:42:31 +0100 Subject: [Freeswitch-users] FS SIP audio quality? In-Reply-To: <499AAEFD.3050307@3c.co.uk> References: <49977FFD.1020002@versafon.com> <191c3a030902151643n62c2b1a0k2c341eb79c1393ea@mail.gmail.com> <4998C99E.9060706@versafon.com> <499A3006.6070208@versafon.com> <499AAEFD.3050307@3c.co.uk> Message-ID: <499ABEC7.2040703@post.cz> David Knell napsal(a): > There's a long history of people in A/B listening tests reporting louder > as sounding > better on the same source material - even if the additional volume isn't > detectable > as such. > Yes, you are right :-) And therefor a lot of (nearly all of) European TelCo operator (TDM, not VoIP) normalize their messages to -3 dB instead of -6 dB. And one of them (yes, you guess it right, it is Telefonica O2 :-) normalize recordings to 0 dB. And it is VERY loud :-) Best regards, kokoska.rokoska From edpimentl at gmail.com Tue Feb 17 05:50:50 2009 From: edpimentl at gmail.com (EdPimentl) Date: Tue, 17 Feb 2009 08:50:50 -0500 Subject: [Freeswitch-users] Anyone running FS from a Thumb Flash USB? Message-ID: <9dc4a1670902170550r3071f65dn76112fdbebaeb6ba@mail.gmail.com> Hello FS Members, Are there any example of FS running on a Thumb Flash USB? Thanks in advance, -E -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090217/44ace346/attachment-0002.html From anthony.minessale at gmail.com Tue Feb 17 05:54:51 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 17 Feb 2009 07:54:51 -0600 Subject: [Freeswitch-users] FS SIP audio quality? In-Reply-To: <499AAEFD.3050307@3c.co.uk> References: <49977FFD.1020002@versafon.com> <191c3a030902151643n62c2b1a0k2c341eb79c1393ea@mail.gmail.com> <4998C99E.9060706@versafon.com> <499A3006.6070208@versafon.com> <499AAEFD.3050307@3c.co.uk> Message-ID: <191c3a030902170554g20dc665bq2c4933ef2ec1becb@mail.gmail.com> Maybe the sox script brian uses to downsample the files has a problem. What if you download the 48k package (original) and listen to that? On Tue, Feb 17, 2009 at 6:35 AM, David Knell wrote: > Paul D. wrote: > > I re-tested calls to VM replacing some of FS prompts with * ones, and it > > appears that * sounds were recorded with a better quality/higher volume, > > so FS itself has nothing to do with that. That's solved. :-) > > > There's a long history of people in A/B listening tests reporting louder > as sounding > better on the same source material - even if the additional volume isn't > detectable > as such. > > Which, I guess, explains my 25 years of going to Motorhead gigs. > > --Dave > > -- > David Knell, Director, 3C Limited > T: 020 8114 5002 F: 020 3002 7257 M: 07773 800623 > http://www.3c.co.uk > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090217/90747390/attachment-0002.html From krice at freeswitch.org Tue Feb 17 06:00:49 2009 From: krice at freeswitch.org (Ken Rice) Date: Tue, 17 Feb 2009 08:00:49 -0600 Subject: [Freeswitch-users] Anyone running FS from a Thumb Flash USB? In-Reply-To: <9dc4a1670902170550r3071f65dn76112fdbebaeb6ba@mail.gmail.com> Message-ID: Could be done ed... FS itself isnt that big From: EdPimentl Reply-To: Date: Tue, 17 Feb 2009 08:50:50 -0500 To: Subject: [Freeswitch-users] Anyone running FS from a Thumb Flash USB? Hello FS Members, Are there any example of FS running on a Thumb Flash USB? Thanks in advance, -E _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090217/e58da0bf/attachment-0002.html From edpimentl at gmail.com Tue Feb 17 06:10:30 2009 From: edpimentl at gmail.com (EdPimentl) Date: Tue, 17 Feb 2009 09:10:30 -0500 Subject: [Freeswitch-users] Anyone running FS from a Thumb Flash USB? In-Reply-To: References: <9dc4a1670902170550r3071f65dn76112fdbebaeb6ba@mail.gmail.com> Message-ID: <9dc4a1670902170610p2e3d94c4u3796958f52f5b187@mail.gmail.com> I would be glad to put a bounty for a FS(and Skypiax/softphone) running on Flash Thumb Drive. Project would be documented on Wiki for others to use and improve. -E -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090217/753a1f40/attachment-0002.html From freeswitch-users at lists.rupa.com Tue Feb 17 06:11:38 2009 From: freeswitch-users at lists.rupa.com (Rupa Schomaker (lists)) Date: Tue, 17 Feb 2009 08:11:38 -0600 Subject: [Freeswitch-users] Passing Caller_ID to MOD_LCR In-Reply-To: References: <499A4497.9080001@lists.rupa.com> Message-ID: <499AC59A.4050209@lists.rupa.com> On 2/17/2009 1:18 AM, Tchavdar Paskov wrote: > I also have another question. > When i insert my custom query it looks like the profile is loaded > successfully but then when i place a call or use > > lcr ########## default /which is where i defined the sql query/ and > check the console output turns out that the switch is using the > default sql guery . lcr_admin show profiles should show you the profiles loaded and the profile's settings. "default" is a reserved profile name -- I should probably prevent that from loading. > How i can make sure that the custom sql is the only one that is to be > executed. > /All required fields in the custom sql are in accordance with the > requirements - it returns the exact required names and returned field > number is also correct/. > > Thank you agai for your time > Chav Try running with debug logging turned on (f8 on the console). This will show the sql being passed to the database. == Regarding passing the callerid to the custom sql, let me see what I can come up with... From kerrada2003 at yahoo.com Tue Feb 17 06:51:52 2009 From: kerrada2003 at yahoo.com (Ali Al-Rubaie) Date: Tue, 17 Feb 2009 06:51:52 -0800 (PST) Subject: [Freeswitch-users] Realm Value In-Reply-To: Message-ID: <350067.9862.qm@web33701.mail.mud.yahoo.com> Thanks Brian, Actually we're using freeswitch ver 1.0.2. Regards, Message: 5 Date: Thu, 12 Feb 2009 15:48:00 -0600 From: Brian West Subject: Re: [Freeswitch-users] Realm value To: freeswitch-users at lists.freeswitch.org Message-ID: Content-Type: text/plain; charset=US-ASCII; format=flowed; delsp=yes What SVN rev? /b On Feb 12, 2009, at 3:41 PM, Ali Al-Rubaie wrote: > Hi, > > How can the default value of "realm" be changed? I had changed the > command: > > > > in the file internal.xml but FS still uses the server IP address as > the challenge realm. > > Thanks in advance! > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090217/af260081/attachment-0002.html From brian at freeswitch.org Tue Feb 17 07:02:54 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 17 Feb 2009 09:02:54 -0600 Subject: [Freeswitch-users] Playing a G729 file as ringback In-Reply-To: <088801c990c1$3e139b50$ba3ad1f0$@com> References: <088801c990c1$3e139b50$ba3ad1f0$@com> Message-ID: <8730408C-CD63-4CF5-B505-11CE3AB03310@freeswitch.org> Its currently not possible. /b On Feb 16, 2009, at 11:33 PM, Cesar Cepeda wrote: > Hi, > > I?m using FS with g279 on passthrough mode and I?m trying to play a > g729 file as ringback to the A-leg while bridging a call. As far as > I understand it should go something like this: > > ? originate {channel_vars}dialstring > ? set some combination of values on ?ringback?, > ?transfer_ringback?, ?instant_ringback? > ? bridge > > The bridge works correctly, but no matter what combination of values > I try for the ?xxxringback? vars I never hear the ringback on the A- > leg. > > Can you tell me what I?m missing? > > Thanks. > > Cesar Cepeda. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090217/3349148f/attachment-0002.html From mrene_lists at avgs.ca Tue Feb 17 07:05:07 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Tue, 17 Feb 2009 10:05:07 -0500 Subject: [Freeswitch-users] Options for configure script In-Reply-To: <247f8100902170529v79ed65a4p7df7241b4c11cb2d@mail.gmail.com> References: <247f8100902170529v79ed65a4p7df7241b4c11cb2d@mail.gmail.com> Message-ID: You can specify those as runtime argument. $ ./freeswitch -h these are the optional arguments you can pass to freeswitch -nf -- no forking -u [user] -- specify user to switch to -g [group] -- specify group to switch to -help -- this message -core -- dump cores -hp -- enable high priority settings -vg -- run under valgrind -nosql -- disable internal sql scoreboard -stop -- stop freeswitch -nc -- do not output to a console and background -c -- output to a console and stay in the foreground -conf [confdir] -- specify an alternate config dir -log [logdir] -- specify an alternate log dir -db [dbdir] -- specify an alternate db dir -mod [moddir] -- specify an alternate mod dir -htdocs [htdocsdir] -- specify an alternate htdocs dir -scripts [scriptsdir] -- specify an alternate scripts dir But in my opinion, you should keep all files at the same place and make a symbolic link if you really want stuf to be accessible from /var/log Mathieu -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090217/7b35685e/attachment-0002.html From pablosaro at gmail.com Tue Feb 17 08:54:01 2009 From: pablosaro at gmail.com (Pablo Hernan Saro) Date: Tue, 17 Feb 2009 14:54:01 -0200 Subject: [Freeswitch-users] Options for configure script In-Reply-To: References: <247f8100902170529v79ed65a4p7df7241b4c11cb2d@mail.gmail.com> Message-ID: <247f8100902170854w291a5ed8ja1090c8a573a5179@mail.gmail.com> Thanks Mathieu. So, it is not possible to set this at build time by passing a parameter to the configure script... IMHO, a symbolic link is not a good idea in production environments. BR Pablo On Tue, Feb 17, 2009 at 1:05 PM, Mathieu Rene wrote: > You can specify those as runtime argument. > > $ ./freeswitch -h > these are the optional arguments you can pass to freeswitch > -nf -- no forking > -u [user] -- specify user to switch to > -g [group] -- specify group to switch to > -help -- this message > -core -- dump cores > -hp -- enable high priority settings > -vg -- run under valgrind > -nosql -- disable internal sql scoreboard > -stop -- stop freeswitch > -nc -- do not output to a console and background > -c -- output to a console and stay in the foreground > -conf [confdir] -- specify an alternate config dir > -log [logdir] -- specify an alternate log dir > -db [dbdir] -- specify an alternate db dir > -mod [moddir] -- specify an alternate mod dir > -htdocs [htdocsdir] -- specify an alternate htdocs dir > -scripts [scriptsdir] -- specify an alternate scripts dir > > But in my opinion, you should keep all files at the same place and make a > symbolic link if you really want stuf to be accessible from /var/log > > > Mathieu > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From mkarp at securesilence.com Tue Feb 17 09:09:41 2009 From: mkarp at securesilence.com (Maxim Karp) Date: Tue, 17 Feb 2009 09:09:41 -0800 Subject: [Freeswitch-users] Voicemail prompts playback too quickly Message-ID: <007301c99122$86438fa0$92caaee0$@com> Hello, Our voicemail prompts playback much too quickly on FS v1.0.1. Any suggestions to slow them down? Thanks, Maxim. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090217/886edbca/attachment-0002.html From cjonesmo at gmail.com Mon Feb 16 18:24:46 2009 From: cjonesmo at gmail.com (Chris Jones) Date: Mon, 16 Feb 2009 20:24:46 -0600 Subject: [Freeswitch-users] SIPX/FS Auto attendant Message-ID: I'm having a problem that calls to the auto-attendant won't transfer. I know this has been a problem in the past but thought that it was fixed. Whenever I enter an extension (or press a key to transfer me to one), the call just hangs up. I ran Freeswitch on the console, but all I see happening is this: 2009-02-16 20:22:51 [NOTICE] mod_sofia.c:1338 sofia_receive_message() Ring-Ready sofia/papayaecs.bluewavecorp.local/8323160383 at 64.2.142.116! 2009-02-16 20:22:51 [NOTICE] mod_sofia.c:1338 sofia_receive_message() Pre-Answer sofia/papayaecs.bluewavecorp.local/8323160383 at 64.2.142.116! 2009-02-16 20:22:51 [NOTICE] mod_dptools.c:600 answer_function() Channel [sofia/papayaecs.bluewavecorp.local/8323160383 at 64.2.142.116] has been answered 2009-02-16 20:23:07 [NOTICE] sofia.c:3179 sofia_handle_sip_i_state() Hangup sofia/papayaecs.bluewavecorp.local/8323160383 at 64.2.142.116 [CS_EXECUTE] [NORMAL_UNSPECIFIED] 2009-02-16 20:23:07 [NOTICE] switch_core_session.c:960 switch_core_session_thread() Session 1 ( sofia/papayaecs.bluewavecorp.local/8323160383 at 64.2.142.116) Ended 2009-02-16 20:23:07 [NOTICE] switch_core_session.c:962 switch_core_session_thread() Close Channel sofia/papayaecs.bluewavecorp.local/8323160383 at 64.2.142.116 [CS_HANGUP] Does anyone have an idea as to how to fix this? The call is coming from a SIP Trunk on vitelity. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090216/e33b9e70/attachment-0002.html From msc at freeswitch.org Tue Feb 17 09:18:51 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 17 Feb 2009 09:18:51 -0800 Subject: [Freeswitch-users] Voicemail prompts playback too quickly In-Reply-To: <007301c99122$86438fa0$92caaee0$@com> References: <007301c99122$86438fa0$92caaee0$@com> Message-ID: <87f2f3b90902170918oc86f4dm4548a68085565ddb@mail.gmail.com> > Our voicemail prompts playback much too quickly on FS v1.0.1. Any > suggestions to slow them down? At this point the best thing for you to do is to update to the latest trunk. We are 99.99% ready to tag 1.0.3RC2 which has significant improvements over 1.0.1. -MC P.S. - you might find this page useful: http://wiki.freeswitch.org/wiki/Reporting_Bugs If you see anything missing from this page please let me know. I'm trying to make it as friendly and easy to use as possible. From brian at freeswitch.org Tue Feb 17 09:21:10 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 17 Feb 2009 11:21:10 -0600 Subject: [Freeswitch-users] SIPX/FS Auto attendant In-Reply-To: References: Message-ID: You're missing some key information to help us answer your question. First off we will need to know the SVN rev, Then you might want to press F8 and check out the debug log. Chances are it'll tell you exactly why. What concerns me is the fact that a .local domain is in there. I wonder if maybe it can't resolve this at some point... again just me guessing here. Please crank up the debug and lets see if we can see any details in that log. /b On Feb 16, 2009, at 8:24 PM, Chris Jones wrote: > I'm having a problem that calls to the auto-attendant won't > transfer. I know this has been a problem in the past but thought > that it was fixed. Whenever I enter an extension (or press a key to > transfer me to one), the call just hangs up. I ran Freeswitch on the > console, but all I see happening is this: > > 2009-02-16 20:22:51 [NOTICE] mod_sofia.c:1338 > sofia_receive_message() Ring-Ready sofia/papayaecs.bluewavecorp.local/8323160383 at 64.2.142.116 > ! > 2009-02-16 20:22:51 [NOTICE] mod_sofia.c:1338 > sofia_receive_message() Pre-Answer sofia/papayaecs.bluewavecorp.local/8323160383 at 64.2.142.116 > ! > 2009-02-16 20:22:51 [NOTICE] mod_dptools.c:600 answer_function() > Channel [sofia/papayaecs.bluewavecorp.local/8323160383 at 64.2.142.116] > has been answered > 2009-02-16 20:23:07 [NOTICE] sofia.c:3179 sofia_handle_sip_i_state() > Hangup sofia/papayaecs.bluewavecorp.local/8323160383 at 64.2.142.116 > [CS_EXECUTE] [NORMAL_UNSPECIFIED] > 2009-02-16 20:23:07 [NOTICE] switch_core_session.c:960 > switch_core_session_thread() Session 1 (sofia/papayaecs.bluewavecorp.local/8323160383 at 64.2.142.116 > ) Ended > 2009-02-16 20:23:07 [NOTICE] switch_core_session.c:962 > switch_core_session_thread() Close Channel sofia/papayaecs.bluewavecorp.local/8323160383 at 64.2.142.116 > [CS_HANGUP] > Does anyone have an idea as to how to fix this? The call is coming > from a SIP Trunk on vitelity. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090217/5f187060/attachment-0002.html From msc at freeswitch.org Tue Feb 17 09:34:39 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 17 Feb 2009 09:34:39 -0800 Subject: [Freeswitch-users] Options for configure script In-Reply-To: <247f8100902170854w291a5ed8ja1090c8a573a5179@mail.gmail.com> References: <247f8100902170529v79ed65a4p7df7241b4c11cb2d@mail.gmail.com> <247f8100902170854w291a5ed8ja1090c8a573a5179@mail.gmail.com> Message-ID: <87f2f3b90902170934w6853809anb7c06b9fc92bf32e@mail.gmail.com> On Tue, Feb 17, 2009 at 8:54 AM, Pablo Hernan Saro wrote: > Thanks Mathieu. > So, it is not possible to set this at build time by passing a > parameter to the configure script... > IMHO, a symbolic link is not a good idea in production environments. > BR > You can also specify these in the config files. You can put the logs and/or CDR files wherever you'd like. If you open up conf/autoload_configs/logfile.conf.xml you'll see that there is a line commented out - this line will let you choose the default log file path. In fact, I think you'll approve of the location already specified on that line. :) -MC From brian at freeswitch.org Tue Feb 17 09:39:58 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 17 Feb 2009 11:39:58 -0600 Subject: [Freeswitch-users] Realm Value In-Reply-To: <350067.9862.qm@web33701.mail.mud.yahoo.com> References: <350067.9862.qm@web33701.mail.mud.yahoo.com> Message-ID: <2ADB8FA0-3E87-4C99-9A6F-F09C6E0F5DB8@freeswitch.org> I'm trying to get a clear picture of what you're trying to accomplish. Why would you need/want to set a static realm? Anyway can you collect sip traces? /b On Feb 17, 2009, at 8:51 AM, Ali Al-Rubaie wrote: > Thanks Brian, > > Actually we're using freeswitch ver 1.0.2. > > > Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090217/e801928a/attachment-0002.html From pablosaro at gmail.com Tue Feb 17 09:59:09 2009 From: pablosaro at gmail.com (Pablo Hernan Saro) Date: Tue, 17 Feb 2009 15:59:09 -0200 Subject: [Freeswitch-users] Options for configure script In-Reply-To: <87f2f3b90902170934w6853809anb7c06b9fc92bf32e@mail.gmail.com> References: <247f8100902170529v79ed65a4p7df7241b4c11cb2d@mail.gmail.com> <247f8100902170854w291a5ed8ja1090c8a573a5179@mail.gmail.com> <87f2f3b90902170934w6853809anb7c06b9fc92bf32e@mail.gmail.com> Message-ID: <247f8100902170959m30c321e9md1979d328dcaf846@mail.gmail.com> Hi Michael, Thank you very much for your help. It meets my needs. I was thinking in something like: ./configure --prefix=/opt/freeswitch --localstatedir=/var/log/freeswitch But I can get same results changing the default configuration of conf/autoload_configs/logfile.conf.xml and conf/autoload_configs/xml_cdr.conf.xml. Regards, Pablo On Tue, Feb 17, 2009 at 3:34 PM, Michael Collins wrote: > On Tue, Feb 17, 2009 at 8:54 AM, Pablo Hernan Saro wrote: >> Thanks Mathieu. >> So, it is not possible to set this at build time by passing a >> parameter to the configure script... >> IMHO, a symbolic link is not a good idea in production environments. >> BR >> > > You can also specify these in the config files. You can put the logs > and/or CDR files wherever you'd like. If you open up > conf/autoload_configs/logfile.conf.xml you'll see that there is a line > commented out - this line will let you choose the default log file > path. In fact, I think you'll approve of the location already > specified on that line. :) > > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From mike at jerris.com Tue Feb 17 10:24:03 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 17 Feb 2009 13:24:03 -0500 Subject: [Freeswitch-users] Options for configure script In-Reply-To: <247f8100902170959m30c321e9md1979d328dcaf846@mail.gmail.com> References: <247f8100902170529v79ed65a4p7df7241b4c11cb2d@mail.gmail.com> <247f8100902170854w291a5ed8ja1090c8a573a5179@mail.gmail.com> <87f2f3b90902170934w6853809anb7c06b9fc92bf32e@mail.gmail.com> <247f8100902170959m30c321e9md1979d328dcaf846@mail.gmail.com> Message-ID: We don't yet support localstatedir configure option. I expect we will soon. Mike On Feb 17, 2009, at 12:59 PM, Pablo Hernan Saro wrote: > Hi Michael, > > Thank you very much for your help. It meets my needs. > I was thinking in something like: > ./configure --prefix=/opt/freeswitch --localstatedir=/var/log/ > freeswitch > But I can get same results changing the default configuration of > conf/autoload_configs/logfile.conf.xml and > conf/autoload_configs/xml_cdr.conf.xml. > Regards, > > Pablo From nik.middleton at noblesolutions.co.uk Tue Feb 17 10:23:35 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Tue, 17 Feb 2009 18:23:35 -0000 Subject: [Freeswitch-users] Big delays in playing audio files Message-ID: Having spent the last week developing a small js app, I ran some tests today. With just 5 calls going on, I'm seeing huge delays from when the call is answered to when the audio file is played. Sometimes it doesn't even play at all!! Example 3 calls and the matching playbacks 2009-02-17 15:41:04 [NOTICE] voice.js:1 console_log() ready: Start DTMF 2009-02-17 15:41:08 [NOTICE] voice.js:1 console_log() ready: Start DTMF 2009-02-17 15:41:22 [NOTICE] voice.js:1 console_log() ready: Start DTMF 2009-02-17 15:41:26 [NOTICE] voice.js:1 console_log() Playing message: message.wav 2009-02-17 15:41:26 [NOTICE] voice.js:1 console_log() Playing message: message.wav 2009-02-17 15:41:26 [NOTICE] voice.js:1 console_log() Playing message: message.wav That's 22 seconds for the first one!! Anyone any ideas as to what's going on here? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090217/b4c93f4f/attachment-0002.html From anthony.minessale at gmail.com Tue Feb 17 10:34:00 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 17 Feb 2009 12:34:00 -0600 Subject: [Freeswitch-users] Big delays in playing audio files In-Reply-To: References: Message-ID: <191c3a030902171034o4ebd68f8w690092763019ced@mail.gmail.com> we would need to see your script. On Tue, Feb 17, 2009 at 12:23 PM, Nik Middleton < nik.middleton at noblesolutions.co.uk> wrote: > Having spent the last week developing a small js app, I ran some tests > today. With just 5 calls going on, I'm seeing huge delays from when the call > is answered to when the audio file is played. Sometimes it doesn't even > play at all!! > > > > Example 3 calls and the matching playbacks > > > > 2009-02-17 15:41:04 [NOTICE] voice.js:1 console_log() ready: Start DTMF > > 2009-02-17 15:41:08 [NOTICE] voice.js:1 console_log() ready: Start DTMF > > 2009-02-17 15:41:22 [NOTICE] voice.js:1 console_log() ready: Start DTMF > > > > 2009-02-17 15:41:26 [NOTICE] voice.js:1 console_log() Playing message: > message.wav > > 2009-02-17 15:41:26 [NOTICE] voice.js:1 console_log() Playing message: > message.wav > > 2009-02-17 15:41:26 [NOTICE] voice.js:1 console_log() Playing message: > message.wav > > > > That's 22 seconds for the first one!! > > > > Anyone any ideas as to what's going on here? > > > > Regards > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090217/d5651da8/attachment-0002.html From nik.middleton at noblesolutions.co.uk Tue Feb 17 11:05:10 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Tue, 17 Feb 2009 19:05:10 -0000 Subject: [Freeswitch-users] Big delays in playing audio files In-Reply-To: <191c3a030902171034o4ebd68f8w690092763019ced@mail.gmail.com> References: <191c3a030902171034o4ebd68f8w690092763019ced@mail.gmail.com> Message-ID: if (first_session.ready()) { console_log("notice","Session state=[" + first_session.state + "] \n"); consoleLog("NOTICE", "ready: Start DTMF\n"); first_session.execute("start_dtmf"); first_session.answer( ); Disposition = "ANS"; first_session.sleep(1500); console_log("notice", "Playing message: " + recording + "\n"); first_session.streamFile(recording, on_event); if (first_session.ready()) { consoleLog("err", "ready: Waiting for input\n"); first_session.streamFile("4.wav",on_event, "dtmf"); consoleLog("err", "ready: Timeout on input\n"); first_session.execute("stop_tone_detect"); //disp_call() first_session.hangup() first_session.execute("sleep", "2000"); consoleLog("NOTICE", "EXITING\n"); exit(); } } ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: 17 February 2009 18:34 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Big delays in playing audio files we would need to see your script. On Tue, Feb 17, 2009 at 12:23 PM, Nik Middleton wrote: Having spent the last week developing a small js app, I ran some tests today. With just 5 calls going on, I'm seeing huge delays from when the call is answered to when the audio file is played. Sometimes it doesn't even play at all!! Example 3 calls and the matching playbacks 2009-02-17 15:41:04 [NOTICE] voice.js:1 console_log() ready: Start DTMF 2009-02-17 15:41:08 [NOTICE] voice.js:1 console_log() ready: Start DTMF 2009-02-17 15:41:22 [NOTICE] voice.js:1 console_log() ready: Start DTMF 2009-02-17 15:41:26 [NOTICE] voice.js:1 console_log() Playing message: message.wav 2009-02-17 15:41:26 [NOTICE] voice.js:1 console_log() Playing message: message.wav 2009-02-17 15:41:26 [NOTICE] voice.js:1 console_log() Playing message: message.wav That's 22 seconds for the first one!! Anyone any ideas as to what's going on here? Regards _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090217/3bb675ae/attachment-0002.html From msc at freeswitch.org Tue Feb 17 11:24:42 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 17 Feb 2009 11:24:42 -0800 Subject: [Freeswitch-users] Big delays in playing audio files In-Reply-To: References: <191c3a030902171034o4ebd68f8w690092763019ced@mail.gmail.com> Message-ID: <87f2f3b90902171124h5a5d4448mb872ca4aded94efa@mail.gmail.com> Is this the entire script?! -MC On Tue, Feb 17, 2009 at 11:05 AM, Nik Middleton wrote: > if (first_session.ready()) { > > console_log("notice","Session state=[" + > first_session.state + "] \n"); > > > > consoleLog("NOTICE", "ready: Start DTMF\n"); > > > > first_session.execute("start_dtmf"); > > first_session.answer( ); > > > > Disposition = "ANS"; > > > > first_session.sleep(1500); > > console_log("notice", "Playing message: " + > recording + "\n"); > > first_session.streamFile(recording, on_event); > > > > if (first_session.ready()) { > > consoleLog("err", "ready: Waiting for input\n"); > > first_session.streamFile("4.wav",on_event, "dtmf"); > > consoleLog("err", "ready: Timeout on input\n"); > > first_session.execute("stop_tone_detect"); > > > > //disp_call() > > first_session.hangup() > > first_session.execute("sleep", "2000"); > > consoleLog("NOTICE", "EXITING\n"); > > exit(); > > } > > } > > > > ________________________________ > > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony > Minessale > Sent: 17 February 2009 18:34 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Big delays in playing audio files > > > > we would need to see your script. > > On Tue, Feb 17, 2009 at 12:23 PM, Nik Middleton > wrote: > > Having spent the last week developing a small js app, I ran some tests > today. With just 5 calls going on, I'm seeing huge delays from when the call > is answered to when the audio file is played. Sometimes it doesn't even > play at all!! > > > > Example 3 calls and the matching playbacks > > > > 2009-02-17 15:41:04 [NOTICE] voice.js:1 console_log() ready: Start DTMF > > 2009-02-17 15:41:08 [NOTICE] voice.js:1 console_log() ready: Start DTMF > > 2009-02-17 15:41:22 [NOTICE] voice.js:1 console_log() ready: Start DTMF > > > > 2009-02-17 15:41:26 [NOTICE] voice.js:1 console_log() Playing message: > message.wav > > 2009-02-17 15:41:26 [NOTICE] voice.js:1 console_log() Playing message: > message.wav > > 2009-02-17 15:41:26 [NOTICE] voice.js:1 console_log() Playing message: > message.wav > > > > That's 22 seconds for the first one!! > > > > Anyone any ideas as to what's going on here? > > > > Regards > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From nik.middleton at noblesolutions.co.uk Tue Feb 17 12:11:19 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Tue, 17 Feb 2009 20:11:19 -0000 Subject: [Freeswitch-users] Big delays in playing audio files In-Reply-To: <87f2f3b90902171124h5a5d4448mb872ca4aded94efa@mail.gmail.com> References: <191c3a030902171034o4ebd68f8w690092763019ced@mail.gmail.com> <87f2f3b90902171124h5a5d4448mb872ca4aded94efa@mail.gmail.com> Message-ID: Pretty much I haven't included the on-event hooks as it never gets to the point where they're called. Only other thing is the dial it's self, attached below. However, I notice in the default dial plan, if I call extension 1001 from 1000 it takes about 2-3 seconds for the phone to ring. Is that normal? //build dial string var dial_string = "{absolute_codec_string=PCMA," + "accountcode=" + account_code + ",ignore_early_media=true" + " ,origination_caller_id_number=" + caller_id + ",originate_timeout=25}" + "sofia/gateway/" + "mygateway/" + dial_num + "' " var first_session = new Session(dial_string); // Trap for call failure if (!first_session.ready()) { consoleLog("err", "Disposition: " + first_session.cause + "\n"); if (first_session.cause == "USER_BUSY") { Disposition = "BUSY"; } else if (first_session.cause == "NO_ROUTE_DESTINATION") { Disposition = "DCN"; } else if (first_session.cause == "NO_ANSWER") { Disposition = "NA"; } disp_call() exit(); } //set the on_hangup function to be called when this session is hungup first_session.setHangupHook(on_hangup,"hup"); -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 17 February 2009 19:25 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Big delays in playing audio files Is this the entire script?! -MC On Tue, Feb 17, 2009 at 11:05 AM, Nik Middleton wrote: > if (first_session.ready()) { > > console_log("notice","Session state=[" + > first_session.state + "] \n"); > > > > consoleLog("NOTICE", "ready: Start DTMF\n"); > > > > first_session.execute("start_dtmf"); > > first_session.answer( ); > > > > Disposition = "ANS"; > > > > first_session.sleep(1500); > > console_log("notice", "Playing message: " + > recording + "\n"); > > first_session.streamFile(recording, on_event); > > > > if (first_session.ready()) { > > consoleLog("err", "ready: Waiting for input\n"); > > first_session.streamFile("4.wav",on_event, "dtmf"); > > consoleLog("err", "ready: Timeout on input\n"); > > first_session.execute("stop_tone_detect"); > > > > //disp_call() > > first_session.hangup() > > first_session.execute("sleep", "2000"); > > consoleLog("NOTICE", "EXITING\n"); > > exit(); > > } > > } > > > > ________________________________ > > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony > Minessale > Sent: 17 February 2009 18:34 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Big delays in playing audio files > > > > we would need to see your script. > > On Tue, Feb 17, 2009 at 12:23 PM, Nik Middleton > wrote: > > Having spent the last week developing a small js app, I ran some tests > today. With just 5 calls going on, I'm seeing huge delays from when the call > is answered to when the audio file is played. Sometimes it doesn't even > play at all!! > > > > Example 3 calls and the matching playbacks > > > > 2009-02-17 15:41:04 [NOTICE] voice.js:1 console_log() ready: Start DTMF > > 2009-02-17 15:41:08 [NOTICE] voice.js:1 console_log() ready: Start DTMF > > 2009-02-17 15:41:22 [NOTICE] voice.js:1 console_log() ready: Start DTMF > > > > 2009-02-17 15:41:26 [NOTICE] voice.js:1 console_log() Playing message: > message.wav > > 2009-02-17 15:41:26 [NOTICE] voice.js:1 console_log() Playing message: > message.wav > > 2009-02-17 15:41:26 [NOTICE] voice.js:1 console_log() Playing message: > message.wav > > > > That's 22 seconds for the first one!! > > > > Anyone any ideas as to what's going on here? > > > > Regards > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From kokoska.rokoska at post.cz Tue Feb 17 12:17:06 2009 From: kokoska.rokoska at post.cz (kokoska rokoska) Date: Tue, 17 Feb 2009 21:17:06 +0100 Subject: [Freeswitch-users] call goes to wrong context Message-ID: <499B1B42.7090000@post.cz> Hi all, I have just "upgraded" to current trunk (before an hour or so), configuration remain the same (served through mod_xml_curl), but something has changed and I don'nt know "where", "what" and "why" :-) What's going on: I have few sofia profiles and each of them has its own context. When call arrives, http POST is made by FreeSWITCH to get the dialplan, but Caller-Context and Hunt-Context variables are always set to "default" regardless what contex I set in sofia profile the call comes in through. May be I miss something, but really have no idea where :-) Any hint is very appreciated. Best regards, kokoska.rokoska From brian at freeswitch.org Tue Feb 17 12:19:36 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 17 Feb 2009 14:19:36 -0600 Subject: [Freeswitch-users] call goes to wrong context In-Reply-To: <499B1B42.7090000@post.cz> References: <499B1B42.7090000@post.cz> Message-ID: <4432BDD8-67A4-45B4-A013-7BB7C7E35D7F@freeswitch.org> Make sure on outbound registrations/gateways you have the context and extension params set. /b On Feb 17, 2009, at 2:17 PM, kokoska rokoska wrote: > > Hi all, > > I have just "upgraded" to current trunk (before an hour or so), > configuration remain the same (served through mod_xml_curl), but > something has changed and I don'nt know "where", "what" and "why" :-) > > What's going on: > I have few sofia profiles and each of them has its own context. When > call arrives, http POST is made by FreeSWITCH to get the dialplan, but > Caller-Context and Hunt-Context variables are always set to "default" > regardless what contex I set in sofia profile the call comes in > through. > > May be I miss something, but really have no idea where :-) > > Any hint is very appreciated. > > Best regards, > > kokoska.rokoska From kerrada2003 at yahoo.com Tue Feb 17 12:21:39 2009 From: kerrada2003 at yahoo.com (Ali Al-Rubaie) Date: Tue, 17 Feb 2009 12:21:39 -0800 (PST) Subject: [Freeswitch-users] Realm Value In-Reply-To: Message-ID: <326635.86195.qm@web33706.mail.mud.yahoo.com> I have to use a specific softphone, HelpCaster, but it can not pass the authentication stage. However it can authenticate with OpenSips server! What I had noticed is that it uses static realm with OpenSips therefore I'm trying to do the same. recv 292 bytes from udp/[209.82.10.250]:3458 at 16:35:24.758862: ?? ------------------------------------------------------------------------ ?? REGISTER sip:209.82.10.235 SIP/2.0 ?? Via: SIP/2.0/UDP 209.82.10.250:1059 ?? From: sip:1001 at 209.82.10.235 ?? To: sip:1001 at 209.82.10.235 ?? Contact: sip:1001 at 209.82.10.250:1059 ?? Call-ID: fc383368-e3e0-4c32-b87b-c3127d0cc2d8 at 192.168.10.23 ?? CSeq: 597775306 REGISTER ?? Content-Length: 0 ?? Expires: 3600 ? ?? ------------------------------------------------------------------------ send 582 bytes to udp/[209.82.10.250]:1059 at 16:35:24.763948: ?? ------------------------------------------------------------------------ ?? SIP/2.0 401 Unauthorized ?? Via: SIP/2.0/UDP 209.82.10.250:1059 ?? From: sip:1001 at 209.82.10.235 ?? To: ;tag=7yam2F01ZH3vH ?? Call-ID: fc383368-e3e0-4c32-b87b-c3127d0cc2d8 at 192.168.10.23 ?? CSeq: 597775306 REGISTER ?? User-Agent: FreeSWITCH-mod_sofia/1.0.2-hacked ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH ?? Supported: timer, precondition, path, replaces ?? WWW-Authenticate: Digest realm="209.82.10.235", nonce="40b63193-85c2-4ed9-874e-c03f81be313d", algorithm=MD5, qop="auth" ?? Content-Length: 0 ? ?? ------------------------------------------------------------------------ recv 466 bytes from udp/[209.82.10.250]:3458 at 16:35:24.772834: ?? ------------------------------------------------------------------------ ?? REGISTER sip:209.82.10.235 SIP/2.0 ?? Via: SIP/2.0/UDP 209.82.10.250:1059 ?? From: sip:1001 at 209.82.10.235 ?? To: sip:1001 at 209.82.10.235 ?? Contact: sip:1001 at 209.82.10.250:1059 ?? Call-ID: fc383368-e3e0-4c32-b87b-c3127d0cc2d8 at 192.168.10.23 ?? CSeq: 597775307 REGISTER ?? Content-Length: 0 ?? Expires: 3600 ?? Authorization: Digest username="1001",realm="209.82.10.235",nonce="40b63193-85c2-4ed9-874e-c03f81be313d",response="eebe0ea43319e82cc5f6dba5877de706",uri="sip:209.82.10.235" ? ?? ------------------------------------------------------------------------ send 458 bytes to udp/[209.82.10.250]:1059 at 16:35:24.774354: ?? ------------------------------------------------------------------------ ?? SIP/2.0 403 Forbidden ?? Via: SIP/2.0/UDP 209.82.10.250:1059 ?? From: sip:1001 at 209.82.10.235 ?? To: ;tag=873c4aH5vtSFD ?? Call-ID: fc383368-e3e0-4c32-b87b-c3127d0cc2d8 at 192.168.10.23 ?? CSeq: 597775307 REGISTER ?? User-Agent: FreeSWITCH-mod_sofia/1.0.2-hacked ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH ?? Supported: timer, precondition, path, replaces ?? Content-Length: 0 ? ?? ------------------------------------------------------------------------ Thanks! ? Message: 2 Date: Tue, 17 Feb 2009 11:39:58 -0600 From: Brian West Subject: Re: [Freeswitch-users] Realm Value To: freeswitch-users at lists.freeswitch.org Message-ID: <2ADB8FA0-3E87-4C99-9A6F-F09C6E0F5DB8 at freeswitch.org> Content-Type: text/plain; charset="us-ascii" I'm trying to get a clear picture of what you're trying to accomplish. Why would you need/want to set a static realm? Anyway can you collect sip traces? /b On Feb 17, 2009, at 8:51 AM, Ali Al-Rubaie wrote: > Thanks Brian, > > Actually we're using freeswitch ver 1.0.2. > > > Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090217/e801928a/attachment-0001.html -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090217/8a59c8fc/attachment-0002.html From msc at freeswitch.org Tue Feb 17 12:28:05 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 17 Feb 2009 12:28:05 -0800 Subject: [Freeswitch-users] Big delays in playing audio files In-Reply-To: References: <191c3a030902171034o4ebd68f8w690092763019ced@mail.gmail.com> <87f2f3b90902171124h5a5d4448mb872ca4aded94efa@mail.gmail.com> Message-ID: <87f2f3b90902171228u7ea87a57h7dab1738370c0970@mail.gmail.com> On Tue, Feb 17, 2009 at 12:11 PM, Nik Middleton wrote: > Pretty much > > I haven't included the on-event hooks as it never gets to the point > where they're called. > > Only other thing is the dial it's self, attached below. However, I > notice in the default dial plan, if I call extension 1001 from 1000 it > takes about 2-3 seconds for the phone to ring. Is that normal? By "ring" do you mean caller hears ringback tone or target phone audibly rings? I haven't noticed a long delay when dialing but I haven't been looking for one either. > > > //build dial string > var dial_string = "{absolute_codec_string=PCMA," + > "accountcode=" + account_code > + > ",ignore_early_media=true" > + > " > ,origination_caller_id_number=" + > caller_id > + > ",originate_timeout=25}" > + > "sofia/gateway/" > + > "mygateway/" > + > dial_num + "' " > > var first_session = new Session(dial_string); > > // Trap for call failure > if (!first_session.ready()) { > consoleLog("err", "Disposition: " + first_session.cause > + "\n"); > if (first_session.cause == "USER_BUSY") { > Disposition = "BUSY"; > } > else if (first_session.cause == > "NO_ROUTE_DESTINATION") { > Disposition = "DCN"; > } > > else if (first_session.cause == "NO_ANSWER") { > Disposition = "NA"; > } > > disp_call() > exit(); > } > > > //set the on_hangup function to be called when this session is > hungup > first_session.setHangupHook(on_hangup,"hup"); > > Can I just say how much I hate JavaScript for stuff like this? :) I can't test this right now but I will lab it up as soon as I can. In the meantime I would have you turn on debugging and capture the results from start to finish for a successful call. Save that for future reference. Then try to recreate the symptoms, also with debug turned on, capturing output. It will be A LOT of output, so you might want to consider rolling log files. (see the "Reporting Bugs" wiki page for hints on how to do that and more) Has anybody else out there used js for something like this, or otherwise have any input on why js seems to be acting up in this case? -MC From nik.middleton at noblesolutions.co.uk Tue Feb 17 12:30:02 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Tue, 17 Feb 2009 20:30:02 -0000 Subject: [Freeswitch-users] Big delays in playing audio files In-Reply-To: References: <191c3a030902171034o4ebd68f8w690092763019ced@mail.gmail.com><87f2f3b90902171124h5a5d4448mb872ca4aded94efa@mail.gmail.com> Message-ID: I'm starting to think it's a thread/DTMF issue. Ran 15 lines to my office number (using latest trunk) 2009-02-17 20:19:38 [CRIT] switch_core_state_machine.c:259 handle_fatality() Caught signal 11 for unmapped thread!Aborted (core dumped) Also then I had tone detect on, I'd often get this freeswitch: src/switch_ivr_async.c:1328: switch_ivr_tone_detect_session: Assertion `read_codec != ((void *)0)' failed. Hardware, HP DL360 G4. Centos 5.2, 4 GB ram. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Nik Middleton Sent: 17 February 2009 20:11 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Big delays in playing audio files Pretty much I haven't included the on-event hooks as it never gets to the point where they're called. Only other thing is the dial it's self, attached below. However, I notice in the default dial plan, if I call extension 1001 from 1000 it takes about 2-3 seconds for the phone to ring. Is that normal? //build dial string var dial_string = "{absolute_codec_string=PCMA," + "accountcode=" + account_code + ",ignore_early_media=true" + " ,origination_caller_id_number=" + caller_id + ",originate_timeout=25}" + "sofia/gateway/" + "mygateway/" + dial_num + "' " var first_session = new Session(dial_string); // Trap for call failure if (!first_session.ready()) { consoleLog("err", "Disposition: " + first_session.cause + "\n"); if (first_session.cause == "USER_BUSY") { Disposition = "BUSY"; } else if (first_session.cause == "NO_ROUTE_DESTINATION") { Disposition = "DCN"; } else if (first_session.cause == "NO_ANSWER") { Disposition = "NA"; } disp_call() exit(); } //set the on_hangup function to be called when this session is hungup first_session.setHangupHook(on_hangup,"hup"); -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 17 February 2009 19:25 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Big delays in playing audio files Is this the entire script?! -MC On Tue, Feb 17, 2009 at 11:05 AM, Nik Middleton wrote: > if (first_session.ready()) { > > console_log("notice","Session state=[" + > first_session.state + "] \n"); > > > > consoleLog("NOTICE", "ready: Start DTMF\n"); > > > > first_session.execute("start_dtmf"); > > first_session.answer( ); > > > > Disposition = "ANS"; > > > > first_session.sleep(1500); > > console_log("notice", "Playing message: " + > recording + "\n"); > > first_session.streamFile(recording, on_event); > > > > if (first_session.ready()) { > > consoleLog("err", "ready: Waiting for input\n"); > > first_session.streamFile("4.wav",on_event, "dtmf"); > > consoleLog("err", "ready: Timeout on input\n"); > > first_session.execute("stop_tone_detect"); > > > > //disp_call() > > first_session.hangup() > > first_session.execute("sleep", "2000"); > > consoleLog("NOTICE", "EXITING\n"); > > exit(); > > } > > } > > > > ________________________________ > > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony > Minessale > Sent: 17 February 2009 18:34 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Big delays in playing audio files > > > > we would need to see your script. > > On Tue, Feb 17, 2009 at 12:23 PM, Nik Middleton > wrote: > > Having spent the last week developing a small js app, I ran some tests > today. With just 5 calls going on, I'm seeing huge delays from when the call > is answered to when the audio file is played. Sometimes it doesn't even > play at all!! > > > > Example 3 calls and the matching playbacks > > > > 2009-02-17 15:41:04 [NOTICE] voice.js:1 console_log() ready: Start DTMF > > 2009-02-17 15:41:08 [NOTICE] voice.js:1 console_log() ready: Start DTMF > > 2009-02-17 15:41:22 [NOTICE] voice.js:1 console_log() ready: Start DTMF > > > > 2009-02-17 15:41:26 [NOTICE] voice.js:1 console_log() Playing message: > message.wav > > 2009-02-17 15:41:26 [NOTICE] voice.js:1 console_log() Playing message: > message.wav > > 2009-02-17 15:41:26 [NOTICE] voice.js:1 console_log() Playing message: > message.wav > > > > That's 22 seconds for the first one!! > > > > Anyone any ideas as to what's going on here? > > > > Regards > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From mark at markehle.net Tue Feb 17 12:33:21 2009 From: mark at markehle.net (Mark) Date: Tue, 17 Feb 2009 15:33:21 -0500 Subject: [Freeswitch-users] (OT) SPA-922 unlock In-Reply-To: References: <191c3a030902171034o4ebd68f8w690092763019ced@mail.gmail.com> <87f2f3b90902171124h5a5d4448mb872ca4aded94efa@mail.gmail.com> Message-ID: <20090217153321.16395204b8l8b604@markehle.net> Hello, folks - I hope that I can reach someone who knows the answer to this one: I bought 2 Linksys SPA-922 phones from a guy on ebay. The phones are locked by Webnet global Communications. From what I can tell, this company went bankrupt, and the ebay seller bought the phones from a bankruptcy auction. He does not know the admin username or password. Nowhere on the linksys site is there a solution to how to unlock these phones. Is there a way, or did I buy 2 interesting looking doorstops? Other than the password thing, they function fine. Thanks - Library Mark From msc at freeswitch.org Tue Feb 17 12:35:25 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 17 Feb 2009 12:35:25 -0800 Subject: [Freeswitch-users] Big delays in playing audio files In-Reply-To: References: <191c3a030902171034o4ebd68f8w690092763019ced@mail.gmail.com> <87f2f3b90902171124h5a5d4448mb872ca4aded94efa@mail.gmail.com> Message-ID: <87f2f3b90902171235k56ff0357vd4d0da45613f68ae@mail.gmail.com> okay, can you do the usual stuff and report a bug on jira? Not sure if it's really bug but having you collect all of the data and submit a bug report will assist us greatly. -MC On Tue, Feb 17, 2009 at 12:30 PM, Nik Middleton wrote: > I'm starting to think it's a thread/DTMF issue. Ran 15 lines to my > office number (using latest trunk) > > 2009-02-17 20:19:38 [CRIT] switch_core_state_machine.c:259 > handle_fatality() Caught signal 11 for unmapped thread!Aborted (core > dumped) > > Also then I had tone detect on, I'd often get this > > freeswitch: src/switch_ivr_async.c:1328: switch_ivr_tone_detect_session: > Assertion `read_codec != ((void *)0)' failed. > > Hardware, HP DL360 G4. Centos 5.2, 4 GB ram. > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Nik > Middleton > Sent: 17 February 2009 20:11 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Big delays in playing audio files > > Pretty much > > I haven't included the on-event hooks as it never gets to the point > where they're called. > > Only other thing is the dial it's self, attached below. However, I > notice in the default dial plan, if I call extension 1001 from 1000 it > takes about 2-3 seconds for the phone to ring. Is that normal? > > > //build dial string > var dial_string = "{absolute_codec_string=PCMA," + > "accountcode=" + account_code > + > ",ignore_early_media=true" > + > " > ,origination_caller_id_number=" + > caller_id > + > ",originate_timeout=25}" > + > "sofia/gateway/" > + > "mygateway/" > + > dial_num + "' " > > var first_session = new Session(dial_string); > > // Trap for call failure > if (!first_session.ready()) { > consoleLog("err", "Disposition: " + first_session.cause > + "\n"); > if (first_session.cause == "USER_BUSY") { > Disposition = "BUSY"; > } > else if (first_session.cause == > "NO_ROUTE_DESTINATION") { > Disposition = "DCN"; > } > > else if (first_session.cause == "NO_ANSWER") { > Disposition = "NA"; > } > > disp_call() > exit(); > } > > > //set the on_hangup function to be called when this session is > hungup > first_session.setHangupHook(on_hangup,"hup"); > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Michael Collins > Sent: 17 February 2009 19:25 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Big delays in playing audio files > > Is this the entire script?! > -MC > > On Tue, Feb 17, 2009 at 11:05 AM, Nik Middleton > wrote: >> if (first_session.ready()) { >> >> console_log("notice","Session state=[" + >> first_session.state + "] \n"); >> >> >> >> consoleLog("NOTICE", "ready: Start DTMF\n"); >> >> >> >> first_session.execute("start_dtmf"); >> >> first_session.answer( ); >> >> >> >> Disposition = "ANS"; >> >> >> >> first_session.sleep(1500); >> >> console_log("notice", "Playing message: " + >> recording + "\n"); >> >> first_session.streamFile(recording, on_event); >> >> >> >> if (first_session.ready()) { >> >> consoleLog("err", "ready: Waiting for > input\n"); >> >> first_session.streamFile("4.wav",on_event, > "dtmf"); >> >> consoleLog("err", "ready: Timeout on > input\n"); >> >> first_session.execute("stop_tone_detect"); >> >> >> >> //disp_call() >> >> first_session.hangup() >> >> first_session.execute("sleep", "2000"); >> >> consoleLog("NOTICE", "EXITING\n"); >> >> exit(); >> >> } >> >> } >> >> >> >> ________________________________ >> >> From: freeswitch-users-bounces at lists.freeswitch.org >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Anthony >> Minessale >> Sent: 17 February 2009 18:34 >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] Big delays in playing audio files >> >> >> >> we would need to see your script. >> >> On Tue, Feb 17, 2009 at 12:23 PM, Nik Middleton >> wrote: >> >> Having spent the last week developing a small js app, I ran some tests >> today. With just 5 calls going on, I'm seeing huge delays from when > the call >> is answered to when the audio file is played. Sometimes it doesn't > even >> play at all!! >> >> >> >> Example 3 calls and the matching playbacks >> >> >> >> 2009-02-17 15:41:04 [NOTICE] voice.js:1 console_log() ready: Start > DTMF >> >> 2009-02-17 15:41:08 [NOTICE] voice.js:1 console_log() ready: Start > DTMF >> >> 2009-02-17 15:41:22 [NOTICE] voice.js:1 console_log() ready: Start > DTMF >> >> >> >> 2009-02-17 15:41:26 [NOTICE] voice.js:1 console_log() Playing message: >> message.wav >> >> 2009-02-17 15:41:26 [NOTICE] voice.js:1 console_log() Playing message: >> message.wav >> >> 2009-02-17 15:41:26 [NOTICE] voice.js:1 console_log() Playing message: >> message.wav >> >> >> >> That's 22 seconds for the first one!! >> >> >> >> Anyone any ideas as to what's going on here? >> >> >> >> Regards >> >> >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:213-799-1400 >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From brian at freeswitch.org Tue Feb 17 12:36:51 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 17 Feb 2009 14:36:51 -0600 Subject: [Freeswitch-users] Realm Value In-Reply-To: <326635.86195.qm@web33706.mail.mud.yahoo.com> References: <326635.86195.qm@web33706.mail.mud.yahoo.com> Message-ID: <090382A2-AF83-4635-90CF-35749F50E0FA@freeswitch.org> Very sorry to hear you have to use Broken Software. But some good has come of this if you update to rev 12113 or great you'll be 100% OK. /b On Feb 17, 2009, at 2:21 PM, Ali Al-Rubaie wrote: > > I have to use a specific softphone, HelpCaster, but it can not pass > the authentication stage. However it can authenticate with OpenSips > server! What I had noticed is that it uses static realm with > OpenSips therefore I'm trying to do the same. From anthony.minessale at gmail.com Tue Feb 17 12:37:44 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 17 Feb 2009 14:37:44 -0600 Subject: [Freeswitch-users] Big delays in playing audio files In-Reply-To: References: <191c3a030902171034o4ebd68f8w690092763019ced@mail.gmail.com> <87f2f3b90902171124h5a5d4448mb872ca4aded94efa@mail.gmail.com> Message-ID: <191c3a030902171237w4dff4c45s4a10aa0deb0ece56@mail.gmail.com> 1) turn off crash protection. 2) you cant manipulate more that one call per script, design the script to be run from the application interface so you originate the call with the api interface and transfer the call to the script so each one has it's own copy of the script. On Tue, Feb 17, 2009 at 2:30 PM, Nik Middleton < nik.middleton at noblesolutions.co.uk> wrote: > I'm starting to think it's a thread/DTMF issue. Ran 15 lines to my > office number (using latest trunk) > > 2009-02-17 20:19:38 [CRIT] switch_core_state_machine.c:259 > handle_fatality() Caught signal 11 for unmapped thread!Aborted (core > dumped) > > Also then I had tone detect on, I'd often get this > > freeswitch: src/switch_ivr_async.c:1328: switch_ivr_tone_detect_session: > Assertion `read_codec != ((void *)0)' failed. > > Hardware, HP DL360 G4. Centos 5.2, 4 GB ram. > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Nik > Middleton > Sent: 17 February 2009 20:11 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Big delays in playing audio files > > Pretty much > > I haven't included the on-event hooks as it never gets to the point > where they're called. > > Only other thing is the dial it's self, attached below. However, I > notice in the default dial plan, if I call extension 1001 from 1000 it > takes about 2-3 seconds for the phone to ring. Is that normal? > > > //build dial string > var dial_string = "{absolute_codec_string=PCMA," + > "accountcode=" + account_code > + > ",ignore_early_media=true" > + > " > ,origination_caller_id_number=" + > caller_id > + > ",originate_timeout=25}" > + > "sofia/gateway/" > + > "mygateway/" > + > dial_num + "' " > > var first_session = new Session(dial_string); > > // Trap for call failure > if (!first_session.ready()) { > consoleLog("err", "Disposition: " + first_session.cause > + "\n"); > if (first_session.cause == "USER_BUSY") { > Disposition = "BUSY"; > } > else if (first_session.cause == > "NO_ROUTE_DESTINATION") { > Disposition = "DCN"; > } > > else if (first_session.cause == "NO_ANSWER") { > Disposition = "NA"; > } > > disp_call() > exit(); > } > > > //set the on_hangup function to be called when this session is > hungup > first_session.setHangupHook(on_hangup,"hup"); > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Michael Collins > Sent: 17 February 2009 19:25 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Big delays in playing audio files > > Is this the entire script?! > -MC > > On Tue, Feb 17, 2009 at 11:05 AM, Nik Middleton > wrote: > > if (first_session.ready()) { > > > > console_log("notice","Session state=[" + > > first_session.state + "] \n"); > > > > > > > > consoleLog("NOTICE", "ready: Start DTMF\n"); > > > > > > > > first_session.execute("start_dtmf"); > > > > first_session.answer( ); > > > > > > > > Disposition = "ANS"; > > > > > > > > first_session.sleep(1500); > > > > console_log("notice", "Playing message: " + > > recording + "\n"); > > > > first_session.streamFile(recording, on_event); > > > > > > > > if (first_session.ready()) { > > > > consoleLog("err", "ready: Waiting for > input\n"); > > > > first_session.streamFile("4.wav",on_event, > "dtmf"); > > > > consoleLog("err", "ready: Timeout on > input\n"); > > > > first_session.execute("stop_tone_detect"); > > > > > > > > //disp_call() > > > > first_session.hangup() > > > > first_session.execute("sleep", "2000"); > > > > consoleLog("NOTICE", "EXITING\n"); > > > > exit(); > > > > } > > > > } > > > > > > > > ________________________________ > > > > From: freeswitch-users-bounces at lists.freeswitch.org > > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Anthony > > Minessale > > Sent: 17 February 2009 18:34 > > To: freeswitch-users at lists.freeswitch.org > > Subject: Re: [Freeswitch-users] Big delays in playing audio files > > > > > > > > we would need to see your script. > > > > On Tue, Feb 17, 2009 at 12:23 PM, Nik Middleton > > wrote: > > > > Having spent the last week developing a small js app, I ran some tests > > today. With just 5 calls going on, I'm seeing huge delays from when > the call > > is answered to when the audio file is played. Sometimes it doesn't > even > > play at all!! > > > > > > > > Example 3 calls and the matching playbacks > > > > > > > > 2009-02-17 15:41:04 [NOTICE] voice.js:1 console_log() ready: Start > DTMF > > > > 2009-02-17 15:41:08 [NOTICE] voice.js:1 console_log() ready: Start > DTMF > > > > 2009-02-17 15:41:22 [NOTICE] voice.js:1 console_log() ready: Start > DTMF > > > > > > > > 2009-02-17 15:41:26 [NOTICE] voice.js:1 console_log() Playing message: > > message.wav > > > > 2009-02-17 15:41:26 [NOTICE] voice.js:1 console_log() Playing message: > > message.wav > > > > 2009-02-17 15:41:26 [NOTICE] voice.js:1 console_log() Playing message: > > message.wav > > > > > > > > That's 22 seconds for the first one!! > > > > > > > > Anyone any ideas as to what's going on here? > > > > > > > > Regards > > > > > > > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090217/fc8b7947/attachment-0002.html From nik.middleton at noblesolutions.co.uk Tue Feb 17 12:38:00 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Tue, 17 Feb 2009 20:38:00 -0000 Subject: [Freeswitch-users] Big delays in playing audio files In-Reply-To: <87f2f3b90902171228u7ea87a57h7dab1738370c0970@mail.gmail.com> References: <191c3a030902171034o4ebd68f8w690092763019ced@mail.gmail.com><87f2f3b90902171124h5a5d4448mb872ca4aded94efa@mail.gmail.com> <87f2f3b90902171228u7ea87a57h7dab1738370c0970@mail.gmail.com> Message-ID: I'm talking of the time when I hit the dial button to the phone 3 ft away starting to ring Regards, -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 17 February 2009 20:28 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Big delays in playing audio files On Tue, Feb 17, 2009 at 12:11 PM, Nik Middleton wrote: > Pretty much > > I haven't included the on-event hooks as it never gets to the point > where they're called. > > Only other thing is the dial it's self, attached below. However, I > notice in the default dial plan, if I call extension 1001 from 1000 it > takes about 2-3 seconds for the phone to ring. Is that normal? By "ring" do you mean caller hears ringback tone or target phone audibly rings? I haven't noticed a long delay when dialing but I haven't been looking for one either. > > > //build dial string > var dial_string = "{absolute_codec_string=PCMA," + > "accountcode=" + account_code > + > ",ignore_early_media=true" > + > " > ,origination_caller_id_number=" + > caller_id > + > ",originate_timeout=25}" > + > "sofia/gateway/" > + > "mygateway/" > + > dial_num + "' " > > var first_session = new Session(dial_string); > > // Trap for call failure > if (!first_session.ready()) { > consoleLog("err", "Disposition: " + first_session.cause > + "\n"); > if (first_session.cause == "USER_BUSY") { > Disposition = "BUSY"; > } > else if (first_session.cause == > "NO_ROUTE_DESTINATION") { > Disposition = "DCN"; > } > > else if (first_session.cause == "NO_ANSWER") { > Disposition = "NA"; > } > > disp_call() > exit(); > } > > > //set the on_hangup function to be called when this session is > hungup > first_session.setHangupHook(on_hangup,"hup"); > > Can I just say how much I hate JavaScript for stuff like this? :) I can't test this right now but I will lab it up as soon as I can. In the meantime I would have you turn on debugging and capture the results from start to finish for a successful call. Save that for future reference. Then try to recreate the symptoms, also with debug turned on, capturing output. It will be A LOT of output, so you might want to consider rolling log files. (see the "Reporting Bugs" wiki page for hints on how to do that and more) Has anybody else out there used js for something like this, or otherwise have any input on why js seems to be acting up in this case? -MC _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From brian at freeswitch.org Tue Feb 17 12:40:14 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 17 Feb 2009 14:40:14 -0600 Subject: [Freeswitch-users] (OT) SPA-922 unlock In-Reply-To: <20090217153321.16395204b8l8b604@markehle.net> References: <191c3a030902171034o4ebd68f8w690092763019ced@mail.gmail.com> <87f2f3b90902171124h5a5d4448mb872ca4aded94efa@mail.gmail.com> <20090217153321.16395204b8l8b604@markehle.net> Message-ID: <0A1321E0-8EDA-434C-BB47-009E0B8A0BBD@freeswitch.org> Mark, Sorry to say but I think you're pretty much SOL. I would check voipsupply or the like for a replacement. On a side note you hijacked the "Big delays in playing audio files" thread by clicking reply on one of those messages then changing the subject and body... in the future please try not to do this as it can cause your request to be overlooked depending on how the reader threads the messages. Maybe someone else has more input on your SPA's... Good Luck, /b On Feb 17, 2009, at 2:33 PM, Mark wrote: > Hello, folks - I hope that I can reach someone who knows the answer to > this one: > > I bought 2 Linksys SPA-922 phones from a guy on ebay. The phones are > locked by Webnet global Communications. From what I can tell, this > company went bankrupt, and the ebay seller bought the phones from a > bankruptcy auction. He does not know the admin username or password. > Nowhere on the linksys site is there a solution to how to unlock these > phones. > > Is there a way, or did I buy 2 interesting looking doorstops? Other > than the password thing, they function fine. > > Thanks - > > Library Mark From gkuri at ieee.org Tue Feb 17 12:41:42 2009 From: gkuri at ieee.org (Gabriel Kuri) Date: Tue, 17 Feb 2009 12:41:42 -0800 Subject: [Freeswitch-users] (OT) SPA-922 unlock In-Reply-To: <20090217153321.16395204b8l8b604@markehle.net> References: <191c3a030902171034o4ebd68f8w690092763019ced@mail.gmail.com> <87f2f3b90902171124h5a5d4448mb872ca4aded94efa@mail.gmail.com> <20090217153321.16395204b8l8b604@markehle.net> Message-ID: <499B2106.20102@ieee.org> Have you tried resetting the phone via the built-in IVR menu? Pick up the handset and dial ****73738# This should reset the phone to factory defaults, assuming that company didn't lock this feature out. Gabe Mark wrote: > Hello, folks - I hope that I can reach someone who knows the answer to > this one: > > I bought 2 Linksys SPA-922 phones from a guy on ebay. The phones are > locked by Webnet global Communications. From what I can tell, this > company went bankrupt, and the ebay seller bought the phones from a > bankruptcy auction. He does not know the admin username or password. > Nowhere on the linksys site is there a solution to how to unlock these > phones. > > Is there a way, or did I buy 2 interesting looking doorstops? Other > than the password thing, they function fine. > > Thanks - > > Library Mark > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mark at markehle.net Tue Feb 17 12:47:56 2009 From: mark at markehle.net (Mark) Date: Tue, 17 Feb 2009 15:47:56 -0500 Subject: [Freeswitch-users] (OT) SPA-922 unlock In-Reply-To: <499B2106.20102@ieee.org> References: <191c3a030902171034o4ebd68f8w690092763019ced@mail.gmail.com> <87f2f3b90902171124h5a5d4448mb872ca4aded94efa@mail.gmail.com> <20090217153321.16395204b8l8b604@markehle.net> <499B2106.20102@ieee.org> Message-ID: <20090217154756.744658d4l2acjnok@markehle.net> Sadly, ****73738# does not work. Is there a jumper on the board or some other hardware fix for this? Quoting "Gabriel Kuri" : > Have you tried resetting the phone via the built-in IVR menu? > > Pick up the handset and dial ****73738# > > This should reset the phone to factory defaults, assuming that company > didn't lock this feature out. > > Gabe > > > > Mark wrote: >> Hello, folks - I hope that I can reach someone who knows the answer to >> this one: >> >> I bought 2 Linksys SPA-922 phones from a guy on ebay. The phones are >> locked by Webnet global Communications. From what I can tell, this >> company went bankrupt, and the ebay seller bought the phones from a >> bankruptcy auction. He does not know the admin username or password. >> Nowhere on the linksys site is there a solution to how to unlock these >> phones. >> >> Is there a way, or did I buy 2 interesting looking doorstops? Other >> than the password thing, they function fine. >> >> Thanks - >> >> Library Mark >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From nik.middleton at noblesolutions.co.uk Tue Feb 17 12:48:51 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Tue, 17 Feb 2009 20:48:51 -0000 Subject: [Freeswitch-users] Big delays in playing audio files In-Reply-To: <191c3a030902171237w4dff4c45s4a10aa0deb0ece56@mail.gmail.com> References: <191c3a030902171034o4ebd68f8w690092763019ced@mail.gmail.com><87f2f3b90902171124h5a5d4448mb872ca4aded94efa@mail.gmail.com> <191c3a030902171237w4dff4c45s4a10aa0deb0ece56@mail.gmail.com> Message-ID: 1, OK, 2. Right now I have a php script calling bgapi via and event socket with the call parameters. Is that what you mean? If not, can you give me a pointer? I had assumed that every time I called bgapi it with the script in it, it would get it's own copy. Regards, ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: 17 February 2009 20:38 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Big delays in playing audio files 1) turn off crash protection. 2) you cant manipulate more that one call per script, design the script to be run from the application interface so you originate the call with the api interface and transfer the call to the script so each one has it's own copy of the script. On Tue, Feb 17, 2009 at 2:30 PM, Nik Middleton wrote: I'm starting to think it's a thread/DTMF issue. Ran 15 lines to my office number (using latest trunk) 2009-02-17 20:19:38 [CRIT] switch_core_state_machine.c:259 handle_fatality() Caught signal 11 for unmapped thread!Aborted (core dumped) Also then I had tone detect on, I'd often get this freeswitch: src/switch_ivr_async.c:1328: switch_ivr_tone_detect_session: Assertion `read_codec != ((void *)0)' failed. Hardware, HP DL360 G4. Centos 5.2, 4 GB ram. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Nik Middleton Sent: 17 February 2009 20:11 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Big delays in playing audio files Pretty much I haven't included the on-event hooks as it never gets to the point where they're called. Only other thing is the dial it's self, attached below. However, I notice in the default dial plan, if I call extension 1001 from 1000 it takes about 2-3 seconds for the phone to ring. Is that normal? //build dial string var dial_string = "{absolute_codec_string=PCMA," + "accountcode=" + account_code + ",ignore_early_media=true" + " ,origination_caller_id_number=" + caller_id + ",originate_timeout=25}" + "sofia/gateway/" + "mygateway/" + dial_num + "' " var first_session = new Session(dial_string); // Trap for call failure if (!first_session.ready()) { consoleLog("err", "Disposition: " + first_session.cause + "\n"); if (first_session.cause == "USER_BUSY") { Disposition = "BUSY"; } else if (first_session.cause == "NO_ROUTE_DESTINATION") { Disposition = "DCN"; } else if (first_session.cause == "NO_ANSWER") { Disposition = "NA"; } disp_call() exit(); } //set the on_hangup function to be called when this session is hungup first_session.setHangupHook(on_hangup,"hup"); -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 17 February 2009 19:25 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Big delays in playing audio files Is this the entire script?! -MC On Tue, Feb 17, 2009 at 11:05 AM, Nik Middleton wrote: > if (first_session.ready()) { > > console_log("notice","Session state=[" + > first_session.state + "] \n"); > > > > consoleLog("NOTICE", "ready: Start DTMF\n"); > > > > first_session.execute("start_dtmf"); > > first_session.answer( ); > > > > Disposition = "ANS"; > > > > first_session.sleep(1500); > > console_log("notice", "Playing message: " + > recording + "\n"); > > first_session.streamFile(recording, on_event); > > > > if (first_session.ready()) { > > consoleLog("err", "ready: Waiting for input\n"); > > first_session.streamFile("4.wav",on_event, "dtmf"); > > consoleLog("err", "ready: Timeout on input\n"); > > first_session.execute("stop_tone_detect"); > > > > //disp_call() > > first_session.hangup() > > first_session.execute("sleep", "2000"); > > consoleLog("NOTICE", "EXITING\n"); > > exit(); > > } > > } > > > > ________________________________ > > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony > Minessale > Sent: 17 February 2009 18:34 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Big delays in playing audio files > > > > we would need to see your script. > > On Tue, Feb 17, 2009 at 12:23 PM, Nik Middleton > wrote: > > Having spent the last week developing a small js app, I ran some tests > today. With just 5 calls going on, I'm seeing huge delays from when the call > is answered to when the audio file is played. Sometimes it doesn't even > play at all!! > > > > Example 3 calls and the matching playbacks > > > > 2009-02-17 15:41:04 [NOTICE] voice.js:1 console_log() ready: Start DTMF > > 2009-02-17 15:41:08 [NOTICE] voice.js:1 console_log() ready: Start DTMF > > 2009-02-17 15:41:22 [NOTICE] voice.js:1 console_log() ready: Start DTMF > > > > 2009-02-17 15:41:26 [NOTICE] voice.js:1 console_log() Playing message: > message.wav > > 2009-02-17 15:41:26 [NOTICE] voice.js:1 console_log() Playing message: > message.wav > > 2009-02-17 15:41:26 [NOTICE] voice.js:1 console_log() Playing message: > message.wav > > > > That's 22 seconds for the first one!! > > > > Anyone any ideas as to what's going on here? > > > > Regards > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090217/622f060e/attachment-0002.html From msc at freeswitch.org Tue Feb 17 12:56:41 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 17 Feb 2009 12:56:41 -0800 Subject: [Freeswitch-users] Big delays in playing audio files In-Reply-To: References: <191c3a030902171034o4ebd68f8w690092763019ced@mail.gmail.com> <87f2f3b90902171124h5a5d4448mb872ca4aded94efa@mail.gmail.com> <191c3a030902171237w4dff4c45s4a10aa0deb0ece56@mail.gmail.com> Message-ID: <87f2f3b90902171256m3d4da7efo9189794d29a87aa3@mail.gmail.com> On Tue, Feb 17, 2009 at 12:48 PM, Nik Middleton wrote: > 1, OK, > > > > 2. Right now I have a php script calling bgapi via and event socket with the > call parameters. Is that what you mean? If not, can you give me a pointer? > I had assumed that every time I called bgapi it with the script in it, it > would get it's own copy. yes, bgapi counts as an API call. Ken Rice thinks this might be related to a spidermonky concurrency issue... -MC From gkuri at ieee.org Tue Feb 17 12:58:08 2009 From: gkuri at ieee.org (Gabriel Kuri) Date: Tue, 17 Feb 2009 12:58:08 -0800 Subject: [Freeswitch-users] (OT) SPA-922 unlock In-Reply-To: <20090217154756.744658d4l2acjnok@markehle.net> References: <191c3a030902171034o4ebd68f8w690092763019ced@mail.gmail.com> <87f2f3b90902171124h5a5d4448mb872ca4aded94efa@mail.gmail.com> <20090217153321.16395204b8l8b604@markehle.net> <499B2106.20102@ieee.org> <20090217154756.744658d4l2acjnok@markehle.net> Message-ID: <499B24E0.4080204@ieee.org> I believe you need to make sure the Ethernet cable is unplugged from the phone when trying to dial that string. Now I've never tried this, but it should theoretically be possible ... Sniff the traffic of the phone and see where it's attempting to pickup the config file. Then setup a local network with your own DNS server, and re-direct the phone (via DNS) to your own web server (assuming it's picking up the config via http) and have a config file on the web server with a username and password you specify to reset the config and get into the phone. Let's hope they didn't setup the phone to provision via https, otherwise you're really SOL If you need help generating a config for the phone, with Linksys' special config tool, contact me offlist. Gabe Mark wrote: > Sadly, ****73738# does not work. > > Is there a jumper on the board or some other hardware fix for this? > > Quoting "Gabriel Kuri" : > >> Have you tried resetting the phone via the built-in IVR menu? >> >> Pick up the handset and dial ****73738# >> >> This should reset the phone to factory defaults, assuming that company >> didn't lock this feature out. >> >> Gabe >> >> >> >> Mark wrote: >>> Hello, folks - I hope that I can reach someone who knows the answer to >>> this one: >>> >>> I bought 2 Linksys SPA-922 phones from a guy on ebay. The phones are >>> locked by Webnet global Communications. From what I can tell, this >>> company went bankrupt, and the ebay seller bought the phones from a >>> bankruptcy auction. He does not know the admin username or password. >>> Nowhere on the linksys site is there a solution to how to unlock these >>> phones. >>> >>> Is there a way, or did I buy 2 interesting looking doorstops? Other >>> than the password thing, they function fine. >>> >>> Thanks - >>> >>> Library Mark >>> >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Tue Feb 17 13:00:20 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 17 Feb 2009 15:00:20 -0600 Subject: [Freeswitch-users] (OT) SPA-922 unlock In-Reply-To: <499B24E0.4080204@ieee.org> References: <191c3a030902171034o4ebd68f8w690092763019ced@mail.gmail.com> <87f2f3b90902171124h5a5d4448mb872ca4aded94efa@mail.gmail.com> <20090217153321.16395204b8l8b604@markehle.net> <499B2106.20102@ieee.org> <20090217154756.744658d4l2acjnok@markehle.net> <499B24E0.4080204@ieee.org> Message-ID: <8D48CE5C-DEF9-4B49-9B2F-BE189C9F95EA@freeswitch.org> Don't they cryptographically sign the config also? /b On Feb 17, 2009, at 2:58 PM, Gabriel Kuri wrote: > If you need help generating a config for the phone, with Linksys' > special config tool, contact me offlist. > > Gabe From kokoska.rokoska at post.cz Tue Feb 17 13:05:21 2009 From: kokoska.rokoska at post.cz (kokoska.rokoska) Date: Tue, 17 Feb 2009 22:05:21 +0100 Subject: [Freeswitch-users] call goes to wrong context In-Reply-To: <4432BDD8-67A4-45B4-A013-7BB7C7E35D7F@freeswitch.org> References: <499B1B42.7090000@post.cz> <4432BDD8-67A4-45B4-A013-7BB7C7E35D7F@freeswitch.org> Message-ID: <499B2691.70003@post.cz> Brian West napsal(a): > Make sure on outbound registrations/gateways you have the context and > extension params set. > Thank you very much, Brian, for your suggestion! I had context defined on all sofia profiles, but I didn't have extension param set on gateways (but it works till I upgraded to current FS). So I add the extension param to all gateways, but I doesn't help. mod_xml_curl still asks for the "default" context instead of the context defined in sofia profile... If you have more hints, I be very happy :-) Best regards, kokoska.rokoska From brian at freeswitch.org Tue Feb 17 13:09:18 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 17 Feb 2009 15:09:18 -0600 Subject: [Freeswitch-users] call goes to wrong context In-Reply-To: <499B2691.70003@post.cz> References: <499B1B42.7090000@post.cz> <4432BDD8-67A4-45B4-A013-7BB7C7E35D7F@freeswitch.org> <499B2691.70003@post.cz> Message-ID: No problem. Just join us on IRC.. things move faster on there. /b On Feb 17, 2009, at 3:05 PM, kokoska.rokoska wrote: > If you have more hints, I be very happy :-) From mark at markehle.net Tue Feb 17 13:10:22 2009 From: mark at markehle.net (Mark) Date: Tue, 17 Feb 2009 16:10:22 -0500 Subject: [Freeswitch-users] (OT) SPA-922 unlock In-Reply-To: <499B24E0.4080204@ieee.org> References: <191c3a030902171034o4ebd68f8w690092763019ced@mail.gmail.com> <87f2f3b90902171124h5a5d4448mb872ca4aded94efa@mail.gmail.com> <20090217153321.16395204b8l8b604@markehle.net> <499B2106.20102@ieee.org> <20090217154756.744658d4l2acjnok@markehle.net> <499B24E0.4080204@ieee.org> Message-ID: <20090217161022.13903dpm9s716484@markehle.net> I did unplug the ethernet cable. I have never been able to make the IVR work on any of the Linksys phones that I have. I must be doing something wrong. I will try to sniff the traffic on the phone when I start it up. I will report back when I do. Thanks so much - Library Mark Quoting "Gabriel Kuri" : > I believe you need to make sure the Ethernet cable is unplugged from the > phone when trying to dial that string. > > Now I've never tried this, but it should theoretically be possible ... > > Sniff the traffic of the phone and see where it's attempting to pickup > the config file. Then setup a local network with your own DNS server, > and re-direct the phone (via DNS) to your own web server (assuming it's > picking up the config via http) and have a config file on the web server > with a username and password you specify to reset the config and get > into the phone. Let's hope they didn't setup the phone to provision via > https, otherwise you're really SOL > > If you need help generating a config for the phone, with Linksys' > special config tool, contact me offlist. > > Gabe > > Mark wrote: >> Sadly, ****73738# does not work. >> >> Is there a jumper on the board or some other hardware fix for this? >> >> Quoting "Gabriel Kuri" : >> >>> Have you tried resetting the phone via the built-in IVR menu? >>> >>> Pick up the handset and dial ****73738# >>> >>> This should reset the phone to factory defaults, assuming that company >>> didn't lock this feature out. >>> >>> Gabe >>> >>> >>> >>> Mark wrote: >>>> Hello, folks - I hope that I can reach someone who knows the answer to >>>> this one: >>>> >>>> I bought 2 Linksys SPA-922 phones from a guy on ebay. The phones are >>>> locked by Webnet global Communications. From what I can tell, this >>>> company went bankrupt, and the ebay seller bought the phones from a >>>> bankruptcy auction. He does not know the admin username or password. >>>> Nowhere on the linksys site is there a solution to how to unlock these >>>> phones. >>>> >>>> Is there a way, or did I buy 2 interesting looking doorstops? Other >>>> than the password thing, they function fine. >>>> >>>> Thanks - >>>> >>>> Library Mark >>>> >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From ivan at myrvold.org Tue Feb 17 13:19:58 2009 From: ivan at myrvold.org (Ivan C Myrvold) Date: Tue, 17 Feb 2009 22:19:58 +0100 Subject: [Freeswitch-users] Skypiax on OS X Message-ID: s it possible to run Skypiax on OS X? The wiki says Linux and Windows, but says nothing about OS X. I have been running FreeSWITCH on OS X for a couple of years now, and love it. Adding Skype gateway would be really sweet. Are there any plans for adding Skypiax to trunk, or do we have to build it only from svn branch? Ivan From nik.middleton at noblesolutions.co.uk Tue Feb 17 13:24:23 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Tue, 17 Feb 2009 21:24:23 -0000 Subject: [Freeswitch-users] Big delays in playing audio files In-Reply-To: <87f2f3b90902171256m3d4da7efo9189794d29a87aa3@mail.gmail.com> References: <191c3a030902171034o4ebd68f8w690092763019ced@mail.gmail.com><87f2f3b90902171124h5a5d4448mb872ca4aded94efa@mail.gmail.com><191c3a030902171237w4dff4c45s4a10aa0deb0ece56@mail.gmail.com> <87f2f3b90902171256m3d4da7efo9189794d29a87aa3@mail.gmail.com> Message-ID: > yes, bgapi counts as an API call. Ken Rice thinks this might be > related to a spidermonky concurrency issue... Well that kinda fits, as I see the audio files stacking up, seems like they're being queued. Question is, what's my alternative, lua? -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 17 February 2009 20:57 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Big delays in playing audio files On Tue, Feb 17, 2009 at 12:48 PM, Nik Middleton wrote: > 1, OK, > > > > 2. Right now I have a php script calling bgapi via and event socket with the > call parameters. Is that what you mean? If not, can you give me a pointer? > I had assumed that every time I called bgapi it with the script in it, it > would get it's own copy. From lfurrea at gmail.com Tue Feb 17 13:26:35 2009 From: lfurrea at gmail.com (Luis F Urrea) Date: Tue, 17 Feb 2009 15:26:35 -0600 Subject: [Freeswitch-users] xml_cdr Call Flow for attended transfer In-Reply-To: References: Message-ID: any hints? what would be the best way to report CDRs for attended transfers?? We are using C with libxml to create a binary that can be called from a script to rotate xml_cdrs and insert them on SQLite and would gladly submit the code to your revision, advice and maybe even potential use. I appreciate your advice. On Mon, Feb 16, 2009 at 5:52 PM, Luis F Urrea wrote: > Hi all, > > > I am trying to understand xml_cdr for an attended (consultative) transfer, > I was thinking that the A-leg that initially > originated the call would remain untouched but I see that it's global tags > get replaced. > > I have a test call that goes as follows: > > 201 originates a call and talks to 203 -----> A-leg(1) and > B-leg(1) > > 203 puts 201 on hold and calls 202 (attended) ------> A-leg(2) and > B-leg(2) > > 203 transfers the call > > 201 and 202 are talking ------> > A-leg(1) w/ B-leg(2) ??? > > Here are the relevant captures: > > A-leg(1) > http://pastebin.freeswitch.org/7253 > > B-leg(1) > http://pastebin.freeswitch.org/7254 > > A-leg(2) > http://pastebin.freeswitch.org/7252 > > B-leg(2) > http://pastebin.freeswitch.org/7255 > > I was expecting A-leg(1) to have corresponding to 201 which is > the original A-leg but it seems that on the transfer, it reverts and 202 > appears as the A-leg and 201 as the B-leg. > > Can someone shed some light on how that transfer gets logged in terms of > A-leg and B-leg? > > TIA > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090217/940eb515/attachment-0002.html From intralanman at freeswitch.org Tue Feb 17 13:27:04 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Tue, 17 Feb 2009 16:27:04 -0500 Subject: [Freeswitch-users] (OT) SPA-922 unlock In-Reply-To: <8D48CE5C-DEF9-4B49-9B2F-BE189C9F95EA@freeswitch.org> References: <191c3a030902171034o4ebd68f8w690092763019ced@mail.gmail.com> <87f2f3b90902171124h5a5d4448mb872ca4aded94efa@mail.gmail.com> <20090217153321.16395204b8l8b604@markehle.net> <499B2106.20102@ieee.org> <20090217154756.744658d4l2acjnok@markehle.net> <499B24E0.4080204@ieee.org> <8D48CE5C-DEF9-4B49-9B2F-BE189C9F95EA@freeswitch.org> Message-ID: <499B2BA8.1070509@freeswitch.org> Brian West wrote: > Don't they cryptographically sign the config also? > > it's an option in the device... some providers do, some don't. but it shouldn't matter too much if they're using https or not, as long as the ata doesn't authenticate via certificate or something. -Ray From msc at freeswitch.org Tue Feb 17 13:30:10 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 17 Feb 2009 13:30:10 -0800 Subject: [Freeswitch-users] Big delays in playing audio files In-Reply-To: References: <191c3a030902171034o4ebd68f8w690092763019ced@mail.gmail.com> <87f2f3b90902171124h5a5d4448mb872ca4aded94efa@mail.gmail.com> <191c3a030902171237w4dff4c45s4a10aa0deb0ece56@mail.gmail.com> <87f2f3b90902171256m3d4da7efo9189794d29a87aa3@mail.gmail.com> Message-ID: <87f2f3b90902171330v181cd5dej79c1e7b95fb599a5@mail.gmail.com> >> yes, bgapi counts as an API call. Ken Rice thinks this might be >> related to a spidermonky concurrency issue... > > Well that kinda fits, as I see the audio files stacking up, seems like > they're being queued. > > Question is, what's my alternative, lua? > Lua or C/C++, but Lua is the consensus for quick and easy. If you join IRC you can ask user "hmmhesays" about his experiences. He's got like 400+ concurrent calls with Lua scripts. -MC From brian at freeswitch.org Tue Feb 17 13:31:26 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 17 Feb 2009 15:31:26 -0600 Subject: [Freeswitch-users] Skypiax on OS X In-Reply-To: References: Message-ID: I think Ctrix is working on mod_airpe in his branch for OS X. /b On Feb 17, 2009, at 3:19 PM, Ivan C Myrvold wrote: > s it possible to run Skypiax on OS X? The wiki says Linux and Windows, > but says nothing about OS X. > I have been running FreeSWITCH on OS X for a couple of years now, and > love it. Adding Skype gateway would be really sweet. > > Are there any plans for adding Skypiax to trunk, or do we have to > build it only from svn branch? > > Ivan From gkuri at ieee.org Tue Feb 17 13:45:36 2009 From: gkuri at ieee.org (Gabriel Kuri) Date: Tue, 17 Feb 2009 13:45:36 -0800 Subject: [Freeswitch-users] (OT) SPA-922 unlock In-Reply-To: <8D48CE5C-DEF9-4B49-9B2F-BE189C9F95EA@freeswitch.org> References: <191c3a030902171034o4ebd68f8w690092763019ced@mail.gmail.com> <87f2f3b90902171124h5a5d4448mb872ca4aded94efa@mail.gmail.com> <20090217153321.16395204b8l8b604@markehle.net> <499B2106.20102@ieee.org> <20090217154756.744658d4l2acjnok@markehle.net> <499B24E0.4080204@ieee.org> <8D48CE5C-DEF9-4B49-9B2F-BE189C9F95EA@freeswitch.org> Message-ID: <499B3000.10306@ieee.org> It depends on the whether you pass the option to the Linksys/Cisco Profile Compiler to generate the config file. In any case, that shouldn't be an issue. Gabe Brian West wrote: > Don't they cryptographically sign the config also? > > /b > > On Feb 17, 2009, at 2:58 PM, Gabriel Kuri wrote: > >> If you need help generating a config for the phone, with Linksys' >> special config tool, contact me offlist. >> >> Gabe > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From gkuri at ieee.org Tue Feb 17 13:51:09 2009 From: gkuri at ieee.org (Gabriel Kuri) Date: Tue, 17 Feb 2009 13:51:09 -0800 Subject: [Freeswitch-users] (OT) SPA-922 unlock In-Reply-To: <20090217161022.13903dpm9s716484@markehle.net> References: <191c3a030902171034o4ebd68f8w690092763019ced@mail.gmail.com> <87f2f3b90902171124h5a5d4448mb872ca4aded94efa@mail.gmail.com> <20090217153321.16395204b8l8b604@markehle.net> <499B2106.20102@ieee.org> <20090217154756.744658d4l2acjnok@markehle.net> <499B24E0.4080204@ieee.org> <20090217161022.13903dpm9s716484@markehle.net> Message-ID: <499B314D.2030302@ieee.org> On the slight chance they're not doing remote provisioning and the phone is just simply locked with a username/password, you'll need to feed the phone a TFTP server via DHCP Option 66 and setup a config file on that tftp server with the name spa922.cfg. Contact me off list about generating a config file for the phone. Gabe Mark wrote: > I did unplug the ethernet cable. I have never been able to make the > IVR work on any of the Linksys phones that I have. I must be doing > something wrong. > > I will try to sniff the traffic on the phone when I start it up. I > will report back when I do. > > Thanks so much - > > Library Mark > > Quoting "Gabriel Kuri" : > >> I believe you need to make sure the Ethernet cable is unplugged from the >> phone when trying to dial that string. >> >> Now I've never tried this, but it should theoretically be possible ... >> >> Sniff the traffic of the phone and see where it's attempting to pickup >> the config file. Then setup a local network with your own DNS server, >> and re-direct the phone (via DNS) to your own web server (assuming it's >> picking up the config via http) and have a config file on the web server >> with a username and password you specify to reset the config and get >> into the phone. Let's hope they didn't setup the phone to provision via >> https, otherwise you're really SOL >> >> If you need help generating a config for the phone, with Linksys' >> special config tool, contact me offlist. >> >> Gabe >> >> Mark wrote: >>> Sadly, ****73738# does not work. >>> >>> Is there a jumper on the board or some other hardware fix for this? >>> >>> Quoting "Gabriel Kuri" : >>> >>>> Have you tried resetting the phone via the built-in IVR menu? >>>> >>>> Pick up the handset and dial ****73738# >>>> >>>> This should reset the phone to factory defaults, assuming that company >>>> didn't lock this feature out. >>>> >>>> Gabe >>>> >>>> >>>> >>>> Mark wrote: >>>>> Hello, folks - I hope that I can reach someone who knows the answer to >>>>> this one: >>>>> >>>>> I bought 2 Linksys SPA-922 phones from a guy on ebay. The phones are >>>>> locked by Webnet global Communications. From what I can tell, this >>>>> company went bankrupt, and the ebay seller bought the phones from a >>>>> bankruptcy auction. He does not know the admin username or password. >>>>> Nowhere on the linksys site is there a solution to how to unlock these >>>>> phones. >>>>> >>>>> Is there a way, or did I buy 2 interesting looking doorstops? Other >>>>> than the password thing, they function fine. >>>>> >>>>> Thanks - >>>>> >>>>> Library Mark >>>>> >>>>> >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From gkuri at ieee.org Tue Feb 17 13:55:25 2009 From: gkuri at ieee.org (Gabriel Kuri) Date: Tue, 17 Feb 2009 13:55:25 -0800 Subject: [Freeswitch-users] (OT) SPA-922 unlock In-Reply-To: <499B2BA8.1070509@freeswitch.org> References: <191c3a030902171034o4ebd68f8w690092763019ced@mail.gmail.com> <87f2f3b90902171124h5a5d4448mb872ca4aded94efa@mail.gmail.com> <20090217153321.16395204b8l8b604@markehle.net> <499B2106.20102@ieee.org> <20090217154756.744658d4l2acjnok@markehle.net> <499B24E0.4080204@ieee.org> <8D48CE5C-DEF9-4B49-9B2F-BE189C9F95EA@freeswitch.org> <499B2BA8.1070509@freeswitch.org> Message-ID: <499B324D.8010603@ieee.org> I'm about 99% positive that if https is enabled for remote provisioning, the web server needs an SSL certificate signed by the Linksys Enterprise CA, otherwise the phone will reject it. Gabe > but it shouldn't matter too much if they're using https or not, as long > as the ata doesn't authenticate via certificate or something. From freeswitch-users at digitaldan.com Tue Feb 17 14:40:06 2009 From: freeswitch-users at digitaldan.com (Dan) Date: Tue, 17 Feb 2009 15:40:06 -0700 (MST) Subject: [Freeswitch-users] Debian rules Message-ID: <1496906.31234910406670.JavaMail.root@zimbra> Hi guys, I noticed that the debian build is missing lines for shout.conf.xml and does not install mod_flite (if its built) . This can be fixed by adding the following lines to debian/freeswitch.install opt/freeswitch/mod/mod_flite* opt/freeswitch/conf/autoload_configs/shout.conf.xml and the following lines to debian/freeswitch.conffiles /opt/freeswitch/conf/autoload_configs/shout.conf.xml thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090217/60597ae6/attachment-0002.html From brian at freeswitch.org Tue Feb 17 14:51:37 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 17 Feb 2009 16:51:37 -0600 Subject: [Freeswitch-users] Debian rules In-Reply-To: <1496906.31234910406670.JavaMail.root@zimbra> References: <1496906.31234910406670.JavaMail.root@zimbra> Message-ID: Please submit all patches and changes via jira if possible http://jira.freeswitch.org Thanks, Brian On Feb 17, 2009, at 4:40 PM, Dan wrote: > Hi guys, > I noticed that the debian build is missing lines for shout.conf.xml > and does not install mod_flite (if its built) . This can be fixed > by adding the following lines to debian/freeswitch.install > > opt/freeswitch/mod/mod_flite* > opt/freeswitch/conf/autoload_configs/shout.conf.xml > > and the following lines to debian/freeswitch.conffiles > > /opt/freeswitch/conf/autoload_configs/shout.conf.xml > > thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090217/844b6886/attachment-0002.html From msc at freeswitch.org Tue Feb 17 14:57:54 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 17 Feb 2009 14:57:54 -0800 Subject: [Freeswitch-users] Debian rules In-Reply-To: <1496906.31234910406670.JavaMail.root@zimbra> References: <1496906.31234910406670.JavaMail.root@zimbra> Message-ID: <87f2f3b90902171457h37c4790crf8fc6ea78d06f0d3@mail.gmail.com> Oh, and thanks for the info! -MC From nik.middleton at noblesolutions.co.uk Tue Feb 17 15:11:48 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Tue, 17 Feb 2009 23:11:48 -0000 Subject: [Freeswitch-users] AddBody to events in lua Message-ID: Hi Guys, I'm having real problems doing something trivial, and there doesn't seem to be any docs on this issue In js I do this //Disposition = disp; //Create Custom event custom_msg = "call_disposition: " + Disposition + "\n" + "called_number: " + dial_num + "\n" ; e = new Event("custom", "dialer::dialer-result"); e.addBody(custom_msg); e.fire(); And it works In lua I try this --Disposition = disp; --Create Custom event custom_msg = "call_disposition: " .. Disposition .. "\n" .. "called_number: " .. dial_num .."