- FreeSWITCH-users - FreeSWITCH Mailing Lists

User who answer the bridge in a execute_answer
by Albano Daniele Salvatore - Lavoro 24 Nov '09

24 Nov '09

Hi, i'm writing some dialplan parts that get executed on execute_on_answer. In this dialplan that get executed i need to make a directory to handle recordings for record_session and my folder structure is: USER/YEAR/MONTH/HOUR-MINUTE-SECOND-CALLER_NUMBER.wav ------ <action application="system" data="mkdir -p $${base_dir}/recordings/${sip_from_user}/${strftime(%Y)}/${strftime(%m)}/" /> <action application="bind_meta_app" data="1 a s record_session::$${base_dir}/recordings/${sip_from_user}/${strftime(%Y)}/${strftime(%m)}/${strftime(%H_%M_%S)}-${caller_id_number}.wav" /> ------ The call flow is: Call from external -> IVR -> Transfer to Group -> Execute on Answer -> system/bind_meta_app Pratically, i need the number (or better the user) that answered the call: what variable should i check? I tried with sip_from_user, callee_id_number and some other. Thank for your help, Best Regards, Daniele

2 2

DTMF javasript
by Baskar 24 Nov '09

24 Nov '09

* Hi,* * * *I want to check value given to the javascript with conditions whether it is voicefile, extension or mobile Number when i press the dtmf value.* * * *Steps i need to check in javascript:* * * *When i Press the DTMF value 1 it should check the 3 condition* * * If the Value for argv[2]=vfsurya means it is a voice file so it should play the Voice file *If the Value for argv[2]=1001 means it is a extension. The call should Bridge the extension* *If the Value for argv[2]=9841799874 means it is a Mobile number. The call should Bridge the Mobile number* * * *var exit = false;* *var dtmf_digits = "";* *var repeat = 0;* *var argv[2]=vfsurya; // or var argv[2]=1001 or var argv[2]=Mobile Number* * * * * *function onInput( session, type, data, arg ) * *{* * if ( type == "dtmf" ) * * {* * console_log( "info", "Got digit " + data.digit + "\n" );* * if ( data.digit == "1" ) * * **{* * if(argv[2].startswith("vf"))* * **{* * **var voice2=voice.substring(2)+"<br />"* * **session.streamFile("/usr/local/freeswitch/sounds/en/us/callie/"+voice2+".wav", onInput );* * **}* * **else if(argv[2].length==4)* * **{* * **console_log( "info", "Got voicefile " + argv[2] + "\n" );* * **session.execute("bridge", "sofia/internal/"+argv[2]+"%192.168.1.2", onInput ); * * **}* * **else* * **{* * **session.execute("bridge", "sofia/default/sip:"+argv[2]+"@ 192.168.1.135:5066", onInput ); * * **}* * }* * }* *}* * * *But if 1 is pressed there is no event trigger but it get the dtmf value as 1 in freeswitch console. * * * *can any one specify what is the error or correct me where i am wrong.* * -- Thanks with Regards, N.Baskar *

2 1

os x compile failure
by James Budge 24 Nov '09

24 Nov '09

The compile fails after this. i686-apple-darwin10-gcc-4.2.1: -bundle not allowed with -dynamiclib

