From god.nirvana at gmail.com Wed Jul 1 00:15:58 2009 From: god.nirvana at gmail.com (qian ma) Date: Wed, 1 Jul 2009 15:15:58 +0800 Subject: [Freeswitch-users] Fwd: freeswitch support PCMU only? In-Reply-To: <49161ef40906302348o44b79b2au9349caec5cbb8e16@mail.gmail.com> References: <49161ef40906302348o44b79b2au9349caec5cbb8e16@mail.gmail.com> Message-ID: <49161ef40907010015yae188a2s6809485fca566f5@mail.gmail.com> ---------- Forwarded message ---------- From: qian ma Date: 2009/7/1 Subject: freeswitch support PCMU only? To: "Freeswitch-users at lists.freeswitch.org" < Freeswitch-users at lists.freeswitch.org> hi all freeswitch support PCMU only? i follow the http://wiki.freeswitch.org/wiki/Codecs#sofia.conf.xml, but freeswitch still support PCMU only, below is the trace: 2009-07-01 14:39:35.571364 [DEBUG] sofia_glue.c:3070 Audio Codec Compare [PCMA:8:8000:0]/[PCMU:0:8000:20] 2009-07-01 14:39:35.571364 [DEBUG] sofia_glue.c:3030 Set 2833 dtmf payload to 101 2009-07-01 14:39:35.571364 [DEBUG] sofia_glue.c:3070 Audio Codec Compare [telephone-event:101:8000:0]/[PCMU:0:8000:20] 2009-07-01 14:39:35.571364 [NOTICE] sofia.c:3423 Hangup sofia/maq/9876 at 58.212.219.104 [CS_NEW] [INCOMPATIBLE_DESTINATION] 2009-07-01 14:39:35.571364 [DEBUG] switch_channel.c:1683 Send signal sofia/maq/9876 at 58.212.219.104 [KILL] 2009-07-01 14:39:35.571364 [DEBUG] switch_core_session.c:933 Send signal sofia/maq/9876 at 58.212.219.104 [BREAK] 2009-07-01 14:39:35.572807 [DEBUG] switch_core_state_machine.c:397 ( sofia/maq/9876 at 58.212.219.104) Running State Change CS_HANGUP 2009-07-01 14:39:35.572807 [DEBUG] switch_core_state_machine.c:433 ( sofia/maq/9876 at 58.212.219.104) State HANGUP how to configure the freeswitch?? support more codecs??? thx! m.q -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090701/94a3166c/attachment.html From gcd at i.ph Wed Jul 1 00:18:27 2009 From: gcd at i.ph (Nandy Dagondon) Date: Wed, 1 Jul 2009 15:18:27 +0800 Subject: [Freeswitch-users] freeswitch support PCMU only? In-Reply-To: <49161ef40906302348o44b79b2au9349caec5cbb8e16@mail.gmail.com> References: <49161ef40906302348o44b79b2au9349caec5cbb8e16@mail.gmail.com> Message-ID: <7d0bfd8c0907010018m1de64e51qc69e8a319792db41@mail.gmail.com> fs can support lots of codecs. you can find the ff variables defined in vars.xml: global_codec_prefs outbound_codec_prefs then look for "inbound_codec_negotiation" in sip_profiles/internal.xml,sip_profiles/external.xml if you want your codec_prefs to set priority or not. -nandy On Wed, Jul 1, 2009 at 2:48 PM, qian ma wrote: > hi all > freeswitch support PCMU only? > i follow the http://wiki.freeswitch.org/wiki/Codecs#sofia.conf.xml, > but freeswitch still support PCMU only, > below is the trace: > > 2009-07-01 14:39:35.571364 [DEBUG] sofia_glue.c:3070 Audio Codec > Compare [PCMA:8:8000:0]/[PCMU:0:8000:20] > 2009-07-01 14:39:35.571364 [DEBUG] sofia_glue.c:3030 Set 2833 dtmf payload > to 101 > 2009-07-01 14:39:35.571364 [DEBUG] sofia_glue.c:3070 Audio Codec Compare > [telephone-event:101:8000:0]/[PCMU:0:8000:20] > 2009-07-01 14:39:35.571364 [NOTICE] sofia.c:3423 Hangup > sofia/maq/9876 at 58.212.219.104 [CS_NEW] [INCOMPATIBLE_DESTINATION] > 2009-07-01 14:39:35.571364 [DEBUG] switch_channel.c:1683 Send signal > sofia/maq/9876 at 58.212.219.104 [KILL] > 2009-07-01 14:39:35.571364 [DEBUG] switch_core_session.c:933 Send signal > sofia/maq/9876 at 58.212.219.104 [BREAK] > 2009-07-01 14:39:35.572807 [DEBUG] switch_core_state_machine.c:397 ( > sofia/maq/9876 at 58.212.219.104) Running State Change CS_HANGUP > 2009-07-01 14:39:35.572807 [DEBUG] switch_core_state_machine.c:433 ( > sofia/maq/9876 at 58.212.219.104) State HANGUP > > > > how to configure the freeswitch?? > support more codecs??? > > thx! > > m.q > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090701/1afcc2c9/attachment.html From dujinfang at gmail.com Wed Jul 1 00:19:32 2009 From: dujinfang at gmail.com (seven) Date: Wed, 1 Jul 2009 15:19:32 +0800 Subject: [Freeswitch-users] freeswitch support PCMU only? In-Reply-To: <49161ef40906302348o44b79b2au9349caec5cbb8e16@mail.gmail.com> References: <49161ef40906302348o44b79b2au9349caec5cbb8e16@mail.gmail.com> Message-ID: <391A7903-3BEF-45FE-8134-321D2865A6A6@gmail.com> absolutely not. codec negotiate depending on your conf. do you have a sip trace? On Jul 1, 2009, at 2:48 PM, qian ma wrote: > hi all > freeswitch support PCMU only? > i follow the http://wiki.freeswitch.org/wiki/Codecs#sofia.conf.xml > , but freeswitch still support PCMU only, > below is the trace: > > 2009-07-01 14:39:35.571364 [DEBUG] sofia_glue.c:3070 Audio > Codec Compare [PCMA:8:8000:0]/[PCMU:0:8000:20] > 2009-07-01 14:39:35.571364 [DEBUG] sofia_glue.c:3030 Set 2833 dtmf > payload to 101 > 2009-07-01 14:39:35.571364 [DEBUG] sofia_glue.c:3070 Audio Codec > Compare [telephone-event:101:8000:0]/[PCMU:0:8000:20] > 2009-07-01 14:39:35.571364 [NOTICE] sofia.c:3423 Hangup sofia/maq/9876 at 58.212.219.104 > [CS_NEW] [INCOMPATIBLE_DESTINATION] > 2009-07-01 14:39:35.571364 [DEBUG] switch_channel.c:1683 Send signal sofia/maq/9876 at 58.212.219.104 > [KILL] > 2009-07-01 14:39:35.571364 [DEBUG] switch_core_session.c:933 Send > signal sofia/maq/9876 at 58.212.219.104 [BREAK] > 2009-07-01 14:39:35.572807 [DEBUG] switch_core_state_machine.c:397 (sofia/maq/9876 at 58.212.219.104 > ) Running State Change CS_HANGUP > 2009-07-01 14:39:35.572807 [DEBUG] switch_core_state_machine.c:433 (sofia/maq/9876 at 58.212.219.104 > ) State HANGUP > > > > how to configure the freeswitch?? > support more codecs??? > > thx! > m > .q > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090701/a7f37063/attachment.html From msc at freeswitch.org Wed Jul 1 00:22:44 2009 From: msc at freeswitch.org (Michael S Collins) Date: Wed, 1 Jul 2009 00:22:44 -0700 Subject: [Freeswitch-users] Error in OpenZap In-Reply-To: References: <1246354661.19445.13.camel@raul-laptop> Message-ID: <734E4C94-CCB9-46C4-BB0F-A8D212A20CBD@freeswitch.org> Look more closely at the output. It looks like mod_libpri.so didn't get installed properly. I think this is a bug in the ozmod_libpri build. For now just locate that missing .so file in your oz build environment and copy it to the freeswitch/mod directory and try again. -MC Sent from my iPhone On Jun 30, 2009, at 11:29 PM, Baskar wrote: > Hi, > > i have changed the openzap.conf file but still i get the same error > > [span wanpipe 1] > number => 1 > trunk_type => e1 > b-channel => 1:1-15 > d-channel => 1:16 > b-channel => 1:17-31 > > freeswitch at localhost.localdomain> load mod_libpri > API CALL [load(mod_libpri)] output: > -ERR [module load file routine returned an error] > > 2009-07-01 11:27:51 [CRIT] switch_loadable_module.c:871 > switch_loadable_module_load_file() Error Loading module /usr/local/ > freeswitch/mod/mod_libpri.so > **/usr/local/freeswitch/mod/mod_libpri.so: cannot open shared object > file: No such file or directory** > freeswitch at localhost.localdomain> load mod_openzap > 2009-07-01 11:28:04 [NOTICE] zap_io.c:2626 zap_global_init() Modules > configured: 1 > 2009-07-01 11:28:04 [ERR] zap_io.c:2365 load_config() unknown param > [] 'name' / 'OpenZAP' > 2009-07-01 11:28:04 [ERR] zap_io.c:2365 load_config() unknown param > [] 'number' / '1' > API CALL [load(mod_openzap)] output: > -ERR [module load file routine returned an error] > > 2009-07-01 11:28:04 [ERR] zap_io.c:2365 load_config() unknown param > [] 'trunk_type' / 'E1' > 2009-07-01 11:28:04 [ERR] zap_io.c:2365 load_config() unknown param > [] 'b-channel' / '1:1-15' > 2009-07-01 11:28:04 [ERR] zap_io.c:2365 load_config() unknown param > [] 'd-channel' / '1:16' > 2009-07-01 11:28:04 [ERR] zap_io.c:2365 load_config() unknown param > [] 'b-channel' / '1:17-31' > 2009-07-01 11:28:04 [ERR] zap_io.c:2365 load_config() unknown param > [] 'name' / 'OpenZAP' > 2009-07-01 11:28:04 [ERR] zap_io.c:2365 load_config() unknown param > [] 'number' / '2' > 2009-07-01 11:28:04 [ERR] zap_io.c:2365 load_config() unknown param > [] 'trunk_type' / 'E1' > 2009-07-01 11:28:04 [ERR] zap_io.c:2365 load_config() unknown param > [] 'b-channel' / '2:1-15' > 2009-07-01 11:28:04 [ERR] zap_io.c:2365 load_config() unknown param > [] 'd-channel' / '2:16' > 2009-07-01 11:28:04 [ERR] zap_io.c:2365 load_config() unknown param > [] 'b-channel' / '2:17-31' > 2009-07-01 11:28:04 [INFO] zap_io.c:2370 load_config() Configured 0 > channel(s) > 2009-07-01 11:28:04 [ERR] zap_io.c:2633 zap_global_init() No modules > configured! > 2009-07-01 11:28:04 [ERR] mod_openzap.c:2401 mod_openzap_load() > Error loading OpenZAP > 2009-07-01 11:28:04 [CRIT] switch_loadable_module.c:871 > switch_loadable_module_load_file() Error Loading module /usr/local/ > freeswitch/mod/mod_openzap.so > **Module load routine returned an error** > > > -- > Thanks with 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090701/f285025a/attachment-0001.html From god.nirvana at gmail.com Wed Jul 1 00:44:31 2009 From: god.nirvana at gmail.com (qian ma) Date: Wed, 1 Jul 2009 15:44:31 +0800 Subject: [Freeswitch-users] freeswitch support PCMU only? In-Reply-To: <391A7903-3BEF-45FE-8134-321D2865A6A6@gmail.com> References: <49161ef40906302348o44b79b2au9349caec5cbb8e16@mail.gmail.com> <391A7903-3BEF-45FE-8134-321D2865A6A6@gmail.com> Message-ID: <49161ef40907010044t415df111qbacd95680e06d005@mail.gmail.com> thanks for your replies. my var.xml: below is the sip trace: recv 781 bytes from udp/[58.212.219.104]:40508 at 07:42:37.086711: ------------------------------------------------------------------------ INVITE sip:123456 at 58.212.219.104 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.241:8422 ;branch=z9hG4bK-d8754z-94087229c9446f5f-1---d8754z-;rport Max-Forwards: 70 Contact: To: "123456"> From: "9876" >;tag=057de365 Call-ID: M2VlMmEyYWU1MmNhMjEyMGI2NDFkMmRkNjYxYmRmYjY. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: eyeBeam release 1102u stamp 52345 Content-Length: 237 v=0 o=- 6 2 IN IP4 192.168.1.241 s=CounterPath eyeBeam 1.5 c=IN IP4 192.168.1.241 t=0 0 m=audio 57862 RTP/AVP 8 101 a=alt:1 1 : vX+dFqlN ruK7YVXO 192.168.1.241 57862 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv ------------------------------------------------------------------------ send 380 bytes to udp/[58.212.219.104]:40508 at 07:42:37.087505: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.241:8422 ;branch=z9hG4bK-d8754z-94087229c9446f5f-1---d8754z-;rport=40508;received=58.212.219.104 From: "9876" >;tag=057de365 To: "123456"> Call-ID: M2VlMmEyYWU1MmNhMjEyMGI2NDFkMmRkNjYxYmRmYjY. CSeq: 1 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13923M Content-Length: 0 ------------------------------------------------------------------------ 2009-07-01 15:42:37.87800 [NOTICE] switch_channel.c:602 New Channel sofia/maq/9876 at 58.212.219.104 [574766e1-e71f-4a62-a659-1d43bea135bd] 2009-07-01 15:42:37.87800 [DEBUG] sofia.c:3214 Channel sofia/maq/9876 at 58.212.219.104 entering state [received][100] 2009-07-01 15:42:37.87800 [DEBUG] sofia.c:3221 Remote SDP: v=0 o=- 6 2 IN IP4 192.168.1.241 s=CounterPath eyeBeam 1.5 c=IN IP4 192.168.1.241 t=0 0 m=audio 57862 RTP/AVP 8 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=alt:1 1 : vX+dFqlN ruK7YVXO 192.168.1.241 57862 2009-07-01 15:42:37.87800 [DEBUG] sofia_glue.c:3070 Audio Codec Compare [PCMA:8:8000:0]/[PCMU:0:8000:20] 2009-07-01 15:42:37.87800 [DEBUG] sofia_glue.c:3030 Set 2833 dtmf payload to 101 2009-07-01 15:42:37.87800 [DEBUG] sofia_glue.c:3070 Audio Codec Compare [telephone-event:101:8000:0]/[PCMU:0:8000:20] 2009-07-01 15:42:37.87800 [NOTICE] sofia.c:3423 Hangup sofia/maq/9876 at 58.212.219.104 [CS_NEW] [INCOMPATIBLE_DESTINATION] 2009-07-01 15:42:37.87800 [DEBUG] switch_channel.c:1683 Send signal sofia/maq/9876 at 58.212.219.104 [KILL] 2009-07-01 15:42:37.87800 [DEBUG] switch_core_session.c:933 Send signal sofia/maq/9876 at 58.212.219.104 [BREAK] 2009-07-01 15:42:37.87800 [DEBUG] switch_core_state_machine.c:397 ( sofia/maq/9876 at 58.212.219.104) Running State Change CS_HANGUP 2009-07-01 15:42:37.88976 [DEBUG] switch_core_state_machine.c:433 ( sofia/maq/9876 at 58.212.219.104) State HANGUP 2009-07-01 15:42:37.88976 [DEBUG] mod_sofia.c:338 Channel sofia/maq/9876 at 58.212.219.104 hanging up, cause: INCOMPATIBLE_DESTINATION 2009-07-01 15:42:37.88976 [DEBUG] mod_sofia.c:414 Responding to INVITE with: 488 send 676 bytes to udp/[58.212.219.104]:40508 at 07:42:37.089766: ------------------------------------------------------------------------ SIP/2.0 488 Not Acceptable Here Via: SIP/2.0/UDP 192.168.1.241:8422 ;branch=z9hG4bK-d8754z-94087229c9446f5f-1---d8754z-;rport=40508;received=58.212.219.104 From: "9876" >;tag=057de365 To: "123456" >;tag=28Q0QB73Bm35K Call-ID: M2VlMmEyYWU1MmNhMjEyMGI2NDFkMmRkNjYxYmRmYjY. CSeq: 1 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13923M Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: timer, precondition, path, replaces Allow-Events: talk, refer Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION" Content-Length: 0 ------------------------------------------------------------------------ 2009-07-01 15:42:37.89978 [DEBUG] switch_core_state_machine.c:46 sofia/maq/9876 at 58.212.219.104 Standard HANGUP, cause: INCOMPATIBLE_DESTINATION 2009-07-01 15:42:37.89978 [DEBUG] switch_core_state_machine.c:433 ( sofia/maq/9876 at 58.212.219.104) State HANGUP going to sleep 2009-07-01 15:42:37.89978 [DEBUG] switch_core_state_machine.c:475 ( sofia/maq/9876 at 58.212.219.104) State Change CS_HANGUP -> CS_REPORTING 2009-07-01 15:42:37.89978 [DEBUG] switch_core_session.c:933 Send signal sofia/maq/9876 at 58.212.219.104 [BREAK] 2009-07-01 15:42:37.89978 [DEBUG] switch_core_state_machine.c:397 ( sofia/maq/9876 at 58.212.219.104) Running State Change CS_REPORTING 2009-07-01 15:42:37.89978 [DEBUG] switch_core_state_machine.c:607 ( sofia/maq/9876 at 58.212.219.104) State REPORTING recv 334 bytes from udp/[58.212.219.104]:40508 at 07:42:37.093591: ------------------------------------------------------------------------ ACK sip:123456 at 58.212.219.104 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.241:8422 ;branch=z9hG4bK-d8754z-94087229c9446f5f-1---d8754z-;rport To: "123456" >;tag=28Q0QB73Bm35K From: "9876" >;tag=057de365 Call-ID: M2VlMmEyYWU1MmNhMjEyMGI2NDFkMmRkNjYxYmRmYjY. CSeq: 1 ACK Content-Length: 0 ------------------------------------------------------------------------ 2009-07-01 15:42:37.133976 [DEBUG] switch_core_state_machine.c:53 sofia/maq/9876 at 58.212.219.104 Standard REPORTING, cause: INCOMPATIBLE_DESTINATION 2009-07-01 15:42:37.133976 [DEBUG] switch_core_state_machine.c:607 ( sofia/maq/9876 at 58.212.219.104) State REPORTING going to sleep 2009-07-01 15:42:37.133976 [DEBUG] switch_core_state_machine.c:410 ( sofia/maq/9876 at 58.212.219.104) State Change CS_REPORTING -> CS_DESTROY 2009-07-01 15:42:37.133976 [DEBUG] switch_core_session.c:1067 Session 3 ( sofia/maq/9876 at 58.212.219.104) Locked, Waiting on external entities 2009-07-01 15:42:37.133976 [NOTICE] switch_core_session.c:1085 Session 3 ( sofia/maq/9876 at 58.212.219.104) Ended 2009-07-01 15:42:37.133976 [NOTICE] switch_core_session.c:1087 Close Channel sofia/maq/9876 at 58.212.219.104 [CS_DESTROY] 2009-07-01 15:42:37.133976 [DEBUG] switch_core_state_machine.c:559 ( sofia/maq/9876 at 58.212.219.104) State DESTROY 2009-07-01 15:42:37.133976 [DEBUG] mod_sofia.c:255 sofia/maq/9876 at 58.212.219.104 SOFIA DESTROY 2009-07-01 15:42:37.133976 [DEBUG] switch_core_state_machine.c:60 sofia/maq/9876 at 58.212.219.104 Standard DESTROY 2009-07-01 15:42:37.133976 [DEBUG] switch_core_state_machine.c:559 ( sofia/maq/9876 at 58.212.219.104) State DESTROY going to sleep the freeswitch not accept PCMA 2009/7/1 seven > absolutely not. > codec negotiate depending on your conf. do you have a sip trace? > > On Jul 1, 2009, at 2:48 PM, qian ma wrote: > > hi all > freeswitch support PCMU only? > i follow the http://wiki.freeswitch.org/wiki/Codecs#sofia.conf.xml, > but freeswitch still support PCMU only, > below is the trace: > > 2009-07-01 14:39:35.571364 [DEBUG] sofia_glue.c:3070 Audio Codec > Compare [PCMA:8:8000:0]/[PCMU:0:8000:20] > 2009-07-01 14:39:35.571364 [DEBUG] sofia_glue.c:3030 Set 2833 dtmf payload > to 101 > 2009-07-01 14:39:35.571364 [DEBUG] sofia_glue.c:3070 Audio Codec Compare > [telephone-event:101:8000:0]/[PCMU:0:8000:20] > 2009-07-01 14:39:35.571364 [NOTICE] sofia.c:3423 Hangup > sofia/maq/9876 at 58.212.219.104 [CS_NEW] [INCOMPATIBLE_DESTINATION] > 2009-07-01 14:39:35.571364 [DEBUG] switch_channel.c:1683 Send signal > sofia/maq/9876 at 58.212.219.104 [KILL] > 2009-07-01 14:39:35.571364 [DEBUG] switch_core_session.c:933 Send signal > sofia/maq/9876 at 58.212.219.104 [BREAK] > 2009-07-01 14:39:35.572807 [DEBUG] switch_core_state_machine.c:397 ( > sofia/maq/9876 at 58.212.219.104) Running State Change CS_HANGUP > 2009-07-01 14:39:35.572807 [DEBUG] switch_core_state_machine.c:433 ( > sofia/maq/9876 at 58.212.219.104) State HANGUP > > > > how to configure the freeswitch?? > support more codecs??? > > thx! > > m.q > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090701/a2971cc3/attachment.html From gcd at i.ph Wed Jul 1 00:57:24 2009 From: gcd at i.ph (Nandy Dagondon) Date: Wed, 1 Jul 2009 15:57:24 +0800 Subject: [Freeswitch-users] freeswitch support PCMU only? In-Reply-To: <391A7903-3BEF-45FE-8134-321D2865A6A6@gmail.com> References: <49161ef40906302348o44b79b2au9349caec5cbb8e16@mail.gmail.com> <391A7903-3BEF-45FE-8134-321D2865A6A6@gmail.com> Message-ID: <7d0bfd8c0907010057x5a05d7b4m1088ad54ffb6101f@mail.gmail.com> seven, of course, codec negotiation depends on the order of codecs in the *_codec_prefs variables. but, the opposite end has also it's own codecs prefs, too. fs can accept the other end's prefs (inbound_codec_negotiation=generous) or imposes it's own prefs (=greedy). you must include the codec in the *_codec_prefs to activate it. is this correct? -nandy On Wed, Jul 1, 2009 at 3:19 PM, seven wrote: > absolutely not. > codec negotiate depending on your conf. do you have a sip trace? > > On Jul 1, 2009, at 2:48 PM, qian ma wrote: > > hi all > freeswitch support PCMU only? > i follow the http://wiki.freeswitch.org/wiki/Codecs#sofia.conf.xml, > but freeswitch still support PCMU only, > below is the trace: > > 2009-07-01 14:39:35.571364 [DEBUG] sofia_glue.c:3070 Audio Codec > Compare [PCMA:8:8000:0]/[PCMU:0:8000:20] > 2009-07-01 14:39:35.571364 [DEBUG] sofia_glue.c:3030 Set 2833 dtmf payload > to 101 > 2009-07-01 14:39:35.571364 [DEBUG] sofia_glue.c:3070 Audio Codec Compare > [telephone-event:101:8000:0]/[PCMU:0:8000:20] > 2009-07-01 14:39:35.571364 [NOTICE] sofia.c:3423 Hangup > sofia/maq/9876 at 58.212.219.104 [CS_NEW] [INCOMPATIBLE_DESTINATION] > 2009-07-01 14:39:35.571364 [DEBUG] switch_channel.c:1683 Send signal > sofia/maq/9876 at 58.212.219.104 [KILL] > 2009-07-01 14:39:35.571364 [DEBUG] switch_core_session.c:933 Send signal > sofia/maq/9876 at 58.212.219.104 [BREAK] > 2009-07-01 14:39:35.572807 [DEBUG] switch_core_state_machine.c:397 ( > sofia/maq/9876 at 58.212.219.104) Running State Change CS_HANGUP > 2009-07-01 14:39:35.572807 [DEBUG] switch_core_state_machine.c:433 ( > sofia/maq/9876 at 58.212.219.104) State HANGUP > > > > how to configure the freeswitch?? > support more codecs??? > > thx! > > m.q > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090701/17cf98a7/attachment-0001.html From fdelawarde at wirelessmundi.com Wed Jul 1 01:24:10 2009 From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde) Date: Wed, 01 Jul 2009 10:24:10 +0200 Subject: [Freeswitch-users] Any advances on T.38 support for FS? In-Reply-To: <71483C2D-335B-4801-BEF1-AAA8F84BE060@jerris.com> References: <1246353349.30167.83.camel@luna.tc.commsmundi.com> <71483C2D-335B-4801-BEF1-AAA8F84BE060@jerris.com> Message-ID: <1246436650.30167.92.camel@luna.tc.commsmundi.com> Is there any work planned for T.38 termination (in mod_fax)? Fran?ois. On Tue, 2009-06-30 at 12:07 -0400, Michael Jerris wrote: > We currently support t.38 passthrough only using proxy_media mode. T. > 38 gateway is on the roadmap but not yet close to complete. > > Mike > > On Jun 30, 2009, at 5:15 AM, Fran?ois Delawarde wrote: > > > Many issues on Asterisk's T.38 (or probably just on T.38?)... > > > > Could it convince those relying on this "modern" version of a 50yo > > technology to switch to and with FreeSwitch? > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.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 Wed Jul 1 01:23:32 2009 From: jason at jasonjgw.net (Jason White) Date: Wed, 1 Jul 2009 18:23:32 +1000 Subject: [Freeswitch-users] Could this be a bug in the SIP registry? In-Reply-To: References: Message-ID: <20090701082332.GA10823@jdc.jasonjgw.net> Mitchel Constantin wrote: > 5. My phones now register using the correct domain name (i.e. weavver.com) > instead of the IP address (205.134.225.20) as the domain. > 6. Now the problem... My originate command no longer works using the new > syntax: originate sofia/internal/mythicalbox%weavver.comsofia/internal/johndoe% > weavver.com > > The phones do show up as registered when I type "sofia status profile > internal": What happens if you use the following syntax? originate user/phone at domain extension e.g. originate user/1000 at example.com 3000 to connext user at example.com to extension 3000. My other advice would be to read the FreeSWITCH log files carefully. Also, use the sofia_contact command to find out how the registered users will be called when the syntax mentioned above is used. Make sure that everything will go where you want it. From jason at jasonjgw.net Wed Jul 1 01:38:55 2009 From: jason at jasonjgw.net (Jason White) Date: Wed, 1 Jul 2009 18:38:55 +1000 Subject: [Freeswitch-users] Any advances on T.38 support for FS? In-Reply-To: <1246436650.30167.92.camel@luna.tc.commsmundi.com> References: <1246353349.30167.83.camel@luna.tc.commsmundi.com> <71483C2D-335B-4801-BEF1-AAA8F84BE060@jerris.com> <1246436650.30167.92.camel@luna.tc.commsmundi.com> Message-ID: <20090701083855.GA11498@jdc.jasonjgw.net> Fran?ois Delawarde wrote: > Is there any work planned for T.38 termination (in mod_fax)? Yes, as discussed on the mailing list recently. If you're volunteering to help, I'm sure the FreeSWITCH developers would appreciate contributions of code. From jason at jasonjgw.net Wed Jul 1 01:45:47 2009 From: jason at jasonjgw.net (Jason White) Date: Wed, 1 Jul 2009 18:45:47 +1000 Subject: [Freeswitch-users] Could this be a bug in the SIP registry? In-Reply-To: <20090701082332.GA10823@jdc.jasonjgw.net> References: <20090701082332.GA10823@jdc.jasonjgw.net> Message-ID: <20090701084547.GA11743@jdc.jasonjgw.net> Jason White wrote: > originate user/1000 at example.com 3000 > to connext user at example.com to extension 3000. That should read "to connect 1000 at example.com to extension 3000". From lubimov at neolant.ru Wed Jul 1 01:56:10 2009 From: lubimov at neolant.ru (Alexey Lubimov) Date: Wed, 01 Jul 2009 12:56:10 +0400 Subject: [Freeswitch-users] Cant register a pointer. What wrong? In-Reply-To: References: <4A4A1F52.2000307@neolant.ru> Message-ID: <4A4B24AA.70901@neolant.ru> grep -ir 111 * default/bad.xml: default/bad.xml: default/bad.xml: default.xml: default.xml: Michael Jerris ?????: > you have a pointer somewhere in your directory for that user, hard to > see without seeing the whole config, but grep for 111 and see what > else you find. > > Mike > > On Jun 30, 2009, at 10:21 AM, Alexey Lubimov wrote: > > >> I sofia_reg.c:1765 have two user records - good #110 and bad #111. >> >> bad.xml: >> >> >> >> >> >> >> >> >> > value="domestic,international,local"/> >> >> >> >> >> > value="$${outbound_caller_name}"/> >> > value="$${outbound_caller_id}"/> >> >> >> >> >> >> good.xml: >> >> >> >> >> >> >> >> >> > value="domestic,international,local"/> >> >> >> >> >> > value="$${outbound_caller_name}"/> >> > value="$${outbound_caller_id}"/> >> >> >> >> >> >> Good user 110 work without any problem. But user "Bad" user 111 can't >> register to freeswitch. >> >> In log I can see only one message - 2009-06-30 16:31:36.590970 >> [WARNING] sofia_reg.c:1765 Cant register a pointer. >> >> good user exists: >> freeswitch at internal> user_exists id 110 neolant.ru >> true >> >> and bad user exists! >> freeswitch at internal> user_exists id 111 neolant.ru >> true >> >> good user don't have attr type: >> >> freeswitch at internal> user_data 110 at neolant.ru attr type >> -ERR no reply >> >> but bad user have attr type! : >> >> freeswitch at internal> user_data 111 at neolant.ru attr type >> pointer >> >> >> Good user have password: >> >> freeswitch at internal> user_data 110 at neolant.ru param password >> 123456 >> >> But bad user no have param pasword! >> >> freeswitch at internal> user_data 111 at neolant.ru param password >> -ERR no reply >> >> >> What's wrong in these configuration? How I can debug and resolve these >> problems? >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From gcd at i.ph Wed Jul 1 01:59:42 2009 From: gcd at i.ph (Nandy Dagondon) Date: Wed, 1 Jul 2009 16:59:42 +0800 Subject: [Freeswitch-users] freeswitch support PCMU only? In-Reply-To: <49161ef40907010044t415df111qbacd95680e06d005@mail.gmail.com> References: <49161ef40906302348o44b79b2au9349caec5cbb8e16@mail.gmail.com> <391A7903-3BEF-45FE-8134-321D2865A6A6@gmail.com> <49161ef40907010044t415df111qbacd95680e06d005@mail.gmail.com> Message-ID: <7d0bfd8c0907010159l416d2585p2463b198d8741c6@mail.gmail.com> you FS doesn't accept PCMU. try to add "PCMU" on both variables. On Wed, Jul 1, 2009 at 3:44 PM, qian ma wrote: > thanks for your replies. > my var.xml: > > > > > below is the sip trace: > recv 781 bytes from udp/[58.212.219.104]:40508 at 07:42:37.086711: > ------------------------------------------------------------------------ > INVITE sip:123456 at 58.212.219.104 SIP/2.0 > Via: SIP/2.0/UDP 192.168.1.241:8422 > ;branch=z9hG4bK-d8754z-94087229c9446f5f-1---d8754z-;rport > Max-Forwards: 70 > Contact: > To: "123456"> > From: "9876" > >;tag=057de365 > Call-ID: M2VlMmEyYWU1MmNhMjEyMGI2NDFkMmRkNjYxYmRmYjY. > CSeq: 1 INVITE > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, > SUBSCRIBE, INFO > Content-Type: application/sdp > User-Agent: eyeBeam release 1102u stamp 52345 > Content-Length: 237 > > v=0 > o=- 6 2 IN IP4 192.168.1.241 > s=CounterPath eyeBeam 1.5 > c=IN IP4 192.168.1.241 > t=0 0 > m=audio 57862 RTP/AVP 8 101 > a=alt:1 1 : vX+dFqlN ruK7YVXO 192.168.1.241 57862 > a=fmtp:101 0-15 > a=rtpmap:101 telephone-event/8000 > a=sendrecv > ------------------------------------------------------------------------ > send 380 bytes to udp/[58.212.219.104]:40508 at 07:42:37.087505: > ------------------------------------------------------------------------ > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 192.168.1.241:8422 > ;branch=z9hG4bK-d8754z-94087229c9446f5f-1---d8754z-;rport=40508;received=58.212.219.104 > From: "9876" > >;tag=057de365 > To: "123456"> > Call-ID: M2VlMmEyYWU1MmNhMjEyMGI2NDFkMmRkNjYxYmRmYjY. > CSeq: 1 INVITE > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13923M > Content-Length: 0 > > ------------------------------------------------------------------------ > 2009-07-01 15:42:37.87800 [NOTICE] switch_channel.c:602 New Channel > sofia/maq/9876 at 58.212.219.104 [574766e1-e71f-4a62-a659-1d43bea135bd] > 2009-07-01 15:42:37.87800 [DEBUG] sofia.c:3214 Channel > sofia/maq/9876 at 58.212.219.104 entering state [received][100] > 2009-07-01 15:42:37.87800 [DEBUG] sofia.c:3221 Remote SDP: > v=0 > o=- 6 2 IN IP4 192.168.1.241 > s=CounterPath eyeBeam 1.5 > c=IN IP4 192.168.1.241 > t=0 0 > m=audio 57862 RTP/AVP 8 101 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=alt:1 1 : vX+dFqlN ruK7YVXO 192.168.1.241 57862 > 2009-07-01 15:42:37.87800 [DEBUG] sofia_glue.c:3070 Audio Codec Compare > [PCMA:8:8000:0]/[PCMU:0:8000:20] > 2009-07-01 15:42:37.87800 [DEBUG] sofia_glue.c:3030 Set 2833 dtmf payload > to 101 > 2009-07-01 15:42:37.87800 [DEBUG] sofia_glue.c:3070 Audio Codec Compare > [telephone-event:101:8000:0]/[PCMU:0:8000:20] > 2009-07-01 15:42:37.87800 [NOTICE] sofia.c:3423 Hangup > sofia/maq/9876 at 58.212.219.104 [CS_NEW] [INCOMPATIBLE_DESTINATION] > 2009-07-01 15:42:37.87800 [DEBUG] switch_channel.c:1683 Send signal > sofia/maq/9876 at 58.212.219.104 [KILL] > 2009-07-01 15:42:37.87800 [DEBUG] switch_core_session.c:933 Send signal > sofia/maq/9876 at 58.212.219.104 [BREAK] > 2009-07-01 15:42:37.87800 [DEBUG] switch_core_state_machine.c:397 ( > sofia/maq/9876 at 58.212.219.104) Running State Change CS_HANGUP > 2009-07-01 15:42:37.88976 [DEBUG] switch_core_state_machine.c:433 ( > sofia/maq/9876 at 58.212.219.104) State HANGUP > 2009-07-01 15:42:37.88976 [DEBUG] mod_sofia.c:338 Channel > sofia/maq/9876 at 58.212.219.104 hanging up, cause: INCOMPATIBLE_DESTINATION > 2009-07-01 15:42:37.88976 [DEBUG] mod_sofia.c:414 Responding to INVITE > with: 488 > send 676 bytes to udp/[58.212.219.104]:40508 at 07:42:37.089766: > ------------------------------------------------------------------------ > SIP/2.0 488 Not Acceptable Here > Via: SIP/2.0/UDP 192.168.1.241:8422 > ;branch=z9hG4bK-d8754z-94087229c9446f5f-1---d8754z-;rport=40508;received=58.212.219.104 > From: "9876" > >;tag=057de365 > To: "123456" > >;tag=28Q0QB73Bm35K > Call-ID: M2VlMmEyYWU1MmNhMjEyMGI2NDFkMmRkNjYxYmRmYjY. > CSeq: 1 INVITE > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13923M > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO > Supported: timer, precondition, path, replaces > Allow-Events: talk, refer > Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION" > Content-Length: 0 > > ------------------------------------------------------------------------ > 2009-07-01 15:42:37.89978 [DEBUG] switch_core_state_machine.c:46 > sofia/maq/9876 at 58.212.219.104 Standard HANGUP, cause: > INCOMPATIBLE_DESTINATION > 2009-07-01 15:42:37.89978 [DEBUG] switch_core_state_machine.c:433 ( > sofia/maq/9876 at 58.212.219.104) State HANGUP going to sleep > 2009-07-01 15:42:37.89978 [DEBUG] switch_core_state_machine.c:475 ( > sofia/maq/9876 at 58.212.219.104) State Change CS_HANGUP -> CS_REPORTING > 2009-07-01 15:42:37.89978 [DEBUG] switch_core_session.c:933 Send signal > sofia/maq/9876 at 58.212.219.104 [BREAK] > 2009-07-01 15:42:37.89978 [DEBUG] switch_core_state_machine.c:397 ( > sofia/maq/9876 at 58.212.219.104) Running State Change CS_REPORTING > 2009-07-01 15:42:37.89978 [DEBUG] switch_core_state_machine.c:607 ( > sofia/maq/9876 at 58.212.219.104) State REPORTING > recv 334 bytes from udp/[58.212.219.104]:40508 at 07:42:37.093591: > ------------------------------------------------------------------------ > ACK sip:123456 at 58.212.219.104 SIP/2.0 > Via: SIP/2.0/UDP 192.168.1.241:8422 > ;branch=z9hG4bK-d8754z-94087229c9446f5f-1---d8754z-;rport > To: "123456" > >;tag=28Q0QB73Bm35K > From: "9876" > >;tag=057de365 > Call-ID: M2VlMmEyYWU1MmNhMjEyMGI2NDFkMmRkNjYxYmRmYjY. > CSeq: 1 ACK > Content-Length: 0 > > ------------------------------------------------------------------------ > 2009-07-01 15:42:37.133976 [DEBUG] switch_core_state_machine.c:53 > sofia/maq/9876 at 58.212.219.104 Standard REPORTING, cause: > INCOMPATIBLE_DESTINATION > 2009-07-01 15:42:37.133976 [DEBUG] switch_core_state_machine.c:607 ( > sofia/maq/9876 at 58.212.219.104) State REPORTING going to sleep > 2009-07-01 15:42:37.133976 [DEBUG] switch_core_state_machine.c:410 ( > sofia/maq/9876 at 58.212.219.104) State Change CS_REPORTING -> CS_DESTROY > 2009-07-01 15:42:37.133976 [DEBUG] switch_core_session.c:1067 Session 3 ( > sofia/maq/9876 at 58.212.219.104) Locked, Waiting on external entities > 2009-07-01 15:42:37.133976 [NOTICE] switch_core_session.c:1085 Session 3 ( > sofia/maq/9876 at 58.212.219.104) Ended > 2009-07-01 15:42:37.133976 [NOTICE] switch_core_session.c:1087 Close > Channel sofia/maq/9876 at 58.212.219.104 [CS_DESTROY] > 2009-07-01 15:42:37.133976 [DEBUG] switch_core_state_machine.c:559 ( > sofia/maq/9876 at 58.212.219.104) State DESTROY > 2009-07-01 15:42:37.133976 [DEBUG] mod_sofia.c:255 > sofia/maq/9876 at 58.212.219.104 SOFIA DESTROY > 2009-07-01 15:42:37.133976 [DEBUG] switch_core_state_machine.c:60 > sofia/maq/9876 at 58.212.219.104 Standard DESTROY > 2009-07-01 15:42:37.133976 [DEBUG] switch_core_state_machine.c:559 ( > sofia/maq/9876 at 58.212.219.104) State DESTROY going to sleep > > > the freeswitch not accept PCMA > > > > 2009/7/1 seven > > absolutely not. >> codec negotiate depending on your conf. do you have a sip trace? >> >> On Jul 1, 2009, at 2:48 PM, qian ma wrote: >> >> hi all >> freeswitch support PCMU only? >> i follow the http://wiki.freeswitch.org/wiki/Codecs#sofia.conf.xml, >> but freeswitch still support PCMU only, >> below is the trace: >> >> 2009-07-01 14:39:35.571364 [DEBUG] sofia_glue.c:3070 Audio Codec >> Compare [PCMA:8:8000:0]/[PCMU:0:8000:20] >> 2009-07-01 14:39:35.571364 [DEBUG] sofia_glue.c:3030 Set 2833 dtmf payload >> to 101 >> 2009-07-01 14:39:35.571364 [DEBUG] sofia_glue.c:3070 Audio Codec Compare >> [telephone-event:101:8000:0]/[PCMU:0:8000:20] >> 2009-07-01 14:39:35.571364 [NOTICE] sofia.c:3423 Hangup >> sofia/maq/9876 at 58.212.219.104 [CS_NEW] [INCOMPATIBLE_DESTINATION] >> 2009-07-01 14:39:35.571364 [DEBUG] switch_channel.c:1683 Send signal >> sofia/maq/9876 at 58.212.219.104 [KILL] >> 2009-07-01 14:39:35.571364 [DEBUG] switch_core_session.c:933 Send signal >> sofia/maq/9876 at 58.212.219.104 [BREAK] >> 2009-07-01 14:39:35.572807 [DEBUG] switch_core_state_machine.c:397 ( >> sofia/maq/9876 at 58.212.219.104) Running State Change CS_HANGUP >> 2009-07-01 14:39:35.572807 [DEBUG] switch_core_state_machine.c:433 ( >> sofia/maq/9876 at 58.212.219.104) State HANGUP >> >> >> >> how to configure the freeswitch?? >> support more codecs??? >> >> thx! >> >> m.q >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090701/810d868c/attachment-0001.html From brad.tuan at gmail.com Wed Jul 1 02:00:18 2009 From: brad.tuan at gmail.com (Brad Tuan) Date: Wed, 1 Jul 2009 17:00:18 +0800 Subject: [Freeswitch-users] How to know my gateway registering is successed?? Message-ID: As title ,Does FS keep the status of gateways?? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090701/77e98c07/attachment.html From gservat at gmail.com Wed Jul 1 02:10:18 2009 From: gservat at gmail.com (Gonzalo Servat) Date: Wed, 1 Jul 2009 19:10:18 +1000 Subject: [Freeswitch-users] Cant register a pointer. What wrong? In-Reply-To: <4A4B24AA.70901@neolant.ru> References: <4A4A1F52.2000307@neolant.ru> <4A4B24AA.70901@neolant.ru> Message-ID: On Wed, Jul 1, 2009 at 6:56 PM, Alexey Lubimov wrote: > > grep -ir 111 * > > default/bad.xml: > default/bad.xml: > default/bad.xml: > default.xml: > [..snip..] This is probably a long shot, but for bad user you didn't close off the (I'm assuming you just didn't paste it but it's closed off, but you never know...) - Gonzalo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090701/58f592f5/attachment.html From jason at jasonjgw.net Wed Jul 1 02:17:09 2009 From: jason at jasonjgw.net (Jason White) Date: Wed, 1 Jul 2009 19:17:09 +1000 Subject: [Freeswitch-users] How to know my gateway registering is successed?? In-Reply-To: References: Message-ID: <20090701091709.GA12637@jdc.jasonjgw.net> Brad Tuan wrote: > As title ,Does FS keep the status of gateways?? sofia status gateway From lubimov at neolant.ru Wed Jul 1 02:22:48 2009 From: lubimov at neolant.ru (Alexey Lubimov) Date: Wed, 01 Jul 2009 13:22:48 +0400 Subject: [Freeswitch-users] Cant register a pointer. What wrong? In-Reply-To: References: <4A4A1F52.2000307@neolant.ru> <4A4B24AA.70901@neolant.ru> Message-ID: <4A4B2AE8.5040208@neolant.ru> No, tag is exists. Gonzalo Servat ?????: > On Wed, Jul 1, 2009 at 6:56 PM, Alexey Lubimov > wrote: > > > grep -ir 111 * > > default/bad.xml: > default/bad.xml: > default/bad.xml: value="111"/> > default.xml: > > > [..snip..] > > This is probably a long shot, but for bad user you didn't close off > the (I'm assuming you just didn't paste it but it's closed > off, but you never know...) > > - Gonzalo > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From lubimov at neolant.ru Wed Jul 1 02:52:08 2009 From: lubimov at neolant.ru (Alexey Lubimov) Date: Wed, 01 Jul 2009 13:52:08 +0400 Subject: [Freeswitch-users] Cant register a pointer. What wrong? In-Reply-To: <4A4B2AE8.5040208@neolant.ru> References: <4A4A1F52.2000307@neolant.ru> <4A4B24AA.70901@neolant.ru> <4A4B2AE8.5040208@neolant.ru> Message-ID: <4A4B31C8.2070303@neolant.ru> Thank You, Gonzalo! freeswitch at internal> reloadxml +OK [Success] 2009-07-01 13:40:37.205835 [ERR] switch_xml.c:1282 Couldnt open /opt/freeswitch/conf/directory/default/bad.xml (Permission denied) ls -l -rw-r----- 1 root root 756 2009-06-30 16:39 bad.xml -rw-r----- 1 freeswitch daemon 761 2009-06-19 12:02 good.xml After chown freeswitch:daemon, problem was resolved. From Prometheus001 at gmx.net Wed Jul 1 03:16:00 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Wed, 01 Jul 2009 12:16:00 +0200 Subject: [Freeswitch-users] How to know my gateway registering is successed?? In-Reply-To: <20090701091709.GA12637@jdc.jasonjgw.net> References: <20090701091709.GA12637@jdc.jasonjgw.net> Message-ID: <4A4B3760.6050405@gmx.net> or simply sofia status for all gateways Jason White schrieb: > Brad Tuan wrote: > >> As title ,Does FS keep the status of gateways?? >> > > sofia status gateway > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From fdelawarde at wirelessmundi.com Wed Jul 1 03:23:06 2009 From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde) Date: Wed, 01 Jul 2009 12:23:06 +0200 Subject: [Freeswitch-users] Any advances on T.38 support for FS? In-Reply-To: <20090701083855.GA11498@jdc.jasonjgw.net> References: <1246353349.30167.83.camel@luna.tc.commsmundi.com> <71483C2D-335B-4801-BEF1-AAA8F84BE060@jerris.com> <1246436650.30167.92.camel@luna.tc.commsmundi.com> <20090701083855.GA11498@jdc.jasonjgw.net> Message-ID: <1246443786.30167.100.camel@luna.tc.commsmundi.com> Great news! For what I could read, the most famous DSP programmer worldwide (Steve) seems to be helping out for mod_fax. I guess I should register to freeswitch-dev to monitor this closely. Thanks, Fran?ois. On Wed, 2009-07-01 at 18:38 +1000, Jason White wrote: > Fran?ois Delawarde wrote: > > Is there any work planned for T.38 termination (in mod_fax)? > > Yes, as discussed on the mailing list recently. > > If you're volunteering to help, I'm sure the FreeSWITCH developers would > appreciate contributions of code. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From regs at kinetix.gr Wed Jul 1 03:21:44 2009 From: regs at kinetix.gr (Apostolos Pantsiopoulos) Date: Wed, 01 Jul 2009 13:21:44 +0300 Subject: [Freeswitch-users] Testing Freeswitch performance led to strange behavior In-Reply-To: <4A28D8E8.4040000@kinetix.gr> References: <4A27DAF6.30005@kinetix.gr> <24B4D21B-D7CF-456A-95FD-EC69C87D8967@freeswitch.org> <4A27E1E2.6070905@kinetix.gr> <87f2f3b90906040938u5c2c680fx3bdcd086e0bdc73c@mail.gmail.com> <4A2808A9.8070409@kinetix.gr> <191c3a030906041508k8743508ne80aa0052992dc0a@mail.gmail.com> <4A28D8E8.4040000@kinetix.gr> Message-ID: <4A4B38B8.4090608@kinetix.gr> I am writing this to let you know that this behavior persists in the 1.0.4pre9. Could the calls/sec issue be due to the single threaded nature of Sofia? Because I am getting the feeling that the number of simultaneous channels doesn't really burdens FS, but many Calls/sec does. Apostolos Pantsiopoulos wrote: > Anthony Minessale wrote: >> FS uses async rtp timers so you may want to set rtp-timer-name=none in >> the profile param to simulate asterisk conditions. > > I tried that - although I am not using rtp in my scenario - with the > same results. > >> Also keep in mind that asterisk as an atvantage in a tiny crappy 32 bit >> single cpu box because that was what was popular when it was designed >> and the chance for race conditions is minimal because there is only 1 >> cpu. As you scale up to a 8 core 64 bit xeon you will set a drastic >> difference. > > Yes I know that this machine is not well suited for today's test needs. > But the issue occurs in every machine as long as it is pushed near (but > not quite near) to its limits. I have the same odd durations using a 64 > bit low end server. In this case I could achieve a better call/sec rate > than that of the crappy PC but around 50-60 calls/sec the same problem > showed up. I also used a Mosso Cloud Server (quad core - 64-bit) and the > same thing happened at a higher rate. > > >> I will be happy to investigate this issue a bit if you'd like but i do >> not have any box like you describe so if I can't find anything >> you may have to lend us your lab. > > I would appreciate it if you did. After all there this might be a > problem that has not surfaced yet but someday will as more and more > production boxes start using FS. So it would be better to investigate it > now. > I don't think lending you access to my old P4 PC would help you very much :) > If you have access to a normal 2-4 core system you can easily start > raising the sipp parameters until it starts happening. However if you > really think it is appropriate to use my test machines I'd be happy to > grant access to our low-end Opteron machine (just send me a personal > email). I cannot grant you access to larger systems because they are > used in production. > > I used the embedded sipp scenarios : > > on the UAS side : > > sipp -i -mi -ci -mp 8000 -sn uas > > on the UAC side : > > sipp :5060 -s 44050505-i -mi -ci -r 70 > -d 5000 -l 500 -m 2000 -sn uac > > The dialplan : > > > > > > > > > > data="absolute_codec_string=PCMU"/> > > data="sofia/gateway/sipp01/$1"/> > > > > > > > If you need anything else from the config just notify me. > > In order to verify that at some point the calls start having a > duration larger than the scenario's 5secs you can tcpdump on the sipp > machine or turn on the cdrs logging (I know that it degrades > performance, but as I said it is not a matter of when exactly it > starts happening, it is a matter that it DOES start happening). > > >> >> On Thu, Jun 4, 2009 at 12:47 PM, regs at kinetix.gr >> > wrote: >> >> Michael Collins wrote: >> > >> > >> > The dialplan : >> > >> > >> > >> > >> > >> > >> > >> > > > expression="^.*$"> >> > >> > >> > You forgot the parens around .* >> > It should be expression="^(.*)$" if you plan to use $1 later in the >> > extension... >> > >> > >> > >> > >> > > > data="absolute_codec_string=PCMA"/> >> > > > data="sofia/gateway/sipp01/$1"/> >> > >> > ... like here ^^^^^^^ >> > :) >> > -MC >> >> You are right! Although, I don't think that would change the outcome of >> my test :) >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> ------------------------------------------------------------------------ >> > >> > _______________________________________________ >> > Freeswitch-users mailing list >> > Freeswitch-users at lists.freeswitch.org >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> 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 > > -- ------------------------------------------- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: regs at kinetix.gr ------------------------------------------- From jason at jasonjgw.net Wed Jul 1 03:26:27 2009 From: jason at jasonjgw.net (Jason White) Date: Wed, 1 Jul 2009 20:26:27 +1000 Subject: [Freeswitch-users] How to know my gateway registering is successed?? In-Reply-To: <4A4B3760.6050405@gmx.net> References: <20090701091709.GA12637@jdc.jasonjgw.net> <4A4B3760.6050405@gmx.net> Message-ID: <20090701102627.GA16779@jdc.jasonjgw.net> Peter P GMX wrote: > or simply > sofia status > for all gateways and, from the shell, fs_cli -x help > helpfile fs_cli -x sofia help >> helpfile and any others you need so as to obtain synopses of all the commands that you might need. From god.nirvana at gmail.com Wed Jul 1 05:27:18 2009 From: god.nirvana at gmail.com (qian ma) Date: Wed, 1 Jul 2009 20:27:18 +0800 Subject: [Freeswitch-users] freeswitch support PCMU only? In-Reply-To: <7d0bfd8c0907010159l416d2585p2463b198d8741c6@mail.gmail.com> References: <49161ef40906302348o44b79b2au9349caec5cbb8e16@mail.gmail.com> <391A7903-3BEF-45FE-8134-321D2865A6A6@gmail.com> <49161ef40907010044t415df111qbacd95680e06d005@mail.gmail.com> <7d0bfd8c0907010159l416d2585p2463b198d8741c6@mail.gmail.com> Message-ID: <49161ef40907010527i1f2b79b1h6172cc802fa06aa@mail.gmail.com> i want the fs accept the PCMA not PCMU. i add "PCMA" in global_codec_prefs and outbound_codec_prefs, it doesn't work. FS only accept PCMU. why?? 2009/7/1 Nandy Dagondon > you FS doesn't accept PCMU. try to add "PCMU" on both variables. > > > On Wed, Jul 1, 2009 at 3:44 PM, qian ma wrote: > >> thanks for your replies. >> > my var.xml: >> >> >> >> >> below is the sip trace: >> recv 781 bytes from udp/[58.212.219.104]:40508 at 07:42:37.086711: >> >> ------------------------------------------------------------------------ >> INVITE sip:123456 at 58.212.219.104 SIP/2.0 >> Via: SIP/2.0/UDP 192.168.1.241:8422 >> ;branch=z9hG4bK-d8754z-94087229c9446f5f-1---d8754z-;rport >> Max-Forwards: 70 >> Contact: >> To: "123456"> >> From: "9876" >> >;tag=057de365 >> Call-ID: M2VlMmEyYWU1MmNhMjEyMGI2NDFkMmRkNjYxYmRmYjY. >> CSeq: 1 INVITE >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, >> SUBSCRIBE, INFO >> Content-Type: application/sdp >> User-Agent: eyeBeam release 1102u stamp 52345 >> Content-Length: 237 >> >> v=0 >> o=- 6 2 IN IP4 192.168.1.241 >> s=CounterPath eyeBeam 1.5 >> c=IN IP4 192.168.1.241 >> t=0 0 >> m=audio 57862 RTP/AVP 8 101 >> a=alt:1 1 : vX+dFqlN ruK7YVXO 192.168.1.241 57862 >> a=fmtp:101 0-15 >> a=rtpmap:101 telephone-event/8000 >> a=sendrecv >> >> ------------------------------------------------------------------------ >> send 380 bytes to udp/[58.212.219.104]:40508 at 07:42:37.087505: >> >> ------------------------------------------------------------------------ >> SIP/2.0 100 Trying >> Via: SIP/2.0/UDP 192.168.1.241:8422 >> ;branch=z9hG4bK-d8754z-94087229c9446f5f-1---d8754z-;rport=40508;received=58.212.219.104 >> From: "9876" >> >;tag=057de365 >> To: "123456"> >> Call-ID: M2VlMmEyYWU1MmNhMjEyMGI2NDFkMmRkNjYxYmRmYjY. >> CSeq: 1 INVITE >> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13923M >> Content-Length: 0 >> >> >> ------------------------------------------------------------------------ >> 2009-07-01 15:42:37.87800 [NOTICE] switch_channel.c:602 New Channel >> sofia/maq/9876 at 58.212.219.104 [574766e1-e71f-4a62-a659-1d43bea135bd] >> 2009-07-01 15:42:37.87800 [DEBUG] sofia.c:3214 Channel >> sofia/maq/9876 at 58.212.219.104 entering state [received][100] >> 2009-07-01 15:42:37.87800 [DEBUG] sofia.c:3221 Remote SDP: >> v=0 >> o=- 6 2 IN IP4 192.168.1.241 >> s=CounterPath eyeBeam 1.5 >> c=IN IP4 192.168.1.241 >> t=0 0 >> m=audio 57862 RTP/AVP 8 101 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-15 >> a=alt:1 1 : vX+dFqlN ruK7YVXO 192.168.1.241 57862 >> 2009-07-01 15:42:37.87800 [DEBUG] sofia_glue.c:3070 Audio Codec Compare >> [PCMA:8:8000:0]/[PCMU:0:8000:20] >> 2009-07-01 15:42:37.87800 [DEBUG] sofia_glue.c:3030 Set 2833 dtmf payload >> to 101 >> 2009-07-01 15:42:37.87800 [DEBUG] sofia_glue.c:3070 Audio Codec Compare >> [telephone-event:101:8000:0]/[PCMU:0:8000:20] >> 2009-07-01 15:42:37.87800 [NOTICE] sofia.c:3423 Hangup >> sofia/maq/9876 at 58.212.219.104 [CS_NEW] [INCOMPATIBLE_DESTINATION] >> 2009-07-01 15:42:37.87800 [DEBUG] switch_channel.c:1683 Send signal >> sofia/maq/9876 at 58.212.219.104 [KILL] >> 2009-07-01 15:42:37.87800 [DEBUG] switch_core_session.c:933 Send signal >> sofia/maq/9876 at 58.212.219.104 [BREAK] >> 2009-07-01 15:42:37.87800 [DEBUG] switch_core_state_machine.c:397 ( >> sofia/maq/9876 at 58.212.219.104) Running State Change CS_HANGUP >> 2009-07-01 15:42:37.88976 [DEBUG] switch_core_state_machine.c:433 ( >> sofia/maq/9876 at 58.212.219.104) State HANGUP >> 2009-07-01 15:42:37.88976 [DEBUG] mod_sofia.c:338 Channel >> sofia/maq/9876 at 58.212.219.104 hanging up, cause: INCOMPATIBLE_DESTINATION >> 2009-07-01 15:42:37.88976 [DEBUG] mod_sofia.c:414 Responding to INVITE >> with: 488 >> send 676 bytes to udp/[58.212.219.104]:40508 at 07:42:37.089766: >> >> ------------------------------------------------------------------------ >> SIP/2.0 488 Not Acceptable Here >> Via: SIP/2.0/UDP 192.168.1.241:8422 >> ;branch=z9hG4bK-d8754z-94087229c9446f5f-1---d8754z-;rport=40508;received=58.212.219.104 >> From: "9876" >> >;tag=057de365 >> To: "123456" >> >;tag=28Q0QB73Bm35K >> Call-ID: M2VlMmEyYWU1MmNhMjEyMGI2NDFkMmRkNjYxYmRmYjY. >> CSeq: 1 INVITE >> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13923M >> Accept: application/sdp >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, >> NOTIFY, REFER, UPDATE, REGISTER, INFO >> Supported: timer, precondition, path, replaces >> Allow-Events: talk, refer >> Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION" >> Content-Length: 0 >> >> >> ------------------------------------------------------------------------ >> 2009-07-01 15:42:37.89978 [DEBUG] switch_core_state_machine.c:46 >> sofia/maq/9876 at 58.212.219.104 Standard HANGUP, cause: >> INCOMPATIBLE_DESTINATION >> 2009-07-01 15:42:37.89978 [DEBUG] switch_core_state_machine.c:433 ( >> sofia/maq/9876 at 58.212.219.104) State HANGUP going to sleep >> 2009-07-01 15:42:37.89978 [DEBUG] switch_core_state_machine.c:475 ( >> sofia/maq/9876 at 58.212.219.104) State Change CS_HANGUP -> CS_REPORTING >> 2009-07-01 15:42:37.89978 [DEBUG] switch_core_session.c:933 Send signal >> sofia/maq/9876 at 58.212.219.104 [BREAK] >> 2009-07-01 15:42:37.89978 [DEBUG] switch_core_state_machine.c:397 ( >> sofia/maq/9876 at 58.212.219.104) Running State Change CS_REPORTING >> 2009-07-01 15:42:37.89978 [DEBUG] switch_core_state_machine.c:607 ( >> sofia/maq/9876 at 58.212.219.104) State REPORTING >> recv 334 bytes from udp/[58.212.219.104]:40508 at 07:42:37.093591: >> >> ------------------------------------------------------------------------ >> ACK sip:123456 at 58.212.219.104 SIP/2.0 >> Via: SIP/2.0/UDP 192.168.1.241:8422 >> ;branch=z9hG4bK-d8754z-94087229c9446f5f-1---d8754z-;rport >> To: "123456" >> >;tag=28Q0QB73Bm35K >> From: "9876" >> >;tag=057de365 >> Call-ID: M2VlMmEyYWU1MmNhMjEyMGI2NDFkMmRkNjYxYmRmYjY. >> CSeq: 1 ACK >> Content-Length: 0 >> >> >> ------------------------------------------------------------------------ >> 2009-07-01 15:42:37.133976 [DEBUG] switch_core_state_machine.c:53 >> sofia/maq/9876 at 58.212.219.104 Standard REPORTING, cause: >> INCOMPATIBLE_DESTINATION >> 2009-07-01 15:42:37.133976 [DEBUG] switch_core_state_machine.c:607 ( >> sofia/maq/9876 at 58.212.219.104) State REPORTING going to sleep >> 2009-07-01 15:42:37.133976 [DEBUG] switch_core_state_machine.c:410 ( >> sofia/maq/9876 at 58.212.219.104) State Change CS_REPORTING -> CS_DESTROY >> 2009-07-01 15:42:37.133976 [DEBUG] switch_core_session.c:1067 Session 3 ( >> sofia/maq/9876 at 58.212.219.104) Locked, Waiting on external entities >> 2009-07-01 15:42:37.133976 [NOTICE] switch_core_session.c:1085 Session 3 ( >> sofia/maq/9876 at 58.212.219.104) Ended >> 2009-07-01 15:42:37.133976 [NOTICE] switch_core_session.c:1087 Close >> Channel sofia/maq/9876 at 58.212.219.104 [CS_DESTROY] >> 2009-07-01 15:42:37.133976 [DEBUG] switch_core_state_machine.c:559 ( >> sofia/maq/9876 at 58.212.219.104) State DESTROY >> 2009-07-01 15:42:37.133976 [DEBUG] mod_sofia.c:255 >> sofia/maq/9876 at 58.212.219.104 SOFIA DESTROY >> 2009-07-01 15:42:37.133976 [DEBUG] switch_core_state_machine.c:60 >> sofia/maq/9876 at 58.212.219.104 Standard DESTROY >> 2009-07-01 15:42:37.133976 [DEBUG] switch_core_state_machine.c:559 ( >> sofia/maq/9876 at 58.212.219.104) State DESTROY going to sleep >> >> >> the freeswitch not accept PCMA >> >> >> >> 2009/7/1 seven >> >> absolutely not. >>> codec negotiate depending on your conf. do you have a sip trace? >>> >>> On Jul 1, 2009, at 2:48 PM, qian ma wrote: >>> >>> hi all >>> freeswitch support PCMU only? >>> i follow the >>> http://wiki.freeswitch.org/wiki/Codecs#sofia.conf.xml, but freeswitch >>> still support PCMU only, >>> below is the trace: >>> >>> 2009-07-01 14:39:35.571364 [DEBUG] sofia_glue.c:3070 Audio Codec >>> Compare [PCMA:8:8000:0]/[PCMU:0:8000:20] >>> 2009-07-01 14:39:35.571364 [DEBUG] sofia_glue.c:3030 Set 2833 dtmf >>> payload to 101 >>> 2009-07-01 14:39:35.571364 [DEBUG] sofia_glue.c:3070 Audio Codec Compare >>> [telephone-event:101:8000:0]/[PCMU:0:8000:20] >>> 2009-07-01 14:39:35.571364 [NOTICE] sofia.c:3423 Hangup >>> sofia/maq/9876 at 58.212.219.104 [CS_NEW] [INCOMPATIBLE_DESTINATION] >>> 2009-07-01 14:39:35.571364 [DEBUG] switch_channel.c:1683 Send signal >>> sofia/maq/9876 at 58.212.219.104 [KILL] >>> 2009-07-01 14:39:35.571364 [DEBUG] switch_core_session.c:933 Send signal >>> sofia/maq/9876 at 58.212.219.104 [BREAK] >>> 2009-07-01 14:39:35.572807 [DEBUG] switch_core_state_machine.c:397 ( >>> sofia/maq/9876 at 58.212.219.104) Running State Change CS_HANGUP >>> 2009-07-01 14:39:35.572807 [DEBUG] switch_core_state_machine.c:433 ( >>> sofia/maq/9876 at 58.212.219.104) State HANGUP >>> >>> >>> >>> how to configure the freeswitch?? >>> support more codecs??? >>> >>> thx! >>> >>> m.q >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090701/34111989/attachment.html From danishmoosa at gmail.com Wed Jul 1 05:29:53 2009 From: danishmoosa at gmail.com (Muhammad Danish Moosa) Date: Wed, 1 Jul 2009 18:29:53 +0600 Subject: [Freeswitch-users] Freeswitch memory usage is too high Message-ID: Hi Freeswitch is being used in a scenario where two endpoints are running traffic with bypass media mode. Performance is good and all things are smooth. But as the time goes after starting freeswitch, it starts consuming almost whole of memory. Note , freeswitch is being started with -core option, is it related? If this 99% memory consumption is any red alert, as we can see calls are still connecting fine and all is going as usual. -- Muhammad Danish Moosa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090701/c72f3ad9/attachment-0001.html From intralanman at freeswitch.org Wed Jul 1 06:10:39 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Wed, 01 Jul 2009 09:10:39 -0400 Subject: [Freeswitch-users] Freeswitch memory usage is too high In-Reply-To: References: Message-ID: <4A4B604F.1070302@freeswitch.org> On 07/01/2009 08:29 AM, Muhammad Danish Moosa wrote: > Hi > > Freeswitch is being used in a scenario where two endpoints are running > traffic with bypass media mode. Performance is good and all things are > smooth. > > But as the time goes after starting freeswitch, it starts consuming > almost whole of memory. How much is the "whole"? You should see the memory usage level off, it won't keep growing forever. -Ray From gcd at i.ph Wed Jul 1 06:14:52 2009 From: gcd at i.ph (Nandy Dagondon) Date: Wed, 1 Jul 2009 21:14:52 +0800 Subject: [Freeswitch-users] freeswitch support PCMU only? In-Reply-To: <49161ef40907010527i1f2b79b1h6172cc802fa06aa@mail.gmail.com> References: <49161ef40906302348o44b79b2au9349caec5cbb8e16@mail.gmail.com> <391A7903-3BEF-45FE-8134-321D2865A6A6@gmail.com> <49161ef40907010044t415df111qbacd95680e06d005@mail.gmail.com> <7d0bfd8c0907010159l416d2585p2463b198d8741c6@mail.gmail.com> <49161ef40907010527i1f2b79b1h6172cc802fa06aa@mail.gmail.com> Message-ID: <7d0bfd8c0907010614y1bca1fddya0b445d005a4c78c@mail.gmail.com> check the value of "inbound_codec_negotiation" in the sip_profiles/*.xml files. is it "generous" or "greedy"? you should also check if the endpoint is offering PCMU. On Wed, Jul 1, 2009 at 8:27 PM, qian ma wrote: > i want the fs accept the PCMA not PCMU. > i add "PCMA" in global_codec_prefs and outbound_codec_prefs, it doesn't > work. FS only accept PCMU. > why?? > > > > > 2009/7/1 Nandy Dagondon > > you FS doesn't accept PCMU. try to add "PCMU" on both variables. >> >> >> On Wed, Jul 1, 2009 at 3:44 PM, qian ma wrote: >> >>> thanks for your replies. >>> >> my var.xml: >>> >>> >>> >>> >>> below is the sip trace: >>> recv 781 bytes from udp/[58.212.219.104]:40508 at 07:42:37.086711: >>> >>> ------------------------------------------------------------------------ >>> INVITE sip:123456 at 58.212.219.104 SIP/2.0 >>> Via: SIP/2.0/UDP 192.168.1.241:8422 >>> ;branch=z9hG4bK-d8754z-94087229c9446f5f-1---d8754z-;rport >>> Max-Forwards: 70 >>> Contact: >>> To: "123456"> >>> From: "9876" >>> >;tag=057de365 >>> Call-ID: M2VlMmEyYWU1MmNhMjEyMGI2NDFkMmRkNjYxYmRmYjY. >>> CSeq: 1 INVITE >>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, >>> SUBSCRIBE, INFO >>> Content-Type: application/sdp >>> User-Agent: eyeBeam release 1102u stamp 52345 >>> Content-Length: 237 >>> >>> v=0 >>> o=- 6 2 IN IP4 192.168.1.241 >>> s=CounterPath eyeBeam 1.5 >>> c=IN IP4 192.168.1.241 >>> t=0 0 >>> m=audio 57862 RTP/AVP 8 101 >>> a=alt:1 1 : vX+dFqlN ruK7YVXO 192.168.1.241 57862 >>> a=fmtp:101 0-15 >>> a=rtpmap:101 telephone-event/8000 >>> a=sendrecv >>> >>> ------------------------------------------------------------------------ >>> send 380 bytes to udp/[58.212.219.104]:40508 at 07:42:37.087505: >>> >>> ------------------------------------------------------------------------ >>> SIP/2.0 100 Trying >>> Via: SIP/2.0/UDP 192.168.1.241:8422 >>> ;branch=z9hG4bK-d8754z-94087229c9446f5f-1---d8754z-;rport=40508;received=58.212.219.104 >>> From: "9876" >>> >;tag=057de365 >>> To: "123456"> >>> Call-ID: M2VlMmEyYWU1MmNhMjEyMGI2NDFkMmRkNjYxYmRmYjY. >>> CSeq: 1 INVITE >>> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13923M >>> Content-Length: 0 >>> >>> >>> ------------------------------------------------------------------------ >>> 2009-07-01 15:42:37.87800 [NOTICE] switch_channel.c:602 New Channel >>> sofia/maq/9876 at 58.212.219.104 [574766e1-e71f-4a62-a659-1d43bea135bd] >>> 2009-07-01 15:42:37.87800 [DEBUG] sofia.c:3214 Channel >>> sofia/maq/9876 at 58.212.219.104 entering state [received][100] >>> 2009-07-01 15:42:37.87800 [DEBUG] sofia.c:3221 Remote SDP: >>> v=0 >>> o=- 6 2 IN IP4 192.168.1.241 >>> s=CounterPath eyeBeam 1.5 >>> c=IN IP4 192.168.1.241 >>> t=0 0 >>> m=audio 57862 RTP/AVP 8 101 >>> a=rtpmap:101 telephone-event/8000 >>> a=fmtp:101 0-15 >>> a=alt:1 1 : vX+dFqlN ruK7YVXO 192.168.1.241 57862 >>> 2009-07-01 15:42:37.87800 [DEBUG] sofia_glue.c:3070 Audio Codec Compare >>> [PCMA:8:8000:0]/[PCMU:0:8000:20] >>> 2009-07-01 15:42:37.87800 [DEBUG] sofia_glue.c:3030 Set 2833 dtmf payload >>> to 101 >>> 2009-07-01 15:42:37.87800 [DEBUG] sofia_glue.c:3070 Audio Codec Compare >>> [telephone-event:101:8000:0]/[PCMU:0:8000:20] >>> 2009-07-01 15:42:37.87800 [NOTICE] sofia.c:3423 Hangup >>> sofia/maq/9876 at 58.212.219.104 [CS_NEW] [INCOMPATIBLE_DESTINATION] >>> 2009-07-01 15:42:37.87800 [DEBUG] switch_channel.c:1683 Send signal >>> sofia/maq/9876 at 58.212.219.104 [KILL] >>> 2009-07-01 15:42:37.87800 [DEBUG] switch_core_session.c:933 Send signal >>> sofia/maq/9876 at 58.212.219.104 [BREAK] >>> 2009-07-01 15:42:37.87800 [DEBUG] switch_core_state_machine.c:397 ( >>> sofia/maq/9876 at 58.212.219.104) Running State Change CS_HANGUP >>> 2009-07-01 15:42:37.88976 [DEBUG] switch_core_state_machine.c:433 ( >>> sofia/maq/9876 at 58.212.219.104) State HANGUP >>> 2009-07-01 15:42:37.88976 [DEBUG] mod_sofia.c:338 Channel >>> sofia/maq/9876 at 58.212.219.104 hanging up, cause: >>> INCOMPATIBLE_DESTINATION >>> 2009-07-01 15:42:37.88976 [DEBUG] mod_sofia.c:414 Responding to INVITE >>> with: 488 >>> send 676 bytes to udp/[58.212.219.104]:40508 at 07:42:37.089766: >>> >>> ------------------------------------------------------------------------ >>> SIP/2.0 488 Not Acceptable Here >>> Via: SIP/2.0/UDP 192.168.1.241:8422 >>> ;branch=z9hG4bK-d8754z-94087229c9446f5f-1---d8754z-;rport=40508;received=58.212.219.104 >>> From: "9876" >>> >;tag=057de365 >>> To: "123456" >>> >;tag=28Q0QB73Bm35K >>> Call-ID: M2VlMmEyYWU1MmNhMjEyMGI2NDFkMmRkNjYxYmRmYjY. >>> CSeq: 1 INVITE >>> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13923M >>> Accept: application/sdp >>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, >>> NOTIFY, REFER, UPDATE, REGISTER, INFO >>> Supported: timer, precondition, path, replaces >>> Allow-Events: talk, refer >>> Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION" >>> Content-Length: 0 >>> >>> >>> ------------------------------------------------------------------------ >>> 2009-07-01 15:42:37.89978 [DEBUG] switch_core_state_machine.c:46 >>> sofia/maq/9876 at 58.212.219.104 Standard HANGUP, cause: >>> INCOMPATIBLE_DESTINATION >>> 2009-07-01 15:42:37.89978 [DEBUG] switch_core_state_machine.c:433 ( >>> sofia/maq/9876 at 58.212.219.104) State HANGUP going to sleep >>> 2009-07-01 15:42:37.89978 [DEBUG] switch_core_state_machine.c:475 ( >>> sofia/maq/9876 at 58.212.219.104) State Change CS_HANGUP -> CS_REPORTING >>> 2009-07-01 15:42:37.89978 [DEBUG] switch_core_session.c:933 Send signal >>> sofia/maq/9876 at 58.212.219.104 [BREAK] >>> 2009-07-01 15:42:37.89978 [DEBUG] switch_core_state_machine.c:397 ( >>> sofia/maq/9876 at 58.212.219.104) Running State Change CS_REPORTING >>> 2009-07-01 15:42:37.89978 [DEBUG] switch_core_state_machine.c:607 ( >>> sofia/maq/9876 at 58.212.219.104) State REPORTING >>> recv 334 bytes from udp/[58.212.219.104]:40508 at 07:42:37.093591: >>> >>> ------------------------------------------------------------------------ >>> ACK sip:123456 at 58.212.219.104 SIP/2.0 >>> Via: SIP/2.0/UDP 192.168.1.241:8422 >>> ;branch=z9hG4bK-d8754z-94087229c9446f5f-1---d8754z-;rport >>> To: "123456" >>> >;tag=28Q0QB73Bm35K >>> From: "9876" >>> >;tag=057de365 >>> Call-ID: M2VlMmEyYWU1MmNhMjEyMGI2NDFkMmRkNjYxYmRmYjY. >>> CSeq: 1 ACK >>> Content-Length: 0 >>> >>> >>> ------------------------------------------------------------------------ >>> 2009-07-01 15:42:37.133976 [DEBUG] switch_core_state_machine.c:53 >>> sofia/maq/9876 at 58.212.219.104 Standard REPORTING, cause: >>> INCOMPATIBLE_DESTINATION >>> 2009-07-01 15:42:37.133976 [DEBUG] switch_core_state_machine.c:607 ( >>> sofia/maq/9876 at 58.212.219.104) State REPORTING going to sleep >>> 2009-07-01 15:42:37.133976 [DEBUG] switch_core_state_machine.c:410 ( >>> sofia/maq/9876 at 58.212.219.104) State Change CS_REPORTING -> CS_DESTROY >>> 2009-07-01 15:42:37.133976 [DEBUG] switch_core_session.c:1067 Session 3 ( >>> sofia/maq/9876 at 58.212.219.104) Locked, Waiting on external entities >>> 2009-07-01 15:42:37.133976 [NOTICE] switch_core_session.c:1085 Session 3 >>> (sofia/maq/9876 at 58.212.219.104) Ended >>> 2009-07-01 15:42:37.133976 [NOTICE] switch_core_session.c:1087 Close >>> Channel sofia/maq/9876 at 58.212.219.104 [CS_DESTROY] >>> 2009-07-01 15:42:37.133976 [DEBUG] switch_core_state_machine.c:559 ( >>> sofia/maq/9876 at 58.212.219.104) State DESTROY >>> 2009-07-01 15:42:37.133976 [DEBUG] mod_sofia.c:255 >>> sofia/maq/9876 at 58.212.219.104 SOFIA DESTROY >>> 2009-07-01 15:42:37.133976 [DEBUG] switch_core_state_machine.c:60 >>> sofia/maq/9876 at 58.212.219.104 Standard DESTROY >>> 2009-07-01 15:42:37.133976 [DEBUG] switch_core_state_machine.c:559 ( >>> sofia/maq/9876 at 58.212.219.104) State DESTROY going to sleep >>> >>> >>> the freeswitch not accept PCMA >>> >>> >>> >>> 2009/7/1 seven >>> >>> absolutely not. >>>> codec negotiate depending on your conf. do you have a sip trace? >>>> >>>> On Jul 1, 2009, at 2:48 PM, qian ma wrote: >>>> >>>> hi all >>>> freeswitch support PCMU only? >>>> i follow the >>>> http://wiki.freeswitch.org/wiki/Codecs#sofia.conf.xml, but freeswitch >>>> still support PCMU only, >>>> below is the trace: >>>> >>>> 2009-07-01 14:39:35.571364 [DEBUG] sofia_glue.c:3070 Audio Codec >>>> Compare [PCMA:8:8000:0]/[PCMU:0:8000:20] >>>> 2009-07-01 14:39:35.571364 [DEBUG] sofia_glue.c:3030 Set 2833 dtmf >>>> payload to 101 >>>> 2009-07-01 14:39:35.571364 [DEBUG] sofia_glue.c:3070 Audio Codec Compare >>>> [telephone-event:101:8000:0]/[PCMU:0:8000:20] >>>> 2009-07-01 14:39:35.571364 [NOTICE] sofia.c:3423 Hangup >>>> sofia/maq/9876 at 58.212.219.104 [CS_NEW] [INCOMPATIBLE_DESTINATION] >>>> 2009-07-01 14:39:35.571364 [DEBUG] switch_channel.c:1683 Send signal >>>> sofia/maq/9876 at 58.212.219.104 [KILL] >>>> 2009-07-01 14:39:35.571364 [DEBUG] switch_core_session.c:933 Send signal >>>> sofia/maq/9876 at 58.212.219.104 [BREAK] >>>> 2009-07-01 14:39:35.572807 [DEBUG] switch_core_state_machine.c:397 ( >>>> sofia/maq/9876 at 58.212.219.104) Running State Change CS_HANGUP >>>> 2009-07-01 14:39:35.572807 [DEBUG] switch_core_state_machine.c:433 ( >>>> sofia/maq/9876 at 58.212.219.104) State HANGUP >>>> >>>> >>>> >>>> how to configure the freeswitch?? >>>> support more codecs??? >>>> >>>> thx! >>>> >>>> m.q >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090701/4a577c41/attachment-0001.html From gcd at i.ph Wed Jul 1 06:20:59 2009 From: gcd at i.ph (Nandy Dagondon) Date: Wed, 1 Jul 2009 21:20:59 +0800 Subject: [Freeswitch-users] freeswitch support PCMU only? In-Reply-To: <7d0bfd8c0907010614y1bca1fddya0b445d005a4c78c@mail.gmail.com> References: <49161ef40906302348o44b79b2au9349caec5cbb8e16@mail.gmail.com> <391A7903-3BEF-45FE-8134-321D2865A6A6@gmail.com> <49161ef40907010044t415df111qbacd95680e06d005@mail.gmail.com> <7d0bfd8c0907010159l416d2585p2463b198d8741c6@mail.gmail.com> <49161ef40907010527i1f2b79b1h6172cc802fa06aa@mail.gmail.com> <7d0bfd8c0907010614y1bca1fddya0b445d005a4c78c@mail.gmail.com> Message-ID: <7d0bfd8c0907010620q4e77a8a3i9c4266e8ea599368@mail.gmail.com> sorry. i mean check the x-lite client if PCMA is enabled? On Wed, Jul 1, 2009 at 9:14 PM, Nandy Dagondon wrote: > check the value of "inbound_codec_negotiation" in the sip_profiles/*.xml > files. is it "generous" or "greedy"? you should also check if the endpoint > is offering PCMU. > > > > On Wed, Jul 1, 2009 at 8:27 PM, qian ma wrote: > >> i want the fs accept the PCMA not PCMU. >> i add "PCMA" in global_codec_prefs and outbound_codec_prefs, it doesn't >> work. FS only accept PCMU. >> why?? >> >> >> >> >> 2009/7/1 Nandy Dagondon >> >> you FS doesn't accept PCMU. try to add "PCMU" on both variables. >>> >>> >>> On Wed, Jul 1, 2009 at 3:44 PM, qian ma wrote: >>> >>>> thanks for your replies. >>>> >>> my var.xml: >>>> >>>> >>>> >>>> >>>> below is the sip trace: >>>> recv 781 bytes from udp/[58.212.219.104]:40508 at 07:42:37.086711: >>>> >>>> ------------------------------------------------------------------------ >>>> INVITE sip:123456 at 58.212.219.104 SIP/2.0 >>>> Via: SIP/2.0/UDP 192.168.1.241:8422 >>>> ;branch=z9hG4bK-d8754z-94087229c9446f5f-1---d8754z-;rport >>>> Max-Forwards: 70 >>>> Contact: >>>> To: "123456" >>>> > >>>> From: "9876" >>>> >;tag=057de365 >>>> Call-ID: M2VlMmEyYWU1MmNhMjEyMGI2NDFkMmRkNjYxYmRmYjY. >>>> CSeq: 1 INVITE >>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, >>>> SUBSCRIBE, INFO >>>> Content-Type: application/sdp >>>> User-Agent: eyeBeam release 1102u stamp 52345 >>>> Content-Length: 237 >>>> >>>> v=0 >>>> o=- 6 2 IN IP4 192.168.1.241 >>>> s=CounterPath eyeBeam 1.5 >>>> c=IN IP4 192.168.1.241 >>>> t=0 0 >>>> m=audio 57862 RTP/AVP 8 101 >>>> a=alt:1 1 : vX+dFqlN ruK7YVXO 192.168.1.241 57862 >>>> a=fmtp:101 0-15 >>>> a=rtpmap:101 telephone-event/8000 >>>> a=sendrecv >>>> >>>> ------------------------------------------------------------------------ >>>> send 380 bytes to udp/[58.212.219.104]:40508 at 07:42:37.087505: >>>> >>>> ------------------------------------------------------------------------ >>>> SIP/2.0 100 Trying >>>> Via: SIP/2.0/UDP 192.168.1.241:8422 >>>> ;branch=z9hG4bK-d8754z-94087229c9446f5f-1---d8754z-;rport=40508;received=58.212.219.104 >>>> From: "9876" >>>> >;tag=057de365 >>>> To: "123456" >>>> > >>>> Call-ID: M2VlMmEyYWU1MmNhMjEyMGI2NDFkMmRkNjYxYmRmYjY. >>>> CSeq: 1 INVITE >>>> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13923M >>>> Content-Length: 0 >>>> >>>> >>>> ------------------------------------------------------------------------ >>>> 2009-07-01 15:42:37.87800 [NOTICE] switch_channel.c:602 New Channel >>>> sofia/maq/9876 at 58.212.219.104 [574766e1-e71f-4a62-a659-1d43bea135bd] >>>> 2009-07-01 15:42:37.87800 [DEBUG] sofia.c:3214 Channel >>>> sofia/maq/9876 at 58.212.219.104 entering state [received][100] >>>> 2009-07-01 15:42:37.87800 [DEBUG] sofia.c:3221 Remote SDP: >>>> v=0 >>>> o=- 6 2 IN IP4 192.168.1.241 >>>> s=CounterPath eyeBeam 1.5 >>>> c=IN IP4 192.168.1.241 >>>> t=0 0 >>>> m=audio 57862 RTP/AVP 8 101 >>>> a=rtpmap:101 telephone-event/8000 >>>> a=fmtp:101 0-15 >>>> a=alt:1 1 : vX+dFqlN ruK7YVXO 192.168.1.241 57862 >>>> 2009-07-01 15:42:37.87800 [DEBUG] sofia_glue.c:3070 Audio Codec Compare >>>> [PCMA:8:8000:0]/[PCMU:0:8000:20] >>>> 2009-07-01 15:42:37.87800 [DEBUG] sofia_glue.c:3030 Set 2833 dtmf >>>> payload to 101 >>>> 2009-07-01 15:42:37.87800 [DEBUG] sofia_glue.c:3070 Audio Codec Compare >>>> [telephone-event:101:8000:0]/[PCMU:0:8000:20] >>>> 2009-07-01 15:42:37.87800 [NOTICE] sofia.c:3423 Hangup >>>> sofia/maq/9876 at 58.212.219.104 [CS_NEW] [INCOMPATIBLE_DESTINATION] >>>> 2009-07-01 15:42:37.87800 [DEBUG] switch_channel.c:1683 Send signal >>>> sofia/maq/9876 at 58.212.219.104 [KILL] >>>> 2009-07-01 15:42:37.87800 [DEBUG] switch_core_session.c:933 Send signal >>>> sofia/maq/9876 at 58.212.219.104 [BREAK] >>>> 2009-07-01 15:42:37.87800 [DEBUG] switch_core_state_machine.c:397 ( >>>> sofia/maq/9876 at 58.212.219.104) Running State Change CS_HANGUP >>>> 2009-07-01 15:42:37.88976 [DEBUG] switch_core_state_machine.c:433 ( >>>> sofia/maq/9876 at 58.212.219.104) State HANGUP >>>> 2009-07-01 15:42:37.88976 [DEBUG] mod_sofia.c:338 Channel >>>> sofia/maq/9876 at 58.212.219.104 hanging up, cause: >>>> INCOMPATIBLE_DESTINATION >>>> 2009-07-01 15:42:37.88976 [DEBUG] mod_sofia.c:414 Responding to INVITE >>>> with: 488 >>>> send 676 bytes to udp/[58.212.219.104]:40508 at 07:42:37.089766: >>>> >>>> ------------------------------------------------------------------------ >>>> SIP/2.0 488 Not Acceptable Here >>>> Via: SIP/2.0/UDP 192.168.1.241:8422 >>>> ;branch=z9hG4bK-d8754z-94087229c9446f5f-1---d8754z-;rport=40508;received=58.212.219.104 >>>> From: "9876" >>>> >;tag=057de365 >>>> To: "123456" >>>> >;tag=28Q0QB73Bm35K >>>> Call-ID: M2VlMmEyYWU1MmNhMjEyMGI2NDFkMmRkNjYxYmRmYjY. >>>> CSeq: 1 INVITE >>>> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13923M >>>> Accept: application/sdp >>>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, >>>> NOTIFY, REFER, UPDATE, REGISTER, INFO >>>> Supported: timer, precondition, path, replaces >>>> Allow-Events: talk, refer >>>> Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION" >>>> Content-Length: 0 >>>> >>>> >>>> ------------------------------------------------------------------------ >>>> 2009-07-01 15:42:37.89978 [DEBUG] switch_core_state_machine.c:46 >>>> sofia/maq/9876 at 58.212.219.104 Standard HANGUP, cause: >>>> INCOMPATIBLE_DESTINATION >>>> 2009-07-01 15:42:37.89978 [DEBUG] switch_core_state_machine.c:433 ( >>>> sofia/maq/9876 at 58.212.219.104) State HANGUP going to sleep >>>> 2009-07-01 15:42:37.89978 [DEBUG] switch_core_state_machine.c:475 ( >>>> sofia/maq/9876 at 58.212.219.104) State Change CS_HANGUP -> CS_REPORTING >>>> 2009-07-01 15:42:37.89978 [DEBUG] switch_core_session.c:933 Send signal >>>> sofia/maq/9876 at 58.212.219.104 [BREAK] >>>> 2009-07-01 15:42:37.89978 [DEBUG] switch_core_state_machine.c:397 ( >>>> sofia/maq/9876 at 58.212.219.104) Running State Change CS_REPORTING >>>> 2009-07-01 15:42:37.89978 [DEBUG] switch_core_state_machine.c:607 ( >>>> sofia/maq/9876 at 58.212.219.104) State REPORTING >>>> recv 334 bytes from udp/[58.212.219.104]:40508 at 07:42:37.093591: >>>> >>>> ------------------------------------------------------------------------ >>>> ACK sip:123456 at 58.212.219.104 SIP/2.0 >>>> Via: SIP/2.0/UDP 192.168.1.241:8422 >>>> ;branch=z9hG4bK-d8754z-94087229c9446f5f-1---d8754z-;rport >>>> To: "123456" >>>> >;tag=28Q0QB73Bm35K >>>> From: "9876" >>>> >;tag=057de365 >>>> Call-ID: M2VlMmEyYWU1MmNhMjEyMGI2NDFkMmRkNjYxYmRmYjY. >>>> CSeq: 1 ACK >>>> Content-Length: 0 >>>> >>>> >>>> ------------------------------------------------------------------------ >>>> 2009-07-01 15:42:37.133976 [DEBUG] switch_core_state_machine.c:53 >>>> sofia/maq/9876 at 58.212.219.104 Standard REPORTING, cause: >>>> INCOMPATIBLE_DESTINATION >>>> 2009-07-01 15:42:37.133976 [DEBUG] switch_core_state_machine.c:607 ( >>>> sofia/maq/9876 at 58.212.219.104) State REPORTING going to sleep >>>> 2009-07-01 15:42:37.133976 [DEBUG] switch_core_state_machine.c:410 ( >>>> sofia/maq/9876 at 58.212.219.104) State Change CS_REPORTING -> CS_DESTROY >>>> 2009-07-01 15:42:37.133976 [DEBUG] switch_core_session.c:1067 Session 3 >>>> (sofia/maq/9876 at 58.212.219.104) Locked, Waiting on external entities >>>> 2009-07-01 15:42:37.133976 [NOTICE] switch_core_session.c:1085 Session 3 >>>> (sofia/maq/9876 at 58.212.219.104) Ended >>>> 2009-07-01 15:42:37.133976 [NOTICE] switch_core_session.c:1087 Close >>>> Channel sofia/maq/9876 at 58.212.219.104 [CS_DESTROY] >>>> 2009-07-01 15:42:37.133976 [DEBUG] switch_core_state_machine.c:559 ( >>>> sofia/maq/9876 at 58.212.219.104) State DESTROY >>>> 2009-07-01 15:42:37.133976 [DEBUG] mod_sofia.c:255 >>>> sofia/maq/9876 at 58.212.219.104 SOFIA DESTROY >>>> 2009-07-01 15:42:37.133976 [DEBUG] switch_core_state_machine.c:60 >>>> sofia/maq/9876 at 58.212.219.104 Standard DESTROY >>>> 2009-07-01 15:42:37.133976 [DEBUG] switch_core_state_machine.c:559 ( >>>> sofia/maq/9876 at 58.212.219.104) State DESTROY going to sleep >>>> >>>> >>>> the freeswitch not accept PCMA >>>> >>>> >>>> >>>> 2009/7/1 seven >>>> >>>> absolutely not. >>>>> codec negotiate depending on your conf. do you have a sip trace? >>>>> >>>>> On Jul 1, 2009, at 2:48 PM, qian ma wrote: >>>>> >>>>> hi all >>>>> freeswitch support PCMU only? >>>>> i follow the >>>>> http://wiki.freeswitch.org/wiki/Codecs#sofia.conf.xml, but >>>>> freeswitch still support PCMU only, >>>>> below is the trace: >>>>> >>>>> 2009-07-01 14:39:35.571364 [DEBUG] sofia_glue.c:3070 Audio Codec >>>>> Compare [PCMA:8:8000:0]/[PCMU:0:8000:20] >>>>> 2009-07-01 14:39:35.571364 [DEBUG] sofia_glue.c:3030 Set 2833 dtmf >>>>> payload to 101 >>>>> 2009-07-01 14:39:35.571364 [DEBUG] sofia_glue.c:3070 Audio Codec >>>>> Compare [telephone-event:101:8000:0]/[PCMU:0:8000:20] >>>>> 2009-07-01 14:39:35.571364 [NOTICE] sofia.c:3423 Hangup >>>>> sofia/maq/9876 at 58.212.219.104 [CS_NEW] [INCOMPATIBLE_DESTINATION] >>>>> 2009-07-01 14:39:35.571364 [DEBUG] switch_channel.c:1683 Send signal >>>>> sofia/maq/9876 at 58.212.219.104 [KILL] >>>>> 2009-07-01 14:39:35.571364 [DEBUG] switch_core_session.c:933 Send >>>>> signal sofia/maq/9876 at 58.212.219.104 [BREAK] >>>>> 2009-07-01 14:39:35.572807 [DEBUG] switch_core_state_machine.c:397 ( >>>>> sofia/maq/9876 at 58.212.219.104) Running State Change CS_HANGUP >>>>> 2009-07-01 14:39:35.572807 [DEBUG] switch_core_state_machine.c:433 ( >>>>> sofia/maq/9876 at 58.212.219.104) State HANGUP >>>>> >>>>> >>>>> >>>>> how to configure the freeswitch?? >>>>> support more codecs??? >>>>> >>>>> thx! >>>>> >>>>> m.q >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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/20090701/9b1c352f/attachment-0001.html From god.nirvana at gmail.com Wed Jul 1 06:25:41 2009 From: god.nirvana at gmail.com (qian ma) Date: Wed, 1 Jul 2009 21:25:41 +0800 Subject: [Freeswitch-users] freeswitch support PCMU only? In-Reply-To: <7d0bfd8c0907010620q4e77a8a3i9c4266e8ea599368@mail.gmail.com> References: <49161ef40906302348o44b79b2au9349caec5cbb8e16@mail.gmail.com> <391A7903-3BEF-45FE-8134-321D2865A6A6@gmail.com> <49161ef40907010044t415df111qbacd95680e06d005@mail.gmail.com> <7d0bfd8c0907010159l416d2585p2463b198d8741c6@mail.gmail.com> <49161ef40907010527i1f2b79b1h6172cc802fa06aa@mail.gmail.com> <7d0bfd8c0907010614y1bca1fddya0b445d005a4c78c@mail.gmail.com> <7d0bfd8c0907010620q4e77a8a3i9c4266e8ea599368@mail.gmail.com> Message-ID: <49161ef40907010625xaf3679am5c02f1a7699268e4@mail.gmail.com> inbound_codec_negotiation is generous and the xlite PCMU is enabled. my var.xml.conf: 2009/7/1 Nandy Dagondon > sorry. i mean check the x-lite client if PCMA is enabled? > > > On Wed, Jul 1, 2009 at 9:14 PM, Nandy Dagondon wrote: > >> check the value of "inbound_codec_negotiation" in the sip_profiles/*.xml >> files. is it "generous" or "greedy"? you should also check if the endpoint >> is offering PCMU. >> >> >> >> On Wed, Jul 1, 2009 at 8:27 PM, qian ma wrote: >> >>> i want the fs accept the PCMA not PCMU. >>> i add "PCMA" in global_codec_prefs and outbound_codec_prefs, it doesn't >>> work. FS only accept PCMU. >>> why?? >>> >>> >>> >>> >>> 2009/7/1 Nandy Dagondon >>> >>> you FS doesn't accept PCMU. try to add "PCMU" on both variables. >>>> >>>> >>>> On Wed, Jul 1, 2009 at 3:44 PM, qian ma wrote: >>>> >>>>> thanks for your replies. >>>>> >>>> my var.xml: >>>>> >>>>> >>>>> >>>>> >>>>> below is the sip trace: >>>>> recv 781 bytes from udp/[58.212.219.104]:40508 at 07:42:37.086711: >>>>> >>>>> ------------------------------------------------------------------------ >>>>> INVITE sip:123456 at 58.212.219.104 SIP/2.0 >>>>> Via: SIP/2.0/UDP 192.168.1.241:8422 >>>>> ;branch=z9hG4bK-d8754z-94087229c9446f5f-1---d8754z-;rport >>>>> Max-Forwards: 70 >>>>> Contact: >>>>> To: "123456" >>>>> > >>>>> From: "9876" >>>>> >;tag=057de365 >>>>> Call-ID: M2VlMmEyYWU1MmNhMjEyMGI2NDFkMmRkNjYxYmRmYjY. >>>>> CSeq: 1 INVITE >>>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, >>>>> SUBSCRIBE, INFO >>>>> Content-Type: application/sdp >>>>> User-Agent: eyeBeam release 1102u stamp 52345 >>>>> Content-Length: 237 >>>>> >>>>> v=0 >>>>> o=- 6 2 IN IP4 192.168.1.241 >>>>> s=CounterPath eyeBeam 1.5 >>>>> c=IN IP4 192.168.1.241 >>>>> t=0 0 >>>>> m=audio 57862 RTP/AVP 8 101 >>>>> a=alt:1 1 : vX+dFqlN ruK7YVXO 192.168.1.241 57862 >>>>> a=fmtp:101 0-15 >>>>> a=rtpmap:101 telephone-event/8000 >>>>> a=sendrecv >>>>> >>>>> ------------------------------------------------------------------------ >>>>> send 380 bytes to udp/[58.212.219.104]:40508 at 07:42:37.087505: >>>>> >>>>> ------------------------------------------------------------------------ >>>>> SIP/2.0 100 Trying >>>>> Via: SIP/2.0/UDP 192.168.1.241:8422 >>>>> ;branch=z9hG4bK-d8754z-94087229c9446f5f-1---d8754z-;rport=40508;received=58.212.219.104 >>>>> From: "9876" >>>>> >;tag=057de365 >>>>> To: "123456" >>>>> > >>>>> Call-ID: M2VlMmEyYWU1MmNhMjEyMGI2NDFkMmRkNjYxYmRmYjY. >>>>> CSeq: 1 INVITE >>>>> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13923M >>>>> Content-Length: 0 >>>>> >>>>> >>>>> ------------------------------------------------------------------------ >>>>> 2009-07-01 15:42:37.87800 [NOTICE] switch_channel.c:602 New Channel >>>>> sofia/maq/9876 at 58.212.219.104 [574766e1-e71f-4a62-a659-1d43bea135bd] >>>>> 2009-07-01 15:42:37.87800 [DEBUG] sofia.c:3214 Channel >>>>> sofia/maq/9876 at 58.212.219.104 entering state [received][100] >>>>> 2009-07-01 15:42:37.87800 [DEBUG] sofia.c:3221 Remote SDP: >>>>> v=0 >>>>> o=- 6 2 IN IP4 192.168.1.241 >>>>> s=CounterPath eyeBeam 1.5 >>>>> c=IN IP4 192.168.1.241 >>>>> t=0 0 >>>>> m=audio 57862 RTP/AVP 8 101 >>>>> a=rtpmap:101 telephone-event/8000 >>>>> a=fmtp:101 0-15 >>>>> a=alt:1 1 : vX+dFqlN ruK7YVXO 192.168.1.241 57862 >>>>> 2009-07-01 15:42:37.87800 [DEBUG] sofia_glue.c:3070 Audio Codec Compare >>>>> [PCMA:8:8000:0]/[PCMU:0:8000:20] >>>>> 2009-07-01 15:42:37.87800 [DEBUG] sofia_glue.c:3030 Set 2833 dtmf >>>>> payload to 101 >>>>> 2009-07-01 15:42:37.87800 [DEBUG] sofia_glue.c:3070 Audio Codec Compare >>>>> [telephone-event:101:8000:0]/[PCMU:0:8000:20] >>>>> 2009-07-01 15:42:37.87800 [NOTICE] sofia.c:3423 Hangup >>>>> sofia/maq/9876 at 58.212.219.104 [CS_NEW] [INCOMPATIBLE_DESTINATION] >>>>> 2009-07-01 15:42:37.87800 [DEBUG] switch_channel.c:1683 Send signal >>>>> sofia/maq/9876 at 58.212.219.104 [KILL] >>>>> 2009-07-01 15:42:37.87800 [DEBUG] switch_core_session.c:933 Send signal >>>>> sofia/maq/9876 at 58.212.219.104 [BREAK] >>>>> 2009-07-01 15:42:37.87800 [DEBUG] switch_core_state_machine.c:397 ( >>>>> sofia/maq/9876 at 58.212.219.104) Running State Change CS_HANGUP >>>>> 2009-07-01 15:42:37.88976 [DEBUG] switch_core_state_machine.c:433 ( >>>>> sofia/maq/9876 at 58.212.219.104) State HANGUP >>>>> 2009-07-01 15:42:37.88976 [DEBUG] mod_sofia.c:338 Channel >>>>> sofia/maq/9876 at 58.212.219.104 hanging up, cause: >>>>> INCOMPATIBLE_DESTINATION >>>>> 2009-07-01 15:42:37.88976 [DEBUG] mod_sofia.c:414 Responding to INVITE >>>>> with: 488 >>>>> send 676 bytes to udp/[58.212.219.104]:40508 at 07:42:37.089766: >>>>> >>>>> ------------------------------------------------------------------------ >>>>> SIP/2.0 488 Not Acceptable Here >>>>> Via: SIP/2.0/UDP 192.168.1.241:8422 >>>>> ;branch=z9hG4bK-d8754z-94087229c9446f5f-1---d8754z-;rport=40508;received=58.212.219.104 >>>>> From: "9876" >>>>> >;tag=057de365 >>>>> To: "123456" >>>>> >;tag=28Q0QB73Bm35K >>>>> Call-ID: M2VlMmEyYWU1MmNhMjEyMGI2NDFkMmRkNjYxYmRmYjY. >>>>> CSeq: 1 INVITE >>>>> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13923M >>>>> Accept: application/sdp >>>>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, >>>>> NOTIFY, REFER, UPDATE, REGISTER, INFO >>>>> Supported: timer, precondition, path, replaces >>>>> Allow-Events: talk, refer >>>>> Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION" >>>>> Content-Length: 0 >>>>> >>>>> >>>>> ------------------------------------------------------------------------ >>>>> 2009-07-01 15:42:37.89978 [DEBUG] switch_core_state_machine.c:46 >>>>> sofia/maq/9876 at 58.212.219.104 Standard HANGUP, cause: >>>>> INCOMPATIBLE_DESTINATION >>>>> 2009-07-01 15:42:37.89978 [DEBUG] switch_core_state_machine.c:433 ( >>>>> sofia/maq/9876 at 58.212.219.104) State HANGUP going to sleep >>>>> 2009-07-01 15:42:37.89978 [DEBUG] switch_core_state_machine.c:475 ( >>>>> sofia/maq/9876 at 58.212.219.104) State Change CS_HANGUP -> CS_REPORTING >>>>> 2009-07-01 15:42:37.89978 [DEBUG] switch_core_session.c:933 Send signal >>>>> sofia/maq/9876 at 58.212.219.104 [BREAK] >>>>> 2009-07-01 15:42:37.89978 [DEBUG] switch_core_state_machine.c:397 ( >>>>> sofia/maq/9876 at 58.212.219.104) Running State Change CS_REPORTING >>>>> 2009-07-01 15:42:37.89978 [DEBUG] switch_core_state_machine.c:607 ( >>>>> sofia/maq/9876 at 58.212.219.104) State REPORTING >>>>> recv 334 bytes from udp/[58.212.219.104]:40508 at 07:42:37.093591: >>>>> >>>>> ------------------------------------------------------------------------ >>>>> ACK sip:123456 at 58.212.219.104 SIP/2.0 >>>>> Via: SIP/2.0/UDP 192.168.1.241:8422 >>>>> ;branch=z9hG4bK-d8754z-94087229c9446f5f-1---d8754z-;rport >>>>> To: "123456" >>>>> >;tag=28Q0QB73Bm35K >>>>> From: "9876" >>>>> >;tag=057de365 >>>>> Call-ID: M2VlMmEyYWU1MmNhMjEyMGI2NDFkMmRkNjYxYmRmYjY. >>>>> CSeq: 1 ACK >>>>> Content-Length: 0 >>>>> >>>>> >>>>> ------------------------------------------------------------------------ >>>>> 2009-07-01 15:42:37.133976 [DEBUG] switch_core_state_machine.c:53 >>>>> sofia/maq/9876 at 58.212.219.104 Standard REPORTING, cause: >>>>> INCOMPATIBLE_DESTINATION >>>>> 2009-07-01 15:42:37.133976 [DEBUG] switch_core_state_machine.c:607 ( >>>>> sofia/maq/9876 at 58.212.219.104) State REPORTING going to sleep >>>>> 2009-07-01 15:42:37.133976 [DEBUG] switch_core_state_machine.c:410 ( >>>>> sofia/maq/9876 at 58.212.219.104) State Change CS_REPORTING -> CS_DESTROY >>>>> 2009-07-01 15:42:37.133976 [DEBUG] switch_core_session.c:1067 Session 3 >>>>> (sofia/maq/9876 at 58.212.219.104) Locked, Waiting on external entities >>>>> 2009-07-01 15:42:37.133976 [NOTICE] switch_core_session.c:1085 Session >>>>> 3 (sofia/maq/9876 at 58.212.219.104) Ended >>>>> 2009-07-01 15:42:37.133976 [NOTICE] switch_core_session.c:1087 Close >>>>> Channel sofia/maq/9876 at 58.212.219.104 [CS_DESTROY] >>>>> 2009-07-01 15:42:37.133976 [DEBUG] switch_core_state_machine.c:559 ( >>>>> sofia/maq/9876 at 58.212.219.104) State DESTROY >>>>> 2009-07-01 15:42:37.133976 [DEBUG] mod_sofia.c:255 >>>>> sofia/maq/9876 at 58.212.219.104 SOFIA DESTROY >>>>> 2009-07-01 15:42:37.133976 [DEBUG] switch_core_state_machine.c:60 >>>>> sofia/maq/9876 at 58.212.219.104 Standard DESTROY >>>>> 2009-07-01 15:42:37.133976 [DEBUG] switch_core_state_machine.c:559 ( >>>>> sofia/maq/9876 at 58.212.219.104) State DESTROY going to sleep >>>>> >>>>> >>>>> the freeswitch not accept PCMA >>>>> >>>>> >>>>> >>>>> 2009/7/1 seven >>>>> >>>>> absolutely not. >>>>>> codec negotiate depending on your conf. do you have a sip trace? >>>>>> >>>>>> On Jul 1, 2009, at 2:48 PM, qian ma wrote: >>>>>> >>>>>> hi all >>>>>> freeswitch support PCMU only? >>>>>> i follow the >>>>>> http://wiki.freeswitch.org/wiki/Codecs#sofia.conf.xml, but >>>>>> freeswitch still support PCMU only, >>>>>> below is the trace: >>>>>> >>>>>> 2009-07-01 14:39:35.571364 [DEBUG] sofia_glue.c:3070 Audio >>>>>> Codec Compare [PCMA:8:8000:0]/[PCMU:0:8000:20] >>>>>> 2009-07-01 14:39:35.571364 [DEBUG] sofia_glue.c:3030 Set 2833 dtmf >>>>>> payload to 101 >>>>>> 2009-07-01 14:39:35.571364 [DEBUG] sofia_glue.c:3070 Audio Codec >>>>>> Compare [telephone-event:101:8000:0]/[PCMU:0:8000:20] >>>>>> 2009-07-01 14:39:35.571364 [NOTICE] sofia.c:3423 Hangup >>>>>> sofia/maq/9876 at 58.212.219.104 [CS_NEW] [INCOMPATIBLE_DESTINATION] >>>>>> 2009-07-01 14:39:35.571364 [DEBUG] switch_channel.c:1683 Send signal >>>>>> sofia/maq/9876 at 58.212.219.104 [KILL] >>>>>> 2009-07-01 14:39:35.571364 [DEBUG] switch_core_session.c:933 Send >>>>>> signal sofia/maq/9876 at 58.212.219.104 [BREAK] >>>>>> 2009-07-01 14:39:35.572807 [DEBUG] switch_core_state_machine.c:397 ( >>>>>> sofia/maq/9876 at 58.212.219.104) Running State Change CS_HANGUP >>>>>> 2009-07-01 14:39:35.572807 [DEBUG] switch_core_state_machine.c:433 ( >>>>>> sofia/maq/9876 at 58.212.219.104) State HANGUP >>>>>> >>>>>> >>>>>> >>>>>> how to configure the freeswitch?? >>>>>> support more codecs??? >>>>>> >>>>>> thx! >>>>>> >>>>>> m.q >>>>>> _______________________________________________ >>>>>> Freeswitch-users mailing list >>>>>> Freeswitch-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>>> >>>>>> _______________________________________________ >>>>>> Freeswitch-users mailing list >>>>>> Freeswitch-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090701/f6401a96/attachment-0001.html From yudha2008 at gmail.com Wed Jul 1 06:28:32 2009 From: yudha2008 at gmail.com (Baskar) Date: Wed, 1 Jul 2009 18:58:32 +0530 Subject: [Freeswitch-users] Javascript session Recording Message-ID: *Hi, I have configured outbound call through JavaScript it is working fine but i want the conversation to be recorded . Javascript: sessionA = new Session("{ignore_early_media=true, origination_uuid="+argv[0]+"}sofia/default/sip:"+argv[0]+"@ 192.168.1.135:5066"); sessionB = new Session("sofia/internal/"+ argv[1] +"%192.168.1.77"); rtn = sessionA .recordFile("/tmp/"+ argv[0] +".wav", "", "", 400000000, 500, 3); bridge(sessionA, sessionB); i have 2 legs one is sessionA and sessionB if i record the sessionA leg i get only the sessionA voice recorded. How can i merge both the call (**sessionA and sessionB)** into single file. can any one assist me to resolve this problem. -- Thanks with Regards, N.Baskar * -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090701/e17294eb/attachment.html From brian at freeswitch.org Wed Jul 1 06:37:39 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 1 Jul 2009 08:37:39 -0500 Subject: [Freeswitch-users] Fwd: [UniMRCP] PocketSphinx Plugin Available References: <38406.33226.qm@web111316.mail.gq1.yahoo.com> Message-ID: <5FBF1AE5-62B7-4025-A264-1C0E513EEA7C@freeswitch.org> I'm about to wire Arsen the donated money for his work. Remember if you haven't sent me what you have said you would send in please paypal brian at freeswitch.org so I can wire it to Arsen. Thanks, Brian Begin forwarded message: > I would like to announce the availability of PocketSphinx ASR plugin > for UniMRCP server. > > PocketSphinx UniMRCP server can be used with an MRCP compliant > client, which supports JSGF grammar. > Currently supported ASR features are as follows: > > Methods: > DEFINE-GRAMMAR > RECOGNIZE > GET-RESULT > START-INPUT-TIMERS > STOP > Events: > START-OF-INPUT > RECOGNITION-COMPLETE > Headers: > Noinput-Timeout > Recognition-Timeout > Completion-Cause > Completion-Reason > Save-Waveform > Grammar: JSGF > > For the instructions on how to build and configure PocketSphinx with > UniMRCP refer to > http://code.google.com/p/unimrcp/wiki/PocketSphinxPlugin > > Please note, everything is working now, nevertheless this is basic > availability only. > I have mostly tested the integrated solution in the following setup > SipPhone -> FreeSWITCH/UniMRCPClient -> UniMRCPServer/PocketSphinx > > However it requires further testing in different environments from > different speakers, e.t.c. > In other words, your feedback is welcome. > > Thanks, > -- > Arsen Chaloyan > The author of UniMRCP > http://www.unimrcp.