\n" ; local e = freeswitch.Event("custom", "dialer::dialer-result"); e.addBody(custom_msg); e:fire(e); This doesn't work, I get an error : Error in addBody expected 2..2 args, got 1 What are the arguments? It seems to be looking for a pointer for the first one, but there's nothing on the wiki on this. Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090217/223bd566/attachment-0002.html From freeswitch-users at digitaldan.com Tue Feb 17 15:17:52 2009 From: freeswitch-users at digitaldan.com (Dan) Date: Tue, 17 Feb 2009 16:17:52 -0700 (MST) Subject: [Freeswitch-users] Debian rules In-Reply-To: <87f2f3b90902171457h37c4790crf8fc6ea78d06f0d3@mail.gmail.com> Message-ID: <1141554.61234912672229.JavaMail.root@zimbra> no problem, its now in jira, key fsbuild-124. D- ----- Original Message ----- From: "Michael Collins" To: freeswitch-users at lists.freeswitch.org Sent: Tuesday, February 17, 2009 3:57:54 PM GMT -07:00 US/Canada Mountain Subject: Re: [Freeswitch-users] Debian rules Oh, and thanks for the info! -MC _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090217/06fdfeb2/attachment-0002.html From msc at freeswitch.org Tue Feb 17 15:23:13 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 17 Feb 2009 15:23:13 -0800 Subject: [Freeswitch-users] AddBody to events in lua In-Reply-To: References: Message-ID: <87f2f3b90902171523o5e014ce9w4fae9f05f378b861@mail.gmail.com> > local e = freeswitch.Event("custom", > "dialer::dialer-result"); > > e.addBody(custom_msg); > > e:fire(e); The wiki page (http://wiki.freeswitch.org/wiki/Lua#event:fire) shows that you fire thusly: e:fire(); --No "e" in the parens. Can you try it and report back? -MC From anthony.minessale at gmail.com Tue Feb 17 15:25:11 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 17 Feb 2009 17:25:11 -0600 Subject: [Freeswitch-users] AddBody to events in lua In-Reply-To: References: Message-ID: <191c3a030902171525y17166787ud37ab5243c287008@mail.gmail.com> in lua you call methods with a colon : e:addBody(blah); calling with a . implies you are going to supply the obj too e.addBody(e, blah); On Tue, Feb 17, 2009 at 5:11 PM, Nik Middleton < nik.middleton at noblesolutions.co.uk> wrote: > Hi Guys, > > > > I'm having real problems doing something trivial, and there doesn't seem to > be any docs on this issue > > > > In js I do this > > > > //Disposition = disp; > > //Create Custom event > > > > custom_msg = > > "call_disposition: " + Disposition + > "\n" + > > "called_number: " + dial_num + "\n" > ; > > > > e = new Event("custom", > "dialer::dialer-result"); > > e.addBody(custom_msg); > > e.fire(); > > > > And it works > > > > In lua I try this > > > > --Disposition = disp; > > --Create Custom event > > > > custom_msg = "call_disposition: " .. Disposition .. "\n" .. > > "called_number: " .. dial_num > .."\n" ; > > > > local e = freeswitch.Event("custom", > "dialer::dialer-result"); > > e.addBody(custom_msg); > > e:fire(e); > > > > This doesn't work, I get an error : Error in addBody expected 2..2 args, > got 1 > > > > What are the arguments? It seems to be looking for a pointer for the first > one, but there's nothing on the wiki on this. > > > > > > Regards, > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090217/6e79ecb9/attachment-0002.html From msc at freeswitch.org Tue Feb 17 15:29:48 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 17 Feb 2009 15:29:48 -0800 Subject: [Freeswitch-users] AddBody to events in lua In-Reply-To: <191c3a030902171525y17166787ud37ab5243c287008@mail.gmail.com> References: <191c3a030902171525y17166787ud37ab5243c287008@mail.gmail.com> Message-ID: <87f2f3b90902171529w1d5c32e5ydc6e6479d03ecab6@mail.gmail.com> On Tue, Feb 17, 2009 at 3:25 PM, Anthony Minessale wrote: > in lua you call methods with a colon : > > e:addBody(blah); > > calling with a . implies you are going to supply the obj too > > e.addBody(e, blah); > Also, there is an explicit example here: http://wiki.freeswitch.org/wiki/Lua#event:addBody It looks exactly like what you're trying to do. -MC From nik.middleton at noblesolutions.co.uk Tue Feb 17 15:30:48 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Tue, 17 Feb 2009 23:30:48 -0000 Subject: [Freeswitch-users] AddBody to events in lua In-Reply-To: <87f2f3b90902171523o5e014ce9w4fae9f05f378b861@mail.gmail.com> References: <87f2f3b90902171523o5e014ce9w4fae9f05f378b861@mail.gmail.com> Message-ID: I've got it working now thanks I've also added a working example to the Wiki (lua/addBody) which was empty Regards, -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 17 February 2009 23:23 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] AddBody to events in lua > local e = freeswitch.Event("custom", > "dialer::dialer-result"); > > e.addBody(custom_msg); > > e:fire(e); The wiki page (http://wiki.freeswitch.org/wiki/Lua#event:fire) shows that you fire thusly: e:fire(); --No "e" in the parens. Can you try it and report back? -MC _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From nik.middleton at noblesolutions.co.uk Tue Feb 17 15:33:46 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Tue, 17 Feb 2009 23:33:46 -0000 Subject: [Freeswitch-users] AddBody to events in lua In-Reply-To: <87f2f3b90902171529w1d5c32e5ydc6e6479d03ecab6@mail.gmail.com> References: <191c3a030902171525y17166787ud37ab5243c287008@mail.gmail.com> <87f2f3b90902171529w1d5c32e5ydc6e6479d03ecab6@mail.gmail.com> Message-ID: Err, that's what I just posted :) Regards, -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 17 February 2009 23:30 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] AddBody to events in lua On Tue, Feb 17, 2009 at 3:25 PM, Anthony Minessale wrote: > in lua you call methods with a colon : > > e:addBody(blah); > > calling with a . implies you are going to supply the obj too > > e.addBody(e, blah); > Also, there is an explicit example here: http://wiki.freeswitch.org/wiki/Lua#event:addBody It looks exactly like what you're trying to do. -MC _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From brian at freeswitch.org Tue Feb 17 15:35:54 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 17 Feb 2009 17:35:54 -0600 Subject: [Freeswitch-users] AddBody to events in lua In-Reply-To: References: <191c3a030902171525y17166787ud37ab5243c287008@mail.gmail.com> <87f2f3b90902171529w1d5c32e5ydc6e6479d03ecab6@mail.gmail.com> Message-ID: Good... keep up the good work adding more docs. ;) /b On Feb 17, 2009, at 5:33 PM, Nik Middleton wrote: > Err, that's what I just posted :) > > Regards, From msc at freeswitch.org Tue Feb 17 15:39:49 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 17 Feb 2009 15:39:49 -0800 Subject: [Freeswitch-users] AddBody to events in lua In-Reply-To: References: <191c3a030902171525y17166787ud37ab5243c287008@mail.gmail.com> <87f2f3b90902171529w1d5c32e5ydc6e6479d03ecab6@mail.gmail.com> Message-ID: <87f2f3b90902171539u29954bf7q2777a96c581a5b88@mail.gmail.com> On Tue, Feb 17, 2009 at 3:33 PM, Nik Middleton wrote: > Err, that's what I just posted :) > oops, hehe, that would explain why I thought my browser cache was messing with me. Nice work. Please definitely add to the Lua wiki page as you gain experience. Hopefully your pain will be other Lua users' gain. ;) -MC From kristian.kielhofner at gmail.com Tue Feb 17 15:43:31 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Tue, 17 Feb 2009 18:43:31 -0500 Subject: [Freeswitch-users] Anyone running FS from a Thumb Flash USB? In-Reply-To: <9dc4a1670902170550r3071f65dn76112fdbebaeb6ba@mail.gmail.com> References: <9dc4a1670902170550r3071f65dn76112fdbebaeb6ba@mail.gmail.com> Message-ID: <2d9149cd0902171543t1d1a5c9dqb23210ab8a2f3eeb@mail.gmail.com> FreeSWITCH now compiles in AsLinux: http://www.astlinux.org AstLinux with the new bootloader Runnix (or you could just use syslinux) boots from flash. It also boots from PXE, ISO, disk, etc. Pretty much anything :). FreeSWITCH (compiled against uClibc) is about 3.3MB and as Tony pointed out, mod_sofia is 1.2MB of that. The sample configs and sounds are much larger. Luckily the sounds compress well with something like squashfs. Put it this way: - Default AstLinux install (quite a bit of stuff these days) - FreeSWITCH (default mods + mod_xml_curl, -spidermonkey, but w/ lua, snom, vmd, and others) - Sample configs (pretty big too but also compress well) - 8k Sounds (HUGE, but compress well) - Native sounds (G723, G729, GSM, PCMU, PCMA all 8K obviously) Results in a squashfs disk image of about 41MB. You could run off a 64MB flash drive and have plenty left over for your union filesystem (configs, etc). :) Be aware that if you are going to run from such a config we recommend the default (which is to run AstLinux from RAM). Otherwise you take quite a hit reading audio from a squashfs filesystem. If you want an ISO to boot on a generic machine (VMware, virtualbox, etc work too) let me know. On Tue, Feb 17, 2009 at 8:50 AM, EdPimentl wrote: > Hello FS Members, > > Are there any example of FS running on a Thumb Flash USB? > Thanks in advance, > -E > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From brian at freeswitch.org Tue Feb 17 15:48:22 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 17 Feb 2009 17:48:22 -0600 Subject: [Freeswitch-users] Anyone running FS from a Thumb Flash USB? In-Reply-To: <2d9149cd0902171543t1d1a5c9dqb23210ab8a2f3eeb@mail.gmail.com> References: <9dc4a1670902170550r3071f65dn76112fdbebaeb6ba@mail.gmail.com> <2d9149cd0902171543t1d1a5c9dqb23210ab8a2f3eeb@mail.gmail.com> Message-ID: Great news!!! Good Job! /b On Feb 17, 2009, at 5:43 PM, Kristian Kielhofner wrote: > FreeSWITCH now compiles in AsLinux: > > http://www.astlinux.org > > AstLinux with the new bootloader Runnix (or you could just use > syslinux) boots from flash. It also boots from PXE, ISO, disk, etc. > Pretty much anything :). > > FreeSWITCH (compiled against uClibc) is about 3.3MB and as Tony > pointed out, mod_sofia is 1.2MB of that. The sample configs and > sounds are much larger. Luckily the sounds compress well with > something like squashfs. > > Put it this way: > > - Default AstLinux install (quite a bit of stuff these days) > - FreeSWITCH (default mods + mod_xml_curl, -spidermonkey, but w/ lua, > snom, vmd, and others) > - Sample configs (pretty big too but also compress well) > - 8k Sounds (HUGE, but compress well) > - Native sounds (G723, G729, GSM, PCMU, PCMA all 8K obviously) > > Results in a squashfs disk image of about 41MB. You could run off a > 64MB flash drive and have plenty left over for your union filesystem > (configs, etc). :) > > Be aware that if you are going to run from such a config we > recommend the default (which is to run AstLinux from RAM). Otherwise > you take quite a hit reading audio from a squashfs filesystem. > > If you want an ISO to boot on a generic machine (VMware, virtualbox, > etc work too) let me know. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090217/5bc53bff/attachment-0002.html From jaybinks at gmail.com Tue Feb 17 15:58:17 2009 From: jaybinks at gmail.com (jay binks) Date: Wed, 18 Feb 2009 09:58:17 +1000 Subject: [Freeswitch-users] Anyone running FS from a Thumb Flash USB? In-Reply-To: <2d9149cd0902171543t1d1a5c9dqb23210ab8a2f3eeb@mail.gmail.com> References: <9dc4a1670902170550r3071f65dn76112fdbebaeb6ba@mail.gmail.com> <2d9149cd0902171543t1d1a5c9dqb23210ab8a2f3eeb@mail.gmail.com> Message-ID: wow... this is awesome ! good job mate. On Wed, Feb 18, 2009 at 9:43 AM, Kristian Kielhofner < kristian.kielhofner at gmail.com> wrote: > FreeSWITCH now compiles in AsLinux: > > http://www.astlinux.org > > AstLinux with the new bootloader Runnix (or you could just use > syslinux) boots from flash. It also boots from PXE, ISO, disk, etc. > Pretty much anything :). > > FreeSWITCH (compiled against uClibc) is about 3.3MB and as Tony > pointed out, mod_sofia is 1.2MB of that. The sample configs and > sounds are much larger. Luckily the sounds compress well with > something like squashfs. > > Put it this way: > > - Default AstLinux install (quite a bit of stuff these days) > - FreeSWITCH (default mods + mod_xml_curl, -spidermonkey, but w/ lua, > snom, vmd, and others) > - Sample configs (pretty big too but also compress well) > - 8k Sounds (HUGE, but compress well) > - Native sounds (G723, G729, GSM, PCMU, PCMA all 8K obviously) > > Results in a squashfs disk image of about 41MB. You could run off a > 64MB flash drive and have plenty left over for your union filesystem > (configs, etc). :) > > Be aware that if you are going to run from such a config we > recommend the default (which is to run AstLinux from RAM). Otherwise > you take quite a hit reading audio from a squashfs filesystem. > > If you want an ISO to boot on a generic machine (VMware, virtualbox, > etc work too) let me know. > > On Tue, Feb 17, 2009 at 8:50 AM, EdPimentl wrote: > > Hello FS Members, > > > > Are there any example of FS running on a Thumb Flash USB? > > Thanks in advance, > > -E > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > -- > Kristian Kielhofner > http://blog.krisk.org > http://www.submityoursip.com > http://www.astlinux.org > http://www.star2star.com > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Sincerely Jay -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090218/8c9e4a6d/attachment-0002.html From nik.middleton at noblesolutions.co.uk Tue Feb 17 16:07:05 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Wed, 18 Feb 2009 00:07:05 -0000 Subject: [Freeswitch-users] AddBody to events in lua In-Reply-To: References: <191c3a030902171525y17166787ud37ab5243c287008@mail.gmail.com><87f2f3b90902171529w1d5c32e5ydc6e6479d03ecab6@mail.gmail.com> Message-ID: I'll shortly post some docs on the php fs_sock. There's also a couple of bugs in it that I've fixed. I ran 10,000 events, which completed in around 20 seconds, all received and processed flawlessly. A new one on me was arrayshift. To think that I messed around in C for ages with circular buffers, this is so simple. Regards, -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: 17 February 2009 23:36 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] AddBody to events in lua Good... keep up the good work adding more docs. ;) /b On Feb 17, 2009, at 5:33 PM, Nik Middleton wrote: > Err, that's what I just posted :) > > Regards, _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From msc at freeswitch.org Tue Feb 17 16:10:17 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 17 Feb 2009 16:10:17 -0800 Subject: [Freeswitch-users] Anyone running FS from a Thumb Flash USB? In-Reply-To: <2d9149cd0902171543t1d1a5c9dqb23210ab8a2f3eeb@mail.gmail.com> References: <9dc4a1670902170550r3071f65dn76112fdbebaeb6ba@mail.gmail.com> <2d9149cd0902171543t1d1a5c9dqb23210ab8a2f3eeb@mail.gmail.com> Message-ID: <87f2f3b90902171610r37679769h2b4f1c033e7b3e94@mail.gmail.com> On Tue, Feb 17, 2009 at 3:43 PM, Kristian Kielhofner wrote: > FreeSWITCH now compiles in AsLinux: Nice work! I'll go tell our friends over in the Yahoo financial forums - I'm sure they're dying to hear about it! ;) -MC From msc at freeswitch.org Tue Feb 17 16:12:28 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 17 Feb 2009 16:12:28 -0800 Subject: [Freeswitch-users] AddBody to events in lua In-Reply-To: References: <191c3a030902171525y17166787ud37ab5243c287008@mail.gmail.com> <87f2f3b90902171529w1d5c32e5ydc6e6479d03ecab6@mail.gmail.com> Message-ID: <87f2f3b90902171612p11dfd5e1u92287e76e264e10d@mail.gmail.com> > I ran 10,000 events, which completed in around 20 seconds, all received > and processed flawlessly. A new one on me was arrayshift. To think that > I messed around in C for ages with circular buffers, this is so simple. Excellent! You're officially deputized to add any Lua examples you create. We can use examples on the wiki and sample scripts in the contrib directory. Nice work. -MC From brian at freeswitch.org Tue Feb 17 16:14:39 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 17 Feb 2009 18:14:39 -0600 Subject: [Freeswitch-users] AddBody to events in lua In-Reply-To: References: <191c3a030902171525y17166787ud37ab5243c287008@mail.gmail.com><87f2f3b90902171529w1d5c32e5ydc6e6479d03ecab6@mail.gmail.com> Message-ID: <677AB3FE-E0B9-46A8-8808-74FE1A6D5C3D@freeswitch.org> And you ran this in lua? /b On Feb 17, 2009, at 6:07 PM, Nik Middleton wrote: > > I ran 10,000 events, which completed in around 20 seconds, all > received > and processed flawlessly. A new one on me was arrayshift. To think > that > I messed around in C for ages with circular buffers, this is so > simple. From gkuri at ieee.org Tue Feb 17 16:16:05 2009 From: gkuri at ieee.org (Gabriel Kuri) Date: Tue, 17 Feb 2009 16:16:05 -0800 Subject: [Freeswitch-users] Anyone running FS from a Thumb Flash USB? In-Reply-To: <2d9149cd0902171543t1d1a5c9dqb23210ab8a2f3eeb@mail.gmail.com> References: <9dc4a1670902170550r3071f65dn76112fdbebaeb6ba@mail.gmail.com> <2d9149cd0902171543t1d1a5c9dqb23210ab8a2f3eeb@mail.gmail.com> Message-ID: <499B5345.1010701@ieee.org> awesome work! on a slightly related [embedded] note, do you know if any work has been done to port FS to any of the Analog Blackfin MCUs? I'd be interested in hearing if anyone has had any such luck. Gabe Kristian Kielhofner wrote: > FreeSWITCH now compiles in AsLinux: > > http://www.astlinux.org > > AstLinux with the new bootloader Runnix (or you could just use > syslinux) boots from flash. It also boots from PXE, ISO, disk, etc. > Pretty much anything :). > > FreeSWITCH (compiled against uClibc) is about 3.3MB and as Tony > pointed out, mod_sofia is 1.2MB of that. The sample configs and > sounds are much larger. Luckily the sounds compress well with > something like squashfs. > > Put it this way: > > - Default AstLinux install (quite a bit of stuff these days) > - FreeSWITCH (default mods + mod_xml_curl, -spidermonkey, but w/ lua, > snom, vmd, and others) > - Sample configs (pretty big too but also compress well) > - 8k Sounds (HUGE, but compress well) > - Native sounds (G723, G729, GSM, PCMU, PCMA all 8K obviously) > > Results in a squashfs disk image of about 41MB. You could run off a > 64MB flash drive and have plenty left over for your union filesystem > (configs, etc). :) > > Be aware that if you are going to run from such a config we > recommend the default (which is to run AstLinux from RAM). Otherwise > you take quite a hit reading audio from a squashfs filesystem. > > If you want an ISO to boot on a generic machine (VMware, virtualbox, > etc work too) let me know. > > On Tue, Feb 17, 2009 at 8:50 AM, EdPimentl wrote: >> Hello FS Members, >> >> Are there any example of FS running on a Thumb Flash USB? >> Thanks in advance, >> -E >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > From kristian.kielhofner at gmail.com Tue Feb 17 16:18:35 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Tue, 17 Feb 2009 19:18:35 -0500 Subject: [Freeswitch-users] Anyone running FS from a Thumb Flash USB? In-Reply-To: <87f2f3b90902171610r37679769h2b4f1c033e7b3e94@mail.gmail.com> References: <9dc4a1670902170550r3071f65dn76112fdbebaeb6ba@mail.gmail.com> <2d9149cd0902171543t1d1a5c9dqb23210ab8a2f3eeb@mail.gmail.com> <87f2f3b90902171610r37679769h2b4f1c033e7b3e94@mail.gmail.com> Message-ID: <2d9149cd0902171618s47640fb2vff3cc139b74c8f64@mail.gmail.com> Ah yes, the "pinnacle of online discussion"! ;) On Tue, Feb 17, 2009 at 7:10 PM, Michael Collins wrote: > On Tue, Feb 17, 2009 at 3:43 PM, Kristian Kielhofner > wrote: >> FreeSWITCH now compiles in AsLinux: > > Nice work! I'll go tell our friends over in the Yahoo financial forums > - I'm sure they're dying to hear about it! ;) > -MC > -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From kristian.kielhofner at gmail.com Tue Feb 17 16:20:01 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Tue, 17 Feb 2009 19:20:01 -0500 Subject: [Freeswitch-users] Anyone running FS from a Thumb Flash USB? In-Reply-To: <499B5345.1010701@ieee.org> References: <9dc4a1670902170550r3071f65dn76112fdbebaeb6ba@mail.gmail.com> <2d9149cd0902171543t1d1a5c9dqb23210ab8a2f3eeb@mail.gmail.com> <499B5345.1010701@ieee.org> Message-ID: <2d9149cd0902171620m28f82962sbdc087c0adbca70e@mail.gmail.com> I don't think so but something tells me that FreeSWITCH won't do too well without an MMU and the external libs and modules could cause quite a problem. Not that it is impossible but the uh, performance, would be interesting... Can anyone call me out on this assumption? On Tue, Feb 17, 2009 at 7:16 PM, Gabriel Kuri wrote: > awesome work! on a slightly related [embedded] note, do you know if any > work has been done to port FS to any of the Analog Blackfin MCUs? I'd be > interested in hearing if anyone has had any such luck. > > Gabe > -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From raul at etellicom.com Tue Feb 17 16:20:55 2009 From: raul at etellicom.com (Raul Fragoso) Date: Tue, 17 Feb 2009 21:20:55 -0300 Subject: [Freeswitch-users] Deployment information and use cases Message-ID: <1234916455.16581.49.camel@raul-laptop> Hello FreeSWITCHERS, My company is currently creating a suite of applications which uses FreeSWITCH as the back-end for an IP-PBX solution. We currently have a prospect to have our first customer installation - a governmental department. That is a tender to have an IP-PBX installation to connect their four office branches, each one with about 300 users - which I am sure FreeSWITCH is able to handle. Since this is an official tender, it's part of their protocol to ask about real sites using the product. Having said that, would you mind sharing some information about your experience with FreeSWITCH deployments ? No need to give many details, but a short summary with company name (if possible), when it was deployed, server equipment, number of users, number of concurrent calls, what kind of functions and services are used and overall capacity of the system. I would really appreciate if you can share that information. And if you guys agree (and explicitly manifest your agreement), I can compile the information in the FreeSWITCH wiki under a "Use Cases" page so it can serve as a common reference as well. Kind regards, Raul Fragoso From nik.middleton at noblesolutions.co.uk Tue Feb 17 16:21:09 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Wed, 18 Feb 2009 00:21:09 -0000 Subject: [Freeswitch-users] AddBody to events in lua In-Reply-To: <677AB3FE-E0B9-46A8-8808-74FE1A6D5C3D@freeswitch.org> References: <191c3a030902171525y17166787ud37ab5243c287008@mail.gmail.com><87f2f3b90902171529w1d5c32e5ydc6e6479d03ecab6@mail.gmail.com> <677AB3FE-E0B9-46A8-8808-74FE1A6D5C3D@freeswitch.org> Message-ID: No, js, I was trying to break the fs_sock.php, though I found the time was dependant on how much I echoed to the screen. I expect lua to be even faster Regards, -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: 18 February 2009 00:15 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] AddBody to events in lua And you ran this in lua? /b On Feb 17, 2009, at 6:07 PM, Nik Middleton wrote: > > I ran 10,000 events, which completed in around 20 seconds, all > received > and processed flawlessly. A new one on me was arrayshift. To think > that > I messed around in C for ages with circular buffers, this is so > simple. _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From nik.middleton at noblesolutions.co.uk Tue Feb 17 16:25:26 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Wed, 18 Feb 2009 00:25:26 -0000 Subject: [Freeswitch-users] Anyone running FS from a Thumb Flash USB? In-Reply-To: <2d9149cd0902171618s47640fb2vff3cc139b74c8f64@mail.gmail.com> References: <9dc4a1670902170550r3071f65dn76112fdbebaeb6ba@mail.gmail.com><2d9149cd0902171543t1d1a5c9dqb23210ab8a2f3eeb@mail.gmail.com><87f2f3b90902171610r37679769h2b4f1c033e7b3e94@mail.gmail.com> <2d9149cd0902171618s47640fb2vff3cc139b74c8f64@mail.gmail.com> Message-ID: Kristian, You're my hero, if I hadn't come across astlinux 3 years ago, I wouldn't be doing this stuff right now. Not too sure if that's a good thing though ;) -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Kristian Kielhofner Sent: 18 February 2009 00:19 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Anyone running FS from a Thumb Flash USB? Ah yes, the "pinnacle of online discussion"! ;) On Tue, Feb 17, 2009 at 7:10 PM, Michael Collins wrote: > On Tue, Feb 17, 2009 at 3:43 PM, Kristian Kielhofner > wrote: >> FreeSWITCH now compiles in AsLinux: > > Nice work! I'll go tell our friends over in the Yahoo financial forums > - I'm sure they're dying to hear about it! ;) > -MC > -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From fax at virgintechnologies.com Tue Feb 17 15:51:30 2009 From: fax at virgintechnologies.com (fax at virgintechnologies.com) Date: Tue, 17 Feb 2009 23:51:30 +0000 Subject: [Freeswitch-users] Sending media streams to a media gateway Message-ID: I have Freeswitch running successfully with a fairly basic config. Nat traversal is working well on both the client and server side. I want to start running all RTP streams through a media gateway, and use Freeswitch for SIP registrations and signalling only. I believe that I need to have Freeswitch invite the SIP phone to send the RTP stream directly to the media gateway when a call starts. Where can I start with this? Does anyone have any example configs? Justin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090217/3d53695c/attachment-0002.html From anthony.minessale at gmail.com Tue Feb 17 16:34:54 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 17 Feb 2009 18:34:54 -0600 Subject: [Freeswitch-users] Sending media streams to a media gateway In-Reply-To: References: Message-ID: <191c3a030902171634t4dc2e15etfa02d118e8fe3322@mail.gmail.com> you could set the variable bypass_media to true before you call bridge that will negotiate a point to point media connection between the caller and callee On Tue, Feb 17, 2009 at 5:51 PM, wrote: > I have Freeswitch running successfully with a fairly basic config. Nat > traversal is working well on both the client and server side. I want to > start running all RTP streams through a media gateway, and use Freeswitch > for SIP registrations and signalling only. > I believe that I need to have Freeswitch invite the SIP phone to send the > RTP stream directly to the media gateway when a call starts. Where can I > start with this? Does anyone have any example configs? > Justin > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090217/3688e75e/attachment-0002.html From gkuri at ieee.org Tue Feb 17 16:44:10 2009 From: gkuri at ieee.org (Gabriel Kuri) Date: Tue, 17 Feb 2009 16:44:10 -0800 Subject: [Freeswitch-users] Anyone running FS from a Thumb Flash USB? In-Reply-To: <2d9149cd0902171620m28f82962sbdc087c0adbca70e@mail.gmail.com> References: <9dc4a1670902170550r3071f65dn76112fdbebaeb6ba@mail.gmail.com> <2d9149cd0902171543t1d1a5c9dqb23210ab8a2f3eeb@mail.gmail.com> <499B5345.1010701@ieee.org> <2d9149cd0902171620m28f82962sbdc087c0adbca70e@mail.gmail.com> Message-ID: <499B59DA.3000806@ieee.org> That and the lack of an FPU, I'm curious how that would affect FS, especially with transcoding. At one point I was interested in building a little embedded PBX running FS on the Blackfin MCU. Since Analog Devices seems fairly open with the Blackfin, I thought it might be a good choice, but I'm not sure how the lack of an MMU or FPU would affect FS. Gabe Kristian Kielhofner wrote: > I don't think so but something tells me that FreeSWITCH won't do too > well without an MMU and the external libs and modules could cause > quite a problem. Not that it is impossible but the uh, performance, > would be interesting... > > Can anyone call me out on this assumption? > > On Tue, Feb 17, 2009 at 7:16 PM, Gabriel Kuri wrote: >> awesome work! on a slightly related [embedded] note, do you know if any >> work has been done to port FS to any of the Analog Blackfin MCUs? I'd be >> interested in hearing if anyone has had any such luck. >> >> Gabe >> > From fax at virgintechnologies.com Tue Feb 17 16:45:05 2009 From: fax at virgintechnologies.com (fax at virgintechnologies.com) Date: Wed, 18 Feb 2009 00:45:05 +0000 Subject: [Freeswitch-users] Sending media streams to a media gateway Message-ID: I looked at that, but I think that will cause issues with the NAT traversal. Our phones will all be in external networks. I forgot to mention that. -----Original Message----- From: Anthony Minessale [mailto:anthony.minessale at gmail.com] Sent: Tuesday, February 17, 2009 05:34 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Sending media streams to a media gateway you could set the variable bypass_media to true before you call bridge that will negotiate a point to point media connection between the caller and callee On Tue, Feb 17, 2009 at 5:51 PM, wrote: I have Freeswitch running successfully with a fairly basic config. Nat traversal is working well on both the client and server side. I want to start running all RTP streams through a media gateway, and use Freeswitch for SIP registrations and signalling only. I believe that I need to have Freeswitch invite the SIP phone to send the RTP stream directly to the media gateway when a call starts. Where can I start with this? Does anyone have any example configs? Justin _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm mailto:MSN%3Aanthony_minessale at hotmail.com GTALK/JABBER/mailto:PAYPAL%3Aanthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference mailto:sip%3A888 at conference.freeswitch.org http://iax:guest at conference.freeswitch.org/888 mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090218/b2b679b1/attachment-0002.html From brian at freeswitch.org Tue Feb 17 17:36:39 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 17 Feb 2009 19:36:39 -0600 Subject: [Freeswitch-users] Anyone running FS from a Thumb Flash USB? In-Reply-To: <2d9149cd0902171620m28f82962sbdc087c0adbca70e@mail.gmail.com> References: <9dc4a1670902170550r3071f65dn76112fdbebaeb6ba@mail.gmail.com> <2d9149cd0902171543t1d1a5c9dqb23210ab8a2f3eeb@mail.gmail.com> <499B5345.1010701@ieee.org> <2d9149cd0902171620m28f82962sbdc087c0adbca70e@mail.gmail.com> Message-ID: So you sticking with the Astlinux name? or switching it to something more general? /b On Feb 17, 2009, at 6:20 PM, Kristian Kielhofner wrote: > I don't think so but something tells me that FreeSWITCH won't do too > well without an MMU and the external libs and modules could cause > quite a problem. Not that it is impossible but the uh, performance, > would be interesting... From kristian.kielhofner at gmail.com Tue Feb 17 17:41:56 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Tue, 17 Feb 2009 20:41:56 -0500 Subject: [Freeswitch-users] AstLinux-FreeSWITCH ISO available for download Message-ID: <2d9149cd0902171741h709cae3ckc9ff6ea00e268710@mail.gmail.com> I really need to work on that name but in the meantime it seems like people are interested. Check it out: http://www.astlinux.org/node/41 It's just a little ISO, download it and give it a shot! ;) -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From krice at freeswitch.org Tue Feb 17 17:48:08 2009 From: krice at freeswitch.org (Ken Rice) Date: Tue, 17 Feb 2009 19:48:08 -0600 Subject: [Freeswitch-users] Happy B-Day BKW! Message-ID: Ok Tomorrow is BKW's b-day... Lets as a community see if we cant pull together and get him something nice off his Amazon wish list... Visit his wish list here... http://www.amazon.com/gp/registry/wishlist/1BWDJUX5LYQE0 If you want to chip in contact me on IRC (I'm SwK) or send something via the donate link at http://www.tollfreegateway.com/bailmeout.html be sure to put "For BKW" in the comment or drop me an off list email... If you want to send something via paypal direct to brian his paypal ID is brian at freeswitch.org Come on guys he works day and night on FS to help bring us arguably the best OpenSource Telephony platform out there!! Happy B'Day Brian!! Ken From sprice at gmail.com Tue Feb 17 17:55:20 2009 From: sprice at gmail.com (SP) Date: Tue, 17 Feb 2009 19:55:20 -0600 Subject: [Freeswitch-users] Happy B-Day BKW! In-Reply-To: References: Message-ID: <7e2ac3270902171755g7774d04fhc659962ba1610d6a@mail.gmail.com> We could ban him from IRC for the day... that would be a gift :) On Tue, Feb 17, 2009 at 19:48, Ken Rice wrote: > Ok Tomorrow is BKW's b-day... > > Lets as a community see if we cant pull together and get him something nice > off his Amazon wish list... > > Visit his wish list here... > http://www.amazon.com/gp/registry/wishlist/1BWDJUX5LYQE0 > > If you want to chip in contact me on IRC (I'm SwK) or send something via the > donate link at http://www.tollfreegateway.com/bailmeout.html be sure to put > "For BKW" in the comment or drop me an off list email... > > If you want to send something via paypal direct to brian his paypal ID is > brian at freeswitch.org > > Come on guys he works day and night on FS to help bring us arguably the best > OpenSource Telephony platform out there!! > > Happy B'Day Brian!! > > Ken > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Shannon From jaybinks at gmail.com Tue Feb 17 18:04:05 2009 From: jaybinks at gmail.com (jay binks) Date: Wed, 18 Feb 2009 12:04:05 +1000 Subject: [Freeswitch-users] AstLinux-FreeSWITCH ISO available for download In-Reply-To: <2d9149cd0902171741h709cae3ckc9ff6ea00e268710@mail.gmail.com> References: <2d9149cd0902171741h709cae3ckc9ff6ea00e268710@mail.gmail.com> Message-ID: awwww man.... geez im interested in this .. I hope it ends up kicking ass ! :) Congrats, you are awesome. Jay On Wed, Feb 18, 2009 at 11:41 AM, Kristian Kielhofner < kristian.kielhofner at gmail.com> wrote: > I really need to work on that name but in the meantime it seems like > people are interested. Check it out: > > http://www.astlinux.org/node/41 > > It's just a little ISO, download it and give it a shot! ;) > > -- > Kristian Kielhofner > http://blog.krisk.org > http://www.submityoursip.com > http://www.astlinux.org > http://www.star2star.com > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Sincerely Jay -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090218/7f861c91/attachment-0002.html From kristian.kielhofner at gmail.com Tue Feb 17 18:15:08 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Tue, 17 Feb 2009 21:15:08 -0500 Subject: [Freeswitch-users] Anyone running FS from a Thumb Flash USB? In-Reply-To: References: <9dc4a1670902170550r3071f65dn76112fdbebaeb6ba@mail.gmail.com> <2d9149cd0902171543t1d1a5c9dqb23210ab8a2f3eeb@mail.gmail.com> <499B5345.1010701@ieee.org> <2d9149cd0902171620m28f82962sbdc087c0adbca70e@mail.gmail.com> Message-ID: <2d9149cd0902171815q4a25b42n447efffb251574f@mail.gmail.com> Brian, We'll figure something out... On Tue, Feb 17, 2009 at 8:36 PM, Brian West wrote: > So you sticking with the Astlinux name? or switching it to something > more general? > > /b > -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From intralanman at freeswitch.org Tue Feb 17 18:41:22 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Tue, 17 Feb 2009 21:41:22 -0500 Subject: [Freeswitch-users] AddBody to events in lua In-Reply-To: References: <191c3a030902171525y17166787ud37ab5243c287008@mail.gmail.com><87f2f3b90902171529w1d5c32e5ydc6e6479d03ecab6@mail.gmail.com> Message-ID: <499B7552.3010104@freeswitch.org> Nik Middleton wrote: > I'll shortly post some docs on the php fs_sock. don't waste your time... There's a php .so for ESL now, and i'll probably be removing the fs_sock from tree sometime very soon... maybe replacing it with some specific api classes... i'm not sure on that part yet. -Ray From mike at jerris.com Tue Feb 17 20:16:22 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 17 Feb 2009 23:16:22 -0500 Subject: [Freeswitch-users] Anyone running FS from a Thumb Flash USB? In-Reply-To: <499B59DA.3000806@ieee.org> References: <9dc4a1670902170550r3071f65dn76112fdbebaeb6ba@mail.gmail.com> <2d9149cd0902171543t1d1a5c9dqb23210ab8a2f3eeb@mail.gmail.com> <499B5345.1010701@ieee.org> <2d9149cd0902171620m28f82962sbdc087c0adbca70e@mail.gmail.com> <499B59DA.3000806@ieee.org> Message-ID: <007ACD53-6B53-4E4D-BB44-38845B8E6B26@jerris.com> Pretty much all the codecs (mod_voipcodecs, mod_speex, mod_ilbc, mod_g722_1, mod_celt) and the resampler all have fixed point implementations (in tree) as well. Mike On Feb 17, 2009, at 7:44 PM, Gabriel Kuri wrote: > That and the lack of an FPU, I'm curious how that would affect FS, > especially with transcoding. At one point I was interested in > building a > little embedded PBX running FS on the Blackfin MCU. Since Analog > Devices > seems fairly open with the Blackfin, I thought it might be a good > choice, but I'm not sure how the lack of an MMU or FPU would affect > FS. > > Gabe > > Kristian Kielhofner wrote: >> I don't think so but something tells me that FreeSWITCH won't do too >> well without an MMU and the external libs and modules could cause >> quite a problem. Not that it is impossible but the uh, performance, >> would be interesting... >> >> Can anyone call me out on this assumption? >> >> On Tue, Feb 17, 2009 at 7:16 PM, Gabriel Kuri wrote: >>> awesome work! on a slightly related [embedded] note, do you know >>> if any >>> work has been done to port FS to any of the Analog Blackfin MCUs? >>> I'd be >>> interested in hearing if anyone has had any such luck. >>> >>> Gabe >>> From kristian.kielhofner at gmail.com Tue Feb 17 23:09:58 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Wed, 18 Feb 2009 02:09:58 -0500 Subject: [Freeswitch-users] Anyone running FS from a Thumb Flash USB? In-Reply-To: References: <9dc4a1670902170550r3071f65dn76112fdbebaeb6ba@mail.gmail.com> <2d9149cd0902171543t1d1a5c9dqb23210ab8a2f3eeb@mail.gmail.com> <87f2f3b90902171610r37679769h2b4f1c033e7b3e94@mail.gmail.com> <2d9149cd0902171618s47640fb2vff3cc139b74c8f64@mail.gmail.com> Message-ID: <2d9149cd0902172309i1aa93fcej8fce68133028ee93@mail.gmail.com> Nik, Thanks but I'm not sure I want to take the credit (blame?) for that! ;) On Tue, Feb 17, 2009 at 7:25 PM, Nik Middleton wrote: > Kristian, > > You're my hero, if I hadn't come across astlinux 3 years ago, I wouldn't > be doing this stuff right now. Not too sure if that's a good thing > though ;) > -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From uv at yuvalhertzog.com Wed Feb 18 03:55:35 2009 From: uv at yuvalhertzog.com (UV) Date: Wed, 18 Feb 2009 22:55:35 +1100 Subject: [Freeswitch-users] Anyone running FS from a Thumb Flash USB? In-Reply-To: <2d9149cd0902172309i1aa93fcej8fce68133028ee93@mail.gmail.com> References: <9dc4a1670902170550r3071f65dn76112fdbebaeb6ba@mail.gmail.com><2d9149cd0902171543t1d1a5c9dqb23210ab8a2f3eeb@mail.gmail.com><87f2f3b90902171610r37679769h2b4f1c033e7b3e94@mail.gmail.com><2d9149cd0902171618s47640fb2vff3cc139b74c8f64@mail.gmail.com> <2d9149cd0902172309i1aa93fcej8fce68133028ee93@mail.gmail.com> Message-ID: <05B5115F398B44C28AFDB2DE0E1A9281@UVix> Awesome work, Kristian! And very much needed for the Freeswitch platform (to me, at least). A suggestion: if the FS team doesn't mind (after getting over the naming issue), it would be a good idea to put Kristian's latest blog entry on the FS Wiki. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Kristian Kielhofner Sent: Wednesday, February 18, 2009 6:10 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Anyone running FS from a Thumb Flash USB? Nik, Thanks but I'm not sure I want to take the credit (blame?) for that! ;) On Tue, Feb 17, 2009 at 7:25 PM, Nik Middleton wrote: > Kristian, > > You're my hero, if I hadn't come across astlinux 3 years ago, I wouldn't > be doing this stuff right now. Not too sure if that's a good thing > though ;) > -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org No virus found in this incoming message. Checked by AVG - www.avg.com Version: 8.0.237 / Virus Database: 270.10.25/1956 - Release Date: 02/17/09 07:07:00 No virus found in this outgoing message. Checked by AVG - www.avg.com Version: 8.0.237 / Virus Database: 270.10.25/1956 - Release Date: 02/17/09 07:07:00 From gmaruzz at celliax.org Wed Feb 18 03:58:12 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Wed, 18 Feb 2009 12:58:12 +0100 Subject: [Freeswitch-users] Skypiax on OS X In-Reply-To: References: Message-ID: <7b197bef0902180358s7c3ca246kf354beef2e72f4e6@mail.gmail.com> > On Feb 17, 2009, at 3:19 PM, Ivan C Myrvold wrote: >> Is it possible to run Skypiax on OS X? The wiki says Linux and Windows, >> but says nothing about OS X. On Tue, Feb 17, 2009 at 10:31 PM, Brian West wrote: > I think Ctrix is working on mod_airpe in his branch for OS X. At the moment is not possible to run Skypiax on OSX. I have no OSX machines at the moment, sorry. I would like to add OSX support to Skypiax tough. I'll have someone lend me a machine in the future, if nobody else sends patches. In the mean time, as bkw wrote, you can try the mod_airpe from Ctrix. Sincerely, Giovanni Maruzzelli ========================================= www.celliax.org via Pierlombardo 9, 20135 Milano Italy gmaruzz at celliax dot org Cell : +39-347-2665618 Fax : +39-02-87390039 On Tue, Feb 17, 2009 at 10:31 PM, Brian West wrote: > I think Ctrix is working on mod_airpe in his branch for OS X. > > /b > > On Feb 17, 2009, at 3:19 PM, Ivan C Myrvold wrote: > >> s it possible to run Skypiax on OS X? The wiki says Linux and Windows, >> but says nothing about OS X. >> I have been running FreeSWITCH on OS X for a couple of years now, and >> love it. Adding Skype gateway would be really sweet. >> >> Are there any plans for adding Skypiax to trunk, or do we have to >> build it only from svn branch? >> >> Ivan > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From leon at scarlet-internet.nl Wed Feb 18 06:03:52 2009 From: leon at scarlet-internet.nl (Leon de Rooij) Date: Wed, 18 Feb 2009 15:03:52 +0100 Subject: [Freeswitch-users] ipauth - directory In-Reply-To: <191c3a030902160604r77089a6s5b3b9f3d07914218@mail.gmail.com> References: <496E0187.8090805@kinetix.gr> <23DEEC73-4DC7-4056-902F-D74DD6644385@freeswitch.org> <496E068B.6050404@kinetix.gr> <191c3a030902160604r77089a6s5b3b9f3d07914218@mail.gmail.com> Message-ID: Hi Anthony, I tried your second option, but how does it work with xml-curl then ? As far as I understand it, this doesn't work by doing a user-directory xml lookup at INVITE time, or does it ? Or does it want to generate an ACL at FS startup and filling up all the allow-nodes by polling the entire domain, filtering out all users with CIDR entry and putting those in the ACL itself ? If so, is that the reason why FS tries (at startup) to POST to the webserver with: hostname = test §ion = directory &tag_name = domain &key_name=name&key_value=test.com&domain=test.com&purpose=network-list ? Thanks & regards, Leon On Feb 16, 2009, at 3:04 PM, Anthony Minessale wrote: > you have 3 options. > on authenticated users, every tag in his account will be > set on each call from that authenticated user. > > 1) make them register, this sets the variables automatically > 2) use the ACL list with cidr= from> this has the same effect with no auth needed. > 3) use some other way to differentiate the user and use the set_user > application in the dialplan to inherit that user's variables. > > > > On Mon, Feb 16, 2009 at 6:49 AM, Leon de Rooij > wrote: > Hi all, > > I'd really like to know more about this too. > > Currently, I have two sip_profiles: > > - residential (where users can do authenticated registers and invites) > - transit (where other users can do un-authenticated invites) > > Right now, FS is not aware of *who* is accessing the transit profile > except for an acl that is set on this profile so unauthorized use is > not possible. > > But what should I do when I want to allow multiple parties (from > different IP addresses) to send their invites to the transit > profile, and still be able to differentiate between them ? > > I'd like to set some variables, like an accountcode for example, on > the basis of what IP address the INVITE originates from. > > So, is it possible to not use digest authentication, but still use a > dialplan-directory user with IP= field or some such ? > > thanks a lot & kind regards, > > Leon de Rooij > > > > On Jan 14, 2009, at 4:36 PM, Apostolos Pantsiopoulos wrote: > >> Yes I know that. But what does the "ip=" setting do? >> >> Brian West wrote: >>> >>> cidr= and the domains acl in acl.conf.xml then apply that ACL to the >>> sofia profile. >>> >>> /b >>> >>> On Jan 14, 2009, at 9:15 AM, Apostolos Pantsiopoulos wrote: >>> >>> >>>> I noticed an "ip=" setting in the brian.xml sample file. >>>> The comments state that this is used for ipauth (IP based >>>> authentication?) >>>> >>>> What exactly is this setting. I cannot find anything in the wiki >>>> about it. >>>> Does it replace the use of the >>>> >>>> + ACL >>>> >>>> mechanism for IP authentication? >>>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> -- >> ------------------------------------------- >> Apostolos Pantsiopoulos >> Kinetix Tele.com R & D >> email: regs at kinetix.gr >> ------------------------------------------- >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090218/17eecf9a/attachment-0002.html From anthony.minessale at gmail.com Wed Feb 18 06:09:27 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 18 Feb 2009 08:09:27 -0600 Subject: [Freeswitch-users] call goes to wrong context In-Reply-To: References: <499B1B42.7090000@post.cz> <4432BDD8-67A4-45B4-A013-7BB7C7E35D7F@freeswitch.org> <499B2691.70003@post.cz> Message-ID: <191c3a030902180609m52ad6b2es2aa84f87b994e3e2@mail.gmail.com> if the inbound calls are coming from a registration to a provider you will have to set a context param in the gateway itself. All inbound calls from a gateway registration are now associated with the gateway they were registered with and inherit the context from there. Maybe i'll change the default context of a gateway to be the default context of it's host profile to avoid this issue. On Tue, Feb 17, 2009 at 3:09 PM, Brian West wrote: > No problem. Just join us on IRC.. things move faster on there. > > /b > > On Feb 17, 2009, at 3:05 PM, kokoska.rokoska wrote: > > > If you have more hints, I be very happy :-) > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090218/d7e1aaeb/attachment-0002.html From kawarod at laposte.net Wed Feb 18 06:10:14 2009 From: kawarod at laposte.net (rod) Date: Wed, 18 Feb 2009 18:10:14 +0400 Subject: [Freeswitch-users] mod_fax and sending a fax In-Reply-To: <49919822.3030101@laposte.net> References: <49919822.3030101@laposte.net> Message-ID: <499C16C6.1000006@laposte.net> Hi all, I'm able to receive a fax with mod_fax, but I still don't understand how to send fax. I don't understand how to send the fax through a specific profile/IP. Is mod_fax limited to an openzap interface?? when I dial to the extension with tx_fax, I get a tone then hangup, but what I'd like to do is send the fax call through a specific peer (as the bridge application). For those who'd like to trigger an event on received fax (convert to pdf, send a mail...), I think a tool like this could help: http://projects.l3ib.org/trac/fsniper I did not try it at this time, if I'm successful I will update the wiki. regards, rod. rod wrote: > Hi all, > > I don't understand how to use the fax commands for sending a fax. In the > wiki I saw this: > > > > > > > > > > my question is how to specify the gateway/profile that will handle the call. > For a call I can use the bridge application like this, but for the txfax ?? > data="sofia/external/${destination_number}@10.10.10.10"/> > > regards, > rod > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > From anthony.minessale at gmail.com Wed Feb 18 06:13:32 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 18 Feb 2009 08:13:32 -0600 Subject: [Freeswitch-users] call goes to wrong context In-Reply-To: <191c3a030902180609m52ad6b2es2aa84f87b994e3e2@mail.gmail.com> References: <499B1B42.7090000@post.cz> <4432BDD8-67A4-45B4-A013-7BB7C7E35D7F@freeswitch.org> <499B2691.70003@post.cz> <191c3a030902180609m52ad6b2es2aa84f87b994e3e2@mail.gmail.com> Message-ID: <191c3a030902180613x2f3b66av336f1bbcedf41703@mail.gmail.com> done, r12138 should give you the correct behavior On Wed, Feb 18, 2009 at 8:09 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > if the inbound calls are coming from a registration to a provider you will > have to set a context param in the gateway itself. > All inbound calls from a gateway registration are now associated with the > gateway they were registered with and inherit > the context from there. > > Maybe i'll change the default context of a gateway to be the default > context of it's host profile to avoid this issue. > > > > > On Tue, Feb 17, 2009 at 3:09 PM, Brian West wrote: > >> No problem. Just join us on IRC.. things move faster on there. >> >> /b >> >> On Feb 17, 2009, at 3:05 PM, kokoska.rokoska wrote: >> >> > If you have more hints, I be very happy :-) >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090218/b49a6ef7/attachment-0002.html From anthony.minessale at gmail.com Wed Feb 18 06:14:20 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 18 Feb 2009 08:14:20 -0600 Subject: [Freeswitch-users] ipauth - directory In-Reply-To: References: <496E0187.8090805@kinetix.gr> <23DEEC73-4DC7-4056-902F-D74DD6644385@freeswitch.org> <496E068B.6050404@kinetix.gr> <191c3a030902160604r77089a6s5b3b9f3d07914218@mail.gmail.com> Message-ID: <191c3a030902180614p5de79174r56b21fe52cb0fe33@mail.gmail.com> yes that is correct. On Wed, Feb 18, 2009 at 8:03 AM, Leon de Rooij wrote: > Hi Anthony, > I tried your second option, but how does it work with xml-curl then ? As > far as I understand it, this doesn't work by doing a user-directory xml > lookup at INVITE time, or does it ? > > Or does it want to generate an ACL at FS startup and filling up all the > allow-nodes by polling the entire domain, filtering out all users with CIDR > entry and putting those in the ACL itself ? > > If so, is that the reason why FS tries (at startup) to POST to the > webserver with: > hostname=test§ion=directory&tag_name=domain&key_name=name&key_value= > test.com&domain=test.com&purpose=network-list > > ? > > Thanks & regards, > > Leon > > > On Feb 16, 2009, at 3:04 PM, Anthony Minessale wrote: > > you have 3 options. > on authenticated users, every tag in his account will be set on > each call from that authenticated user. > > 1) make them register, this sets the variables automatically > 2) use the ACL list with cidr= this > has the same effect with no auth needed. > 3) use some other way to differentiate the user and use the set_user > application in the dialplan to inherit that user's variables. > > > > On Mon, Feb 16, 2009 at 6:49 AM, Leon de Rooij wrote: > >> Hi all, >> >> I'd really like to know more about this too. >> >> Currently, I have two sip_profiles: >> >> - residential (where users can do authenticated registers and invites) >> - transit (where other users can do un-authenticated invites) >> >> Right now, FS is not aware of *who* is accessing the transit profile >> except for an acl that is set on this profile so unauthorized use is not >> possible. >> >> But what should I do when I want to allow multiple parties (from different >> IP addresses) to send their invites to the transit profile, and still be >> able to differentiate between them ? >> >> I'd like to set some variables, like an accountcode for example, on the >> basis of what IP address the INVITE originates from. >> >> So, is it possible to not use digest authentication, but still use a >> dialplan-directory user with IP= field or some such ? >> >> thanks a lot & kind regards, >> >> Leon de Rooij >> >> >> >> On Jan 14, 2009, at 4:36 PM, Apostolos Pantsiopoulos wrote: >> >> Yes I know that. But what does the "ip=" setting do? >> >> Brian West wrote: >> >> cidr= and the domains acl in acl.conf.xml then apply that ACL to the >> sofia profile. >> >> /b >> >> On Jan 14, 2009, at 9:15 AM, Apostolos Pantsiopoulos wrote: >> >> >> >> I noticed an "ip=" setting in the brian.xml sample file. >> The comments state that this is used for ipauth (IP based >> authentication?) >> >> What exactly is this setting. I cannot find anything in the wiki >> about it. >> Does it replace the use of the >> >> + ACL >> >> mechanism for IP authentication? >> >> >> _______________________________________________ >> Freeswitch-users mailing listFreeswitch-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >> >> >> >> -- >> ------------------------------------------- >> Apostolos Pantsiopoulos >> Kinetix Tele.com R & D >> email: regs at kinetix.gr >> ------------------------------------------- >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090218/b0c289a9/attachment-0002.html From edpimentl at gmail.com Wed Feb 18 06:21:24 2009 From: edpimentl at gmail.com (EdPimentl) Date: Wed, 18 Feb 2009 09:21:24 -0500 Subject: [Freeswitch-users] Skypiax on OS X In-Reply-To: <7b197bef0902180358s7c3ca246kf354beef2e72f4e6@mail.gmail.com> References: <7b197bef0902180358s7c3ca246kf354beef2e72f4e6@mail.gmail.com> Message-ID: <9dc4a1670902180621u350ed70bt6aa92731790014d0@mail.gmail.com> OSX can be loaded on any new Intel machines.. -E http://TwiTR.Me -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090218/1a6798e7/attachment-0002.html From kokoska.rokoska at post.cz Wed Feb 18 06:24:51 2009 From: kokoska.rokoska at post.cz (kokoska rokoska) Date: Wed, 18 Feb 2009 15:24:51 +0100 Subject: [Freeswitch-users] call goes to wrong context In-Reply-To: <191c3a030902180609m52ad6b2es2aa84f87b994e3e2@mail.gmail.com> References: <499B1B42.7090000@post.cz> <4432BDD8-67A4-45B4-A013-7BB7C7E35D7F@freeswitch.org> <499B2691.70003@post.cz> <191c3a030902180609m52ad6b2es2aa84f87b994e3e2@mail.gmail.com> Message-ID: <499C1A33.8070002@post.cz> Anthony Minessale napsal(a): > if the inbound calls are coming from a registration to a provider you > will have to set a context param in the gateway itself. > All inbound calls from a gateway registration are now associated with > the gateway they were registered with and inherit > the context from there. > Thank you very much, Anthony, for your explanation! I got advice to setup context in gateway definition from Brian on IRC channel and it works, so I asume the the reason is some internal FS change :-) > Maybe i'll change the default context of a gateway to be the default > context of it's host profile to avoid this issue. > You'll be very glad :-) Because it is how it works in the past (at about 3-4 weeks old FS svn trunk) and even more - it is a little bit strange if xml_curl looks for "default" context which I don't have either... BTW: Is "default" context mandatory for FreeSWITCH (hardcoded somewhere in the code) or it is up to users decision how to name contexts? Tahnks once more, Anthony! Best regards, kokoska.rokoska From kokoska.rokoska at post.cz Wed Feb 18 06:26:45 2009 From: kokoska.rokoska at post.cz (kokoska rokoska) Date: Wed, 18 Feb 2009 15:26:45 +0100 Subject: [Freeswitch-users] call goes to wrong context In-Reply-To: <191c3a030902180613x2f3b66av336f1bbcedf41703@mail.gmail.com> References: <499B1B42.7090000@post.cz> <4432BDD8-67A4-45B4-A013-7BB7C7E35D7F@freeswitch.org> <499B2691.70003@post.cz> <191c3a030902180609m52ad6b2es2aa84f87b994e3e2@mail.gmail.com> <191c3a030902180613x2f3b66av336f1bbcedf41703@mail.gmail.com> Message-ID: <499C1AA5.5080807@post.cz> Anthony Minessale napsal(a): > done, > > r12138 should give you the correct behavior > Thank you very much, Anthony! Incredible speed :-) Best regards, kokoska.rokoska From moizchinoy at gmail.com Wed Feb 18 07:09:47 2009 From: moizchinoy at gmail.com (Moiz Chinoy) Date: Wed, 18 Feb 2009 07:09:47 -0800 (PST) Subject: [Freeswitch-users] FreeSwitch - PcoketSphinx Prompt Playback & Recognition Issue... Message-ID: <22080033.post@talk.nabble.com> Hi, I have downloaded and build the Freeswitch from http://files.freeswitch.org/freeswitch-1.0.3RC1.tar.gz on Windows XP. Everything built successfully. Then I configured PocketSphinx as described at http://wiki.freeswitch.org/wiki/Mod_pocketsphinx. The problem is that prompts (wave files) are not being played properly while testing the Pizza demo i.e it plays, stops then start playing again... I am using heaset for testing. Moreover, the recognition seems to be very poor. Any clue what might be the issue? Moiz Chinoy. Following is snippet from log: 2009-02-18 16:59:59 [DEBUG] switch_core_session.c:513 switch_core_session_perform_receive_message() Send signal sofia/internal/1000 at 192.168.16.63 [BREAK] 2009-02-18 16:59:59 [NOTICE] mod_spidermonkey.c:2041 session_answer() Channel [sofia/internal/1000 at 192.168.16.63] has been answered 2009-02-18 16:59:59 [DEBUG] switch_channel.c:179 switch_channel_audio_sync() sofia/internal/1000 at 192.168.16.63 receive message [AUDIO_SYNC] 2009-02-18 16:59:59 [DEBUG] sofia.c:2672 sofia_handle_sip_i_state() Channel sofia/internal/1000 at 192.168.16.63 entering state [completed] 2009-02-18 16:59:59 [DEBUG] sofia.c:2672 sofia_handle_sip_i_state() Channel sofia/internal/1000 at 192.168.16.63 entering state [ready] 2009-02-18 16:59:59 [DEBUG] switch_ivr_play_say.c:968 switch_ivr_play_file() Codec Activated L16 at 8000hz 1 channels 20ms 2009-02-18 16:59:59 [DEBUG] switch_core_io.c:654 switch_core_session_write_frame() sofia/internal/1000 at 192.168.16.63 receive message [TRANSCODING_NECESSARY] 2009-02-18 17:00:00 [INFO] switch_rtp.c:1422 rtp_common_read() Auto Changing port from 127.0.0.1:49194 to 192.168.16.63:49194 2009-02-18 17:00:02 [DEBUG] switch_ivr_play_say.c:1258 switch_ivr_play_file() done playing file 2009-02-18 17:00:11 [WARNING] switch_scheduler.c:114 task_thread_loop() Task was executed late by 2 seconds 1 heartbeat (core) 2009-02-18 17:00:18 [DEBUG] switch_core_media_bug.c:284 switch_core_media_bug_add() Attaching BUG to sofia/internal/1000 at 192.168.16.63 2009-02-18 17:00:18 [DEBUG] switch_core_io.c:234 switch_core_session_read_frame() sofia/internal/1000 at 192.168.16.63 receive message [TRANSCODING_NECESSARY] 2009-02-18 17:00:51 [WARNING] switch_scheduler.c:114 task_thread_loop() Task was executed late by 20 seconds 1 heartbeat (core) 2009-02-18 17:00:51 [DEBUG] switch_ivr_play_say.c:968 switch_ivr_play_file() Codec Activated L16 at 8000hz 1 channels 20ms 2009-02-18 17:00:58 [DEBUG] switch_ivr_play_say.c:1258 switch_ivr_play_file() done playing file 2009-02-18 17:00:58 [DEBUG] switch_core_io.c:234 switch_core_session_read_frame() sofia/internal/1000 at 192.168.16.63 receive message [TRANSCODING_NECESSARY] 2009-02-18 17:01:00 [DEBUG] mod_pocketsphinx.c:386 pocketsphinx_asr_get_results() Recognized: ????????????????, Score: 100 2009-02-18 17:01:00 [DEBUG] js_modules/SpeechTools.jm:150 console_log() ----XML: ???????????????? ???????????????? 2009-02-18 17:01:00 [DEBUG] js_modules/SpeechTools.jm:150 console_log() ----Heard [????????????????] 2009-02-18 17:01:00 [DEBUG] js_modules/SpeechTools.jm:150 console_log() ----Hit score 100/40/70 2009-02-18 17:01:00 [INFO] js_modules/SpeechTools.jm:150 console_log() ----???????????????? 2009-02-18 17:01:00 [DEBUG] js_modules/SpeechTools.jm:365 console_log() ----Testing [0] [0] ???????????????? =~ [Delivery:::Delivery] 2009-02-18 17:01:00 [DEBUG] js_modules/SpeechTools.jm:365 console_log() ----Testing [0] [1] ???????????????? =~ [Takeout:::Pickup] 2009-02-18 17:01:00 [DEBUG] js_modules/SpeechTools.jm:365 console_log() ----Testing [0] [2] ???????????????? =~ [Pickup:::Pickup] 2009-02-18 17:01:02 [DEBUG] mod_pocketsphinx.c:342 pocketsphinx_asr_resume() Manually Resuming 2009-02-18 17:01:02 [DEBUG] switch_ivr_play_say.c:968 switch_ivr_play_file() Codec Activated L16 at 8000hz 1 channels 20ms 2009-02-18 17:01:02 [DEBUG] switch_ivr_play_say.c:1258 switch_ivr_play_file() done playing file 2009-02-18 17:01:02 [DEBUG] switch_core_io.c:234 switch_core_session_read_frame() sofia/internal/1000 at 192.168.16.63 receive message [TRANSCODING_NECESSARY] 2009-02-18 17:01:03 [DEBUG] switch_ivr_play_say.c:968 switch_ivr_play_file() Codec Activated L16 at 8000hz 1 channels 20ms 2009-02-18 17:01:06 [DEBUG] switch_ivr_play_say.c:1258 switch_ivr_play_file() done playing file -- View this message in context: http://www.nabble.com/FreeSwitch---PcoketSphinx-Prompt-Playback---Recognition-Issue...-tp22080033p22080033.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From moizchinoy at gmail.com Wed Feb 18 07:17:03 2009 From: moizchinoy at gmail.com (Moiz Chinoy) Date: Wed, 18 Feb 2009 07:17:03 -0800 (PST) Subject: [Freeswitch-users] FreeSwitch - PcoketSphinx Prompt Playback & Recognition Issue... In-Reply-To: <22080033.post@talk.nabble.com> References: <22080033.post@talk.nabble.com> Message-ID: <22080857.post@talk.nabble.com> And one more thing... As soon as it recognizez TAKEOUT, freeswitch crashes... 2009-02-18 19:03:51 [DEBUG] js_modules/SpeechTools.jm:150 console_log() ----Heard [????] 2009-02-18 19:03:51 [DEBUG] js_modules/SpeechTools.jm:150 console_log() ----Hit score 98/40/70 2009-02-18 19:03:51 [INFO] js_modules/SpeechTools.jm:150 console_log() ----???? 2009-02-18 19:03:51 [DEBUG] switch_ivr_play_say.c:1258 switch_ivr_play_file() done playing file 2009-02-18 19:03:51 [DEBUG] js_modules/SpeechTools.jm:365 console_log() ----Testing [0] [0] ???? =~ [Delivery:::Delivery] 2009-02-18 19:03:51 [DEBUG] js_modules/SpeechTools.jm:365 console_log() ----Testing [0] [1] ???? =~ [Takeout:::Pickup] 2009-02-18 19:03:51 [DEBUG] js_modules/SpeechTools.jm:365 console_log() ----Testing [0] [2] ???? =~ [Pickup:::Pickup] 2009-02-18 19:03:51 [DEBUG] switch_core_io.c:234 switch_core_session_read_frame() sofia/internal/1000 at 192.168.16.63 receive message [TRANSCODING_NECESSARY] 2009-02-18 19:03:53 [DEBUG] mod_pocketsphinx.c:342 pocketsphinx_asr_resume() Manually Resuming 2009-02-18 19:03:53 [DEBUG] switch_ivr_play_say.c:968 switch_ivr_play_file() Codec Activated L16 at 8000hz 1 channels 20ms 2009-02-18 19:03:58 [DEBUG] switch_ivr_play_say.c:1258 switch_ivr_play_file() done playing file 2009-02-18 19:03:58 [DEBUG] switch_core_io.c:234 switch_core_session_read_frame() sofia/internal/1000 at 192.168.16.63 receive message [TRANSCODING_NECESSARY] 2009-02-18 19:03:59 [DEBUG] switch_ivr_play_say.c:968 switch_ivr_play_file() Codec Activated L16 at 8000hz 1 channels 20ms 2009-02-18 19:04:02 [DEBUG] switch_ivr_play_say.c:1258 switch_ivr_play_file() done playing file 2009-02-18 19:04:02 [DEBUG] switch_core_io.c:234 switch_core_session_read_frame() sofia/internal/1000 at 192.168.16.63 receive message [TRANSCODING_NECESSARY] 2009-02-18 19:04:09 [DEBUG] mod_pocketsphinx.c:342 pocketsphinx_asr_resume() Manually Resuming 2009-02-18 19:04:09 [DEBUG] switch_ivr_play_say.c:968 switch_ivr_play_file() Codec Activated L16 at 8000hz 1 channels 20ms 2009-02-18 19:04:14 [DEBUG] switch_ivr_play_say.c:1258 switch_ivr_play_file() done playing file 2009-02-18 19:04:14 [DEBUG] switch_core_io.c:234 switch_core_session_read_frame() sofia/internal/1000 at 192.168.16.63 receive message [TRANSCODING_NECESSARY] 2009-02-18 19:04:15 [DEBUG] switch_ivr_play_say.c:968 switch_ivr_play_file() Codec Activated L16 at 8000hz 1 channels 20ms 2009-02-18 19:04:16 [DEBUG] mod_pocketsphinx.c:386 pocketsphinx_asr_get_results() Recognized: ?????|, Score: 100 2009-02-18 19:04:16 [DEBUG] js_modules/SpeechTools.jm:150 console_log() ----XML: ?????| ?????| 2009-02-18 19:04:16 [DEBUG] js_modules/SpeechTools.jm:150 console_log() ----Heard [?????|] 2009-02-18 19:04:16 [DEBUG] js_modules/SpeechTools.jm:150 console_log() ----Hit score 100/40/70 2009-02-18 19:04:16 [INFO] js_modules/SpeechTools.jm:150 console_log() ----?????| 2009-02-18 19:04:16 [DEBUG] switch_ivr_play_say.c:1258 switch_ivr_play_file() done playing file 2009-02-18 19:04:16 [DEBUG] js_modules/SpeechTools.jm:365 console_log() ----Testing [0] [0] ?????| =~ [Delivery:::Delivery] 2009-02-18 19:04:16 [DEBUG] js_modules/SpeechTools.jm:365 console_log() ----Testing [0] [1] ?????| =~ [Takeout:::Pickup] 2009-02-18 19:04:16 [DEBUG] js_modules/SpeechTools.jm:365 console_log() ----Testing [0] [2] ?????| =~ [Pickup:::Pickup] 2009-02-18 19:04:16 [DEBUG] switch_core_io.c:234 switch_core_session_read_frame() sofia/internal/1000 at 192.168.16.63 receive message [TRANSCODING_NECESSARY] 2009-02-18 19:04:18 [DEBUG] mod_pocketsphinx.c:342 pocketsphinx_asr_resume() Manually Resuming 2009-02-18 19:04:18 [DEBUG] switch_ivr_play_say.c:968 switch_ivr_play_file() Codec Activated L16 at 8000hz 1 channels 20ms 2009-02-18 19:04:18 [DEBUG] switch_ivr_play_say.c:1258 switch_ivr_play_file() done playing file 2009-02-18 19:04:18 [DEBUG] switch_core_io.c:234 switch_core_session_read_frame() sofia/internal/1000 at 192.168.16.63 receive message [TRANSCODING_NECESSARY] 2009-02-18 19:04:19 [DEBUG] switch_ivr_play_say.c:968 switch_ivr_play_file() Codec Activated L16 at 8000hz 1 channels 20ms 2009-02-18 19:04:20 [DEBUG] mod_pocketsphinx.c:386 pocketsphinx_asr_get_results() Recognized: TAKEOUT, Score: 72 2009-02-18 19:04:20 [DEBUG] js_modules/SpeechTools.jm:150 console_log() ----XML: TAKEOUT TAKEOUT 2009-02-18 19:04:20 [DEBUG] js_modules/SpeechTools.jm:150 console_log() ----Heard [TAKEOUT] 2009-02-18 19:04:20 [DEBUG] js_modules/SpeechTools.jm:150 console_log() ----Hit score 72/40/70 2009-02-18 19:04:20 [INFO] js_modules/SpeechTools.jm:150 console_log() ----TAKEOUT 2009-02-18 19:04:20 [DEBUG] switch_ivr_play_say.c:1258 switch_ivr_play_file() done playing file 2009-02-18 19:04:20 [DEBUG] js_modules/SpeechTools.jm:365 console_log() ----Testing [0] [0] TAKEOUT =~ [Delivery:::Delivery] 2009-02-18 19:04:20 [DEBUG] js_modules/SpeechTools.jm:365 console_log() ----Testing [0] [1] TAKEOUT =~ [Takeout:::Pickup] 2009-02-18 19:04:20 [DEBUG] js_modules/SpeechTools.jm:365 console_log() ----Adding Pickup 2009-02-18 19:04:20 [DEBUG] js_modules/SpeechTools.jm:365 console_log() ----Testing [0] [2] TAKEOUT =~ [Pickup:::Pickup] 2009-02-18 19:04:20 [DEBUG] switch_core_io.c:234 switch_core_session_read_frame() sofia/internal/1000 at 192.168.16.63 receive message [TRANSCODING_NECESSARY] 2009-02-18 19:04:21 [DEBUG] js_modules/SpeechTools.jm:109 console_log() Unloading grammar pizza_order Moiz Chinoy. Moiz Chinoy wrote: > > Hi, > > I have downloaded and build the Freeswitch from > http://files.freeswitch.org/freeswitch-1.0.3RC1.tar.gz on Windows XP. > > Everything built successfully. > Then I configured PocketSphinx as described at > http://wiki.freeswitch.org/wiki/Mod_pocketsphinx. > The problem is that prompts (wave files) are not being played properly > while testing the Pizza demo i.e it plays, stops then start playing > again... > > I am using heaset for testing. > > Moreover, the recognition seems to be very poor. > > Any clue what might be the issue? > > Moiz Chinoy. > > Following is snippet from log: > > 2009-02-18 16:59:59 [DEBUG] switch_core_session.c:513 > switch_core_session_perform_receive_message() Send signal > sofia/internal/1000 at 192.168.16.63 [BREAK] > 2009-02-18 16:59:59 [NOTICE] mod_spidermonkey.c:2041 session_answer() > Channel [sofia/internal/1000 at 192.168.16.63] has been answered > 2009-02-18 16:59:59 [DEBUG] switch_channel.c:179 > switch_channel_audio_sync() sofia/internal/1000 at 192.168.16.63 receive > message [AUDIO_SYNC] > 2009-02-18 16:59:59 [DEBUG] sofia.c:2672 sofia_handle_sip_i_state() > Channel sofia/internal/1000 at 192.168.16.63 entering state [completed] > 2009-02-18 16:59:59 [DEBUG] sofia.c:2672 sofia_handle_sip_i_state() > Channel sofia/internal/1000 at 192.168.16.63 entering state [ready] > 2009-02-18 16:59:59 [DEBUG] switch_ivr_play_say.c:968 > switch_ivr_play_file() Codec Activated L16 at 8000hz 1 channels 20ms > 2009-02-18 16:59:59 [DEBUG] switch_core_io.c:654 > switch_core_session_write_frame() sofia/internal/1000 at 192.168.16.63 > receive message [TRANSCODING_NECESSARY] > 2009-02-18 17:00:00 [INFO] switch_rtp.c:1422 rtp_common_read() Auto > Changing port from 127.0.0.1:49194 to 192.168.16.63:49194 > 2009-02-18 17:00:02 [DEBUG] switch_ivr_play_say.c:1258 > switch_ivr_play_file() done playing file > 2009-02-18 17:00:11 [WARNING] switch_scheduler.c:114 task_thread_loop() > Task was executed late by 2 seconds 1 heartbeat (core) > 2009-02-18 17:00:18 [DEBUG] switch_core_media_bug.c:284 > switch_core_media_bug_add() Attaching BUG to > sofia/internal/1000 at 192.168.16.63 > 2009-02-18 17:00:18 [DEBUG] switch_core_io.c:234 > switch_core_session_read_frame() sofia/internal/1000 at 192.168.16.63 receive > message [TRANSCODING_NECESSARY] > 2009-02-18 17:00:51 [WARNING] switch_scheduler.c:114 task_thread_loop() > Task was executed late by 20 seconds 1 heartbeat (core) > 2009-02-18 17:00:51 [DEBUG] switch_ivr_play_say.c:968 > switch_ivr_play_file() Codec Activated L16 at 8000hz 1 channels 20ms > 2009-02-18 17:00:58 [DEBUG] switch_ivr_play_say.c:1258 > switch_ivr_play_file() done playing file > 2009-02-18 17:00:58 [DEBUG] switch_core_io.c:234 > switch_core_session_read_frame() sofia/internal/1000 at 192.168.16.63 receive > message [TRANSCODING_NECESSARY] > 2009-02-18 17:01:00 [DEBUG] mod_pocketsphinx.c:386 > pocketsphinx_asr_get_results() Recognized: ????????????????, Score: 100 > 2009-02-18 17:01:00 [DEBUG] js_modules/SpeechTools.jm:150 console_log() > ----XML: > > ???????????????? > ???????????????? > > 2009-02-18 17:01:00 [DEBUG] js_modules/SpeechTools.jm:150 console_log() > ----Heard [????????????????] > 2009-02-18 17:01:00 [DEBUG] js_modules/SpeechTools.jm:150 console_log() > ----Hit score 100/40/70 > 2009-02-18 17:01:00 [INFO] js_modules/SpeechTools.jm:150 console_log() > ----???????????????? > 2009-02-18 17:01:00 [DEBUG] js_modules/SpeechTools.jm:365 console_log() > ----Testing [0] [0] ???????????????? =~ [Delivery:::Delivery] > 2009-02-18 17:01:00 [DEBUG] js_modules/SpeechTools.jm:365 console_log() > ----Testing [0] [1] ???????????????? =~ [Takeout:::Pickup] > 2009-02-18 17:01:00 [DEBUG] js_modules/SpeechTools.jm:365 console_log() > ----Testing [0] [2] ???????????????? =~ [Pickup:::Pickup] > 2009-02-18 17:01:02 [DEBUG] mod_pocketsphinx.c:342 > pocketsphinx_asr_resume() Manually Resuming > 2009-02-18 17:01:02 [DEBUG] switch_ivr_play_say.c:968 > switch_ivr_play_file() Codec Activated L16 at 8000hz 1 channels 20ms > 2009-02-18 17:01:02 [DEBUG] switch_ivr_play_say.c:1258 > switch_ivr_play_file() done playing file > 2009-02-18 17:01:02 [DEBUG] switch_core_io.c:234 > switch_core_session_read_frame() sofia/internal/1000 at 192.168.16.63 receive > message [TRANSCODING_NECESSARY] > 2009-02-18 17:01:03 [DEBUG] switch_ivr_play_say.c:968 > switch_ivr_play_file() Codec Activated L16 at 8000hz 1 channels 20ms > 2009-02-18 17:01:06 [DEBUG] switch_ivr_play_say.c:1258 > switch_ivr_play_file() done playing file > -- View this message in context: http://www.nabble.com/FreeSwitch---PcoketSphinx-Prompt-Playback---Recognition-Issue...-tp22080033p22080857.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From brian at freeswitch.org Wed Feb 18 07:31:32 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 18 Feb 2009 09:31:32 -0600 Subject: [Freeswitch-users] FreeSwitch - PcoketSphinx Prompt Playback & Recognition Issue... In-Reply-To: <22080033.post@talk.nabble.com> References: <22080033.post@talk.nabble.com> Message-ID: Please update to SVN Trunk and try again... what are the specs on your machine? I have been testing PocketSphinx the past couple of days on linux again and its fine. /b On Feb 18, 2009, at 9:09 AM, Moiz Chinoy wrote: > > Hi, > > I have downloaded and build the Freeswitch from > http://files.freeswitch.org/freeswitch-1.0.3RC1.tar.gz on Windows XP. > > Everything built successfully. > Then I configured PocketSphinx as described at > http://wiki.freeswitch.org/wiki/Mod_pocketsphinx. > The problem is that prompts (wave files) are not being played > properly while > testing the Pizza demo i.e it plays, stops then start playing again... > > I am using heaset for testing. > > Moreover, the recognition seems to be very poor. > > Any clue what might be the issue? > > Moiz Chinoy. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090218/1e214424/attachment-0002.html From moizchinoy at gmail.com Wed Feb 18 07:49:29 2009 From: moizchinoy at gmail.com (Moiz Chinoy) Date: Wed, 18 Feb 2009 07:49:29 -0800 (PST) Subject: [Freeswitch-users] FreeSwitch - PcoketSphinx Prompt Playback & Recognition Issue... In-Reply-To: References: <22080033.post@talk.nabble.com> Message-ID: <22081606.post@talk.nabble.com> System specs: - Intel Core 2 Duo - 2.00 GHZ CPU - 1 Gb Ram I will download the latest from here http://files.freeswitch.org/freeswitch-snapshot.tar.gz and will try it. Moiz Chinoy. Brian West-3 wrote: > > Please update to SVN Trunk and try again... what are the specs on your > machine? > > I have been testing PocketSphinx the past couple of days on linux > again and its fine. > > /b > > > On Feb 18, 2009, at 9:09 AM, Moiz Chinoy wrote: > >> >> Hi, >> >> I have downloaded and build the Freeswitch from >> http://files.freeswitch.org/freeswitch-1.0.3RC1.tar.gz on Windows XP. >> >> Everything built successfully. >> Then I configured PocketSphinx as described at >> http://wiki.freeswitch.org/wiki/Mod_pocketsphinx. >> The problem is that prompts (wave files) are not being played >> properly while >> testing the Pizza demo i.e it plays, stops then start playing again... >> >> I am using heaset for testing. >> >> Moreover, the recognition seems to be very poor. >> >> Any clue what might be the issue? >> >> Moiz Chinoy. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/FreeSwitch---PcoketSphinx-Prompt-Playback---Recognition-Issue...-tp22080033p22081606.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From brian at freeswitch.org Wed Feb 18 07:55:11 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 18 Feb 2009 09:55:11 -0600 Subject: [Freeswitch-users] FreeSwitch - PcoketSphinx Prompt Playback & Recognition Issue... In-Reply-To: <22081606.post@talk.nabble.com> References: <22080033.post@talk.nabble.com> <22081606.post@talk.nabble.com> Message-ID: <3FEB93CB-9E5B-40FA-8DF7-1CEC8A364732@freeswitch.org> Please go get an SVN client for windows... svn update vs downloading the tarball every day will save bandwidth. ;) /b On Feb 18, 2009, at 9:49 AM, Moiz Chinoy wrote: > > System specs: > > - Intel Core 2 Duo > - 2.00 GHZ CPU > - 1 Gb Ram > > I will download the latest from here > http://files.freeswitch.org/freeswitch-snapshot.tar.gz and will try > it. > > Moiz Chinoy. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090218/8a40aff2/attachment-0002.html From msc at freeswitch.org Wed Feb 18 07:58:10 2009 From: msc at freeswitch.org (Michael S Collins) Date: Wed, 18 Feb 2009 07:58:10 -0800 Subject: [Freeswitch-users] Anyone running FS from a Thumb Flash USB? In-Reply-To: <05B5115F398B44C28AFDB2DE0E1A9281@UVix> References: <9dc4a1670902170550r3071f65dn76112fdbebaeb6ba@mail.gmail.com><2d9149cd0902171543t1d1a5c9dqb23210ab8a2f3eeb@mail.gmail.com><87f2f3b90902171610r37679769h2b4f1c033e7b3e94@mail.gmail.com><2d9149cd0902171618s47640fb2vff3cc139b74c8f64@mail.gmail.com> <2d9149cd0902172309i1aa93fcej8fce68133028ee93@mail.gmail.com> <05B5115F398B44C28AFDB2DE0E1A9281@UVix> Message-ID: <45834C32-CF49-4973-9C7C-637D92C44EF4@freeswitch.org> Sent from my iPhone On Feb 18, 2009, at 3:55 AM, "UV" wrote: > Awesome work, Kristian! > And very much needed for the Freeswitch platform (to me, at least). > > A suggestion: if the FS team doesn't mind (after getting over the > naming > issue), it would be a good idea to put Kristian's latest blog entry > on the > FS Wiki. > Not a problem at all. We already link to Kristian's blog from our main page. I will give KK's new ISO a test drive and then put some directions on the wiki. -MC > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Kristian > Kielhofner > Sent: Wednesday, February 18, 2009 6:10 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Anyone running FS from a Thumb Flash > USB? > > Nik, > > Thanks but I'm not sure I want to take the credit (blame?) for > that! ;) > > On Tue, Feb 17, 2009 at 7:25 PM, Nik Middleton > wrote: >> Kristian, >> >> You're my hero, if I hadn't come across astlinux 3 years ago, I >> wouldn't >> be doing this stuff right now. Not too sure if that's a good thing >> though ;) >> > > -- > Kristian Kielhofner > http://blog.krisk.org > http://www.submityoursip.com > http://www.astlinux.org > http://www.star2star.com > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > No virus found in this incoming message. > Checked by AVG - www.avg.com > Version: 8.0.237 / Virus Database: 270.10.25/1956 - Release Date: > 02/17/09 > 07:07:00 > > No virus found in this outgoing message. > Checked by AVG - www.avg.com > Version: 8.0.237 / Virus Database: 270.10.25/1956 - Release Date: > 02/17/09 > 07:07:00 > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From gmaruzz at celliax.org Wed Feb 18 08:29:07 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Wed, 18 Feb 2009 17:29:07 +0100 Subject: [Freeswitch-users] Skypiax on OS X In-Reply-To: <9dc4a1670902180621u350ed70bt6aa92731790014d0@mail.gmail.com> References: <7b197bef0902180358s7c3ca246kf354beef2e72f4e6@mail.gmail.com> <9dc4a1670902180621u350ed70bt6aa92731790014d0@mail.gmail.com> Message-ID: <7b197bef0902180829h1f7c7c88od2c6f6d4a8f7aa29@mail.gmail.com> On Wed, Feb 18, 2009 at 3:21 PM, EdPimentl wrote: > OSX can be loaded on any new Intel machines.. > -E That's nice! How I can do it, I mean, in an easy way that will let me develop on it? It's just like installing a distro, or involves black magic? -gm On Wed, Feb 18, 2009 at 3:21 PM, EdPimentl wrote: > OSX can be loaded on any new Intel machines.. > -E > http://TwiTR.Me > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From javieraristizabal at gmail.com Wed Feb 18 09:35:39 2009 From: javieraristizabal at gmail.com (=?ISO-8859-1?Q?Javier_Aristiz=E1bal?=) Date: Wed, 18 Feb 2009 12:35:39 -0500 Subject: [Freeswitch-users] mod_fax and sending a fax In-Reply-To: <499C16C6.1000006@laposte.net> References: <49919822.3030101@laposte.net> <499C16C6.1000006@laposte.net> Message-ID: Hi Rod, i just play with rx_fax and work for me. I didn't work with tx_fax but i understand, that you need a .tiff file to send passthrough the rx_fax. Maybe that can help you regards javar -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090218/fa28bcb2/attachment-0002.html From red.rain.seven at gmail.com Wed Feb 18 10:09:33 2009 From: red.rain.seven at gmail.com (Henry Huang) Date: Wed, 18 Feb 2009 10:09:33 -0800 Subject: [Freeswitch-users] Deployment information and use cases In-Reply-To: <1234916455.16581.49.camel@raul-laptop> References: <1234916455.16581.49.camel@raul-laptop> Message-ID: <59ad9ca10902181009p7cbc0fd1s9860aacde7f3a30c@mail.gmail.com> bandwidth.com has a service called phonebooth which is developed upon freeswitch. On Tue, Feb 17, 2009 at 4:20 PM, Raul Fragoso wrote: > Hello FreeSWITCHERS, > > My company is currently creating a suite of applications which uses > FreeSWITCH as the back-end for an IP-PBX solution. We currently have a > prospect to have our first customer installation - a governmental > department. That is a tender to have an IP-PBX installation to connect > their four office branches, each one with about 300 users - which I am > sure FreeSWITCH is able to handle. Since this is an official tender, > it's part of their protocol to ask about real sites using the product. > > Having said that, would you mind sharing some information about your > experience with FreeSWITCH deployments ? > > No need to give many details, but a short summary with company name (if > possible), when it was deployed, server equipment, number of users, > number of concurrent calls, what kind of functions and services are used > and overall capacity of the system. > > I would really appreciate if you can share that information. And if you > guys agree (and explicitly manifest your agreement), I can compile the > information in the FreeSWITCH wiki under a "Use Cases" page so it can > serve as a common reference as well. > > Kind regards, > > Raul Fragoso > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Henry Huang UniC Solution - Communication Unified VoIP & Open Source software Consultant -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090218/a17c9b49/attachment-0002.html From pablosaro at gmail.com Wed Feb 18 10:19:37 2009 From: pablosaro at gmail.com (Pablo Hernan Saro) Date: Wed, 18 Feb 2009 16:19:37 -0200 Subject: [Freeswitch-users] Deployment information and use cases In-Reply-To: <59ad9ca10902181009p7cbc0fd1s9860aacde7f3a30c@mail.gmail.com> References: <1234916455.16581.49.camel@raul-laptop> <59ad9ca10902181009p7cbc0fd1s9860aacde7f3a30c@mail.gmail.com> Message-ID: <247f8100902181019m61ce803m5ed504d9b198dfb6@mail.gmail.com> Hi Raul, In my company (http://www.globant.com) we're using FreeSWITCH for high quality conference services, integrated with OpenSIPS (http://www.opensips.org) and Asterisk. Its performance is pretty good. Pablo On Wed, Feb 18, 2009 at 4:09 PM, Henry Huang wrote: > bandwidth.com has a service called phonebooth which is developed upon > freeswitch. > > > On Tue, Feb 17, 2009 at 4:20 PM, Raul Fragoso wrote: >> >> Hello FreeSWITCHERS, >> >> My company is currently creating a suite of applications which uses >> FreeSWITCH as the back-end for an IP-PBX solution. We currently have a >> prospect to have our first customer installation - a governmental >> department. That is a tender to have an IP-PBX installation to connect >> their four office branches, each one with about 300 users - which I am >> sure FreeSWITCH is able to handle. Since this is an official tender, >> it's part of their protocol to ask about real sites using the product. >> >> Having said that, would you mind sharing some information about your >> experience with FreeSWITCH deployments ? >> >> No need to give many details, but a short summary with company name (if >> possible), when it was deployed, server equipment, number of users, >> number of concurrent calls, what kind of functions and services are used >> and overall capacity of the system. >> >> I would really appreciate if you can share that information. And if you >> guys agree (and explicitly manifest your agreement), I can compile the >> information in the FreeSWITCH wiki under a "Use Cases" page so it can >> serve as a common reference as well. >> >> Kind regards, >> >> Raul Fragoso >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > -- > Henry Huang > UniC Solution - Communication Unified > VoIP & Open Source software Consultant > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From nik.middleton at noblesolutions.co.uk Wed Feb 18 11:26:48 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Wed, 18 Feb 2009 19:26:48 -0000 Subject: [Freeswitch-users] Originate and bridge with lua Message-ID: Hi Guys, It's not clear from the docs how I can originate a call from within an lua script This what works in js, Question. How do I instantiate a new session, do I use the execute to dial, and same for bridge? Regards, if (!first_session.ready()) var new_session = new Session(tdial-string); if (!first_session.ready()) { disp_call(DROP) exit(); new_session.answer(); if (new_session.ready()) { bridge(first_session, new_session); } -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090218/999668e0/attachment-0002.html From msc at freeswitch.org Wed Feb 18 11:41:07 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 18 Feb 2009 11:41:07 -0800 Subject: [Freeswitch-users] Originate and bridge with lua In-Reply-To: References: Message-ID: <87f2f3b90902181141q180680d5i4b54272eec4bf3de@mail.gmail.com> Nik, What are you building? I'm wondering if this is the correct approach for your application. You might be better off using the even socket and controlling your calls from a central point. -MC On Wed, Feb 18, 2009 at 11:26 AM, Nik Middleton wrote: > Hi Guys, > > > > It's not clear from the docs how I can originate a call from within an lua > script > > > > This what works in js, > > > > Question. How do I instantiate a new session, do I use the execute to dial, > and same for bridge? > > > > Regards, > > > > if (!first_session.ready()) > > > > var new_session = new Session(tdial-string); > > > > if (!first_session.ready()) { > > disp_call(DROP) > > exit(); > > > > > > > > new_session.answer(); > > > > if (new_session.ready()) { > > bridge(first_session, new_session); > > } > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From fax at virgintechnologies.com Wed Feb 18 11:46:04 2009 From: fax at virgintechnologies.com (Justin Miller) Date: Wed, 18 Feb 2009 19:46:04 +0000 Subject: [Freeswitch-users] Sending media streams to a media gateway Message-ID: Anyone else have any ideas on this? -----Original Message----- From: fax at virgintechnologies.com [mailto:fax at virgintechnologies.com] Sent: Tuesday, February 17, 2009 05:45 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Sending media streams to a media gateway I looked at that, but I think that will cause issues with the NAT traversal. Our phones will all be in external networks. I forgot to mention that. -----Original Message----- From: Anthony Minessale [mailto:anthony.minessale at gmail.com] Sent: Tuesday, February 17, 2009 05:34 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Sending media streams to a media gateway you could set the variable bypass_media to true before you call bridge that will negotiate a point to point media connection between the caller and callee On Tue, Feb 17, 2009 at 5:51 PM, wrote: I have Freeswitch running successfully with a fairly basic config. Nat traversal is working well on both the client and server side. I want to start running all RTP streams through a media gateway, and use Freeswitch for SIP registrations and signalling only. I believe that I need to have Freeswitch invite the SIP phone to send the RTP stream directly to the media gateway when a call starts. Where can I start with this? Does anyone have any example configs? Justin _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm mailto:MSN%3Aanthony_minessale at hotmail.com GTALK/JABBER/mailto:PAYPAL%3Aanthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference mailto:sip%3A888 at conference.freeswitch.org http://iax:guest at conference.freeswitch.org/888 mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090218/dc5ccdf2/attachment-0002.html From msc at freeswitch.org Wed Feb 18 11:48:05 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 18 Feb 2009 11:48:05 -0800 Subject: [Freeswitch-users] FreeSwitch - PcoketSphinx Prompt Playback & Recognition Issue... In-Reply-To: <3FEB93CB-9E5B-40FA-8DF7-1CEC8A364732@freeswitch.org> References: <22080033.post@talk.nabble.com> <22081606.post@talk.nabble.com> <3FEB93CB-9E5B-40FA-8DF7-1CEC8A364732@freeswitch.org> Message-ID: <87f2f3b90902181148r798c5ed5n4ab30877a8e06c00@mail.gmail.com> On Wed, Feb 18, 2009 at 7:55 AM, Brian West wrote: > Please go get an SVN client for windows... svn update vs downloading the > tarball every day will save bandwidth. ;) > /b Use this for windows: http://tortoisesvn.tigris.org/ -MC From anthony.minessale at gmail.com Wed Feb 18 11:51:05 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 18 Feb 2009 13:51:05 -0600 Subject: [Freeswitch-users] Sending media streams to a media gateway In-Reply-To: References: Message-ID: <191c3a030902181151o4c635d04qd7e1adebea390e2a@mail.gmail.com> In that case you would need a sip proxy in place to rewrite the packets for the nat issue. There's nothing else we can really do. We have a way to do what you want but you are using it under circumstances we can't control. On Wed, Feb 18, 2009 at 1:46 PM, Justin Miller wrote: > Anyone else have any ideas on this? > > -----Original Message----- > *From:* fax at virgintechnologies.com [mailto:fax at virgintechnologies.com] > *Sent:* Tuesday, February 17, 2009 05:45 PM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Sending media streams to a media gateway > > I looked at that, but I think that will cause issues with the NAT > traversal. Our phones will all be in external networks. I forgot to > mention that. > > -----Original Message----- > *From:* Anthony Minessale [mailto:anthony.minessale at gmail.com] > *Sent:* Tuesday, February 17, 2009 05:34 PM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Sending media streams to a media gateway > > you could set the variable bypass_media to true before you call bridge > > > > that will negotiate a point to point media connection between the caller > and callee > > > On Tue, Feb 17, 2009 at 5:51 PM, wrote: > >> I have Freeswitch running successfully with a fairly basic config. Nat >> traversal is working well on both the client and server side. I want to >> start running all RTP streams through a media gateway, and use Freeswitch >> for SIP registrations and signalling only. >> I believe that I need to have Freeswitch invite the SIP phone to send the >> RTP stream directly to the media gateway when a call starts. Where can I >> start with this? Does anyone have any example configs? >> Justin >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > mailto:MSN%3Aanthony_minessale at hotmail.com > GTALK/JABBER/mailto:PAYPAL%3Aanthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > mailto:sip%3A888 at conference.freeswitch.org > http://iax:guest at conference.freeswitch.org/888 > mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org > pstn:213-799-1400 > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090218/9c0e6d83/attachment-0002.html From nik.middleton at noblesolutions.co.uk Wed Feb 18 11:53:51 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Wed, 18 Feb 2009 19:53:51 -0000 Subject: [Freeswitch-users] Originate and bridge with lua In-Reply-To: <87f2f3b90902181141q180680d5i4b54272eec4bf3de@mail.gmail.com> References: <87f2f3b90902181141q180680d5i4b54272eec4bf3de@mail.gmail.com> Message-ID: I'm trying to build an emergency broadcasting solution. So I place a call, and have ivr in the lua script. But I also want to give them the option of speaking to someone. If they hit the option to speak to someone, while I can fire an event to originate a call, I'm not sure how I could bridge the 2 call legs. Regards, -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 18 February 2009 19:41 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Originate and bridge with lua Nik, What are you building? I'm wondering if this is the correct approach for your application. You might be better off using the even socket and controlling your calls from a central point. -MC On Wed, Feb 18, 2009 at 11:26 AM, Nik Middleton wrote: > Hi Guys, > > > > It's not clear from the docs how I can originate a call from within an lua > script > > > > This what works in js, > > > > Question. How do I instantiate a new session, do I use the execute to dial, > and same for bridge? > > > > Regards, > > > > if (!first_session.ready()) > > > > var new_session = new Session(tdial-string); > > > > if (!first_session.ready()) { > > disp_call(DROP) > > exit(); > > > > > > > > new_session.answer(); > > > > if (new_session.ready()) { > > bridge(first_session, new_session); > > } > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From msc at freeswitch.org Wed Feb 18 13:09:26 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 18 Feb 2009 13:09:26 -0800 Subject: [Freeswitch-users] Originate and bridge with lua In-Reply-To: References: <87f2f3b90902181141q180680d5i4b54272eec4bf3de@mail.gmail.com> Message-ID: <87f2f3b90902181309r38e0c480u5e3ba64b838f8985@mail.gmail.com> On Wed, Feb 18, 2009 at 11:53 AM, Nik Middleton wrote: > I'm trying to build an emergency broadcasting solution. > > So I place a call, and have ivr in the lua script. But I also want to > give them the option of speaking to someone. > > If they hit the option to speak to someone, while I can fire an event to > originate a call, I'm not sure how I could bridge the 2 call legs. > > Regards, So really, it's just an outbound IVR, no? Just for a specific purpose. I would recommend using the event socket and bgapi originate commands from a central program/script/controller thingy. Generate the calls and then drop them into a dialplan or script that controls them. I like to use the dialpan but it really does not matter. Using a script lets you make changes without doing a reloadxml command. In any case, your originate commands could be something like this: bgapi originate {myvar='myval',myvar2='myval2'}sofia/gateway/mygateway/user at domain 5555 Have extension 5555 do the gruntwork of confirming that you actually had a successful call, got a human on the line, etc. It can also handle failures that are not handled by the originate itself. (Depends on whether or not you ignore early media.) In any case, you've got a single dp entry that handles the mundane call handling. Then, if there is a human on the line, you can do something like this: Now you can write a plain Lua script that only has to handle the delivery of the message. You can handle a DTMF event and the callback function could use session:execute("bridge","agent") to connect the called party with your agent. Hope that helps. -MC From nik.middleton at noblesolutions.co.uk Wed Feb 18 13:27:54 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Wed, 18 Feb 2009 21:27:54 -0000 Subject: [Freeswitch-users] Originate and bridge with lua In-Reply-To: <87f2f3b90902181309r38e0c480u5e3ba64b838f8985@mail.gmail.com> References: <87f2f3b90902181141q180680d5i4b54272eec4bf3de@mail.gmail.com> <87f2f3b90902181309r38e0c480u5e3ba64b838f8985@mail.gmail.com> Message-ID: Hi Michael, Yes that's exactly what it boils down to, an outbound ivr. Everything is working perfectly, except the bridge to another number. Because of the nature of the beast the bridge needs to dial an external number (ie sofia/gateway/Mygateway/num) What I'm getting is: attempt to perform arithmetic on global 'sofia' (a nil value) regards -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 18 February 2009 21:09 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Originate and bridge with lua On Wed, Feb 18, 2009 at 11:53 AM, Nik Middleton wrote: > I'm trying to build an emergency broadcasting solution. > > So I place a call, and have ivr in the lua script. But I also want to > give them the option of speaking to someone. > > If they hit the option to speak to someone, while I can fire an event to > originate a call, I'm not sure how I could bridge the 2 call legs. > > Regards, So really, it's just an outbound IVR, no? Just for a specific purpose. I would recommend using the event socket and bgapi originate commands from a central program/script/controller thingy. Generate the calls and then drop them into a dialplan or script that controls them. I like to use the dialpan but it really does not matter. Using a script lets you make changes without doing a reloadxml command. In any case, your originate commands could be something like this: bgapi originate {myvar='myval',myvar2='myval2'}sofia/gateway/mygateway/user at domain 5555 Have extension 5555 do the gruntwork of confirming that you actually had a successful call, got a human on the line, etc. It can also handle failures that are not handled by the originate itself. (Depends on whether or not you ignore early media.) In any case, you've got a single dp entry that handles the mundane call handling. Then, if there is a human on the line, you can do something like this: Now you can write a plain Lua script that only has to handle the delivery of the message. You can handle a DTMF event and the callback function could use session:execute("bridge","agent") to connect the called party with your agent. Hope that helps. -MC _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From msc at freeswitch.org Wed Feb 18 13:43:18 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 18 Feb 2009 13:43:18 -0800 Subject: [Freeswitch-users] Originate and bridge with lua In-Reply-To: References: <87f2f3b90902181141q180680d5i4b54272eec4bf3de@mail.gmail.com> <87f2f3b90902181309r38e0c480u5e3ba64b838f8985@mail.gmail.com> Message-ID: <87f2f3b90902181343m6e807edfv6280cf01a70af9a8@mail.gmail.com> > Everything is working perfectly, except the bridge to another number. > Because of the nature of the beast the bridge needs to dial an external > number (ie sofia/gateway/Mygateway/num) What I'm getting is: > > attempt to perform arithmetic on global 'sofia' (a nil value) > Can you pastebin your Lua script? -MC From edpimentl at gmail.com Wed Feb 18 13:53:06 2009 From: edpimentl at gmail.com (EdPimentl) Date: Wed, 18 Feb 2009 16:53:06 -0500 Subject: [Freeswitch-users] Skypiax on OS X In-Reply-To: <7b197bef0902180829h1f7c7c88od2c6f6d4a8f7aa29@mail.gmail.com> References: <7b197bef0902180358s7c3ca246kf354beef2e72f4e6@mail.gmail.com> <9dc4a1670902180621u350ed70bt6aa92731790014d0@mail.gmail.com> <7b197bef0902180829h1f7c7c88od2c6f6d4a8f7aa29@mail.gmail.com> Message-ID: <9dc4a1670902181353s271e82a8u3f54b6e51a4c6f2d@mail.gmail.com> http://www.tech-recipes.com/rx/964/install_osx_tiger_on_intel_usb_drives_windows/ -E -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090218/43dac7bc/attachment-0002.html From nik.middleton at noblesolutions.co.uk Wed Feb 18 13:56:12 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Wed, 18 Feb 2009 21:56:12 -0000 Subject: [Freeswitch-users] Originate and bridge with lua In-Reply-To: <87f2f3b90902181343m6e807edfv6280cf01a70af9a8@mail.gmail.com> References: <87f2f3b90902181141q180680d5i4b54272eec4bf3de@mail.gmail.com><87f2f3b90902181309r38e0c480u5e3ba64b838f8985@mail.gmail.com> <87f2f3b90902181343m6e807edfv6280cf01a70af9a8@mail.gmail.com> Message-ID: Sorted now thanks, it needed to be in the format session:execute("bridge", "{params}sofia/gateway/Mygateway/number"); key change was '"' Now I've converted my js script to lua going to run some tests tomorrow. I sincerely hope it'll handle more than the 10 calls js would break at. Here's my current setup External prog generates bgapi calls via socket and calls originate with name of lua script also passed. Lua does IVR and then bridges where required. It also fires back an event to show result of call. Astererisk happily does around 200 calls, I'm hoping FS will do better or I've just been wasting my time. Is there a more efficient way of doing this? Regards, -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 18 February 2009 21:43 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Originate and bridge with lua > Everything is working perfectly, except the bridge to another number. > Because of the nature of the beast the bridge needs to dial an external > number (ie sofia/gateway/Mygateway/num) What I'm getting is: > > attempt to perform arithmetic on global 'sofia' (a nil value) > Can you pastebin your Lua script? -MC _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From brian at freeswitch.org Wed Feb 18 14:02:01 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 18 Feb 2009 16:02:01 -0600 Subject: [Freeswitch-users] Originate and bridge with lua In-Reply-To: References: <87f2f3b90902181141q180680d5i4b54272eec4bf3de@mail.gmail.com><87f2f3b90902181309r38e0c480u5e3ba64b838f8985@mail.gmail.com> <87f2f3b90902181343m6e807edfv6280cf01a70af9a8@mail.gmail.com> Message-ID: <3BEB0C42-9F44-497E-9CEB-F9049BD7E147@freeswitch.org> Learn C and write it all in C. /b On Feb 18, 2009, at 3:56 PM, Nik Middleton wrote: > Astererisk happily does around 200 calls, I'm hoping FS will do better > or I've just been wasting my time. Is there a more efficient way of > doing this? From anthony.minessale at gmail.com Wed Feb 18 14:06:33 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 18 Feb 2009 16:06:33 -0600 Subject: [Freeswitch-users] Originate and bridge with lua In-Reply-To: References: <87f2f3b90902181141q180680d5i4b54272eec4bf3de@mail.gmail.com> <87f2f3b90902181309r38e0c480u5e3ba64b838f8985@mail.gmail.com> <87f2f3b90902181343m6e807edfv6280cf01a70af9a8@mail.gmail.com> Message-ID: <191c3a030902181406g659a51d9j89a3dcf4502d9b40@mail.gmail.com> You want to make it even more efficient? when they press 1, session:execute("transfer", ""); Then, put an extension in your dialplan to match and do the bridge. Then you can exit the script and only run the script when you need it. Your problem with js was the same issue, you should have been doing something similar there too. BTW, If you make another comparison to asterisk comment, I will never answer another email from you again I don't have time for that crap. On Wed, Feb 18, 2009 at 3:56 PM, Nik Middleton < nik.middleton at noblesolutions.co.uk> wrote: > Sorted now thanks, it needed to be in the format > > session:execute("bridge", "{params}sofia/gateway/Mygateway/number"); > > key change was '"' > > Now I've converted my js script to lua going to run some tests tomorrow. > > I sincerely hope it'll handle more than the 10 calls js would break at. > > > Here's my current setup > > External prog generates bgapi calls via socket and calls originate with > name of lua script also passed. > > Lua does IVR and then bridges where required. It also fires back an > event to show result of call. > > Astererisk happily does around 200 calls, I'm hoping FS will do better > or I've just been wasting my time. Is there a more efficient way of > doing this? > > > Regards, > > > > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Michael Collins > Sent: 18 February 2009 21:43 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Originate and bridge with lua > > > Everything is working perfectly, except the bridge to another number. > > Because of the nature of the beast the bridge needs to dial an > external > > number (ie sofia/gateway/Mygateway/num) What I'm getting is: > > > > attempt to perform arithmetic on global 'sofia' (a nil value) > > > Can you pastebin your Lua script? > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090218/dcb3e7bb/attachment-0002.html From anthony.minessale at gmail.com Wed Feb 18 15:39:14 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 18 Feb 2009 17:39:14 -0600 Subject: [Freeswitch-users] Originate and bridge with lua In-Reply-To: <191c3a030902181406g659a51d9j89a3dcf4502d9b40@mail.gmail.com> References: <87f2f3b90902181141q180680d5i4b54272eec4bf3de@mail.gmail.com> <87f2f3b90902181309r38e0c480u5e3ba64b838f8985@mail.gmail.com> <87f2f3b90902181343m6e807edfv6280cf01a70af9a8@mail.gmail.com> <191c3a030902181406g659a51d9j89a3dcf4502d9b40@mail.gmail.com> Message-ID: <191c3a030902181539g637b01fke2582e830f602033@mail.gmail.com> i replied to your last private message and it was returned as undeliverable. overzealous spam server? Can you add my account to your whitelist? On Wed, Feb 18, 2009 at 4:06 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > You want to make it even more efficient? > when they press 1, > session:execute("transfer", ""); > > Then, put an extension in your dialplan to match and do the > bridge. > Then you can exit the script and only run the script when you need it. > > Your problem with js was the same issue, you should have been doing > something similar there too. > > BTW, > If you make another comparison to asterisk comment, I will never answer > another email from you again I don't have time for that crap. > > > > > > On Wed, Feb 18, 2009 at 3:56 PM, Nik Middleton < > nik.middleton at noblesolutions.co.uk> wrote: > >> Sorted now thanks, it needed to be in the format >> >> session:execute("bridge", "{params}sofia/gateway/Mygateway/number"); >> >> key change was '"' >> >> Now I've converted my js script to lua going to run some tests tomorrow. >> >> I sincerely hope it'll handle more than the 10 calls js would break at. >> >> >> Here's my current setup >> >> External prog generates bgapi calls via socket and calls originate with >> name of lua script also passed. >> >> Lua does IVR and then bridges where required. It also fires back an >> event to show result of call. >> >> Astererisk happily does around 200 calls, I'm hoping FS will do better >> or I've just been wasting my time. Is there a more efficient way of >> doing this? >> >> >> Regards, >> >> >> >> >> >> -----Original Message----- >> From: freeswitch-users-bounces at lists.freeswitch.org >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of >> Michael Collins >> Sent: 18 February 2009 21:43 >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] Originate and bridge with lua >> >> > Everything is working perfectly, except the bridge to another number. >> > Because of the nature of the beast the bridge needs to dial an >> external >> > number (ie sofia/gateway/Mygateway/num) What I'm getting is: >> > >> > attempt to perform arithmetic on global 'sofia' (a nil value) >> > >> Can you pastebin your Lua script? >> -MC >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090218/6f945f3a/attachment-0002.html From nik.middleton at noblesolutions.co.uk Wed Feb 18 15:56:53 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Wed, 18 Feb 2009 23:56:53 -0000 Subject: [Freeswitch-users] Originate and bridge with lua In-Reply-To: <191c3a030902181539g637b01fke2582e830f602033@mail.gmail.com> References: <87f2f3b90902181141q180680d5i4b54272eec4bf3de@mail.gmail.com><87f2f3b90902181309r38e0c480u5e3ba64b838f8985@mail.gmail.com><87f2f3b90902181343m6e807edfv6280cf01a70af9a8@mail.gmail.com><191c3a030902181406g659a51d9j89a3dcf4502d9b40@mail.gmail.com> <191c3a030902181539g637b01fke2582e830f602033@mail.gmail.com> Message-ID: Done Seems it had a spam score of 2 for some reason Regards, ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: 18 February 2009 23:39 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Originate and bridge with lua i replied to your last private message and it was returned as undeliverable. overzealous spam server? Can you add my account to your whitelist? On Wed, Feb 18, 2009 at 4:06 PM, Anthony Minessale wrote: You want to make it even more efficient? when they press 1, session:execute("transfer", ""); Then, put an extension in your dialplan to match and do the bridge. Then you can exit the script and only run the script when you need it. Your problem with js was the same issue, you should have been doing something similar there too. BTW, If you make another comparison to asterisk comment, I will never answer another email from you again I don't have time for that crap. On Wed, Feb 18, 2009 at 3:56 PM, Nik Middleton wrote: Sorted now thanks, it needed to be in the format session:execute("bridge", "{params}sofia/gateway/Mygateway/number"); key change was '"' Now I've converted my js script to lua going to run some tests tomorrow. I sincerely hope it'll handle more than the 10 calls js would break at. Here's my current setup External prog generates bgapi calls via socket and calls originate with name of lua script also passed. Lua does IVR and then bridges where required. It also fires back an event to show result of call. Astererisk happily does around 200 calls, I'm hoping FS will do better or I've just been wasting my time. Is there a more efficient way of doing this? Regards, -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 18 February 2009 21:43 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Originate and bridge with lua > Everything is working perfectly, except the bridge to another number. > Because of the nature of the beast the bridge needs to dial an external > number (ie sofia/gateway/Mygateway/num) What I'm getting is: > > attempt to perform arithmetic on global 'sofia' (a nil value) > Can you pastebin your Lua script? -MC _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090218/cfc7b385/attachment-0002.html From philip.patterson at gmail.com Wed Feb 18 18:00:07 2009 From: philip.patterson at gmail.com (Philip Patterson) Date: Wed, 18 Feb 2009 22:00:07 -0400 Subject: [Freeswitch-users] Missing file for 1.0.3 Message-ID: Hi All. Have a fresh server and going to install FS on it. Went to the download page (http://wiki.freeswitch.org/wiki/Installation_Guide) and tried to download the "Phoenix" build, which is supposed to be found at http://files.freeswitch.org/freeswitch-1.0.3.tar.gz but that file is nowhere to be found. Did the Wiki get updated before the file was uploaded, or is there something else going on? Philip -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090218/5a36e6cc/attachment-0002.html From carlos.talbot at gmail.com Wed Feb 18 18:53:29 2009 From: carlos.talbot at gmail.com (Carlos Talbot) Date: Wed, 18 Feb 2009 20:53:29 -0600 Subject: [Freeswitch-users] Skypiax (Skype Network endpoint) for FreeSWITCH In-Reply-To: <7b197bef0902131015k456dee87x8f0f1b7059f1b49c@mail.gmail.com> References: <7b197bef0902131015k456dee87x8f0f1b7059f1b49c@mail.gmail.com> Message-ID: <5800526b0902181853h32de5535l6c0f9d7067a69cf0@mail.gmail.com> Giovannia, great work on mod_skypiax. I've been testing it under Windows and it sounds great including PSTN calls. I plan to include it as part of the Windows MSI build. One question I have, is ringback suppose to work with mod_skypiax? Whenever I dial a number I get a few seconds of dead air before the call is answered. I've tried adding ringback and transfer_ringback into the dialplan just before the bridge command but no go. Am I missing something? Thanks. regards, Carlos -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090218/f4740bd6/attachment-0002.html From brian at freeswitch.org Wed Feb 18 18:55:50 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 18 Feb 2009 20:55:50 -0600 Subject: [Freeswitch-users] Missing file for 1.0.3 In-Reply-To: References: Message-ID: Looks like someone jumped the gun... just get SVN trunk... we are in the process of release right now. /b On Feb 18, 2009, at 8:00 PM, Philip Patterson wrote: > Hi All. > > Have a fresh server and going to install FS on it. Went to the > download page (http://wiki.freeswitch.org/wiki/Installation_Guide) > and tried to download the "Phoenix" build, which is supposed to be > found at http://files.freeswitch.org/freeswitch-1.0.3.tar.gz but > that file is nowhere to be found. Did the Wiki get updated before > the file was uploaded, or is there something else going on? > > Philip -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090218/54efa178/attachment-0002.html From brian at freeswitch.org Wed Feb 18 18:57:37 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 18 Feb 2009 20:57:37 -0600 Subject: [Freeswitch-users] Skypiax (Skype Network endpoint) for FreeSWITCH In-Reply-To: <5800526b0902181853h32de5535l6c0f9d7067a69cf0@mail.gmail.com> References: <7b197bef0902131015k456dee87x8f0f1b7059f1b49c@mail.gmail.com> <5800526b0902181853h32de5535l6c0f9d7067a69cf0@mail.gmail.com> Message-ID: <9D0D74A5-7E6C-41FD-9A52-0430101C909E@freeswitch.org> Thats one I think Anthm will need to chime in on... maybe skypiax isn't sending the right indications to cause the core to trigger the ringback. /b On Feb 18, 2009, at 8:53 PM, Carlos Talbot wrote: > Giovannia, > > great work on mod_skypiax. I've been testing it under Windows and it > sounds great including PSTN calls. I plan to include it as part of > the Windows MSI build. > > One question I have, is ringback suppose to work with mod_skypiax? > Whenever I dial a number I get a few seconds of dead air before the > call is answered. I've tried adding ringback and transfer_ringback > into the dialplan just before the bridge command but no go. Am I > missing something? Thanks. > > regards, > > Carlos > > From raul at etellicom.com Wed Feb 18 20:13:13 2009 From: raul at etellicom.com (Raul Fragoso) Date: Thu, 19 Feb 2009 01:13:13 -0300 Subject: [Freeswitch-users] Deployment information and use cases In-Reply-To: <247f8100902181019m61ce803m5ed504d9b198dfb6@mail.gmail.com> References: <1234916455.16581.49.camel@raul-laptop> <59ad9ca10902181009p7cbc0fd1s9860aacde7f3a30c@mail.gmail.com> <247f8100902181019m61ce803m5ed504d9b198dfb6@mail.gmail.com> Message-ID: <1235016793.22050.0.camel@raul-laptop> Thanks guys, this is very useful information. Anyone else willing to share your experience ? Regards, Raul On Wed, 2009-02-18 at 16:19 -0200, Pablo Hernan Saro wrote: > Hi Raul, > > In my company (http://www.globant.com) we're using FreeSWITCH for high > quality conference services, integrated with OpenSIPS > (http://www.opensips.org) and Asterisk. Its performance is pretty > good. > > Pablo > > On Wed, Feb 18, 2009 at 4:09 PM, Henry Huang wrote: > > bandwidth.com has a service called phonebooth which is developed upon > > freeswitch. > > > > > > On Tue, Feb 17, 2009 at 4:20 PM, Raul Fragoso wrote: > >> > >> Hello FreeSWITCHERS, > >> > >> My company is currently creating a suite of applications which uses > >> FreeSWITCH as the back-end for an IP-PBX solution. We currently have a > >> prospect to have our first customer installation - a governmental > >> department. That is a tender to have an IP-PBX installation to connect > >> their four office branches, each one with about 300 users - which I am > >> sure FreeSWITCH is able to handle. Since this is an official tender, > >> it's part of their protocol to ask about real sites using the product. > >> > >> Having said that, would you mind sharing some information about your > >> experience with FreeSWITCH deployments ? > >> > >> No need to give many details, but a short summary with company name (if > >> possible), when it was deployed, server equipment, number of users, > >> number of concurrent calls, what kind of functions and services are used > >> and overall capacity of the system. > >> > >> I would really appreciate if you can share that information. And if you > >> guys agree (and explicitly manifest your agreement), I can compile the > >> information in the FreeSWITCH wiki under a "Use Cases" page so it can > >> serve as a common reference as well. > >> > >> Kind regards, > >> > >> Raul Fragoso > >> > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > > > -- > > Henry Huang > > UniC Solution - Communication Unified > > VoIP & Open Source software Consultant > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From msc at freeswitch.org Wed Feb 18 20:37:49 2009 From: msc at freeswitch.org (Michael S Collins) Date: Wed, 18 Feb 2009 20:37:49 -0800 Subject: [Freeswitch-users] Missing file for 1.0.3 Message-ID: <9384F444-E628-4769-A507-3693C06BB985@freeswitch.org> Sent from my iPhone On Feb 18, 2009, at 6:00 PM, Philip Patterson wrote: > Hi All. > > Have a fresh server and going to install FS on it. Went to the > download page (http://wiki.freeswitch.org/wiki/Installation_Guide) > and tried to download the "Phoenix" build, which is supposed to be > found at http://files.freeswitch.org/freeswitch-1.0.3.tar.gz but > that file is nowhere to be found. Did the Wiki get updated before > the file was uploaded, or is there something else going on? Oops, my bad. That's exactly what happened. The file is actually 1.0.3RC1.tar.gz, although 1.0.3 should hit the server in the next day or so. Stay tuned! -MC > > > Philip > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090218/c914b016/attachment-0002.html From msc at freeswitch.org Wed Feb 18 21:39:29 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 18 Feb 2009 21:39:29 -0800 Subject: [Freeswitch-users] Skypiax (Skype Network endpoint) for FreeSWITCH In-Reply-To: <9D0D74A5-7E6C-41FD-9A52-0430101C909E@freeswitch.org> References: <7b197bef0902131015k456dee87x8f0f1b7059f1b49c@mail.gmail.com> <5800526b0902181853h32de5535l6c0f9d7067a69cf0@mail.gmail.com> <9D0D74A5-7E6C-41FD-9A52-0430101C909E@freeswitch.org> Message-ID: <87f2f3b90902182139q66393955r1cd6ef12bb496291@mail.gmail.com> On Wed, Feb 18, 2009 at 6:57 PM, Brian West wrote: > Thats one I think Anthm will need to chime in on... maybe skypiax > isn't sending the right indications to cause the core to trigger the > ringback. > > /b > Out of curiosity, you might try this trick: See also: http://wiki.freeswitch.org/wiki/Channel_Variables#instant_ringback I'm curious to know how that works with your setup. -MC From carlos.talbot at gmail.com Wed Feb 18 22:00:00 2009 From: carlos.talbot at gmail.com (Carlos Talbot) Date: Thu, 19 Feb 2009 00:00:00 -0600 Subject: [Freeswitch-users] Skypiax (Skype Network endpoint) for FreeSWITCH In-Reply-To: <87f2f3b90902182139q66393955r1cd6ef12bb496291@mail.gmail.com> References: <7b197bef0902131015k456dee87x8f0f1b7059f1b49c@mail.gmail.com> <5800526b0902181853h32de5535l6c0f9d7067a69cf0@mail.gmail.com> <9D0D74A5-7E6C-41FD-9A52-0430101C909E@freeswitch.org> <87f2f3b90902182139q66393955r1cd6ef12bb496291@mail.gmail.com> Message-ID: <5800526b0902182200l4da4be2dyfebfc3542aef67b4@mail.gmail.com> That did it! I had to add both lines below in order for it to work: Now, suppose I call a number that's busy...do I hear a ringback followed by a busy signal? On Wed, Feb 18, 2009 at 11:39 PM, Michael Collins wrote: > On Wed, Feb 18, 2009 at 6:57 PM, Brian West wrote: > > Thats one I think Anthm will need to chime in on... maybe skypiax > > isn't sending the right indications to cause the core to trigger the > > ringback. > > > > /b > > > Out of curiosity, you might try this trick: > > See also: > http://wiki.freeswitch.org/wiki/Channel_Variables#instant_ringback > > I'm curious to know how that works with your setup. > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090219/b081dea4/attachment-0002.html From gmaruzz at celliax.org Wed Feb 18 22:07:36 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Thu, 19 Feb 2009 07:07:36 +0100 Subject: [Freeswitch-users] Skypiax (Skype Network endpoint) for FreeSWITCH In-Reply-To: <87f2f3b90902182139q66393955r1cd6ef12bb496291@mail.gmail.com> References: <7b197bef0902131015k456dee87x8f0f1b7059f1b49c@mail.gmail.com> <5800526b0902181853h32de5535l6c0f9d7067a69cf0@mail.gmail.com> <9D0D74A5-7E6C-41FD-9A52-0430101C909E@freeswitch.org> <87f2f3b90902182139q66393955r1cd6ef12bb496291@mail.gmail.com> Message-ID: <7b197bef0902182207q2b8a1b3bj1abac4aad768aa7@mail.gmail.com> Carlos, Maybe the solution Michael is suggesting could work. For sure you are not missing anything' Brian is right: rimgback and early media are to be added to skypiax. They're on the TODO section of the wiki :-) I'll be all day at a customer's premise, I'll add it this evening, late afternoon for you. Would be *very* nice to have skypiax in MSI, thank you! On 2/19/09, Michael Collins wrote: > On Wed, Feb 18, 2009 at 6:57 PM, Brian West wrote: >> Thats one I think Anthm will need to chime in on... maybe skypiax >> isn't sending the right indications to cause the core to trigger the >> ringback. >> >> /b >> > Out of curiosity, you might try this trick: > > See also: > http://wiki.freeswitch.org/wiki/Channel_Variables#instant_ringback > > I'm curious to know how that works with your setup. > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Sent from my mobile device Sincerely, Giovanni Maruzzelli ========================================= www.celliax.org via Pierlombardo 9, 20135 Milano Italy gmaruzz at celliax dot org Cell : +39-347-2665618 Fax : +39-02-87390039 From brian at freeswitch.org Wed Feb 18 22:42:37 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 19 Feb 2009 00:42:37 -0600 Subject: [Freeswitch-users] Missing file for 1.0.3 In-Reply-To: <9384F444-E628-4769-A507-3693C06BB985@freeswitch.org> References: <9384F444-E628-4769-A507-3693C06BB985@freeswitch.org> Message-ID: <72414A49-FB9B-43A1-99AB-5937D8721F25@freeswitch.org> go try now! ;) /b From brian at freeswitch.org Wed Feb 18 22:52:15 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 19 Feb 2009 00:52:15 -0600 Subject: [Freeswitch-users] Skypiax (Skype Network endpoint) for FreeSWITCH In-Reply-To: <7b197bef0902182207q2b8a1b3bj1abac4aad768aa7@mail.gmail.com> References: <7b197bef0902131015k456dee87x8f0f1b7059f1b49c@mail.gmail.com> <5800526b0902181853h32de5535l6c0f9d7067a69cf0@mail.gmail.com> <9D0D74A5-7E6C-41FD-9A52-0430101C909E@freeswitch.org> <87f2f3b90902182139q66393955r1cd6ef12bb496291@mail.gmail.com> <7b197bef0902182207q2b8a1b3bj1abac4aad768aa7@mail.gmail.com> Message-ID: <0E2F27BF-565D-42F5-B453-6A34D0EBEEB2@freeswitch.org> It has to be in trunk to be in the MSI... I don't want to cause confusion ... Now that 1.0.3 is tagged we can put it in trunk? /b On Feb 19, 2009, at 12:07 AM, Giovanni Maruzzelli wrote: > Would be *very* nice to have skypiax in MSI, thank you! From moizchinoy at gmail.com Wed Feb 18 23:29:41 2009 From: moizchinoy at gmail.com (Moiz Chinoy) Date: Thu, 19 Feb 2009 11:29:41 +0400 Subject: [Freeswitch-users] FreeSwitch - PcoketSphinx Prompt Playback & Recognition Issue... In-Reply-To: <87f2f3b90902181148r798c5ed5n4ab30877a8e06c00@mail.gmail.com> References: <22080033.post@talk.nabble.com> <22081606.post@talk.nabble.com> <3FEB93CB-9E5B-40FA-8DF7-1CEC8A364732@freeswitch.org> <87f2f3b90902181148r798c5ed5n4ab30877a8e06c00@mail.gmail.com> Message-ID: <29b888f80902182329o578a66d5x66454355d7575f48@mail.gmail.com> Thanks for your help.... I have downloaded the latest build and tried... Often in the log I see cryptic characters in the XML part returned by ASR. Is it silence or nose?? If yes is there any way we can control it? Prompt playback problem is still there... So far I am only able to get TAKEOUT and YES recognized and then the application crashes with Windows error: AppName: freeswitch.exe AppVer: 0.0.0.0 ModName: sphinxbase.dll ModVer: 0.0.0.0 Offset: 00053791 Below is log snippet: o=FreeSWITCH 1235002217 1235002218 IN IP4 192.168.16.63 s=FreeSWITCH c=IN IP4 192.168.16.63 t=0 0 m=audio 25190 RTP/AVP 3 101 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 2009-02-19 11:10:07 [DEBUG] switch_core_session.c:523 switch_core_session_perform_receive_message() Send signal sofia/internal/1000 at 192.168.16.63 [BREAK] 2009-02-19 11:10:07 [NOTICE] mod_spidermonkey.c:2041 session_answer() Channel [sofia/internal/1000 at 192.168.16.63] has been answered 2009-02-19 11:10:07 [DEBUG] switch_channel.c:179 switch_channel_audio_sync() sofia/internal/1000 at 192.168.16.63 receive message [AUDIO_SYNC] 2009-02-19 11:10:07 [DEBUG] switch_ivr_play_say.c:989 switch_ivr_play_file() Codec Activated L16 at 8000hz 1 channels 20ms 2009-02-19 11:10:07 [DEBUG] switch_core_io.c:652 switch_core_session_write_frame() sofia/internal/1000 at 192.168.16.63 receive message [TRANSCODING_NECESSARY] 2009-02-19 11:10:07 [DEBUG] sofia.c:2725 sofia_handle_sip_i_state() Channel sofia/internal/1000 at 192.168.16.63 entering state [completed] 2009-02-19 11:10:07 [DEBUG] sofia.c:2725 sofia_handle_sip_i_state() Channel sofia/internal/1000 at 192.168.16.63 entering state [ready] 2009-02-19 11:10:07 [INFO] switch_rtp.c:1422 rtp_common_read() Auto Changing port from 127.0.0.1:49166 to 192.168.16.63:49166 2009-02-19 11:10:09 [DEBUG] switch_ivr_play_say.c:1279 switch_ivr_play_file() done playing file 2009-02-19 11:10:12 [DEBUG] switch_core_media_bug.c:297 switch_core_media_bug_add() Attaching BUG to sofia/internal/1000 at 192.168.16.63 2009-02-19 11:10:12 [DEBUG] switch_core_io.c:234 switch_core_session_read_frame() sofia/internal/1000 at 192.168.16.63 receive message [TRANSCODING_NECESSARY] 2009-02-19 11:10:12 [DEBUG] switch_ivr_play_say.c:989 switch_ivr_play_file() Codec Activated L16 at 8000hz 1 channels 20ms 2009-02-19 11:10:15 [DEBUG] switch_ivr_play_say.c:1279 switch_ivr_play_file() done playing file 2009-02-19 11:10:15 [DEBUG] switch_core_io.c:234 switch_core_session_read_frame() sofia/internal/1000 at 192.168.16.63 receive message [TRANSCODING_NECESSARY] 2009-02-19 11:10:18 [DEBUG] mod_pocketsphinx.c:387 pocketsphinx_asr_get_results() Recognized: TAKEOUT, Score: 62 2009-02-19 11:10:18 [DEBUG] js_modules/SpeechTools.jm:150 console_log() ----XML: TAKEOUT TAKEOUT 2009-02-19 11:10:18 [DEBUG] js_modules/SpeechTools.jm:150 console_log() ----Heard [TAKEOUT] 2009-02-19 11:10:18 [DEBUG] js_modules/SpeechTools.jm:150 console_log() ----Hit score 62/40/70 2009-02-19 11:10:18 [INFO] js_modules/SpeechTools.jm:150 console_log() ----TAKEOUT 2009-02-19 11:10:18 [DEBUG] js_modules/SpeechTools.jm:365 console_log() ----We need to confirm this one 2009-02-19 11:10:18 [DEBUG] js_modules/SpeechTools.jm:365 console_log() ----Testing [1] [0] TAKEOUT =~ [Delivery:::Delivery] 2009-02-19 11:10:18 [DEBUG] js_modules/SpeechTools.jm:365 console_log() ----Testing [1] [1] TAKEOUT =~ [Takeout:::Pickup] 2009-02-19 11:10:18 [DEBUG] js_modules/SpeechTools.jm:365 console_log() ----Adding Pickup 2009-02-19 11:10:18 [DEBUG] js_modules/SpeechTools.jm:365 console_log() ----Testing [1] [2] TAKEOUT =~ [Pickup:::Pickup] 2009-02-19 11:10:19 [DEBUG] js_modules/SpeechTools.jm:109 console_log() Unloading grammar pizza_order 2009-02-19 11:10:21 [DEBUG] switch_ivr_play_say.c:989 switch_ivr_play_file() Codec Activated L16 at 8000hz 1 channels 20ms 2009-02-19 11:10:22 [DEBUG] mod_pocketsphinx.c:387 pocketsphinx_asr_get_results() Recognized: ????, Score: 100 2009-02-19 11:10:22 [DEBUG] js_modules/SpeechTools.jm:150 console_log() ----XML: ???? ???? 2009-02-19 11:10:22 [DEBUG] js_modules/SpeechTools.jm:150 console_log() ----Heard [????] 2009-02-19 11:10:22 [DEBUG] js_modules/SpeechTools.jm:150 console_log() ----Hit score 100/40/20 2009-02-19 11:10:22 [INFO] js_modules/SpeechTools.jm:150 console_log() ----???? 2009-02-19 11:10:22 [DEBUG] switch_ivr_play_say.c:1279 switch_ivr_play_file() done playing file 2009-02-19 11:10:22 [DEBUG] js_modules/SpeechTools.jm:365 console_log() ----Testing [0] [0] ???? =~ [^yes:::yes] 2009-02-19 11:10:22 [DEBUG] js_modules/SpeechTools.jm:365 console_log() ----Testing [0] [1] ???? =~ [^correct:::yes] 2009-02-19 11:10:22 [DEBUG] js_modules/SpeechTools.jm:365 console_log() ----Testing [0] [2] ???? =~ [^no:::no] 2009-02-19 11:10:22 [DEBUG] switch_core_io.c:234 switch_core_session_read_frame() sofia/internal/1000 at 192.168.16.63 receive message [TRANSCODING_NECESSARY] 2009-02-19 11:10:23 [DEBUG] mod_pocketsphinx.c:343 pocketsphinx_asr_resume() Manually Resuming 2009-02-19 11:10:23 [DEBUG] switch_ivr_play_say.c:989 switch_ivr_play_file() Codec Activated L16 at 8000hz 1 channels 20ms 2009-02-19 11:10:25 [DEBUG] switch_ivr_play_say.c:1279 switch_ivr_play_file() done playing file 2009-02-19 11:10:25 [DEBUG] switch_core_io.c:234 switch_core_session_read_frame() sofia/internal/1000 at 192.168.16.63 receive message [TRANSCODING_NECESSARY] 2009-02-19 11:10:26 [DEBUG] switch_ivr_play_say.c:989 switch_ivr_play_file() Codec Activated L16 at 8000hz 1 channels 20ms 2009-02-19 11:10:29 [DEBUG] switch_ivr_play_say.c:1279 switch_ivr_play_file() done playing file 2009-02-19 11:10:29 [DEBUG] switch_core_io.c:234 switch_core_session_read_frame() sofia/internal/1000 at 192.168.16.63 receive message [TRANSCODING_NECESSARY] 2009-02-19 11:10:31 [DEBUG] switch_ivr_play_say.c:989 switch_ivr_play_file() Codec Activated L16 at 8000hz 1 channels 20ms 2009-02-19 11:10:32 [DEBUG] mod_pocketsphinx.c:387 pocketsphinx_asr_get_results() Recognized: YES, Score: 100 2009-02-19 11:10:32 [DEBUG] js_modules/SpeechTools.jm:150 console_log() ----XML: YES YES 2009-02-19 11:10:32 [DEBUG] js_modules/SpeechTools.jm:150 console_log() ----Heard [YES] 2009-02-19 11:10:32 [DEBUG] js_modules/SpeechTools.jm:150 console_log() ----Hit score 100/40/20 2009-02-19 11:10:32 [INFO] js_modules/SpeechTools.jm:150 console_log() ----YES 2009-02-19 11:10:32 [DEBUG] switch_ivr_play_say.c:1279 switch_ivr_play_file() done playing file 2009-02-19 11:10:32 [DEBUG] js_modules/SpeechTools.jm:365 console_log() ----Testing [0] [0] YES =~ [^yes:::yes] 2009-02-19 11:10:32 [DEBUG] js_modules/SpeechTools.jm:365 console_log() ----Adding yes 2009-02-19 11:10:32 [DEBUG] js_modules/SpeechTools.jm:365 console_log() ----Testing [0] [1] YES =~ [^correct:::yes] 2009-02-19 11:10:32 [DEBUG] js_modules/SpeechTools.jm:365 console_log() ----Testing [0] [2] YES =~ [^no:::no] 2009-02-19 11:10:32 [DEBUG] switch_core_io.c:234 switch_core_session_read_frame() sofia/internal/1000 at 192.168.16.63 receive message [TRANSCODING_NECESSARY] 2009-02-19 11:10:33 [DEBUG] js_modules/SpeechTools.jm:109 console_log() Unloading grammar pizza_yesno On Wed, Feb 18, 2009 at 11:48 PM, Michael Collins wrote: > On Wed, Feb 18, 2009 at 7:55 AM, Brian West wrote: >> Please go get an SVN client for windows... svn update vs downloading the >> tarball every day will save bandwidth. ;) >> /b > > Use this for windows: > http://tortoisesvn.tigris.org/ > > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Regards, Moiz Chinoy. Mobile: 055-8527492 From gmaruzz at celliax.org Wed Feb 18 23:35:37 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Thu, 19 Feb 2009 08:35:37 +0100 Subject: [Freeswitch-users] Skypiax (Skype Network endpoint) for FreeSWITCH In-Reply-To: <0E2F27BF-565D-42F5-B453-6A34D0EBEEB2@freeswitch.org> References: <7b197bef0902131015k456dee87x8f0f1b7059f1b49c@mail.gmail.com> <5800526b0902181853h32de5535l6c0f9d7067a69cf0@mail.gmail.com> <9D0D74A5-7E6C-41FD-9A52-0430101C909E@freeswitch.org> <87f2f3b90902182139q66393955r1cd6ef12bb496291@mail.gmail.com> <7b197bef0902182207q2b8a1b3bj1abac4aad768aa7@mail.gmail.com> <0E2F27BF-565D-42F5-B453-6A34D0EBEEB2@freeswitch.org> Message-ID: <7b197bef0902182335h6faa1753w18fd59643039b01a@mail.gmail.com> Yes, I'd like it in trunk. There are still some rough edges, but I'll iron out in the trunk. Sincerely, Giovanni Maruzzelli ========================================= www.celliax.org via Pierlombardo 9, 20135 Milano Italy gmaruzz at celliax dot org Cell : +39-347-2665618 Fax : +39-02-87390039 On Thu, Feb 19, 2009 at 7:52 AM, Brian West wrote: > It has to be in trunk to be in the MSI... I don't want to cause > confusion ... Now that 1.0.3 is tagged we can put it in trunk? > > /b > > On Feb 19, 2009, at 12:07 AM, Giovanni Maruzzelli wrote: > >> Would be *very* nice to have skypiax in MSI, thank you! > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From brian at freeswitch.org Wed Feb 18 23:43:54 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 19 Feb 2009 01:43:54 -0600 Subject: [Freeswitch-users] FreeSwitch - PcoketSphinx Prompt Playback & Recognition Issue... In-Reply-To: <29b888f80902182329o578a66d5x66454355d7575f48@mail.gmail.com> References: <22080033.post@talk.nabble.com> <22081606.post@talk.nabble.com> <3FEB93CB-9E5B-40FA-8DF7-1CEC8A364732@freeswitch.org> <87f2f3b90902181148r798c5ed5n4ab30877a8e06c00@mail.gmail.com> <29b888f80902182329o578a66d5x66454355d7575f48@mail.gmail.com> Message-ID: <590097F8-ABB3-49B6-8574-E3A3B7ADD134@freeswitch.org> No clue, I haven't ever seen that behavior on linux. Maybe you can try to narrow it down and report it on jira.. chances are its a bug in the pocketsphinx libs. /b On Feb 19, 2009, at 1:29 AM, Moiz Chinoy wrote: > > Often in the log I see cryptic characters in the XML part returned by > ASR. Is it silence or nose?? > If yes is there any way we can control it? From gcd at i.ph Thu Feb 19 00:59:06 2009 From: gcd at i.ph (Nandy Dagondon) Date: Thu, 19 Feb 2009 16:59:06 +0800 Subject: [Freeswitch-users] Default IVR action Message-ID: <7d0bfd8c0902190059y8262895h3a470ccfa4f6c602@mail.gmail.com> hi everybody, i'm looking for a default action in an IVR if the caller doesn't press any key. for example, the caller will be transferred to the operator (or fifo) if no key is received after, let's say 5 seconds. is this available in the IVR? pls show a sample. tks, -nandy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090219/2a081326/attachment-0002.html From moizchinoy at gmail.com Thu Feb 19 01:29:06 2009 From: moizchinoy at gmail.com (Moiz Chinoy) Date: Thu, 19 Feb 2009 13:29:06 +0400 Subject: [Freeswitch-users] FreeSwitch - PcoketSphinx Prompt Playback & Recognition Issue... In-Reply-To: <590097F8-ABB3-49B6-8574-E3A3B7ADD134@freeswitch.org> References: <22080033.post@talk.nabble.com> <22081606.post@talk.nabble.com> <3FEB93CB-9E5B-40FA-8DF7-1CEC8A364732@freeswitch.org> <87f2f3b90902181148r798c5ed5n4ab30877a8e06c00@mail.gmail.com> <29b888f80902182329o578a66d5x66454355d7575f48@mail.gmail.com> <590097F8-ABB3-49B6-8574-E3A3B7ADD134@freeswitch.org> Message-ID: <29b888f80902190129k377cd710v1fe525659ac68fcb@mail.gmail.com> Can anyone please explain the following fields from pocketsphinx.conf.xml: Moiz Chinoy. On Thu, Feb 19, 2009 at 11:43 AM, Brian West wrote: > No clue, I haven't ever seen that behavior on linux. Maybe you can > try to narrow it down and report it on jira.. chances are its a bug in > the pocketsphinx libs. > > /b > > On Feb 19, 2009, at 1:29 AM, Moiz Chinoy wrote: > >> >> Often in the log I see cryptic characters in the XML part returned by >> ASR. Is it silence or nose?? >> If yes is there any way we can control it? > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Regards, Moiz Chinoy. Mobile: 055-8527492 From Claudio.Cavalera at italtel.it Thu Feb 19 02:02:30 2009 From: Claudio.Cavalera at italtel.it (Cavalera Claudio Luigi) Date: Thu, 19 Feb 2009 11:02:30 +0100 Subject: [Freeswitch-users] howto master logging in fs_cli Message-ID: Hello, I'm trying to clarify behaviour of fs_cli http://wiki.freeswitch.org/wiki/Fs_cli After some experiments I'm still not sure on how to deal with logging. For example: --- root at lallobox:/usr/local/freeswitch/bin# ./fs_cli -d 6 _____ ____ ____ _ ___ | ___/ ___| / ___| | |_ _| | |_ \___ \ | | | | | | | _| ___) | | |___| |___ | | |_| |____/ \____|_____|___| ***************************************************** * Anthony Minessale II, Ken Rice, Michael Jerris * * FreeSWITCH (http://www.freeswitch.org) * * Brought to you by ClueCon http://www.cluecon.com/ * ***************************************************** Type /help to see a list of commands [INFO] libs/esl/fs_cli.c:726 main() FS CLI Ready. enter /help for a list of commands. freeswitch at internal> --- So I assume now I'm logging at level 6 in fs_cli, instead I still see debug messages like this as in loglevel 7: 2009-02-19 10:53:29 [DEBUG] mod_event_socket.c:1856 listener_run() Connection Open from 127.0.0.1:48400 2009-02-19 10:53:29 [DEBUG] mod_event_socket.c:1979 listener_run() Session complete, waiting for children To achieve a info loglevel i have to start fs_cli like this ./fs_cli -l info or type /log info in the fs_cli console So could you please someone more expert with me clarify the difference between -d and -l options so that I can update the wiki which is now wrong/incomplete ? -l, --loglevel=command Log Level -q, --quiet Disable logging -d, --debug=level Debug Level (0 - 7) Thanks, Claudio Internet Email Confidentiality Footer ----------------------------------------------------------------------------------------------------- La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. ----------------------------------------------------------------------------------------------------- From jason at jasonjgw.net Thu Feb 19 02:17:29 2009 From: jason at jasonjgw.net (Jason White) Date: Thu, 19 Feb 2009 21:17:29 +1100 Subject: [Freeswitch-users] howto master logging in fs_cli In-Reply-To: References: Message-ID: <20090219101729.GA1868@jdc.jasonjgw.net> Cavalera Claudio Luigi wrote: > So could you please someone more expert with me clarify the difference > between -d and -l options so that I can update the wiki which is now > wrong/incomplete ? > > -l, --loglevel=command Log Level > -q, --quiet Disable logging > -d, --debug=level Debug Level (0 - 7) The difference is that -d controls the level of debugging output generated by fs_cli itself. The log level controls which log messages from your running FreeSWITCH daemon are printed to the fs_cli console. From alex at sinapticode.ro Thu Feb 19 02:36:08 2009 From: alex at sinapticode.ro (Alexandru Nedelcu) Date: Thu, 19 Feb 2009 12:36:08 +0200 Subject: [Freeswitch-users] origination_caller_id_number used instead of effective_caller_id_number Message-ID: <1235039768.4537.17.camel@gathern.lan> I don't know what I'm doing wrong. origination_caller_id_number is used on the B-leg of the bridge, although I also specify the effective_caller_id_number. The thing is that it originally worked in my tests, and I can't figure out what changed in the meantime. The code is something like this ... session = new Session("{originate_retry_sleep_ms=30000,ignore_early_media=true,is_callcenter=0,origination_caller_id_number=+40722333444,effective_caller_id_number=+40711222333}sofia/gateway/provider/"); if (session.ready()) { new_session = new Session("sofia/gateway/provider/", session); if (new_session.ready()) bridge(session, new_session); } Thanks, From alex at sinapticode.ro Thu Feb 19 03:03:09 2009 From: alex at sinapticode.ro (Alexandru Nedelcu) Date: Thu, 19 Feb 2009 13:03:09 +0200 Subject: [Freeswitch-users] origination_caller_id_number used instead of effective_caller_id_number In-Reply-To: <1235039768.4537.17.camel@gathern.lan> References: <1235039768.4537.17.camel@gathern.lan> Message-ID: <1235041389.4537.22.camel@gathern.lan> Just a follow up, I execute the following command: originate {effective_caller_id_number=40722333444}sofia/gateway/provider/ &bridge('sofia/gateway/provider/') And it worked, but when I add origination_caller_id_number ... it overrides effective_caller_id_number. My provider's setup is nothing fancy, something like: On Thu, 2009-02-19 at 12:36 +0200, Alexandru Nedelcu wrote: > I don't know what I'm doing wrong. origination_caller_id_number is used > on the B-leg of the bridge, although I also specify the > effective_caller_id_number. > > The thing is that it originally worked in my tests, and I can't figure > out what changed in the meantime. > > The code is something like this ... > > session = new > Session("{originate_retry_sleep_ms=30000,ignore_early_media=true,is_callcenter=0,origination_caller_id_number=+40722333444,effective_caller_id_number=+40711222333}sofia/gateway/provider/"); > > if (session.ready()) { > > new_session = new Session("sofia/gateway/provider/", > session); > if (new_session.ready()) > bridge(session, new_session); > } > > Thanks, From Claudio.Cavalera at italtel.it Thu Feb 19 03:07:59 2009 From: Claudio.Cavalera at italtel.it (Cavalera Claudio Luigi) Date: Thu, 19 Feb 2009 12:07:59 +0100 Subject: [Freeswitch-users] Random problems with cepstral text to speech Message-ID: Hello list, sometimes when issue a say command for Cepstral TTS in a conference I get this error: [CRIT] mod_local_stream.c:237 read_stream_thread() Leaking stream handle! [conference_play_file() mod_conference.c:2431] and no audio is played to the conference. I have to restart fs to make it work again, any hint on what it could be? BRs, Claudio Internet Email Confidentiality Footer ----------------------------------------------------------------------------------------------------- La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. ----------------------------------------------------------------------------------------------------- From Claudio.Cavalera at italtel.it Thu Feb 19 03:12:25 2009 From: Claudio.Cavalera at italtel.it (Cavalera Claudio Luigi) Date: Thu, 19 Feb 2009 12:12:25 +0100 Subject: [Freeswitch-users] howto master logging in fs_cli In-Reply-To: <20090219101729.GA1868@jdc.jasonjgw.net> Message-ID: freeswitch-users-bounces at lists.freeswitch.org wrote: > Cavalera Claudio Luigi wrote: >> So could you please someone more expert with me clarify the >> difference between -d and -l options so that I can update the wiki >> which is now wrong/incomplete ? >> >> -l, --loglevel=command Log Level >> -q, --quiet Disable logging >> -d, --debug=level Debug Level (0 - 7) > > The difference is that -d controls the level of debugging output > generated by fs_cli itself. The log level controls which log messages > from your running FreeSWITCH daemon are printed to the fs_cli console. > Ah thanks, it was a bit confusing for me also because /log expects a key word such as "info" instead of the number "6". Claudio Internet Email Confidentiality Footer ----------------------------------------------------------------------------------------------------- La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. ----------------------------------------------------------------------------------------------------- From moizchinoy at gmail.com Thu Feb 19 04:50:40 2009 From: moizchinoy at gmail.com (Moiz Chinoy) Date: Thu, 19 Feb 2009 16:50:40 +0400 Subject: [Freeswitch-users] FreeSwitch - PcoketSphinx Prompt Playback & Recognition Issue... In-Reply-To: <29b888f80902190129k377cd710v1fe525659ac68fcb@mail.gmail.com> References: <22080033.post@talk.nabble.com> <22081606.post@talk.nabble.com> <3FEB93CB-9E5B-40FA-8DF7-1CEC8A364732@freeswitch.org> <87f2f3b90902181148r798c5ed5n4ab30877a8e06c00@mail.gmail.com> <29b888f80902182329o578a66d5x66454355d7575f48@mail.gmail.com> <590097F8-ABB3-49B6-8574-E3A3B7ADD134@freeswitch.org> <29b888f80902190129k377cd710v1fe525659ac68fcb@mail.gmail.com> Message-ID: <29b888f80902190450u29b383c5i46a555c0fed82bbb@mail.gmail.com> Can anyone please help me sort it out. Below is the call stack when sphinxbase.dll crashes: sphinxbase.dll!logmath_get_base(logmath_s * lmath=0x00000000) Line 370 + 0x3 bytes C sphinxbase.dll!ngram_model_set_init(cmd_ln_s * config=0x01f01068, ngram_model_s * * models=0x04a9e1dc, char * * names=0x047601ac, const float * weights=0x00000000, int n_models=1) Line 140 + 0x12 bytes C pocketsphinx.dll!ngram_search_init(cmd_ln_s * config=0x01f01068, acmod_s * acmod=0x035add20, dict_s * dict=0x0373ed08) Line 209 + 0x19 bytes C pocketsphinx.dll!ps_reinit(ps_decoder_s * ps=0x03587490, cmd_ln_s * config=0x01f01068) Line 179 + 0x17 bytes C mod_pocketsphinx.dll!pocketsphinx_asr_load_grammar(switch_asr_handle * ah=0x0365a3a0, const char * grammar=0x036d1428, const char * path=0x038622b0) Line 168 + 0x15 bytes C FreeSwitch.dll!switch_core_asr_load_grammar(switch_asr_handle * ah=0x0365a3a0, const char * grammar=0x036d1428, const char * path=0x038622b0) Line 94 + 0x18 bytes C FreeSwitch.dll!switch_ivr_detect_speech_load_grammar(switch_core_session * session=0x0353a308, char * grammar=0x036d1428, char * path=0x00000000) Line 1973 + 0x14 bytes C mod_dptools.dll!detect_speech_function(switch_core_session * session=0x0353a308, const char * data=0x036fa6b8) Line 97 + 0x14 bytes C FreeSwitch.dll!switch_core_session_exec(switch_core_session * session=0x0353a308, const switch_application_interface * application_interface=0x01f0c6c8, const char * arg=0x036fa6b8) Line 1342 + 0x12 bytes C mod_spidermonkey.dll!session_execute(JSContext * cx=0x035433b0, JSObject * obj=0x035b0138, unsigned int argc=2, long * argv=0x03627dd8, long * rval=0x04a9e610) Line 2232 + 0x16 bytes C js32.dll!js_Invoke(JSContext * cx=0x035433b0, unsigned int argc=2, unsigned int flags=0) Line 1181 + 0x20 bytes C js32.dll!js_Interpret(JSContext * cx=0x035433b0, unsigned char * pc=0x03630e56, long * result=0x04a9f014) Line 3571 + 0xf bytes C js32.dll!js_Execute(JSContext * cx=0x035433b0, JSObject * chain=0x035ae7c8, JSScript * script=0x03626fe8, JSStackFrame * down=0x00000000, unsigned int flags=0, long * result=0x04a9f0e8) Line 1427 + 0x13 bytes C js32.dll!JS_ExecuteScript(JSContext * cx=0x035433b0, JSObject * obj=0x035ae7c8, JSScript * script=0x03626fe8, long * rval=0x04a9f0e8) Line 4035 + 0x19 bytes C mod_spidermonkey.dll!eval_some_js(const char * code=0x035421d8, JSContext * cx=0x035433b0, JSObject * obj=0x035ae7c8, long * rval=0x04a9f0e8) Line 103 + 0x15 bytes C mod_spidermonkey.dll!js_parse_and_execute(switch_core_session * session=0x0353a308, const char * input_code=0x035421d8, request_obj * ro=0x00000000) Line 3582 + 0x1e bytes C mod_spidermonkey.dll!js_dp_function(switch_core_session * session=0x0353a308, const char * data=0x035421d8) Line 3591 + 0xf bytes C FreeSwitch.dll!switch_core_session_exec(switch_core_session * session=0x0353a308, const switch_application_interface * application_interface=0x020713f0, const char * arg=0x035421d8) Line 1342 + 0x12 bytes C FreeSwitch.dll!switch_core_session_execute_application(switch_core_session * session=0x0353a308, const char * app=0x035421c8, const char * arg=0x035421d8) Line 1266 C FreeSwitch.dll!switch_core_standard_on_execute(switch_core_session * session=0x0353a308) Line 157 + 0x16 bytes C FreeSwitch.dll!switch_core_session_run(switch_core_session * session=0x0353a308) Line 464 + 0x204 bytes C FreeSwitch.dll!switch_core_session_thread(apr_thread_t * thread=0x020778d0, void * obj=0x0353a308) Line 951 C libapr.dll!dummy_worker(void * opaque=0x020778d0) Line 80 C msvcr90d.dll!1023dfd3() [Frames below may be incorrect and/or missing, no symbols loaded for msvcr90d.dll] msvcr90d.dll!1023df69() kernel32.dll!7c80b683() Moiz Chinoy. On Thu, Feb 19, 2009 at 1:29 PM, Moiz Chinoy wrote: > Can anyone please explain the following fields from pocketsphinx.conf.xml: > > > > > > > > > Moiz Chinoy. > > On Thu, Feb 19, 2009 at 11:43 AM, Brian West wrote: >> No clue, I haven't ever seen that behavior on linux. Maybe you can >> try to narrow it down and report it on jira.. chances are its a bug in >> the pocketsphinx libs. >> >> /b >> >> On Feb 19, 2009, at 1:29 AM, Moiz Chinoy wrote: >> >>> >>> Often in the log I see cryptic characters in the XML part returned by >>> ASR. Is it silence or nose?? >>> If yes is there any way we can control it? >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Regards, > Moiz Chinoy. > Mobile: 055-8527492 > -- Regards, Moiz Chinoy. Mobile: 055-8527492 From frank at impactfax.com Thu Feb 19 04:58:25 2009 From: frank at impactfax.com (Frank @ Impact) Date: Thu, 19 Feb 2009 07:58:25 -0500 Subject: [Freeswitch-users] Voice pattern detection Message-ID: <20b801c99291$c0dc5c30$33014c0a@ws4> I am trying to detect if a caller is an automated greeting voice. And if so, take an action. I have samples of the caller recording that I am looking to match. So this is like a really complex tone detection I guess. It would work like this. - Call comes in - We answer/bridge the call - We start to listen for about 10 seconds. - During this time we are trying to match a snippet of a sound sample (say 2 or 3 seconds worth) we have recorded on a file on the server. We are trying to match this sound sample to the caller side only. - If we hear a match, we take the action. Or if we don't hear the match, we might take a different action. Any of this sound doable? Any guidance on how to accomplish this? -Frank -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090219/f5440595/attachment-0002.html From r.pankratz at fh-wolfenbuettel.de Thu Feb 19 05:04:11 2009 From: r.pankratz at fh-wolfenbuettel.de (Rene Pankratz) Date: Thu, 19 Feb 2009 14:04:11 +0100 Subject: [Freeswitch-users] mod_portaudio: Do not accept next call after Hangup Message-ID: <499D58CB.9080405@fh-wolfenbuettel.de> Hello, when hanging up a call with portaudio automatically the next call that is incoming or held is accepted. Is it possible to configure PA that way, that after hanging up (doesn't matter whether caller or callee) no call is activated automatically? I want to choose if I accept the next call or not. Thanks in advance Ren? From mrene_lists at avgs.ca Thu Feb 19 05:42:33 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Thu, 19 Feb 2009 08:42:33 -0500 Subject: [Freeswitch-users] origination_caller_id_number used instead of effective_caller_id_number In-Reply-To: <1235041389.4537.22.camel@gathern.lan> References: <1235039768.4537.17.camel@gathern.lan> <1235041389.4537.22.camel@gathern.lan> Message-ID: Clarification on the 2 vars....you set origination_caller_id_number on the call leg directly. you set effective_caller_id_number on any leg that will get bridged to something else. Internally, the core will look if effective_caller_id_number is set on the A-leg to see if it can use it. This said: originate {origination_caller_id_number=12223334444,effective_caller_id_number=13334445555}sofia/gateway/blah/12345 &bridge(sofia/gateway/blah/54321) 12345 is called and sees 12223334444 as callerid, then the call is bridged to 54321 which sees 13334445555 because bridge looks up the variable in the a-leg. It would be the same as doing originate {origination_caller_id_number=12223334444}sofia/gateway/blah/12345 &bridge({origination_caller_id_number=13334445555}sofia/gateway/blah/54321) Having it as effective_caller_id_number only saves you the extra work of setting it on all B-legs Mathieu On Thu, Feb 19, 2009 at 6:03 AM, Alexandru Nedelcu wrote: > Just a follow up, I execute the following command: > > originate > {effective_caller_id_number=40722333444}sofia/gateway/provider/ > &bridge('sofia/gateway/provider/') > > And it worked, but when I add origination_caller_id_number ... it > overrides effective_caller_id_number. > > My provider's setup is nothing fancy, something like: > > > > > > > > > > > > > On Thu, 2009-02-19 at 12:36 +0200, Alexandru Nedelcu wrote: > > I don't know what I'm doing wrong. origination_caller_id_number is used > > on the B-leg of the bridge, although I also specify the > > effective_caller_id_number. > > > > The thing is that it originally worked in my tests, and I can't figure > > out what changed in the meantime. > > > > The code is something like this ... > > > > session = new > > > Session("{originate_retry_sleep_ms=30000,ignore_early_media=true,is_callcenter=0,origination_caller_id_number=+40722333444,effective_caller_id_number=+40711222333}sofia/gateway/provider/"); > > > > if (session.ready()) { > > > > new_session = new Session("sofia/gateway/provider/", > > session); > > if (new_session.ready()) > > bridge(session, new_session); > > } > > > > Thanks, > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090219/524241a5/attachment-0002.html From Tim.Meade at millicorp.com Thu Feb 19 04:23:49 2009 From: Tim.Meade at millicorp.com (Tim Meade) Date: Thu, 19 Feb 2009 07:23:49 -0500 Subject: [Freeswitch-users] xml_cdr setup and use questions. Message-ID: <7832D4F0FEC057488FD3BA68CA25A6FD0CD1E05C@postman.millicorp.com> Greetings all. I'm fairly new to freeswitch, but I've been watching here for a while. We've finally decided to take the plunge and try to integrate this into our working environments, so I've spent the last couple days playing around with it. My immediate question has to do with xml_cdr. I want to have it contact our web system so that I can put away the specifics of each call. This should be perfect for what I need to do. Here is my current configuration. The call to the web server is being executed twice. Any ideas why? One has more fields than the other. Both have what seems to be identical , but one has several other sections as well. Thanks in advance. Tim -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090219/ed0dfadc/attachment-0002.html From krice at suspicious.org Thu Feb 19 05:48:42 2009 From: krice at suspicious.org (Ken Rice) Date: Thu, 19 Feb 2009 07:48:42 -0600 Subject: [Freeswitch-users] xml_cdr setup and use questions. In-Reply-To: <7832D4F0FEC057488FD3BA68CA25A6FD0CD1E05C@postman.millicorp.com> Message-ID: A-Leg and B-leg? From: Tim Meade Reply-To: Date: Thu, 19 Feb 2009 07:23:49 -0500 To: "freeswitch-users at lists.freeswitch.org" Subject: [Freeswitch-users] xml_cdr setup and use questions. Greetings all. I'm fairly new to freeswitch, but I've been watching here for a while. We've finally decided to take the plunge and try to integrate this into our working environments, so I've spent the last couple days playing around with it. My immediate question has to do with xml_cdr. I want to have it contact our web system so that I can put away the specifics of each call. This should be perfect for what I need to do. Here is my current configuration. The call to the web server is being executed twice. Any ideas why? One has more fields than the other. Both have what seems to be identical , but one has several other sections as well. Thanks in advance. Tim _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090219/49dd952d/attachment-0002.html From Tim.Meade at millicorp.com Thu Feb 19 06:01:29 2009 From: Tim.Meade at millicorp.com (Tim Meade) Date: Thu, 19 Feb 2009 09:01:29 -0500 Subject: [Freeswitch-users] xml_cdr setup and use questions. In-Reply-To: References: <7832D4F0FEC057488FD3BA68CA25A6FD0CD1E05C@postman.millicorp.com> Message-ID: <7832D4F0FEC057488FD3BA68CA25A6FD0E001FCD@postman.millicorp.com> Not real certain on that. If I setup for it to drop a log file, it only drops a single file, but it seems to be calling the http twice. I've got the web page setup to send an email with the form vars and I'm getting two of them. One just one with that and several other sections. Thanks ... From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice Sent: Thursday, February 19, 2009 8:49 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] xml_cdr setup and use questions. A-Leg and B-leg? ________________________________ From: Tim Meade Reply-To: Date: Thu, 19 Feb 2009 07:23:49 -0500 To: "freeswitch-users at lists.freeswitch.org" Subject: [Freeswitch-users] xml_cdr setup and use questions. Greetings all. I'm fairly new to freeswitch, but I've been watching here for a while. We've finally decided to take the plunge and try to integrate this into our working environments, so I've spent the last couple days playing around with it. My immediate question has to do with xml_cdr. I want to have it contact our web system so that I can put away the specifics of each call. This should be perfect for what I need to do. Here is my current configuration. The call to the web server is being executed twice. Any ideas why? One has more fields than the other. Both have what seems to be identical , but one has several other sections as well. Thanks in advance. Tim ________________________________ _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090219/979cde41/attachment-0002.html From Claudio.Cavalera at italtel.it Thu Feb 19 05:59:02 2009 From: Claudio.Cavalera at italtel.it (Cavalera Claudio Luigi) Date: Thu, 19 Feb 2009 14:59:02 +0100 Subject: [Freeswitch-users] Strange error load testing mod_xml_rpc Message-ID: Hello, doing a bit of load and stress I'm sending request like originate &bridge originate &playback to mod_xml_rpc and I get these errors in freeswitch_http.log ?? getpeername() failed. errno=107 (Transport endpoint is not connected) - no_user - [19/Feb/2009:14:45:25 -0100] "GET" 200 150 ?? getpeername() failed. errno=107 (Transport endpoint is not connected) - no_user - [19/Feb/2009:14:45:31 -0100] "GET" 200 0 Any idea on what it means? Could it be because of the seagull load runner I'm using? I have found this old JIRA with a similar message http://jira.freeswitch.org/browse/MDXMLINT-28 but I'm not sure it's related. BRs, Claudio Internet Email Confidentiality Footer ----------------------------------------------------------------------------------------------------- La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. ----------------------------------------------------------------------------------------------------- From krice at suspicious.org Thu Feb 19 06:11:27 2009 From: krice at suspicious.org (Ken Rice) Date: Thu, 19 Feb 2009 08:11:27 -0600 Subject: [Freeswitch-users] xml_cdr setup and use questions. In-Reply-To: <7832D4F0FEC057488FD3BA68CA25A6FD0E001FCD@postman.millicorp.com> Message-ID: Tim, Try this param and see if it helps in you xml_cdr.conf file From: Tim Meade Reply-To: Date: Thu, 19 Feb 2009 09:01:29 -0500 To: "freeswitch-users at lists.freeswitch.org" Subject: Re: [Freeswitch-users] xml_cdr setup and use questions. Not real certain on that. If I setup for it to drop a log file, it only drops a single file, but it seems to be calling the http twice. I've got the web page setup to send an email with the form vars and I'm getting two of them. One just one with that and several other sections. Thanks ... From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice Sent: Thursday, February 19, 2009 8:49 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] xml_cdr setup and use questions. A-Leg and B-leg? From: Tim Meade Reply-To: Date: Thu, 19 Feb 2009 07:23:49 -0500 To: "freeswitch-users at lists.freeswitch.org" Subject: [Freeswitch-users] xml_cdr setup and use questions. Greetings all. I'm fairly new to freeswitch, but I've been watching here for a while. We've finally decided to take the plunge and try to integrate this into our working environments, so I've spent the last couple days playing around with it. My immediate question has to do with xml_cdr. I want to have it contact our web system so that I can put away the specifics of each call. This should be perfect for what I need to do. Here is my current configuration. The call to the web server is being executed twice. Any ideas why? One has more fields than the other. Both have what seems to be identical , but one has several other sections as well. Thanks in advance. Tim _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090219/89c3b5a7/attachment-0002.html From anthony.minessale at gmail.com Thu Feb 19 06:16:42 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 19 Feb 2009 08:16:42 -0600 Subject: [Freeswitch-users] Random problems with cepstral text to speech In-Reply-To: References: Message-ID: <191c3a030902190616pae78f47x757702108c760264@mail.gmail.com> Are you using cepstral 5.1? There is a known issue with that release and it's closed source so we cannot do much about it. Cepstral 4.x works fine. 2009/2/19 Cavalera Claudio Luigi > Hello list, > sometimes when issue a say command for Cepstral TTS in a conference I > get this error: > [CRIT] mod_local_stream.c:237 read_stream_thread() Leaking stream > handle! [conference_play_file() mod_conference.c:2431] > > and no audio is played to the conference. > > I have to restart fs to make it work again, any hint on what it could > be? > > BRs, > Claudio > > > Internet Email Confidentiality Footer > > ----------------------------------------------------------------------------------------------------- > La presente comunicazione, con le informazioni in essa contenute e ogni > documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' > indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete > i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, > comunicazione, divulgazione o simili basate sul contenuto di tali > informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., > D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se > avete ricevuto questa comunicazione per errore, vi preghiamo di darne > immediata notizia al mittente e di distruggere il messaggio originale e ogni > file allegato senza farne copia alcuna o riprodurne in alcun modo il > contenuto. > > This e-mail and its attachments are intended for the addressee(s) only and > are confidential and/or may contain legally privileged information. If you > have received this message by mistake or are not one of the addressees > above, you may take no action based on it, and you may not copy or show it > to anyone; please reply to this e-mail and point out the error which has > occurred. > > ----------------------------------------------------------------------------------------------------- > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090219/e27f3a83/attachment-0002.html From anthony.minessale at gmail.com Thu Feb 19 06:19:27 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 19 Feb 2009 08:19:27 -0600 Subject: [Freeswitch-users] Strange error load testing mod_xml_rpc In-Reply-To: References: Message-ID: <191c3a030902190619y176f62bdtd0e3329c0c9b1337@mail.gmail.com> Yes we already see you have reported this issue in jira and we are working on it. you do not need to report it here as well. You may want to make yourself available on irc or some other im today so we can contact you for more details. 