3 4

Problems with Voicemail
by Peter P GMX 24 Nov '09

24 Nov '09

Hello, i have a couple of problems with voicemail. Voicemails are recorded but not played in any way. 1) when I call my voicemail, I can hear the number of new messages, but I canot not hear the recorded files itself. I hear the following * "You have 1 urgent new message in forder inbox" * "You have 7 new messages in forder inbox" * "New message number 1 Jan 011970 at 1 am" * (message is NOT played) * "You have 1 urgent new message in forder inbox" * "You have 7 new messages in forder inbox" * "press 1 to listen, press 2...." * (I press 1) * "New message number 1 Jan 011970 at 1 am" * (message is NOT played) * "You have 1 urgent new message in forder inbox" * "You have 7 new messages in forder inbox" * ... The voicemail files are stored in the file system as wav files and I can play them manually from the file system - so there is sound inside. 2) Another strange thing is that all recorded calls are announced with a date of 01.Jan.1970 although the databse shows correct values. 3) Alternatively playing it on the web Gui on http://fs.ip:8080/api/voicemail/web doesn't work either. Date is again 01.Jan.1970 and shown length of the file is always 00:00:00, although the database shows the correct number of seconds 4) Just to note that whenever I expect a recorded file to be played I see the following on the console 2009-11-21 00:17:02.511110 [ERR] mod_native_file.c:68 Error opening /usr/local/freeswitch/sounds/en/us/callie/inbox.PCMA In my installation Freeswitch is running in a cluster and voicemails are stored in a mysql database. read_epoch is always 0, so file seems that Freeswitch never reads and updates an entry. Grepping mysql however shows a number of queries against the database and also the filenames are correctly read (output of ngrep): select * from voicemail_msgs where username='200' and domain='sip11.mydomain.com' and read_epoch=0 order by read_flags, created_epoch 1258748304.0.200.sip11.mydomain.com$db2801c4-d611-11de-8c58-554df1d6d322.Gor Nico.061035013113.inboxr/usr/local/freeswitch/storage/voicemail/default/sip… 1258746833.0.200.sip11.mydomain.com$6e486a2e-d60e-11de-bb97-eb22f15930a0.Gor Nico.061035013113.inboxr/usr/local/freeswitch/storage/voicemail/default/sip… 1258748679.0.200.sip11.mydomain.com$bac4dd0c-d612-11de-9618-afbc82bc409a.Gor Nico.061035013113.inboxr/usr/local/freeswitch/storage/voicemail/default/sip… 1258749095.0.200.sip11.mydomain.com$b2376082-d613-11de-80e8-89d0ee29138d.Gor Nico.061035013113.inboxr/usr/local/freeswitch/storage/voicemail/default/sip… 1258749417.0.200.sip11.mydomain.com$726b375c-d614-11de-bb4c-6d51cf20cc23.Gor Nico.061035013113.inboxr/usr/local/freeswitch/storage/voicemail/default/sip… 1258750260.0.200.sip11.mydomain.com$68cecedc-d616-11de-b8c8-69b0064d633e.Gor Nico.061035013113.inboxr/usr/local/freeswitch/storage/voicemail/default/sip… 1258753767.0.200.sip11.mydomain.com$93657422-d61e-11de-b8c8-69b0064d633e.Gor Nico.061035013113.inboxr/usr/local/freeswitch/storage/voicemail/default/sip… Here's the debug log: EXECUTE sofia/internal/200(a)sip1.mydomain.com send_display(VM 200) 2009-11-20 23:16:36.392353 [DEBUG] mod_dptools.c:703 sofia/internal/200(a)sip1.mydomain.com receive message [DISPLAY] EXECUTE sofia/internal/200(a)sip1.mydomain.com voicemail(check default sip11.mydomain.com 200) 2009-11-20 23:16:36.392353 [DEBUG] mod_voicemail.c:799 [default] rwlock 2009-11-20 23:16:36.392353 [DEBUG] switch_ivr_play_say.c:118 No language specified - Using [en] 2009-11-20 23:16:36.392353 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[voicemail/vm-hello.wav] (en:en) 2009-11-20 23:16:36.392353 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16@8000hz 1 channels 20ms 2009-11-20 23:16:36.392353 [DEBUG] switch_core_io.c:660 sofia/internal/200(a)sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:37.612349 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:37.712349 [DEBUG] switch_channel.c:182 sofia/internal/200(a)sip1.mydomain.com receive message [AUDIO_SYNC] 2009-11-20 23:16:37.812353 [DEBUG] switch_channel.c:182 sofia/internal/200(a)sip1.mydomain.com receive message [AUDIO_SYNC] 2009-11-20 23:16:37.942376 [DEBUG] switch_channel.c:182 sofia/internal/200(a)sip1.mydomain.com receive message [AUDIO_SYNC] 2009-11-20 23:16:38.062368 [DEBUG] switch_ivr_play_say.c:118 No language specified - Using [en] 2009-11-20 23:16:38.062368 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[voicemail/vm-you_have.wav] (en:en) 2009-11-20 23:16:38.062368 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16@8000hz 1 channels 20ms 2009-11-20 23:16:38.062368 [DEBUG] switch_core_io.c:660 sofia/internal/200(a)sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:38.612348 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:38.712354 [DEBUG] switch_ivr_play_say.c:273 Handle say:[1] (en:en) 2009-11-20 23:16:38.712354 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16@8000hz 1 channels 20ms 2009-11-20 23:16:38.712354 [DEBUG] switch_core_io.c:660 sofia/internal/200(a)sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:39.172350 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:39.272353 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[voicemail/vm-urgent-new.wav] (en:en) 2009-11-20 23:16:39.272353 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16@8000hz 1 channels 20ms 2009-11-20 23:16:39.272353 [DEBUG] switch_core_io.c:660 sofia/internal/200(a)sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:40.052349 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:40.152353 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[voicemail/vm-message.wav] (en:en) 2009-11-20 23:16:40.152353 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16@8000hz 1 channels 20ms 2009-11-20 23:16:40.152353 [DEBUG] switch_core_io.c:660 