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090701/3ef320b1/attachment.html From mike at jerris.com Wed Jul 1 07:01:05 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 1 Jul 2009 10:01:05 -0400 Subject: [Freeswitch-users] Any advances on T.38 support for FS? In-Reply-To: <1246436650.30167.92.camel@luna.tc.commsmundi.com> References: <1246353349.30167.83.camel@luna.tc.commsmundi.com> <71483C2D-335B-4801-BEF1-AAA8F84BE060@jerris.com> <1246436650.30167.92.camel@luna.tc.commsmundi.com> Message-ID: <3F09019C-326B-4502-A41F-C509A58C1D7E@jerris.com> There was a bit of work towards it but no one has worked on it lately On Jul 1, 2009, at 4:24 AM, Fran?ois Delawarde wrote: > Is there any work planned for T.38 termination (in mod_fax)? > > Fran?ois. > > On Tue, 2009-06-30 at 12:07 -0400, Michael Jerris wrote: >> We currently support t.38 passthrough only using proxy_media mode. >> T. >> 38 gateway is on the roadmap but not yet close to complete. >> >> Mike >> >> On Jun 30, 2009, at 5:15 AM, Fran?ois Delawarde wrote: >> >>> Many issues on Asterisk's T.38 (or probably just on T.38?)... >>> >>> Could it convince those relying on this "modern" version of a 50yo >>> technology to switch to and with FreeSwitch? >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.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 Wed Jul 1 07:11:28 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 1 Jul 2009 10:11:28 -0400 Subject: [Freeswitch-users] Freeswitch memory usage is too high In-Reply-To: References: Message-ID: <71F036B3-B47B-4792-A865-F778BF28CCBA@jerris.com> How much memory is it using? Can you use memstat to see where the memory is allocated. Mike On Jul 1, 2009, at 8:29 AM, Muhammad Danish Moosa wrote: > Hi > > Freeswitch is being used in a scenario where two endpoints are > running traffic with bypass media mode. Performance is good and all > things are smooth. > > But as the time goes after starting freeswitch, it starts consuming > almost whole of memory. Note , freeswitch is being started with - > core option, is it related? > > If this 99% memory consumption is any red alert, as we can see calls > are still connecting fine and all is going as usual. > > > -- > Muhammad Danish Moosa > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.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 Jul 1 07:18:57 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 1 Jul 2009 09:18:57 -0500 Subject: [Freeswitch-users] Freeswitch memory usage is too high In-Reply-To: References: Message-ID: You also have a jira http://jira.freeswitch.org/browse/MODAPP-298, It looks like you're using lua sql and the backtrace you attached to the jira was cut off right before the data I needed to see... can you follow up on that ASAP? It looks like a crash in libmysql from the last line but again I can't see the rest of it. /b On Jul 1, 2009, at 7:29 AM, Muhammad Danish Moosa wrote: > Hi > > Freeswitch is being used in a scenario where two endpoints are > running traffic with bypass media mode. Performance is good and all > things are smooth. > > But as the time goes after starting freeswitch, it starts consuming > almost whole of memory. Note , freeswitch is being started with - > core option, is it related? > > If this 99% memory consumption is any red alert, as we can see calls > are still connecting fine and all is going as usual. > > > -- > Muhammad Danish Moosa > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.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 Jul 1 09:25:29 2009 From: Claudio.Cavalera at italtel.it (Cavalera Claudio Luigi) Date: Wed, 1 Jul 2009 18:25:29 +0200 Subject: [Freeswitch-users] How to check if a channel variable is present or NULL Message-ID: Hello, I'm trying to implement this kind of logic in the dialplan if the channel variable sip_refer_to matches regexp than do action else if sip_refer_to exists (not NULL) but does not match regexp than do anti-action else if sip_refer_to does not exist as a channel_variable (NULL ??) than do another action is there a way to check if a channel variable exists or not? Maybe a special regexp to match? 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 gcd at i.ph Wed Jul 1 09:29:51 2009 From: gcd at i.ph (Nandy Dagondon) Date: Thu, 2 Jul 2009 00:29:51 +0800 Subject: [Freeswitch-users] freeswitch support PCMU only? In-Reply-To: <49161ef40907010625xaf3679am5c02f1a7699268e4@mail.gmail.com> References: <49161ef40906302348o44b79b2au9349caec5cbb8e16@mail.gmail.com> <391A7903-3BEF-45FE-8134-321D2865A6A6@gmail.com> <49161ef40907010044t415df111qbacd95680e06d005@mail.gmail.com> <7d0bfd8c0907010159l416d2585p2463b198d8741c6@mail.gmail.com> <49161ef40907010527i1f2b79b1h6172cc802fa06aa@mail.gmail.com> <7d0bfd8c0907010614y1bca1fddya0b445d005a4c78c@mail.gmail.com> <7d0bfd8c0907010620q4e77a8a3i9c4266e8ea599368@mail.gmail.com> <49161ef40907010625xaf3679am5c02f1a7699268e4@mail.gmail.com> Message-ID: <7d0bfd8c0907010929y11e595fbj2a50d2df47a8ee50@mail.gmail.com> is PCMA enabled in X-Lite, too? On Wed, Jul 1, 2009 at 9:25 PM, qian ma wrote: > > inbound_codec_negotiation is generous > and the xlite PCMU is enabled. > > my var.xml.conf: > > > > > > 2009/7/1 Nandy Dagondon > >> sorry. i mean check the x-lite client if PCMA is enabled? >> >> >> On Wed, Jul 1, 2009 at 9:14 PM, Nandy Dagondon wrote: >> >>> check the value of "inbound_codec_negotiation" in the sip_profiles/*.xml >>> files. is it "generous" or "greedy"? you should also check if the endpoint >>> is offering PCMU. >>> >>> >>> >>> On Wed, Jul 1, 2009 at 8:27 PM, qian ma wrote: >>> >>>> i want the fs accept the PCMA not PCMU. >>>> i add "PCMA" in global_codec_prefs and outbound_codec_prefs, it doesn't >>>> work. FS only accept PCMU. >>>> why?? >>>> >>>> >>>> >>>> >>>> 2009/7/1 Nandy Dagondon >>>> >>>> you FS doesn't accept PCMU. try to add "PCMU" on both variables. >>>>> >>>>> >>>>> On Wed, Jul 1, 2009 at 3:44 PM, qian ma wrote: >>>>> >>>>>> thanks for your replies. >>>>>> >>>>> my var.xml: >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> below is the sip trace: >>>>>> recv 781 bytes from udp/[58.212.219.104]:40508 at 07:42:37.086711: >>>>>> >>>>>> ------------------------------------------------------------------------ >>>>>> INVITE sip:123456 at 58.212.219.104 SIP/2.0 >>>>>> Via: SIP/2.0/UDP 192.168.1.241:8422 >>>>>> ;branch=z9hG4bK-d8754z-94087229c9446f5f-1---d8754z-;rport >>>>>> Max-Forwards: 70 >>>>>> Contact: >>>>>> To: "123456" >>>>>> > >>>>>> From: "9876" >>>>>> >;tag=057de365 >>>>>> Call-ID: M2VlMmEyYWU1MmNhMjEyMGI2NDFkMmRkNjYxYmRmYjY. >>>>>> CSeq: 1 INVITE >>>>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, >>>>>> SUBSCRIBE, INFO >>>>>> Content-Type: application/sdp >>>>>> User-Agent: eyeBeam release 1102u stamp 52345 >>>>>> Content-Length: 237 >>>>>> >>>>>> v=0 >>>>>> o=- 6 2 IN IP4 192.168.1.241 >>>>>> s=CounterPath eyeBeam 1.5 >>>>>> c=IN IP4 192.168.1.241 >>>>>> t=0 0 >>>>>> m=audio 57862 RTP/AVP 8 101 >>>>>> a=alt:1 1 : vX+dFqlN ruK7YVXO 192.168.1.241 57862 >>>>>> a=fmtp:101 0-15 >>>>>> a=rtpmap:101 telephone-event/8000 >>>>>> a=sendrecv >>>>>> >>>>>> ------------------------------------------------------------------------ >>>>>> send 380 bytes to udp/[58.212.219.104]:40508 at 07:42:37.087505: >>>>>> >>>>>> ------------------------------------------------------------------------ >>>>>> SIP/2.0 100 Trying >>>>>> Via: SIP/2.0/UDP 192.168.1.241:8422 >>>>>> ;branch=z9hG4bK-d8754z-94087229c9446f5f-1---d8754z-;rport=40508;received=58.212.219.104 >>>>>> From: "9876" >>>>>> >;tag=057de365 >>>>>> To: "123456" >>>>>> > >>>>>> Call-ID: M2VlMmEyYWU1MmNhMjEyMGI2NDFkMmRkNjYxYmRmYjY. >>>>>> CSeq: 1 INVITE >>>>>> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13923M >>>>>> Content-Length: 0 >>>>>> >>>>>> >>>>>> ------------------------------------------------------------------------ >>>>>> 2009-07-01 15:42:37.87800 [NOTICE] switch_channel.c:602 New Channel >>>>>> sofia/maq/9876 at 58.212.219.104 [574766e1-e71f-4a62-a659-1d43bea135bd] >>>>>> 2009-07-01 15:42:37.87800 [DEBUG] sofia.c:3214 Channel >>>>>> sofia/maq/9876 at 58.212.219.104 entering state [received][100] >>>>>> 2009-07-01 15:42:37.87800 [DEBUG] sofia.c:3221 Remote SDP: >>>>>> v=0 >>>>>> o=- 6 2 IN IP4 192.168.1.241 >>>>>> s=CounterPath eyeBeam 1.5 >>>>>> c=IN IP4 192.168.1.241 >>>>>> t=0 0 >>>>>> m=audio 57862 RTP/AVP 8 101 >>>>>> a=rtpmap:101 telephone-event/8000 >>>>>> a=fmtp:101 0-15 >>>>>> a=alt:1 1 : vX+dFqlN ruK7YVXO 192.168.1.241 57862 >>>>>> 2009-07-01 15:42:37.87800 [DEBUG] sofia_glue.c:3070 Audio Codec >>>>>> Compare [PCMA:8:8000:0]/[PCMU:0:8000:20] >>>>>> 2009-07-01 15:42:37.87800 [DEBUG] sofia_glue.c:3030 Set 2833 dtmf >>>>>> payload to 101 >>>>>> 2009-07-01 15:42:37.87800 [DEBUG] sofia_glue.c:3070 Audio Codec >>>>>> Compare [telephone-event:101:8000:0]/[PCMU:0:8000:20] >>>>>> 2009-07-01 15:42:37.87800 [NOTICE] sofia.c:3423 Hangup >>>>>> sofia/maq/9876 at 58.212.219.104 [CS_NEW] [INCOMPATIBLE_DESTINATION] >>>>>> 2009-07-01 15:42:37.87800 [DEBUG] switch_channel.c:1683 Send signal >>>>>> sofia/maq/9876 at 58.212.219.104 [KILL] >>>>>> 2009-07-01 15:42:37.87800 [DEBUG] switch_core_session.c:933 Send >>>>>> signal sofia/maq/9876 at 58.212.219.104 [BREAK] >>>>>> 2009-07-01 15:42:37.87800 [DEBUG] switch_core_state_machine.c:397 ( >>>>>> sofia/maq/9876 at 58.212.219.104) Running State Change CS_HANGUP >>>>>> 2009-07-01 15:42:37.88976 [DEBUG] switch_core_state_machine.c:433 ( >>>>>> sofia/maq/9876 at 58.212.219.104) State HANGUP >>>>>> 2009-07-01 15:42:37.88976 [DEBUG] mod_sofia.c:338 Channel >>>>>> sofia/maq/9876 at 58.212.219.104 hanging up, cause: >>>>>> INCOMPATIBLE_DESTINATION >>>>>> 2009-07-01 15:42:37.88976 [DEBUG] mod_sofia.c:414 Responding to INVITE >>>>>> with: 488 >>>>>> send 676 bytes to udp/[58.212.219.104]:40508 at 07:42:37.089766: >>>>>> >>>>>> ------------------------------------------------------------------------ >>>>>> SIP/2.0 488 Not Acceptable Here >>>>>> Via: SIP/2.0/UDP 192.168.1.241:8422 >>>>>> ;branch=z9hG4bK-d8754z-94087229c9446f5f-1---d8754z-;rport=40508;received=58.212.219.104 >>>>>> From: "9876" >>>>>> >;tag=057de365 >>>>>> To: "123456" >>>>>> >;tag=28Q0QB73Bm35K >>>>>> Call-ID: M2VlMmEyYWU1MmNhMjEyMGI2NDFkMmRkNjYxYmRmYjY. >>>>>> CSeq: 1 INVITE >>>>>> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13923M >>>>>> Accept: application/sdp >>>>>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, >>>>>> SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO >>>>>> Supported: timer, precondition, path, replaces >>>>>> Allow-Events: talk, refer >>>>>> Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION" >>>>>> Content-Length: 0 >>>>>> >>>>>> >>>>>> ------------------------------------------------------------------------ >>>>>> 2009-07-01 15:42:37.89978 [DEBUG] switch_core_state_machine.c:46 >>>>>> sofia/maq/9876 at 58.212.219.104 Standard HANGUP, cause: >>>>>> INCOMPATIBLE_DESTINATION >>>>>> 2009-07-01 15:42:37.89978 [DEBUG] switch_core_state_machine.c:433 ( >>>>>> sofia/maq/9876 at 58.212.219.104) State HANGUP going to sleep >>>>>> 2009-07-01 15:42:37.89978 [DEBUG] switch_core_state_machine.c:475 ( >>>>>> sofia/maq/9876 at 58.212.219.104) State Change CS_HANGUP -> CS_REPORTING >>>>>> 2009-07-01 15:42:37.89978 [DEBUG] switch_core_session.c:933 Send >>>>>> signal sofia/maq/9876 at 58.212.219.104 [BREAK] >>>>>> 2009-07-01 15:42:37.89978 [DEBUG] switch_core_state_machine.c:397 ( >>>>>> sofia/maq/9876 at 58.212.219.104) Running State Change CS_REPORTING >>>>>> 2009-07-01 15:42:37.89978 [DEBUG] switch_core_state_machine.c:607 ( >>>>>> sofia/maq/9876 at 58.212.219.104) State REPORTING >>>>>> recv 334 bytes from udp/[58.212.219.104]:40508 at 07:42:37.093591: >>>>>> >>>>>> ------------------------------------------------------------------------ >>>>>> ACK sip:123456 at 58.212.219.104 SIP/2.0 >>>>>> Via: SIP/2.0/UDP 192.168.1.241:8422 >>>>>> ;branch=z9hG4bK-d8754z-94087229c9446f5f-1---d8754z-;rport >>>>>> To: "123456" >>>>>> >;tag=28Q0QB73Bm35K >>>>>> From: "9876" >>>>>> >;tag=057de365 >>>>>> Call-ID: M2VlMmEyYWU1MmNhMjEyMGI2NDFkMmRkNjYxYmRmYjY. >>>>>> CSeq: 1 ACK >>>>>> Content-Length: 0 >>>>>> >>>>>> >>>>>> ------------------------------------------------------------------------ >>>>>> 2009-07-01 15:42:37.133976 [DEBUG] switch_core_state_machine.c:53 >>>>>> sofia/maq/9876 at 58.212.219.104 Standard REPORTING, cause: >>>>>> INCOMPATIBLE_DESTINATION >>>>>> 2009-07-01 15:42:37.133976 [DEBUG] switch_core_state_machine.c:607 ( >>>>>> sofia/maq/9876 at 58.212.219.104) State REPORTING going to sleep >>>>>> 2009-07-01 15:42:37.133976 [DEBUG] switch_core_state_machine.c:410 ( >>>>>> sofia/maq/9876 at 58.212.219.104) State Change CS_REPORTING -> >>>>>> CS_DESTROY >>>>>> 2009-07-01 15:42:37.133976 [DEBUG] switch_core_session.c:1067 Session >>>>>> 3 (sofia/maq/9876 at 58.212.219.104) Locked, Waiting on external >>>>>> entities >>>>>> 2009-07-01 15:42:37.133976 [NOTICE] switch_core_session.c:1085 Session >>>>>> 3 (sofia/maq/9876 at 58.212.219.104) Ended >>>>>> 2009-07-01 15:42:37.133976 [NOTICE] switch_core_session.c:1087 Close >>>>>> Channel sofia/maq/9876 at 58.212.219.104 [CS_DESTROY] >>>>>> 2009-07-01 15:42:37.133976 [DEBUG] switch_core_state_machine.c:559 ( >>>>>> sofia/maq/9876 at 58.212.219.104) State DESTROY >>>>>> 2009-07-01 15:42:37.133976 [DEBUG] mod_sofia.c:255 >>>>>> sofia/maq/9876 at 58.212.219.104 SOFIA DESTROY >>>>>> 2009-07-01 15:42:37.133976 [DEBUG] switch_core_state_machine.c:60 >>>>>> sofia/maq/9876 at 58.212.219.104 Standard DESTROY >>>>>> 2009-07-01 15:42:37.133976 [DEBUG] switch_core_state_machine.c:559 ( >>>>>> sofia/maq/9876 at 58.212.219.104) State DESTROY going to sleep >>>>>> >>>>>> >>>>>> the freeswitch not accept PCMA >>>>>> >>>>>> >>>>>> >>>>>> 2009/7/1 seven >>>>>> >>>>>> absolutely not. >>>>>>> codec negotiate depending on your conf. do you have a sip trace? >>>>>>> >>>>>>> On Jul 1, 2009, at 2:48 PM, qian ma wrote: >>>>>>> >>>>>>> hi all >>>>>>> freeswitch support PCMU only? >>>>>>> i follow the >>>>>>> http://wiki.freeswitch.org/wiki/Codecs#sofia.conf.xml, but >>>>>>> freeswitch still support PCMU only, >>>>>>> below is the trace: >>>>>>> >>>>>>> 2009-07-01 14:39:35.571364 [DEBUG] sofia_glue.c:3070 Audio >>>>>>> Codec Compare [PCMA:8:8000:0]/[PCMU:0:8000:20] >>>>>>> 2009-07-01 14:39:35.571364 [DEBUG] sofia_glue.c:3030 Set 2833 dtmf >>>>>>> payload to 101 >>>>>>> 2009-07-01 14:39:35.571364 [DEBUG] sofia_glue.c:3070 Audio Codec >>>>>>> Compare [telephone-event:101:8000:0]/[PCMU:0:8000:20] >>>>>>> 2009-07-01 14:39:35.571364 [NOTICE] sofia.c:3423 Hangup >>>>>>> sofia/maq/9876 at 58.212.219.104 [CS_NEW] [INCOMPATIBLE_DESTINATION] >>>>>>> 2009-07-01 14:39:35.571364 [DEBUG] switch_channel.c:1683 Send signal >>>>>>> sofia/maq/9876 at 58.212.219.104 [KILL] >>>>>>> 2009-07-01 14:39:35.571364 [DEBUG] switch_core_session.c:933 Send >>>>>>> signal sofia/maq/9876 at 58.212.219.104 [BREAK] >>>>>>> 2009-07-01 14:39:35.572807 [DEBUG] switch_core_state_machine.c:397 ( >>>>>>> sofia/maq/9876 at 58.212.219.104) Running State Change CS_HANGUP >>>>>>> 2009-07-01 14:39:35.572807 [DEBUG] switch_core_state_machine.c:433 ( >>>>>>> sofia/maq/9876 at 58.212.219.104) State HANGUP >>>>>>> >>>>>>> >>>>>>> >>>>>>> how to configure the freeswitch?? >>>>>>> support more codecs??? >>>>>>> >>>>>>> thx! >>>>>>> >>>>>>> m.q >>>>>>> _______________________________________________ >>>>>>> Freeswitch-users mailing list >>>>>>> Freeswitch-users at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE: >>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>>> >>>>>>> >>>>>>> _______________________________________________ >>>>>>> Freeswitch-users mailing list >>>>>>> Freeswitch-users at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE: >>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>>> >>>>>> >>>>>> _______________________________________________ >>>>>> Freeswitch-users mailing list >>>>>> Freeswitch-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090702/7740cccb/attachment-0001.html From chris at fowler.cc Wed Jul 1 09:35:39 2009 From: chris at fowler.cc (Chris Fowler) Date: Wed, 1 Jul 2009 12:35:39 -0400 Subject: [Freeswitch-users] Freeswitch memory usage is too high In-Reply-To: <4A4B604F.1070302@freeswitch.org> References: <4A4B604F.1070302@freeswitch.org> Message-ID: <7454A296C7EDE34EA57199FAA401E2F115F5D09972@VMBX113.ihostexchange.net> Hi Ray, This was a problem some time ago (couple of months ago). Are you running the latest build? Chris. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Raymond Chandler Sent: Wednesday, July 01, 2009 6:11 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Freeswitch memory usage is too high On 07/01/2009 08:29 AM, Muhammad Danish Moosa wrote: > Hi > > Freeswitch is being used in a scenario where two endpoints are running > traffic with bypass media mode. Performance is good and all things are > smooth. > > But as the time goes after starting freeswitch, it starts consuming > almost whole of memory. How much is the "whole"? You should see the memory usage level off, it won't keep growing forever. -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 From mattdfong at gmail.com Wed Jul 1 09:46:53 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Wed, 1 Jul 2009 09:46:53 -0700 Subject: [Freeswitch-users] Freeswitch memory usage is too high In-Reply-To: References: Message-ID: <4256bf830907010946r71656946y1839c9b8478b32d2@mail.gmail.com> bkw, you said "Downgrading. I suspect its an issue with your lua sql module not linking to the thread safe client." in the Jira ticket. I'm curious how one would go about doing this. I use luasql (the default ubuntu apt-get install) and have a similar memory problem. I suppose I would need to compile luasql with some sort of flag? --matt On Wed, Jul 1, 2009 at 7:18 AM, Brian West wrote: > You also have a jira http://jira.freeswitch.org/browse/MODAPP-298, It > looks like you're using lua sql and the backtrace you attached to the > jira was cut off right before the data I needed to see... can you > follow up on that ASAP? > > It looks like a crash in libmysql from the last line but again I can't > see the rest of it. > > /b > > On Jul 1, 2009, at 7:29 AM, Muhammad Danish Moosa wrote: > >> Hi >> >> Freeswitch is being used in a scenario where two endpoints are >> running traffic with bypass media mode. Performance is good and all >> things are smooth. >> >> But as the time goes after starting freeswitch, it starts consuming >> almost whole of memory. Note , freeswitch is being started with - >> core option, is it related? >> >> If this 99% memory consumption is any red alert, as we can see calls >> are still connecting fine and all is going as usual. >> >> >> -- >> Muhammad Danish Moosa >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From mrene_lists at avgs.ca Wed Jul 1 10:00:03 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Wed, 1 Jul 2009 13:00:03 -0400 Subject: [Freeswitch-users] How to check if a channel variable is present or NULL In-Reply-To: References: Message-ID: <469F2465-8396-4992-85FF-7F5390129833@avgs.ca> it will match ^$ if the var isnt defined Math On 1-Jul-09, at 12:25 PM, Cavalera Claudio Luigi wrote: > Hello, > I'm trying to implement this kind of logic in the dialplan > > if the channel variable sip_refer_to matches regexp > than do action > else if sip_refer_to exists (not NULL) but does not match regexp > than do anti-action > else if sip_refer_to does not exist as a channel_variable (NULL ??) > than do another action > > is there a way to check if a channel variable exists or not? > Maybe a special regexp to match? > > 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. > ----------------------------------------------------------------------------------------------------- > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mitcheloc at gmail.com Wed Jul 1 10:04:03 2009 From: mitcheloc at gmail.com (mitcheloc) Date: Wed, 1 Jul 2009 10:04:03 -0700 Subject: [Freeswitch-users] Could this be a bug in the SIP registry? In-Reply-To: <20090701084547.GA11743@jdc.jasonjgw.net> References: <20090701082332.GA10823@jdc.jasonjgw.net> <20090701084547.GA11743@jdc.jasonjgw.net> Message-ID: Jason, Thanks for the reply. I tried the commands as suggested: freeswitch at internal> originate user/mythicalbox at weavver.com 3000 -ERR SUBSCRIBER_ABSENT 2009-07-01 09:43:16 [ERR] switch_xml.c:1555 switch_xml_locate() Error[[error near line 1]: root tag missing] freeswitch at internal> 2009-07-01 09:43:16 [WARNING] mod_dptools.c:2364 user_outgoing_channel() Can't find user [mythicalbox at weavver.com] 2009-07-01 09:43:16 [ERR] switch_ivr_originate.c:1494 switch_ivr_originate() Cannot create outgoing channel of type [user] cause: [SUBSCRIBER_ABSENT] 2009-07-01 09:43:16 [DEBUG] switch_ivr_originate.c:2101 switch_ivr_originate() Originate Resulted in Error Cause: 20 [SUBSCRIBER_ABSENT] Trying the sofia_contact function: freeswitch at internal> expand echo ${sofia_contact(profile/ mythicalbox at weavver.com)} error/facility_not_subscribed Here is some output to show that parts of FreeSWITCH do think that the phone is registered: freeswitch at internal> sofia status profile internal ================================================================================================= Name internal Domain Name N/A DBName sofia_reg_internal Pres Hosts Dialplan XML Context public Challenge Realm auto_from RTP-IP 205.134.225.20 SIP-IP 205.134.225.20 URL sip:mod_sofia at 205.134.225.20:5060 BIND-URL sip:mod_sofia at 205.134.225.20:5060 HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG false PROXY-MEDIA false AGGRESSIVENAT true STUN-ENABLED true STUN-AUTO-DISABLE false CALLS-IN 9 FAILED-CALLS-IN 3 CALLS-OUT 8 FAILED-CALLS-OUT 18 Registrations: ================================================================================================= Call-ID: ZDg4NDU3MjI2ODVlZmZiNGYzZDYzNmRkOTYxMmNhMDY. User: mythicalbox at weavver.com Contact: "mythicalbox" Agent: eyeBeam release 1102u stamp 52344 Status: Registered(TCP-NAT)(unknown) EXP(2009-07-01 11:33:18) Host: duck.weavver.com IP: 64.183.110.250 Port: 8443 Auth-User: mythicalbox Auth-Realm: weavver.com ================================================================================================= FreeSWITCH and my Softphone (eyeBeam) shows me as registered when polling for registrations but when trying to connect the call FreeSWITCH is not seeing it as registered. Any more ideas? Thanks again! On Wed, Jul 1, 2009 at 1:45 AM, Jason White wrote: > Jason White wrote: > > originate user/1000 at example.com 3000 > > to connext user at example.com to extension 3000. > > That should read "to connect 1000 at example.com to extension 3000". > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Mitchel Constantin Weavver, Inc. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090701/540613b6/attachment.html From msc at freeswitch.org Wed Jul 1 10:10:46 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 1 Jul 2009 10:10:46 -0700 Subject: [Freeswitch-users] Could this be a bug in the SIP registry? In-Reply-To: References: <20090701082332.GA10823@jdc.jasonjgw.net> <20090701084547.GA11743@jdc.jasonjgw.net> Message-ID: <87f2f3b90907011010p7e73269agf0c6ee93195b41b3@mail.gmail.com> On Wed, Jul 1, 2009 at 10:04 AM, mitcheloc wrote: > Jason, > > Thanks for the reply. I tried the commands as suggested: > > freeswitch at internal> originate user/mythicalbox at weavver.com 3000 > -ERR SUBSCRIBER_ABSENT > I suspect the following line is a clue: > > 2009-07-01 09:43:16 [ERR] switch_xml.c:1555 switch_xml_locate() > Error[[error near line 1]: root tag missing] > Can you confirm the XML that is getting read? -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090701/1c8eae8e/attachment.html From brian at freeswitch.org Wed Jul 1 10:14:17 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 1 Jul 2009 12:14:17 -0500 Subject: [Freeswitch-users] Could this be a bug in the SIP registry? In-Reply-To: References: <20090701082332.GA10823@jdc.jasonjgw.net> <20090701084547.GA11743@jdc.jasonjgw.net> Message-ID: <5BAB85E8-C0C3-45C4-9C70-657961E0C0F1@freeswitch.org> What does 'sofia status' say? expand echo ${sofia_contact(internal/mythicalbox at weavver.com)} <-- notice I put the profile name instead of the word "profile" /b On Jul 1, 2009, at 12:04 PM, mitcheloc wrote: > Jason, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090701/afea5400/attachment-0001.html From mitcheloc at gmail.com Wed Jul 1 11:07:54 2009 From: mitcheloc at gmail.com (mitcheloc) Date: Wed, 1 Jul 2009 11:07:54 -0700 Subject: [Freeswitch-users] Could this be a bug in the SIP registry? In-Reply-To: <5BAB85E8-C0C3-45C4-9C70-657961E0C0F1@freeswitch.org> References: <20090701082332.GA10823@jdc.jasonjgw.net> <20090701084547.GA11743@jdc.jasonjgw.net> <5BAB85E8-C0C3-45C4-9C70-657961E0C0F1@freeswitch.org> Message-ID: Brian, Oh yay! Good catch.. it gave me output this time, and I could make a call using it: sofia/internal/sip:mythicalbox at 64.183.110.250:9136 ;rinstance=24c9b78f5fc6c759;transport=TCP;fs_nat=yes;fs_path=sip%3Amythicalbox%4064.183.110.250%3A9136%3Brinstance%3D24c9b78f5fc6c759%3Btransport%3DTCP That is definitely not what I'd been trying! Here is sofia status in case you still want it: freeswitch at internal> sofia status Name Type Data State ================================================================================================= internal profile sip:mod_sofia at 205.134.225.20:5060 RUNNING (0) external profile sip:mod_sofia at 205.134.225.20:5080 RUNNING (0) example.com gateway sip:joeuser at example.com NOREG internal-ipv6 profile sip:mod_sofia@[::1]:5060 RUNNING (0) default alias internal ALIASED nat alias external ALIASED outbound alias external ALIASED 205.134.225.20 alias external ALIASED ================================================================================================= 3 profiles 4 aliases Thank you!! On Wed, Jul 1, 2009 at 10:14 AM, Brian West wrote: > What does 'sofia status' say? > expand echo ${sofia_contact(internal/mythicalbox at weavver.com)} <-- notice > I put the profile name instead of the word "profile" > > /b > > > > On Jul 1, 2009, at 12:04 PM, mitcheloc wrote: > > 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 > > -- Mitchel Constantin Weavver, Inc. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090701/a715a368/attachment.html From danishmoosa at gmail.com Wed Jul 1 10:42:04 2009 From: danishmoosa at gmail.com (Muhammad Danish Moosa) Date: Wed, 1 Jul 2009 23:42:04 +0600 Subject: [Freeswitch-users] Freeswitch memory usage is too high Message-ID: Hi Brian My customer is now using lua odbc ( not lua mysql anymore) and problem mentioned in jira is resolved now. * http://www.mail-archive.com/freeswitch-dev at lists.freeswitch.org/msg01352.html * * *This seems to answer my question, rite? BTW , starting FS with following 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 ulimit -a Date: Wed, 1 Jul 2009 09:18:57 -0500 From: Brian West Subject: Re: [Freeswitch-users] Freeswitch memory usage is too high To: freeswitch-users at lists.freeswitch.org Message-ID: Content-Type: text/plain; charset=US-ASCII; format=flowed; delsp=yes You also have a jira http://jira.freeswitch.org/browse/MODAPP-298, It looks like you're using lua sql and the backtrace you attached to the jira was cut off right before the data I needed to see... can you follow up on that ASAP? It looks like a crash in libmysql from the last line but again I can't see the rest of it. /b On Jul 1, 2009, at 7:29 AM, Muhammad Danish Moosa wrote: > Hi > > Freeswitch is being used in a scenario where two endpoints are > running traffic with bypass media mode. Performance is good and all > things are smooth. > > But as the time goes after starting freeswitch, it starts consuming > almost whole of memory. Note , freeswitch is being started with - > core option, is it related? > > If this 99% memory consumption is any red alert, as we can see calls > are still connecting fine and all is going as usual. > > > -- > Muhammad Danish Moosa > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Muhammad Danish Moosa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090701/b65ed61b/attachment.html From brian at freeswitch.org Wed Jul 1 12:27:43 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 1 Jul 2009 14:27:43 -0500 Subject: [Freeswitch-users] Freeswitch memory usage is too high In-Reply-To: References: Message-ID: <96DAA879-CB53-4B76-B6B0-8684D5A40DE2@freeswitch.org> If the problem is resolved please follow up on the jira. /b On Jul 1, 2009, at 12:42 PM, Muhammad Danish Moosa wrote: > My customer is now using lua odbc ( not lua mysql anymore) and > problem mentioned in jira is resolved now. > From god.nirvana at gmail.com Wed Jul 1 11:33:07 2009 From: god.nirvana at gmail.com (qian ma) Date: Thu, 2 Jul 2009 02:33:07 +0800 Subject: [Freeswitch-users] freeswitch support PCMU only? In-Reply-To: <7d0bfd8c0907010929y11e595fbj2a50d2df47a8ee50@mail.gmail.com> References: <49161ef40906302348o44b79b2au9349caec5cbb8e16@mail.gmail.com> <391A7903-3BEF-45FE-8134-321D2865A6A6@gmail.com> <49161ef40907010044t415df111qbacd95680e06d005@mail.gmail.com> <7d0bfd8c0907010159l416d2585p2463b198d8741c6@mail.gmail.com> <49161ef40907010527i1f2b79b1h6172cc802fa06aa@mail.gmail.com> <7d0bfd8c0907010614y1bca1fddya0b445d005a4c78c@mail.gmail.com> <7d0bfd8c0907010620q4e77a8a3i9c4266e8ea599368@mail.gmail.com> <49161ef40907010625xaf3679am5c02f1a7699268e4@mail.gmail.com> <7d0bfd8c0907010929y11e595fbj2a50d2df47a8ee50@mail.gmail.com> Message-ID: <49161ef40907011133s6627a82fn6de4d73504a2201b@mail.gmail.com> yes,PCMA enabled in x-lite. doesn't work. FS accept PCMU only. 2009/7/2 Nandy Dagondon > is PCMA enabled in X-Lite, too? > > > On Wed, Jul 1, 2009 at 9:25 PM, qian ma wrote: > >> >> inbound_codec_negotiation is generous >> and the xlite PCMU is enabled. >> >> my var.xml.conf: >> >> >> >> >> >> 2009/7/1 Nandy Dagondon >> >>> sorry. i mean check the x-lite client if PCMA is enabled? >>> >>> >>> On Wed, Jul 1, 2009 at 9:14 PM, Nandy Dagondon wrote: >>> >>>> check the value of "inbound_codec_negotiation" in the >>>> sip_profiles/*.xml files. is it "generous" or "greedy"? you should also >>>> check if the endpoint is offering PCMU. >>>> >>>> >>>> >>>> On Wed, Jul 1, 2009 at 8:27 PM, qian ma wrote: >>>> >>>>> i want the fs accept the PCMA not PCMU. >>>>> i add "PCMA" in global_codec_prefs and outbound_codec_prefs, it doesn't >>>>> work. FS only accept PCMU. >>>>> why?? >>>>> >>>>> >>>>> >>>>> >>>>> 2009/7/1 Nandy Dagondon >>>>> >>>>> you FS doesn't accept PCMU. try to add "PCMU" on both variables. >>>>>> >>>>>> >>>>>> On Wed, Jul 1, 2009 at 3:44 PM, qian ma wrote: >>>>>> >>>>>>> thanks for your replies. >>>>>>> >>>>>> my var.xml: >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> below is the sip trace: >>>>>>> recv 781 bytes from udp/[58.212.219.104]:40508 at 07:42:37.086711: >>>>>>> >>>>>>> ------------------------------------------------------------------------ >>>>>>> INVITE sip:123456 at 58.212.219.104 SIP/2.0 >>>>>>> Via: SIP/2.0/UDP 192.168.1.241:8422 >>>>>>> ;branch=z9hG4bK-d8754z-94087229c9446f5f-1---d8754z-;rport >>>>>>> Max-Forwards: 70 >>>>>>> Contact: >>>>>>> To: "123456" >>>>>>> > >>>>>>> From: "9876" >>>>>>> >;tag=057de365 >>>>>>> Call-ID: M2VlMmEyYWU1MmNhMjEyMGI2NDFkMmRkNjYxYmRmYjY. >>>>>>> CSeq: 1 INVITE >>>>>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, >>>>>>> SUBSCRIBE, INFO >>>>>>> Content-Type: application/sdp >>>>>>> User-Agent: eyeBeam release 1102u stamp 52345 >>>>>>> Content-Length: 237 >>>>>>> >>>>>>> v=0 >>>>>>> o=- 6 2 IN IP4 192.168.1.241 >>>>>>> s=CounterPath eyeBeam 1.5 >>>>>>> c=IN IP4 192.168.1.241 >>>>>>> t=0 0 >>>>>>> m=audio 57862 RTP/AVP 8 101 >>>>>>> a=alt:1 1 : vX+dFqlN ruK7YVXO 192.168.1.241 57862 >>>>>>> a=fmtp:101 0-15 >>>>>>> a=rtpmap:101 telephone-event/8000 >>>>>>> a=sendrecv >>>>>>> >>>>>>> ------------------------------------------------------------------------ >>>>>>> send 380 bytes to udp/[58.212.219.104]:40508 at 07:42:37.087505: >>>>>>> >>>>>>> ------------------------------------------------------------------------ >>>>>>> SIP/2.0 100 Trying >>>>>>> Via: SIP/2.0/UDP 192.168.1.241:8422 >>>>>>> ;branch=z9hG4bK-d8754z-94087229c9446f5f-1---d8754z-;rport=40508;received=58.212.219.104 >>>>>>> From: "9876" >>>>>>> >;tag=057de365 >>>>>>> To: "123456" >>>>>>> > >>>>>>> Call-ID: M2VlMmEyYWU1MmNhMjEyMGI2NDFkMmRkNjYxYmRmYjY. >>>>>>> CSeq: 1 INVITE >>>>>>> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13923M >>>>>>> Content-Length: 0 >>>>>>> >>>>>>> >>>>>>> ------------------------------------------------------------------------ >>>>>>> 2009-07-01 15:42:37.87800 [NOTICE] switch_channel.c:602 New Channel >>>>>>> sofia/maq/9876 at 58.212.219.104 [574766e1-e71f-4a62-a659-1d43bea135bd] >>>>>>> 2009-07-01 15:42:37.87800 [DEBUG] sofia.c:3214 Channel >>>>>>> sofia/maq/9876 at 58.212.219.104 entering state [received][100] >>>>>>> 2009-07-01 15:42:37.87800 [DEBUG] sofia.c:3221 Remote SDP: >>>>>>> v=0 >>>>>>> o=- 6 2 IN IP4 192.168.1.241 >>>>>>> s=CounterPath eyeBeam 1.5 >>>>>>> c=IN IP4 192.168.1.241 >>>>>>> t=0 0 >>>>>>> m=audio 57862 RTP/AVP 8 101 >>>>>>> a=rtpmap:101 telephone-event/8000 >>>>>>> a=fmtp:101 0-15 >>>>>>> a=alt:1 1 : vX+dFqlN ruK7YVXO 192.168.1.241 57862 >>>>>>> 2009-07-01 15:42:37.87800 [DEBUG] sofia_glue.c:3070 Audio Codec >>>>>>> Compare [PCMA:8:8000:0]/[PCMU:0:8000:20] >>>>>>> 2009-07-01 15:42:37.87800 [DEBUG] sofia_glue.c:3030 Set 2833 dtmf >>>>>>> payload to 101 >>>>>>> 2009-07-01 15:42:37.87800 [DEBUG] sofia_glue.c:3070 Audio Codec >>>>>>> Compare [telephone-event:101:8000:0]/[PCMU:0:8000:20] >>>>>>> 2009-07-01 15:42:37.87800 [NOTICE] sofia.c:3423 Hangup >>>>>>> sofia/maq/9876 at 58.212.219.104 [CS_NEW] [INCOMPATIBLE_DESTINATION] >>>>>>> 2009-07-01 15:42:37.87800 [DEBUG] switch_channel.c:1683 Send signal >>>>>>> sofia/maq/9876 at 58.212.219.104 [KILL] >>>>>>> 2009-07-01 15:42:37.87800 [DEBUG] switch_core_session.c:933 Send >>>>>>> signal sofia/maq/9876 at 58.212.219.104 [BREAK] >>>>>>> 2009-07-01 15:42:37.87800 [DEBUG] switch_core_state_machine.c:397 ( >>>>>>> sofia/maq/9876 at 58.212.219.104) Running State Change CS_HANGUP >>>>>>> 2009-07-01 15:42:37.88976 [DEBUG] switch_core_state_machine.c:433 ( >>>>>>> sofia/maq/9876 at 58.212.219.104) State HANGUP >>>>>>> 2009-07-01 15:42:37.88976 [DEBUG] mod_sofia.c:338 Channel >>>>>>> sofia/maq/9876 at 58.212.219.104 hanging up, cause: >>>>>>> INCOMPATIBLE_DESTINATION >>>>>>> 2009-07-01 15:42:37.88976 [DEBUG] mod_sofia.c:414 Responding to >>>>>>> INVITE with: 488 >>>>>>> send 676 bytes to udp/[58.212.219.104]:40508 at 07:42:37.089766: >>>>>>> >>>>>>> ------------------------------------------------------------------------ >>>>>>> SIP/2.0 488 Not Acceptable Here >>>>>>> Via: SIP/2.0/UDP 192.168.1.241:8422 >>>>>>> ;branch=z9hG4bK-d8754z-94087229c9446f5f-1---d8754z-;rport=40508;received=58.212.219.104 >>>>>>> From: "9876" >>>>>>> >;tag=057de365 >>>>>>> To: "123456" >>>>>>> >;tag=28Q0QB73Bm35K >>>>>>> Call-ID: M2VlMmEyYWU1MmNhMjEyMGI2NDFkMmRkNjYxYmRmYjY. >>>>>>> CSeq: 1 INVITE >>>>>>> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13923M >>>>>>> Accept: application/sdp >>>>>>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, >>>>>>> SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO >>>>>>> Supported: timer, precondition, path, replaces >>>>>>> Allow-Events: talk, refer >>>>>>> Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION" >>>>>>> Content-Length: 0 >>>>>>> >>>>>>> >>>>>>> ------------------------------------------------------------------------ >>>>>>> 2009-07-01 15:42:37.89978 [DEBUG] switch_core_state_machine.c:46 >>>>>>> sofia/maq/9876 at 58.212.219.104 Standard HANGUP, cause: >>>>>>> INCOMPATIBLE_DESTINATION >>>>>>> 2009-07-01 15:42:37.89978 [DEBUG] switch_core_state_machine.c:433 ( >>>>>>> sofia/maq/9876 at 58.212.219.104) State HANGUP going to sleep >>>>>>> 2009-07-01 15:42:37.89978 [DEBUG] switch_core_state_machine.c:475 ( >>>>>>> sofia/maq/9876 at 58.212.219.104) State Change CS_HANGUP -> >>>>>>> CS_REPORTING >>>>>>> 2009-07-01 15:42:37.89978 [DEBUG] switch_core_session.c:933 Send >>>>>>> signal sofia/maq/9876 at 58.212.219.104 [BREAK] >>>>>>> 2009-07-01 15:42:37.89978 [DEBUG] switch_core_state_machine.c:397 ( >>>>>>> sofia/maq/9876 at 58.212.219.104) Running State Change CS_REPORTING >>>>>>> 2009-07-01 15:42:37.89978 [DEBUG] switch_core_state_machine.c:607 ( >>>>>>> sofia/maq/9876 at 58.212.219.104) State REPORTING >>>>>>> recv 334 bytes from udp/[58.212.219.104]:40508 at 07:42:37.093591: >>>>>>> >>>>>>> ------------------------------------------------------------------------ >>>>>>> ACK sip:123456 at 58.212.219.104 SIP/2.0 >>>>>>> Via: SIP/2.0/UDP 192.168.1.241:8422 >>>>>>> ;branch=z9hG4bK-d8754z-94087229c9446f5f-1---d8754z-;rport >>>>>>> To: "123456" >>>>>>> >;tag=28Q0QB73Bm35K >>>>>>> From: "9876" >>>>>>> >;tag=057de365 >>>>>>> Call-ID: M2VlMmEyYWU1MmNhMjEyMGI2NDFkMmRkNjYxYmRmYjY. >>>>>>> CSeq: 1 ACK >>>>>>> Content-Length: 0 >>>>>>> >>>>>>> >>>>>>> ------------------------------------------------------------------------ >>>>>>> 2009-07-01 15:42:37.133976 [DEBUG] switch_core_state_machine.c:53 >>>>>>> sofia/maq/9876 at 58.212.219.104 Standard REPORTING, cause: >>>>>>> INCOMPATIBLE_DESTINATION >>>>>>> 2009-07-01 15:42:37.133976 [DEBUG] switch_core_state_machine.c:607 ( >>>>>>> sofia/maq/9876 at 58.212.219.104) State REPORTING going to sleep >>>>>>> 2009-07-01 15:42:37.133976 [DEBUG] switch_core_state_machine.c:410 ( >>>>>>> sofia/maq/9876 at 58.212.219.104) State Change CS_REPORTING -> >>>>>>> CS_DESTROY >>>>>>> 2009-07-01 15:42:37.133976 [DEBUG] switch_core_session.c:1067 Session >>>>>>> 3 (sofia/maq/9876 at 58.212.219.104) Locked, Waiting on external >>>>>>> entities >>>>>>> 2009-07-01 15:42:37.133976 [NOTICE] switch_core_session.c:1085 >>>>>>> Session 3 (sofia/maq/9876 at 58.212.219.104) Ended >>>>>>> 2009-07-01 15:42:37.133976 [NOTICE] switch_core_session.c:1087 Close >>>>>>> Channel sofia/maq/9876 at 58.212.219.104 [CS_DESTROY] >>>>>>> 2009-07-01 15:42:37.133976 [DEBUG] switch_core_state_machine.c:559 ( >>>>>>> sofia/maq/9876 at 58.212.219.104) State DESTROY >>>>>>> 2009-07-01 15:42:37.133976 [DEBUG] mod_sofia.c:255 >>>>>>> sofia/maq/9876 at 58.212.219.104 SOFIA DESTROY >>>>>>> 2009-07-01 15:42:37.133976 [DEBUG] switch_core_state_machine.c:60 >>>>>>> sofia/maq/9876 at 58.212.219.104 Standard DESTROY >>>>>>> 2009-07-01 15:42:37.133976 [DEBUG] switch_core_state_machine.c:559 ( >>>>>>> sofia/maq/9876 at 58.212.219.104) State DESTROY going to sleep >>>>>>> >>>>>>> >>>>>>> the freeswitch not accept PCMA >>>>>>> >>>>>>> >>>>>>> >>>>>>> 2009/7/1 seven >>>>>>> >>>>>>> absolutely not. >>>>>>>> codec negotiate depending on your conf. do you have a sip trace? >>>>>>>> >>>>>>>> On Jul 1, 2009, at 2:48 PM, qian ma wrote: >>>>>>>> >>>>>>>> hi all >>>>>>>> freeswitch support PCMU only? >>>>>>>> i follow the >>>>>>>> http://wiki.freeswitch.org/wiki/Codecs#sofia.conf.xml, but >>>>>>>> freeswitch still support PCMU only, >>>>>>>> below is the trace: >>>>>>>> >>>>>>>> 2009-07-01 14:39:35.571364 [DEBUG] sofia_glue.c:3070 Audio >>>>>>>> Codec Compare [PCMA:8:8000:0]/[PCMU:0:8000:20] >>>>>>>> 2009-07-01 14:39:35.571364 [DEBUG] sofia_glue.c:3030 Set 2833 dtmf >>>>>>>> payload to 101 >>>>>>>> 2009-07-01 14:39:35.571364 [DEBUG] sofia_glue.c:3070 Audio Codec >>>>>>>> Compare [telephone-event:101:8000:0]/[PCMU:0:8000:20] >>>>>>>> 2009-07-01 14:39:35.571364 [NOTICE] sofia.c:3423 Hangup >>>>>>>> sofia/maq/9876 at 58.212.219.104 [CS_NEW] [INCOMPATIBLE_DESTINATION] >>>>>>>> 2009-07-01 14:39:35.571364 [DEBUG] switch_channel.c:1683 Send signal >>>>>>>> sofia/maq/9876 at 58.212.219.104 [KILL] >>>>>>>> 2009-07-01 14:39:35.571364 [DEBUG] switch_core_session.c:933 Send >>>>>>>> signal sofia/maq/9876 at 58.212.219.104 [BREAK] >>>>>>>> 2009-07-01 14:39:35.572807 [DEBUG] switch_core_state_machine.c:397 ( >>>>>>>> sofia/maq/9876 at 58.212.219.104) Running State Change CS_HANGUP >>>>>>>> 2009-07-01 14:39:35.572807 [DEBUG] switch_core_state_machine.c:433 ( >>>>>>>> sofia/maq/9876 at 58.212.219.104) State HANGUP >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> how to configure the freeswitch?? >>>>>>>> support more codecs??? >>>>>>>> >>>>>>>> thx! >>>>>>>> >>>>>>>> m.q >>>>>>>> _______________________________________________ >>>>>>>> Freeswitch-users mailing list >>>>>>>> Freeswitch-users at lists.freeswitch.org >>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>> UNSUBSCRIBE: >>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>> http://www.freeswitch.org >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> _______________________________________________ >>>>>>>> Freeswitch-users mailing list >>>>>>>> Freeswitch-users at lists.freeswitch.org >>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>> UNSUBSCRIBE: >>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>> http://www.freeswitch.org >>>>>>>> >>>>>>>> >>>>>>> >>>>>>> _______________________________________________ >>>>>>> Freeswitch-users mailing list >>>>>>> Freeswitch-users at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE: >>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>>> >>>>>> >>>>>> _______________________________________________ >>>>>> Freeswitch-users mailing list >>>>>> Freeswitch-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090702/a800d55f/attachment-0001.html From brian at freeswitch.org Wed Jul 1 12:35:16 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 1 Jul 2009 14:35:16 -0500 Subject: [Freeswitch-users] freeswitch support PCMU only? In-Reply-To: <49161ef40907011133s6627a82fn6de4d73504a2201b@mail.gmail.com> References: <49161ef40906302348o44b79b2au9349caec5cbb8e16@mail.gmail.com> <391A7903-3BEF-45FE-8134-321D2865A6A6@gmail.com> <49161ef40907010044t415df111qbacd95680e06d005@mail.gmail.com> <7d0bfd8c0907010159l416d2585p2463b198d8741c6@mail.gmail.com> <49161ef40907010527i1f2b79b1h6172cc802fa06aa@mail.gmail.com> <7d0bfd8c0907010614y1bca1fddya0b445d005a4c78c@mail.gmail.com> <7d0bfd8c0907010620q4e77a8a3i9c4266e8ea599368@mail.gmail.com> <49161ef40907010625xaf3679am5c02f1a7699268e4@mail.gmail.com> <7d0bfd8c0907010929y11e595fbj2a50d2df47a8ee50@mail.gmail.com> <49161ef40907011133s6627a82fn6de4d73504a2201b@mail.gmail.com> Message-ID: <00860BFC-4E84-403A-AB42-972330C01F18@freeswitch.org> Thats bull... I just did PCMA all morning testing! Your config is wrong. /b On Jul 1, 2009, at 1:33 PM, qian ma wrote: > yes,PCMA enabled in x-lite. > > doesn't work. > > FS accept PCMU only. From dftoro at yahoo.com Wed Jul 1 12:41:01 2009 From: dftoro at yahoo.com (Diego Toro) Date: Wed, 1 Jul 2009 12:41:01 -0700 (PDT) Subject: [Freeswitch-users] sound_prefix is changed Message-ID: <431342.62083.qm@web33507.mail.mud.yahoo.com> hi all, I am working with release Pre9, I have a problem now?with say module, the sound_prefix var is changed to SWITCH_GLOBAL_dirs.base_dir/sounds/es for default_language to 'es'. ? I checked c code on switch_ivr_say the value of sound_prefix is changed always ? Diego -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090701/b14586ad/attachment.html From nik.middleton at noblesolutions.co.uk Wed Jul 1 12:51:11 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Wed, 1 Jul 2009 20:51:11 +0100 Subject: [Freeswitch-users] freeswitch support PCMU only? In-Reply-To: <00860BFC-4E84-403A-AB42-972330C01F18@freeswitch.org> References: <49161ef40906302348o44b79b2au9349caec5cbb8e16@mail.gmail.com><391A7903-3BEF-45FE-8134-321D2865A6A6@gmail.com><49161ef40907010044t415df111qbacd95680e06d005@mail.gmail.com><7d0bfd8c0907010159l416d2585p2463b198d8741c6@mail.gmail.com><49161ef40907010527i1f2b79b1h6172cc802fa06aa@mail.gmail.com><7d0bfd8c0907010614y1bca1fddya0b445d005a4c78c@mail.gmail.com><7d0bfd8c0907010620q4e77a8a3i9c4266e8ea599368@mail.gmail.com><49161ef40907010625xaf3679am5c02f1a7699268e4@mail.gmail.com><7d0bfd8c0907010929y11e595fbj2a50d2df47a8ee50@mail.gmail.com><49161ef40907011133s6627a82fn6de4d73504a2201b@mail.gmail.com> <00860BFC-4E84-403A-AB42-972330C01F18@freeswitch.org> Message-ID: I'm ONLY use PCMA, so I would agree with Brian -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: 01 July 2009 20:35 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] freeswitch support PCMU only? Thats bull... I just did PCMA all morning testing! Your config is wrong. /b On Jul 1, 2009, at 1:33 PM, qian ma wrote: > yes,PCMA enabled in x-lite. > > doesn't work. > > FS accept PCMU only. _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From pjintheusa at gmail.com Wed Jul 1 13:22:14 2009 From: pjintheusa at gmail.com (Phillip Jones) Date: Wed, 1 Jul 2009 16:22:14 -0400 Subject: [Freeswitch-users] SIP re-invite / bypass_media Message-ID: <367751820907011322s1b17192ct3d7303632456db87@mail.gmail.com> Hi there, I was wondering whether it is possible to have FreeSwitch go into bypass_media mode on demand? For instance, leg a bridges to leg b - leg b is invited to accept the call by pressing 1. I want to go to bypass_media (do a SIP reinvite to reroute the media) after the one is pressed. Currently I am issuing the following from my js script that prompts for the 1: session.apiExecute("uuid_media",session.uuid); Not working however. Any help to get me going would be appreciated. Thanks Phillip Jones. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090701/dc94fb97/attachment.html From anthony.minessale at gmail.com Wed Jul 1 13:36:33 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 1 Jul 2009 15:36:33 -0500 Subject: [Freeswitch-users] sound_prefix is changed In-Reply-To: <431342.62083.qm@web33507.mail.mud.yahoo.com> References: <431342.62083.qm@web33507.mail.mud.yahoo.com> Message-ID: <191c3a030907011336o391d11c6y53b3c73ea6dd5eb3@mail.gmail.com> what is the problem exactly? what do you want it to change to? On Wed, Jul 1, 2009 at 2:41 PM, Diego Toro wrote: > hi all, > I am working with release Pre9, I have a problem now with say module, the > sound_prefix var is changed to SWITCH_GLOBAL_dirs.base_dir/sounds/es for > default_language to 'es'. > > I checked c code on switch_ivr_say the value of sound_prefix is changed > always > > Diego > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference 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/20090701/57e539cc/attachment.html From shaheryarkh at googlemail.com Wed Jul 1 14:04:34 2009 From: shaheryarkh at googlemail.com (Muhammad Shahzad) Date: Thu, 2 Jul 2009 03:04:34 +0600 Subject: [Freeswitch-users] G723 timer problem Message-ID: Hi, I am using FS svn revision 14046 and trying to send call from SIP Dialer to a SIP gateway using G723 in passthrough mode. Everything works perfect and destination rings but then call drops with following error on FS CLI, 2009-07-02 02:39:28.790508 [WARNING] mod_sofia.c:807 We were told to use ptime 30 but what they meant to say was 60 This issue has so far been identified to happen on the following broken platforms/devices: Linksys/Sipura aka Cisco ShoreTel Sonus/L3 We will try to fix it but some of the devices on this list are so broken who knows what will happen.. 2009-07-02 02:39:28.790508 [WARNING] switch_core_codec.c:499 Codec G723 Exists but not at the desired implementation. 8000hz 60ms Is there any work around for this or i have downgrade my server back to Asterisk. :'-( Thank you. -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090702/45c95653/attachment.html From dftoro at yahoo.com Wed Jul 1 14:10:31 2009 From: dftoro at yahoo.com (Diego Toro) Date: Wed, 1 Jul 2009 14:10:31 -0700 (PDT) Subject: [Freeswitch-users] sound_prefix is changed Message-ID: <665906.28719.qm@web33503.mail.mud.yahoo.com> Hello, the problem is that sound_prefix value is ignored and it's changed to SWITCH_GLOBAL_dirs.base_dir/sounds/language? (with language=en), so audio files are not found . Diego ? --- On Wed, 7/1/09, Anthony Minessale wrote: From: Anthony Minessale Subject: Re: [Freeswitch-users] sound_prefix is changed To: freeswitch-users at lists.freeswitch.org Date: Wednesday, July 1, 2009, 3:36 PM what is the problem exactly? what do you want it to change to? On Wed, Jul 1, 2009 at 2:41 PM, Diego Toro wrote: hi all, I am working with release Pre9, I have a problem now?with say module, the sound_prefix var is changed to SWITCH_GLOBAL_dirs.base_dir/sounds/es for default_language to 'es'. ? I checked c code on switch_ivr_say the value of sound_prefix is changed always ? Diego _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -----Inline Attachment Follows----- _______________________________________________ Freeswitch-users mailing list Freeswitch-users at 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/20090701/61d80c71/attachment-0001.html From brian at freeswitch.org Wed Jul 1 14:13:00 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 1 Jul 2009 16:13:00 -0500 Subject: [Freeswitch-users] G723 timer problem In-Reply-To: References: Message-ID: You have two choices... set codec neg. to scrooge or get a provider that doesn't lie about the ptime in their SDP. /b On Jul 1, 2009, at 4:04 PM, Muhammad Shahzad wrote: > Hi, > > I am using FS svn revision 14046 and trying to send call from SIP > Dialer to a SIP gateway using G723 in passthrough mode. Everything > works perfect and destination rings but then call drops with > following error on FS CLI, > > > 2009-07-02 02:39:28.790508 [WARNING] mod_sofia.c:807 We were told to > use ptime 30 but what they meant to say was 60 > This issue has so far been identified to happen on the following > broken platforms/devices: > Linksys/Sipura aka Cisco > ShoreTel > Sonus/L3 > We will try to fix it but some of the devices on this list are so > broken who knows what will happen.. > 2009-07-02 02:39:28.790508 [WARNING] switch_core_codec.c:499 Codec > G723 Exists but not at the desired implementation. 8000hz 60ms > > > Is there any work around for this or i have downgrade my server back > to Asterisk. :'-( > > Thank you. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090701/00663137/attachment.html From brian at freeswitch.org Wed Jul 1 14:29:55 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 1 Jul 2009 16:29:55 -0500 Subject: [Freeswitch-users] G723 timer problem In-Reply-To: References: Message-ID: <97A687D1-E73F-4743-A83C-FF94BB61F211@freeswitch.org> I'm sorry about my response... I had overlooked that I only did one 30ms implementation in mod_g723_1.c, Anthony added some more to the list so it might actually work correctly. Thanks, Brian On Jul 1, 2009, at 4:04 PM, Muhammad Shahzad wrote: > Hi, > > I am using FS svn revision 14046 and trying to send call from SIP > Dialer to a SIP gateway using G723 in passthrough mode. Everything > works perfect and destination rings but then call drops with > following error on FS CLI, > > > 2009-07-02 02:39:28.790508 [WARNING] mod_sofia.c:807 We were told to > use ptime 30 but what they meant to say was 60 > This issue has so far been identified to happen on the following > broken platforms/devices: > Linksys/Sipura aka Cisco > ShoreTel > Sonus/L3 > We will try to fix it but some of the devices on this list are so > broken who knows what will happen.. > 2009-07-02 02:39:28.790508 [WARNING] switch_core_codec.c:499 Codec > G723 Exists but not at the desired implementation. 8000hz 60ms > > > Is there any work around for this or i have downgrade my server back > to Asterisk. :'-( > > Thank you. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090701/2c3a02f2/attachment.html From anthony.minessale at gmail.com Wed Jul 1 14:31:02 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 1 Jul 2009 16:31:02 -0500 Subject: [Freeswitch-users] sound_prefix is changed In-Reply-To: <665906.28719.qm@web33503.mail.mud.yahoo.com> References: <665906.28719.qm@web33503.mail.mud.yahoo.com> Message-ID: <191c3a030907011431s3978c417g970a0180dcc1acde@mail.gmail.com> did you put them in that spot? the say app needs them to be in that spot. On Wed, Jul 1, 2009 at 4:10 PM, Diego Toro wrote: > Hello, the problem is that sound_prefix value is ignored and it's changed > to SWITCH_GLOBAL_dirs.base_dir/sounds/language (with language=en), so audio > files are not found . > Diego > > > --- On *Wed, 7/1/09, Anthony Minessale *wrote: > > > From: Anthony Minessale > Subject: Re: [Freeswitch-users] sound_prefix is changed > To: freeswitch-users at lists.freeswitch.org > Date: Wednesday, July 1, 2009, 3:36 PM > > > what is the problem exactly? what do you want it to change to? > > > On Wed, Jul 1, 2009 at 2:41 PM, Diego Toro > > wrote: > >> hi all, >> I am working with release Pre9, I have a problem now with say module, the >> sound_prefix var is changed to SWITCH_GLOBAL_dirs.base_dir/sounds/es for >> default_language to 'es'. >> >> I checked c code on switch_ivr_say the value of sound_prefix is changed >> always >> >> Diego >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > -----Inline Attachment Follows----- > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference 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/20090701/381b7e42/attachment.html From dftoro at yahoo.com Wed Jul 1 14:48:40 2009 From: dftoro at yahoo.com (Diego Toro) Date: Wed, 1 Jul 2009 14:48:40 -0700 (PDT) Subject: [Freeswitch-users] sound_prefix is changed Message-ID: <887826.56879.qm@web33503.mail.mud.yahoo.com> The audio files are for instance on ...us/callie/digits/8000, but the last 1.0.4 pre9 version has changed, it ignores the var sound_prefix setting to SWITCH_GLOBAL_dirs.base_dir/sounds/en the path to audio files. ? ?That change is on the switch_ivr_say function ? Diego. --- On Wed, 7/1/09, Anthony Minessale wrote: From: Anthony Minessale Subject: Re: [Freeswitch-users] sound_prefix is changed To: freeswitch-users at lists.freeswitch.org Date: Wednesday, July 1, 2009, 4:31 PM did you put them in that spot? the say app needs them to be in that spot. On Wed, Jul 1, 2009 at 4:10 PM, Diego Toro wrote: Hello, the problem is that sound_prefix value is ignored and it's changed to SWITCH_GLOBAL_dirs.base_dir/sounds/language? (with language=en), so audio files are not found . Diego ? --- On Wed, 7/1/09, Anthony Minessale wrote: From: Anthony Minessale Subject: Re: [Freeswitch-users] sound_prefix is changed To: freeswitch-users at lists.freeswitch.org Date: Wednesday, July 1, 2009, 3:36 PM what is the problem exactly? what do you want it to change to? On Wed, Jul 1, 2009 at 2:41 PM, Diego Toro wrote: hi all, I am working with release Pre9, I have a problem now?with say module, the sound_prefix var is changed to SWITCH_GLOBAL_dirs.base_dir/sounds/es for default_language to 'es'. ? I checked c code on switch_ivr_say the value of sound_prefix is changed always ? Diego _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -----Inline Attachment Follows----- _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -----Inline Attachment Follows----- _______________________________________________ Freeswitch-users mailing list Freeswitch-users at 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/20090701/d54a8463/attachment-0001.html From evilla at chipoly.com Wed Jul 1 14:54:28 2009 From: evilla at chipoly.com (Ing. Edwin Villarreal) Date: Wed, 1 Jul 2009 15:54:28 -0600 Subject: [Freeswitch-users] FreeSWITCH with DimDim Integration Message-ID: <006e01c9fa96$839aabc0$8ad00340$@com> Hello ! I?ve been trying to integrate DimDim with FreeSWITCH. What I?d like is to create a SIP leg into a FreeSWITCH audio conference room, so meetings running can be listened in conference room as well. After reading their API and ?understanding? how dimdim Works (I think), this configuration may fit for FreeSWITCH. Please see img attached. With this scenario I can add DimDim to a VoIP conference room, which can host hundreads or users. Have any of you tried this before? Do you have a better idea? Thanks Edwin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090701/dde588be/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: SIP-UA.jpg Type: image/jpeg Size: 63188 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090701/dde588be/attachment-0001.jpg From anthony.minessale at gmail.com Wed Jul 1 16:06:00 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 1 Jul 2009 18:06:00 -0500 Subject: [Freeswitch-users] SIP re-invite / bypass_media In-Reply-To: <367751820907011322s1b17192ct3d7303632456db87@mail.gmail.com> References: <367751820907011322s1b17192ct3d7303632456db87@mail.gmail.com> Message-ID: <191c3a030907011606m537b6895la167d2cfb79fac7d@mail.gmail.com> try apiExecute("uuid_media", "off " + session.uuid); On Wed, Jul 1, 2009 at 3:22 PM, Phillip Jones wrote: > Hi there, > > I was wondering whether it is possible to have FreeSwitch go into > bypass_media mode on demand? > > For instance, leg a bridges to leg b - leg b is invited to accept the call > by pressing 1. I want to go to bypass_media (do a SIP reinvite to reroute > the media) after the one is pressed. > > Currently I am issuing the following from my js script that prompts for the > 1: > > session.apiExecute("uuid_media",session.uuid); > > Not working however. > > Any help to get me going would be appreciated. > > Thanks > > Phillip Jones. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference 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/20090701/8af66408/attachment.html From mfedyk at mikefedyk.com Wed Jul 1 16:16:00 2009 From: mfedyk at mikefedyk.com (Mike Fedyk) Date: Wed, 1 Jul 2009 16:16:00 -0700 Subject: [Freeswitch-users] PCMU fallback for T.38 In-Reply-To: <49C45508.402@coppice.org> References: <49C40A83.1050003@ieee.org> <49C45508.402@coppice.org> Message-ID: <93cdabd20907011616k1dd85b17ib3f7918ea6259c71@mail.gmail.com> On Fri, Mar 20, 2009 at 7:46 PM, Steve Underwood wrote: > Gabriel Kuri wrote: > > once the FAX tone is detected on the PSTN side, FS receives a T.38 > > re-INVITE from the provider and FS sends back a 488/Not Acceptable > > (proxy_media=false). at that point the provider than attempts fallback > > to PCMU with another reINVITE ... > > > > This part is interesting, and the subject of a discussion we had > recently. A number of systems try that second re-invite after a 488, but > the SIP specs say the call is pretty much dead after the 488 message is > exchanged. Are they just hoping that maybe the other end will be > non-compliant enough to keep the call alive, and recover its media mode, > or haven't they read the specs? > > Steve I am interested in this later document. From what I can see there is rfc3261 and at least one other RFC (and one draft that I have found after about 30 minutes of searching) that support that a 488 response can be recovered from when it is a response to a reinvite (ie, the dialog is already in place and there is something to fall back to). Where does it say that a 488 has to end a dialog? From what I understand there are not any 4xx codes that by themselves terminate a dialog (except where it terminates the last leg of a call -- much like unlink() in unix). draft-ietf-sipping-realtimefax-01 says: > 6.2. Unsuccessful T.38 fax scenario - > > - 488/606 rsp & G.711 fallback > > > This section represents an unsuccessful SIP T.38 fax call: when the > emitting gateway does not support T.38 fax relay, it SHOULD respond > with either a ??488 Not Acceptable Here?? response or a ??606 Not > > Acceptable?? response to indicate that some aspects of the session > description are not acceptable. The terminating gateway SHOULD > react by proposing a fallback to G.711 fax pass-through with special > > codec characteristics - > -silence suppression OFF. The message details > in this section make use of the generic SDP attribute silenceSupp > defined in RFC3108 > > rfc3261 section 3 says: > During the session, either Alice or Bob may decide to change the > > characteristics of the media session. This is accomplished by > sending a re-INVITE containing a new media description. This re- > INVITE references the existing dialog so that the other party knows > that it is to modify an existing session instead of establishing a > > new session. The other party sends a 200 (OK) to accept the change. > The requestor responds to the 200 (OK) with an ACK. If the other > party does not accept the change, he sends an error response such as > > 488 (Not Acceptable Here), which also receives an ACK. However, the > failure of the re-INVITE does not cause the existing call to fail - > the session continues using the previously negotiated > characteristics. Full details on session modification are in Section > > 14. > > section 14.1 says: > If a UA receives a non-2xx final response to a re-INVITE, the session > > parameters MUST remain unchanged, as if no re-INVITE had been issued. > Note that, as stated in Section 12.2.1.2, if the non-2xx final > response is a 481 (Call/Transaction Does Not Exist), or a 408 > (Request Timeout), or no response at all is received for the re- > > INVITE (that is, a timeout is returned by the INVITE client > transaction), the UAC will terminate the dialog. > > rfc4497 says: > 8.5. Request to Change Media Characteristics > > If after a call has been successfully established the gateway > receives a SIP INVITE request to change the media characteristics of > the call in a way that would be incompatible with the bearer > capability in use within the PISN, the gateway SHALL send back a SIP > 488 (Not Acceptable Here) response and SHALL NOT change the media > characteristics of the existing call. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090701/516b49c4/attachment.html From msc at freeswitch.org Wed Jul 1 17:08:01 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 1 Jul 2009 17:08:01 -0700 Subject: [Freeswitch-users] Language Handling: call for assistance Message-ID: <87f2f3b90907011708t161e310em833fc728c7dfdbee@mail.gmail.com> Hello all! There's been some discussion lately on how to handle multiple languages, specifically with the *say* application. We would like some input from the community on how to handle multiple languages and sound files. Anthony notes that the say application needs to build the path to the sound files by using the ${sound_prefix} and ${lang} variables. Some have asked about countries or language variants, like European Portugese vs. Brazilian Portugese. These are good questions. >From the community we need input. If you have experience with multiple languages in a telephony environment then please give us your suggestions. How would you like to see the say application handle various languages and dialects? Please give us your helpful suggestions. Thanks, Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090701/c8a8b8e0/attachment.html From brad.tuan at gmail.com Wed Jul 1 18:05:54 2009 From: brad.tuan at gmail.com (Brad Tuan) Date: Thu, 2 Jul 2009 09:05:54 +0800 Subject: [Freeswitch-users] How to know my gateway registering is successed?? Message-ID: Another question, Where does FS keep these information?? In *.db or somewhere?? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090702/4e467323/attachment.html From msc at freeswitch.org Wed Jul 1 18:54:06 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 1 Jul 2009 18:54:06 -0700 Subject: [Freeswitch-users] How to know my gateway registering is successed?? In-Reply-To: References: Message-ID: <87f2f3b90907011854y3098bdfav2293c3d652420094@mail.gmail.com> Are you looking for something more than what "sofia status" at the CLI shows? -MC On Wed, Jul 1, 2009 at 6:05 PM, Brad Tuan wrote: > Another question, Where does FS keep these information?? > > In *.db or somewhere?? > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090701/0256761f/attachment.html From jason at jasonjgw.net Wed Jul 1 20:05:04 2009 From: jason at jasonjgw.net (Jason White) Date: Thu, 2 Jul 2009 13:05:04 +1000 Subject: [Freeswitch-users] How to know my gateway registering is successed?? In-Reply-To: References: Message-ID: <20090702030504.GA26790@jdc.jasonjgw.net> Brad Tuan wrote: > Another question, Where does FS keep these information?? > > In *.db or somewhere?? It's a hash table in memory. See sofia_reg_find_gateway__ in sofia_reg.c for the code that performs the hash table lookup and returns a pointer to the structure with all of the fields in it. From mythicalbox at weavver.com Wed Jul 1 22:05:10 2009 From: mythicalbox at weavver.com (Mitchel Constantin) Date: Wed, 1 Jul 2009 22:05:10 -0700 Subject: [Freeswitch-users] How to remove the IP from the SIP caller id number Message-ID: Hello, I'm working on configuring my FreeSWITCH and would like to set the caller id number like this in dialplan/default.xml: I wonder if this is a problem with eyeBeam.. When the call is received the CID is like this: John Doe johndoe at weavver.com@205.134.225.20 205.134.225.20 is the EXT IP of the switch I'd like to remove the 205.134.225.20 from the Caller ID if possible. From what I understand the CID without the IP would still point to the correct server. TIA, -- Mitchel Constantin Weavver, Inc. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090701/a61155ae/attachment-0001.html From jason at jasonjgw.net Wed Jul 1 22:56:01 2009 From: jason at jasonjgw.net (Jason White) Date: Thu, 2 Jul 2009 15:56:01 +1000 Subject: [Freeswitch-users] How to remove the IP from the SIP caller id number In-Reply-To: References: Message-ID: <20090702055601.GA13328@jdc.jasonjgw.net> Mitchel Constantin wrote: > I'm working on configuring my FreeSWITCH and would like to set the caller id > number like this in dialplan/default.xml: > > > > > I wonder if this is a problem with eyeBeam.. When the call is received the > CID is like this: > > John Doe > johndoe at weavver.com@205.134.225.20 > > 205.134.225.20 is the EXT IP of the switch I suspect the other end (whatever device you are calling from FreeSWITCH) is adding the IP address to the caller id. However, I am no SIP expert and may be wrong, but you can confirm this by doing a SIP trace on the device that receives the call (or on its local network via packet capture) to discover what FreeSWITCH is sending out. From igor at 3gnt.net Thu Jul 2 03:01:34 2009 From: igor at 3gnt.net (Igor Neves) Date: Thu, 02 Jul 2009 11:01:34 +0100 Subject: [Freeswitch-users] Language Handling: call for assistance In-Reply-To: <87f2f3b90907011708t161e310em833fc728c7dfdbee@mail.gmail.com> References: <87f2f3b90907011708t161e310em833fc728c7dfdbee@mail.gmail.com> Message-ID: <4A4C857E.3060805@3gnt.net> Hi, Michael Collins wrote: > Hello all! > > There's been some discussion lately on how to handle multiple > languages, specifically with the *say* application. We would like some > input from the community on how to handle multiple languages and sound > files. Anthony notes that the say application needs to build the path > to the sound files by using the ${sound_prefix} and ${lang} variables. > Some have asked about countries or language variants, like European > Portugese vs. Brazilian Portugese. These are good questions. What it's the problem about Portuguese VS Brazilian? Can't we just use "PT_pt" and "PT_br" in ${lang}, just like a lot of others softwares do? What about ${sound_prefix} = ${lang}, since ${lang} should always be unique, and you make the path's automatically language organized? > > From the community we need input. If you have experience with multiple > languages in a telephony environment then please give us your > suggestions. How would you like to see the say application handle > various languages and dialects? Please give us your helpful suggestions. > > Thanks, > Michael Sorry if I miss understood something. Cheers, -- Igor Neves 3GNTW - Tecnologias de Informa??o, Lda SIP: igor at 3gnt.net JID: igor at 3gnt.net ICQ: 249075444 MSN: igor at 3gnt.net TLM: 00351914503611 PSTN: 00351252377120 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090702/a9f317fb/attachment.html From darklion11 at yahoo.com Thu Jul 2 03:08:43 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Thu, 2 Jul 2009 03:08:43 -0700 (PDT) Subject: [Freeswitch-users] Connecting to FS through text or SMS? Message-ID: <24304261.post@talk.nabble.com> Hi everyone, Can you give me some proper instructions what I will do to enable texting or SMS through freeswitch using a GSM gateway? Thanks for your cooperation, Edmar -- View this message in context: http://www.nabble.com/Connecting-to-FS-through-text-or-SMS--tp24304261p24304261.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From Prometheus001 at gmx.net Thu Jul 2 04:11:37 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Thu, 02 Jul 2009 13:11:37 +0200 Subject: [Freeswitch-users] Freswitch closes UDP port after OPTIONS with TLS? Message-ID: <4A4C95E9.7010706@gmx.net> Hello, I have the following problem: Every call stops after 30 seconds when TLS is enabled. SIP/RTP and SIP/SRTP works but not TLS/SRTP. The phones are behind NAT. So I expect, that every 30 seconds an Options request is sent. Wiresharking the traffic I can see * that there are ongoing UDP packets. * Then a TSLv1 packet ist sent from FS to the Phone. * This is acknowleged by the phone * Next the phone send another UDP packet to the same FS port as before * Then the Phone receives an ICMP request that the FS port is closed. Anybody has a clue about this? Best regards Peter From Prometheus001 at gmx.net Thu Jul 2 04:42:24 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Thu, 02 Jul 2009 13:42:24 +0200 Subject: [Freeswitch-users] Freswitch closes UDP port after OPTIONS with TLS? In-Reply-To: <4A4C95E9.7010706@gmx.net> References: <4A4C95E9.7010706@gmx.net> Message-ID: <4A4C9D20.9050405@gmx.net> Some additions: TLS/RTP instead of SRTP does also not work. There are no logs on the debug console except the message that the call is being terminated 2009-07-02 12:06:45.252177 [DEBUG] sofia.c:3100 Channel sofia/internal/835333 at sip.mydomain.de entering state [terminating][0] and later cause: NORMAL_UNSPECIFIED Best regards Peter Peter P GMX schrieb: > Hello, > > I have the following problem: Every call stops after 30 seconds when TLS > is enabled. SIP/RTP and SIP/SRTP works but not TLS/SRTP. > The phones are behind NAT. So I expect, that every 30 seconds an Options > request is sent. > > Wiresharking the traffic I can see > > * that there are ongoing UDP packets. > * Then a TSLv1 packet ist sent from FS to the Phone. > * This is acknowleged by the phone > * Next the phone send another UDP packet to the same FS port as before > * Then the Phone receives an ICMP request that the FS port is closed. > > > Anybody has a clue about this? > > Best regards > Peter > > > > > > From tayeb.meftah at gmail.com Thu Jul 2 07:19:48 2009 From: tayeb.meftah at gmail.com (Meftah Tayeb) Date: Thu, 02 Jul 2009 14:19:48 +0000 Subject: [Freeswitch-users] Jingle (Mod_Dingaling Dialplan Extention Message-ID: <4A4CC204.4010101@gmail.com> hello, i asked about mod_dingaling usage befor finally, i configured my XMPP acount with my FS, connected to it and is ready / Online now i need to call / recev call from / to GTalk (Jingle) cool anyone give me a sample Dialplan Extention? thanks __________ Information from ESET NOD32 Antivirus, version of virus signature database 4209 (20090702) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com From brian at freeswitch.org Thu Jul 2 06:35:47 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 2 Jul 2009 08:35:47 -0500 Subject: [Freeswitch-users] Freswitch closes UDP port after OPTIONS with TLS? In-Reply-To: <4A4C95E9.7010706@gmx.net> References: <4A4C95E9.7010706@gmx.net> Message-ID: <10292506-A353-4E33-AA98-87833A40E224@freeswitch.org> If its TLS you don't need options packets in the first place. Your client should do the keep alive NOT FreeSWITCH. TLS is over TCP and Options over UDP... doesn't make much sense. /b On Jul 2, 2009, at 6:11 AM, Peter P GMX wrote: > Hello, > > I have the following problem: Every call stops after 30 seconds when > TLS > is enabled. SIP/RTP and SIP/SRTP works but not TLS/SRTP. > The phones are behind NAT. So I expect, that every 30 seconds an > Options > request is sent. > > Wiresharking the traffic I can see > > * that there are ongoing UDP packets. > * Then a TSLv1 packet ist sent from FS to the Phone. > * This is acknowleged by the phone > * Next the phone send another UDP packet to the same FS port as > before > * Then the Phone receives an ICMP request that the FS port is > closed. > > > Anybody has a clue about this? > > Best regards > Peter > From brian at freeswitch.org Thu Jul 2 06:37:57 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 2 Jul 2009 08:37:57 -0500 Subject: [Freeswitch-users] How to remove the IP from the SIP caller id number In-Reply-To: <20090702055601.GA13328@jdc.jasonjgw.net> References: <20090702055601.GA13328@jdc.jasonjgw.net> Message-ID: set the variable sip_invite_domain /b On Jul 2, 2009, at 12:56 AM, Jason White wrote: > I suspect the other end (whatever device you are calling from > FreeSWITCH) is > adding the IP address to the caller id. However, I am no SIP expert > and may be > wrong, but you can confirm this by doing a SIP trace on the device > that > receives the call (or on its local network via packet capture) to > discover > what FreeSWITCH is sending out. From dujinfang at gmail.com Thu Jul 2 07:45:58 2009 From: dujinfang at gmail.com (Seven Du) Date: Thu, 2 Jul 2009 22:45:58 +0800 Subject: [Freeswitch-users] Jingle (Mod_Dingaling Dialplan Extention In-Reply-To: <4A4CC204.4010101@gmail.com> References: <4A4CC204.4010101@gmail.com> Message-ID: for outbound: for inbount: you should be able to set context and extension in your dingaling profile, and set the dialplan accordingly. as always you can press F8 on console to see dialplan information On Jul 2, 2009, at 10:19 PM, Meftah Tayeb wrote: > hello, > i asked about mod_dingaling usage befor > finally, i configured my XMPP acount with my FS, connected to it and > is > ready / Online > now i need to call / recev call from / to GTalk (Jingle) > cool anyone give me a sample Dialplan Extention? > thanks > > > __________ Information from ESET NOD32 Antivirus, version of virus > signature database 4209 (20090702) __________ > > The message was checked by ESET NOD32 Antivirus. > > http://www.eset.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 pjintheusa at gmail.com Thu Jul 2 08:41:16 2009 From: pjintheusa at gmail.com (Phillip Jones) Date: Thu, 2 Jul 2009 11:41:16 -0400 Subject: [Freeswitch-users] SIP re-invite / bypass_media In-Reply-To: <191c3a030907011606m537b6895la167d2cfb79fac7d@mail.gmail.com> References: <367751820907011322s1b17192ct3d7303632456db87@mail.gmail.com> <191c3a030907011606m537b6895la167d2cfb79fac7d@mail.gmail.com> Message-ID: <367751820907020841j379749cblef2302e7dcae32da@mail.gmail.com> Thanks for that. That seems to successfully re-invite and re-route the the B leg - but does not reinvite the A leg and then immediately issues a "bye" on both legs. Do I have to do something to reinvite that A leg? On Wed, Jul 1, 2009 at 7:06 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > try > apiExecute("uuid_media", "off " + session.uuid); > > > > On Wed, Jul 1, 2009 at 3:22 PM, Phillip Jones wrote: > >> Hi there, >> >> I was wondering whether it is possible to have FreeSwitch go into >> bypass_media mode on demand? >> >> For instance, leg a bridges to leg b - leg b is invited to accept the call >> by pressing 1. I want to go to bypass_media (do a SIP reinvite to reroute >> the media) after the one is pressed. >> >> Currently I am issuing the following from my js script that prompts for >> the 1: >> >> session.apiExecute("uuid_media",session.uuid); >> >> Not working however. >> >> Any help to get me going would be appreciated. >> >> Thanks >> >> Phillip Jones. >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > 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/20090702/8876a495/attachment-0001.html From brian at freeswitch.org Thu Jul 2 08:51:22 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 2 Jul 2009 10:51:22 -0500 Subject: [Freeswitch-users] Baby Update! Message-ID: <3EC5EF44-BF81-4C23-AD0B-B4AAB5F7DFCE@freeswitch.org> FreeSWITCHers, Kaiden Anthony Chandler will arrive sometime Friday July 3rd 2009!!! So to help out with any last minute expenses and help ease things up for Ray and Samantha and remove some of the worry I'm going to donate $100 dollars myself to the cause... never know diapers and various other expenses that come up. Be sure to select the "personal" option on paypal so they don't take any money from the transaction. Paypal: intralanman at gmail.com And congratulations to Ray and Samantha on their first Boy! Thanks everyone you're a great community! /b From anthony.minessale at gmail.com Thu Jul 2 09:19:48 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 2 Jul 2009 11:19:48 -0500 Subject: [Freeswitch-users] SIP re-invite / bypass_media In-Reply-To: <367751820907020841j379749cblef2302e7dcae32da@mail.gmail.com> References: <367751820907011322s1b17192ct3d7303632456db87@mail.gmail.com> <191c3a030907011606m537b6895la167d2cfb79fac7d@mail.gmail.com> <367751820907020841j379749cblef2302e7dcae32da@mail.gmail.com> Message-ID: <191c3a030907020919i5be162b3x69b86b5bcc2eb8e4@mail.gmail.com> I would need to know more details about what you are doing. you could set the variable bypass_media_after_bridge=true on the a leg before you call the b leg and use the group_confirm feature to get the caller to press the key. On Thu, Jul 2, 2009 at 10:41 AM, Phillip Jones wrote: > Thanks for that. > > That seems to successfully re-invite and re-route the the B leg - but does > not reinvite the A leg and then immediately issues a "bye" on both legs. > > Do I have to do something to reinvite that A leg? > > On Wed, Jul 1, 2009 at 7:06 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> try >> apiExecute("uuid_media", "off " + session.uuid); >> >> >> >> On Wed, Jul 1, 2009 at 3:22 PM, Phillip Jones wrote: >> >>> Hi there, >>> >>> I was wondering whether it is possible to have FreeSwitch go into >>> bypass_media mode on demand? >>> >>> For instance, leg a bridges to leg b - leg b is invited to accept the >>> call by pressing 1. I want to go to bypass_media (do a SIP reinvite to >>> reroute the media) after the one is pressed. >>> >>> Currently I am issuing the following from my js script that prompts for >>> the 1: >>> >>> session.apiExecute("uuid_media",session.uuid); >>> >>> Not working however. >>> >>> Any help to get me going would be appreciated. >>> >>> Thanks >>> >>> Phillip Jones. >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> 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/20090702/b51e63cd/attachment.html From tayeb.meftah at gmail.com Thu Jul 2 10:23:23 2009 From: tayeb.meftah at gmail.com (Meftah Tayeb) Date: Thu, 02 Jul 2009 17:23:23 +0000 Subject: [Freeswitch-users] Jingle (Mod_Dingaling Dialplan Extention In-Reply-To: References: <4A4CC204.4010101@gmail.com> Message-ID: <4A4CED0B.9060907@gmail.com> hello, thank you for this informations for inbound, i say the extention to 3001 if i get call, is droped automatikaly thanks Seven Du wrote: > for outbound: > > expression="^(.*@gmail.com)"> > > > > > for inbount: > you should be able to set context and extension in your dingaling > profile, and set the dialplan accordingly. as always you can press F8 > on console to see dialplan information > > On Jul 2, 2009, at 10:19 PM, Meftah Tayeb wrote: > >> hello, >> i asked about mod_dingaling usage befor >> finally, i configured my XMPP acount with my FS, connected to it and >> is >> ready / Online >> now i need to call / recev call from / to GTalk (Jingle) >> cool anyone give me a sample Dialplan Extention? >> thanks >> >> >> __________ Information from ESET NOD32 Antivirus, version of virus >> signature database 4209 (20090702) __________ >> >> The message was checked by ESET NOD32 Antivirus. >> >> http://www.eset.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 > > > __________ Information from ESET NOD32 Antivirus, version of virus signature database 4209 (20090702) __________ > > The message was checked by ESET NOD32 Antivirus. > > http://www.eset.com > > > __________ Information from ESET NOD32 Antivirus, version of virus signature database 4209 (20090702) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090702/ede07ef8/attachment.html From pjintheusa at gmail.com Thu Jul 2 09:53:44 2009 From: pjintheusa at gmail.com (Phillip Jones) Date: Thu, 2 Jul 2009 12:53:44 -0400 Subject: [Freeswitch-users] SIP re-invite / bypass_media In-Reply-To: <191c3a030907020919i5be162b3x69b86b5bcc2eb8e4@mail.gmail.com> References: <367751820907011322s1b17192ct3d7303632456db87@mail.gmail.com> <191c3a030907011606m537b6895la167d2cfb79fac7d@mail.gmail.com> <367751820907020841j379749cblef2302e7dcae32da@mail.gmail.com> <191c3a030907020919i5be162b3x69b86b5bcc2eb8e4@mail.gmail.com> Message-ID: <367751820907020953l34363158md0f16e069e5a9d14@mail.gmail.com> Thanks for responding and for your help. The xml and confirm.js are attached below. Basically trying to bypass_media after the leg B presses 1 to accept the call. I tried, using bypass_media_after_bridge=true, but the re-invite appears to be done before the confirm.js, So the media is successfully rerouted, but BEFORE the leg b never gets hear a prompt or gets the opportunity to press 1. To get round this I am trying to manually bypass_media in the confirm.js script with apiExecute("uuid_media", "off " + session.uuid);. However only the B leg is reinvited (and media is routed correctly). I don't see the A leg reinvite, and then a BYE is issueed on both legs. < This is the confirm.js: // confirm.js - FreeSwitch call confirmation script // (c) 2009 - St?phane Alnet // License: GPL2 or above console_log("info", "Destination: "+ session.destination + "\n"); if(!session.getVariable('leg_confirm')) { console_log("info", "No need to confirm, connect the call!\n"); exit(); } var confirmed = false; var confirmation_digit = "1"; var try_count = 6; var prompt_file = "prompts/ToAcceptThisCallPress1.wav"; function onInput( session, type, data, arg ) { if ( type == "dtmf" ) { console_log( "info", "Got digit " + data.digit + "\n" ); if ( data.digit == confirmation_digit ) { confirmed = true; console_log( "info", "Confirming session..\n" ); return(false); } } return(true); } if ( session.ready() ) { session.answer(); session.flushDigits(); console_log("info", "Starting confirmation\n"); var count = try_count; while( session.ready() && ! confirmed && count-- > 0 ) { session.execute("sleep","200"); session.streamFile( prompt_file, onInput ); } if( ! confirmed ) { console_log("info", "Not confirmed\n"); session.hangup(); } else { *apiExecute("uuid_media", "off " + session.uuid);* console_log("info", "Confirmed\n"); } } else { console_log("info", "Session is not ready.\n"); } On Thu, Jul 2, 2009 at 12:19 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > I would need to know more details about what you are doing. > > you could set the variable bypass_media_after_bridge=true on the a leg > before you call the b leg and use the group_confirm feature to get the > caller > to press the key. > > > > On Thu, Jul 2, 2009 at 10:41 AM, Phillip Jones wrote: > >> Thanks for that. >> >> That seems to successfully re-invite and re-route the the B leg - but does >> not reinvite the A leg and then immediately issues a "bye" on both legs. >> >> Do I have to do something to reinvite that A leg? >> >> On Wed, Jul 1, 2009 at 7:06 PM, Anthony Minessale < >> anthony.minessale at gmail.com> wrote: >> >>> try >>> apiExecute("uuid_media", "off " + session.uuid); >>> >>> >>> >>> On Wed, Jul 1, 2009 at 3:22 PM, Phillip Jones wrote: >>> >>>> Hi there, >>>> >>>> I was wondering whether it is possible to have FreeSwitch go into >>>> bypass_media mode on demand? >>>> >>>> For instance, leg a bridges to leg b - leg b is invited to accept the >>>> call by pressing 1. I want to go to bypass_media (do a SIP reinvite to >>>> reroute the media) after the one is pressed. >>>> >>>> Currently I am issuing the following from my js script that prompts for >>>> the 1: >>>> >>>> session.apiExecute("uuid_media",session.uuid); >>>> >>>> Not working however. >>>> >>>> Any help to get me going would be appreciated. >>>> >>>> Thanks >>>> >>>> Phillip Jones. >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> Anthony Minessale II >>> >>> FreeSWITCH http://www.freeswitch.org/ >>> ClueCon http://www.cluecon.com/ >>> >>> AIM: anthm >>> MSN:anthony_minessale at hotmail.com >>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>> IRC: irc.freenode.net #freeswitch >>> >>> FreeSWITCH Developer Conference >>> 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/20090702/20f9a8e0/attachment-0001.html From anthony.minessale at gmail.com Thu Jul 2 10:05:31 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 2 Jul 2009 12:05:31 -0500 Subject: [Freeswitch-users] SIP re-invite / bypass_media In-Reply-To: <367751820907020953l34363158md0f16e069e5a9d14@mail.gmail.com> References: <367751820907011322s1b17192ct3d7303632456db87@mail.gmail.com> <191c3a030907011606m537b6895la167d2cfb79fac7d@mail.gmail.com> <367751820907020841j379749cblef2302e7dcae32da@mail.gmail.com> <191c3a030907020919i5be162b3x69b86b5bcc2eb8e4@mail.gmail.com> <367751820907020953l34363158md0f16e069e5a9d14@mail.gmail.com> Message-ID: <191c3a030907021005r2762e48fu97310abbc8b75b92@mail.gmail.com> try setting bypass_media_after_bridge=true on the session in your confirm script On Thu, Jul 2, 2009 at 11:53 AM, Phillip Jones wrote: > Thanks for responding and for your help. > > The xml and confirm.js are attached below. Basically trying to bypass_media > after the leg B presses 1 to accept the call. I tried, > using bypass_media_after_bridge=true, but the re-invite appears to be done > before the confirm.js, So the media is successfully rerouted, but BEFORE the > leg b never gets hear a prompt or gets the opportunity to press 1. > > To get round this I am trying to manually bypass_media in the confirm.js > script with apiExecute("uuid_media", "off " + session.uuid);. However only > the B leg is reinvited (and media is routed correctly). I don't see the A > leg reinvite, and then a BYE is issueed on both legs. > > > > > > > > < > > > > > > > This is the confirm.js: > > // confirm.js - FreeSwitch call confirmation script > // (c) 2009 - St?phane Alnet > // License: GPL2 or above > console_log("info", "Destination: "+ session.destination + "\n"); > if(!session.getVariable('leg_confirm')) > { > console_log("info", "No need to confirm, connect the call!\n"); > exit(); > } > var confirmed = false; > var confirmation_digit = "1"; > var try_count = 6; > var prompt_file = "prompts/ToAcceptThisCallPress1.wav"; > function onInput( session, type, data, arg ) { > if ( type == "dtmf" ) { > console_log( "info", "Got digit " + data.digit + "\n" ); > if ( data.digit == confirmation_digit ) { > confirmed = true; > console_log( "info", "Confirming session..\n" ); > return(false); > } > } > return(true); > } > if ( session.ready() ) > { > session.answer(); > session.flushDigits(); > console_log("info", "Starting confirmation\n"); > var count = try_count; > while( session.ready() && ! confirmed && count-- > 0 ) > { > session.execute("sleep","200"); > session.streamFile( prompt_file, onInput ); > } > > if( ! confirmed ) > { > console_log("info", "Not confirmed\n"); > session.hangup(); > } > else > { > *apiExecute("uuid_media", "off " + session.uuid);* > console_log("info", "Confirmed\n"); > } > } > else > { > console_log("info", "Session is not ready.