2009/2/19 Cavalera Claudio Luigi > Hello, > doing a bit of load and stress I'm sending request like > > originate &bridge > originate &playback > > to mod_xml_rpc and I get these errors in freeswitch_http.log > > ?? getpeername() failed. errno=107 (Transport endpoint is not > connected) - no_user - [19/Feb/2009:14:45:25 -0100] "GET" 200 150 > ?? getpeername() failed. errno=107 (Transport endpoint is not > connected) - no_user - [19/Feb/2009:14:45:31 -0100] "GET" 200 0 > > Any idea on what it means? > Could it be because of the seagull load runner I'm using? > > I have found this old JIRA with a similar message > http://jira.freeswitch.org/browse/MDXMLINT-28 > but I'm not sure it's related. > > BRs, > Claudio > > > Internet Email Confidentiality Footer > > ----------------------------------------------------------------------------------------------------- > La presente comunicazione, con le informazioni in essa contenute e ogni > documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' > indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete > i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, > comunicazione, divulgazione o simili basate sul contenuto di tali > informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., > D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se > avete ricevuto questa comunicazione per errore, vi preghiamo di darne > immediata notizia al mittente e di distruggere il messaggio originale e ogni > file allegato senza farne copia alcuna o riprodurne in alcun modo il > contenuto. > > This e-mail and its attachments are intended for the addressee(s) only and > are confidential and/or may contain legally privileged information. If you > have received this message by mistake or are not one of the addressees > above, you may take no action based on it, and you may not copy or show it > to anyone; please reply to this e-mail and point out the error which has > occurred. > > ----------------------------------------------------------------------------------------------------- > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090219/96d52df1/attachment-0002.html From alex at sinapticode.ro Thu Feb 19 06:19:45 2009 From: alex at sinapticode.ro (Alexandru Nedelcu) Date: Thu, 19 Feb 2009 16:19:45 +0200 Subject: [Freeswitch-users] origination_caller_id_number used instead of effective_caller_id_number In-Reply-To: References: <1235039768.4537.17.camel@gathern.lan> <1235041389.4537.22.camel@gathern.lan> Message-ID: <1235053185.4537.59.camel@gathern.lan> OK, so effective_caller_id_number is the same as origination_caller_id_number set on the B-leg (cool). Unfortunately origination_caller_id_number on the A-leg overides the origination_caller_id_number on the B-leg (I tried setting it with both effective_caller_id on the A-leg and origination_caller_id on the B-leg). It doesn't work. If I'm setting caller-id on the B-leg only, then it works. Has this something to do with my SIP provider maybe? On Thu, 2009-02-19 at 08:42 -0500, Mathieu Rene wrote: > Clarification on the 2 vars.... > you set origination_caller_id_number on the call leg directly. > you set effective_caller_id_number on any leg that will get bridged to > something else. > > > Internally, the core will look if effective_caller_id_number is set on > the A-leg to see if it can use it. > > > This said: originate > {origination_caller_id_number=12223334444,effective_caller_id_number=13334445555}sofia/gateway/blah/12345 &bridge(sofia/gateway/blah/54321) > > > 12345 is called and sees 12223334444 as callerid, then the call is > bridged to 54321 which sees 13334445555 because bridge looks up the > variable in the a-leg. > > > It would be the same as doing > originate > {origination_caller_id_number=12223334444}sofia/gateway/blah/12345 > &bridge({origination_caller_id_number=13334445555}sofia/gateway/blah/54321) > > > Having it as effective_caller_id_number only saves you the extra work > of setting it on all B-legs > > > Mathieu From Tim.Meade at millicorp.com Thu Feb 19 06:23:02 2009 From: Tim.Meade at millicorp.com (Tim Meade) Date: Thu, 19 Feb 2009 09:23:02 -0500 Subject: [Freeswitch-users] xml_cdr setup and use questions. In-Reply-To: References: <7832D4F0FEC057488FD3BA68CA25A6FD0E001FCD@postman.millicorp.com> Message-ID: <7832D4F0FEC057488FD3BA68CA25A6FD0E001FCE@postman.millicorp.com> That seemed to do it.... Noob question: reloadXML didn't reload the change. I still got the two emails. So to be sure, I shutdown and restarted. Now I'm only getting the one email. Shouldn't the reloadXML reload that module also? Or do I have to reload the modules directly as they are "outside" fs? Tim From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice Sent: Thursday, February 19, 2009 9:11 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] xml_cdr setup and use questions. Tim, Try this param and see if it helps in you xml_cdr.conf file ________________________________ From: Tim Meade Reply-To: Date: Thu, 19 Feb 2009 09:01:29 -0500 To: "freeswitch-users at lists.freeswitch.org" Subject: Re: [Freeswitch-users] xml_cdr setup and use questions. Not real certain on that. If I setup for it to drop a log file, it only drops a single file, but it seems to be calling the http twice. I've got the web page setup to send an email with the form vars and I'm getting two of them. One just one with that and several other sections. Thanks ... From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice Sent: Thursday, February 19, 2009 8:49 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] xml_cdr setup and use questions. A-Leg and B-leg? ________________________________ From: Tim Meade Reply-To: Date: Thu, 19 Feb 2009 07:23:49 -0500 To: "freeswitch-users at lists.freeswitch.org" Subject: [Freeswitch-users] xml_cdr setup and use questions. Greetings all. I'm fairly new to freeswitch, but I've been watching here for a while. We've finally decided to take the plunge and try to integrate this into our working environments, so I've spent the last couple days playing around with it. My immediate question has to do with xml_cdr. I want to have it contact our web system so that I can put away the specifics of each call. This should be perfect for what I need to do. Here is my current configuration. The call to the web server is being executed twice. Any ideas why? One has more fields than the other. Both have what seems to be identical , but one has several other sections as well. Thanks in advance. Tim ________________________________ _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ________________________________ _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090219/d7148c35/attachment-0002.html From krice at freeswitch.org Thu Feb 19 06:27:32 2009 From: krice at freeswitch.org (Ken Rice) Date: Thu, 19 Feb 2009 08:27:32 -0600 Subject: [Freeswitch-users] xml_cdr setup and use questions. In-Reply-To: <7832D4F0FEC057488FD3BA68CA25A6FD0E001FCE@postman.millicorp.com> Message-ID: You reloadxml just reparses the config it does not automattically tell specific modules to reload their configuration From: Tim Meade Reply-To: Date: Thu, 19 Feb 2009 09:23:02 -0500 To: "freeswitch-users at lists.freeswitch.org" Subject: Re: [Freeswitch-users] xml_cdr setup and use questions. That seemed to do it.... Noob question: reloadXML didn't reload the change. I still got the two emails. So to be sure, I shutdown and restarted. Now I'm only getting the one email. Shouldn't the reloadXML reload that module also? Or do I have to reload the modules directly as they are "outside" fs? Tim From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice Sent: Thursday, February 19, 2009 9:11 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] xml_cdr setup and use questions. Tim, Try this param and see if it helps in you xml_cdr.conf file From: Tim Meade Reply-To: Date: Thu, 19 Feb 2009 09:01:29 -0500 To: "freeswitch-users at lists.freeswitch.org" Subject: Re: [Freeswitch-users] xml_cdr setup and use questions. Not real certain on that. If I setup for it to drop a log file, it only drops a single file, but it seems to be calling the http twice. I've got the web page setup to send an email with the form vars and I'm getting two of them. One just one with that and several other sections. Thanks ... From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice Sent: Thursday, February 19, 2009 8:49 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] xml_cdr setup and use questions. A-Leg and B-leg? From: Tim Meade Reply-To: Date: Thu, 19 Feb 2009 07:23:49 -0500 To: "freeswitch-users at lists.freeswitch.org" Subject: [Freeswitch-users] xml_cdr setup and use questions. Greetings all. I'm fairly new to freeswitch, but I've been watching here for a while. We've finally decided to take the plunge and try to integrate this into our working environments, so I've spent the last couple days playing around with it. My immediate question has to do with xml_cdr. I want to have it contact our web system so that I can put away the specifics of each call. This should be perfect for what I need to do. Here is my current configuration. The call to the web server is being executed twice. Any ideas why? One has more fields than the other. Both have what seems to be identical , but one has several other sections as well. Thanks in advance. Tim _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090219/2acdb419/attachment-0002.html From Tim.Meade at millicorp.com Thu Feb 19 07:01:41 2009 From: Tim.Meade at millicorp.com (Tim Meade) Date: Thu, 19 Feb 2009 10:01:41 -0500 Subject: [Freeswitch-users] xml_cdr setup and use questions. In-Reply-To: References: <7832D4F0FEC057488FD3BA68CA25A6FD0E001FCE@postman.millicorp.com> Message-ID: <7832D4F0FEC057488FD3BA68CA25A6FD0E001FD0@postman.millicorp.com> Thanks Ken. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice Sent: Thursday, February 19, 2009 9:28 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] xml_cdr setup and use questions. You reloadxml just reparses the config it does not automattically tell specific modules to reload their configuration ________________________________ From: Tim Meade Reply-To: Date: Thu, 19 Feb 2009 09:23:02 -0500 To: "freeswitch-users at lists.freeswitch.org" Subject: Re: [Freeswitch-users] xml_cdr setup and use questions. That seemed to do it.... Noob question: reloadXML didn't reload the change. I still got the two emails. So to be sure, I shutdown and restarted. Now I'm only getting the one email. Shouldn't the reloadXML reload that module also? Or do I have to reload the modules directly as they are "outside" fs? Tim From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice Sent: Thursday, February 19, 2009 9:11 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] xml_cdr setup and use questions. Tim, Try this param and see if it helps in you xml_cdr.conf file ________________________________ From: Tim Meade Reply-To: Date: Thu, 19 Feb 2009 09:01:29 -0500 To: "freeswitch-users at lists.freeswitch.org" Subject: Re: [Freeswitch-users] xml_cdr setup and use questions. Not real certain on that. If I setup for it to drop a log file, it only drops a single file, but it seems to be calling the http twice. I've got the web page setup to send an email with the form vars and I'm getting two of them. One just one with that and several other sections. Thanks ... From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice Sent: Thursday, February 19, 2009 8:49 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] xml_cdr setup and use questions. A-Leg and B-leg? ________________________________ From: Tim Meade Reply-To: Date: Thu, 19 Feb 2009 07:23:49 -0500 To: "freeswitch-users at lists.freeswitch.org" Subject: [Freeswitch-users] xml_cdr setup and use questions. Greetings all. I'm fairly new to freeswitch, but I've been watching here for a while. We've finally decided to take the plunge and try to integrate this into our working environments, so I've spent the last couple days playing around with it. My immediate question has to do with xml_cdr. I want to have it contact our web system so that I can put away the specifics of each call. This should be perfect for what I need to do. Here is my current configuration. The call to the web server is being executed twice. Any ideas why? One has more fields than the other. Both have what seems to be identical , but one has several other sections as well. Thanks in advance. Tim ________________________________ _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ________________________________ _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ________________________________ _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090219/3d9b6cc7/attachment-0002.html From mike at jerris.com Thu Feb 19 07:03:35 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 19 Feb 2009 10:03:35 -0500 Subject: [Freeswitch-users] origination_caller_id_number used instead of effective_caller_id_number In-Reply-To: <1235053185.4537.59.camel@gathern.lan> References: <1235039768.4537.17.camel@gathern.lan> <1235041389.4537.22.camel@gathern.lan> <1235053185.4537.59.camel@gathern.lan> Message-ID: <2CEAC5BC-D5F8-4253-A710-C5E60DAC7EB8@jerris.com> Can you re-test this with current svn trunk. I believe this was fixed yesterday. Mike On Feb 19, 2009, at 9:19 AM, Alexandru Nedelcu wrote: > OK, so effective_caller_id_number is the same as > origination_caller_id_number set on the B-leg (cool). > > Unfortunately origination_caller_id_number on the A-leg overides the > origination_caller_id_number on the B-leg (I tried setting it with > both > effective_caller_id on the A-leg and origination_caller_id on the > B-leg). It doesn't work. > > If I'm setting caller-id on the B-leg only, then it works. > > Has this something to do with my SIP provider maybe? > > > On Thu, 2009-02-19 at 08:42 -0500, Mathieu Rene wrote: >> Clarification on the 2 vars.... >> you set origination_caller_id_number on the call leg directly. >> you set effective_caller_id_number on any leg that will get bridged >> to >> something else. >> >> >> Internally, the core will look if effective_caller_id_number is set >> on >> the A-leg to see if it can use it. >> >> >> This said: originate >> {origination_caller_id_number >> =12223334444,effective_caller_id_number=13334445555}sofia/gateway/ >> blah/12345 &bridge(sofia/gateway/blah/54321) >> >> >> 12345 is called and sees 12223334444 as callerid, then the call is >> bridged to 54321 which sees 13334445555 because bridge looks up the >> variable in the a-leg. >> >> >> It would be the same as doing >> originate >> {origination_caller_id_number=12223334444}sofia/gateway/blah/12345 >> &bridge({origination_caller_id_number=13334445555}sofia/gateway/ >> blah/54321) >> >> >> Having it as effective_caller_id_number only saves you the extra work >> of setting it on all B-legs >> >> >> Mathieu From kerrada2003 at yahoo.com Thu Feb 19 07:05:05 2009 From: kerrada2003 at yahoo.com (Ali Al-Rubaie) Date: Thu, 19 Feb 2009 07:05:05 -0800 (PST) Subject: [Freeswitch-users] Realm Value In-Reply-To: Message-ID: <472895.39120.qm@web33708.mail.mud.yahoo.com> Thanks Brian, Sorry if the question looks primitive but in which file I can find the rvn? Is there any tarball with the latest revisions? Thanks, --- On Tue, 2/17/09, freeswitch-users-request at lists.freeswitch.org wrote: Message: 5 Date: Tue, 17 Feb 2009 14:36:51 -0600 From: Brian West Subject: Re: [Freeswitch-users] Realm Value To: freeswitch-users at lists.freeswitch.org Message-ID: <090382A2-AF83-4635-90CF-35749F50E0FA at freeswitch.org> Content-Type: text/plain; charset=US-ASCII; format=flowed; delsp=yes Very sorry to hear you have to use Broken Software. But some good has come of this if you update to rev 12113 or great you'll be 100% OK. /b On Feb 17, 2009, at 2:21 PM, Ali Al-Rubaie wrote: > > I have to use a specific softphone, HelpCaster, but it can not pass > the authentication stage. However it can authenticate with OpenSips > server! What I had noticed is that it uses static realm with > OpenSips therefore I'm trying to do the same. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090219/07b6169f/attachment-0002.html From intralanman at freeswitch.org Thu Feb 19 07:26:15 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Thu, 19 Feb 2009 10:26:15 -0500 Subject: [Freeswitch-users] Realm Value In-Reply-To: <472895.39120.qm@web33708.mail.mud.yahoo.com> References: <472895.39120.qm@web33708.mail.mud.yahoo.com> Message-ID: <499D7A17.7030202@freeswitch.org> Ali Al-Rubaie wrote: > Thanks Brian, > > Sorry if the question looks primitive but in which file I can find the > rvn? > you can find the revision from typing "version" at the fs_cli > Is there any tarball with the latest revisions? > the 1.0.3 tarball was just rolled yesterday, give it a shot -Ray -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090219/89abc2b0/attachment-0002.html From Claudio.Cavalera at italtel.it Thu Feb 19 07:47:27 2009 From: Claudio.Cavalera at italtel.it (Cavalera Claudio Luigi) Date: Thu, 19 Feb 2009 16:47:27 +0100 Subject: [Freeswitch-users] Random problems with cepstral text to speech In-Reply-To: <191c3a030902190616pae78f47x757702108c760264@mail.gmail.com> Message-ID: From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale > Are you using cepstral 5.1? > There is a known issue with that release and it's closed source so we cannot do much about it. > Cepstral 4.x works fine. Yes 5.1, my fault. I have added an initial warning here on the wiki http://wiki.freeswitch.org/wiki/Mod_cepstral although it also speaks about 5.1 and Ubuntu... Thanks, Claudio Internet E. Mail Confidentiality Footer ----------------------------------------------------------------------------------------------------- La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. From Claudio.Cavalera at italtel.it Thu Feb 19 08:00:44 2009 From: Claudio.Cavalera at italtel.it (Cavalera Claudio Luigi) Date: Thu, 19 Feb 2009 17:00:44 +0100 Subject: [Freeswitch-users] Strange error load testing mod_xml_rpc In-Reply-To: <191c3a030902190619y176f62bdtd0e3329c0c9b1337@mail.gmail.com> Message-ID: From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale > Yes we already see you have reported this issue in jira and we are working on it. > you do not need to report it here as well. > You may want to make yourself available on irc or some other im today so we can contact you for more details. I'm sorry I reported this message here because I did not think it's related with my JIRA reports. They are about two segmentation faults, while this message is about load testing not going very well, although it comes out during the same load tests :-) I'm starting to think that most people are using the sofia SIP stack in fs to get rate of 100cps while the other interfaces such as event socket and mod_xml_rpc are not well "engineered" yet. I'm going deeper into this and I hope my results will be of help for the community. At the moment I'm trying to understand the cause of TCP retransmissions I see in the snoop, it seems that fs lags a little bit in sending the TCP Ack back to the load runner, even with the machine doing nothing and a really tiny amount of load. BRs, Claudio Internet Email Confidentiality Footer ----------------------------------------------------------------------------------------------------- La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. ----------------------------------------------------------------------------------------------------- From Claudio.Cavalera at italtel.it Thu Feb 19 09:00:19 2009 From: Claudio.Cavalera at italtel.it (Cavalera Claudio Luigi) Date: Thu, 19 Feb 2009 18:00:19 +0100 Subject: [Freeswitch-users] Strange error load testing mod_xml_rpc In-Reply-To: Message-ID: Here is a snoop: http://pastebin.freeswitch.org/7351 thx, cla Internet Email Confidentiality Footer ----------------------------------------------------------------------------------------------------- La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. ----------------------------------------------------------------------------------------------------- From anthony.minessale at gmail.com Thu Feb 19 09:30:01 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 19 Feb 2009 11:30:01 -0600 Subject: [Freeswitch-users] Strange error load testing mod_xml_rpc In-Reply-To: References: Message-ID: <191c3a030902190930x121eba94jb644c25922cbdb7b@mail.gmail.com> api calls by default are blocking, it will not return until the result of the originate is determined. you must pre-empt the originate command with the bgapi api command which is similar to the event_socket bgapi command so that it tells the task to run in a dedicated thread. also the web interface was meant to use an xml rpc client. I have generated at least 400cps on event socket before. 2009/2/19 Cavalera Claudio Luigi > Here is a snoop: > http://pastebin.freeswitch.org/7351 > thx, > cla > > > Internet Email Confidentiality Footer > > ----------------------------------------------------------------------------------------------------- > La presente comunicazione, con le informazioni in essa contenute e ogni > documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' > indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete > i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, > comunicazione, divulgazione o simili basate sul contenuto di tali > informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., > D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se > avete ricevuto questa comunicazione per errore, vi preghiamo di darne > immediata notizia al mittente e di distruggere il messaggio originale e ogni > file allegato senza farne copia alcuna o riprodurne in alcun modo il > contenuto. > > This e-mail and its attachments are intended for the addressee(s) only and > are confidential and/or may contain legally privileged information. If you > have received this message by mistake or are not one of the addressees > above, you may take no action based on it, and you may not copy or show it > to anyone; please reply to this e-mail and point out the error which has > occurred. > > ----------------------------------------------------------------------------------------------------- > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090219/0c42b1d8/attachment-0002.html From brian at freeswitch.org Thu Feb 19 10:44:22 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 19 Feb 2009 12:44:22 -0600 Subject: [Freeswitch-users] ESL Message-ID: FreeSWITCHers, Not sure anyone is paying attention or not but Anthony wrapped the ESL library up so you can use it from Perl, Python, Lua, Ruby and PHP. What I'm requesting from our community is to help flex it out.. write examples and populate the Wiki page with information about it. http://wiki.freeswitch.org/wiki/Esl Collins and I are going to start filling in the page but I want someone thats good with Ruby, Python, PHP to help in those areas.. kick in some lua and perl if you like. It works with OES and IES... (Outbound Event Socket and Inbound Event Socket) Not sure those names are official but we have been calling them that ;) Thanks, Brian West From kerrada2003 at yahoo.com Thu Feb 19 12:06:54 2009 From: kerrada2003 at yahoo.com (Ali Al-Rubaie) Date: Thu, 19 Feb 2009 12:06:54 -0800 (PST) Subject: [Freeswitch-users] Realm Value In-Reply-To: Message-ID: <97318.60826.qm@web33708.mail.mud.yahoo.com> Thanks Ray but unfortunately the 1.0.3 tarball compilation results in the following errors: Compiling src/switch_odbc.c ... In file included from src/switch_odbc.c:33: ./src/include/switch_odbc.h:36:17: error: sql.h: No such file or directory ./src/include/switch_odbc.h:43:20: error: sqlext.h: No such file or directory ./src/include/switch_odbc.h:45:22: error: sqltypes.h: No such file or directory In file included from src/switch_odbc.c:33: ./src/include/switch_odbc.h:66: error: expected declaration specifiers or '...' before 'SQLHSTMT' ./src/include/switch_odbc.h:96: error: expected declaration specifiers or '...' before 'SQLHSTMT' src/switch_odbc.c:43: error: expected specifier-qualifier-list before 'SQLHENV' src/switch_odbc.c: In function 'switch_odbc_handle_new': src/switch_odbc.c:76: error: 'switch_odbc_handle_t' has no member named 'env' src/switch_odbc.c:76: error: 'SQL_NULL_HANDLE' undeclared (first use in this function) src/switch_odbc.c:76: error: (Each undeclared identifier is reported only once src/switch_odbc.c:76: error: for each function it appears in.) src/switch_odbc.c:77: error: 'switch_odbc_handle_t' has no member named 'state' src/switch_odbc.c: In function 'switch_odbc_handle_disconnect': src/switch_odbc.c:96: error: 'switch_odbc_handle_t' has no member named 'state' cc1: warnings being treated as errors src/switch_odbc.c:97: warning: implicit declaration of function 'SQLDisconnect' src/switch_odbc.c:97: error: 'switch_odbc_handle_t' has no member named 'con' src/switch_odbc.c:105: error: 'switch_odbc_handle_t' has no member named 'state' src/switch_odbc.c: In function 'switch_odbc_handle_connect': src/switch_odbc.c:113: error: 'SQLINTEGER' undeclared (first use in this function) src/switch_odbc.c:113: error: expected ';' before 'err' src/switch_odbc.c:116: error: 'SQLSMALLINT' undeclared (first use in this function) src/switch_odbc.c:116: error: expected ';' before 'valueLength' src/switch_odbc.c:119: error: 'switch_odbc_handle_t' has no member named 'env' src/switch_odbc.c:119: error: 'SQL_NULL_HANDLE' undeclared (first use in this function) src/switch_odbc.c:120: warning: implicit declaration of function 'SQLAllocHandle' src/switch_odbc.c:120: error: 'SQL_HANDLE_ENV' undeclared (first use in this function) src/switch_odbc.c:120: error: 'switch_odbc_handle_t' has no member named 'env' src/switch_odbc.c:122: error: 'SQL_SUCCESS' undeclared (first use in this function) src/switch_odbc.c:122: error: 'SQL_SUCCESS_WITH_INFO' undeclared (first use in this function) src/switch_odbc.c:127: warning: implicit declaration of function 'SQLSetEnvAttr' src/switch_odbc.c:127: error: 'switch_odbc_handle_t' has no member named 'env' src/switch_odbc.c:127: error: 'SQL_ATTR_ODBC_VERSION' undeclared (first use in this function) src/switch_odbc.c:127: error: 'SQL_OV_ODBC3' undeclared (first use in this function) src/switch_odbc.c:131: warning: implicit declaration of function 'SQLFreeHandle' src/switch_odbc.c:131: error: 'switch_odbc_handle_t' has no member named 'env' src/switch_odbc.c:135: error: 'SQL_HANDLE_DBC' undeclared (first use in this function) src/switch_odbc.c:135: error: 'switch_odbc_handle_t' has no member named 'env' src/switch_odbc.c:135: error: 'switch_odbc_handle_t' has no member named 'con' src/switch_odbc.c:139: error: 'switch_odbc_handle_t' has no member named 'env' src/switch_odbc.c:142: warning: implicit declaration of function 'SQLSetConnectAttr' src/switch_odbc.c:142: error: 'switch_odbc_handle_t' has no member named 'con' src/switch_odbc.c:142: error: 'SQL_LOGIN_TIMEOUT' undeclared (first use in this function) src/switch_odbc.c:142: error: 'SQLPOINTER' undeclared (first use in this function) src/switch_odbc.c:142: error: expected expression before ')' token src/switch_odbc.c:144: error: 'switch_odbc_handle_t' has no member named 'state' src/switch_odbc.c:152: warning: implicit declaration of function 'SQLConnect' src/switch_odbc.c:152: error: 'switch_odbc_handle_t' has no member named 'con' src/switch_odbc.c:152: error: 'SQLCHAR' undeclared (first use in this function) src/switch_odbc.c:152: error: expected expression before ')' token src/switch_odbc.c:154: error: expected ';' before 'outstr' src/switch_odbc.c:155: error: expected ';' before 'outstrlen' src/switch_odbc.c:157: warning: implicit declaration of function 'SQLDriverConnect' src/switch_odbc.c:157: error: 'switch_odbc_handle_t' has no member named 'con' src/switch_odbc.c:157: error: expected expression before ')' token src/switch_odbc.c:163: error: too many arguments to function 'switch_odbc_handle_get_error' src/switch_odbc.c:167: warning: implicit declaration of function 'SQLGetDiagRec' src/switch_odbc.c:167: error: 'switch_odbc_handle_t' has no member named 'con' src/switch_odbc.c:167: error: 'err' undeclared (first use in this function) src/switch_odbc.c:170: error: 'switch_odbc_handle_t' has no member named 'env' src/switch_odbc.c:174: warning: implicit declaration of function 'SQLGetInfo' src/switch_odbc.c:174: error: 'switch_odbc_handle_t' has no member named 'con' src/switch_odbc.c:174: error: 'SQL_DRIVER_NAME' undeclared (first use in this function) src/switch_odbc.c:174: error: expected expression before ')' token src/switch_odbc.c:176: error: 'valueLength' undeclared (first use in this function) src/switch_odbc.c:177: error: 'switch_odbc_handle_t' has no member named 'odbc_driver' src/switch_odbc.c:177: error: 'switch_odbc_handle_t' has no member named 'odbc_driver' src/switch_odbc.c:177: error: 'switch_odbc_handle_t' has no member named 'odbc_driver' src/switch_odbc.c:177: error: 'switch_odbc_handle_t' has no member named 'odbc_driver' src/switch_odbc.c:177: error: 'switch_odbc_handle_t' has no member named 'odbc_driver' src/switch_odbc.c:177: error: 'switch_odbc_handle_t' has no member named 'odbc_driver' src/switch_odbc.c:180: error: 'switch_odbc_handle_t' has no member named 'odbc_driver' src/switch_odbc.c:180: error: 'switch_odbc_handle_t' has no member named 'odbc_driver' src/switch_odbc.c:180: error: 'switch_odbc_handle_t' has no member named 'odbc_driver' src/switch_odbc.c:181: error: 'switch_odbc_handle_t' has no member named 'is_firebird' src/switch_odbc.c:183: error: 'switch_odbc_handle_t' has no member named 'is_firebird' src/switch_odbc.c:187: error: 'switch_odbc_handle_t' has no member named 'state' src/switch_odbc.c: In function 'db_is_up': src/switch_odbc.c:194: error: 'SQLHSTMT' undeclared (first use in this function) src/switch_odbc.c:194: error: expected ';' before 'stmt' src/switch_odbc.c:195: error: 'SQLLEN' undeclared (first use in this function) src/switch_odbc.c:195: error: expected ';' before 'm' src/switch_odbc.c:200: error: 'SQLCHAR' undeclared (first use in this function) src/switch_odbc.c:200: error: expected ';' before 'sql' src/switch_odbc.c:203: error: 'SQLRETURN' undeclared (first use in this function) src/switch_odbc.c:203: error: expected ';' before 'rc' src/switch_odbc.c:204: error: 'SQLSMALLINT' undeclared (first use in this function) src/switch_odbc.c:204: error: expected ';' before 'nresultcols' src/switch_odbc.c:213: error: 'switch_odbc_handle_t' has no member named 'is_firebird' src/switch_odbc.c:214: error: 'sql' undeclared (first use in this function) src/switch_odbc.c:219: error: 'SQL_HANDLE_STMT' undeclared (first use in this function) src/switch_odbc.c:219: error: 'switch_odbc_handle_t' has no member named 'con' src/switch_odbc.c:219: error: 'stmt' undeclared (first use in this function) src/switch_odbc.c:219: error: 'SQL_SUCCESS' undeclared (first use in this function) src/switch_odbc.c:223: warning: implicit declaration of function 'SQLPrepare' src/switch_odbc.c:223: error: 'SQL_NTS' undeclared (first use in this function) src/switch_odbc.c:227: warning: implicit declaration of function 'SQLExecute' src/switch_odbc.c:229: error: 'SQL_SUCCESS_WITH_INFO' undeclared (first use in this function) src/switch_odbc.c:233: warning: implicit declaration of function 'SQLRowCount' src/switch_odbc.c:233: error: 'm' undeclared (first use in this function) src/switch_odbc.c:234: error: 'rc' undeclared (first use in this function) src/switch_odbc.c:234: warning: implicit declaration of function 'SQLNumResultCols' src/switch_odbc.c:234: error: 'nresultcols' undeclared (first use in this function) src/switch_odbc.c:248: error: too many arguments to function 'switch_odbc_handle_get_error' src/switch_odbc.c: At top level: src/switch_odbc.c:293: error: expected declaration specifiers or '...' before 'SQLHSTMT' src/switch_odbc.c: In function 'switch_odbc_handle_exec': src/switch_odbc.c:295: error: 'SQLHSTMT' undeclared (first use in this function) src/switch_odbc.c:295: error: expected ';' before 'stmt' src/switch_odbc.c:302: error: 'SQL_HANDLE_STMT' undeclared (first use in this function) src/switch_odbc.c:302: error: 'switch_odbc_handle_t' has no member named 'con' src/switch_odbc.c:302: error: 'stmt' undeclared (first use in this function) src/switch_odbc.c:302: error: 'SQL_SUCCESS' undeclared (first use in this function) src/switch_odbc.c:306: error: 'SQL_NTS' undeclared (first use in this function) src/switch_odbc.c:312: error: 'SQL_SUCCESS_WITH_INFO' undeclared (first use in this function) src/switch_odbc.c:316: error: 'rstmt' undeclared (first use in this function) src/switch_odbc.c: In function 'switch_odbc_handle_callback_exec_detailed': src/switch_odbc.c:337: error: 'SQLHSTMT' undeclared (first use in this function) src/switch_odbc.c:337: error: expected ';' before 'stmt' src/switch_odbc.c:338: error: 'SQLSMALLINT' undeclared (first use in this function) src/switch_odbc.c:338: error: expected ';' before 'c' src/switch_odbc.c:339: error: 'SQLLEN' undeclared (first use in this function) src/switch_odbc.c:339: error: expected ';' before 'm' src/switch_odbc.c:350: error: 'SQL_HANDLE_STMT' undeclared (first use in this function) src/switch_odbc.c:350: error: 'switch_odbc_handle_t' has no member named 'con' src/switch_odbc.c:350: error: 'stmt' undeclared (first use in this function) src/switch_odbc.c:350: error: 'SQL_SUCCESS' undeclared (first use in this function) src/switch_odbc.c:355: error: 'SQL_NTS' undeclared (first use in this function) src/switch_odbc.c:362: error: 'SQL_SUCCESS_WITH_INFO' undeclared (first use in this function) src/switch_odbc.c:366: error: 'c' undeclared (first use in this function) src/switch_odbc.c:367: error: 'm' undeclared (first use in this function) src/switch_odbc.c:369: error: 't' undeclared (first use in this function) src/switch_odbc.c:376: warning: implicit declaration of function 'SQLFetch' src/switch_odbc.c:390: error: 'x' undeclared (first use in this function) src/switch_odbc.c:391: error: expected ';' before 'NameLength' src/switch_odbc.c:392: error: 'SQLULEN' undeclared (first use in this function) src/switch_odbc.c:392: error: expected ';' before 'ColumnSize' src/switch_odbc.c:396: warning: implicit declaration of function 'SQLDescribeCol' src/switch_odbc.c:396: error: 'SQLCHAR' undeclared (first use in this function) src/switch_odbc.c:396: error: expected expression before ')' token src/switch_odbc.c:397: error: 'ColumnSize' undeclared (first use in this function) src/switch_odbc.c:401: warning: implicit declaration of function 'SQLGetData' src/switch_odbc.c:401: error: 'SQL_C_CHAR' undeclared (first use in this function) src/switch_odbc.c:401: error: expected expression before ')' token src/switch_odbc.c:436: error: too many arguments to function 'switch_odbc_handle_get_error' src/switch_odbc.c: In function 'switch_odbc_handle_destroy': src/switch_odbc.c:459: error: 'SQL_HANDLE_DBC' undeclared (first use in this function) src/switch_odbc.c:459: error: 'switch_odbc_handle_t' has no member named 'con' src/switch_odbc.c:460: error: 'SQL_HANDLE_ENV' undeclared (first use in this function) src/switch_odbc.c:460: error: 'switch_odbc_handle_t' has no member named 'env' src/switch_odbc.c: In function 'switch_odbc_handle_get_state': src/switch_odbc.c:471: error: 'switch_odbc_handle_t' has no member named 'state' src/switch_odbc.c: At top level: src/switch_odbc.c:474: error: expected declaration specifiers or '...' before 'SQLHSTMT' src/switch_odbc.c: In function 'switch_odbc_handle_get_error': src/switch_odbc.c:476: error: 'SQL_MAX_MESSAGE_LENGTH' undeclared (first use in this function) src/switch_odbc.c:477: error: 'SQL_SQLSTATE_SIZE' undeclared (first use in this function) src/switch_odbc.c:478: error: 'SQLINTEGER' undeclared (first use in this function) src/switch_odbc.c:478: error: expected ';' before 'sqlcode' src/switch_odbc.c:479: error: 'SQLSMALLINT' undeclared (first use in this function) src/switch_odbc.c:479: error: expected ';' before 'length' src/switch_odbc.c:482: warning: implicit declaration of function 'SQLError' src/switch_odbc.c:482: error: 'switch_odbc_handle_t' has no member named 'env' src/switch_odbc.c:482: error: 'switch_odbc_handle_t' has no member named 'con' src/switch_odbc.c:482: error: 'stmt' undeclared (first use in this function) src/switch_odbc.c:482: error: 'SQLCHAR' undeclared (first use in this function) src/switch_odbc.c:482: error: expected expression before ')' token src/switch_odbc.c:482: error: 'SQL_SUCCESS' undeclared (first use in this function) src/switch_odbc.c:483: error: 'sqlcode' undeclared (first use in this function) src/switch_odbc.c:477: warning: unused variable 'sqlstate' src/switch_odbc.c:476: warning: unused variable 'buffer' make[2]: *** [libfreeswitch_la-switch_odbc.lo] Error 1 Making all in src Making all in mod making all mod_amr make[5]: *** No rule to make target `/usr/src/FreeSwitch/freeswitch-1.0.3/libfreeswitch.la', needed by `mod_amr.so'.? Stop. make[4]: *** [all] Error 1 make[3]: *** [mod_amr-all] Error 1 make[2]: *** [all-recursive] Error 1 Making all in build ?+-------- FreeSWITCH Build Complete -----------+ ?+ FreeSWITCH has been successfully built.????? + ?+ Install by running:????????????????????????? + ?+????????????????????????????????????????????? + ?+?????????????? make install?????????????????? + ?+----------------------------------------------+ make[1]: *** [all-recursive] Error 1 make: *** [all] Error 2 --- On Thu, 2/19/09, freeswitch-users-request at lists.freeswitch.org wrote: Message: 3 Date: Thu, 19 Feb 2009 10:26:15 -0500 From: Raymond Chandler Subject: Re: [Freeswitch-users] Realm Value To: freeswitch-users at lists.freeswitch.org Message-ID: <499D7A17.7030202 at freeswitch.org> Content-Type: text/plain; charset="iso-8859-1" Ali Al-Rubaie wrote: > Thanks Brian, > > Sorry if the question looks primitive but in which file I can find the > rvn? > you can find the revision from typing "version" at the fs_cli > Is there any tarball with the latest revisions? > the 1.0.3 tarball was just rolled yesterday, give it a shot -Ray -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090219/89abc2b0/attachment-0001.html ------------------------------ Message: 4 Date: Thu, 19 Feb 2009 16:47:27 +0100 From: "Cavalera Claudio Luigi" Subject: Re: [Freeswitch-users] Random problems with cepstral text to speech To: Message-ID: Content-Type: text/plain; charset="us-ascii" From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale > Are you using cepstral 5.1? > There is a known issue with that release and it's closed source so we cannot do much about it. > Cepstral 4.x works fine. Yes 5.1, my fault. I have added an initial warning here on the wiki http://wiki.freeswitch.org/wiki/Mod_cepstral although it also speaks about 5.1 and Ubuntu... Thanks, Claudio Internet E. Mail Confidentiality Footer ----------------------------------------------------------------------------------------------------- La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. ------------------------------ Message: 5 Date: Thu, 19 Feb 2009 17:00:44 +0100 From: "Cavalera Claudio Luigi" Subject: Re: [Freeswitch-users] Strange error load testing mod_xml_rpc To: Message-ID: Content-Type: text/plain; charset="us-ascii" From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale > Yes we already see you have reported this issue in jira and we are working on it. > you do not need to report it here as well. > You may want to make yourself available on irc or some other im today so we can contact you for more details. I'm sorry I reported this message here because I did not think it's related with my JIRA reports. They are about two segmentation faults, while this message is about load testing not going very well, although it comes out during the same load tests :-) I'm starting to think that most people are using the sofia SIP stack in fs to get rate of 100cps while the other interfaces such as event socket and mod_xml_rpc are not well "engineered" yet. I'm going deeper into this and I hope my results will be of help for the community. At the moment I'm trying to understand the cause of TCP retransmissions I see in the snoop, it seems that fs lags a little bit in sending the TCP Ack back to the load runner, even with the machine doing nothing and a really tiny amount of load. BRs, Claudio Internet Email Confidentiality Footer ----------------------------------------------------------------------------------------------------- La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. ----------------------------------------------------------------------------------------------------- ------------------------------ Message: 6 Date: Thu, 19 Feb 2009 18:00:19 +0100 From: "Cavalera Claudio Luigi" Subject: Re: [Freeswitch-users] Strange error load testing mod_xml_rpc To: Message-ID: Content-Type: text/plain; charset="us-ascii" Here is a snoop: http://pastebin.freeswitch.org/7351 thx, cla Internet Email Confidentiality Footer ----------------------------------------------------------------------------------------------------- La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. ----------------------------------------------------------------------------------------------------- ------------------------------ Message: 7 Date: Thu, 19 Feb 2009 11:30:01 -0600 From: Anthony Minessale Subject: Re: [Freeswitch-users] Strange error load testing mod_xml_rpc To: freeswitch-users at lists.freeswitch.org Message-ID: <191c3a030902190930x121eba94jb644c25922cbdb7b at mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" api calls by default are blocking, it will not return until the result of the originate is determined. you must pre-empt the originate command with the bgapi api command which is similar to the event_socket bgapi command so that it tells the task to run in a dedicated thread. also the web interface was meant to use an xml rpc client. I have generated at least 400cps on event socket before. 2009/2/19 Cavalera Claudio Luigi > Here is a snoop: > http://pastebin.freeswitch.org/7351 > thx, > cla > > > Internet Email Confidentiality Footer > > ----------------------------------------------------------------------------------------------------- > La presente comunicazione, con le informazioni in essa contenute e ogni > documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' > indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete > i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, > comunicazione, divulgazione o simili basate sul contenuto di tali > informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., > D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se > avete ricevuto questa comunicazione per errore, vi preghiamo di darne > immediata notizia al mittente e di distruggere il messaggio originale e ogni > file allegato senza farne copia alcuna o riprodurne in alcun modo il > contenuto. > > This e-mail and its attachments are intended for the addressee(s) only and > are confidential and/or may contain legally privileged information. If you > have received this message by mistake or are not one of the addressees > above, you may take no action based on it, and you may not copy or show it > to anyone; please reply to this e-mail and point out the error which has > occurred. > > ----------------------------------------------------------------------------------------------------- > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090219/0c42b1d8/attachment.html ------------------------------ _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org End of Freeswitch-users Digest, Vol 32, Issue 166 ************************************************* -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090219/67854f71/attachment-0002.html From intralanman at freeswitch.org Thu Feb 19 12:17:21 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Thu, 19 Feb 2009 15:17:21 -0500 Subject: [Freeswitch-users] Realm Value In-Reply-To: <97318.60826.qm@web33708.mail.mud.yahoo.com> References: <97318.60826.qm@web33708.mail.mud.yahoo.com> Message-ID: <499DBE51.5010700@freeswitch.org> did you ./configure --enable-core-odbc-suport... those errors reek of that flag with no unixODBC-devel package installed -Ray From msc at freeswitch.org Thu Feb 19 13:47:09 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 19 Feb 2009 13:47:09 -0800 Subject: [Freeswitch-users] Voice pattern detection In-Reply-To: <20b801c99291$c0dc5c30$33014c0a@ws4> References: <20b801c99291$c0dc5c30$33014c0a@ws4> Message-ID: <87f2f3b90902191347g1750a132k4a71159c44ef54b1@mail.gmail.com> This is really advanced stuff. You're going to need to pay someone who really understands DSP and programming. You might want to start with consulting at freeswitch.org. -MC On Thu, Feb 19, 2009 at 4:58 AM, Frank @ Impact wrote: > I am trying to detect if a caller is an automated greeting voice. And if > so, take an action. > > > > I have samples of the caller recording that I am looking to match. So > this is like a really complex tone detection I guess. > > > > It would work like this. > > - Call comes in > > - We answer/bridge the call > > - We start to listen for about 10 seconds. > > - During this time we are trying to match a snippet of a sound sample (say 2 > or 3 seconds worth) we have recorded on a file on the server. We are trying > to match this sound sample to the caller side only. > > - If we hear a match, we take the action. Or if we don't hear the match, we > might take a different action. > > > > Any of this sound doable? Any guidance on how to accomplish this? > > > > -Frank > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From BenHoltsclaw at averyschools.net Thu Feb 19 12:23:26 2009 From: BenHoltsclaw at averyschools.net (Ben Holtsclaw) Date: Thu, 19 Feb 2009 15:23:26 -0500 Subject: [Freeswitch-users] Deployment information and use cases In-Reply-To: <1235016793.22050.0.camel@raul-laptop> References: <1234916455.16581.49.camel@raul-laptop> <59ad9ca10902181009p7cbc0fd1s9860aacde7f3a30c@mail.gmail.com> <247f8100902181019m61ce803m5ed504d9b198dfb6@mail.gmail.com> <1235016793.22050.0.camel@raul-laptop> Message-ID: <499D796E.45B7.0079.0@averyschools.net> Raul, I am in the process of rolling out a FreeSWITCH IP PBX solution similar to what you describe. When I was trying to procure funds for a FreeSWITCH solution, I looked for the same information you're after, but came up with little. I'll briefly describe what we're trying to accomplish, and the tools I'm using to do it. This is probably more information than what you are looking for, but maybe it will also benefit someone else. We had several schools with aging or dying PBX's or KSU's. Each site had something different system, and was supported by a different VAR. Of course, the VAR's charged some outlandish fee to make onsite repair visits. Some number of Centrex lines supplied each school's dial tone. All in all, we had a very outdated and financially draining mess. Our district's long term goal had been to move to a more unified phone system. That made sense for many reasons, the chief of which was cost. We already had a strong fiber WAN in place. Why not use that for trunking and eliminate the monthly cost of the Centrex lines? That's the path we started down. Being a public entity, we had to be sure to explore all possible avenues. We looked at everything from traditional PBX's with IP add-on modules for trunking to a full blown Cisco CallManager solution. With third party proprietary systems, we were just never able to find the sweet spot between required feature set and cost. Would Cisco have been a workable solution? Absolutely. Could our small, rural, K12 public school district afford that? Not in a million years. I looked at several software packages -- some open source, some not -- but always came back to FreeSWITCH. The scalability and active development community were major factors for us. Server Hardware. Each of our five sites has a dedicated FreeSWITCH server. For hardware, we went with Dell PowerEdge 1950's with dual quad core Xeon 2.33 GHz processors, and 4GB of RAM. I have mirrored disks set up with enough space to accommodate users' voicemail. Each server will average only about 60 voicemail boxes, and we're storing sound as MP3. Disk space shouldn't be an issue. We have always been a Novell shop, so SLES is naturally our Linux distribution of choice. We chose to go with server hardware at each site so that in the event of a WAN outage, we would still at least have intra-building and emergency communication (see below). Telephony Hardware. Each of our servers includes Sangoma hardware. We actually looked at doing IP trunking to a carrier from our network core, but decided to use telco provided PRI's instead. Presently, we have two PRI's that connect to a FreeSWITCH server at the center of our network via a Sangoma A102 dual port telephony card. All calls to and from the PSTN traverse this primary server. Servers at each remote site include one of Sangoma's A200 analog cards. Emergency calls to 911 route out over this analog card through one of at least two POTS lines that remain connected at each site. Not only does this provide some redundancy in the event of a WAN outage, but it ensures proper caller location is delivered to the 911 dispatcher. Granted, there are some other solutions for the latter, but this seemed to be the most cost effective solution for us. Telephone Desksets. We chose to go with Aastra for the telephones. The standard phone that we will place in each classroom and office is the 9143i. This is an attractive phone with an adequate feature set at a price we can afford. The person that is primarily responsible for answering the phone at each site will have an Aastra 57i and some number of 560M expansion modules. We have purchased roughly 300 Aastra desksets. Logical Layout. As new sites come online, their primary phone number is being moved from the Centrex to our PRI group. All inbound calls hit our primary server, and then FreeSWITCH bridges to the appropriate secondary server based on the DID it received. On the reverse, each servers dial plan is set up to route outbound calls (save 911) to the primary server where FreeSWITCH bridges with Openzap. Site to site calls, accomplished via four digit dialing, do not hit the primary server. Outbound calls to the PSTN deliver the site's DID as the calling number. In other words, if a user from site two calls my cell phone, I see site two's published telephone number on my caller ID. Our dial plans are set up so that receptionists at each site still answer all outside calls. If not answered, the call fails over to an IVR. Should we ever decide to do so, we are now perfectly positioned to have all inbound calls to the district answered by one operator or IVR. "Welcome, and thank you for calling Avery County Schools." Stumbling Blocks. Problems we've faced so far have primarily surrounded Openzap and the Sangoma Wanpipe driver. FreeSWITCH developers won't mind telling you that this is an area that is currently not well "funded" and not 100% complete. There is some known issue where voice channels on the PRI get stuck in the wrong state and become unusable. We have experienced this a couple of times and have not been able to make or receive calls. Bouncing the Wanpipe driver has fixed this each time. We have also had trouble with DTMF detection across the PRI. If a user hits the IVR, it is oftentimes difficult to get it to properly recognize the digits that are being keyed in by the caller. This can be very, very frustrating to a caller that doesn't want to deal with an IVR anyway. The developers have suggested to me that this is a problem with the Sangoma's echo cancellation goofing up Openzap's ability to interpret the DTMF. The Sangoma hardware does have its own DTMF decoder and API, but the Openzap code currently does not make use of it. I have created a patch that makes use of the hardware decoder. We have been running it in production for a couple of weeks, and that does seem to have helped the problem. The problem hasn't gone away altogether. Those have been our two biggest issues, but we haven't let them hold us up. Conclusion. Of the five sites that will be on this system, one is fully functional with calls inbound and outbound from the PSTN. A second site is up and running with full outbound PSTN access. Their inbound DID is scheduled to move over to the PRI in one week. The server has been worked up for a third site, and the phones are starting to roll out. Sites four and five should come online by the end of April. Currently, I don't have numbers compiled for things like concurrent calls. At this point in my project, it is just not important. I really don't think our implementation will ever push FreeSWITCH's abilities in that regard. I base that statement primarily on other users' benchmarks, and what I've heard some are doing in carrier class environments. FreeSWITCH has made our project viable. An open source solution was the only way we could meet all of the project goals and stay within our budget. FreeSWITCH has proven to have all the features we require in a district wide phone system. It has not locked us into annual support contracts with third party vendors. I could go on with the accolades. However, I'll end this terribly lengthy post by saying that, overall, we have been very pleased with our choice to go with FreeSWITCH. The information in this email will seem very elementary to most people on this list, but having a message of this nature in hand would have made me feel much more confident the first time I ever went to my supervisor to mention something called FreeSWITCH. :-) Thanks Tony, Brian, and Mike for a great product! Ben Holtsclaw Network Engineer Avery County Schools Ph: 828.733.3567 x2301 >>> On 2/18/2009 at 11:13 PM, Raul Fragoso wrote: Thanks guys, this is very useful information. Anyone else willing to share your experience ? Regards, Raul On Wed, 2009-02-18 at 16:19 -0200, Pablo Hernan Saro wrote: > Hi Raul, > > In my company (http://www.globant.com) we're using FreeSWITCH for high > quality conference services, integrated with OpenSIPS > (http://www.opensips.org) and Asterisk. Its performance is pretty > good. > > Pablo > > On Wed, Feb 18, 2009 at 4:09 PM, Henry Huang wrote: > > bandwidth.com has a service called phonebooth which is developed upon > > freeswitch. > > > > > > On Tue, Feb 17, 2009 at 4:20 PM, Raul Fragoso wrote: > >> > >> Hello FreeSWITCHERS, > >> > >> My company is currently creating a suite of applications which uses > >> FreeSWITCH as the back-end for an IP-PBX solution. We currently have a > >> prospect to have our first customer installation - a governmental > >> department. That is a tender to have an IP-PBX installation to connect > >> their four office branches, each one with about 300 users - which I am > >> sure FreeSWITCH is able to handle. Since this is an official tender, > >> it's part of their protocol to ask about real sites using the product. > >> > >> Having said that, would you mind sharing some information about your > >> experience with FreeSWITCH deployments ? > >> > >> No need to give many details, but a short summary with company name (if > >> possible), when it was deployed, server equipment, number of users, > >> number of concurrent calls, what kind of functions and services are used > >> and overall capacity of the system. > >> > >> I would really appreciate if you can share that information. And if you > >> guys agree (and explicitly manifest your agreement), I can compile the > >> information in the FreeSWITCH wiki under a "Use Cases" page so it can > >> serve as a common reference as well. > >> > >> Kind regards, > >> > >> Raul Fragoso > >> > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > > > -- > > Henry Huang > > UniC Solution - Communication Unified > > VoIP & Open Source software Consultant > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090219/e0bf6b3e/attachment-0002.html From brian at freeswitch.org Thu Feb 19 13:50:34 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 19 Feb 2009 15:50:34 -0600 Subject: [Freeswitch-users] Voice pattern detection In-Reply-To: <87f2f3b90902191347g1750a132k4a71159c44ef54b1@mail.gmail.com> References: <20b801c99291$c0dc5c30$33014c0a@ws4> <87f2f3b90902191347g1750a132k4a71159c44ef54b1@mail.gmail.com> Message-ID: Plus its not an exact science in the first place. /b On Feb 19, 2009, at 3:47 PM, Michael Collins wrote: > This is really advanced stuff. You're going to need to pay someone who > really understands DSP and programming. You might want to start with > consulting at freeswitch.org. > > -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090219/5a405300/attachment-0002.html From frank at impactfax.com Thu Feb 19 14:12:44 2009 From: frank at impactfax.com (Frank @ Impact) Date: Thu, 19 Feb 2009 17:12:44 -0500 Subject: [Freeswitch-users] Voice pattern detection In-Reply-To: <87f2f3b90902191347g1750a132k4a71159c44ef54b1@mail.gmail.com> Message-ID: <253f01c992df$30d5a580$33014c0a@ws4> Ok. Maybe it is more like answering machine detection in reverse? Detection on the caller leg instead of the called leg. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Thursday, February 19, 2009 4:47 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Voice pattern detection This is really advanced stuff. You're going to need to pay someone who really understands DSP and programming. You might want to start with consulting at freeswitch.org. From jaugenstine at gmail.com Thu Feb 19 14:13:39 2009 From: jaugenstine at gmail.com (jonathan augenstine) Date: Thu, 19 Feb 2009 14:13:39 -0800 Subject: [Freeswitch-users] Pika development Message-ID: <207e7a5e0902191413j4e7ac724uf6abc4ee94f2d45b@mail.gmail.com> I have heard a rumor that Pika support was being developed for Freeswitch. Is that still going on? Can someone tell me if the rumor is true or not, and if so, what is the status of the development? Thank you. Jonathan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090219/e66cc465/attachment-0002.html From brian at freeswitch.org Thu Feb 19 14:19:53 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 19 Feb 2009 16:19:53 -0600 Subject: [Freeswitch-users] Pika development In-Reply-To: <207e7a5e0902191413j4e7ac724uf6abc4ee94f2d45b@mail.gmail.com> References: <207e7a5e0902191413j4e7ac724uf6abc4ee94f2d45b@mail.gmail.com> Message-ID: <3C32E446-8E6B-4738-AAE5-4422CD782979@freeswitch.org> Pika hardware already works with OpenZAP. /b On Feb 19, 2009, at 4:13 PM, jonathan augenstine wrote: > I have heard a rumor that Pika support was being developed for > Freeswitch. Is that still going on? Can someone tell me if the > rumor is true or not, and if so, what is the status of the > development? > > Thank you. > Jonathan From msc at freeswitch.org Thu Feb 19 14:22:31 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 19 Feb 2009 14:22:31 -0800 Subject: [Freeswitch-users] Pika development In-Reply-To: <207e7a5e0902191413j4e7ac724uf6abc4ee94f2d45b@mail.gmail.com> References: <207e7a5e0902191413j4e7ac724uf6abc4ee94f2d45b@mail.gmail.com> Message-ID: <87f2f3b90902191422w66834fe5m6764508ae08e2dd8@mail.gmail.com> On Thu, Feb 19, 2009 at 2:13 PM, jonathan augenstine wrote: > I have heard a rumor that Pika support was being developed for Freeswitch. > Is that still going on? Can someone tell me if the rumor is true or not, > and if so, what is the status of the development? Well, the PIKA cards work with FS and they have an appliance they were showing off at ClueCon last year... not sure what else is in the pipeline. -MC From msc at freeswitch.org Thu Feb 19 14:35:29 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 19 Feb 2009 14:35:29 -0800 Subject: [Freeswitch-users] Realm Value In-Reply-To: <499DBE51.5010700@freeswitch.org> References: <97318.60826.qm@web33708.mail.mud.yahoo.com> <499DBE51.5010700@freeswitch.org> Message-ID: <87f2f3b90902191435p1c9c03aend3303dfb013495b1@mail.gmail.com> On Thu, Feb 19, 2009 at 12:17 PM, Raymond Chandler wrote: > did you ./configure --enable-core-odbc-suport... those errors reek of > that flag with no unixODBC-devel package installed > > -Ray > Anthony described this as a false positive on detecting ODBC. If you are in Linux you can install the ODBC-devel package and be done with it. -MC From anthony.minessale at gmail.com Thu Feb 19 16:56:44 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 19 Feb 2009 18:56:44 -0600 Subject: [Freeswitch-users] Pika development In-Reply-To: <87f2f3b90902191422w66834fe5m6764508ae08e2dd8@mail.gmail.com> References: <207e7a5e0902191413j4e7ac724uf6abc4ee94f2d45b@mail.gmail.com> <87f2f3b90902191422w66834fe5m6764508ae08e2dd8@mail.gmail.com> Message-ID: <191c3a030902191656l788940c8uabe8077e3c81ad17@mail.gmail.com> If you get a pika card to play with on FS, please inform them that it is for that purpose so they can help make sure you get it going. On Thu, Feb 19, 2009 at 4:22 PM, Michael Collins wrote: > On Thu, Feb 19, 2009 at 2:13 PM, jonathan augenstine > wrote: > > I have heard a rumor that Pika support was being developed for > Freeswitch. > > Is that still going on? Can someone tell me if the rumor is true or not, > > and if so, what is the status of the development? > > Well, the PIKA cards work with FS and they have an appliance they were > showing off at ClueCon last year... not sure what else is in the > pipeline. > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090219/4be497a9/attachment-0002.html From gmaruzz at celliax.org Thu Feb 19 17:32:46 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Fri, 20 Feb 2009 02:32:46 +0100 Subject: [Freeswitch-users] Skypiax (Skype Network endpoint) for FreeSWITCH In-Reply-To: <5800526b0902181853h32de5535l6c0f9d7067a69cf0@mail.gmail.com> References: <7b197bef0902131015k456dee87x8f0f1b7059f1b49c@mail.gmail.com> <5800526b0902181853h32de5535l6c0f9d7067a69cf0@mail.gmail.com> Message-ID: <7b197bef0902191732i6fead849uace0ac906a9437b0@mail.gmail.com> On Thu, Feb 19, 2009 at 3:53 AM, Carlos Talbot wrote: > One question I have, is ringback suppose to work with mod_skypiax? Whenever > I dial a number I get a few seconds of dead air before the call is answered. > I've tried adding ringback and transfer_ringback into the dialplan just > before the bridge command but no go. Am I missing something? Thanks. Carlos, ringback now works without tricks, and Skypiax is in trunk. Both remote ringing and early media are treated as remote ringing right now (eg: no early media, just ringing). I'll add early media support in the near future. Thanks a lot for testing and exercising skypiax, and please let me know any hint, suggestion, feature request, etc Sincerely, Giovanni Maruzzelli ========================================= www.celliax.org via Pierlombardo 9, 20135 Milano Italy gmaruzz at celliax dot org Cell : +39-347-2665618 Fax : +39-02-87390039 On Thu, Feb 19, 2009 at 3:53 AM, Carlos Talbot wrote: > Giovannia, > > great work on mod_skypiax. I've been testing it under Windows and it sounds > great including PSTN calls. I plan to include it as part of the Windows MSI > build. > > One question I have, is ringback suppose to work with mod_skypiax? Whenever > I dial a number I get a few seconds of dead air before the call is answered. > I've tried adding ringback and transfer_ringback into the dialplan just > before the bridge command but no go. Am I missing something? Thanks. > > regards, > > Carlos > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From jaugenstine at gmail.com Thu Feb 19 18:29:33 2009 From: jaugenstine at gmail.com (jonathan augenstine) Date: Thu, 19 Feb 2009 18:29:33 -0800 Subject: [Freeswitch-users] Pika development In-Reply-To: <191c3a030902191656l788940c8uabe8077e3c81ad17@mail.gmail.com> References: <207e7a5e0902191413j4e7ac724uf6abc4ee94f2d45b@mail.gmail.com> <87f2f3b90902191422w66834fe5m6764508ae08e2dd8@mail.gmail.com> <191c3a030902191656l788940c8uabe8077e3c81ad17@mail.gmail.com> Message-ID: <207e7a5e0902191829o6f23e2dej6362461f212ad939@mail.gmail.com> Anthony/Michael/Brian, Thank you for all the input. I appreciate the responses. I will certainly make sure they are aware of the application if I get the green light. Jonathan On Thu, Feb 19, 2009 at 4:56 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > If you get a pika card to play with on FS, please inform them that it is > for that purpose so they > can help make sure you get it going. > > > > On Thu, Feb 19, 2009 at 4:22 PM, Michael Collins wrote: > >> On Thu, Feb 19, 2009 at 2:13 PM, jonathan augenstine >> wrote: >> > I have heard a rumor that Pika support was being developed for >> Freeswitch. >> > Is that still going on? Can someone tell me if the rumor is true or >> not, >> > and if so, what is the status of the development? >> >> Well, the PIKA cards work with FS and they have an appliance they were >> showing off at ClueCon last year... not sure what else is in the >> pipeline. >> -MC >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090219/0d8df332/attachment-0002.html From cesar.bermudez at gmail.com Thu Feb 19 18:38:31 2009 From: cesar.bermudez at gmail.com (Cesar Bermudez) Date: Fri, 20 Feb 2009 03:38:31 +0100 Subject: [Freeswitch-users] Pika development In-Reply-To: <207e7a5e0902191829o6f23e2dej6362461f212ad939@mail.gmail.com> References: <207e7a5e0902191413j4e7ac724uf6abc4ee94f2d45b@mail.gmail.com> <87f2f3b90902191422w66834fe5m6764508ae08e2dd8@mail.gmail.com> <191c3a030902191656l788940c8uabe8077e3c81ad17@mail.gmail.com> <207e7a5e0902191829o6f23e2dej6362461f212ad939@mail.gmail.com> Message-ID: this is for the pika warp http://svn.pikatech.com/pads/distro/branches/freeswitch-1.0.0/ On Fri, Feb 20, 2009 at 3:29 AM, jonathan augenstine wrote: > Anthony/Michael/Brian, > > Thank you for all the input. I appreciate the responses. I will certainly > make sure they are aware of the application if I get the green light. > > Jonathan > > > On Thu, Feb 19, 2009 at 4:56 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> If you get a pika card to play with on FS, please inform them that it is >> for that purpose so they >> can help make sure you get it going. >> >> >> >> On Thu, Feb 19, 2009 at 4:22 PM, Michael Collins wrote: >> >>> On Thu, Feb 19, 2009 at 2:13 PM, jonathan augenstine >>> wrote: >>> > I have heard a rumor that Pika support was being developed for >>> Freeswitch. >>> > Is that still going on? Can someone tell me if the rumor is true or >>> not, >>> > and if so, what is the status of the development? >>> >>> Well, the PIKA cards work with FS and they have an appliance they were >>> showing off at ClueCon last year... not sure what else is in the >>> pipeline. >>> -MC >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:213-799-1400 >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090220/6b95ee8e/attachment-0002.html From jason at jasonjgw.net Thu Feb 19 18:52:09 2009 From: jason at jasonjgw.net (Jason White) Date: Fri, 20 Feb 2009 13:52:09 +1100 Subject: [Freeswitch-users] Calling an IPv6 host with a SIP URI from portAudio Message-ID: <20090220025209.GA12844@jdc.jasonjgw.net> I notice that pa call sip:nnnn at host fails if the host in question only has an IPv6 address (i.e., an AAAA record in DNS). The logs show that FreeSWITCH is trying to use the internal profile, and failing. If I write a dialplan extension that accesses the same address using the internal-ipv6 profile, it succeeds. In the supplied default.xml dial plan, the SIP URI is processed thus: I can't find any documentation of use_profile on the wiki, but clearly it takes the value "internal" in this case. What would be the best way to fix this so that it will work regardless of whether the host is reachable over IPv4 or IPv6, or both? I could rewrite the extension to try multiple SIP profiles, but there could be a better way - hence the question. From brian at freeswitch.org Thu Feb 19 18:55:39 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 19 Feb 2009 20:55:39 -0600 Subject: [Freeswitch-users] Calling an IPv6 host with a SIP URI from portAudio In-Reply-To: <20090220025209.GA12844@jdc.jasonjgw.net> References: <20090220025209.GA12844@jdc.jasonjgw.net> Message-ID: <4F757A97-34E5-48A0-94D2-07883BFE23C3@freeswitch.org> Just make sure you send the call out an ipv6 profile and it'll work. /b On Feb 19, 2009, at 8:52 PM, Jason White wrote: > > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090219/aa71b4bd/attachment-0002.html From jason at jasonjgw.net Thu Feb 19 19:53:17 2009 From: jason at jasonjgw.net (Jason White) Date: Fri, 20 Feb 2009 14:53:17 +1100 Subject: [Freeswitch-users] Calling an IPv6 host with a SIP URI from portAudio In-Reply-To: <4F757A97-34E5-48A0-94D2-07883BFE23C3@freeswitch.org> References: <20090220025209.GA12844@jdc.jasonjgw.net> <4F757A97-34E5-48A0-94D2-07883BFE23C3@freeswitch.org> Message-ID: <20090220035317.GA13566@jdc.jasonjgw.net> I have it working now. The relevant changes were as follows. From raul at etellicom.com Thu Feb 19 20:18:00 2009 From: raul at etellicom.com (Raul Fragoso) Date: Fri, 20 Feb 2009 01:18:00 -0300 Subject: [Freeswitch-users] Deployment information and use cases In-Reply-To: <499D796E.45B7.0079.0@averyschools.net> References: <1234916455.16581.49.camel@raul-laptop> <59ad9ca10902181009p7cbc0fd1s9860aacde7f3a30c@mail.gmail.com> <247f8100902181019m61ce803m5ed504d9b198dfb6@mail.gmail.com> <1235016793.22050.0.camel@raul-laptop> <499D796E.45B7.0079.0@averyschools.net> Message-ID: <1235103480.30511.346.camel@raul-laptop> Ben, Wow !!! Thank you very much for such descriptive and detailed information ! Indeed, this is really more than I expected, and once again I thank you for your collaboration. It's very cheering and inspiring to hear such successful story regarding FreeSWITCH. Kind regards, Raul On Thu, 2009-02-19 at 15:23 -0500, Ben Holtsclaw wrote: > Raul, > > I am in the process of rolling out a FreeSWITCH IP PBX solution > similar to what you describe. When I was trying to procure funds for a > FreeSWITCH solution, I looked for the same information you're after, > but came up with little. I'll briefly describe what we're trying to > accomplish, and the tools I'm using to do it. This is probably more > information than what you are looking for, but maybe it will also > benefit someone else. > > We had several schools with aging or dying PBX's or KSU's. Each site > had something different system, and was supported by a different > VAR. Of course, the VAR's charged some outlandish fee to make onsite > repair visits. Some number of Centrex lines supplied each school's > dial tone. All in all, we had a very outdated and financially draining > mess. Our district's long term goal had been to move to a more unified > phone system. That made sense for many reasons, the chief of which was > cost. We already had a strong fiber WAN in place. Why not use that for > trunking and eliminate the monthly cost of the Centrex lines? That's > the path we started down. > > Being a public entity, we had to be sure to explore all possible > avenues. We looked at everything from traditional PBX's with IP add-on > modules for trunking to a full blown Cisco CallManager solution. With > third party proprietary systems, we were just never able to find the > sweet spot between required feature set and cost. Would Cisco have > been a workable solution? Absolutely. Could our small, rural, K12 > public school district afford that? Not in a million years. I looked > at several software packages -- some open source, some not -- but > always came back to FreeSWITCH. The scalability and active development > community were major factors for us. > > Server Hardware. Each of our five sites has a dedicated FreeSWITCH > server. For hardware, we went with Dell PowerEdge 1950's with dual > quad core Xeon 2.33 GHz processors, and 4GB of RAM. I have mirrored > disks set up with enough space to accommodate users' voicemail. Each > server will average only about 60 voicemail boxes, and we're storing > sound as MP3. Disk space shouldn't be an issue. We have always been a > Novell shop, so SLES is naturally our Linux distribution of choice. We > chose to go with server hardware at each site so that in the event of > a WAN outage, we would still at least have intra-building and > emergency communication (see below). > > Telephony Hardware. Each of our servers includes Sangoma hardware. We > actually looked at doing IP trunking to a carrier from our network > core, but decided to use telco provided PRI's instead. Presently, we > have two PRI's that connect to a FreeSWITCH server at the center of > our network via a Sangoma A102 dual port telephony card. All calls to > and from the PSTN traverse this primary server. Servers at each remote > site include one of Sangoma's A200 analog cards. Emergency calls to > 911 route out over this analog card through one of at least two POTS > lines that remain connected at each site. Not only does this provide > some redundancy in the event of a WAN outage, but it ensures proper > caller location is delivered to the 911 dispatcher. Granted, there are > some other solutions for the latter, but this seemed to be the most > cost effective solution for us. > > Telephone Desksets. We chose to go with Aastra for the telephones. The > standard phone that we will place in each classroom and office is the > 9143i. This is an attractive phone with an adequate feature set at a > price we can afford. The person that is primarily responsible for > answering the phone at each site will have an Aastra 57i and some > number of 560M expansion modules. We have purchased roughly 300 Aastra > desksets. > > Logical Layout. As new sites come online, their primary phone number > is being moved from the Centrex to our PRI group. All inbound calls > hit our primary server, and then FreeSWITCH bridges to the appropriate > secondary server based on the DID it received. On the reverse, each > servers dial plan is set up to route outbound calls (save 911) to the > primary server where FreeSWITCH bridges with Openzap. Site to site > calls, accomplished via four digit dialing, do not hit the primary > server. Outbound calls to the PSTN deliver the site's DID as the > calling number. In other words, if a user from site two calls my cell > phone, I see site two's published telephone number on my caller ID. > Our dial plans are set up so that receptionists at each site still > answer all outside calls. If not answered, the call fails over to an > IVR. Should we ever decide to do so, we are now perfectly positioned > to have all inbound calls to the district answered by one operator or > IVR. "Welcome, and thank you for calling Avery County Schools." > > Stumbling Blocks. Problems we've faced so far have primarily > surrounded Openzap and the Sangoma Wanpipe driver. FreeSWITCH > developers won't mind telling you that this is an area that is > currently not well "funded" and not 100% complete. There is some known > issue where voice channels on the PRI get stuck in the wrong state and > become unusable. We have experienced this a couple of times and have > not been able to make or receive calls. Bouncing the Wanpipe driver > has fixed this each time. We have also had trouble with DTMF detection > across the PRI. If a user hits the IVR, it is oftentimes difficult to > get it to properly recognize the digits that are being keyed in by the > caller. This can be very, very frustrating to a caller that doesn't > want to deal with an IVR anyway. The developers have suggested to me > that this is a problem with the Sangoma's echo cancellation goofing up > Openzap's ability to interpret the DTMF. The Sangoma hardware does > have its own DTMF decoder and API, but the Openzap code currently does > not make use of it. I have created a patch that makes use of the > hardware decoder. We have been running it in production for a couple > of weeks, and that does seem to have helped the problem. The problem > hasn't gone away altogether. Those have been our two biggest issues, > but we haven't let them hold us up. > > Conclusion. Of the five sites that will be on this system, one is > fully functional with calls inbound and outbound from the PSTN. A > second site is up and running with full outbound PSTN access. Their > inbound DID is scheduled to move over to the PRI in one week. The > server has been worked up for a third site, and the phones are > starting to roll out. Sites four and five should come online by the > end of April. Currently, I don't have numbers compiled for things like > concurrent calls. At this point in my project, it is just not > important. I really don't think our implementation will ever push > FreeSWITCH's abilities in that regard. I base that statement primarily > on other users' benchmarks, and what I've heard some are doing in > carrier class environments. > > FreeSWITCH has made our project viable. An open source solution was > the only way we could meet all of the project goals and stay within > our budget. FreeSWITCH has proven to have all the features we require > in a district wide phone system. It has not locked us into annual > support contracts with third party vendors. I could go on with the > accolades. However, I'll end this terribly lengthy post by saying > that, overall, we have been very pleased with our choice to go with > FreeSWITCH. > > The information in this email will seem very elementary to most people > on this list, but having a message of this nature in hand would have > made me feel much more confident the first time I ever went to my > supervisor to mention something called FreeSWITCH. :-) Thanks Tony, > Brian, and Mike for a great product! > > > Ben Holtsclaw > Network Engineer > Avery County Schools > Ph: 828.733.3567 x2301 > > > >>> On 2/18/2009 at 11:13 PM, Raul Fragoso wrote: > > Thanks guys, this is very useful information. > > Anyone else willing to share your experience ? > > Regards, > > Raul > > On Wed, 2009-02-18 at 16:19 -0200, Pablo Hernan Saro wrote: > > Hi Raul, > > > > In my company (http://www.globant.com) we're using FreeSWITCH for > high > > quality conference services, integrated with OpenSIPS > > (http://www.opensips.org) and Asterisk. Its performance is pretty > > good. > > > > Pablo > > > > On Wed, Feb 18, 2009 at 4:09 PM, Henry Huang > wrote: > > > bandwidth.com has a service called phonebooth which is developed > upon > > > freeswitch. > > > > > > > > > On Tue, Feb 17, 2009 at 4:20 PM, Raul Fragoso > wrote: > > >> > > >> Hello FreeSWITCHERS, > > >> > > >> My company is currently creating a suite of applications which > uses > > >> FreeSWITCH as the back-end for an IP-PBX solution. We currently > have a > > >> prospect to have our first customer installation - a governmental > > >> department. That is a tender to have an IP-PBX installation to > connect > > >> their four office branches, each one with about 300 users - which > I am > > >> sure FreeSWITCH is able to handle. Since this is an official > tender, > > >> it's part of their protocol to ask about real sites using the > product. > > >> > > >> Having said that, would you mind sharing some information about > your > > >> experience with FreeSWITCH deployments ? > > >> > > >> No need to give many details, but a short summary with company > name (if > > >> possible), when it was deployed, server equipment, number of > users, > > >> number of concurrent calls, what kind of functions and services > are used > > >> and overall capacity of the system. > > >> > > >> I would really appreciate if you can share that information. And > if you > > >> guys agree (and explicitly manifest your agreement), I can > compile the > > >> information in the FreeSWITCH wiki under a "Use Cases" page so it > can > > >> serve as a common reference as well. > > >> > > >> Kind regards, > > >> > > >> Raul Fragoso > > >> > > >> > > >> _______________________________________________ > > >> Freeswitch-users mailing list > > >> Freeswitch-users at lists.freeswitch.org > > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > >> http://www.freeswitch.org > > > > > > > > > > > > -- > > > Henry Huang > > > UniC Solution - Communication Unified > > > VoIP & Open Source software Consultant > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From stevecrozz at gmail.com Thu Feb 19 21:27:05 2009 From: stevecrozz at gmail.com (Stephen Crosby) Date: Thu, 19 Feb 2009 21:27:05 -0800 Subject: [Freeswitch-users] js session.streamFile() interrupt slowness Message-ID: <11990ade0902192127l414ecfeaia04fb80f626c35d5@mail.gmail.com> I have a few scripts that use the javascript session.streamFile('somefile.wav', onDtmf); where onDtmf is a function that returns false to interrupt the streaming file. There is a short delay between the time when I press a key and the time the file stops playing. Is there anything I can adjust that would affect that? It's only maybe 2-3 seconds, but it "feels" too long to me. --Stephen From msc at freeswitch.org Thu Feb 19 22:22:19 2009 From: msc at freeswitch.org (Michael S Collins) Date: Thu, 19 Feb 2009 22:22:19 -0800 Subject: [Freeswitch-users] Calling an IPv6 host with a SIP URI from portAudio In-Reply-To: <20090220035317.GA13566@jdc.jasonjgw.net> References: <20090220025209.GA12844@jdc.jasonjgw.net> <4F757A97-34E5-48A0-94D2-07883BFE23C3@freeswitch.org> <20090220035317.GA13566@jdc.jasonjgw.net> Message-ID: <9426570E-EE76-4BAD-9E44-1AA5622FFA8E@freeswitch.org> On Feb 19, 2009, at 7:53 PM, Jason White wrote: > I have it working now. The relevant changes were as follows. > > > > > > > Jason, I like this approach. Good use of the many Dialplan tools. -MC From pmhshz at gmail.com Fri Feb 20 02:18:29 2009 From: pmhshz at gmail.com (shehzad p) Date: Fri, 20 Feb 2009 02:18:29 -0800 (PST) Subject: [Freeswitch-users] Suggestion for xml_curl performance Message-ID: <22118122.post@talk.nabble.com> Hi all, Recently I faced some performance bottleneck by using Javascript. Now I am testing xml_curl for next setup, In terms of performance and stability will any body give me some information, what are the pros & cons of using xml_curl, what is precaution for using it, or any other recommendations... Thanks, msp -- View this message in context: http://www.nabble.com/Suggestion-for-xml_curl-performance-tp22118122p22118122.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From pmhshz at gmail.com Fri Feb 20 02:59:24 2009 From: pmhshz at gmail.com (shehzad p) Date: Fri, 20 Feb 2009 02:59:24 -0800 (PST) Subject: [Freeswitch-users] Suggestion for xml_curl performance In-Reply-To: <22118122.post@talk.nabble.com> References: <22118122.post@talk.nabble.com> Message-ID: <22118614.post@talk.nabble.com> My setup of the system is like: When calls come from Originator gateway I route the call to Terminator Gateway based on database lookup. FS works as switching platform. In previous setup using JavaScript, JavaScript caused the performance bottleneck when call traffic increases. Now I am testing xml_curl so asking for any suggestion, if some one has experienced... shehzad p wrote: > > Hi all, > > Recently I faced some performance bottleneck by using Javascript. > > Now I am testing xml_curl for next setup, > In terms of performance and stability will any body give me some > information, what are the pros & cons of using xml_curl, what is > precaution for using it, or any other recommendations... > > Thanks, > msp > -- View this message in context: http://www.nabble.com/Suggestion-for-xml_curl-performance-tp22118122p22118614.