sofia/internal/200(a)sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:40.732349 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:40.832349 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[voicemail/vm-in_folder.wav] (en:en) 2009-11-20 23:16:40.832349 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16@8000hz 1 channels 20ms 2009-11-20 23:16:40.832349 [DEBUG] switch_core_io.c:660 sofia/internal/200(a)sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:42.012350 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:42.112350 [DEBUG] switch_ivr_play_say.c:118 No language specified - Using [en] 2009-11-20 23:16:42.112350 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[voicemail/vm-you_have.wav] (en:en) 2009-11-20 23:16:42.112350 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16@8000hz 1 channels 20ms 2009-11-20 23:16:42.112350 [DEBUG] switch_core_io.c:660 sofia/internal/200(a)sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:42.672354 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:42.772362 [DEBUG] switch_ivr_play_say.c:273 Handle say:[7] (en:en) 2009-11-20 23:16:42.772362 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16@8000hz 1 channels 20ms 2009-11-20 23:16:42.772362 [DEBUG] switch_core_io.c:660 sofia/internal/200(a)sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:43.312349 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:43.412353 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[voicemail/vm-new.wav] (en:en) 2009-11-20 23:16:43.412353 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16@8000hz 1 channels 20ms 2009-11-20 23:16:43.412353 [DEBUG] switch_core_io.c:660 sofia/internal/200(a)sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:43.752353 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:43.872362 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[voicemail/vm-messages.wav] (en:en) 2009-11-20 23:16:43.872362 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16@8000hz 1 channels 20ms 2009-11-20 23:16:43.872362 [DEBUG] switch_core_io.c:660 sofia/internal/200(a)sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:44.532350 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:44.652368 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[voicemail/vm-in_folder.wav] (en:en) 2009-11-20 23:16:44.652368 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16@8000hz 1 channels 20ms 2009-11-20 23:16:44.652368 [DEBUG] switch_core_io.c:660 sofia/internal/200(a)sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:45.832349 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:45.932353 [DEBUG] switch_channel.c:182 sofia/internal/200(a)sip1.mydomain.com receive message [AUDIO_SYNC] 2009-11-20 23:16:46.062365 [DEBUG] switch_ivr_play_say.c:118 No language specified - Using [en] 2009-11-20 23:16:46.062365 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[voicemail/vm-new.wav] (en:en) 2009-11-20 23:16:46.062365 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16@8000hz 1 channels 20ms 2009-11-20 23:16:46.062365 [DEBUG] switch_core_io.c:660 sofia/internal/200(a)sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:46.392349 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:46.492349 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[voicemail/vm-message_number.wav] (en:en) 2009-11-20 23:16:46.492349 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16@8000hz 1 channels 20ms 2009-11-20 23:16:46.492349 [DEBUG] switch_core_io.c:660 sofia/internal/200(a)sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:47.312349 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:47.412349 [DEBUG] switch_ivr_play_say.c:273 Handle say:[1] (en:en) 2009-11-20 23:16:47.412349 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16@8000hz 1 channels 20ms 2009-11-20 23:16:47.412349 [DEBUG] switch_core_io.c:660 sofia/internal/200(a)sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:47.872372 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:47.992349 [DEBUG] switch_ivr_play_say.c:118 No language specified - Using [en] 2009-11-20 23:16:47.992349 [DEBUG] switch_ivr_play_say.c:273 Handle say:[25] (en:en) 2009-11-20 23:16:47.992349 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16@8000hz 1 channels 20ms 2009-11-20 23:16:47.992349 [DEBUG] switch_core_io.c:660 sofia/internal/200(a)sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:48.632353 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:48.632353 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16@8000hz 1 channels 20ms 2009-11-20 23:16:48.632353 [DEBUG] switch_core_io.c:660 sofia/internal/200(a)sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:49.152353 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:49.152353 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16@8000hz 1 channels 20ms 2009-11-20 23:16:49.152353 [DEBUG] switch_core_io.c:660 sofia/internal/200(a)sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:49.612349 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:49.612349 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16@8000hz 1 channels 20ms 2009-11-20 23:16:49.612349 [DEBUG] switch_core_io.c:660 sofia/internal/200(a)sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:50.092349 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:50.092349 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16@8000hz 1 channels 20ms 2009-11-20 23:16:50.092349 [DEBUG] switch_core_io.c:660 sofia/internal/200(a)sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:50.532349 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:50.532349 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16@8000hz 1 channels 20ms 2009-11-20 23:16:50.532349 [DEBUG] switch_core_io.c:660 sofia/internal/200(a)sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:51.012350 