\n"); > } > > > > > On Thu, Jul 2, 2009 at 12:19 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> I would need to know more details about what you are doing. >> >> you could set the variable bypass_media_after_bridge=true on the a leg >> before you call the b leg and use the group_confirm feature to get the >> caller >> to press the key. >> >> >> >> On Thu, Jul 2, 2009 at 10:41 AM, Phillip Jones wrote: >> >>> Thanks for that. >>> >>> That seems to successfully re-invite and re-route the the B leg - but >>> does not reinvite the A leg and then immediately issues a "bye" on both >>> legs. >>> >>> Do I have to do something to reinvite that A leg? >>> >>> On Wed, Jul 1, 2009 at 7:06 PM, Anthony Minessale < >>> anthony.minessale at gmail.com> wrote: >>> >>>> try >>>> apiExecute("uuid_media", "off " + session.uuid); >>>> >>>> >>>> >>>> On Wed, Jul 1, 2009 at 3:22 PM, Phillip Jones wrote: >>>> >>>>> Hi there, >>>>> >>>>> I was wondering whether it is possible to have FreeSwitch go into >>>>> bypass_media mode on demand? >>>>> >>>>> For instance, leg a bridges to leg b - leg b is invited to accept the >>>>> call by pressing 1. I want to go to bypass_media (do a SIP reinvite to >>>>> reroute the media) after the one is pressed. >>>>> >>>>> Currently I am issuing the following from my js script that prompts for >>>>> the 1: >>>>> >>>>> session.apiExecute("uuid_media",session.uuid); >>>>> >>>>> Not working however. >>>>> >>>>> Any help to get me going would be appreciated. >>>>> >>>>> Thanks >>>>> >>>>> Phillip Jones. >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> -- >>>> Anthony Minessale II >>>> >>>> FreeSWITCH http://www.freeswitch.org/ >>>> ClueCon http://www.cluecon.com/ >>>> >>>> AIM: anthm >>>> MSN:anthony_minessale at hotmail.com >>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>>> IRC: irc.freenode.net #freeswitch >>>> >>>> FreeSWITCH Developer Conference >>>> 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/20090702/95d72779/attachment.html From msc at freeswitch.org Thu Jul 2 10:07:39 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 2 Jul 2009 10:07:39 -0700 Subject: [Freeswitch-users] Language Handling: call for assistance In-Reply-To: <4A4C857E.3060805@3gnt.net> References: <87f2f3b90907011708t161e310em833fc728c7dfdbee@mail.gmail.com> <4A4C857E.3060805@3gnt.net> Message-ID: <87f2f3b90907021007x7033f2bcjeb087a808a9e8c52@mail.gmail.com> On Thu, Jul 2, 2009 at 3:01 AM, Igor Neves wrote: > Hi, > > Michael Collins wrote: > > Hello all! > > There's been some discussion lately on how to handle multiple languages, > specifically with the *say* application. We would like some input from the > community on how to handle multiple languages and sound files. Anthony notes > that the say application needs to build the path to the sound files by using > the ${sound_prefix} and ${lang} variables. Some have asked about countries > or language variants, like European Portugese vs. Brazilian Portugese. These > are good questions. > > > What it's the problem about Portuguese VS Brazilian? > > Can't we just use "PT_pt" and "PT_br" in ${lang}, just like a lot of others > softwares do? > > What about ${sound_prefix} = ${lang}, since ${lang} should always be > unique, and you make the path's automatically language organized? > This is reasonable to me, but it would be nice to have a consensus, just to be sure. > > > > >From the community we need input. If you have experience with multiple > languages in a telephony environment then please give us your suggestions. > How would you like to see the say application handle various languages and > dialects? Please give us your helpful suggestions. > > Thanks, > Michael > > > Sorry if I miss understood something. > Cheers, > Believe, the moment we put this into place we will have someone purporting to be an expert offering a completely new solution. That's why we asked for input now, before Tony spends a lot of time working on it. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090702/ca185bdc/attachment.html From anthony.minessale at gmail.com Thu Jul 2 10:19:39 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 2 Jul 2009 12:19:39 -0500 Subject: [Freeswitch-users] Language Handling: call for assistance In-Reply-To: <87f2f3b90907021007x7033f2bcjeb087a808a9e8c52@mail.gmail.com> References: <87f2f3b90907011708t161e310em833fc728c7dfdbee@mail.gmail.com> <4A4C857E.3060805@3gnt.net> <87f2f3b90907021007x7033f2bcjeb087a808a9e8c52@mail.gmail.com> Message-ID: <191c3a030907021019l3807bc4am1759cb6a1d1e0e16@mail.gmail.com> sound prefix should not be used for lang just "language" the sound_prefix is automatically built as ${base}/sounds/${language}/ each time you execute say. and restored to its previous val when the say is over. On Thu, Jul 2, 2009 at 12:07 PM, Michael Collins wrote: > > > On Thu, Jul 2, 2009 at 3:01 AM, Igor Neves wrote: > >> Hi, >> >> Michael Collins wrote: >> >> Hello all! >> >> There's been some discussion lately on how to handle multiple languages, >> specifically with the *say* application. We would like some input from >> the community on how to handle multiple languages and sound files. Anthony >> notes that the say application needs to build the path to the sound files by >> using the ${sound_prefix} and ${lang} variables. Some have asked about >> countries or language variants, like European Portugese vs. Brazilian >> Portugese. These are good questions. >> >> >> What it's the problem about Portuguese VS Brazilian? >> >> Can't we just use "PT_pt" and "PT_br" in ${lang}, just like a lot of >> others softwares do? >> >> What about ${sound_prefix} = ${lang}, since ${lang} should always be >> unique, and you make the path's automatically language organized? >> > > This is reasonable to me, but it would be nice to have a consensus, just to > be sure. > > >> >> >> >> >From the community we need input. If you have experience with multiple >> languages in a telephony environment then please give us your suggestions. >> How would you like to see the say application handle various languages and >> dialects? Please give us your helpful suggestions. >> >> Thanks, >> Michael >> >> >> Sorry if I miss understood something. >> Cheers, >> > > Believe, the moment we put this into place we will have someone purporting > to be an expert offering a completely new solution. That's why we asked for > input now, before Tony spends a lot of time working on it. > -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/20090702/71e2bb3e/attachment-0001.html From pjintheusa at gmail.com Thu Jul 2 10:27:56 2009 From: pjintheusa at gmail.com (Phillip Jones) Date: Thu, 2 Jul 2009 13:27:56 -0400 Subject: [Freeswitch-users] SIP re-invite / bypass_media In-Reply-To: <191c3a030907021005r2762e48fu97310abbc8b75b92@mail.gmail.com> References: <367751820907011322s1b17192ct3d7303632456db87@mail.gmail.com> <191c3a030907011606m537b6895la167d2cfb79fac7d@mail.gmail.com> <367751820907020841j379749cblef2302e7dcae32da@mail.gmail.com> <191c3a030907020919i5be162b3x69b86b5bcc2eb8e4@mail.gmail.com> <367751820907020953l34363158md0f16e069e5a9d14@mail.gmail.com> <191c3a030907021005r2762e48fu97310abbc8b75b92@mail.gmail.com> Message-ID: <367751820907021027rcb09956j9be2e1837b29b4ff@mail.gmail.com> Used: session.execute("set","bypass_media_after_bridge=true"); in the confirm.js script and that works perfectly! Thank you for you help! On Thu, Jul 2, 2009 at 1:05 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > try setting bypass_media_after_bridge=true on the session in your confirm > script > > > > On Thu, Jul 2, 2009 at 11:53 AM, Phillip Jones wrote: > >> Thanks for responding and for your help. >> >> The xml and confirm.js are attached below. Basically trying to >> bypass_media after the leg B presses 1 to accept the call. I tried, >> using bypass_media_after_bridge=true, but the re-invite appears to be done >> before the confirm.js, So the media is successfully rerouted, but BEFORE the >> leg b never gets hear a prompt or gets the opportunity to press 1. >> >> To get round this I am trying to manually bypass_media in the confirm.js >> script with apiExecute("uuid_media", "off " + session.uuid);. However only >> the B leg is reinvited (and media is routed correctly). I don't see the A >> leg reinvite, and then a BYE is issueed on both legs. >> >> >> >> >> >> >> >> < >> >> >> >> >> >> >> This is the confirm.js: >> >> // confirm.js - FreeSwitch call confirmation script >> // (c) 2009 - St?phane Alnet >> // License: GPL2 or above >> console_log("info", "Destination: "+ session.destination + "\n"); >> if(!session.getVariable('leg_confirm')) >> { >> console_log("info", "No need to confirm, connect the call!\n"); >> exit(); >> } >> var confirmed = false; >> var confirmation_digit = "1"; >> var try_count = 6; >> var prompt_file = "prompts/ToAcceptThisCallPress1.wav"; >> function onInput( session, type, data, arg ) { >> if ( type == "dtmf" ) { >> console_log( "info", "Got digit " + data.digit + "\n" ); >> if ( data.digit == confirmation_digit ) { >> confirmed = true; >> console_log( "info", "Confirming session..\n" ); >> return(false); >> } >> } >> return(true); >> } >> if ( session.ready() ) >> { >> session.answer(); >> session.flushDigits(); >> console_log("info", "Starting confirmation\n"); >> var count = try_count; >> while( session.ready() && ! confirmed && count-- > 0 ) >> { >> session.execute("sleep","200"); >> session.streamFile( prompt_file, onInput ); >> } >> >> if( ! confirmed ) >> { >> console_log("info", "Not confirmed\n"); >> session.hangup(); >> } >> else >> { >> *apiExecute("uuid_media", "off " + session.uuid);* >> console_log("info", "Confirmed\n"); >> } >> } >> else >> { >> console_log("info", "Session is not ready.\n"); >> } >> >> >> >> >> On Thu, Jul 2, 2009 at 12:19 PM, Anthony Minessale < >> anthony.minessale at gmail.com> wrote: >> >>> I would need to know more details about what you are doing. >>> >>> you could set the variable bypass_media_after_bridge=true on the a leg >>> before you call the b leg and use the group_confirm feature to get the >>> caller >>> to press the key. >>> >>> >>> >>> On Thu, Jul 2, 2009 at 10:41 AM, Phillip Jones wrote: >>> >>>> Thanks for that. >>>> >>>> That seems to successfully re-invite and re-route the the B leg - but >>>> does not reinvite the A leg and then immediately issues a "bye" on both >>>> legs. >>>> >>>> Do I have to do something to reinvite that A leg? >>>> >>>> On Wed, Jul 1, 2009 at 7:06 PM, Anthony Minessale < >>>> anthony.minessale at gmail.com> wrote: >>>> >>>>> try >>>>> apiExecute("uuid_media", "off " + session.uuid); >>>>> >>>>> >>>>> >>>>> On Wed, Jul 1, 2009 at 3:22 PM, Phillip Jones wrote: >>>>> >>>>>> Hi there, >>>>>> >>>>>> I was wondering whether it is possible to have FreeSwitch go into >>>>>> bypass_media mode on demand? >>>>>> >>>>>> For instance, leg a bridges to leg b - leg b is invited to accept the >>>>>> call by pressing 1. I want to go to bypass_media (do a SIP reinvite to >>>>>> reroute the media) after the one is pressed. >>>>>> >>>>>> Currently I am issuing the following from my js script that prompts >>>>>> for the 1: >>>>>> >>>>>> session.apiExecute("uuid_media",session.uuid); >>>>>> >>>>>> Not working however. >>>>>> >>>>>> Any help to get me going would be appreciated. >>>>>> >>>>>> Thanks >>>>>> >>>>>> Phillip Jones. >>>>>> >>>>>> _______________________________________________ >>>>>> Freeswitch-users mailing list >>>>>> Freeswitch-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>> >>>>> >>>>> -- >>>>> Anthony Minessale II >>>>> >>>>> FreeSWITCH http://www.freeswitch.org/ >>>>> ClueCon http://www.cluecon.com/ >>>>> >>>>> AIM: anthm >>>>> MSN:anthony_minessale at hotmail.com >>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>>>> IRC: irc.freenode.net #freeswitch >>>>> >>>>> FreeSWITCH Developer Conference >>>>> 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/20090702/db946465/attachment.html From steveu at coppice.org Thu Jul 2 10:29:13 2009 From: steveu at coppice.org (Steve Underwood) Date: Fri, 03 Jul 2009 01:29:13 +0800 Subject: [Freeswitch-users] Language Handling: call for assistance In-Reply-To: <87f2f3b90907021007x7033f2bcjeb087a808a9e8c52@mail.gmail.com> References: <87f2f3b90907011708t161e310em833fc728c7dfdbee@mail.gmail.com> <4A4C857E.3060805@3gnt.net> <87f2f3b90907021007x7033f2bcjeb087a808a9e8c52@mail.gmail.com> Message-ID: <4A4CEE69.8050108@coppice.org> Michael Collins wrote: > > > On Thu, Jul 2, 2009 at 3:01 AM, Igor Neves > wrote: > > Hi, > > > Michael Collins wrote: >> Hello all! >> >> There's been some discussion lately on how to handle multiple >> languages, specifically with the *say* application. We would like >> some input from the community on how to handle multiple languages >> and sound files. Anthony notes that the say application needs to >> build the path to the sound files by using the ${sound_prefix} >> and ${lang} variables. Some have asked about countries or >> language variants, like European Portugese vs. Brazilian >> Portugese. These are good questions. > > What it's the problem about Portuguese VS Brazilian? > > Can't we just use "PT_pt" and "PT_br" in ${lang}, just like a lot > of others softwares do? > > What about ${sound_prefix} = ${lang}, since ${lang} should always > be unique, and you make the path's automatically language organized? > > > This is reasonable to me, but it would be nice to have a consensus, > just to be sure. > > > > >> >> >From the community we need input. If you have experience with >> multiple languages in a telephony environment then please give us >> your suggestions. How would you like to see the say application >> handle various languages and dialects? Please give us your >> helpful suggestions. >> >> Thanks, >> Michael > > Sorry if I miss understood something. > Cheers, > > > Believe, the moment we put this into place we will have someone > purporting to be an expert offering a completely new solution. That's > why we asked for input now, before Tony spends a lot of time working > on it. > -MC The PT_pt format is for written languages, rather than spoken languages. There is often a difference. The SSML 1.1 spec references http://www.ietf.org/rfc/bcp/bcp47.txt as a definition of how to identify a language and accent for speech. I'm not clear if its really works, though. Steve From anthony.minessale at gmail.com Thu Jul 2 10:39:39 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 2 Jul 2009 12:39:39 -0500 Subject: [Freeswitch-users] SIP re-invite / bypass_media In-Reply-To: <367751820907021027rcb09956j9be2e1837b29b4ff@mail.gmail.com> References: <367751820907011322s1b17192ct3d7303632456db87@mail.gmail.com> <191c3a030907011606m537b6895la167d2cfb79fac7d@mail.gmail.com> <367751820907020841j379749cblef2302e7dcae32da@mail.gmail.com> <191c3a030907020919i5be162b3x69b86b5bcc2eb8e4@mail.gmail.com> <367751820907020953l34363158md0f16e069e5a9d14@mail.gmail.com> <191c3a030907021005r2762e48fu97310abbc8b75b92@mail.gmail.com> <367751820907021027rcb09956j9be2e1837b29b4ff@mail.gmail.com> Message-ID: <191c3a030907021039u7aa74943u496d8992b5901180@mail.gmail.com> no problem On Thu, Jul 2, 2009 at 12:27 PM, Phillip Jones wrote: > Used: > > session.execute("set","bypass_media_after_bridge=true"); > in the confirm.js script and that works perfectly! > > Thank you for you help! > On Thu, Jul 2, 2009 at 1:05 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> try setting bypass_media_after_bridge=true on the session in your confirm >> script >> >> >> >> On Thu, Jul 2, 2009 at 11:53 AM, Phillip Jones wrote: >> >>> Thanks for responding and for your help. >>> >>> The xml and confirm.js are attached below. Basically trying to >>> bypass_media after the leg B presses 1 to accept the call. I tried, >>> using bypass_media_after_bridge=true, but the re-invite appears to be done >>> before the confirm.js, So the media is successfully rerouted, but BEFORE the >>> leg b never gets hear a prompt or gets the opportunity to press 1. >>> >>> To get round this I am trying to manually bypass_media in the confirm.js >>> script with apiExecute("uuid_media", "off " + session.uuid);. However only >>> the B leg is reinvited (and media is routed correctly). I don't see the A >>> leg reinvite, and then a BYE is issueed on both legs. >>> >>> >>> >>> >>> >>> >>> >>> < >>> >>> >>> >>> >>> >>> >>> This is the confirm.js: >>> >>> // confirm.js - FreeSwitch call confirmation script >>> // (c) 2009 - St?phane Alnet >>> // License: GPL2 or above >>> console_log("info", "Destination: "+ session.destination + "\n"); >>> if(!session.getVariable('leg_confirm')) >>> { >>> console_log("info", "No need to confirm, connect the call!\n"); >>> exit(); >>> } >>> var confirmed = false; >>> var confirmation_digit = "1"; >>> var try_count = 6; >>> var prompt_file = "prompts/ToAcceptThisCallPress1.wav"; >>> function onInput( session, type, data, arg ) { >>> if ( type == "dtmf" ) { >>> console_log( "info", "Got digit " + data.digit + "\n" ); >>> if ( data.digit == confirmation_digit ) { >>> confirmed = true; >>> console_log( "info", "Confirming session..\n" ); >>> return(false); >>> } >>> } >>> return(true); >>> } >>> if ( session.ready() ) >>> { >>> session.answer(); >>> session.flushDigits(); >>> console_log("info", "Starting confirmation\n"); >>> var count = try_count; >>> while( session.ready() && ! confirmed && count-- > 0 ) >>> { >>> session.execute("sleep","200"); >>> session.streamFile( prompt_file, onInput ); >>> } >>> >>> if( ! confirmed ) >>> { >>> console_log("info", "Not confirmed\n"); >>> session.hangup(); >>> } >>> else >>> { >>> *apiExecute("uuid_media", "off " + session.uuid);* >>> console_log("info", "Confirmed\n"); >>> } >>> } >>> else >>> { >>> console_log("info", "Session is not ready.\n"); >>> } >>> >>> >>> >>> >>> On Thu, Jul 2, 2009 at 12:19 PM, Anthony Minessale < >>> anthony.minessale at gmail.com> wrote: >>> >>>> I would need to know more details about what you are doing. >>>> >>>> you could set the variable bypass_media_after_bridge=true on the a leg >>>> before you call the b leg and use the group_confirm feature to get the >>>> caller >>>> to press the key. >>>> >>>> >>>> >>>> On Thu, Jul 2, 2009 at 10:41 AM, Phillip Jones wrote: >>>> >>>>> Thanks for that. >>>>> >>>>> That seems to successfully re-invite and re-route the the B leg - but >>>>> does not reinvite the A leg and then immediately issues a "bye" on both >>>>> legs. >>>>> >>>>> Do I have to do something to reinvite that A leg? >>>>> >>>>> On Wed, Jul 1, 2009 at 7:06 PM, Anthony Minessale < >>>>> anthony.minessale at gmail.com> wrote: >>>>> >>>>>> try >>>>>> apiExecute("uuid_media", "off " + session.uuid); >>>>>> >>>>>> >>>>>> >>>>>> On Wed, Jul 1, 2009 at 3:22 PM, Phillip Jones >>>>> > wrote: >>>>>> >>>>>>> Hi there, >>>>>>> >>>>>>> I was wondering whether it is possible to have FreeSwitch go into >>>>>>> bypass_media mode on demand? >>>>>>> >>>>>>> For instance, leg a bridges to leg b - leg b is invited to accept the >>>>>>> call by pressing 1. I want to go to bypass_media (do a SIP reinvite to >>>>>>> reroute the media) after the one is pressed. >>>>>>> >>>>>>> Currently I am issuing the following from my js script that prompts >>>>>>> for the 1: >>>>>>> >>>>>>> session.apiExecute("uuid_media",session.uuid); >>>>>>> >>>>>>> Not working however. >>>>>>> >>>>>>> Any help to get me going would be appreciated. >>>>>>> >>>>>>> Thanks >>>>>>> >>>>>>> Phillip Jones. >>>>>>> >>>>>>> _______________________________________________ >>>>>>> Freeswitch-users mailing list >>>>>>> Freeswitch-users at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE: >>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> -- >>>>>> Anthony Minessale II >>>>>> >>>>>> FreeSWITCH http://www.freeswitch.org/ >>>>>> ClueCon http://www.cluecon.com/ >>>>>> >>>>>> AIM: anthm >>>>>> MSN:anthony_minessale at hotmail.com >>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>>>>> IRC: irc.freenode.net #freeswitch >>>>>> >>>>>> FreeSWITCH Developer Conference >>>>>> 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/20090702/7b227aef/attachment-0001.html From Prometheus001 at gmx.net Thu Jul 2 11:16:52 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Thu, 02 Jul 2009 20:16:52 +0200 Subject: [Freeswitch-users] Freswitch closes UDP port after OPTIONS with TLS? In-Reply-To: <10292506-A353-4E33-AA98-87833A40E224@freeswitch.org> References: <4A4C95E9.7010706@gmx.net> <10292506-A353-4E33-AA98-87833A40E224@freeswitch.org> Message-ID: <4A4CF994.7060609@gmx.net> Hello Brian, ok, I got it. Any other idea why the UDP port is closed after the TLS packet? Best regards Peter Brian West schrieb: > If its TLS you don't need options packets in the first place. Your > client should do the keep alive NOT FreeSWITCH. TLS is over TCP and > Options over UDP... doesn't make much sense. > > /b > > On Jul 2, 2009, at 6:11 AM, Peter P GMX wrote: > > >> Hello, >> >> I have the following problem: Every call stops after 30 seconds when >> TLS >> is enabled. SIP/RTP and SIP/SRTP works but not TLS/SRTP. >> The phones are behind NAT. So I expect, that every 30 seconds an >> Options >> request is sent. >> >> Wiresharking the traffic I can see >> >> * that there are ongoing UDP packets. >> * Then a TSLv1 packet ist sent from FS to the Phone. >> * This is acknowleged by the phone >> * Next the phone send another UDP packet to the same FS port as >> before >> * Then the Phone receives an ICMP request that the FS port is >> closed. >> >> >> Anybody has a clue about this? >> >> Best regards >> Peter >> >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.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 Thu Jul 2 11:47:59 2009 From: raul at etellicom.com (Raul Fragoso) Date: Thu, 02 Jul 2009 15:47:59 -0300 Subject: [Freeswitch-users] Language Handling: call for assistance In-Reply-To: <4A4CEE69.8050108@coppice.org> References: <87f2f3b90907011708t161e310em833fc728c7dfdbee@mail.gmail.com> <4A4C857E.3060805@3gnt.net> <87f2f3b90907021007x7033f2bcjeb087a808a9e8c52@mail.gmail.com> <4A4CEE69.8050108@coppice.org> Message-ID: <1246560479.3082.18.camel@raul-laptop> On Fri, 2009-07-03 at 01:29 +0800, Steve Underwood wrote: > Michael Collins wrote: > > > > > > On Thu, Jul 2, 2009 at 3:01 AM, Igor Neves > > wrote: > > > > Hi, > > > > > > Michael Collins wrote: > >> Hello all! > >> > >> There's been some discussion lately on how to handle multiple > >> languages, specifically with the *say* application. We would like > >> some input from the community on how to handle multiple languages > >> and sound files. Anthony notes that the say application needs to > >> build the path to the sound files by using the ${sound_prefix} > >> and ${lang} variables. Some have asked about countries or > >> language variants, like European Portugese vs. Brazilian > >> Portugese. These are good questions. > > > > What it's the problem about Portuguese VS Brazilian? > > > > Can't we just use "PT_pt" and "PT_br" in ${lang}, just like a lot > > of others softwares do? > > > > What about ${sound_prefix} = ${lang}, since ${lang} should always > > be unique, and you make the path's automatically language organized? > > > > > > This is reasonable to me, but it would be nice to have a consensus, > > just to be sure. > > > > > > > > > >> > >> >From the community we need input. If you have experience with > >> multiple languages in a telephony environment then please give us > >> your suggestions. How would you like to see the say application > >> handle various languages and dialects? Please give us your > >> helpful suggestions. > >> > >> Thanks, > >> Michael > > > > Sorry if I miss understood something. > > Cheers, > > > > > > Believe, the moment we put this into place we will have someone > > purporting to be an expert offering a completely new solution. That's > > why we asked for input now, before Tony spends a lot of time working > > on it. > > -MC > The PT_pt format is for written languages, rather than spoken languages. > There is often a difference. > > The SSML 1.1 spec references http://www.ietf.org/rfc/bcp/bcp47.txt as a > definition of how to identify a language and accent for speech. I'm not > clear if its really works, though. > > Steve I think that would be overkill. The usual way of using i.e. "pt-br" (two letters for the main language, dash and then two more letters for the dialect/variation) would be enough. Regards, Raul From dave at 3c.co.uk Thu Jul 2 12:07:51 2009 From: dave at 3c.co.uk (David Knell) Date: Thu, 02 Jul 2009 16:07:51 -0300 Subject: [Freeswitch-users] Language Handling: call for assistance In-Reply-To: <1246560479.3082.18.camel@raul-laptop> References: <87f2f3b90907011708t161e310em833fc728c7dfdbee@mail.gmail.com> <4A4C857E.3060805@3gnt.net> <87f2f3b90907021007x7033f2bcjeb087a808a9e8c52@mail.gmail.com> <4A4CEE69.8050108@coppice.org> <1246560479.3082.18.camel@raul-laptop> Message-ID: <1246561671.32598.7.camel@dk-d820> On Thu, 2009-07-02 at 15:47 -0300, Raul Fragoso wrote: > On Fri, 2009-07-03 at 01:29 +0800, Steve Underwood wrote: > > Michael Collins wrote: > > > > > > > > > On Thu, Jul 2, 2009 at 3:01 AM, Igor Neves > > > wrote: > > > > > > Hi, > > > > > > > > > Michael Collins wrote: > > >> Hello all! > > >> > > >> There's been some discussion lately on how to handle multiple > > >> languages, specifically with the *say* application. We would like > > >> some input from the community on how to handle multiple languages > > >> and sound files. Anthony notes that the say application needs to > > >> build the path to the sound files by using the ${sound_prefix} > > >> and ${lang} variables. Some have asked about countries or > > >> language variants, like European Portugese vs. Brazilian > > >> Portugese. These are good questions. > > > > > > What it's the problem about Portuguese VS Brazilian? > > > > > > Can't we just use "PT_pt" and "PT_br" in ${lang}, just like a lot > > > of others softwares do? > > > > > > What about ${sound_prefix} = ${lang}, since ${lang} should always > > > be unique, and you make the path's automatically language organized? > > > > > > > > > This is reasonable to me, but it would be nice to have a consensus, > > > just to be sure. The - thing would appear to be the obvious choice; what's slightly less obvious is what to do about fallback. Three choices: - None - i.e. I say en-gb, I either get en-gb or nothing; - Best guess - I want en-gb, but I'm quite happy with en-us or en-au. I could specify en-gb and, if en-gb's not available but en-something else is, whatever's available gets chosen for me; - Preference list - I specify a (possibly) wildcarded list of what I want - e.g. en-gb,en-us,en* - and whichever match comes first is what I get. --Dave -- David Knell, Director, 3C Limited T: +44 20 3298 2000 E: dave at 3c.co.uk W: http://www.3c.co.uk From tzury.by at gmail.com Thu Jul 2 12:17:05 2009 From: tzury.by at gmail.com (Tzury Bar Yochay) Date: Thu, 2 Jul 2009 22:17:05 +0300 Subject: [Freeswitch-users] seeking for a freeswitch expert Message-ID: <2f5ea7490907021217g4d736670i64359772bbfc3220@mail.gmail.com> Hi, We are looking to deploy a secured sip environment whereas all traffic, signalling and rtp encrypted etc. according to the rfc's. We are looking for an expert with deep understanding in both fields the freeSwitch and crypto world so to config and compile this for us. you may contact me at tzury.by at reguluslabs.com. kindly provide information about your experience with freeSwitch deployment. -- Tzury Bar Yochay Regulus Labs ltd. http://reguluslabs.com +972 52 5133399 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090702/0201a3e2/attachment.html From mythicalbox at weavver.com Thu Jul 2 12:26:56 2009 From: mythicalbox at weavver.com (Mitchel Constantin) Date: Thu, 2 Jul 2009 12:26:56 -0700 Subject: [Freeswitch-users] How to remove the IP from the SIP caller id number In-Reply-To: References: <20090702055601.GA13328@jdc.jasonjgw.net> Message-ID: Thanks Brian, that worked like a charm. :) On Thu, Jul 2, 2009 at 6:37 AM, Brian West wrote: > set the variable sip_invite_domain > > /b > > On Jul 2, 2009, at 12:56 AM, Jason White wrote: > > > I suspect the other end (whatever device you are calling from > > FreeSWITCH) is > > adding the IP address to the caller id. However, I am no SIP expert > > and may be > > wrong, but you can confirm this by doing a SIP trace on the device > > that > > receives the call (or on its local network via packet capture) to > > discover > > what FreeSWITCH is sending out. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Mitchel Constantin Weavver, Inc. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090702/77402919/attachment.html From brian at freeswitch.org Thu Jul 2 12:33:26 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 2 Jul 2009 14:33:26 -0500 Subject: [Freeswitch-users] How to remove the IP from the SIP caller id number In-Reply-To: References: <20090702055601.GA13328@jdc.jasonjgw.net> Message-ID: <6A9F8D14-C6F5-4CD2-86F3-D54A61FCA2ED@freeswitch.org> For that you now have to attend cluecon! ;) /b On Jul 2, 2009, at 2:26 PM, Mitchel Constantin wrote: > Thanks Brian, that worked like a charm. :) From steveu at coppice.org Thu Jul 2 16:50:29 2009 From: steveu at coppice.org (Steve Underwood) Date: Fri, 03 Jul 2009 07:50:29 +0800 Subject: [Freeswitch-users] Language Handling: call for assistance In-Reply-To: <1246560479.3082.18.camel@raul-laptop> References: <87f2f3b90907011708t161e310em833fc728c7dfdbee@mail.gmail.com> <4A4C857E.3060805@3gnt.net> <87f2f3b90907021007x7033f2bcjeb087a808a9e8c52@mail.gmail.com> <4A4CEE69.8050108@coppice.org> <1246560479.3082.18.camel@raul-laptop> Message-ID: <4A4D47C5.7060205@coppice.org> Raul Fragoso wrote: > On Fri, 2009-07-03 at 01:29 +0800, Steve Underwood wrote: > >> Michael Collins wrote: >> >>> On Thu, Jul 2, 2009 at 3:01 AM, Igor Neves >> > wrote: >>> >>> Hi, >>> >>> >>> Michael Collins wrote: >>> >>>> Hello all! >>>> >>>> There's been some discussion lately on how to handle multiple >>>> languages, specifically with the *say* application. We would like >>>> some input from the community on how to handle multiple languages >>>> and sound files. Anthony notes that the say application needs to >>>> build the path to the sound files by using the ${sound_prefix} >>>> and ${lang} variables. Some have asked about countries or >>>> language variants, like European Portugese vs. Brazilian >>>> Portugese. These are good questions. >>>> >>> What it's the problem about Portuguese VS Brazilian? >>> >>> Can't we just use "PT_pt" and "PT_br" in ${lang}, just like a lot >>> of others softwares do? >>> >>> What about ${sound_prefix} = ${lang}, since ${lang} should always >>> be unique, and you make the path's automatically language organized? >>> >>> >>> This is reasonable to me, but it would be nice to have a consensus, >>> just to be sure. >>> >>> >>> >>> >>> >>>> >From the community we need input. If you have experience with >>>> multiple languages in a telephony environment then please give us >>>> your suggestions. How would you like to see the say application >>>> handle various languages and dialects? Please give us your >>>> helpful suggestions. >>>> >>>> Thanks, >>>> Michael >>>> >>> Sorry if I miss understood something. >>> Cheers, >>> >>> >>> Believe, the moment we put this into place we will have someone >>> purporting to be an expert offering a completely new solution. That's >>> why we asked for input now, before Tony spends a lot of time working >>> on it. >>> -MC >>> >> The PT_pt format is for written languages, rather than spoken languages. >> There is often a difference. >> >> The SSML 1.1 spec references http://www.ietf.org/rfc/bcp/bcp47.txt as a >> definition of how to identify a language and accent for speech. I'm not >> clear if its really works, though. >> >> Steve >> > > > I think that would be overkill. The usual way of using i.e. "pt-br" (two > letters for the main language, dash and then two more letters for the > dialect/variation) would be enough. > If by "the usual way" you mean the standard 2 + 2 letter codes we are used to on computers, that just doesn't work. As I said before, those are for written languages, not spoken languages. There are no standard codes for many spoken languages. For example, the standard codes for Chinese are zh_cn for mainland China, zh_tw for Taiwan, zh_hk for Hong Kong. However, in GuangDong you will probably want to offer Cantonese as well as Mandarin voice prompts, so you will want a zh_gd, or something, which you won't find among the standard 2 + 2 letter codes. That's why the SSML people had a hard time coming up with a language scheme, and SSML 1.0 didn't even reference one. The more you look around the world, the most complex the issue of language variants becomes. If you don't face that at the beginning it just gets messier later on. Steve From mike at jerris.com Thu Jul 2 16:58:17 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 2 Jul 2009 19:58:17 -0400 Subject: [Freeswitch-users] Language Handling: call for assistance In-Reply-To: <4A4D47C5.7060205@coppice.org> References: <87f2f3b90907011708t161e310em833fc728c7dfdbee@mail.gmail.com> <4A4C857E.3060805@3gnt.net> <87f2f3b90907021007x7033f2bcjeb087a808a9e8c52@mail.gmail.com> <4A4CEE69.8050108@coppice.org> <1246560479.3082.18.camel@raul-laptop> <4A4D47C5.7060205@coppice.org> Message-ID: On Jul 2, 2009, at 7:50 PM, Steve Underwood wrote: >> > If by "the usual way" you mean the standard 2 + 2 letter codes we are > used to on computers, that just doesn't work. As I said before, those > are for written languages, not spoken languages. There are no standard > codes for many spoken languages. For example, the standard codes for > Chinese are zh_cn for mainland China, zh_tw for Taiwan, zh_hk for Hong > Kong. However, in GuangDong you will probably want to offer > Cantonese as > well as Mandarin voice prompts, so you will want a zh_gd, or > something, > which you won't find among the standard 2 + 2 letter codes. That's why > the SSML people had a hard time coming up with a language scheme, and > SSML 1.0 didn't even reference one. The more you look around the > world, > the most complex the issue of language variants becomes. If you don't > face that at the beginning it just gets messier later on. > > Steve Do we know that the language model at least always pairs with the first 2 letter code? So zh_* we can use mod_say_zh for? or do we need to address different language rules for different dialects as well? Mike From jaybinks at gmail.com Thu Jul 2 18:40:20 2009 From: jaybinks at gmail.com (Jay Binks) Date: Fri, 03 Jul 2009 11:40:20 +1000 Subject: [Freeswitch-users] BOUNTY - extended RTP stats - RFC3611 Message-ID: <1246585220.5330.26.camel@jay-desktop.home.gateway> howdy all.. ive added some extra $$$ to this existing ( seemingly closed ) bounty http://wiki.freeswitch.org/wiki/Bounty#RFC_3611_-_RTP_Control_Protocol_Extended_Reports_.28RTCP_XR.29_support http://jira.freeswitch.org/browse/BOUNTY-4 it would be sweet if a) a Jira admin would re-open the bounty ticket b) other users would get on board and post some $$$ towards this cause quality monitoring in your network must be important to more than a few of us, so lets show the core devs and offer some support to get some awesome quality QOS Stats into FS. Jay -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090703/906ac4e3/attachment-0001.html From jmesquita at gmail.com Thu Jul 2 20:11:53 2009 From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Fri, 3 Jul 2009 00:11:53 -0300 Subject: [Freeswitch-users] Language Handling: call for assistance In-Reply-To: References: <87f2f3b90907011708t161e310em833fc728c7dfdbee@mail.gmail.com> <4A4C857E.3060805@3gnt.net> <87f2f3b90907021007x7033f2bcjeb087a808a9e8c52@mail.gmail.com> <4A4CEE69.8050108@coppice.org> <1246560479.3082.18.camel@raul-laptop> <4A4D47C5.7060205@coppice.org> Message-ID: <5a8712120907022011w5b174adbx3e02f0f7926c749f@mail.gmail.com> Guys, I don't know if I really get the problem here. I mean, I do get that the 2+2 model does not work not even for where I live. I really hate the fact that all spanish south american dialects (some within the same country) are put in the same bag as it wouldn't matter to ppl so I am with you Steve on this one to find an alternative to the 2+2 model. So, in summary, what I am asking is: What would be the problem with mod_say_es_ar_ba for Porte?o dialect spoken in Buenos Aires, Argentina besides the verbosity of it and the limited amount of "levels" we have? Do we know any country that has a sub-dialect from a dialect? jmesquita PS: Please, forgive me if I totally misunderstood it. Afterall, I do have I high fever. On Thu, Jul 2, 2009 at 8:58 PM, Michael Jerris wrote: > > On Jul 2, 2009, at 7:50 PM, Steve Underwood wrote: > > >> > > If by "the usual way" you mean the standard 2 + 2 letter codes we are > > used to on computers, that just doesn't work. As I said before, those > > are for written languages, not spoken languages. There are no standard > > codes for many spoken languages. For example, the standard codes for > > Chinese are zh_cn for mainland China, zh_tw for Taiwan, zh_hk for Hong > > Kong. However, in GuangDong you will probably want to offer > > Cantonese as > > well as Mandarin voice prompts, so you will want a zh_gd, or > > something, > > which you won't find among the standard 2 + 2 letter codes. That's why > > the SSML people had a hard time coming up with a language scheme, and > > SSML 1.0 didn't even reference one. The more you look around the > > world, > > the most complex the issue of language variants becomes. If you don't > > face that at the beginning it just gets messier later on. > > > > Steve > > Do we know that the language model at least always pairs with the > first 2 letter code? So zh_* we can use mod_say_zh for? or do we > need to address different language rules for different dialects as well? > > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090703/385eb979/attachment.html From darklion11 at yahoo.com Thu Jul 2 20:12:32 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Thu, 2 Jul 2009 20:12:32 -0700 (PDT) Subject: [Freeswitch-users] -ERR SERVICE_NOT_IMPLEMENTED on mobiles and landlines Message-ID: <24316987.post@talk.nabble.com> I have a GSM gateway. The issue is sometimes the calls failed what is the cause of the error this is my logs? This is on my freeswitch logs... 09-06-25 10:21:50 [DEBUG] sofia_glue.c:1913 sofia_glue_tech_set_codec() Set Codec sofia/internal/1001 at 116.541.23.11 PCMU/8000 20 ms 160 samples 2009-06-25 10:21:50 [DEBUG] sofia_glue.c:2915 sofia_glue_negotiate_sdp() Set 2833 dtmf payload to 101 2009-06-25 10:21:50 [DEBUG] sofia.c:3203 sofia_handle_sip_i_state() (sofia/internal/1001 at 116.541.23.11) State Change CS_NEW -> CS_INIT 2009-06-25 10:21:50 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/internal/1001 at 116.541.23.11 [BREAK] 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) Running State Change CS_INIT 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:480 switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) State INIT 2009-06-25 10:21:50 [DEBUG] mod_sofia.c:83 sofia_on_init() sofia/internal/1001 at 116.541.23.12 SOFIA INIT 2009-06-25 10:21:50 [DEBUG] mod_sofia.c:111 sofia_on_init() (sofia/internal/1001 at 116.541.23.11) State Change CS_INIT -> CS_ROUTING 2009-06-25 10:21:50 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/internal/1001 at 116.541.23.11 [BREAK] 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:480 switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) State INIT going to sleep 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) Running State Change CS_ROUTING 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:483 switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) State ROUTING 2009-06-25 10:21:50 [DEBUG] mod_sofia.c:130 sofia_on_routing() sofia/internal/1001 at 116.541.23.11 SOFIA ROUTING 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:78 switch_core_standard_on_routing() sofia/internal/1001 at 116.541.23.11 Standard ROUTING 2009-06-25 10:21:50 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() Processing Edmar->639273642511 in context public Dialplan: sofia/internal/1001 at 116.541.23.11 parsing [public->unloop] continue=false Dialplan: sofia/internal/1001 at 116.541.23.11 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/internal/1001 at 116.541.23.11 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: sofia/internal/1001 at 116.541.23.11 parsing [public->outside_call] continue=true Dialplan: sofia/internal/1001 at 116.541.23.11 Absolute Condition [outside_call] Dialplan: sofia/internal/1001 at 116.541.23.11 Action set(outside_call=true) Dialplan: sofia/internal/1001 at 116.541.23.11 parsing [public->call_debug] continue=true Dialplan: sofia/internal/1001 at 116.541.23.11 Regex (FAIL) [call_debug] ${call_debug}(false) =~ /^true$/ break=never Dialplan: sofia/internal/1001 at 116.541.23.11 parsing [public->public_extensions] continue=false Dialplan: sofia/internal/1001 at 116.541.23.11 Regex (FAIL) [public_extensions] destination_number(639273642511) =~ /^(10[01][0-9])$/ break=on-false Dialplan: sofia/internal/1001 at 116.541.23.11 parsing [public->public_did] continue=false Dialplan: sofia/internal/1001 at 116.541.23.11 Regex (FAIL) [public_did] destination_number(639273642511) =~ /^(5551212)$/ break=on-false 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:114 switch_core_standard_on_routing() (sofia/internal/1001 at 116.541.23.11) State Change CS_ROUTING -> CS_EXECUTE 2009-06-25 10:21:50 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/internal/1001 at 116.541.23.11 [BREAK] 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:483 switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) State ROUTING going to sleep 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) Running State Change CS_EXECUTE 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:490 switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) State EXECUTE 2009-06-25 10:21:50 [DEBUG] mod_sofia.c:173 sofia_on_execute() sofia/internal/1001 at 116.541.23.11 SOFIA EXECUTE 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:151 switch_core_standard_on_execute() sofia/internal/1001 at 116.541.23.11 Standard EXECUTE EXECUTE sofia/internal/1001 at 116.541.23.11 set(outside_call=true) 2009-06-25 10:21:50 [DEBUG] mod_dptools.c:748 set_function() sofia/internal/1001 at 116.541.23.11 SET [outside_call]=[true] 2009-06-25 10:21:50 [NOTICE] switch_core_state_machine.c:179 switch_core_standard_on_execute() Hangup sofia/internal/1001 at 116.541.23.11 [CS_EXECUTE] [NORMAL_CLEARING] 2009-06-25 10:21:50 [DEBUG] switch_channel.c:1660 switch_channel_perform_hangup() Send signal sofia/internal/1001 at 116.541.23.11 [KILL] 2009-06-25 10:21:50 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/internal/1001 at 116.541.23.11 [BREAK] 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:490 switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) State EXECUTE going to sleep 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) Running State Change CS_HANGUP 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:433 switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) State HANGUP 2009-06-25 10:21:50 [DEBUG] mod_sofia.c:323 sofia_on_hangup() Channel sofia/internal/1001 at 116.541.23.11 hanging up, cause: NORMAL_CLEARING 2009-06-25 10:21:50 [DEBUG] mod_sofia.c:399 sofia_on_hangup() Responding to INVITE with: 480 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:46 switch_core_standard_on_hangup() sofia/internal/1001 at 116.541.23.11 Standard HANGUP, cause: NORMAL_CLEARING 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:433 switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) State HANGUP going to sleep 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:475 switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) State Change CS_HANGUP -> CS_REPORTING 2009-06-25 10:21:50 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/internal/1001 at 116.541.23.11 [BREAK] 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) Running State Change CS_REPORTING 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:607 switch_core_session_reporting_state() (sofia/internal/1001 at 116.541.23.11) State REPORTING 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:53 switch_core_standard_on_reporting() sofia/internal/1001 at 116.541.23.11 Standard REPORTING, cause: NORMAL_CLEARING 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:607 switch_core_session_reporting_state() (sofia/internal/1001 at 116.541.23.11) State REPORTING going to sleep 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:410 switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) State Change CS_REPORTING -> CS_DESTROY 2009-06-25 10:21:50 [DEBUG] switch_core_session.c:1067 switch_core_session_thread() Session 3 (sofia/internal/1001 at 116.541.23.11) Locked, Waiting on external entities 2009-06-25 10:21:50 [NOTICE] switch_core_session.c:1085 switch_core_session_thread() Session 3 (sofia/internal/1001 at 116.541.23.11) Ended 2009-06-25 10:21:50 [NOTICE] switch_core_session.c:1087 switch_core_session_thread() Close Channel sofia/internal/1001 at 116.541.23.11 [CS_DESTROY] 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:559 switch_core_session_destroy_state() (sofia/internal/1001 at 116.541.23.11) State DESTROY 2009-06-25 10:21:50 [DEBUG] mod_sofia.c:240 sofia_on_destroy() sofia/internal/1001 at 116.541.23.11 SOFIA DESTROY 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:60 switch_core_standard_on_destroy() sofia/internal/1001 at 116.541.23.11 Standard DESTROY 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:559 switch_core_session_destroy_state() (sofia/internal/1001 at 116.541.23.11) State DESTROY going to sleep Thanks hope u can help me -- View this message in context: http://www.nabble.com/-ERR-SERVICE_NOT_IMPLEMENTED-on-mobiles-and-landlines-tp24316987p24316987.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From mike at jerris.com Thu Jul 2 21:05:10 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 3 Jul 2009 00:05:10 -0400 Subject: [Freeswitch-users] -ERR SERVICE_NOT_IMPLEMENTED on mobiles and landlines In-Reply-To: <24316987.post@talk.nabble.com> References: <24316987.post@talk.nabble.com> Message-ID: Try enabling 3pcc in the sip profile. On Jul 2, 2009, at 11:12 PM, Edmar Cruz wrote: > > I have a GSM gateway. The issue is sometimes the calls failed what > is the > cause of the error this is my logs? > > This is on my freeswitch logs... > > > 09-06-25 10:21:50 [DEBUG] sofia_glue.c:1913 > sofia_glue_tech_set_codec() Set > Codec sofia/internal/1001 at 116.541.23.11 PCMU/8000 20 ms 160 samples > 2009-06-25 10:21:50 [DEBUG] sofia_glue.c:2915 > sofia_glue_negotiate_sdp() Set > 2833 dtmf payload to 101 > 2009-06-25 10:21:50 [DEBUG] sofia.c:3203 sofia_handle_sip_i_state() > (sofia/internal/1001 at 116.541.23.11) State Change CS_NEW -> CS_INIT > 2009-06-25 10:21:50 [DEBUG] switch_core_session.c:933 > switch_core_session_signal_state_change() Send signal > sofia/internal/1001 at 116.541.23.11 [BREAK] > 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:397 > switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) > Running State > Change CS_INIT > 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:480 > switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) State > INIT > 2009-06-25 10:21:50 [DEBUG] mod_sofia.c:83 sofia_on_init() > sofia/internal/1001 at 116.541.23.12 SOFIA INIT > 2009-06-25 10:21:50 [DEBUG] mod_sofia.c:111 sofia_on_init() > (sofia/internal/1001 at 116.541.23.11) State Change CS_INIT -> CS_ROUTING > 2009-06-25 10:21:50 [DEBUG] switch_core_session.c:933 > switch_core_session_signal_state_change() Send signal > sofia/internal/1001 at 116.541.23.11 [BREAK] > 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:480 > switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) State > INIT > going to sleep > 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:397 > switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) > Running State > Change CS_ROUTING > 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:483 > switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) State > ROUTING > 2009-06-25 10:21:50 [DEBUG] mod_sofia.c:130 sofia_on_routing() > sofia/internal/1001 at 116.541.23.11 SOFIA ROUTING > 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:78 > switch_core_standard_on_routing() sofia/internal/1001 at 116.541.23.11 > Standard > ROUTING > 2009-06-25 10:21:50 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() > Processing > Edmar->639273642511 in context public > Dialplan: sofia/internal/1001 at 116.541.23.11 parsing [public->unloop] > continue=false > Dialplan: sofia/internal/1001 at 116.541.23.11 Regex (PASS) [unloop] > ${unroll_loops}(true) =~ /^true$/ break=on-false > Dialplan: sofia/internal/1001 at 116.541.23.11 Regex (FAIL) [unloop] > ${sip_looped_call}() =~ /^true$/ break=on-false > Dialplan: sofia/internal/1001 at 116.541.23.11 parsing [public- > >outside_call] > continue=true > Dialplan: sofia/internal/1001 at 116.541.23.11 Absolute Condition > [outside_call] > Dialplan: sofia/internal/1001 at 116.541.23.11 Action > set(outside_call=true) > Dialplan: sofia/internal/1001 at 116.541.23.11 parsing [public- > >call_debug] > continue=true > Dialplan: sofia/internal/1001 at 116.541.23.11 Regex (FAIL) [call_debug] > ${call_debug}(false) =~ /^true$/ break=never > Dialplan: sofia/internal/1001 at 116.541.23.11 parsing > [public->public_extensions] continue=false > Dialplan: sofia/internal/1001 at 116.541.23.11 Regex (FAIL) > [public_extensions] > destination_number(639273642511) =~ /^(10[01][0-9])$/ break=on-false > Dialplan: sofia/internal/1001 at 116.541.23.11 parsing [public- > >public_did] > continue=false > Dialplan: sofia/internal/1001 at 116.541.23.11 Regex (FAIL) [public_did] > destination_number(639273642511) =~ /^(5551212)$/ break=on-false > 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:114 > switch_core_standard_on_routing() (sofia/internal/ > 1001 at 116.541.23.11) State > Change CS_ROUTING -> CS_EXECUTE > 2009-06-25 10:21:50 [DEBUG] switch_core_session.c:933 > switch_core_session_signal_state_change() Send signal > sofia/internal/1001 at 116.541.23.11 [BREAK] > 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:483 > switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) State > ROUTING > going to sleep > 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:397 > switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) > Running State > Change CS_EXECUTE > 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:490 > switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) State > EXECUTE > 2009-06-25 10:21:50 [DEBUG] mod_sofia.c:173 sofia_on_execute() > sofia/internal/1001 at 116.541.23.11 SOFIA EXECUTE > 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:151 > switch_core_standard_on_execute() sofia/internal/1001 at 116.541.23.11 > Standard > EXECUTE > EXECUTE sofia/internal/1001 at 116.541.23.11 set(outside_call=true) > 2009-06-25 10:21:50 [DEBUG] mod_dptools.c:748 set_function() > sofia/internal/1001 at 116.541.23.11 SET [outside_call]=[true] > 2009-06-25 10:21:50 [NOTICE] switch_core_state_machine.c:179 > switch_core_standard_on_execute() Hangup sofia/internal/1001 at 116.541.23.11 > [CS_EXECUTE] [NORMAL_CLEARING] > 2009-06-25 10:21:50 [DEBUG] switch_channel.c:1660 > switch_channel_perform_hangup() Send signal > sofia/internal/1001 at 116.541.23.11 [KILL] > 2009-06-25 10:21:50 [DEBUG] switch_core_session.c:933 > switch_core_session_signal_state_change() Send signal > sofia/internal/1001 at 116.541.23.11 [BREAK] > 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:490 > switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) State > EXECUTE > going to sleep > 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:397 > switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) > Running State > Change CS_HANGUP > 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:433 > switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) State > HANGUP > 2009-06-25 10:21:50 [DEBUG] mod_sofia.c:323 sofia_on_hangup() Channel > sofia/internal/1001 at 116.541.23.11 hanging up, cause: NORMAL_CLEARING > 2009-06-25 10:21:50 [DEBUG] mod_sofia.c:399 sofia_on_hangup() > Responding to > INVITE with: 480 > 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:46 > switch_core_standard_on_hangup() sofia/internal/1001 at 116.541.23.11 > Standard > HANGUP, cause: NORMAL_CLEARING > 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:433 > switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) State > HANGUP > going to sleep > 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:475 > switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) State > Change > CS_HANGUP -> CS_REPORTING > 2009-06-25 10:21:50 [DEBUG] switch_core_session.c:933 > switch_core_session_signal_state_change() Send signal > sofia/internal/1001 at 116.541.23.11 [BREAK] > 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:397 > switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) > Running State > Change CS_REPORTING > 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:607 > switch_core_session_reporting_state() (sofia/internal/1001 at 116.541.23.11 > ) > State REPORTING > 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:53 > switch_core_standard_on_reporting() sofia/internal/1001 at 116.541.23.11 > Standard REPORTING, cause: NORMAL_CLEARING > 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:607 > switch_core_session_reporting_state() (sofia/internal/1001 at 116.541.23.11 > ) > State REPORTING going to sleep > 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:410 > switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) State > Change > CS_REPORTING -> CS_DESTROY > 2009-06-25 10:21:50 [DEBUG] switch_core_session.c:1067 > switch_core_session_thread() Session 3 (sofia/internal/1001 at 116.541.23.11 > ) > Locked, Waiting on external entities > 2009-06-25 10:21:50 [NOTICE] switch_core_session.c:1085 > switch_core_session_thread() Session 3 (sofia/internal/1001 at 116.541.23.11 > ) > Ended > 2009-06-25 10:21:50 [NOTICE] switch_core_session.c:1087 > switch_core_session_thread() Close Channel sofia/internal/1001 at 116.541.23.11 > [CS_DESTROY] > 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:559 > switch_core_session_destroy_state() (sofia/internal/ > 1001 at 116.541.23.11) > State DESTROY > 2009-06-25 10:21:50 [DEBUG] mod_sofia.c:240 sofia_on_destroy() > sofia/internal/1001 at 116.541.23.11 SOFIA DESTROY > 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:60 > switch_core_standard_on_destroy() sofia/internal/1001 at 116.541.23.11 > Standard > DESTROY > 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:559 > switch_core_session_destroy_state() (sofia/internal/ > 1001 at 116.541.23.11) > State DESTROY going to sleep > > > Thanks hope u can help me > -- > View this message in context: http://www.nabble.com/-ERR-SERVICE_NOT_IMPLEMENTED-on-mobiles-and-landlines-tp24316987p24316987.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 darklion11 at yahoo.com Thu Jul 2 22:24:52 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Thu, 2 Jul 2009 22:24:52 -0700 (PDT) Subject: [Freeswitch-users] -ERR SERVICE_NOT_IMPLEMENTED on mobiles and landlines In-Reply-To: References: <24316987.post@talk.nabble.com> Message-ID: <24317784.post@talk.nabble.com> The same issue... Michael Jerris wrote: > > Try enabling 3pcc in the sip profile. > > On Jul 2, 2009, at 11:12 PM, Edmar Cruz wrote: > >> >> I have a GSM gateway. The issue is sometimes the calls failed what >> is the >> cause of the error this is my logs? >> >> This is on my freeswitch logs... >> >> >> 09-06-25 10:21:50 [DEBUG] sofia_glue.c:1913 >> sofia_glue_tech_set_codec() Set >> Codec sofia/internal/1001 at 116.541.23.11 PCMU/8000 20 ms 160 samples >> 2009-06-25 10:21:50 [DEBUG] sofia_glue.c:2915 >> sofia_glue_negotiate_sdp() Set >> 2833 dtmf payload to 101 >> 2009-06-25 10:21:50 [DEBUG] sofia.c:3203 sofia_handle_sip_i_state() >> (sofia/internal/1001 at 116.541.23.11) State Change CS_NEW -> CS_INIT >> 2009-06-25 10:21:50 [DEBUG] switch_core_session.c:933 >> switch_core_session_signal_state_change() Send signal >> sofia/internal/1001 at 116.541.23.11 [BREAK] >> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:397 >> switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) >> Running State >> Change CS_INIT >> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:480 >> switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) State >> INIT >> 2009-06-25 10:21:50 [DEBUG] mod_sofia.c:83 sofia_on_init() >> sofia/internal/1001 at 116.541.23.12 SOFIA INIT >> 2009-06-25 10:21:50 [DEBUG] mod_sofia.c:111 sofia_on_init() >> (sofia/internal/1001 at 116.541.23.11) State Change CS_INIT -> CS_ROUTING >> 2009-06-25 10:21:50 [DEBUG] switch_core_session.c:933 >> switch_core_session_signal_state_change() Send signal >> sofia/internal/1001 at 116.541.23.11 [BREAK] >> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:480 >> switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) State >> INIT >> going to sleep >> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:397 >> switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) >> Running State >> Change CS_ROUTING >> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:483 >> switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) State >> ROUTING >> 2009-06-25 10:21:50 [DEBUG] mod_sofia.c:130 sofia_on_routing() >> sofia/internal/1001 at 116.541.23.11 SOFIA ROUTING >> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:78 >> switch_core_standard_on_routing() sofia/internal/1001 at 116.541.23.11 >> Standard >> ROUTING >> 2009-06-25 10:21:50 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() >> Processing >> Edmar->639273642511 in context public >> Dialplan: sofia/internal/1001 at 116.541.23.11 parsing [public->unloop] >> continue=false >> Dialplan: sofia/internal/1001 at 116.541.23.11 Regex (PASS) [unloop] >> ${unroll_loops}(true) =~ /^true$/ break=on-false >> Dialplan: sofia/internal/1001 at 116.541.23.11 Regex (FAIL) [unloop] >> ${sip_looped_call}() =~ /^true$/ break=on-false >> Dialplan: sofia/internal/1001 at 116.541.23.11 parsing [public- >> >outside_call] >> continue=true >> Dialplan: sofia/internal/1001 at 116.541.23.11 Absolute Condition >> [outside_call] >> Dialplan: sofia/internal/1001 at 116.541.23.11 Action >> set(outside_call=true) >> Dialplan: sofia/internal/1001 at 116.541.23.11 parsing [public- >> >call_debug] >> continue=true >> Dialplan: sofia/internal/1001 at 116.541.23.11 Regex (FAIL) [call_debug] >> ${call_debug}(false) =~ /^true$/ break=never >> Dialplan: sofia/internal/1001 at 116.541.23.11 parsing >> [public->public_extensions] continue=false >> Dialplan: sofia/internal/1001 at 116.541.23.11 Regex (FAIL) >> [public_extensions] >> destination_number(639273642511) =~ /^(10[01][0-9])$/ break=on-false >> Dialplan: sofia/internal/1001 at 116.541.23.11 parsing [public- >> >public_did] >> continue=false >> Dialplan: sofia/internal/1001 at 116.541.23.11 Regex (FAIL) [public_did] >> destination_number(639273642511) =~ /^(5551212)$/ break=on-false >> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:114 >> switch_core_standard_on_routing() (sofia/internal/ >> 1001 at 116.541.23.11) State >> Change CS_ROUTING -> CS_EXECUTE >> 2009-06-25 10:21:50 [DEBUG] switch_core_session.c:933 >> switch_core_session_signal_state_change() Send signal >> sofia/internal/1001 at 116.541.23.11 [BREAK] >> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:483 >> switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) State >> ROUTING >> going to sleep >> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:397 >> switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) >> Running State >> Change CS_EXECUTE >> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:490 >> switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) State >> EXECUTE >> 2009-06-25 10:21:50 [DEBUG] mod_sofia.c:173 sofia_on_execute() >> sofia/internal/1001 at 116.541.23.11 SOFIA EXECUTE >> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:151 >> switch_core_standard_on_execute() sofia/internal/1001 at 116.541.23.11 >> Standard >> EXECUTE >> EXECUTE sofia/internal/1001 at 116.541.23.11 set(outside_call=true) >> 2009-06-25 10:21:50 [DEBUG] mod_dptools.c:748 set_function() >> sofia/internal/1001 at 116.541.23.11 SET [outside_call]=[true] >> 2009-06-25 10:21:50 [NOTICE] switch_core_state_machine.c:179 >> switch_core_standard_on_execute() Hangup >> sofia/internal/1001 at 116.541.23.11 >> [CS_EXECUTE] [NORMAL_CLEARING] >> 2009-06-25 10:21:50 [DEBUG] switch_channel.c:1660 >> switch_channel_perform_hangup() Send signal >> sofia/internal/1001 at 116.541.23.11 [KILL] >> 2009-06-25 10:21:50 [DEBUG] switch_core_session.c:933 >> switch_core_session_signal_state_change() Send signal >> sofia/internal/1001 at 116.541.23.11 [BREAK] >> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:490 >> switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) State >> EXECUTE >> going to sleep >> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:397 >> switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) >> Running State >> Change CS_HANGUP >> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:433 >> switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) State >> HANGUP >> 2009-06-25 10:21:50 [DEBUG] mod_sofia.c:323 sofia_on_hangup() Channel >> sofia/internal/1001 at 116.541.23.11 hanging up, cause: NORMAL_CLEARING >> 2009-06-25 10:21:50 [DEBUG] mod_sofia.c:399 sofia_on_hangup() >> Responding to >> INVITE with: 480 >> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:46 >> switch_core_standard_on_hangup() sofia/internal/1001 at 116.541.23.11 >> Standard >> HANGUP, cause: NORMAL_CLEARING >> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:433 >> switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) State >> HANGUP >> going to sleep >> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:475 >> switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) State >> Change >> CS_HANGUP -> CS_REPORTING >> 2009-06-25 10:21:50 [DEBUG] switch_core_session.c:933 >> switch_core_session_signal_state_change() Send signal >> sofia/internal/1001 at 116.541.23.11 [BREAK] >> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:397 >> switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) >> Running State >> Change CS_REPORTING >> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:607 >> switch_core_session_reporting_state() (sofia/internal/1001 at 116.541.23.11 >> ) >> State REPORTING >> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:53 >> switch_core_standard_on_reporting() sofia/internal/1001 at 116.541.23.11 >> Standard REPORTING, cause: NORMAL_CLEARING >> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:607 >> switch_core_session_reporting_state() (sofia/internal/1001 at 116.541.23.11 >> ) >> State REPORTING going to sleep >> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:410 >> switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) State >> Change >> CS_REPORTING -> CS_DESTROY >> 2009-06-25 10:21:50 [DEBUG] switch_core_session.c:1067 >> switch_core_session_thread() Session 3 (sofia/internal/1001 at 116.541.23.11 >> ) >> Locked, Waiting on external entities >> 2009-06-25 10:21:50 [NOTICE] switch_core_session.c:1085 >> switch_core_session_thread() Session 3 (sofia/internal/1001 at 116.541.23.11 >> ) >> Ended >> 2009-06-25 10:21:50 [NOTICE] switch_core_session.c:1087 >> switch_core_session_thread() Close Channel >> sofia/internal/1001 at 116.541.23.11 >> [CS_DESTROY] >> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:559 >> switch_core_session_destroy_state() (sofia/internal/ >> 1001 at 116.541.23.11) >> State DESTROY >> 2009-06-25 10:21:50 [DEBUG] mod_sofia.c:240 sofia_on_destroy() >> sofia/internal/1001 at 116.541.23.11 SOFIA DESTROY >> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:60 >> switch_core_standard_on_destroy() sofia/internal/1001 at 116.541.23.11 >> Standard >> DESTROY >> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:559 >> switch_core_session_destroy_state() (sofia/internal/ >> 1001 at 116.541.23.11) >> State DESTROY going to sleep >> >> >> Thanks hope u can help me >> -- >> View this message in context: >> http://www.nabble.com/-ERR-SERVICE_NOT_IMPLEMENTED-on-mobiles-and-landlines-tp24316987p24316987.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/-ERR-SERVICE_NOT_IMPLEMENTED-on-mobiles-and-landlines-tp24316987p24317784.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From darklion11 at yahoo.com Thu Jul 2 22:25:18 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Thu, 2 Jul 2009 22:25:18 -0700 (PDT) Subject: [Freeswitch-users] -ERR SERVICE_NOT_IMPLEMENTED on mobiles and landlines Message-ID: <24317784.post@talk.nabble.com> The same issue... param name="enable-3pcc" value="true"/> Michael Jerris wrote: > > Try enabling 3pcc in the sip profile. > > On Jul 2, 2009, at 11:12 PM, Edmar Cruz wrote: > >> >> I have a GSM gateway. The issue is sometimes the calls failed what >> is the >> cause of the error this is my logs? >> >> This is on my freeswitch logs... >> >> >> 09-06-25 10:21:50 [DEBUG] sofia_glue.c:1913 >> sofia_glue_tech_set_codec() Set >> Codec sofia/internal/1001 at 116.541.23.11 PCMU/8000 20 ms 160 samples >> 2009-06-25 10:21:50 [DEBUG] sofia_glue.c:2915 >> sofia_glue_negotiate_sdp() Set >> 2833 dtmf payload to 101 >> 2009-06-25 10:21:50 [DEBUG] sofia.c:3203 sofia_handle_sip_i_state() >> (sofia/internal/1001 at 116.541.23.11) State Change CS_NEW -> CS_INIT >> 2009-06-25 10:21:50 [DEBUG] switch_core_session.c:933 >> switch_core_session_signal_state_change() Send signal >> sofia/internal/1001 at 116.541.23.11 [BREAK] >> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:397 >> switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) >> Running State >> Change CS_INIT >> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:480 >> switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) State >> INIT >> 2009-06-25 10:21:50 [DEBUG] mod_sofia.c:83 sofia_on_init() >> sofia/internal/1001 at 116.541.23.12 SOFIA INIT >> 2009-06-25 10:21:50 [DEBUG] mod_sofia.c:111 sofia_on_init() >> (sofia/internal/1001 at 116.541.23.11) State Change CS_INIT -> CS_ROUTING >> 2009-06-25 10:21:50 [DEBUG] switch_core_session.c:933 >> switch_core_session_signal_state_change() Send signal >> sofia/internal/1001 at 116.541.23.11 [BREAK] >> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:480 >> switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) State >> INIT >> going to sleep >> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:397 >> switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) >> Running State >> Change CS_ROUTING >> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:483 >> switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) State >> ROUTING >> 2009-06-25 10:21:50 [DEBUG] mod_sofia.c:130 sofia_on_routing() >> sofia/internal/1001 at 116.541.23.11 SOFIA ROUTING >> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:78 >> switch_core_standard_on_routing() sofia/internal/1001 at 116.541.23.11 >> Standard >> ROUTING >> 2009-06-25 10:21:50 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() >> Processing >> Edmar->639273642511 in context public >> Dialplan: sofia/internal/1001 at 116.541.23.11 parsing [public->unloop] >> continue=false >> Dialplan: sofia/internal/1001 at 116.541.23.11 Regex (PASS) [unloop] >> ${unroll_loops}(true) =~ /^true$/ break=on-false >> Dialplan: sofia/internal/1001 at 116.541.23.11 Regex (FAIL) [unloop] >> ${sip_looped_call}() =~ /^true$/ break=on-false >> Dialplan: sofia/internal/1001 at 116.541.23.11 parsing [public- >> >outside_call] >> continue=true >> Dialplan: sofia/internal/1001 at 116.541.23.11 Absolute Condition >> [outside_call] >> Dialplan: sofia/internal/1001 at 116.541.23.11 Action >> set(outside_call=true) >> Dialplan: sofia/internal/1001 at 116.541.23.11 parsing [public- >> >call_debug] >> continue=true >> Dialplan: sofia/internal/1001 at 116.541.23.11 Regex (FAIL) [call_debug] >> ${call_debug}(false) =~ /^true$/ break=never >> Dialplan: sofia/internal/1001 at 116.541.23.11 parsing >> [public->public_extensions] continue=false >> Dialplan: sofia/internal/1001 at 116.541.23.11 Regex (FAIL) >> [public_extensions] >> destination_number(639273642511) =~ /^(10[01][0-9])$/ break=on-false >> Dialplan: sofia/internal/1001 at 116.541.23.11 parsing [public- >> >public_did] >> continue=false >> Dialplan: sofia/internal/1001 at 116.541.23.11 Regex (FAIL) [public_did] >> destination_number(639273642511) =~ /^(5551212)$/ break=on-false >> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:114 >> switch_core_standard_on_routing() (sofia/internal/ >> 1001 at 116.541.23.11) State >> Change CS_ROUTING -> CS_EXECUTE >> 2009-06-25 10:21:50 [DEBUG] switch_core_session.c:933 >> switch_core_session_signal_state_change() Send signal >> sofia/internal/1001 at 116.541.23.11 [BREAK] >> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:483 >> switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) State >> ROUTING >> going to sleep >> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:397 >> switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) >> Running State >> Change CS_EXECUTE >> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:490 >> switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) State >> EXECUTE >> 2009-06-25 10:21:50 [DEBUG] mod_sofia.c:173 sofia_on_execute() >> sofia/internal/1001 at 116.541.23.11 SOFIA EXECUTE >> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:151 >> switch_core_standard_on_execute() sofia/internal/1001 at 116.541.23.11 >> Standard >> EXECUTE >> EXECUTE sofia/internal/1001 at 116.541.23.11 set(outside_call=true) >> 2009-06-25 10:21:50 [DEBUG] mod_dptools.c:748 set_function() >> sofia/internal/1001 at 116.541.23.11 SET [outside_call]=[true] >> 2009-06-25 10:21:50 [NOTICE] switch_core_state_machine.c:179 >> switch_core_standard_on_execute() Hangup >> sofia/internal/1001 at 116.541.23.11 >> [CS_EXECUTE] [NORMAL_CLEARING] >> 2009-06-25 10:21:50 [DEBUG] switch_channel.c:1660 >> switch_channel_perform_hangup() Send signal >> sofia/internal/1001 at 116.541.23.11 [KILL] >> 2009-06-25 10:21:50 [DEBUG] switch_core_session.c:933 >> switch_core_session_signal_state_change() Send signal >> sofia/internal/1001 at 116.541.23.11 [BREAK] >> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:490 >> switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) State >> EXECUTE >> going to sleep >> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:397 >> switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) >> Running State >> Change CS_HANGUP >> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:433 >> switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) State >> HANGUP >> 2009-06-25 10:21:50 [DEBUG] mod_sofia.c:323 sofia_on_hangup() Channel >> sofia/internal/1001 at 116.541.23.11 hanging up, cause: NORMAL_CLEARING >> 2009-06-25 10:21:50 [DEBUG] mod_sofia.c:399 sofia_on_hangup() >> Responding to >> INVITE with: 480 >> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:46 >> switch_core_standard_on_hangup() sofia/internal/1001 at 116.541.23.11 >> Standard >> HANGUP, cause: NORMAL_CLEARING >> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:433 >> switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) State >> HANGUP >> going to sleep >> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:475 >> switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) State >> Change >> CS_HANGUP -> CS_REPORTING >> 2009-06-25 10:21:50 [DEBUG] switch_core_session.c:933 >> switch_core_session_signal_state_change() Send signal >> sofia/internal/1001 at 116.541.23.11 [BREAK] >> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:397 >> switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) >> Running State >> Change CS_REPORTING >> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:607 >> switch_core_session_reporting_state() (sofia/internal/1001 at 116.541.23.11 >> ) >> State REPORTING >> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:53 >> switch_core_standard_on_reporting() sofia/internal/1001 at 116.541.23.11 >> Standard REPORTING, cause: NORMAL_CLEARING >> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:607 >> switch_core_session_reporting_state() (sofia/internal/1001 at 116.541.23.11 >> ) >> State REPORTING going to sleep >> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:410 >> switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) State >> Change >> CS_REPORTING -> CS_DESTROY >> 2009-06-25 10:21:50 [DEBUG] switch_core_session.c:1067 >> switch_core_session_thread() Session 3 (sofia/internal/1001 at 116.541.23.11 >> ) >> Locked, Waiting on external entities >> 2009-06-25 10:21:50 [NOTICE] switch_core_session.c:1085 >> switch_core_session_thread() Session 3 (sofia/internal/1001 at 116.541.23.11 >> ) >> Ended >> 2009-06-25 10:21:50 [NOTICE] switch_core_session.c:1087 >> switch_core_session_thread() Close Channel >> sofia/internal/1001 at 116.541.23.11 >> [CS_DESTROY] >> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:559 >> switch_core_session_destroy_state() (sofia/internal/ >> 1001 at 116.541.23.11) >> State DESTROY >> 2009-06-25 10:21:50 [DEBUG] mod_sofia.c:240 sofia_on_destroy() >> sofia/internal/1001 at 116.541.23.11 SOFIA DESTROY >> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:60 >> switch_core_standard_on_destroy() sofia/internal/1001 at 116.541.23.11 >> Standard >> DESTROY >> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:559 >> switch_core_session_destroy_state() (sofia/internal/ >> 1001 at 116.541.23.11) >> State DESTROY going to sleep >> >> >> Thanks hope u can help me >> -- >> View this message in context: >> http://www.nabble.com/-ERR-SERVICE_NOT_IMPLEMENTED-on-mobiles-and-landlines-tp24316987p24316987.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/-ERR-SERVICE_NOT_IMPLEMENTED-on-mobiles-and-landlines-tp24316987p24317784.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From mike at jerris.com Thu Jul 2 23:13:41 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 3 Jul 2009 02:13:41 -0400 Subject: [Freeswitch-users] -ERR SERVICE_NOT_IMPLEMENTED on mobiles and landlines In-Reply-To: <24317784.post@talk.nabble.com> References: <24317784.post@talk.nabble.com> Message-ID: Looking closer, it looks like it ran the dialplan, executed some actions to set vars and ran out of actions to do so it hung up. On Jul 3, 2009, at 1:25 AM, Edmar Cruz wrote: > > The same issue... > param name="enable-3pcc" value="true"/> > > Michael Jerris wrote: >> >> Try enabling 3pcc in the sip profile. >> >> On Jul 2, 2009, at 11:12 PM, Edmar Cruz wrote: >> >>> >>> I have a GSM gateway. The issue is sometimes the calls failed what >>> is the >>> cause of the error this is my logs? >>> >>> This is on my freeswitch logs... >>> >>> >>> 09-06-25 10:21:50 [DEBUG] sofia_glue.c:1913 >>> sofia_glue_tech_set_codec() Set >>> Codec sofia/internal/1001 at 116.541.23.11 PCMU/8000 20 ms 160 samples >>> 2009-06-25 10:21:50 [DEBUG] sofia_glue.c:2915 >>> sofia_glue_negotiate_sdp() Set >>> 2833 dtmf payload to 101 >>> 2009-06-25 10:21:50 [DEBUG] sofia.c:3203 sofia_handle_sip_i_state() >>> (sofia/internal/1001 at 116.541.23.11) State Change CS_NEW -> CS_INIT >>> 2009-06-25 10:21:50 [DEBUG] switch_core_session.c:933 >>> switch_core_session_signal_state_change() Send signal >>> sofia/internal/1001 at 116.541.23.11 [BREAK] >>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:397 >>> switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) >>> Running State >>> Change CS_INIT >>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:480 >>> switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) State >>> INIT >>> 2009-06-25 10:21:50 [DEBUG] mod_sofia.c:83 sofia_on_init() >>> sofia/internal/1001 at 116.541.23.12 SOFIA INIT >>> 2009-06-25 10:21:50 [DEBUG] mod_sofia.c:111 sofia_on_init() >>> (sofia/internal/1001 at 116.541.23.11) State Change CS_INIT -> >>> CS_ROUTING >>> 2009-06-25 10:21:50 [DEBUG] switch_core_session.c:933 >>> switch_core_session_signal_state_change() Send signal >>> sofia/internal/1001 at 116.541.23.11 [BREAK] >>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:480 >>> switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) State >>> INIT >>> going to sleep >>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:397 >>> switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) >>> Running State >>> Change CS_ROUTING >>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:483 >>> switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) State >>> ROUTING >>> 2009-06-25 10:21:50 [DEBUG] mod_sofia.c:130 sofia_on_routing() >>> sofia/internal/1001 at 116.541.23.11 SOFIA ROUTING >>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:78 >>> switch_core_standard_on_routing() sofia/internal/1001 at 116.541.23.11 >>> Standard >>> ROUTING >>> 2009-06-25 10:21:50 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() >>> Processing >>> Edmar->639273642511 in context public >>> Dialplan: sofia/internal/1001 at 116.541.23.11 parsing [public->unloop] >>> continue=false >>> Dialplan: sofia/internal/1001 at 116.541.23.11 Regex (PASS) [unloop] >>> ${unroll_loops}(true) =~ /^true$/ break=on-false >>> Dialplan: sofia/internal/1001 at 116.541.23.11 Regex (FAIL) [unloop] >>> ${sip_looped_call}() =~ /^true$/ break=on-false >>> Dialplan: sofia/internal/1001 at 116.541.23.11 parsing [public- >>>> outside_call] >>> continue=true >>> Dialplan: sofia/internal/1001 at 116.541.23.11 Absolute Condition >>> [outside_call] >>> Dialplan: sofia/internal/1001 at 116.541.23.11 Action >>> set(outside_call=true) >>> Dialplan: sofia/internal/1001 at 116.541.23.11 parsing [public- >>>> call_debug] >>> continue=true >>> Dialplan: sofia/internal/1001 at 116.541.23.11 Regex (FAIL) >>> [call_debug] >>> ${call_debug}(false) =~ /^true$/ break=never >>> Dialplan: sofia/internal/1001 at 116.541.23.11 parsing >>> [public->public_extensions] continue=false >>> Dialplan: sofia/internal/1001 at 116.541.23.11 Regex (FAIL) >>> [public_extensions] >>> destination_number(639273642511) =~ /^(10[01][0-9])$/ break=on-false >>> Dialplan: sofia/internal/1001 at 116.541.23.11 parsing [public- >>>> public_did] >>> continue=false >>> Dialplan: sofia/internal/1001 at 116.541.23.11 Regex (FAIL) >>> [public_did] >>> destination_number(639273642511) =~ /^(5551212)$/ break=on-false >>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:114 >>> switch_core_standard_on_routing() (sofia/internal/ >>> 1001 at 116.541.23.11) State >>> Change CS_ROUTING -> CS_EXECUTE >>> 2009-06-25 10:21:50 [DEBUG] switch_core_session.c:933 >>> switch_core_session_signal_state_change() Send signal >>> sofia/internal/1001 at 116.541.23.11 [BREAK] >>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:483 >>> switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) State >>> ROUTING >>> going to sleep >>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:397 >>> switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) >>> Running State >>> Change CS_EXECUTE >>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:490 >>> switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) State >>> EXECUTE >>> 2009-06-25 10:21:50 [DEBUG] mod_sofia.c:173 sofia_on_execute() >>> sofia/internal/1001 at 116.541.23.11 SOFIA EXECUTE >>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:151 >>> switch_core_standard_on_execute() sofia/internal/1001 at 116.541.23.11 >>> Standard >>> EXECUTE >>> EXECUTE sofia/internal/1001 at 116.541.23.11 set(outside_call=true) >>> 2009-06-25 10:21:50 [DEBUG] mod_dptools.c:748 set_function() >>> sofia/internal/1001 at 116.541.23.11 SET [outside_call]=[true] >>> 2009-06-25 10:21:50 [NOTICE] switch_core_state_machine.c:179 >>> switch_core_standard_on_execute() Hangup >>> sofia/internal/1001 at 116.541.23.11 >>> [CS_EXECUTE] [NORMAL_CLEARING] >>> 2009-06-25 10:21:50 [DEBUG] switch_channel.c:1660 >>> switch_channel_perform_hangup() Send signal >>> sofia/internal/1001 at 116.541.23.11 [KILL] >>> 2009-06-25 10:21:50 [DEBUG] switch_core_session.c:933 >>> switch_core_session_signal_state_change() Send signal >>> sofia/internal/1001 at 116.541.23.11 [BREAK] >>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:490 >>> switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) State >>> EXECUTE >>> going to sleep >>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:397 >>> switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) >>> Running State >>> Change CS_HANGUP >>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:433 >>> switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) State >>> HANGUP >>> 2009-06-25 10:21:50 [DEBUG] mod_sofia.c:323 sofia_on_hangup() >>> Channel >>> sofia/internal/1001 at 116.541.23.11 hanging up, cause: NORMAL_CLEARING >>> 2009-06-25 10:21:50 [DEBUG] mod_sofia.c:399 sofia_on_hangup() >>> Responding to >>> INVITE with: 480 >>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:46 >>> switch_core_standard_on_hangup() sofia/internal/1001 at 116.541.23.11 >>> Standard >>> HANGUP, cause: NORMAL_CLEARING >>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:433 >>> switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) State >>> HANGUP >>> going to sleep >>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:475 >>> switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) State >>> Change >>> CS_HANGUP -> CS_REPORTING >>> 2009-06-25 10:21:50 [DEBUG] switch_core_session.c:933 >>> switch_core_session_signal_state_change() Send signal >>> sofia/internal/1001 at 116.541.23.11 [BREAK] >>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:397 >>> switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) >>> Running State >>> Change CS_REPORTING >>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:607 >>> switch_core_session_reporting_state() (sofia/internal/1001 at 116.541.23.11 >>> ) >>> State REPORTING >>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:53 >>> switch_core_standard_on_reporting() sofia/internal/ >>> 1001 at 116.541.23.11 >>> Standard REPORTING, cause: NORMAL_CLEARING >>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:607 >>> switch_core_session_reporting_state() (sofia/internal/1001 at 116.541.23.11 >>> ) >>> State REPORTING going to sleep >>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:410 >>> switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) State >>> Change >>> CS_REPORTING -> CS_DESTROY >>> 2009-06-25 10:21:50 [DEBUG] switch_core_session.c:1067 >>> switch_core_session_thread() Session 3 (sofia/internal/1001 at 116.541.23.11 >>> ) >>> Locked, Waiting on external entities >>> 2009-06-25 10:21:50 [NOTICE] switch_core_session.c:1085 >>> switch_core_session_thread() Session 3 (sofia/internal/1001 at 116.541.23.11 >>> ) >>> Ended >>> 2009-06-25 10:21:50 [NOTICE] switch_core_session.c:1087 >>> switch_core_session_thread() Close Channel >>> sofia/internal/1001 at 116.541.23.11 >>> [CS_DESTROY] >>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:559 >>> switch_core_session_destroy_state() (sofia/internal/ >>> 1001 at 116.541.23.11) >>> State DESTROY >>> 2009-06-25 10:21:50 [DEBUG] mod_sofia.c:240 sofia_on_destroy() >>> sofia/internal/1001 at 116.541.23.11 SOFIA DESTROY >>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:60 >>> switch_core_standard_on_destroy() sofia/internal/1001 at 116.541.23.11 >>> Standard >>> DESTROY >>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:559 >>> switch_core_session_destroy_state() (sofia/internal/ >>> 1001 at 116.541.23.11) >>> State DESTROY going to sleep >>> >>> >>> Thanks hope u can help me >>> -- >>> View this message in context: >>> http://www.nabble.com/-ERR-SERVICE_NOT_IMPLEMENTED-on-mobiles-and-landlines-tp24316987p24316987.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/-ERR-SERVICE_NOT_IMPLEMENTED-on-mobiles-and-landlines-tp24316987p24317784.