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From kokoska.rokoska at post.cz Fri Feb 20 03:24:20 2009 From: kokoska.rokoska at post.cz (kokoska rokoska) Date: Fri, 20 Feb 2009 12:24:20 +0100 Subject: [Freeswitch-users] SIP dump to DB Message-ID: <499E92E4.5010503@post.cz> Hi all, I'm facing the problem I need all SIP messages "going thru" FreeSWITCH (I know FS i B2BUA - so, better to say just "all SIP messages") logged somewhere and this log have to be "searchable" (by call-id etc) and I should be able to simply delete "old" messages... And more over - it should be done on not trivial SIP messages amount - say hundreds messages per second. My questions are: 1. Do you have any suggestion how to do it with FreeSWITCH? 2. Or it is not possible now, and "bounty" is necessary? :-) 3. How hard it will be to implement? ---------- FYI: I think something like SER/Kamailio/OpenSIPS siptrace is what I'am (probably) looking for: http://www.kamailio.org/docs/modules/1.4.x/siptrace.html It could go (based on my tests) up to 7-10.000 messages per second till MySQL "dies"... ---------- Thanks for your time :-) Best regards, kokoska.rokoska From anthony.minessale at gmail.com Fri Feb 20 05:54:59 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 20 Feb 2009 07:54:59 -0600 Subject: [Freeswitch-users] SIP dump to DB In-Reply-To: <499E92E4.5010503@post.cz> References: <499E92E4.5010503@post.cz> Message-ID: <191c3a030902200554s4abda94g76ba4e2975fffa98@mail.gmail.com> what exact info do you need? That's likely to be a challenge with any database to store at that speed. On Fri, Feb 20, 2009 at 5:24 AM, kokoska rokoska wrote: > > Hi all, > > I'm facing the problem I need all SIP messages "going thru" FreeSWITCH > (I know FS i B2BUA - so, better to say just "all SIP messages") logged > somewhere and this log have to be "searchable" (by call-id etc) and I > should be able to simply delete "old" messages... > And more over - it should be done on not trivial SIP messages amount - > say hundreds messages per second. > > My questions are: > 1. Do you have any suggestion how to do it with FreeSWITCH? > 2. Or it is not possible now, and "bounty" is necessary? :-) > 3. How hard it will be to implement? > > ---------- > FYI: > I think something like SER/Kamailio/OpenSIPS siptrace is what I'am > (probably) looking for: > http://www.kamailio.org/docs/modules/1.4.x/siptrace.html > It could go (based on my tests) up to 7-10.000 messages per second till > MySQL "dies"... > ---------- > > Thanks for your time :-) > > Best regards, > > kokoska.rokoska > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090220/af47481d/attachment-0002.html From anthony.minessale at gmail.com Fri Feb 20 06:21:08 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 20 Feb 2009 08:21:08 -0600 Subject: [Freeswitch-users] Calling an IPv6 host with a SIP URI from portAudio In-Reply-To: <9426570E-EE76-4BAD-9E44-1AA5622FFA8E@freeswitch.org> References: <20090220025209.GA12844@jdc.jasonjgw.net> <4F757A97-34E5-48A0-94D2-07883BFE23C3@freeswitch.org> <20090220035317.GA13566@jdc.jasonjgw.net> <9426570E-EE76-4BAD-9E44-1AA5622FFA8E@freeswitch.org> Message-ID: <191c3a030902200621i37e61312r36998c73164dc7da@mail.gmail.com> another way would be to make the original condition have no actions then use another condition under that with an ip6 specific regex and use action and anti-action to differentiate On Fri, Feb 20, 2009 at 12:22 AM, Michael S Collins wrote: > > On Feb 19, 2009, at 7:53 PM, Jason White wrote: > > > I have it working now. The relevant changes were as follows. > > > > > > > > > > > > > > > > Jason, > > I like this approach. Good use of the many Dialplan tools. > > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090220/415adecb/attachment-0002.html From brian at freeswitch.org Fri Feb 20 06:22:54 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 20 Feb 2009 08:22:54 -0600 Subject: [Freeswitch-users] Suggestion for xml_curl performance In-Reply-To: <22118122.post@talk.nabble.com> References: <22118122.post@talk.nabble.com> Message-ID: it all depends on what you're doing.. can you elaborate? /b On Feb 20, 2009, at 4:18 AM, shehzad p wrote: > Recently I faced some performance bottleneck by using Javascript. From ajlong at worldlink.net Fri Feb 20 06:31:52 2009 From: ajlong at worldlink.net (Adam Long) Date: Fri, 20 Feb 2009 09:31:52 -0500 Subject: [Freeswitch-users] SIP dump to DB In-Reply-To: <191c3a030902200554s4abda94g76ba4e2975fffa98@mail.gmail.com> References: <499E92E4.5010503@post.cz> <191c3a030902200554s4abda94g76ba4e2975fffa98@mail.gmail.com> Message-ID: <03a601c99367$f92f48f0$eb8ddad0$@net> MySQL MEMORY/HEAP table might be ideal for this. This data is prob not critical and is probably being used for diagnosing peer connectivity issues anyway. If it is critical well then... there are always trade offs right :) I think in general what he is speaking of is just some sort of temporary SIP trace setup that can log that can be controlled or filtered. So its not just all or nothing. I wonder is it possible to enable the current sip trace functionality via a variable. For example something like this. This of course only helps for bridged B2BUA calls. But if inbound sip tracing is required a user param like sip_trace could address that, yes? That could be a good starting point, then perhaps I could help roll a mod_xml_siptrace module based on the mod_xml_cdr design/concept that could somehow link into the existing sip trace logging. Just some thoughts, no idea how much of this exists today. Regards, -Adam From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Friday, February 20, 2009 8:55 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] SIP dump to DB what exact info do you need? That's likely to be a challenge with any database to store at that speed. On Fri, Feb 20, 2009 at 5:24 AM, kokoska rokoska wrote: Hi all, I'm facing the problem I need all SIP messages "going thru" FreeSWITCH (I know FS i B2BUA - so, better to say just "all SIP messages") logged somewhere and this log have to be "searchable" (by call-id etc) and I should be able to simply delete "old" messages... And more over - it should be done on not trivial SIP messages amount - say hundreds messages per second. My questions are: 1. Do you have any suggestion how to do it with FreeSWITCH? 2. Or it is not possible now, and "bounty" is necessary? :-) 3. How hard it will be to implement? ---------- FYI: I think something like SER/Kamailio/OpenSIPS siptrace is what I'am (probably) looking for: http://www.kamailio.org/docs/modules/1.4.x/siptrace.html It could go (based on my tests) up to 7-10.000 messages per second till MySQL "dies"... ---------- Thanks for your time :-) Best regards, kokoska.rokoska _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 From kokoska.rokoska at post.cz Fri Feb 20 06:39:04 2009 From: kokoska.rokoska at post.cz (kokoska rokoska) Date: Fri, 20 Feb 2009 15:39:04 +0100 Subject: [Freeswitch-users] SIP dump to DB In-Reply-To: <191c3a030902200554s4abda94g76ba4e2975fffa98@mail.gmail.com> References: <499E92E4.5010503@post.cz> <191c3a030902200554s4abda94g76ba4e2975fffa98@mail.gmail.com> Message-ID: <499EC088.8080900@post.cz> Anthony Minessale napsal(a): > what exact info do you need? That's likely to be a challenge with any > database to store at that speed. > Thank you very much, Anthony, for your reply! I should say: Personally I don't need it (I see preformance penalty), but few people around me need to store somewhere ALL sip messages going through the server. And ALL means "really all" (well, it will be very helpful if I can skip OPTIONS and other nat-keep-alive related messages). So I need something like "sipgrep dump" but I should be able to simply corelate messages to user (in case of MESSAGE, REGISTER etc.) and to user+call (in case of INVITE, BYE, CANCEL etc.). ------------- BTW: I'm sure it will be challenge for DB - and thus I made some tests with mentioned siptrace. One SIP call is about 18-21 SIP messages, so for 100 cps I should make a little bit over 2.000 INSERTs per second. 900 REGISTERs per second will generate about 5.500 INSERTs per second => I need to fire cca 8.000 INSERTs per second... ------------- But, may be, better solution exists - without DB. Any hint is very appreciated :-) Best regards, kokoska.rokoska > On Fri, Feb 20, 2009 at 5:24 AM, kokoska rokoska > > wrote: > > > Hi all, > > I'm facing the problem I need all SIP messages "going thru" FreeSWITCH > (I know FS i B2BUA - so, better to say just "all SIP messages") logged > somewhere and this log have to be "searchable" (by call-id etc) and I > should be able to simply delete "old" messages... > And more over - it should be done on not trivial SIP messages amount - > say hundreds messages per second. > > My questions are: > 1. Do you have any suggestion how to do it with FreeSWITCH? > 2. Or it is not possible now, and "bounty" is necessary? :-) > 3. How hard it will be to implement? > > ---------- > FYI: > I think something like SER/Kamailio/OpenSIPS siptrace is what I'am > (probably) looking for: > http://www.kamailio.org/docs/modules/1.4.x/siptrace.html > It could go (based on my tests) up to 7-10.000 messages per second till > MySQL "dies"... > ---------- > > Thanks for your time :-) > > Best regards, > > kokoska.rokoska > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From kokoska.rokoska at post.cz Fri Feb 20 06:51:06 2009 From: kokoska.rokoska at post.cz (kokoska rokoska) Date: Fri, 20 Feb 2009 15:51:06 +0100 Subject: [Freeswitch-users] SIP dump to DB In-Reply-To: <03a601c99367$f92f48f0$eb8ddad0$@net> References: <499E92E4.5010503@post.cz> <191c3a030902200554s4abda94g76ba4e2975fffa98@mail.gmail.com> <03a601c99367$f92f48f0$eb8ddad0$@net> Message-ID: <499EC35A.1060808@post.cz> Adam Long napsal(a): > MySQL MEMORY/HEAP table might be ideal for this. This data is prob not > critical and is probably > being used for diagnosing peer connectivity issues anyway. > If it is critical well then... there are always trade offs right :) > Thank you very much, Adam, for interest! You are right - it is for diagnosis. And while the data is not critical, I can't leave them in MySQL MEMORY table. Based on my counts I have to store and maintain about 70-90 GiB of SIP messages and I simply don't have enough RAM :-) > I think in general what he is speaking of is just some sort of temporary SIP > trace setup that can log > that can be controlled or filtered. So its not just all or nothing. > Exactly! :-) > I wonder is it possible to enable the current sip trace functionality via a > variable. > For example something like this. > > data="{sip_trace=on}sofia/public/XXXXXXXXXX at 10.10.10.1" /> > This of course only helps for bridged B2BUA calls. > It will be very helpful... > But if inbound sip tracing is required a user param like sip_trace could > address that, yes? > Yes, I need all SIP packets except nat-keep-alives... BTW: How to recognize them? I know how could I do it on proxy (SER like), but what about on FreeSWITCH? Anyway - for me it shouldn't be an issue, because I filter then on loadbalancer. > That could be a good starting point, then perhaps I could help roll a > mod_xml_siptrace module > based on the mod_xml_cdr design/concept that could somehow link into the > existing sip trace logging. > It will be very powerfull, but I'm affraid it can't scale to thousands request per second. > Just some thoughts, no idea how much of this exists today. > Thanks once more, Adam! Best regards, kokoska.rokoska From sicfslist at gmail.com Fri Feb 20 06:58:36 2009 From: sicfslist at gmail.com (Shelby Ramsey) Date: Fri, 20 Feb 2009 08:58:36 -0600 Subject: [Freeswitch-users] SIP dump to DB In-Reply-To: <499EC35A.1060808@post.cz> References: <499E92E4.5010503@post.cz> <191c3a030902200554s4abda94g76ba4e2975fffa98@mail.gmail.com> <03a601c99367$f92f48f0$eb8ddad0$@net> <499EC35A.1060808@post.cz> Message-ID: <35b355e90902200658k48ba850ele3ede371ebe96631@mail.gmail.com> Why not just use NGREP and then dump the packets at a more reasonable pace? You aren't going to be able to analysis in real time anyway. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090220/75b5878e/attachment-0002.html From kokoska.rokoska at post.cz Fri Feb 20 07:27:46 2009 From: kokoska.rokoska at post.cz (kokoska rokoska) Date: Fri, 20 Feb 2009 16:27:46 +0100 Subject: [Freeswitch-users] SIP dump to DB In-Reply-To: <35b355e90902200658k48ba850ele3ede371ebe96631@mail.gmail.com> References: <499E92E4.5010503@post.cz> <191c3a030902200554s4abda94g76ba4e2975fffa98@mail.gmail.com> <03a601c99367$f92f48f0$eb8ddad0$@net> <499EC35A.1060808@post.cz> <35b355e90902200658k48ba850ele3ede371ebe96631@mail.gmail.com> Message-ID: <499ECBF2.4030606@post.cz> Shelby Ramsey napsal(a): > Why not just use NGREP and then dump the packets at a more reasonable > pace? Thank you very much, Shelby, for your interest! I can't use ngrep because the dump is not searchable IMO :-) See below, please. You aren't going to be able to analysis in real time anyway. > Some basic diagnosis should by doable realtime. For example: 1. find whole SIP trace for selected call (both a-leg, b-leg) 2. find all succeful user REGISTERs for given period ... And I have no idea how to do it on 50-100 GiB plain text file. Best regards, kokoska.rokoska From anthony.minessale at gmail.com Fri Feb 20 07:33:52 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 20 Feb 2009 09:33:52 -0600 Subject: [Freeswitch-users] SIP dump to DB In-Reply-To: <499ECBF2.4030606@post.cz> References: <499E92E4.5010503@post.cz> <191c3a030902200554s4abda94g76ba4e2975fffa98@mail.gmail.com> <03a601c99367$f92f48f0$eb8ddad0$@net> <499EC35A.1060808@post.cz> <35b355e90902200658k48ba850ele3ede371ebe96631@mail.gmail.com> <499ECBF2.4030606@post.cz> Message-ID: <191c3a030902200733y795626edq8eef6f06bcd70571@mail.gmail.com> you could try sippcapdump but i hear it needs work but it snoops the wire and tries to make individual files out of each call. On Fri, Feb 20, 2009 at 9:27 AM, kokoska rokoska wrote: > > > > Shelby Ramsey napsal(a): > > Why not just use NGREP and then dump the packets at a more reasonable > > pace? > > Thank you very much, Shelby, for your interest! > > I can't use ngrep because the dump is not searchable IMO :-) > See below, please. > > You aren't going to be able to analysis in real time anyway. > > > > Some basic diagnosis should by doable realtime. For example: > 1. find whole SIP trace for selected call (both a-leg, b-leg) > 2. find all succeful user REGISTERs for given period > ... > > And I have no idea how to do it on 50-100 GiB plain text file. > > Best regards, > > kokoska.rokoska > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090220/f8086de7/attachment-0002.html From sicfslist at gmail.com Fri Feb 20 07:38:17 2009 From: sicfslist at gmail.com (Shelby Ramsey) Date: Fri, 20 Feb 2009 09:38:17 -0600 Subject: [Freeswitch-users] SIP dump to DB In-Reply-To: <499ECBF2.4030606@post.cz> References: <499E92E4.5010503@post.cz> <191c3a030902200554s4abda94g76ba4e2975fffa98@mail.gmail.com> <03a601c99367$f92f48f0$eb8ddad0$@net> <499EC35A.1060808@post.cz> <35b355e90902200658k48ba850ele3ede371ebe96631@mail.gmail.com> <499ECBF2.4030606@post.cz> Message-ID: <35b355e90902200738n3f21dfdn50ccc038326ae5f1@mail.gmail.com> Sorry .. I didn't give enough detail. My point was to dump it via NGREP ... parse it using something else to get it into a database where it would be usable. Then you can match calls from the CDR (using the UUID) to the database. The benefit is that you don't have to put the burden on your FS boxes to do it ... Just monitor from another device and then dump it into the database. Of course you better have a beast of database if you want to do 10,000 writes per second :) or be running something like NDB that scales well. There are other tools like scapy as well that can be quite useful in fact. Shelby -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090220/628b6fca/attachment-0002.html From jaugenstine at gmail.com Fri Feb 20 07:50:22 2009 From: jaugenstine at gmail.com (jonathan augenstine) Date: Fri, 20 Feb 2009 07:50:22 -0800 Subject: [Freeswitch-users] SIP dump to DB In-Reply-To: <191c3a030902200733y795626edq8eef6f06bcd70571@mail.gmail.com> References: <499E92E4.5010503@post.cz> <191c3a030902200554s4abda94g76ba4e2975fffa98@mail.gmail.com> <03a601c99367$f92f48f0$eb8ddad0$@net> <499EC35A.1060808@post.cz> <35b355e90902200658k48ba850ele3ede371ebe96631@mail.gmail.com> <499ECBF2.4030606@post.cz> <191c3a030902200733y795626edq8eef6f06bcd70571@mail.gmail.com> Message-ID: <207e7a5e0902200750t2ac9f21avfb5ae88791e4efa8@mail.gmail.com> You can tcpdump and then use wireshark to graph the calls. When the dump is displayed in wireshark, select 'Statistics' -> VoIP Calls. You will see a display of all VoIP calls. Select the one you want graphed, or select them all and you will see REINVITE and REFER interaction as well as RTP streams. Jonathan On Fri, Feb 20, 2009 at 7:33 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > you could try sippcapdump but i hear it needs work but it snoops the wire > and tries to make individual files out of each call. > > > > On Fri, Feb 20, 2009 at 9:27 AM, kokoska rokoska wrote: > >> >> >> >> Shelby Ramsey napsal(a): >> > Why not just use NGREP and then dump the packets at a more reasonable >> > pace? >> >> Thank you very much, Shelby, for your interest! >> >> I can't use ngrep because the dump is not searchable IMO :-) >> See below, please. >> >> You aren't going to be able to analysis in real time anyway. >> > >> >> Some basic diagnosis should by doable realtime. For example: >> 1. find whole SIP trace for selected call (both a-leg, b-leg) >> 2. find all succeful user REGISTERs for given period >> ... >> >> And I have no idea how to do it on 50-100 GiB plain text file. >> >> Best regards, >> >> kokoska.rokoska >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090220/d0089afd/attachment-0002.html From kokoska.rokoska at post.cz Fri Feb 20 07:57:15 2009 From: kokoska.rokoska at post.cz (kokoska rokoska) Date: Fri, 20 Feb 2009 16:57:15 +0100 Subject: [Freeswitch-users] SIP dump to DB In-Reply-To: <191c3a030902200733y795626edq8eef6f06bcd70571@mail.gmail.com> References: <499E92E4.5010503@post.cz> <191c3a030902200554s4abda94g76ba4e2975fffa98@mail.gmail.com> <03a601c99367$f92f48f0$eb8ddad0$@net> <499EC35A.1060808@post.cz> <35b355e90902200658k48ba850ele3ede371ebe96631@mail.gmail.com> <499ECBF2.4030606@post.cz> <191c3a030902200733y795626edq8eef6f06bcd70571@mail.gmail.com> Message-ID: <499ED2DB.90807@post.cz> Anthony Minessale napsal(a): > you could try sippcapdump but i hear it needs work but it snoops the > wire and tries to make individual files out of each call. > > Thank you very much, Anthony, for the suggestion! I will look at it. Best regards, kokoska.rokoska From kokoska.rokoska at post.cz Fri Feb 20 08:09:56 2009 From: kokoska.rokoska at post.cz (kokoska rokoska) Date: Fri, 20 Feb 2009 17:09:56 +0100 Subject: [Freeswitch-users] SIP dump to DB In-Reply-To: <35b355e90902200738n3f21dfdn50ccc038326ae5f1@mail.gmail.com> References: <499E92E4.5010503@post.cz> <191c3a030902200554s4abda94g76ba4e2975fffa98@mail.gmail.com> <03a601c99367$f92f48f0$eb8ddad0$@net> <499EC35A.1060808@post.cz> <35b355e90902200658k48ba850ele3ede371ebe96631@mail.gmail.com> <499ECBF2.4030606@post.cz> <35b355e90902200738n3f21dfdn50ccc038326ae5f1@mail.gmail.com> Message-ID: <499ED5D4.5010603@post.cz> Shelby Ramsey napsal(a): > Sorry .. I didn't give enough detail. My point was to dump it via NGREP > ... parse it using something else to get it into a database where it > would be usable. This is good point! Thank you very much, Shelby! > Then you can match calls from the CDR (using the UUID) > to the database. This is exactly what I try to accomplish :-) > The benefit is that you don't have to put the burden > on your FS boxes to do it ... Just monitor from another device and then > dump it into the database. I think of 2 possibilities: 1. On router replicate whole traffic to dedicated machine and process dump there (but it probably kills that machine, because it gets all RTP traffic) 2. Modify FreeSWITCH/Sofia (I have no idea how hard it will be, or if it is even possible) to duplicate all SIP messages to given URI - main benefit of this scenario is that I have only SIP messages on logging machine and that I can use SERlike proxy to parse messages and store them to DB. > Of course you better have a beast of > database if you want to do 10,000 writes per second :) or be running > something like NDB that scales well. > NDB is overkill for logging :-) What I think of is some kind of "caching" on FreeSWITCH side. I.e. store mesasges to DB in bigger chukns (whole call etc.) It will significatnly reduce DB utilization... > There are other tools like scapy as well that can be quite useful in fact. > I will look at it, thank you Shelby! Best regards, kokoska.roksoka From kokoska.rokoska at post.cz Fri Feb 20 08:15:51 2009 From: kokoska.rokoska at post.cz (kokoska rokoska) Date: Fri, 20 Feb 2009 17:15:51 +0100 Subject: [Freeswitch-users] SIP dump to DB In-Reply-To: <207e7a5e0902200750t2ac9f21avfb5ae88791e4efa8@mail.gmail.com> References: <499E92E4.5010503@post.cz> <191c3a030902200554s4abda94g76ba4e2975fffa98@mail.gmail.com> <03a601c99367$f92f48f0$eb8ddad0$@net> <499EC35A.1060808@post.cz> <35b355e90902200658k48ba850ele3ede371ebe96631@mail.gmail.com> <499ECBF2.4030606@post.cz> <191c3a030902200733y795626edq8eef6f06bcd70571@mail.gmail.com> <207e7a5e0902200750t2ac9f21avfb5ae88791e4efa8@mail.gmail.com> Message-ID: <499ED737.3020401@post.cz> jonathan augenstine napsal(a): > You can tcpdump and then use wireshark to graph the calls. When the > dump is displayed in wireshark, select 'Statistics' -> VoIP Calls. You > will see a display of all VoIP calls. Select the one you want graphed, > or select them all and you will see REINVITE and REFER interaction as > well as RTP streams. > Thank you very much, jonathan, for your interest! I use ngrep+wireshark many times a day, but I'm affraid it is not suitable for that amount of data. Even with few hundreds MiBs of pcap file wireshark becoms very slow and I can't imagine how to load 50-100 GiB file with milions of calls and try to search for one of them :-) And, even worse, I should "rotate" the file and, don't end with call divided to multiple files... Best regards, kokoska.rokoska From leon at scarlet-internet.nl Fri Feb 20 08:19:25 2009 From: leon at scarlet-internet.nl (Leon de Rooij) Date: Fri, 20 Feb 2009 17:19:25 +0100 Subject: [Freeswitch-users] mod_erlang_event compile problem In-Reply-To: <20090109230650.GF5210@hijacked.us> References: <20090109230650.GF5210@hijacked.us> Message-ID: <4A496EA6-D036-4BB5-9347-5536AD5FD0FC@scarlet-internet.nl> Hi, I wanted to try out the mod_erlang_event module. I have Erlang R12B5 compiled and it's in the same location as the Makefile specifies (/usr/ local/lib/erlang/...), but running make in the src/mod/event_handlers/ mod_erlang_event goes wrong: Compiling handle_msg.c... cc1: warnings being treated as errors handle_msg.c: In function 'handle_msg_sendmsg': handle_msg.c:429: warning: the address of 'uuid' will always evaluate as 'true' handle_msg.c: In function 'handle_msg_handlecall': handle_msg.c:541: warning: the address of 'uuid_str' will always evaluate as 'true' make[1]: *** [handle_msg.o] Error 1 make: *** [all] Error 1 At line 429 in handle_msg.c it says: if (!switch_strlen_zero(uuid) && (session = switch_core_session_locate(uuid))) { (at line 541 is the same problem) Is this a bug ? I tried removing the first part "! switch_strlen_zero(uuid) &&" after which it compiles fine, but since I don't fully understand what's going on, I'm sure this is not the solution.. Also, after this, FS goes haywire after loading the module and spews out these messages continuously: 2009-02-20 14:15:48 [ERR] mod_erlang_event.c:1417 mod_erlang_event_runtime() Failed to start empd manually 2009-02-20 14:15:48 [DEBUG] mod_erlang_event.c:1401 mod_erlang_event_runtime() Socket up listening on 127.0.0.1:8031 2009-02-20 14:15:48 [WARNING] mod_erlang_event.c:1415 mod_erlang_event_runtime() Failed to publish port to empd, trying to start empd manually 2009-02-20 14:15:48 [ERR] mod_erlang_event.c:1417 mod_erlang_event_runtime() Failed to start empd manually 2009-02-20 14:15:48 [DEBUG] mod_erlang_event.c:1401 mod_erlang_event_runtime() Socket up listening on 127.0.0.1:8031 2009-02-20 14:15:48 [WARNING] mod_erlang_event.c:1415 mod_erlang_event_runtime() Failed to publish port to empd, trying to start empd manually etc.. Can someone help me and point out what's wrong ? thanks & kind regards, Leon de Rooij From msc at freeswitch.org Fri Feb 20 08:36:29 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 20 Feb 2009 08:36:29 -0800 Subject: [Freeswitch-users] js session.streamFile() interrupt slowness In-Reply-To: <11990ade0902192127l414ecfeaia04fb80f626c35d5@mail.gmail.com> References: <11990ade0902192127l414ecfeaia04fb80f626c35d5@mail.gmail.com> Message-ID: <87f2f3b90902200836o638fa345qec10888f39e46414@mail.gmail.com> On Thu, Feb 19, 2009 at 9:27 PM, Stephen Crosby wrote: > I have a few scripts that use the javascript > session.streamFile('somefile.wav', onDtmf); where onDtmf is a function > that returns false to interrupt the streaming file. There is a short > delay between the time when I press a key and the time the file stops > playing. Is there anything I can adjust that would affect that? It's > only maybe 2-3 seconds, but it "feels" too long to me. > > --Stephen Could you pastebin your entire script plus the relevant dialplan entry? Also, could you tell us which operating system and FS revision? -MC From andrew at hijacked.us Fri Feb 20 11:08:11 2009 From: andrew at hijacked.us (Andrew Thompson) Date: Fri, 20 Feb 2009 14:08:11 -0500 Subject: [Freeswitch-users] mod_erlang_event compile problem In-Reply-To: <4A496EA6-D036-4BB5-9347-5536AD5FD0FC@scarlet-internet.nl> References: <20090109230650.GF5210@hijacked.us> <4A496EA6-D036-4BB5-9347-5536AD5FD0FC@scarlet-internet.nl> Message-ID: <20090220190811.GC29511@hijacked.us> On Fri, Feb 20, 2009 at 05:19:25PM +0100, Leon de Rooij wrote: > Hi, > > I wanted to try out the mod_erlang_event module. I have Erlang R12B5 > compiled and it's in the same location as the Makefile specifies (/usr/ > local/lib/erlang/...), but running make in the src/mod/event_handlers/ > mod_erlang_event goes wrong: > Yeah, this was a gcc4 thing, I've done most of my testing on gcc3 so it didn't show up for me. Thanks to MikeJ for the fix suggestion. > Also, after this, FS goes haywire after loading the module > and spews out these messages continuously: > You don't have the erlang port mapper daemon running (epmd). mod_erlang_event needs it to be running in order to be able to register itself as an erlang node. On your system; epmd isn't in $PATH so my system() call that tries to start it fails. I've made the module init system fail properly instead of looping indefinitely as well as print a slightly more helpful error message now. Let me know if you have any better luck :) The fix is in-tree as of r12192. Thanks again for the bug report. Andrew From andrew at hijacked.us Fri Feb 20 11:19:11 2009 From: andrew at hijacked.us (Andrew Thompson) Date: Fri, 20 Feb 2009 14:19:11 -0500 Subject: [Freeswitch-users] [ANN] Spice Telephony - an open source FreeSWITCH/Erlang callcenter platform In-Reply-To: <22015518.post@talk.nabble.com> References: <22015518.post@talk.nabble.com> Message-ID: <20090220191911.GD29511@hijacked.us> On Sat, Feb 14, 2009 at 03:04:01PM -0800, JCATS wrote: > > Have you planned any predictive dialer features ( like VICIDIAL )? > As Ken Rice mentioned, this isn't really the focus of the project - it's more for inbound and directed outbound (calling campaigns to specific people/businesses - not everyone in the phonebook). Primary focus is inbound (multi brand, skill based routing, dynamic wrapup times, etc). Expect a new release sometime soonish that actually does something useful (accepts and routes inbound calls from FreeSWITCH to an agent). Also; public source control. There's just some additional corporate nonsense that I have to sort out (again) before that can go live. Andrew From freeswitch-users at lists.rupa.com Fri Feb 20 14:08:40 2009 From: freeswitch-users at lists.rupa.com (Rupa Schomaker (lists)) Date: Fri, 20 Feb 2009 16:08:40 -0600 Subject: [Freeswitch-users] Passing Caller_ID to MOD_LCR In-Reply-To: <499AC59A.4050209@lists.rupa.com> References: <499A4497.9080001@lists.rupa.com> <499AC59A.4050209@lists.rupa.com> Message-ID: <499F29E8.4020100@lists.rupa.com> > "default" is a reserved profile name -- I should probably prevent that > from loading. correction, default is not reserved... > Regarding passing the callerid to the custom sql, let me see what I can > come up with... You can now specify channel variables in your custom sql. So, you should be able to pass CID or a subset of CID to your custom sql. I've updated the wiki with info on this. beware: channel vars only work when called in the context of a session. Using a profile that uses a custom sql with channel variables from the commandline will result in an error. -Rupa From leon at scarlet-internet.nl Fri Feb 20 15:28:08 2009 From: leon at scarlet-internet.nl (Leon de Rooij) Date: Sat, 21 Feb 2009 00:28:08 +0100 Subject: [Freeswitch-users] mod_erlang_event compile problem In-Reply-To: <20090220190811.GC29511@hijacked.us> References: <20090109230650.GF5210@hijacked.us> <4A496EA6-D036-4BB5-9347-5536AD5FD0FC@scarlet-internet.nl> <20090220190811.GC29511@hijacked.us> Message-ID: <8D43B57D-27ED-4517-BAD1-00EE526006A3@scarlet-internet.nl> Hi Andrew, Thanks! it compiles fine now.. Also thanks for the tip about empd, got it running without errors now :-) regards, Leon On Feb 20, 2009, at 8:08 PM, Andrew Thompson wrote: > On Fri, Feb 20, 2009 at 05:19:25PM +0100, Leon de Rooij wrote: >> Hi, >> >> I wanted to try out the mod_erlang_event module. I have Erlang R12B5 >> compiled and it's in the same location as the Makefile specifies (/ >> usr/ >> local/lib/erlang/...), but running make in the src/mod/ >> event_handlers/ >> mod_erlang_event goes wrong: >> > > Yeah, this was a gcc4 thing, I've done most of my testing on gcc3 so > it > didn't show up for me. Thanks to MikeJ for the fix suggestion. > >> Also, after this, FS goes haywire after loading the module >> and spews out these messages continuously: >> > > You don't have the erlang port mapper daemon running (epmd). > mod_erlang_event needs it to be running in order to be able to > register > itself as an erlang node. On your system; epmd isn't in $PATH so my > system() call that tries to start it fails. I've made the module init > system fail properly instead of looping indefinitely as well as > print a > slightly more helpful error message now. Let me know if you have any > better luck :) The fix is in-tree as of r12192. > > Thanks again for the bug report. > > Andrew > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From pmhshz at gmail.com Fri Feb 20 22:16:36 2009 From: pmhshz at gmail.com (shehzad p) Date: Fri, 20 Feb 2009 22:16:36 -0800 (PST) Subject: [Freeswitch-users] Suggestion for xml_curl performance In-Reply-To: References: <22118122.post@talk.nabble.com> Message-ID: <22133185.post@talk.nabble.com> Hi Brian, My setup is to use FS as basic calls routing. 1. Calls are coming to FS from more than one customer Gateways, and I need to authenticate them and check for enough balance based on database, [Caller Gateways] ===> [FreeSWITCH] ===> [Provider Gateways] 2. After knowing that Caller Gateways is valid, then based on dialed number it search in database for Provider Gateway and bridge the call there. 3. After call finish CDR is inserted back into database. My old setup was using Javascript which works fine in traffic of 10 to 20 calls, but then increase of traffic causes many problems. Now I eliminate use of any of the script (javascript or any other) for call routing, and route calls directly from dialplan, So I have setup test system using xml-curl to generate dynamic dialplan, I used below xml_curl PHP example as reference: http://wiki.freeswitch.org/wiki/Mod_xml_curl_PHP_example For CDR processing I used xml_cdr, with help of the example in FS source :scripts/contrib/trixter/xml-cdr. Waiting for any better suggestions, any comments... thanks msp. Brian West-3 wrote: > > it all depends on what you're doing.. can you elaborate? > > /b > > On Feb 20, 2009, at 4:18 AM, shehzad p wrote: > >> Recently I faced some performance bottleneck by using Javascript. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/Suggestion-for-xml_curl-performance-tp22118122p22133185.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From jmesquita at gmail.com Sat Feb 21 05:56:41 2009 From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Sat, 21 Feb 2009 11:56:41 -0200 Subject: [Freeswitch-users] Deployment information and use cases In-Reply-To: <499D796E.45B7.0079.0@averyschools.net> References: <1234916455.16581.49.camel@raul-laptop> <59ad9ca10902181009p7cbc0fd1s9860aacde7f3a30c@mail.gmail.com> <247f8100902181019m61ce803m5ed504d9b198dfb6@mail.gmail.com> <1235016793.22050.0.camel@raul-laptop> <499D796E.45B7.0079.0@averyschools.net> Message-ID: <7405656F-1AAA-4FF7-9DC3-4CE694D7B0AB@gmail.com> Ben, thank you for your story. I would very much like to add this to the wiki if you don't mind and everyone else agrees. What do you think guys? Use cases are _ALWAYS_ a good thing for new users. Mesquita On Feb 19, 2009, at 6:23 PM, Ben Holtsclaw wrote: > Raul, > > I am in the process of rolling out a FreeSWITCH IP PBX solution > similar to what you describe. When I was trying to procure funds for > a FreeSWITCH solution, I looked for the same information you're > after, but came up with little. I'll briefly describe what we're > trying to accomplish, and the tools I'm using to do it. This is > probably more information than what you are looking for, but maybe > it will also benefit someone else. > > We had several schools with aging or dying PBX's or KSU's. Each site > had something different system, and was supported by a different > VAR. Of course, the VAR's charged some outlandish fee to make onsite > repair visits. Some number of Centrex lines supplied each school's > dial tone. All in all, we had a very outdated and financially > draining mess. Our district's long term goal had been to move to a > more unified phone system. That made sense for many reasons, the > chief of which was cost. We already had a strong fiber WAN in place. > Why not use that for trunking and eliminate the monthly cost of the > Centrex lines? That's the path we started down. > > Being a public entity, we had to be sure to explore all possible > avenues. We looked at everything from traditional PBX's with IP add- > on modules for trunking to a full blown Cisco CallManager solution. > With third party proprietary systems, we were just never able to > find the sweet spot between required feature set and cost. Would > Cisco have been a workable solution? Absolutely. Could our small, > rural, K12 public school district afford that? Not in a million > years. I looked at several software packages -- some open source, > some not -- but always came back to FreeSWITCH. The scalability and > active development community were major factors for us. > > Server Hardware. Each of our five sites has a dedicated FreeSWITCH > server. For hardware, we went with Dell PowerEdge 1950's with dual > quad core Xeon 2.33 GHz processors, and 4GB of RAM. I have mirrored > disks set up with enough space to accommodate users' voicemail. Each > server will average only about 60 voicemail boxes, and we're storing > sound as MP3. Disk space shouldn't be an issue. We have always been > a Novell shop, so SLES is naturally our Linux distribution of > choice. We chose to go with server hardware at each site so that in > the event of a WAN outage, we would still at least have intra- > building and emergency communication (see below). > > Telephony Hardware. Each of our servers includes Sangoma hardware. > We actually looked at doing IP trunking to a carrier from our > network core, but decided to use telco provided PRI's instead. > Presently, we have two PRI's that connect to a FreeSWITCH server at > the center of our network via a Sangoma A102 dual port telephony > card. All calls to and from the PSTN traverse this primary server. > Servers at each remote site include one of Sangoma's A200 analog > cards. Emergency calls to 911 route out over this analog card > through one of at least two POTS lines that remain connected at each > site. Not only does this provide some redundancy in the event of a > WAN outage, but it ensures proper caller location is delivered to > the 911 dispatcher. Granted, there are some other solutions for the > latter, but this seemed to be the most cost effective solution for us. > > Telephone Desksets. We chose to go with Aastra for the telephones. > The standard phone that we will place in each classroom and office > is the 9143i. This is an attractive phone with an adequate feature > set at a price we can afford. The person that is primarily > responsible for answering the phone at each site will have an Aastra > 57i and some number of 560M expansion modules. We have purchased > roughly 300 Aastra desksets. > > Logical Layout. As new sites come online, their primary phone number > is being moved from the Centrex to our PRI group. All inbound calls > hit our primary server, and then FreeSWITCH bridges to the > appropriate secondary server based on the DID it received. On the > reverse, each servers dial plan is set up to route outbound calls > (save 911) to the primary server where FreeSWITCH bridges with > Openzap. Site to site calls, accomplished via four digit dialing, do > not hit the primary server. Outbound calls to the PSTN deliver the > site's DID as the calling number. In other words, if a user from > site two calls my cell phone, I see site two's published telephone > number on my caller ID. Our dial plans are set up so that > receptionists at each site still answer all outside calls. If not > answered, the call fails over to an IVR. Should we ever decide to do > so, we are now perfectly positioned to have all inbound calls to the > district answered by one operator or IVR. "Welcome, and thank you > for calling Avery County Schools." > > Stumbling Blocks. Problems we've faced so far have primarily > surrounded Openzap and the Sangoma Wanpipe driver. FreeSWITCH > developers won't mind telling you that this is an area that is > currently not well "funded" and not 100% complete. There is some > known issue where voice channels on the PRI get stuck in the wrong > state and become unusable. We have experienced this a couple of > times and have not been able to make or receive calls. Bouncing the > Wanpipe driver has fixed this each time. We have also had trouble > with DTMF detection across the PRI. If a user hits the IVR, it is > oftentimes difficult to get it to properly recognize the digits that > are being keyed in by the caller. This can be very, very frustrating > to a caller that doesn't want to deal with an IVR anyway. The > developers have suggested to me that this is a problem with the > Sangoma's echo cancellation goofing up Openzap's ability to > interpret the DTMF. The Sangoma hardware does have its own DTMF > decoder and API, but the Openzap code currently does not make use of > it. I have created a patch that makes use of the hardware decoder. > We have been running it in production for a couple of weeks, and > that does seem to have helped the problem. The problem hasn't gone > away altogether. Those have been our two biggest issues, but we > haven't let them hold us up. > > Conclusion. Of the five sites that will be on this system, one is > fully functional with calls inbound and outbound from the PSTN. A > second site is up and running with full outbound PSTN access. Their > inbound DID is scheduled to move over to the PRI in one week. The > server has been worked up for a third site, and the phones are > starting to roll out. Sites four and five should come online by the > end of April. Currently, I don't have numbers compiled for things > like concurrent calls. At this point in my project, it is just not > important. I really don't think our implementation will ever push > FreeSWITCH's abilities in that regard. I base that statement > primarily on other users' benchmarks, and what I've heard some are > doing in carrier class environments. > > FreeSWITCH has made our project viable. An open source solution was > the only way we could meet all of the project goals and stay within > our budget. FreeSWITCH has proven to have all the features we > require in a district wide phone system. It has not locked us into > annual support contracts with third party vendors. I could go on > with the accolades. However, I'll end this terribly lengthy post by > saying that, overall, we have been very pleased with our choice to > go with FreeSWITCH. > > The information in this email will seem very elementary to most > people on this list, but having a message of this nature in hand > would have made me feel much more confident the first time I ever > went to my supervisor to mention something called FreeSWITCH. :-) > Thanks Tony, Brian, and Mike for a great product! > > > Ben Holtsclaw > Network Engineer > Avery County Schools > Ph: 828.733.3567 x2301 > > >>> On 2/18/2009 at 11:13 PM, Raul Fragoso wrote: > Thanks guys, this is very useful information. > > Anyone else willing to share your experience ? > > Regards, > > Raul > > On Wed, 2009-02-18 at 16:19 -0200, Pablo Hernan Saro wrote: > > Hi Raul, > > > > In my company (http://www.globant.com) we're using FreeSWITCH for > high > > quality conference services, integrated with OpenSIPS > > (http://www.opensips.org) and Asterisk. Its performance is pretty > > good. > > > > Pablo > > > > On Wed, Feb 18, 2009 at 4:09 PM, Henry Huang > wrote: > > > bandwidth.com has a service called phonebooth which is developed > upon > > > freeswitch. > > > > > > > > > On Tue, Feb 17, 2009 at 4:20 PM, Raul Fragoso > wrote: > > >> > > >> Hello FreeSWITCHERS, > > >> > > >> My company is currently creating a suite of applications which > uses > > >> FreeSWITCH as the back-end for an IP-PBX solution. We currently > have a > > >> prospect to have our first customer installation - a governmental > > >> department. That is a tender to have an IP-PBX installation to > connect > > >> their four office branches, each one with about 300 users - > which I am > > >> sure FreeSWITCH is able to handle. Since this is an official > tender, > > >> it's part of their protocol to ask about real sites using the > product. > > >> > > >> Having said that, would you mind sharing some information about > your > > >> experience with FreeSWITCH deployments ? > > >> > > >> No need to give many details, but a short summary with company > name (if > > >> possible), when it was deployed, server equipment, number of > users, > > >> number of concurrent calls, what kind of functions and services > are used > > >> and overall capacity of the system. > > >> > > >> I would really appreciate if you can share that information. > And if you > > >> guys agree (and explicitly manifest your agreement), I can > compile the > > >> information in the FreeSWITCH wiki under a "Use Cases" page so > it can > > >> serve as a common reference as well. > > >> > > >> Kind regards, > > >> > > >> Raul Fragoso > > >> > > >> > > >> _______________________________________________ > > >> Freeswitch-users mailing list > > >> Freeswitch-users at lists.freeswitch.org > > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > >> http://www.freeswitch.org > > > > > > > > > > > > -- > > > Henry Huang > > > UniC Solution - Communication Unified > > > VoIP & Open Source software Consultant > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090221/3cdd410f/attachment-0002.html From leon at scarlet-internet.nl Sat Feb 21 06:42:24 2009 From: leon at scarlet-internet.nl (Leon de Rooij) Date: Sat, 21 Feb 2009 15:42:24 +0100 Subject: [Freeswitch-users] mod_erlang_event compile problem In-Reply-To: <20090220190811.GC29511@hijacked.us> References: <20090109230650.GF5210@hijacked.us> <4A496EA6-D036-4BB5-9347-5536AD5FD0FC@scarlet-internet.nl> <20090220190811.GC29511@hijacked.us> Message-ID: <90DDD253-BA8D-4A79-9ACA-1560F2E18688@scarlet-internet.nl> Hi Andrew, Thanks for your help so far, I hope you can help me a bit further as I don't get any reply from the FS erlang node, or so it seems.. Here is what I've done: - The erlang_event.conf.xml is unchanged: - mod_erlang_event is not loaded in FS. - First I start "epmd -d -d" epmd: Sat Feb 21 13:12:56 2009: epmd running - daemon = 0 epmd: Sat Feb 21 13:12:56 2009: try to initiate listening port 4369 epmd: Sat Feb 21 13:12:56 2009: starting epmd: Sat Feb 21 13:12:56 2009: entering the main select() loop - After that I "load mod_erlang_event" in FS: 2009-02-21 13:13:36 [DEBUG] mod_erlang_event.c:1324 mod_erlang_event_load() sections 16 2009-02-21 13:13:36 [CONSOLE] switch_loadable_module.c:858 switch_loadable_module_load_file() Successfully Loaded [mod_erlang_event] 2009-02-21 13:13:36 [NOTICE] switch_loadable_module.c:240 switch_loadable_module_process() Adding Application 'erlang' 2009-02-21 13:13:36 [NOTICE] switch_loadable_module.c:260 switch_loadable_module_process() Adding API Function 'erlang' 2009-02-21 13:13:36 [DEBUG] mod_erlang_event.c:1401 mod_erlang_event_runtime() Socket up listening on 127.0.0.1:8031 2009-02-21 13:13:36 [DEBUG] mod_erlang_event.c:1426 mod_erlang_event_runtime() Connected and published erlang cnode at freeswitch at erlyfs - For which epmd gives the following output: epmd: Sat Feb 21 13:13:36 2009: opening connection on file descriptor 4 epmd: Sat Feb 21 13:13:36 2009: got 25 bytes ***** 00000000 00 17 78 1f 5f 68 00 00 05 00 01 00 0a 66 72 65 |..x._h.......fre| ***** 00000010 65 73 77 69 74 63 68 00 00 | eswitch..| epmd: Sat Feb 21 13:13:36 2009: ** got ALIVE2_REQ epmd: Sat Feb 21 13:13:36 2009: registering 'freeswitch:1', port 8031 epmd: Sat Feb 21 13:13:36 2009: type 104 proto 0 highvsn 5 lowvsn 1 epmd: Sat Feb 21 13:13:36 2009: got 4 bytes ***** 00000000 79 00 00 01 |y...| epmd: Sat Feb 21 13:13:36 2009: ** sent ALIVE2_RESP for "freeswitch" - Then I start an erl shell on that same machine with "erl -sname ldr - setcookie ClueCon". Output of epmd: epmd: Sat Feb 21 13:16:24 2009: opening connection on file descriptor 5 epmd: Sat Feb 21 13:16:24 2009: got 18 bytes ***** 00000000 00 10 78 8e 2c 4d 00 00 05 00 05 00 03 6c 64 72 |..x.,M.......ldr| ***** 00000010 00 00 |..| epmd: Sat Feb 21 13:16:24 2009: ** got ALIVE2_REQ epmd: Sat Feb 21 13:16:24 2009: registering 'ldr:1', port 36396 epmd: Sat Feb 21 13:16:24 2009: type 77 proto 0 highvsn 5 lowvsn 5 epmd: Sat Feb 21 13:16:24 2009: got 4 bytes ***** 00000000 79 00 00 01 |y...| epmd: Sat Feb 21 13:16:24 2009: ** sent ALIVE2_RESP for "ldr" As far as I understand the freeswitch at erlyfs node cannot be seen with nodes() ? So does that mean that I also cannot net_adm:ping() it ? Anyway, I tried sending some tuples as is shown on the wiki, but I get no reply: (ldr at erlyfs)1> {foo, freeswitch at erlyfs} ! {api, status, ""}, receive X -> X after 1000 -> timeout end. timeout (ldr at erlyfs)2> - Epmd gives some logs: epmd: Sat Feb 21 13:19:09 2009: opening connection on file descriptor 6 epmd: Sat Feb 21 13:19:09 2009: got 13 bytes ***** 00000000 00 0b 7a 66 72 65 65 73 77 69 74 63 68 |..zfreeswitch| epmd: Sat Feb 21 13:19:09 2009: ** got PORT2_REQ epmd: Sat Feb 21 13:19:09 2009: got 23 bytes ***** 00000000 77 00 1f 5f 68 00 00 05 00 01 00 0a 66 72 65 65 | w.._h.......free| ***** 00000010 73 77 69 74 63 68 00 | switch.| epmd: Sat Feb 21 13:19:09 2009: ** sent PORT2_RESP (ok) for "freeswitch" epmd: Sat Feb 21 13:19:09 2009: closing connection on file descriptor 6 - And in tcpdump on lo, I see that epmd is contacted after which some traffic was sent to FS: 13:19:09.535293 IP 172.31.0.13.34678 > 172.31.0.13.4369: S 2875169966:2875169966(0) win 32792 ... 13:19:09.536834 IP 172.31.0.13.4369 > 172.31.0.13.34678: . ack 15 win 512 13:19:09.536923 IP 172.31.0.13.47054 > 172.31.0.13.8031: S 2868322908:2868322908(0) win 32792 13:19:09.536935 IP 172.31.0.13.8031 > 172.31.0.13.47054: R 0:0(0) ack 2868322909 win 0 Shouldn't FS then send a message back to the process in my erl shell ? I tried logging all events in fs_cli, by entering "/event plain all", but I see no events at all coming from erlang, just some heartbeats.. Also, I recompiled the module with EI_DEBUG defined as suggested on the wiki. Still I don't see anything in the CLI when set to debug logging. Thanks again, Leon On Feb 20, 2009, at 8:08 PM, Andrew Thompson wrote: > On Fri, Feb 20, 2009 at 05:19:25PM +0100, Leon de Rooij wrote: >> Hi, >> >> I wanted to try out the mod_erlang_event module. I have Erlang R12B5 >> compiled and it's in the same location as the Makefile specifies (/ >> usr/ >> local/lib/erlang/...), but running make in the src/mod/ >> event_handlers/ >> mod_erlang_event goes wrong: >> > > Yeah, this was a gcc4 thing, I've done most of my testing on gcc3 so > it > didn't show up for me. Thanks to MikeJ for the fix suggestion. > >> Also, after this, FS goes haywire after loading the module >> and spews out these messages continuously: >> > > You don't have the erlang port mapper daemon running (epmd). > mod_erlang_event needs it to be running in order to be able to > register > itself as an erlang node. On your system; epmd isn't in $PATH so my > system() call that tries to start it fails. I've made the module init > system fail properly instead of looping indefinitely as well as > print a > slightly more helpful error message now. Let me know if you have any > better luck :) The fix is in-tree as of r12192. > > Thanks again for the bug report. > > Andrew > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From shannon at sacredhearts.us Sat Feb 21 07:11:00 2009 From: shannon at sacredhearts.us (Shannon) Date: Sat, 21 Feb 2009 09:11:00 -0600 Subject: [Freeswitch-users] Suggestion for xml_curl performance In-Reply-To: <22133185.post@talk.nabble.com> References: <22118122.post@talk.nabble.com> <22133185.post@talk.nabble.com> Message-ID: <7e2ac3270902210711u3ea54b79j5868aef6b59d4800@mail.gmail.com> I'd recommend having a look at fastcgi as well. On 2/21/09, shehzad p wrote: > > Hi Brian, > > My setup is to use FS as basic calls routing. > 1. Calls are coming to FS from more than one customer Gateways, and I need > to authenticate them and check for enough balance based on database, > [Caller Gateways] ===> [FreeSWITCH] ===> > [Provider Gateways] > 2. After knowing that Caller Gateways is valid, then based on dialed number > it search in database for Provider Gateway and bridge the call there. > 3. After call finish CDR is inserted back into database. > > My old setup was using Javascript which works fine in traffic of 10 to 20 > calls, but then increase of traffic causes many problems. > > Now I eliminate use of any of the script (javascript or any other) for call > routing, and route calls directly from dialplan, > So I have setup test system using xml-curl to generate dynamic dialplan, > I used below xml_curl PHP example as reference: > http://wiki.freeswitch.org/wiki/Mod_xml_curl_PHP_example > For CDR processing I used xml_cdr, with help of the example in FS source > :scripts/contrib/trixter/xml-cdr. > > > Waiting for any better suggestions, any comments... > > thanks > msp. > > Brian West-3 wrote: >> >> it all depends on what you're doing.. can you elaborate? >> >> /b >> >> On Feb 20, 2009, at 4:18 AM, shehzad p wrote: >> >>> Recently I faced some performance bottleneck by using Javascript. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -- > View this message in context: > http://www.nabble.com/Suggestion-for-xml_curl-performance-tp22118122p22133185.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Shannon From freeswitch at servercorps.com Sat Feb 21 07:57:08 2009 From: freeswitch at servercorps.com (Addison Martin) Date: Sat, 21 Feb 2009 09:57:08 -0600 Subject: [Freeswitch-users] Suggestion for xml_curl performance In-Reply-To: <7e2ac3270902210711u3ea54b79j5868aef6b59d4800@mail.gmail.com> References: <22118122.post@talk.nabble.com> <22133185.post@talk.nabble.com> <7e2ac3270902210711u3ea54b79j5868aef6b59d4800@mail.gmail.com> Message-ID: <92e7d2090902210757u5ec034bew53353b7838c01b78@mail.gmail.com> Is this a good application for the new ESL (Event Socket Library)interface? -anm On Sat, Feb 21, 2009 at 9:11 AM, Shannon wrote: > I'd recommend having a look at fastcgi as well. > > On 2/21/09, shehzad p wrote: >> >> Hi Brian, >> >> My setup is to use FS as basic calls routing. >> 1. Calls are coming to FS from more than one customer Gateways, and I need >> to authenticate them and check for enough balance based on database, >> [Caller Gateways] ===> [FreeSWITCH] ===> >> [Provider Gateways] >> 2. After knowing that Caller Gateways is valid, then based on dialed number >> it search in database for Provider Gateway and bridge the call there. >> 3. After call finish CDR is inserted back into database. >> >> My old setup was using Javascript which works fine in traffic of 10 to 20 >> calls, but then increase of traffic causes many problems. >> >> Now I eliminate use of any of the script (javascript or any other) for call >> routing, and route calls directly from dialplan, >> So I have setup test system using xml-curl to generate dynamic dialplan, >> I used below xml_curl PHP example as reference: >> http://wiki.freeswitch.org/wiki/Mod_xml_curl_PHP_example >> For CDR processing I used xml_cdr, with help of the example in FS source >> :scripts/contrib/trixter/xml-cdr. >> >> >> Waiting for any better suggestions, any comments... >> >> thanks >> msp. >> >> Brian West-3 wrote: >>> >>> it all depends on what you're doing.. can you elaborate? >>> >>> /b >>> >>> On Feb 20, 2009, at 4:18 AM, shehzad p wrote: >>> >>>> Recently I faced some performance bottleneck by using Javascript. >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> -- >> View this message in context: >> http://www.nabble.com/Suggestion-for-xml_curl-performance-tp22118122p22133185.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > -- > Shannon > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From freeswitch at servercorps.com Sat Feb 21 07:59:49 2009 From: freeswitch at servercorps.com (Addison Martin) Date: Sat, 21 Feb 2009 09:59:49 -0600 Subject: [Freeswitch-users] ESL In-Reply-To: References: Message-ID: <92e7d2090902210759v6f934421y9873a3cb526dd245@mail.gmail.com> I have ported all the Perl samples to python, and they appear to be working fine. They are available in svn rev 12210 and > -anm On Thu, Feb 19, 2009 at 12:44 PM, Brian West wrote: > FreeSWITCHers, > Not sure anyone is paying attention or not but Anthony wrapped the > ESL library up so you can use it from Perl, Python, Lua, Ruby and > PHP. What I'm requesting from our community is to help flex it out.. > write examples and populate the Wiki page with information about it. > > http://wiki.freeswitch.org/wiki/Esl > > Collins and I are going to start filling in the page but I want > someone thats good with Ruby, Python, PHP to help in those areas.. > kick in some lua and perl if you like. > > It works with OES and IES... (Outbound Event Socket and Inbound Event > Socket) Not sure those names are official but we have been calling > them that ;) > > Thanks, > Brian West > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From freeswitch at servercorps.com Sat Feb 21 09:00:39 2009 From: freeswitch at servercorps.com (Addison Martin) Date: Sat, 21 Feb 2009 11:00:39 -0600 Subject: [Freeswitch-users] Deployment information and use cases In-Reply-To: <1234916455.16581.49.camel@raul-laptop> References: <1234916455.16581.49.camel@raul-laptop> Message-ID: <92e7d2090902210900u7da015c3n7a815ab4beaf3b00@mail.gmail.com> We're still in the construction and design phase, but my company is building a multi-tenant Freeswitch based PBX for a Research Park in South Alabama. We expect to handle about 120 concurrent calls, and 6-700 registered UAs. The system will be based on commodity house-built SuperMicro servers, with mod_xml_curl handling all configuration. We will have PRIs for fax and 911, and SIP trunks to upstream ITSPs for most call volume. -anm On Tue, Feb 17, 2009 at 6:20 PM, Raul Fragoso wrote: > Hello FreeSWITCHERS, > > My company is currently creating a suite of applications which uses > FreeSWITCH as the back-end for an IP-PBX solution. We currently have a > prospect to have our first customer installation - a governmental > department. That is a tender to have an IP-PBX installation to connect > their four office branches, each one with about 300 users - which I am > sure FreeSWITCH is able to handle. Since this is an official tender, > it's part of their protocol to ask about real sites using the product. > > Having said that, would you mind sharing some information about your > experience with FreeSWITCH deployments ? > > No need to give many details, but a short summary with company name (if > possible), when it was deployed, server equipment, number of users, > number of concurrent calls, what kind of functions and services are used > and overall capacity of the system. > > I would really appreciate if you can share that information. And if you > guys agree (and explicitly manifest your agreement), I can compile the > information in the FreeSWITCH wiki under a "Use Cases" page so it can > serve as a common reference as well. > > Kind regards, > > Raul Fragoso > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From stevecrozz at gmail.com Sat Feb 21 11:31:19 2009 From: stevecrozz at gmail.com (Stephen Crosby) Date: Sat, 21 Feb 2009 11:31:19 -0800 Subject: [Freeswitch-users] js session.streamFile() interrupt slowness In-Reply-To: <87f2f3b90902200836o638fa345qec10888f39e46414@mail.gmail.com> References: <11990ade0902192127l414ecfeaia04fb80f626c35d5@mail.gmail.com> <87f2f3b90902200836o638fa345qec10888f39e46414@mail.gmail.com> Message-ID: <11990ade0902211131i134b892q71f18626b7ae54b5@mail.gmail.com> Sure, I've stripped down the script somewhat to something smaller that still produces this effect and you can see it at: http://pastebin.freeswitch.org/7388 The file sound 'VR1' continues to play for a short time after I interrupt it with a DTMF event. It does interrupt, but it sounds a little awkward because of the delay. I was probably wrong in my estimate of the delay which seems to be about a full second, not two or three. I'm hoping I can adjust it somehow to feel more immediate. Any ideas? --Stephen On Fri, Feb 20, 2009 at 8:36 AM, Michael Collins wrote: > On Thu, Feb 19, 2009 at 9:27 PM, Stephen Crosby wrote: >> I have a few scripts that use the javascript >> session.streamFile('somefile.wav', onDtmf); where onDtmf is a function >> that returns false to interrupt the streaming file. There is a short >> delay between the time when I press a key and the time the file stops >> playing. Is there anything I can adjust that would affect that? It's >> only maybe 2-3 seconds, but it "feels" too long to me. >> >> --Stephen > > Could you pastebin your entire script plus the relevant dialplan > entry? Also, could you tell us which operating system and FS revision? > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From stevecrozz at gmail.com Sat Feb 21 11:33:11 2009 From: stevecrozz at gmail.com (Stephen Crosby) Date: Sat, 21 Feb 2009 11:33:11 -0800 Subject: [Freeswitch-users] js session.streamFile() interrupt slowness In-Reply-To: <11990ade0902211131i134b892q71f18626b7ae54b5@mail.gmail.com> References: <11990ade0902192127l414ecfeaia04fb80f626c35d5@mail.gmail.com> <87f2f3b90902200836o638fa345qec10888f39e46414@mail.gmail.com> <11990ade0902211131i134b892q71f18626b7ae54b5@mail.gmail.com> Message-ID: <11990ade0902211133n5fba7a27k4ae9b2e90978abf1@mail.gmail.com> There was a small error in that last script I sent, please test using this version: http://pastebin.freeswitch.org/7388 Thanks. On Sat, Feb 21, 2009 at 11:31 AM, Stephen Crosby wrote: > Sure, I've stripped down the script somewhat to something smaller that > still produces this effect and you can see it at: > http://pastebin.freeswitch.org/7388 > > The file sound 'VR1' continues to play for a short time after I > interrupt it with a DTMF event. It does interrupt, but it sounds a > little awkward because of the delay. I was probably wrong in my > estimate of the delay which seems to be about a full second, not two > or three. I'm hoping I can adjust it somehow to feel more immediate. > Any ideas? > > --Stephen > > On Fri, Feb 20, 2009 at 8:36 AM, Michael Collins wrote: >> On Thu, Feb 19, 2009 at 9:27 PM, Stephen Crosby wrote: >>> I have a few scripts that use the javascript >>> session.streamFile('somefile.wav', onDtmf); where onDtmf is a function >>> that returns false to interrupt the streaming file. There is a short >>> delay between the time when I press a key and the time the file stops >>> playing. Is there anything I can adjust that would affect that? It's >>> only maybe 2-3 seconds, but it "feels" too long to me. >>> >>> --Stephen >> >> Could you pastebin your entire script plus the relevant dialplan >> entry? Also, could you tell us which operating system and FS revision? >> -MC >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > From stevecrozz at gmail.com Sat Feb 21 11:35:07 2009 From: stevecrozz at gmail.com (Stephen Crosby) Date: Sat, 21 Feb 2009 11:35:07 -0800 Subject: [Freeswitch-users] js session.streamFile() interrupt slowness In-Reply-To: <11990ade0902211133n5fba7a27k4ae9b2e90978abf1@mail.gmail.com> References: <11990ade0902192127l414ecfeaia04fb80f626c35d5@mail.gmail.com> <87f2f3b90902200836o638fa345qec10888f39e46414@mail.gmail.com> <11990ade0902211131i134b892q71f18626b7ae54b5@mail.gmail.com> <11990ade0902211133n5fba7a27k4ae9b2e90978abf1@mail.gmail.com> Message-ID: <11990ade0902211135o7180f06fj3d9c1fa6cdbb59b8@mail.gmail.com> Verry sorry for the list spam, this is the link to the corrected script: http://pastebin.freeswitch.org/7389 --Stephen On Sat, Feb 21, 2009 at 11:33 AM, Stephen Crosby wrote: > There was a small error in that last script I sent, please test using > this version: http://pastebin.freeswitch.org/7388 > > Thanks. > > On Sat, Feb 21, 2009 at 11:31 AM, Stephen Crosby wrote: >> Sure, I've stripped down the script somewhat to something smaller that >> still produces this effect and you can see it at: >> http://pastebin.freeswitch.org/7388 >> >> The file sound 'VR1' continues to play for a short time after I >> interrupt it with a DTMF event. It does interrupt, but it sounds a >> little awkward because of the delay. I was probably wrong in my >> estimate of the delay which seems to be about a full second, not two >> or three. I'm hoping I can adjust it somehow to feel more immediate. >> Any ideas? >> >> --Stephen >> >> On Fri, Feb 20, 2009 at 8:36 AM, Michael Collins wrote: >>> On Thu, Feb 19, 2009 at 9:27 PM, Stephen Crosby wrote: >>>> I have a few scripts that use the javascript >>>> session.streamFile('somefile.wav', onDtmf); where onDtmf is a function >>>> that returns false to interrupt the streaming file. There is a short >>>> delay between the time when I press a key and the time the file stops >>>> playing. Is there anything I can adjust that would affect that? It's >>>> only maybe 2-3 seconds, but it "feels" too long to me. >>>> >>>> --Stephen >>> >>> Could you pastebin your entire script plus the relevant dialplan >>> entry? Also, could you tell us which operating system and FS revision? >>> -MC >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> > From chavpaskov at shaw.ca Sat Feb 21 11:40:20 2009 From: chavpaskov at shaw.ca (Chav Paskov) Date: Sat, 21 Feb 2009 11:40:20 -0800 Subject: [Freeswitch-users] Passing Caller_ID to MOD_LCR In-Reply-To: <499F29E8.4020100@lists.rupa.com> References: <499A4497.9080001@lists.rupa.com> <499AC59A.4050209@lists.rupa.com> <499F29E8.4020100@lists.rupa.com> Message-ID: <49A058A4.7000609@shaw.ca> Rupa Schomaker (lists) wrote: >> "default" is a reserved profile name -- I should probably prevent that >> from loading. >> > > correction, default is not reserved... > > >> Regarding passing the callerid to the custom sql, let me see what I can >> come up with... >> > > You can now specify channel variables in your custom sql. So, you > should be able to pass CID or a subset of CID to your custom sql. I've > updated the wiki with info on this. > > beware: channel vars only work when called in the context of a session. > Using a profile that uses a custom sql with channel variables from the > commandline will result in an error. > > -Rupa > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > do i have to upgrade to the latest trunk in order to pass channel variables to mod_lcr? Currently the version used is 1.0.trunk (12134M) Regards Chav From freeswitch-users at lists.rupa.com Sat Feb 21 13:46:02 2009 From: freeswitch-users at lists.rupa.com (Rupa Schomaker (lists)) Date: Sat, 21 Feb 2009 15:46:02 -0600 Subject: [Freeswitch-users] Passing Caller_ID to MOD_LCR In-Reply-To: <49A058A4.7000609@shaw.ca> References: <499A4497.9080001@lists.rupa.com> <499AC59A.4050209@lists.rupa.com> <499F29E8.4020100@lists.rupa.com> <49A058A4.7000609@shaw.ca> Message-ID: <49A0761A.8090101@lists.rupa.com> > do i have to upgrade to the latest trunk in order to pass channel > variables to mod_lcr? > Currently the version used is 1.0.trunk (12134M) > Regards > Chav > You need at least 12204. From chavpaskov at shaw.ca Sat Feb 21 14:53:27 2009 From: chavpaskov at shaw.ca (Chav Paskov) Date: Sat, 21 Feb 2009 14:53:27 -0800 Subject: [Freeswitch-users] Passing Caller_ID to MOD_LCR In-Reply-To: <49A0761A.8090101@lists.rupa.com> References: <499A4497.9080001@lists.rupa.com> <499AC59A.4050209@lists.rupa.com> <499F29E8.4020100@lists.rupa.com> <49A058A4.7000609@shaw.ca> <49A0761A.8090101@lists.rupa.com> Message-ID: <49A085E7.3000506@shaw.ca> Rupa Schomaker (lists) wrote: >> do i have to upgrade to the latest trunk in order to pass channel >> variables to mod_lcr? >> Currently the version used is 1.0.trunk (12134M) >> Regards >> Chav >> >> > > You need at least 12204. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > Thank you very much. I'll try it and will keep you postged. Chav From anthony.minessale at gmail.com Sat Feb 21 15:29:50 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sat, 21 Feb 2009 17:29:50 -0600 Subject: [Freeswitch-users] js session.streamFile() interrupt slowness In-Reply-To: <11990ade0902211135o7180f06fj3d9c1fa6cdbb59b8@mail.gmail.com> References: <11990ade0902192127l414ecfeaia04fb80f626c35d5@mail.gmail.com> <87f2f3b90902200836o638fa345qec10888f39e46414@mail.gmail.com> <11990ade0902211131i134b892q71f18626b7ae54b5@mail.gmail.com> <11990ade0902211133n5fba7a27k4ae9b2e90978abf1@mail.gmail.com> <11990ade0902211135o7180f06fj3d9c1fa6cdbb59b8@mail.gmail.com> Message-ID: <191c3a030902211529j2637494dqc4c476a461e8cc5f@mail.gmail.com> Try this one http://pastebin.freeswitch.org/7391 I just tested this on latest trunk and it stopped instantly. On Sat, Feb 21, 2009 at 1:35 PM, Stephen Crosby wrote: > Verry sorry for the list spam, this is the link to the corrected script: > http://pastebin.freeswitch.org/7389 > > --Stephen > > On Sat, Feb 21, 2009 at 11:33 AM, Stephen Crosby > wrote: > > There was a small error in that last script I sent, please test using > > this version: http://pastebin.freeswitch.org/7388 > > > > Thanks. > > > > On Sat, Feb 21, 2009 at 11:31 AM, Stephen Crosby > wrote: > >> Sure, I've stripped down the script somewhat to something smaller that > >> still produces this effect and you can see it at: > >> http://pastebin.freeswitch.org/7388 > >> > >> The file sound 'VR1' continues to play for a short time after I > >> interrupt it with a DTMF event. It does interrupt, but it sounds a > >> little awkward because of the delay. I was probably wrong in my > >> estimate of the delay which seems to be about a full second, not two > >> or three. I'm hoping I can adjust it somehow to feel more immediate. > >> Any ideas? > >> > >> --Stephen > >> > >> On Fri, Feb 20, 2009 at 8:36 AM, Michael Collins > wrote: > >>> On Thu, Feb 19, 2009 at 9:27 PM, Stephen Crosby > wrote: > >>>> I have a few scripts that use the javascript > >>>> session.streamFile('somefile.wav', onDtmf); where onDtmf is a function > >>>> that returns false to interrupt the streaming file. There is a short > >>>> delay between the time when I press a key and the time the file stops > >>>> playing. Is there anything I can adjust that would affect that? It's > >>>> only maybe 2-3 seconds, but it "feels" too long to me. > >>>> > >>>> --Stephen > >>> > >>> Could you pastebin your entire script plus the relevant dialplan > >>> entry? Also, could you tell us which operating system and FS revision? > >>> -MC > >>> > >>> _______________________________________________ > >>> Freeswitch-users mailing list > >>> Freeswitch-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> http://www.freeswitch.org > >>> > >> > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090221/8aa4340b/attachment-0002.html From chavpaskov at shaw.ca Sat Feb 21 15:37:26 2009 From: chavpaskov at shaw.ca (Chav Paskov) Date: Sat, 21 Feb 2009 15:37:26 -0800 Subject: [Freeswitch-users] recommended settings for max-proceeding param Message-ID: <49A09036.2080905@shaw.ca> Hi Everybody, if it is not too much of a trouble can somebody point to a recommended value for max-proceeding in sofia.conf.xml ? If there is no recommended value what should be taken under consideration in order to determine one. I dug into the archives and discovered a thread called "Freeswitch freezes under increased call load" and there together with session per sec and max allowed sessions was recommended max-proceeding under sofia.conf.xml to be changed. I've just installed 1.0.3 version and checked the sofia.conf.xml file . I was not able to find a default setting for max-proceeding , when i added it i started getting cored umps. Regards Chav From alex at sinapticode.ro Sun Feb 22 07:30:37 2009 From: alex at sinapticode.ro (Alexandru Nedelcu) Date: Sun, 22 Feb 2009 17:30:37 +0200 Subject: [Freeswitch-users] origination_caller_id_number used instead of effective_caller_id_number In-Reply-To: <2CEAC5BC-D5F8-4253-A710-C5E60DAC7EB8@jerris.com> References: <1235039768.4537.17.camel@gathern.lan> <1235041389.4537.22.camel@gathern.lan> <1235053185.4537.59.camel@gathern.lan> <2CEAC5BC-D5F8-4253-A710-C5E60DAC7EB8@jerris.com> Message-ID: Yeap, it's fixed. On Thu, Feb 19, 2009 at 5:03 PM, Michael Jerris wrote: > Can you re-test this with current svn trunk. I believe this was fixed > yesterday. > > Mike > -- Alexandru Nedelcu Software Developer, Sinapticode From sicfslist at gmail.com Sun Feb 22 10:30:00 2009 From: sicfslist at gmail.com (Shelby Ramsey) Date: Sun, 22 Feb 2009 12:30:00 -0600 Subject: [Freeswitch-users] Compile Errors ... Message-ID: <35b355e90902221030n589cf95bxc33f5a57ab921046@mail.gmail.com> Hello, I'm getting this all over the place today: make[5]: *** No rule to make target `/usr/src/freeswitch/libfreeswitch.la', needed by `mod_commands.so'. Stop. make[4]: *** [all] Error 1 make[3]: *** [mod_commands-all] Error 1 make[2]: *** [all-recursive] Error 1 I normally do this: svn checkout http://svn.freeswitch.org/svn/freeswitch/trunk freeswitch.trunk ./bootstrap.sh ./configure make make install Thx! SDR -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090222/bfda4658/attachment-0002.html From mike at jerris.com Sun Feb 22 10:58:12 2009 From: mike at jerris.com (Michael Jerris) Date: Sun, 22 Feb 2009 13:58:12 -0500 Subject: [Freeswitch-users] Compile Errors ... In-Reply-To: <35b355e90902221030n589cf95bxc33f5a57ab921046@mail.gmail.com> References: <35b355e90902221030n589cf95bxc33f5a57ab921046@mail.gmail.com> Message-ID: <53236046-2332-440E-9B1C-DA693464836B@jerris.com> Look up furthur, there will be an error around where it builds or links the core. Try typing make core. Mike On Feb 22, 2009, at 1:30 PM, Shelby Ramsey wrote: > Hello, > > I'm getting this all over the place today: > > make[5]: *** No rule to make target `/usr/src/freeswitch/ > libfreeswitch.la', needed by `mod_commands.so'. Stop. > make[4]: *** [all] Error 1 > make[3]: *** [mod_commands-all] Error 1 > make[2]: *** [all-recursive] Error 1 > > I normally do this: > > svn checkout http://svn.freeswitch.org/svn/freeswitch/trunk > freeswitch.trunk > ./bootstrap.sh > ./configure > make > make install > > Thx! > > SDR > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090222/949901e3/attachment-0002.html From sicfslist at gmail.com Sun Feb 22 11:26:21 2009 From: sicfslist at gmail.com (Shelby Ramsey) Date: Sun, 22 Feb 2009 13:26:21 -0600 Subject: [Freeswitch-users] Compile Errors ... In-Reply-To: <53236046-2332-440E-9B1C-DA693464836B@jerris.com> References: <35b355e90902221030n589cf95bxc33f5a57ab921046@mail.gmail.com> <53236046-2332-440E-9B1C-DA693464836B@jerris.com> Message-ID: <35b355e90902221126r4ac44a00k690499546c04ae7b@mail.gmail.com> Mike, This is what I get when I run make core (after I did a checkout on latest svn trunk, ./bootstrap.sh, ./configure): src/switch_console.c:35:28: error: switch_version.h: No such file or directory src/switch_console.c: In function 'switch_console_process': src/switch_console.c:233: error: 'SWITCH_VERSION_FULL' undeclared (first use in this function) src/switch_console.c:233: error: (Each undeclared identifier is reported only once src/switch_console.c:233: error: for each function it appears in.) make[1]: *** [libfreeswitch_la-switch_console.lo] Error 1 make: *** [core] Error 2 Thanks! SDR -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090222/f19c1c28/attachment-0002.html From mike at jerris.com Sun Feb 22 11:55:28 2009 From: mike at jerris.com (Michael Jerris) Date: Sun, 22 Feb 2009 14:55:28 -0500 Subject: [Freeswitch-users] Compile Errors ... In-Reply-To: <35b355e90902221126r4ac44a00k690499546c04ae7b@mail.gmail.com> References: <35b355e90902221030n589cf95bxc33f5a57ab921046@mail.gmail.com> <53236046-2332-440E-9B1C-DA693464836B@jerris.com> <35b355e90902221126r4ac44a00k690499546c04ae7b@mail.gmail.com> Message-ID: There should be more errors above that? Are you cutting off some of them in your paste? On Feb 22, 2009, at 2:26 PM, Shelby Ramsey wrote: > Mike, > > This is what I get when I run make core (after I did a checkout on > latest svn trunk, ./bootstrap.sh, ./configure): > > src/switch_console.c:35:28: error: switch_version.h: No such file or > directory > src/switch_console.c: In function 'switch_console_process': > src/switch_console.c:233: error: 'SWITCH_VERSION_FULL' undeclared > (first use in this function) > src/switch_console.c:233: error: (Each undeclared identifier is > reported only once > src/switch_console.c:233: error: for each function it appears in.) > make[1]: *** [libfreeswitch_la-switch_console.lo] Error 1 > make: *** [core] Error 2 > > Thanks! > > SDR > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From sicfslist at gmail.com Sun Feb 22 12:09:45 2009 From: sicfslist at gmail.com (Shelby Ramsey) Date: Sun, 22 Feb 2009 14:09:45 -0600 Subject: [Freeswitch-users] Compile Errors ... In-Reply-To: References: <35b355e90902221030n589cf95bxc33f5a57ab921046@mail.gmail.com> <53236046-2332-440E-9B1C-DA693464836B@jerris.com> <35b355e90902221126r4ac44a00k690499546c04ae7b@mail.gmail.com> Message-ID: <35b355e90902221209t329e6819oae727031ce78a502@mail.gmail.com> Mike, the entire output can be seen @ http://www.sipinterchange.com/downloads/fs_compile_err.txt I don't see anything else. It's really odd ... SDR -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090222/4e49b439/attachment-0002.html From gkuri at ieee.org Sun Feb 22 13:01:41 2009 From: gkuri at ieee.org (Gabriel Kuri) Date: Sun, 22 Feb 2009 13:01:41 -0800 Subject: [Freeswitch-users] Compile Errors ... In-Reply-To: <35b355e90902221209t329e6819oae727031ce78a502@mail.gmail.com> References: <35b355e90902221030n589cf95bxc33f5a57ab921046@mail.gmail.com> <53236046-2332-440E-9B1C-DA693464836B@jerris.com> <35b355e90902221126r4ac44a00k690499546c04ae7b@mail.gmail.com> <35b355e90902221209t329e6819oae727031ce78a502@mail.gmail.com> Message-ID: <49A1BD35.4000308@ieee.org> It sounds like your build environment is whacked, is this a fresh checkout of trunk or did you overwrite an existing directory? You might want to scrap that directory and try a fresh checkout. I just freshly checked out a copy of trunk into a new dir and ran ./bootstrap, ./configure, and make without a problem. Gabe Shelby Ramsey wrote: > Mike, > > the entire output can be seen @ > http://www.sipinterchange.com/downloads/fs_compile_err.txt > > I don't see anything else. It's really odd ... > > SDR > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From sicfslist at gmail.com Sun Feb 22 13:13:01 2009 From: sicfslist at gmail.com (Shelby Ramsey) Date: Sun, 22 Feb 2009 15:13:01 -0600 Subject: [Freeswitch-users] Compile Errors ... In-Reply-To: <49A1BD35.4000308@ieee.org> References: <35b355e90902221030n589cf95bxc33f5a57ab921046@mail.gmail.com> <53236046-2332-440E-9B1C-DA693464836B@jerris.com> <35b355e90902221126r4ac44a00k690499546c04ae7b@mail.gmail.com> <35b355e90902221209t329e6819oae727031ce78a502@mail.gmail.com> <49A1BD35.4000308@ieee.org> Message-ID: <35b355e90902221313u6af1ad75kd751050457b51ad3@mail.gmail.com> Yeah ... that's what I did when I first got the error. I'll try it again. Thanks for the help! SDR -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090222/feb8f6f7/attachment-0002.html From sicfslist at gmail.com Sun Feb 22 14:52:43 2009 From: sicfslist at gmail.com (Shelby Ramsey) Date: Sun, 22 Feb 2009 16:52:43 -0600 Subject: [Freeswitch-users] Compile Errors ... In-Reply-To: <35b355e90902221313u6af1ad75kd751050457b51ad3@mail.gmail.com> References: <35b355e90902221030n589cf95bxc33f5a57ab921046@mail.gmail.com> <53236046-2332-440E-9B1C-DA693464836B@jerris.com> <35b355e90902221126r4ac44a00k690499546c04ae7b@mail.gmail.com> <35b355e90902221209t329e6819oae727031ce78a502@mail.gmail.com> <49A1BD35.4000308@ieee.org> <35b355e90902221313u6af1ad75kd751050457b51ad3@mail.gmail.com> Message-ID: <35b355e90902221452g3a9f705bnfb76f4c3095949ab@mail.gmail.com> Did this (to make sure I started from scratch): rm -rf /usr/src/freeswitch.trunk rm -rf /usr/local/freeswitch svn checkout http://svn.freeswitch.org/svn/freeswitch/trunk freeswitch.trunk cd freeswitch.trunk/ ./bootstrap.sh ./configure make ... and then it pukes all over the place with errors compiling odbc support ... but I thought that odbc was disabled by default. Errors look like this: Compiling src/switch_odbc.c ... In file included from src/switch_odbc.c:33: ./src/include/switch_odbc.h:36:17: error: sql.h: No such file or directory ./src/include/switch_odbc.h:43:20: error: sqlext.h: No such file or directory ./src/include/switch_odbc.h:45:22: error: sqltypes.h: No such file or directory In file included from src/switch_odbc.c:33: ./src/include/switch_odbc.h:66: error: expected declaration specifiers or '...' before 'SQLHSTMT' ./src/include/switch_odbc.h:96: error: expected declaration specifiers or '...' before 'SQLHSTMT' src/switch_odbc.c:43: error: expected specifier-qualifier-list before 'SQLHENV' src/switch_odbc.c: In function 'switch_odbc_handle_new': src/switch_odbc.c:76: error: 'switch_odbc_handle_t' has no member named 'env' src/switch_odbc.c:76: error: 'SQL_NULL_HANDLE' undeclared (first use in this function) src/switch_odbc.c:76: error: (Each undeclared identifier is reported only once src/switch_odbc.c:76: error: for each function it appears in.) src/switch_odbc.c:77: error: 'switch_odbc_handle_t' has no member named 'state' src/switch_odbc.c: In function 'switch_odbc_handle_disconnect': src/switch_odbc.c:96: error: 'switch_odbc_handle_t' has no member named 'state' cc1: warnings being treated as errors src/switch_odbc.c:97: warning: implicit declaration of function 'SQLDisconnect' src/switch_odbc.c:97: error: 'switch_odbc_handle_t' has no member named 'con' src/switch_odbc.c:105: error: 'switch_odbc_handle_t' has no member named 'state' src/switch_odbc.c: In function 'switch_odbc_handle_connect': src/switch_odbc.c:113: error: 'SQLINTEGER' undeclared (first use in this function) src/switch_odbc.c:113: error: expected ';' before 'err' src/switch_odbc.c:116: error: 'SQLSMALLINT' undeclared (first use in this function) src/switch_odbc.c:116: error: expected ';' before 'valueLength' SDR -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090222/13e59eb6/attachment-0002.html From gkuri at ieee.org Sun Feb 22 15:12:49 2009 From: gkuri at ieee.org (Gabriel Kuri) Date: Sun, 22 Feb 2009 15:12:49 -0800 Subject: [Freeswitch-users] Compile Errors ... In-Reply-To: <35b355e90902221452g3a9f705bnfb76f4c3095949ab@mail.gmail.com> References: <35b355e90902221030n589cf95bxc33f5a57ab921046@mail.gmail.com> <53236046-2332-440E-9B1C-DA693464836B@jerris.com> <35b355e90902221126r4ac44a00k690499546c04ae7b@mail.gmail.com> <35b355e90902221209t329e6819oae727031ce78a502@mail.gmail.com> <49A1BD35.4000308@ieee.org> <35b355e90902221313u6af1ad75kd751050457b51ad3@mail.gmail.com> <35b355e90902221452g3a9f705bnfb76f4c3095949ab@mail.gmail.com> Message-ID: <49A1DBF1.8000702@ieee.org> Someone can problem correct me if I'm wrong, however I believe a recent change was made to the configure script to try and autodetect ODBC. The configure script may be hitting a false positive as described in this similar thread from a few days ago ... http://lists.freeswitch.org/pipermail/freeswitch-users/2009-February/011762.html Gabe Shelby Ramsey wrote: > Did this (to make sure I started from scratch): > > rm -rf /usr/src/freeswitch.trunk > rm -rf /usr/local/freeswitch > svn checkout http://svn.freeswitch.org/svn/freeswitch/trunk freeswitch.trunk > cd freeswitch.trunk/ > ./bootstrap.sh > ./configure > make > > ... and then it pukes all over the place with errors compiling odbc > support ... but I thought that odbc was disabled by default. > > Errors look like this: > > Compiling src/switch_odbc.c ... > In file included from src/switch_odbc.c:33: > ./src/include/switch_odbc.h:36:17: error: sql.h: No such file or directory > ./src/include/switch_odbc.h:43:20: error: sqlext.h: No such file or > directory > ./src/include/switch_odbc.h:45:22: error: sqltypes.h: No such file or > directory > In file included from src/switch_odbc.c:33: > ./src/include/switch_odbc.h:66: error: expected declaration specifiers > or '...' before 'SQLHSTMT' > ./src/include/switch_odbc.h:96: error: expected declaration specifiers > or '...' before 'SQLHSTMT' > src/switch_odbc.c:43: error: expected specifier-qualifier-list before > 'SQLHENV' > src/switch_odbc.c: In function 'switch_odbc_handle_new': > src/switch_odbc.c:76: error: 'switch_odbc_handle_t' has no member named > 'env' > src/switch_odbc.c:76: error: 'SQL_NULL_HANDLE' undeclared (first use in > this function) > src/switch_odbc.c:76: error: (Each undeclared identifier is reported > only once > src/switch_odbc.c:76: error: for each function it appears in.) > src/switch_odbc.c:77: error: 'switch_odbc_handle_t' has no member named > 'state' > src/switch_odbc.c: In function 'switch_odbc_handle_disconnect': > src/switch_odbc.c:96: error: 'switch_odbc_handle_t' has no member named > 'state' > cc1: warnings being treated as errors > src/switch_odbc.c:97: warning: implicit declaration of function > 'SQLDisconnect' > src/switch_odbc.c:97: error: 'switch_odbc_handle_t' has no member named > 'con' > src/switch_odbc.c:105: error: 'switch_odbc_handle_t' has no member named > 'state' > src/switch_odbc.c: In function 'switch_odbc_handle_connect': > src/switch_odbc.c:113: error: 'SQLINTEGER' undeclared (first use in this > function) > src/switch_odbc.c:113: error: expected ';' before 'err' > src/switch_odbc.c:116: error: 'SQLSMALLINT' undeclared (first use in > this function) > src/switch_odbc.c:116: error: expected ';' before 'valueLength' > > > SDR > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From andrew at hijacked.us Sun Feb 22 15:32:16 2009 From: andrew at hijacked.us (Andrew Thompson) Date: Sun, 22 Feb 2009 18:32:16 -0500 Subject: [Freeswitch-users] mod_erlang_event compile problem In-Reply-To: <90DDD253-BA8D-4A79-9ACA-1560F2E18688@scarlet-internet.nl> References: <20090109230650.GF5210@hijacked.us> <4A496EA6-D036-4BB5-9347-5536AD5FD0FC@scarlet-internet.nl> <20090220190811.GC29511@hijacked.us> <90DDD253-BA8D-4A79-9ACA-1560F2E18688@scarlet-internet.nl> Message-ID: <20090222233215.GB13957@hijacked.us> On Sat, Feb 21, 2009 at 03:42:24PM +0100, Leon de Rooij wrote: > Hi Andrew, > > Thanks for your help so far, I hope you can help me a bit further as I > don't get any reply from the FS erlang node, or so it seems.. Here is > what I've done: > > - The erlang_event.conf.xml is unchanged: > > > > > > > > > > > You actually installed this to the right place? It's not installed by default... The defaults *should* be sane anyway, but I'm just checking. > > - mod_erlang_event is not loaded in FS. > > - First I start "epmd -d -d" > > epmd: Sat Feb 21 13:12:56 2009: epmd running - daemon = 0 > epmd: Sat Feb 21 13:12:56 2009: try to initiate listening port 4369 > epmd: Sat Feb 21 13:12:56 2009: starting > epmd: Sat Feb 21 13:12:56 2009: entering the main select() loop > > - After that I "load mod_erlang_event" in FS: > > 2009-02-21 13:13:36 [DEBUG] mod_erlang_event.c:1324 > mod_erlang_event_load() sections 16 > 2009-02-21 13:13:36 [CONSOLE] switch_loadable_module.c:858 > switch_loadable_module_load_file() Successfully Loaded > [mod_erlang_event] > 2009-02-21 13:13:36 [NOTICE] switch_loadable_module.c:240 > switch_loadable_module_process() Adding Application 'erlang' > 2009-02-21 13:13:36 [NOTICE] switch_loadable_module.c:260 > switch_loadable_module_process() Adding API Function 'erlang' > 2009-02-21 13:13:36 [DEBUG] mod_erlang_event.c:1401 > mod_erlang_event_runtime() Socket up listening on 127.0.0.1:8031 > 2009-02-21 13:13:36 [DEBUG] mod_erlang_event.c:1426 > mod_erlang_event_runtime() Connected and published erlang cnode at > freeswitch at erlyfs > > - For which epmd gives the following output: > > epmd: Sat Feb 21 13:13:36 2009: opening connection on file descriptor 4 > epmd: Sat Feb 21 13:13:36 2009: got 25 bytes > ***** 00000000 00 17 78 1f 5f 68 00 00 05 00 01 00 0a 66 72 65 > |..x._h.......fre| > ***** 00000010 65 73 77 69 74 63 68 00 00 | > eswitch..