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:51.012350 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16@8000hz 1 channels 20ms 2009-11-20 23:16:51.012350 [DEBUG] switch_core_io.c:660 sofia/internal/200(a)sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:51.572349 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:51.572349 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16@8000hz 1 channels 20ms 2009-11-20 23:16:51.572349 [DEBUG] switch_core_io.c:660 sofia/internal/200(a)sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:51.852353 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:51.852353 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16@8000hz 1 channels 20ms 2009-11-20 23:16:51.852353 [DEBUG] switch_core_io.c:660 sofia/internal/200(a)sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:52.312350 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:52.312350 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16@8000hz 1 channels 20ms 2009-11-20 23:16:52.312350 [DEBUG] switch_core_io.c:660 sofia/internal/200(a)sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:52.892356 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:52.892356 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16@8000hz 1 channels 20ms 2009-11-20 23:16:52.892356 [DEBUG] switch_core_io.c:660 sofia/internal/200(a)sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:53.472353 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:53.592349 [ERR] mod_native_file.c:68 Error opening /usr/local/freeswitch/sounds/en/us/callie/inbox.PCMA 2009-11-20 23:16:53.602359 [DEBUG] switch_channel.c:182 sofia/internal/200(a)sip1.mydomain.com receive message [AUDIO_SYNC] 2009-11-20 23:16:53.722367 [DEBUG] switch_ivr_play_say.c:118 No language specified - Using [en] 2009-11-20 23:16:53.722367 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[voicemail/vm-you_have.wav] (en:en) 2009-11-20 23:16:53.722367 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16@8000hz 1 channels 20ms 2009-11-20 23:16:53.722367 [DEBUG] switch_core_io.c:660 sofia/internal/200(a)sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:54.272349 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:54.372353 [DEBUG] switch_ivr_play_say.c:273 Handle say:[1] (en:en) 2009-11-20 23:16:54.372353 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16@8000hz 1 channels 20ms 2009-11-20 23:16:54.372353 [DEBUG] switch_core_io.c:660 sofia/internal/200(a)sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:54.832441 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:54.943764 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[voicemail/vm-urgent-new.wav] (en:en) 2009-11-20 23:16:54.943764 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16@8000hz 1 channels 20ms 2009-11-20 23:16:54.943764 [DEBUG] switch_core_io.c:660 sofia/internal/200(a)sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:55.732350 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:55.847095 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[voicemail/vm-message.wav] (en:en) 2009-11-20 23:16:55.847095 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16@8000hz 1 channels 20ms 2009-11-20 23:16:55.847095 [DEBUG] switch_core_io.c:660 sofia/internal/200(a)sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:56.432351 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:56.532354 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[voicemail/vm-in_folder.wav] (en:en) 2009-11-20 23:16:56.532354 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16@8000hz 1 channels 20ms 2009-11-20 23:16:56.532354 [DEBUG] switch_core_io.c:660 sofia/internal/200(a)sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:57.712350 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:57.812353 [DEBUG] switch_ivr_play_say.c:118 No language specified - Using [en] 2009-11-20 23:16:57.812353 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[voicemail/vm-you_have.wav] (en:en) 2009-11-20 23:16:57.812353 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16@8000hz 1 channels 20ms 2009-11-20 23:16:57.812353 [DEBUG] switch_core_io.c:660 sofia/internal/200(a)sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:58.372350 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:58.472356 [DEBUG] switch_ivr_play_say.c:273 Handle say:[7] (en:en) 2009-11-20 23:16:58.472356 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16@8000hz 1 channels 20ms 2009-11-20 23:16:58.472356 [DEBUG] switch_core_io.c:660 sofia/internal/200(a)sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:59.012350 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:59.112350 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[voicemail/vm-new.wav] (en:en) 2009-11-20 23:16:59.112350 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16@8000hz 1 channels 20ms 2009-11-20 23:16:59.112350 [DEBUG] switch_core_io.c:660 sofia/internal/200(a)sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:16:59.452351 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:16:59.572350 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[voicemail/vm-messages.wav] (en:en) 2009-11-20 23:16:59.572350 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16@8000hz 1 channels 20ms 2009-11-20 23:16:59.572350 [DEBUG] switch_core_io.c:660 sofia/internal/200(a)sip1.mydomain.com receive message [TRANSCODING_NECESSARY] 2009-11-20 23:17:00.232350 [DEBUG] switch_ivr_play_say.c:1428 done playing file 2009-11-20 23:17:00.332371 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[voicemail/vm-in_folder.wav] (en:en) 2009-11-20 23:17:00.332371 [DEBUG] switch_ivr_play_say.c:1136 Codec Activated L16@8000hz 1 channels 20ms 2009-11-20 23:17:00.332371 [DEBUG] switch_core_io.c:660 sofia/internal/200(a)sip1.mydomain.com receive message [TRANSCODING_NECESSARY]