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 darklion11 at yahoo.com Thu Jul 2 23:32:29 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Thu, 2 Jul 2009 23:32:29 -0700 (PDT) Subject: [Freeswitch-users] -ERR SERVICE_NOT_IMPLEMENTED on mobiles and landlines In-Reply-To: References: <24316987.post@talk.nabble.com> <24317784.post@talk.nabble.com> Message-ID: <24318288.post@talk.nabble.com> What do you think I shall do? Michael Jerris wrote: > > Looking closer, it looks like it ran the dialplan, executed some > actions to set vars and ran out of actions to do so it hung up. > > On Jul 3, 2009, at 1:25 AM, Edmar Cruz wrote: > >> >> The same issue... >> param name="enable-3pcc" value="true"/> >> >> Michael Jerris wrote: >>> >>> Try enabling 3pcc in the sip profile. >>> >>> On Jul 2, 2009, at 11:12 PM, Edmar Cruz wrote: >>> >>>> >>>> I have a GSM gateway. The issue is sometimes the calls failed what >>>> is the >>>> cause of the error this is my logs? >>>> >>>> This is on my freeswitch logs... >>>> >>>> >>>> 09-06-25 10:21:50 [DEBUG] sofia_glue.c:1913 >>>> sofia_glue_tech_set_codec() Set >>>> Codec sofia/internal/1001 at 116.541.23.11 PCMU/8000 20 ms 160 samples >>>> 2009-06-25 10:21:50 [DEBUG] sofia_glue.c:2915 >>>> sofia_glue_negotiate_sdp() Set >>>> 2833 dtmf payload to 101 >>>> 2009-06-25 10:21:50 [DEBUG] sofia.c:3203 sofia_handle_sip_i_state() >>>> (sofia/internal/1001 at 116.541.23.11) State Change CS_NEW -> CS_INIT >>>> 2009-06-25 10:21:50 [DEBUG] switch_core_session.c:933 >>>> switch_core_session_signal_state_change() Send signal >>>> sofia/internal/1001 at 116.541.23.11 [BREAK] >>>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:397 >>>> switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) >>>> Running State >>>> Change CS_INIT >>>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:480 >>>> switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) State >>>> INIT >>>> 2009-06-25 10:21:50 [DEBUG] mod_sofia.c:83 sofia_on_init() >>>> sofia/internal/1001 at 116.541.23.12 SOFIA INIT >>>> 2009-06-25 10:21:50 [DEBUG] mod_sofia.c:111 sofia_on_init() >>>> (sofia/internal/1001 at 116.541.23.11) State Change CS_INIT -> >>>> CS_ROUTING >>>> 2009-06-25 10:21:50 [DEBUG] switch_core_session.c:933 >>>> switch_core_session_signal_state_change() Send signal >>>> sofia/internal/1001 at 116.541.23.11 [BREAK] >>>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:480 >>>> switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) State >>>> INIT >>>> going to sleep >>>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:397 >>>> switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) >>>> Running State >>>> Change CS_ROUTING >>>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:483 >>>> switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) State >>>> ROUTING >>>> 2009-06-25 10:21:50 [DEBUG] mod_sofia.c:130 sofia_on_routing() >>>> sofia/internal/1001 at 116.541.23.11 SOFIA ROUTING >>>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:78 >>>> switch_core_standard_on_routing() sofia/internal/1001 at 116.541.23.11 >>>> Standard >>>> ROUTING >>>> 2009-06-25 10:21:50 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() >>>> Processing >>>> Edmar->639273642511 in context public >>>> Dialplan: sofia/internal/1001 at 116.541.23.11 parsing [public->unloop] >>>> continue=false >>>> Dialplan: sofia/internal/1001 at 116.541.23.11 Regex (PASS) [unloop] >>>> ${unroll_loops}(true) =~ /^true$/ break=on-false >>>> Dialplan: sofia/internal/1001 at 116.541.23.11 Regex (FAIL) [unloop] >>>> ${sip_looped_call}() =~ /^true$/ break=on-false >>>> Dialplan: sofia/internal/1001 at 116.541.23.11 parsing [public- >>>>> outside_call] >>>> continue=true >>>> Dialplan: sofia/internal/1001 at 116.541.23.11 Absolute Condition >>>> [outside_call] >>>> Dialplan: sofia/internal/1001 at 116.541.23.11 Action >>>> set(outside_call=true) >>>> Dialplan: sofia/internal/1001 at 116.541.23.11 parsing [public- >>>>> call_debug] >>>> continue=true >>>> Dialplan: sofia/internal/1001 at 116.541.23.11 Regex (FAIL) >>>> [call_debug] >>>> ${call_debug}(false) =~ /^true$/ break=never >>>> Dialplan: sofia/internal/1001 at 116.541.23.11 parsing >>>> [public->public_extensions] continue=false >>>> Dialplan: sofia/internal/1001 at 116.541.23.11 Regex (FAIL) >>>> [public_extensions] >>>> destination_number(639273642511) =~ /^(10[01][0-9])$/ break=on-false >>>> Dialplan: sofia/internal/1001 at 116.541.23.11 parsing [public- >>>>> public_did] >>>> continue=false >>>> Dialplan: sofia/internal/1001 at 116.541.23.11 Regex (FAIL) >>>> [public_did] >>>> destination_number(639273642511) =~ /^(5551212)$/ break=on-false >>>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:114 >>>> switch_core_standard_on_routing() (sofia/internal/ >>>> 1001 at 116.541.23.11) State >>>> Change CS_ROUTING -> CS_EXECUTE >>>> 2009-06-25 10:21:50 [DEBUG] switch_core_session.c:933 >>>> switch_core_session_signal_state_change() Send signal >>>> sofia/internal/1001 at 116.541.23.11 [BREAK] >>>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:483 >>>> switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) State >>>> ROUTING >>>> going to sleep >>>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:397 >>>> switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) >>>> Running State >>>> Change CS_EXECUTE >>>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:490 >>>> switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) State >>>> EXECUTE >>>> 2009-06-25 10:21:50 [DEBUG] mod_sofia.c:173 sofia_on_execute() >>>> sofia/internal/1001 at 116.541.23.11 SOFIA EXECUTE >>>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:151 >>>> switch_core_standard_on_execute() sofia/internal/1001 at 116.541.23.11 >>>> Standard >>>> EXECUTE >>>> EXECUTE sofia/internal/1001 at 116.541.23.11 set(outside_call=true) >>>> 2009-06-25 10:21:50 [DEBUG] mod_dptools.c:748 set_function() >>>> sofia/internal/1001 at 116.541.23.11 SET [outside_call]=[true] >>>> 2009-06-25 10:21:50 [NOTICE] switch_core_state_machine.c:179 >>>> switch_core_standard_on_execute() Hangup >>>> sofia/internal/1001 at 116.541.23.11 >>>> [CS_EXECUTE] [NORMAL_CLEARING] >>>> 2009-06-25 10:21:50 [DEBUG] switch_channel.c:1660 >>>> switch_channel_perform_hangup() Send signal >>>> sofia/internal/1001 at 116.541.23.11 [KILL] >>>> 2009-06-25 10:21:50 [DEBUG] switch_core_session.c:933 >>>> switch_core_session_signal_state_change() Send signal >>>> sofia/internal/1001 at 116.541.23.11 [BREAK] >>>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:490 >>>> switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) State >>>> EXECUTE >>>> going to sleep >>>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:397 >>>> switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) >>>> Running State >>>> Change CS_HANGUP >>>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:433 >>>> switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) State >>>> HANGUP >>>> 2009-06-25 10:21:50 [DEBUG] mod_sofia.c:323 sofia_on_hangup() >>>> Channel >>>> sofia/internal/1001 at 116.541.23.11 hanging up, cause: NORMAL_CLEARING >>>> 2009-06-25 10:21:50 [DEBUG] mod_sofia.c:399 sofia_on_hangup() >>>> Responding to >>>> INVITE with: 480 >>>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:46 >>>> switch_core_standard_on_hangup() sofia/internal/1001 at 116.541.23.11 >>>> Standard >>>> HANGUP, cause: NORMAL_CLEARING >>>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:433 >>>> switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) State >>>> HANGUP >>>> going to sleep >>>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:475 >>>> switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) State >>>> Change >>>> CS_HANGUP -> CS_REPORTING >>>> 2009-06-25 10:21:50 [DEBUG] switch_core_session.c:933 >>>> switch_core_session_signal_state_change() Send signal >>>> sofia/internal/1001 at 116.541.23.11 [BREAK] >>>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:397 >>>> switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) >>>> Running State >>>> Change CS_REPORTING >>>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:607 >>>> switch_core_session_reporting_state() >>>> (sofia/internal/1001 at 116.541.23.11 >>>> ) >>>> State REPORTING >>>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:53 >>>> switch_core_standard_on_reporting() sofia/internal/ >>>> 1001 at 116.541.23.11 >>>> Standard REPORTING, cause: NORMAL_CLEARING >>>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:607 >>>> switch_core_session_reporting_state() >>>> (sofia/internal/1001 at 116.541.23.11 >>>> ) >>>> State REPORTING going to sleep >>>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:410 >>>> switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) State >>>> Change >>>> CS_REPORTING -> CS_DESTROY >>>> 2009-06-25 10:21:50 [DEBUG] switch_core_session.c:1067 >>>> switch_core_session_thread() Session 3 >>>> (sofia/internal/1001 at 116.541.23.11 >>>> ) >>>> Locked, Waiting on external entities >>>> 2009-06-25 10:21:50 [NOTICE] switch_core_session.c:1085 >>>> switch_core_session_thread() Session 3 >>>> (sofia/internal/1001 at 116.541.23.11 >>>> ) >>>> Ended >>>> 2009-06-25 10:21:50 [NOTICE] switch_core_session.c:1087 >>>> switch_core_session_thread() Close Channel >>>> sofia/internal/1001 at 116.541.23.11 >>>> [CS_DESTROY] >>>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:559 >>>> switch_core_session_destroy_state() (sofia/internal/ >>>> 1001 at 116.541.23.11) >>>> State DESTROY >>>> 2009-06-25 10:21:50 [DEBUG] mod_sofia.c:240 sofia_on_destroy() >>>> sofia/internal/1001 at 116.541.23.11 SOFIA DESTROY >>>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:60 >>>> switch_core_standard_on_destroy() sofia/internal/1001 at 116.541.23.11 >>>> Standard >>>> DESTROY >>>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:559 >>>> switch_core_session_destroy_state() (sofia/internal/ >>>> 1001 at 116.541.23.11) >>>> State DESTROY going to sleep >>>> >>>> >>>> Thanks hope u can help me >>>> -- >>>> View this message in context: >>>> http://www.nabble.com/-ERR-SERVICE_NOT_IMPLEMENTED-on-mobiles-and-landlines-tp24316987p24316987.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/-ERR-SERVICE_NOT_IMPLEMENTED-on-mobiles-and-landlines-tp24316987p24317784.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/-ERR-SERVICE_NOT_IMPLEMENTED-on-mobiles-and-landlines-tp24316987p24318288.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From darklion11 at yahoo.com Fri Jul 3 03:30:45 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Fri, 3 Jul 2009 03:30:45 -0700 (PDT) Subject: [Freeswitch-users] Database for Nibble Rates? Message-ID: <24320955.post@talk.nabble.com> Is there any options make nibble rates to MySQL database? -- View this message in context: http://www.nabble.com/Database-for-Nibble-Rates--tp24320955p24320955.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From shaheryarkh at googlemail.com Fri Jul 3 04:43:29 2009 From: shaheryarkh at googlemail.com (Muhammad Shahzad) Date: Fri, 3 Jul 2009 17:43:29 +0600 Subject: [Freeswitch-users] Baby Update! In-Reply-To: <3EC5EF44-BF81-4C23-AD0B-B4AAB5F7DFCE@freeswitch.org> References: <3EC5EF44-BF81-4C23-AD0B-B4AAB5F7DFCE@freeswitch.org> Message-ID: Congratulations to Ray and Samantha. Lets see what new features and bug fixes we will get in their "new version"..! ;-) Thank you. On 7/2/09, Brian West wrote: > > FreeSWITCHers, > > Kaiden Anthony Chandler will arrive sometime Friday July 3rd > 2009!!! > So to help out with any last minute expenses and help ease things up > for Ray and Samantha and remove some of the worry I'm going to donate > $100 dollars myself to the cause... never know diapers and various > other expenses that come up. Be sure to select the "personal" option > on paypal so they don't take any money from the transaction. Paypal: > intralanman at gmail.com > > And congratulations to Ray and Samantha on their first Boy! > > Thanks everyone you're a great community! > > /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 > -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090703/7df7aabe/attachment.html From asannucci at gmail.com Fri Jul 3 06:11:55 2009 From: asannucci at gmail.com (bakko) Date: Fri, 3 Jul 2009 15:11:55 +0200 Subject: [Freeswitch-users] Database for Nibble Rates? In-Reply-To: <24320955.post@talk.nabble.com> References: <24320955.post@talk.nabble.com> Message-ID: <400F7AB6F7804720978E7F77B4D0C868@voztovoice> look at mod_lcr From brian at freeswitch.org Fri Jul 3 06:27:13 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 3 Jul 2009 08:27:13 -0500 Subject: [Freeswitch-users] Baby Update! In-Reply-To: References: <3EC5EF44-BF81-4C23-AD0B-B4AAB5F7DFCE@freeswitch.org> Message-ID: <4BAB1A87-4967-46C3-852A-CC990AA927CE@freeswitch.org> HAHAHAH love it! /b On Jul 3, 2009, at 6:43 AM, Muhammad Shahzad wrote: > Congratulations to Ray and Samantha. Lets see what new features and > bug fixes we will get in their "new version"..! ;-) > > Thank you. From edpimentl at gmail.com Fri Jul 3 07:01:09 2009 From: edpimentl at gmail.com (EdPimentl) Date: Fri, 3 Jul 2009 10:01:09 -0400 Subject: [Freeswitch-users] Database for Nibble Rates? In-Reply-To: <400F7AB6F7804720978E7F77B4D0C868@voztovoice> References: <24320955.post@talk.nabble.com> <400F7AB6F7804720978E7F77B4D0C868@voztovoice> Message-ID: <9dc4a1670907030701q1804409as63e157a913ab718c@mail.gmail.com> CouchDB -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090703/612f4b81/attachment.html From dave at 3c.co.uk Fri Jul 3 06:58:58 2009 From: dave at 3c.co.uk (David Knell) Date: Fri, 03 Jul 2009 10:58:58 -0300 Subject: [Freeswitch-users] Baby Update! In-Reply-To: References: <3EC5EF44-BF81-4C23-AD0B-B4AAB5F7DFCE@freeswitch.org> Message-ID: <1246629538.4185.9.camel@dk-d820> On Fri, 2009-07-03 at 17:43 +0600, Muhammad Shahzad wrote: > Congratulations to Ray and Samantha. Lets see what new features and > bug fixes we will get in their "new version"..! ;-) Bug fixes..?! I'd refer you to Philip Larkin (went to my school, a bit before my time, poet, deceased, recently voted "Britain's favourite poet") whose "This Be The Verse" suggests otherwise: http://www.artofeurope.com/larkin/lar2.htm [as a recent father myself, I'm trying to prove him wrong..] --Dave -- David Knell, Director, 3C Limited T: +44 20 3298 2000 E: dave at 3c.co.uk W: http://www.3c.co.uk From brian at freeswitch.org Fri Jul 3 07:06:48 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 3 Jul 2009 09:06:48 -0500 Subject: [Freeswitch-users] Database for Nibble Rates? In-Reply-To: <9dc4a1670907030701q1804409as63e157a913ab718c@mail.gmail.com> References: <24320955.post@talk.nabble.com> <400F7AB6F7804720978E7F77B4D0C868@voztovoice> <9dc4a1670907030701q1804409as63e157a913ab718c@mail.gmail.com> Message-ID: <78DD27A7-7E47-418B-88E5-2E5997B1BF24@freeswitch.org> Is that a database for lazy people? :P /b On Jul 3, 2009, at 9:01 AM, EdPimentl wrote: > CouchDB From mike at jerris.com Fri Jul 3 07:24:06 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 3 Jul 2009 10:24:06 -0400 Subject: [Freeswitch-users] -ERR SERVICE_NOT_IMPLEMENTED on mobiles and landlines In-Reply-To: <24318288.post@talk.nabble.com> References: <24316987.post@talk.nabble.com> <24317784.post@talk.nabble.com> <24318288.post@talk.nabble.com> Message-ID: I would guess your looking to do more than set variables, you should make it do those other actions in the dialplan On Jul 3, 2009, at 2:32 AM, Edmar Cruz wrote: > > What do you think I shall do? > > Michael Jerris wrote: >> >> Looking closer, it looks like it ran the dialplan, executed some >> actions to set vars and ran out of actions to do so it hung up. >> >> On Jul 3, 2009, at 1:25 AM, Edmar Cruz wrote: >> >>> >>> The same issue... >>> param name="enable-3pcc" value="true"/> >>> >>> Michael Jerris wrote: >>>> >>>> Try enabling 3pcc in the sip profile. >>>> >>>> On Jul 2, 2009, at 11:12 PM, Edmar Cruz >>>> wrote: >>>> >>>>> >>>>> I have a GSM gateway. The issue is sometimes the calls failed what >>>>> is the >>>>> cause of the error this is my logs? >>>>> >>>>> This is on my freeswitch logs... >>>>> >>>>> >>>>> 09-06-25 10:21:50 [DEBUG] sofia_glue.c:1913 >>>>> sofia_glue_tech_set_codec() Set >>>>> Codec sofia/internal/1001 at 116.541.23.11 PCMU/8000 20 ms 160 >>>>> samples >>>>> 2009-06-25 10:21:50 [DEBUG] sofia_glue.c:2915 >>>>> sofia_glue_negotiate_sdp() Set >>>>> 2833 dtmf payload to 101 >>>>> 2009-06-25 10:21:50 [DEBUG] sofia.c:3203 >>>>> sofia_handle_sip_i_state() >>>>> (sofia/internal/1001 at 116.541.23.11) State Change CS_NEW -> CS_INIT >>>>> 2009-06-25 10:21:50 [DEBUG] switch_core_session.c:933 >>>>> switch_core_session_signal_state_change() Send signal >>>>> sofia/internal/1001 at 116.541.23.11 [BREAK] >>>>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:397 >>>>> switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) >>>>> Running State >>>>> Change CS_INIT >>>>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:480 >>>>> switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) >>>>> State >>>>> INIT >>>>> 2009-06-25 10:21:50 [DEBUG] mod_sofia.c:83 sofia_on_init() >>>>> sofia/internal/1001 at 116.541.23.12 SOFIA INIT >>>>> 2009-06-25 10:21:50 [DEBUG] mod_sofia.c:111 sofia_on_init() >>>>> (sofia/internal/1001 at 116.541.23.11) State Change CS_INIT -> >>>>> CS_ROUTING >>>>> 2009-06-25 10:21:50 [DEBUG] switch_core_session.c:933 >>>>> switch_core_session_signal_state_change() Send signal >>>>> sofia/internal/1001 at 116.541.23.11 [BREAK] >>>>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:480 >>>>> switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) >>>>> State >>>>> INIT >>>>> going to sleep >>>>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:397 >>>>> switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) >>>>> Running State >>>>> Change CS_ROUTING >>>>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:483 >>>>> switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) >>>>> State >>>>> ROUTING >>>>> 2009-06-25 10:21:50 [DEBUG] mod_sofia.c:130 sofia_on_routing() >>>>> sofia/internal/1001 at 116.541.23.11 SOFIA ROUTING >>>>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:78 >>>>> switch_core_standard_on_routing() sofia/internal/ >>>>> 1001 at 116.541.23.11 >>>>> Standard >>>>> ROUTING >>>>> 2009-06-25 10:21:50 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() >>>>> Processing >>>>> Edmar->639273642511 in context public >>>>> Dialplan: sofia/internal/1001 at 116.541.23.11 parsing [public- >>>>> >unloop] >>>>> continue=false >>>>> Dialplan: sofia/internal/1001 at 116.541.23.11 Regex (PASS) [unloop] >>>>> ${unroll_loops}(true) =~ /^true$/ break=on-false >>>>> Dialplan: sofia/internal/1001 at 116.541.23.11 Regex (FAIL) [unloop] >>>>> ${sip_looped_call}() =~ /^true$/ break=on-false >>>>> Dialplan: sofia/internal/1001 at 116.541.23.11 parsing [public- >>>>>> outside_call] >>>>> continue=true >>>>> Dialplan: sofia/internal/1001 at 116.541.23.11 Absolute Condition >>>>> [outside_call] >>>>> Dialplan: sofia/internal/1001 at 116.541.23.11 Action >>>>> set(outside_call=true) >>>>> Dialplan: sofia/internal/1001 at 116.541.23.11 parsing [public- >>>>>> call_debug] >>>>> continue=true >>>>> Dialplan: sofia/internal/1001 at 116.541.23.11 Regex (FAIL) >>>>> [call_debug] >>>>> ${call_debug}(false) =~ /^true$/ break=never >>>>> Dialplan: sofia/internal/1001 at 116.541.23.11 parsing >>>>> [public->public_extensions] continue=false >>>>> Dialplan: sofia/internal/1001 at 116.541.23.11 Regex (FAIL) >>>>> [public_extensions] >>>>> destination_number(639273642511) =~ /^(10[01][0-9])$/ break=on- >>>>> false >>>>> Dialplan: sofia/internal/1001 at 116.541.23.11 parsing [public- >>>>>> public_did] >>>>> continue=false >>>>> Dialplan: sofia/internal/1001 at 116.541.23.11 Regex (FAIL) >>>>> [public_did] >>>>> destination_number(639273642511) =~ /^(5551212)$/ break=on-false >>>>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:114 >>>>> switch_core_standard_on_routing() (sofia/internal/ >>>>> 1001 at 116.541.23.11) State >>>>> Change CS_ROUTING -> CS_EXECUTE >>>>> 2009-06-25 10:21:50 [DEBUG] switch_core_session.c:933 >>>>> switch_core_session_signal_state_change() Send signal >>>>> sofia/internal/1001 at 116.541.23.11 [BREAK] >>>>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:483 >>>>> switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) >>>>> State >>>>> ROUTING >>>>> going to sleep >>>>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:397 >>>>> switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) >>>>> Running State >>>>> Change CS_EXECUTE >>>>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:490 >>>>> switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) >>>>> State >>>>> EXECUTE >>>>> 2009-06-25 10:21:50 [DEBUG] mod_sofia.c:173 sofia_on_execute() >>>>> sofia/internal/1001 at 116.541.23.11 SOFIA EXECUTE >>>>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:151 >>>>> switch_core_standard_on_execute() sofia/internal/ >>>>> 1001 at 116.541.23.11 >>>>> Standard >>>>> EXECUTE >>>>> EXECUTE sofia/internal/1001 at 116.541.23.11 set(outside_call=true) >>>>> 2009-06-25 10:21:50 [DEBUG] mod_dptools.c:748 set_function() >>>>> sofia/internal/1001 at 116.541.23.11 SET [outside_call]=[true] >>>>> 2009-06-25 10:21:50 [NOTICE] switch_core_state_machine.c:179 >>>>> switch_core_standard_on_execute() Hangup >>>>> sofia/internal/1001 at 116.541.23.11 >>>>> [CS_EXECUTE] [NORMAL_CLEARING] >>>>> 2009-06-25 10:21:50 [DEBUG] switch_channel.c:1660 >>>>> switch_channel_perform_hangup() Send signal >>>>> sofia/internal/1001 at 116.541.23.11 [KILL] >>>>> 2009-06-25 10:21:50 [DEBUG] switch_core_session.c:933 >>>>> switch_core_session_signal_state_change() Send signal >>>>> sofia/internal/1001 at 116.541.23.11 [BREAK] >>>>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:490 >>>>> switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) >>>>> State >>>>> EXECUTE >>>>> going to sleep >>>>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:397 >>>>> switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) >>>>> Running State >>>>> Change CS_HANGUP >>>>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:433 >>>>> switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) >>>>> State >>>>> HANGUP >>>>> 2009-06-25 10:21:50 [DEBUG] mod_sofia.c:323 sofia_on_hangup() >>>>> Channel >>>>> sofia/internal/1001 at 116.541.23.11 hanging up, cause: >>>>> NORMAL_CLEARING >>>>> 2009-06-25 10:21:50 [DEBUG] mod_sofia.c:399 sofia_on_hangup() >>>>> Responding to >>>>> INVITE with: 480 >>>>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:46 >>>>> switch_core_standard_on_hangup() sofia/internal/1001 at 116.541.23.11 >>>>> Standard >>>>> HANGUP, cause: NORMAL_CLEARING >>>>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:433 >>>>> switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) >>>>> State >>>>> HANGUP >>>>> going to sleep >>>>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:475 >>>>> switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) >>>>> State >>>>> Change >>>>> CS_HANGUP -> CS_REPORTING >>>>> 2009-06-25 10:21:50 [DEBUG] switch_core_session.c:933 >>>>> switch_core_session_signal_state_change() Send signal >>>>> sofia/internal/1001 at 116.541.23.11 [BREAK] >>>>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:397 >>>>> switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) >>>>> Running State >>>>> Change CS_REPORTING >>>>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:607 >>>>> switch_core_session_reporting_state() >>>>> (sofia/internal/1001 at 116.541.23.11 >>>>> ) >>>>> State REPORTING >>>>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:53 >>>>> switch_core_standard_on_reporting() sofia/internal/ >>>>> 1001 at 116.541.23.11 >>>>> Standard REPORTING, cause: NORMAL_CLEARING >>>>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:607 >>>>> switch_core_session_reporting_state() >>>>> (sofia/internal/1001 at 116.541.23.11 >>>>> ) >>>>> State REPORTING going to sleep >>>>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:410 >>>>> switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) >>>>> State >>>>> Change >>>>> CS_REPORTING -> CS_DESTROY >>>>> 2009-06-25 10:21:50 [DEBUG] switch_core_session.c:1067 >>>>> switch_core_session_thread() Session 3 >>>>> (sofia/internal/1001 at 116.541.23.11 >>>>> ) >>>>> Locked, Waiting on external entities >>>>> 2009-06-25 10:21:50 [NOTICE] switch_core_session.c:1085 >>>>> switch_core_session_thread() Session 3 >>>>> (sofia/internal/1001 at 116.541.23.11 >>>>> ) >>>>> Ended >>>>> 2009-06-25 10:21:50 [NOTICE] switch_core_session.c:1087 >>>>> switch_core_session_thread() Close Channel >>>>> sofia/internal/1001 at 116.541.23.11 >>>>> [CS_DESTROY] >>>>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:559 >>>>> switch_core_session_destroy_state() (sofia/internal/ >>>>> 1001 at 116.541.23.11) >>>>> State DESTROY >>>>> 2009-06-25 10:21:50 [DEBUG] mod_sofia.c:240 sofia_on_destroy() >>>>> sofia/internal/1001 at 116.541.23.11 SOFIA DESTROY >>>>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:60 >>>>> switch_core_standard_on_destroy() sofia/internal/ >>>>> 1001 at 116.541.23.11 >>>>> Standard >>>>> DESTROY >>>>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:559 >>>>> switch_core_session_destroy_state() (sofia/internal/ >>>>> 1001 at 116.541.23.11) >>>>> State DESTROY going to sleep >>>>> >>>>> >>>>> Thanks hope u can help me >>>>> -- >>>>> View this message in context: >>>>> http://www.nabble.com/-ERR-SERVICE_NOT_IMPLEMENTED-on-mobiles-and-landlines-tp24316987p24316987.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/-ERR-SERVICE_NOT_IMPLEMENTED-on-mobiles-and-landlines-tp24316987p24317784.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/-ERR-SERVICE_NOT_IMPLEMENTED-on-mobiles-and-landlines-tp24316987p24318288.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 mike at jerris.com Fri Jul 3 07:27:58 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 3 Jul 2009 10:27:58 -0400 Subject: [Freeswitch-users] Database for Nibble Rates? In-Reply-To: <24320955.post@talk.nabble.com> References: <24320955.post@talk.nabble.com> Message-ID: <6DB183A1-5E35-4CA0-BF19-FC42D7EEEC81@jerris.com> It can go to anything that works with odbc theoretically. On Jul 3, 2009, at 6:30 AM, Edmar Cruz wrote: > > Is there any options make nibble rates to MySQL database? > -- > View this message in context: http://www.nabble.com/Database-for-Nibble-Rates--tp24320955p24320955.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 gustavodartagnan at yahoo.com Fri Jul 3 08:58:05 2009 From: gustavodartagnan at yahoo.com (Gustavo Dartagnan Xavier) Date: Fri, 3 Jul 2009 08:58:05 -0700 (PDT) Subject: [Freeswitch-users] Trying to confirm answer directly on Originate command Message-ID: <143896.88647.qm@web57106.mail.re3.yahoo.com> I'm trying to build a click to call app using the FreeSWITCH webapi. Maybe I'm trying the wrong way, but, I couldn't understand why it's playing the confirmation wave only on the second leg, and not only on the first one. Is this originate command correct? http://myFreeSWITCH:8080/webapi/originate?{forked_dial=false,ignore_early_media=true,origination_caller_id_name=aleg,origination_caller_id_number=aleg,originate_timeout=30}[group_confirm_key=5,group_confirm_file=playback/path/to/press5.wav]sofia/gateway/agateway/usera at adomaind%20&bridge(sofia/gateway/bgateway/buser at bdomain) Is therea an easy way to do it? Thanks in advance, Gustavo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090703/8f1bbf08/attachment.html From darklion11 at yahoo.com Fri Jul 3 09:24:02 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Fri, 3 Jul 2009 09:24:02 -0700 (PDT) Subject: [Freeswitch-users] -ERR SERVICE_NOT_IMPLEMENTED on mobiles and landlines In-Reply-To: References: <24316987.post@talk.nabble.com> <24317784.post@talk.nabble.com> <24318288.post@talk.nabble.com> Message-ID: <24325570.post@talk.nabble.com> Can I have some examples? Michael Jerris wrote: > > I would guess your looking to do more than set variables, you should > make it do those other actions in the dialplan > > On Jul 3, 2009, at 2:32 AM, Edmar Cruz wrote: > >> >> What do you think I shall do? >> >> Michael Jerris wrote: >>> >>> Looking closer, it looks like it ran the dialplan, executed some >>> actions to set vars and ran out of actions to do so it hung up. >>> >>> On Jul 3, 2009, at 1:25 AM, Edmar Cruz wrote: >>> >>>> >>>> The same issue... >>>> param name="enable-3pcc" value="true"/> >>>> >>>> Michael Jerris wrote: >>>>> >>>>> Try enabling 3pcc in the sip profile. >>>>> >>>>> On Jul 2, 2009, at 11:12 PM, Edmar Cruz >>>>> wrote: >>>>> >>>>>> >>>>>> I have a GSM gateway. The issue is sometimes the calls failed what >>>>>> is the >>>>>> cause of the error this is my logs? >>>>>> >>>>>> This is on my freeswitch logs... >>>>>> >>>>>> >>>>>> 09-06-25 10:21:50 [DEBUG] sofia_glue.c:1913 >>>>>> sofia_glue_tech_set_codec() Set >>>>>> Codec sofia/internal/1001 at 116.541.23.11 PCMU/8000 20 ms 160 >>>>>> samples >>>>>> 2009-06-25 10:21:50 [DEBUG] sofia_glue.c:2915 >>>>>> sofia_glue_negotiate_sdp() Set >>>>>> 2833 dtmf payload to 101 >>>>>> 2009-06-25 10:21:50 [DEBUG] sofia.c:3203 >>>>>> sofia_handle_sip_i_state() >>>>>> (sofia/internal/1001 at 116.541.23.11) State Change CS_NEW -> CS_INIT >>>>>> 2009-06-25 10:21:50 [DEBUG] switch_core_session.c:933 >>>>>> switch_core_session_signal_state_change() Send signal >>>>>> sofia/internal/1001 at 116.541.23.11 [BREAK] >>>>>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:397 >>>>>> switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) >>>>>> Running State >>>>>> Change CS_INIT >>>>>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:480 >>>>>> switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) >>>>>> State >>>>>> INIT >>>>>> 2009-06-25 10:21:50 [DEBUG] mod_sofia.c:83 sofia_on_init() >>>>>> sofia/internal/1001 at 116.541.23.12 SOFIA INIT >>>>>> 2009-06-25 10:21:50 [DEBUG] mod_sofia.c:111 sofia_on_init() >>>>>> (sofia/internal/1001 at 116.541.23.11) State Change CS_INIT -> >>>>>> CS_ROUTING >>>>>> 2009-06-25 10:21:50 [DEBUG] switch_core_session.c:933 >>>>>> switch_core_session_signal_state_change() Send signal >>>>>> sofia/internal/1001 at 116.541.23.11 [BREAK] >>>>>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:480 >>>>>> switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) >>>>>> State >>>>>> INIT >>>>>> going to sleep >>>>>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:397 >>>>>> switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) >>>>>> Running State >>>>>> Change CS_ROUTING >>>>>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:483 >>>>>> switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) >>>>>> State >>>>>> ROUTING >>>>>> 2009-06-25 10:21:50 [DEBUG] mod_sofia.c:130 sofia_on_routing() >>>>>> sofia/internal/1001 at 116.541.23.11 SOFIA ROUTING >>>>>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:78 >>>>>> switch_core_standard_on_routing() sofia/internal/ >>>>>> 1001 at 116.541.23.11 >>>>>> Standard >>>>>> ROUTING >>>>>> 2009-06-25 10:21:50 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() >>>>>> Processing >>>>>> Edmar->639273642511 in context public >>>>>> Dialplan: sofia/internal/1001 at 116.541.23.11 parsing [public- >>>>>> >unloop] >>>>>> continue=false >>>>>> Dialplan: sofia/internal/1001 at 116.541.23.11 Regex (PASS) [unloop] >>>>>> ${unroll_loops}(true) =~ /^true$/ break=on-false >>>>>> Dialplan: sofia/internal/1001 at 116.541.23.11 Regex (FAIL) [unloop] >>>>>> ${sip_looped_call}() =~ /^true$/ break=on-false >>>>>> Dialplan: sofia/internal/1001 at 116.541.23.11 parsing [public- >>>>>>> outside_call] >>>>>> continue=true >>>>>> Dialplan: sofia/internal/1001 at 116.541.23.11 Absolute Condition >>>>>> [outside_call] >>>>>> Dialplan: sofia/internal/1001 at 116.541.23.11 Action >>>>>> set(outside_call=true) >>>>>> Dialplan: sofia/internal/1001 at 116.541.23.11 parsing [public- >>>>>>> call_debug] >>>>>> continue=true >>>>>> Dialplan: sofia/internal/1001 at 116.541.23.11 Regex (FAIL) >>>>>> [call_debug] >>>>>> ${call_debug}(false) =~ /^true$/ break=never >>>>>> Dialplan: sofia/internal/1001 at 116.541.23.11 parsing >>>>>> [public->public_extensions] continue=false >>>>>> Dialplan: sofia/internal/1001 at 116.541.23.11 Regex (FAIL) >>>>>> [public_extensions] >>>>>> destination_number(639273642511) =~ /^(10[01][0-9])$/ break=on- >>>>>> false >>>>>> Dialplan: sofia/internal/1001 at 116.541.23.11 parsing [public- >>>>>>> public_did] >>>>>> continue=false >>>>>> Dialplan: sofia/internal/1001 at 116.541.23.11 Regex (FAIL) >>>>>> [public_did] >>>>>> destination_number(639273642511) =~ /^(5551212)$/ break=on-false >>>>>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:114 >>>>>> switch_core_standard_on_routing() (sofia/internal/ >>>>>> 1001 at 116.541.23.11) State >>>>>> Change CS_ROUTING -> CS_EXECUTE >>>>>> 2009-06-25 10:21:50 [DEBUG] switch_core_session.c:933 >>>>>> switch_core_session_signal_state_change() Send signal >>>>>> sofia/internal/1001 at 116.541.23.11 [BREAK] >>>>>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:483 >>>>>> switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) >>>>>> State >>>>>> ROUTING >>>>>> going to sleep >>>>>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:397 >>>>>> switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) >>>>>> Running State >>>>>> Change CS_EXECUTE >>>>>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:490 >>>>>> switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) >>>>>> State >>>>>> EXECUTE >>>>>> 2009-06-25 10:21:50 [DEBUG] mod_sofia.c:173 sofia_on_execute() >>>>>> sofia/internal/1001 at 116.541.23.11 SOFIA EXECUTE >>>>>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:151 >>>>>> switch_core_standard_on_execute() sofia/internal/ >>>>>> 1001 at 116.541.23.11 >>>>>> Standard >>>>>> EXECUTE >>>>>> EXECUTE sofia/internal/1001 at 116.541.23.11 set(outside_call=true) >>>>>> 2009-06-25 10:21:50 [DEBUG] mod_dptools.c:748 set_function() >>>>>> sofia/internal/1001 at 116.541.23.11 SET [outside_call]=[true] >>>>>> 2009-06-25 10:21:50 [NOTICE] switch_core_state_machine.c:179 >>>>>> switch_core_standard_on_execute() Hangup >>>>>> sofia/internal/1001 at 116.541.23.11 >>>>>> [CS_EXECUTE] [NORMAL_CLEARING] >>>>>> 2009-06-25 10:21:50 [DEBUG] switch_channel.c:1660 >>>>>> switch_channel_perform_hangup() Send signal >>>>>> sofia/internal/1001 at 116.541.23.11 [KILL] >>>>>> 2009-06-25 10:21:50 [DEBUG] switch_core_session.c:933 >>>>>> switch_core_session_signal_state_change() Send signal >>>>>> sofia/internal/1001 at 116.541.23.11 [BREAK] >>>>>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:490 >>>>>> switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) >>>>>> State >>>>>> EXECUTE >>>>>> going to sleep >>>>>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:397 >>>>>> switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) >>>>>> Running State >>>>>> Change CS_HANGUP >>>>>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:433 >>>>>> switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) >>>>>> State >>>>>> HANGUP >>>>>> 2009-06-25 10:21:50 [DEBUG] mod_sofia.c:323 sofia_on_hangup() >>>>>> Channel >>>>>> sofia/internal/1001 at 116.541.23.11 hanging up, cause: >>>>>> NORMAL_CLEARING >>>>>> 2009-06-25 10:21:50 [DEBUG] mod_sofia.c:399 sofia_on_hangup() >>>>>> Responding to >>>>>> INVITE with: 480 >>>>>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:46 >>>>>> switch_core_standard_on_hangup() sofia/internal/1001 at 116.541.23.11 >>>>>> Standard >>>>>> HANGUP, cause: NORMAL_CLEARING >>>>>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:433 >>>>>> switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) >>>>>> State >>>>>> HANGUP >>>>>> going to sleep >>>>>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:475 >>>>>> switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) >>>>>> State >>>>>> Change >>>>>> CS_HANGUP -> CS_REPORTING >>>>>> 2009-06-25 10:21:50 [DEBUG] switch_core_session.c:933 >>>>>> switch_core_session_signal_state_change() Send signal >>>>>> sofia/internal/1001 at 116.541.23.11 [BREAK] >>>>>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:397 >>>>>> switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) >>>>>> Running State >>>>>> Change CS_REPORTING >>>>>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:607 >>>>>> switch_core_session_reporting_state() >>>>>> (sofia/internal/1001 at 116.541.23.11 >>>>>> ) >>>>>> State REPORTING >>>>>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:53 >>>>>> switch_core_standard_on_reporting() sofia/internal/ >>>>>> 1001 at 116.541.23.11 >>>>>> Standard REPORTING, cause: NORMAL_CLEARING >>>>>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:607 >>>>>> switch_core_session_reporting_state() >>>>>> (sofia/internal/1001 at 116.541.23.11 >>>>>> ) >>>>>> State REPORTING going to sleep >>>>>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:410 >>>>>> switch_core_session_run() (sofia/internal/1001 at 116.541.23.11) >>>>>> State >>>>>> Change >>>>>> CS_REPORTING -> CS_DESTROY >>>>>> 2009-06-25 10:21:50 [DEBUG] switch_core_session.c:1067 >>>>>> switch_core_session_thread() Session 3 >>>>>> (sofia/internal/1001 at 116.541.23.11 >>>>>> ) >>>>>> Locked, Waiting on external entities >>>>>> 2009-06-25 10:21:50 [NOTICE] switch_core_session.c:1085 >>>>>> switch_core_session_thread() Session 3 >>>>>> (sofia/internal/1001 at 116.541.23.11 >>>>>> ) >>>>>> Ended >>>>>> 2009-06-25 10:21:50 [NOTICE] switch_core_session.c:1087 >>>>>> switch_core_session_thread() Close Channel >>>>>> sofia/internal/1001 at 116.541.23.11 >>>>>> [CS_DESTROY] >>>>>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:559 >>>>>> switch_core_session_destroy_state() (sofia/internal/ >>>>>> 1001 at 116.541.23.11) >>>>>> State DESTROY >>>>>> 2009-06-25 10:21:50 [DEBUG] mod_sofia.c:240 sofia_on_destroy() >>>>>> sofia/internal/1001 at 116.541.23.11 SOFIA DESTROY >>>>>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:60 >>>>>> switch_core_standard_on_destroy() sofia/internal/ >>>>>> 1001 at 116.541.23.11 >>>>>> Standard >>>>>> DESTROY >>>>>> 2009-06-25 10:21:50 [DEBUG] switch_core_state_machine.c:559 >>>>>> switch_core_session_destroy_state() (sofia/internal/ >>>>>> 1001 at 116.541.23.11) >>>>>> State DESTROY going to sleep >>>>>> >>>>>> >>>>>> Thanks hope u can help me >>>>>> -- >>>>>> View this message in context: >>>>>> http://www.nabble.com/-ERR-SERVICE_NOT_IMPLEMENTED-on-mobiles-and-landlines-tp24316987p24316987.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/-ERR-SERVICE_NOT_IMPLEMENTED-on-mobiles-and-landlines-tp24316987p24317784.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/-ERR-SERVICE_NOT_IMPLEMENTED-on-mobiles-and-landlines-tp24316987p24318288.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/-ERR-SERVICE_NOT_IMPLEMENTED-on-mobiles-and-landlines-tp24316987p24325570.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From brian at freeswitch.org Fri Jul 3 09:27:01 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 3 Jul 2009 11:27:01 -0500 Subject: [Freeswitch-users] -ERR SERVICE_NOT_IMPLEMENTED on mobiles and landlines In-Reply-To: <24325570.post@talk.nabble.com> References: <24316987.post@talk.nabble.com> <24317784.post@talk.nabble.com> <24318288.post@talk.nabble.com> <24325570.post@talk.nabble.com> Message-ID: Yes create an entry to route 639273642511 in the dialplan. /b On Jul 3, 2009, at 11:24 AM, Edmar Cruz wrote: > > Can I have some examples? From brian at freeswitch.org Fri Jul 3 13:31:49 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 3 Jul 2009 15:31:49 -0500 Subject: [Freeswitch-users] Baby Update! In-Reply-To: <1246629538.4185.9.camel@dk-d820> References: <3EC5EF44-BF81-4C23-AD0B-B4AAB5F7DFCE@freeswitch.org> <1246629538.4185.9.camel@dk-d820> Message-ID: <41646A56-C830-489D-99D7-CE5C1218E194@freeswitch.org> Kaiden Anthony Chandler arrived Friday July 3rd at 1411 EDT 7lbs 10oz YAY... Congrats mr Lanman! /b On Jul 3, 2009, at 8:58 AM, David Knell wrote: > On Fri, 2009-07-03 at 17:43 +0600, Muhammad Shahzad wrote: >> Congratulations to Ray and Samantha. Lets see what new features and >> bug fixes we will get in their "new version"..! ;-) > > Bug fixes..?! I'd refer you to Philip Larkin (went to my school, a > bit > before my time, poet, deceased, recently voted "Britain's favourite > poet") whose "This Be The Verse" suggests otherwise: > http://www.artofeurope.com/larkin/lar2.htm > > [as a recent father myself, I'm trying to prove him wrong..] > > --Dave > > -- > David Knell, Director, 3C Limited > T: +44 20 3298 2000 > E: dave at 3c.co.uk > W: 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090703/4137d235/attachment-0001.html From gmaruzz at celliax.org Fri Jul 3 16:27:26 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Sat, 4 Jul 2009 01:27:26 +0200 Subject: [Freeswitch-users] Baby Update! In-Reply-To: <41646A56-C830-489D-99D7-CE5C1218E194@freeswitch.org> References: <3EC5EF44-BF81-4C23-AD0B-B4AAB5F7DFCE@freeswitch.org> <1246629538.4185.9.camel@dk-d820> <41646A56-C830-489D-99D7-CE5C1218E194@freeswitch.org> Message-ID: <7b197bef0907031627o4e061c4by745a4492f2de0608@mail.gmail.com> Yeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeh! On Fri, Jul 3, 2009 at 10:31 PM, Brian West wrote: > Kaiden Anthony Chandler arrived Friday July 3rd at 1411 EDT 7lbs 10oz > YAY... Congrats mr Lanman! > /b > On Jul 3, 2009, at 8:58 AM, David Knell wrote: > > On Fri, 2009-07-03 at 17:43 +0600, Muhammad Shahzad wrote: > > Congratulations to Ray and Samantha. Lets see what new features and > > bug fixes we will get in their "new version"..! ;-) > > Bug fixes..?! ?I'd refer you to Philip Larkin (went to my school, a bit > before my time, poet, deceased, recently voted "Britain's favourite > poet") whose "This Be The Verse" suggests otherwise: > http://www.artofeurope.com/larkin/lar2.htm > > [as a recent father myself, I'm trying to prove him wrong..] > > --Dave > > -- > David Knell, Director, 3C Limited > T: +44 20 3298 2000 > E: dave at 3c.co.uk > W: 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 > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.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 Jul 3 16:35:05 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 3 Jul 2009 18:35:05 -0500 Subject: [Freeswitch-users] Baby Update! In-Reply-To: <7b197bef0907031627o4e061c4by745a4492f2de0608@mail.gmail.com> References: <3EC5EF44-BF81-4C23-AD0B-B4AAB5F7DFCE@freeswitch.org> <1246629538.4185.9.camel@dk-d820> <41646A56-C830-489D-99D7-CE5C1218E194@freeswitch.org> <7b197bef0907031627o4e061c4by745a4492f2de0608@mail.gmail.com> Message-ID: <23FF9919-E96F-4067-832F-6E19A83A4F1F@freeswitch.org> Remember send him a little something to help out with the last minute expenses! ;) Btw lanboy will be at ClueCon ;) As will lanwife! /b On Jul 3, 2009, at 6:27 PM, Giovanni Maruzzelli wrote: > Yeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeh! > > > On Fri, Jul 3, 2009 at 10:31 PM, Brian West > wrote: >> Kaiden Anthony Chandler arrived Friday July 3rd at 1411 EDT 7lbs 10oz >> YAY... Congrats mr Lanman! >> /b >> On Jul 3, 2009, at 8:58 AM, David Knell wrote: >> >> On Fri, 2009-07-03 at 17:43 +0600, Muhammad Shahzad wrote: >> >> Congratulations to Ray and Samantha. Lets see what new features and >> >> bug fixes we will get in their "new version"..! ;-) >> >> Bug fixes..?! I'd refer you to Philip Larkin (went to my school, a >> bit >> before my time, poet, deceased, recently voted "Britain's favourite >> poet") whose "This Be The Verse" suggests otherwise: >> http://www.artofeurope.com/larkin/lar2.htm >> >> [as a recent father myself, I'm trying to prove him wrong..] >> >> --Dave >> >> -- >> David Knell, Director, 3C Limited >> T: +44 20 3298 2000 >> E: dave at 3c.co.uk >> W: 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 >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From diego.viola at gmail.com Fri Jul 3 17:46:25 2009 From: diego.viola at gmail.com (Diego Viola) Date: Fri, 3 Jul 2009 20:46:25 -0400 Subject: [Freeswitch-users] Baby Update! In-Reply-To: <23FF9919-E96F-4067-832F-6E19A83A4F1F@freeswitch.org> References: <3EC5EF44-BF81-4C23-AD0B-B4AAB5F7DFCE@freeswitch.org> <1246629538.4185.9.camel@dk-d820> <41646A56-C830-489D-99D7-CE5C1218E194@freeswitch.org> <7b197bef0907031627o4e061c4by745a4492f2de0608@mail.gmail.com> <23FF9919-E96F-4067-832F-6E19A83A4F1F@freeswitch.org> Message-ID: <86a32abc0907031746x61996750k2a423d3ef0db70da@mail.gmail.com> Congrats! On Fri, Jul 3, 2009 at 7:35 PM, Brian West wrote: > Remember send him a little something to help out with the last minute > expenses! ;) > > Btw lanboy will be at ClueCon ;) As will lanwife! > > /b > > On Jul 3, 2009, at 6:27 PM, Giovanni Maruzzelli wrote: > > > Yeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeh! > > > > > > On Fri, Jul 3, 2009 at 10:31 PM, Brian West > > wrote: > >> Kaiden Anthony Chandler arrived Friday July 3rd at 1411 EDT 7lbs 10oz > >> YAY... Congrats mr Lanman! > >> /b > >> On Jul 3, 2009, at 8:58 AM, David Knell wrote: > >> > >> On Fri, 2009-07-03 at 17:43 +0600, Muhammad Shahzad wrote: > >> > >> Congratulations to Ray and Samantha. Lets see what new features and > >> > >> bug fixes we will get in their "new version"..! ;-) > >> > >> Bug fixes..?! I'd refer you to Philip Larkin (went to my school, a > >> bit > >> before my time, poet, deceased, recently voted "Britain's favourite > >> poet") whose "This Be The Verse" suggests otherwise: > >> http://www.artofeurope.com/larkin/lar2.htm > >> > >> [as a recent father myself, I'm trying to prove him wrong..] > >> > >> --Dave > >> > >> -- > >> David Knell, Director, 3C Limited > >> T: +44 20 3298 2000 > >> E: dave at 3c.co.uk > >> W: 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 > >> > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > >> > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090703/df137151/attachment.html From darklion11 at yahoo.com Fri Jul 3 18:52:24 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Fri, 3 Jul 2009 18:52:24 -0700 (PDT) Subject: [Freeswitch-users] -ERR SERVICE_NOT_IMPLEMENTED on mobiles and landlines In-Reply-To: References: <24316987.post@talk.nabble.com> <24317784.post@talk.nabble.com> <24318288.post@talk.nabble.com> <24325570.post@talk.nabble.com> Message-ID: <24330577.post@talk.nabble.com> Yes that a run but I am not calling a registered user that number is an example Brian West-3 wrote: > > Yes create an entry to route 639273642511 in the dialplan. > > /b > > On Jul 3, 2009, at 11:24 AM, Edmar Cruz wrote: > >> >> Can I have some examples? > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/-ERR-SERVICE_NOT_IMPLEMENTED-on-mobiles-and-landlines-tp24316987p24330577.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From brian at freeswitch.org Fri Jul 3 19:03:34 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 3 Jul 2009 21:03:34 -0500 Subject: [Freeswitch-users] -ERR SERVICE_NOT_IMPLEMENTED on mobiles and landlines In-Reply-To: <24330577.post@talk.nabble.com> References: <24316987.post@talk.nabble.com> <24317784.post@talk.nabble.com> <24318288.post@talk.nabble.com> <24325570.post@talk.nabble.com> <24330577.post@talk.nabble.com> Message-ID: <3EAAAEA3-65A3-4507-A5BC-50D5BBD5F550@freeswitch.org> Your logs showed someone was hunting for that number in context public. was that not the case? /b On Jul 3, 2009, at 8:52 PM, Edmar Cruz wrote: > > Yes that a run but I am not calling a registered user that number > is an > example From darklion11 at yahoo.com Fri Jul 3 19:48:14 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Fri, 3 Jul 2009 19:48:14 -0700 (PDT) Subject: [Freeswitch-users] -ERR SERVICE_NOT_IMPLEMENTED on mobiles and landlines In-Reply-To: <3EAAAEA3-65A3-4507-A5BC-50D5BBD5F550@freeswitch.org> References: <24316987.post@talk.nabble.com> <24317784.post@talk.nabble.com> <24318288.post@talk.nabble.com> <24325570.post@talk.nabble.com> <24330577.post@talk.nabble.com> <3EAAAEA3-65A3-4507-A5BC-50D5BBD5F550@freeswitch.org> Message-ID: <24330845.post@talk.nabble.com> Nope Brian West-3 wrote: > > Your logs showed someone was hunting for that number in context > public. was that not the case? > > /b > > On Jul 3, 2009, at 8:52 PM, Edmar Cruz wrote: > >> >> Yes that a run but I am not calling a registered user that number >> is an >> example > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/-ERR-SERVICE_NOT_IMPLEMENTED-on-mobiles-and-landlines-tp24316987p24330845.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From darklion11 at yahoo.com Fri Jul 3 20:30:17 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Fri, 3 Jul 2009 20:30:17 -0700 (PDT) Subject: [Freeswitch-users] -ERR SERVICE_NOT_IMPLEMENTED on mobiles and landlines In-Reply-To: <24330845.post@talk.nabble.com> References: <24316987.post@talk.nabble.com> <24317784.post@talk.nabble.com> <24318288.post@talk.nabble.com> <24325570.post@talk.nabble.com> <24330577.post@talk.nabble.com> <3EAAAEA3-65A3-4507-A5BC-50D5BBD5F550@freeswitch.org> <24330845.post@talk.nabble.com> Message-ID: <24331058.post@talk.nabble.com> I think the problem is on the bridge Am not dialing a registered user so I put $ instead of % Is this correct? Am calling a number 639273642511 SERVICE_NOT_IMPLEMENTED What is cause of the issue? Edmar Cruz wrote: > > Nope > > Brian West-3 wrote: >> >> Your logs showed someone was hunting for that number in context >> public. was that not the case? >> >> /b >> >> On Jul 3, 2009, at 8:52 PM, Edmar Cruz wrote: >> >>> >>> Yes that a run but I am not calling a registered user that number >>> is an >>> example >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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/-ERR-SERVICE_NOT_IMPLEMENTED-on-mobiles-and-landlines-tp24316987p24331058.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From mike at jerris.com Fri Jul 3 20:35:56 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 3 Jul 2009 23:35:56 -0400 Subject: [Freeswitch-users] -ERR SERVICE_NOT_IMPLEMENTED on mobiles and landlines In-Reply-To: <24331058.post@talk.nabble.com> References: <24316987.post@talk.nabble.com> <24317784.post@talk.nabble.com> <24318288.post@talk.nabble.com> <24325570.post@talk.nabble.com> <24330577.post@talk.nabble.com> <3EAAAEA3-65A3-4507-A5BC-50D5BBD5F550@freeswitch.org> <24330845.post@talk.nabble.com> <24331058.post@talk.nabble.com> Message-ID: The log you had in this thread never called bridge at all. On Jul 3, 2009, at 11:30 PM, Edmar Cruz wrote: > > I think the problem is on the bridge > > > Am not dialing a registered user so I put $ instead of % > > Is this correct? > > Am calling a number 639273642511 > > SERVICE_NOT_IMPLEMENTED > > What is cause of the issue? > > Edmar Cruz wrote: >> >> Nope >> >> Brian West-3 wrote: >>> >>> Your logs showed someone was hunting for that number in context >>> public. was that not the case? >>> >>> /b >>> >>> On Jul 3, 2009, at 8:52 PM, Edmar Cruz wrote: >>> >>>> >>>> Yes that a run but I am not calling a registered user that number >>>> is an >>>> example >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at 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/-ERR-SERVICE_NOT_IMPLEMENTED-on-mobiles-and-landlines-tp24316987p24331058.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 jason at jasonjgw.net Fri Jul 3 21:00:32 2009 From: jason at jasonjgw.net (Jason White) Date: Sat, 4 Jul 2009 14:00:32 +1000 Subject: [Freeswitch-users] -ERR SERVICE_NOT_IMPLEMENTED on mobiles and landlines In-Reply-To: <24316987.post@talk.nabble.com> References: <24316987.post@talk.nabble.com> Message-ID: <20090704040032.GA6720@jdc.jasonjgw.net> Edmar Cruz wrote: > > I have a GSM gateway. The issue is sometimes the calls failed what is the > cause of the error this is my logs? The cause of the error is that you are searching the dial plan for 639273642511 in context public, and no dial plan entry matches, so FreeSWITCH terminates the call. It's your task to look at your configuration and work out why this is happening, since clearly it isn't what you intended. From jason at jasonjgw.net Fri Jul 3 21:08:23 2009 From: jason at jasonjgw.net (Jason White) Date: Sat, 4 Jul 2009 14:08:23 +1000 Subject: [Freeswitch-users] -ERR SERVICE_NOT_IMPLEMENTED on mobiles and landlines In-Reply-To: <24331058.post@talk.nabble.com> References: <24317784.post@talk.nabble.com> <24318288.post@talk.nabble.com> <24325570.post@talk.nabble.com> <24330577.post@talk.nabble.com> <3EAAAEA3-65A3-4507-A5BC-50D5BBD5F550@freeswitch.org> <24330845.post@talk.nabble.com> <24331058.post@talk.nabble.com> Message-ID: <20090704040823.GA10745@jdc.jasonjgw.net> Edmar Cruz wrote: > > I think the problem is on the bridge No, it's in the fact that FreeSWITCH fails to match the destination number in the public context. If you've placed the user that is making the call in the public context, and the dial plan entry that you want to match the destination number is in the default context, it isn't going to work. From darklion11 at yahoo.com Fri Jul 3 21:16:59 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Fri, 3 Jul 2009 21:16:59 -0700 (PDT) Subject: [Freeswitch-users] -ERR SERVICE_NOT_IMPLEMENTED on mobiles and landlines In-Reply-To: <20090704040032.GA6720@jdc.jasonjgw.net> References: <24316987.post@talk.nabble.com> <20090704040032.GA6720@jdc.jasonjgw.net> Message-ID: <24331250.post@talk.nabble.com> Sometimes it works but someyimes not? SERVICE_ NOT_IMPLEMENTED This is my dialpla/public/00_test.xml Actually am connecting Freeswitch and Asterisks Do you think the issue is the codec? Asterisks has a G729 License while Freeswitch has none an received an error passthrough mode only available? Jason White-14 wrote: > > Edmar Cruz wrote: >> >> I have a GSM gateway. The issue is sometimes the calls failed what is the >> cause of the error this is my logs? > > The cause of the error is that you are searching the dial plan for > 639273642511 in context public, and no dial plan entry matches, so > FreeSWITCH > terminates the call. > > It's your task to look at your configuration and work out why this is > happening, since clearly it isn't what you intended. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/-ERR-SERVICE_NOT_IMPLEMENTED-on-mobiles-and-landlines-tp24316987p24331250.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From jason at jasonjgw.net Fri Jul 3 23:17:51 2009 From: jason at jasonjgw.net (Jason White) Date: Sat, 4 Jul 2009 16:17:51 +1000 Subject: [Freeswitch-users] -ERR SERVICE_NOT_IMPLEMENTED on mobiles and landlines In-Reply-To: <24331250.post@talk.nabble.com> References: <24316987.post@talk.nabble.com> <20090704040032.GA6720@jdc.jasonjgw.net> <24331250.post@talk.nabble.com> Message-ID: <20090704061751.GA26199@jdc.jasonjgw.net> Edmar Cruz wrote: > This is my dialpla/public/00_test.xml > Where's your tag to open the extension element and give it a name? > > data="effective_caller_id_name=${effective_caller_id_name}"/> > data="effective_caller_id_number=${effective_caller_id_number}"/> The above two lines won't do anything - you're setting the variables to values that they already have. > You now need to close the elements: > Actually am connecting Freeswitch and Asterisks > > Do you think the issue is the codec? No. I've explained this already: the issue in the log that you provided is a failure of FreeSWITCH to match your dial plan extension. From norm at goes.com Sat Jul 4 04:42:08 2009 From: norm at goes.com (Norman Brandinger) Date: Sat, 4 Jul 2009 07:42:08 -0400 (EDT) Subject: [Freeswitch-users] Baby Update! Message-ID: <46295.75.97.76.77.1246707728.squirrel@mail.goes.com> Congrats, Hope mom and baby are doing well. Let's see some photos. Norm > Kaiden Anthony Chandler arrived Friday July 3rd at 1411 EDT 7lbs 10oz > > YAY... Congrats mr Lanman! > > /b > > On Jul 3, 2009, at 8:58 AM, David Knell wrote: > >> On Fri, 2009-07-03 at 17:43 +0600, Muhammad Shahzad wrote: >>> Congratulations to Ray and Samantha. Lets see what new features and >>> bug fixes we will get in their "new version"..! ;-) >> >> Bug fixes..?! I'd refer you to Philip Larkin (went to my school, a >> bit >> before my time, poet, deceased, recently voted "Britain's favourite >> poet") whose "This Be The Verse" suggests otherwise: >> http://www.artofeurope.com/larkin/lar2.htm >> >> [as a recent father myself, I'm trying to prove him wrong..] >> >> --Dave >> >> -- >> David Knell, Director, 3C Limited >> T: +44 20 3298 2000 >> E: dave at 3c.co.uk >> W: 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 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > Global Online (GOES) 271 Main St., Suite C Hackettstown, NJ 07840-2032 (908) 813-0600 x8105 From codecomplete at free.fr Sat Jul 4 13:05:12 2009 From: codecomplete at free.fr (Fred-145) Date: Sat, 4 Jul 2009 13:05:12 -0700 (PDT) Subject: [Freeswitch-users] Cheapest, most compact FreeSwitch appliance? Message-ID: <24337599.post@talk.nabble.com> Hello Atcom's IP appliance is nice but it uses a BlackFish CPU, which is not supported by FreeSwitch yet. What FreeSwitch-appliance do you is the cheapest, most compact out there? If possible, I'd rather a device that has room for an SDD or 2.5" HD so as to be able to use a standard *nix distro. Thank you. -- View this message in context: http://www.nabble.com/Cheapest%2C-most-compact-FreeSwitch-appliance--tp24337599p24337599.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From brian at freeswitch.org Sat Jul 4 13:18:53 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 4 Jul 2009 15:18:53 -0500 Subject: [Freeswitch-users] Cheapest, most compact FreeSwitch appliance? In-Reply-To: <24337599.post@talk.nabble.com> References: <24337599.post@talk.nabble.com> Message-ID: I use one of the intel atom boxes at home. /b On Jul 4, 2009, at 3:05 PM, Fred-145 wrote: > What FreeSwitch-appliance do you is the cheapest, most compact out > there? If > possible, I'd rather a device that has room for an SDD or 2.5" HD so > as to > be able to use a standard *nix distro. From jay.fenton at howlertech.com Sat Jul 4 13:59:26 2009 From: jay.fenton at howlertech.com (Jay Fenton) Date: Sat, 4 Jul 2009 22:59:26 +0200 Subject: [Freeswitch-users] Cheapest, most compact FreeSwitch appliance? In-Reply-To: References: <24337599.post@talk.nabble.com> Message-ID: > I use one of the intel atom boxes at home. A BeagleBoard (http://beagleboard.org/) would make for a decent FreeSWITCH appliance (tiny and only 2 watts). -- Regards, Jay Fenton, CTO Howler Technologies Ltd, 11 Charlotte Mews, London, W1T 4EQ tel: +44 207 099 7095 fax: +44 207 099 7098 http://www.howlertech.com/ http://www.linkedin.com/in/jfenton Registered in England & Wales, Company No. 06285634 From brian at freeswitch.org Sat Jul 4 14:06:50 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 4 Jul 2009 16:06:50 -0500 Subject: [Freeswitch-users] Cheapest, most compact FreeSwitch appliance? In-Reply-To: References: <24337599.post@talk.nabble.com> Message-ID: The problem is what case options do you have for such a device? /b On Jul 4, 2009, at 3:59 PM, Jay Fenton wrote: > >> I use one of the intel atom boxes at home. > > A BeagleBoard (http://beagleboard.org/) would make for a decent > FreeSWITCH appliance (tiny and only 2 watts). > > -- > Regards, > > Jay Fenton, CTO > Howler Technologies Ltd, 11 Charlotte Mews, London, W1T 4EQ > tel: +44 207 099 7095 fax: +44 207 099 7098 > http://www.howlertech.com/ > http://www.linkedin.com/in/jfenton > > Registered in England & Wales, Company No. 06285634 > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.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 Jul 4 14:07:08 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 4 Jul 2009 16:07:08 -0500 Subject: [Freeswitch-users] Cheapest, most compact FreeSwitch appliance? In-Reply-To: <24337599.post@talk.nabble.com> References: <24337599.post@talk.nabble.com> Message-ID: <09DDA996-8374-4CC7-BC79-89B9F7BDCE07@freeswitch.org> http://www.marvell.com/products/embedded_processors/developer/kirkwood/sheevaplug.jsp That would work better. /b On Jul 4, 2009, at 3:05 PM, Fred-145 wrote: > > Hello > > Atcom's IP appliance is nice but it uses a BlackFish CPU, which is not > supported by FreeSwitch yet. > > What FreeSwitch-appliance do you is the cheapest, most compact out > there? If > possible, I'd rather a device that has room for an SDD or 2.5" HD so > as to > be able to use a standard *nix distro. > > Thank you. > -- > View this message in context: http://www.nabble.com/Cheapest%2C-most-compact-FreeSwitch-appliance--tp24337599p24337599.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 william.suffill at gmail.com Sat Jul 4 15:21:10 2009 From: william.suffill at gmail.com (William Suffill) Date: Sat, 4 Jul 2009 18:21:10 -0400 Subject: [Freeswitch-users] Cheapest, most compact FreeSwitch appliance? In-Reply-To: <09DDA996-8374-4CC7-BC79-89B9F7BDCE07@freeswitch.org> References: <24337599.post@talk.nabble.com> <09DDA996-8374-4CC7-BC79-89B9F7BDCE07@freeswitch.org> Message-ID: <6b65470d0907041521t2f79d079x481e2759bfd6a16a@mail.gmail.com> Ya I have a SheevaPlug but yet to have anything interesting to report about making it do anything. The potential is there tho. -- W From gcd at i.ph Sat Jul 4 19:39:16 2009 From: gcd at i.ph (Nandy Dagondon) Date: Sun, 5 Jul 2009 10:39:16 +0800 Subject: [Freeswitch-users] Cheapest, most compact FreeSwitch appliance? In-Reply-To: <6b65470d0907041521t2f79d079x481e2759bfd6a16a@mail.gmail.com> References: <24337599.post@talk.nabble.com> <09DDA996-8374-4CC7-BC79-89B9F7BDCE07@freeswitch.org> <6b65470d0907041521t2f79d079x481e2759bfd6a16a@mail.gmail.com> Message-ID: <7d0bfd8c0907041939h5d49729dwab8f937b3862b3f5@mail.gmail.com> we have a forum on compact,fanless last may. On Sun, Jul 5, 2009 at 6:21 AM, William Suffill wrote: > Ya I have a SheevaPlug but yet to have anything interesting to report > about making it do anything. The potential is there tho. > > -- W > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090705/555d3afe/attachment.html From gcd at i.ph Sat Jul 4 19:41:18 2009 From: gcd at i.ph (Nandy Dagondon) Date: Sun, 5 Jul 2009 10:41:18 +0800 Subject: [Freeswitch-users] Compact, fanless appliance? In-Reply-To: <23430873.post@talk.nabble.com> References: <23193738.post@talk.nabble.com> <1241015506.11362.1.camel@portable-evil> <23317579.post@talk.nabble.com> <7d0bfd8c0904301908o7bca18b5gfe8a830f1f54b41e@mail.gmail.com> <23364535.post@talk.nabble.com> <49FEA513.8020109@mctelefonia.com> <23366596.post@talk.nabble.com> <49FEDA25.2050703@mctelefonia.com> <23430873.post@talk.nabble.com> Message-ID: <7d0bfd8c0907041941s75e541dex982fae2195858ea4@mail.gmail.com> just bumping this topic. -nandy On Fri, May 8, 2009 at 12:44 AM, Fred-145 wrote: > > > Antonio Gallo wrote: > > Alix cases are like 6/9 ? from their shop site. I think its easy to find > > someone who work with aluminium that can make for you custom boxes for > > like like 6/20 ? at pcs > > Unfortunately, none of the PCEngines cases ( > www.pcengines.ch/order1.php?c=2) > allow for a PCI slot, either on top of the mobo, or away from it :-/ > > I'll see if I can get those from Soekris (http://soekris.eu/shop/cases_en/ > ) > allow this, and if I can get a good price for a case + PSU. > > Thank you. > -- > View this message in context: > http://www.nabble.com/Compact%2C-fanless-appliance--tp23193738p23430873.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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090705/1bb0745b/attachment.html From krice at suspicious.org Sat Jul 4 19:49:20 2009 From: krice at suspicious.org (Ken Rice) Date: Sat, 04 Jul 2009 21:49:20 -0500 Subject: [Freeswitch-users] Compact, fanless appliance? In-Reply-To: <7d0bfd8c0907041941s75e541dex982fae2195858ea4@mail.gmail.com> Message-ID: No need to bump these things as this is a mailing list and it annoys quite a few people when you do that From: Nandy Dagondon Reply-To: Date: Sun, 5 Jul 2009 10:41:18 +0800 To: Subject: Re: [Freeswitch-users] Compact, fanless appliance? just bumping this topic. -nandy On Fri, May 8, 2009 at 12:44 AM, Fred-145 wrote: > > > Antonio Gallo wrote: >> > Alix cases are like 6/9 ? from their shop site. I think its easy to find >> > someone who work with aluminium that can make for you custom boxes for >> > like like 6/20 ? at pcs > > Unfortunately, none of the PCEngines cases (www.pcengines.ch/order1.php?c=2 > ) > allow for a PCI slot, either on top of the mobo, or away from it :-/ > > I'll see if I can get those from Soekris (http://soekris.eu/shop/cases_en/) > allow this, and if I can get a good price for a case + PSU. > > Thank you. > -- > View this message in context: > http://www.nabble.com/Compact%2C-fanless-appliance--tp23193738p23430873.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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090704/c5c69221/attachment.html From jay.fenton at howlertech.com Sun Jul 5 03:20:37 2009 From: jay.fenton at howlertech.com (Jay Fenton) Date: Sun, 5 Jul 2009 12:20:37 +0200 Subject: [Freeswitch-users] Cheapest, most compact FreeSwitch appliance? In-Reply-To: References: <24337599.post@talk.nabble.com> Message-ID: <7C0A0FFD-EC03-46DE-A277-AE793017D617@howlertech.com> > The problem is what case options do you have for such a device? https://specialcomp.com/beagleboard/order.htm If you look on there there's a clear acrylic case for the BeagleBoard - I haven't seen any others available. -- Regards, Jay Fenton, CTO Howler Technologies Ltd, 11 Charlotte Mews, London, W1T 4EQ tel: +44 207 099 7095 fax: +44 207 099 7098 http://www.howlertech.com/ http://www.linkedin.com/in/jfenton Registered in England & Wales, Company No. 06285634 From gustavodartagnan at yahoo.com Sun Jul 5 10:43:43 2009 From: gustavodartagnan at yahoo.com (Gustavo Dartagnan Xavier) Date: Sun, 5 Jul 2009 10:43:43 -0700 (PDT) Subject: [Freeswitch-users] Trying to confirm answer directly on Originate command Message-ID: <537628.53680.qm@web57108.mail.re3.yahoo.com> I'm trying to build a click to call app using the FreeSWITCH webapi. Maybe I'm trying the wrong way, but, I couldn't understand why it's playing the confirmation audio only on the second leg, and not only on the first one. Is this originate command correct? http://10.141.1.137:8080/webapi/originate?{forked_dial=false,ignore_early_media=true,origination_caller_id_name=4140637770,origination_caller_id_number=4140637770,originate_timeout=30}[group_confirm_key=5,group_confirm_file=playback/usr/share/sounds/alsa/Noise.wav]sofia/gateway/cs2k/141130252530 at 10.140.131.208%20&bridge({ignore_early_media=false,origination_caller_id_name=4140637770,origination_caller_id_number=4140637770,call_timeout=60}sofia/gateway/cs2k/141140636805 at 10.140.131.208) Is there an easy way to do it? Thanks in advance, Gustavo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090705/0472d112/attachment.html From codecomplete at free.fr Sun Jul 5 12:11:20 2009 From: codecomplete at free.fr (Fred-145) Date: Sun, 5 Jul 2009 12:11:20 -0700 (PDT) Subject: [Freeswitch-users] Cheapest, most compact FreeSwitch appliance? In-Reply-To: <7C0A0FFD-EC03-46DE-A277-AE793017D617@howlertech.com> References: <24337599.post@talk.nabble.com> <7C0A0FFD-EC03-46DE-A277-AE793017D617@howlertech.com> Message-ID: <24345928.post@talk.nabble.com> jfenton wrote: > If you look on there there's a clear acrylic case for the BeagleBoard > - I haven't seen any others available. That's often the problem with appliances: Once the CPU/modo, RAM, SDD/HD, PSU, case, and a PCI FXO board are computed, we end up with something as expensive as a regular PC. Is acrylic easy to work with, eg. using a Dremel? If it is, I could save money on the case, and stick a PCI card beneath the mobo and have a stand-alone, compact thingy while still using a standard Linux distro. -- View this message in context: http://www.nabble.com/Cheapest%2C-most-compact-FreeSwitch-appliance--tp24337599p24345928.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From tzury.by at reguluslabs.com Sun Jul 5 00:04:47 2009 From: tzury.by at reguluslabs.com (Tzury Bar Yochay) Date: Sun, 5 Jul 2009 10:04:47 +0300 Subject: [Freeswitch-users] Full Redundant Environment For FS In-Reply-To: <10128ef10907042256o7ff8887euf1046915aba3017d@mail.gmail.com> References: <10128ef10907042256o7ff8887euf1046915aba3017d@mail.gmail.com> Message-ID: <10128ef10907050004i721bb406r38e4659e698fe46e@mail.gmail.com> Hi all, I was wondering whether this is possible to achieve with FS of not. And if so what are the best practices for this. We wish to have our environment fully redundant, that is, live session should continue uninterrupted when an FS server goes down. Is there a document somewhere describe how to do it? From mgg at giagnocavo.net Sun Jul 5 14:06:48 2009 From: mgg at giagnocavo.net (Michael Giagnocavo) Date: Sun, 5 Jul 2009 17:06:48 -0400 Subject: [Freeswitch-users] Full Redundant Environment For FS In-Reply-To: <10128ef10907050004i721bb406r38e4659e698fe46e@mail.gmail.com> References: <10128ef10907042256o7ff8887euf1046915aba3017d@mail.gmail.com> <10128ef10907050004i721bb406r38e4659e698fe46e@mail.gmail.com> Message-ID: <6E8D2069C08AA84A83D336E996AE4C67027C3AD362@mse17be1.mse17.exchange.ms> In case of FS or server failure, this is not available yet. Search archives for more info, but in summary, it's a lot of work. However, you can do things like virtualization to have planned live migration. -Michael -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Tzury Bar Yochay Sent: Sunday, July 05, 2009 1:05 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Full Redundant Environment For FS Hi all, I was wondering whether this is possible to achieve with FS of not. And if so what are the best practices for this. We wish to have our environment fully redundant, that is, live session should continue uninterrupted when an FS server goes down. Is there a document somewhere describe how to do 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 geoffreymina at gmail.com Sun Jul 5 15:29:05 2009 From: geoffreymina at gmail.com (geoffreymina at gmail.com) Date: Sun, 05 Jul 2009 22:29:05 +0000 Subject: [Freeswitch-users] Contemplating a jump from Asterisk to Freeswitch Message-ID: <000e0cd4043ed3da65046dfcea6d@google.com> Hello, I have been reading through the on-line info as well as some reviews of the FreeSwitch platform. I am fairly convinced at this point that FreeSwitch is at least something I need to carefully look into. Our company utilizes asterisk to support our SaaS ACD/VPD/IVR platform. We currently support many thousands of concurrent agents (inbound and outbound). I have spent a lot of time trouble shooting bugs and working through 'issues' with asterisk. While I have tamed the beast, I am still not thrilled with the performance, nor am I very excited about the direction the project appears to be heading. It seems like every time a 'fix' is committed to SVN, it breaks something else. It's kind of like the wild-wild-west over there... and it certainly doesn't give me the warm/fuzzies when thinking about the future of my company. One of the benefits of our architecture is that our business logic is completely abstracted from the asterisk system. We use a combination of FastAGI and AMI to control channels on the asterisk server. We have a Java based server which interfaces with the higher level call routing engines. It looks to me like the Mod_event_socket would probably satisfy my requirements for controlling the calls via an external process, although it doesn't look as cut/dry as the FastAGI model. I haven't seen anything which would let me know the equivalent of the FastAGI 'script' being requested. The other thing I haven't seen is how to dynamically create conferences on the fly and redirect channels into them. We use app_conference on asterisk to avoid the ztdummy issue. Once the higher level intelligence engine determines two channels need to speak with each other, they are both redirected via AMI Redirect into a dynamic Conference created just for that particular call. Also - what is the status of call progress on FreeSwitch? Some things that are important to me are answering machine detection as well as detecting SIT intercept tones in the early media stream... any love here? I have a ton more questions, but this seems like a good start. Thanks! Geoff -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090705/de712507/attachment.html From edpimentl at gmail.com Sun Jul 5 16:17:16 2009 From: edpimentl at gmail.com (EdPimentl) Date: Sun, 5 Jul 2009 19:17:16 -0400 Subject: [Freeswitch-users] Cheapest, most compact FreeSwitch appliance? In-Reply-To: <24345928.post@talk.nabble.com> References: <24337599.post@talk.nabble.com> <7C0A0FFD-EC03-46DE-A277-AE793017D617@howlertech.com> <24345928.post@talk.nabble.com> Message-ID: <9dc4a1670907051617k3c112eeal2356d3df55283f89@mail.gmail.com> Back in April I posted these links on the list, in regards to a similar question http://www.logicsupply.com/products/dex4501 http://www.logicsupply.com/products/de945_fl http://en.wikipedia.org/wiki/NSLU2 http://en.wikipedia.org/wiki/Gumstix http://www.cheaprouter.us http://www.avm.de/en/Produkte/FRITZBox/FRITZ_Box_Fon_WLAN_7270/index.php http://www.plugcomputer.org/ http://forum.openwrt.org/viewtopic.php?pid=83701#p83701 http://www.pikatechnologies.com/ http://www.pikatechnologies.com/english/View.asp?x=608 http://fit-pc2.com http://www.cappuccinopc.com/solutions/fanless.asp http://www.wdlsystems.com/ebox/ebox.shtml http://www.linuxdevices.com/articles/AT2016997232.html -E http://Gpro.ws http://WatchNtweet.Me (Watch and Chat/Tweet) SocialTV http://TwebEX.com (Twitter Based Online Web Conference Platform) http://DatR.Ws (Cloud Computing Media Sharing, Access and Publish) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090705/a513cc8a/attachment.html From gcd at i.ph Sun Jul 5 16:21:03 2009 From: gcd at i.ph (Nandy Dagondon) Date: Mon, 6 Jul 2009 07:21:03 +0800 Subject: [Freeswitch-users] Compact, fanless appliance? In-Reply-To: References: <7d0bfd8c0907041941s75e541dex982fae2195858ea4@mail.gmail.com> Message-ID: <7d0bfd8c0907051621u1c3f553fh96a9df8557952e47@mail.gmail.com> ok. w/ my apologies. - nandy On Sun, Jul 5, 2009 at 10:49 AM, Ken Rice wrote: > No need to bump these things as this is a mailing list and it annoys > quite a few people when you do that > > > ------------------------------ > *From: *Nandy Dagondon > *Reply-To: * > *Date: *Sun, 5 Jul 2009 10:41:18 +0800 > *To: * > *Subject: *Re: [Freeswitch-users] Compact, fanless appliance? > > just bumping this topic. > -nandy > > On Fri, May 8, 2009 at 12:44 AM, Fred-145 wrote: > > > > Antonio Gallo wrote: > > Alix cases are like 6/9 ? from their shop site. I think its easy to find > > someone who work with aluminium that can make for you custom boxes for > > like like 6/20 ? at pcs > > Unfortunately, none of the PCEngines cases ( > www.pcengines.ch/order1.php?c=2 ) > allow for a PCI slot, either on top of the mobo, or away from it :-/ > > I'll see if I can get those from Soekris ( > http://soekris.eu/shop/cases_en/) > allow this, and if I can get a good price for a case + PSU. > > Thank you. > -- > View this message in context: > http://www.nabble.com/Compact%2C-fanless-appliance--tp23193738p23430873.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 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090706/d9824e10/attachment.html From dave at 3c.co.uk Sun Jul 5 16:22:49 2009 From: dave at 3c.co.uk (David Knell) Date: Sun, 05 Jul 2009 20:22:49 -0300 Subject: [Freeswitch-users] Contemplating a jump from Asterisk to Freeswitch In-Reply-To: <000e0cd4043ed3da65046dfcea6d@google.com> References: <000e0cd4043ed3da65046dfcea6d@google.com> Message-ID: <1246836169.31637.6.camel@dk-d820> Hi Geoff, > One of the benefits of our architecture is that our business logic is > completely abstracted from the asterisk system. We use a combination > of FastAGI and AMI to control channels on the asterisk server. We have > a Java based server which interfaces with the higher level call > routing engines. It looks to me like the Mod_event_socket would > probably satisfy my requirements for controlling the calls via an > external process, although it doesn't look as cut/dry as the FastAGI > model. I haven't seen anything which would let me know the equivalent > of the FastAGI 'script' being requested. Three possibilities spring to mind:- * have each distinct 'script' listen on a different socket; * set a variable in the dialplan to a script name or other identifier before making the outbound socket connection; * have your event socket handler work out what to do itself based on the dialled number, or whatever other criteria you'd use. > The other thing I haven't seen is how to dynamically create > conferences on the fly and redirect channels into them. We use > app_conference on asterisk to avoid the ztdummy issue. Once the higher > level intelligence engine determines two channels need to speak with > each other, they are both redirected via AMI Redirect into a dynamic > Conference created just for that particular call. Choose a (unique) conference ID, and execute conference on each of the channels. > Also - what is the status of call progress on FreeSwitch? Some things > that are important to me are answering machine detection as well as > detecting SIT intercept tones in the early media stream... any love > here? Not sure on these, but I'm *am* sure that someone else will be ;-) Cheers -- Dave -- David Knell, Director, 3C Limited T: +44 20 3298 2000 E: dave at 3c.co.uk W: http://www.3c.co.uk From codecomplete at free.fr Sun Jul 5 18:22:26 2009 From: codecomplete at free.fr (Fred-145) Date: Sun, 5 Jul 2009 18:22:26 -0700 (PDT) Subject: [Freeswitch-users] Cheapest, most compact FreeSwitch appliance? In-Reply-To: <9dc4a1670907051617k3c112eeal2356d3df55283f89@mail.gmail.com> References: <24337599.post@talk.nabble.com> <7C0A0FFD-EC03-46DE-A277-AE793017D617@howlertech.com> <24345928.post@talk.nabble.com> <9dc4a1670907051617k3c112eeal2356d3df55283f89@mail.gmail.com> Message-ID: <24348506.post@talk.nabble.com> EdPimentl wrote: > Back in April I posted these links on the list, in regards to a similar > question Thanks Ed, but the problem with all those, is: - they typically have so little RAM/Flash RAM that they can't run a regular Linux distro, which means that we're stuck with whatever software is available with the customized distro for the appliance - they don't have room for a PCI card, which means that we have to have an external VoIP gateway to connect the appliance to the POTS - they're as expensive or more expensive than a regular PC At this point, there doesn't seem to be any appliance that can run FreeSwitch and handle a POTS line in a compact, sub-$200 price-range. -- View this message in context: http://www.nabble.com/Cheapest%2C-most-compact-FreeSwitch-appliance--tp24337599p24348506.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From kristian.kielhofner at gmail.com Sun Jul 5 18:28:50 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Sun, 5 Jul 2009 21:28:50 -0400 Subject: [Freeswitch-users] Contemplating a jump from Asterisk to Freeswitch In-Reply-To: <000e0cd4043ed3da65046dfcea6d@google.com> References: <000e0cd4043ed3da65046dfcea6d@google.com> Message-ID: <2d9149cd0907051828g66360c5r52c63bca41aaff21@mail.gmail.com> On Sun, Jul 5, 2009 at 6:29 PM, wrote: > > Also - what is the status of call progress on FreeSwitch? Some things that > are important to me are answering machine detection as well as detecting SIT > intercept tones in the early media stream... any love here? > Not my specialty but I'll try... Answering machine detection can be done with mod_vmd: http://wiki.freeswitch.org/wiki/Mod_vmd Tone detection: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_tone_detect -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From msc at freeswitch.org Sun Jul 5 22:43:55 2009 From: msc at freeswitch.org (Michael S Collins) Date: Sun, 5 Jul 2009 22:43:55 -0700 Subject: [Freeswitch-users] Contemplating a jump from Asterisk to Freeswitch In-Reply-To: <000e0cd4043ed3da65046dfcea6d@google.com> References: <000e0cd4043ed3da65046dfcea6d@google.com> Message-ID: A few questions for you if I may: FreeSWITCH doesn't yet have a GUI -are you okay with XML config files? Do you have TDM circuits for your outbound traffic or are you using a SIP provider? BTW, mod_vmd is used to detect an answering machine beep, but it does not detect human vs. machine. For that you'll need mod_amd which isn't free but is available at a reasonable price. (email consulting at FreeSWITCH.org ) FYI, detecting SIT tones is always a challenge if you telco forces you to listen inband. You'll need a little processing power and the tone_detect app. I've done it on a PRI and cheap Tormenta 2 clone and it actually works pretty well. -MC Sent from my iPhone On Jul 5, 2009, at 3:29 PM, geoffreymina at gmail.com wrote: > Hello, > I have been reading through the on-line info as well as some reviews > of the FreeSwitch platform. I am fairly convinced at this point that > FreeSwitch is at least something I need to carefully look into. > > Our company utilizes asterisk to support our SaaS ACD/VPD/IVR > platform. We currently support many thousands of concurrent agents > (inbound and outbound). I have spent a lot of time trouble shooting > bugs and working through 'issues' with asterisk. While I have tamed > the beast, I am still not thrilled with the performance, nor am I > very excited about the direction the project appears to be heading. > It seems like every time a 'fix' is committed to SVN, it breaks > something else. It's kind of like the wild-wild-west over there... > and it certainly doesn't give me the warm/fuzzies when thinking > about the future of my company. > > One of the benefits of our architecture is that our business logic > is completely abstracted from the asterisk system. We use a > combination of FastAGI and AMI to control channels on the asterisk > server. We have a Java based server which interfaces with the higher > level call routing engines. It looks to me like the Mod_event_socket > would probably satisfy my requirements for controlling the calls via > an external process, although it doesn't look as cut/dry as the > FastAGI model. I haven't seen anything which would let me know the > equivalent of the FastAGI 'script' being requested. > > The other thing I haven't seen is how to dynamically create > conferences on the fly and redirect channels into them. We use > app_conference on asterisk to avoid the ztdummy issue. Once the > higher level intelligence engine determines two channels need to > speak with each other, they are both redirected via AMI Redirect > into a dynamic Conference created just for that particular call. > > Also - what is the status of call progress on FreeSwitch? Some > things that are important to me are answering machine detection as > well as detecting SIT intercept tones in the early media stream... > any love here? > > I have a ton more questions, but this seems like a good start. > > Thanks! > Geoff > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From talk2ram at gmail.com Mon Jul 6 02:35:13 2009 From: talk2ram at gmail.com (ram) Date: Mon, 6 Jul 2009 15:05:13 +0530 Subject: [Freeswitch-users] Real time Integration with Opensips Message-ID: Hi I am using Opensips as registrar and proxy * boxes as PSTN and VoIP Termination I would like to try replacing the * boxes with FS box in my lab so how can i make realtime Users integration with FS ( i use to do with Views in Mysql for Opensips and * boxes) Any URL and example available success people . just suggest me. Ram -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090706/0ade903b/attachment.html From jaybinks at gmail.com Mon Jul 6 03:23:51 2009 From: jaybinks at gmail.com (Jay Binks) Date: Mon, 06 Jul 2009 20:23:51 +1000 Subject: [Freeswitch-users] Real time Integration with Opensips In-Reply-To: References: Message-ID: <1246875831.5330.48.camel@jay-desktop.home.gateway> sounds like the simplest way would be to use a web application ( PHP or something similar ) that handles the users Directory.. that way you can keep your DB exactly the same and just pull the required fields. Jay On Mon, 2009-07-06 at 15:05 +0530, ram wrote: > Hi > > I am using Opensips as registrar and proxy > * boxes as PSTN and VoIP Termination > > I would like to try replacing the * boxes with FS box in my lab > > so how can i make realtime Users integration with FS ( i use to do > with Views in Mysql for Opensips and * boxes) > > Any URL and example available > > success people . just suggest me. > > Ram > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090706/87f3b701/attachment.html From talk2ram at gmail.com Mon Jul 6 04:03:46 2009 From: talk2ram at gmail.com (ram) Date: Mon, 6 Jul 2009 16:33:46 +0530 Subject: [Freeswitch-users] Real time Integration with Opensips In-Reply-To: <1246875831.5330.48.camel@jay-desktop.home.gateway> References: <1246875831.5330.48.camel@jay-desktop.home.gateway> Message-ID: On Mon, Jul 6, 2009 at 3:53 PM, Jay Binks wrote: > sounds like the simplest way would be to use a web application ( PHP or > something similar ) > that handles the users Directory.. that way you can keep your DB exactly > the same and just pull the required fields. > thanks for quick reply but if it grows 10K+ users maintaining folders will be tough. can some one point me mysql Internal and external schema for mysql iam trying to search not found Ram -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090706/3572d052/attachment.html From jaybinks at gmail.com Mon Jul 6 04:28:51 2009 From: jaybinks at gmail.com (Jay Binks) Date: Mon, 06 Jul 2009 21:28:51 +1000 Subject: [Freeswitch-users] Real time Integration with Opensips In-Reply-To: References: <1246875831.5330.48.camel@jay-desktop.home.gateway> Message-ID: <1246879731.5330.51.camel@jay-desktop.home.gateway> hmmm ok... was concerned my terminology may confuse. in Freeswitch sip_users are stored in a "User Directory" ( nothing to do with the filesystem ) it is a directory of all users ... ( like yellow pages is a directory .. ) look here .. http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide Freeswitch can get this user directory ( list of users ) in many ways.. one way is using CURL, to ask a web server to generate an XML file . this XML file can be created with PHP from your existing DB Structure this is what I was suggesting.. another way is to look at using OBDC to connect to mysql, however Im not a fan of OBDC. Jay On Mon, 2009-07-06 at 16:33 +0530, ram wrote: > > > > On Mon, Jul 6, 2009 at 3:53 PM, Jay Binks wrote: > > sounds like the simplest way would be to use a web application > ( PHP or something similar ) > that handles the users Directory.. that way you can keep your > DB exactly the same and just pull the required fields. > > thanks for quick reply > > but if it grows 10K+ users maintaining folders will be tough. > > can some one point me mysql Internal and external schema for mysql > > iam trying to search not found > > Ram > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090706/af57855a/attachment.html From geoffreymina at gmail.com Mon Jul 6 05:14:36 2009 From: geoffreymina at gmail.com (Geoffrey Mina) Date: Mon, 6 Jul 2009 08:14:36 -0400 Subject: [Freeswitch-users] Contemplating a jump from Asterisk to Freeswitch In-Reply-To: References: <000e0cd4043ed3da65046dfcea6d@google.com> Message-ID: <9f9c89f50907060514l417e1497x4f81a534c4306d13@mail.gmail.com> I love the fact that there is no GUI. I have never used any GUI for asterisk, so that is certainly not a problem. XML is fine with me. We are a pure VoIP environment. I have many wholesale SIP providers whom I interface with. AMD and SIT detection are very important to me. Because of that, I am exploring a relationship with Sangoma for their SIP based CPD product to satisfy those requirements. There are a couple things which I don't like... namely that it only runs on windows, but I may be able to ignore that for the time being. thanks. On Mon, Jul 6, 2009 at 1:43 AM, Michael S Collins wrote: > A few questions for you if I may: > FreeSWITCH doesn't yet have a GUI -are you okay with XML config files? > > Do you have TDM circuits for your outbound traffic or are you using a > SIP provider? > > BTW, mod_vmd is used to detect an answering machine beep, but it does > not detect human vs. machine. For that you'll need mod_amd which isn't > free but is available at a reasonable price. (email consulting at FreeSWITCH.org > ) > > FYI, detecting SIT tones is always a challenge if you telco forces you > to listen inband. You'll need a little processing power and the > tone_detect app. I've done it on a PRI and cheap Tormenta 2 clone and > it actually works pretty well. > > -MC > > Sent from my iPhone > > On Jul 5, 2009, at 3:29 PM, geoffreymina at gmail.com wrote: > >> Hello, >> I have been reading through the on-line info as well as some reviews >> of the FreeSwitch platform. I am fairly convinced at this point that >> FreeSwitch is at least something I need to carefully look into. >> >> Our company utilizes asterisk to support our SaaS ACD/VPD/IVR >> platform. We currently support many thousands of concurrent agents >> (inbound and outbound). I have spent a lot of time trouble shooting >> bugs and working through 'issues' with asterisk. While I have tamed >> the beast, I am still not thrilled with the performance, nor am I >> very excited about the direction the project appears to be heading. >> It seems like every time a 'fix' is committed to SVN, it breaks >> something else. It's kind of like the wild-wild-west over there... >> and it certainly doesn't give me the warm/fuzzies when thinking >> about the future of my company. >> >> One of the benefits of our architecture is that our business logic >> is completely abstracted from the asterisk system. We use a >> combination of FastAGI and AMI to control channels on the asterisk >> server. We have a Java based server which interfaces with the higher >> level call routing engines. It looks to me like the Mod_event_socket >> would probably satisfy my requirements for controlling the calls via >> an external process, although it doesn't look as cut/dry as the >> FastAGI model. I haven't seen anything which would let me know the >> equivalent of the FastAGI 'script' being requested. >> >> The other thing I haven't seen is how to dynamically create >> conferences on the fly and redirect channels into them. We use >> app_conference on asterisk to avoid the ztdummy issue. ?Once the >> higher level intelligence engine determines two channels need to >> speak with each other, they are both redirected via AMI Redirect >> into a dynamic Conference created just for that particular call. >> >> Also - what is the status of call progress on FreeSwitch? Some >> things that are important to me are answering machine detection as >> well as detecting SIT intercept tones in the early media stream... >> any love here? >> >> I have a ton more questions, but this seems like a good start. >> >> Thanks! >> Geoff >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From david.goodenough at linkchoose.co.uk Mon Jul 6 02:23:21 2009 From: david.goodenough at linkchoose.co.uk (David Goodenough) Date: Mon, 6 Jul 2009 10:23:21 +0100 Subject: [Freeswitch-users] Cheapest, most compact FreeSwitch appliance? In-Reply-To: <24348506.post@talk.nabble.com> References: <24337599.post@talk.nabble.com> <9dc4a1670907051617k3c112eeal2356d3df55283f89@mail.gmail.com> <24348506.post@talk.nabble.com> Message-ID: <200907061023.21388.david.goodenough@linkchoose.co.uk> On Monday 06 July 2009, Fred-145 wrote: > EdPimentl wrote: > > Back in April I posted these links on the list, in regards to a similar > > question > > Thanks Ed, but the problem with all those, is: > - they typically have so little RAM/Flash RAM that they can't run a regular > Linux distro, which means that we're stuck with whatever software is > available with the customized distro for the appliance > - they don't have room for a PCI card, which means that we have to have an > external VoIP gateway to connect the appliance to the POTS > - they're as expensive or more expensive than a regular PC > > At this point, there doesn't seem to be any appliance that can run > FreeSwitch and handle a POTS line in a compact, sub-$200 price-range. I though that I had read somewhere about someone using the Marvel ShevaPlug and a 2 line USB POTS adapter. That should be under $200. David From anthony.minessale at gmail.com Mon Jul 6 08:43:57 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 6 Jul 2009 10:43:57 -0500 Subject: [Freeswitch-users] Contemplating a jump from Asterisk to Freeswitch In-Reply-To: <000e0cd4043ed3da65046dfcea6d@google.com> References: <000e0cd4043ed3da65046dfcea6d@google.com> Message-ID: <191c3a030907060843k30bd1b79q5dcf3056a71b9890@mail.gmail.com> The best way to describe event socket to someone familiar with asterisk is that its a combination of AGI and AMI which can be used bidirectional. You can: connect one inbound socket from a client and control every call at once using events. connect one inbound socket then latch on to an existing single call and control it. connect one outbound socket to your application per call and control it. In all cases you have the option for full control which allows you to gain access to log, event, and FSAPI commands (the equiv of cli commands in asterisk) You can have your script listen on a dedicated port or use the ivrd example which is a daemon written in C that gets the desired script name from a channel variable and executes it on the remote end of the socket using STDIN/STDOUT as the socket. The other big difference besides that the single protocol does all these things is that we have a BSD licensed client library in our source tree called ESL. its in the libs/esl directory. This can be use to write clients in C or several other higher level languages using swig. fs_cli that is built with FS is written using ESL. Perl, Ruby, Python, Lua, PHP are all working and there is the beginning of a JAVA one which is stubbed out but just needs a little bit of work to finish it off and you could have that too. On Sun, Jul 5, 2009 at 5:29 PM, wrote: > Hello, > I have been reading through the on-line info as well as some reviews of the > FreeSwitch platform. I am fairly convinced at this point that FreeSwitch is > at least something I need to carefully look into. > > Our company utilizes asterisk to support our SaaS ACD/VPD/IVR platform. We > currently support many thousands of concurrent agents (inbound and > outbound). I have spent a lot of time trouble shooting bugs and working > through 'issues' with asterisk. While I have tamed the beast, I am still not > thrilled with the performance, nor am I very excited about the direction the > project appears to be heading. It seems like every time a 'fix' is committed > to SVN, it breaks something else. It's kind of like the wild-wild-west over > there... and it certainly doesn't give me the warm/fuzzies when thinking > about the future of my company. > > One of the benefits of our architecture is that our business logic is > completely abstracted from the asterisk system. We use a combination of > FastAGI and AMI to control channels on the asterisk server. We have a Java > based server which interfaces with the higher level call routing engines. It > looks to me like the Mod_event_socket would probably satisfy my requirements > for controlling the calls via an external process, although it doesn't look > as cut/dry as the FastAGI model. I haven't seen anything which would let me > know the equivalent of the FastAGI 'script' being requested. > > The other thing I haven't seen is how to dynamically create conferences on > the fly and redirect channels into them. We use app_conference on asterisk > to avoid the ztdummy issue. Once the higher level intelligence engine > determines two channels need to speak with each other, they are both > redirected via AMI Redirect into a dynamic Conference created just for that > particular call. > > Also - what is the status of call progress on FreeSwitch? Some things that > are important to me are answering machine detection as well as detecting SIT > intercept tones in the early media stream... any love here? > > I have a ton more questions, but this seems like a good start. > > Thanks! > Geoff > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference 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/20090706/c9aacd0c/attachment-0001.html From lordwizard007 at gmail.com Mon Jul 6 09:25:30 2009 From: lordwizard007 at gmail.com (lw) Date: Mon, 6 Jul 2009 12:25:30 -0400 Subject: [Freeswitch-users] Baby Update! In-Reply-To: <86a32abc0907031746x61996750k2a423d3ef0db70da@mail.gmail.com> References: <3EC5EF44-BF81-4C23-AD0B-B4AAB5F7DFCE@freeswitch.org> <1246629538.4185.9.camel@dk-d820> <41646A56-C830-489D-99D7-CE5C1218E194@freeswitch.org> <7b197bef0907031627o4e061c4by745a4492f2de0608@mail.gmail.com> <23FF9919-E96F-4067-832F-6E19A83A4F1F@freeswitch.org> <86a32abc0907031746x61996750k2a423d3ef0db70da@mail.gmail.com> Message-ID: Congrats! On Fri, Jul 3, 2009 at 8:46 PM, Diego Viola wrote: > Congrats! > > > On Fri, Jul 3, 2009 at 7:35 PM, Brian West wrote: > >> Remember send him a little something to help out with the last minute >> expenses! ;) >> >> Btw lanboy will be at ClueCon ;) As will lanwife! >> >> /b >> >> On Jul 3, 2009, at 6:27 PM, Giovanni Maruzzelli wrote: >> >> > Yeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeh! >> > >> > >> > On Fri, Jul 3, 2009 at 10:31 PM, Brian West >> > wrote: >> >> Kaiden Anthony Chandler arrived Friday July 3rd at 1411 EDT 7lbs 10oz >> >> YAY... Congrats mr Lanman! >> >> /b >> >> On Jul 3, 2009, at 8:58 AM, David Knell wrote: >> >> >> >> On Fri, 2009-07-03 at 17:43 +0600, Muhammad Shahzad wrote: >> >> >> >> Congratulations to Ray and Samantha. Lets see what new features and >> >> >> >> bug fixes we will get in their "new version"..! ;-) >> >> >> >> Bug fixes..?! I'd refer you to Philip Larkin (went to my school, a >> >> bit >> >> before my time, poet, deceased, recently voted "Britain's favourite >> >> poet") whose "This Be The Verse" suggests otherwise: >> >> http://www.artofeurope.com/larkin/lar2.htm >> >> >> >> [as a recent father myself, I'm trying to prove him wrong..] >> >> >> >> --Dave >> >> >> >> -- >> >> David Knell, Director, 3C Limited >> >> T: +44 20 3298 2000 >> >> E: dave at 3c.co.uk >> >> W: 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 >> >> >> >> >> >> _______________________________________________ >> >> Freeswitch-users mailing list >> >> Freeswitch-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> >> >> >> >> > >> > _______________________________________________ >> > Freeswitch-users mailing list >> > Freeswitch-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090706/30884091/attachment.html From jgarland at gmail.com Mon Jul 6 12:45:39 2009 From: jgarland at gmail.com (Jason Garland) Date: Mon, 06 Jul 2009 15:45:39 -0400 Subject: [Freeswitch-users] Cluecon 2009 hotel deals get a free $50 prepaid mastercard, and $10 off each night with a coupon code Message-ID: <4A525463.6000109@gmail.com> Use the following link to get a free $50 prepaid mastercard via Expedia: Expedia July Sale: Book a qualified 3+ night hotel stay & get a $50 Prepaid MasterCard? card for gas! - Expires 7/31/09 Then enter coupon code: 09JUL10 I think this coupon code expires on July 10th. If anyone manages to find some better coupon codes please post them. This is the best I could find. Expedia seems to have the best rates, and they count towards the minimum hotel room requirements imposed by the hotel for the fine folks running Cluecon. My total price after all the taxes and discounts for 3 nights ended up at $511.20 And that doesn't include the $50 prepaid mastercard. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090706/40dc032d/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image-3370729-10655845 Type: image/gif Size: 50 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090706/40dc032d/attachment.gif From ronmccar at gmail.com Mon Jul 6 15:53:04 2009 From: ronmccar at gmail.com (Ron McCarthy) Date: Mon, 6 Jul 2009 15:53:04 -0700 Subject: [Freeswitch-users] Best OS for FreeSwitch? Message-ID: <3885f4fe0907061553s39eb723as4782bf87097a041@mail.gmail.com> Hey list, We are running FS on FreeBSD 7.2 right now and cannot get it to push over 48 CPS on a Dual core Xeon (2.4 ghz), we start running into issues, max concurrent calls is around 500 as well. Asterisk can do this so I would FS could out perform this! We run into major PDD issues more then anything, where FS takes 10+ seconds to respond to a invite, it's very weird and bad. I have seen some issues with FreeBSD and FS so id like to try a different OS and see what our results are. What does everyone recommend for just raw performance and speed? I use to always run Slackware until we moved everything over to FreeBSD, would that be a good choice again? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090706/da2c4b12/attachment.html From brian at freeswitch.org Mon Jul 6 16:03:25 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 6 Jul 2009 18:03:25 -0500 Subject: [Freeswitch-users] Best OS for FreeSwitch? In-Reply-To: <3885f4fe0907061553s39eb723as4782bf87097a041@mail.gmail.com> References: <3885f4fe0907061553s39eb723as4782bf87097a041@mail.gmail.com> Message-ID: CentOS 5.3 /b On Jul 6, 2009, at 5:53 PM, Ron McCarthy wrote: > Hey list, > > We are running FS on FreeBSD 7.2 right now and cannot get it to push > over 48 CPS on a Dual core Xeon (2.4 ghz), we start running into > issues, max concurrent calls is around 500 as well. Asterisk can do > this so I would FS could out perform this! We run into major PDD > issues more then anything, where FS takes 10+ seconds to respond to > a invite, it's very weird and bad. > > I have seen some issues with FreeBSD and FS so id like to try a > different OS and see what our results are. What does everyone > recommend for just raw performance and speed? I use to always run > Slackware until we moved everything over to FreeBSD, would that be a > good choice again? > > Thanks From ronmccar at gmail.com Mon Jul 6 16:04:20 2009 From: ronmccar at gmail.com (Ron McCarthy) Date: Mon, 6 Jul 2009 16:04:20 -0700 Subject: [Freeswitch-users] Best OS for FreeSwitch? In-Reply-To: References: <3885f4fe0907061553s39eb723as4782bf87097a041@mail.gmail.com> Message-ID: <3885f4fe0907061604s154e71a9jb06ce5a4d970f8c@mail.gmail.com> 64bit is better as well correct? On Mon, Jul 6, 2009 at 4:03 PM, Brian West wrote: > CentOS 5.3 > > /b > > On Jul 6, 2009, at 5:53 PM, Ron McCarthy wrote: > > > Hey list, > > > > We are running FS on FreeBSD 7.2 right now and cannot get it to push > > over 48 CPS on a Dual core Xeon (2.4 ghz), we start running into > > issues, max concurrent calls is around 500 as well. Asterisk can do > > this so I would FS could out perform this! We run into major PDD > > issues more then anything, where FS takes 10+ seconds to respond to > > a invite, it's very weird and bad. > > > > I have seen some issues with FreeBSD and FS so id like to try a > > different OS and see what our results are. What does everyone > > recommend for just raw performance and speed? I use to always run > > Slackware until we moved everything over to FreeBSD, would that be a > > good choice again? > > > > 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/20090706/a219fd90/attachment.html From brian at freeswitch.org Mon Jul 6 16:06:43 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 6 Jul 2009 18:06:43 -0500 Subject: [Freeswitch-users] Best OS for FreeSwitch? In-Reply-To: <3885f4fe0907061604s154e71a9jb06ce5a4d970f8c@mail.gmail.com> References: <3885f4fe0907061553s39eb723as4782bf87097a041@mail.gmail.com> <3885f4fe0907061604s154e71a9jb06ce5a4d970f8c@mail.gmail.com> Message-ID: Yes, 32bit can only serve as a boat anchor in my opinion.... also don't run a 32bit OS on a 64bit CPU, you might as well just paypal me half the money you spent on the CPU's in the first place and snagged a couple of P4's :P /b On Jul 6, 2009, at 6:04 PM, Ron McCarthy wrote: > 64bit is better as well correct? > > > On Mon, Jul 6, 2009 at 4:03 PM, Brian West > wrote: > CentOS 5.3 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090706/b3f940a7/attachment-0001.html From hyppolite72 at yahoo.com Mon Jul 6 16:20:05 2009 From: hyppolite72 at yahoo.com (Jean-Marc Hyppolite) Date: Mon, 6 Jul 2009 16:20:05 -0700 (PDT) Subject: [Freeswitch-users] Controlling MOH from a java application Message-ID: <219371.37062.qm@web35605.mail.mud.yahoo.com> Hello, ? First of all, I would like to thank Anthony, Brian and all the developers for this wonderful piece of software. Very good job. ? I would like to know how I can start and stop Music On Hold from a JAVA script (using mod_java) similar to the StartMusicOnHold and StopMusicHold functions found in AGI (Asterisk-Java). ? I am using FreeSWITCH as an IVR server. I would like to be able to put the caller on hold while doing some other stuff. ? Thanks in advance. ? Jean-Marc. __________________________________________________________________ Yahoo! Canada Toolbar: Search from anywhere on the web, and bookmark your favourite sites. Download it now http://ca.toolbar.yahoo.com. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090706/27778bcd/attachment.html From max.bridgewater at gmail.com Mon Jul 6 16:24:06 2009 From: max.bridgewater at gmail.com (Max Bridgewater) Date: Mon, 6 Jul 2009 19:24:06 -0400 Subject: [Freeswitch-users] Start Freeswitch as Daemon on CentOS 5 Message-ID: Hi, I have a server that i log into using SSH. Then in my local SSH terminal, i start Freeswitch with: /usr/local/freeswitch/bin/freeswitch -nonat & Yet, when i close the terminal window, Freeswitch also dies. I was hoping that the ampersand would make it run as a dameon process that would live pass the lifetime of the terminal. Any trick? Thanks, Max. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090706/679d400f/attachment.html From brian at freeswitch.org Mon Jul 6 16:27:22 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 6 Jul 2009 18:27:22 -0500 Subject: [Freeswitch-users] Controlling MOH from a java application In-Reply-To: <219371.37062.qm@web35605.mail.mud.yahoo.com> References: <219371.37062.qm@web35605.mail.mud.yahoo.com> Message-ID: uuid_hold uuid_hold off These two api's will do it. /b On Jul 6, 2009, at 6:20 PM, Jean-Marc Hyppolite wrote: > Hello, > > First of all, I would like to thank Anthony, Brian and all the > developers for this wonderful piece of software. Very good job. > > I would like to know how I can start and stop Music On Hold from a > JAVA script (using mod_java) similar to the StartMusicOnHold and > StopMusicHold functions found in AGI (Asterisk-Java). > > I am using FreeSWITCH as an IVR server. I would like to be able to > put the caller on hold while doing some other stuff. > > Thanks in advance. > > Jean-Marc. > > > Yahoo! Canada Toolbar : Search from anywhere on the web and > bookmark your favourite sites. Download it now! > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090706/3fc83eb3/attachment.html From brian at freeswitch.org Mon Jul 6 16:27:39 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 6 Jul 2009 18:27:39 -0500 Subject: [Freeswitch-users] Start Freeswitch as Daemon on CentOS 5 In-Reply-To: References: Message-ID: add a -nc in there too. > /usr/local/freeswitch/bin/freeswitch -nonat -nc & /b On Jul 6, 2009, at 6:24 PM, Max Bridgewater wrote: > Hi, > > I have a server that i log into using SSH. Then in my local SSH > terminal, i start Freeswitch with: > > /usr/local/freeswitch/bin/freeswitch -nonat & > > > Yet, when i close the terminal window, Freeswitch also dies. I was > hoping that the ampersand would make it run as a dameon process that > would live pass the lifetime of the terminal. > > Any trick? > > Thanks, > > Max. From sicfslist at gmail.com Mon Jul 6 16:30:03 2009 From: sicfslist at gmail.com (Shelby Ramsey) Date: Mon, 6 Jul 2009 18:30:03 -0500 Subject: [Freeswitch-users] Start Freeswitch as Daemon on CentOS 5 In-Reply-To: References: Message-ID: <35b355e90907061630l4b4c738codd12a998bbb39333@mail.gmail.com> use the -nc flag ... that will do the trick -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090706/8622502f/attachment.html From max.bridgewater at gmail.com Mon Jul 6 16:31:28 2009 From: max.bridgewater at gmail.com (Max Bridgewater) Date: Mon, 6 Jul 2009 19:31:28 -0400 Subject: [Freeswitch-users] Start Freeswitch as Daemon on CentOS 5 In-Reply-To: References: Message-ID: Cool! Thanks. On Mon, Jul 6, 2009 at 7:27 PM, Brian West wrote: > add a -nc in there too. > > > /usr/local/freeswitch/bin/freeswitch -nonat -nc & > > > /b > > On Jul 6, 2009, at 6:24 PM, Max Bridgewater wrote: > > > Hi, > > > > I have a server that i log into using SSH. Then in my local SSH > > terminal, i start Freeswitch with: > > > > /usr/local/freeswitch/bin/freeswitch -nonat & > > > > > > Yet, when i close the terminal window, Freeswitch also dies. I was > > hoping that the ampersand would make it run as a dameon process that > > would live pass the lifetime of the terminal. > > > > Any trick? > > > > Thanks, > > > > Max. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090706/f2625f06/attachment.html From jens at vegeby.nu Mon Jul 6 16:35:24 2009 From: jens at vegeby.nu (Jens Vegeby) Date: Tue, 7 Jul 2009 01:35:24 +0200 Subject: [Freeswitch-users] Start Freeswitch as Daemon on CentOS 5 In-Reply-To: References: Message-ID: <30ee97110907061635pdb8978bvb3f3effb114c7fc8@mail.gmail.com> Use the -nc (no console) command line parameter. There is a centos init script somewhere in the svn source tree. /Jens On 7/7/09, Max Bridgewater wrote: > Hi, > > I have a server that i log into using SSH. Then in my local SSH terminal, i > start Freeswitch with: > > /usr/local/freeswitch/bin/freeswitch -nonat & > > > Yet, when i close the terminal window, Freeswitch also dies. I was hoping > that the ampersand would make it run as a dameon process that would live > pass the lifetime of the terminal. > > Any trick? > > Thanks, > > Max. > -- Sent from my mobile device Mvh/Regards Jens From brian at freeswitch.org Mon Jul 6 16:38:15 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 6 Jul 2009 18:38:15 -0500 Subject: [Freeswitch-users] Start Freeswitch as Daemon on CentOS 5 In-Reply-To: <30ee97110907061635pdb8978bvb3f3effb114c7fc8@mail.gmail.com> References: <30ee97110907061635pdb8978bvb3f3effb114c7fc8@mail.gmail.com> Message-ID: <54E2126D-86BD-49D6-8C18-D34F8F8D60E1@freeswitch.org> build/freeswitch.init.redhat /b On Jul 6, 2009, at 6:35 PM, Jens Vegeby wrote: > Use the -nc (no console) command line parameter. > > There is a centos init script somewhere in the svn source tree. > > /Jens > > On 7/7/09, Max Bridgewater wrote: >> Hi, >> >> I have a server that i log into using SSH. Then in my local SSH >> terminal, i >> start Freeswitch with: >> >> /usr/local/freeswitch/bin/freeswitch -nonat & >> >> >> Yet, when i close the terminal window, Freeswitch also dies. I was >> hoping >> that the ampersand would make it run as a dameon process that would >> live >> pass the lifetime of the terminal. >> >> Any trick? >> >> Thanks, >> >> Max. >> > > -- > Sent from my mobile device > > Mvh/Regards Jens > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brad.tuan at gmail.com Mon Jul 6 19:01:49 2009 From: brad.tuan at gmail.com (Brad Tuan) Date: Tue, 7 Jul 2009 10:01:49 +0800 Subject: [Freeswitch-users] How to modify my INVITE msg?? Message-ID: For example, send a "INVITE 1001123 at xxx.xxx.xxx.xxx" to my FS user 1001 How to do this?? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090707/94277ed9/attachment.html From brian at freeswitch.org Mon Jul 6 19:23:21 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 6 Jul 2009 21:23:21 -0500 Subject: [Freeswitch-users] How to modify my INVITE msg?? In-Reply-To: References: Message-ID: Try this: /b On Jul 6, 2009, at 9:01 PM, Brad Tuan wrote: > For example, send a "INVITE 1001123 at xxx.xxx.xxx.xxx" to my FS user > 1001 > > How to do this?? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090706/4ff987d1/attachment-0001.html From brad.tuan at gmail.com Mon Jul 6 19:48:04 2009 From: brad.tuan at gmail.com (Brad Tuan) Date: Tue, 7 Jul 2009 10:48:04 +0800 Subject: [Freeswitch-users] How to modify my INVITE msg?? Message-ID: Useless , the dialplan was changed like this: but when 1003 call 1001 ,the request is still "Request-Line: INVITE sip:1001 at 192.168.141.182 SIP/2.0" Dialplan: sofia/internal/1003 at 192.168.141.182 parsing [default->transfer_to_516003] continue=false Dialplan: sofia/internal/1003 at 192.168.141.182 Regex (FAIL) [transfer_to_516003] destination_number(1001) =~ /^516001$/ break=on-false Dialplan: sofia/internal/1003 at 192.168.141.182 parsing [default->transfer_to_516003] continue=false Dialplan: sofia/internal/1003 at 192.168.141.182 Regex (FAIL) [transfer_to_516003] destination_number(1001) =~ /^516002$/ break=on-false Dialplan: sofia/internal/1003 at 192.168.141.182 parsing [default->skype_to_1001] continue=false Dialplan: sofia/internal/1003 at 192.168.141.182 Regex (PASS) [skype_to_1001] destination_number(1001) =~ /^1001$/ break=on-false Dialplan: sofia/internal/1003 at 192.168.141.182 Action bridge(sofia/profile/1001123${regex(${sofia_contact( 1001@${domain})}|^[^\@]+(.*)|%1)}) 2009-07-07 10:38:46 [DEBUG] switch_core_state_machine.c:114 switch_core_standard_on_routing() (sofia/internal/1003 at 192.168.141.182) State Change CS_ROUTING -> CS_EXECUTE 2009-07-07 10:38:46 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/internal/1003 at 192.168.141.182 [BREAK] 2009-07-07 10:38:46 [DEBUG] switch_core_state_machine.c:483 switch_core_session_run() (sofia/internal/1003 at 192.168.141.182) State ROUTING going to sleep 2009-07-07 10:38:46 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/internal/1003 at 192.168.141.182) Running State Change CS_EXECUTE 2009-07-07 10:38:46 [DEBUG] switch_core_state_machine.c:490 switch_core_session_run() (sofia/internal/1003 at 192.168.141.182) State EXECUTE 2009-07-07 10:38:46 [DEBUG] mod_sofia.c:173 sofia_on_execute() sofia/internal/1003 at 192.168.141.182 SOFIA EXECUTE 2009-07-07 10:38:46 [DEBUG] switch_core_state_machine.c:151 switch_core_standard_on_execute() sofia/internal/1003 at 192.168.141.182Standard EXECUTE EXECUTE sofia/internal/1003 at 192.168.141.182 bridge( sofia/profile/1001123 at 192.168.141.182:29084;rinstance=b4a8ae8884b9ed6b) 2009-07-07 10:38:46 [ERR] mod_sofia.c:2681 sofia_outgoing_channel() Invalid Profile FS return it is a Invalid Profile....Why?? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090707/8775fcaf/attachment.html From brad.tuan at gmail.com Mon Jul 6 19:56:30 2009 From: brad.tuan at gmail.com (Brad Tuan) Date: Tue, 7 Jul 2009 10:56:30 +0800 Subject: [Freeswitch-users] How to modify my INVITE msg?? Message-ID: Useless , the dialplan was changed like this: but when 1003 call 1001 ,the request is still "Request-Line: INVITE sip:1001 at 192.168.141.182 SIP/2.0" Dialplan: sofia/internal/1003 at 192.168.141.182 parsing [default->transfer_to_516003] continue=false Dialplan: sofia/internal/1003 at 192.168.141.182 Regex (FAIL) [transfer_to_516003] destination_number(1001) =~ /^516001$/ break=on-false Dialplan: sofia/internal/1003 at 192.168.141.182 parsing [default->transfer_to_516003] continue=false Dialplan: sofia/internal/1003 at 192.168.141.182 Regex (FAIL) [transfer_to_516003] destination_number(1001) =~ /^516002$/ break=on-false Dialplan: sofia/internal/1003 at 192.168.141.182 parsing [default->skype_to_1001] continue=false Dialplan: sofia/internal/1003 at 192.168.141.182 Regex (PASS) [skype_to_1001] destination_number(1001) =~ /^1001$/ break=on-false Dialplan: sofia/internal/1003 at 192.168.141.182 Action bridge(sofia/profile/1001123${regex(${sofia_contact( 1001@${domain})}|^[^\@]+(.*)|%1)}) 2009-07-07 10:38:46 [DEBUG] switch_core_state_machine.c:114 switch_core_standard_on_routing() (sofia/internal/1003 at 192.168.141.182) State Change CS_ROUTING -> CS_EXECUTE 2009-07-07 10:38:46 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/internal/1003 at 192.168.141.182 [BREAK] 2009-07-07 10:38:46 [DEBUG] switch_core_state_machine.c:483 switch_core_session_run() (sofia/internal/1003 at 192.168.141.182) State ROUTING going to sleep 2009-07-07 10:38:46 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/internal/1003 at 192.168.141.182) Running State Change CS_EXECUTE 2009-07-07 10:38:46 [DEBUG] switch_core_state_machine.c:490 switch_core_session_run() (sofia/internal/1003 at 192.168.141.182) State EXECUTE 2009-07-07 10:38:46 [DEBUG] mod_sofia.c:173 sofia_on_execute() sofia/internal/1003 at 192.168.141.182 SOFIA EXECUTE 2009-07-07 10:38:46 [DEBUG] switch_core_state_machine.c:151 switch_core_standard_on_execute() sofia/internal/1003 at 192.168.141.182Standard EXECUTE EXECUTE sofia/internal/1003 at 192.168.141.182 bridge( sofia/profile/1001123 at 192.168.141.182:29084;rinstance=b4a8ae8884b9ed6b) 2009-07-07 10:38:46 [ERR] mod_sofia.c:2681 sofia_outgoing_channel() Invalid Profile FS return it is a Invalid Profile....Why?? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090707/66412b35/attachment.html From hyppolite72 at yahoo.com Mon Jul 6 20:21:31 2009 From: hyppolite72 at yahoo.com (Jean-Marc Hyppolite) Date: Mon, 6 Jul 2009 20:21:31 -0700 (PDT) Subject: [Freeswitch-users] Controlling MOH from a java application Message-ID: <443906.62890.qm@web35604.mail.mud.yahoo.com> Hello Brian, ? Thank you for your quick answer. I tried the two API functions but with no result. The caller is not able to hear any music. But, when I use two extensions (one calling the other), MOH does work. ? My code on the JAVA side (for test purposes) ? session.answer(); session.sleep(500); ? session.execute("eval", "uuid_hold " + session.get_uuid()); ? java_function(); // lasts 30 to 40 seconds ? session.execute("eval", "uuid_hold off " + session.get_uuid()); ? session.sleep(500); session.hangup(); ? Thanks for the help. ? Jean-Marc. ? ? ? --- On Mon, 7/6/09, Brian West wrote: From: Brian West Subject: Re: [Freeswitch-users] Controlling MOH from a java application To: freeswitch-users at lists.freeswitch.org Received: Monday, July 6, 2009, 7:27 PM uuid_hold uuid_hold off These two api's will do it. /b On Jul 6, 2009, at 6:20 PM, Jean-Marc Hyppolite wrote: Hello, ? First of all, I would like to thank Anthony, Brian and all the developers for this wonderful piece of software. Very good job. ? I would like to know how I can start and stop Music On Hold from a JAVA script (using mod_java) similar to the StartMusicOnHold and StopMusicHold functions found in AGI (Asterisk-Java). ? I am using FreeSWITCH as an IVR server. I would like to be able to put the caller on hold while doing some other stuff. ? Thanks in advance. ? Jean-Marc. Yahoo! Canada Toolbar : Search from anywhere on the web and bookmark your favourite sites. Download it now! _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -----Inline Attachment Follows----- _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org __________________________________________________________________ Yahoo! Canada Toolbar: Search from anywhere on the web, and bookmark your favourite sites. Download it now http://ca.toolbar.yahoo.com. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090706/fb0e8db9/attachment.html From elihayun at gmail.com Mon Jul 6 21:53:37 2009 From: elihayun at gmail.com (Eli Hayun) Date: Tue, 07 Jul 2009 07:53:37 +0300 Subject: [Freeswitch-users] How to get the hook state? Message-ID: <4A52D4D1.1010805@gmail.com> Hi I am a newbie in FreeSwitch and my question is: When I am calling to an extension, how should I know in advance what is the hook status. I tried to find out a variable that can get me this information but with no success. any help? From brad.tuan at gmail.com Mon Jul 6 23:15:27 2009 From: brad.tuan at gmail.com (Brad Tuan) Date: Tue, 7 Jul 2009 14:15:27 +0800 Subject: [Freeswitch-users] How to modify the Subject and Body when sending voicemail?? Message-ID: As title, How to custom the Subject and Body and ... of the mail ?? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090707/0af57d8a/attachment-0001.html From jason at jasonjgw.net Mon Jul 6 23:42:15 2009 From: jason at jasonjgw.net (Jason White) Date: Tue, 7 Jul 2009 16:42:15 +1000 Subject: [Freeswitch-users] How to modify the Subject and Body when sending voicemail?? In-Reply-To: References: Message-ID: <20090707064215.GA21128@jdc.jasonjgw.net> Brad Tuan wrote: > As title, How to custom the Subject and Body and ... of the mail ?? Have a look at the notify-voicemail.tpl and voicemail.tpl files, and the template parameters in voicemail.conf.xml in the default FreeSWITCH configuration to see how it all works and to decide what to edit. From dujinfang at gmail.com Tue Jul 7 00:06:16 2009 From: dujinfang at gmail.com (seven) Date: Tue, 7 Jul 2009 15:06:16 +0800 Subject: [Freeswitch-users] dp tools record to 1 channel but uuid_record record to 2 channels which cannot been playback Message-ID: <0DA560FF-D5F1-4E86-B8CD-6DBE0B9C9B47@gmail.com> # play test.wav Input File : 'test.wav' Sample Size : 16-bit (2 bytes) Sample Encoding: signed (2's complement) Channels : 2 Sample Rate : 16000 1) is it the default behavior that uuid_record record with 2 channels 2) is it reasonable that FS can record to 2 channels but cannot playback? 3) do I need to set RECORD_STEREO=false before uuid_record? Thanks for help. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090707/6d1a41cd/attachment.html From jason at jasonjgw.net Tue Jul 7 00:16:32 2009 From: jason at jasonjgw.net (Jason White) Date: Tue, 7 Jul 2009 17:16:32 +1000 Subject: [Freeswitch-users] dp tools record to 1 channel but uuid_record record to 2 channels which cannot been playback In-Reply-To: <0DA560FF-D5F1-4E86-B8CD-6DBE0B9C9B47@gmail.com> References: <0DA560FF-D5F1-4E86-B8CD-6DBE0B9C9B47@gmail.com> Message-ID: <20090707071632.GA22797@jdc.jasonjgw.net> seven wrote: > 1) is it the default behavior that uuid_record record with 2 channels Yes. > 2) is it reasonable that FS can record to 2 channels but cannot > playback? Could you explain what happens when you play back the files? > 3) do I need to set RECORD_STEREO=false before uuid_record? That depends on whether you want two-channel output files or not. From dujinfang at gmail.com Tue Jul 7 00:30:27 2009 From: dujinfang at gmail.com (seven) Date: Tue, 7 Jul 2009 15:30:27 +0800 Subject: [Freeswitch-users] dp tools record to 1 channel but uuid_record record to 2 channels which cannot been playback In-Reply-To: <20090707071632.GA22797@jdc.jasonjgw.net> References: <0DA560FF-D5F1-4E86-B8CD-6DBE0B9C9B47@gmail.com> <20090707071632.GA22797@jdc.jasonjgw.net> Message-ID: <05D41406-17C3-43DF-BA94-3753B191B7BC@gmail.com> > >> 2) is it reasonable that FS can record to 2 channels but cannot >> playback? > > Could you explain what happens when you play back the files? yes, it's here: http://pastebin.freeswitch.org/9641 > >> 3) do I need to set RECORD_STEREO=false before uuid_record? > > That depends on whether you want two-channel output files or not. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.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 Tue Jul 7 00:56:33 2009 From: jason at jasonjgw.net (Jason White) Date: Tue, 7 Jul 2009 17:56:33 +1000 Subject: [Freeswitch-users] dp tools record to 1 channel but uuid_record record to 2 channels which cannot been playback In-Reply-To: <05D41406-17C3-43DF-BA94-3753B191B7BC@gmail.com> References: <0DA560FF-D5F1-4E86-B8CD-6DBE0B9C9B47@gmail.com> <20090707071632.GA22797@jdc.jasonjgw.net> <05D41406-17C3-43DF-BA94-3753B191B7BC@gmail.com> Message-ID: <20090707075633.GA25395@jdc.jasonjgw.net> seven wrote: > yes, it's here: http://pastebin.freeswitch.org/9641 Judging by the error message, it's a known limitation. You are welcome to work on a fix, or pay the develoeprs to fix it, or offer a bounty that might encourage someone to work on it, or wait until it gets fixed. Meanwhile, convert the file to mono and try again. Sox should be able to do this, for example. From dujinfang at gmail.com Tue Jul 7 01:11:00 2009 From: dujinfang at gmail.com (seven) Date: Tue, 7 Jul 2009 16:11:00 +0800 Subject: [Freeswitch-users] dp tools record to 1 channel but uuid_record record to 2 channels which cannot been playback In-Reply-To: <20090707075633.GA25395@jdc.jasonjgw.net> References: <0DA560FF-D5F1-4E86-B8CD-6DBE0B9C9B47@gmail.com> <20090707071632.GA22797@jdc.jasonjgw.net> <05D41406-17C3-43DF-BA94-3753B191B7BC@gmail.com> <20090707075633.GA25395@jdc.jasonjgw.net> Message-ID: <2E6E3DB3-471A-4D87-8BD6-7FA61B3ED9F9@gmail.com> On Jul 7, 2009, at 3:56 PM, Jason White wrote: > seven wrote: > >> yes, it's here: http://pastebin.freeswitch.org/9641 > > Judging by the error message, it's a known limitation. You are > welcome to work > on a fix, or pay the develoeprs to fix it, or offer a bounty that > might > encourage someone to work on it, or wait until it gets fixed. > > Meanwhile, convert the file to mono and try again. > > Sox should be able to do this, for example. > > Thanks, I'm using sox. > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mariusz_kolo at wp.pl Tue Jul 7 03:35:02 2009 From: mariusz_kolo at wp.pl (=?ISO-8859-2?Q?Mariusz_Ko=B3odziejczyk?=) Date: Tue, 07 Jul 2009 12:35:02 +0200 Subject: [Freeswitch-users] Record_session cutting wav files Message-ID: <4A5324D6.3070600@wp.pl> Hello I saw a strange behavior when i'm using record_session for outbound call. Recorded file is 24:20 time length, but in logs should have about 25:18. Here ma log: start recording about 2009-07-07 10:56:15 2009-07-07 10:56:15.108447 [DEBUG] mod_dptools.c:748 sofia/internal/1062 SET [czas]=[2009-07-07-10-56-15] - variable $czas = "2009-07-07-10-56-15" i use it in filename below EXECUTE sofia/internal/1062 record_session(/records/2009-07-07/069630d2-6ad4-11de-aa6e-e31e1f9431c0.10XX.07135XXXXX.2009-07-07-10-56-15.out.ISDN.wav) ..... stop recording 2009-07-07 11:21:33.291177 [DEBUG] switch_ivr_async.c:444 Stop recording file /records/2009-07-07/069630d2-6ad4-11de-aa6e-e31e1f9431c0.10XX.07135XXXXX.2009-07-07-10-56-15.out.ISDN.wav file should have about: 25:18 time length when i listen a file it's really cut My piece of dialplan: freeswitch version: FreeSWITCH Version 1.0.trunk (14013) linux: Linux pbx2 2.6.26-1-686 #1 SMP Fri Mar 13 18:08:45 UTC 2009 i686 GNU/Linux Thanks From jalsot at gmail.com Tue Jul 7 03:48:20 2009 From: jalsot at gmail.com (Tamas) Date: Tue, 07 Jul 2009 12:48:20 +0200 Subject: [Freeswitch-users] Record_session cutting wav files In-Reply-To: <4A5324D6.3070600@wp.pl> References: <4A5324D6.3070600@wp.pl> Message-ID: <4A5327F4.1040206@gmail.com> Hello, please try out FS >= r14143 as there were some fixes around call recording and media bugs. Please let us know the results. Regards, Tamas Mariusz Ko?odziejczyk ?rta: > Hello > > I saw a strange behavior when i'm using record_session for outbound > call. Recorded file is 24:20 time length, but in logs should have about > 25:18. > > Here ma log: > > start recording about 2009-07-07 10:56:15 > 2009-07-07 10:56:15.108447 [DEBUG] mod_dptools.c:748 sofia/internal/1062 > SET [czas]=[2009-07-07-10-56-15] - variable $czas = > "2009-07-07-10-56-15" i use it in filename below > EXECUTE sofia/internal/1062 > record_session(/records/2009-07-07/069630d2-6ad4-11de-aa6e-e31e1f9431c0.10XX.07135XXXXX.2009-07-07-10-56-15.out.ISDN.wav) > ..... > stop recording > 2009-07-07 11:21:33.291177 [DEBUG] switch_ivr_async.c:444 Stop recording > file > /records/2009-07-07/069630d2-6ad4-11de-aa6e-e31e1f9431c0.10XX.07135XXXXX.2009-07-07-10-56-15.out.ISDN.wav > > file should have about: 25:18 time length > when i listen a file it's really cut > > My piece of dialplan: > > > > > > > > > > > > > > > > > > > data="/records/${dir}/${uuid}.${caller_id_number}.$1.${czas}.out.ISDN.wav"/> > > > data="{origination_caller_id_number=${cti_gateway_number},effective_caller_id_number=${cti_gateway_number}}openzap/1/A/$1"/> > > > freeswitch version: FreeSWITCH Version 1.0.trunk (14013) > linux: Linux pbx2 2.6.26-1-686 #1 SMP Fri Mar 13 18:08:45 UTC 2009 i686 > GNU/Linux > > 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 > From jgarland at gmail.com Tue Jul 7 04:44:35 2009 From: jgarland at gmail.com (Jason Garland) Date: Tue, 7 Jul 2009 07:44:35 -0400 Subject: [Freeswitch-users] Cheapest, most compact FreeSwitch appliance? In-Reply-To: <200907061023.21388.david.goodenough@linkchoose.co.uk> References: <24337599.post@talk.nabble.com> <9dc4a1670907051617k3c112eeal2356d3df55283f89@mail.gmail.com> <24348506.post@talk.nabble.com> <200907061023.21388.david.goodenough@linkchoose.co.uk> Message-ID: <1CCA942A-B45E-4EC3-AF26-89FF832B9D8C@gmail.com> I have FreeSwitch running on a $50 Linksys NSLU2 Sent from my iPhone On Jul 6, 2009, at 5:23 AM, David Goodenough wrote: > On Monday 06 July 2009, Fred-145 wrote: >> EdPimentl wrote: >>> Back in April I posted these links on the list, in regards to a >>> similar >>> question >> >> Thanks Ed, but the problem with all those, is: >> - they typically have so little RAM/Flash RAM that they can't run a >> regular >> Linux distro, which means that we're stuck with whatever software is >> available with the customized distro for the appliance >> - they don't have room for a PCI card, which means that we have to >> have an >> external VoIP gateway to connect the appliance to the POTS >> - they're as expensive or more expensive than a regular PC >> >> At this point, there doesn't seem to be any appliance that can run >> FreeSwitch and handle a POTS line in a compact, sub-$200 price- >> range. > > I though that I had read somewhere about someone using the Marvel > ShevaPlug and a 2 line USB POTS adapter. That should be under $200. > > David > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.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 Jul 7 06:18:58 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 7 Jul 2009 08:18:58 -0500 Subject: [Freeswitch-users] How to modify my INVITE msg?? In-Reply-To: References: Message-ID: <4D91CAF3-8BA4-4EBB-96EC-112D612B786B@freeswitch.org> Well put the right profile name in there... instead of just "profile" /b On Jul 6, 2009, at 9:48 PM, Brad Tuan wrote: > 2009-07-07 10:38:46 [ERR] mod_sofia.c:2681 sofia_outgoing_channel() > Invalid Profile > > FS return it is a Invalid Profile....Why?? > __________________________________ From brian at freeswitch.org Tue Jul 7 06:23:50 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 7 Jul 2009 08:23:50 -0500 Subject: [Freeswitch-users] dp tools record to 1 channel but uuid_record record to 2 channels which cannot been playback In-Reply-To: <20090707075633.GA25395@jdc.jasonjgw.net> References: <0DA560FF-D5F1-4E86-B8CD-6DBE0B9C9B47@gmail.com> <20090707071632.GA22797@jdc.jasonjgw.net> <05D41406-17C3-43DF-BA94-3753B191B7BC@gmail.com> <20090707075633.GA25395@jdc.jasonjgw.net> Message-ID: <2D58177C-7A5B-49BA-A9F1-422F1E120BB1@freeswitch.org> OK let me comment ion this. Your voip connection is a single channel mono. The recording is two channel stereo.. the caller is in the left and the callee is in the right. This is a very helpful tool for call centers when your agent gets into a fight with the caller. FreeSWITCH can not shove a stereo signal down a mono line. Its rather obvious that it has to mux it into one channel... which it does... the other alternative is to just hang up the call and say WOOPS can't do it. /b On Jul 7, 2009, at 2:56 AM, Jason White wrote: > seven wrote: > >> yes, it's here: http://pastebin.freeswitch.org/9641 > > Judging by the error message, it's a known limitation. You are > welcome to work > on a fix, or pay the develoeprs to fix it, or offer a bounty that > might > encourage someone to work on it, or wait until it gets fixed. > > Meanwhile, convert the file to mono and try again. > > Sox should be able to do this, for example. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090707/e81454d1/attachment.html From brian at freeswitch.org Tue Jul 7 06:41:35 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 7 Jul 2009 08:41:35 -0500 Subject: [Freeswitch-users] How to get the hook state? In-Reply-To: <4A52D4D1.1010805@gmail.com> References: <4A52D4D1.1010805@gmail.com> Message-ID: What are you trying to accomplish? /b On Jul 6, 2009, at 11:53 PM, Eli Hayun wrote: > Hi > I am a newbie in FreeSwitch and my question is: > When I am calling to an extension, how should I know in advance what > is > the hook status. I tried to find out a variable that can get me this > information but with no success. > any help? From lubimov at neolant.ru Tue Jul 7 06:53:02 2009 From: lubimov at neolant.ru (Alexey Lubimov) Date: Tue, 07 Jul 2009 17:53:02 +0400 Subject: [Freeswitch-users] freeswitch & sipnet.ru & caller id Message-ID: <4A53533E.9050305@neolant.ru> Good day. I have external sip gateway on sipnet.ru Logs from sipnet.ru contain my ip address (193.112.5.111) instead actual number (111 at 193.112.5.111) in field "caller id". Is it possible to set caller id in actual number instead ip address? 07/07/09 14:08 74959345610 193.112.5.111 Russia Moscow 00381982 0:14 0.01750 0.00408 From brian at freeswitch.org Tue Jul 7 07:01:55 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 7 Jul 2009 09:01:55 -0500 Subject: [Freeswitch-users] freeswitch & sipnet.ru & caller id In-Reply-To: <4A53533E.9050305@neolant.ru> References: <4A53533E.9050305@neolant.ru> Message-ID: I would need to see the sip packets to know why its doing that. /b On Jul 7, 2009, at 8:53 AM, Alexey Lubimov wrote: > Good day. > > I have external sip gateway on sipnet.ru > > Logs from sipnet.ru contain my ip address (193.112.5.111) instead > actual number (111 at 193.112.5.111) in field "caller id". > > Is it possible to set caller id in actual number instead ip address? > > > > 07/07/09 14:08 74959345610 193.112.5.111 Russia Moscow 00381982 0:14 > 0.01750 0.00408 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090707/5528f9c5/attachment.html From mcampbellsmith at gmail.com Tue Jul 7 07:11:38 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Tue, 7 Jul 2009 22:11:38 +0800 Subject: [Freeswitch-users] 2 voicemail questions Message-ID: <33c87fa30907070711gca33ba5vd251d6ae1e89b5a2@mail.gmail.com> Hi! I have 2 questions regarding voicemail ... 1. Can I email the voicemail message to multiple email addresses? If so, what format is this in? 2. How can I make Freeswitch dial a number AFTER a voicemail is left? Thanks! From lfurrea at gmail.com Tue Jul 7 08:25:38 2009 From: lfurrea at gmail.com (Luis F Urrea) Date: Tue, 7 Jul 2009 09:25:38 -0600 Subject: [Freeswitch-users] mod_fifo: Selection of consumers in a certain order. Message-ID: Hi all, We are in the need of a certain application using mod fifo. Basically we are doing the following as described in the wiki: {call_timeout=30,fifo_member_wait=nowait}user/1009@$${domain} {call_timeout=30,fifo_member_wait=nowait}user/1008@$${domain} Things work fine, but we have noticed that the consumers are selected in kind of a load balancing basis, so that if member 1009 answered the last call then the next call goes to 1008 even if 1009 is available. We would like to know if there is a way to select the next consumer available in a different fashion such as in strict order setting a preference for each member. Say member 1 preference=1 member 2 preference=2 so that only if member 1 is on the phone the call rolls to member 2. Hope it makes sense. All input on how to achieve this is appreciated. Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090707/0d3f2c3e/attachment.html From brian at freeswitch.org Tue Jul 7 08:28:35 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 7 Jul 2009 10:28:35 -0500 Subject: [Freeswitch-users] Fwd: [UniMRCP] Flite Plugin Available References: <892298.53232.qm@web111312.mail.gq1.yahoo.com> Message-ID: <1442D19E-110B-4757-B061-898C960272C2@freeswitch.org> Arsen just keeps em coming... Thank you! /b Begin forwarded message: > From: Arsen Chaloyan > Date: July 7, 2009 10:20:13 AM CDT > To: Brian West > Subject: Fw: [UniMRCP] Flite Plugin Available > > > > ----- Forwarded Message ---- > From: Arsen Chaloyan > To: UniMRCP > Cc: unimrcp-announcements at googlegroups.com > Sent: Tuesday, July 7, 2009 8:16:16 PM > Subject: [UniMRCP] Flite Plugin Available > > I would like to announce the availability of Flite TTS plugin for > UniMRCP server. > Special thanks goes to Garmt, who initially contributed and helped > develop the plugin. > > Currently supported TTS features are as follows: > > English voices: > awb > kal > rms > slt > Methods: > SPEAK > STOP > PAUSE > RESUME > BARGE-IN-OCCURRED > Events: > SPEAK-COMPLETE > Synthesizer Speech Data: > text/plain > > For the instructions on how to build and configure Flite with > UniMRCP refer to > http://code.google.com/p/unimrcp/wiki/FlitePlugin > > Please note, everything is working now, nevertheless this is basic > availability only. > I have mostly tested the integrated solution in the following setup > SIPPhone -> FreeSWITCH/UniMRCPClient -> UniMRCPServer/Flite > > > Feedback is welcome. > Thanks, > > -- > Arsen Chaloyan > The author of UniMRCP > http://www.unimrcp.org > > --~--~---------~--~----~------------~-------~--~----~ > You received this message because you are subscribed to the Google > Groups "UniMRCP" group. > To post to this group, send email to unimrcp at googlegroups.com > To unsubscribe from this group, send email to unimrcp+unsubscribe at googlegroups.com > For more options, visit this group at http://groups.google.com/group/unimrcp?hl=en > -~----------~----~----~----~------~----~------~--~--- > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090707/f00595ab/attachment-0001.html From brian at freeswitch.org Tue Jul 7 08:30:39 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 7 Jul 2009 10:30:39 -0500 Subject: [Freeswitch-users] 2 voicemail questions In-Reply-To: <33c87fa30907070711gca33ba5vd251d6ae1e89b5a2@mail.gmail.com> References: <33c87fa30907070711gca33ba5vd251d6ae1e89b5a2@mail.gmail.com> Message-ID: <839078F3-48A6-4EF5-BC9F-22475CA54644@freeswitch.org> On Jul 7, 2009, at 9:11 AM, Mark Campbell-Smith wrote: > Hi! > > I have 2 questions regarding voicemail ... > > 1. Can I email the voicemail message to multiple email addresses? If > so, what format is this in? > Try a comma sep. list. Not sure if it will work. > > 2. How can I make Freeswitch dial a number AFTER a voicemail is left? api Hangup hook? > > Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090707/87004e3a/attachment.html From dujinfang at gmail.com Tue Jul 7 08:49:08 2009 From: dujinfang at gmail.com (Seven Du) Date: Tue, 7 Jul 2009 23:49:08 +0800 Subject: [Freeswitch-users] mod_fifo: Selection of consumers in a certain order. In-Reply-To: References: Message-ID: <696807A5-33BE-4E48-8AEF-584D0A4F5FF5@gmail.com> On Jul 7, 2009, at 11:25 PM, Luis F Urrea wrote: > Hi all, > > We are in the need of a certain application using mod fifo. > Basically we are doing the following as described in the wiki: > > > lag="5">{call_timeout=30,fifo_member_wait=nowait}user/1009@$$ > {domain} > lag="5">{call_timeout=30,fifo_member_wait=nowait}user/1008@$$ > {domain} > > > > > > > > > > > > > > > > > > > Things work fine, but we have noticed that the consumers are > selected in kind of a load balancing basis, so that if member 1009 > answered the last call then the next call goes to 1008 even if 1009 > is available. > > > We would like to know if there is a way to select the next consumer > available in a different fashion such as in strict order setting a > preference > for each member. No. Maybe you'd like to patch it or add a wishlist to jira or add a bounty. But, depending on how many agents you have, a '|' separated dialstring might do the trick: > lag="5">{leg_timeout=30,fifo_member_wait=nowait}user/1009|user/1008| > user/1007... > > > Say > member 1 preference=1 > member 2 preference=2 > > > so that only if member 1 is on the phone the call rolls to member 2. > > Hope it makes sense. > > All input on how to achieve this is appreciated. > > 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 mctch at yahoo.com Tue Jul 7 10:42:16 2009 From: mctch at yahoo.com (Mark Crane) Date: Tue, 7 Jul 2009 10:42:16 -0700 (PDT) Subject: [Freeswitch-users] 2 voicemail questions Message-ID: <348845.78768.qm@web56402.mail.re3.yahoo.com> 1. Can I email the voicemail message to multiple email addresses?? If so, what format is this in? ? ? ? I've been doing this successfully for quite some time using the mailer script that I wrote I just updated the wiki so that it would show the version I have been using that allows you to send multiple emails: You can use a comma or semi-colon between emails and send as many as you want. example: or http://wiki.freeswitch.org/wiki/PHP_email#mailer_app.php The script can send to a mail server to send over plain smtp, smtp authentication or even smtp tls which works with gmail. Mark J Crane mctch at yahoo.com --- On Tue, 7/7/09, Mark Campbell-Smith wrote: From: Mark Campbell-Smith Subject: [Freeswitch-users] 2 voicemail questions To: freeswitch-users at lists.freeswitch.org Date: Tuesday, July 7, 2009, 8:11 AM Hi! I have 2 questions regarding voicemail ... 1. Can I email the voicemail message to multiple email addresses?? If so, what format is this in? ? ? ? 2. How can I make Freeswitch dial a number AFTER a voicemail is left? 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/20090707/399ec050/attachment.html From max.bridgewater at gmail.com Tue Jul 7 10:47:58 2009 From: max.bridgewater at gmail.com (Max Bridgewater) Date: Tue, 7 Jul 2009 13:47:58 -0400 Subject: [Freeswitch-users] Failed DTMF Sanity check Message-ID: Hi Guys, I keep getting this message printed in red on my consolde; and whenevr i get it, DTMF will stop being trasmitting in that session. [ERR] switch_rtp.c:2013 Failed DTMF sanity check. What does that mean and how can i prevent this from occurring? I'm using the socket API to send DTMF signals to freeswitch. Thanks, Max. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090707/f9596ba1/attachment.html From brian at freeswitch.org Tue Jul 7 10:55:28 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 7 Jul 2009 12:55:28 -0500 Subject: [Freeswitch-users] Failed DTMF Sanity check In-Reply-To: References: Message-ID: <7A3C7717-10DD-4436-8A21-403E34E6FB72@freeswitch.org> That usually means your device you're using is broken for sending rfc2833.... can you tell me what device are you using? /b On Jul 7, 2009, at 12:47 PM, Max Bridgewater wrote: > Hi Guys, > > I keep getting this message printed in red on my consolde; and > whenevr i get it, DTMF will stop being trasmitting in that session. > > [ERR] switch_rtp.c:2013 Failed DTMF sanity check. > > What does that mean and how can i prevent this from occurring? I'm > using the socket API to send DTMF signals to freeswitch. > > Thanks, > Max. From msc at freeswitch.org Tue Jul 7 10:59:00 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 7 Jul 2009 10:59:00 -0700 Subject: [Freeswitch-users] Follow FreeSWITCH_wire on Twitter! Message-ID: <87f2f3b90907071059j3a45a7d0x1d727e0f51c35b39@mail.gmail.com> Okay everyone, spread the word: we have a Twitter channel for everyone to follow. It's called "FreeSWITCH_wire" and it's were you'll see up-to-the minute updates on FreeSWITCH development, new releases, important news and the like. Everyone go follow FreeSWITCH_wire right now, and don't forget to tell everyone in the VoIP world that they need to follow it as well! -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090707/a8c3d823/attachment.html From max.bridgewater at gmail.com Tue Jul 7 11:07:54 2009 From: max.bridgewater at gmail.com (Max Bridgewater) Date: Tue, 7 Jul 2009 14:07:54 -0400 Subject: [Freeswitch-users] Failed DTMF Sanity check In-Reply-To: <7A3C7717-10DD-4436-8A21-403E34E6FB72@freeswitch.org> References: <7A3C7717-10DD-4436-8A21-403E34E6FB72@freeswitch.org> Message-ID: Actually I'm using Voxeo to generate DTMFs. They have the following construct that allows me to play DTMF: I think it's not standard VXML. How can i track this easily or at least capture the RTP stream so i can send it to them? Max. On Tue, Jul 7, 2009 at 1:55 PM, Brian West wrote: > That usually means your device you're using is broken for sending > rfc2833.... can you tell me what device are you using? > > /b > > On Jul 7, 2009, at 12:47 PM, Max Bridgewater wrote: > > > Hi Guys, > > > > I keep getting this message printed in red on my consolde; and > > whenevr i get it, DTMF will stop being trasmitting in that session. > > > > [ERR] switch_rtp.c:2013 Failed DTMF sanity check. > > > > What does that mean and how can i prevent this from occurring? I'm > > using the socket API to send DTMF signals to freeswitch. > > > > Thanks, > > Max. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090707/133e5510/attachment.html From brian at freeswitch.org Tue Jul 7 11:12:44 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 7 Jul 2009 13:12:44 -0500 Subject: [Freeswitch-users] Failed DTMF Sanity check In-Reply-To: References: <7A3C7717-10DD-4436-8A21-403E34E6FB72@freeswitch.org> Message-ID: Can you get me an RTP trace bet they might be doing it wrong... seems to be common. /b On Jul 7, 2009, at 1:07 PM, Max Bridgewater wrote: > Actually I'm using Voxeo to generate DTMFs. They have the following > construct that allows me to play DTMF: > > > > I think it's not standard VXML. How can i track this easily or at > least capture the RTP stream so i can send it to them? > > Max. From msc at freeswitch.org Tue Jul 7 11:15:03 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 7 Jul 2009 11:15:03 -0700 Subject: [Freeswitch-users] Failed DTMF Sanity check In-Reply-To: References: <7A3C7717-10DD-4436-8A21-403E34E6FB72@freeswitch.org> Message-ID: <87f2f3b90907071115n1664d204tcb50cfbe7e5a9195@mail.gmail.com> There are some troubleshooting tips here: http://wiki.freeswitch.org/wiki/Reporting_Bugs#Debugging_Specific_Technologies There are several ways of capturing packets on your system and the above link explains how to set them up. -MC On Tue, Jul 7, 2009 at 11:07 AM, Max Bridgewater wrote: > Actually I'm using Voxeo to generate DTMFs. They have the following > construct that allows me to play DTMF: > > > > I think it's not standard VXML. How can i track this easily or at least > capture the RTP stream so i can send it to them? > > Max. > > > > On Tue, Jul 7, 2009 at 1:55 PM, Brian West wrote: > >> That usually means your device you're using is broken for sending >> rfc2833.... can you tell me what device are you using? >> >> /b >> >> On Jul 7, 2009, at 12:47 PM, Max Bridgewater wrote: >> >> > Hi Guys, >> > >> > I keep getting this message printed in red on my consolde; and >> > whenevr i get it, DTMF will stop being trasmitting in that session. >> > >> > [ERR] switch_rtp.c:2013 Failed DTMF sanity check. >> > >> > What does that mean and how can i prevent this from occurring? I'm >> > using the socket API to send DTMF signals to freeswitch. >> > >> > Thanks, >> > Max. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090707/3dd29d8c/attachment.html From max.bridgewater at gmail.com Tue Jul 7 11:19:31 2009 From: max.bridgewater at gmail.com (Max Bridgewater) Date: Tue, 7 Jul 2009 14:19:31 -0400 Subject: [Freeswitch-users] Failed DTMF Sanity check In-Reply-To: References: <7A3C7717-10DD-4436-8A21-403E34E6FB72@freeswitch.org> Message-ID: > Can you get me an RTP trace bet they might be doing it wrong... seems > to be common. > Hmm Sorry. Can i activate RTP traces in Freeswitch somehow or do i need to run Something like Wireshark? Pascal. > > /b > > On Jul 7, 2009, at 1:07 PM, Max Bridgewater wrote: > > > Actually I'm using Voxeo to generate DTMFs. They have the following > > construct that allows me to play DTMF: > > > > > > > > I think it's not standard VXML. How can i track this easily or at > > least capture the RTP stream so i can send it to them? > > > > Max. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090707/1c556506/attachment.html From msc at freeswitch.org Tue Jul 7 11:40:54 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 7 Jul 2009 11:40:54 -0700 Subject: [Freeswitch-users] Failed DTMF Sanity check In-Reply-To: References: <7A3C7717-10DD-4436-8A21-403E34E6FB72@freeswitch.org> Message-ID: <87f2f3b90907071140l3cde3a79se5a0b4adc1eeab81@mail.gmail.com> You'll need Wireshark or similar. Some tips can be found here: http://wiki.freeswitch.org/wiki/Reporting_Bugs#Debugging_Specific_Technologies -MC On Tue, Jul 7, 2009 at 11:19 AM, Max Bridgewater wrote: > > Can you get me an RTP trace bet they might be doing it wrong... seems >> to be common. >> > > > Hmm Sorry. Can i activate RTP traces in Freeswitch somehow or do i need to > run Something like Wireshark? > > Pascal. > >> >> /b >> >> On Jul 7, 2009, at 1:07 PM, Max Bridgewater wrote: >> >> > Actually I'm using Voxeo to generate DTMFs. They have the following >> > construct that allows me to play DTMF: >> > >> > >> > >> > I think it's not standard VXML. How can i track this easily or at >> > least capture the RTP stream so i can send it to them? >> > >> > Max. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090707/3b0306c4/attachment.html From max.bridgewater at gmail.com Tue Jul 7 11:48:24 2009 From: max.bridgewater at gmail.com (Max Bridgewater) Date: Tue, 7 Jul 2009 14:48:24 -0400 Subject: [Freeswitch-users] Failed DTMF Sanity check In-Reply-To: <87f2f3b90907071115n1664d204tcb50cfbe7e5a9195@mail.gmail.com> References: <7A3C7717-10DD-4436-8A21-403E34E6FB72@freeswitch.org> <87f2f3b90907071115n1664d204tcb50cfbe7e5a9195@mail.gmail.com> Message-ID: Thanks! What is the best way to send you the 4M pcap file? On Tue, Jul 7, 2009 at 2:15 PM, Michael Collins wrote: > There are some troubleshooting tips here: > > http://wiki.freeswitch.org/wiki/Reporting_Bugs#Debugging_Specific_Technologies > > There are several ways of capturing packets on your system and the above > link explains how to set them up. > -MC > > > On Tue, Jul 7, 2009 at 11:07 AM, Max Bridgewater < > max.bridgewater at gmail.com> wrote: > >> Actually I'm using Voxeo to generate DTMFs. They have the following >> construct that allows me to play DTMF: >> >> >> >> I think it's not standard VXML. How can i track this easily or at least >> capture the RTP stream so i can send it to them? >> >> Max. >> >> >> >> On Tue, Jul 7, 2009 at 1:55 PM, Brian West wrote: >> >>> That usually means your device you're using is broken for sending >>> rfc2833.... can you tell me what device are you using? >>> >>> /b >>> >>> On Jul 7, 2009, at 12:47 PM, Max Bridgewater wrote: >>> >>> > Hi Guys, >>> > >>> > I keep getting this message printed in red on my consolde; and >>> > whenevr i get it, DTMF will stop being trasmitting in that session. >>> > >>> > [ERR] switch_rtp.c:2013 Failed DTMF sanity check. >>> > >>> > What does that mean and how can i prevent this from occurring? I'm >>> > using the socket API to send DTMF signals to freeswitch. >>> > >>> > Thanks, >>> > Max. >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090707/b757abdd/attachment.html From brian at freeswitch.org Tue Jul 7 11:52:26 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 7 Jul 2009 13:52:26 -0500 Subject: [Freeswitch-users] Failed DTMF Sanity check In-Reply-To: References: <7A3C7717-10DD-4436-8A21-403E34E6FB72@freeswitch.org> <87f2f3b90907071115n1664d204tcb50cfbe7e5a9195@mail.gmail.com> Message-ID: <64908434-88C2-4B98-8C1C-15CD2585EB53@freeswitch.org> email it directly to me off list please. /b On Jul 7, 2009, at 1:48 PM, Max Bridgewater wrote: > Thanks! What is the best way to send you the 4M pcap file? From msc at freeswitch.org Tue Jul 7 11:53:43 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 7 Jul 2009 11:53:43 -0700 Subject: [Freeswitch-users] Failed DTMF Sanity check In-Reply-To: References: <7A3C7717-10DD-4436-8A21-403E34E6FB72@freeswitch.org> <87f2f3b90907071115n1664d204tcb50cfbe7e5a9195@mail.gmail.com> Message-ID: <87f2f3b90907071153u3212d5c0vcb49270d1031b0fe@mail.gmail.com> Put it out on a webserver where one of the devs can grab it with a browser. -MC On Tue, Jul 7, 2009 at 11:48 AM, Max Bridgewater wrote: > Thanks! What is the best way to send you the 4M pcap file? > > > On Tue, Jul 7, 2009 at 2:15 PM, Michael Collins wrote: > >> There are some troubleshooting tips here: >> >> http://wiki.freeswitch.org/wiki/Reporting_Bugs#Debugging_Specific_Technologies >> >> There are several ways of capturing packets on your system and the above >> link explains how to set them up. >> -MC >> >> >> On Tue, Jul 7, 2009 at 11:07 AM, Max Bridgewater < >> max.bridgewater at gmail.com> wrote: >> >>> Actually I'm using Voxeo to generate DTMFs. They have the following >>> construct that allows me to play DTMF: >>> >>> >>> >>> I think it's not standard VXML. How can i track this easily or at least >>> capture the RTP stream so i can send it to them? >>> >>> Max. >>> >>> >>> >>> On Tue, Jul 7, 2009 at 1:55 PM, Brian West wrote: >>> >>>> That usually means your device you're using is broken for sending >>>> rfc2833.... can you tell me what device are you using? >>>> >>>> /b >>>> >>>> On Jul 7, 2009, at 12:47 PM, Max Bridgewater wrote: >>>> >>>> > Hi Guys, >>>> > >>>> > I keep getting this message printed in red on my consolde; and >>>> > whenevr i get it, DTMF will stop being trasmitting in that session. >>>> > >>>> > [ERR] switch_rtp.c:2013 Failed DTMF sanity check. >>>> > >>>> > What does that mean and how can i prevent this from occurring? I'm >>>> > using the socket API to send DTMF signals to freeswitch. >>>> > >>>> > Thanks, >>>> > Max. >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090707/fcf82335/attachment-0001.html From jalsot at gmail.com Tue Jul 7 12:01:28 2009 From: jalsot at gmail.com (Tamas) Date: Tue, 07 Jul 2009 21:01:28 +0200 Subject: [Freeswitch-users] Failed DTMF Sanity check In-Reply-To: <87f2f3b90907071153u3212d5c0vcb49270d1031b0fe@mail.gmail.com> References: <7A3C7717-10DD-4436-8A21-403E34E6FB72@freeswitch.org> <87f2f3b90907071115n1664d204tcb50cfbe7e5a9195@mail.gmail.com> <87f2f3b90907071153u3212d5c0vcb49270d1031b0fe@mail.gmail.com> Message-ID: <4A539B88.6070407@gmail.com> http://filebin.ca/ (up to 50MB) Tamas Michael Collins ?rta: > Put it out on a webserver where one of the devs can grab it with a > browser. > -MC > > On Tue, Jul 7, 2009 at 11:48 AM, Max Bridgewater > > wrote: > > Thanks! What is the best way to send you the 4M pcap file? > > > On Tue, Jul 7, 2009 at 2:15 PM, Michael Collins > > wrote: > > There are some troubleshooting tips here: > > http://wiki.freeswitch.org/wiki/Reporting_Bugs#Debugging_Specific_Technologies > > There are several ways of capturing packets on your system and > the above link explains how to set them up. > -MC > > > On Tue, Jul 7, 2009 at 11:07 AM, Max Bridgewater > > > wrote: > > Actually I'm using Voxeo to generate DTMFs. They have the > following construct that allows me to play DTMF: > > > > I think it's not standard VXML. How can i track this > easily or at least capture the RTP stream so i can send it > to them? > > Max. > > > > On Tue, Jul 7, 2009 at 1:55 PM, Brian West > > wrote: > > That usually means your device you're using is broken > for sending > rfc2833.... can you tell me what device are you using? > > /b > > On Jul 7, 2009, at 12:47 PM, Max Bridgewater wrote: > > > Hi Guys, > > > > I keep getting this message printed in red on my > consolde; and > > whenevr i get it, DTMF will stop being trasmitting > in that session. > > > > [ERR] switch_rtp.c:2013 Failed DTMF sanity check. > > > > What does that mean and how can i prevent this from > occurring? I'm > > using the socket API to send DTMF signals to freeswitch. > > > > Thanks, > > Max. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From marketing at cluecon.com Tue Jul 7 12:14:04 2009 From: marketing at cluecon.com (Michael Collins) Date: Tue, 7 Jul 2009 12:14:04 -0700 Subject: [Freeswitch-users] ClueCon 2009 Updates - July 7, 2009 Message-ID: <87f2f3b90907071214w60f0d1b0x21d39b7e5ad661ac@mail.gmail.com> Hello folks! We have a few more updates. First of all, if you haven't already heard, we've extended the early bird sign up to go through July 21. That's only two weeks away, so if you haven't already registered then please call us at 877.742.CLUE and we'll get you set up. Secondly, there are some updates on the ClueCon blog: http://cluecon.com/blog/1 The breakfast and lunch menus have been posted. (Subject to change, of course. :) Also, we have a synopsis for Irv Shapiro's talk, entitled "Cloud Telephony" on the latest blog post. Can't wait to see you all in Chicago! -Michael Collins http://www.cluecon.com 877.742.CLUE -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090707/a8639017/attachment.html From anthony.minessale at gmail.com Tue Jul 7 13:37:54 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 7 Jul 2009 15:37:54 -0500 Subject: [Freeswitch-users] Controlling MOH from a java application In-Reply-To: <443906.62890.qm@web35604.mail.mud.yahoo.com> References: <443906.62890.qm@web35604.mail.mud.yahoo.com> Message-ID: <191c3a030907071337i6eb2ce09i387d9912a05d36bd@mail.gmail.com> FSAPI commands are accessed via the API obj api = new API(); api.execute("uuid_hold", session.get_uuid()); ... api.execute("uuid_hold", "off " + session.get_uuid()); On Mon, Jul 6, 2009 at 10:21 PM, Jean-Marc Hyppolite wrote: > Hello Brian, > > Thank you for your quick answer. I tried the two API functions but with no > result. The caller is not able to hear any music. But, when I use two > extensions (one calling the other), MOH does work. > > My code on the JAVA side (for test purposes) > > session.answer(); > session.sleep(500); > > session.execute("eval", "uuid_hold " + session.get_uuid()); > > java_function(); // lasts 30 to 40 seconds > > session.execute("eval", "uuid_hold off " + session.get_uuid()); > > session.sleep(500); > session.hangup(); > > Thanks for the help. > > Jean-Marc. > > > > > --- On *Mon, 7/6/09, Brian West * wrote: > > > From: Brian West > Subject: Re: [Freeswitch-users] Controlling MOH from a java application > To: freeswitch-users at lists.freeswitch.org > Received: Monday, July 6, 2009, 7:27 PM > > > uuid_hold > uuid_hold off > > These two api's will do it. > > /b > > > > On Jul 6, 2009, at 6:20 PM, Jean-Marc Hyppolite wrote: > > Hello, > > First of all, I would like to thank Anthony, Brian and all the developers > for this wonderful piece of software. Very good job. > > I would like to know how I can start and stop Music On Hold from a JAVA > script (using mod_java) similar to the StartMusicOnHold and StopMusicHold > functions found in AGI (Asterisk-Java). > > I am using FreeSWITCH as an IVR server. I would like to be able to put the > caller on hold while doing some other stuff. > > Thanks in advance. > > Jean-Marc. > > > ------------------------------ > > *Yahoo! Canada Toolbar :* Search from anywhere on the web and bookmark > your favourite sites. Download it now! > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -----Inline Attachment Follows----- > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > ------------------------------ > *All new Yahoo! Mail - * Get > a sneak peak at messages with a handy reading pane. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference 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/20090707/db4c86d2/attachment.html From kees at mroffice.org Tue Jul 7 14:26:38 2009 From: kees at mroffice.org (Kees Varekamp) Date: Wed, 8 Jul 2009 09:26:38 +1200 Subject: [Freeswitch-users] Leaking stream handle Message-ID: <98d38dcf0907071426m581d07bcr27702c5e5bd3b574@mail.gmail.com> I am testing Freeswitch as an alternative to Asterisk. So far, so good, except for the following: - I have a lua channel listening to: - session:streamFile('local_stream://moh') - I have a socket bridging this channel to a sip gateway: - SendMsg 1e0cc726-6b33-11de-bae1-5fd843059ad5 - call-command: execute - execute-app-name: bridge - execute-app-arg: sofia/gateway// - This all works well, but the console says: - [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1026] Is this something I should be worried about? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090708/6c8b1e9e/attachment.html From vladislaus at gmail.com Tue Jul 7 14:29:01 2009 From: vladislaus at gmail.com (Andres Gomez) Date: Tue, 7 Jul 2009 16:29:01 -0500 Subject: [Freeswitch-users] Clustering Freeswitch Message-ID: <31da44070907071429g3d31687bj714b827b0ec69f02@mail.gmail.com> Hello to all Exist any solution to clustering. Any load balancing appliance o heartbeat test?. Regards Andres Gomez. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090707/1fd6c9b7/attachment.html From sprice at gmail.com Tue Jul 7 14:39:06 2009 From: sprice at gmail.com (SP) Date: Tue, 7 Jul 2009 16:39:06 -0500 Subject: [Freeswitch-users] Clustering Freeswitch In-Reply-To: <31da44070907071429g3d31687bj714b827b0ec69f02@mail.gmail.com> References: <31da44070907071429g3d31687bj714b827b0ec69f02@mail.gmail.com> Message-ID: <7e2ac3270907071439o7bf90e45g6c6017b48f6dfa64@mail.gmail.com> openser/kamailio/ser/opensips you pick a name DNS SRV On Tue, Jul 7, 2009 at 16:29, Andres Gomez wrote: > Hello to all > > Exist any solution to clustering. Any load balancing appliance o heartbeat > test?. > > Regards > > Andres Gomez. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090707/13feee7b/attachment-0001.html From sicfslist at gmail.com Tue Jul 7 14:39:59 2009 From: sicfslist at gmail.com (Shelby Ramsey) Date: Tue, 7 Jul 2009 16:39:59 -0500 Subject: [Freeswitch-users] Clustering Freeswitch In-Reply-To: <31da44070907071429g3d31687bj714b827b0ec69f02@mail.gmail.com> References: <31da44070907071429g3d31687bj714b827b0ec69f02@mail.gmail.com> Message-ID: <35b355e90907071439r698907b8o3b2dae2ef1b22620@mail.gmail.com> Andres, OpenSIP's works very well as a load balancer. You could also use DNS SRV (if the clients support it), round robin DNS .... Really just depends on what you are trying to accomplish. SDR -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090707/289a80b5/attachment.html From anthony.minessale at gmail.com Tue Jul 7 14:53:13 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 7 Jul 2009 16:53:13 -0500 Subject: [Freeswitch-users] Leaking stream handle In-Reply-To: <98d38dcf0907071426m581d07bcr27702c5e5bd3b574@mail.gmail.com> References: <98d38dcf0907071426m581d07bcr27702c5e5bd3b574@mail.gmail.com> Message-ID: <191c3a030907071453g6a3c5eb3ha4ed77d721403b1a@mail.gmail.com> you should stop playing the file first. or transfer the call to the bridge app instead of executing it direct. send this over event_socket (replacing uuid of course) api uuid_transfer 1e0cc726-6b33-11de-bae1-5fd843059ad5 bridge:sofia/gateway// inline On Tue, Jul 7, 2009 at 4:26 PM, Kees Varekamp wrote: > I am testing Freeswitch as an alternative to Asterisk. So far, so good, > except for the following: > > > - I have a lua channel listening to: > - session:streamFile('local_stream://moh') > - I have a socket bridging this channel to a sip gateway: > - SendMsg 1e0cc726-6b33-11de-bae1-5fd843059ad5 > - call-command: execute > - execute-app-name: bridge > - execute-app-arg: sofia/gateway// > - This all works well, but the console says: > - [CRIT] mod_local_stream.c:234 Leaking stream handle! > [switch_ivr_play_file() src/switch_ivr_play_say.c:1026] > > > Is this something I should be worried about? 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/20090707/e44e6c53/attachment.html From raul at etellicom.com Tue Jul 7 18:14:40 2009 From: raul at etellicom.com (Raul Fragoso) Date: Tue, 07 Jul 2009 22:14:40 -0300 Subject: [Freeswitch-users] Cheapest, most compact FreeSwitch appliance? In-Reply-To: <24348506.post@talk.nabble.com> References: <24337599.post@talk.nabble.com> <7C0A0FFD-EC03-46DE-A277-AE793017D617@howlertech.com> <24345928.post@talk.nabble.com> <9dc4a1670907051617k3c112eeal2356d3df55283f89@mail.gmail.com> <24348506.post@talk.nabble.com> Message-ID: <1247015680.4515.15.camel@raul-laptop> I'm about to order one of these eBox systems: http://www.wdlsystems.com/modperl/view_services.cgi?r=detail&prod_num=1EBX33J&aisle_id=1073 They sell for under $200, have decent specs and some models come with a mini-PCI slot, which can be used to attach a POTS card. Regards, Raul On Sun, 2009-07-05 at 18:22 -0700, Fred-145 wrote: > > EdPimentl wrote: > > Back in April I posted these links on the list, in regards to a similar > > question > > Thanks Ed, but the problem with all those, is: > - they typically have so little RAM/Flash RAM that they can't run a regular > Linux distro, which means that we're stuck with whatever software is > available with the customized distro for the appliance > - they don't have room for a PCI card, which means that we have to have an > external VoIP gateway to connect the appliance to the POTS > - they're as expensive or more expensive than a regular PC > > At this point, there doesn't seem to be any appliance that can run > FreeSwitch and handle a POTS line in a compact, sub-$200 price-range. From brian at freeswitch.org Tue Jul 7 18:26:54 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 7 Jul 2009 20:26:54 -0500 Subject: [Freeswitch-users] Cheapest, most compact FreeSwitch appliance? In-Reply-To: <1247015680.4515.15.camel@raul-laptop> References: <24337599.post@talk.nabble.com> <7C0A0FFD-EC03-46DE-A277-AE793017D617@howlertech.com> <24345928.post@talk.nabble.com> <9dc4a1670907051617k3c112eeal2356d3df55283f89@mail.gmail.com> <24348506.post@talk.nabble.com> <1247015680.4515.15.camel@raul-laptop> Message-ID: <318EF937-FF6E-4D40-A432-1BFA797C2040@freeswitch.org> I'll take two please! ;) /b On Jul 7, 2009, at 8:14 PM, Raul Fragoso wrote: > I'm about to order one of these eBox systems: > http://www.wdlsystems.com/modperl/view_services.cgi?r=detail&prod_num=1EBX33J&aisle_id=1073 > > They sell for under $200, have decent specs and some models come > with a > mini-PCI slot, which can be used to attach a POTS card. > > Regards, > > Raul > > On Sun, 2009-07-05 at 18:22 -0700, Fred-145 wrote: >> >> EdPimentl wrote: >>> Back in April I posted these links on the list, in regards to a >>> similar >>> question >> >> Thanks Ed, but the problem with all those, is: >> - they typically have so little RAM/Flash RAM that they can't run a >> regular >> Linux distro, which means that we're stuck with whatever software is >> available with the customized distro for the appliance >> - they don't have room for a PCI card, which means that we have to >> have an >> external VoIP gateway to connect the appliance to the POTS >> - they're as expensive or more expensive than a regular PC >> >> At this point, there doesn't seem to be any appliance that can run >> FreeSwitch and handle a POTS line in a compact, sub-$200 price- >> range. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From intralanman at freeswitch.org Tue Jul 7 18:28:56 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Tue, 07 Jul 2009 21:28:56 -0400 Subject: [Freeswitch-users] Baby Update! In-Reply-To: <41646A56-C830-489D-99D7-CE5C1218E194@freeswitch.org> References: <3EC5EF44-BF81-4C23-AD0B-B4AAB5F7DFCE@freeswitch.org> <1246629538.4185.9.camel@dk-d820> <41646A56-C830-489D-99D7-CE5C1218E194@freeswitch.org> Message-ID: <4A53F658.2030400@freeswitch.org> Brian West wrote: > Kaiden Anthony Chandler arrived Friday July 3rd at 1411 EDT 7lbs 10oz > > YAY... Congrats mr Lanman! > > /b THANKS BRIAN!!!! And the few others that sent money. The hospital is a little over an hour drive, so the money definitely helped out with gas, etc... You really are a great community... -Ray -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090707/9f1c891c/attachment.html From hads at nice.net.nz Tue Jul 7 18:52:59 2009 From: hads at nice.net.nz (Hadley Rich) Date: Wed, 08 Jul 2009 13:52:59 +1200 Subject: [Freeswitch-users] Cheapest, most compact FreeSwitch appliance? In-Reply-To: References: <24337599.post@talk.nabble.com> Message-ID: <1247017979.30254.11.camel@lithium.nice.net.nz> On Sat, 2009-07-04 at 15:18 -0500, Brian West wrote: > I use one of the intel atom boxes at home. These guys have a case/PCI-riser for an Intel Atom board which would make a nice little appliance. http://www.mini-box.com/I-O-shield-and-riser-card-for-D945GSEJT hads -- http://nicegear.co.nz New Zealand's Open Source Hardware Supplier From brian at freeswitch.org Tue Jul 7 19:02:11 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 7 Jul 2009 21:02:11 -0500 Subject: [Freeswitch-users] Cheapest, most compact FreeSwitch appliance? In-Reply-To: <1247017979.30254.11.camel@lithium.nice.net.nz> References: <24337599.post@talk.nabble.com> <1247017979.30254.11.camel@lithium.nice.net.nz> Message-ID: <3ACD5095-3594-4990-8360-3131EF374D32@freeswitch.org> I'll take two of those too! :) /b On Jul 7, 2009, at 8:52 PM, Hadley Rich wrote: > On Sat, 2009-07-04 at 15:18 -0500, Brian West wrote: >> I use one of the intel atom boxes at home. > > These guys have a case/PCI-riser for an Intel Atom board which would > make a nice little appliance. > > http://www.mini-box.com/I-O-shield-and-riser-card-for-D945GSEJT > > hads > -- > http://nicegear.co.nz > New Zealand's Open Source Hardware Supplier > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090707/98ecd69e/attachment.html From craig at overthewire.com.au Tue Jul 7 19:11:16 2009 From: craig at overthewire.com.au (Craig Askings) Date: Wed, 8 Jul 2009 12:11:16 +1000 Subject: [Freeswitch-users] Cheapest, most compact FreeSwitch appliance? In-Reply-To: <1247015680.4515.15.camel@raul-laptop> References: <24337599.post@talk.nabble.com> <7C0A0FFD-EC03-46DE-A277-AE793017D617@howlertech.com> <24345928.post@talk.nabble.com> <9dc4a1670907051617k3c112eeal2356d3df55283f89@mail.gmail.com> <24348506.post@talk.nabble.com> <1247015680.4515.15.camel@raul-laptop> Message-ID: <8cc991dd0907071911v1c26861ck43f54d43c2f055ce@mail.gmail.com> Does the mini-pci slot have external access to route the POTS cable? 2009/7/8 Raul Fragoso : > I'm about to order one of these eBox systems: > http://www.wdlsystems.com/modperl/view_services.cgi?r=detail&prod_num=1EBX33J&aisle_id=1073 > > They sell for under $200, have decent specs and some models come with a > mini-PCI slot, which can be used to attach a POTS card. > > Regards, > > Raul -- Craig Askings Network Engineer | Over the Wire Pty Ltd craig at overthewire.com.au | www.overthewire.com.au Phone: 07 3847 9292 | Fax: 07 3847 9696 | Mobile: 0404 019 365 From hyppolite72 at yahoo.com Tue Jul 7 19:22:10 2009 From: hyppolite72 at yahoo.com (Jean-Marc Hyppolite) Date: Tue, 7 Jul 2009 19:22:10 -0700 (PDT) Subject: [Freeswitch-users] Controlling MOH from a java application Message-ID: <602658.55818.qm@web35601.mail.mud.yahoo.com> Hello, ? Thank you Anthony. ? My problem now is when the call is put on hold, no music is heard from the caller. ? Thank you again. ? Jean-Marc. --- On Tue, 7/7/09, Anthony Minessale wrote: From: Anthony Minessale Subject: Re: [Freeswitch-users] Controlling MOH from a java application To: freeswitch-users at lists.freeswitch.org Received: Tuesday, July 7, 2009, 4:37 PM FSAPI commands are accessed via the API obj api = new API(); api.execute("uuid_hold", session.get_uuid()); ... api.execute("uuid_hold", "off " + session.get_uuid()); On Mon, Jul 6, 2009 at 10:21 PM, Jean-Marc Hyppolite wrote: Hello Brian, ? Thank you for your quick answer. I tried the two API functions but with no result. The caller is not able to hear any music. But, when I use two extensions (one calling the other), MOH does work. ? My code on the JAVA side (for test purposes) ? session.answer(); session.sleep(500); ? session.execute("eval", "uuid_hold " + session.get_uuid()); ? java_function(); // lasts 30 to 40 seconds ? session.execute("eval", "uuid_hold off " + session.get_uuid()); ? session.sleep(500); session.hangup(); ? Thanks for the help. ? Jean-Marc. ? ? ? --- On Mon, 7/6/09, Brian West wrote: From: Brian West Subject: Re: [Freeswitch-users] Controlling MOH from a java application To: freeswitch-users at lists.freeswitch.org Received: Monday, July 6, 2009, 7:27 PM uuid_hold uuid_hold off These two api's will do it. /b On Jul 6, 2009, at 6:20 PM, Jean-Marc Hyppolite wrote: Hello, ? First of all, I would like to thank Anthony, Brian and all the developers for this wonderful piece of software. Very good job. ? I would like to know how I can start and stop Music On Hold from a JAVA script (using mod_java) similar to the StartMusicOnHold and StopMusicHold functions found in AGI (Asterisk-Java). ? I am using FreeSWITCH as an IVR server. I would like to be able to put the caller on hold while doing some other stuff. ? Thanks in advance. ? Jean-Marc. Yahoo! Canada Toolbar : Search from anywhere on the web and bookmark your favourite sites. Download it now! _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -----Inline Attachment Follows----- _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org All new Yahoo! Mail - Get a sneak peak at messages with a handy reading pane. _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -----Inline Attachment Follows----- _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org __________________________________________________________________ Yahoo! Canada Toolbar: Search from anywhere on the web, and bookmark your favourite sites. Download it now http://ca.toolbar.yahoo.com. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090707/fdf9ed2e/attachment-0001.html From brian at freeswitch.org Tue Jul 7 19:28:08 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 7 Jul 2009 21:28:08 -0500 Subject: [Freeswitch-users] Controlling MOH from a java application In-Reply-To: <602658.55818.qm@web35601.mail.mud.yahoo.com> References: <602658.55818.qm@web35601.mail.mud.yahoo.com> Message-ID: <33BE7599-9BA8-4743-A725-C260E89BE8C8@freeswitch.org> Can you open a jira please. /b On Jul 7, 2009, at 9:22 PM, Jean-Marc Hyppolite wrote: > Hello, > > Thank you Anthony. > > My problem now is when the call is put on hold, no music is heard > from the caller. > > Thank you again. > > Jean-Marc. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090707/2c3f3de6/attachment.html From Nick.Lemberger at lkfd.net Tue Jul 7 19:46:15 2009 From: Nick.Lemberger at lkfd.net (Nick Lemberger) Date: Tue, 07 Jul 2009 21:46:15 -0500 Subject: [Freeswitch-users] Force SUBSCRIBE or sendevent NOTIFY without subscription Message-ID: <4A53C228020000FE00009B5D@application-tr-fa-1.lakefield.telco> Is it possible to force a sofia profile to subscribe to an event or use sendevent to force send a NOTIFY to a SIP endpoint? I'm trying to use FreeSwitch as a voicemail server but the sending switch doesn't send SIP SUBSCRIBE messages. I'd like to send unsolicited SIP notifies to turn on MWI indicators as that's what the switch expects. Any ideas, or is this even possible with FreeSwitch? Thanks, Nick From hyppolite72 at yahoo.com Tue Jul 7 19:54:46 2009 From: hyppolite72 at yahoo.com (Jean-Marc Hyppolite) Date: Tue, 7 Jul 2009 19:54:46 -0700 (PDT) Subject: [Freeswitch-users] Controlling MOH from a java application Message-ID: <141988.11413.qm@web35607.mail.mud.yahoo.com> Thank you Brian. --- On Tue, 7/7/09, Brian West wrote: From: Brian West Subject: Re: [Freeswitch-users] Controlling MOH from a java application To: freeswitch-users at lists.freeswitch.org Received: Tuesday, July 7, 2009, 10:28 PM Can you open a jira please. /b On Jul 7, 2009, at 9:22 PM, Jean-Marc Hyppolite wrote: Hello, ? Thank you Anthony. ? My problem now is when the call is put on hold, no music is heard from the caller. ? Thank you again. ? Jean-Marc. -----Inline Attachment Follows----- _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org __________________________________________________________________ Make your browsing faster, safer, and easier with the new Internet Explorer? 8. Optimized for Yahoo! Get it Now for Free! at http://downloads.yahoo.com/ca/internetexplorer/ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090707/86d8fbeb/attachment.html From brian at freeswitch.org Tue Jul 7 19:59:09 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 7 Jul 2009 21:59:09 -0500 Subject: [Freeswitch-users] Force SUBSCRIBE or sendevent NOTIFY without subscription In-Reply-To: <4A53C228020000FE00009B5D@application-tr-fa-1.lakefield.telco> References: <4A53C228020000FE00009B5D@application-tr-fa-1.lakefield.telco> Message-ID: <3B3DB3A0-5EB9-475F-A456-1B34952DFA13@freeswitch.org> Yes you can sendevent NOTIFY Here is the headers you'll need.. some are optional const char *profile_name = switch_event_get_header(event, "profile"); const char *ct = switch_event_get_header(event, "content-type"); const char *es = switch_event_get_header(event, "event-string"); const char *user = switch_event_get_header(event, "user"); const char *host = switch_event_get_header(event, "host"); const char *call_id = switch_event_get_header(event, "call-id"); const char *uuid = switch_event_get_header(event, "uuid"); const char *body = switch_event_get_body(event); const char *to_uri = switch_event_get_header(event, "to-uri"); const char *from_uri = switch_event_get_header(event, "from-uri"); See mod_sofia.c line 2887 /b PS: you also have SEND_MESSAGE as an event you can send below that in mod_sofia.c On Jul 7, 2009, at 9:46 PM, Nick Lemberger wrote: > Is it possible to force a sofia profile to subscribe to an event or > use sendevent to force send a NOTIFY to a SIP endpoint? > > I'm trying to use FreeSwitch as a voicemail server but the sending > switch doesn't send SIP SUBSCRIBE messages. I'd like to send > unsolicited SIP notifies to turn on MWI indicators as that's what > the switch expects. > > Any ideas, or is this even possible with FreeSwitch? > > Thanks, > Nick From yehavi.bourvine at gmail.com Tue Jul 7 21:40:36 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Wed, 8 Jul 2009 07:40:36 +0300 Subject: [Freeswitch-users] How to get the hook state? In-Reply-To: References: <4A52D4D1.1010805@gmail.com> Message-ID: Hello, The problem we are trying to solve here is handling a busy state according to the user's prefference (some want a busy to be heard, some want the call to go to voicemail, and some want to get the second call). The first step is finding that an extension is busy. It would be nice in the future to know also other states of an extension (like - not registered, etc.). Thanks, __Yehavi: 2009/7/7 Brian West > What are you trying to accomplish? > > /b > > On Jul 6, 2009, at 11:53 PM, Eli Hayun wrote: > > > Hi > > I am a newbie in FreeSwitch and my question is: > > When I am calling to an extension, how should I know in advance what > > is > > the hook status. I tried to find out a variable that can get me this > > information but with no success. > > any help? > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090708/66f98d37/attachment.html From shiyanov at gmail.com Tue Jul 7 23:39:15 2009 From: shiyanov at gmail.com (Artem Shiyanov) Date: Wed, 8 Jul 2009 10:39:15 +0400 Subject: [Freeswitch-users] Can't mute SIP channel with "receiveonly" in SDP Message-ID: Hy all! With Asterisk I can mute SIP channel using re-INVITE with "a=receiveonly" in media description. But this feature doesn't work with Freeswitch. For sure, there is old good method: transfer both legs to the conference room where one leg is able to listen/talk, the other one - only to listen, but this is unwanted workaround for me.. So I wonder: is there any other (preferably through the SIP) way to "mute" given SIP channel with Freeswitch? Thanks for all, Artem Shiyanov -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090708/32e2a3f9/attachment.html From Claudio.Cavalera at italtel.it Wed Jul 8 01:53:39 2009 From: Claudio.Cavalera at italtel.it (Cavalera Claudio Luigi) Date: Wed, 8 Jul 2009 10:53:39 +0200 Subject: [Freeswitch-users] ClueCon 2009 Updates - July 7, 2009 In-Reply-To: <87f2f3b90907071214w60f0d1b0x21d39b7e5ad661ac@mail.gmail.com> Message-ID: I'm really missing this event! Can't we organize a ClueCon Winter edition in Europe too? :-) BRs, Claudio ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Tuesday, July 07, 2009 9:14 PM To: marketing at cluecon.com Subject: [Freeswitch-users] ClueCon 2009 Updates - July 7, 2009 Hello folks! We have a few more updates. First of all, if you haven't already heard, we've extended the early bird sign up to go through July 21. That's only two weeks away, so if you haven't already registered then please call us at 877.742.CLUE and we'll get you set up. Secondly, there are some updates on the ClueCon blog: http://cluecon.com/blog/1 The breakfast and lunch menus have been posted. (Subject to change, of course. :) Also, we have a synopsis for Irv Shapiro's talk, entitled "Cloud Telephony" on the latest blog post. Can't wait to see you all in Chicago! -Michael Collins http://www.cluecon.com 877.742.CLUE 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. ----------------------------------------------------------------------------------------------------- -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090708/82ee3da4/attachment-0001.html From michal.bielicki at halo2.pl Wed Jul 8 02:13:59 2009 From: michal.bielicki at halo2.pl (Michal Bielicki) Date: Wed, 8 Jul 2009 11:13:59 +0200 Subject: [Freeswitch-users] ClueCon 2009 Updates - July 7, 2009 In-Reply-To: References: Message-ID: <04909F90-2B3C-4796-BB85-7DB13D5D6493@halo2.pl> The GUUG will be doing something along those lines in fall 2010 in Germany. Am 08.07.2009 um 10:53 schrieb Cavalera Claudio Luigi: > I'm really missing this event! > Can't we organize a ClueCon Winter edition in Europe too? :-) > BRs, > Claudio > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Michael Collins > Sent: Tuesday, July 07, 2009 9:14 PM > To: marketing at cluecon.com > Subject: [Freeswitch-users] ClueCon 2009 Updates - July 7, 2009 > > Hello folks! > > We have a few more updates. First of all, if you haven't already > heard, we've extended the early bird sign up to go through July 21. > That's only two weeks away, so if you haven't already registered > then please call us at 877.742.CLUE and we'll get you set up. > Secondly, there are some updates on the ClueCon blog: > http://cluecon.com/blog/1 > > The breakfast and lunch menus have been posted. (Subject to change, > of course. :) Also, we have a synopsis for Irv Shapiro's talk, > entitled "Cloud Telephony" on the latest blog post. > > Can't wait to see you all in Chicago! > -Michael Collins > http://www.cluecon.com > 877.742.CLUE > > 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 Michal Bielicki Leiter der Niederlassung HaloKwadrat Sp. z o.o. Niederlassung Kleinmachnow Eingetragen im Handelsregister beim Amtsgericht Potsdam, HRB21422P Ust.Id.: DE261885536 Geschaeftsfuehrer: Aleksander Wiercinski Meiereifeld 2b, 14532 Kleinmachnow t. +49 33203 263220 f. +49 33203 263229 sip. info at halokwadrat.de e. michal.bielicki at halokwadrat.de | w. www.halokwadrat.de Hauptgesch?ftsstelle: Halo Kwadrat Sp. z o.o. ul. Polna 46/14 00-644 Warszawa, Polen EIngetragen im HRB Warszawa, KRS 0000153539 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090708/95b1a406/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/pkcs7-signature Size: 2453 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090708/95b1a406/attachment.bin From mariusz_kolo at wp.pl Wed Jul 8 04:23:50 2009 From: mariusz_kolo at wp.pl (=?ISO-8859-2?Q?Mariusz_Ko=B3odziejczyk?