| > epmd: Sat Feb 21 13:13:36 2009: ** got ALIVE2_REQ > epmd: Sat Feb 21 13:13:36 2009: registering 'freeswitch:1', port 8031 > epmd: Sat Feb 21 13:13:36 2009: type 104 proto 0 highvsn 5 lowvsn 1 > epmd: Sat Feb 21 13:13:36 2009: got 4 bytes > ***** 00000000 79 00 00 01 |y...| > epmd: Sat Feb 21 13:13:36 2009: ** sent ALIVE2_RESP for "freeswitch" > > - Then I start an erl shell on that same machine with "erl -sname ldr - > setcookie ClueCon". Output of epmd: > > epmd: Sat Feb 21 13:16:24 2009: opening connection on file descriptor 5 > epmd: Sat Feb 21 13:16:24 2009: got 18 bytes > ***** 00000000 00 10 78 8e 2c 4d 00 00 05 00 05 00 03 6c 64 72 > |..x.,M.......ldr| > ***** 00000010 00 00 |..| > epmd: Sat Feb 21 13:16:24 2009: ** got ALIVE2_REQ > epmd: Sat Feb 21 13:16:24 2009: registering 'ldr:1', port 36396 > epmd: Sat Feb 21 13:16:24 2009: type 77 proto 0 highvsn 5 lowvsn 5 > epmd: Sat Feb 21 13:16:24 2009: got 4 bytes > ***** 00000000 79 00 00 01 |y...| > epmd: Sat Feb 21 13:16:24 2009: ** sent ALIVE2_RESP for "ldr" > > As far as I understand the freeswitch at erlyfs node cannot be seen with > nodes() ? So does that mean that I also cannot net_adm:ping() it ? Yes, it's a 'hidden' node, as all non-erlang nodes are. However, it should be visible in the output of epmd -names. > > Anyway, I tried sending some tuples as is shown on the wiki, but I get > no reply: > > (ldr at erlyfs)1> {foo, freeswitch at erlyfs} ! {api, status, ""}, receive X > -> X after 1000 -> timeout end. > timeout > (ldr at erlyfs)2> > > - Epmd gives some logs: > > epmd: Sat Feb 21 13:19:09 2009: opening connection on file descriptor 6 > epmd: Sat Feb 21 13:19:09 2009: got 13 bytes > ***** 00000000 00 0b 7a 66 72 65 65 73 77 69 74 63 68 > |..zfreeswitch| > epmd: Sat Feb 21 13:19:09 2009: ** got PORT2_REQ > epmd: Sat Feb 21 13:19:09 2009: got 23 bytes > ***** 00000000 77 00 1f 5f 68 00 00 05 00 01 00 0a 66 72 65 65 | > w.._h.......free| > ***** 00000010 73 77 69 74 63 68 00 | > switch.| > epmd: Sat Feb 21 13:19:09 2009: ** sent PORT2_RESP (ok) for "freeswitch" > epmd: Sat Feb 21 13:19:09 2009: closing connection on file descriptor 6 > > - And in tcpdump on lo, I see that epmd is contacted after which some > traffic was sent to FS: > > 13:19:09.535293 IP 172.31.0.13.34678 > 172.31.0.13.4369: S > 2875169966:2875169966(0) win 32792 17946545 0,nop,wscale 6> > ... > 13:19:09.536834 IP 172.31.0.13.4369 > 172.31.0.13.34678: . ack 15 win > 512 > 13:19:09.536923 IP 172.31.0.13.47054 > 172.31.0.13.8031: S > 2868322908:2868322908(0) win 32792 17946546 0,nop,wscale 6> > 13:19:09.536935 IP 172.31.0.13.8031 > 172.31.0.13.47054: R 0:0(0) ack > 2868322909 win 0 > > Shouldn't FS then send a message back to the process in my erl shell ? > > I tried logging all events in fs_cli, by entering "/event plain all", > but I see no events at all coming from erlang, just some heartbeats.. > > Also, I recompiled the module with EI_DEBUG defined as suggested on > the wiki. Still I don't see anything in the CLI when set to debug > logging. This is the part that's confusing me, you should be seeing *something*, especially with EI_DEBUG on, because in that case you see *everything* that is sent to or received by the erlang module. Let me to a 'make current' and doublecheck something didn't get broken along the way. Andrew From andrew at hijacked.us Sun Feb 22 16:12:12 2009 From: andrew at hijacked.us (Andrew Thompson) Date: Sun, 22 Feb 2009 19:12:12 -0500 Subject: [Freeswitch-users] mod_erlang_event compile problem In-Reply-To: <90DDD253-BA8D-4A79-9ACA-1560F2E18688@scarlet-internet.nl> References: <20090109230650.GF5210@hijacked.us> <4A496EA6-D036-4BB5-9347-5536AD5FD0FC@scarlet-internet.nl> <20090220190811.GC29511@hijacked.us> <90DDD253-BA8D-4A79-9ACA-1560F2E18688@scarlet-internet.nl> Message-ID: <20090223001211.GC13957@hijacked.us> Leon, I can't replicate your issue, at the very least I'd expect you to see the "Ignorable error in ei_accept - probable bad client version, bad cookie or bad nodename" warning. What OS/Erlang version are you using? Andrew From hochlehnert at hotmail.com Sun Feb 22 16:04:35 2009 From: hochlehnert at hotmail.com (Klaus Hochlehnert) Date: Mon, 23 Feb 2009 01:04:35 +0100 Subject: [Freeswitch-users] Question about BLF... Message-ID: Hi, I'm just playing around with FreeSWITCH and I have 2 questions about BLF (with SNOM phones): - When I played around with the sample dial plan I found out that BLF works better than Asterisk, but not 100% right: > When phone 1000 gets a call the BLF lamp on phone 1001 blinks and after phone 1000 takes the call the lamp on phone 1001 is on > But when phone 1000 gets a second call, takes it and hangs up the lamp on phone 1001 turns off even if the first call is still active > Is that a problem or did I do something wrong??? - Second question is how can I set up BLF if I want to have my dial plan completely in a perl script (no XML besides calling the perl script)? Thanks, Klaus From sicfslist at gmail.com Sun Feb 22 16:46:04 2009 From: sicfslist at gmail.com (Shelby Ramsey) Date: Sun, 22 Feb 2009 18:46:04 -0600 Subject: [Freeswitch-users] Compile Errors ... In-Reply-To: <49A1DBF1.8000702@ieee.org> References: <35b355e90902221030n589cf95bxc33f5a57ab921046@mail.gmail.com> <53236046-2332-440E-9B1C-DA693464836B@jerris.com> <35b355e90902221126r4ac44a00k690499546c04ae7b@mail.gmail.com> <35b355e90902221209t329e6819oae727031ce78a502@mail.gmail.com> <49A1BD35.4000308@ieee.org> <35b355e90902221313u6af1ad75kd751050457b51ad3@mail.gmail.com> <35b355e90902221452g3a9f705bnfb76f4c3095949ab@mail.gmail.com> <49A1DBF1.8000702@ieee.org> Message-ID: <35b355e90902221646m5f1cd40u9037d7755eea8f48@mail.gmail.com> Thanks for the help. That did the trick. SDR -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090222/0add7f9c/attachment-0002.html From mike at jerris.com Sun Feb 22 20:45:47 2009 From: mike at jerris.com (Michael Jerris) Date: Sun, 22 Feb 2009 23:45:47 -0500 Subject: [Freeswitch-users] Compile Errors ... In-Reply-To: <35b355e90902221646m5f1cd40u9037d7755eea8f48@mail.gmail.com> References: <35b355e90902221030n589cf95bxc33f5a57ab921046@mail.gmail.com> <53236046-2332-440E-9B1C-DA693464836B@jerris.com> <35b355e90902221126r4ac44a00k690499546c04ae7b@mail.gmail.com> <35b355e90902221209t329e6819oae727031ce78a502@mail.gmail.com> <49A1BD35.4000308@ieee.org> <35b355e90902221313u6af1ad75kd751050457b51ad3@mail.gmail.com> <35b355e90902221452g3a9f705bnfb76f4c3095949ab@mail.gmail.com> <49A1DBF1.8000702@ieee.org> <35b355e90902221646m5f1cd40u9037d7755eea8f48@mail.gmail.com> Message-ID: <10314107-A6F5-4E16-B8F4-CF604B4A0F45@jerris.com> can you please contact me off list and get me information to access your box so I can try to correct this for good in tree. Thanks Mike On Feb 22, 2009, at 7:46 PM, Shelby Ramsey wrote: > Thanks for the help. That did the trick. > > SDR From leon at scarlet-internet.nl Mon Feb 23 02:09:28 2009 From: leon at scarlet-internet.nl (Leon de Rooij) Date: Mon, 23 Feb 2009 11:09:28 +0100 Subject: [Freeswitch-users] mod_erlang_event compile problem In-Reply-To: <20090223001211.GC13957@hijacked.us> References: <20090109230650.GF5210@hijacked.us> <4A496EA6-D036-4BB5-9347-5536AD5FD0FC@scarlet-internet.nl> <20090220190811.GC29511@hijacked.us> <90DDD253-BA8D-4A79-9ACA-1560F2E18688@scarlet-internet.nl> <20090223001211.GC13957@hijacked.us> Message-ID: <0DEC394E-4739-4055-B524-E4217593DD3C@scarlet-internet.nl> Hi Andrew, Everything is running on an Ubuntu Hardy Xen domu with kernel 2.6.24-23-xen. Erlang is version R12B5 and was compiled from source with options -- enable-hipe, --enable-smp-support en --enable-threads. FS is trunk version 12197. I did copy the configuration file to ~freeswitch/conf/autoload_configs Also, I just checked the 'empd -names', after both FS and an erl shell have been started: root at erlyfs:~# epmd -names epmd: up and running on port 4369 with data: name ldr at port 57114 name freeswitch at port 8031 So that should be fine.. I also tried loading mod_erlang_event from modules.conf, and starting FS as root, but - not surprisingly - that didn't make any difference. I've been looking in wireshark, what exactly is going over the line, and the strange thing is, that erl opens a TCP connection, a SYN packet is sent to FS, after which FS immediately returns an RST/ACK packet and thus closes the connection.. I still don't see anything in the FS CLI. Is there anything I can do to get more verbose output from FS - esp info about why the connection was closed ? thanks, Leon On Feb 23, 2009, at 1:12 AM, Andrew Thompson wrote: > Leon, > > I can't replicate your issue, at the very least I'd expect you to see > the "Ignorable error in ei_accept - probable bad client version, bad > cookie or bad nodename" warning. What OS/Erlang version are you using? > > Andrew > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From Claudio.Cavalera at italtel.it Mon Feb 23 03:09:26 2009 From: Claudio.Cavalera at italtel.it (Cavalera Claudio Luigi) Date: Mon, 23 Feb 2009 12:09:26 +0100 Subject: [Freeswitch-users] Random problems with cepstral text to speech In-Reply-To: Message-ID: freeswitch-users-bounces at lists.freeswitch.org wrote: > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Anthony Minessale > >> Are you using cepstral 5.1? >> There is a known issue with that release and it's closed > source so we > cannot do much about it. >> Cepstral 4.x works fine. > > Yes 5.1, my fault. > I have added an initial warning here on the wiki > http://wiki.freeswitch.org/wiki/Mod_cepstral > although it also speaks about 5.1 and Ubuntu... Hello, if Cepstral 4.x is the way to go does anybody know where to get the demo version? BRs, Claudio Internet Email Confidentiality Footer ----------------------------------------------------------------------------------------------------- La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. ----------------------------------------------------------------------------------------------------- From pekkis50 at gmail.com Mon Feb 23 04:28:29 2009 From: pekkis50 at gmail.com (Pekka Kurki) Date: Mon, 23 Feb 2009 13:28:29 +0100 Subject: [Freeswitch-users] undefined symbol: krb5_auth_con_getrcache** Message-ID: <49A2966D.7010004@gmail.com> got this error when starting freeswitch -latest svn 2009-02-23 13:22:50 [CRIT] switch_loadable_module.c:840 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_sofia.so **/usr/local/freeswitch/mod/mod_sofia.so: undefined symbol: krb5_auth_con_getrcache** no ccompile errors, krb5libs installed, config with and w/o libcurl. br /pekka From kerrada2003 at yahoo.com Mon Feb 23 06:42:12 2009 From: kerrada2003 at yahoo.com (Ali Al-Rubaie) Date: Mon, 23 Feb 2009 06:42:12 -0800 (PST) Subject: [Freeswitch-users] Realm Value In-Reply-To: Message-ID: <711428.49972.qm@web33708.mail.mud.yahoo.com> I could compile and install FS 1.0.2 successsfully so, do I need to install ODBC-devel package for 1.0.3 version? Thanks, Message: 5 Date: Thu, 19 Feb 2009 14:35:29 -0800 From: Michael Collins Subject: Re: [Freeswitch-users] Realm Value To: freeswitch-users at lists.freeswitch.org Message-ID: <87f2f3b90902191435p1c9c03aend3303dfb013495b1 at mail.gmail.com> Content-Type: text/plain; charset=ISO-8859-1 On Thu, Feb 19, 2009 at 12:17 PM, Raymond Chandler wrote: > did you ./configure --enable-core-odbc-suport... those errors reek of > that flag with no unixODBC-devel package installed > > -Ray > Anthony described this as a false positive on detecting ODBC. If you are in Linux you can install the ODBC-devel package and be done with it. -MC ********************** -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090223/72ca12ae/attachment-0002.html From helmut.kuper at ewetel.de Mon Feb 23 06:44:45 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Mon, 23 Feb 2009 15:44:45 +0100 Subject: [Freeswitch-users] OpenZAP codec Question: Why L16@8000 codec for incoming calls Message-ID: <49A2B65D.2080003@ewetel.de> Hello, today I found in FS logfile lines like this: 2009-02-23 15:27:12 [DEBUG] switch_ivr_originate.c:1605 switch_ivr_originate() Raw Codec Activation Success L16 at 8000hz 1 channel 20ms It looks like L16 codec is used for incoming calls: 2009-02-23 15:27:03 [DEBUG] switch_core_session.c:523 switch_core_session_perform_receive_message() Send signal OpenZAP/1:18/2799 [BREAK] 2009-02-23 15:27:03 [NOTICE] switch_ivr_originate.c:1588 switch_ivr_originate() Pre-Answer OpenZAP/1:18/2799! 2009-02-23 15:27:03 [DEBUG] switch_ivr_originate.c:1605 switch_ivr_originate() Raw Codec Activation Success L16 at 8000hz 1 channel 20ms 2009-02-23 15:27:03 [DEBUG] switch_ivr_originate.c:1664 switch_ivr_originate() Play Ringback Tone [%(1000, 4000, 425.0, 0)] 2009-02-23 15:27:03 [DEBUG] sofia.c:2725 sofia_handle_sip_i_state() Channel sofia/internal/sip:2799 at 85.16.245.254:5060;user=phone;transport=udp entering state [proceeding] 2009-02-23 15:27:03 [NOTICE] sofia.c:2779 sofia_handle_sip_i_state() Ring-Ready sofia/internal/sip:2799 at 85.16.245.254:5060;user=phone;transport=udp! 2009-02-23 15:27:03 [DEBUG] switch_core_io.c:652 switch_core_session_write_frame() OpenZAP/1:18/2799 receive message [TRANSCODING_NECESSARY] 2009-02-23 15:27:07 [DEBUG] Span:1 Q.931() Timer 0 of call 6 (CRV: 61, State: 0) timed out 2009-02-23 15:27:12 [DEBUG] sofia.c:2725 sofia_handle_sip_i_state() Channel sofia/internal/sip:2799 at 85.16.245.254:5060;user=phone;transport=udp entering state [ready] 2009-02-23 15:27:12 [DEBUG] sofia.c:2729 sofia_handle_sip_i_state() Remote SDP: v=0^M o=2799 121183017 121183017 IN IP4 85.16.245.254^M s=ATA186 Call^M c=IN IP4 85.16.245.254^M t=0 0^M m=audio 16384 RTP/AVP 8 101^M a=rtpmap:8 PCMA/8000/1^M a=rtpmap:101 telephone-event/8000^M a=fmtp:101 0-15^M 2009-02-23 15:27:12 [DEBUG] sofia_glue.c:2549 sofia_glue_negotiate_sdp() Audio Codec Compare [PCMA:8:8000:0]/[PCMA:8:8000:20] 2009-02-23 15:27:12 [DEBUG] sofia_glue.c:1684 sofia_glue_tech_set_codec() Set Codec sofia/internal/sip:2799 at 85.16.245.254:5060;user=phone;transport=udp PCMA/8000 20 ms 160 samples The audio codec compare function finds slightly different codecs for A and B party. The dialplan for incoming calls via openzap is this. I set the codec to use in extensions "bridge" line: In my vars.xml config I have these codecs configured: So where can I disable the L16 codec, or why is a transcoding necessary? regards Helmut From brian at freeswitch.org Mon Feb 23 07:16:30 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 23 Feb 2009 09:16:30 -0600 Subject: [Freeswitch-users] Realm Value In-Reply-To: <711428.49972.qm@web33708.mail.mud.yahoo.com> References: <711428.49972.qm@web33708.mail.mud.yahoo.com> Message-ID: You have something on your system thats causing the audio detect to see you have odbc installed.. easiest way to get around this is to just install the devel headers. http://lists.freeswitch.org/pipermail/freeswitch-users/2009-February/011762.html /b On Feb 23, 2009, at 8:42 AM, Ali Al-Rubaie wrote: > > I could compile and install FS 1.0.2 successsfully so, do I need to > install ODBC-devel package for 1.0.3 version? > > Thanks, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090223/694a3fed/attachment-0002.html From mike at jerris.com Mon Feb 23 07:40:25 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 23 Feb 2009 10:40:25 -0500 Subject: [Freeswitch-users] OpenZAP codec Question: Why L16@8000 codec for incoming calls In-Reply-To: <49A2B65D.2080003@ewetel.de> References: <49A2B65D.2080003@ewetel.de> Message-ID: <6352F00A-EC49-4A12-91ED-2EA56C2A09C5@jerris.com> On Feb 23, 2009, at 9:44 AM, Helmut Kuper wrote: > Hello, > > today I found in FS logfile lines like this: > > 2009-02-23 15:27:12 [DEBUG] switch_ivr_originate.c:1605 > switch_ivr_originate() Raw Codec Activation Success L16 at 8000hz 1 > channel > 20ms > > > It looks like L16 codec is used for incoming calls: > > 2009-02-23 15:27:03 [DEBUG] switch_core_session.c:523 > switch_core_session_perform_receive_message() Send signal > OpenZAP/1:18/2799 [BREAK] > 2009-02-23 15:27:03 [NOTICE] switch_ivr_originate.c:1588 > switch_ivr_originate() Pre-Answer OpenZAP/1:18/2799! > 2009-02-23 15:27:03 [DEBUG] switch_ivr_originate.c:1605 > switch_ivr_originate() Raw Codec Activation Success L16 at 8000hz 1 > channel > 20ms > 2009-02-23 15:27:03 [DEBUG] switch_ivr_originate.c:1664 > switch_ivr_originate() Play Ringback Tone [%(1000, 4000, 425.0, 0)] > 2009-02-23 15:27:03 [DEBUG] sofia.c:2725 sofia_handle_sip_i_state() > Channel > sofia/internal/sip:2799 at 85.16.245.254:5060;user=phone;transport=udp > entering state [proceeding] > 2009-02-23 15:27:03 [NOTICE] sofia.c:2779 sofia_handle_sip_i_state() > Ring-Ready > sofia/internal/sip:2799 at 85.16.245.254:5060;user=phone;transport=udp! > 2009-02-23 15:27:03 [DEBUG] switch_core_io.c:652 > switch_core_session_write_frame() OpenZAP/1:18/2799 receive message > [TRANSCODING_NECESSARY] > 2009-02-23 15:27:07 [DEBUG] Span:1 Q.931() Timer 0 of call 6 (CRV: 61, > State: 0) timed out > 2009-02-23 15:27:12 [DEBUG] sofia.c:2725 sofia_handle_sip_i_state() > Channel > sofia/internal/sip:2799 at 85.16.245.254:5060;user=phone;transport=udp > entering state [ready] > 2009-02-23 15:27:12 [DEBUG] sofia.c:2729 sofia_handle_sip_i_state() > Remote SDP: > v=0^M > o=2799 121183017 121183017 IN IP4 85.16.245.254^M > s=ATA186 Call^M > c=IN IP4 85.16.245.254^M > t=0 0^M > m=audio 16384 RTP/AVP 8 101^M > a=rtpmap:8 PCMA/8000/1^M > a=rtpmap:101 telephone-event/8000^M > a=fmtp:101 0-15^M > > 2009-02-23 15:27:12 [DEBUG] sofia_glue.c:2549 > sofia_glue_negotiate_sdp() > Audio Codec Compare [PCMA:8:8000:0]/[PCMA:8:8000:20] > 2009-02-23 15:27:12 [DEBUG] sofia_glue.c:1684 > sofia_glue_tech_set_codec() Set Codec > sofia/internal/sip:2799 at 85.16.245.254:5060;user=phone;transport=udp > PCMA/8000 20 ms 160 samples > > The audio codec compare function finds slightly different codecs for A > and B party. > > The dialplan for incoming calls via openzap is this. I set the codec > to > use in extensions "bridge" line: > > > expression="(491[0-9]|492[0-8])$"> > > > data="nolocal:sip_secure_media=${user_data(${dialed_extension}@$ > {domain_name} > var sip_secure_media)}"/> > data="{absolute_codec_string=PCMA}user/$1@$${domain}"/> > > > > > In my vars.xml config I have these codecs configured: > > > > > So where can I disable the L16 codec, or why is a transcoding > necessary? Your playing a tone, we need to encode that tone into the codec of the channel. You could make it stop transcoding by not providing ringback but we are still doing some transcoding for the tone detection in openzap that you won't see via log messages. Why is this transcoding a problem? Mike From mike at jerris.com Mon Feb 23 07:41:38 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 23 Feb 2009 10:41:38 -0500 Subject: [Freeswitch-users] Realm Value In-Reply-To: References: <711428.49972.qm@web33708.mail.mud.yahoo.com> Message-ID: <2A6B9260-50A4-4DDE-808C-1F68AF2EBF45@jerris.com> I need someone with this issue to provide me ssh access to their box so I can fix this problem for everyone. No one has done so yet. Please find me on irc if you can provide access. Mike On Feb 23, 2009, at 10:16 AM, Brian West wrote: > You have something on your system thats causing the audio detect to > see you have odbc installed.. easiest way to get around this is to > just install the devel headers. > > http://lists.freeswitch.org/pipermail/freeswitch-users/2009-February/011762.html > > /b > > > On Feb 23, 2009, at 8:42 AM, Ali Al-Rubaie wrote: > >> >> I could compile and install FS 1.0.2 successsfully so, do I need to >> install ODBC-devel package for 1.0.3 version? >> >> Thanks, > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090223/30f0c6ab/attachment-0002.html From carlos.talbot at gmail.com Mon Feb 23 08:26:54 2009 From: carlos.talbot at gmail.com (Carlos Talbot) Date: Mon, 23 Feb 2009 10:26:54 -0600 Subject: [Freeswitch-users] Skypiax (Skype Network endpoint) for FreeSWITCH In-Reply-To: <7b197bef0902191732i6fead849uace0ac906a9437b0@mail.gmail.com> References: <7b197bef0902131015k456dee87x8f0f1b7059f1b49c@mail.gmail.com> <5800526b0902181853h32de5535l6c0f9d7067a69cf0@mail.gmail.com> <7b197bef0902191732i6fead849uace0ac906a9437b0@mail.gmail.com> Message-ID: <5800526b0902230826m255e0f4fmeeece95ed44e8cb4@mail.gmail.com> Thanks Giovanni. Were you planning to check in the sample skype.conf.xml into the default FreeSWITCH conf folder? If so, just be aware the default config causes freeswitch to hang right after a "load mod_skypiax" (if you do not have skype running or specify a nonexistant skype user). regards, Carlos On Thu, Feb 19, 2009 at 7:32 PM, Giovanni Maruzzelli wrote: > On Thu, Feb 19, 2009 at 3:53 AM, Carlos Talbot > wrote: > > > One question I have, is ringback suppose to work with mod_skypiax? > Whenever > > I dial a number I get a few seconds of dead air before the call is > answered. > > I've tried adding ringback and transfer_ringback into the dialplan just > > before the bridge command but no go. Am I missing something? Thanks. > > Carlos, > > ringback now works without tricks, and Skypiax is in trunk. > > Both remote ringing and early media are treated as remote ringing > right now (eg: no early media, just ringing). > > I'll add early media support in the near future. > > Thanks a lot for testing and exercising skypiax, and please let me > know any hint, suggestion, feature request, etc > > > > Sincerely, > > Giovanni Maruzzelli > ========================================= > www.celliax.org > via Pierlombardo 9, 20135 Milano > Italy > gmaruzz at celliax dot org > Cell : +39-347-2665618 > Fax : +39-02-87390039 > > > > > On Thu, Feb 19, 2009 at 3:53 AM, Carlos Talbot > wrote: > > Giovannia, > > > > great work on mod_skypiax. I've been testing it under Windows and it > sounds > > great including PSTN calls. I plan to include it as part of the Windows > MSI > > build. > > > > One question I have, is ringback suppose to work with mod_skypiax? > Whenever > > I dial a number I get a few seconds of dead air before the call is > answered. > > I've tried adding ringback and transfer_ringback into the dialplan just > > before the bridge command but no go. Am I missing something? Thanks. > > > > regards, > > > > Carlos > > > > > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090223/dfd925ef/attachment-0002.html From andrew at hijacked.us Mon Feb 23 09:13:41 2009 From: andrew at hijacked.us (Andrew Thompson) Date: Mon, 23 Feb 2009 12:13:41 -0500 Subject: [Freeswitch-users] mod_erlang_event compile problem In-Reply-To: <0DEC394E-4739-4055-B524-E4217593DD3C@scarlet-internet.nl> References: <20090109230650.GF5210@hijacked.us> <4A496EA6-D036-4BB5-9347-5536AD5FD0FC@scarlet-internet.nl> <20090220190811.GC29511@hijacked.us> <90DDD253-BA8D-4A79-9ACA-1560F2E18688@scarlet-internet.nl> <20090223001211.GC13957@hijacked.us> <0DEC394E-4739-4055-B524-E4217593DD3C@scarlet-internet.nl> Message-ID: <20090223171340.GD13957@hijacked.us> On Mon, Feb 23, 2009 at 11:09:28AM +0100, Leon de Rooij wrote: > Everything is running on an Ubuntu Hardy Xen domu with kernel > 2.6.24-23-xen. Oh, this might explain some things.. > > Erlang is version R12B5 and was compiled from source with options -- > enable-hipe, --enable-smp-support en --enable-threads. > I'm running this too. > FS is trunk version 12197. > Fine too. > I did copy the configuration file to ~freeswitch/conf/autoload_configs > > Also, I just checked the 'empd -names', after both FS and an erl shell > have been started: > > root at erlyfs:~# epmd -names > epmd: up and running on port 4369 with data: > name ldr at port 57114 > name freeswitch at port 8031 > > So that should be fine.. > Yes, that's correct. > I also tried loading mod_erlang_event from modules.conf, and > starting FS as root, but - not surprisingly - that didn't make any > difference. > > I've been looking in wireshark, what exactly is going over the line, > and the strange thing is, that erl opens a TCP connection, a SYN > packet is sent to FS, after which FS immediately returns an RST/ACK > packet and thus closes the connection.. I still don't see anything in > the FS CLI. > > Is there anything I can do to get more verbose output from FS - esp > info about why the connection was closed ? > It looks like ei_accept_tmo is the one resetting the connection, not my code. I can't even get an error out of it when, for example, I telnet to port 8031, it just closes the connection instantly with no error to the console. Is it possible that something is screwy with the loopback device in a xen guest? Can you get normal erlang nodes on that host to net_adm:ping each other? Andrew From helmut.kuper at ewetel.de Mon Feb 23 09:28:53 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Mon, 23 Feb 2009 18:28:53 +0100 Subject: [Freeswitch-users] OpenZAP codec Question: Why L16@8000 codec for incoming calls In-Reply-To: <6352F00A-EC49-4A12-91ED-2EA56C2A09C5@jerris.com> References: <49A2B65D.2080003@ewetel.de> <6352F00A-EC49-4A12-91ED-2EA56C2A09C5@jerris.com> Message-ID: <49A2DCD5.5090105@ewetel.de> Hi Mike, thx. Today we had some failing test fax sessions (g711/PCMA) and my first thought was that it could be caused by FS during transcoding. So I looked into FS logfile and found those hints about transcoding. But ringback shouldn't be the problem. Fax path was from FAX device (source) through a voip cpe, through a SBC through a SS7 PSTN device into PSTN. Then from PSTN through AVAYA PBX trough sangoma A104d into freeswitch. From there RTP goes to a Cisco ATA to be converted to TDM and consumed by a FAX device (Target). So we have a lot of points to look at ... regrads Helmut > Your playing a tone, we need to encode that tone into the codec of the > channel. You could make it stop transcoding by not providing ringback > but we are still doing some transcoding for the tone detection in > openzap that you won't see via log messages. Why is this transcoding > a problem? > > Mike > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > -- Mit freundlichen Gr??en Helmut Kuper Finanzdienstleistungen und Entwicklung Telefax: (0441) 8000-2799 mailto:helmut.kuper at ewetel.de ___________________________________ EWE TEL GmbH Cloppenburger Stra?e 310 26133 Oldenburg EWE TEL GmbH Handelsregister Amtsgericht Oldenburg HRB 3723 Vorsitzender des Aufsichtsrates: Heiko Harms Gesch?ftsf?hrung: Hans-Joachim Iken (Vorsitzender), Dr. Norbert Schulz, Dirk Thole Homepage: http://www.ewetel.de ___________________________________ From helmut.kuper at ewetel.de Mon Feb 23 09:29:04 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Mon, 23 Feb 2009 18:29:04 +0100 Subject: [Freeswitch-users] OpenZAP codec Question: Why L16@8000 codec for incoming calls In-Reply-To: <6352F00A-EC49-4A12-91ED-2EA56C2A09C5@jerris.com> References: <49A2B65D.2080003@ewetel.de> <6352F00A-EC49-4A12-91ED-2EA56C2A09C5@jerris.com> Message-ID: <49A2DCE0.7090204@ewetel.de> Hi Mike, thx. Today we had some failing test fax sessions (g711/PCMA) and my first thought was that it could be caused by FS during transcoding. So I looked into FS logfile and found those hints about transcoding. But ringback shouldn't be the problem. Fax path was from FAX device (source) through a voip cpe, through a SBC through a SS7 PSTN device into PSTN. Then from PSTN through AVAYA PBX trough sangoma A104d into freeswitch. From there RTP goes to a Cisco ATA to be converted to TDM and consumed by a FAX device (Target). So we have a lot of points to look at ... regrads Helmut > Your playing a tone, we need to encode that tone into the codec of the > channel. You could make it stop transcoding by not providing ringback > but we are still doing some transcoding for the tone detection in > openzap that you won't see via log messages. Why is this transcoding > a problem? > > Mike > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > From BenHoltsclaw at averyschools.net Mon Feb 23 09:52:11 2009 From: BenHoltsclaw at averyschools.net (Ben Holtsclaw) Date: Mon, 23 Feb 2009 12:52:11 -0500 Subject: [Freeswitch-users] Deployment information and use cases In-Reply-To: <7405656F-1AAA-4FF7-9DC3-4CE694D7B0AB@gmail.com> References: <1234916455.16581.49.camel@raul-laptop> <59ad9ca10902181009p7cbc0fd1s9860aacde7f3a30c@mail.gmail.com> <247f8100902181019m61ce803m5ed504d9b198dfb6@mail.gmail.com> <1235016793.22050.0.camel@raul-laptop> <499D796E.45B7.0079.0@averyschools.net> <7405656F-1AAA-4FF7-9DC3-4CE694D7B0AB@gmail.com> Message-ID: <49A29BFB.45B7.0079.0@averyschools.net> Mesquita, Relatively speaking, I feel like we are near the end of our project roll out. Perhaps the case would be stronger once everything is completed. At that time, I will be very glad to share the story on the wiki -- and hopefully elsewhere! Ben >>> On 2/21/2009 at 8:56 AM, Jo?o Mesquita wrote: Ben, thank you for your story. I would very much like to add this to the wiki if you don't mind and everyone else agrees. What do you think guys? Use cases are _ALWAYS_ a good thing for new users. Mesquita On Feb 19, 2009, at 6:23 PM, Ben Holtsclaw wrote: Raul, I am in the process of rolling out a FreeSWITCH IP PBX solution similar to what you describe. When I was trying to procure funds for a FreeSWITCH solution, I looked for the same information you're after, but came up with little. I'll briefly describe what we're trying to accomplish, and the tools I'm using to do it. This is probably more information than what you are looking for, but maybe it will also benefit someone else. We had several schools with aging or dying PBX's or KSU's. Each site had something different system, and was supported by a different VAR. Of course, the VAR's charged some outlandish fee to make onsite repair visits. Some number of Centrex lines supplied each school's dial tone. All in all, we had a very outdated and financially draining mess. Our district's long term goal had been to move to a more unified phone system. That made sense for many reasons, the chief of which was cost. We already had a strong fiber WAN in place. Why not use that for trunking and eliminate the monthly cost of the Centrex lines? That's the path we started down. Being a public entity, we had to be sure to explore all possible avenues. We looked at everything from traditional PBX's with IP add-on modules for trunking to a full blown Cisco CallManager solution. With third party proprietary systems, we were just never able to find the sweet spot between required feature set and cost. Would Cisco have been a workable solution? Absolutely. Could our small, rural, K12 public school district afford that? Not in a million years. I looked at several software packages -- some open source, some not -- but always came back to FreeSWITCH. The scalability and active development community were major factors for us. Server Hardware. Each of our five sites has a dedicated FreeSWITCH server. For hardware, we went with Dell PowerEdge 1950's with dual quad core Xeon 2.33 GHz processors, and 4GB of RAM. I have mirrored disks set up with enough space to accommodate users' voicemail. Each server will average only about 60 voicemail boxes, and we're storing sound as MP3. Disk space shouldn't be an issue. We have always been a Novell shop, so SLES is naturally our Linux distribution of choice. We chose to go with server hardware at each site so that in the event of a WAN outage, we would still at least have intra-building and emergency communication (see below). Telephony Hardware. Each of our servers includes Sangoma hardware. We actually looked at doing IP trunking to a carrier from our network core, but decided to use telco provided PRI's instead. Presently, we have two PRI's that connect to a FreeSWITCH server at the center of our network via a Sangoma A102 dual port telephony card. All calls to and from the PSTN traverse this primary server. Servers at each remote site include one of Sangoma's A200 analog cards. Emergency calls to 911 route out over this analog card through one of at least two POTS lines that remain connected at each site. Not only does this provide some redundancy in the event of a WAN outage, but it ensures proper caller location is delivered to the 911 dispatcher. Granted, there are some other solutions for the latter, but this seemed to be the most cost effective solution for us. Telephone Desksets. We chose to go with Aastra for the telephones. The standard phone that we will place in each classroom and office is the 9143i. This is an attractive phone with an adequate feature set at a price we can afford. The person that is primarily responsible for answering the phone at each site will have an Aastra 57i and some number of 560M expansion modules. We have purchased roughly 300 Aastra desksets. Logical Layout. As new sites come online, their primary phone number is being moved from the Centrex to our PRI group. All inbound calls hit our primary server, and then FreeSWITCH bridges to the appropriate secondary server based on the DID it received. On the reverse, each servers dial plan is set up to route outbound calls (save 911) to the primary server where FreeSWITCH bridges with Openzap. Site to site calls, accomplished via four digit dialing, do not hit the primary server. Outbound calls to the PSTN deliver the site's DID as the calling number. In other words, if a user from site two calls my cell phone, I see site two's published telephone number on my caller ID. Our dial plans are set up so that receptionists at each site still answer all outside calls. If not answered, the call fails over to an IVR. Should we ever decide to do so, we are now perfectly positioned to have all inbound calls to the district answered by one operator or IVR. "Welcome, and thank you for calling Avery County Schools." Stumbling Blocks. Problems we've faced so far have primarily surrounded Openzap and the Sangoma Wanpipe driver. FreeSWITCH developers won't mind telling you that this is an area that is currently not well "funded" and not 100% complete. There is some known issue where voice channels on the PRI get stuck in the wrong state and become unusable. We have experienced this a couple of times and have not been able to make or receive calls. Bouncing the Wanpipe driver has fixed this each time. We have also had trouble with DTMF detection across the PRI. If a user hits the IVR, it is oftentimes difficult to get it to properly recognize the digits that are being keyed in by the caller. This can be very, very frustrating to a caller that doesn't want to deal with an IVR anyway. The developers have suggested to me that this is a problem with the Sangoma's echo cancellation goofing up Openzap's ability to interpret the DTMF. The Sangoma hardware does have its own DTMF decoder and API, but the Openzap code currently does not make use of it. I have created a patch that makes use of the hardware decoder. We have been running it in production for a couple of weeks, and that does seem to have helped the problem. The problem hasn't gone away altogether. Those have been our two biggest issues, but we haven't let them hold us up. Conclusion. Of the five sites that will be on this system, one is fully functional with calls inbound and outbound from the PSTN. A second site is up and running with full outbound PSTN access. Their inbound DID is scheduled to move over to the PRI in one week. The server has been worked up for a third site, and the phones are starting to roll out. Sites four and five should come online by the end of April. Currently, I don't have numbers compiled for things like concurrent calls. At this point in my project, it is just not important. I really don't think our implementation will ever push FreeSWITCH's abilities in that regard. I base that statement primarily on other users' benchmarks, and what I've heard some are doing in carrier class environments. FreeSWITCH has made our project viable. An open source solution was the only way we could meet all of the project goals and stay within our budget. FreeSWITCH has proven to have all the features we require in a district wide phone system. It has not locked us into annual support contracts with third party vendors. I could go on with the accolades. However, I'll end this terribly lengthy post by saying that, overall, we have been very pleased with our choice to go with FreeSWITCH. The information in this email will seem very elementary to most people on this list, but having a message of this nature in hand would ha ve made me feel much more confident the first time I ever went to my supervisor to mention something called FreeSWITCH. :-) Thanks Tony, Brian, and Mike for a great product! Ben Holtsclaw Network Engineer Avery County Schools Ph: 828.733.3567 x2301 >>> On 2/18/2009 at 11:13 PM, Raul Fragoso wrote: Thanks guys, this is very useful information. Anyone else willing to share your experience ? Regards, Raul On Wed, 2009-02-18 at 16:19 -0200, Pablo Hernan Saro wrote: > Hi Raul, > > In my company (http://www.globant.com) we're using FreeSWITCH for high > quality conference services, integrated with OpenSIPS > (http://www.opensips.org) and Asterisk. Its performance is pretty > good. > > Pablo > > On Wed, Feb 18, 2009 at 4:09 PM, Henry Huang wrote: > > bandwidth.com has a service called phonebooth which is developed upon > > freeswitch. > > > > > > On Tue, Feb 17, 2009 at 4:20 PM, Raul Fragoso wrote: > >> > >> Hello FreeSWITCHERS, > >> > >> My company is currently creating a suite of applications which uses > >> FreeSWITCH as the back-end for an IP-PBX solution. We currently have a > >> prospect to have our first customer installation - a governmental > >> department. That is a tender to have an IP-PBX installation to connect > >> their four office branches, each one with about 300 users - which I am > >> sure FreeSWITCH is able to handle. Since this is an official tender, > >> it's part of their protocol to ask about real sites using the product. > >> > >> Having said that, would you mind sharing some information about your > >> experience with FreeSWITCH deployments ? > >> > >> No need to give many details, but a short summary with company name (if > >> possible), when it was deployed, server equipment, number of users, > >> number of concurrent calls, what kind of functions and services are used > >> and overall capacity of the system. > >> > >> I would really appreciate if you can share that information. And if you > >> guys agree (and explicitly manifest your agreement), I can compile the > >> information in the FreeSWITCH wiki under a "Use Cases" page so it can > >> serve as a common reference as well. > >> > >> Kind regards, > >> > >> Raul Fragoso > >> > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > > > -- > > Henry Huang > > UniC Solution - Communication Unified > > VoIP & Open Source software Consultant > > _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090223/232e3701/attachment-0002.html From msc at freeswitch.org Mon Feb 23 10:08:01 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 23 Feb 2009 10:08:01 -0800 Subject: [Freeswitch-users] mod_portaudio: Do not accept next call after Hangup In-Reply-To: <499D58CB.9080405@fh-wolfenbuettel.de> References: <499D58CB.9080405@fh-wolfenbuettel.de> Message-ID: <87f2f3b90902231008n73ad33fbvb05d7c87f5aa2a0c@mail.gmail.com> On Thu, Feb 19, 2009 at 5:04 AM, Rene Pankratz wrote: > Hello, > when hanging up a call with portaudio automatically the next call that > is incoming or held is accepted. > Is it possible to configure PA that way, that after hanging up (doesn't > matter whether caller or callee) no call is activated automatically? I > want to choose if I accept the next call or not. > > Thanks in advance > Ren? > Just following up - did this get resolved? -MC From msc at freeswitch.org Mon Feb 23 10:23:47 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 23 Feb 2009 10:23:47 -0800 Subject: [Freeswitch-users] Random problems with cepstral text to speech In-Reply-To: References: Message-ID: <87f2f3b90902231023t3578cc4dg515dc604fc2f3eae@mail.gmail.com> > Hello, > if Cepstral 4.x is the way to go does anybody know where to get the demo > version? > > BRs, > Claudio I think you'll have to contact Cepstral on this one. I've tried to find older revisions on their site and I can't find any way to get any voices prior to 5.1. -MC From oseslija at gmail.com Mon Feb 23 10:55:19 2009 From: oseslija at gmail.com (Ognjen Seslija) Date: Mon, 23 Feb 2009 19:55:19 +0100 Subject: [Freeswitch-users] Deployment information and use cases In-Reply-To: <1234916455.16581.49.camel@raul-laptop> References: <1234916455.16581.49.camel@raul-laptop> Message-ID: <4468a6770902231055o7b235245tb41529b55cfb76c4@mail.gmail.com> Hello, I run FreeSWITCH as a PBX solution for several companies, all sharing a single server in a "vritual pbx" deployment. Dialplans and user directories are all separate and handled per domains. Currently, there is about 250 phones set to use it, about 200 more will be migrated soon from Asterisk (I'm still using it as a PSTN PRI gateway). Everything is designed per domain, so it's easy to add more servers, add more sites into a company's dialplan, lcr etc. I really love FS as it saved me a lot of trouble I had with Asterisk. Ognjen (sekil) On Wed, Feb 18, 2009 at 1:20 AM, Raul Fragoso wrote: > Hello FreeSWITCHERS, > > My company is currently creating a suite of applications which uses > FreeSWITCH as the back-end for an IP-PBX solution. We currently have a > prospect to have our first customer installation - a governmental > department. That is a tender to have an IP-PBX installation to connect > their four office branches, each one with about 300 users - which I am > sure FreeSWITCH is able to handle. Since this is an official tender, > it's part of their protocol to ask about real sites using the product. > > Having said that, would you mind sharing some information about your > experience with FreeSWITCH deployments ? > > No need to give many details, but a short summary with company name (if > possible), when it was deployed, server equipment, number of users, > number of concurrent calls, what kind of functions and services are used > and overall capacity of the system. > > I would really appreciate if you can share that information. And if you > guys agree (and explicitly manifest your agreement), I can compile the > information in the FreeSWITCH wiki under a "Use Cases" page so it can > serve as a common reference as well. > > Kind regards, > > Raul Fragoso > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090223/6b46967f/attachment-0002.html From nik.middleton at noblesolutions.co.uk Mon Feb 23 10:58:28 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Mon, 23 Feb 2009 18:58:28 -0000 Subject: [Freeswitch-users] Help debuging core dump Message-ID: Hi Guys I'm having problems with seg faults about every 10 mins with call loads > 200. I've processed the core dump (http://pastebin.freeswitch.org/7436) but I'm unsure what I should be looking for. I don't see the point where the crash occurred. Can someone point me to where I should be looking? FreeSWITCH Version 1.0.trunk (12246) Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090223/9ec17c37/attachment-0002.html From anthony.minessale at gmail.com Mon Feb 23 11:38:33 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 23 Feb 2009 13:38:33 -0600 Subject: [Freeswitch-users] Help debuging core dump In-Reply-To: References: Message-ID: <191c3a030902231138l56b751acrbfa06ec2b2a2b8cf@mail.gmail.com> It looks like a file rewind operation. does the lua script use the input callback to rewind a file? It maybe be a race in some other thread can you paste a "thread apply all bt" from the same core to look at the other threads. On Mon, Feb 23, 2009 at 12:58 PM, Nik Middleton < nik.middleton at noblesolutions.co.uk> wrote: > Hi Guys > > > > I'm having problems with seg faults about every 10 mins with call loads > > 200. I've processed the core dump (http://pastebin.freeswitch.org/7436) > but I'm unsure what I should be looking for. I don't see the point where the > crash occurred. Can someone point me to where I should be looking? > > > > > > FreeSWITCH Version 1.0.trunk (12246) > > > > Regards, > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090223/e7fd79f1/attachment-0002.html From josephbajin at gmail.com Mon Feb 23 12:10:51 2009 From: josephbajin at gmail.com (Joseph Bajin) Date: Mon, 23 Feb 2009 15:10:51 -0500 Subject: [Freeswitch-users] SIP dump to DB In-Reply-To: <499ED737.3020401@post.cz> References: <499E92E4.5010503@post.cz> <191c3a030902200554s4abda94g76ba4e2975fffa98@mail.gmail.com> <03a601c99367$f92f48f0$eb8ddad0$@net> <499EC35A.1060808@post.cz> <35b355e90902200658k48ba850ele3ede371ebe96631@mail.gmail.com> <499ECBF2.4030606@post.cz> <191c3a030902200733y795626edq8eef6f06bcd70571@mail.gmail.com> <207e7a5e0902200750t2ac9f21avfb5ae88791e4efa8@mail.gmail.com> <499ED737.3020401@post.cz> Message-ID: <1dce11f20902231210u4c128b4ch491b49c75e6e58af@mail.gmail.com> Basically, you are trying to build what Empirix has with their Hammer tool. You can create an application that is basically a mix of tshark and a database feeder. You sniff with tshark and going to basically pipe it to another application that will read the pcap file, parse it, and load it into the db for you. There are plenty of modules out there that will read pcap for you. On Fri, Feb 20, 2009 at 11:15 AM, kokoska rokoska wrote: > > > > jonathan augenstine napsal(a): > > You can tcpdump and then use wireshark to graph the calls. When the > > dump is displayed in wireshark, select 'Statistics' -> VoIP Calls. You > > will see a display of all VoIP calls. Select the one you want graphed, > > or select them all and you will see REINVITE and REFER interaction as > > well as RTP streams. > > > > Thank you very much, jonathan, for your interest! > > I use ngrep+wireshark many times a day, but I'm affraid it is not > suitable for that amount of data. > > Even with few hundreds MiBs of pcap file wireshark becoms very slow and > I can't imagine how to load 50-100 GiB file with milions of calls and > try to search for one of them :-) > > And, even worse, I should "rotate" the file and, don't end with call > divided to multiple files... > > > Best regards, > > kokoska.rokoska > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090223/c21f50fa/attachment-0002.html From nik.middleton at noblesolutions.co.uk Mon Feb 23 12:14:10 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Mon, 23 Feb 2009 20:14:10 -0000 Subject: [Freeswitch-users] Help debuging core dump In-Reply-To: <191c3a030902231138l56b751acrbfa06ec2b2a2b8cf@mail.gmail.com> References: <191c3a030902231138l56b751acrbfa06ec2b2a2b8cf@mail.gmail.com> Message-ID: There's 160 threads, but I don't want to post it on the pastebin as it has real phone numbers. I'm sending as an attachment Regards, ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: 23 February 2009 19:39 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Help debuging core dump It looks like a file rewind operation. does the lua script use the input callback to rewind a file? It maybe be a race in some other thread can you paste a "thread apply all bt" from the same core to look at the other threads. On Mon, Feb 23, 2009 at 12:58 PM, Nik Middleton < nik.middleton at noblesolutions.co.uk> wrote: Hi Guys I'm having problems with seg faults about every 10 mins with call loads > 200. I've processed the core dump ( http://pastebin.freeswitch.org/7436) but I'm unsure what I should be looking for. I don't see the point where the crash occurred. Can someone point me to where I should be looking? FreeSWITCH Version 1.0.trunk (12246) Regards, _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... 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Name: threads.txt Url: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090223/47eb6d4d/attachment-0002.txt From kokoska.rokoska at post.cz Mon Feb 23 14:32:26 2009 From: kokoska.rokoska at post.cz (kokoska.rokoska) Date: Mon, 23 Feb 2009 23:32:26 +0100 Subject: [Freeswitch-users] SIP dump to DB In-Reply-To: <1dce11f20902231210u4c128b4ch491b49c75e6e58af@mail.gmail.com> References: <499E92E4.5010503@post.cz> <191c3a030902200554s4abda94g76ba4e2975fffa98@mail.gmail.com> <03a601c99367$f92f48f0$eb8ddad0$@net> <499EC35A.1060808@post.cz> <35b355e90902200658k48ba850ele3ede371ebe96631@mail.gmail.com> <499ECBF2.4030606@post.cz> <191c3a030902200733y795626edq8eef6f06bcd70571@mail.gmail.com> <207e7a5e0902200750t2ac9f21avfb5ae88791e4efa8@mail.gmail.com> <499ED737.3020401@post.cz> <1dce11f20902231210u4c128b4ch491b49c75e6e58af@mail.gmail.com> Message-ID: <49A323FA.8000802@post.cz> Joseph Bajin napsal(a): > Basically, you are trying to build what Empirix has with their Hammer tool. > Thank you very much, Joseph, for your interest! I have never heard about Empirix (I'll look at it), but what I'm trying to build is something like SER/Kamailio/OpenSIPS sip_trace module. > You can create an application that is basically a mix of tshark and a > database feeder. > You sniff with tshark and going to basically pipe it to another > application that will read the pcap file, parse it, and load it into the > db for you. There are plenty of modules out there that will read pcap > for you. > Thank you once more, Joseph, for suggestion! I think about it - it will be challenge for me to write robust and still fast enough (thousands messages per second) SIP parser + DB feeder :-) Best regards, kokoska.rokoska From swalker at SONASEARCH.com Mon Feb 23 14:47:13 2009 From: swalker at SONASEARCH.com (Stephen Walker) Date: Mon, 23 Feb 2009 14:47:13 -0800 Subject: [Freeswitch-users] FREESwitch on Windows Server 2003 Message-ID: <3B93E0500B57D04CBAE85520B750CFF04CA6CE@exchange.sonasearch.com> Hello: I have successfully loaded the Windows implementation (SVN 11602 - 02/02/09) from your site and it runs fine. I configured a Linksys SPA 2102 and have acquired dial tone and the '999X' tests work. I have not been able to establish connection with either FreeWorldDialup or Broadvoice as of yet. Which files do I need to edit and what are the proper entries to enable connection to FreeWorldDialup and Broadvoice? Example files and where they reside in the file structure would be very much appreciated. Thank you All the Best, Steve Steve Walker President SONASEARCH, INC 425/883-1984 NOTICE: The information contained in this document is intended by Sonasearch, Inc. or one of its subsidiaries for the use of the named individuals or entities to which it is addressed and may contain information that is privileged or otherwise confidential. It is not intended for transmission to, or receipt by, any individual or entity other than the named addressee (or a person authorized to deliver it to the named addressee) except as otherwise expressly permitted in this document. If you have received this document in error, please destroy it without copying or forwarding it, and notify the sender of the error by calling Sonasearch at (425) 883-1984. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090223/7d514817/attachment-0002.html From andrew at hijacked.us Mon Feb 23 16:22:08 2009 From: andrew at hijacked.us (Andrew Thompson) Date: Mon, 23 Feb 2009 19:22:08 -0500 Subject: [Freeswitch-users] mod_erlang_event compile problem In-Reply-To: <0DEC394E-4739-4055-B524-E4217593DD3C@scarlet-internet.nl> References: <20090109230650.GF5210@hijacked.us> <4A496EA6-D036-4BB5-9347-5536AD5FD0FC@scarlet-internet.nl> <20090220190811.GC29511@hijacked.us> <90DDD253-BA8D-4A79-9ACA-1560F2E18688@scarlet-internet.nl> <20090223001211.GC13957@hijacked.us> <0DEC394E-4739-4055-B524-E4217593DD3C@scarlet-internet.nl> Message-ID: <20090224002207.GF13957@hijacked.us> Leon, I think I found the problem. I shouldn't have been defaulting to binding to 127.0.0.1, instead the default should be 0.0.0.0. I've patched the module to actually bind to 0.0.0.0 correctly and made it the default in the config file. Erlang nodes by default bind to 0.0.0.0, so I decided to make mod_erlang_event follow suit. Please give that a shot and see if it fixes things. Andrew From carlos.talbot at gmail.com Mon Feb 23 18:20:20 2009 From: carlos.talbot at gmail.com (Carlos Talbot) Date: Mon, 23 Feb 2009 20:20:20 -0600 Subject: [Freeswitch-users] FREESwitch on Windows Server 2003 In-Reply-To: <3B93E0500B57D04CBAE85520B750CFF04CA6CE@exchange.sonasearch.com> References: <3B93E0500B57D04CBAE85520B750CFF04CA6CE@exchange.sonasearch.com> Message-ID: <5800526b0902231820u468908c6ia11191ccf8e37767@mail.gmail.com> On Mon, Feb 23, 2009 at 4:47 PM, Stephen Walker wrote: > > Which files do I need to edit and what are the proper entries to enable > connection to FreeWorldDialup and Broadvoice? Example files and where they > reside in the file structure would be very much appreciated. > You'll need to place a gateway configuration for Broadvoice in conf/sip_profiles/external similar to this example: http://wiki.freeswitch.org/wiki/SIP_Provider_Examples#Broadvoice The same applies to FWD. http://wiki.freeswitch.org/wiki/SIP_Provider_Examples#Free_World_Dialup_.28FWD.29 Once the gateways are configured you'll need to modify the default dial plan to recognize these gateways: http://wiki.freeswitch.org/wiki/Getting_Started_Guide#Dialing_out_via_Gatewayfor dialing out and http://wiki.freeswitch.org/wiki/Getting_Started_Guide#Receiving_an_inbound_call_from_a_Gatewayfor incoming. Most of this is actually covered here: http://wiki.freeswitch.org/wiki/Installation_Guide#Windows_quick_start regards, Carlos -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090223/9bef760f/attachment-0002.html From josephbajin at gmail.com Mon Feb 23 20:44:04 2009 From: josephbajin at gmail.com (Joseph Bajin) Date: Mon, 23 Feb 2009 23:44:04 -0500 Subject: [Freeswitch-users] SIP dump to DB In-Reply-To: <49A323FA.8000802@post.cz> References: <499E92E4.5010503@post.cz> <03a601c99367$f92f48f0$eb8ddad0$@net> <499EC35A.1060808@post.cz> <35b355e90902200658k48ba850ele3ede371ebe96631@mail.gmail.com> <499ECBF2.4030606@post.cz> <191c3a030902200733y795626edq8eef6f06bcd70571@mail.gmail.com> <207e7a5e0902200750t2ac9f21avfb5ae88791e4efa8@mail.gmail.com> <499ED737.3020401@post.cz> <1dce11f20902231210u4c128b4ch491b49c75e6e58af@mail.gmail.com> <49A323FA.8000802@post.cz> Message-ID: <1dce11f20902232044u85259f4hf369da49ce00b46b@mail.gmail.com> If you write it correctly it will work just fine. That is how most of all the other correlation engines work. Your setup is not going to be bigger than some of the large telecoms that use these systems today. On 2/23/09, kokoska.rokoska wrote: > Joseph Bajin napsal(a): >> Basically, you are trying to build what Empirix has with their Hammer >> tool. >> > > Thank you very much, Joseph, for your interest! > > I have never heard about Empirix (I'll look at it), but what I'm trying > to build is something like SER/Kamailio/OpenSIPS sip_trace module. > >> You can create an application that is basically a mix of tshark and a >> database feeder. >> You sniff with tshark and going to basically pipe it to another >> application that will read the pcap file, parse it, and load it into the >> db for you. There are plenty of modules out there that will read pcap >> for you. >> > > Thank you once more, Joseph, for suggestion! > I think about it - it will be challenge for me to write robust and still > fast enough (thousands messages per second) SIP parser + DB feeder :-) > > Best regards, > > kokoska.rokoska > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Sent from my mobile device --Joe From kokoska.rokoska at post.cz Mon Feb 23 22:13:52 2009 From: kokoska.rokoska at post.cz (kokoska.rokoska) Date: Tue, 24 Feb 2009 07:13:52 +0100 Subject: [Freeswitch-users] SIP dump to DB In-Reply-To: <1dce11f20902232044u85259f4hf369da49ce00b46b@mail.gmail.com> References: <499E92E4.5010503@post.cz> <03a601c99367$f92f48f0$eb8ddad0$@net> <499EC35A.1060808@post.cz> <35b355e90902200658k48ba850ele3ede371ebe96631@mail.gmail.com> <499ECBF2.4030606@post.cz> <191c3a030902200733y795626edq8eef6f06bcd70571@mail.gmail.com> <207e7a5e0902200750t2ac9f21avfb5ae88791e4efa8@mail.gmail.com> <499ED737.3020401@post.cz> <1dce11f20902231210u4c128b4ch491b49c75e6e58af@mail.gmail.com> <49A323FA.8000802@post.cz> <1dce11f20902232044u85259f4hf369da49ce00b46b@mail.gmail.com> Message-ID: <49A39020.3020808@post.cz> Joseph Bajin napsal(a): > If you write it correctly it will work just fine. Yes, this is challenge I have talked about :-) > That is how most of > all the other correlation engines work. I don't have enough informations but from what I heard from friendly "competitors" they are usualy log (SIP|ISUP) messages after they are parsed by their "routing" servers and not run separate tshark+parser+logger. Or they duplicate (just) SIP messages to separate machine and parse and log them there (SERlike server + sip_trace). > Your setup is not going to be > bigger than some of the large telecoms that use these systems today. > I hope so :-) Thanks once more, Joseph, for your info! Best regards, kokoska.rokoska From r.pankratz at fh-wolfenbuettel.de Mon Feb 23 23:27:02 2009 From: r.pankratz at fh-wolfenbuettel.de (Rene Pankratz) Date: Tue, 24 Feb 2009 08:27:02 +0100 Subject: [Freeswitch-users] mod_portaudio: Do not accept next call after Hangup In-Reply-To: <87f2f3b90902231008n73ad33fbvb05d7c87f5aa2a0c@mail.gmail.com> References: <499D58CB.9080405@fh-wolfenbuettel.de> <87f2f3b90902231008n73ad33fbvb05d7c87f5aa2a0c@mail.gmail.com> Message-ID: <49A3A146.8050001@fh-wolfenbuettel.de> No, unfortunately the problem still persists. Portaudio still automatically accepts/takes the next call. Ren? > On Thu, Feb 19, 2009 at 5:04 AM, Rene Pankratz > wrote: > >> Hello, >> when hanging up a call with portaudio automatically the next call that >> is incoming or held is accepted. >> Is it possible to configure PA that way, that after hanging up (doesn't >> matter whether caller or callee) no call is activated automatically? I >> want to choose if I accept the next call or not. >> >> Thanks in advance >> Ren? >> >> > Just following up - did this get resolved? > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From helmut.kuper at ewetel.de Tue Feb 24 00:33:42 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Tue, 24 Feb 2009 09:33:42 +0100 Subject: [Freeswitch-users] Patch for openzap concerning finding a free channel. Message-ID: <49A3B0E6.80408@ewetel.de> Hello, today I uploaded a little patch for openzap into trunk (r667). It marks now inbound channels as "inUse" which is conform with outbound channel handling. This should solve some problems finding a free channel in ozmod_isdn.c for inbound and outbound calls. regards Helmut From helmut.kuper at ewetel.de Tue Feb 24 01:03:29 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Tue, 24 Feb 2009 10:03:29 +0100 Subject: [Freeswitch-users] mod_openzap stops working after some calls Update In-Reply-To: <49941E24.2070002@ewetel.de> References: <498C2DC3.40701@ewetel.de> <191c3a030902060802r424f8a22tb4887dff7c6aead5@mail.gmail.com> <49907327.6010703@ewetel.de> <7CEFE4F8-7461-4DF3-ABE0-C0B818466AD0@jerris.com> <4990789B.40405@ewetel.de> <4992F295.4070809@ewetel.de> <87f2f3b90902110943q12c550e4qf8e6ecae1ecc6bec@mail.gmail.com> <49941E24.2070002@ewetel.de> Message-ID: <49A3B7E1.1080009@ewetel.de> Hello, just to keep you informed about this problem. As mentioned I added a hack to free allocated channels depending on last event time. I enhanced oz dump as well to display "last event time" and "InUse"-Flag. What I found is this: 1. InUse channel flag wasn't set for inbound calls. I fixed that as far as I understood the openzap code ;) and I tested the patch successfully for 7 days now... 2. In my setup (AVAYA as remote end for a E1) channels tend to hang in a state <> DOWN after terminating a call. Then I found TOMANYCALLS entries in FS log. I had to restart FS resp. openzap module. The hack I added is in production and works for 7 days now. No channels hanging anymore. Of course, the hack is not the final solution, but it seems to solve at least my problems in production until openzap has state timers. If the board wants, I can upload the hack as well. regards Helmut On 12.02.2009 14:03, Helmut Kuper wrote: > Hi Mike, > > at least for incoming calls this shouldn't be too brutal, cause far > end seems to know that the channel should be free otherwise it never > would allocate it. By now the hack works at least for me quite good. > Nobody from AVAYA side moaned about it, yet. But I have to wait one or > two further days to be sure ... I guess I have to talk to stkn in irc > to get an idea how long I have to use it. > > regards > helmut -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090224/bc1e80bd/attachment-0002.html From leon at scarlet-internet.nl Tue Feb 24 01:38:29 2009 From: leon at scarlet-internet.nl (Leon de Rooij) Date: Tue, 24 Feb 2009 10:38:29 +0100 Subject: [Freeswitch-users] mod_erlang_event compile problem In-Reply-To: <20090224002207.GF13957@hijacked.us> References: <20090109230650.GF5210@hijacked.us> <4A496EA6-D036-4BB5-9347-5536AD5FD0FC@scarlet-internet.nl> <20090220190811.GC29511@hijacked.us> <90DDD253-BA8D-4A79-9ACA-1560F2E18688@scarlet-internet.nl> <20090223001211.GC13957@hijacked.us> <0DEC394E-4739-4055-B524-E4217593DD3C@scarlet-internet.nl> <20090224002207.GF13957@hijacked.us> Message-ID: <4BD8A505-CC3D-45C0-9C1D-37983657DFC1@scarlet-internet.nl> Andrew, I think you're right, packets are indeed sent to 172.31.0.13 while mod_erlang_event is listening at 127.0.0.1 ! Why didn't I see that ! ;-) Will test it now and let you know how it goes.. regards, Leon On Feb 24, 2009, at 1:22 AM, Andrew Thompson wrote: > Leon, > > I think I found the problem. I shouldn't have been defaulting to > binding > to 127.0.0.1, instead the default should be 0.0.0.0. I've patched the > module to actually bind to 0.0.0.0 correctly and made it the default > in > the config file. Erlang nodes by default bind to 0.0.0.0, so I decided > to make mod_erlang_event follow suit. > > Please give that a shot and see if it fixes things. > > Andrew > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From leon at scarlet-internet.nl Tue Feb 24 01:49:24 2009 From: leon at scarlet-internet.nl (Leon de Rooij) Date: Tue, 24 Feb 2009 10:49:24 +0100 Subject: [Freeswitch-users] mod_erlang_event compile problem In-Reply-To: <20090224002207.GF13957@hijacked.us> References: <20090109230650.GF5210@hijacked.us> <4A496EA6-D036-4BB5-9347-5536AD5FD0FC@scarlet-internet.nl> <20090220190811.GC29511@hijacked.us> <90DDD253-BA8D-4A79-9ACA-1560F2E18688@scarlet-internet.nl> <20090223001211.GC13957@hijacked.us> <0DEC394E-4739-4055-B524-E4217593DD3C@scarlet-internet.nl> <20090224002207.GF13957@hijacked.us> Message-ID: <714AD975-7224-43F8-A8D2-3381379237D3@scarlet-internet.nl> Well, this works, I feel a bit stupid now :-] Now it's time to play with it.. Thanks a lot ! kind regards, Leon On Feb 24, 2009, at 1:22 AM, Andrew Thompson wrote: > Leon, > > I think I found the problem. I shouldn't have been defaulting to > binding > to 127.0.0.1, instead the default should be 0.0.0.0. I've patched the > module to actually bind to 0.0.0.0 correctly and made it the default > in > the config file. Erlang nodes by default bind to 0.0.0.0, so I decided > to make mod_erlang_event follow suit. > > Please give that a shot and see if it fixes things. > > Andrew > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From kawarod at laposte.net Tue Feb 24 06:05:53 2009 From: kawarod at laposte.net (rod) Date: Tue, 24 Feb 2009 18:05:53 +0400 Subject: [Freeswitch-users] mod_fax and sending a fax In-Reply-To: References: <49919822.3030101@laposte.net> <499C16C6.1000006@laposte.net> Message-ID: <49A3FEC1.5090300@laposte.net> Hi, the clue for sending fax is to use the originate command in the CLI: originate sofia/example/100 at 10.10.10.10 &txfax(/path_to_fax_file) this command will send the fax file via profile example to fax machine 100 reachable via 10.10.10.10 Hope this could help others :p regards, rod. Javier Aristiz?bal wrote: > Hi Rod, i just play with rx_fax and work for me. I didn't work with > tx_fax but i understand, that you need a .tiff file to send > passthrough the rx_fax. Maybe that can help you > > regards > javar > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From eric at rf.com Tue Feb 24 01:33:08 2009 From: eric at rf.com (Eric Chamberlain) Date: Tue, 24 Feb 2009 01:33:08 -0800 Subject: [Freeswitch-users] Skypiax, same skype user, multiple channels Message-ID: I was reading through the Skypiax documentation and saw the comment that it's not possible to run multiple skype clients on the same linux machine, all using the same skype user account. It's possible to run multiple skype clients with the same skype user account, as long as the skype clients are not accessing the same Skype dbpath. We use runuser to run multiple skype clients. All the clients use the same skype user, but each instance uses a different home directory, each with its own .Skype folder. In such a configuration, will Skypiax support multiple channels using the same skype username? -- Eric Chamberlain, Founder RF.com - http://RF.com/ From mike at jerris.com Tue Feb 24 06:47:27 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 24 Feb 2009 09:47:27 -0500 Subject: [Freeswitch-users] mod_portaudio: Do not accept next call after Hangup In-Reply-To: <49A3A146.8050001@fh-wolfenbuettel.de> References: <499D58CB.9080405@fh-wolfenbuettel.de> <87f2f3b90902231008n73ad33fbvb05d7c87f5aa2a0c@mail.gmail.com> <49A3A146.8050001@fh-wolfenbuettel.de> Message-ID: Please report this bug to jira.freeswitch.org. On Feb 24, 2009, at 2:27 AM, Rene Pankratz wrote: > No, unfortunately the problem still persists. Portaudio still > automatically accepts/takes the next call. > > Ren? >> On Thu, Feb 19, 2009 at 5:04 AM, Rene Pankratz >> wrote: >> >>> Hello, >>> when hanging up a call with portaudio automatically the next call >>> that >>> is incoming or held is accepted. >>> Is it possible to configure PA that way, that after hanging up >>> (doesn't >>> matter whether caller or callee) no call is activated >>> automatically? I >>> want to choose if I accept the next call or not. >>> >>> Thanks in advance >>> Ren? >>> >>> >> Just following up - did this get resolved? >> -MC >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From anthony.minessale at gmail.com Tue Feb 24 07:22:40 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 24 Feb 2009 09:22:40 -0600 Subject: [Freeswitch-users] mod_portaudio: Do not accept next call after Hangup In-Reply-To: <49A3A146.8050001@fh-wolfenbuettel.de> References: <499D58CB.9080405@fh-wolfenbuettel.de> <87f2f3b90902231008n73ad33fbvb05d7c87f5aa2a0c@mail.gmail.com> <49A3A146.8050001@fh-wolfenbuettel.de> Message-ID: <191c3a030902240722q30bae77cmd60cdea825011fb6@mail.gmail.com> What direction is the original call? Are you sure you do not have the auto_answer enabled? On Tue, Feb 24, 2009 at 1:27 AM, Rene Pankratz < r.pankratz at fh-wolfenbuettel.de> wrote: > No, unfortunately the problem still persists. Portaudio still > automatically accepts/takes the next call. > > Ren? > > On Thu, Feb 19, 2009 at 5:04 AM, Rene Pankratz > > wrote: > > > >> Hello, > >> when hanging up a call with portaudio automatically the next call that > >> is incoming or held is accepted. > >> Is it possible to configure PA that way, that after hanging up (doesn't > >> matter whether caller or callee) no call is activated automatically? I > >> want to choose if I accept the next call or not. > >> > >> Thanks in advance > >> Ren? > >> > >> > > Just following up - did this get resolved? > > -MC > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090224/007b9339/attachment-0002.html From andrew at hijacked.us Tue Feb 24 09:33:00 2009 From: andrew at hijacked.us (Andrew Thompson) Date: Tue, 24 Feb 2009 12:33:00 -0500 Subject: [Freeswitch-users] mod_erlang_event compile problem In-Reply-To: <714AD975-7224-43F8-A8D2-3381379237D3@scarlet-internet.nl> References: <20090109230650.GF5210@hijacked.us> <4A496EA6-D036-4BB5-9347-5536AD5FD0FC@scarlet-internet.nl> <20090220190811.GC29511@hijacked.us> <90DDD253-BA8D-4A79-9ACA-1560F2E18688@scarlet-internet.nl> <20090223001211.GC13957@hijacked.us> <0DEC394E-4739-4055-B524-E4217593DD3C@scarlet-internet.nl> <20090224002207.GF13957@hijacked.us> <714AD975-7224-43F8-A8D2-3381379237D3@scarlet-internet.nl> Message-ID: <20090224173259.GH13957@hijacked.us> On Tue, Feb 24, 2009 at 10:49:24AM +0100, Leon de Rooij wrote: > Well, this works, I feel a bit stupid now :-] Now it's time to play > with it.. > Nah, bad choice of defaults on my part. Defaulting to 0.0.0.0 is much more consistant and compatible. For some reason I was trying to emulate the event socket, not an erlang node. Thanks for finally making me solve the problem instead of just working around it. Andrew From kerrada2003 at yahoo.com Tue Feb 24 09:24:07 2009 From: kerrada2003 at yahoo.com (Ali Al-Rubaie) Date: Tue, 24 Feb 2009 09:24:07 -0800 (PST) Subject: [Freeswitch-users] file directory.conf.xml In-Reply-To: Message-ID: <57884.44323.qm@web33703.mail.mud.yahoo.com> Hi, The file directory.conf.xml had been mentioned in the documentation many times but there is not such file in the conf folder. Do you mean default.xml in directory folder? Thanks! --- On Tue, 2/24/09, freeswitch-users-request at lists.freeswitch.org wrote: From: freeswitch-users-request at lists.freeswitch.org Subject: Freeswitch-users Digest, Vol 32, Issue 181 To: freeswitch-users at lists.freeswitch.org Date: Tuesday, February 24, 2009, 3:34 AM Send Freeswitch-users mailing list submissions to freeswitch-users at lists.freeswitch.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.freeswitch.org/mailman/listinfo/freeswitch-users or, via email, send a message with subject or body 'help' to freeswitch-users-request at lists.freeswitch.org You can reach the person managing the list at freeswitch-users-owner at lists.freeswitch.org When replying, please edit your Subject line so it is more specific than "Re: Contents of Freeswitch-users digest..." Today's Topics: 1. Re: SIP dump to DB (kokoska.rokoska) 2. FREESwitch on Windows Server 2003 (Stephen Walker) 3. Re: mod_erlang_event compile problem (Andrew Thompson) 4. Re: FREESwitch on Windows Server 2003 (Carlos Talbot) 5. Re: SIP dump to DB (Joseph Bajin) 6. Re: SIP dump to DB (kokoska.rokoska) 7. Re: mod_portaudio: Do not accept next call after Hangup (Rene Pankratz) 8. Patch for openzap concerning finding a free channel. (Helmut Kuper) ---------------------------------------------------------------------- Message: 1 Date: Mon, 23 Feb 2009 23:32:26 +0100 From: "kokoska.rokoska" Subject: Re: [Freeswitch-users] SIP dump to DB To: freeswitch-users at lists.freeswitch.org Message-ID: <49A323FA.8000802 at post.cz> Content-Type: text/plain; charset=ISO-8859-1 Joseph Bajin napsal(a): > Basically, you are trying to build what Empirix has with their Hammer tool. > Thank you very much, Joseph, for your interest! I have never heard about Empirix (I'll look at it), but what I'm trying to build is something like SER/Kamailio/OpenSIPS sip_trace module. > You can create an application that is basically a mix of tshark and a > database feeder. > You sniff with tshark and going to basically pipe it to another > application that will read the pcap file, parse it, and load it into the > db for you. There are plenty of modules out there that will read pcap > for you. > Thank you once more, Joseph, for suggestion! I think about it - it will be challenge for me to write robust and still fast enough (thousands messages per second) SIP parser + DB feeder :-) Best regards, kokoska.rokoska ------------------------------ Message: 2 Date: Mon, 23 Feb 2009 14:47:13 -0800 From: "Stephen Walker" Subject: [Freeswitch-users] FREESwitch on Windows Server 2003 To: Message-ID: <3B93E0500B57D04CBAE85520B750CFF04CA6CE at exchange.sonasearch.com> Content-Type: text/plain; charset="us-ascii" Hello: I have successfully loaded the Windows implementation (SVN 11602 - 02/02/09) from your site and it runs fine. I configured a Linksys SPA 2102 and have acquired dial tone and the '999X' tests work. I have not been able to establish connection with either FreeWorldDialup or Broadvoice as of yet. Which files do I need to edit and what are the proper entries to enable connection to FreeWorldDialup and Broadvoice? Example files and where they reside in the file structure would be very much appreciated. Thank you All the Best, Steve Steve Walker President SONASEARCH, INC 425/883-1984 NOTICE: The information contained in this document is intended by Sonasearch, Inc. or one of its subsidiaries for the use of the named individuals or entities to which it is addressed and may contain information that is privileged or otherwise confidential. It is not intended for transmission to, or receipt by, any individual or entity other than the named addressee (or a person authorized to deliver it to the named addressee) except as otherwise expressly permitted in this document. If you have received this document in error, please destroy it without copying or forwarding it, and notify the sender of the error by calling Sonasearch at (425) 883-1984. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090223/7d514817/attachment-0001.html ------------------------------ Message: 3 Date: Mon, 23 Feb 2009 19:22:08 -0500 From: Andrew Thompson Subject: Re: [Freeswitch-users] mod_erlang_event compile problem To: freeswitch-users at lists.freeswitch.org Message-ID: <20090224002207.GF13957 at hijacked.us> Content-Type: text/plain; charset=us-ascii Leon, I think I found the problem. I shouldn't have been defaulting to binding to 127.0.0.1, instead the default should be 0.0.0.0. I've patched the module to actually bind to 0.0.0.0 correctly and made it the default in the config file. Erlang nodes by default bind to 0.0.0.0, so I decided to make mod_erlang_event follow suit. Please give that a shot and see if it fixes things. Andrew ------------------------------ Message: 4 Date: Mon, 23 Feb 2009 20:20:20 -0600 From: Carlos Talbot Subject: Re: [Freeswitch-users] FREESwitch on Windows Server 2003 To: freeswitch-users at lists.freeswitch.org Message-ID: <5800526b0902231820u468908c6ia11191ccf8e37767 at mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" On Mon, Feb 23, 2009 at 4:47 PM, Stephen Walker wrote: > > Which files do I need to edit and what are the proper entries to enable > connection to FreeWorldDialup and Broadvoice? Example files and where they > reside in the file structure would be very much appreciated. > You'll need to place a gateway configuration for Broadvoice in conf/sip_profiles/external similar to this example: http://wiki.freeswitch.org/wiki/SIP_Provider_Examples#Broadvoice The same applies to FWD. http://wiki.freeswitch.org/wiki/SIP_Provider_Examples#Free_World_Dialup_.28FWD.29 Once the gateways are configured you'll need to modify the default dial plan to recognize these gateways: http://wiki.freeswitch.org/wiki/Getting_Started_Guide#Dialing_out_via_Gatewayfor dialing out and http://wiki.freeswitch.org/wiki/Getting_Started_Guide#Receiving_an_inbound_call_from_a_Gatewayfor incoming. Most of this is actually covered here: http://wiki.freeswitch.org/wiki/Installation_Guide#Windows_quick_start regards, Carlos -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090223/9bef760f/attachment-0001.html ------------------------------ Message: 5 Date: Mon, 23 Feb 2009 23:44:04 -0500 From: Joseph Bajin Subject: Re: [Freeswitch-users] SIP dump to DB To: freeswitch-users at lists.freeswitch.org Message-ID: <1dce11f20902232044u85259f4hf369da49ce00b46b at mail.gmail.com> Content-Type: text/plain; charset=ISO-8859-1 If you write it correctly it will work just fine. That is how most of all the other correlation engines work. Your setup is not going to be bigger than some of the large telecoms that use these systems today. On 2/23/09, kokoska.rokoska wrote: > Joseph Bajin napsal(a): >> Basically, you are trying to build what Empirix has with their Hammer >> tool. >> > > Thank you very much, Joseph, for your interest! > > I have never heard about Empirix (I'll look at it), but what I'm trying > to build is something like SER/Kamailio/OpenSIPS sip_trace module. > >> You can create an application that is basically a mix of tshark and a >> database feeder. >> You sniff with tshark and going to basically pipe it to another >> application that will read the pcap file, parse it, and load it into the >> db for you. There are plenty of modules out there that will read pcap >> for you. >> > > Thank you once more, Joseph, for suggestion! > I think about it - it will be challenge for me to write robust and still > fast enough (thousands messages per second) SIP parser + DB feeder :-) > > Best regards, > > kokoska.rokoska > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Sent from my mobile device --Joe ------------------------------ Message: 6 Date: Tue, 24 Feb 2009 07:13:52 +0100 From: "kokoska.rokoska" Subject: Re: [Freeswitch-users] SIP dump to DB To: freeswitch-users at lists.freeswitch.org Message-ID: <49A39020.3020808 at post.cz> Content-Type: text/plain; charset=ISO-8859-1 Joseph Bajin napsal(a): > If you write it correctly it will work just fine. Yes, this is challenge I have talked about :-) > That is how most of > all the other correlation engines work. I don't have enough informations but from what I heard from friendly "competitors" they are usualy log (SIP|ISUP) messages after they are parsed by their "routing" servers and not run separate tshark+parser+logger. Or they duplicate (just) SIP messages to separate machine and parse and log them there (SERlike server + sip_trace). > Your setup is not going to be > bigger than some of the large telecoms that use these systems today. > I hope so :-) Thanks once more, Joseph, for your info! Best regards, kokoska.rokoska ------------------------------ Message: 7 Date: Tue, 24 Feb 2009 08:27:02 +0100 From: Rene Pankratz Subject: Re: [Freeswitch-users] mod_portaudio: Do not accept next call after Hangup To: freeswitch-users at lists.freeswitch.org Message-ID: <49A3A146.8050001 at fh-wolfenbuettel.de> Content-Type: text/plain; charset=ISO-8859-1; format=flowed No, unfortunately the problem still persists. Portaudio still automatically accepts/takes the next call. Ren? > On Thu, Feb 19, 2009 at 5:04 AM, Rene Pankratz > wrote: > >> Hello, >> when hanging up a call with portaudio automatically the next call that >> is incoming or held is accepted. >> Is it possible to configure PA that way, that after hanging up (doesn't >> matter whether caller or callee) no call is activated automatically? I >> want to choose if I accept the next call or not. >> >> Thanks in advance >> Ren? >> >> > Just following up - did this get resolved? > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > ------------------------------ Message: 8 Date: Tue, 24 Feb 2009 09:33:42 +0100 From: Helmut Kuper Subject: [Freeswitch-users] Patch for openzap concerning finding a free channel. To: freeswitch-users at lists.freeswitch.org Message-ID: <49A3B0E6.80408 at ewetel.de> Content-Type: text/plain; charset=ISO-8859-1 Hello, today I uploaded a little patch for openzap into trunk (r667). It marks now inbound channels as "inUse" which is conform with outbound channel handling. This should solve some problems finding a free channel in ozmod_isdn.c for inbound and outbound calls. regards Helmut ------------------------------ _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org End of Freeswitch-users Digest, Vol 32, Issue 181 ************************************************* -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090224/cf3cb945/attachment-0002.html From alex.gusak at gmail.com Tue Feb 24 09:28:29 2009 From: alex.gusak at gmail.com (Alex Gusak) Date: Tue, 24 Feb 2009 19:28:29 +0200 Subject: [Freeswitch-users] new ilbc lib Message-ID: Hello. After upgrade to version 1.0.3 we have a problem with the codec iLBC (I think that this is due to the transition to a new ilbc libs 1 week ago). Very poor quality for calls to the codec iLBC mode=20 (crack in the dynamic). iLBC mode=30 works well. Tested with phones and Zoiper SJPhone. After a rollback to the old version of FreeSWITCH 1.0.2 this is not a problem, iLBC works fine in both modes (mode = 20 and mode = 30). What could be the problem? -- Alex Gusak From brian at freeswitch.org Tue Feb 24 09:32:38 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 24 Feb 2009 11:32:38 -0600 Subject: [Freeswitch-users] new ilbc lib In-Reply-To: References: Message-ID: The problem comes up that the default is 30... the chances are that your phone doesn't set the mode= line so we default to 30 when this takes place. Not setting the mode= line in the FMTP usually means 30ms... which is the default. So to force this always to 30 you can allow iLBC at 30i, because if you invite to me with 20 and I 200 ok you 30.. you are to use 30 no exceptions. Most phones do not obey this rule. /b On Feb 24, 2009, at 11:28 AM, Alex Gusak wrote: > Hello. > > After upgrade to version 1.0.3 we have a problem with the codec iLBC > (I think that this is due to the transition to a new ilbc libs 1 week > ago). > Very poor quality for calls to the codec iLBC mode=20 (crack in the > dynamic). iLBC mode=30 works well. > Tested with phones and Zoiper SJPhone. > > After a rollback to the old version of FreeSWITCH 1.0.2 this is not a > problem, iLBC works fine in both modes (mode = 20 and mode = 30). > > What could be the problem? From freeswitch-users at digitaldan.com Tue Feb 24 09:49:51 2009 From: freeswitch-users at digitaldan.com (Dan) Date: Tue, 24 Feb 2009 10:49:51 -0700 (MST) Subject: [Freeswitch-users] Recording and outbound rtp Message-ID: <12581186.4501235497785860.JavaMail.daniel@osxlaptop> Hi, I have a small javascript application that accepts a call, retrieves some dtmf digits and then records the call to an icecast server. This works great. The problem I'm having is that when the call is being recorded freeswitch is no longer sending rtp packets back to the originating caller, in my case a Cisco 5300 with a bunch of T1 voice circuits in it. This makes sense, since no voice data back is being generated. Unfortunately my Cisco gear has rtp inactivity timers set up to hang up a call after 3 minutes of no incoming rtp packets, this is a global setting that cannot be configured for a single dial peer. Does anyone have a suggestion to generate rtp packets every once in a while? I tried setting comfort noise which did not seem to send anything. I could try playing a empty/short wav file every minute or so but the javascript call session.record is blocking, would a traditional javascript timer and callback to play a wav file be my best bet or is there a better approach? I'm using FreeSWITCH Version 1.0.trunk (12108M) on debian etch. Thanks! Dan- -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090224/edcd3f58/attachment-0002.html From msc at freeswitch.org Tue Feb 24 10:16:44 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 24 Feb 2009 10:16:44 -0800 Subject: [Freeswitch-users] file directory.conf.xml In-Reply-To: <57884.44323.qm@web33703.mail.mud.yahoo.com> References: <57884.44323.qm@web33703.mail.mud.yahoo.com> Message-ID: <87f2f3b90902241016w5f31675bmf9cf20d13e552650@mail.gmail.com> On Tue, Feb 24, 2009 at 9:24 AM, Ali Al-Rubaie wrote: > Hi, > > The file directory.conf.xml had been mentioned in the documentation many > times but there is not such file in the conf folder. Do you mean default.xml > in directory folder? > > Thanks! Can you tell me where you see that file name listed? It's possible that it should be "dialplan_directory.conf.xml" but I don't know for sure. I will check it out. -MC From anthony.minessale at gmail.com Tue Feb 24 11:05:33 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 24 Feb 2009 13:05:33 -0600 Subject: [Freeswitch-users] Recording and outbound rtp In-Reply-To: <12581186.4501235497785860.JavaMail.daniel@osxlaptop> References: <12581186.4501235497785860.JavaMail.daniel@osxlaptop> Message-ID: <191c3a030902241105t5d57b1abt17555c68faf16263@mail.gmail.com> is it during a bridged call? On Tue, Feb 24, 2009 at 11:49 AM, Dan wrote: > Hi, > > I have a small javascript application that accepts a call, retrieves some > dtmf digits and then records the call to an icecast server. This works > great. > > The problem I'm having is that when the call is being recorded freeswitch > is no longer sending rtp packets back to the originating caller, in my case > a Cisco 5300 with a bunch of T1 voice circuits in it. This makes sense, > since no voice data back is being generated. Unfortunately my Cisco gear > has rtp inactivity timers set up to hang up a call after 3 minutes of no > incoming rtp packets, this is a global setting that cannot be configured for > a single dial peer. Does anyone have a suggestion to generate rtp packets > every once in a while? I tried setting comfort noise which did not seem to > send anything. I could try playing a empty/short wav file every minute or > so but the javascript call session.record is blocking, would a traditional > javascript timer and callback to play a wav file be my best bet or is there > a better approach? I'm using FreeSWITCH Version 1.0.trunk (12108M) on debian > etch. > > Thanks! > Dan- > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090224/82514870/attachment-0002.html From codecomplete at free.fr Tue Feb 24 11:08:06 2009 From: codecomplete at free.fr (Fred) Date: Tue, 24 Feb 2009 20:08:06 +0100 Subject: [Freeswitch-users] Web-based forum? Message-ID: <7.0.1.0.2.20090224200706.02496ca0@fredshack.com> Hello Maybe this question has been raised before, but if not: There's so much traffic in this mailing list that I was wondering if adding a web-based forum on the site was in the works? Cheers, From mike at jerris.com Tue Feb 24 11:19:33 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 24 Feb 2009 14:19:33 -0500 Subject: [Freeswitch-users] Web-based forum? In-Reply-To: <7.0.1.0.2.20090224200706.02496ca0@fredshack.com> References: <7.0.1.0.2.20090224200706.02496ca0@fredshack.com> Message-ID: <2836579B-EA7A-4CC8-859E-3C0A439176C9@jerris.com> The web version of this list is available at: http://www.nabble.com/Freeswitch-users-f32209.html Mike On Feb 24, 2009, at 2:08 PM, Fred wrote: > Hello > > Maybe this question has been raised before, but if not: There's so > much traffic in this mailing list that I was wondering if adding a > web-based forum on the site was in the works? > > Cheers, > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From msc at freeswitch.org Tue Feb 24 11:19:44 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 24 Feb 2009 11:19:44 -0800 Subject: [Freeswitch-users] Web-based forum? In-Reply-To: <7.0.1.0.2.20090224200706.02496ca0@fredshack.com> References: <7.0.1.0.2.20090224200706.02496ca0@fredshack.com> Message-ID: <87f2f3b90902241119x5fe2f1cek57b4ee42c6b47525@mail.gmail.com> > Maybe this question has been raised before, but if not: There's so > much traffic in this mailing list that I was wondering if adding a > web-based forum on the site was in the works? We are upgrading the freeswitch.org site soon to drupal 6.9. We are considering turning on the forum feature there. No definitive decision has been made but this request has come in several times. However, we are trying to make it so that the devs don't have yet another place to have to monitor for user questions, etc. so we will need to figure out a way to make it easy to use for the experts... -MC From mszlazak at aol.com Tue Feb 24 11:36:44 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Tue, 24 Feb 2009 14:36:44 -0500 Subject: [Freeswitch-users] New build gives error message for default grammar file?? Message-ID: <8CB64CE5BC62C9B-8D0-13DC@WEBMAIL-MC11.sysops.aol.com> I'm getting this error message trying out the pizza demo in FS 1.0.3: ?"Can't open dictionary C:\Program Files\FreeSWITCH\grammar\default.dic" I didn't have this before where there was no default.dic file. Is there some place a path has to be set now? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090224/eb925b37/attachment-0002.html From brian at freeswitch.org Tue Feb 24 11:45:11 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 24 Feb 2009 13:45:11 -0600 Subject: [Freeswitch-users] New build gives error message for default grammar file?? In-Reply-To: <8CB64CE5BC62C9B-8D0-13DC@WEBMAIL-MC11.sysops.aol.com> References: <8CB64CE5BC62C9B-8D0-13DC@WEBMAIL-MC11.sysops.aol.com> Message-ID: <89CDF21E-B363-49B4-BB71-22D97F2CF89F@freeswitch.org> You're not in 1.0.3 you're in SVN trunk... The reason I know this is that wasn't changed till AFTER 1.0.3 was tagged and a new file set was used... please go into your libs dir and wipe out pocketsphinx and sphinx base.. then let it redownload them. I'll make you a new tarball of the new grammar files which are in the jsgf format. An example was added to scripts yes_no.gram ... word of warning the new pocketsphinx loads the entire dictionary on port open which on a machine of any speed should load the entire thing in 2 seconds or less... but here is the WARNING WARNING WARNING... If you happen to use words that aren't in the dictionary pocketsphinx WILL CRASH.. I am already on this with the dev's to work out a fix for this its been a long standing bug apparently in the lib. /b On Feb 24, 2009, at 1:36 PM, mszlazak at aol.com wrote: > I'm getting this error message trying out the pizza demo in FS 1.0.3: > > "Can't open dictionary C:\Program Files\FreeSWITCH\grammar > \default.dic" > > I didn't have this before where there was no default.dic file. > > Is there some place a path has to be set now? > > Thanks. > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090224/455dad39/attachment-0002.html From brian at freeswitch.org Tue Feb 24 11:46:34 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 24 Feb 2009 13:46:34 -0600 Subject: [Freeswitch-users] New build gives error message for default grammar file?? In-Reply-To: <8CB64CE5BC62C9B-8D0-13DC@WEBMAIL-MC11.sysops.aol.com> References: <8CB64CE5BC62C9B-8D0-13DC@WEBMAIL-MC11.sysops.aol.com> Message-ID: <730B80E0-6DD8-4A74-9EF2-04E1C851815A@freeswitch.org> http://www.bkw.org/pizza_gram.tar.gz /b On Feb 24, 2009, at 1:36 PM, mszlazak at aol.com wrote: > I'm getting this error message trying out the pizza demo in FS 1.0.3: > > "Can't open dictionary C:\Program Files\FreeSWITCH\grammar > \default.dic" > > I didn't have this before where there was no default.dic file. > > Is there some place a path has to be set now? > > Thanks. > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090224/6dc0d4ce/attachment-0002.html From freeswitch-users at digitaldan.com Tue Feb 24 12:02:19 2009 From: freeswitch-users at digitaldan.com (freeswitch-users at digitaldan.com) Date: Tue, 24 Feb 2009 13:02:19 -0700 (MST) Subject: [Freeswitch-users] Recording and outbound rtp In-Reply-To: <17338844.9091235504936529.JavaMail.daniel@radio> Message-ID: <611594.9131235505720885.JavaMail.daniel@radio> no, I'm matching the incoming sip call via the destination number in my public context and executing the javascript appliaction. This app directly answers the call and records it until the user hangs up. D- ----- Original Message ----- From: "Anthony Minessale" To: freeswitch-users at lists.freeswitch.org Sent: Tuesday, February 24, 2009 12:05:33 PM GMT -07:00 US/Canada Mountain Subject: Re: [Freeswitch-users] Recording and outbound rtp is it during a bridged call? On Tue, Feb 24, 2009 at 11:49 AM, Dan < freeswitch-users at digitaldan.com > wrote: Hi, I have a small javascript application that accepts a call, retrieves some dtmf digits and then records the call to an icecast server. This works great. The problem I'm having is that when the call is being recorded freeswitch is no longer sending rtp packets back to the originating caller, in my case a Cisco 5300 with a bunch of T1 voice circuits in it. This makes sense, since no voice data back is being generated. Unfortunately my Cisco gear has rtp inactivity timers set up to hang up a call after 3 minutes of no incoming rtp packets, this is a global setting that cannot be configured for a single dial peer. Does anyone have a suggestion to generate rtp packets every once in a while? I tried setting comfort noise which did not seem to send anything. I could try playing a empty/short wav file every minute or so but the javascript call session.record is blocking, would a traditional javascript timer and callback to play a wav file be my best bet or is there a better approach? I'm using FreeSWITCH Version 1.0.trunk (12108M) on debian etch. Thanks! Dan- _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/ PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090224/e8956712/attachment-0002.html From mszlazak at aol.com Tue Feb 24 12:18:54 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Tue, 24 Feb 2009 15:18:54 -0500 Subject: [Freeswitch-users] New build gives error message for default grammar file?? In-Reply-To: <89CDF21E-B363-49B4-BB71-22D97F2CF89F@freeswitch.org> References: <8CB64CE5BC62C9B-8D0-13DC@WEBMAIL-MC11.sysops.aol.com> <89CDF21E-B363-49B4-BB71-22D97F2CF89F@freeswitch.org> Message-ID: <8CB64D43FA7B70B-8D0-16EF@WEBMAIL-MC11.sysops.aol.com> Hi Brian, It sounds like I'd be better off with 1.0.3 than SVN and will waiting for the fix? But thanks for the files and info. Mark. -----Original Message----- From: Brian West To: freeswitch-users at lists.freeswitch.org Sent: Tue, 24 Feb 2009 11:45 am Subject: Re: [Freeswitch-users] New build gives error message for default grammar file?? You're not in 1.0.3 you're in SVN trunk... The reason I know this is that wasn't changed till AFTER 1.0.3 was tagged and a new file set was used... please go into your libs dir and wipe out pocketsphinx and sphinx base.. then let it redownload them. ?I'll make you a new tarball of the new grammar files which are in the jsgf format. ?An example was added to scripts yes_no.gram ... word of warning the new pocketsphinx loads the entire dictionary on port open which on a machine of any speed should load the entire thing in 2 seconds or less... but here is the WARNING WARNING WARNING... If you happen to use words that aren't in the dictionary pocketsphinx WILL CRASH.. I am already on this with the dev's to work out a fix for this its been a long standing bug apparently in the lib. /b On Feb 24, 2009, at 1:36 PM, mszlazak at aol.com wrote: I'm getting this error message trying out the pizza demo in FS 1.0.3: ?"Can't open dictionary C:\Program Files\FreeSWITCH\grammar\default.dic" I didn't have this before where there was no default.dic file. Is there some place a path has to be set now? Thanks. = _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090224/0a673569/attachment-0002.html From egghunt at gmail.com Tue Feb 24 12:40:17 2009 From: egghunt at gmail.com (Arnaldo de Moraes Pereira) Date: Tue, 24 Feb 2009 17:40:17 -0300 Subject: [Freeswitch-users] Web-based forum? In-Reply-To: <87f2f3b90902241119x5fe2f1cek57b4ee42c6b47525@mail.gmail.com> References: <7.0.1.0.2.20090224200706.02496ca0@fredshack.com> <87f2f3b90902241119x5fe2f1cek57b4ee42c6b47525@mail.gmail.com> Message-ID: On Tue, Feb 24, 2009 at 4:19 PM, Michael Collins wrote: > > Maybe this question has been raised before, but if not: There's so > > much traffic in this mailing list that I was wondering if adding a > > web-based forum on the site was in the works? > > We are upgrading the freeswitch.org site soon to drupal 6.9. We are > considering turning on the forum feature there. No definitive decision > has been made but this request has come in several times. However, we > are trying to make it so that the devs don't have yet another place to > have to monitor for user questions, etc. so we will need to figure out > a way to make it easy to use for the experts... -1 for a forum. > > > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Arnaldo M Pereira -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090224/5caa564a/attachment-0002.html From alexander at degreiff.com Tue Feb 24 13:09:50 2009 From: alexander at degreiff.com (Alexander de Greiff) Date: Tue, 24 Feb 2009 22:09:50 +0100 (CET) Subject: [Freeswitch-users] howto originate fs call from webapp (python) Message-ID: <13600111.821235509735422.JavaMail.alexanderdegreiff@AdG-Mac> hi all, i come from asterisk an i am new to freeswitch. after my with days with freeswitch i am very excited! but trying to migrate our deployment i have three challenges. one of them is: i need to call freeswitch from a webapp (e.g. python) and pass number1 and number2. i then need freeswitch to call number1. as soon as it is picked up say a short confirmaton text, call number2 and bridge the two. my first approach was to call via xml_rpc like described in the wiki but when i call like server.freeswitch.api("originate","sofia/gategay/gateway1/{number1} &bridge(sofia/gateway/gateway2/{number2})") but in this case both numbers are called in parallel and the first number to pick up gets a ringback tone until the other number picks up. how can i get the sequence described above? thanks for your help alex From freeswitch-users at lists.rupa.com Tue Feb 24 13:28:05 2009 From: freeswitch-users at lists.rupa.com (Rupa Schomaker (lists)) Date: Tue, 24 Feb 2009 15:28:05 -0600 Subject: [Freeswitch-users] howto originate fs call from webapp (python) In-Reply-To: <13600111.821235509735422.JavaMail.alexanderdegreiff@AdG-Mac> References: <13600111.821235509735422.JavaMail.alexanderdegreiff@AdG-Mac> Message-ID: <49A46665.5030904@lists.rupa.com> > my first approach was to call via xml_rpc like described in the wiki > but when i call like > > server.freeswitch.api("originate","sofia/gategay/gateway1/{number1} > &bridge(sofia/gateway/gateway2/{number2})") > > but in this case both numbers are called in parallel and the first > number to pick up gets a ringback tone until the other number picks > up. how can i get the sequence described above? > > thanks for your help alex You are probably getting early media when dialing number 1. Try : server.freeswitch.api("originate","{ignore_early_media=true}sofia/gategay/gateway1/{number1} &bridge(sofia/gateway/gateway2/{number2})") From msc at freeswitch.org Tue Feb 24 13:43:11 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 24 Feb 2009 13:43:11 -0800 Subject: [Freeswitch-users] howto originate fs call from webapp (python) In-Reply-To: <13600111.821235509735422.JavaMail.alexanderdegreiff@AdG-Mac> References: <13600111.821235509735422.JavaMail.alexanderdegreiff@AdG-Mac> Message-ID: <87f2f3b90902241343w68078c38oc54ffa4a88ad32de@mail.gmail.com> On Tue, Feb 24, 2009 at 1:09 PM, Alexander de Greiff wrote: > hi all, > > i come from asterisk an i am new to freeswitch. after my with days with freeswitch i am very excited! Welcome to FreeSWITCH! > > but trying to migrate our deployment i have three challenges. one of them is: > > i need to call freeswitch from a webapp (e.g. python) and pass number1 and number2. i then need freeswitch to call number1. as soon as it is picked up say a short confirmaton text, call number2 and bridge the two. > > my first approach was to call via xml_rpc like described in the wiki but when i call like > > ?server.freeswitch.api("originate","sofia/gategay/gateway1/{number1} &bridge(sofia/gateway/gateway2/{number2})") > > but in this case both numbers are called in parallel and the first number to pick up gets a ringback tone until the other number picks up. how can i get the sequence described above? > > thanks for your help > alex Do you have any other requirements? For example, what happens if the first bridge fails? Does your Python app need to "do anything"? Just curious. Thanks, MC From mszlazak at aol.com Tue Feb 24 22:51:51 2009 From: mszlazak at aol.com (mszlazak at aol.com) Date: Wed, 25 Feb 2009 01:51:51 -0500 Subject: [Freeswitch-users] New build gives error message for default grammar file?? In-Reply-To: <89CDF21E-B363-49B4-BB71-22D97F2CF89F@freeswitch.org> References: <8CB64CE5BC62C9B-8D0-13DC@WEBMAIL-MC11.sysops.aol.com> <89CDF21E-B363-49B4-BB71-22D97F2CF89F@freeswitch.org> Message-ID: <8CB652CABEF943F-DE0-3729@WEBMAIL-DF13.sysops.aol.com> Hey Brian, Where abouts do you keep the Window MSI 1.0.3 build that isn't in SVN trunk. Installing from the wiki installation page gets me a build with the same error. Thanks. Mark. -----Original Message----- From: Brian West To: freeswitch-users at lists.freeswitch.org Sent: Tue, 24 Feb 2009 11:45 am Subject: Re: [Freeswitch-users] New build gives error message for default grammar file?? You're not in 1.0.3 you're in SVN trunk... The reason I know this is that wasn't changed till AFTER 1.0.3 was tagged and a new file set was used... please go into your libs dir and wipe out pocketsphinx and sphinx base.. then let it redownload them. ?I'll make you a new tarball of the new grammar files which are in the jsgf format. ?An example was added to scripts yes_no.gram ... word of warning the new pocketsphinx loads the entire dictionary on port open which on a machine of any speed should load the entire thing in 2 seconds or less... but here is the WARNING WARNING WARNING... If you happen to use words that aren't in the dictionary pocketsphinx WILL CRASH.. I am already on this with the dev's to work out a fix for this its been a long standing bug apparently in the lib. /b On Feb 24, 2009, at 1:36 PM, mszlazak at aol.com wrote: I'm getting this error message trying out the pizza demo in FS 1.0.3: ?"Can't open dictionary C:\Program Files\FreeSWITCH\grammar\default.dic" I didn't have this before where there was no default.dic file. Is there some place a path has to be set now? Thanks. = _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090225/fc339b8d/attachment-0002.html From gmaruzz at celliax.org Tue Feb 24 23:49:52 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Wed, 25 Feb 2009 08:49:52 +0100 Subject: [Freeswitch-users] Skypiax (Skype Network endpoint) for FreeSWITCH In-Reply-To: <5800526b0902230826m255e0f4fmeeece95ed44e8cb4@mail.gmail.com> References: <7b197bef0902131015k456dee87x8f0f1b7059f1b49c@mail.gmail.com> <5800526b0902181853h32de5535l6c0f9d7067a69cf0@mail.gmail.com> <7b197bef0902191732i6fead849uace0ac906a9437b0@mail.gmail.com> <5800526b0902230826m255e0f4fmeeece95ed44e8cb4@mail.gmail.com> Message-ID: <7b197bef0902242349o36f153a4y583c5c76685a95e0@mail.gmail.com> On Mon, Feb 23, 2009 at 5:26 PM, Carlos Talbot wrote: > Were you planning to check in the sample skype.conf.xml into the default > FreeSWITCH conf folder? If so, just be aware the default config causes > freeswitch to hang right after a "load mod_skypiax" (if you do not have > skype running or specify a nonexistant skype user). Carlos, many thanks for reporting! I'll fix this this evening, if you have time to file a Jira for it would be wonderful. ciao for now, giovanni > > regards, > > > Carlos > On Thu, Feb 19, 2009 at 7:32 PM, Giovanni Maruzzelli > wrote: >> >> On Thu, Feb 19, 2009 at 3:53 AM, Carlos Talbot >> wrote: >> >> > One question I have, is ringback suppose to work with mod_skypiax? >> > Whenever >> > I dial a number I get a few seconds of dead air before the call is >> > answered. >> > I've tried adding ringback and transfer_ringback into the dialplan just >> > before the bridge command but no go. Am I missing something? Thanks. >> >> Carlos, >> >> ringback now works without tricks, and Skypiax is in trunk. >> >> Both remote ringing and early media are treated as remote ringing >> right now (eg: no early media, just ringing). >> >> I'll add early media support in the near future. >> >> Thanks a lot for testing and exercising skypiax, and please let me >> know any hint, suggestion, feature request, etc >> >> >> >> Sincerely, >> >> Giovanni Maruzzelli >> ========================================= >> www.celliax.org >> via Pierlombardo 9, 20135 Milano >> Italy >> gmaruzz at celliax dot org >> Cell : +39-347-2665618 >> Fax : +39-02-87390039 >> >> >> >> >> On Thu, Feb 19, 2009 at 3:53 AM, Carlos Talbot >> wrote: >> > Giovannia, >> > >> > great work on mod_skypiax. I've been testing it under Windows and it >> > sounds >> > great including PSTN calls. I plan to include it as part of the Windows >> > MSI >> > build. >> > >> > One question I have, is ringback suppose to work with mod_skypiax? >> > Whenever >> > I dial a number I get a few seconds of dead air before the call is >> > answered. >> > I've tried adding ringback and transfer_ringback into the dialplan just >> > before the bridge command but no go. Am I missing something? Thanks. >> > >> > regards, >> > >> > Carlos >> > >> > >> > >> > >> > _______________________________________________ >> > Freeswitch-users mailing list >> > Freeswitch-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From gmaruzz at celliax.org Tue Feb 24 23:55:23 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Wed, 25 Feb 2009 08:55:23 +0100 Subject: [Freeswitch-users] Skypiax, same skype user, multiple channels In-Reply-To: References: Message-ID: <7b197bef0902242355p1475be00l98e6a5e0596fe4ca@mail.gmail.com> On Tue, Feb 24, 2009 at 10:33 AM, Eric Chamberlain wrote: > I was reading through the Skypiax documentation and saw the comment > that it's not possible to run multiple skype clients on the same linux > machine, all using the same skype user account. > > It's possible to run multiple skype clients with the same skype user > account, as long as the skype clients are not accessing the same Skype > dbpath. > > We use runuser to run multiple skype clients. ?All the clients use the > same skype user, but each instance uses a different home directory, > each with its own .Skype folder. > > In such a configuration, will Skypiax support multiple channels using > the same skype username? Hi Eric, yes, definitely yes. If you give me more details I would like to integrate this use case both in the docs and in my testings. BTW: I'm about to move on your previous *very useful* suggestions and feature requests, please continue to send it :-) Sincerely, Giovanni Maruzzelli ========================================= www.celliax.org via Pierlombardo 9, 20135 Milano Italy gmaruzz at celliax dot org Cell : +39-347-2665618 Fax : +39-02-87390039 > > -- > Eric Chamberlain, Founder > RF.com - http://RF.com/ > > > > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From alexander at degreiff.com Wed Feb 25 00:30:47 2009 From: alexander at degreiff.com (Alexander de Greiff) Date: Wed, 25 Feb 2009 09:30:47 +0100 (CET) Subject: [Freeswitch-users] howto originate fs call from webapp (python) In-Reply-To: <29008165.8511235550562928.JavaMail.root@h1376493.stratoserver.net> Message-ID: <26996569.8531235550647802.JavaMail.root@h1376493.stratoserver.net> hi, oops, i must have been very tired when i wrote my first mail to the list... thanks for your replies. {ignore_early_media=true} really worked for me. i try very hard to "unlearn" asterisk. with asterisk i did not do much more with the python script, but i would like the pthon script to interact more with freeswitch like: - call number1 - say a welcome message with cepstral voice - call number2 - bridge other scenario: enter telephone number in webapp python script have fs to call number say "please enter the pin code from the website" validate dtmf code pass back to webapp: correct or not correct unfortunately just from reading the wiki i don't know how to do it in my python script. can you share your experience? thanks alex From alexander at degreiff.com Wed Feb 25 02:35:27 2009 From: alexander at degreiff.com (Alexander de Greiff) Date: Wed, 25 Feb 2009 11:35:27 +0100 (CET) Subject: [Freeswitch-users] problem speaking with cepstral voice in xml ivr menu In-Reply-To: <16711643.881235557974513.JavaMail.alexanderdegreiff@AdG-Mac.local> Message-ID: <3236119.901235558070484.JavaMail.alexanderdegreiff@AdG-Mac.local> hi all, here is my second problem trying to migrate from * to fs: i can speak with cepstral voices from my dialplan, but when i implement an ivr menu with cepstral voices like this: i get the following errors: [ERR] mod_native_file.c:68 native_file_file_open() Error opening /usr/local/freeswitch/sounds/en/us/callie/say:text to speak.GSM can you point me in the right direction? thanks alex --- freeswitch 1.0.3 build 12166 From anthony.minessale at gmail.com Wed Feb 25 06:09:02 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 25 Feb 2009 08:09:02 -0600 Subject: [Freeswitch-users] howto originate fs call from webapp (python) In-Reply-To: <26996569.8531235550647802.JavaMail.root@h1376493.stratoserver.net> References: <29008165.8511235550562928.JavaMail.root@h1376493.stratoserver.net> <26996569.8531235550647802.JavaMail.root@h1376493.stratoserver.net> Message-ID: <191c3a030902250609j777fqb4464c994dfa953@mail.gmail.com> from the freeswitch build root cd libs/esl if you have python-devel or the equiv make pymod from there if you cd python you will see a python module you can use to control freeswitch. On Wed, Feb 25, 2009 at 2:30 AM, Alexander de Greiff wrote: > hi, > > oops, i must have been very tired when i wrote my first mail to the list... > > thanks for your replies. {ignore_early_media=true} really worked for me. > > i try very hard to "unlearn" asterisk. > > with asterisk i did not do much more with the python script, but i would > like the pthon script to interact more with freeswitch like: > > - call number1 > - say a welcome message with cepstral voice > - call number2 > - bridge > > > other scenario: > > enter telephone number in webapp > python script have fs to call number > say "please enter the pin code from the website" > validate dtmf code > pass back to webapp: correct or not correct > > unfortunately just from reading the wiki i don't know how to do it in my > python script. > > can you share your experience? > > thanks > alex > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090225/cf04cd3d/attachment-0002.html From anthony.minessale at gmail.com Wed Feb 25 06:19:50 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 25 Feb 2009 08:19:50 -0600 Subject: [Freeswitch-users] Recording and outbound rtp In-Reply-To: <611594.9131235505720885.JavaMail.daniel@radio> References: <17338844.9091235504936529.JavaMail.daniel@radio> <611594.9131235505720885.JavaMail.daniel@radio> Message-ID: <191c3a030902250619i5496929ap1e0d5c0d40c2b6a@mail.gmail.com> We would have to code in a feature to purposely write silence back during a recording that does not currently exist. You could perhaps post it on the bounty section in jira. On Tue, Feb 24, 2009 at 2:02 PM, wrote: > no, I'm matching the incoming sip call via the destination number in my > public context and executing the javascript appliaction. This app directly > answers the call and records it until the user hangs up. > D- > > > ----- Original Message ----- > From: "Anthony Minessale" > To: freeswitch-users at lists.freeswitch.org > Sent: Tuesday, February 24, 2009 12:05:33 PM GMT -07:00 US/Canada Mountain > Subject: Re: [Freeswitch-users] Recording and outbound rtp > > is it during a bridged call? > > > On Tue, Feb 24, 2009 at 11:49 AM, Dan wrote: > >> Hi, >> >> I have a small javascript application that accepts a call, retrieves some >> dtmf digits and then records the call to an icecast server. This works >> great. >> >> The problem I'm having is that when the call is being recorded freeswitch >> is no longer sending rtp packets back to the originating caller, in my case >> a Cisco 5300 with a bunch of T1 voice circuits in it. This makes sense, >> since no voice data back is being generated. Unfortunately my Cisco gear >> has rtp inactivity timers set up to hang up a call after 3 minutes of no >> incoming rtp packets, this is a global setting that cannot be configured for >> a single dial peer. Does anyone have a suggestion to generate rtp packets >> every once in a while? I tried setting comfort noise which did not seem to >> send anything. I could try playing a empty/short wav file every minute or >> so but the javascript call session.record is blocking, would a traditional >> javascript timer and callback to play a wav file be my best bet or is there >> a better approach? I'm using FreeSWITCH Version 1.0.trunk (12108M) on debian >> etch. >> >> Thanks! >> Dan- >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ Freeswitch-users mailing > list Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090225/133784e9/attachment-0002.html From jforman at wcgltd.com Wed Feb 25 07:22:24 2009 From: jforman at wcgltd.com (Josh Forman) Date: Wed, 25 Feb 2009 10:22:24 -0500 Subject: [Freeswitch-users] Adding an info digit to sip from header Message-ID: <63C69E8D-3ED4-4FEC-8F21-2738A1A194DC@wcgltd.com> I'm trying to edit the sip headers to make the from field look like this: From: ;tag=gK0a00d6ea. I know that to read that data on an incoming sip message it is in $ {sip_from_params}, but how can I add the ;isup-oli=27 part on an outgoing message? Thanks Josh From brian at freeswitch.org Wed Feb 25 07:29:26 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 25 Feb 2009 09:29:26 -0600 Subject: [Freeswitch-users] Adding an info digit to sip from header In-Reply-To: <63C69E8D-3ED4-4FEC-8F21-2738A1A194DC@wcgltd.com> References: <63C69E8D-3ED4-4FEC-8F21-2738A1A194DC@wcgltd.com> Message-ID: <0E1E9F62-64C5-4B70-9C67-C7B5728DB111@freeswitch.org> You can do something like this "sofia/blah/somenumber at someip: 5060;this=rocks" /b On Feb 25, 2009, at 9:22 AM, Josh Forman wrote: > I'm trying to edit the sip headers to make the from field look like > this: > > From: ;tag=gK0a00d6ea. > > I know that to read that data on an incoming sip message it is in $ > {sip_from_params}, but how can I add the ;isup-oli=27 part on an > outgoing message? > > Thanks > > Josh -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090225/13b8feaf/attachment-0002.html From freeswitch-users at digitaldan.com Wed Feb 25 08:00:45 2009 From: freeswitch-users at digitaldan.com (Dan) Date: Wed, 25 Feb 2009 09:00:45 -0700 (MST) Subject: [Freeswitch-users] Recording and outbound rtp In-Reply-To: <191c3a030902250619i5496929ap1e0d5c0d40c2b6a@mail.gmail.com> Message-ID: <18165784.9951235577626769.JavaMail.daniel@radio> Thanks, I will look around and see if I can come up with a solution. I'll post back here and on the wiki if I find one. D- ----- Original Message ----- From: "Anthony Minessale" To: freeswitch-users at lists.freeswitch.org Sent: Wednesday, February 25, 2009 7:19:50 AM GMT -07:00 US/Canada Mountain Subject: Re: [Freeswitch-users] Recording and outbound rtp We would have to code in a feature to purposely write silence back during a recording that does not currently exist. You could perhaps post it on the bounty section in jira. On Tue, Feb 24, 2009 at 2:02 PM, < freeswitch-users at digitaldan.com > wrote: no, I'm matching the incoming sip call via the destination number in my public context and executing the javascript appliaction. This app directly answers the call and records it until the user hangs up. D- ----- Original Message ----- From: "Anthony Minessale" < anthony.minessale at gmail.com > To: freeswitch-users at lists.freeswitch.org Sent: Tuesday, February 24, 2009 12:05:33 PM GMT -07:00 US/Canada Mountain Subject: Re: [Freeswitch-users] Recording and outbound rtp is it during a bridged call? On Tue, Feb 24, 2009 at 11:49 AM, Dan < freeswitch-users at digitaldan.com > wrote: Hi, I have a small javascript application that accepts a call, retrieves some dtmf digits and then records the call to an icecast server. This works great. The problem I'm having is that when the call is being recorded freeswitch is no longer sending rtp packets back to the originating caller, in my case a Cisco 5300 with a bunch of T1 voice circuits in it. This makes sense, since no voice data back is being generated. Unfortunately my Cisco gear has rtp inactivity timers set up to hang up a call after 3 minutes of no incoming rtp packets, this is a global setting that cannot be configured for a single dial peer. Does anyone have a suggestion to generate rtp packets every once in a while? I tried setting comfort noise which did not seem to send anything. I could try playing a empty/short wav file every minute or so but the javascript call session.record is blocking, would a traditional javascript timer and callback to play a wav file be my best bet or is there a better approach? I'm using FreeSWITCH Version 1.0.trunk (12108M) on debian etch. Thanks! Dan- _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/ PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/ PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090225/4a33ace5/attachment-0002.html From jforman at wcgltd.com Wed Feb 25 08:34:08 2009 From: jforman at wcgltd.com (Josh Forman) Date: Wed, 25 Feb 2009 11:34:08 -0500 Subject: [Freeswitch-users] Adding an info digit to sip from header In-Reply-To: References: Message-ID: <3E49EAB9-0333-4E99-AC99-9BAE34CAC500@wcgltd.com> Which variable would I need to set via the dialplan to do this though? Your example looks like it would be the dialstring for the bridge application but if that works it would probably be added to the To header instead of the From, right? I can't be sure since nothing I've tried has had any affect. Between looking at the wiki and random experimenting I haven't found anything that works thus far. On Feb 25, 2009, at 11:01 AM, freeswitch-users-request at lists.freeswitch.org wrote: > Message: 3 > Date: Wed, 25 Feb 2009 09:29:26 -0600 > From: Brian West > Subject: Re: [Freeswitch-users] Adding an info digit to sip from > header > To: freeswitch-users at lists.freeswitch.org > Message-ID: <0E1E9F62-64C5-4B70-9C67-C7B5728DB111 at freeswitch.org> > Content-Type: text/plain; charset="us-ascii" > > You can do something like this "sofia/blah/somenumber at someip: > 5060;this=rocks" > > /b > > On Feb 25, 2009, at 9:22 AM, Josh Forman wrote: > >> I'm trying to edit the sip headers to make the from field look like >> this: >> >> From: ;tag=gK0a00d6ea. >> >> I know that to read that data on an incoming sip message it is in $ >> {sip_from_params}, but how can I add the ;isup-oli=27 part on an >> outgoing message? >> >> Thanks >> >> Josh From brian at freeswitch.org Wed Feb 25 08:50:17 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 25 Feb 2009 10:50:17 -0600 Subject: [Freeswitch-users] problem speaking with cepstral voice in xml ivr menu In-Reply-To: <3236119.901235558070484.JavaMail.alexanderdegreiff@AdG-Mac.local> References: <3236119.901235558070484.JavaMail.alexanderdegreiff@AdG-Mac.local> Message-ID: Alex, If you want to update to svn trunk the tts-engine and tts-voice are now valid options on the menu. They were not before (But the wiki said they were). So to cut confusion I made them work... if you do not wish to upgrade you'll need to set the tts_engine and tts_voice variables before you call the IVR application and it will work with the code you already have. I highly recommend a "make curret" ;) Committed revision 12278. /b On Feb 25, 2009, at 4:35 AM, Alexander de Greiff wrote: > hi all, > > here is my second problem trying to migrate from * to fs: > > i can speak with cepstral voices from my dialplan, but when i > implement an ivr menu with cepstral voices like this: > > greet-long="say:text to speak" > greet-short="say:main menu" > invalid-sound="say:invalid entry" > exit-sound="say:goodbye" > timeout ="10000" > max-failures="3" > tts-engine="cepstral" > tts-voice="allison" > phrase_lang="en"> > > > > > > i get the following errors: > > [ERR] mod_native_file.c:68 native_file_file_open() Error opening / > usr/local/freeswitch/sounds/en/us/callie/say:text to speak.GSM > > > can you point me in the right direction? > > thanks > alex > > > --- > freeswitch 1.0.3 build 12166 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Wed Feb 25 08:53:39 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 25 Feb 2009 10:53:39 -0600 Subject: [Freeswitch-users] Adding an info digit to sip from header In-Reply-To: <3E49EAB9-0333-4E99-AC99-9BAE34CAC500@wcgltd.com> References: <3E49EAB9-0333-4E99-AC99-9BAE34CAC500@wcgltd.com> Message-ID: It will actually add it to both places INVITE sip:1235 at conference.freeswitch.org;this=rocks SIP/2.0 Via: SIP/2.0/UDP 99.185.85.3;rport;branch=z9hG4bK0Kaa1322U42eK Max-Forwards: 69 From: "1004" ;tag=1SparjgraS69m To: I verified it does indeed add it in both places. /b On Feb 25, 2009, at 10:34 AM, Josh Forman wrote: > Which variable would I need to set via the dialplan to do this > though? Your example looks like it would be the dialstring for the > bridge application but if that works it would probably be added to the > To header instead of the From, right? I can't be sure since nothing > I've tried has had any affect. > Between looking at the wiki and random experimenting I haven't found > anything that works thus far. From msc at freeswitch.org Wed Feb 25 10:11:30 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 25 Feb 2009 10:11:30 -0800 Subject: [Freeswitch-users] howto originate fs call from webapp (python) In-Reply-To: <26996569.8531235550647802.JavaMail.root@h1376493.stratoserver.net> References: <29008165.8511235550562928.JavaMail.root@h1376493.stratoserver.net> <26996569.8531235550647802.JavaMail.root@h1376493.stratoserver.net> Message-ID: <87f2f3b90902251011l3bc1a806gdcc75a781e090ec9@mail.gmail.com> > enter telephone number in webapp > python script have fs to call number > say "please enter the pin code from the website" > validate dtmf code > pass back to webapp: correct or not correct > > unfortunately just from reading the wiki i don't know how to do it in my python script. > > can you share your experience? You definitely need to become familiar with the event socket. However, to become familiar with the event socket you need also to become familiar with some of the basic FreeSWITCH API functions, like "bgapi" and "originate" as well as what kinds of events come over the event socket. Here is some recommended reading: #1 - The reporting bugs page on the wiki. It may sound crazy, but I promise you that if you at least skim over it then it will save you time when you start having to debug things. http://wiki.freeswitch.org/wiki/Reporting_Bugs #2 - The event socket page on the wiki: http://wiki.freeswitch.org/wiki/Mod_event_socket #3 - The commands page on the wiki. Pay special attention to the "originate," "bridge," and "bgapi" commands because they will be extremely useful to you in your application: http://wiki.freeswitch.org/wiki/Mod_commands #4 - The Asterisk/FreeSWITCH Rosetta Stone wiki page. In some cases you can leverage your Asterisk knowledge. This page gives you some tips on how to do stuff in FS that you already know how to do with Asterisk: http://wiki.freeswitch.org/wiki/Rosetta_stone You have lots of reading to do! :) You will also need to start doing test phone calls. Make test calls and see how things work. Watch the debug information on the CLI to see what FS is doing with each call. It's very interesting. Join us on IRC when you have questions and want to talk in real-time. -MC (IRC: mercutioviz) From alexander at degreiff.com Wed Feb 25 10:27:34 2009 From: alexander at degreiff.com (Alexander de Greiff) Date: Wed, 25 Feb 2009 19:27:34 +0100 (CET) Subject: [Freeswitch-users] problem speaking with cepstral voice in xml ivr menu In-Reply-To: Message-ID: <7349336.921235586397898.JavaMail.alexanderdegreiff@AdG-Mac> brian, thanks for your help. i really apreciate the active support. upgrading to the current trunk solved the problem. i can hear the cepstal voices in the ivr menus now. (with the current trunk i have all sorts of other compile problems (mod_fax, python) but i will work this out following the build instructions again). i only wonder because these worked with my last version of last week. so far i am a happy camper with freeswitch. this is a different snack bracket than asterisk... kind regards alex ----- Urspr?ngliche Mail ----- Alex, If you want to update to svn trunk the tts-engine and tts-voice are now valid options on the menu. They were not before (But the wiki said they were). So to cut confusion I made them work... if you do not wish to upgrade you'll need to set the tts_engine and tts_voice variables before you call the IVR application and it will work with the code you already have. I highly recommend a "make curret" ;) Committed revision 12278. /b On Feb 25, 2009, at 4:35 AM, Alexander de Greiff wrote: > hi all, > > here is my second problem trying to migrate from * to fs: > > i can speak with cepstral voices from my dialplan, but when i > implement an ivr menu with cepstral voices like this: > > greet-long="say:text to speak" > greet-short="say:main menu" > invalid-sound="say:invalid entry" > exit-sound="say:goodbye" > timeout ="10000" > max-failures="3" > tts-engine="cepstral" > tts-voice="allison" > phrase_lang="en"> > > > > > > i get the following errors: > > [ERR] mod_native_file.c:68 native_file_file_open() Error opening / > usr/local/freeswitch/sounds/en/us/callie/say:text to speak.GSM > > > can you point me in the right direction? > > thanks > alex > > > --- > freeswitch 1.0.3 build 12166 > From msc at freeswitch.org Wed Feb 25 10:28:00 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 25 Feb 2009 10:28:00 -0800 Subject: [Freeswitch-users] Question about BLF... In-Reply-To: References: Message-ID: <87f2f3b90902251028k1079bdb8s436e81a856c6256e@mail.gmail.com> Klaus, Can you update us on where you are with this? -MC On Sun, Feb 22, 2009 at 4:04 PM, Klaus Hochlehnert wrote: > Hi, > > I'm just playing around with FreeSWITCH and I have 2 questions about BLF > (with SNOM phones): > > - When I played around with the sample dial plan I found out that BLF works > better than Asterisk, but not 100% right: > ?> When phone 1000 gets a call the BLF lamp on phone 1001 blinks and after > phone 1000 takes the call the lamp on phone 1001 is on > ?> But when phone 1000 gets a second call, takes it and hangs up the lamp > on phone 1001 turns off even if the first call is still active > ?> Is that a problem or did I do something wrong??? > > > - Second question is how can I set up BLF if I want to have my dial plan > completely in a perl script (no XML besides calling the perl script)? > > Thanks, Klaus > From msc at freeswitch.org Wed Feb 25 10:31:24 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 25 Feb 2009 10:31:24 -0800 Subject: [Freeswitch-users] problem speaking with cepstral voice in xml ivr menu In-Reply-To: <7349336.921235586397898.JavaMail.alexanderdegreiff@AdG-Mac> References: <7349336.921235586397898.JavaMail.alexanderdegreiff@AdG-Mac> Message-ID: <87f2f3b90902251031t627f3822w7fa8733cfb52f7b8@mail.gmail.com> > so far i am a happy camper with freeswitch. this is a different snack bracket than asterisk... If you don't mind telling us, where did you hear about FS and what made you decide to try it? Are you unhappy with Asterisk or are you simply looking for something a bit different? Just curious. Thanks, MC From brian at freeswitch.org Wed Feb 25 11:10:09 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 25 Feb 2009 13:10:09 -0600 Subject: [Freeswitch-users] problem speaking with cepstral voice in xml ivr menu In-Reply-To: <7349336.921235586397898.JavaMail.alexanderdegreiff@AdG-Mac> References: <7349336.921235586397898.JavaMail.alexanderdegreiff@AdG-Mac> Message-ID: Can you report these issues in jira? http://jira.freeswitch.org /b On Feb 25, 2009, at 12:27 PM, Alexander de Greiff wrote: > (with the current trunk i have all sorts of other compile problems > (mod_fax, python) but i will work this out following the build > instructions again). From rex.alex345 at yahoo.com Wed Feb 25 10:42:31 2009 From: rex.alex345 at yahoo.com (Rex Alex) Date: Wed, 25 Feb 2009 10:42:31 -0800 (PST) Subject: [Freeswitch-users] ESL Wrapper Message-ID: <558004.60211.qm@web59511.mail.ac4.yahoo.com> Hi All, I am new to freeswitch but installed(freeswitch version 1.0.3), configured and tested successfully. Now I want to do the dialling funtions through a php script. Read about Event Socket Library(ESL). How to implement the same in freeswitch. Please assist. Thanks, Rex From mrene_lists at avgs.ca Wed Feb 25 11:31:37 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Wed, 25 Feb 2009 14:31:37 -0500 Subject: [Freeswitch-users] ESL Wrapper In-Reply-To: <558004.60211.qm@web59511.mail.ac4.yahoo.com> References: <558004.60211.qm@web59511.mail.ac4.yahoo.com> Message-ID: <78A818D3-6C4F-420D-A922-751A69E7E080@avgs.ca> FreeSWITCH will listen on a socket allowing clients to send commands / receive events. ESL is a library to ease the creation of applications connecting to that socket. To install the php ESL module, cd into libs/esl and type "make phpmod" A sample php file is included in the libs/esl/php directory. Mathieu On 25-Feb-09, at 1:42 PM, Rex Alex wrote: > > Hi All, > > I am new to freeswitch but installed(freeswitch version 1.0.3), > configured and tested successfully. Now I want to do the dialling > funtions through a php script. Read about Event Socket Library(ESL). > How to implement the same in freeswitch. > > Please assist. > > Thanks, > Rex > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Wed Feb 25 11:34:59 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 25 Feb 2009 13:34:59 -0600 Subject: [Freeswitch-users] ESL Wrapper In-Reply-To: <78A818D3-6C4F-420D-A922-751A69E7E080@avgs.ca> References: <558004.60211.qm@web59511.mail.ac4.yahoo.com> <78A818D3-6C4F-420D-A922-751A69E7E080@avgs.ca> Message-ID: <6CE58813-715A-43DB-877B-638B5CE7E6E9@freeswitch.org> If he's on 1.0.3 I don't think it has php in it.. /b On Feb 25, 2009, at 1:31 PM, Mathieu Rene wrote: > FreeSWITCH will listen on a socket allowing clients to send commands / > receive events. > ESL is a library to ease the creation of applications connecting to > that socket. > > To install the php ESL module, cd into libs/esl and type "make phpmod" > > A sample php file is included in the libs/esl/php directory. > > Mathieu From jforman at wcgltd.com Wed Feb 25 11:51:00 2009 From: jforman at wcgltd.com (Josh Forman) Date: Wed, 25 Feb 2009 14:51:00 -0500 Subject: [Freeswitch-users] Adding an info digit to sip from In-Reply-To: References: Message-ID: The problem here is that what you are showing me produces: From: "1004" ;tag=1SparjgraS69m To: when what I need to output would look like this: From: "1004" ;tag=1SparjgraS69m To: with the "this=rocks" in the FROM field, not the TO field. I know that you can change parts of the from field by setting effective_caller_id_name and effective_caller_id_number, but I don't know how I would add that bit of data to the end of the SIP URI inside the < > Is there a variable that I could set or perhaps some method similar to overwriting the To header shown at http://wiki.freeswitch.org/wiki/Sofia#Modifying_the_To :_header that can be used to accomplish this? Thanks Josh On Feb 25, 2009, at 2:35 PM, freeswitch-users-request at lists.freeswitch.org wrote: >> Message: 3 >> Date: Wed, 25 Feb 2009 09:29:26 -0600 >> From: Brian West >> Subject: Re: [Freeswitch-users] Adding an info digit to sip from >> header >> To: freeswitch-users at lists.freeswitch.org >> Message-ID: <0E1E9F62-64C5-4B70-9C67-C7B5728DB111 at freeswitch.org> >> Content-Type: text/plain; charset="us-ascii" >> >> You can do something like this "sofia/blah/somenumber at someip: >> 5060;this=rocks" >> >> /b >> >> On Feb 25, 2009, at 9:22 AM, Josh Forman wrote: >> >>> I'm trying to edit the sip headers to make the from field look like >>> this: >>> >>> From: ;tag=gK0a00d6ea. >>> >>> I know that to read that data on an incoming sip message it is in $ >>> {sip_from_params}, but how can I add the ;isup-oli=27 part on an >>> outgoing message? >>> >>> Thanks >>> >>> Josh > > > > ------------------------------ > > Message: 3 > Date: Wed, 25 Feb 2009 10:53:39 -0600 > From: Brian West > Subject: Re: [Freeswitch-users] Adding an info digit to sip from > header > To: freeswitch-users at lists.freeswitch.org > Message-ID: > Content-Type: text/plain; charset=US-ASCII; format=flowed > > It will actually add it to both places > > INVITE sip:1235 at conference.freeswitch.org;this=rocks SIP/2.0 > Via: SIP/2.0/UDP 99.185.85.3;rport;branch=z9hG4bK0Kaa1322U42eK > Max-Forwards: 69 > From: "1004" ;tag=1SparjgraS69m > To: > > I verified it does indeed add it in both places. > > /b > > > > On Feb 25, 2009, at 10:34 AM, Josh Forman wrote: > >> Which variable would I need to set via the dialplan to do this >> though? Your example looks like it would be the dialstring for the >> bridge application but if that works it would probably be added to >> the >> To header instead of the From, right? I can't be sure since nothing >> I've tried has had any affect. >> Between looking at the wiki and random experimenting I haven't found >> anything that works thus far. > From alexander at degreiff.com Wed Feb 25 12:12:03 2009 From: alexander at degreiff.com (Alexander de Greiff) Date: Wed, 25 Feb 2009 21:12:03 +0100 (CET) Subject: [Freeswitch-users] problem speaking with cepstral voice in xml ivr menu In-Reply-To: <1262470.981235592644063.JavaMail.alexanderdegreiff@AdG-Mac> Message-ID: <9100461.1001235592664872.JavaMail.alexanderdegreiff@AdG-Mac> michael, i googled for "asterisk alternative" and voila... the trigger was that every now and then i renew servers in the infrastructure and the one with asterisk was overdue. i wasn't really unhappy with asterisk, but these things bothered me (maybe i am not up to date): - dialplan gets messy - no conferences without hardware (rented remote server!) - ivr with cepstral voices: sometimes get hickups so far i like the fs approach very much. stable sip channels, no hickups with voices. kind regards alex ----- Urspr?ngliche Mail ----- Von: "Michael Collins" An: freeswitch-users at lists.freeswitch.org Gesendet: Mittwoch, 25. Februar 2009 19:31:24 GMT +01:00 Amsterdam/Berlin/Bern/Rom/Stockholm/Wien Betreff: Re: [Freeswitch-users] problem speaking with cepstral voice in xml ivr menu > so far i am a happy camper with freeswitch. this is a different snack bracket than asterisk... If you don't mind telling us, where did you hear about FS and what made you decide to try it? Are you unhappy with Asterisk or are you simply looking for something a bit different? Just curious. Thanks, MC _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From sprice at gmail.com Wed Feb 25 12:13:53 2009 From: sprice at gmail.com (SP) Date: Wed, 25 Feb 2009 14:13:53 -0600 Subject: [Freeswitch-users] Adding an info digit to sip from In-Reply-To: References: Message-ID: <7e2ac3270902251213j39bd8d9bme98172fe44305303@mail.gmail.com> On Wed, Feb 25, 2009 at 13:51, Josh Forman wrote: > The problem here is that what you are showing me produces: > > From: "1004" ;tag=1SparjgraS69m > To: > > when what I need to output would look like this: > > From: "1004" ;tag=1SparjgraS69m > To: > > with the "this=rocks" in the FROM field, not the TO field. > I know that you can change parts of the from field by setting > effective_caller_id_name and effective_caller_id_number, but I don't > know how I would add that bit of data to the end of the SIP URI inside > the < > > Is there a variable that I could set or perhaps some method similar to > overwriting the To header shown at http://wiki.freeswitch.org/wiki/Sofia#Modifying_the_To > :_header that can be used to accomplish this? > > Thanks > Josh > > On Feb 25, 2009, at 2:35 PM, freeswitch-users-request at lists.freeswitch.org > ?wrote: > >>> Message: 3 >>> Date: Wed, 25 Feb 2009 09:29:26 -0600 >>> From: Brian West >>> Subject: Re: [Freeswitch-users] Adding an info digit to sip from >>> ? ? ? header >>> To: freeswitch-users at lists.freeswitch.org >>> Message-ID: <0E1E9F62-64C5-4B70-9C67-C7B5728DB111 at freeswitch.org> >>> Content-Type: text/plain; charset="us-ascii" >>> >>> You can do something like this "sofia/blah/somenumber at someip: >>> 5060;this=rocks" >>> >>> /b >>> >>> On Feb 25, 2009, at 9:22 AM, Josh Forman wrote: >>> >>>> I'm trying to edit the sip headers to make the from field look like >>>> this: >>>> >>>> From: ;tag=gK0a00d6ea. >>>> >>>> I know that to read that data on an incoming sip message it is in $ >>>> {sip_from_params}, but how can I add the ;isup-oli=27 part on an >>>> outgoing message? >>>> >>>> Thanks >>>> >>>> Josh >> >> >> >> ------------------------------ >> >> Message: 3 >> Date: Wed, 25 Feb 2009 10:53:39 -0600 >> From: Brian West >> Subject: Re: [Freeswitch-users] Adding an info digit to sip from >> ? ? ? ?header >> To: freeswitch-users at lists.freeswitch.org >> Message-ID: >> Content-Type: text/plain; charset=US-ASCII; format=flowed >> >> It will actually add it to both places >> >> INVITE sip:1235 at conference.freeswitch.org;this=rocks SIP/2.0 >> Via: SIP/2.0/UDP 99.185.85.3;rport;branch=z9hG4bK0Kaa1322U42eK >> Max-Forwards: 69 >> From: "1004" ;tag=1SparjgraS69m >> To: >> >> I verified it does indeed add it in both places. >> >> /b >> >> >> >> On Feb 25, 2009, at 10:34 AM, Josh Forman wrote: >> >>> Which variable would I need to set via the dialplan to do this >>> though? ?Your example looks like it would be the dialstring for the >>> bridge application but if that works it would probably be added to >>> the >>> To header instead of the From, right? ?I can't be sure since nothing >>> I've tried has had any affect. >>> Between looking at the wiki and random experimenting I haven't found >>> anything that works thus far. >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Shannon From brian at freeswitch.org Wed Feb 25 12:27:12 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 25 Feb 2009 14:27:12 -0600 Subject: [Freeswitch-users] Adding an info digit to sip from In-Reply-To: <7e2ac3270902251213j39bd8d9bme98172fe44305303@mail.gmail.com> References: <7e2ac3270902251213j39bd8d9bme98172fe44305303@mail.gmail.com> Message-ID: SP, That won't go into the from. You can't add params to the from unless you have svn rev 12287 or higher. I added the ability to set "sip_invite_params, sip_invite_to_params, sip_invite_from_params" to sofia_glue.c, I added two lines and changed two lines to make this possible. So to be clear: sip_invite_params will set params on the request URI, sip_invite_to_params will set params on the to URI, sip_invite_from_params will set params on the from URI. (someone wiki this) http://fisheye.freeswitch.org/browse/FreeSWITCH/src/mod/endpoints/mod_sofia/sofia_glue.c?r1=12235&r2=12287 You brought up a good point that it wasn't possible but when I looked at the code it was fairly simple to add support for it so I did. Please check out that Donate button on the home page! ;) /b PS: Its MikeJ's birthday today! On Feb 25, 2009, at 2:13 PM, SP wrote: > data="sip_invite_domain=some.domain;this=rocks"/> > From brian at freeswitch.org Wed Feb 25 12:34:14 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 25 Feb 2009 14:34:14 -0600 Subject: [Freeswitch-users] Adding an info digit to sip from In-Reply-To: <7e2ac3270902251213j39bd8d9bme98172fe44305303@mail.gmail.com> References: <7e2ac3270902251213j39bd8d9bme98172fe44305303@mail.gmail.com> Message-ID: <5E93A503-391B-4B4D-B5D1-27CA1A147A33@freeswitch.org> I also realized I broke backwards compatibility for anyone using sip_invite_params so I corrected that in rev 12288 /b On Feb 25, 2009, at 2:13 PM, SP wrote: > data="sip_invite_domain=some.domain;this=rocks"/> From kristian.kielhofner at gmail.com Wed Feb 25 13:00:55 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Wed, 25 Feb 2009 16:00:55 -0500 Subject: [Freeswitch-users] SheevaPlug Development Kit Message-ID: <2d9149cd0902251300y336e5e35hb2b9a1f9d30d6f3f@mail.gmail.com> Hello everyone, I just ordered one of these: http://www.marvell.com/products/embedded_processors/developer/kirkwood/sheevaplug.jsp Just over $110 with shipping but they are expecting the price to come down quite a bit (perhaps as low as $50): - 1.2Ghz ARM5 - 512MB RAM - Multiple flash storage options - Gigabit ethernet - USB 2.0 - 5 watt power usage They probably won't be shipping until late March but I thought I'd get my order in early. Who knows how practical it will be but needless to say I'm going to get FreeSWITCH to run on it! -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com From brian at freeswitch.org Wed Feb 25 13:07:44 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 25 Feb 2009 15:07:44 -0600 Subject: [Freeswitch-users] SheevaPlug Development Kit In-Reply-To: <2d9149cd0902251300y336e5e35hb2b9a1f9d30d6f3f@mail.gmail.com> References: <2d9149cd0902251300y336e5e35hb2b9a1f9d30d6f3f@mail.gmail.com> Message-ID: I seen that yesterday... looks interesting. /b On Feb 25, 2009, at 3:00 PM, Kristian Kielhofner wrote: > Hello everyone, > > I just ordered one of these: > > http://www.marvell.com/products/embedded_processors/developer/kirkwood/sheevaplug.jsp > > Just over $110 with shipping but they are expecting the price to > come down quite a bit (perhaps as low as $50): > > - 1.2Ghz ARM5 > - 512MB RAM > - Multiple flash storage options > - Gigabit ethernet > - USB 2.0 > - 5 watt power usage > > They probably won't be shipping until late March but I thought I'd > get my order in early. > > Who knows how practical it will be but needless to say I'm going to > get FreeSWITCH to run on it! > > -- > Kristian Kielhofner > http://blog.krisk.org > http://www.submityoursip.com > http://www.astlinux.org > http://www.star2star.com > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From erik at erikwickstrom.com Wed Feb 25 13:11:23 2009 From: erik at erikwickstrom.com (Erik Wickstrom) Date: Wed, 25 Feb 2009 13:11:23 -0800 Subject: [Freeswitch-users] Thin Client VOIP setup? Message-ID: <3d381e170902251311i3d3a4205j117d472228c30219@mail.gmail.com> Hi, I've deployed Freeswitch as our phone system at work. We now want to use our new phonesystem in a phone room with thin clients (Terminal Server, possibly LTSP) for each agent. Ideally, we'd like to use x-lite or another softphone for each agent. The desired workflow for the agents is as follows: 1) A web based CRM with click to dial. (and customer data card etc) 2) Agent clicks dial button and is connected to customer 3) Interact with CRM... >From what I've read so far, there are some challenges that need to be overcome in deploying softphones over thin clients. Has anyone here had any success in setting up a system like this? I'm I asking for trouble trying to use softphones with thin clients (should I just use hardware phones? Do they support click to dial?) Thanks! Erik -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090225/a80bad4b/attachment-0002.html From msc at freeswitch.org Wed Feb 25 13:42:14 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 25 Feb 2009 13:42:14 -0800 Subject: [Freeswitch-users] ESL Wrapper In-Reply-To: <6CE58813-715A-43DB-877B-638B5CE7E6E9@freeswitch.org> References: <558004.60211.qm@web59511.mail.ac4.yahoo.com> <78A818D3-6C4F-420D-A922-751A69E7E080@avgs.ca> <6CE58813-715A-43DB-877B-638B5CE7E6E9@freeswitch.org> Message-ID: <87f2f3b90902251342q1e393127ha7fbdbebf6d83dac@mail.gmail.com> On Wed, Feb 25, 2009 at 11:34 AM, Brian West wrote: > If he's on 1.0.3 I don't think it has php in it.. Can't he do the whole bootstrap process? svn up && ./bootstrap.sh && ./configure && make install And then do Mathieu's suggestion? -MC From mashudiflexi at telkom.co.id Wed Feb 25 20:21:37 2009 From: mashudiflexi at telkom.co.id (mashudi) Date: Thu, 26 Feb 2009 11:21:37 +0700 Subject: [Freeswitch-users] Session Timer Message-ID: <49A618D1.2070900@telkom.co.id> Hi Folks, in case of FreeSwitch sip message response for UPDATE message wih SIP/2.0 200 OK, how to change the session timer value from 120 to 300 ? Session-Expires: 120;refresher=uac. thank's for help and suggestion. mashudi ***************************************** Sekarang Gratis Nelpon SLJJ Flexi diperluas ke Yogya ***************************************** From mrene_lists at avgs.ca Wed Feb 25 20:10:23 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Wed, 25 Feb 2009 23:10:23 -0500 Subject: [Freeswitch-users] Session Timer In-Reply-To: <49A618D1.2070900@telkom.co.id> References: <49A618D1.2070900@telkom.co.id> Message-ID: <21585264-712C-4673-BD91-43D63435330A@avgs.ca> In the sip profile: Math On 25-Feb-09, at 11:21 PM, mashudi wrote: > Hi Folks, > > in case of FreeSwitch sip message response for UPDATE message wih > SIP/2.0 200 OK, how to change the session timer value from 120 to > 300 ? > > Session-Expires: 120;refresher=uac. > > thank's for help and suggestion. > > mashudi > > > ***************************************** > Sekarang Gratis Nelpon SLJJ Flexi diperluas ke > Yogya > ***************************************** > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mashudiflexi at telkom.co.id Wed Feb 25 20:43:28 2009 From: mashudiflexi at telkom.co.id (mashudi) Date: Thu, 26 Feb 2009 11:43:28 +0700 Subject: [Freeswitch-users] Session Timer In-Reply-To: <21585264-712C-4673-BD91-43D63435330A@avgs.ca> References: <49A618D1.2070900@telkom.co.id> <21585264-712C-4673-BD91-43D63435330A@avgs.ca> Message-ID: <49A61DF0.8070204@telkom.co.id> Dear Mathieu Rene, Thanks for you response. I already change the value in the sip profile to 300 as prerquisite by our external gateway, for responds to INVITE message it's work, but no for response to UPDATE message, the session-timer still use default value, namely 120. Mathieu Rene wrote: > In the sip profile: > > > > Math > > On 25-Feb-09, at 11:21 PM, mashudi wrote: > > >> Hi Folks, >> >> in case of FreeSwitch sip message response for UPDATE message wih >> SIP/2.0 200 OK, how to change the session timer value from 120 to >> 300 ? >> >> Session-Expires: 120;refresher=uac. >> >> thank's for help and suggestion. >> >> mashudi >> >> >> ***************************************** >> Sekarang Gratis Nelpon SLJJ Flexi diperluas ke >> Yogya >> ***************************************** >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > ***************************************** > Sekarang Gratis Nelpon SLJJ Flexi diperluas ke > Yogya > ***************************************** > > ***************************************** Sekarang Gratis Nelpon SLJJ Flexi diperluas ke Yogya ***************************************** From rex.alex345 at yahoo.com Thu Feb 26 03:25:34 2009 From: rex.alex345 at yahoo.com (Rex_Alex) Date: Thu, 26 Feb 2009 03:25:34 -0800 (PST) Subject: [Freeswitch-users] ESL Wrapper In-Reply-To: <87f2f3b90902251342q1e393127ha7fbdbebf6d83dac@mail.gmail.com> References: <558004.60211.qm@web59511.mail.ac4.yahoo.com> <78A818D3-6C4F-420D-A922-751A69E7E080@avgs.ca> <6CE58813-715A-43DB-877B-638B5CE7E6E9@freeswitch.org> <87f2f3b90902251342q1e393127ha7fbdbebf6d83dac@mail.gmail.com> Message-ID: <1235647534150-2389093.post@n2.nabble.com> Hi All, I tried svn up && ./bootstrap.sh && ./configure && make install and did Mathieu's suggestion but getting error as below, [root at server esl]# make phpmod make MYLIB="../libesl.a" SOLINK="-shared -Xlinker -x" CFLAGS="-I/root/freeswitch-1.0.3/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2 -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes" CXXFLAGS="-I/root/freeswitch-1.0.3/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC" CXX_CFLAGS="" -C php make[1]: php-config: Command not found make[1]: Entering directory `/root/freeswitch-1.0.3/libs/esl/php' g++ -I/root/freeswitch-1.0.3/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -c esl_wrap.cpp -o esl_wrap.o esl_wrap.cpp:717:18: error: zend.h: No such file or directory esl_wrap.cpp:718:22: error: zend_API.h: No such file or directory esl_wrap.cpp:719:17: error: php.h: No such file or directory esl_wrap.cpp:973:21: error: php_ini.h: No such file or directory esl_wrap.cpp:974:31: error: ext/standard/info.h: No such file or directory esl_wrap.cpp:767: error: ?E_ERROR? was not declared in this scope esl_wrap.cpp:788: error: ISO C++ forbids declaration of ?ZEND_RSRC_DTOR_FUNC? with no type esl_wrap.cpp:788: error: ?SWIG_landfill? was not declared in this scope esl_wrap.cpp:788: error: expected ?,? or ?;? before ?{? token esl_wrap.cpp:793: error: variable or field ?SWIG_ZTS_SetPointerZval? declared void esl_wrap.cpp:793: error: ?zval? was not declared in this scope esl_wrap.cpp:793: error: ?z? was not declared in this scope esl_wrap.cpp:793: error: expected primary-expression before ?void? esl_wrap.cpp:793: error: expected primary-expression before ?*? token esl_wrap.cpp:793: error: ?type? was not declared in this scope esl_wrap.cpp:793: error: expected primary-expression before ?int? esl_wrap.cpp:793: error: initializer expression list treated as compound expression esl_wrap.cpp:793: error: expected ?,? or ?;? before ?{? token make[1]: *** [esl_wrap.o] Error 1 make[1]: Leaving directory `/root/freeswitch-1.0.3/libs/esl/php' make: *** [phpmod] Error 2 [root at server esl]# Please tell me where am i wrong? Thanks, Rex mercutioviz wrote: > > On Wed, Feb 25, 2009 at 11:34 AM, Brian West wrote: >> If he's on 1.0.3 I don't think it has php in it.. > > Can't he do the whole bootstrap process? > svn up && ./bootstrap.sh && ./configure && make install > > And then do Mathieu's suggestion? > > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://n2.nabble.com/ESL-Wrapper-tp2385651p2389093.html Sent from the freeswitch-users mailing list archive at Nabble.com. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090226/d53b1775/attachment-0002.html From yudha2008 at gmail.com Thu Feb 26 03:45:06 2009 From: yudha2008 at gmail.com (Baskar) Date: Thu, 26 Feb 2009 17:15:06 +0530 Subject: [Freeswitch-users] Cant get Disposition status in Javascript Message-ID: Hi, I am using javascript to store uuid, phone_no, endpoint_disposition and hangup cause in my MYSQL. I can get the session UUID , Phone_no, endpoint disposition but i cant get the originate disposition. Javascript : session.setVariable("session.uuid", "ses_uuid: " + session.uuid); session.setVariable("phone", "phone_no: " +argv[0]); result = session.getVariable("endpoint_disposition") hangup = session.getVariable("originate_disposition") OUTPUT: S_UUID PHONE_NO RESULT HANGUP_STATE f579cb15-5145-4eb1-a080-03b9e53b90f739841799874 39841799874 ANSWER for "originate_disposition" i did not get any value stored in the Table. So how can get the originate_disposition ???? -- Warm Regards, N.Baskar -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090226/8505d026/attachment-0002.html From yudha2008 at gmail.com Thu Feb 26 04:44:50 2009 From: yudha2008 at gmail.com (Baskar) Date: Thu, 26 Feb 2009 18:14:50 +0530 Subject: [Freeswitch-users] Cant get Disposition status in Javascript In-Reply-To: References: Message-ID: Hi, One thing i forget to tell i can able to get this Disconnection causeand Disconnection code in Freeswitch console but when i set in the variable i did not get the cause or cause code in javascript. I use these line in Javascript i get the output in the Freeswitch console. console_log("notice", "Disconnect cause: " + session.cause + "\n"); console_log("notice", "Disconnect cause: " + session.causecode + "\n"); OUTPUT: (For the Above line) 2009-02-26 18:08:57 [NOTICE] odbc1.js:1 console_log() Disconnect cause: NORMAL_CLEARING 2009-02-26 18:08:57 [NOTICE] odbc1.js:1 console_log() Disconnect cause: 16 But same session cause and code if i set in the variable i did not get output session.setVariable("session.causecode", "discause: " + session.causecode+ "\n); session.setVariable("notice", "Disconnect cause: " + session.cause + "\n"); OUTPUT: variable_session.causecode: [discause: 0] variable_notice: [Disconnect cause: NONE] Correct me where i am wrong how can i get the disconnection cause in variable. Please help to solve the problem. -- Warm Regards, N.Baskar -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090226/5764d9c5/attachment-0002.html From mrene_lists at avgs.ca Thu Feb 26 05:49:31 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Thu, 26 Feb 2009 08:49:31 -0500 Subject: [Freeswitch-users] ESL Wrapper In-Reply-To: <1235647534150-2389093.post@n2.nabble.com> References: <558004.60211.qm@web59511.mail.ac4.yahoo.com> <78A818D3-6C4F-420D-A922-751A69E7E080@avgs.ca> <6CE58813-715A-43DB-877B-638B5CE7E6E9@freeswitch.org> <87f2f3b90902251342q1e393127ha7fbdbebf6d83dac@mail.gmail.com> <1235647534150-2389093.post@n2.nabble.com> Message-ID: You need your distro's php dev pakage. On 26-Feb-09, at 6:25 AM, Rex_Alex wrote: > Hi All, I tried svn up && ./bootstrap.sh && ./configure && make > install and did Mathieu's suggestion but getting error as below, > [root at server esl]# make phpmod make MYLIB="../libesl.a" SOLINK="- > shared -Xlinker -x" CFLAGS="-I/root/freeswitch-1.0.3/libs/esl/src/ > include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2 > -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings -Wstrict- > prototypes -Wmissing-prototypes" CXXFLAGS="-I/root/freeswitch-1.0.3/ > libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/ > src/ -fPIC" CXX_CFLAGS="" -C php make[1]: php-config: Command not > found make[1]: Entering directory `/root/freeswitch-1.0.3/libs/esl/ > php' g++ -I/root/freeswitch-1.0.3/libs/esl/src/include - > DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -c > esl_wrap.cpp -o esl_wrap.o esl_wrap.cpp:717:18: error: zend.h: No > such file or directory esl_wrap.cpp:718:22: error: zend_API.h: No > such file or directory esl_wrap.cpp:719:17: error: php.h: No such > file or directory esl_wrap.cpp:973:21: error: php_ini.h: No such > file or directory esl_wrap.cpp:974:31: error: ext/standard/info.h: > No such file or directory esl_wrap.cpp:767: error: ?E_ERROR? was not > declared in this scope esl_wrap.cpp:788: error: ISO C++ forbids > declaration of ?ZEND_RSRC_DTOR_FUNC? with no type esl_wrap.cpp:788: > error: ?SWIG_landfill? was not declared in this scope esl_wrap.cpp: > 788: error: expected ?,? or ?;? before ?{? token esl_wrap.cpp:793: > error: variable or field ?SWIG_ZTS_SetPointerZval? declared void > esl_wrap.cpp:793: error: ?zval? was not declared in this scope > esl_wrap.cpp:793: error: ?z? was not declared in this scope > esl_wrap.cpp:793: error: expected primary-expression before ?void? > esl_wrap.cpp:793: error: expected primary-expression before ?*? > token esl_wrap.cpp:793: error: ?type? was not declared in this scope > esl_wrap.cpp:793: error: expected primary-expression before ?int? > esl_wrap.cpp:793: error: initializer expression list treated as > compound expression esl_wrap.cpp:793: error: expected ?,? or ?;? > before ?{? token make[1]: *** [esl_wrap.o] Error 1 make[1]: Leaving > directory `/root/freeswitch-1.0.3/libs/esl/php' make: *** [phpmod] > Error 2 [root at server esl]# Please tell me where am i wrong? Thanks, > Rex > mercutioviz wrote: > On Wed, Feb 25, 2009 at 11:34 AM, Brian West wrote: > If he's on > 1.0.3 I don't think it has php in it.. Can't he do the whole > bootstrap process? svn up && ./bootstrap.sh && ./configure && make > install And then do Mathieu's suggestion? -MC > _______________________________________________ Freeswitch-users > mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > View this message in context: Re: ESL Wrapper > Sent from the freeswitch-users mailing list archive at Nabble.com. > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090226/8003a394/attachment-0002.html From anthony.minessale at gmail.com Thu Feb 26 06:11:55 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 26 Feb 2009 08:11:55 -0600 Subject: [Freeswitch-users] Cant get Disposition status in Javascript In-Reply-To: References: Message-ID: <191c3a030902260611p5ee885e1ud422b8c0bcd17502@mail.gmail.com> both of your set lines are wrong: session.cause and session.causecode are attributes session.setVariable("session.causecode", "discause: " + session.causecode+ "\n); session.setVariable("notice", "Disconnect cause: " + session.cause + "\n"); session.setVariable("cause_code", session.causecode); session.setVariable("cause_name", session.cause); would be more appropriate. Also originate_disposition is only on outgoing legs. Doing this sort of thing in the same script is not a good plan. You should really be doing it in the CDR engine where you can get records for both legs of the call in a relaxed environment. On Thu, Feb 26, 2009 at 6:44 AM, Baskar wrote: > Hi, > One thing i forget to tell i can able to get this Disconnection causeand Disconnection > code in Freeswitch console but when i set in the variable i did not get > the cause or cause code in javascript. > > I use these line in Javascript i get the output in the Freeswitch console. > > console_log("notice", "Disconnect cause: " + session.cause + "\n"); > console_log("notice", "Disconnect cause: " + session.causecode + "\n"); > > OUTPUT: (For the Above line) > > 2009-02-26 18:08:57 [NOTICE] odbc1.js:1 console_log() Disconnect cause: > NORMAL_CLEARING > 2009-02-26 18:08:57 [NOTICE] odbc1.js:1 console_log() Disconnect cause: 16 > > But same session cause and code if i set in the variable i did not get > output > > session.setVariable("session.causecode", "discause: " + session.causecode+ > "\n); > session.setVariable("notice", "Disconnect cause: " + session.cause + "\n"); > > OUTPUT: > > variable_session.causecode: [discause: 0] > variable_notice: [Disconnect cause: NONE] > > Correct me where i am wrong how can i get the disconnection cause in > variable. > > Please help to solve the problem. > > > > -- > Warm Regards, > N.Baskar > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090226/75a71d97/attachment-0002.html From alexander at degreiff.com Thu Feb 26 06:56:07 2009 From: alexander at degreiff.com (Alexander de Greiff) Date: Thu, 26 Feb 2009 15:56:07 +0100 (CET) Subject: [Freeswitch-users] dialplan condition regex question Message-ID: <1520981.41235660110667.JavaMail.alexanderdegreiff@AdG-Mac.local> hi all, i am dialing the number 123456789 (example) reaching fs via inbound sip gateway and hitting following dialplan: ... ... via info i can see that the variable my_dialed_extension is populated ok with 789 but somehow the second condition is not met. when i change that to match (.*) the actions gets executed and the my_dialed_extension inside is correct. any suggestions? kind regards alex From yudha2008 at gmail.com Thu Feb 26 06:55:49 2009 From: yudha2008 at gmail.com (Baskar) Date: Thu, 26 Feb 2009 20:25:49 +0530 Subject: [Freeswitch-users] Cant get Disposition status in Javascript In-Reply-To: <191c3a030902260611p5ee885e1ud422b8c0bcd17502@mail.gmail.com> References: <191c3a030902260611p5ee885e1ud422b8c0bcd17502@mail.gmail.com> Message-ID: Hi Anthony Minessale, I have added these lines in my javascript with your *guidance. *But still i did not get any status like busy , no answer, etc . session.setVariable("cause_code", session.causecode); session.setVariable("cause_name", session.cause); I Get this output only for all the call: variable_cause_code: [0] variable_cause_name: [NONE] -- Warm Regards, N.Baskar -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090226/295d9234/attachment-0002.html From saigop at gmail.com Thu Feb 26 07:11:03 2009 From: saigop at gmail.com (Gopalakrishnan A.N) Date: Thu, 26 Feb 2009 20:41:03 +0530 Subject: [Freeswitch-users] session orignate freeswitch 1.0.3 segmentation error Message-ID: <2ea4d47e0902260711m21fe581ge3a12288808359fa@mail.gmail.com> Hi, I have installed Freeswitch 1.0.3. I am using event socket with Javascript. When I try to dial the script with below command, the call is not going thru it seems to be idle. and segmentation fault core dump error, (freeswitch hangs).....[?] new_session = new Session.originate(session, "sofia/default/@foo.com"); bridge(session, new_session); I saw in the wiki http://wiki.freeswitch.org/wiki/FreeSwitch_Javascript_Session that the session is depreciated, earlier I was using like this in Freeswitch 1.0.2, it works fine....:) session = new Session(); session.originate(session, "{ignore_early_media=true}sofia/default/@foo.com"); So something I am missing, please let me know where I am wrong? -- Thank you with regards, Gopal, PeopleTech Systems Private Limited www.peopletech.co.in -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090226/ab99f74e/attachment-0002.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/gif Size: 100 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090226/ab99f74e/attachment-0002.gif From alexander at degreiff.com Thu Feb 26 07:19:33 2009 From: alexander at degreiff.com (Alexander de Greiff) Date: Thu, 26 Feb 2009 16:19:33 +0100 (CET) Subject: [Freeswitch-users] switch voices in ivr menus Message-ID: <15479302.61235661515638.JavaMail.alexanderdegreiff@AdG-Mac.local> another question: - in the ivr application i have cepstral voice matthias read the main menu ok. - i select submenu1 and cepstral voice katrin reads the submenu1 correctly. - i go back to the main menu and the voice is not switched back to the specified voice matthias. each voice is explicitly specified in each menu. any suggestions? kind regards alex From brian at freeswitch.org Thu Feb 26 07:31:00 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 26 Feb 2009 09:31:00 -0600 Subject: [Freeswitch-users] dialplan condition regex question In-Reply-To: <1520981.41235660110667.JavaMail.alexanderdegreiff@AdG-Mac.local> References: <1520981.41235660110667.JavaMail.alexanderdegreiff@AdG-Mac.local> Message-ID: Read this http://wiki.freeswitch.org/wiki/Dialplan_XML#About_Dialplan_Variables You can't condition on a variable you set in the same extension because the set happens later. Thus its not possible to do what you're doing... Once you understand the dialplan is just a list of instructions that is compiled before its installed on the session and sent into execute. You're trying to have soup before the chicken has hatched. /b On Feb 26, 2009, at 8:56 AM, Alexander de Greiff wrote: > hi all, > > i am dialing the number 123456789 (example) reaching fs via inbound > sip gateway and hitting following dialplan: > > > > > > > ... > data="{ignore_early_media=true}user/${my_dialed_extension}@$$ > {domain}"/> > ... > > > via info i can see that the variable my_dialed_extension is > populated ok with 789 but somehow the second condition is not met. > when i change that to match (.*) the actions gets executed and the > my_dialed_extension inside is correct. > > any suggestions? From brian at freeswitch.org Thu Feb 26 07:35:00 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 26 Feb 2009 09:35:00 -0600 Subject: [Freeswitch-users] session orignate freeswitch 1.0.3 segmentation error In-Reply-To: <2ea4d47e0902260711m21fe581ge3a12288808359fa@mail.gmail.com> References: <2ea4d47e0902260711m21fe581ge3a12288808359fa@mail.gmail.com> Message-ID: can you include the backtrace? We might have already fixed this one. http://wiki.freeswitch.org/wiki/Reporting_Bugs /b On Feb 26, 2009, at 9:11 AM, Gopalakrishnan A.N wrote: > Hi, > > I have installed Freeswitch 1.0.3. I am using event socket with > Javascript. When I try to dial the script with below command, the > call is not going thru it seems to be idle. and segmentation fault > core dump error, (freeswitch hangs).....