2 5

FIFO Orgination_caller_id
by Adam Ford 24 Nov '09

24 Nov '09

Is there any way to set the origination_caller_id for a FIFO outbound call to an on-hook agent? I can't find anything in the wiki about a FIFO or member variable to set this. It seems to be set to 'Queue' by default, and appears to be hardcoded in the module source. It would be nice to be able to change per FIFO queue. That way agents that handle multiple companies can more easily see which queue is calling and answer accordingly. It is not a big deal, since it does automatically set the origination_caller_id_number to 'fifo+<fifo name>'. However, depending on the phone, the caller ID number is not always readily shown, and must be looked for. Thanks to anyone who has some insight on this, -Adam

5 9

Memory leak with mod_local_stream
by Rob Forman 24 Nov '09

24 Nov '09

Hey guys, Having a problem with mod_local_stream. I recently did a "make current" from 15334 to the latest trunk (15630). After restarting, there now appears to be a memory leak. On a test system (CentOS 5.4, 64-bit) with no calls or registrations, Freeswitch gradually consumes all of the host memory (rate of about 200K/second), then swaps out, eventually rendering the system useless. I isolated it to mod_local_stream. If I unload mod_local_stream, the memory use stops climbing. If I re-load mod_local_stream, it starts again. I would submit the logs except they aren't any besides it starting. The system is just sitting there idle. Even valgrind didn't show much (http://pastebin.freeswitch.org/11238) Maybe I'm using it wrong? I ran it: valgrind --tool=memcheck --log-file-exactly=vg.log --leak- check=full --leak-resolution=high --show-reachable=yes .libs/ freeswitch -vg Questions: * has anyone else seen this? * what is the best way I can assist troubleshooting this? I saw a patch to mod_local_stream (rev 15431) a few weeks back. Could that have anything to do with it? Rob

3 4

conference digits and conference control
by god.nirvana 24 Nov '09

24 Nov '09

hi all how can i get the digits when users in the conference?? and,in conference.conf.xml <control action="mute" digits="0"/> the "action" will set another value?e.g:transfer? thanks 2009-10-15 god.nirvana