=) Date: Wed, 08 Jul 2009 13:23:50 +0200 Subject: [Freeswitch-users] Record_session cutting wav files In-Reply-To: <4A5324D6.3070600@wp.pl> References: <4A5324D6.3070600@wp.pl> Message-ID: <4A5481C6.2060207@wp.pl> Hello After upgrade to latest version wavs recording is OK Thanks a lot >Hello, >please try out FS >= r14143 as there were some fixes around call >recording and media bugs. >Please let us know the results. >Regards, > Tamas Mariusz Ko?odziejczyk pisze: > Hello > > I saw a strange behavior when i'm using record_session for outbound > call. Recorded file is 24:20 time length, but in logs should have > about 25:18. > > Here ma log: > > start recording about 2009-07-07 10:56:15 > 2009-07-07 10:56:15.108447 [DEBUG] mod_dptools.c:748 > sofia/internal/1062 SET [czas]=[2009-07-07-10-56-15] - variable $czas > = "2009-07-07-10-56-15" i use it in filename below > EXECUTE sofia/internal/1062 > record_session(/records/2009-07-07/069630d2-6ad4-11de-aa6e-e31e1f9431c0.10XX.07135XXXXX.2009-07-07-10-56-15.out.ISDN.wav) > > ..... > stop recording > 2009-07-07 11:21:33.291177 [DEBUG] switch_ivr_async.c:444 Stop > recording file > /records/2009-07-07/069630d2-6ad4-11de-aa6e-e31e1f9431c0.10XX.07135XXXXX.2009-07-07-10-56-15.out.ISDN.wav > > > file should have about: 25:18 time length > when i listen a file it's really cut > > My piece of dialplan: > > > > > > > > > > > > > > > data="czas=${strftime(%Y-%m-%d-%H-%M-%S)}"/> > > > data="/records/${dir}/${uuid}.${caller_id_number}.$1.${czas}.out.ISDN.wav"/> > > > > data="{origination_caller_id_number=${cti_gateway_number},effective_caller_id_number=${cti_gateway_number}}openzap/1/A/$1"/> > > > > freeswitch version: FreeSWITCH Version 1.0.trunk (14013) > linux: Linux pbx2 2.6.26-1-686 #1 SMP Fri Mar 13 18:08:45 UTC 2009 > i686 GNU/Linux > > Thanks > From mariusz_kolo at wp.pl Wed Jul 8 04:24:08 2009 From: mariusz_kolo at wp.pl (=?ISO-8859-2?Q?Mariusz_Ko=B3odziejczyk?=) Date: Wed, 08 Jul 2009 13:24:08 +0200 Subject: [Freeswitch-users] Record_session cutting wav files In-Reply-To: <4A5324D6.3070600@wp.pl> References: <4A5324D6.3070600@wp.pl> Message-ID: <4A5481D8.3090204@wp.pl> Hello After upgrade to latest version wavs recording is OK Thanks a lot >Hello, >please try out FS >= r14143 as there were some fixes around call >recording and media bugs. >Please let us know the results. >Regards, > Tamas Mariusz Ko?odziejczyk pisze: > Hello > > I saw a strange behavior when i'm using record_session for outbound > call. Recorded file is 24:20 time length, but in logs should have > about 25:18. > > Here ma log: > > start recording about 2009-07-07 10:56:15 > 2009-07-07 10:56:15.108447 [DEBUG] mod_dptools.c:748 > sofia/internal/1062 SET [czas]=[2009-07-07-10-56-15] - variable $czas > = "2009-07-07-10-56-15" i use it in filename below > EXECUTE sofia/internal/1062 > record_session(/records/2009-07-07/069630d2-6ad4-11de-aa6e-e31e1f9431c0.10XX.07135XXXXX.2009-07-07-10-56-15.out.ISDN.wav) > > ..... > stop recording > 2009-07-07 11:21:33.291177 [DEBUG] switch_ivr_async.c:444 Stop > recording file > /records/2009-07-07/069630d2-6ad4-11de-aa6e-e31e1f9431c0.10XX.07135XXXXX.2009-07-07-10-56-15.out.ISDN.wav > > > file should have about: 25:18 time length > when i listen a file it's really cut > > My piece of dialplan: > > > > > > > > > > > > > > > data="czas=${strftime(%Y-%m-%d-%H-%M-%S)}"/> > > > data="/records/${dir}/${uuid}.${caller_id_number}.$1.${czas}.out.ISDN.wav"/> > > > > data="{origination_caller_id_number=${cti_gateway_number},effective_caller_id_number=${cti_gateway_number}}openzap/1/A/$1"/> > > > > freeswitch version: FreeSWITCH Version 1.0.trunk (14013) > linux: Linux pbx2 2.6.26-1-686 #1 SMP Fri Mar 13 18:08:45 UTC 2009 > i686 GNU/Linux > > Thanks > From rupa at rupa.com Wed Jul 8 05:14:52 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Wed, 8 Jul 2009 07:14:52 -0500 Subject: [Freeswitch-users] How to get the hook state? In-Reply-To: References: <4A52D4D1.1010805@gmail.com> Message-ID: I think you can (mostly) solved this using mod_limit and some dialplan work. Something like: if you want the second call to ring, limit should be set to 2 or higher. for either VM or busy, limit should be set to 1 if limit was reached and the user wants busy, send back busy if limit was reached and the user wants VM, transfer to voicemail Remember that you have to check the limit both on outbound calls and on inbound calls so that you get the desired behavior. Also, you'll have to special handle outbound calls so they don't fail if the limit is reached (think transfer). On Tue, Jul 7, 2009 at 11:40 PM, Yehavi Bourvine wrote: > Hello, > > The problem we are trying to solve here is handling a busy state > according to the user's prefference (some want a busy to be heard, some want > the call to go to voicemail, and some want to get the second call). > > The first step is finding that an extension is busy. It would be nice in > the future to know also other states of an extension (like - not registered, > etc.). > > Thanks, __Yehavi: > > 2009/7/7 Brian West > > What are you trying to accomplish? >> >> /b >> >> On Jul 6, 2009, at 11:53 PM, Eli Hayun wrote: >> >> > Hi >> > I am a newbie in FreeSwitch and my question is: >> > When I am calling to an extension, how should I know in advance what >> > is >> > the hook status. I tried to find out a variable that can get me this >> > information but with no success. >> > any help? >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090708/306aa6bb/attachment-0001.html From brian at freeswitch.org Wed Jul 8 05:15:18 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 8 Jul 2009 07:15:18 -0500 Subject: [Freeswitch-users] Can't mute SIP channel with "receiveonly" in SDP In-Reply-To: References: Message-ID: <9BD68E4E-F7C7-4E48-AF0E-D7E46F8F6194@freeswitch.org> You could put a feature request bounty on jira. /b On Jul 8, 2009, at 1:39 AM, Artem Shiyanov wrote: > Hy all! > > With Asterisk I can mute SIP channel using re-INVITE with > "a=receiveonly" in media description. But this feature doesn't work > with Freeswitch. For sure, there is old good method: transfer both > legs to the conference room where one leg is able to listen/talk, > the other one - only to listen, but this is unwanted workaround for > me.. > So I wonder: is there any other (preferably through the SIP) way to > "mute" given SIP channel with Freeswitch? > > Thanks for all, > Artem Shiyanov From brian at freeswitch.org Wed Jul 8 05:20:08 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 8 Jul 2009 07:20:08 -0500 Subject: [Freeswitch-users] How to get the hook state? In-Reply-To: References: <4A52D4D1.1010805@gmail.com> Message-ID: continue_on_fail=user_busy (not 100% reliable because some devices won't say that the user is busy and ring the second line) facility_not_subscribed is what you'll get if they aren't registered. /b On Jul 7, 2009, at 11:40 PM, Yehavi Bourvine wrote: > Hello, > > The problem we are trying to solve here is handling a busy state > according to the user's prefference (some want a busy to be heard, > some want the call to go to voicemail, and some want to get the > second call). > > The first step is finding that an extension is busy. It would be > nice in the future to know also other states of an extension (like - > not registered, etc.). > > Thanks, __Yehavi: From maxim.tsvetov at gmail.com Wed Jul 8 05:42:03 2009 From: maxim.tsvetov at gmail.com (Maxim Tsvetov) Date: Wed, 8 Jul 2009 16:42:03 +0400 Subject: [Freeswitch-users] Freeswitch architecture Message-ID: <89c9bbf80907080542y39f2b234tff9163ea34c0b968@mail.gmail.com> Hi All Where can I get information about internal Freeswitch architecture: 1) how modules interoperates with each other (maybe using corba or com objects or something else) 2) how core interoperates with other modules 3) how javascript function is translated to internal commands. In addition if you cand send me some schemas of Freeswitch architecture that will be great. Regards, Maxim Tsvetov -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090708/7488203c/attachment.html From maxim.tsvetov at gmail.com Wed Jul 8 05:53:43 2009 From: maxim.tsvetov at gmail.com (Maxim Tsvetov) Date: Wed, 8 Jul 2009 16:53:43 +0400 Subject: [Freeswitch-users] CallID Message-ID: <89c9bbf80907080553jf7a7fbbga6c963599f4dabb2@mail.gmail.com> Hi All Can somebody explain me algorithm of assigning CallID to new calls in Freeswitch. Are they unique? If not - how frequent they become duplicated. Regards, Maxim Tsvetov -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090708/9a9f3a1b/attachment.html From brian at freeswitch.org Wed Jul 8 06:29:28 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 8 Jul 2009 08:29:28 -0500 Subject: [Freeswitch-users] CallID In-Reply-To: <89c9bbf80907080553jf7a7fbbga6c963599f4dabb2@mail.gmail.com> References: <89c9bbf80907080553jf7a7fbbga6c963599f4dabb2@mail.gmail.com> Message-ID: They are uuid's and yes they are unique. /b On Jul 8, 2009, at 7:53 AM, Maxim Tsvetov wrote: > Hi All > Can somebody explain me algorithm of assigning CallID to new calls > in Freeswitch. Are they unique? If not - how frequent they become > duplicated. > Regards, > Maxim Tsvetov From brian at freeswitch.org Wed Jul 8 06:29:50 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 8 Jul 2009 08:29:50 -0500 Subject: [Freeswitch-users] Freeswitch architecture In-Reply-To: <89c9bbf80907080542y39f2b234tff9163ea34c0b968@mail.gmail.com> References: <89c9bbf80907080542y39f2b234tff9163ea34c0b968@mail.gmail.com> Message-ID: <347F12AB-EA05-4090-A203-BDC011F02411@freeswitch.org> http://wiki.freeswitch.org /b On Jul 8, 2009, at 7:42 AM, Maxim Tsvetov wrote: > Hi All > > Where can I get information about internal Freeswitch architecture: > 1) how modules interoperates with each other (maybe using corba or > com > objects or something else) > 2) how core interoperates with other modules > 3) how javascript function is translated to internal commands. > > In addition if you cand send me some schemas of Freeswitch > architecture > that will be great. > > Regards, > Maxim Tsvetov > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.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 Jul 8 07:01:24 2009 From: Claudio.Cavalera at italtel.it (Cavalera Claudio Luigi) Date: Wed, 8 Jul 2009 16:01:24 +0200 Subject: [Freeswitch-users] ClueCon 2009 Updates - July 7, 2009 In-Reply-To: <04909F90-2B3C-4796-BB85-7DB13D5D6493@halo2.pl> Message-ID: Advertise it a bit when it will happen! :-) ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michal Bielicki Sent: Wednesday, July 08, 2009 11:14 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] ClueCon 2009 Updates - July 7, 2009 The GUUG will be doing something along those lines in fall 2010 in Germany. Am 08.07.2009 um 10:53 schrieb Cavalera Claudio Luigi: I'm really missing this event! Can't we organize a ClueCon Winter edition in Europe too? :-) BRs, Claudio ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Tuesday, July 07, 2009 9:14 PM To: marketing at cluecon.com Subject: [Freeswitch-users] ClueCon 2009 Updates - July 7, 2009 Hello folks! We have a few more updates. First of all, if you haven't already heard, we've extended the early bird sign up to go through July 21. That's only two weeks away, so if you haven't already registered then please call us at 877.742.CLUE and we'll get you set up. Secondly, there are some updates on the ClueCon blog: http://cluecon.com/blog/1 The breakfast and lunch menus have been posted. (Subject to change, of course. :) Also, we have a synopsis for Irv Shapiro's talk, entitled "Cloud Telephony" on the latest blog post. Can't wait to see you all in Chicago! -Michael Collins http://www.cluecon.com 877.742.CLUE 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 Michal Bielicki Leiter der Niederlassung HaloKwadrat Sp. z o.o. Niederlassung Kleinmachnow Eingetragen im Handelsregister beim Amtsgericht Potsdam, HRB21422P Ust.Id.: DE261885536 Geschaeftsfuehrer: Aleksander Wiercinski Meiereifeld 2b, 14532 Kleinmachnow t. +49 33203 263220 f. +49 33203 263229 sip. info at halokwadrat.de e. michal.bielicki at halokwadrat.de | w. www.halokwadrat.de Hauptgesch?ftsstelle: Halo Kwadrat Sp. z o.o. ul. Polna 46/14 00-644 Warszawa, Polen EIngetragen im HRB Warszawa, KRS 0000153539 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090708/ad9b346a/attachment-0001.html From sprice at gmail.com Wed Jul 8 07:11:02 2009 From: sprice at gmail.com (SP) Date: Wed, 8 Jul 2009 09:11:02 -0500 Subject: [Freeswitch-users] Freeswitch architecture In-Reply-To: <89c9bbf80907080542y39f2b234tff9163ea34c0b968@mail.gmail.com> References: <89c9bbf80907080542y39f2b234tff9163ea34c0b968@mail.gmail.com> Message-ID: <7e2ac3270907080711i48ebaf9coed950556e712c8aa@mail.gmail.com> http://cluecon.com/node/3 On Wed, Jul 8, 2009 at 07:42, Maxim Tsvetov wrote: > Hi All > > Where can I get information about internal Freeswitch architecture: > 1) how modules interoperates with each other (maybe using corba or com > objects or something else) > 2) how core interoperates with other modules > 3) how javascript function is translated to internal commands. > > In addition if you cand send me some schemas of Freeswitch architecture > that will be great. > > Regards, > Maxim Tsvetov > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090708/2ec4f18e/attachment.html From Prometheus001 at gmx.net Wed Jul 8 08:50:36 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Wed, 08 Jul 2009 17:50:36 +0200 Subject: [Freeswitch-users] Sangoma 108 and libpri problems - only distortion sound Message-ID: <4A54C04C.7080804@gmx.net> Hello, I installed a Sangoma A108 with openzap and libpri. Signalling (E1) works sometimes (inbound and outbound calls are connected) but not always. Sound is just distortion but connection is stable. 2 questions: 1) What is the best way to go with Sangoma? OpenZAP with libpri or without libpri? (I remember there are some timer problems in openzap when not using libpri but this might have changed) However Sangoma recommends OpenZAP on their wiki. 2) What might cause the distortion? I crosschecked the config files and had a look at the interrupts (<2k/sec). Seems to be ok. ACPI and APIC is turned on. Freeswitch starts successfully with all the channels enabled. "oz dump" shows D-Channel up. "oz libpri debug 1 all" shows debugging messages with no special warnings. Best regards Peter Some confs for 1 Channel: Wanpipe1.conf [devices] wanpipe1 = WAN_AFT_TE1, Comment [interfaces] w1g1 = wanpipe1, , TDM_VOICE_API, Comment [wanpipe1] CARD_TYPE = AFT S514CPU = A CommPort = PRI AUTO_PCISLOT = NO PCISLOT = 4 PCIBUS = 11 FE_MEDIA = E1 FE_LCODE = HDB3 FE_FRAME = CRC4 FE_LINE = 1 TE_CLOCK = NORMAL TE_REF_CLOCK = 0 TE_SIG_MODE = CCS TE_HIGHIMPEDANCE = NO LBO = 120OH FE_TXTRISTATE = NO MTU = 1500 UDPPORT = 9000 TTL = 255 IGNORE_FRONT_END = NO TDMV_SPAN = 1 TDMV_DCHAN = 16 TDMV_HW_DTMF = YES TDMV_HW_FAX_DETECT = YES [w1g1] ACTIVE_CH = ALL TDMV_HWEC = YES MTU = 80 openzap.conf [span wanpipe PRI_1] name => OpenZAP number => 1 trunk_type => e1 b-channel => 1:1-15 d-channel => 1:16 b-channel => 1:17-31 openzap.conf.xml From paparoga at mailinator.com Wed Jul 8 07:41:40 2009 From: paparoga at mailinator.com (paparoga at mailinator.com) Date: Wed, 8 Jul 2009 16:41:40 +0200 Subject: [Freeswitch-users] simple originate / bridge js Message-ID: <200907081641.40381.paparoga@mailinator.com> Hi all, I'm attempting to setup a simple alarm handling machine. It should be triggered by an external event, dial a phone number (depending on the alarm type), and play a few wav files indicating the failure happened. Using Free, up to now, I've created a simple IVR and connected it to my EXT. 118. Connecting a softphone to the ext. 1001 and dialing the ext 118 the IVR is ok. Also using the following Free command from the console all is ok: originate sofia/zz.xxx.200.29/1001 118 My softphone at ext 1001 get ringed and then connected to the IVR at ext. 118. I cannot get the same from a simple js. I tried: ===================================== session = new Session("sofia/zz.xxx.200.29/1001"); //session = new Session(); //session.originate(session, "sofia/zz.yyy.200.29/118"); session.execute("bridge", "sofia/default/118"); ===================================== and almost all possibles variations, but I'get this result: 2009-07-08 16:23:00 [NOTICE] sofia.c:3220 sofia_handle_sip_i_state() Channel [sofia/internal/1001] has been answered 2009-07-08 16:23:00 [WARNING] mod_sofia.c:2495 sofia_outgoing_channel() Cannot locate registered user 118 at default 2009-07-08 16:23:00 [NOTICE] mod_sofia.c:2591 sofia_outgoing_channel() Close Channel N/A [CS_NEW] How can I tell to the script that the EXT. 118 is an IVR and not a registered USER? By the way, I attempted also the 'transfer' function, but I get the following: 2009-07-08 16:35:43 [NOTICE] sofia.c:2782 sofia_handle_sip_i_state() Ring-Ready sofia/internal/1001! 2009-07-08 16:35:50 [NOTICE] sofia.c:3220 sofia_handle_sip_i_state() Channel [sofia/internal/1001] has been answered 2009-07-08 16:35:50 [NOTICE] switch_ivr.c:1343 switch_ivr_session_transfer() Transfer sofia/internal/1001 to XML[sofia/default/118 at default] 2009-07-08 16:35:50 [NOTICE] mod_spidermonkey.c:2994 session_destroy() Hangup sofia/internal/1001 [CS_ROUTING] [NORMAL_CLEARING] 2009-07-08 16:35:50 [NOTICE] switch_core_session.c:970 switch_core_session_thread() Session 8 (sofia/internal/1001) Ended 2009-07-08 16:35:50 [NOTICE] switch_core_session.c:972 switch_core_session_thread() Close Channel sofia/internal/1001 [CS_HANGUP] Any suggestion? Regards Kowalsky From larclap at yahoo.com Wed Jul 8 09:33:34 2009 From: larclap at yahoo.com (Lars Zeb) Date: Wed, 8 Jul 2009 09:33:34 -0700 Subject: [Freeswitch-users] Can't understand documentation Message-ID: <009301c9ffe9$d6424560$82c6d020$@com> On http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session, at the bottom of "Activating via DTMF", it states: The other party doesn't hear the DTMFs but maybe its comfort noisy is disappearing for a very short time. When that sip client *starts* a call the above dialplan forbids activating recording. Can someone explain what these two sentences mean? Thanks, Lars -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090708/243a181c/attachment.html From anthony.minessale at gmail.com Wed Jul 8 09:40:11 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 8 Jul 2009 11:40:11 -0500 Subject: [Freeswitch-users] simple originate / bridge js In-Reply-To: <200907081641.40381.paparoga@mailinator.com> References: <200907081641.40381.paparoga@mailinator.com> Message-ID: <191c3a030907080940x32dd2469w7ba5e32601bb1e9c@mail.gmail.com> in the first example you are transferring to 118 in the script you are calling a sip address but you are not supplying the domain try var my_domain = "1.2.3.4" session.execute("bridge", "sofia/default/118%" + my_domain); where you set my_domain to whatever domain your phone registered with. On Wed, Jul 8, 2009 at 9:41 AM, wrote: > Hi all, > > I'm attempting to setup a simple alarm handling machine. > > It should be triggered by an external event, dial a phone number (depending > on > the alarm type), and play a few wav files indicating the failure happened. > > Using Free, up to now, I've created a simple IVR and connected it to my > EXT. > 118. > > Connecting a softphone to the ext. 1001 and dialing the ext 118 the IVR is > ok. > > Also using the following Free command from the console all is ok: > > originate sofia/zz.xxx.200.29/1001 118 > > My softphone at ext 1001 get ringed and then connected to the IVR at ext. > 118. > > I cannot get the same from a simple js. > > I tried: > ===================================== > session = new Session("sofia/zz.xxx.200.29/1001"); > //session = new Session(); > //session.originate(session, "sofia/zz.yyy.200.29/118"); > session.execute("bridge", "sofia/default/118"); > ===================================== > and almost all possibles variations, but I'get this result: > > 2009-07-08 16:23:00 [NOTICE] sofia.c:3220 sofia_handle_sip_i_state() > Channel > [sofia/internal/1001] has been answered > 2009-07-08 16:23:00 [WARNING] mod_sofia.c:2495 sofia_outgoing_channel() > Cannot > locate registered user 118 at default > 2009-07-08 16:23:00 [NOTICE] mod_sofia.c:2591 sofia_outgoing_channel() > Close > Channel N/A [CS_NEW] > > How can I tell to the script that the EXT. 118 is an IVR and not a > registered > USER? > > By the way, I attempted also the 'transfer' function, but I get the > following: > > 2009-07-08 16:35:43 [NOTICE] sofia.c:2782 sofia_handle_sip_i_state() > Ring-Ready sofia/internal/1001! > 2009-07-08 16:35:50 [NOTICE] sofia.c:3220 sofia_handle_sip_i_state() > Channel > [sofia/internal/1001] has been answered > 2009-07-08 16:35:50 [NOTICE] switch_ivr.c:1343 > switch_ivr_session_transfer() > Transfer sofia/internal/1001 to XML[sofia/default/118 at default] > 2009-07-08 16:35:50 [NOTICE] mod_spidermonkey.c:2994 session_destroy() > Hangup > sofia/internal/1001 [CS_ROUTING] [NORMAL_CLEARING] > 2009-07-08 16:35:50 [NOTICE] switch_core_session.c:970 > switch_core_session_thread() Session 8 (sofia/internal/1001) Ended > 2009-07-08 16:35:50 [NOTICE] switch_core_session.c:972 > switch_core_session_thread() Close Channel sofia/internal/1001 [CS_HANGUP] > > > Any suggestion? > > Regards > > Kowalsky > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference 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/20090708/7dc0ff01/attachment.html From msc at freeswitch.org Wed Jul 8 09:48:08 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 8 Jul 2009 09:48:08 -0700 Subject: [Freeswitch-users] simple originate / bridge js In-Reply-To: <200907081641.40381.paparoga@mailinator.com> References: <200907081641.40381.paparoga@mailinator.com> Message-ID: <87f2f3b90907080948v27f7c1d6r7784da1d2353a2f4@mail.gmail.com> On Wed, Jul 8, 2009 at 7:41 AM, wrote: > Hi all, > > I'm attempting to setup a simple alarm handling machine. > > It should be triggered by an external event, dial a phone number (depending > on > the alarm type), and play a few wav files indicating the failure happened. > > Using Free, up to now, I've created a simple IVR and connected it to my > EXT. > 118. > > Connecting a softphone to the ext. 1001 and dialing the ext 118 the IVR is > ok. > > Also using the following Free command from the console all is ok: > > originate sofia/zz.xxx.200.29/1001 118 > > My softphone at ext 1001 get ringed and then connected to the IVR at ext. > 118. > > I cannot get the same from a simple js. > > I tried: > ===================================== > session = new Session("sofia/zz.xxx.200.29/1001"); > //session = new Session(); > //session.originate(session, "sofia/zz.yyy.200.29/118"); > session.execute("bridge", "sofia/default/118"); > ===================================== > and almost all possibles variations, but I'get this result: > > 2009-07-08 16:23:00 [NOTICE] sofia.c:3220 sofia_handle_sip_i_state() > Channel > [sofia/internal/1001] has been answered > 2009-07-08 16:23:00 [WARNING] mod_sofia.c:2495 sofia_outgoing_channel() > Cannot > locate registered user 118 at default > 2009-07-08 16:23:00 [NOTICE] mod_sofia.c:2591 sofia_outgoing_channel() > Close > Channel N/A [CS_NEW] > > How can I tell to the script that the EXT. 118 is an IVR and not a > registered > USER? It looks like 118 is defined in your dialplan so use transfer instead of bridge: session.execute("transfer", "118 XML default"); -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090708/eb3464bb/attachment-0001.html From msc at freeswitch.org Wed Jul 8 10:07:11 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 8 Jul 2009 10:07:11 -0700 Subject: [Freeswitch-users] Can't understand documentation In-Reply-To: <009301c9ffe9$d6424560$82c6d020$@com> References: <009301c9ffe9$d6424560$82c6d020$@com> Message-ID: <87f2f3b90907081007h73cc07b5s63eecd0f26710bf3@mail.gmail.com> We'll clean it up a bit, but for reference here's what they mean: #1 - the other party won't hear the DTMFs, but the DTMFs might disrupt the comfort noise generation (CNG). Perhaps instead of comfort noise the part might hear a short period of complete silence #2 - this sentence is awkwardly written but I believe it is referring to the bind_meta_app settings in the example dialplan, which are on the B leg only; if the SIP client in question is the A leg then the bind_meta_app would not be available. -MC On Wed, Jul 8, 2009 at 9:33 AM, Lars Zeb wrote: > On http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session, > at the bottom of ?Activating via DTMF?, it states: > > > > The other party doesn't hear the DTMFs but maybe its comfort noisy is > disappearing for a very short time. When that sip client *starts* a call the > above dialplan forbids activating recording. > > > > Can someone explain what these two sentences mean? > > > > Thanks, Lars > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090708/11a194f4/attachment.html From Nick.Lemberger at lkfd.net Wed Jul 8 11:24:41 2009 From: Nick.Lemberger at lkfd.net (Nick Lemberger) Date: Wed, 08 Jul 2009 13:24:41 -0500 Subject: [Freeswitch-users] LUA Event generation Message-ID: <4A549E23.2C9A.00FE.0@lkfd.net> When I create an even with mod_lua, I need to add the even variable name as the 1st argument too all of the event statements - this is not how it's listed in the examples in the Wiki, is this new and should I update the WIKI or am I doing something wrong? The way I needed to write it to get it to work looks like this: ie: http://pastebin.freeswitch.org/9654 The examples in the wiki are just missing the first argument for addHeader(), addBody() and fire(). ie: http://wiki.freeswitch.org/wiki/Mod_lua#Events -Nick From paparoga at mailinator.com Wed Jul 8 13:15:57 2009 From: paparoga at mailinator.com (paparoga at mailinator.com) Date: Wed, 8 Jul 2009 22:15:57 +0200 Subject: [Freeswitch-users] simple originate / bridge js Message-ID: <200907082215.57328.paparoga@mailinator.com> First of all I apologise for my long post. In the meantime I reworked the simple js as suggested (adding the domain or using transfer instead of bridge) but yet the script doesn't work. Let me add some other info. I just cloned the '5000' demo_ivr and reworked a little to reach my target. Next I added: into the default.xml dialplan. Now all is ok if I connect Ekiga to FreeSwitch as user 1001 (for example) and than I dial '118'. The IVR works fine. Next from the console: originate sofia/my.freeswitch.address/1001 118 I get my Ekiga Phone ringing and connecter to the ext. 118 (the ivr) That is the log: ========================================================= originate sofia/10.0.0.33/1001 118 2009-07-08 21:56:57 [NOTICE] switch_channel.c:567 switch_channel_set_name() New Channel sofia/internal/1001 [7e660b20-6bf9-11de-b2fa-f3963e050c84] 2009-07-08 21:56:57 [NOTICE] sofia.c:2782 sofia_handle_sip_i_state() Ring-Ready sofia/internal/1001! 2009-07-08 21:57:00 [NOTICE] sofia.c:3220 sofia_handle_sip_i_state() Channel [sofia/internal/1001] has been answered 2009-07-08 21:57:00 [NOTICE] switch_ivr.c:1343 switch_ivr_session_transfer() Transfer sofia/internal/1001 to XML[118 at default] API CALL [originate(sofia/10.0.0.33/1001 118)] output: +OK 7e660b20-6bf9-11de-b2fa-f3963e050c84 freeswitch at Linux61> 2009-07-08 21:57:00 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing FreeSWITCH->118 in context default 2009-07-08 21:57:02 [WARNING] switch_core_file.c:119 switch_core_perform_file_open() Sample rate doesn't match 2009-07-08 21:57:15 [WARNING] switch_core_file.c:119 switch_core_perform_file_open() Sample rate doesn't match 2009-07-08 21:57:21 [NOTICE] sofia.c:3279 sofia_handle_sip_i_state() Hangup sofia/internal/1001 [CS_EXECUTE] [NORMAL_CLEARING] 2009-07-08 21:57:25 [NOTICE] switch_core_session.c:970 switch_core_session_thread() Session 33 (sofia/internal/1001) Ended 2009-07-08 21:57:25 [NOTICE] switch_core_session.c:972 switch_core_session_thread() Close Channel sofia/internal/1001 [CS_HANGUP] ==================================================== Using the js instead it looks like if the bridge commands looks for a REGISTERED user at ext. 118, and so it fails. Using the transfer option, as suggested, the Ekiga user (1001) get to be connected to the ext 118, bur the connection drops immediately after. ==================================================== freeswitch at Linux61> jsrun alarm.js API CALL [jsrun(alarm.js)] output: OK freeswitch at Linux61> 2009-07-08 22:03:47 [NOTICE] switch_channel.c:567 switch_channel_set_name() New Channel sofia/internal/1001 [7267f936-6bfa-11de-b2fa-f3963e050c84] 2009-07-08 22:03:47 [NOTICE] sofia.c:2782 sofia_handle_sip_i_state() Ring-Ready sofia/internal/1001! 2009-07-08 22:03:51 [NOTICE] sofia.c:3220 sofia_handle_sip_i_state() Channel [sofia/internal/1001] has been answered 2009-07-08 22:03:51 [NOTICE] switch_ivr.c:1343 switch_ivr_session_transfer() Transfer sofia/internal/1001 to XML[118 at default] 2009-07-08 22:03:51 [NOTICE] mod_spidermonkey.c:2994 session_destroy() Hangup sofia/internal/1001 [CS_ROUTING] [NORMAL_CLEARING] 2009-07-08 22:03:51 [NOTICE] switch_core_session.c:970 switch_core_session_thread() Session 35 (sofia/internal/1001) Ended 2009-07-08 22:03:51 [NOTICE] switch_core_session.c:972 switch_core_session_thread() Close Channel sofia/internal/1001 [CS_HANGUP] ================================================== May be I'm on the wrong way, but this is my target: 1) From the external Perl freeswitch interface call a simple js 2) Make this script dial a sip/pstn phone number at the assistance location 3) Connect the just dialled assistance location to the IVR at ext 118 and let the support people hear some info about the raising fault. Thanks in advance for any suggestion. Roberto From anthony.minessale at gmail.com Wed Jul 8 13:29:46 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 8 Jul 2009 15:29:46 -0500 Subject: [Freeswitch-users] Less Than a Month Until ClueCon Message-ID: <191c3a030907081329q53ec3609i37b9ee87a31c9247@mail.gmail.com> Hi everyone, We are less than a month away from ClueCon 2009 and I would like to urge anyone who is considering attending to sign up ASAP to make sure you are properly counted in the food totals and the early bird pricing. Bring your laptops and all your gizmos and get ready to dive into telephony for 3 fun-filled days. We will be having FULL BREAKFAST on the first 2 mornings (continental on day 3) FULL LUNCH, and OPEN BAR for 2 hours the first 2 nights for ALL atendees included in your attendance fee. We also will be giving away several prizes provided by our various sponsors. Every paid registration gives you a chance to win one of many goodies such as phones/tdm cards etc. REGISTER NOW http://www.cluecon.com or CALL (877) 742 -CLUE or INSTALL FreeSWITCH and dial 5000 and choose the "register for cluecon" option on the ivr. or E-MAIL marketing at cluecon.com to discuss sponsoring ot participating in helping out with logistics etc for a reduced attendance fee. -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference 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/20090708/88206d6e/attachment.html From diego.viola at gmail.com Wed Jul 8 13:32:34 2009 From: diego.viola at gmail.com (Diego Viola) Date: Wed, 8 Jul 2009 16:32:34 -0400 Subject: [Freeswitch-users] FreeSWITCH article for OSnews Message-ID: <86a32abc0907081332m3e25c1c7x73d8753e8b1cc616@mail.gmail.com> Hey guys. I wrote the OSnews staff about the possibility to post some FreeSWITCH articles in the OSnews site, as they have published some Asterisk articles before, I thought that it would be nice to post something about FreeSWITCH as well, since it deserves more attention. This is what they said: "I don?t think any of the staff have any knowledge on this topic to write an article, so the best we could do is a page 2 item. As I see it, there?s two things you can do here: 1. Send the item to us using the "Submit News" link at the top of the page, it will appear in the back end for all of us to see and one of the staff may pick it up. 2. Write an article about FreeSWITCH, what it is, and how it differs from Asterisk using your knowledge an submit it in the usual manner, you?re much more likely to get on the front page then. Kind regards, Kroc." So if you guys are interested and want to help me to write a FreeSWITCH article, maybe we could send them so they publish it there. Diego -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090708/3c5073d6/attachment.html From anthony.minessale at gmail.com Wed Jul 8 13:42:57 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 8 Jul 2009 15:42:57 -0500 Subject: [Freeswitch-users] simple originate / bridge js In-Reply-To: <200907082215.57328.paparoga@mailinator.com> References: <200907082215.57328.paparoga@mailinator.com> Message-ID: <191c3a030907081342l3275021k1025900592221e20@mail.gmail.com> you also have to set session.setAutoHangup(0); or it will hangup as soon as it exits the script. On Wed, Jul 8, 2009 at 3:15 PM, wrote: > First of all I apologise for my long post. > > In the meantime I reworked the simple js as suggested (adding the domain or > using transfer instead of bridge) but yet the script doesn't work. > > Let me add some other info. > > I just cloned the '5000' demo_ivr and reworked a little to reach my target. > > Next I added: > > > > > > > > > > > into the default.xml dialplan. > > Now all is ok if I connect Ekiga to FreeSwitch as user 1001 (for example) > and > than I dial '118'. > > The IVR works fine. > > Next from the console: > > originate sofia/my.freeswitch.address/1001 118 > > I get my Ekiga Phone ringing and connecter to the ext. 118 (the ivr) > > That is the log: > ========================================================= > originate sofia/10.0.0.33/1001 118 > > 2009-07-08 21:56:57 [NOTICE] switch_channel.c:567 switch_channel_set_name() > New Channel sofia/internal/1001 [7e660b20-6bf9-11de-b2fa-f3963e050c84] > 2009-07-08 21:56:57 [NOTICE] sofia.c:2782 sofia_handle_sip_i_state() > Ring-Ready sofia/internal/1001! > 2009-07-08 21:57:00 [NOTICE] sofia.c:3220 sofia_handle_sip_i_state() > Channel > [sofia/internal/1001] has been answered > 2009-07-08 21:57:00 [NOTICE] switch_ivr.c:1343 > switch_ivr_session_transfer() > Transfer sofia/internal/1001 to XML[118 at default] > API CALL [originate(sofia/10.0.0.33/1001 118)] output: > +OK 7e660b20-6bf9-11de-b2fa-f3963e050c84 > > freeswitch at Linux61> 2009-07-08 21:57:00 [INFO] mod_dialplan_xml.c:233 > dialplan_hunt() Processing FreeSWITCH->118 in context default > 2009-07-08 21:57:02 [WARNING] switch_core_file.c:119 > switch_core_perform_file_open() Sample rate doesn't match > 2009-07-08 21:57:15 [WARNING] switch_core_file.c:119 > switch_core_perform_file_open() Sample rate doesn't match > 2009-07-08 21:57:21 [NOTICE] sofia.c:3279 sofia_handle_sip_i_state() Hangup > sofia/internal/1001 [CS_EXECUTE] [NORMAL_CLEARING] > 2009-07-08 21:57:25 [NOTICE] switch_core_session.c:970 > switch_core_session_thread() Session 33 (sofia/internal/1001) Ended > 2009-07-08 21:57:25 [NOTICE] switch_core_session.c:972 > switch_core_session_thread() Close Channel sofia/internal/1001 [CS_HANGUP] > ==================================================== > > Using the js instead it looks like if the bridge commands looks for a > REGISTERED user at ext. 118, and so it fails. > > Using the transfer option, as suggested, the Ekiga user (1001) get to be > connected to the ext 118, bur the connection drops immediately after. > > ==================================================== > freeswitch at Linux61> jsrun alarm.js > API CALL [jsrun(alarm.js)] output: > OK > > freeswitch at Linux61> 2009-07-08 22:03:47 [NOTICE] switch_channel.c:567 > switch_channel_set_name() New Channel sofia/internal/1001 > [7267f936-6bfa-11de-b2fa-f3963e050c84] > 2009-07-08 22:03:47 [NOTICE] sofia.c:2782 sofia_handle_sip_i_state() > Ring-Ready sofia/internal/1001! > 2009-07-08 22:03:51 [NOTICE] sofia.c:3220 sofia_handle_sip_i_state() > Channel > [sofia/internal/1001] has been answered > 2009-07-08 22:03:51 [NOTICE] switch_ivr.c:1343 > switch_ivr_session_transfer() > Transfer sofia/internal/1001 to XML[118 at default] > 2009-07-08 22:03:51 [NOTICE] mod_spidermonkey.c:2994 session_destroy() > Hangup > sofia/internal/1001 [CS_ROUTING] [NORMAL_CLEARING] > 2009-07-08 22:03:51 [NOTICE] switch_core_session.c:970 > switch_core_session_thread() Session 35 (sofia/internal/1001) Ended > 2009-07-08 22:03:51 [NOTICE] switch_core_session.c:972 > switch_core_session_thread() Close Channel sofia/internal/1001 [CS_HANGUP] > ================================================== > > May be I'm on the wrong way, but this is my target: > > 1) From the external Perl freeswitch interface call a simple js > 2) Make this script dial a sip/pstn phone number at the assistance location > 3) Connect the just dialled assistance location to the IVR at ext 118 and > let > the support people hear some info about the raising fault. > > Thanks in advance for any suggestion. > > > > Roberto > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference 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/20090708/6d62c035/attachment-0001.html From larclap at yahoo.com Wed Jul 8 14:37:51 2009 From: larclap at yahoo.com (Lars Zeb) Date: Wed, 8 Jul 2009 14:37:51 -0700 Subject: [Freeswitch-users] Error in my dialplan Message-ID: <00ff01ca0014$58a20700$09e61500$@com> I receive an error on an inbound call from my dialplan. I don't have a clue what it means. Can someone help? from log: 2009-07-08 09:54:50.172590 [DEBUG] switch_core_state_machine.c:78 sofia/external/+13105551212 at 66.53.188.187 Standard ROUTING 2009-07-08 09:54:50.172590 [INFO] mod_dialplan_xml.c:310 Processing +13105551212->1000 in context default .... Dialplan: sofia/external/+13105551212 at 66.53.188.187 parsing [default->Local_Extension_Lars] continue=false Dialplan: sofia/external/+13105551212 at 66.53.188.187 Regex (PASS) [Local_Extension_Lars] destination_number(1000) =~ /^(100[0-9])$/ break=on-false Dialplan: sofia/external/+13105551212 at 66.53.188.187 Action set(dialed_ext=1000) Dialplan: sofia/external/+13105551212 at 66.53.188.187 Action export(dialed_ext=1000) 2009-07-08 09:54:50.173586 [ERR] switch_regex.c:101 COMPILE ERROR: 1 [nothing to repeat][^+13105551212$] Dialplan: sofia/external/+13105551212 at 66.53.188.187 Regex (FAIL) [Local_Extension_Lars] destination_number(1000) =~ /^+13105551212$/ break=on-false Dialplan: sofia/external/+13104647614 at 66.53.188.187 ANTI-Action bind_meta_app(1 b s execute_extension::dx XML features) Dialplan: sofia/external/+13104647614 at 66.53.188.187 ANTI-Action bind_meta_app(2 b s record_session::/usr/local/freeswitch/recordings/${caller_id_number}.${strft ime(%Y-%m-%d-%H-%M-%S)}.wav) .... Caller-Dialplan: [XML] Caller-Caller-ID-Name: [+13105551212] Caller-Caller-ID-Number: [+13105551212] My dialplan: ... .... Thanks, Lars -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090708/037e29a1/attachment.html From brian at freeswitch.org Wed Jul 8 14:46:44 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 8 Jul 2009 16:46:44 -0500 Subject: [Freeswitch-users] Error in my dialplan In-Reply-To: <00ff01ca0014$58a20700$09e61500$@com> References: <00ff01ca0014$58a20700$09e61500$@com> Message-ID: <92324E8F-B612-4328-8F30-197562963EB1@freeswitch.org> You have to escape the + with \+ /b On Jul 8, 2009, at 4:37 PM, Lars Zeb wrote: > 2009-07-08 09:54:50.173586 [ERR] switch_regex.c:101 COMPILE ERROR: 1 > [nothing to repeat][^+13105551212$] -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090708/b9154a63/attachment.html From rupa at rupa.com Wed Jul 8 14:48:42 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Wed, 8 Jul 2009 16:48:42 -0500 Subject: [Freeswitch-users] Error in my dialplan In-Reply-To: <92324E8F-B612-4328-8F30-197562963EB1@freeswitch.org> References: <00ff01ca0014$58a20700$09e61500$@com> <92324E8F-B612-4328-8F30-197562963EB1@freeswitch.org> Message-ID: the problem is the + is coming from the network... On Wed, Jul 8, 2009 at 4:46 PM, Brian West wrote: > You have to escape the + with \+ > /b > > On Jul 8, 2009, at 4:37 PM, Lars Zeb wrote: > > 2009-07-08 09:54:50.173586 [ERR] switch_regex.c:101 COMPILE ERROR: 1 > [nothing to repeat][^+13105551212$] > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090708/6b1f8a80/attachment.html From brian at freeswitch.org Wed Jul 8 14:50:53 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 8 Jul 2009 16:50:53 -0500 Subject: [Freeswitch-users] Error in my dialplan In-Reply-To: References: <00ff01ca0014$58a20700$09e61500$@com> <92324E8F-B612-4328-8F30-197562963EB1@freeswitch.org> Message-ID: Ah yes this line We have since removed that ability to login without password from the default configs. /b On Jul 8, 2009, at 4:48 PM, Rupa Schomaker wrote: > the problem is the + is coming from the network... > > On Wed, Jul 8, 2009 at 4:46 PM, Brian West > wrote: > You have to escape the + with \+ > > /b > > On Jul 8, 2009, at 4:37 PM, Lars Zeb wrote: > >> 2009-07-08 09:54:50.173586 [ERR] switch_regex.c:101 COMPILE ERROR: >> 1 [nothing to repeat][^+13105551212$] > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > -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 From larclap at yahoo.com Wed Jul 8 15:14:37 2009 From: larclap at yahoo.com (Lars Zeb) Date: Wed, 8 Jul 2009 15:14:37 -0700 Subject: [Freeswitch-users] Error in my dialplan In-Reply-To: References: <00ff01ca0014$58a20700$09e61500$@com> <92324E8F-B612-4328-8F30-197562963EB1@freeswitch.org> Message-ID: <011901ca0019$7b2f72d0$718e5870$@com> Still lost. What is the solution? 1) Remove the ability to login without password (and the comparison between destination_number and ${caller_id_number}, 2) Create a condition which strips the + sign and creates a new variable like caller_id_number, or 3) ??? > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch- > users-bounces at lists.freeswitch.org] On Behalf Of Brian West > Sent: Wednesday, July 08, 2009 2:51 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Error in my dialplan > > Ah yes this line > > We have since removed that ability to login without password from the > default configs. > > /b > > On Jul 8, 2009, at 4:48 PM, Rupa Schomaker wrote: > > > the problem is the + is coming from the network... > > > > On Wed, Jul 8, 2009 at 4:46 PM, Brian West > > wrote: > > You have to escape the + with \+ > > > > /b > > > > On Jul 8, 2009, at 4:37 PM, Lars Zeb wrote: > > > >> 2009-07-08 09:54:50.173586 [ERR] switch_regex.c:101 COMPILE ERROR: > >> 1 [nothing to repeat][^+13105551212$] > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > > http://www.freeswitch.org > > > > > > > > > > -- > > -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 > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From rupa at rupa.com Wed Jul 8 15:37:19 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Wed, 8 Jul 2009 17:37:19 -0500 Subject: [Freeswitch-users] Error in my dialplan In-Reply-To: <011901ca0019$7b2f72d0$718e5870$@com> References: <00ff01ca0014$58a20700$09e61500$@com> <92324E8F-B612-4328-8F30-197562963EB1@freeswitch.org> <011901ca0019$7b2f72d0$718e5870$@com> Message-ID: Your choice, but I'd suggest 1 -- you don't want to trust the network callerid for authenticating voicemail access. On Wed, Jul 8, 2009 at 5:14 PM, Lars Zeb wrote: > Still lost. What is the solution? 1) Remove the ability to login without > password (and the comparison between destination_number and > ${caller_id_number}, 2) Create a condition which strips the + sign and > creates a new variable like caller_id_number, or 3) ??? > > > -----Original Message----- > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch- > > users-bounces at lists.freeswitch.org] On Behalf Of Brian West > > Sent: Wednesday, July 08, 2009 2:51 PM > > To: freeswitch-users at lists.freeswitch.org > > Subject: Re: [Freeswitch-users] Error in my dialplan > > > > Ah yes this line > > > > We have since removed that ability to login without password from the > > default configs. > > > > /b > > > > On Jul 8, 2009, at 4:48 PM, Rupa Schomaker wrote: > > > > > the problem is the + is coming from the network... > > > > > > On Wed, Jul 8, 2009 at 4:46 PM, Brian West > > > wrote: > > > You have to escape the + with \+ > > > > > > /b > > > > > > On Jul 8, 2009, at 4:37 PM, Lars Zeb wrote: > > > > > >> 2009-07-08 09:54:50.173586 [ERR] switch_regex.c:101 COMPILE ERROR: > > >> 1 [nothing to repeat][^+13105551212$] > > > > > > > > > _______________________________________________ > > > Freeswitch-users mailing list > > > Freeswitch-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > > users > > > http://www.freeswitch.org > > > > > > > > > > > > > > > -- > > > -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 > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > > users > > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090708/74dedc87/attachment.html From brian at freeswitch.org Wed Jul 8 15:47:19 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 8 Jul 2009 17:47:19 -0500 Subject: [Freeswitch-users] Error in my dialplan In-Reply-To: References: <00ff01ca0014$58a20700$09e61500$@com> <92324E8F-B612-4328-8F30-197562963EB1@freeswitch.org> <011901ca0019$7b2f72d0$718e5870$@com> Message-ID: <45A1B0D7-5018-4989-9ED5-64E738EB1755@freeswitch.org> Hence the reason I removed it from the defaults. :p /b On Jul 8, 2009, at 5:37 PM, Rupa Schomaker wrote: > Your choice, but I'd suggest 1 -- you don't want to trust the > network callerid for authenticating voicemail access. From msc at freeswitch.org Wed Jul 8 16:31:31 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 8 Jul 2009 16:31:31 -0700 Subject: [Freeswitch-users] FreeSWITCH article for OSnews In-Reply-To: <86a32abc0907081332m3e25c1c7x73d8753e8b1cc616@mail.gmail.com> References: <86a32abc0907081332m3e25c1c7x73d8753e8b1cc616@mail.gmail.com> Message-ID: <87f2f3b90907081631x563dc073n10235c5a0f24c81c@mail.gmail.com> Thanks for taking the initiative. I'll catch up with you offline and we'll work up a gameplan. -MC On Wed, Jul 8, 2009 at 1:32 PM, Diego Viola wrote: > Hey guys. > > I wrote the OSnews staff about the possibility to post some FreeSWITCH > articles in the OSnews site, as they have published some Asterisk articles > before, I thought that it would be nice to post something about FreeSWITCH > as well, since it deserves more attention. > > This is what they said: > > "I don?t think any of the staff have any knowledge on this topic to write > an article, so the best we could do is a page 2 item. As I see it, there?s > two things you can do here: > > 1. Send the item to us using the "Submit News" link at the top of the page, > it will appear in the back end for all of us to see and one of the staff may > pick it up. > > 2. Write an article about FreeSWITCH, what it is, and how it differs from > Asterisk using your knowledge an submit it in the usual manner, you?re much > more likely to get on the front page then. > > Kind regards, > Kroc." > > So if you guys are interested and want to help me to write a FreeSWITCH > article, maybe we could send them so they publish it there. > > Diego > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090708/10ef69eb/attachment.html From kees at mroffice.org Wed Jul 8 18:21:03 2009 From: kees at mroffice.org (Kees Varekamp) Date: Thu, 9 Jul 2009 13:21:03 +1200 Subject: [Freeswitch-users] play file and stop play file on 2 channels Message-ID: <98d38dcf0907081821x797dc2b9q7ff9f1a9e64a34af@mail.gmail.com> Hi there, I am trying to play a file to both channels of a bridged conversation through a socket. This can be done with: api uuid_broadcast both However, I would like to be able to stop playing the file as well. This can be done with: api uuid_displace start 3600 mux and api uuid_displace stop But that one seems to work on only one channel. The other channel can mix its output but can not hear the file. Are there any other options? Thanks, Kees -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090709/d9b834cc/attachment.html From mashudiflexi at telkom.co.id Wed Jul 8 19:35:04 2009 From: mashudiflexi at telkom.co.id (mashudi) Date: Thu, 09 Jul 2009 09:35:04 +0700 Subject: [Freeswitch-users] FreeSwitch & Sangoma Q.SIG Message-ID: <4A555758.9000006@telkom.co.id> Dear All, do FreeSwitch and Sangoma card can support Q.SIG signalling protocol? thank you in advance. best regards, mashudi ==================================== Mau GRATIS TELPON LOKAL, DISCOUNT 50% SMS, DISCOUNT 20% SLJJ, dan DISCOUNT FLEXI MILIS? Ikuti Dahsyatnya FLEXI KOMUNITAS. Ketik CREATE[NAMA GRUP], sms ke 345. Contoh: CREATE SMU2, sms ke 345. Informasi selanjutnya: - hubungi 147 - http://www.telkomflexi.com - ketik INFO, sms ke 345. From krivushinme at rn-inform.tomsk.ru Thu Jul 9 00:41:26 2009 From: krivushinme at rn-inform.tomsk.ru (=?utf-8?b?0JrRgNC40LLRg9GI0LjQvSDQnNC40YXQsNC40Ls=?=) Date: Thu, 9 Jul 2009 14:41:26 +0700 Subject: [Freeswitch-users] Conference, ask to unmute Message-ID: <200907091441.26456.krivushinme@rn-inform.tomsk.ru> Hello! Is any ability to ask to unmute in conference? -- ? ?????????, ???????? ?????? ??????? ?????????? ?????? ????????????????, ??? "??-??????" ?????? ? ?.??????, ?. ????? ???. +7 913 865 78 66 icq: 218 744 127 xmpp: KrivushinME at jabber.ru mail: KrivushinME at rn-inform.tomsk.ru ?????????? ????: 86 099 192726 From krivushinme at rn-inform.tomsk.ru Thu Jul 9 02:25:26 2009 From: krivushinme at rn-inform.tomsk.ru (=?utf-8?b?0JrRgNC40LLRg9GI0LjQvSDQnNC40YXQsNC40Ls=?=) Date: Thu, 9 Jul 2009 16:25:26 +0700 Subject: [Freeswitch-users] Conference, ask to unmute Message-ID: <200907091625.26630.krivushinme@rn-inform.tomsk.ru> Hello! Is any ability to ask to unmute in conference? -- ? ?????????, ???????? ?????? ??????? ?????????? ?????? ????????????????, ??? "??-??????" ?????? ? ?.??????, ?. ????? ???. +7 913 865 78 66 icq: 218 744 127 xmpp: KrivushinME at jabber.ru mail: KrivushinME at rn-inform.tomsk.ru ?????????? ????: 86 099 192726 From anthony.minessale at gmail.com Thu Jul 9 06:55:03 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 9 Jul 2009 08:55:03 -0500 Subject: [Freeswitch-users] play file and stop play file on 2 channels In-Reply-To: <98d38dcf0907081821x797dc2b9q7ff9f1a9e64a34af@mail.gmail.com> References: <98d38dcf0907081821x797dc2b9q7ff9f1a9e64a34af@mail.gmail.com> Message-ID: <191c3a030907090655x574907d7td2f0403e7408a66f@mail.gmail.com> you can use the first way then send the "break" api command to each leg individually On Wed, Jul 8, 2009 at 8:21 PM, Kees Varekamp wrote: > Hi there, > > I am trying to play a file to both channels of a bridged conversation > through a socket. This can be done with: > > api uuid_broadcast both > > However, I would like to be able to stop playing the file as well. This can > be done with: > > api uuid_displace start 3600 mux > > and > > api uuid_displace stop > > But that one seems to work on only one channel. The other channel can mix > its output but can not hear the file. Are there any other options? > > Thanks, > > Kees > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference 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/20090709/43d0f2e2/attachment-0001.html From larclap at yahoo.com Thu Jul 9 07:08:31 2009 From: larclap at yahoo.com (Lars Zeb) Date: Thu, 9 Jul 2009 07:08:31 -0700 Subject: [Freeswitch-users] Documentation error? Message-ID: <003801ca009e$bda23f00$38e6bd00$@com> In http://wiki.freeswitch.org/wiki/Mod_cidlookup, at the third line after CLI, it states: which would the options from your config file: Should this be?: which would use the options from your config file: Thanks, Lars -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090709/7f38892b/attachment.html From brian at freeswitch.org Thu Jul 9 07:11:42 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 9 Jul 2009 09:11:42 -0500 Subject: [Freeswitch-users] Documentation error? In-Reply-To: <003801ca009e$bda23f00$38e6bd00$@com> References: <003801ca009e$bda23f00$38e6bd00$@com> Message-ID: <9CB32C44-9C6D-4D42-8ED4-FD395A761968@freeswitch.org> If it makes better sense to you then please login and fix it! ;) /b On Jul 9, 2009, at 9:08 AM, Lars Zeb wrote: > In http://wiki.freeswitch.org/wiki/Mod_cidlookup, at the third line > after CLI, it states: > > which would the options from your config file: > > Should this be?: > > which would use the options from your config file: > > Thanks, Lars -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090709/3b8f0c5c/attachment.html From larclap at yahoo.com Thu Jul 9 07:43:23 2009 From: larclap at yahoo.com (Lars Zeb) Date: Thu, 9 Jul 2009 07:43:23 -0700 Subject: [Freeswitch-users] Documentation error? In-Reply-To: <9CB32C44-9C6D-4D42-8ED4-FD395A761968@freeswitch.org> References: <003801ca009e$bda23f00$38e6bd00$@com> <9CB32C44-9C6D-4D42-8ED4-FD395A761968@freeswitch.org> Message-ID: <004901ca00a3$9c653c70$d52fb550$@com> Done. Not knowing much, I'm reluctant to make changes without asking. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Thursday, July 09, 2009 7:12 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Documentation error? If it makes better sense to you then please login and fix it! ;) /b On Jul 9, 2009, at 9:08 AM, Lars Zeb wrote: In http://wiki.freeswitch.org/wiki/Mod_cidlookup, at the third line after CLI, it states: which would the options from your config file: Should this be?: which would use the options from your config file: Thanks, Lars -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090709/2df62404/attachment.html From msc at freeswitch.org Thu Jul 9 08:20:38 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 9 Jul 2009 08:20:38 -0700 Subject: [Freeswitch-users] Conference, ask to unmute In-Reply-To: <200907091441.26456.krivushinme@rn-inform.tomsk.ru> References: <200907091441.26456.krivushinme@rn-inform.tomsk.ru> Message-ID: <87f2f3b90907090820g2edcc38oa2a1a25c2c982b64@mail.gmail.com> 2009/7/9 ???????? ?????? > Hello! > > Is any ability to ask to unmute in conference? > Not sure if I understand the question. Are you talking about the caller pressing zero to mute/unmute? -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090709/934bb104/attachment.html From mike at jerris.com Thu Jul 9 08:28:45 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 9 Jul 2009 11:28:45 -0400 Subject: [Freeswitch-users] Conference, ask to unmute In-Reply-To: <87f2f3b90907090820g2edcc38oa2a1a25c2c982b64@mail.gmail.com> References: <200907091441.26456.krivushinme@rn-inform.tomsk.ru> <87f2f3b90907090820g2edcc38oa2a1a25c2c982b64@mail.gmail.com> Message-ID: mute/unmute is a toggle. Mike On Jul 9, 2009, at 11:20 AM, Michael Collins wrote: > > > 2009/7/9 ???????? ?????? inform.tomsk.ru> > Hello! > > Is any ability to ask to unmute in conference? > > Not sure if I understand the question. Are you talking about the > caller pressing zero to mute/unmute? > > -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/20090709/8404bee1/attachment.html From krivushinme at rn-inform.tomsk.ru Thu Jul 9 08:47:10 2009 From: krivushinme at rn-inform.tomsk.ru (=?utf-8?b?0JrRgNC40LLRg9GI0LjQvSDQnNC40YXQsNC40Ls=?=) Date: Thu, 9 Jul 2009 22:47:10 +0700 Subject: [Freeswitch-users] Conference, ask to unmute In-Reply-To: <87f2f3b90907090820g2edcc38oa2a1a25c2c982b64@mail.gmail.com> References: <200907091441.26456.krivushinme@rn-inform.tomsk.ru> <87f2f3b90907090820g2edcc38oa2a1a25c2c982b64@mail.gmail.com> Message-ID: <200907092247.10336.krivushinme@rn-inform.tomsk.ru> My guys want to work with operator - I wrote an WEB-face for conferencing. And he wants to mute all participants, and give voice by order. May be say caller_id on press any button to all conference. I think to make execute_application + say caller_id. I will try to introspect channel vars in "exec_app" in conference context tomorrow. Finally - we always has esl to make anything we want! : ) -- ? ?????????, ???????? ?????? ??????? ?????????? ?????? ????????????????, ??? "??-??????" ?????? ? ?.??????, ?. ????? ???. +7 913 865 78 66 icq: 218 744 127 xmpp: KrivushinME at jabber.ru mail: KrivushinME at rn-inform.tomsk.ru ?????????? ????: 86 099 192726 From rupa at rupa.com Thu Jul 9 08:56:23 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Thu, 9 Jul 2009 10:56:23 -0500 Subject: [Freeswitch-users] Documentation error? In-Reply-To: <004901ca00a3$9c653c70$d52fb550$@com> References: <003801ca009e$bda23f00$38e6bd00$@com> <9CB32C44-9C6D-4D42-8ED4-FD395A761968@freeswitch.org> <004901ca00a3$9c653c70$d52fb550$@com> Message-ID: Thanks. At least a few of us monitor every change to the wiki. I especially monitor stuff related to my modules. So... feel free to make changes, we can always roll back to an earlier version of the page or clarify further. The wiki is a community resource, everyone should feel like they can make (reasonable) changes to it. On Thu, Jul 9, 2009 at 9:43 AM, Lars Zeb wrote: > Done. > > > > Not knowing much, I?m reluctant to make changes without asking. > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Brian West > *Sent:* Thursday, July 09, 2009 7:12 AM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Documentation error? > > > > If it makes better sense to you then please login and fix it! ;) > > > > /b > > > > On Jul 9, 2009, at 9:08 AM, Lars Zeb wrote: > > > > In http://wiki.freeswitch.org/wiki/Mod_cidlookup, at the third line after > CLI, it states: > > > > which would the options from your config file: > > > > Should this be?: > > > > which would use the options from your config file: > > > > Thanks, Lars > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090709/549f01c5/attachment.html From anthony.minessale at gmail.com Thu Jul 9 09:01:29 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 9 Jul 2009 11:01:29 -0500 Subject: [Freeswitch-users] Conference, ask to unmute In-Reply-To: <200907092247.10336.krivushinme@rn-inform.tomsk.ru> References: <200907091441.26456.krivushinme@rn-inform.tomsk.ru> <87f2f3b90907090820g2edcc38oa2a1a25c2c982b64@mail.gmail.com> <200907092247.10336.krivushinme@rn-inform.tomsk.ru> Message-ID: <191c3a030907090901q2c8fdf0at9c1238b54b59bda1@mail.gmail.com> conference xyz mute all On Thu, Jul 9, 2009 at 10:47 AM, ???????? ?????? < krivushinme at rn-inform.tomsk.ru> wrote: > My guys want to work with operator - I wrote an WEB-face for conferencing. > And he wants to mute all participants, and give voice by order. > May be say caller_id on press any button to all conference. > I think to make execute_application + say caller_id. I will try to > introspect > channel vars in "exec_app" in conference context tomorrow. > > Finally - we always has esl to make anything we want! : ) > > -- > ? ?????????, ???????? ?????? > ??????? ?????????? ?????? ????????????????, > ??? "??-??????" ?????? ? ?.??????, > ?. ????? ???. +7 913 865 78 66 > icq: 218 744 127 > xmpp: KrivushinME at jabber.ru > mail: KrivushinME at rn-inform.tomsk.ru > ?????????? ????: 86 099 192726 > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference 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/20090709/c2ac7e2c/attachment.html From pjintheusa at gmail.com Thu Jul 9 09:06:33 2009 From: pjintheusa at gmail.com (Phillip Jones) Date: Thu, 9 Jul 2009 12:06:33 -0400 Subject: [Freeswitch-users] mod_say_en directory location Message-ID: <367751820907090906if8493dfoc927b07f8bc86cfd@mail.gmail.com> Hi there, I have a very simply script that speaks back some digits, as so: session:execute("say", "en number iterated 1234"); However, to get this to work successfully I have had to move the 'digits' directory to: C:\Program Files (x86)\Freeswitch\sounds\en from the default: C:\Program Files (x86)\Freeswitch\sounds\en\us\callie\digits\8000 This is a clean install of FreeSWITCH - so I am wondering why I needed to do this, what have not configured correctly? As you can see I am using windows with a resent build (3 days) from svn. Any help appreciated. Thanks Phillip Jones -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090709/0d39366d/attachment.html From msc at freeswitch.org Thu Jul 9 09:07:47 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 9 Jul 2009 09:07:47 -0700 Subject: [Freeswitch-users] Documentation error? In-Reply-To: References: <003801ca009e$bda23f00$38e6bd00$@com> <9CB32C44-9C6D-4D42-8ED4-FD395A761968@freeswitch.org> <004901ca00a3$9c653c70$d52fb550$@com> Message-ID: <87f2f3b90907090907p17aea315ob3d5ade9423c0005@mail.gmail.com> On Thu, Jul 9, 2009 at 8:56 AM, Rupa Schomaker wrote: > Thanks. > > At least a few of us monitor every change to the wiki. I especially > monitor stuff related to my modules. > > So... feel free to make changes, we can always roll back to an earlier > version of the page or clarify further. > > The wiki is a community resource, everyone should feel like they can make > (reasonable) changes to it. Also, several of us check the changes each day so if anything really goofy happens we can always roll it back. Feel free to make changes, especially obvious ones. If you have questions don't hesitate to ask. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090709/a88df4ed/attachment.html From msc at freeswitch.org Thu Jul 9 09:38:31 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 9 Jul 2009 09:38:31 -0700 Subject: [Freeswitch-users] mod_say_en directory location In-Reply-To: <367751820907090906if8493dfoc927b07f8bc86cfd@mail.gmail.com> References: <367751820907090906if8493dfoc927b07f8bc86cfd@mail.gmail.com> Message-ID: <87f2f3b90907090938u53ee6145l825beb30275e1a20@mail.gmail.com> Is the call perhaps at 16kHz and it's looking for non-installed 16kHz sound files? -MC On Thu, Jul 9, 2009 at 9:06 AM, Phillip Jones wrote: > Hi there, > > I have a very simply script that speaks back some digits, as so: > > session:execute("say", "en number iterated 1234"); > > However, to get this to work successfully I have had to move the 'digits' > directory to: > > C:\Program Files (x86)\Freeswitch\sounds\en > > from the default: > > C:\Program Files (x86)\Freeswitch\sounds\en\us\callie\digits\8000 > > > This is a clean install of FreeSWITCH - so I am wondering why I needed to > do this, what have not configured correctly? > > As you can see I am using windows with a resent build (3 days) from svn. > > Any help appreciated. > > Thanks > > > Phillip Jones > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090709/decd2304/attachment.html From brian at freeswitch.org Thu Jul 9 10:19:11 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 9 Jul 2009 12:19:11 -0500 Subject: [Freeswitch-users] Less Than a Month Until ClueCon In-Reply-To: <191c3a030907081329q53ec3609i37b9ee87a31c9247@mail.gmail.com> References: <191c3a030907081329q53ec3609i37b9ee87a31c9247@mail.gmail.com> Message-ID: <09DA50D4-C82E-425D-9CB2-467159B673DB@freeswitch.org> We have info of where the meet and greet will take place here http://www.cluecon.com/node/35 Also info on travel from both O'Hare and Midway will be posted today via taxi or CTA. Thanks, /b On Jul 8, 2009, at 3:29 PM, Anthony Minessale wrote: > Hi everyone, > > We are less than a month away from ClueCon 2009 and I would like to > urge anyone who is considering attending to > sign up ASAP to make sure you are properly counted in the food > totals and the early bird pricing. > Bring your laptops and all your gizmos and get ready to dive into > telephony for 3 fun-filled days. > > We will be having FULL BREAKFAST on the first 2 mornings > (continental on day 3) FULL LUNCH, and OPEN BAR for 2 hours the > first 2 nights for ALL atendees included in your attendance fee. > We also will be giving away several prizes provided by our various > sponsors. Every paid registration gives you a chance to win one of > many goodies such as phones/tdm cards etc. > > REGISTER NOW http://www.cluecon.com > > or CALL (877) 742 -CLUE > or INSTALL FreeSWITCH and dial 5000 and choose the "register for > cluecon" option on the ivr. > or E-MAIL marketing at cluecon.com to discuss sponsoring ot > participating in helping out with logistics etc for a reduced > attendance fee. > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > 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/20090709/1619ce39/attachment-0001.html From msc at freeswitch.org Thu Jul 9 10:47:24 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 9 Jul 2009 10:47:24 -0700 Subject: [Freeswitch-users] Less Than a Month Until ClueCon In-Reply-To: <09DA50D4-C82E-425D-9CB2-467159B673DB@freeswitch.org> References: <191c3a030907081329q53ec3609i37b9ee87a31c9247@mail.gmail.com> <09DA50D4-C82E-425D-9CB2-467159B673DB@freeswitch.org> Message-ID: <87f2f3b90907091047p21c536d5re65acd6c5141ba99@mail.gmail.com> FYI, visit the hotel details page . Scroll to the bottom and there are links both to online maps and directions as well as a handy document that you can download. We have directions for getting to/from the hotel from both airports (O'Hare and Midway) via public transport as well as for those driving into town. -Michael On Thu, Jul 9, 2009 at 10:19 AM, Brian West wrote: > We have info of where the meet and greet will take place here > http://www.cluecon.com/node/35 Also info on travel from both O'Hare and > Midway will be posted today via taxi or CTA. > Thanks, > /b > > On Jul 8, 2009, at 3:29 PM, Anthony Minessale wrote: > > Hi everyone, > > We are less than a month away from ClueCon 2009 and I would like to urge > anyone who is considering attending to > sign up ASAP to make sure you are properly counted in the food totals and > the early bird pricing. > Bring your laptops and all your gizmos and get ready to dive into telephony > for 3 fun-filled days. > > We will be having FULL BREAKFAST on the first 2 mornings (continental on > day 3) FULL LUNCH, and OPEN BAR for 2 hours the first 2 nights for ALL > atendees included in your attendance fee. > We also will be giving away several prizes provided by our various > sponsors. Every paid registration gives you a chance to win one of many > goodies such as phones/tdm cards etc. > > REGISTER NOW http://www.cluecon.com > > or CALL (877) 742 -CLUE > or INSTALL FreeSWITCH and dial 5000 and choose the "register for cluecon" > option on the ivr. > or E-MAIL marketing at cluecon.com to discuss sponsoring ot participating in > helping out with logistics etc for a reduced attendance fee. > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > 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/20090709/0d9cffe0/attachment.html From brian at freeswitch.org Thu Jul 9 10:54:11 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 9 Jul 2009 12:54:11 -0500 Subject: [Freeswitch-users] Less Than a Month Until ClueCon In-Reply-To: <87f2f3b90907091047p21c536d5re65acd6c5141ba99@mail.gmail.com> References: <191c3a030907081329q53ec3609i37b9ee87a31c9247@mail.gmail.com> <09DA50D4-C82E-425D-9CB2-467159B673DB@freeswitch.org> <87f2f3b90907091047p21c536d5re65acd6c5141ba99@mail.gmail.com> Message-ID: Also if you need room share info email Michael or Myself offlist we can help you find a roomie! /b On Jul 9, 2009, at 12:47 PM, Michael Collins wrote: > FYI, visit the hotel details page. Scroll to the bottom and there > are links both to online maps and directions as well as a handy > document that you can download. We have directions for getting to/ > from the hotel from both airports (O'Hare and Midway) via public > transport as well as for those driving into town. > > -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090709/908db0cf/attachment.html From pjintheusa at gmail.com Thu Jul 9 11:46:10 2009 From: pjintheusa at gmail.com (Phillip Jones) Date: Thu, 9 Jul 2009 14:46:10 -0400 Subject: [Freeswitch-users] mod_say_en directory location In-Reply-To: <87f2f3b90907090938u53ee6145l825beb30275e1a20@mail.gmail.com> References: <367751820907090906if8493dfoc927b07f8bc86cfd@mail.gmail.com> <87f2f3b90907090938u53ee6145l825beb30275e1a20@mail.gmail.com> Message-ID: <367751820907091146y61b78b4v2b7525afe38c43a5@mail.gmail.com> Thanks for the response. I don't think so - the trace states: [ERR] mod_sndfile.c: 192 Error Opening File [C:\Program Files (x86)\Freeswitch\sounds\en\digits/1.wav] [System error : The system cannot find the path specified.] I created a 16000 directory to see whether that would help, and it did not. My C:\Program Files (x86)\Freeswitch\conf\lang\en\en.xml contains: Am I correct in thinking this is where the sound file dir for digits would be specified? On Thu, Jul 9, 2009 at 12:38 PM, Michael Collins wrote: > Is the call perhaps at 16kHz and it's looking for non-installed 16kHz sound > files? > -MC > > On Thu, Jul 9, 2009 at 9:06 AM, Phillip Jones wrote: > >> Hi there, >> >> I have a very simply script that speaks back some digits, as so: >> >> session:execute("say", "en number iterated 1234"); >> >> However, to get this to work successfully I have had to move the 'digits' >> directory to: >> >> C:\Program Files (x86)\Freeswitch\sounds\en >> >> from the default: >> >> C:\Program Files (x86)\Freeswitch\sounds\en\us\callie\digits\8000 >> >> >> This is a clean install of FreeSWITCH - so I am wondering why I needed to >> do this, what have not configured correctly? >> >> As you can see I am using windows with a resent build (3 days) from svn. >> >> Any help appreciated. >> >> Thanks >> >> >> Phillip Jones >> >> >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090709/1c8bce4c/attachment.html From pjintheusa at gmail.com Thu Jul 9 13:38:22 2009 From: pjintheusa at gmail.com (Phillip Jones) Date: Thu, 9 Jul 2009 16:38:22 -0400 Subject: [Freeswitch-users] mod_say_en directory location In-Reply-To: <367751820907091146y61b78b4v2b7525afe38c43a5@mail.gmail.com> References: <367751820907090906if8493dfoc927b07f8bc86cfd@mail.gmail.com> <87f2f3b90907090938u53ee6145l825beb30275e1a20@mail.gmail.com> <367751820907091146y61b78b4v2b7525afe38c43a5@mail.gmail.com> Message-ID: <367751820907091338n2e5f8312r50a743ec6d18ec26@mail.gmail.com> Ok - forget this one - I did a fresh install from the pre-compiled windows binary/msi - referenced on the wiki - and every thing is working as it should be. Thanks On Thu, Jul 9, 2009 at 2:46 PM, Phillip Jones wrote: > Thanks for the response. > > I don't think so - the trace states: > > [ERR] mod_sndfile.c: 192 Error Opening File [C:\Program Files > (x86)\Freeswitch\sounds\en\digits/1.wav] [System error : The system cannot > find the path specified.] > > I created a 16000 directory to see whether that would help, and it did not. > > My C:\Program Files (x86)\Freeswitch\conf\lang\en\en.xml contains: > > > tts-engine="cepstral" tts-voice="callie"> > > > > > > > Am I correct in thinking this is where the sound file dir for digits would > be specified? > > > > On Thu, Jul 9, 2009 at 12:38 PM, Michael Collins wrote: > >> Is the call perhaps at 16kHz and it's looking for non-installed 16kHz >> sound files? >> -MC >> >> On Thu, Jul 9, 2009 at 9:06 AM, Phillip Jones wrote: >> >>> Hi there, >>> >>> I have a very simply script that speaks back some digits, as so: >>> >>> session:execute("say", "en number iterated 1234"); >>> >>> However, to get this to work successfully I have had to move the 'digits' >>> directory to: >>> >>> C:\Program Files (x86)\Freeswitch\sounds\en >>> >>> from the default: >>> >>> C:\Program Files (x86)\Freeswitch\sounds\en\us\callie\digits\8000 >>> >>> >>> This is a clean install of FreeSWITCH - so I am wondering why I needed to >>> do this, what have not configured correctly? >>> >>> As you can see I am using windows with a resent build (3 days) from svn. >>> >>> Any help appreciated. >>> >>> Thanks >>> >>> >>> Phillip Jones >>> >>> >>> >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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/20090709/e2eddb63/attachment-0001.html From tayeb.meftah at gmail.com Thu Jul 9 16:58:22 2009 From: tayeb.meftah at gmail.com (Meftah Tayeb) Date: Thu, 09 Jul 2009 23:58:22 +0000 Subject: [Freeswitch-users] Skypiax Parameters Informations Request Message-ID: <4A56841E.4080206@gmail.com> hello, i have the folowing parameter in Skypiax.conf.xml: each call that will to by routed to this destination?? Each Call will to by routed to this destination? each codecs that is pocible to use it with Skypiax? all? speex? this codecs is used beetwan skypiax and the remote peer? thanks __________ Information from ESET NOD32 Antivirus, version of virus signature database 4229 (20090709) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com From andrew at hijacked.us Thu Jul 9 17:21:05 2009 From: andrew at hijacked.us (Andrew Thompson) Date: Thu, 9 Jul 2009 20:21:05 -0400 Subject: [Freeswitch-users] mod_snom dialplan demo not working Message-ID: <20090710002104.GE28401@hijacked.us> I'm trying to replace our aging nortel system with a FreeSWITCH based system using snom 3[267]0 phones. I've been doing okay so far but I'm running into a brick wall when I try to run the mod_snom demo in the dialplan. I've setup the 'line 2' function key to type button with the number being 'message' like it says on the snom wiki and when I dial extension 9000 the button lights up, but pressing the button does absolutely nothing. I can't figure out what I'm missing here. Any advice? Andrew From brian at freeswitch.org Thu Jul 9 18:10:41 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 9 Jul 2009 20:10:41 -0500 Subject: [Freeswitch-users] mod_snom dialplan demo not working In-Reply-To: <20090710002104.GE28401@hijacked.us> References: <20090710002104.GE28401@hijacked.us> Message-ID: <96EB3A77-7230-4EFE-9541-8A0AC747DBB8@freeswitch.org> It needs tons of work its not so demo tastic ... /b On Jul 9, 2009, at 7:21 PM, Andrew Thompson wrote: > I'm trying to replace our aging nortel system with a FreeSWITCH based > system using snom 3[267]0 phones. I've been doing okay so far but I'm > running into a brick wall when I try to run the mod_snom demo in the > dialplan. I've setup the 'line 2' function key to type button with the > number being 'message' like it says on the snom wiki and when I dial > extension 9000 the button lights up, but pressing the button does > absolutely nothing. I can't figure out what I'm missing here. Any > advice? > > 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 andrew at hijacked.us Thu Jul 9 19:00:28 2009 From: andrew at hijacked.us (Andrew Thompson) Date: Thu, 9 Jul 2009 22:00:28 -0400 Subject: [Freeswitch-users] mod_snom dialplan demo not working In-Reply-To: <96EB3A77-7230-4EFE-9541-8A0AC747DBB8@freeswitch.org> References: <20090710002104.GE28401@hijacked.us> <96EB3A77-7230-4EFE-9541-8A0AC747DBB8@freeswitch.org> Message-ID: <20090710020027.GF28401@hijacked.us> On Thu, Jul 09, 2009 at 08:10:41PM -0500, Brian West wrote: > It needs tons of work its not so demo tastic ... > Is it incomplete or just there's no documentation on how to do it? From the sip trace it looks like FreeSWITCH is sending the phone the right stuff just when I hit the button that was programmed nothing happens. I want to be able to *use* this feature of the snom phones, not just play with the demo anyway, so if there's some development needed to make it work I'd be happy to take a stab at it. Andrew From shaheryarkh at googlemail.com Thu Jul 9 22:40:00 2009 From: shaheryarkh at googlemail.com (Muhammad Shahzad) Date: Fri, 10 Jul 2009 11:40:00 +0600 Subject: [Freeswitch-users] Skypiax Parameters Informations Request In-Reply-To: <4A56841E.4080206@gmail.com> References: <4A56841E.4080206@gmail.com> Message-ID: Destination parameter actually specifies the extension on which this Skype user is reachable within FreeSWITCH dialplan for incoming calls. If this parameter is specified in per_interface_settings xml tag then it will override the value of this parameter in global_settings xml tag, otherwise value of this parameter from global_settings xml tag will be used. Here is an example (see below), the user test.01 is reachable on dialplan extension 2000 (since it has its own destination defined in per_interface_settings xml tag), whereas test.02 is reachable on dialplan extension 5000 (since it does not have destination parameter defined and thus it will use value for this parameter in global_settings xml tag). Now the codec, Skype has its own proprietory code for Skype to Skype calls. The codec we specify in Skypiax configuration file is actually used for Skype to/from non-Skype calls. Consider following dial plan example (with skypiax configuration given above), If a remote Skype user dials test.01 from his/her Skype client, then FreeSWITCH will route this call to SIP user 1000 and codecs specified in Skypiax configuration will be offered to destination SIP endpoint (SIP user 1000 in this case). Thank you. On Fri, Jul 10, 2009 at 5:58 AM, Meftah Tayeb wrote: > hello, > i have the folowing parameter in Skypiax.conf.xml: > > > > each call that will to by routed to this destination?? > > > > > > Each Call will to by routed to this destination? > > each codecs that is pocible to use it with Skypiax? all? speex? > this codecs is used beetwan skypiax and the remote peer? > thanks > > > __________ Information from ESET NOD32 Antivirus, version of virus > signature database 4229 (20090709) __________ > > The message was checked by ESET NOD32 Antivirus. > > http://www.eset.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 > -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090710/db9c354a/attachment.html From shaheryarkh at googlemail.com Thu Jul 9 22:55:17 2009 From: shaheryarkh at googlemail.com (Muhammad Shahzad) Date: Fri, 10 Jul 2009 11:55:17 +0600 Subject: [Freeswitch-users] Interactive Connectivity Establishment (ICE) support in FS Message-ID: Hi, Do we have ICE support in FreeSWITCH. If so, any module as example that is using it? If not then i would like to write one for my mod_msn module, do we have any FS API that i would need to implement in this case? Thank you. -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090710/10ceea11/attachment.html From tayeb.meftah at gmail.com Fri Jul 10 02:48:27 2009 From: tayeb.meftah at gmail.com (Meftah Tayeb) Date: Fri, 10 Jul 2009 09:48:27 +0000 Subject: [Freeswitch-users] Skypiax Parameters Informations Request In-Reply-To: References: <4A56841E.4080206@gmail.com> Message-ID: <4A570E6B.1000106@gmail.com> hello Muhammad , thank you what about hig cality audio codec to use? speex is good? thanks Muhammad Shahzad wrote: > Destination parameter actually specifies the extension on which this > Skype user is reachable within FreeSWITCH dialplan for incoming calls. > > If this parameter is specified in per_interface_settings xml tag then > it will override the value of this parameter in global_settings xml > tag, otherwise value of this parameter from global_settings xml tag > will be used. > > Here is an example (see below), the user test.01 is reachable on > dialplan extension 2000 (since it has its own destination defined in > per_interface_settings xml tag), whereas test.02 is reachable on > dialplan extension 5000 (since it does not have destination parameter > defined and thus it will use value for this parameter in > global_settings xml tag). > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > Now the codec, Skype has its own proprietory code for Skype to Skype > calls. The codec we specify in Skypiax configuration file is actually > used for Skype to/from non-Skype calls. Consider following dial plan > example (with skypiax configuration given above), > > > > > > > > If a remote Skype user dials test.01 from his/her Skype client, then > FreeSWITCH will route this call to SIP user 1000 and codecs specified > in Skypiax configuration will be offered to destination SIP endpoint > (SIP user 1000 in this case). > > Thank you. > > > On Fri, Jul 10, 2009 at 5:58 AM, Meftah Tayeb > wrote: > > hello, > i have the folowing parameter in Skypiax.conf.xml: > > > > each call that will to by routed to this destination?? > > > > > > Each Call will to by routed to this destination? > > each codecs that is pocible to use it with Skypiax? all? speex? > this codecs is used beetwan skypiax and the remote peer? > thanks > > > __________ Information from ESET NOD32 Antivirus, version of virus > signature database 4229 (20090709) __________ > > The message was checked by ESET NOD32 Antivirus. > > http://www.eset.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 > > > > > -- > Muhammad Shahzad > ----------------------------------- > CISCO Rich Media Communication Specialist (CRMCS) > CISCO Certified Network Associate (CCNA) > Cell: +92 334 422 40 88 > MSN: shari_786pk at hotmail.com > Email: shaheryarkh at googlemail.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 > > > > __________ Information from ESET NOD32 Antivirus, version of virus signature database 4231 (20090710) __________ > > The message was checked by ESET NOD32 Antivirus. > > http://www.eset.com > > __________ Information from ESET NOD32 Antivirus, version of virus signature database 4231 (20090710) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090710/f68d2fed/attachment-0001.html From mcampbellsmith at gmail.com Fri Jul 10 02:57:55 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Fri, 10 Jul 2009 19:57:55 +1000 Subject: [Freeswitch-users] 2 voicemail questions Message-ID: <33c87fa30907100257w29c628cbnb641d09a75bc22b8@mail.gmail.com> Hi! >> 1. Can I email the voicemail message to multiple email addresses? A comma separated list does not work in the extensions.xml file (1000.xml), but it does work if I hard code the email addresses into the notify-voicemail.tpl file. Could this be added to the switch so that it can handle comma separated lists? >> 2. How can I make Freeswitch dial a number AFTER a voicemail is left? >api Hangup hook? i want the 'voicemail' application to appear to call the extension to notify the user that there is a waiting message. This is an extract from my dialplan.xml: This only works if the B leg (ie voicemail application) hangs up first. This would be an unusual situation and does not achieve what I want... is there any other way to achieve this? Thanks > Hi! > > I have 2 questions regarding voicemail ... > > 1. Can I email the voicemail message to multiple email addresses? If > so, what format is this in? > Try a comma sep. list. Not sure if it will work. > > 2. How can I make Freeswitch dial a number AFTER a voicemail is left? api Hangup hook? I g > From: Brian West > On Jul 7, 2009, at 9:11 AM, Mark Campbell-Smith wrote: > Hi! > > I have 2 questions regarding voicemail ... > > 1. Can I email the voicemail message to multiple email addresses? If > so, what format is this in? > > Try a comma sep. list. Not sure if it will work. From shaheryarkh at googlemail.com Fri Jul 10 04:02:34 2009 From: shaheryarkh at googlemail.com (Muhammad Shahzad) Date: Fri, 10 Jul 2009 17:02:34 +0600 Subject: [Freeswitch-users] Skypiax Parameters Informations Request In-Reply-To: <4A570E6B.1000106@gmail.com> References: <4A56841E.4080206@gmail.com> <4A570E6B.1000106@gmail.com> Message-ID: I think you can use it has long as remote end-point supports it. Thank you. On Fri, Jul 10, 2009 at 3:48 PM, Meftah Tayeb wrote: > hello Muhammad , > thank you > what about hig cality audio codec to use? > speex is good? > thanks > > Muhammad Shahzad wrote: > > Destination parameter actually specifies the extension on which this Skype > user is reachable within FreeSWITCH dialplan for incoming calls. > > If this parameter is specified in per_interface_settings xml tag then it > will override the value of this parameter in global_settings xml tag, > otherwise value of this parameter from global_settings xml tag will be used. > > Here is an example (see below), the user test.01 is reachable on dialplan > extension 2000 (since it has its own destination defined in > per_interface_settings xml tag), whereas test.02 is reachable on dialplan > extension 5000 (since it does not have destination parameter defined and > thus it will use value for this parameter in global_settings xml tag). > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > Now the codec, Skype has its own proprietory code for Skype to Skype calls. > The codec we specify in Skypiax configuration file is actually used for > Skype to/from non-Skype calls. Consider following dial plan example (with > skypiax configuration given above), > > > > > > > > If a remote Skype user dials test.01 from his/her Skype client, then > FreeSWITCH will route this call to SIP user 1000 and codecs specified in > Skypiax configuration will be offered to destination SIP endpoint (SIP user > 1000 in this case). > > Thank you. > > > On Fri, Jul 10, 2009 at 5:58 AM, Meftah Tayeb wrote: > >> hello, >> i have the folowing parameter in Skypiax.conf.xml: >> >> >> >> each call that will to by routed to this destination?? >> >> >> >> >> >> Each Call will to by routed to this destination? >> >> each codecs that is pocible to use it with Skypiax? all? speex? >> this codecs is used beetwan skypiax and the remote peer? >> thanks >> >> >> __________ Information from ESET NOD32 Antivirus, version of virus >> signature database 4229 (20090709) __________ >> >> The message was checked by ESET NOD32 Antivirus. >> >> http://www.eset.