<323.gif> > > > new_session = new Session.originate(session, "sofia/default/ > @foo.com"); > bridge(session, new_session); > > I saw in the wiki http://wiki.freeswitch.org/wiki/FreeSwitch_Javascript_Session > that the session is depreciated, earlier I was using like this in > Freeswitch 1.0.2, it works fine....:) > > session = new Session(); > session.originate(session, "{ignore_early_media=true}sofia/default/ > @foo.com"); > > So something I am missing, please let me know where I am wrong? > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090226/e5d368be/attachment-0002.html From brian at freeswitch.org Thu Feb 26 07:36:19 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 26 Feb 2009 09:36:19 -0600 Subject: [Freeswitch-users] switch voices in ivr menus In-Reply-To: <15479302.61235661515638.JavaMail.alexanderdegreiff@AdG-Mac.local> References: <15479302.61235661515638.JavaMail.alexanderdegreiff@AdG-Mac.local> Message-ID: <6A44C656-4352-4246-9CF1-816E86DACB54@freeswitch.org> What I would need is a debug log... ie press F8, attach that info to a jira http://jira.freeswitch.org and please assign it to me "brian" is the user. /b PS: do not paste logs in the comment box.. Attach them instead. On Feb 26, 2009, at 9:19 AM, Alexander de Greiff wrote: > another question: > > - in the ivr application i have cepstral voice matthias read the > main menu ok. > - i select submenu1 and cepstral voice katrin reads the submenu1 > correctly. > - i go back to the main menu and the voice is not switched back to > the specified voice matthias. > > each voice is explicitly specified in each menu. > > any suggestions? > > kind regards > alex From sunil.d.admin at gmail.com Thu Feb 26 07:36:18 2009 From: sunil.d.admin at gmail.com (Sunil Singh) Date: Thu, 26 Feb 2009 21:06:18 +0530 Subject: [Freeswitch-users] [ANN] Spice Telephony - an open source FreeSWITCH/Erlang callcenter platform In-Reply-To: <20090220191911.GD29511@hijacked.us> References: <22015518.post@talk.nabble.com> <20090220191911.GD29511@hijacked.us> Message-ID: Suggest Try C-Zentrix (from tvtworld.com). It has a very powerful predictive dialer and its also ranked 10th just 2 places below freewsitch. It supports 100 agents on a single box with predictive dialer,IVR, logger and CRM. I don't know how they are doing it but they are running successfully with almost 100% uptime. Try it out . On Sat, Feb 21, 2009 at 12:49 AM, Andrew Thompson wrote: > On Sat, Feb 14, 2009 at 03:04:01PM -0800, JCATS wrote: > > > > Have you planned any predictive dialer features ( like VICIDIAL )? > > > > As Ken Rice mentioned, this isn't really the focus of the project - it's > more for inbound and directed outbound (calling campaigns to specific > people/businesses - not everyone in the phonebook). Primary focus is > inbound (multi brand, skill based routing, dynamic wrapup times, etc). > > Expect a new release sometime soonish that actually does something > useful (accepts and routes inbound calls from FreeSWITCH to an agent). > Also; public source control. There's just some additional corporate > nonsense that I have to sort out (again) before that can go live. > > Andrew > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090226/0635f4d8/attachment-0002.html From brian at freeswitch.org Thu Feb 26 07:57:53 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 26 Feb 2009 09:57:53 -0600 Subject: [Freeswitch-users] switch voices in ivr menus In-Reply-To: <15479302.61235661515638.JavaMail.alexanderdegreiff@AdG-Mac.local> References: <15479302.61235661515638.JavaMail.alexanderdegreiff@AdG-Mac.local> Message-ID: <539F17DE-96C8-45C1-8BEB-273D0AA83A3D@freeswitch.org> Alex, Mine changes voices every time.. can you post your ivr.conf.xml along with the report? /b From alexander at degreiff.com Thu Feb 26 08:52:31 2009 From: alexander at degreiff.com (Alexander de Greiff) Date: Thu, 26 Feb 2009 17:52:31 +0100 (CET) Subject: [Freeswitch-users] switch voices in ivr menus In-Reply-To: <4222480.81235667066314.JavaMail.alexanderdegreiff@AdG-Mac> Message-ID: <7219632.101235667091863.JavaMail.alexanderdegreiff@AdG-Mac> brian, demo3 is the main menu. all submenus change voices correctly. when i go to the main menu via menu-sub (7) then the voice is changes correctly. only when i menu-top (9) to main menu the voice is not changed. how do i produce the debug log? in the cli? this i a remote terminal. f8 is not an option. here is the part of my ivr.conf.xml: From brian at freeswitch.org Thu Feb 26 09:01:33 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 26 Feb 2009 11:01:33 -0600 Subject: [Freeswitch-users] switch voices in ivr menus In-Reply-To: <7219632.101235667091863.JavaMail.alexanderdegreiff@AdG-Mac> References: <7219632.101235667091863.JavaMail.alexanderdegreiff@AdG-Mac> Message-ID: Yes F8 will work on a remote teminal... thats how I do it :P /b On Feb 26, 2009, at 10:52 AM, Alexander de Greiff wrote: > how do i produce the debug log? in the cli? this i a remote > terminal. f8 is not an option. From brian at freeswitch.org Thu Feb 26 09:18:01 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 26 Feb 2009 11:18:01 -0600 Subject: [Freeswitch-users] switch voices in ivr menus In-Reply-To: <7219632.101235667091863.JavaMail.alexanderdegreiff@AdG-Mac> References: <7219632.101235667091863.JavaMail.alexanderdegreiff@AdG-Mac> Message-ID: :P fixed in 12297 /b On Feb 26, 2009, at 10:52 AM, Alexander de Greiff wrote: > brian, > > demo3 is the main menu. all submenus change voices correctly. when i > go to the main menu via menu-sub (7) then the voice is changes > correctly. only when i menu-top (9) to main menu the voice is not > changed. > > how do i produce the debug log? in the cli? this i a remote > terminal. f8 is not an option. > > here is the part of my ivr.conf.xml: From intralanman at freeswitch.org Thu Feb 26 10:39:23 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Thu, 26 Feb 2009 13:39:23 -0500 Subject: [Freeswitch-users] ESL Wrapper In-Reply-To: References: <558004.60211.qm@web59511.mail.ac4.yahoo.com> <78A818D3-6C4F-420D-A922-751A69E7E080@avgs.ca> <6CE58813-715A-43DB-877B-638B5CE7E6E9@freeswitch.org> <87f2f3b90902251342q1e393127ha7fbdbebf6d83dac@mail.gmail.com> <1235647534150-2389093.post@n2.nabble.com> Message-ID: <49A6E1DB.3070806@freeswitch.org> and it will probably be a good idea to do make phpmod-install so that the .so and .php files gets into the correct place to be included -Ray Mathieu Rene wrote: > > You need your distro's php dev pakage. > On 26-Feb-09, at 6:25 AM, Rex_Alex wrote: > >> Hi All, I tried svn up && ./bootstrap.sh && ./configure && make >> install and did Mathieu's suggestion but getting error as below, >> [root at server esl]# make phpmod make MYLIB="../libesl.a" >> SOLINK="-shared -Xlinker -x" >> CFLAGS="-I/root/freeswitch-1.0.3/libs/esl/src/include -DHAVE_EDITLINE >> -g -ggdb -I../../libs/libedit/src/ -fPIC -O2 -ffast-math -Wall >> -Werror -Wunused-variable -Wwrite-strings -Wstrict-prototypes >> -Wmissing-prototypes" >> CXXFLAGS="-I/root/freeswitch-1.0.3/libs/esl/src/include >> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC" >> CXX_CFLAGS="" -C php make[1]: php-config: Command not found make[1]: >> Entering directory `/root/freeswitch-1.0.3/libs/esl/php' g++ >> -I/root/freeswitch-1.0.3/libs/esl/src/include -DHAVE_EDITLINE -g >> -ggdb -I../../libs/libedit/src/ -fPIC -c esl_wrap.cpp -o esl_wrap.o >> esl_wrap.cpp:717:18: error: zend.h: No such file or directory >> esl_wrap.cpp:718:22: error: zend_API.h: No such file or directory >> esl_wrap.cpp:719:17: error: php.h: No such file or directory >> esl_wrap.cpp:973:21: error: php_ini.h: No such file or directory >> esl_wrap.cpp:974:31: error: ext/standard/info.h: No such file or >> directory esl_wrap.cpp:767: error: ?E_ERROR? was not declared in this >> scope esl_wrap.cpp:788: error: ISO C++ forbids declaration of >> ?ZEND_RSRC_DTOR_FUNC? with no type esl_wrap.cpp:788: error: >> ?SWIG_landfill? was not declared in this scope esl_wrap.cpp:788: >> error: expected ?,? or ?;? before ?{? token esl_wrap.cpp:793: error: >> variable or field ?SWIG_ZTS_SetPointerZval? declared void >> esl_wrap.cpp:793: error: ?zval? was not declared in this scope >> esl_wrap.cpp:793: error: ?z? was not declared in this scope >> esl_wrap.cpp:793: error: expected primary-expression before ?void? >> esl_wrap.cpp:793: error: expected primary-expression before ?*? token >> esl_wrap.cpp:793: error: ?type? was not declared in this scope >> esl_wrap.cpp:793: error: expected primary-expression before ?int? >> esl_wrap.cpp:793: error: initializer expression list treated as >> compound expression esl_wrap.cpp:793: error: expected ?,? or ?;? >> before ?{? token make[1]: *** [esl_wrap.o] Error 1 make[1]: Leaving >> directory `/root/freeswitch-1.0.3/libs/esl/php' make: *** [phpmod] >> Error 2 [root at server esl]# Please tell me where am i wrong? Thanks, Rex >> >> mercutioviz wrote: >> On Wed, Feb 25, 2009 at 11:34 AM, Brian West wrote: > If he's on >> 1.0.3 I don't think it has php in it.. Can't he do the whole >> bootstrap process? svn up && ./bootstrap.sh && ./configure && >> make install And then do Mathieu's suggestion? -MC >> _______________________________________________ Freeswitch-users >> mailing list Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> ------------------------------------------------------------------------ >> View this message in context: Re: ESL Wrapper >> >> Sent from the freeswitch-users mailing list archive >> at Nabble.com. >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090226/0e5cecd6/attachment-0002.html From rex.alex345 at yahoo.com Thu Feb 26 10:51:01 2009 From: rex.alex345 at yahoo.com (Rex_Alex) Date: Thu, 26 Feb 2009 10:51:01 -0800 (PST) Subject: [Freeswitch-users] ESL Wrapper In-Reply-To: References: <558004.60211.qm@web59511.mail.ac4.yahoo.com> <78A818D3-6C4F-420D-A922-751A69E7E080@avgs.ca> <6CE58813-715A-43DB-877B-638B5CE7E6E9@freeswitch.org> <87f2f3b90902251342q1e393127ha7fbdbebf6d83dac@mail.gmail.com> <1235647534150-2389093.post@n2.nabble.com> Message-ID: <1235674261134-2391480.post@n2.nabble.com> Hi Mathieu, But other php scripts are working fine. Only when I am tring Single_Command.php with ESP.php, it's not working. Rex. Mathieu Rene wrote: > > > You need your distro's php dev pakage. > On 26-Feb-09, at 6:25 AM, Rex_Alex wrote: > >> Hi All, I tried svn up && ./bootstrap.sh && ./configure && make >> install and did Mathieu's suggestion but getting error as below, >> [root at server esl]# make phpmod make MYLIB="../libesl.a" SOLINK="- >> shared -Xlinker -x" CFLAGS="-I/root/freeswitch-1.0.3/libs/esl/src/ >> include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2 >> -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings -Wstrict- >> prototypes -Wmissing-prototypes" CXXFLAGS="-I/root/freeswitch-1.0.3/ >> libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/ >> src/ -fPIC" CXX_CFLAGS="" -C php make[1]: php-config: Command not >> found make[1]: Entering directory `/root/freeswitch-1.0.3/libs/esl/ >> php' g++ -I/root/freeswitch-1.0.3/libs/esl/src/include - >> DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -c >> esl_wrap.cpp -o esl_wrap.o esl_wrap.cpp:717:18: error: zend.h: No >> such file or directory esl_wrap.cpp:718:22: error: zend_API.h: No >> such file or directory esl_wrap.cpp:719:17: error: php.h: No such >> file or directory esl_wrap.cpp:973:21: error: php_ini.h: No such >> file or directory esl_wrap.cpp:974:31: error: ext/standard/info.h: >> No such file or directory esl_wrap.cpp:767: error: ?E_ERROR? was not >> declared in this scope esl_wrap.cpp:788: error: ISO C++ forbids >> declaration of ?ZEND_RSRC_DTOR_FUNC? with no type esl_wrap.cpp:788: >> error: ?SWIG_landfill? was not declared in this scope esl_wrap.cpp: >> 788: error: expected ?,? or ?;? before ?{? token esl_wrap.cpp:793: >> error: variable or field ?SWIG_ZTS_SetPointerZval? declared void >> esl_wrap.cpp:793: error: ?zval? was not declared in this scope >> esl_wrap.cpp:793: error: ?z? was not declared in this scope >> esl_wrap.cpp:793: error: expected primary-expression before ?void? >> esl_wrap.cpp:793: error: expected primary-expression before ?*? >> token esl_wrap.cpp:793: error: ?type? was not declared in this scope >> esl_wrap.cpp:793: error: expected primary-expression before ?int? >> esl_wrap.cpp:793: error: initializer expression list treated as >> compound expression esl_wrap.cpp:793: error: expected ?,? or ?;? >> before ?{? token make[1]: *** [esl_wrap.o] Error 1 make[1]: Leaving >> directory `/root/freeswitch-1.0.3/libs/esl/php' make: *** [phpmod] >> Error 2 [root at server esl]# Please tell me where am i wrong? Thanks, >> Rex >> mercutioviz wrote: >> On Wed, Feb 25, 2009 at 11:34 AM, Brian West wrote: > If he's on >> 1.0.3 I don't think it has php in it.. Can't he do the whole >> bootstrap process? svn up && ./bootstrap.sh && ./configure && make >> install And then do Mathieu's suggestion? -MC >> _______________________________________________ Freeswitch-users >> mailing list Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> View this message in context: Re: ESL Wrapper >> Sent from the freeswitch-users mailing list archive at Nabble.com. >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://n2.nabble.com/ESL-Wrapper-tp2385651p2391480.html Sent from the freeswitch-users mailing list archive at Nabble.com. From anthony.minessale at gmail.com Thu Feb 26 11:04:44 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 26 Feb 2009 13:04:44 -0600 Subject: [Freeswitch-users] ESL Wrapper In-Reply-To: <1235674261134-2391480.post@n2.nabble.com> References: <558004.60211.qm@web59511.mail.ac4.yahoo.com> <78A818D3-6C4F-420D-A922-751A69E7E080@avgs.ca> <6CE58813-715A-43DB-877B-638B5CE7E6E9@freeswitch.org> <87f2f3b90902251342q1e393127ha7fbdbebf6d83dac@mail.gmail.com> <1235647534150-2389093.post@n2.nabble.com> <1235674261134-2391480.post@n2.nabble.com> Message-ID: <191c3a030902261104n560475acqa924491afd07bd42@mail.gmail.com> the esl php mod is a binary mod which must be compiled so it requires the devel add on for php On Thu, Feb 26, 2009 at 12:51 PM, Rex_Alex wrote: > > Hi Mathieu, > > But other php scripts are working fine. Only when I am tring > Single_Command.php with ESP.php, it's not working. > > Rex. > > Mathieu Rene wrote: > > > > > > You need your distro's php dev pakage. > > On 26-Feb-09, at 6:25 AM, Rex_Alex wrote: > > > >> Hi All, I tried svn up && ./bootstrap.sh && ./configure && make > >> install and did Mathieu's suggestion but getting error as below, > >> [root at server esl]# make phpmod make MYLIB="../libesl.a" SOLINK="- > >> shared -Xlinker -x" CFLAGS="-I/root/freeswitch-1.0.3/libs/esl/src/ > >> include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2 > >> -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings -Wstrict- > >> prototypes -Wmissing-prototypes" CXXFLAGS="-I/root/freeswitch-1.0.3/ > >> libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/ > >> src/ -fPIC" CXX_CFLAGS="" -C php make[1]: php-config: Command not > >> found make[1]: Entering directory `/root/freeswitch-1.0.3/libs/esl/ > >> php' g++ -I/root/freeswitch-1.0.3/libs/esl/src/include - > >> DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -c > >> esl_wrap.cpp -o esl_wrap.o esl_wrap.cpp:717:18: error: zend.h: No > >> such file or directory esl_wrap.cpp:718:22: error: zend_API.h: No > >> such file or directory esl_wrap.cpp:719:17: error: php.h: No such > >> file or directory esl_wrap.cpp:973:21: error: php_ini.h: No such > >> file or directory esl_wrap.cpp:974:31: error: ext/standard/info.h: > >> No such file or directory esl_wrap.cpp:767: error: ?E_ERROR? was not > >> declared in this scope esl_wrap.cpp:788: error: ISO C++ forbids > >> declaration of ?ZEND_RSRC_DTOR_FUNC? with no type esl_wrap.cpp:788: > >> error: ?SWIG_landfill? was not declared in this scope esl_wrap.cpp: > >> 788: error: expected ?,? or ?;? before ?{? token esl_wrap.cpp:793: > >> error: variable or field ?SWIG_ZTS_SetPointerZval? declared void > >> esl_wrap.cpp:793: error: ?zval? was not declared in this scope > >> esl_wrap.cpp:793: error: ?z? was not declared in this scope > >> esl_wrap.cpp:793: error: expected primary-expression before ?void? > >> esl_wrap.cpp:793: error: expected primary-expression before ?*? > >> token esl_wrap.cpp:793: error: ?type? was not declared in this scope > >> esl_wrap.cpp:793: error: expected primary-expression before ?int? > >> esl_wrap.cpp:793: error: initializer expression list treated as > >> compound expression esl_wrap.cpp:793: error: expected ?,? or ?;? > >> before ?{? token make[1]: *** [esl_wrap.o] Error 1 make[1]: Leaving > >> directory `/root/freeswitch-1.0.3/libs/esl/php' make: *** [phpmod] > >> Error 2 [root at server esl]# Please tell me where am i wrong? Thanks, > >> Rex > >> mercutioviz wrote: > >> On Wed, Feb 25, 2009 at 11:34 AM, Brian West wrote: > If he's on > >> 1.0.3 I don't think it has php in it.. Can't he do the whole > >> bootstrap process? svn up && ./bootstrap.sh && ./configure && make > >> install And then do Mathieu's suggestion? -MC > >> _______________________________________________ Freeswitch-users > >> mailing list Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > >> View this message in context: Re: ESL Wrapper > >> Sent from the freeswitch-users mailing list archive at Nabble.com. > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > View this message in context: > http://n2.nabble.com/ESL-Wrapper-tp2385651p2391480.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090226/6bef0e9d/attachment-0002.html From msc at freeswitch.org Thu Feb 26 11:06:22 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 26 Feb 2009 11:06:22 -0800 Subject: [Freeswitch-users] ESL Wrapper In-Reply-To: <1235674261134-2391480.post@n2.nabble.com> References: <558004.60211.qm@web59511.mail.ac4.yahoo.com> <78A818D3-6C4F-420D-A922-751A69E7E080@avgs.ca> <6CE58813-715A-43DB-877B-638B5CE7E6E9@freeswitch.org> <87f2f3b90902251342q1e393127ha7fbdbebf6d83dac@mail.gmail.com> <1235647534150-2389093.post@n2.nabble.com> <1235674261134-2391480.post@n2.nabble.com> Message-ID: <87f2f3b90902261106l6925b035o1a5e19d881ff1584@mail.gmail.com> On Thu, Feb 26, 2009 at 10:51 AM, Rex_Alex wrote: > > Hi Mathieu, > > But other php scripts are working fine. Only when I am tring > Single_Command.php with ESP.php, it's not working. > > Rex. > That may be true but the php-devel package is necessary for building the ESL wrapper for php. The php-devel package is NOT necessary simply to run most php scripts. What OS is it? If it's CentOS or similar then you can just do this: yum install -y php-devel -MC From rex.alex345 at yahoo.com Thu Feb 26 11:20:38 2009 From: rex.alex345 at yahoo.com (Rex_Alex) Date: Thu, 26 Feb 2009 11:20:38 -0800 (PST) Subject: [Freeswitch-users] ESL Wrapper In-Reply-To: <87f2f3b90902261106l6925b035o1a5e19d881ff1584@mail.gmail.com> References: <558004.60211.qm@web59511.mail.ac4.yahoo.com> <78A818D3-6C4F-420D-A922-751A69E7E080@avgs.ca> <6CE58813-715A-43DB-877B-638B5CE7E6E9@freeswitch.org> <87f2f3b90902251342q1e393127ha7fbdbebf6d83dac@mail.gmail.com> <1235647534150-2389093.post@n2.nabble.com> <1235674261134-2391480.post@n2.nabble.com> <87f2f3b90902261106l6925b035o1a5e19d881ff1584@mail.gmail.com> Message-ID: <1235676038843-2391645.post@n2.nabble.com> Hi, Yea, you are right it's CentOS version 5.2. Let me try the same and then I will reply you with status. Thanks, Rex. mercutioviz wrote: > > On Thu, Feb 26, 2009 at 10:51 AM, Rex_Alex wrote: >> >> Hi Mathieu, >> >> But other php scripts are working fine. Only when I am tring >> Single_Command.php with ESP.php, it's not working. >> >> Rex. >> > > That may be true but the php-devel package is necessary for building > the ESL wrapper for php. The php-devel package is NOT necessary simply > to run most php scripts. What OS is it? If it's CentOS or similar then > you can just do this: > yum install -y php-devel > > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://n2.nabble.com/ESL-Wrapper-tp2385651p2391645.html Sent from the freeswitch-users mailing list archive at Nabble.com. From chavpaskov at shaw.ca Thu Feb 26 12:56:10 2009 From: chavpaskov at shaw.ca (Tchavdar Paskov) Date: Thu, 26 Feb 2009 12:56:10 -0800 Subject: [Freeswitch-users] Variables from failed call to be exported to a a new B leg Message-ID: Hi Everybody, this is what i' trying to do / unsuccessfully / so far: ?? ????? ???????? ???????? ???????? ???????? ? -> at this point i'd like to collect some sip Vars from the failed call ???????? - from what i red in wiki i think this is the way to export the var to the Bleg ???????? ????? ??? the gw_2? does not? seem to receive the? sip_hangup_phrase. pls? help me to figure out what i'm doing wrong. thank you in advance. regards Chav -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090226/fdc264ff/attachment-0002.html From msc at freeswitch.org Thu Feb 26 13:06:31 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 26 Feb 2009 13:06:31 -0800 Subject: [Freeswitch-users] Variables from failed call to be exported to a a new B leg In-Reply-To: References: Message-ID: <87f2f3b90902261306t1efc6440y41bacef2a100b9ce@mail.gmail.com> > ???????? Why are you using $0 here? Is that a typo? -MC From nik.middleton at noblesolutions.co.uk Thu Feb 26 13:09:46 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Thu, 26 Feb 2009 21:09:46 -0000 Subject: [Freeswitch-users] Cant get Disposition status in Javascript In-Reply-To: References: <191c3a030902260611p5ee885e1ud422b8c0bcd17502@mail.gmail.com> Message-ID: Works for me, see snippet below var first_session = new Session(dial_string); // Trap for call failure if (!first_session.ready()) { consoleLog("err", "Disposition: " + first_session.cause + "\n"); if (first_session.cause == "USER_BUSY") { Disposition = "BUSY"; } else if (first_session.cause == "NO_ROUTE_DESTINATION") { Disposition = "DCN"; } else if (first_session.cause == "NO_ANSWER") { Disposition = "NA"; } exit(); } ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Baskar Sent: 26 February 2009 14:56 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Cant get Disposition status in Javascript Hi Anthony Minessale, I have added these lines in my javascript with your guidance. But still i did not get any status like busy , no answer, etc . session.setVariable("cause_code", session.causecode); session.setVariable("cause_name", session.cause); I Get this output only for all the call: variable_cause_code: [0] variable_cause_name: [NONE] -- Warm Regards, N.Baskar -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090226/0cd4cb79/attachment-0002.html From chavpaskov at shaw.ca Thu Feb 26 13:09:55 2009 From: chavpaskov at shaw.ca (Tchavdar Paskov) Date: Thu, 26 Feb 2009 13:09:55 -0800 Subject: [Freeswitch-users] Variables from failed call to be exported to a a new B leg In-Reply-To: <87f2f3b90902261306t1efc6440y41bacef2a100b9ce@mail.gmail.com> References: <87f2f3b90902261306t1efc6440y41bacef2a100b9ce@mail.gmail.com> Message-ID: Yes it was typo.My Bad Chav ----- Original Message ----- From: Michael Collins Date: Thursday, February 26, 2009 1:07 pm Subject: Re: [Freeswitch-users] Variables from failed call to be exported to a a new B leg To: freeswitch-users at lists.freeswitch.org > > ???????? data="sofia/gateway/gw_2/$0" /> > > Why are you using $0 here? Is that a typo? > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090226/d4a4583f/attachment-0002.html From brian at freeswitch.org Thu Feb 26 13:14:39 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 26 Feb 2009 15:14:39 -0600 Subject: [Freeswitch-users] Variables from failed call to be exported to a a new B leg In-Reply-To: References: Message-ID: You might also want to actually SET a value to the variable. /b On Feb 26, 2009, at 2:56 PM, Tchavdar Paskov wrote: > From chavpaskov at shaw.ca Thu Feb 26 13:18:36 2009 From: chavpaskov at shaw.ca (Tchavdar Paskov) Date: Thu, 26 Feb 2009 13:18:36 -0800 Subject: [Freeswitch-users] Variables from failed call to be exported to a a new B leg In-Reply-To: References: Message-ID: isn't this already done because of the info application called before ? Chav ----- Original Message ----- From: Brian West Date: Thursday, February 26, 2009 1:14 pm Subject: Re: [Freeswitch-users] Variables from failed call to be exported to a a new B leg To: freeswitch-users at lists.freeswitch.org > You might also want to actually SET a value to the variable. > > /b > > On Feb 26, 2009, at 2:56 PM, Tchavdar Paskov wrote: > > > data="nolocal:sip_hangup_phrase" /> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090226/3d4b890c/attachment-0002.html From msc at freeswitch.org Thu Feb 26 13:35:25 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 26 Feb 2009 13:35:25 -0800 Subject: [Freeswitch-users] Variables from failed call to be exported to a a new B leg In-Reply-To: References: Message-ID: <87f2f3b90902261335g337d1bb5w6191cb97b3e90602@mail.gmail.com> On Thu, Feb 26, 2009 at 1:18 PM, Tchavdar Paskov wrote: > isn't this already done because of the info application called before ? > Chav > to make sure that there is indeed a value and that it gets exported to the second b-leg try this: see if my_var is populated on the new b-leg. -MC From brian at freeswitch.org Thu Feb 26 13:47:02 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 26 Feb 2009 15:47:02 -0600 Subject: [Freeswitch-users] Variables from failed call to be exported to a a new B leg In-Reply-To: References: Message-ID: <797DE9FB-7B89-4FD1-A90B-2A09EDB96831@freeswitch.org> You then don't use the export command... you set the variable /b On Feb 26, 2009, at 3:18 PM, Tchavdar Paskov wrote: > isn't this already done because of the info application called > before ? > Chav From mrene_lists at avgs.ca Thu Feb 26 13:48:01 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Thu, 26 Feb 2009 16:48:01 -0500 Subject: [Freeswitch-users] Variables from failed call to be exported to a a new B leg In-Reply-To: References: Message-ID: <07CCDD39-4836-4114-8966-DE81C629D70F@avgs.ca> On 26-Feb-09, at 3:56 PM, Tchavdar Paskov wrote: > Hi Everybody, > > this is what i' trying to do / unsuccessfully / so far: > > > > break="never"> > > > > -> at this point i'd like to > collect some sip Vars from the failed call > data="nolocal:sip_hangup_phrase" /> - from what i red in wiki i > think this is the way to export the var to the Bleg > > > > > the gw_2 does not seem to receive the sip_hangup_phrase. > pls help me to figure out what i'm doing wrong. > > thank you in advance. > regards > Chav > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Thu Feb 26 13:48:32 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 26 Feb 2009 15:48:32 -0600 Subject: [Freeswitch-users] Variables from failed call to be exported to a a new B leg In-Reply-To: <87f2f3b90902261335g337d1bb5w6191cb97b3e90602@mail.gmail.com> References: <87f2f3b90902261335g337d1bb5w6191cb97b3e90602@mail.gmail.com> Message-ID: <5D84AF46-1C77-45D3-96C4-E670A1AFBE8C@freeswitch.org> You can also do it like this: I don't think what MC pointed out works. I'll have to double check. /b On Feb 26, 2009, at 3:35 PM, Michael Collins wrote: > > > to make sure that there is indeed a value and that it gets exported to > the second b-leg try this: > > > > see if my_var is populated on the new b-leg. > -MC From mrene_lists at avgs.ca Thu Feb 26 13:51:26 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Thu, 26 Feb 2009 16:51:26 -0500 Subject: [Freeswitch-users] Variables from failed call to be exported to a a new B leg In-Reply-To: References: Message-ID: <9FA332A7-0566-489B-9C1B-9E5A2CC04A5A@avgs.ca> Oh I just remembered something, You can set the "failed_xml_cdr_prefix" variable (on the A-leg). And it will copy ALL variables from the B-leg if the call fails. Then you should have providerA_hangup_cause, providerB_hangup_cause. Mathieu On 26-Feb-09, at 3:56 PM, Tchavdar Paskov wrote: > Hi Everybody, > > this is what i' trying to do / unsuccessfully / so far: > > > > break="never"> > > > > -> at this point i'd like to > collect some sip Vars from the failed call > data="nolocal:sip_hangup_phrase" /> - from what i red in wiki i > think this is the way to export the var to the Bleg > > > > > the gw_2 does not seem to receive the sip_hangup_phrase. > pls help me to figure out what i'm doing wrong. > > thank you in advance. > regards > Chav > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Thu Feb 26 13:52:29 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 26 Feb 2009 15:52:29 -0600 Subject: [Freeswitch-users] Variables from failed call to be exported to a a new B leg In-Reply-To: <07CCDD39-4836-4114-8966-DE81C629D70F@avgs.ca> References: <07CCDD39-4836-4114-8966-DE81C629D70F@avgs.ca> Message-ID: <8F49C3FE-3F8A-4BA2-8F67-09150DCBFC99@freeswitch.org> OK I think we have covered BOTH directions now ;) /b On Feb 26, 2009, at 3:48 PM, Mathieu Rene wrote: > > > > From msc at freeswitch.org Thu Feb 26 14:03:38 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 26 Feb 2009 14:03:38 -0800 Subject: [Freeswitch-users] Variables from failed call to be exported to a a new B leg In-Reply-To: <87f2f3b90902261335g337d1bb5w6191cb97b3e90602@mail.gmail.com> References: <87f2f3b90902261335g337d1bb5w6191cb97b3e90602@mail.gmail.com> Message-ID: <87f2f3b90902261403v50b3a5cfrf352115d40511cb0@mail.gmail.com> > > to make sure that there is indeed a value and that it gets exported to > the second b-leg try this: > oops, that set line should have been: That "nolocal:" was extraneous from a lazy copy & paste > > > see if my_var is populated on the new b-leg. > -MC > The most elegant solution is the one Brian gave: So use it, please, and forget what I wrote. :) -MC From brian at freeswitch.org Thu Feb 26 14:32:53 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 26 Feb 2009 16:32:53 -0600 Subject: [Freeswitch-users] Variables from failed call to be exported to a a new B leg In-Reply-To: <87f2f3b90902261403v50b3a5cfrf352115d40511cb0@mail.gmail.com> References: <87f2f3b90902261335g337d1bb5w6191cb97b3e90602@mail.gmail.com> <87f2f3b90902261403v50b3a5cfrf352115d40511cb0@mail.gmail.com> Message-ID: <60652253-9B46-4CBF-94E2-5E61BCC996D2@freeswitch.org> You can use nolocal: with export. Just not set. /b On Feb 26, 2009, at 4:03 PM, Michael Collins wrote: > That "nolocal:" was extraneous from a lazy copy & paste From anthony.minessale at gmail.com Thu Feb 26 14:34:53 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 26 Feb 2009 16:34:53 -0600 Subject: [Freeswitch-users] Variables from failed call to be exported to a a new B leg In-Reply-To: <87f2f3b90902261403v50b3a5cfrf352115d40511cb0@mail.gmail.com> References: <87f2f3b90902261335g337d1bb5w6191cb97b3e90602@mail.gmail.com> <87f2f3b90902261403v50b3a5cfrf352115d40511cb0@mail.gmail.com> Message-ID: <191c3a030902261434k325de0bcm4b30dcb960d6fd75@mail.gmail.com> set the var failed_xml_cdr_prefix=foo before you call bridge and the failed calls will have a complete xml cdr saved in foo_X where X is an incrementing number from 1 upwards. On Thu, Feb 26, 2009 at 4:03 PM, Michael Collins wrote: > > > > to make sure that there is indeed a value and that it gets exported to > > the second b-leg try this: > > > > oops, that set line should have been: > > That "nolocal:" was extraneous from a lazy copy & paste > > > > > > > see if my_var is populated on the new b-leg. > > -MC > > > > The most elegant solution is the one Brian gave: > > > So use it, please, and forget what I wrote. :) > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090226/a9b5c3d1/attachment-0002.html From nik.middleton at noblesolutions.co.uk Thu Feb 26 14:55:22 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Thu, 26 Feb 2009 22:55:22 -0000 Subject: [Freeswitch-users] Console messages Message-ID: Hi Guys, Is there a way of displaying a console message not related to a log level? I've got the console only reporting errors now, but it would be nice to be able to display a message when a given condition exists. Yes, I could set it as an error level message, but I'd rather not do that. Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090226/d60d04b0/attachment-0002.html From b_ball_henry at hotmail.com Thu Feb 26 16:32:30 2009 From: b_ball_henry at hotmail.com (Henry Huang) Date: Thu, 26 Feb 2009 16:32:30 -0800 Subject: [Freeswitch-users] snd_dummy setting for skype Message-ID: <59ad9ca10902261632s8a50903gba865a093c892bd9@mail.gmail.com> I went through the wiki on mod_skypiax and see there should be a script to make skype work without sound card in linux. Does anyone know where to obtain that script to make sound work "without sound card"? I am currently creating a /etc/asound.conf for skype to load the "fake" sound driver. I do hear sound, but it's not perfect, it's very choppy and it gives me error message when starting skype. The following is my asound.conf setting. Hopefully someone can shed some light : pcm.plugfile{ type plug slave { pcm infile format S16_LE channels 1 rate 16000 } } pcm.infile { type file slave { pcm null } file /dev/dsp infile /dev/dsp } by using this configuration. skype spit out error messages as follow but still works: ALSA lib control.c:909:(snd_ctl_open_noupdate) Invalid CTL plugfile ALSA lib control.c:909:(snd_ctl_open_noupdate) Invalid CTL infile -- Henry Huang UniC Solution - Communication Unified VoIP & Open Source software Consultant -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090226/a6524f53/attachment-0002.html From msc at freeswitch.org Thu Feb 26 16:41:25 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 26 Feb 2009 16:41:25 -0800 Subject: [Freeswitch-users] Console messages In-Reply-To: References: Message-ID: <87f2f3b90902261641s30f2036bg6800e0b49222b514@mail.gmail.com> > Is there a way of displaying a console message not related to a log level? > I?ve got the console only reporting errors now, but it would be nice to be > able to display a message when a given condition exists.? Yes, I could set > it as an error level message, but I?d rather not do that. What is the condition? That will probably determine how you proceed. -MC From anthony.minessale at gmail.com Thu Feb 26 16:49:54 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 26 Feb 2009 18:49:54 -0600 Subject: [Freeswitch-users] Console messages In-Reply-To: References: Message-ID: <191c3a030902261649q7bc16ddaxfef5e2e57ff5efa6@mail.gmail.com> You could use level "console" which will always print or use "err" or "crit". On Thu, Feb 26, 2009 at 4:55 PM, Nik Middleton < nik.middleton at noblesolutions.co.uk> wrote: > Hi Guys, > > > > Is there a way of displaying a console message not related to a log level? > I?ve got the console only reporting errors now, but it would be nice to be > able to display a message when a given condition exists. Yes, I could set > it as an error level message, but I?d rather not do that. > > > > Regards, > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090226/68a33e98/attachment-0002.html From mike at jerris.com Thu Feb 26 18:29:22 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 26 Feb 2009 21:29:22 -0500 Subject: [Freeswitch-users] Console messages In-Reply-To: <87f2f3b90902261641s30f2036bg6800e0b49222b514@mail.gmail.com> References: <87f2f3b90902261641s30f2036bg6800e0b49222b514@mail.gmail.com> Message-ID: <416BE849-131D-4A28-8381-1107AC9C19BF@jerris.com> You should be able to do loglevel of console Mike On Feb 26, 2009, at 7:41 PM, Michael Collins wrote: >> Is there a way of displaying a console message not related to a log >> level? >> I?ve got the console only reporting errors now, but it would be ni >> ce to be >> able to display a message when a given condition exists. Yes, I >> could set >> it as an error level message, but I?d rather not do that. > > What is the condition? That will probably determine how you proceed. > -MC > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mashudiflexi at telkom.co.id Thu Feb 26 20:52:24 2009 From: mashudiflexi at telkom.co.id (mashudi) Date: Fri, 27 Feb 2009 11:52:24 +0700 Subject: [Freeswitch-users] change session-timer value Message-ID: <49A77188.4070105@telkom.co.id> Hi Folks , I have problem to change the session timer value that automatically created by Freeswitch for 120, I can change the value of session-timer in /sip_profiles/internal.xml become and it work only for INVITE message response, but for UPDATE message response still use default value of 120, please help me to solved this problem. Thank you in advanced. regards mashudi pls help me to figure out what i'm doing wrong. > ***************************************** Sekarang Gratis Nelpon SLJJ Flexi diperluas ke Yogya ***************************************** From pmhshz at gmail.com Thu Feb 26 22:01:12 2009 From: pmhshz at gmail.com (shehzad p) Date: Thu, 26 Feb 2009 22:01:12 -0800 (PST) Subject: [Freeswitch-users] Suggestion for xml_curl performance In-Reply-To: <7e2ac3270902210711u3ea54b79j5868aef6b59d4800@mail.gmail.com> References: <22118122.post@talk.nabble.com> <22133185.post@talk.nabble.com> <7e2ac3270902210711u3ea54b79j5868aef6b59d4800@mail.gmail.com> Message-ID: <22240044.post@talk.nabble.com> Thanks for your response, I am studying FastCGI as Shannon recommend. Is there any other possibility of making this setup better, OR making it better requires change in architecture, If yes please anybody comment and point me out to that direction. Thanks msp Shannon-27 wrote: > > I'd recommend having a look at fastcgi as well. > > On 2/21/09, shehzad p wrote: >> >> Hi Brian, >> >> My setup is to use FS as basic calls routing. >> 1. Calls are coming to FS from more than one customer Gateways, and I >> need >> to authenticate them and check for enough balance based on database, >> [Caller Gateways] ===> [FreeSWITCH] ===> >> [Provider Gateways] >> 2. After knowing that Caller Gateways is valid, then based on dialed >> number >> it search in database for Provider Gateway and bridge the call there. >> 3. After call finish CDR is inserted back into database. >> >> My old setup was using Javascript which works fine in traffic of 10 to 20 >> calls, but then increase of traffic causes many problems. >> >> Now I eliminate use of any of the script (javascript or any other) for >> call >> routing, and route calls directly from dialplan, >> So I have setup test system using xml-curl to generate dynamic dialplan, >> I used below xml_curl PHP example as reference: >> http://wiki.freeswitch.org/wiki/Mod_xml_curl_PHP_example >> For CDR processing I used xml_cdr, with help of the example in FS source >> :scripts/contrib/trixter/xml-cdr. >> >> >> Waiting for any better suggestions, any comments... >> >> thanks >> msp. >> >> Brian West-3 wrote: >>> >>> it all depends on what you're doing.. can you elaborate? >>> >>> /b >>> >>> On Feb 20, 2009, at 4:18 AM, shehzad p wrote: >>> >>>> Recently I faced some performance bottleneck by using Javascript. >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> -- >> View this message in context: >> http://www.nabble.com/Suggestion-for-xml_curl-performance-tp22118122p22133185.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > -- > Shannon > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/Suggestion-for-xml_curl-performance-tp22118122p22240044.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From gopal2krishnan at gmail.com Thu Feb 26 22:11:23 2009 From: gopal2krishnan at gmail.com (Gopal krishnan) Date: Fri, 27 Feb 2009 11:41:23 +0530 Subject: [Freeswitch-users] session orignate freeswitch 1.0.3 segmentation error In-Reply-To: References: <2ea4d47e0902260711m21fe581ge3a12288808359fa@mail.gmail.com> Message-ID: <2ea4d47e0902262211y2e4b243ey52eb7bda04329dc7@mail.gmail.com> Hi Brian, Please find the attached backtrace files attached. And 1. SVN revision number (or binary file) - FreeSWITCH Version 1.0.3 (exported) 2. Operating System and revision - CentOS 5.2 3. Hardware information - 32 bit with 512 MB RAM 4. I am using Event socket 5. Language - Javascript One more thing in the same machine earlier I was using freeswitch 1.0.2. when the segmentation fault happens the core file was generated in the older version freeswitch bin. In 1.0.3 there is no bin directory. Is that could be the prob? On Thu, Feb 26, 2009 at 9:05 PM, Brian West wrote: > can you include the backtrace? We might have already fixed this one. > http://wiki.freeswitch.org/wiki/Reporting_Bugs > > /b > > > On Feb 26, 2009, at 9:11 AM, Gopalakrishnan A.N wrote: > > Hi, > > I have installed Freeswitch 1.0.3. I am using event socket with > Javascript. When I try to dial the script with below command, the call is > not going thru it seems to be idle. and segmentation fault core dump error, > (freeswitch hangs).....<323.gif> > > > new_session = new Session.originate(session, > "sofia/default/@foo.com"); > bridge(session, new_session); > > I saw in the wiki > http://wiki.freeswitch.org/wiki/FreeSwitch_Javascript_Session > that the session is depreciated, earlier I was using like this in > Freeswitch 1.0.2, it works fine....:) > > session = new Session(); > session.originate(session, > "{ignore_early_media=true}sofia/default/@foo.com"); > > So something I am missing, please let me know where I am wrong? > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Thank you with regards, Gopal, PeopleTech Systems Private Limited www.peopletech.co.in -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090227/6daf4869/attachment-0002.html -------------- next part -------------- (gdb) bt #0 session_originate (cx=0x856e988, obj=0x85d59b0, argc=2, argv=0x85e44c8, rval=0xaad1a25c) at mod_spidermonkey.c:2855 #1 0x00806231 in js_Invoke () from /usr/local/freeswitch/lib/libjs.so.1 #2 0x007f9418 in js_Interpret () from /usr/local/freeswitch/lib/libjs.so.1 #3 0x00805976 in js_Execute () from /usr/local/freeswitch/lib/libjs.so.1 #4 0x007c503a in JS_ExecuteScript () from /usr/local/freeswitch/lib/libjs.so.1 #5 0x0070bfe4 in eval_some_js (code=0x84c3656 "new1.js", cx=0x856e988, obj=0x85d3fc8, rval=0xaad1b278) at mod_spidermonkey.h:103 #6 0x0070c592 in js_parse_and_execute (session=0x0, input_code=0x84c3656 "new1.js", ro=0xaad1b2a0) at mod_spidermonkey.c:3583 #7 0x0070c8b1 in jsapi_function (cmd=0x84c3656 "new1.js", session=0x0, stream=0xaad1b328) at mod_spidermonkey.c:3663 #8 0x00edb91d in switch_api_execute (cmd=0x84c3650 "jsapi", arg=0x84c3656 "new1.js", session=0x0, stream=0xaad1b328) at src/switch_loadable_module.c:1524 #9 0x00ec0c06 in switch_console_process (cmd=0x84c3650 "jsapi", rec=0) at src/switch_console.c:254 #10 0x00ec0e3a in console_thread (thread=0x8535980, obj=0x85358f8) at src/switch_console.c:454 #11 0x00f370d6 in dummy_worker (opaque=0x8535980) at threadproc/unix/thread.c:138 #12 0x00caf462 in start_thread () from /lib/i686/nosegneg/libpthread.so.0 #13 0x00c062ce in clone () from /lib/i686/nosegneg/libc.so.6 -------------- next part -------------- An embedded and charset-unspecified text was scrubbed... Name: bt_full.txt Url: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090227/6daf4869/attachment-0002.txt -------------- next part -------------- [root at localhost bin]# cat /proc/cpuinfo processor : 0 vendor_id : GenuineIntel cpu family : 15 model : 4 model name : Intel(R) Celeron(R) CPU 2.66GHz stepping : 9 cpu MHz : 2659.202 cache size : 256 KB fdiv_bug : no hlt_bug : no f00f_bug : no coma_bug : no fpu : yes fpu_exception : yes cpuid level : 5 wp : yes flags : fpu tsc msr pae mce cx8 apic mtrr mca cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe nx lm constant_tsc up pni monitor ds_cpl tm2 cid cx16 xtpr lahf_lm bogomips : 6651.17 [root at localhost bin]# -------------- next part -------------- [root at localhost bin]# uname -a Linux localhost.localdomain 2.6.18-53.el5xen #1 SMP Mon Nov 12 03:26:12 EST 2007 i686 i686 i386 GNU/Linux [root at localhost bin]# From gmaruzz at celliax.org Fri Feb 27 04:15:47 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Fri, 27 Feb 2009 13:15:47 +0100 Subject: [Freeswitch-users] snd_dummy setting for skype In-Reply-To: <59ad9ca10902261632s8a50903gba865a093c892bd9@mail.gmail.com> References: <59ad9ca10902261632s8a50903gba865a093c892bd9@mail.gmail.com> Message-ID: <7b197bef0902270415g6bd06a96sa3dc9c7b694babb5@mail.gmail.com> On Fri, Feb 27, 2009 at 1:32 AM, Henry Huang wrote: > I went through the wiki on mod_skypiax and see there should be a script to > make skype work without sound card in linux. Does anyone know where to > obtain that script to make sound work "without sound card"? Dear Henry, I apologize if the wiki page was not clear. snd-dummy is an ALSA driver (loadable module for the linux kernel) that you load like the other ALSA modules using the 'modprobe' command, no need at all to create an asound.conf file. You can find an example on how to load snd-dummy in the first lines of the script mod_skypiax/configs/startskype.sh I modified the wiki page, could you check is now clear? Thanks for reporting this, please continue to help us! Sincerely, Giovanni Maruzzelli ========================================= www.celliax.org via Pierlombardo 9, 20135 Milano Italy gmaruzz at celliax dot org Cell : +39-347-2665618 Fax : +39-02-87390039 On Fri, Feb 27, 2009 at 1:32 AM, Henry Huang wrote: > I went through the wiki on mod_skypiax and see there should be a script to > make skype work without sound card in linux. Does anyone know where to > obtain that script to make sound work "without sound card"? > > I am currently creating a /etc/asound.conf for skype to load the "fake" > sound driver. I do hear sound, but it's not perfect, it's very choppy and it > gives me error message when starting skype. The following is my asound.conf > setting. Hopefully someone can shed some light : > pcm.plugfile{ > ??? type plug > ??? slave { > ??????? pcm infile > ??????? format S16_LE > ??????? channels 1 > ??????? rate 16000 > ??? } > } > > pcm.infile { > ??? type file > ??? slave { > ??????? pcm null > ??? } > ??? file /dev/dsp > ??? infile /dev/dsp > } > > by using this configuration. skype spit out error messages as follow but > still works: > ALSA lib control.c:909:(snd_ctl_open_noupdate) Invalid CTL plugfile > ALSA lib control.c:909:(snd_ctl_open_noupdate) Invalid CTL infile > > > > > -- > Henry Huang > UniC Solution - Communication Unified > VoIP & Open Source software Consultant > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From rex.alex345 at yahoo.com Fri Feb 27 05:13:12 2009 From: rex.alex345 at yahoo.com (Rex_Alex) Date: Fri, 27 Feb 2009 05:13:12 -0800 (PST) Subject: [Freeswitch-users] ESL Wrapper In-Reply-To: <49A6E1DB.3070806@freeswitch.org> References: <558004.60211.qm@web59511.mail.ac4.yahoo.com> <78A818D3-6C4F-420D-A922-751A69E7E080@avgs.ca> <6CE58813-715A-43DB-877B-638B5CE7E6E9@freeswitch.org> <87f2f3b90902251342q1e393127ha7fbdbebf6d83dac@mail.gmail.com> <1235647534150-2389093.post@n2.nabble.com> <49A6E1DB.3070806@freeswitch.org> Message-ID: <1235740392995-2395557.post@n2.nabble.com> Hi All, I did what you have all suggested. Now its working perfectly. Thanks a lot for all your assistance. Rex. Raymond Chandler wrote: > > and it will probably be a good idea to do > make phpmod-install > so that the .so and .php files gets into the correct place to be included > > -Ray > > Mathieu Rene wrote: >> >> You need your distro's php dev pakage. >> On 26-Feb-09, at 6:25 AM, Rex_Alex wrote: >> >>> Hi All, I tried svn up && ./bootstrap.sh && ./configure && make >>> install and did Mathieu's suggestion but getting error as below, >>> [root at server esl]# make phpmod make MYLIB="../libesl.a" >>> SOLINK="-shared -Xlinker -x" >>> CFLAGS="-I/root/freeswitch-1.0.3/libs/esl/src/include -DHAVE_EDITLINE >>> -g -ggdb -I../../libs/libedit/src/ -fPIC -O2 -ffast-math -Wall >>> -Werror -Wunused-variable -Wwrite-strings -Wstrict-prototypes >>> -Wmissing-prototypes" >>> CXXFLAGS="-I/root/freeswitch-1.0.3/libs/esl/src/include >>> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC" >>> CXX_CFLAGS="" -C php make[1]: php-config: Command not found make[1]: >>> Entering directory `/root/freeswitch-1.0.3/libs/esl/php' g++ >>> -I/root/freeswitch-1.0.3/libs/esl/src/include -DHAVE_EDITLINE -g >>> -ggdb -I../../libs/libedit/src/ -fPIC -c esl_wrap.cpp -o esl_wrap.o >>> esl_wrap.cpp:717:18: error: zend.h: No such file or directory >>> esl_wrap.cpp:718:22: error: zend_API.h: No such file or directory >>> esl_wrap.cpp:719:17: error: php.h: No such file or directory >>> esl_wrap.cpp:973:21: error: php_ini.h: No such file or directory >>> esl_wrap.cpp:974:31: error: ext/standard/info.h: No such file or >>> directory esl_wrap.cpp:767: error: ?E_ERROR? was not declared in this >>> scope esl_wrap.cpp:788: error: ISO C++ forbids declaration of >>> ?ZEND_RSRC_DTOR_FUNC? with no type esl_wrap.cpp:788: error: >>> ?SWIG_landfill? was not declared in this scope esl_wrap.cpp:788: >>> error: expected ?,? or ?;? before ?{? token esl_wrap.cpp:793: error: >>> variable or field ?SWIG_ZTS_SetPointerZval? declared void >>> esl_wrap.cpp:793: error: ?zval? was not declared in this scope >>> esl_wrap.cpp:793: error: ?z? was not declared in this scope >>> esl_wrap.cpp:793: error: expected primary-expression before ?void? >>> esl_wrap.cpp:793: error: expected primary-expression before ?*? token >>> esl_wrap.cpp:793: error: ?type? was not declared in this scope >>> esl_wrap.cpp:793: error: expected primary-expression before ?int? >>> esl_wrap.cpp:793: error: initializer expression list treated as >>> compound expression esl_wrap.cpp:793: error: expected ?,? or ?;? >>> before ?{? token make[1]: *** [esl_wrap.o] Error 1 make[1]: Leaving >>> directory `/root/freeswitch-1.0.3/libs/esl/php' make: *** [phpmod] >>> Error 2 [root at server esl]# Please tell me where am i wrong? Thanks, Rex >>> >>> mercutioviz wrote: >>> On Wed, Feb 25, 2009 at 11:34 AM, Brian West wrote: > If he's on >>> 1.0.3 I don't think it has php in it.. Can't he do the whole >>> bootstrap process? svn up && ./bootstrap.sh && ./configure && >>> make install And then do Mathieu's suggestion? -MC >>> _______________________________________________ Freeswitch-users >>> mailing list Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> ------------------------------------------------------------------------ >>> View this message in context: Re: ESL Wrapper >>> >>> Sent from the freeswitch-users mailing list archive >>> at Nabble.com. >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://n2.nabble.com/ESL-Wrapper-tp2385651p2395557.html Sent from the freeswitch-users mailing list archive at Nabble.com. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090227/c650d3a7/attachment-0002.html From anthony.minessale at gmail.com Fri Feb 27 05:58:29 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 27 Feb 2009 07:58:29 -0600 Subject: [Freeswitch-users] change session-timer value In-Reply-To: <49A77188.4070105@telkom.co.id> References: <49A77188.4070105@telkom.co.id> Message-ID: <191c3a030902270558q2a6e1f55v3b6d118d45ec00e1@mail.gmail.com> Didn't you already start a thread with this same question yesterday? If it doesn't work open a jira http://jira.freeswitch.org under the sofia sip category and we will give it to the sofia developer to look at. On Thu, Feb 26, 2009 at 10:52 PM, mashudi wrote: > Hi Folks , > I have problem to change the session timer value that automatically > created by Freeswitch for 120, I can change the value of session-timer > in /sip_profiles/internal.xml > > > > become > > > > and it work only for INVITE message response, > but for UPDATE message response still use default value of 120, > please help me to solved this problem. > Thank you in advanced. > regards > > mashudi > > pls help me to figure out what i'm doing wrong. > > > > > > > > ***************************************** > Sekarang Gratis Nelpon SLJJ Flexi diperluas ke > Yogya > ***************************************** > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090227/3948338f/attachment-0002.html From helmut.kuper at ewetel.de Fri Feb 27 06:07:15 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Fri, 27 Feb 2009 15:07:15 +0100 Subject: [Freeswitch-users] Problems with record_stereo Message-ID: <49A7F393.6080406@ewetel.de> Hello, I play around with record_session and would like to have caller and callee separated on left and right channel. I found record_stereo is used for this. Unfortunately it doesn't work. A and B leg are still mixed. Additionally I found that B leg is significant louder than A leg, but both legs were local extensions. My Dialplan looks like this: regards helmut From anthony.minessale at gmail.com Fri Feb 27 06:08:54 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 27 Feb 2009 08:08:54 -0600 Subject: [Freeswitch-users] session orignate freeswitch 1.0.3 segmentation error In-Reply-To: <2ea4d47e0902262211y2e4b243ey52eb7bda04329dc7@mail.gmail.com> References: <2ea4d47e0902260711m21fe581ge3a12288808359fa@mail.gmail.com> <2ea4d47e0902262211y2e4b243ey52eb7bda04329dc7@mail.gmail.com> Message-ID: <191c3a030902270608i2e6988d1ld53aa3e3de1b9bd8@mail.gmail.com> why are we doing this on the mailing list. This info belongs in a jira ticket. Please reproduce this issue with SVN trunk and if it persists, report it on http://jira.freeswitch.org 2009/2/27 Gopal krishnan > Hi Brian, > Please find the attached backtrace files attached. > And > 1. SVN revision number (or binary file) - FreeSWITCH Version 1.0.3 > (exported) > 2. Operating System and revision - CentOS 5.2 > 3. Hardware information - 32 bit with 512 MB RAM > 4. I am using Event socket > 5. Language - Javascript > > One more thing in the same machine earlier I was using freeswitch 1.0.2. > when the segmentation fault happens the core file was generated in the older > version freeswitch bin. In 1.0.3 there is no bin directory. Is that could be > the prob? > > On Thu, Feb 26, 2009 at 9:05 PM, Brian West wrote: > >> can you include the backtrace? We might have already fixed this one. >> http://wiki.freeswitch.org/wiki/Reporting_Bugs >> >> /b >> >> >> On Feb 26, 2009, at 9:11 AM, Gopalakrishnan A.N wrote: >> >> Hi, >> >> I have installed Freeswitch 1.0.3. I am using event socket with >> Javascript. When I try to dial the script with below command, the call is >> not going thru it seems to be idle. and segmentation fault core dump error, >> (freeswitch hangs).....<323.gif> >> >> >> new_session = new Session.originate(session, >> "sofia/default/@foo.com"); >> bridge(session, new_session); >> >> I saw in the wiki >> http://wiki.freeswitch.org/wiki/FreeSwitch_Javascript_Session >> that the session is depreciated, earlier I was using like this in >> Freeswitch 1.0.2, it works fine....:) >> >> session = new Session(); >> session.originate(session, >> "{ignore_early_media=true}sofia/default/@foo.com"); >> >> So something I am missing, please let me know where I am wrong? >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Thank you with regards, > Gopal, > PeopleTech Systems Private Limited > www.peopletech.co.in > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090227/5b47f62b/attachment-0002.html From mashudiflexi at telkom.co.id Fri Feb 27 08:29:50 2009 From: mashudiflexi at telkom.co.id (mashudi) Date: Fri, 27 Feb 2009 23:29:50 +0700 Subject: [Freeswitch-users] change session-timer value In-Reply-To: <191c3a030902270558q2a6e1f55v3b6d118d45ec00e1@mail.gmail.com> References: <49A77188.4070105@telkom.co.id> <191c3a030902270558q2a6e1f55v3b6d118d45ec00e1@mail.gmail.com> Message-ID: <49A814FE.402@telkom.co.id> Dear Anthony Minessale, As information, I try install to Freeswitch as inbound conference server and integrated with the Huawei MSCe, I got trouble with the session-timer as response message UPDATE that send by Huawei MSC. Because the session timer value below the MSCe specification, MSCe send bye message after the message UPDATE response from Freeswitch. could you give me the guidance how to solve this? thank you in advanced for your kind support, regards, mashudi Anthony Minessale wrote: > Didn't you already start a thread with this same question yesterday? > > If it doesn't work open a jira http://jira.freeswitch.org under the > sofia sip category and we will > give it to the sofia developer to look at. > > > On Thu, Feb 26, 2009 at 10:52 PM, mashudi > wrote: > > Hi Folks , > I have problem to change the session timer value that automatically > created by Freeswitch for 120, I can change the value of session-timer > in /sip_profiles/internal.xml > > > > become > > > > and it work only for INVITE message response, > but for UPDATE message response still use default value of 120, > please help me to solved this problem. > Thank you in advanced. > regards > > mashudi > > pls help me to figure out what i'm doing wrong. > > > > > > > > ***************************************** > Sekarang Gratis Nelpon SLJJ Flexi diperluas ke > Yogya > ***************************************** > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > ***************************************** > Sekarang Gratis Nelpon SLJJ Flexi diperluas ke > Yogya > ***************************************** > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > ***************************************** Sekarang Gratis Nelpon SLJJ Flexi diperluas ke Yogya ***************************************** From brian at freeswitch.org Fri Feb 27 08:29:08 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 27 Feb 2009 10:29:08 -0600 Subject: [Freeswitch-users] change session-timer value In-Reply-To: <49A814FE.402@telkom.co.id> References: <49A77188.4070105@telkom.co.id> <191c3a030902270558q2a6e1f55v3b6d118d45ec00e1@mail.gmail.com> <49A814FE.402@telkom.co.id> Message-ID: Please open a ticket on http://jira.freeswitch.org, as per his last email. Attach all the information to explain the bug along with sip traces. Attach the info, DO NOT paste the logs or traces in the comment box. Attachments are much easier to download and read for us. /b On Feb 27, 2009, at 10:29 AM, mashudi wrote: > Dear Anthony Minessale, > As information, I try install to Freeswitch as inbound conference > server > and integrated with the Huawei MSCe, I got trouble with the > session-timer as response message UPDATE that send by Huawei MSC. > Because the session timer value below the MSCe specification, MSCe > send > bye message after the message UPDATE response from Freeswitch. > > could you give me the guidance how to solve this? > thank you in advanced for your kind support, > regards, > > mashudi From freeswitch at servercorps.com Fri Feb 27 09:17:04 2009 From: freeswitch at servercorps.com (Addison Martin) Date: Fri, 27 Feb 2009 11:17:04 -0600 Subject: [Freeswitch-users] Console messages In-Reply-To: References: Message-ID: <92e7d2090902270917t1f7865f9r8245b56934b0d70c@mail.gmail.com> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_log I'd set the log level to CONSOLE (0), then do: On Thu, Feb 26, 2009 at 4:55 PM, Nik Middleton wrote: > Hi Guys, > > > > Is there a way of displaying a console message not related to a log level? > I?ve got the console only reporting errors now, but it would be nice to be > able to display a message when a given condition exists.? Yes, I could set > it as an error level message, but I?d rather not do that. > > > > Regards, > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From freeswitch at servercorps.com Fri Feb 27 09:41:59 2009 From: freeswitch at servercorps.com (Addison Martin) Date: Fri, 27 Feb 2009 11:41:59 -0600 Subject: [Freeswitch-users] Thin Client VOIP setup? In-Reply-To: <3d381e170902251311i3d3a4205j117d472228c30219@mail.gmail.com> References: <3d381e170902251311i3d3a4205j117d472228c30219@mail.gmail.com> Message-ID: <92e7d2090902270941o76e68288q246009167323e7ac@mail.gmail.com> With a little bit of scripting + web programming, click to call is readily available with FreeSWITCH. Check out call.php in scripts/ under the freeswitch source directory. The code in that php file can be rolled into your CRM. You'll then need to set a variable based on the user's login that is his/her extension. When he clicks the link, the script will dial the user that clicked, and the number he clicked, and connect the two together. Hope this helps. nik On Wed, Feb 25, 2009 at 3:11 PM, Erik Wickstrom wrote: > Hi, > > I've deployed Freeswitch as our phone system at work.? We now want to use > our new phonesystem in a phone room with thin clients (Terminal Server, > possibly LTSP) for each agent.? Ideally, we'd like to use x-lite or another > softphone for each agent. > > The desired workflow for the agents is as follows: > 1) A web based CRM with click to dial. (and customer data card etc) > 2) Agent clicks dial button and is connected to customer > 3) Interact with CRM... > > From what I've read so far, there are some challenges that need to be > overcome in deploying softphones over thin clients. > > Has anyone here had any success in setting up a system like this?? I'm I > asking for trouble trying to use softphones with thin clients (should I just > use hardware phones?? Do they support click to dial?) > > Thanks! > Erik > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From freeswitch at servercorps.com Fri Feb 27 11:20:33 2009 From: freeswitch at servercorps.com (Addison Martin) Date: Fri, 27 Feb 2009 13:20:33 -0600 Subject: [Freeswitch-users] Thin Client VOIP setup? In-Reply-To: <92e7d2090902270941o76e68288q246009167323e7ac@mail.gmail.com> References: <3d381e170902251311i3d3a4205j117d472228c30219@mail.gmail.com> <92e7d2090902270941o76e68288q246009167323e7ac@mail.gmail.com> Message-ID: <92e7d2090902271120nbe5688di71736b30c96c40dc@mail.gmail.com> I forgot to add, this will work for ANY SIP UA, not just soft phones. nik On Fri, Feb 27, 2009 at 11:41 AM, Addison Martin wrote: > With a little bit of scripting + web programming, click to call is > readily available with FreeSWITCH. ?Check out call.php in scripts/ > under the freeswitch source directory. The code in that php file can > be rolled into your CRM. ? You'll then need to set a variable based on > the user's login that is his/her extension. ?When he clicks the link, > the script will dial the user that clicked, and the number he clicked, > and connect the two together. > > Hope this helps. > > nik > > > On Wed, Feb 25, 2009 at 3:11 PM, Erik Wickstrom wrote: >> Hi, >> >> I've deployed Freeswitch as our phone system at work.? We now want to use >> our new phonesystem in a phone room with thin clients (Terminal Server, >> possibly LTSP) for each agent.? Ideally, we'd like to use x-lite or another >> softphone for each agent. >> >> The desired workflow for the agents is as follows: >> 1) A web based CRM with click to dial. (and customer data card etc) >> 2) Agent clicks dial button and is connected to customer >> 3) Interact with CRM... >> >> From what I've read so far, there are some challenges that need to be >> overcome in deploying softphones over thin clients. >> >> Has anyone here had any success in setting up a system like this?? I'm I >> asking for trouble trying to use softphones with thin clients (should I just >> use hardware phones?? Do they support click to dial?) >> >> Thanks! >> Erik >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > From chavpaskov at shaw.ca Fri Feb 27 14:27:08 2009 From: chavpaskov at shaw.ca (Tchavdar Paskov) Date: Fri, 27 Feb 2009 14:27:08 -0800 Subject: [Freeswitch-users] Variables from failed call to be exported to a a new B leg In-Reply-To: <797DE9FB-7B89-4FD1-A90B-2A09EDB96831@freeswitch.org> References: <797DE9FB-7B89-4FD1-A90B-2A09EDB96831@freeswitch.org> Message-ID: Worked like charm. Thanks everyone for your support. Regards Chav ----- Original Message ----- From: Brian West Date: Thursday, February 26, 2009 1:47 pm Subject: Re: [Freeswitch-users] Variables from failed call to be exported to a a new B leg To: freeswitch-users at lists.freeswitch.org > You then don't use the export command...? you set the > variable application="set" data=""export_vars=sip_hangup_phrase"/> > > /b > > On Feb 26, 2009, at 3:18 PM, Tchavdar Paskov wrote: > > > isn't this already done because of the info application > called? > > before ? > > Chav > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090227/891e76e1/attachment-0002.html From brian at freeswitch.org Fri Feb 27 14:30:58 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 27 Feb 2009 16:30:58 -0600 Subject: [Freeswitch-users] Variables from failed call to be exported to a a new B leg In-Reply-To: References: <797DE9FB-7B89-4FD1-A90B-2A09EDB96831@freeswitch.org> Message-ID: <755E463C-C0DE-4F0F-9F7E-7F3C582AC25E@freeswitch.org> Don't run off so fast. ;) You should join us on IRC... I would like to see the IRC numbers over 200 soon ;) /b On Feb 27, 2009, at 4:27 PM, Tchavdar Paskov wrote: > Worked like charm. > Thanks everyone for your support. > Regards > Chav From chavpaskov at shaw.ca Fri Feb 27 14:34:42 2009 From: chavpaskov at shaw.ca (Tchavdar Paskov) Date: Fri, 27 Feb 2009 14:34:42 -0800 Subject: [Freeswitch-users] Use XML dialplan and mod_perl Message-ID: Hi, i have a quick question. is it possible? to use both? XML dial plan? and mod_perl? together. examlpe: default.xml - used as default context mod_perl? - used to generate the public context If it is possible how i have to set perl.conf.xml? and especially? xml-handler-bindings ? is it possible? in value="dialplan"? to specify the? name of the context? Regards Chav -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090227/1e3f09c1/attachment-0002.html From chavpaskov at shaw.ca Fri Feb 27 14:36:19 2009 From: chavpaskov at shaw.ca (Tchavdar Paskov) Date: Fri, 27 Feb 2009 14:36:19 -0800 Subject: [Freeswitch-users] Variables from failed call to be exported to a a new B leg In-Reply-To: <755E463C-C0DE-4F0F-9F7E-7F3C582AC25E@freeswitch.org> References: <797DE9FB-7B89-4FD1-A90B-2A09EDB96831@freeswitch.org> <755E463C-C0DE-4F0F-9F7E-7F3C582AC25E@freeswitch.org> Message-ID: Just logging into IRC Regards Chav ----- Original Message ----- From: Brian West Date: Friday, February 27, 2009 2:31 pm Subject: Re: [Freeswitch-users] Variables from failed call to be exported to a a new B leg To: freeswitch-users at lists.freeswitch.org > Don't run off so fast.? ;)? You should join us on > IRC... I would like? > to see the IRC numbers over 200 soon ;) > > /b > > On Feb 27, 2009, at 4:27 PM, Tchavdar Paskov wrote: > > > Worked like charm. > > Thanks everyone for your support. > > Regards > > Chav > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090227/965f1f01/attachment-0002.html From msc at freeswitch.org Fri Feb 27 16:24:22 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 27 Feb 2009 16:24:22 -0800 Subject: [Freeswitch-users] Use XML dialplan and mod_perl In-Reply-To: References: Message-ID: <87f2f3b90902271624n25fee20ere016fd04372f3e5e@mail.gmail.com> > Hi, > i have a quick question. > is it possible? to use both? XML dial plan? and mod_perl? together. > > examlpe: > > default.xml - used as default context > mod_perl? - used to generate the public context > > If it is possible how i have to set perl.conf.xml? and especially > xml-handler-bindings ? > > is it possible? in value="dialplan"? to specify the? name of the context? > > Regards > Chav What's your IRC nick? We can discuss it more there. -MC From Prometheus001 at gmx.net Sat Feb 28 04:18:54 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Sat, 28 Feb 2009 13:18:54 +0100 Subject: [Freeswitch-users] pocketsphinx and event socket Message-ID: <49A92BAE.4090907@gmx.net> Hello, I have tried the pizza demo and did it get to work so far. However I would like to use pocketsphinx through event socket. I saw in the wiki that there is a chapter for Speech Synthesis Commands: http://wiki.freeswitch.org/wiki/Mod_commands#Speech_Synthesis_Commands However this is empty. Also http://wiki.freeswitch.org/wiki/ASR didn't give me a hint. As I am not a Java programmer, it's hard for me to determine how the pizza demo actually works. Anybody has a sample how he did it e.g. in Php/Perl or so? (I am working with Ruby) Or back to the basics: Is it possible to use pocketsphinx through event socket? Best regards Peter From codecomplete at free.fr Sat Feb 28 07:47:46 2009 From: codecomplete at free.fr (Fred) Date: Sat, 28 Feb 2009 16:47:46 +0100 Subject: [Freeswitch-users] SIP server? PBX vs. softswitch? Message-ID: <7.0.1.0.2.20090228120132.0285f8f8@fredshack.com> Hello Even though I successfully set up an Asterisk voice server, I'm no telecom expert, and would like some clarification about the following things: - What is an SIP server as opposed to a IP PBX? - What is the different between a PBX like Asterisk and a softswitch? Thank you. From e.schmidbauer at gmail.com Sat Feb 28 08:01:36 2009 From: e.schmidbauer at gmail.com (e schmidbauer) Date: Sat, 28 Feb 2009 11:01:36 -0500 Subject: [Freeswitch-users] freeswitch with celt on windows Message-ID: <2cef777b0902280801o7380d81kf00be14f43c304a9@mail.gmail.com> Hi. I was wondering if it is possible to compile freeswitch with the celt codec on windows? I have been able to compile celt for windows but when i compiled freeswitch i did not see the mod_celt in the solution explorer. If this is possible could someone point me in the right direction. Thank you. From brian at freeswitch.org Sat Feb 28 08:10:57 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 28 Feb 2009 10:10:57 -0600 Subject: [Freeswitch-users] pocketsphinx and event socket In-Reply-To: <49A92BAE.4090907@gmx.net> References: <49A92BAE.4090907@gmx.net> Message-ID: You can accomplish this .... here is an example using ESL in perl http://fisheye.freeswitch.org/browse/FreeSWITCH/libs/esl/perl/server3.pl?r=12344 /b On Feb 28, 2009, at 6:18 AM, Peter P GMX wrote: > Or back to the basics: Is it possible to use pocketsphinx through > event > socket? From codecomplete at free.fr Sat Feb 28 08:30:07 2009 From: codecomplete at free.fr (Fred) Date: Sat, 28 Feb 2009 17:30:07 +0100 Subject: [Freeswitch-users] Using OpenZAP + FXO card just to get CID info? Message-ID: <7.0.1.0.2.20090228172749.027ad548@fredshack.com> Hello I'd like to write a single-host CRM application, so I need to get the CallerID information when a call comes in. I don't actually need a PBX/softswitch. The user will have the FXO cards and a phoneset connected on the same line, and will pick up the phone once the CRM application has picked up the CID info and popped up a dialog box, etc. Is it possible to just use OpenZAP to get this information, or must I install Freeswitch and provide an IP phone as well? Thank you. From mrene_lists at avgs.ca Sat Feb 28 08:32:18 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Sat, 28 Feb 2009 11:32:18 -0500 Subject: [Freeswitch-users] Using OpenZAP + FXO card just to get CID info? In-Reply-To: <7.0.1.0.2.20090228172749.027ad548@fredshack.com> References: <7.0.1.0.2.20090228172749.027ad548@fredshack.com> Message-ID: OpenZAP is just a module accessing the card, you need to use it within freeswitch. Then, you can use event socket to get the callerid. On 28-Feb-09, at 11:30 AM, Fred wrote: > Hello > > I'd like to write a single-host CRM application, so I need to get the > CallerID information when a call comes in. I don't actually need a > PBX/softswitch. The user will have the FXO cards and a phoneset > connected on the same line, and will pick up the phone once the CRM > application has picked up the CID info and popped up a dialog box, > etc. > > Is it possible to just use OpenZAP to get this information, or must I > install Freeswitch and provide an IP phone as well? > > Thank you. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Sat Feb 28 08:34:16 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 28 Feb 2009 10:34:16 -0600 Subject: [Freeswitch-users] freeswitch with celt on windows In-Reply-To: <2cef777b0902280801o7380d81kf00be14f43c304a9@mail.gmail.com> References: <2cef777b0902280801o7380d81kf00be14f43c304a9@mail.gmail.com> Message-ID: <5006814D-09D1-40EB-92C4-CD76AD574ECE@freeswitch.org> Someone just needs to do the work of adding it to the build... I thought Carlos did this already... Are you on SVN Trunk? /b On Feb 28, 2009, at 10:01 AM, e schmidbauer wrote: > Hi. I was wondering if it is possible to compile freeswitch with the > celt codec on windows? I have been able to compile celt for windows > but when i compiled freeswitch i did not see the mod_celt in the > solution explorer. If this is possible could someone point me in the > right direction. Thank you. From brian at freeswitch.org Sat Feb 28 08:37:07 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 28 Feb 2009 10:37:07 -0600 Subject: [Freeswitch-users] SIP server? PBX vs. softswitch? In-Reply-To: <7.0.1.0.2.20090228120132.0285f8f8@fredshack.com> References: <7.0.1.0.2.20090228120132.0285f8f8@fredshack.com> Message-ID: It depends on how you look at it... most will say there is no difference... but last I checked you usually don't run heavy apps on a softswitch. FreeSWITCH can be everything from softphone to softswitch and everything in between including PBX. The default config comes configured as a PBX. /b On Feb 28, 2009, at 9:47 AM, Fred wrote: > Hello > > Even though I successfully set up an Asterisk voice server, I'm no > telecom expert, and would like some clarification about the > following things: > - What is an SIP server as opposed to a IP PBX? > - What is the different between a PBX like Asterisk and a softswitch? > > Thank you. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090228/9aa8feab/attachment-0002.html From e.schmidbauer at gmail.com Sat Feb 28 08:55:25 2009 From: e.schmidbauer at gmail.com (e schmidbauer) Date: Sat, 28 Feb 2009 11:55:25 -0500 Subject: [Freeswitch-users] freeswitch with celt on windows In-Reply-To: <5006814D-09D1-40EB-92C4-CD76AD574ECE@freeswitch.org> References: <2cef777b0902280801o7380d81kf00be14f43c304a9@mail.gmail.com> <5006814D-09D1-40EB-92C4-CD76AD574ECE@freeswitch.org> Message-ID: <2cef777b0902280855sa804542g71d8d6d46e879f10@mail.gmail.com> i used the nightly snapshot... http://files.freeswitch.org/freeswitch-snapshot.tar.gz On Sat, Feb 28, 2009 at 11:34 AM, Brian West wrote: > Someone just needs to do the work of adding it to the build... I > thought Carlos did this already... Are you on SVN Trunk? > > /b > > On Feb 28, 2009, at 10:01 AM, e schmidbauer wrote: > >> Hi. I was wondering if it is possible to compile freeswitch with the >> celt codec on windows? I have been able to compile celt for windows >> but when i compiled freeswitch i did not see the mod_celt in the >> solution explorer. If this is possible could someone point me in the >> right direction. Thank you. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From dyfet at gnutelephony.org Sat Feb 28 09:20:53 2009 From: dyfet at gnutelephony.org (David Sugar) Date: Sat, 28 Feb 2009 12:20:53 -0500 Subject: [Freeswitch-users] SIP server? PBX vs. softswitch? In-Reply-To: References: <7.0.1.0.2.20090228120132.0285f8f8@fredshack.com> Message-ID: <49A97275.6000007@gnutelephony.org> Where this is distinguished, it is not directly at the level that user's experience the end result. In the case of what is called a "softswitch", one answer is found in organizations like the ISC (International Softswitch Consortium) and vendors who built products around their architecture recommendations. These systems tend to be very complex and componetized, where basic functionality operates in self-contained components that then interact with the whole through defined open standards and network protocols, such as SIP. The primary reason for ISC-style architectures is a result of proprietary development, where code and internal operations cannot be shared or modified. Hence, by breaking up functionality into subcomponents, it is possible to replace a component subsystem as a whole while retaining the interfaces. A perfect example is call forwarding. In a "traditional" proprietary (ISC-model) softswitch, call forwarding would be an entirely separate self-contained proprietary "feature" server interacting over SIP. If someone wants to create a different call forwarding behavior, one slips in an alternate server. By contrast, it is far easier in an open source/free software PBX to simply modify the feature code that implements call forwarding directly to create new and specialized versions of that feature. Hence, you do not find or have need for micro-services for tiny features in pbx software that originated as open source and free software or that did not follow the path of proprietary architectures, such as Bayonne, Asterisk, or FreeSwitch. A perfect example of a traditional "softswitch" architecture is SipX, which originated as a proprietary VoIP pbx codebase. However, even at this point, such distinctions I think are still somewhat artificial, as Brian suggests. What does distinguish architectures that may be relevant to end users is whether a IP-PBX solution operates as a B2BUA (back-to-back user agent) or not. A pure B2BUA solution is one where all media as well as signalling goes directly through the central PBX switch. A perfect example of this is how Asterisk traditionally works. This makes it very easy to adapt and connect multi-protocol endpoints, to convert media formats for endpoints who do not have common codecs, etc, since all media endpoints talk to the switch rather than each other. However, since all media goes through a central point, the scalability of such systems can often become "compute-bound", and extra latency is induced. A "pure" network solution by contrast has all media connect directly peer to peer by the user agent endpoints, and the "pbx" really only handles and coordinate independently operating endpoints through signalling. This often requires separate servers for gateways to the PSTN or other protocols. But it does offer better latency and scalability, and the ability to provide end-to-end media security, such as when using ZRTP. This difference, between B2BUA and non-B2BUA, is I think far more relevant today than traditional classifications such as IP-PBX, softswitch, "SIP Server", etc. Brian West wrote: > It depends on how you look at it... most will say there is no > difference... but last I checked you usually don't run heavy apps on a > softswitch. > > FreeSWITCH can be everything from softphone to softswitch and everything > in between including PBX. The default config comes configured as a PBX. > > /b > > On Feb 28, 2009, at 9:47 AM, Fred wrote: > >> Hello >> >> Even though I successfully set up an Asterisk voice server, I'm no >> telecom expert, and would like some clarification about the following >> things: >> - What is an SIP server as opposed to a IP PBX? >> - What is the different between a PBX like Asterisk and a softswitch? >> >> Thank you. > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- A non-text attachment was scrubbed... Name: dyfet.vcf Type: text/x-vcard Size: 177 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090228/dd828db8/attachment-0002.vcf From nik.middleton at noblesolutions.co.uk Sat Feb 28 14:49:39 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Sat, 28 Feb 2009 22:49:39 -0000 Subject: [Freeswitch-users] Orginate: getting status of call fail Message-ID: Hi Guys, I've been running a test script written in lua which now works very well thanks to Anthony's fix to stream file. Right now I'm using an event socket to initiate the call and passing the name of the script along with originate thus: $dialstring = "originate {ignore_early_media=true,origination,originate_timeout=25}sofia/gateway/ Mygw/phonenum '&lua(helloworld.lua )'"; $result = $obj ->bgapi_command($dialstring); The script gets fired (it would appear) on answer. However, if the number is invalid , timed out or was busy, I'm not sure the script gets executed or am I wrong? I want to be able to fire an event back on what happed to the call in the event that it failed for whatever reason. I know I can simply call the originate and pass the number as an argument and execute the dial within the script but I'm led to believe that's not very efficient, or am I completely wrong? Looking for the most FS friendly way here Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090228/3b813f25/attachment-0002.html From davidwdan at gmail.com Sat Feb 28 11:33:53 2009 From: davidwdan at gmail.com (David Dan) Date: Sat, 28 Feb 2009 14:33:53 -0500 Subject: [Freeswitch-users] Problems loading mod_spidermonkey_curl Message-ID: <65bd1c9f0902281133t2d79806bm63461288e4ca0c0f@mail.gmail.com> I'm getting the following error when I try to load the mod_spidermonkey_curl module. I didn't get any errors when I compiled it. I also tried --without-libcurl but I got the same result. Any help would be appreciated. freeswitch at internal> load mod_spidermonkey_curl -ERR [module load file routine returned an error] 2009-02-28 14:17:54 [CRIT] switch_loadable_module.c:839 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_spidermonkey_curl.so **/usr/local/freeswitch/mod/mod_spidermonkey_curl.so: undefined symbol: mod_spidermonkey_curl_module_interface** freeswitch at internal> version FreeSWITCH Version 1.0.3 (exported) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090228/68f593fc/attachment-0002.html From brian at freeswitch.org Sat Feb 28 17:57:40 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 28 Feb 2009 19:57:40 -0600 Subject: [Freeswitch-users] Problems loading mod_spidermonkey_curl In-Reply-To: <65bd1c9f0902281133t2d79806bm63461288e4ca0c0f@mail.gmail.com> References: <65bd1c9f0902281133t2d79806bm63461288e4ca0c0f@mail.gmail.com> Message-ID: <885EAAC5-FDF0-4DA3-A835-EC652EDC2368@freeswitch.org> please open up spidermonkey.conf.xml and add it to the load there... its a sub module of mod_spidermonkey so you can't load it at the CLI /b On Feb 28, 2009, at 1:33 PM, David Dan wrote: > I'm getting the following error when I try to load the > mod_spidermonkey_curl module. I didn't get any errors when I > compiled it. I also tried --without-libcurl but I got the same > result. Any help would be appreciated. > > freeswitch at internal> load mod_spidermonkey_curl > -ERR [module load file routine returned an error] > > 2009-02-28 14:17:54 [CRIT] switch_loadable_module.c:839 > switch_loadable_module_load_file() Error Loading module /usr/local/ > freeswitch/mod/mod_spidermonkey_curl.so > **/usr/local/freeswitch/mod/mod_spidermonkey_curl.so: undefined > symbol: mod_spidermonkey_curl_module_interface** From davidwdan at gmail.com Sat Feb 28 20:08:46 2009 From: davidwdan at gmail.com (David Dan) Date: Sat, 28 Feb 2009 23:08:46 -0500 Subject: [Freeswitch-users] Problems loading mod_spidermonkey_curl In-Reply-To: <885EAAC5-FDF0-4DA3-A835-EC652EDC2368@freeswitch.org> References: <65bd1c9f0902281133t2d79806bm63461288e4ca0c0f@mail.gmail.com> <885EAAC5-FDF0-4DA3-A835-EC652EDC2368@freeswitch.org> Message-ID: <65bd1c9f0902282008o3d6b721bu4735c26d21e61d59@mail.gmail.com> That did it. Thank you On 2/28/09, Brian West wrote: > please open up spidermonkey.conf.xml and add it to the load there... > its a sub module of mod_spidermonkey so you can't load it at the CLI > > /b > > On Feb 28, 2009, at 1:33 PM, David Dan wrote: > >> I'm getting the following error when I try to load the >> mod_spidermonkey_curl module. I didn't get any errors when I >> compiled it. I also tried --without-libcurl but I got the same >> result. Any help would be appreciated. >> >> freeswitch at internal> load mod_spidermonkey_curl >> -ERR [module load file routine returned an error] >> >> 2009-02-28 14:17:54 [CRIT] switch_loadable_module.c:839 >> switch_loadable_module_load_file() Error Loading module /usr/local/ >> freeswitch/mod/mod_spidermonkey_curl.so >> **/usr/local/freeswitch/mod/mod_spidermonkey_curl.so: undefined >> symbol: mod_spidermonkey_curl_module_interface** > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Sent from my mobile device