2 1

tcp call misses sip message
by RobertT 23 Nov '09

23 Nov '09

Hello everyone! I'v got strange problem with incomplete call via tcp transport. When I perform bridged call from one ua (no matter what transport udp or tcp) through FS this call's leg b message sequence (over tcp) lacks finishing SIP message what in it's turn cause the call to be disconnected by the client by timeout. Everything works fine with local calls, so I guess the problem is somewhere between UA and FS. There is no NAT and calls via udp are being established correctly. The problem is with tcp and tls as well. This is the sender's ua SIP trace: TX 1049 bytes Request msg INVITE/cseq=11615 (tdta0486C000) to UDP : RX 348 bytes Response msg 100/INVITE/cseq=11615 (rdata0482806C) from UDP : RX 813 bytes Response msg 407/INVITE/cseq=11615 (rdata0482806C) from UDP : TX 346 bytes Request msg ACK/cseq=11615 (tdta0486EFD0) to UDP : TX 1324 bytes Request msg INVITE/cseq=11616 (tdta0486C000) to UDP : RX 348 bytes Response msg 100/INVITE/cseq=11616 (rdata0482806C) from UDP : RX 1083 bytes Response msg 200/INVITE/cseq=11616 (rdata0482806C) from UDP : TX 360 bytes Request msg ACK/cseq=11616 (tdta04874E38) to UDP : And this is the reciever's SIP trace: RX 1167 bytes Request msg INVITE/cseq=122911315 (rdata04864E10) from tcp : TX 298 bytes Response msg 100/INVITE/cseq=122911315 (tdta0486D010) to tcp : TX 801 bytes Response msg 200/INVITE/cseq=122911315 (tdta0486D010) to tcp : ------ I guess this is where ACK is supposed to arrive Retransmiting Response msg 200/INVITE/cseq=122911315 (tdta0486D010), count=0, restart?=1 TX 801 bytes Response msg 200/INVITE/cseq=122911315 (tdta0486D010) to tcp : Retransmiting Response msg 200/INVITE/cseq=122911315 (tdta0486D010), count=0, restart?=2 TX 801 bytes Response msg 200/INVITE/cseq=122911315 (tdta0486D010) to tcp : Retransmiting Response msg 200/INVITE/cseq=122911315 (tdta0486D010), count=0, restart?=3 TX 801 bytes Response msg 200/INVITE/cseq=122911315 (tdta0486D010) to tcp : .... Sofia profile config: <param name="debug" value="8"/> <param name="sip-trace" value="yes"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="$${external_call_sip_port}"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="$${global_codec_prefs}"/> <param name="hold-music" value="$${hold_music}"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="session-timeout" value="172800"/> <param name="enable-timer" value="false"/> <param name="minimum-session-expires" value="172800"/> <param name="pass-callee-id" value="false"/> <param name="manage-presence" value="true"/> <param name="challenge-realm" value="auto_from"/> <param name="force-register-domain" value="$${domain}"/> <param name="send-message-query-on-register" value="false"/> <param name="aggressive-nat-detection" value="true"/> <param name="inbound-proxy-media" value="true"/> <param name="inbound-bypass-media" value="false"/> <param name="inbound-late-negotiation" value="true"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="true"/> <param name="rtp-timeout-sec" value="1800"/> <param name="accept-blind-auth" value="false"/> <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> <param name="tls" value="$${external_ssl_enable}"/> <param name="tls-bind-params" value="transport=tls"/> <param name="tls-sip-port" value="$${external_call_tls_port}"/> <param name="tls-cert-dir" value="$${external_ssl_dir}"/> <param name="tls-version" value="$${sip_tls_version}"/> and super-smart dialplan <extension name="one2one"> <condition field="destination_number" expression="(.*)"> <action application="bridge" data="sofia/external_call/$1%${domain_name}"/> </condition> </extension> FS 1.0.5pre5 is running on Windows Server 2007SP1 64bit.This issue first occured with 1.0.4 release. Best regards, Robert

3 13

Building in a builddir using --srcdir option but modules still build in srcdir
by Robert Hadley 23 Nov '09

23 Nov '09

I am trying to build in a subdirectory off the Freeswitch source. I can configure successfully and have make working for switch files and the libraries, but I am having trouble with the modules in src/mod. They still compile in the src/mod folders. Any ideas? Thanks, Robert

3 7

Simplest of Conference Setup questions
by Phillip Jones 23 Nov '09

23 Nov '09

Hi there, I have created a simple conference that works great. The only problem is, when a participant press # it exits the call. So when a user enters a conference with a PIN, and by habit they enter 12345 followed by pound, it puts them in and then straight out. So I edited conference.conf.xml so:  and even assigned # to another function: <control action="energy up" digits="#"/> and the same occurs. Pressing # exits the conference. What am I missing here? tia - phil Conf Setup: <extension name="conference.conf"> <condition field="destination_number" expression="^(2125556625)$"> <action application="answer"/> <action application="conference" data="$1-${domain_name}@default +123"/> </condition> </extension>

5 9