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 >> > > > > -- > Muhammad Shahzad > ----------------------------------- > CISCO Rich Media Communication Specialist (CRMCS) > CISCO Certified Network Associate (CCNA) > Cell: +92 334 422 40 88 > MSN: shari_786pk at hotmail.com > Email: shaheryarkh at googlemail.com > > ------------------------------ > > _______________________________________________ > 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 > > > > > __________ Information from ESET NOD32 Antivirus, version of virus signature database 4231 (20090710) __________ > > The message was checked by ESET NOD32 Antivirus. > http://www.eset.com > > > > > __________ Information from ESET NOD32 Antivirus, version of virus > signature database 4231 (20090710) __________ > > The message was checked by ESET NOD32 Antivirus. > > http://www.eset.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 > > -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090710/8dab18a1/attachment.html From helmut.kuper at ewetel.de Fri Jul 10 05:53:50 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Fri, 10 Jul 2009 14:53:50 +0200 Subject: [Freeswitch-users] pocketsphinx Message-ID: <4A5739DE.1080800@ewetel.de> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I try to change pocketsphinx's grammar from default (english) to german. I found this archive (http://www.repository.voxforge1.org/downloads/de/Trunk/AcousticModels/), which contains similar files like those which can be found in grammar/model/communicator directory. Unfortunately FS crashed without writing a core file nor logfile enries as soon as as pizza demo trys to detect speech. Any Ideas? Maybe someone has already working grammar/model files for german language? regards helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) iD8DBQFKVzne4tZeNddg3dwRAthzAJ4hvonJLgaTWc3kCQXhESb2wsTu8QCeP/DD skUOkNgUHLRaKqVGWWk1uM8= =HkcN -----END PGP SIGNATURE----- From anthony.minessale at gmail.com Fri Jul 10 06:25:04 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 10 Jul 2009 08:25:04 -0500 Subject: [Freeswitch-users] Less Than a Month Until ClueCon In-Reply-To: References: <191c3a030907081329q53ec3609i37b9ee87a31c9247@mail.gmail.com> <09DA50D4-C82E-425D-9CB2-467159B673DB@freeswitch.org> <87f2f3b90907091047p21c536d5re65acd6c5141ba99@mail.gmail.com> Message-ID: <191c3a030907100625g77a6fe32ub7a948931144e90f@mail.gmail.com> Yes, This is very important. The rates on the room will soon rise out of control as the conference date nears so it's important to book now or find someone to share with before it's too late. On Thu, Jul 9, 2009 at 12:54 PM, Brian West wrote: > Also if you need room share info email Michael or Myself offlist we can > help you find a roomie! > /b > > On Jul 9, 2009, at 12:47 PM, Michael Collins wrote: > > FYI, visit the hotel details page . Scroll to > the bottom and there are links both to online maps and directions as well as > a handy document that you can download. We have directions for getting > to/from the hotel from both airports (O'Hare and Midway) via public > transport as well as for those driving into town. > > -Michael > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference 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/20090710/876bed77/attachment-0001.html From ney at frota.net Fri Jul 10 00:00:28 2009 From: ney at frota.net (Ney Frota) Date: Fri, 10 Jul 2009 04:00:28 -0300 Subject: [Freeswitch-users] Help Message-ID: <5E41B7C2-B163-4364-A2A9-8B90AD58F47E@frota.net> Help From vkozak at abisoft.spb.ru Fri Jul 10 02:39:28 2009 From: vkozak at abisoft.spb.ru (Kozak Vladimir) Date: Fri, 10 Jul 2009 13:39:28 +0400 Subject: [Freeswitch-users] FS not wait respond from called and send 200 Ok Message-ID: <5F5ABAC4334545CE8323DACB10805DE1@abisoft.biz> Hello, I have the following problem: I send Invite without SDP to Freeswitch on destination_number "xxx_123" And I want Freeswitch to make "bridge", but it doesn't wait respond from "123" and sends 200 Ok with SDP to me. Does nybody know a clue about this? Best regards vkozak -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090710/8ee67c81/attachment.html From velu.technical at gmail.com Fri Jul 10 05:36:14 2009 From: velu.technical at gmail.com (velusamy velu) Date: Fri, 10 Jul 2009 18:06:14 +0530 Subject: [Freeswitch-users] How to configure SIP(Sofia) profiles Message-ID: <1452e2980907100536s15818474k901093b742510064@mail.gmail.com> Dear Friends, I am a newbie for FreeSWITCH. I was installed FreeSWITCH locally. I just wanted to test whether my FreeSWITCH is working fine. I need help from you that how to configure my Softphone(Twinkle) to use FreeSWITCH. I need steps to check my FreeSWITCH working with Twinkle. Please help me in this... Thanks in Advance. Regards, K.Velusamy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090710/66f36fc9/attachment.html From Prometheus001 at gmx.net Fri Jul 10 07:12:46 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Fri, 10 Jul 2009 16:12:46 +0200 Subject: [Freeswitch-users] pocketsphinx In-Reply-To: <4A5739DE.1080800@ewetel.de> References: <4A5739DE.1080800@ewetel.de> Message-ID: <4A574C5E.2000401@gmx.net> Hello Helmut, I looked at these dic files. Their content (look at all the qq's) is quite different from the dic files supplied with freeswitch pocketsphinx. As I remember the CMU dict file format has changed in April 2008. Best regards Peter Helmut Kuper schrieb: > Hi, > > I try to change pocketsphinx's grammar from default (english) to german. > I found this archive > (http://www.repository.voxforge1.org/downloads/de/Trunk/AcousticModels/), > which > contains similar files like those which can be found in > grammar/model/communicator directory. > > Unfortunately FS crashed without writing a core file nor logfile enries > as soon as as pizza demo trys to detect speech. > > Any Ideas? Maybe someone has already working grammar/model files for > german language? > > > 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 brian at freeswitch.org Fri Jul 10 07:24:01 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 10 Jul 2009 09:24:01 -0500 Subject: [Freeswitch-users] pocketsphinx In-Reply-To: <4A574C5E.2000401@gmx.net> References: <4A5739DE.1080800@ewetel.de> <4A574C5E.2000401@gmx.net> Message-ID: <51760CDC-8560-4C2B-B618-2E97A0B3F8C2@freeswitch.org> Yes you have to make sure you use the one that comes with Pocketsphinx and not the 7.x one you download from the website. They aren't compatible last I checked. /b On Jul 10, 2009, at 9:12 AM, Peter P GMX wrote: > Hello Helmut, > > I looked at these dic files. Their content (look at all the qq's) is > quite different from the dic files supplied with freeswitch > pocketsphinx. > As I remember the CMU dict file format has changed in April 2008. > > Best regards > Peter From Prometheus001 at gmx.net Fri Jul 10 07:24:13 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Fri, 10 Jul 2009 16:24:13 +0200 Subject: [Freeswitch-users] pocketsphinx In-Reply-To: <4A5739DE.1080800@ewetel.de> References: <4A5739DE.1080800@ewetel.de> Message-ID: <4A574F0D.7040604@gmx.net> Hello Helmut, I looked at these dic files. Their content (look at all the qq's) is quite different from the dic files supplied with freeswitch pocketsphinx. As I remember the CMU dict file format has changed in April 2008. Maybe there is a converter somewhere? I was thinking of just enhancing the current dict file for some german words I need, but did not test it so far. This should be possible without modifying the underlying grammar. http://en.wikipedia.org/wiki/CMU_Pronouncing_Dictionary I would love to hear when you have had any progress on this. Best regards Peter Helmut Kuper schrieb: > Hi, > > I try to change pocketsphinx's grammar from default (english) to german. > I found this archive > (http://www.repository.voxforge1.org/downloads/de/Trunk/AcousticModels/), > which > contains similar files like those which can be found in > grammar/model/communicator directory. > > Unfortunately FS crashed without writing a core file nor logfile enries > as soon as as pizza demo trys to detect speech. > > Any Ideas? Maybe someone has already working grammar/model files for > german language? > > > 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 dftoro at yahoo.com Fri Jul 10 08:54:34 2009 From: dftoro at yahoo.com (Diego Toro) Date: Fri, 10 Jul 2009 08:54:34 -0700 (PDT) Subject: [Freeswitch-users] mod_say_en directory location Message-ID: <45224.76458.qm@web33505.mail.mud.yahoo.com> Hi all, ? This happens becouse sound_prefix variable is not used to make path to sound files, case "Language Handling: call for assistance" ? Diego --- On Thu, 7/9/09, Phillip Jones wrote: From: Phillip Jones Subject: Re: [Freeswitch-users] mod_say_en directory location To: freeswitch-users at lists.freeswitch.org Date: Thursday, July 9, 2009, 3:38 PM Ok - forget this one - I did a fresh install from the pre-compiled windows binary/msi - referenced on the wiki - and every thing is working as it should be. Thanks On Thu, Jul 9, 2009 at 2:46 PM, Phillip Jones wrote: Thanks for the response. I don't think so - the trace states: [ERR] mod_sndfile.c: 192 Error Opening File [C:\Program Files (x86)\Freeswitch\sounds\en\digits/1.wav] [System error : The system cannot find the path specified.] I created a 16000 directory to see whether that would help, and it did not. My C:\Program Files (x86)\Freeswitch\conf\lang\en\en.xml contains: ? ??? ??? ??? ? ? Am I correct in thinking this is where the sound file dir for digits would be specified? On Thu, Jul 9, 2009 at 12:38 PM, Michael Collins wrote: Is the call perhaps at 16kHz and it's looking for non-installed 16kHz sound files? -MC On Thu, Jul 9, 2009 at 9:06 AM, Phillip Jones wrote: Hi there, I have a very simply script that speaks back some digits, as so: session:execute("say", "en number iterated 1234"); However, to get this to work successfully I have had to move the 'digits' directory to: C:\Program Files (x86)\Freeswitch\sounds\en from the default: C:\Program Files (x86)\Freeswitch\sounds\en\us\callie\digits\8000 This is a clean install of FreeSWITCH - so I am wondering why I needed to do this, what have not configured correctly? As you can see I am using windows with a resent build (3 days) from svn. Any help appreciated. Thanks Phillip Jones _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -----Inline Attachment Follows----- _______________________________________________ Freeswitch-users mailing list Freeswitch-users at 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/20090710/c5f5dfc9/attachment.html From woof at iwoof.org Fri Jul 10 08:56:16 2009 From: woof at iwoof.org (Andy Spitzer) Date: Fri, 10 Jul 2009 11:56:16 -0400 Subject: [Freeswitch-users] Question on auth-calls Message-ID: Woof! It is my understanding, that if I set in a SIP profile, it shouldn't challenge for authentication under any circumstances. However, if an INVITE contains a a Proxy-Authorization header from another proxy, Sofia DOES challenge with a 407. I'm aware one can set accept-blind-auth to work around this, but I'm wondering if my understanding of auth-calls is wrong, or the behavior I'm seeing is wrong. --Woof! From mike at jerris.com Fri Jul 10 09:04:01 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 10 Jul 2009 12:04:01 -0400 Subject: [Freeswitch-users] 2 voicemail questions In-Reply-To: <33c87fa30907100257w29c628cbnb641d09a75bc22b8@mail.gmail.com> References: <33c87fa30907100257w29c628cbnb641d09a75bc22b8@mail.gmail.com> Message-ID: <344FC083-5916-4232-AEB5-3504626566DF@jerris.com> could you post how you tired to do it in dialplan that didn't work? Mike On Jul 10, 2009, at 5:57 AM, Mark Campbell-Smith wrote: > Hi! >>> 1. Can I email the voicemail message to multiple email addresses? > A comma separated list does not work in the extensions.xml file > (1000.xml), but it does work if I hard code the email addresses into > the notify-voicemail.tpl file. > > Could this be added to the switch so that it can handle comma > separated lists? > >>> 2. How can I make Freeswitch dial a number AFTER a voicemail is >>> left? > >> api Hangup hook? > > i want the 'voicemail' application to appear to call the extension to > notify the user that there is a waiting message. This is an extract > from my dialplan.xml: > > > > > > > > > data="user/${dialed_extension}@${domain_name}"/> > > > > > field="${vm_boxcount(${destination_number}@${domain_name})}" > expression="^(1)$"> > > > > This only works if the B leg (ie voicemail application) hangs up > first. This would be an unusual situation and does not achieve what I > want... is there any other way to achieve this? > > Thanks > >> Hi! >> >> I have 2 questions regarding voicemail ... >> >> 1. Can I email the voicemail message to multiple email addresses? If >> so, what format is this in? >> > > Try a comma sep. list. Not sure if it will work. > >> >> 2. How can I make Freeswitch dial a number AFTER a voicemail is left? > > api Hangup hook? > > I g >> From: Brian West > >> On Jul 7, 2009, at 9:11 AM, Mark Campbell-Smith wrote: > >> Hi! >> >> I have 2 questions regarding voicemail ... >> >> 1. Can I email the voicemail message to multiple email addresses? If >> so, what format is this in? >> > >> Try a comma sep. list. Not sure if it will work. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.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 Jul 10 09:06:07 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 10 Jul 2009 11:06:07 -0500 Subject: [Freeswitch-users] Question on auth-calls In-Reply-To: References: Message-ID: <19E3739C-73C3-4B8D-BC54-18079CB65CEA@freeswitch.org> accept-blind-auth is for this scenario.... /b On Jul 10, 2009, at 10:56 AM, Andy Spitzer wrote: > Woof! > > It is my understanding, that if I set > > in a SIP profile, it shouldn't challenge for authentication under > any circumstances. > > However, if an INVITE contains a a Proxy-Authorization header from > another proxy, Sofia DOES challenge with a 407. > > I'm aware one can set accept-blind-auth to work around this, but I'm > wondering if my understanding of auth-calls is wrong, or the > behavior I'm seeing is wrong. > > --Woof! From anthony.minessale at gmail.com Fri Jul 10 09:09:05 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 10 Jul 2009 11:09:05 -0500 Subject: [Freeswitch-users] Question on auth-calls In-Reply-To: References: Message-ID: <191c3a030907100909h6a0031cfv3bbc0f80e0cdbcec@mail.gmail.com> The way it works by default is that if you send a www-authenticate, we *always* try to process it. HOWEVER, we have a accept-blind-auth sofia profile param (in fact it was invented just for sipX) On Fri, Jul 10, 2009 at 10:56 AM, Andy Spitzer wrote: > Woof! > > It is my understanding, that if I set > > in a SIP profile, it shouldn't challenge for authentication under any > circumstances. > > However, if an INVITE contains a a Proxy-Authorization header from another > proxy, Sofia DOES challenge with a 407. > > I'm aware one can set accept-blind-auth to work around this, but I'm > wondering if my understanding of auth-calls is wrong, or the behavior I'm > seeing is wrong. > > --Woof! > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference 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/20090710/8d650d75/attachment.html From dome at tel.co.th Fri Jul 10 09:10:23 2009 From: dome at tel.co.th (Dome Charoenyost) Date: Fri, 10 Jul 2009 23:10:23 +0700 Subject: [Freeswitch-users] How to check hangup cause before try next route in mod_lcr Message-ID: <8ccbff060907100910s2f12827fu18f9ff267b8b8e51@mail.gmail.com> Dear all, I'm testing mod_lcr. i found example in wiki If i want to use ${lcr_route_1} and ${lcr_route_2} What's best way to check hangup cause before try lcr_route_2 ? BG Dome C. From mrene_lists at avgs.ca Fri Jul 10 09:11:22 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Fri, 10 Jul 2009 12:11:22 -0400 Subject: [Freeswitch-users] How to check hangup cause before try next route in mod_lcr In-Reply-To: <8ccbff060907100910s2f12827fu18f9ff267b8b8e51@mail.gmail.com> References: <8ccbff060907100910s2f12827fu18f9ff267b8b8e51@mail.gmail.com> Message-ID: <7E454794-5270-4BA0-A11E-9CF220FA4BE4@avgs.ca> That'll make your dialplan stop if the call is hung up from the B-leg. Math On 10-Jul-09, at 12:10 PM, Dome Charoenyost wrote: > Dear all, > > I'm testing mod_lcr. > i found example in wiki > > > If i want to use ${lcr_route_1} and ${lcr_route_2} What's best way to > check hangup cause before try lcr_route_2 ? > > > BG > > Dome C. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.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 Fri Jul 10 09:12:03 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 10 Jul 2009 12:12:03 -0400 Subject: [Freeswitch-users] FS not wait respond from called and send 200 Ok In-Reply-To: <5F5ABAC4334545CE8323DACB10805DE1@abisoft.biz> References: <5F5ABAC4334545CE8323DACB10805DE1@abisoft.biz> Message-ID: <74C6337E-6F03-4485-B69B-372658E0D79B@jerris.com> Look closer at the logs, we don't send a 200ok in a bridge until we get one from the b leg. Mike On Jul 10, 2009, at 5:39 AM, Kozak Vladimir wrote: > Hello, > > I have the following problem: I send Invite without SDP to > Freeswitch on destination_number "xxx_123" > > > > > > > And I want Freeswitch to make "bridge", but it doesn't wait respond > from "123" and sends 200 Ok with SDP to me. > Does nybody know a clue about this? > > Best regards > vkozak > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090710/a5643c1f/attachment.html From woof at iwoof.org Fri Jul 10 09:17:47 2009 From: woof at iwoof.org (Andy Spitzer) Date: Fri, 10 Jul 2009 12:17:47 -0400 Subject: [Freeswitch-users] Question on auth-calls In-Reply-To: <191c3a030907100909h6a0031cfv3bbc0f80e0cdbcec@mail.gmail.com> References: <191c3a030907100909h6a0031cfv3bbc0f80e0cdbcec@mail.gmail.com> Message-ID: Woof! On Fri, 10 Jul 2009 12:09:05 -0400, Anthony Minessale wrote: > The way it works by default is that if you send a www-authenticate, we > *always* try to process it. > HOWEVER, we have a accept-blind-auth sofia profile param (in fact it was > invented just for sipX) Then my understanding was incorrect. Thanks for the clarification. --Woof! From Christian.Jensen at Teligence.Net Fri Jul 10 09:32:08 2009 From: Christian.Jensen at Teligence.Net (Christian Jensen) Date: Fri, 10 Jul 2009 09:32:08 -0700 Subject: [Freeswitch-users] 302 Redirect Message-ID: Hi everyone! I have a question - I need to change the "From" SIP header during a "Redirect" to make it look like the number that called is a different number. I need to be able to change it but it is not taking - I have tried just about every combination of variable settings that I know of but the SIP message is just not getting the data. Here is my config: In fact - it would appear that no channel variables are making it out the door during a redirect. Any help would be greatly appreciated. Thanks! Christian Jensen Software Development Manager - Back Office Teligence T: 604-629-6055 Ext. 3304 M: 778-996-4283 F: 604-257-5578 christian.jensen at teligence.net www.teligence.net -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090710/f6b41a3d/attachment-0001.html From dome at tel.co.th Fri Jul 10 09:37:29 2009 From: dome at tel.co.th (Dome Charoenyost) Date: Fri, 10 Jul 2009 23:37:29 +0700 Subject: [Freeswitch-users] How to check hangup cause before try next route in mod_lcr In-Reply-To: <7E454794-5270-4BA0-A11E-9CF220FA4BE4@avgs.ca> References: <8ccbff060907100910s2f12827fu18f9ff267b8b8e51@mail.gmail.com> <7E454794-5270-4BA0-A11E-9CF220FA4BE4@avgs.ca> Message-ID: <8ccbff060907100937l319b85e9t4200a1a8a630a1c9@mail.gmail.com> 2009/7/10 Mathieu Rene : > thanks. let's me ask some question. i want to provide callback solution. 1. customer call to FS 2. FS hangup call and check balance (by callerid) 3. Make call to customer number 4. customer answer call and input destination this step when B-leg hangup (Or A-leg put *) customer can input other number until A-leg Hangup Now my solution use javascripts. but someone tell me JS not good for handle call. So now i testing mod_lcr and nibblebill Can someone show me dialplan for do like that. Best Regards. Dome C. > > That'll make your dialplan stop if the call is hung up from the B-leg. > > Math > > On 10-Jul-09, at 12:10 PM, Dome Charoenyost wrote: > >> Dear all, >> >> ? ? ? ? I'm testing mod_lcr. >> i found example in wiki >> ? ? ? >> ? ? ? >> If i want to use ${lcr_route_1} and ${lcr_route_2} What's best way to >> check hangup cause before try lcr_route_2 ? >> >> >> BG >> >> Dome C. >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.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 Jul 10 09:41:03 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 10 Jul 2009 11:41:03 -0500 Subject: [Freeswitch-users] 302 Redirect In-Reply-To: References: Message-ID: <9F8A6292-3BE0-4E29-B331-9EA7A413BE37@freeswitch.org> Ok you need the nice new feature I added to FreeSWITCH that lets you handle all the 302 redirects in your own dialplan/context. Set the param manual-redirect on your sofia profile then you can define sip_redirect_profile, sip_redirect_context, sip_redirect_dialplan or you can create a context called redirected and do anything you wish with the 302'ed call 100% manually. You'll get sip_redirect_contact_X sip_redirected_to sip_redirect_contact_user_X sip_redirect_contact_host_X sip_redirect_contact_params_X sip_redirected_by All of these will help you process this via the dialplan. /b On Jul 10, 2009, at 11:32 AM, Christian Jensen wrote: > Hi everyone! > > I have a question ? I need to change the ?From? SIP header during a > ?Redirect? to make it look like the number that called is a > different number. > > I need to be able to change it but it is not taking ? I have tried > just about every combination of variable settings that I know of but > the SIP message is just not getting the data. > > Here is my config: > > > > > data="sip_from_user_stripped=false"/> > data="sip_from_user=0445555555"/> > data="sip_invite_domain=sip:0446666666 at 192.168.8.2"/> > > > data="myani=0449999999"/> > > > > > > In fact ? it would appear that no channel variables are making it > out the door during a redirect. > > Any help would be greatly appreciated. > > Thanks! > > Christian Jensen > Software Development Manager ? Back Office > Teligence > T: 604-629-6055 Ext. 3304 > M: 778-996-4283 > F: 604-257-5578 > christian.jensen at teligence.net > www.teligence.net > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090710/213ac79a/attachment-0001.html From mrene_lists at avgs.ca Fri Jul 10 09:43:46 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Fri, 10 Jul 2009 12:43:46 -0400 Subject: [Freeswitch-users] 302 Redirect In-Reply-To: <9F8A6292-3BE0-4E29-B331-9EA7A413BE37@freeswitch.org> References: <9F8A6292-3BE0-4E29-B331-9EA7A413BE37@freeswitch.org> Message-ID: You can also set the following variables to control where the call will go once you get a 302: sip_redirect_profile (used to build the dialstrings) sip_redirect_context (the dialplan context to send the call) sip_redirect_dialplan (the dialplan to send the call) Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 10-Jul-09, at 12:41 PM, Brian West wrote: > Ok you need the nice new feature I added to FreeSWITCH that lets you > handle all the 302 redirects in your own dialplan/context. > > Set the param manual-redirect on your sofia profile then you can > define sip_redirect_profile, sip_redirect_context, > sip_redirect_dialplan or you can create a context called redirected > and do anything you wish with the 302'ed call 100% manually. > > You'll get > > sip_redirect_contact_X > sip_redirected_to > sip_redirect_contact_user_X > sip_redirect_contact_host_X > sip_redirect_contact_params_X > sip_redirected_by > > All of these will help you process this via the dialplan. > > /b > > On Jul 10, 2009, at 11:32 AM, Christian Jensen wrote: > >> Hi everyone! >> >> I have a question ? I need to change the ?From? SIP header during a >> ?Redirect? to make it look like the number that called is a >> different number. >> >> I need to be able to change it but it is not taking ? I have tried >> just about every combination of variable settings that I know of >> but the SIP message is just not getting the data. >> >> Here is my config: >> >> >> >> >> > data="sip_from_user_stripped=false"/> >> > data="sip_from_user=0445555555"/> >> > data="sip_invite_domain=sip:0446666666 at 192.168.8.2"/> >> >> >> > data="myani=0449999999"/> >> >> >> >> >> >> In fact ? it would appear that no channel variables are making it >> out the door during a redirect. >> >> Any help would be greatly appreciated. >> >> Thanks! >> >> Christian Jensen >> Software Development Manager ? Back Office >> Teligence >> T: 604-629-6055 Ext. 3304 >> M: 778-996-4283 >> F: 604-257-5578 >> christian.jensen at teligence.net >> www.teligence.net >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090710/1ce4e939/attachment-0001.html From jens at vegeby.nu Fri Jul 10 09:44:34 2009 From: jens at vegeby.nu (Jens Vegeby) Date: Fri, 10 Jul 2009 18:44:34 +0200 Subject: [Freeswitch-users] Help In-Reply-To: <5E41B7C2-B163-4364-A2A9-8B90AD58F47E@frota.net> References: <5E41B7C2-B163-4364-A2A9-8B90AD58F47E@frota.net> Message-ID: <30ee97110907100944y15d1e90fu29b543cd87f12c5f@mail.gmail.com> You might wanna write what you need help with :) On 7/10/09, Ney Frota wrote: > Help > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 Mvh/Regards Jens From brian at freeswitch.org Fri Jul 10 09:46:08 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 10 Jul 2009 11:46:08 -0500 Subject: [Freeswitch-users] 302 Redirect In-Reply-To: References: <9F8A6292-3BE0-4E29-B331-9EA7A413BE37@freeswitch.org> Message-ID: We really need a wiki page for this. This is handy for those people that do CNAM dips via 302's. /b On Jul 10, 2009, at 11:43 AM, Mathieu Rene wrote: > You can also set the following variables to control where the call > will go once you get a 302: > > sip_redirect_profile (used to build the dialstrings) > sip_redirect_context (the dialplan context to send the call) > sip_redirect_dialplan (the dialplan to send the call) > > Mathieu Rene > Avant-Garde Solutions Inc > Office: + 1 (514) 664-1044 x100 > Cell: +1 (514) 664-1044 x200 > mrene at avgs.ca > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090710/d61a885d/attachment.html From Christian.Jensen at Teligence.Net Fri Jul 10 09:52:00 2009 From: Christian.Jensen at Teligence.Net (Christian Jensen) Date: Fri, 10 Jul 2009 09:52:00 -0700 Subject: [Freeswitch-users] 302 Redirect In-Reply-To: <9F8A6292-3BE0-4E29-B331-9EA7A413BE37@freeswitch.org> References: <9F8A6292-3BE0-4E29-B331-9EA7A413BE37@freeswitch.org> Message-ID: I will give it a try. Gracias! Christian Jensen Software Development Manager Back Office ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Friday, July 10, 2009 9:41 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] 302 Redirect Ok you need the nice new feature I added to FreeSWITCH that lets you handle all the 302 redirects in your own dialplan/context. Set the param manual-redirect on your sofia profile then you can define sip_redirect_profile, sip_redirect_context, sip_redirect_dialplan or you can create a context called redirected and do anything you wish with the 302'ed call 100% manually. You'll get sip_redirect_contact_X sip_redirected_to sip_redirect_contact_user_X sip_redirect_contact_host_X sip_redirect_contact_params_X sip_redirected_by All of these will help you process this via the dialplan. /b On Jul 10, 2009, at 11:32 AM, Christian Jensen wrote: Hi everyone! I have a question - I need to change the "From" SIP header during a "Redirect" to make it look like the number that called is a different number. I need to be able to change it but it is not taking - I have tried just about every combination of variable settings that I know of but the SIP message is just not getting the data. Here is my config: In fact - it would appear that no channel variables are making it out the door during a redirect. Any help would be greatly appreciated. Thanks! Christian Jensen Software Development Manager - Back Office Teligence T: 604-629-6055 Ext. 3304 M: 778-996-4283 F: 604-257-5578 christian.jensen at teligence.net www.teligence.net _______________________________________________ Freeswitch-users mailing list Freeswitch-users at 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/20090710/8f974121/attachment-0001.html From msc at freeswitch.org Fri Jul 10 09:52:03 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 10 Jul 2009 09:52:03 -0700 Subject: [Freeswitch-users] How to configure SIP(Sofia) profiles In-Reply-To: <1452e2980907100536s15818474k901093b742510064@mail.gmail.com> References: <1452e2980907100536s15818474k901093b742510064@mail.gmail.com> Message-ID: <87f2f3b90907100952q104339ebqe26999fb38705136@mail.gmail.com> The wiki has some basic information. If you already have FS installed then you will have 20 "users" pre-configured, 1000-1019, all with password of "1234" so you can use that to set up your softphone. Just be sure to have your soft-phone on a different computer than your FreeSWITCH server! (Technically it can work sometimes but we don't recommend it at all.) See here for more info: http://wiki.freeswitch.org/wiki/Getting_Started_Guide#Some_stuff_to_try_out.21 -MC On Fri, Jul 10, 2009 at 5:36 AM, velusamy velu wrote: > Dear Friends, > I am a newbie for FreeSWITCH. I was installed FreeSWITCH locally. I > just wanted to test whether my FreeSWITCH is working fine. I need help from > you that how to configure my Softphone(Twinkle) to use FreeSWITCH. I need > steps to check my FreeSWITCH working with Twinkle. > Please help me in this... > Thanks in Advance. > > Regards, > K.Velusamy > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090710/ba858d92/attachment.html From larclap at yahoo.com Fri Jul 10 10:35:03 2009 From: larclap at yahoo.com (Lars Zeb) Date: Fri, 10 Jul 2009 10:35:03 -0700 Subject: [Freeswitch-users] Intercom call failing with "No Matching gateway found" error Message-ID: <00f001ca0184$c226e440$4674acc0$@com> Trying to make an intercom call (8+extension#) gives me an error. I don't know what I've done wrong, but I think it used to work. I am on Centos 5 with 14196M. Can someone point me in the right direction? The sofia status, dialplan and log are in http://pastebin.freeswitch.org/9681. Thanks, Lars -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090710/9588972a/attachment.html From pjintheusa at gmail.com Fri Jul 10 10:46:02 2009 From: pjintheusa at gmail.com (Phillip Jones) Date: Fri, 10 Jul 2009 13:46:02 -0400 Subject: [Freeswitch-users] managed_mod directories Message-ID: <367751820907101046l7b41a2fdl7cac3d05b37a80f7@mail.gmail.com> Hi there, Using windows with the pre-compiled binary / msi found via the WIKI Using mod_managed with no problems however: mod_managed appears to require I create a directory 'managed' under C:\Program Files (x86)\FreeSWITCH\mod BUT also requires that I place my .dll in C:\Program Files (x86)\FreeSWITCH\mod and NOT C:\Program Files (x86)\FreeSWITCH\mod\managed thus making C:\Program Files (x86)\FreeSWITCH\mod\managed redundant Anyone else seen this behavior? Thanks! Phillip Jones -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090710/69c2baf0/attachment.html From Christian.Jensen at Teligence.Net Fri Jul 10 10:54:55 2009 From: Christian.Jensen at Teligence.Net (Christian Jensen) Date: Fri, 10 Jul 2009 10:54:55 -0700 Subject: [Freeswitch-users] 302 Redirect In-Reply-To: <9F8A6292-3BE0-4E29-B331-9EA7A413BE37@freeswitch.org> References: <9F8A6292-3BE0-4E29-B331-9EA7A413BE37@freeswitch.org> Message-ID: I think I may not have been clear enough on my original post - not unusual for me :-) The information below is for handling incoming redirects - I am certain that it works perfectly from what I see in the source code - however... What I am trying to do is adjust the "From" header in an Outgoing redirect that I am sending to another device (a nextone). Effectively what is happening is I am receiving a call and then telling the originator that I would like them to go somewhere else but at the same time I am changing the "From" field to look like a different number is calling - changing the Caller Id won't cut it. What I need to be able to do is have a parameter on the "redirect" application or have a "From" field override channel variable. Is this doable? I can build and test from source if need be - I am just not familiar enough with the code to do it the "right" way. Thanks! Christian Jensen Software Development Manager Back Office ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Friday, July 10, 2009 9:41 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] 302 Redirect Ok you need the nice new feature I added to FreeSWITCH that lets you handle all the 302 redirects in your own dialplan/context. Set the param manual-redirect on your sofia profile then you can define sip_redirect_profile, sip_redirect_context, sip_redirect_dialplan or you can create a context called redirected and do anything you wish with the 302'ed call 100% manually. You'll get sip_redirect_contact_X sip_redirected_to sip_redirect_contact_user_X sip_redirect_contact_host_X sip_redirect_contact_params_X sip_redirected_by All of these will help you process this via the dialplan. /b On Jul 10, 2009, at 11:32 AM, Christian Jensen wrote: Hi everyone! I have a question - I need to change the "From" SIP header during a "Redirect" to make it look like the number that called is a different number. I need to be able to change it but it is not taking - I have tried just about every combination of variable settings that I know of but the SIP message is just not getting the data. Here is my config: In fact - it would appear that no channel variables are making it out the door during a redirect. Any help would be greatly appreciated. Thanks! Christian Jensen Software Development Manager - Back Office Teligence T: 604-629-6055 Ext. 3304 M: 778-996-4283 F: 604-257-5578 christian.jensen at teligence.net www.teligence.net _______________________________________________ Freeswitch-users mailing list Freeswitch-users at 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/20090710/6375cf26/attachment-0001.html From mrene_lists at avgs.ca Fri Jul 10 10:59:25 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Fri, 10 Jul 2009 13:59:25 -0400 Subject: [Freeswitch-users] 302 Redirect In-Reply-To: References: <9F8A6292-3BE0-4E29-B331-9EA7A413BE37@freeswitch.org> Message-ID: Add that to your gateway: Then set the callerid. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 10-Jul-09, at 1:54 PM, Christian Jensen wrote: > I think I may not have been clear enough on my original post ? not > unusual for me J > > The information below is for handling incoming redirects ? I am > certain that it works perfectly from what I see in the source code ? > however? > > What I am trying to do is adjust the ?From? header in an Outgoing > redirect that I am sending to another device (a nextone). > > Effectively what is happening is I am receiving a call and then > telling the originator that I would like them to go somewhere else > but at the same time I am changing the ?From? field to look like a > different number is calling ? changing the Caller Id won?t cut it. > > What I need to be able to do is have a parameter on the ?redirect? > application or have a ?From? field override channel variable. > > Is this doable? I can build and test from source if need be ? I am > just not familiar enough with the code to do it the ?right? way. > > Thanks! > > Christian Jensen > Software Development Manager > Back Office > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Brian West > Sent: Friday, July 10, 2009 9:41 AM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] 302 Redirect > > Ok you need the nice new feature I added to FreeSWITCH that lets you > handle all the 302 redirects in your own dialplan/context. > > Set the param manual-redirect on your sofia profile then you can > define sip_redirect_profile, sip_redirect_context, > sip_redirect_dialplan or you can create a context called redirected > and do anything you wish with the 302'ed call 100% manually. > > You'll get > > sip_redirect_contact_X > sip_redirected_to > sip_redirect_contact_user_X > sip_redirect_contact_host_X > sip_redirect_contact_params_X > sip_redirected_by > > All of these will help you process this via the dialplan. > > /b > > On Jul 10, 2009, at 11:32 AM, Christian Jensen wrote: > > > Hi everyone! > > I have a question ? I need to change the ?From? SIP header during a > ?Redirect? to make it look like the number that called is a > different number. > > I need to be able to change it but it is not taking ? I have tried > just about every combination of variable settings that I know of but > the SIP message is just not getting the data. > > Here is my config: > > > > > data="sip_from_user_stripped=false"/> > data="sip_from_user=0445555555"/> > data="sip_invite_domain=sip:0446666666 at 192.168.8.2"/> > > > data="myani=0449999999"/> > > > > > > In fact ? it would appear that no channel variables are making it > out the door during a redirect. > > Any help would be greatly appreciated. > > Thanks! > > Christian Jensen > Software Development Manager ? Back Office > Teligence > T: 604-629-6055 Ext. 3304 > M: 778-996-4283 > F: 604-257-5578 > christian.jensen at teligence.net > www.teligence.net > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090710/a40dfeb5/attachment-0001.html From brian at freeswitch.org Fri Jul 10 11:05:35 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 10 Jul 2009 13:05:35 -0500 Subject: [Freeswitch-users] 302 Redirect In-Reply-To: References: <9F8A6292-3BE0-4E29-B331-9EA7A413BE37@freeswitch.org> Message-ID: <76E5EAB2-66A0-40F8-93A5-6534744376B9@freeswitch.org> Can you show me what we send now vs what you want to send in a sip packet? /b On Jul 10, 2009, at 12:54 PM, Christian Jensen wrote: > I think I may not have been clear enough on my original post ? not > unusual for me J > > The information below is for handling incoming redirects ? I am > certain that it works perfectly from what I see in the source code ? > however? > > What I am trying to do is adjust the ?From? header in an Outgoing > redirect that I am sending to another device (a nextone). > > Effectively what is happening is I am receiving a call and then > telling the originator that I would like them to go somewhere else > but at the same time I am changing the ?From? field to look like a > different number is calling ? changing the Caller Id won?t cut it. > > What I need to be able to do is have a parameter on the ?redirect? > application or have a ?From? field override channel variable. > > Is this doable? I can build and test from source if need be ? I am > just not familiar enough with the code to do it the ?right? way. > > Thanks! > > Christian Jensen > Software Development Manager > Back Office -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090710/afe9e380/attachment.html From msc at freeswitch.org Fri Jul 10 11:26:56 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 10 Jul 2009 11:26:56 -0700 Subject: [Freeswitch-users] Intercom call failing with "No Matching gateway found" error In-Reply-To: <00f001ca0184$c226e440$4674acc0$@com> References: <00f001ca0184$c226e440$4674acc0$@com> Message-ID: <87f2f3b90907101126o1c80fe4fyc86aa4c7944a398f@mail.gmail.com> Lars, If I read your dialplan correctly I believe this line is a problem: Try this: Let us know if that works... -MC On Fri, Jul 10, 2009 at 10:35 AM, Lars Zeb wrote: > Trying to make an intercom call (8+extension#) gives me an error. I don?t > know what I?ve done wrong, but I think it used to work. I am on Centos 5 > with 14196M. > > > > Can someone point me in the right direction? The sofia status, dialplan and > log are in http://pastebin.freeswitch.org/9681. > > > > Thanks, Lars > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090710/8cbd5704/attachment.html From dftoro at yahoo.com Fri Jul 10 12:04:10 2009 From: dftoro at yahoo.com (dftoro at yahoo.com) Date: Fri, 10 Jul 2009 12:04:10 -0700 (PDT) Subject: [Freeswitch-users] managed_mod directories Message-ID: <547062.98354.qm@web33504.mail.mud.yahoo.com> Hi, check whether the dll references to other?dll's, in that case you should put the references in managed directory. ? Check this link http://lists.freeswitch.org/pipermail/freeswitch-dev/2009-April/002232.html?, may be your case ? Diego ? ? --- On Fri, 7/10/09, Phillip Jones wrote: From: Phillip Jones Subject: [Freeswitch-users] managed_mod directories To: freeswitch-users at lists.freeswitch.org Date: Friday, July 10, 2009, 12:46 PM Hi there, Using windows with the pre-compiled binary / msi found via the WIKI Using mod_managed with no problems however: mod_managed appears to require I create a directory 'managed' under C:\Program Files (x86)\FreeSWITCH\mod BUT also requires that I place my .dll in C:\Program Files (x86)\FreeSWITCH\mod and NOT C:\Program Files (x86)\FreeSWITCH\mod\managed thus making C:\Program Files (x86)\FreeSWITCH\mod\managed redundant Anyone else seen this behavior? Thanks! Phillip Jones -----Inline Attachment Follows----- _______________________________________________ Freeswitch-users mailing list Freeswitch-users at 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/20090710/41c965ab/attachment.html From larclap at yahoo.com Fri Jul 10 12:04:55 2009 From: larclap at yahoo.com (Lars Zeb) Date: Fri, 10 Jul 2009 12:04:55 -0700 Subject: [Freeswitch-users] Intercom call failing with "No Matching gateway found" error In-Reply-To: <87f2f3b90907101126o1c80fe4fyc86aa4c7944a398f@mail.gmail.com> References: <00f001ca0184$c226e440$4674acc0$@com> <87f2f3b90907101126o1c80fe4fyc86aa4c7944a398f@mail.gmail.com> Message-ID: <013801ca0191$50388ed0$f0a9ac70$@com> Michael, The extension-intercom is from the conf/dialplan/default.xml. I checked the file in the source tree, and it's the same as I originally used. But I did try your suggestion: . The result was the same after reloadxml. To check if it had been reloaded, I opened conf/freeswitch.xml. I was surprised to see only 64 lines in the file. Something is hosed in my configuration. Should I try to rebuild from scratch and move over my changes to the xml files? Lars From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Friday, July 10, 2009 11:27 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Intercom call failing with "No Matching gateway found" error Lars, If I read your dialplan correctly I believe this line is a problem: Try this: Let us know if that works... -MC On Fri, Jul 10, 2009 at 10:35 AM, Lars Zeb wrote: Trying to make an intercom call (8+extension#) gives me an error. I don't know what I've done wrong, but I think it used to work. I am on Centos 5 with 14196M. Can someone point me in the right direction? The sofia status, dialplan and log are in http://pastebin.freeswitch.org/9681. Thanks, Lars _______________________________________________ Freeswitch-users mailing list Freeswitch-users at 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/20090710/33643197/attachment-0001.html From msc at freeswitch.org Fri Jul 10 12:18:53 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 10 Jul 2009 12:18:53 -0700 Subject: [Freeswitch-users] Intercom call failing with "No Matching gateway found" error In-Reply-To: <013801ca0191$50388ed0$f0a9ac70$@com> References: <00f001ca0184$c226e440$4674acc0$@com> <87f2f3b90907101126o1c80fe4fyc86aa4c7944a398f@mail.gmail.com> <013801ca0191$50388ed0$f0a9ac70$@com> Message-ID: <87f2f3b90907101218l1fba314at62fabb596e0f8603@mail.gmail.com> Yeah, you definitely have an issue somewhere. Move your existing configs to a safe place, reinstall the defaults configs, then slowly merge your customizations back in. I recommend creating a separate dialplan file in conf/dialplan/default/ directory so that you can keep track of your custom stuff. -MC On Fri, Jul 10, 2009 at 12:04 PM, Lars Zeb wrote: > Michael, > > > > The extension-intercom is from the conf/dialplan/default.xml. I checked the > file in the source tree, and it?s the same as I originally used. > > > > But I did try your suggestion: data="user/${dialed_extension}"/>. The result was the same after reloadxml. > > > > To check if it had been reloaded, I opened conf/freeswitch.xml. I was > surprised to see only 64 lines in the file. Something is hosed in my > configuration. Should I try to rebuild from scratch and move over my changes > to the xml files? > > > > Lars > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael > Collins > *Sent:* Friday, July 10, 2009 11:27 AM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Intercom call failing with "No Matching > gateway found" error > > > > Lars, > > If I read your dialplan correctly I believe this line is a problem: > > > Try this: > > > Let us know if that works... > > -MC > > On Fri, Jul 10, 2009 at 10:35 AM, Lars Zeb wrote: > > Trying to make an intercom call (8+extension#) gives me an error. I don?t > know what I?ve done wrong, but I think it used to work. I am on Centos 5 > with 14196M. > > > > Can someone point me in the right direction? The sofia status, dialplan and > log are in http://pastebin.freeswitch.org/9681. > > > > Thanks, Lars > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090710/20d124e0/attachment.html From pjintheusa at gmail.com Fri Jul 10 13:06:27 2009 From: pjintheusa at gmail.com (Phillip Jones) Date: Fri, 10 Jul 2009 16:06:27 -0400 Subject: [Freeswitch-users] managed_mod directories In-Reply-To: <547062.98354.qm@web33504.mail.mud.yahoo.com> References: <547062.98354.qm@web33504.mail.mud.yahoo.com> Message-ID: <367751820907101306n39a71912r2e2361b0b2b17d23@mail.gmail.com> Hi, Thanks for the reply. My DDL is working fine. Just not in the mod\managed directory. What is the mod\managed directory for? It is required but not used? Phil On Fri, Jul 10, 2009 at 3:04 PM, wrote: > Hi, check whether the dll references to other dll's, in that case you > should put the references in managed directory. > > Check this link > http://lists.freeswitch.org/pipermail/freeswitch-dev/2009-April/002232.html , > may be your case > > Diego > > > > --- On *Fri, 7/10/09, Phillip Jones * wrote: > > > From: Phillip Jones > Subject: [Freeswitch-users] managed_mod directories > To: freeswitch-users at lists.freeswitch.org > Date: Friday, July 10, 2009, 12:46 PM > > > Hi there, > > Using windows with the pre-compiled binary / msi found via the WIKI > > Using mod_managed with no problems however: > > mod_managed appears to require I create a directory 'managed' under > C:\Program Files (x86)\FreeSWITCH\mod > > BUT also requires that I place my .dll in C:\Program Files > (x86)\FreeSWITCH\mod and NOT C:\Program Files (x86)\FreeSWITCH\mod\managed > > thus making C:\Program Files (x86)\FreeSWITCH\mod\managed redundant > > Anyone else seen this behavior? > > Thanks! > > > Phillip Jones > > -----Inline Attachment Follows----- > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090710/41571b2d/attachment.html From mgg at giagnocavo.net Fri Jul 10 13:48:02 2009 From: mgg at giagnocavo.net (Michael Giagnocavo) Date: Fri, 10 Jul 2009 16:48:02 -0400 Subject: [Freeswitch-users] managed_mod directories In-Reply-To: <367751820907101046l7b41a2fdl7cac3d05b37a80f7@mail.gmail.com> References: <367751820907101046l7b41a2fdl7cac3d05b37a80f7@mail.gmail.com> Message-ID: <6E8D2069C08AA84A83D336E996AE4C67027C6C2A1C@mse17be1.mse17.exchange.ms> You're saying that it requires the managed DLL to be in both the mod and mod\managed directory? What error do you get if it's only in mod? It's been months, but I just checked loader.cs and it looks explicitly in the managed directory to resolve assemblies as well as to scan to load them. -Michael From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Phillip Jones Sent: Friday, July 10, 2009 11:46 AM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] managed_mod directories Hi there, Using windows with the pre-compiled binary / msi found via the WIKI Using mod_managed with no problems however: mod_managed appears to require I create a directory 'managed' under C:\Program Files (x86)\FreeSWITCH\mod BUT also requires that I place my .dll in C:\Program Files (x86)\FreeSWITCH\mod and NOT C:\Program Files (x86)\FreeSWITCH\mod\managed thus making C:\Program Files (x86)\FreeSWITCH\mod\managed redundant Anyone else seen this behavior? Thanks! Phillip Jones -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090710/e77db563/attachment.html From pjintheusa at gmail.com Fri Jul 10 16:22:06 2009 From: pjintheusa at gmail.com (Phillip Jones) Date: Fri, 10 Jul 2009 19:22:06 -0400 Subject: [Freeswitch-users] managed_mod directories In-Reply-To: <6E8D2069C08AA84A83D336E996AE4C67027C6C2A1C@mse17be1.mse17.exchange.ms> References: <367751820907101046l7b41a2fdl7cac3d05b37a80f7@mail.gmail.com> <6E8D2069C08AA84A83D336E996AE4C67027C6C2A1C@mse17be1.mse17.exchange.ms> Message-ID: <367751820907101622u5b561dd1qcd058c6c649f49c7@mail.gmail.com> It is looking in mod. It required the mod\managed directory, but if I place my dll in mod\managed it fails. DLL must be in mod - mod\managed is empty. My app works fine though Phil On Fri, Jul 10, 2009 at 4:48 PM, Michael Giagnocavo wrote: > You?re saying that it requires the managed DLL to be in both the mod and > mod\managed directory? What error do you get if it?s only in mod? It?s been > months, but I just checked loader.cs and it looks explicitly in the managed > directory to resolve assemblies as well as to scan to load them. > > > > -Michael > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Phillip > Jones > *Sent:* Friday, July 10, 2009 11:46 AM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* [Freeswitch-users] managed_mod directories > > > > Hi there, > > > Using windows with the pre-compiled binary / msi found via the WIKI > > Using mod_managed with no problems however: > > mod_managed appears to require I create a directory 'managed' under > C:\Program Files (x86)\FreeSWITCH\mod > > BUT also requires that I place my .dll in C:\Program Files > (x86)\FreeSWITCH\mod and NOT C:\Program Files (x86)\FreeSWITCH\mod\managed > > thus making C:\Program Files (x86)\FreeSWITCH\mod\managed redundant > > Anyone else seen this behavior? > > Thanks! > > > Phillip Jones > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090710/406fa883/attachment-0001.html From velu.technical at gmail.com Fri Jul 10 22:23:51 2009 From: velu.technical at gmail.com (velusamy velu) Date: Sat, 11 Jul 2009 10:53:51 +0530 Subject: [Freeswitch-users] Error in default Sofia profile checking Message-ID: <1452e2980907102223r681f59e9kad0cf6883cda4df2@mail.gmail.com> Dear Friends, When I register my Softphone(Twinkle) with predefined sofia registration("1000" with password "1234"). I have got the following error in FreeSWITCH console. "2009-07-11 09:37:16 [ERR] sofia_reg.c:1135 sofia_reg_handle_sip_i_register() NO CONTACT!" Please help me to solve this problem... Regards, K.Velusamy. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090711/c97d496b/attachment.html From jason at jasonjgw.net Fri Jul 10 23:59:40 2009 From: jason at jasonjgw.net (Jason White) Date: Sat, 11 Jul 2009 16:59:40 +1000 Subject: [Freeswitch-users] Error in default Sofia profile checking In-Reply-To: <1452e2980907102223r681f59e9kad0cf6883cda4df2@mail.gmail.com> References: <1452e2980907102223r681f59e9kad0cf6883cda4df2@mail.gmail.com> Message-ID: <20090711065940.GA28162@jdc.jasonjgw.net> velusamy velu wrote: > When I register my Softphone(Twinkle) with predefined sofia > registration("1000" with password "1234"). I have got the following error > in FreeSWITCH console. > > "2009-07-11 09:37:16 [ERR] sofia_reg.c:1135 > sofia_reg_handle_sip_i_register() NO CONTACT!" Activate sip tracing on the profile (e.g., sofia profile internal siptrace on), try to register again and save the trace. This should help you to solve the problem. From mrene_lists at avgs.ca Sat Jul 11 00:28:56 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Sat, 11 Jul 2009 03:28:56 -0400 Subject: [Freeswitch-users] Error in default Sofia profile checking In-Reply-To: <1452e2980907102223r681f59e9kad0cf6883cda4df2@mail.gmail.com> References: <1452e2980907102223r681f59e9kad0cf6883cda4df2@mail.gmail.com> Message-ID: <3EFB9073-9339-4A06-BADF-130E69A104CF@avgs.ca> Chances are the registering UA didnt provide a Contact header (required by rfc3261) Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 11-Jul-09, at 1:23 AM, velusamy velu wrote: > Dear Friends, > When I register my Softphone(Twinkle) with predefined > sofia registration("1000" with password "1234"). I have got the > following error in FreeSWITCH console. > > "2009-07-11 09:37:16 [ERR] sofia_reg.c:1135 > sofia_reg_handle_sip_i_ > register() NO CONTACT!" > > Please help me to solve this problem... > > Regards, > K.Velusamy. > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.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 Sat Jul 11 00:48:18 2009 From: jason at jasonjgw.net (Jason White) Date: Sat, 11 Jul 2009 17:48:18 +1000 Subject: [Freeswitch-users] Error in default Sofia profile checking In-Reply-To: <3EFB9073-9339-4A06-BADF-130E69A104CF@avgs.ca> References: <1452e2980907102223r681f59e9kad0cf6883cda4df2@mail.gmail.com> <3EFB9073-9339-4A06-BADF-130E69A104CF@avgs.ca> Message-ID: <20090711074818.GA31938@jdc.jasonjgw.net> Mathieu Rene wrote: > Chances are the registering UA didnt provide a Contact header > (required by rfc3261) Just what I thought, hence the suggestion to obtain a sip trace. From velu.technical at gmail.com Sat Jul 11 01:29:13 2009 From: velu.technical at gmail.com (velusamy velu) Date: Sat, 11 Jul 2009 13:59:13 +0530 Subject: [Freeswitch-users] ERROR in Sofia internal profile Message-ID: <1452e2980907110129i145cbc3fpe53028c92d9b127a@mail.gmail.com> Dear Friends, When I reload the mod_sofia I have got the following error. "2009-07-11 13:19:32 [ERR] sofia.c:739 sofia_profile_thread_run() Error Creating SIP UA for profile: internal" Please any one explain about this error and please give any suggestions to solve this problem.. Thanks in Advance.. Regards, K.Velusamy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090711/12547731/attachment.html From dujinfang at gmail.com Sat Jul 11 02:10:55 2009 From: dujinfang at gmail.com (Seven Du) Date: Sat, 11 Jul 2009 17:10:55 +0800 Subject: [Freeswitch-users] ERROR in Sofia internal profile In-Reply-To: <1452e2980907110129i145cbc3fpe53028c92d9b127a@mail.gmail.com> References: <1452e2980907110129i145cbc3fpe53028c92d9b127a@mail.gmail.com> Message-ID: <096BCE35-ABA3-4CA0-AF59-CCA3806F40BE@gmail.com> chances are the tcp/udp port (5060?) already used by other software, are you running softphone on the same computer? On Jul 11, 2009, at 4:29 PM, velusamy velu wrote: > Dear Friends, > When I reload the mod_sofia I have got the following error. > > "2009-07-11 13:19:32 [ERR] sofia.c:739 sofia_profile_thread_run() > Error Creating SIP UA for profile: internal" > > Please any one explain about this error and please give any > suggestions to solve this problem.. > > Thanks in Advance.. > > Regards, > K.Velusamy > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mcampbellsmith at gmail.com Sat Jul 11 06:15:42 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Sat, 11 Jul 2009 23:15:42 +1000 Subject: [Freeswitch-users] 2 voicemail questions In-Reply-To: <344FC083-5916-4232-AEB5-3504626566DF@jerris.com> References: <33c87fa30907100257w29c628cbnb641d09a75bc22b8@mail.gmail.com> <344FC083-5916-4232-AEB5-3504626566DF@jerris.com> Message-ID: <33c87fa30907110615n3b7b7baclc6276d659fda1e85@mail.gmail.com> Hi Mike, This was my dialplan (extracted from my last email): >> 2. How can I make Freeswitch dial a number AFTER a voicemail is left? >api Hangup hook? i want the 'voicemail' application to appear to call the extension to notify the user that there is a waiting message. This is an extract from my dialplan.xml: This only works if the B leg (ie voicemail application) hangs up first. This would be an unusual situation and does not achieve what I want... is there any other way to achieve this? On Sat, Jul 11, 2009 at 2:04 AM, Michael Jerris wrote: > could you post how you tired to do it in dialplan that didn't work? > > Mike > > On Jul 10, 2009, at 5:57 AM, Mark Campbell-Smith wrote: > >> Hi! >>>> 1. Can I email the voicemail message to multiple email addresses? >> A comma separated list does not work in the extensions.xml file >> (1000.xml), but it does work if I hard code the email addresses into >> the notify-voicemail.tpl file. >> >> Could this be added to the switch so that it can handle comma >> separated lists? >> >>>> 2. How can I make Freeswitch dial a number AFTER a voicemail is >>>> left? >> >>> api Hangup hook? >> >> i want the 'voicemail' application to appear to call the extension to >> notify the user that there is a waiting message. ?This is an extract >> from my dialplan.xml: >> >> ? ? ? >> ? ? ? ? >> ? ? ? ? >> ? ? ? ? >> ? ? ? ? >> ? ? ? ? >> ? ? ? ? >> ? ? ? ?> data="user/${dialed_extension}@${domain_name}"/> >> ? ? ? ? >> ? ? ? ? >> ? ? ? ? >> ? ? ? >> ? ? ?> field="${vm_boxcount(${destination_number}@${domain_name})}" >> expression="^(1)$"> >> ? ? ? ? >> ? ? ? ? >> >> This only works if the B leg (ie voicemail application) hangs up >> first. ?This would be an unusual situation and does not achieve what I >> want... is there any other way to achieve this? >> >> Thanks >> >>> Hi! >>> >>> I have 2 questions regarding voicemail ... >>> >>> 1. Can I email the voicemail message to multiple email addresses? ?If >>> so, what format is this in? >>> ? ? >> >> Try a comma sep. list. ?Not sure if it will work. >> >>> >>> 2. How can I make Freeswitch dial a number AFTER a voicemail is left? >> >> api Hangup hook? >> >> I g >>> From: Brian West >> >>> On Jul 7, 2009, at 9:11 AM, Mark Campbell-Smith wrote: >> >>> Hi! >>> >>> I have 2 questions regarding voicemail ... >>> >>> 1. Can I email the voicemail message to multiple email addresses? ?If >>> so, what format is this in? >>> ? ? >> >>> Try a comma sep. list. ?Not sure if it will work. >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From dftoro at yahoo.com Sat Jul 11 06:15:57 2009 From: dftoro at yahoo.com (Diego Toro) Date: Sat, 11 Jul 2009 06:15:57 -0700 (PDT) Subject: [Freeswitch-users] managed_mod directories Message-ID: <509787.62310.qm@web33504.mail.mud.yahoo.com> Hello, ? What error do you get when dll is put on mod/managed ?, I work with dll's on mod/managed although I changed loadfile by loadfrom on loader.cs. ? Diego --- On Fri, 7/10/09, Phillip Jones wrote: From: Phillip Jones Subject: Re: [Freeswitch-users] managed_mod directories To: freeswitch-users at lists.freeswitch.org Date: Friday, July 10, 2009, 6:22 PM It is looking in mod. It required the mod\managed directory, but if I place my dll in mod\managed it fails. DLL must be in mod - mod\managed is empty. My app works fine though Phil On Fri, Jul 10, 2009 at 4:48 PM, Michael Giagnocavo wrote: You?re saying that it requires the managed DLL to be in both the mod and mod\managed directory? ?What error do you get if it?s only in mod? It?s been months, but I just checked loader.cs and it looks explicitly in the managed directory to resolve assemblies as well as to scan to load them. ? -Michael ? From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Phillip Jones Sent: Friday, July 10, 2009 11:46 AM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] managed_mod directories ? Hi there, Using windows with the pre-compiled binary / msi found via the WIKI Using mod_managed with no problems however: mod_managed appears to require I create a directory 'managed' under C:\Program Files (x86)\FreeSWITCH\mod BUT also requires that I place my .dll in C:\Program Files (x86)\FreeSWITCH\mod and NOT C:\Program Files (x86)\FreeSWITCH\mod\managed thus making C:\Program Files (x86)\FreeSWITCH\mod\managed redundant Anyone else seen this behavior? Thanks! Phillip Jones _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -----Inline Attachment Follows----- _______________________________________________ Freeswitch-users mailing list Freeswitch-users at 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/20090711/c95949a3/attachment-0001.html From Prometheus001 at gmx.net Sat Jul 11 06:46:56 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Sat, 11 Jul 2009 15:46:56 +0200 Subject: [Freeswitch-users] Error in default Sofia profile checking In-Reply-To: <3EFB9073-9339-4A06-BADF-130E69A104CF@avgs.ca> References: <1452e2980907102223r681f59e9kad0cf6883cda4df2@mail.gmail.com> <3EFB9073-9339-4A06-BADF-130E69A104CF@avgs.ca> Message-ID: <4A5897D0.7060005@gmx.net> I have several Twinkles running against freeswitch on a locally installed machine (FS acts as a SIP/TLS proxy). So in general Twinkle works (on various Ubuntu machines from 7 upto 9 with various Twinkle versions). It must be some kind of setting in Twinkle. E.g. * set the local Twinkle SIP UDP port to 5062 in general settings * Set the right network interface (e.g. eth0) * In the profile do not set the realm * Allow missing contact header on 200 OK Best regards Peter Mathieu Rene schrieb: > Chances are the registering UA didnt provide a Contact header > (required by rfc3261) > > Mathieu Rene > Avant-Garde Solutions Inc > Office: + 1 (514) 664-1044 x100 > Cell: +1 (514) 664-1044 x200 > mrene at avgs.ca > > > > > On 11-Jul-09, at 1:23 AM, velusamy velu wrote: > > >> Dear Friends, >> When I register my Softphone(Twinkle) with predefined >> sofia registration("1000" with password "1234"). I have got the >> following error in FreeSWITCH console. >> >> "2009-07-11 09:37:16 [ERR] sofia_reg.c:1135 >> sofia_reg_handle_sip_i_ >> register() NO CONTACT!" >> >> Please help me to solve this problem... >> >> Regards, >> K.Velusamy. >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.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 Jul 11 07:07:29 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 11 Jul 2009 09:07:29 -0500 Subject: [Freeswitch-users] Error in default Sofia profile checking In-Reply-To: <4A5897D0.7060005@gmx.net> References: <1452e2980907102223r681f59e9kad0cf6883cda4df2@mail.gmail.com> <3EFB9073-9339-4A06-BADF-130E69A104CF@avgs.ca> <4A5897D0.7060005@gmx.net> Message-ID: <0FE28F43-2AAA-4928-AF72-73845A129801@freeswitch.org> http://jira.freeswitch.org/browse/MODENDP-86 /b On Jul 11, 2009, at 8:46 AM, Peter P GMX wrote: > I have several Twinkles running against freeswitch on a locally > installed machine (FS acts as a SIP/TLS proxy). > So in general Twinkle works (on various Ubuntu machines from 7 upto 9 > with various Twinkle versions). It must be some kind of setting in > Twinkle. E.g. > > * set the local Twinkle SIP UDP port to 5062 in general settings > * Set the right network interface (e.g. eth0) > * In the profile do not set the realm > * Allow missing contact header on 200 OK > > Best regards > Peter > > > > Mathieu Rene schrieb: >> Chances are the registering UA didnt provide a Contact header >> (required by rfc3261) >> >> Mathieu Rene >> Avant-Garde Solutions Inc >> Office: + 1 (514) 664-1044 x100 >> Cell: +1 (514) 664-1044 x200 >> mrene at avgs.ca >> >> >> >> >> On 11-Jul-09, at 1:23 AM, velusamy velu wrote: >> >> >>> Dear Friends, >>> When I register my Softphone(Twinkle) with predefined >>> sofia registration("1000" with password "1234"). I have got the >>> following error in FreeSWITCH console. >>> >>> "2009-07-11 09:37:16 [ERR] sofia_reg.c:1135 >>> sofia_reg_handle_sip_i_ >>> register() NO CONTACT!" >>> >>> Please help me to solve this problem... >>> >>> Regards, >>> K.Velusamy. >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From pjintheusa at gmail.com Sat Jul 11 09:25:28 2009 From: pjintheusa at gmail.com (Phillip Jones) Date: Sat, 11 Jul 2009 12:25:28 -0400 Subject: [Freeswitch-users] managed_mod directories In-Reply-To: <509787.62310.qm@web33504.mail.mud.yahoo.com> References: <509787.62310.qm@web33504.mail.mud.yahoo.com> Message-ID: <367751820907110925o61168b64t1db5b6944e0dac7e@mail.gmail.com> Hi, If I place the DLL in mod\managed I get the following error: [err] mod_managed.cpp:287 Assembly::LoadFrom failed: system.IO.FileNotFoundException: Could not load file or assembly 'file:///c:program files (x86)\Freeswitch\mod\freeSWITCH.Managed.dll' or one of its dependencies. The system could not find the file specified. As I said. When I place freeSWITCH.Managed.dll straight into \mod then everything works fine. Thanks Phil On Sat, Jul 11, 2009 at 9:15 AM, Diego Toro wrote: > Hello, > > What error do you get when dll is put on mod/managed ?, I work with dll's > on mod/managed although I changed loadfile by loadfrom on loader.cs. > > Diego > > > --- On *Fri, 7/10/09, Phillip Jones * wrote: > > > From: Phillip Jones > Subject: Re: [Freeswitch-users] managed_mod directories > To: freeswitch-users at lists.freeswitch.org > Date: Friday, July 10, 2009, 6:22 PM > > > It is looking in mod. > > It required the mod\managed directory, but if I place my dll in mod\managed > it fails. DLL must be in mod - mod\managed is empty. > > My app works fine though > > > Phil > > > On Fri, Jul 10, 2009 at 4:48 PM, Michael Giagnocavo > > wrote: > >> You?re saying that it requires the managed DLL to be in both the mod and >> mod\managed directory? What error do you get if it?s only in mod? It?s been >> months, but I just checked loader.cs and it looks explicitly in the managed >> directory to resolve assemblies as well as to scan to load them. >> >> -Michael >> >> *From:* freeswitch-users-bounces at lists.freeswitch.org[mailto: >> freeswitch-users-bounces at lists.freeswitch.org] >> *On Behalf Of *Phillip Jones >> *Sent:* Friday, July 10, 2009 11:46 AM >> *To:* freeswitch-users at lists.freeswitch.org >> *Subject:* [Freeswitch-users] managed_mod directories >> >> Hi there, >> >> >> Using windows with the pre-compiled binary / msi found via the WIKI >> >> Using mod_managed with no problems however: >> >> mod_managed appears to require I create a directory 'managed' under >> C:\Program Files (x86)\FreeSWITCH\mod >> >> BUT also requires that I place my .dll in C:\Program Files >> (x86)\FreeSWITCH\mod and NOT C:\Program Files (x86)\FreeSWITCH\mod\managed >> >> thus making C:\Program Files (x86)\FreeSWITCH\mod\managed redundant >> >> Anyone else seen this behavior? >> >> Thanks! >> >> >> Phillip Jones >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -----Inline Attachment Follows----- > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090711/c9820dc7/attachment.html From jlenk at frontiernet.net Sat Jul 11 10:43:43 2009 From: jlenk at frontiernet.net (Jeff Lenk) Date: Sat, 11 Jul 2009 10:43:43 -0700 (PDT) Subject: [Freeswitch-users] managed_mod directories In-Reply-To: <367751820907110925o61168b64t1db5b6944e0dac7e@mail.gmail.com> References: <367751820907101046l7b41a2fdl7cac3d05b37a80f7@mail.gmail.com> <509787.62310.qm@web33504.mail.mud.yahoo.com> <367751820907110925o61168b64t1db5b6944e0dac7e@mail.gmail.com> Message-ID: <1247334223093-3243183.post@n2.nabble.com> Hi, The base dll (FreeSWITCH.Managed.dll) is loaded from /mod - additional managed dlls are loaded from /mod/managed. This is designed to allow your dll's to be built and maintained independant of the FS build files. You can simply just drop your dlls into mod/managed and they will be loaded and available for use(this happens at FS startup). The base managed dll (FreeSWITCH.Managed.dll) is only really supposed to be used for loader support and the demo classes - you should place your code in your own dll. - Jeff Phillip Jones-2 wrote: > > Hi, > > If I place the DLL in mod\managed I get the following error: > > [err] mod_managed.cpp:287 Assembly::LoadFrom failed: > system.IO.FileNotFoundException: Could not load file or assembly > 'file:///c:program files (x86)\Freeswitch\mod\freeSWITCH.Managed.dll' or > one > of its dependencies. The system could not find the file specified. > > As I said. When I place freeSWITCH.Managed.dll straight into \mod then > everything works fine. > > Thanks > > > Phil > > > On Sat, Jul 11, 2009 at 9:15 AM, Diego Toro wrote: > >> Hello, >> >> What error do you get when dll is put on mod/managed ?, I work with dll's >> on mod/managed although I changed loadfile by loadfrom on loader.cs. >> >> Diego >> >> >> --- On *Fri, 7/10/09, Phillip Jones * wrote: >> >> >> From: Phillip Jones >> Subject: Re: [Freeswitch-users] managed_mod directories >> To: freeswitch-users at lists.freeswitch.org >> Date: Friday, July 10, 2009, 6:22 PM >> >> >> It is looking in mod. >> >> It required the mod\managed directory, but if I place my dll in >> mod\managed >> it fails. DLL must be in mod - mod\managed is empty. >> >> My app works fine though >> >> >> Phil >> >> >> On Fri, Jul 10, 2009 at 4:48 PM, Michael Giagnocavo >> >> > wrote: >> >>> You?re saying that it requires the managed DLL to be in both the mod >>> and >>> mod\managed directory? What error do you get if it?s only in mod? It?s >>> been >>> months, but I just checked loader.cs and it looks explicitly in the >>> managed >>> directory to resolve assemblies as well as to scan to load them. >>> >>> -Michael >>> >>> *From:* >>> freeswitch-users-bounces at lists.freeswitch.org[mailto: >>> freeswitch-users-bounces at lists.freeswitch.org] >>> *On Behalf Of *Phillip Jones >>> *Sent:* Friday, July 10, 2009 11:46 AM >>> *To:* >>> freeswitch-users at lists.freeswitch.org >>> *Subject:* [Freeswitch-users] managed_mod directories >>> >>> Hi there, >>> >>> >>> Using windows with the pre-compiled binary / msi found via the WIKI >>> >>> Using mod_managed with no problems however: >>> >>> mod_managed appears to require I create a directory 'managed' under >>> C:\Program Files (x86)\FreeSWITCH\mod >>> >>> BUT also requires that I place my .dll in C:\Program Files >>> (x86)\FreeSWITCH\mod and NOT C:\Program Files >>> (x86)\FreeSWITCH\mod\managed >>> >>> thus making C:\Program Files (x86)\FreeSWITCH\mod\managed redundant >>> >>> Anyone else seen this behavior? >>> >>> Thanks! >>> >>> >>> Phillip Jones >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> -----Inline Attachment Follows----- >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/managed_mod-directories-tp3240303p3243183.html Sent from the freeswitch-users mailing list archive at Nabble.com. From nik.middleton at noblesolutions.co.uk Sat Jul 11 12:19:02 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Sat, 11 Jul 2009 20:19:02 +0100 Subject: [Freeswitch-users] Setting channel variables using event socket Message-ID: Hi Guys, Is it possible to set a channel variable while a call is in progress using an outbound event socket? I have a listening process that examines the hang-up events and it would be neat if it could also get some variables that I have set mid call as well. Note: I know it's possible to set them in the originate but that's not what I'm after Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090711/41c4f3df/attachment-0001.html From brian at freeswitch.org Sat Jul 11 12:50:57 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 11 Jul 2009 14:50:57 -0500 Subject: [Freeswitch-users] Setting channel variables using event socket In-Reply-To: References: Message-ID: <7525E12C-1213-4502-8993-F01300D5D644@freeswitch.org> uuid_setvar /b On Jul 11, 2009, at 2:19 PM, Nik Middleton wrote: > Hi Guys, > > Is it possible to set a channel variable while a call is in progress > using an outbound event socket? I have a listening process that > examines the hang-up events and it would be neat if it could also > get some variables that I have set mid call as well. Note: I know > it?s possible to set them in the originate but that?s not what I?m > after > > Regards, > ________ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090711/7e248e5c/attachment.html From nik.middleton at noblesolutions.co.uk Sat Jul 11 14:08:50 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Sat, 11 Jul 2009 22:08:50 +0100 Subject: [Freeswitch-users] Setting channel variables using event socket In-Reply-To: <7525E12C-1213-4502-8993-F01300D5D644@freeswitch.org> References: <7525E12C-1213-4502-8993-F01300D5D644@freeswitch.org> Message-ID: Excellent. Do I need to supply uuid on an outbound socket? Regards ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: 11 July 2009 20:51 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Setting channel variables using event socket uuid_setvar /b On Jul 11, 2009, at 2:19 PM, Nik Middleton wrote: Hi Guys, Is it possible to set a channel variable while a call is in progress using an outbound event socket? I have a listening process that examines the hang-up events and it would be neat if it could also get some variables that I have set mid call as well. Note: I know it's possible to set them in the originate but that's not what I'm after Regards, ________ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090711/0212f069/attachment.html From brian at freeswitch.org Sat Jul 11 14:21:42 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 11 Jul 2009 16:21:42 -0500 Subject: [Freeswitch-users] Setting channel variables using event socket In-Reply-To: References: <7525E12C-1213-4502-8993-F01300D5D644@freeswitch.org> Message-ID: I think you do ... /b On Jul 11, 2009, at 4:08 PM, Nik Middleton wrote: > Excellent. Do I need to supply uuid on an outbound socket? > > Regards > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090711/17242f4c/attachment.html From msc at freeswitch.org Sat Jul 11 14:50:55 2009 From: msc at freeswitch.org (Michael S Collins) Date: Sat, 11 Jul 2009 14:50:55 -0700 Subject: [Freeswitch-users] Setting channel variables using event socket In-Reply-To: References: <7525E12C-1213-4502-8993-F01300D5D644@freeswitch.org> Message-ID: <0E283956-3903-4822-9BD9-684803A15EC4@freeswitch.org> IIRC you need to supply the uuid because the socket doesn't make any assumptions about the APIs you send. -MC Sent from my iPhone On Jul 11, 2009, at 2:21 PM, Brian West wrote: > I think you do ... > > /b > > On Jul 11, 2009, at 4:08 PM, Nik Middleton wrote: > >> Excellent. Do I need to supply uuid on an outbound socket? >> >> 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/20090711/4cde569e/attachment.html From anthony.minessale at gmail.com Sat Jul 11 14:51:03 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sat, 11 Jul 2009 16:51:03 -0500 Subject: [Freeswitch-users] Setting channel variables using event socket In-Reply-To: References: <7525E12C-1213-4502-8993-F01300D5D644@freeswitch.org> Message-ID: <191c3a030907111451u7ddb17c3r66cc4371c58cc63d@mail.gmail.com> I think also sendmsg with command execute works in this case if you are using async socket but uuid_setvar always works in all cases On Jul 11, 2009 4:27 PM, "Brian West" wrote: I think you do ... /b On Jul 11, 2009, at 4:08 PM, Nik Middleton wrote: > Excellent. Do I need to supply uuid on an out... _______________________________________________ Freeswitch-users mailing list Freeswitch-users at 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/20090711/acdb945f/attachment-0001.html From elihayun at gmail.com Sun Jul 12 01:49:20 2009 From: elihayun at gmail.com (Eli Hayun) Date: Sun, 12 Jul 2009 11:49:20 +0300 Subject: [Freeswitch-users] Getting xml_request in LUA Message-ID: <4A59A390.7030000@savion.huji.ac.il> In the Perl example I found: How to access request parameters and how to return data You have two hashes that are populated for you by freeswitch. Those hashes are: * %XML_REQUEST * %XML_DATA I want to use LUA to set the directory and dialplan xml. How do I get the XML_REQUEST/XML_DATA from LUA? Thanks Eli Hayun From info at nalawo.com Sun Jul 12 06:54:57 2009 From: info at nalawo.com (Maarten De Maeyer) Date: Sun, 12 Jul 2009 13:54:57 +0000 (GMT) Subject: [Freeswitch-users] QSIG Message-ID: <1964152093.1076.1247406897795.JavaMail.mail@webmail25> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090712/97dafa9b/attachment.html From dome at tel.co.th Sun Jul 12 09:59:05 2009 From: dome at tel.co.th (Dome Charoenyost) Date: Sun, 12 Jul 2009 23:59:05 +0700 Subject: [Freeswitch-users] Originate in Dial plan Message-ID: <8ccbff060907120959ma7c0ey22adeae196f872c3@mail.gmail.com> Dear sir, I want to user dialplan callback to customer. is posible to to this is dialplan XML ? Now i use javascript. my call flow. 1. customer call 2. FS rining and wait until customer hangup 3. callback to customer number Best Regards. Dome C. From mike at jerris.com Sun Jul 12 09:58:15 2009 From: mike at jerris.com (Michael Jerris) Date: Sun, 12 Jul 2009 12:58:15 -0400 Subject: [Freeswitch-users] QSIG In-Reply-To: <1964152093.1076.1247406897795.JavaMail.mail@webmail25> References: <1964152093.1076.1247406897795.JavaMail.mail@webmail25> Message-ID: We have no qsig support. On Jul 12, 2009, at 9:54 AM, Maarten De Maeyer wrote: > > Hi, > > Can someone tell me how complete QSIG support is in FS ? Is there a > config example available ? I need to connect FS to another pbx with > QSIG. > Any tips/advice more than welcome. > > Thanks. > > MdM > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090712/e7c73975/attachment.html From q.edward at gmail.com Sun Jul 12 13:45:16 2009 From: q.edward at gmail.com (Edward Q.) Date: Sun, 12 Jul 2009 16:45:16 -0400 Subject: [Freeswitch-users] outbound_caller_id dynamic from mysql ? Message-ID: <89313a90907121345v76a1fe41o35fd70753dded41c@mail.gmail.com> Hi guys. I was just wondering if it is possible to have the outbound_caller_id dynamically pulled from MySQL db ? If it is can anyone please point me in the right direction ? Thanks in advanced to all for all your help. Thanks Ed -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090712/1594cec7/attachment.html From mike at jerris.com Sun Jul 12 13:48:46 2009 From: mike at jerris.com (Michael Jerris) Date: Sun, 12 Jul 2009 16:48:46 -0400 Subject: [Freeswitch-users] outbound_caller_id dynamic from mysql ? In-Reply-To: <89313a90907121345v76a1fe41o35fd70753dded41c@mail.gmail.com> References: <89313a90907121345v76a1fe41o35fd70753dded41c@mail.gmail.com> Message-ID: See mod_xml_curl On Jul 12, 2009, at 4:45 PM, "Edward Q." wrote: > Hi guys. > > I was just wondering if it is possible to have the > outbound_caller_id dynamically pulled from MySQL db ? > If it is can anyone please point me in the right direction ? > Thanks in advanced to all for all your help. > Thanks > Ed > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From q.edward at gmail.com Sun Jul 12 13:59:13 2009 From: q.edward at gmail.com (Edward Q.) Date: Sun, 12 Jul 2009 16:59:13 -0400 Subject: [Freeswitch-users] outbound_caller_id dynamic from mysql ? In-Reply-To: References: <89313a90907121345v76a1fe41o35fd70753dded41c@mail.gmail.com> Message-ID: <89313a90907121359n7aa219afoc14e45da55ff60af@mail.gmail.com> Thank you for the prompt reply Michael .. Looking through the wiki now Thanks to all and have all of you a great day. Ed On Sun, Jul 12, 2009 at 4:48 PM, Michael Jerris wrote: > See mod_xml_curl > > On Jul 12, 2009, at 4:45 PM, "Edward Q." wrote: > > > Hi guys. > > > > I was just wondering if it is possible to have the > > outbound_caller_id dynamically pulled from MySQL db ? > > If it is can anyone please point me in the right direction ? > > Thanks in advanced to all for all your help. > > Thanks > > Ed > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090712/13c52525/attachment.html From nik.middleton at noblesolutions.co.uk Sun Jul 12 15:10:37 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Sun, 12 Jul 2009 23:10:37 +0100 Subject: [Freeswitch-users] Setting channel variables using event socket In-Reply-To: <191c3a030907111451u7ddb17c3r66cc4371c58cc63d@mail.gmail.com> References: <7525E12C-1213-4502-8993-F01300D5D644@freeswitch.org> <191c3a030907111451u7ddb17c3r66cc4371c58cc63d@mail.gmail.com> Message-ID: HI Guys, Can't seem to get this to work call-command: execute execute-app-name: uuid_setvar execute-app-arg: cad8eb99-cdcd-4d0d-9453-20b8d9d71859 fred=out_to_lunch Tried various permutations, but still nothing stored when the channel is hung up Can anyone tell me what I'm doing wrong? Regards, ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: 11 July 2009 22:51 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Setting channel variables using event socket I think also sendmsg with command execute works in this case if you are using async socket but uuid_setvar always works in all cases On Jul 11, 2009 4:27 PM, "Brian West" wrote: I think you do ... /b On Jul 11, 2009, at 4:08 PM, Nik Middleton wrote: > Excellent. Do I need to supply uuid on an out... _______________________________________________ Freeswitch-users mailing list Freeswitch-users at 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/20090712/99431a7c/attachment-0001.html From brian at freeswitch.org Sun Jul 12 15:23:45 2009 From: brian at freeswitch.org (Brian West) Date: Sun, 12 Jul 2009 17:23:45 -0500 Subject: [Freeswitch-users] Setting channel variables using event socket In-Reply-To: References: <7525E12C-1213-4502-8993-F01300D5D644@freeswitch.org> <191c3a030907111451u7ddb17c3r66cc4371c58cc63d@mail.gmail.com> Message-ID: If you're going to do it that way you can just use set. uuid_setvar is an api call... /b On Jul 12, 2009, at 5:10 PM, Nik Middleton wrote: > HI Guys, > > Can?t seem to get this to work > > call-command: execute > execute-app-name: uuid_setvar > execute-app-arg: cad8eb99-cdcd-4d0d-9453-20b8d9d71859 > fred=out_to_lunch > > Tried various permutations, but still nothing stored when the > channel is hung up > > Can anyone tell me what I?m doing wrong? > > Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090712/f6a62091/attachment.html From nik.middleton at noblesolutions.co.uk Sun Jul 12 15:33:45 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Sun, 12 Jul 2009 23:33:45 +0100 Subject: [Freeswitch-users] Setting channel variables using event socket In-Reply-To: References: <7525E12C-1213-4502-8993-F01300D5D644@freeswitch.org><191c3a030907111451u7ddb17c3r66cc4371c58cc63d@mail.gmail.com> Message-ID: As in call-command: set joe=out_to_lunch ? ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: 12 July 2009 23:24 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Setting channel variables using event socket If you're going to do it that way you can just use set. uuid_setvar is an api call... /b On Jul 12, 2009, at 5:10 PM, Nik Middleton wrote: HI Guys, Can't seem to get this to work call-command: execute execute-app-name: uuid_setvar execute-app-arg: cad8eb99-cdcd-4d0d-9453-20b8d9d71859 fred=out_to_lunch Tried various permutations, but still nothing stored when the channel is hung up Can anyone tell me what I'm doing wrong? Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090712/fea97d65/attachment.html From brian at freeswitch.org Sun Jul 12 16:00:24 2009 From: brian at freeswitch.org (Brian West) Date: Sun, 12 Jul 2009 18:00:24 -0500 Subject: [Freeswitch-users] Setting channel variables using event socket In-Reply-To: References: <7525E12C-1213-4502-8993-F01300D5D644@freeswitch.org><191c3a030907111451u7ddb17c3r66cc4371c58cc63d@mail.gmail.com> Message-ID: call-command: execute execute-app-name: set execute-app-arg: fred=out_to_lunch On Jul 12, 2009, at 5:33 PM, Nik Middleton wrote: > As in > > call-command: set joe=out_to_lunch ? > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090712/14d15cc6/attachment.html From timuckun at gmail.com Sun Jul 12 21:07:06 2009 From: timuckun at gmail.com (Tim Uckun) Date: Mon, 13 Jul 2009 16:07:06 +1200 Subject: [Freeswitch-users] Dialogic cards Message-ID: <855e4dcf0907122107q392a9ca1h851dfbd57e2518ca@mail.gmail.com> We have some older dialogic cards (D300 series E1 cards) and I am wondering if freeswitch can support these cards. Thanks. From velu.technical at gmail.com Sun Jul 12 22:38:48 2009 From: velu.technical at gmail.com (velusamy velu) Date: Mon, 13 Jul 2009 11:08:48 +0530 Subject: [Freeswitch-users] Error in default Sofia profile checking Message-ID: <1452e2980907122238h1392b903o6b1179e423fb7893@mail.gmail.com> Dear Peter, I have followed your steps, For me my FS and Twinkle running in separate machine. But, I am still receiving the same error "[ERR] sofia_reg.c:1135 sofia_reg_handle_sip_i_register() NO CONTACT!" Please give any suggestions to rectify this error.. Thanks in Advance, Regards, K.Velusamy. > > > On Jul 11, 2009, at 8:46 AM, Peter P GMX wrote: > > > I have several Twinkles running against freeswitch on a locally > > installed machine (FS acts as a SIP/TLS proxy). > > So in general Twinkle works (on various Ubuntu machines from 7 upto 9 > > with various Twinkle versions). It must be some kind of setting in > > Twinkle. E.g. > > > > * set the local Twinkle SIP UDP port to 5062 in general settings > > * Set the right network interface (e.g. eth0) > > * In the profile do not set the realm > > * Allow missing contact header on 200 OK > > > > Best regards > > Peter > > > > > > > > Mathieu Rene schrieb: > >> Chances are the registering UA didnt provide a Contact header > >> (required by rfc3261) > >> > >> Mathieu Rene > >> Avant-Garde Solutions Inc > >> Office: + 1 (514) 664-1044 x100 > >> Cell: +1 (514) 664-1044 x200 > >> mrene at avgs.ca > >> > >> > >> > >> > >> On 11-Jul-09, at 1:23 AM, velusamy velu wrote: > >> > >> > >>> Dear Friends, > >>> When I register my Softphone(Twinkle) with predefined > >>> sofia registration("1000" with password "1234"). I have got the > >>> following error in FreeSWITCH console. > >>> > >>> "2009-07-11 09:37:16 [ERR] sofia_reg.c:1135 > >>> sofia_reg_handle_sip_i_ > >>> register() NO CONTACT!" > >>> > >>> Please help me to solve this problem... > >>> > >>> Regards, > >>> K.Velusamy. > >>> _______________________________________________ > >>> Freeswitch-users mailing list > >>> Freeswitch-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> http://www.freeswitch.org > >>> > >> > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > >> > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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/20090713/54b29a3f/attachment-0001.html From sridhart at alcatel-lucent.com Mon Jul 13 00:07:26 2009 From: sridhart at alcatel-lucent.com (Rajagopal, Sridhar (Sridhar)) Date: Mon, 13 Jul 2009 12:37:26 +0530 Subject: [Freeswitch-users] Help Regarding memory leak with freeswitch Message-ID: <9389DD3DDD6B9144B147CE564C6599B9082FC07020@INBANSXCHMBSA3.in.alcatel-lucent.com> Hi all, I am running freeswitch on powerpc processor. I see memory being allocated for each subsequent REGISTER requests coming to freeswitch. But not all the memory allocated is not freed. If I run the code for two days the system is running out of memory (RAM available to me is very less). The same memory issue is happening even for calls. Please let me know if any body has seen this issue. Please let me know how I can go ahead and debug this issue. Thanks in advance for the help. Regards, Sridhar -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090713/9fc85a3d/attachment.html From helmut.kuper at ewetel.de Mon Jul 13 00:53:51 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Mon, 13 Jul 2009 09:53:51 +0200 Subject: [Freeswitch-users] pocketsphinx In-Reply-To: <4A574F0D.7040604@gmx.net> References: <4A5739DE.1080800@ewetel.de> <4A574F0D.7040604@gmx.net> Message-ID: <4A5AE80F.1000904@ewetel.de> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi Peter, hmmm well, I had the same idea and I tested it! Buuuuut ... you have to make sure that the english grammar/acousticModel is able to cover all german noises. E.g. I tried to detect "Burke", "Jan" and "Gerd". I was able to map Burke successfully in default.dic. Otherwise I had to say "B?hrki" ... I did the same with "Jan" - but when I tried to detect Jan I always got Gerd (with and witout mapping ind default.dic) ... Quite strange and not really usable for (german) customers. But some typical software magic happened on my way: During my tests I had somehow a configuration using the voxforge files which was working within FS. But I can't reproduce it. I configured serveral files at the same time and used for reloading "reloadxml" and "reload mod_pocketsphinx" instead of rebooting FS. When it worked, FS was able to detect "Burke", "Jan" and "Gerd" correctly without modifying the dictionary... Is there any manual about pocketsphinx and its config files, which can explain how PS is working in more detail? Currently I walk with a flashlight in the dark ... regards Helmut On 10.07.2009 16:24, Peter P GMX wrote: > Hello Helmut, > > I looked at these dic files. Their content (look at all the qq's) is > quite different from the dic files supplied with freeswitch pocketsphinx. > As I remember the CMU dict file format has changed in April 2008. > Maybe there is a converter somewhere? > > I was thinking of just enhancing the current dict file for some german > words I need, but did not test it so far. This should be possible > without modifying the underlying grammar. > http://en.wikipedia.org/wiki/CMU_Pronouncing_Dictionary > I would love to hear when you have had any progress on this. > > Best regards > Peter -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) iD8DBQFKWugP4tZeNddg3dwRAhjkAKConTWen4bq5BxSg23F6keZeY2CIACffAks yyOVZOkROr8tfUNGMv4t9o8= =lC+W -----END PGP SIGNATURE----- From nik.middleton at noblesolutions.co.uk Mon Jul 13 01:51:32 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Mon, 13 Jul 2009 09:51:32 +0100 Subject: [Freeswitch-users] Help Regarding memory leak with freeswitch In-Reply-To: <9389DD3DDD6B9144B147CE564C6599B9082FC07020@INBANSXCHMBSA3.in.alcatel-lucent.com> References: <9389DD3DDD6B9144B147CE564C6599B9082FC07020@INBANSXCHMBSA3.in.alcatel-lucent.com> Message-ID: If this is Linux, there's nothing wrong with it using most of the memory, if it starts to use the swap, then there might be an issue. Utilizing the memory does not mean there is a memory leak Regards, ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rajagopal, Sridhar (Sridhar) Sent: 13 July 2009 08:07 To: 'freeswitch-users at lists.freeswitch.org' Subject: [Freeswitch-users] Help Regarding memory leak with freeswitch Hi all, I am running freeswitch on powerpc processor. I see memory being allocated for each subsequent REGISTER requests coming to freeswitch. But not all the memory allocated is not freed. If I run the code for two days the system is running out of memory (RAM available to me is very less). The same memory issue is happening even for calls. Please let me know if any body has seen this issue. Please let me know how I can go ahead and debug this issue. Thanks in advance for the help. Regards, Sridhar -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090713/c870589a/attachment.html From jingwei.yang at gmail.com Mon Jul 13 02:27:21 2009 From: jingwei.yang at gmail.com (Jingwei Yang) Date: Mon, 13 Jul 2009 17:27:21 +0800 Subject: [Freeswitch-users] mod_dingaling no audio In-Reply-To: <507898380906290346w20cfea6bq82114efc63439f52@mail.gmail.com> References: <13529f9d0906240240x55f0c19ao7246de1d34b1f661@mail.gmail.com> <13529f9d0906242042j598a2a21v3f32c2d99a28b58@mail.gmail.com> <507898380906242053p155e16edve8bc797c22f11c12@mail.gmail.com> <13529f9d0906242231s7384908ayb30340f5442b2ab@mail.gmail.com> <507898380906250602r4c666b06xfdeea5943ad0cbff@mail.gmail.com> <13529f9d0906251915m440a1c46s534a5c7c735b8ce8@mail.gmail.com> <507898380906251930v6f82d786l9d1bf0da61212b05@mail.gmail.com> <13529f9d0906252034o5cb0b7b7y8a9ffe750edf2b20@mail.gmail.com> <13529f9d0906290225v4abd1552l4a3b72e0ff1afc66@mail.gmail.com> <507898380906290346w20cfea6bq82114efc63439f52@mail.gmail.com> Message-ID: <13529f9d0907130227u377b7459l8143a37b9ccfc761@mail.gmail.com> Hi Chris, sorry for the late reply. Have been quite busy last few days. I had shifted 888 from default.xml to public.xml and the dialplan is simply having an echo action now. I've turned on dl_debug but unfortunately didn't find anything useful. Logs are attached for your reference. I don't think there's something wrong with the dialplan as two external parties can talk to each other perfectly (with ext-rtp-ip uncommented, at this time my ip was interpreted to be an external one). With ext-rtp-ip commented, I can hear the echo and I saw my ip was translated into an internal one (at this time, external party's audio failed). I tried the method on this wiki page as well: http://wiki.freeswitch.org/wiki/NAT_Traversal (the last FreeSwitch behind NAT portion) but still no luck. Please kindly let me know what other configs I should change. Thanks, -Jingwei On Mon, Jun 29, 2009 at 6:46 PM, Chris Chen wrote: > Jingwei, I don't know if you have the 888 defined in default.xml? also you > have to define $${domain}. > please do " dl_debug on" from fs_cli, and watch the console logs and see > what's going on when you try calling from external. Most likely your > dialplan is not correctly defined. > > Chris > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090713/d5061ea6/attachment-0001.html -------------- next part -------------- freeswitch at localhost.localdomain> originate dingaling/gmail.com/xxxxxx at gmail.com &echo 2009-07-13 15:19:34.950696 [NOTICE] libdingaling.c:1309 SecSEND: ------------------------------------------------------------------------------- Call Me! 2009-07-13 15:19:34.950696 [NOTICE] libdingaling.c:1309 SecSEND: ------------------------------------------------------------------------------- 2009-07-13 15:19:35.643651 [INFO] libdingaling.c:1307 SecRECV: ------------------------------------------------------------------------------- 24 2009-07-13 15:19:35.647665 [NOTICE] switch_channel.c:602 New Channel dingaling/gmail.com/xxxxxx at gmail.com [dde6f921-ed98-46c0-9628-8f2c0d1b2835] 2009-07-13 15:19:35.648943 [NOTICE] mod_dingaling.c:1084 Ring-Ready dingaling/gmail.com/xxxxxx at gmail.com! 2009-07-13 15:19:35.651651 [NOTICE] libdingaling.c:1309 SecSEND: ------------------------------------------------------------------------------- Incoming Call From FreeSWITCH 0000000000 2009-07-13 15:19:35.651651 [NOTICE] libdingaling.c:1309 SecSEND: ------------------------------------------------------------------------------- 2009-07-13 15:19:35.662651 [INFO] libdingaling.c:1307 SecRECV: ------------------------------------------------------------------------------- 24 2009-07-13 15:19:36.458618 [INFO] libdingaling.c:1307 SecRECV: ------------------------------------------------------------------------------- 2009-07-13 15:19:36.631611 [INFO] libdingaling.c:1307 SecRECV: ------------------------------------------------------------------------------- 2009-07-13 15:19:36.657730 [NOTICE] libdingaling.c:1309 SecSEND: ------------------------------------------------------------------------------- 2009-07-13 15:19:36.872615 [INFO] libdingaling.c:1307 SecRECV: ------------------------------------------------------------------------------- 2009-07-13 15:19:36.959598 [NOTICE] libdingaling.c:1309 SecSEND: ------------------------------------------------------------------------------- 2009-07-13 15:19:37.14595 [INFO] libdingaling.c:1307 SecRECV: ------------------------------------------------------------------------------- 2009-07-13 15:19:37.60593 [NOTICE] libdingaling.c:1309 SecSEND: ------------------------------------------------------------------------------- 2009-07-13 15:19:40.791443 [INFO] libdingaling.c:1307 SecRECV: ------------------------------------------------------------------------------- 2009-07-13 15:19:41.680379 [INFO] mod_dingaling.c:974 Stun Success 59.189.194.106:26746 2009-07-13 15:19:41.691378 [NOTICE] mod_dingaling.c:1142 Channel [dingaling/gmail.com/xxxxxx at gmail.com] has been answered API CALL [originate(dingaling/gmail.com/xxxxxx at gmail.com &echo)] output: +OK dde6f921-ed98-46c0-9628-8f2c0d1b2835 freeswitch at localhost.localdomain> 2009-07-13 15:19:41.762376 [NOTICE] libdingaling.c:1309 SecSEND: ------------------------------------------------------------------------------- 2009-07-13 15:19:41.762376 [NOTICE] libdingaling.c:1309 SecSEND: ------------------------------------------------------------------------------- 2009-07-13 15:19:42.502589 [INFO] libdingaling.c:1307 SecRECV: ------------------------------------------------------------------------------- 2009-07-13 15:19:43.672335 [INFO] libdingaling.c:1307 SecRECV: ------------------------------------------------------------------------------- 2009-07-13 15:19:43.762360 [NOTICE] libdingaling.c:1309 SecSEND: ------------------------------------------------------------------------------- 2009-07-13 15:19:44.499288 [INFO] libdingaling.c:1307 SecRECV: ------------------------------------------------------------------------------- 2009-07-13 15:19:44.562353 [NOTICE] libdingaling.c:1309 SecSEND: ------------------------------------------------------------------------------- 2009-07-13 15:19:44.691280 [INFO] libdingaling.c:1307 SecRECV: ------------------------------------------------------------------------------- 2009-07-13 15:19:44.762321 [NOTICE] libdingaling.c:1309 SecSEND: ------------------------------------------------------------------------------- 2009-07-13 15:19:45.498291 [INFO] libdingaling.c:1307 SecRECV: ------------------------------------------------------------------------------- 2009-07-13 15:19:45.562530 [NOTICE] libdingaling.c:1309 SecSEND: ------------------------------------------------------------------------------- 2009-07-13 15:19:52.922946 [INFO] libdingaling.c:1307 SecRECV: ------------------------------------------------------------------------------- 2009-07-13 15:19:52.922946 [NOTICE] mod_dingaling.c:718 Hangup dingaling/gmail.com/xxxxxx at gmail.com [CS_EXECUTE] [NORMAL_CLEARING] 2009-07-13 15:19:52.936942 [NOTICE] switch_core_session.c:1085 Session 2 (dingaling/gmail.com/xxxxxx at gmail.com) Ended 2009-07-13 15:19:52.936942 [NOTICE] switch_core_session.c:1087 Close Channel dingaling/gmail.com/xxxxxx at gmail.com [CS_DESTROY] 2009-07-13 15:19:52.961986 [NOTICE] libdingaling.c:1309 SecSEND: ------------------------------------------------------------------------------- 2009-07-13 15:19:52.961986 [NOTICE] libdingaling.c:1309 SecSEND: ------------------------------------------------------------------------------- 2009-07-13 15:19:53.702945 [INFO] libdingaling.c:1307 SecRECV: ------------------------------------------------------------------------------- unknown session From Prometheus001 at gmx.net Mon Jul 13 02:30:14 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Mon, 13 Jul 2009 11:30:14 +0200 Subject: [Freeswitch-users] Error in default Sofia profile checking In-Reply-To: <1452e2980907122238h1392b903o6b1179e423fb7893@mail.gmail.com> References: <1452e2980907122238h1392b903o6b1179e423fb7893@mail.gmail.com> Message-ID: <4A5AFEA6.9000806@gmx.net> Which twinkle version are you using? I use Twinkle 1.0.1 and my SIP register looks as follows. As you can see, the contact header is there. U 127.0.0.1:5062 -> 127.0.0.1:5060 REGISTER sip:127.0.0.1 SIP/2.0. Via: SIP/2.0/UDP 192.168.178.146:5062;rport;branch=z9hG4bKtvnvzdwy. Max-Forwards: 70. To: "8353310" . From: "8353310" ;tag=avpju. Call-ID: ibubkykiithqlne at 192.168.178.146. CSeq: 5792 REGISTER. Contact: ;expires=60. Authorization: Digest username="8353310",realm="127.0.0.1",nonce="4bcfe1b0-6f8f-11de-bc32-2dff86a04420",uri="sip:127.0.0.1",response="922690317852a402052da6f74f7196df",algorithm=MD5,cnonce="k9662kmk64",qop=auth,nc=00000001. Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO. User-Agent: Twinkle/1.0.1. Content-Length: 0. . # U 127.0.0.1:5060 -> 127.0.0.1:5062 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.178.146:5062;rport=5062;branch=z9hG4bKtvnvzdwy;received=127.0.0.1. From: "8353310" ;tag=avpju. To: "8353310" ;tag=4p5K211F33N2c. Call-ID: ibubkykiithqlne at 192.168.178.146. CSeq: 5792 REGISTER. Contact: ;expires=60. Date: Mon, 13 Jul 2009 09:26:51 GMT. User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-12955M. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. Supported: timer, precondition, path, replaces. Content-Length: 0. Can you ngrep your traffic and port your register request? ngrep -d any port 5060 -W byline Best regards Peter velusamy velu schrieb: > > Dear Peter, > I have followed your steps, For me my FS and Twinkle running > in separate machine. But, I am still receiving the same error > "[ERR] sofia_reg.c:1135 > sofia_reg_handle_sip_i_register() NO CONTACT!" > > Please give any suggestions to rectify this error.. > > Thanks in Advance, > > Regards, > K.Velusamy. > > > > On Jul 11, 2009, at 8:46 AM, Peter P GMX wrote: > > > I have several Twinkles running against freeswitch on a locally > > installed machine (FS acts as a SIP/TLS proxy). > > So in general Twinkle works (on various Ubuntu machines from 7 > upto 9 > > with various Twinkle versions). It must be some kind of setting in > > Twinkle. E.g. > > > > * set the local Twinkle SIP UDP port to 5062 in general settings > > * Set the right network interface (e.g. eth0) > > * In the profile do not set the realm > > * Allow missing contact header on 200 OK > > > > Best regards > > Peter > > > > > > > > Mathieu Rene schrieb: > >> Chances are the registering UA didnt provide a Contact header > >> (required by rfc3261) > >> > >> Mathieu Rene > >> Avant-Garde Solutions Inc > >> Office: + 1 (514) 664-1044 x100 > >> Cell: +1 (514) 664-1044 x200 > >> mrene at avgs.ca > >> > >> > >> > >> > >> On 11-Jul-09, at 1:23 AM, velusamy velu wrote: > >> > >> > >>> Dear Friends, > >>> When I register my Softphone(Twinkle) with predefined > >>> sofia registration("1000" with password "1234"). I have got the > >>> following error in FreeSWITCH console. > >>> > >>> "2009-07-11 09:37:16 [ERR] sofia_reg.c:1135 > >>> sofia_reg_handle_sip_i_ > >>> register() NO CONTACT!" > >>> > >>> Please help me to solve this problem... > >>> > >>> Regards, > >>> K.Velusamy. > >>> _______________________________________________ > >>> Freeswitch-users mailing list > >>> Freeswitch-users at lists.freeswitch.org > > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> http://www.freeswitch.org > >>> > >> > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > >> > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From chris.chen2004 at gmail.com Mon Jul 13 04:29:23 2009 From: chris.chen2004 at gmail.com (Chris Chen) Date: Mon, 13 Jul 2009 07:29:23 -0400 Subject: [Freeswitch-users] mod_dingaling no audio In-Reply-To: <13529f9d0907130227u377b7459l8143a37b9ccfc761@mail.gmail.com> References: <13529f9d0906240240x55f0c19ao7246de1d34b1f661@mail.gmail.com> <507898380906242053p155e16edve8bc797c22f11c12@mail.gmail.com> <13529f9d0906242231s7384908ayb30340f5442b2ab@mail.gmail.com> <507898380906250602r4c666b06xfdeea5943ad0cbff@mail.gmail.com> <13529f9d0906251915m440a1c46s534a5c7c735b8ce8@mail.gmail.com> <507898380906251930v6f82d786l9d1bf0da61212b05@mail.gmail.com> <13529f9d0906252034o5cb0b7b7y8a9ffe750edf2b20@mail.gmail.com> <13529f9d0906290225v4abd1552l4a3b72e0ff1afc66@mail.gmail.com> <507898380906290346w20cfea6bq82114efc63439f52@mail.gmail.com> <13529f9d0907130227u377b7459l8143a37b9ccfc761@mail.gmail.com> Message-ID: <507898380907130429haeb7ef9haeaf31b7e409da5c@mail.gmail.com> Jingwei, can you show your console log when somebody is calling you from gtalk client? Will it really hit 888 in your dialplan? Thanks, Chris On Mon, Jul 13, 2009 at 5:27 AM, Jingwei Yang wrote: > Hi Chris, sorry for the late reply. Have been quite busy last few days. > > I had shifted 888 from default.xml to public.xml and the dialplan is simply > having an echo action now. I've turned on dl_debug but unfortunately didn't > find anything useful. Logs are attached for your reference. > > I don't think there's something wrong with the dialplan as two external > parties can talk to each other perfectly (with ext-rtp-ip uncommented, at > this time my ip was interpreted to be an external one). With ext-rtp-ip > commented, I can hear the echo and I saw my ip was translated into an > internal one (at this time, external party's audio failed). > > I tried the method on this wiki page as well: > http://wiki.freeswitch.org/wiki/NAT_Traversal (the last FreeSwitch behind > NAT portion) but still no luck. Please kindly let me know what other configs > I should change. > > Thanks, > -Jingwei > > On Mon, Jun 29, 2009 at 6:46 PM, Chris Chen wrote: > >> Jingwei, I don't know if you have the 888 defined in default.xml? also >> you have to define $${domain}. >> please do " dl_debug on" from fs_cli, and watch the console logs and see >> what's going on when you try calling from external. Most likely your >> dialplan is not correctly defined. >> >> Chris >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090713/69780b63/attachment.html From Prometheus001 at gmx.net Mon Jul 13 05:55:00 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Mon, 13 Jul 2009 14:55:00 +0200 Subject: [Freeswitch-users] pocketsphinx In-Reply-To: <4A5AE80F.1000904@ewetel.de> References: <4A5739DE.1080800@ewetel.de> <4A574F0D.7040604@gmx.net> <4A5AE80F.1000904@ewetel.de> Message-ID: <4A5B2EA4.8050608@gmx.net> Hello Helmut, the 3 mentioned words are already part of the englisch standard dictionary, so maybe this causes the problem? You may test with words which are outside of the standard grammar files or delete the original ones? So far I have no other documentation available. This part of PocketSphinx is rather poorly documented. And for the FS part I've only got some information from this mailing list. Best regards Peter Helmut Kuper schrieb: > Hi Peter, > > hmmm well, I had the same idea and I tested it! Buuuuut ... you have to > make sure that the english grammar/acousticModel is able to cover all > german noises. E.g. I tried to detect "Burke", "Jan" and "Gerd". I was > able to map Burke successfully in default.dic. Otherwise I had to say > "B?hrki" ... I did the same with "Jan" - but when I tried to detect Jan > I always got Gerd (with and witout mapping ind default.dic) ... Quite > strange and not really usable for (german) customers. > > But some typical software magic happened on my way: > During my tests I had somehow a configuration using the voxforge files > which was working within FS. But I can't reproduce it. I configured > serveral files at the same time and used for reloading "reloadxml" and > "reload mod_pocketsphinx" instead of rebooting FS. When it worked, FS > was able to detect "Burke", "Jan" and "Gerd" correctly without modifying > the dictionary... > > > Is there any manual about pocketsphinx and its config files, which can > explain how PS is working in more detail? Currently I walk with a > flashlight in the dark ... > > regards > Helmut > > > On 10.07.2009 16:24, Peter P GMX wrote: > > Hello Helmut, > > > I looked at these dic files. Their content (look at all the qq's) is > > quite different from the dic files supplied with freeswitch > pocketsphinx. > > As I remember the CMU dict file format has changed in April 2008. > > Maybe there is a converter somewhere? > > > I was thinking of just enhancing the current dict file for some german > > words I need, but did not test it so far. This should be possible > > without modifying the underlying grammar. > > http://en.wikipedia.org/wiki/CMU_Pronouncing_Dictionary > > I would love to hear when you have had any progress on this. > > > Best regards > > Peter _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From andy at fabulous4.co.uk Mon Jul 13 05:59:55 2009 From: andy at fabulous4.co.uk (Andy) Date: Mon, 13 Jul 2009 13:59:55 +0100 Subject: [Freeswitch-users] Problems with Ping and re-registering broken gateways Message-ID: <0324DD608A074940AAC173B85A3978F2@D810> Hi, I'm fairly sure my problem lies with my voip provider VoipTalk but wonder if you could help me understand a couple of things. My config is very simple, I'm using freeswitch to accept incoming calls via a voip gateway and record messages. Here's the problem: - When freeswitch starts the gateways are all created and register correctly - I have the ping parameter set to make sure the gateway stays alive. - The first time freeswitch pings the gateway it fails even though the registration appears intact as calls are still coming through to freeswitch - Freeswitch then tries to re-register the gateway but this fails. The SIP trace shows an Unauthorized message and the actual log entry is 'Registration Failed with status Operation has no matching challenge [904]' - eventually the registration times out with my provider and all is lost. - if I call 'sofia profile external restart' or restart the software this fixes the problem My questions are: 1) Why would the ping fail when the registration appears to be intact? 2) Whay would the auto re-register not work but a restart would? This ones driving me nuts so any help greatly appreciated. regards Andy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090713/f4e8db59/attachment.html From brian at freeswitch.org Mon Jul 13 06:27:25 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 13 Jul 2009 08:27:25 -0500 Subject: [Freeswitch-users] Error in default Sofia profile checking In-Reply-To: <4A5AFEA6.9000806@gmx.net> References: <1452e2980907122238h1392b903o6b1179e423fb7893@mail.gmail.com> <4A5AFEA6.9000806@gmx.net> Message-ID: Its not a bug... its just something we do not support in FreeSWITCH yet... Register with no contact is a fetch operation. /b On Jul 13, 2009, at 4:30 AM, Peter P GMX wrote: > Which twinkle version are you using? I use Twinkle 1.0.1 and my SIP > register looks as follows. As you can see, the contact header is > there. > > U 127.0.0.1:5062 -> 127.0.0.1:5060 > REGISTER sip:127.0.0.1 SIP/2.0. > Via: SIP/2.0/UDP 192.168.178.146:5062;rport;branch=z9hG4bKtvnvzdwy. > Max-Forwards: 70. > To: "8353310" . > From: "8353310" ;tag=avpju. > Call-ID: ibubkykiithqlne at 192.168.178.146. > CSeq: 5792 REGISTER. > Contact: ;expires=60. > Authorization: Digest > username="8353310",realm="127.0.0.1",nonce="4bcfe1b0-6f8f-11de- > bc32-2dff86a04420",uri="sip: > 127.0.0.1 > ",response > = > "922690317852a402052da6f74f7196df > ",algorithm=MD5,cnonce="k9662kmk64",qop=auth,nc=00000001. > Allow: > INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO. > User-Agent: Twinkle/1.0.1. > Content-Length: 0. > . > > # > U 127.0.0.1:5060 -> 127.0.0.1:5062 > SIP/2.0 200 OK. > Via: SIP/2.0/UDP > 192.168.178.146 > :5062;rport=5062;branch=z9hG4bKtvnvzdwy;received=127.0.0.1. > From: "8353310" ;tag=avpju. > To: "8353310" ;tag=4p5K211F33N2c. > Call-ID: ibubkykiithqlne at 192.168.178.146. > CSeq: 5792 REGISTER. > Contact: ;expires=60. > Date: Mon, 13 Jul 2009 09:26:51 GMT. > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-12955M. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. > Supported: timer, precondition, path, replaces. > Content-Length: 0. > > Can you ngrep your traffic and port your register request? > ngrep -d any port 5060 -W byline > > > Best regards > Peter > > velusamy velu schrieb: >> >> Dear Peter, >> I have followed your steps, For me my FS and Twinkle running >> in separate machine. But, I am still receiving the same error >> "[ERR] sofia_reg.c:1135 >> sofia_reg_handle_sip_i_register() NO CONTACT!" >> >> Please give any suggestions to rectify this error.. >> >> Thanks in Advance, >> >> Regards, >> K.Velusamy. >> >> >> >> On Jul 11, 2009, at 8:46 AM, Peter P GMX wrote: >> >>> I have several Twinkles running against freeswitch on a locally >>> installed machine (FS acts as a SIP/TLS proxy). >>> So in general Twinkle works (on various Ubuntu machines from 7 >> upto 9 >>> with various Twinkle versions). It must be some kind of setting in >>> Twinkle. E.g. >>> >>> * set the local Twinkle SIP UDP port to 5062 in general settings >>> * Set the right network interface (e.g. eth0) >>> * In the profile do not set the realm >>> * Allow missing contact header on 200 OK >>> >>> Best regards >>> Peter >>> >>> >>> >>> Mathieu Rene schrieb: >>>> Chances are the registering UA didnt provide a Contact header >>>> (required by rfc3261) >>>> >>>> Mathieu Rene >>>> Avant-Garde Solutions Inc >>>> Office: + 1 (514) 664-1044 x100 >>>> Cell: +1 (514) 664-1044 x200 >>>> mrene at avgs.ca >>>> >>>> >>>> >>>> >>>> On 11-Jul-09, at 1:23 AM, velusamy velu wrote: >>>> >>>> >>>>> Dear Friends, >>>>> When I register my Softphone(Twinkle) with predefined >>>>> sofia registration("1000" with password "1234"). I have got the >>>>> following error in FreeSWITCH console. >>>>> >>>>> "2009-07-11 09:37:16 [ERR] sofia_reg.c:1135 >>>>> sofia_reg_handle_sip_i_ >>>>> register() NO CONTACT!" >>>>> >>>>> Please help me to solve this problem... >>>>> >>>>> Regards, >>>>> K.Velusamy. >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.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 Jul 13 08:08:26 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 13 Jul 2009 10:08:26 -0500 Subject: [Freeswitch-users] Dialogic cards In-Reply-To: <855e4dcf0907122107q392a9ca1h851dfbd57e2518ca@mail.gmail.com> References: <855e4dcf0907122107q392a9ca1h851dfbd57e2518ca@mail.gmail.com> Message-ID: <191c3a030907130808s37e52ddg3ba567d6a5a1034b@mail.gmail.com> Dialogic is coming to ClueCon this year (this aug 4th) and they are sponsoring the conference. I can discuss the possibility of supporting their cards at that time. On Sun, Jul 12, 2009 at 11:07 PM, Tim Uckun wrote: > We have some older dialogic cards (D300 series E1 cards) and I am > wondering if freeswitch can support these cards. > > 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/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference 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/20090713/25e82735/attachment.html From anthony.minessale at gmail.com Mon Jul 13 08:20:53 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 13 Jul 2009 10:20:53 -0500 Subject: [Freeswitch-users] Help Regarding memory leak with freeswitch In-Reply-To: <9389DD3DDD6B9144B147CE564C6599B9082FC07020@INBANSXCHMBSA3.in.alcatel-lucent.com> References: <9389DD3DDD6B9144B147CE564C6599B9082FC07020@INBANSXCHMBSA3.in.alcatel-lucent.com> Message-ID: <191c3a030907130820y46985521y1a01270a98cc2fcf@mail.gmail.com> Which revision of FreeSWITCH are you using? Several memory leaks have been fixed since the last formal release. One specifically in REGISTER. You should probably try SVN trunk or the latest pre-release of 1.0.4 On Mon, Jul 13, 2009 at 2:07 AM, Rajagopal, Sridhar (Sridhar) < sridhart at alcatel-lucent.com> wrote: > Hi all, > > I am running freeswitch on powerpc processor. I see memory being allocated > for each subsequent REGISTER requests coming to freeswitch. But not all the > memory allocated is not freed. If I run the code for two days the system is > running out of memory (RAM available to me is very less). > The same memory issue is happening even for calls. > > Please let me know if any body has seen this issue. Please let me know how > I can go ahead and debug this issue. > > Thanks in advance for the help. > > Regards, > Sridhar > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference 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/20090713/b97f4578/attachment.html From anthony.minessale at gmail.com Mon Jul 13 08:25:10 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 13 Jul 2009 10:25:10 -0500 Subject: [Freeswitch-users] Setting channel variables using event socket In-Reply-To: References: <7525E12C-1213-4502-8993-F01300D5D644@freeswitch.org> <191c3a030907111451u7ddb17c3r66cc4371c58cc63d@mail.gmail.com> Message-ID: <191c3a030907130825q178fdfa9q1f3098258288ed3b@mail.gmail.com> and if you go the uuid_setvar route you do this: api uuid_setvar joe=out_to_lunch On Sun, Jul 12, 2009 at 6:00 PM, Brian West wrote: > call-command: execute > execute-app-name: set > execute-app-arg: fred=out_to_lunch > > > On Jul 12, 2009, at 5:33 PM, Nik Middleton wrote: > > As in > > call-command: set joe=out_to_lunch ? > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference 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/20090713/113023df/attachment.html From steveu at coppice.org Mon Jul 13 08:39:39 2009 From: steveu at coppice.org (Steve Underwood) Date: Mon, 13 Jul 2009 23:39:39 +0800 Subject: [Freeswitch-users] Dialogic cards In-Reply-To: <855e4dcf0907122107q392a9ca1h851dfbd57e2518ca@mail.gmail.com> References: <855e4dcf0907122107q392a9ca1h851dfbd57e2518ca@mail.gmail.com> Message-ID: <4A5B553B.9020000@coppice.org> Tim Uckun wrote: > We have some older dialogic cards (D300 series E1 cards) and I am > wondering if freeswitch can support these cards. > Oh, I like the easy questions. No. It lacks the hardware features to do anything useful with Freeswitch, or Asterisk, or Callweaver, or Yate, or anything else that expects to do two way telephony through the host CPU. Steve From steveu at coppice.org Mon Jul 13 08:39:47 2009 From: steveu at coppice.org (Steve Underwood) Date: Mon, 13 Jul 2009 23:39:47 +0800 Subject: [Freeswitch-users] Dialogic cards In-Reply-To: <191c3a030907130808s37e52ddg3ba567d6a5a1034b@mail.gmail.com> References: <855e4dcf0907122107q392a9ca1h851dfbd57e2518ca@mail.gmail.com> <191c3a030907130808s37e52ddg3ba567d6a5a1034b@mail.gmail.com> Message-ID: <4A5B5543.6080405@coppice.org> Being now a mashup of several CTI companies, there are now a number of disparate things called Dialogic cards. Some, like the cards previously known as Prince..... er, Eicon are perfectly supportable. The old Dialogic cards, like the D300 series, are not duplex to and from the host. They are only duplex across the mezzanine bus. There's nothing you can do with them. Steve Anthony Minessale wrote: > Dialogic is coming to ClueCon this year (this aug 4th) and they are > sponsoring the conference. > I can discuss the possibility of supporting their cards at that time. > > > On Sun, Jul 12, 2009 at 11:07 PM, Tim Uckun > wrote: > > We have some older dialogic cards (D300 series E1 cards) and I am > wondering if freeswitch can support these cards. > > Thanks. > From anthony.minessale at gmail.com Mon Jul 13 08:46:58 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 13 Jul 2009 10:46:58 -0500 Subject: [Freeswitch-users] Dialogic cards In-Reply-To: <4A5B553B.9020000@coppice.org> References: <855e4dcf0907122107q392a9ca1h851dfbd57e2518ca@mail.gmail.com> <4A5B553B.9020000@coppice.org> Message-ID: <191c3a030907130846u4c134fc4n578589c196617557@mail.gmail.com> ok, or we could ask Steve I guess. =D On Mon, Jul 13, 2009 at 10:39 AM, Steve Underwood wrote: > Tim Uckun wrote: > > We have some older dialogic cards (D300 series E1 cards) and I am > > wondering if freeswitch can support these cards. > > > Oh, I like the easy questions. No. It lacks the hardware features to do > anything useful with Freeswitch, or Asterisk, or Callweaver, or Yate, or > anything else that expects to do two way telephony through the host CPU. > > 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/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference 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/20090713/2912d77b/attachment-0001.html From valentin.doroga at pronexus.com Mon Jul 13 08:57:07 2009 From: valentin.doroga at pronexus.com (Valentin Doroga) Date: Mon, 13 Jul 2009 11:57:07 -0400 Subject: [Freeswitch-users] Dialogic cards In-Reply-To: <4A5B5543.6080405@coppice.org> Message-ID: <20090713155712.OYAB273.tomts27-srv.bellnexxia.net@toip37-bus.srvr.bell.ca> Maybe Dialogic would add support for "thin blades", currently used for HMP (DNI series). Val. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steve Underwood Sent: Monday, July 13, 2009 11:40 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Dialogic cards Being now a mashup of several CTI companies, there are now a number of disparate things called Dialogic cards. Some, like the cards previously known as Prince..... er, Eicon are perfectly supportable. The old Dialogic cards, like the D300 series, are not duplex to and from the host. They are only duplex across the mezzanine bus. There's nothing you can do with them. Steve Anthony Minessale wrote: > Dialogic is coming to ClueCon this year (this aug 4th) and they are > sponsoring the conference. > I can discuss the possibility of supporting their cards at that time. > > > On Sun, Jul 12, 2009 at 11:07 PM, Tim Uckun > wrote: > > We have some older dialogic cards (D300 series E1 cards) and I am > wondering if freeswitch can support these cards. > > 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 From anthony.minessale at gmail.com Mon Jul 13 08:59:28 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 13 Jul 2009 10:59:28 -0500 Subject: [Freeswitch-users] Problems with Ping and re-registering broken gateways In-Reply-To: <0324DD608A074940AAC173B85A3978F2@D810> References: <0324DD608A074940AAC173B85A3978F2@D810> Message-ID: <191c3a030907130859y2665da03xf704f9db6070fb3c@mail.gmail.com> Are they ignoring the options packet we send them or are they maybe getting lost behind NAT? we send an OPTIONS and even if we get a error back we consider that a successful reply. We did have a patch into SVN very recently to correct a problem with OPTIONS ping in a NAT situation. Maybe try latest trunk first then capture the console log with sip traffic in place if it still does not work so we can have a look. to capture the log use these 2 commands from the cli. sofia profile internal siptrace on console loglevel debug On Mon, Jul 13, 2009 at 7:59 AM, Andy wrote: > Hi, > > I'm fairly sure my problem lies with my voip provider VoipTalk but wonder > if you could help me understand a couple of things. My config is very > simple, I'm using freeswitch to accept incoming calls via a voip gateway and > record messages. Here's the problem: > > - When freeswitch starts the gateways are all created and register > correctly > > - I have the ping parameter set to make sure the gateway stays alive. > > - The first time freeswitch pings the gateway it fails even though the > registration appears intact as calls are still coming through to freeswitch > > - Freeswitch then tries to re-register the gateway but this fails. The SIP > trace shows an Unauthorized message and the actual log entry is > 'Registration Failed with status Operation has no matching challenge [904]' > > - eventually the registration times out with my provider and all is lost. > > - if I call 'sofia profile external restart' or restart the software this > fixes the problem > > My questions are: > > 1) Why would the ping fail when the registration appears to be intact? > 2) Whay would the auto re-register not work but a restart would? > > This ones driving me nuts so any help greatly appreciated. > > regards > Andy > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference 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/20090713/d54f755e/attachment.html From pjintheusa at gmail.com Mon Jul 13 09:03:17 2009 From: pjintheusa at gmail.com (Phillip Jones) Date: Mon, 13 Jul 2009 12:03:17 -0400 Subject: [Freeswitch-users] managed_mod directories In-Reply-To: <1247334223093-3243183.post@n2.nabble.com> References: <367751820907101046l7b41a2fdl7cac3d05b37a80f7@mail.gmail.com> <509787.62310.qm@web33504.mail.mud.yahoo.com> <367751820907110925o61168b64t1db5b6944e0dac7e@mail.gmail.com> <1247334223093-3243183.post@n2.nabble.com> Message-ID: <367751820907130903p36734e0eyb010da5409dfb765@mail.gmail.com> Got it! Thanks very much for that clarification. Phil On Sat, Jul 11, 2009 at 1:43 PM, Jeff Lenk wrote: > > Hi, > > The base dll (FreeSWITCH.Managed.dll) is loaded from /mod - additional > managed dlls are loaded from /mod/managed. This is designed to allow your > dll's to be built and maintained independant of the FS build files. You can > simply just drop your dlls into mod/managed and they will be loaded and > available for use(this happens at FS startup). > > The base managed dll (FreeSWITCH.Managed.dll) is only really supposed to be > used for loader support and the demo classes - you should place your code > in > your own dll. > > - Jeff > > > Phillip Jones-2 wrote: > > > > Hi, > > > > If I place the DLL in mod\managed I get the following error: > > > > [err] mod_managed.cpp:287 Assembly::LoadFrom failed: > > system.IO.FileNotFoundException: Could not load file or assembly > > 'file:///c:program files (x86)\Freeswitch\mod\freeSWITCH.Managed.dll' or > > one > > of its dependencies. The system could not find the file specified. > > > > As I said. When I place freeSWITCH.Managed.dll straight into \mod then > > everything works fine. > > > > Thanks > > > > > > Phil > > > > > > On Sat, Jul 11, 2009 at 9:15 AM, Diego Toro wrote: > > > >> Hello, > >> > >> What error do you get when dll is put on mod/managed ?, I work with > dll's > >> on mod/managed although I changed loadfile by loadfrom on loader.cs. > >> > >> Diego > >> > >> > >> --- On *Fri, 7/10/09, Phillip Jones * wrote: > >> > >> > >> From: Phillip Jones > >> Subject: Re: [Freeswitch-users] managed_mod directories > >> To: freeswitch-users at lists.freeswitch.org > >> Date: Friday, July 10, 2009, 6:22 PM > >> > >> > >> It is looking in mod. > >> > >> It required the mod\managed directory, but if I place my dll in > >> mod\managed > >> it fails. DLL must be in mod - mod\managed is empty. > >> > >> My app works fine though > >> > >> > >> Phil > >> > >> > >> On Fri, Jul 10, 2009 at 4:48 PM, Michael Giagnocavo > >> http://us.mc335.mail.yahoo.com/mc/compose?to=mgg at giagnocavo.net> > >> > wrote: > >> > >>> You?re saying that it requires the managed DLL to be in both the mod > >>> and > >>> mod\managed directory? What error do you get if it?s only in mod? It?s > >>> been > >>> months, but I just checked loader.cs and it looks explicitly in the > >>> managed > >>> directory to resolve assemblies as well as to scan to load them. > >>> > >>> -Michael > >>> > >>> *From:* > >>> freeswitch-users-bounces at lists.freeswitch.org< > http://us.mc335.mail.yahoo.com/mc/compose?to=freeswitch-users-bounces at lists.freeswitch.org > >[mailto: > >>> freeswitch-users-bounces at lists.freeswitch.org< > http://us.mc335.mail.yahoo.com/mc/compose?to=freeswitch-users-bounces at lists.freeswitch.org > >] > >>> *On Behalf Of *Phillip Jones > >>> *Sent:* Friday, July 10, 2009 11:46 AM > >>> *To:* > >>> freeswitch-users at lists.freeswitch.org< > http://us.mc335.mail.yahoo.com/mc/compose?to=freeswitch-users at lists.freeswitch.org > > > >>> *Subject:* [Freeswitch-users] managed_mod directories > >>> > >>> Hi there, > >>> > >>> > >>> Using windows with the pre-compiled binary / msi found via the WIKI > >>> > >>> Using mod_managed with no problems however: > >>> > >>> mod_managed appears to require I create a directory 'managed' under > >>> C:\Program Files (x86)\FreeSWITCH\mod > >>> > >>> BUT also requires that I place my .dll in C:\Program Files > >>> (x86)\FreeSWITCH\mod and NOT C:\Program Files > >>> (x86)\FreeSWITCH\mod\managed > >>> > >>> thus making C:\Program Files (x86)\FreeSWITCH\mod\managed redundant > >>> > >>> Anyone else seen this behavior? > >>> > >>> Thanks! > >>> > >>> > >>> Phillip Jones > >>> > >>> _______________________________________________ > >>> Freeswitch-users mailing list > >>> Freeswitch-users at lists.freeswitch.org< > http://us.mc335.mail.yahoo.com/mc/compose?to=Freeswitch-users at lists.freeswitch.org > > > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> http://www.freeswitch.org > >>> > >>> > >> > >> -----Inline Attachment Follows----- > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org< > http://us.mc335.mail.yahoo.com/mc/compose?to=Freeswitch-users at lists.freeswitch.org > > > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > >> > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > >> > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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/managed_mod-directories-tp3240303p3243183.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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090713/fbfd9270/attachment-0001.html From excelsio at gmx.net Mon Jul 13 09:26:59 2009 From: excelsio at gmx.net (excelsio at gmx.net) Date: Mon, 13 Jul 2009 18:26:59 +0200 Subject: [Freeswitch-users] DID with 10 numbers Message-ID: <20090713162659.129030@gmx.net> Hi, I purchased a block of 10 did numbers. Base number is 01234/56789. The numbers themselves range from 01234/567890 to 01234/567899 What works? Well, I can dial in to a "hardcoded" 01234/56789 which belongs to user 1000. I can?t dial out. The main problem is, that I do not know how I can assing those numbers to the users. Of course I want to dial out with each user and the corresponding number should be displayed. Also I want to dial in to the appropriate number. I can?t find an example within the wiki, where there a several numbers to be configured. It would be great if you could give me some hints or links. thanks in advance Michael -- Neu: GMX Doppel-FLAT mit Internet-Flatrate + Telefon-Flatrate f?r nur 19,99 Euro/mtl.!* http://portal.gmx.net/de/go/dsl02 From mrene_lists at avgs.ca Mon Jul 13 09:31:26 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Mon, 13 Jul 2009 12:31:26 -0400 Subject: [Freeswitch-users] DID with 10 numbers In-Reply-To: <20090713162659.129030@gmx.net> References: <20090713162659.129030@gmx.net> Message-ID: You need to define variables within the user's entry, you can then re- use those in the dialplan to route the call using the proper line. If you are using a single trunk you can set the effective_caller_id_number to the number you want to call from and it'll set the callerid accordignly. PS: Meld dich bei unserem irc-channel und wir koennen dir einfacher weiter helfen. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 13-Jul-09, at 12:26 PM, excelsio at gmx.net wrote: > Hi, > > I purchased a block of 10 did numbers. Base number is 01234/56789. > The numbers themselves range from 01234/567890 to 01234/567899 > > What works? Well, I can dial in to a "hardcoded" 01234/56789 which > belongs to user 1000. > I can?t dial out. > > The main problem is, that I do not know how I can assing those > numbers to the users. > Of course I want to dial out with each user and the corresponding > number should be displayed. Also I want to dial in to the > appropriate number. > > I can?t find an example within the wiki, where there a several > numbers to be configured. > > It would be great if you could give me some hints or links. > > thanks in advance > > Michael > -- > Neu: GMX Doppel-FLAT mit Internet-Flatrate + Telefon-Flatrate > f?r nur 19,99 Euro/mtl.!* http://portal.gmx.net/de/go/dsl02 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From vkozak at abisoft.spb.ru Mon Jul 13 10:08:20 2009 From: vkozak at abisoft.spb.ru (Kozak Vladimir) Date: Mon, 13 Jul 2009 21:08:20 +0400 Subject: [Freeswitch-users] FS not wait respond from called and send 200Ok References: <5F5ABAC4334545CE8323DACB10805DE1@abisoft.biz> <74C6337E-6F03-4485-B69B-372658E0D79B@jerris.com> Message-ID: Please, look at these logs. Fist invite witchout SDP (from 1007 at uat.agent.starpoundtech.net to vk_1008 at uat.agent.starpoundtech.net) not wait respond from 1008 and send 200 Ok to 1007. Called phone didn't accept call. ip:ports 172.26.200.252:5071 - userAgent (starpound), originator call 172.26.200.252:5080 - FS external profile 172.26.200.252:5090 - FS doubleNat profile 172.26.10.65:38464 - phone1 (1007, registered on FS) 172.26.10.65:17748 - phone2 (1008, registered on FS) ----- Original Message ----- From: Michael Jerris To: freeswitch-users at lists.freeswitch.org Sent: Friday, July 10, 2009 8:12 PM Subject: Re: [Freeswitch-users] FS not wait respond from called and send 200Ok Look closer at the logs, we don't send a 200ok in a bridge until we get one from the b leg. Mike On Jul 10, 2009, at 5:39 AM, Kozak Vladimir wrote: Hello, I have the following problem: I send Invite without SDP to Freeswitch on destination_number "xxx_123" And I want Freeswitch to make "bridge", but it doesn't wait respond from "123" and sends 200 Ok with SDP to me. Does nybody know a clue about this? Best regards vkozak _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ------------------------------------------------------------------------------ _______________________________________________ Freeswitch-users mailing list Freeswitch-users at 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/20090713/abeb13b5/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: fs.log Type: application/octet-stream Size: 56155 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090713/abeb13b5/attachment-0003.obj -------------- next part -------------- A non-text attachment was scrubbed... Name: sip.log Type: application/octet-stream Size: 13359 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090713/abeb13b5/attachment-0004.obj -------------- next part -------------- A non-text attachment was scrubbed... Name: sip4wireshark.log Type: application/octet-stream Size: 14186 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090713/abeb13b5/attachment-0005.obj From brian at freeswitch.org Mon Jul 13 10:19:42 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 13 Jul 2009 12:19:42 -0500 Subject: [Freeswitch-users] FS not wait respond from called and send 200Ok In-Reply-To: References: <5F5ABAC4334545CE8323DACB10805DE1@abisoft.biz> <74C6337E-6F03-4485-B69B-372658E0D79B@jerris.com> Message-ID: <03192D98-85A9-4A7A-8D06-837C6FDB0BBE@freeswitch.org> If you're on SVN trunk you no longer have to use a double nat profile. You can set the local-network-acl and ext-[rtp|sip]-ip settings correctly. /b On Jul 13, 2009, at 12:08 PM, Kozak Vladimir wrote: > Please, look at these logs. > Fist invite witchout SDP (from 1007 at uat.agent.starpoundtech.net to vk_1008 at uat.agent.starpoundtech.net > ) not wait respond from 1008 and send 200 Ok to 1007. Called phone > didn't accept call. > > > > > > > > > ip:ports > 172.26.200.252:5071 - userAgent (starpound), originator call > 172.26.200.252:5080 - FS external profile > 172.26.200.252:5090 - FS doubleNat profile > 172.26.10.65:38464 - phone1 (1007, registered on FS) > 172.26.10.65:17748 - phone2 (1008, registered on FS) > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090713/123fcb00/attachment.html From sridhart at alcatel-lucent.com Mon Jul 13 10:32:12 2009 From: sridhart at alcatel-lucent.com (Rajagopal, Sridhar (Sridhar)) Date: Mon, 13 Jul 2009 23:02:12 +0530 Subject: [Freeswitch-users] Help Regarding memory leak with freeswitch Message-ID: <9389DD3DDD6B9144B147CE564C6599B9082F92A2ED@INBANSXCHMBSA3.in.alcatel-lucent.com> Hi , Thanks very much for the help. Please let me know which module of the code has fix. Do I need to update freeswitch core library or sofia-sip library or is it necessary to update entire freeswitch code. Regards, Sridhar ________________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of freeswitch-users-request at lists.freeswitch.org [freeswitch-users-request at lists.freeswitch.org] Sent: Monday, July 13, 2009 9:17 PM To: freeswitch-users at lists.freeswitch.org Subject: Freeswitch-users Digest, Vol 37, Issue 67 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: Dialogic cards (Anthony Minessale) 2. Re: Help Regarding memory leak with freeswitch (Anthony Minessale) 3. Re: Setting channel variables using event socket (Anthony Minessale) 4. Re: Dialogic cards (Steve Underwood) 5. Re: Dialogic cards (Steve Underwood) 6. Re: Dialogic cards (Anthony Minessale) ---------------------------------------------------------------------- Message: 1 Date: Mon, 13 Jul 2009 10:08:26 -0500 From: Anthony Minessale Subject: Re: [Freeswitch-users] Dialogic cards To: freeswitch-users at lists.freeswitch.org Message-ID: <191c3a030907130808s37e52ddg3ba567d6a5a1034b at mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" Dialogic is coming to ClueCon this year (this aug 4th) and they are sponsoring the conference. I can discuss the possibility of supporting their cards at that time. On Sun, Jul 12, 2009 at 11:07 PM, Tim Uckun wrote: > We have some older dialogic cards (D300 series E1 cards) and I am > wondering if freeswitch can support these cards. > > 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/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference 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/20090713/25e82735/attachment-0001.html ------------------------------ Message: 2 Date: Mon, 13 Jul 2009 10:20:53 -0500 From: Anthony Minessale Subject: Re: [Freeswitch-users] Help Regarding memory leak with freeswitch To: freeswitch-users at lists.freeswitch.org Message-ID: <191c3a030907130820y46985521y1a01270a98cc2fcf at mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" Which revision of FreeSWITCH are you using? Several memory leaks have been fixed since the last formal release. One specifically in REGISTER. You should probably try SVN trunk or the latest pre-release of 1.0.4 On Mon, Jul 13, 2009 at 2:07 AM, Rajagopal, Sridhar (Sridhar) < sridhart at alcatel-lucent.com> wrote: > Hi all, > > I am running freeswitch on powerpc processor. I see memory being allocated > for each subsequent REGISTER requests coming to freeswitch. But not all the > memory allocated is not freed. If I run the code for two days the system is > running out of memory (RAM available to me is very less). > The same memory issue is happening even for calls. > > Please let me know if any body has seen this issue. Please let me know how > I can go ahead and debug this issue. > > Thanks in advance for the help. > > Regards, > Sridhar > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference 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/20090713/b97f4578/attachment-0001.html ------------------------------ Message: 3 Date: Mon, 13 Jul 2009 10:25:10 -0500 From: Anthony Minessale Subject: Re: [Freeswitch-users] Setting channel variables using event socket To: freeswitch-users at lists.freeswitch.org Message-ID: <191c3a030907130825q178fdfa9q1f3098258288ed3b at mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" and if you go the uuid_setvar route you do this: api uuid_setvar joe=out_to_lunch On Sun, Jul 12, 2009 at 6:00 PM, Brian West wrote: > call-command: execute > execute-app-name: set > execute-app-arg: fred=out_to_lunch > > > On Jul 12, 2009, at 5:33 PM, Nik Middleton wrote: > > As in > > call-command: set joe=out_to_lunch ? > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference 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/20090713/113023df/attachment-0001.html ------------------------------ Message: 4 Date: Mon, 13 Jul 2009 23:39:39 +0800 From: Steve Underwood Subject: Re: [Freeswitch-users] Dialogic cards To: freeswitch-users at lists.freeswitch.org Message-ID: <4A5B553B.9020000 at coppice.org> Content-Type: text/plain; charset=ISO-8859-1; format=flowed Tim Uckun wrote: > We have some older dialogic cards (D300 series E1 cards) and I am > wondering if freeswitch can support these cards. > Oh, I like the easy questions. No. It lacks the hardware features to do anything useful with Freeswitch, or Asterisk, or Callweaver, or Yate, or anything else that expects to do two way telephony through the host CPU. Steve ------------------------------ Message: 5 Date: Mon, 13 Jul 2009 23:39:47 +0800 From: Steve Underwood Subject: Re: [Freeswitch-users] Dialogic cards To: freeswitch-users at lists.freeswitch.org Message-ID: <4A5B5543.6080405 at coppice.org> Content-Type: text/plain; charset=ISO-8859-1; format=flowed Being now a mashup of several CTI companies, there are now a number of disparate things called Dialogic cards. Some, like the cards previously known as Prince..... er, Eicon are perfectly supportable. The old Dialogic cards, like the D300 series, are not duplex to and from the host. They are only duplex across the mezzanine bus. There's nothing you can do with them. Steve Anthony Minessale wrote: > Dialogic is coming to ClueCon this year (this aug 4th) and they are > sponsoring the conference. > I can discuss the possibility of supporting their cards at that time. > > > On Sun, Jul 12, 2009 at 11:07 PM, Tim Uckun > wrote: > > We have some older dialogic cards (D300 series E1 cards) and I am > wondering if freeswitch can support these cards. > > Thanks. > ------------------------------ Message: 6 Date: Mon, 13 Jul 2009 10:46:58 -0500 From: Anthony Minessale Subject: Re: [Freeswitch-users] Dialogic cards To: freeswitch-users at lists.freeswitch.org Message-ID: <191c3a030907130846u4c134fc4n578589c196617557 at mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" ok, or we could ask Steve I guess. =D On Mon, Jul 13, 2009 at 10:39 AM, Steve Underwood wrote: > Tim Uckun wrote: > > We have some older dialogic cards (D300 series E1 cards) and I am > > wondering if freeswitch can support these cards. > > > Oh, I like the easy questions. No. It lacks the hardware features to do > anything useful with Freeswitch, or Asterisk, or Callweaver, or Yate, or > anything else that expects to do two way telephony through the host CPU. > > 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/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference 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/20090713/2912d77b/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 37, Issue 67 ************************************************ From anthony.minessale at gmail.com Mon Jul 13 10:50:02 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 13 Jul 2009 12:50:02 -0500 Subject: [Freeswitch-users] Help Regarding memory leak with freeswitch In-Reply-To: <9389DD3DDD6B9144B147CE564C6599B9082F92A2ED@INBANSXCHMBSA3.in.alcatel-lucent.com> References: <9389DD3DDD6B9144B147CE564C6599B9082F92A2ED@INBANSXCHMBSA3.in.alcatel-lucent.com> Message-ID: <191c3a030907131050i7b39ef91s631d3ab9ea189600@mail.gmail.com> I am saying you should update the entire freeswitch code to either latest trunk or at least 1.0.4pre9 On Mon, Jul 13, 2009 at 12:32 PM, Rajagopal, Sridhar (Sridhar) < sridhart at alcatel-lucent.com> wrote: > Hi , > > Thanks very much for the help. > > Please let me know which module of the code has fix. Do I need to update > freeswitch core library or sofia-sip library or is it necessary to update > entire freeswitch code. > > Regards, > Sridhar > > ________________________________________ > From: freeswitch-users-bounces at lists.freeswitch.org [ > freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > freeswitch-users-request at lists.freeswitch.org [ > freeswitch-users-request at lists.freeswitch.org] > Sent: Monday, July 13, 2009 9:17 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Freeswitch-users Digest, Vol 37, Issue 67 > > 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: Dialogic cards (Anthony Minessale) > 2. Re: Help Regarding memory leak with freeswitch (Anthony Minessale) > 3. Re: Setting channel variables using event socket > (Anthony Minessale) > 4. Re: Dialogic cards (Steve Underwood) > 5. Re: Dialogic cards (Steve Underwood) > 6. Re: Dialogic cards (Anthony Minessale) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Mon, 13 Jul 2009 10:08:26 -0500 > From: Anthony Minessale > Subject: Re: [Freeswitch-users] Dialogic cards > To: freeswitch-users at lists.freeswitch.org > Message-ID: > <191c3a030907130808s37e52ddg3ba567d6a5a1034b at mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > Dialogic is coming to ClueCon this year (this aug 4th) and they are > sponsoring the conference. > I can discuss the possibility of supporting their cards at that time. > > > On Sun, Jul 12, 2009 at 11:07 PM, Tim Uckun wrote: > > > We have some older dialogic cards (D300 series E1 cards) and I am > > wondering if freeswitch can support these cards. > > > > 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/ > Twitter: http://twitter.com/FreeSWITCH_wire > > 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 > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090713/25e82735/attachment-0001.html > > ------------------------------ > > Message: 2 > Date: Mon, 13 Jul 2009 10:20:53 -0500 > From: Anthony Minessale > Subject: Re: [Freeswitch-users] Help Regarding memory leak with > freeswitch > To: freeswitch-users at lists.freeswitch.org > Message-ID: > <191c3a030907130820y46985521y1a01270a98cc2fcf at mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > Which revision of FreeSWITCH are you using? Several memory leaks have been > fixed since the last formal release. One specifically in REGISTER. > You should probably try SVN trunk or the latest pre-release of 1.0.4 > > > > > On Mon, Jul 13, 2009 at 2:07 AM, Rajagopal, Sridhar (Sridhar) < > sridhart at alcatel-lucent.com> wrote: > > > Hi all, > > > > I am running freeswitch on powerpc processor. I see memory being > allocated > > for each subsequent REGISTER requests coming to freeswitch. But not all > the > > memory allocated is not freed. If I run the code for two days the system > is > > running out of memory (RAM available to me is very less). > > The same memory issue is happening even for calls. > > > > Please let me know if any body has seen this issue. Please let me know > how > > I can go ahead and debug this issue. > > > > Thanks in advance for the help. > > > > Regards, > > Sridhar > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > 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 > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090713/b97f4578/attachment-0001.html > > ------------------------------ > > Message: 3 > Date: Mon, 13 Jul 2009 10:25:10 -0500 > From: Anthony Minessale > Subject: Re: [Freeswitch-users] Setting channel variables using event > socket > To: freeswitch-users at lists.freeswitch.org > Message-ID: > <191c3a030907130825q178fdfa9q1f3098258288ed3b at mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > and if you go the uuid_setvar route you do this: > > api uuid_setvar joe=out_to_lunch > > > > On Sun, Jul 12, 2009 at 6:00 PM, Brian West wrote: > > > call-command: execute > > execute-app-name: set > > execute-app-arg: fred=out_to_lunch > > > > > > On Jul 12, 2009, at 5:33 PM, Nik Middleton wrote: > > > > As in > > > > call-command: set joe=out_to_lunch ? > > > > > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > 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 > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090713/113023df/attachment-0001.html > > ------------------------------ > > Message: 4 > Date: Mon, 13 Jul 2009 23:39:39 +0800 > From: Steve Underwood > Subject: Re: [Freeswitch-users] Dialogic cards > To: freeswitch-users at lists.freeswitch.org > Message-ID: <4A5B553B.9020000 at coppice.org> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Tim Uckun wrote: > > We have some older dialogic cards (D300 series E1 cards) and I am > > wondering if freeswitch can support these cards. > > > Oh, I like the easy questions. No. It lacks the hardware features to do > anything useful with Freeswitch, or Asterisk, or Callweaver, or Yate, or > anything else that expects to do two way telephony through the host CPU. > > Steve > > > > > > ------------------------------ > > Message: 5 > Date: Mon, 13 Jul 2009 23:39:47 +0800 > From: Steve Underwood > Subject: Re: [Freeswitch-users] Dialogic cards > To: freeswitch-users at lists.freeswitch.org > Message-ID: <4A5B5543.6080405 at coppice.org> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Being now a mashup of several CTI companies, there are now a number of > disparate things called Dialogic cards. Some, like the cards previously > known as Prince..... er, Eicon are perfectly supportable. The old > Dialogic cards, like the D300 series, are not duplex to and from the > host. They are only duplex across the mezzanine bus. There's nothing you > can do with them. > > Steve > > > Anthony Minessale wrote: > > Dialogic is coming to ClueCon this year (this aug 4th) and they are > > sponsoring the conference. > > I can discuss the possibility of supporting their cards at that time. > > > > > > On Sun, Jul 12, 2009 at 11:07 PM, Tim Uckun > > wrote: > > > > We have some older dialogic cards (D300 series E1 cards) and I am > > wondering if freeswitch can support these cards. > > > > Thanks. > > > > > > > > ------------------------------ > > Message: 6 > Date: Mon, 13 Jul 2009 10:46:58 -0500 > From: Anthony Minessale > Subject: Re: [Freeswitch-users] Dialogic cards > To: freeswitch-users at lists.freeswitch.org > Message-ID: > <191c3a030907130846u4c134fc4n578589c196617557 at mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > ok, > > or we could ask Steve I guess. =D > > > On Mon, Jul 13, 2009 at 10:39 AM, Steve Underwood >wrote: > > > Tim Uckun wrote: > > > We have some older dialogic cards (D300 series E1 cards) and I am > > > wondering if freeswitch can support these cards. > > > > > Oh, I like the easy questions. No. It lacks the hardware features to do > > anything useful with Freeswitch, or Asterisk, or Callweaver, or Yate, or > > anything else that expects to do two way telephony through the host CPU. > > > > 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/ > Twitter: http://twitter.com/FreeSWITCH_wire > > 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 > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090713/2912d77b/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 37, Issue 67 > ************************************************ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference 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/20090713/4a541591/attachment-0001.html From fdhege at gmail.com Mon Jul 13 12:04:05 2009 From: fdhege at gmail.com (Dale) Date: Mon, 13 Jul 2009 15:04:05 -0400 Subject: [Freeswitch-users] Gateway Settings from-domain and caller-id-in-from Message-ID: <3DB91FF8-24DA-4031-9A7E-8B5390907D3B@gmail.com> Hello, I have been playing around with gateway settings today and noticed something that I wasn't sure if it was a bug or if its just the way it works. When I have from-domain set in my gateway config it correctly uses the configured from domain. If I then set caller-id-in-from to true the configured from domain is no longer used and it reverts back to the ip address configured for that sip_profile. Is that expected or is that a bug? Thanks, -Dale From brian at freeswitch.org Mon Jul 13 12:09:30 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 13 Jul 2009 14:09:30 -0500 Subject: [Freeswitch-users] Gateway Settings from-domain and caller-id-in-from In-Reply-To: <3DB91FF8-24DA-4031-9A7E-8B5390907D3B@gmail.com> References: <3DB91FF8-24DA-4031-9A7E-8B5390907D3B@gmail.com> Message-ID: <41E67BD3-25A5-41B6-B605-D7A75DA2342B@freeswitch.org> Can you collect up sip traces and open a jira please. /b On Jul 13, 2009, at 2:04 PM, Dale wrote: > > Hello, > > I have been playing around with gateway settings today and noticed > something that I wasn't sure if it was a bug or if its just the way it > works. > > When I have from-domain set in my gateway config it correctly uses the > configured from domain. If I then set caller-id-in-from to true the > configured from domain is no longer used and it reverts back to the ip > address configured for that sip_profile. > > Is that expected or is that a bug? > > Thanks, > > -Dale From saeedahmad1981 at gmail.com Mon Jul 13 12:38:42 2009 From: saeedahmad1981 at gmail.com (Saeed Ahmad) Date: Mon, 13 Jul 2009 21:38:42 +0200 Subject: [Freeswitch-users] Help In-Reply-To: <30ee97110907100944y15d1e90fu29b543cd87f12c5f@mail.gmail.com> References: <5E41B7C2-B163-4364-A2A9-8B90AD58F47E@frota.net> <30ee97110907100944y15d1e90fu29b543cd87f12c5f@mail.gmail.com> Message-ID: helpless On Fri, Jul 10, 2009 at 6:44 PM, Jens Vegeby wrote: > You might wanna write what you need help with :) > > On 7/10/09, Ney Frota wrote: > > Help > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 > > Mvh/Regards Jens > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090713/c2daeb59/attachment.html From siniypin at gmail.com Mon Jul 13 13:53:46 2009 From: siniypin at gmail.com (=?KOI8-R?B?8s/CxdLUIPTXxdLJ1M7F0g==?=) Date: Tue, 14 Jul 2009 00:53:46 +0400 Subject: [Freeswitch-users] port restricted NAT, SRTP problem Message-ID: <2160023e0907131353v39fe88a5i3b1c05ee9ee4dd23@mail.gmail.com> Hi guys! I'm a novice in VoIP world, and may be missing some important concepts, but recently I've faced a problem with client softphone residing behind a port-restricted NAT and a public FS server and can't find an explanation on why it is happening and how to escape it. Okey, the problem is as follows. I have a client residing behind port restricted NAT. It can register at our public server and can issue a looped call and hear itself perfectly well. But when I call to speek to this natted guy from computer exposed to web without any routers he able to hear me for merely a second and then I become muted and he hear nothing. On the other side I can hear this guy quite good, though with slight jittery sound. If I set bypass_media param in our server's external profile to true - everything works as it supposed to - we hear one another. But still there is a problem with call originating from that guy - it is being interrupted after some time (after about 30 sec). Both clients are capable to do STUN and ICE and have these options enabled. Calls are secured with TLS and SRTP enabled on our server. FreeSWITCH is installed on Windows Server 2008 box with open UDP traffic and TCP, UDP ports 5080,5081 opened in order to expose an external profile. As far as I understand, with bypass_media param disabled FreeSWITCH is acting as media proxy and it is unable to do ICE and that should be a reason why that guy can't hear me. In overmentioned peer2peer scenario switching to no media mode is acceptable, but still there is a question whether this call with media flow bypassing FreeSWITCH is secured? I guess not. Cause I don't have any certificates installed on clients. Also, we've plans to use our FreeSWITCH as a media conference server. And of course this guy failes to connect to the testing one. Below are some of configurations: vars.xml ... ... external.xml ... ... public dialplan My SDP: Remote SDP: v=0 o=- 3456517465 3456517465 IN IP4 91.79.44.168 s=pjmedia c=IN IP4 91.79.44.168 t=0 0 a=X-nat:2 m=audio 1142 RTP/SAVP 103 102 104 117 3 0 8 9 101 a=rtpmap:103 speex/16000 ... a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtcp:1143 IN IP4 91.79.44.168 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:ee+Xv9etM5t5w3DH5B1hR+i9lrt7BHQhzJIwFv7d a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:ELeudfX0mgsL+0u7qzGOqdEfg891fMn281BJszkS a=ice-ufrag:70db7dd4 a=ice-pwd:708804fe a=candidate:H 1 UDP 39 91.79.44.168 1142 typ host ... And this is the other guy SDP: Remote SDP: v=0 o=- 3456517469 3456517470 IN IP4 85.140.191.254 s=pjmedia c=IN IP4 85.140.191.254 t=0 0 a=X-nat:8 m=audio 23374 RTP/SAVP 103 101 a=rtpmap:103 speex/16000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtcp:23375 IN IP4 85.140.191.254 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:2m4YBCRid0h+5AIaIGmXaqelQsSuK3HP1jtAMoiG a=ice-ufrag:182f5a61 a=ice-pwd:1d690863 a=candidate:S 1 UDP 31 85.140.191.254 23374 typ srflx raddr 192.168.2.2 rport 2796 ... PS sorry for a long post -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090714/0b021252/attachment.html From timuckun at gmail.com Mon Jul 13 14:33:54 2009 From: timuckun at gmail.com (Tim Uckun) Date: Tue, 14 Jul 2009 09:33:54 +1200 Subject: [Freeswitch-users] Dialogic cards In-Reply-To: <4A5B553B.9020000@coppice.org> References: <855e4dcf0907122107q392a9ca1h851dfbd57e2518ca@mail.gmail.com> <4A5B553B.9020000@coppice.org> Message-ID: <855e4dcf0907131433q33a21d2v935797013046aa81@mail.gmail.com> On Tue, Jul 14, 2009 at 3:39 AM, Steve Underwood wrote: > Tim Uckun wrote: >> We have some older dialogic cards (D300 series E1 cards) and I am >> wondering if freeswitch can support these cards. >> > Oh, I like the easy questions. No. It lacks the hardware features to do > anything useful with Freeswitch, or Asterisk, or Callweaver, or Yate, or > anything else that expects to do two way telephony through the host CPU. What are the recommended cards to be used with freeswitch? From msc at freeswitch.org Mon Jul 13 14:45:40 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 13 Jul 2009 14:45:40 -0700 Subject: [Freeswitch-users] Dialogic cards In-Reply-To: <855e4dcf0907131433q33a21d2v935797013046aa81@mail.gmail.com> References: <855e4dcf0907122107q392a9ca1h851dfbd57e2518ca@mail.gmail.com> <4A5B553B.9020000@coppice.org> <855e4dcf0907131433q33a21d2v935797013046aa81@mail.gmail.com> Message-ID: <87f2f3b90907131445r3bda46e9wa8650ce487c0b62c@mail.gmail.com> > What are the recommended cards to be used with freeswitch? > Sangoma cards and Zaptel/DAHDI compatible cards work well. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090713/07f49276/attachment.html From eweaver at meetingone.com Mon Jul 13 16:35:17 2009 From: eweaver at meetingone.com (Weaver, Eric) Date: Mon, 13 Jul 2009 16:35:17 -0700 Subject: [Freeswitch-users] Preventing disconnect on event_sockte close Message-ID: <7DB369DA44D5CF42B6C23BD7D611A50FDA7E1D5FD6@VA3DIAXVS061.RED001.local> Using mod_event_socket in outbound mode, is there any to prevent a call from being disconnected when the outbound socket is closed ? I would like to handle the initial inbound call using outbound but after the disposition of the call is determined, close the socket and have that call managed using an inbound socket instead. Eric Weaver -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090713/6de37657/attachment-0001.html From msc at freeswitch.org Mon Jul 13 18:08:22 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 13 Jul 2009 18:08:22 -0700 Subject: [Freeswitch-users] Preventing disconnect on event_sockte close In-Reply-To: <7DB369DA44D5CF42B6C23BD7D611A50FDA7E1D5FD6@VA3DIAXVS061.RED001.local> References: <7DB369DA44D5CF42B6C23BD7D611A50FDA7E1D5FD6@VA3DIAXVS061.RED001.local> Message-ID: <87f2f3b90907131808l100afe7dwc95d0f624ff38d50@mail.gmail.com> I don't know if this will work for you but I just tested this scenario with uuid_park. After parking the call I disconnected the socket and the call continued. I did the same thing with uuid_transfer. After the transfer I disconnected the socket and the call continued. How are you handling the call and how is the socket getting disconnected? -MC On Mon, Jul 13, 2009 at 4:35 PM, Weaver, Eric wrote: > Using mod_event_socket in outbound mode, is there any to prevent a call > from being disconnected when the outbound socket is closed ? I would like to > handle the initial inbound call using outbound but after the disposition of > the call is determined, close the socket and have that call managed using an > inbound socket instead. > > > > Eric Weaver > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090713/a143bc38/attachment.html From msc at freeswitch.org Mon Jul 13 18:10:13 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 13 Jul 2009 18:10:13 -0700 Subject: [Freeswitch-users] Originate in Dial plan In-Reply-To: <8ccbff060907120959ma7c0ey22adeae196f872c3@mail.gmail.com> References: <8ccbff060907120959ma7c0ey22adeae196f872c3@mail.gmail.com> Message-ID: <87f2f3b90907131810q219c04f3ra385e0e1d252d31c@mail.gmail.com> What phone number do you call back? I mean, how do you know what the customer's number is? Do you go by the caller id number? -MC On Sun, Jul 12, 2009 at 9:59 AM, Dome Charoenyost wrote: > Dear sir, > I want to user dialplan callback to customer. is posible to > to this is dialplan XML ? > Now i use javascript. > my call flow. > 1. customer call > 2. FS rining and wait until customer hangup > 3. callback to customer number > > > Best Regards. > > Dome C. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090713/4f700998/attachment.html From msc at freeswitch.org Mon Jul 13 18:11:56 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 13 Jul 2009 18:11:56 -0700 Subject: [Freeswitch-users] Getting xml_request in LUA In-Reply-To: <4A59A390.7030000@savion.huji.ac.il> References: <4A59A390.7030000@savion.huji.ac.il> Message-ID: <87f2f3b90907131811w312c802fofe04f2be158c1521@mail.gmail.com> On Sun, Jul 12, 2009 at 1:49 AM, Eli Hayun wrote: > In the Perl example I found: > > > How to access request parameters and how to return data > > You have two hashes that are populated for you by freeswitch. Those > hashes are: > > * %XML_REQUEST > * %XML_DATA > > I want to use LUA to set the directory and dialplan xml. How do I get > the XML_REQUEST/XML_DATA from LUA? Is this the information you are looking for? http://wiki.freeswitch.org/wiki/Mod_lua#For_serving_configuration > > > Thanks > Eli Hayun > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090713/304d2fe8/attachment.html From eweaver at meetingone.com Mon Jul 13 18:47:12 2009 From: eweaver at meetingone.com (Weaver, Eric) Date: Mon, 13 Jul 2009 18:47:12 -0700 Subject: [Freeswitch-users] Preventing disconnect on event_sockte close In-Reply-To: <87f2f3b90907131808l100afe7dwc95d0f624ff38d50@mail.gmail.com> References: <7DB369DA44D5CF42B6C23BD7D611A50FDA7E1D5FD6@VA3DIAXVS061.RED001.local>, <87f2f3b90907131808l100afe7dwc95d0f624ff38d50@mail.gmail.com> Message-ID: <7DB369DA44D5CF42B6C23BD7D611A50FDA7E210D73@VA3DIAXVS061.RED001.local> I noticed it in testing last night using net cat. I killed netcat and the inbound call was disconnected, I'll try your suggestions tonight. Thanks for the reply, eric ________________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins [msc at freeswitch.org] Sent: Monday, July 13, 2009 7:08 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Preventing disconnect on event_sockte close I don't know if this will work for you but I just tested this scenario with uuid_park. After parking the call I disconnected the socket and the call continued. I did the same thing with uuid_transfer. After the transfer I disconnected the socket and the call continued. How are you handling the call and how is the socket getting disconnected? -MC On Mon, Jul 13, 2009 at 4:35 PM, Weaver, Eric > wrote: Using mod_event_socket in outbound mode, is there any to prevent a call from being disconnected when the outbound socket is closed ? I would like to handle the initial inbound call using outbound but after the disposition of the call is determined, close the socket and have that call managed using an inbound socket instead. Eric Weaver _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From klaus.teller at gmx.net Mon Jul 13 19:34:11 2009 From: klaus.teller at gmx.net (Klaus Teller) Date: Tue, 14 Jul 2009 04:34:11 +0200 Subject: [Freeswitch-users] Gafachi no passing caller number Message-ID: <20090714023411.151100@gmx.net> Hi, I tend to believe that we already had this working. Here is my origination string: {effective_caller_id_name=Paul Gascogne,effective_caller_id_number=16478343812}sofia/gateway/sip.gafachi.com/164783486421 The caller number is not being passed to the destination. Is there something i'm missing? Thanks, Klaus. -- GRATIS f?r alle GMX-Mitglieder: Die maxdome Movie-FLAT! Jetzt freischalten unter http://portal.gmx.net/de/go/maxdome01 From mrene_lists at avgs.ca Mon Jul 13 19:36:37 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Mon, 13 Jul 2009 22:36:37 -0400 Subject: [Freeswitch-users] Gafachi no passing caller number In-Reply-To: <20090714023411.151100@gmx.net> References: <20090714023411.151100@gmx.net> Message-ID: You need to escape the spaces with \s in the caller id name. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 13-Jul-09, at 10:34 PM, Klaus Teller wrote: > Hi, > > I tend to believe that we already had this working. Here is my > origination string: > > {effective_caller_id_name=Paul > Gascogne,effective_caller_id_number=16478343812}sofia/gateway/ > sip.gafachi.com/164783486421 > > The caller number is not being passed to the destination. Is there > something i'm missing? > > Thanks, > > Klaus. > > -- > GRATIS f?r alle GMX-Mitglieder: Die maxdome Movie-FLAT! > Jetzt freischalten unter http://portal.gmx.net/de/go/maxdome01 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From klaus.teller at gmx.net Mon Jul 13 19:43:20 2009 From: klaus.teller at gmx.net (Klaus Teller) Date: Tue, 14 Jul 2009 04:43:20 +0200 Subject: [Freeswitch-users] Gafachi no passing caller number In-Reply-To: References: <20090714023411.151100@gmx.net> Message-ID: <20090714024320.198030@gmx.net> It doesn't seem to work though. I tried removing the space completely as well as removing the caller name parameter. -------- Original-Nachricht -------- > Datum: Mon, 13 Jul 2009 22:36:37 -0400 > Von: Mathieu Rene > An: freeswitch-users at lists.freeswitch.org > Betreff: Re: [Freeswitch-users] Gafachi no passing caller number > You need to escape the spaces with \s in the caller id name. > > Mathieu Rene > Avant-Garde Solutions Inc > Office: + 1 (514) 664-1044 x100 > Cell: +1 (514) 664-1044 x200 > mrene at avgs.ca > > > > > On 13-Jul-09, at 10:34 PM, Klaus Teller wrote: > > > Hi, > > > > I tend to believe that we already had this working. Here is my > > origination string: > > > > {effective_caller_id_name=Paul > > Gascogne,effective_caller_id_number=16478343812}sofia/gateway/ > > sip.gafachi.com/164783486421 > > > > The caller number is not being passed to the destination. Is there > > something i'm missing? > > > > Thanks, > > > > Klaus. > > > > -- > > GRATIS f?r alle GMX-Mitglieder: Die maxdome Movie-FLAT! > > Jetzt freischalten unter http://portal.gmx.net/de/go/maxdome01 > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- GRATIS f?r alle GMX-Mitglieder: Die maxdome Movie-FLAT! Jetzt freischalten unter http://portal.gmx.net/de/go/maxdome01 From mrene_lists at avgs.ca Mon Jul 13 19:47:18 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Mon, 13 Jul 2009 22:47:18 -0400 Subject: [Freeswitch-users] Gafachi no passing caller number In-Reply-To: <20090714024320.198030@gmx.net> References: <20090714023411.151100@gmx.net> <20090714024320.198030@gmx.net> Message-ID: <3DD2DE9A-7726-4167-B968-7434A7BEFBEC@avgs.ca> Oh you're using effective_caller_id_number, those vars are only checked when an a-leg exists. Use origination_caller_id_number and origination_caller_id_name. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 13-Jul-09, at 10:43 PM, Klaus Teller wrote: > It doesn't seem to work though. I tried removing the space > completely as well as removing the caller name parameter. > > -------- Original-Nachricht -------- >> Datum: Mon, 13 Jul 2009 22:36:37 -0400 >> Von: Mathieu Rene >> An: freeswitch-users at lists.freeswitch.org >> Betreff: Re: [Freeswitch-users] Gafachi no passing caller number > >> You need to escape the spaces with \s in the caller id name. >> >> Mathieu Rene >> Avant-Garde Solutions Inc >> Office: + 1 (514) 664-1044 x100 >> Cell: +1 (514) 664-1044 x200 >> mrene at avgs.ca >> >> >> >> >> On 13-Jul-09, at 10:34 PM, Klaus Teller wrote: >> >>> Hi, >>> >>> I tend to believe that we already had this working. Here is my >>> origination string: >>> >>> {effective_caller_id_name=Paul >>> Gascogne,effective_caller_id_number=16478343812}sofia/gateway/ >>> sip.gafachi.com/164783486421 >>> >>> The caller number is not being passed to the destination. Is there >>> something i'm missing? >>> >>> Thanks, >>> >>> Klaus. >>> >>> -- >>> GRATIS f?r alle GMX-Mitglieder: Die maxdome Movie-FLAT! >>> Jetzt freischalten unter http://portal.gmx.net/de/go/maxdome01 >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > -- > GRATIS f?r alle GMX-Mitglieder: Die maxdome Movie-FLAT! > Jetzt freischalten unter http://portal.gmx.net/de/go/maxdome01 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.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 Mon Jul 13 19:48:15 2009 From: sicfslist at gmail.com (Shelby Ramsey) Date: Mon, 13 Jul 2009 21:48:15 -0500 Subject: [Freeswitch-users] Gafachi no passing caller number In-Reply-To: <20090714024320.198030@gmx.net> References: <20090714023411.151100@gmx.net> <20090714024320.198030@gmx.net> Message-ID: <35b355e90907131948r651f1fdcle139031d8f6a9521@mail.gmail.com> Klaus, Use ngrep and see if the From / RPID headers are correct in the SIP message. This will let you know if FS is doing the correct thing. SDR -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090713/a8caa856/attachment.html From klaus.teller at gmx.net Mon Jul 13 19:53:15 2009 From: klaus.teller at gmx.net (Klaus Teller) Date: Tue, 14 Jul 2009 04:53:15 +0200 Subject: [Freeswitch-users] Gafachi no passing caller number In-Reply-To: <3DD2DE9A-7726-4167-B968-7434A7BEFBEC@avgs.ca> References: <20090714023411.151100@gmx.net> <20090714024320.198030@gmx.net> <3DD2DE9A-7726-4167-B968-7434A7BEFBEC@avgs.ca> Message-ID: <20090714025315.198040@gmx.net> Thanks folks. Indeed i had to use origination_caller_id_number. Cheers, Klaus. -------- Original-Nachricht -------- > Datum: Mon, 13 Jul 2009 22:47:18 -0400 > Von: Mathieu Rene > An: freeswitch-users at lists.freeswitch.org > Betreff: Re: [Freeswitch-users] Gafachi no passing caller number > Oh you're using effective_caller_id_number, those vars are only > checked when an a-leg exists. > > Use origination_caller_id_number and origination_caller_id_name. > > Mathieu Rene > Avant-Garde Solutions Inc > Office: + 1 (514) 664-1044 x100 > Cell: +1 (514) 664-1044 x200 > mrene at avgs.ca > > > > > On 13-Jul-09, at 10:43 PM, Klaus Teller wrote: > > > It doesn't seem to work though. I tried removing the space > > completely as well as removing the caller name parameter. > > > > -------- Original-Nachricht -------- > >> Datum: Mon, 13 Jul 2009 22:36:37 -0400 > >> Von: Mathieu Rene > >> An: freeswitch-users at lists.freeswitch.org > >> Betreff: Re: [Freeswitch-users] Gafachi no passing caller number > > > >> You need to escape the spaces with \s in the caller id name. > >> > >> Mathieu Rene > >> Avant-Garde Solutions Inc > >> Office: + 1 (514) 664-1044 x100 > >> Cell: +1 (514) 664-1044 x200 > >> mrene at avgs.ca > >> > >> > >> > >> > >> On 13-Jul-09, at 10:34 PM, Klaus Teller wrote: > >> > >>> Hi, > >>> > >>> I tend to believe that we already had this working. Here is my > >>> origination string: > >>> > >>> {effective_caller_id_name=Paul > >>> Gascogne,effective_caller_id_number=16478343812}sofia/gateway/ > >>> sip.gafachi.com/164783486421 > >>> > >>> The caller number is not being passed to the destination. Is there > >>> something i'm missing? > >>> > >>> Thanks, > >>> > >>> Klaus. > >>> > >>> -- > >>> GRATIS f?r alle GMX-Mitglieder: Die maxdome Movie-FLAT! > >>> Jetzt freischalten unter http://portal.gmx.net/de/go/maxdome01 > >>> > >>> _______________________________________________ > >>> Freeswitch-users mailing list > >>> Freeswitch-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> http://www.freeswitch.org > >> > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > -- > > GRATIS f?r alle GMX-Mitglieder: Die maxdome Movie-FLAT! > > Jetzt freischalten unter http://portal.gmx.net/de/go/maxdome01 > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Neu: GMX Doppel-FLAT mit Internet-Flatrate + Telefon-Flatrate f?r nur 19,99 Euro/mtl.!* http://portal.gmx.net/de/go/dsl02 From jingwei.yang at gmail.com Mon Jul 13 20:34:32 2009 From: jingwei.yang at gmail.com (Jingwei Yang) Date: Tue, 14 Jul 2009 11:34:32 +0800 Subject: [Freeswitch-users] mod_dingaling no audio In-Reply-To: <507898380907130429haeb7ef9haeaf31b7e409da5c@mail.gmail.com> References: <13529f9d0906240240x55f0c19ao7246de1d34b1f661@mail.gmail.com> <13529f9d0906242231s7384908ayb30340f5442b2ab@mail.gmail.com> <507898380906250602r4c666b06xfdeea5943ad0cbff@mail.gmail.com> <13529f9d0906251915m440a1c46s534a5c7c735b8ce8@mail.gmail.com> <507898380906251930v6f82d786l9d1bf0da61212b05@mail.gmail.com> <13529f9d0906252034o5cb0b7b7y8a9ffe750edf2b20@mail.gmail.com> <13529f9d0906290225v4abd1552l4a3b72e0ff1afc66@mail.gmail.com> <507898380906290346w20cfea6bq82114efc63439f52@mail.gmail.com> <13529f9d0907130227u377b7459l8143a37b9ccfc761@mail.gmail.com> <507898380907130429haeb7ef9haeaf31b7e409da5c@mail.gmail.com> Message-ID: <13529f9d0907132034j546feeabj2aa7185f9711186a@mail.gmail.com> Hi Chris, I've attached the console logs for your reference. It really hits 888 in the dialplan and the external call can hear the echo without any problem. One thing attracts me is how the ip addresses are translated. Here's the working external example: *(external party's local addr)* * (external party's global addr)* *(server's global addr)* Here's the non-working internal exmaple: *(internal party's local addr)* *(internal party's global addr) * * (server's global addr)* *(internal party's local addr, again!) * *(internal party's global addr) * *(google's global addr)* *(google's global addr) * *(internal party's local addr, 3rd time!) * *(google's global addr)* And finally, when the call was hung up, the internal one showed an error like this: unknown session Regards, -Jingwei On Mon, Jul 13, 2009 at 7:29 PM, Chris Chen wrote: > Jingwei, can you show your console log when somebody is calling you from > gtalk client? Will it really hit 888 in your dialplan? > Thanks, > Chris > > > On Mon, Jul 13, 2009 at 5:27 AM, Jingwei Yang wrote: > >> Hi Chris, sorry for the late reply. Have been quite busy last few days. >> >> I had shifted 888 from default.xml to public.xml and the dialplan is >> simply having an echo action now. I've turned on dl_debug but unfortunately >> didn't find anything useful. Logs are attached for your reference. >> >> I don't think there's something wrong with the dialplan as two external >> parties can talk to each other perfectly (with ext-rtp-ip uncommented, at >> this time my ip was interpreted to be an external one). With ext-rtp-ip >> commented, I can hear the echo and I saw my ip was translated into an >> internal one (at this time, external party's audio failed). >> >> I tried the method on this wiki page as well: >> http://wiki.freeswitch.org/wiki/NAT_Traversal (the last FreeSwitch behind >> NAT portion) but still no luck. Please kindly let me know what other configs >> I should change. >> >> Thanks, >> -Jingwei >> >> On Mon, Jun 29, 2009 at 6:46 PM, Chris Chen wrote: >> >>> Jingwei, I don't know if you have the 888 defined in default.xml? also >>> you have to define $${domain}. >>> please do " dl_debug on" from fs_cli, and watch the console logs and see >>> what's going on when you try calling from external. Most likely your >>> dialplan is not correctly defined. >>> >>> Chris >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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... 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Name: external_call_in.txt Url: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090714/e4a3eb1a/attachment-0001.txt -------------- next part -------------- freeswitch at localhost.localdomain> 2009-07-14 10:20:12.779916 [INFO] libdingaling.c:1307 SecRECV: ------------------------------------------------------------------------------- 2009-07-14 10:20:12.779916 [INFO] libdingaling.c:1307 SecRECV: ------------------------------------------------------------------------------- 2009-07-14 10:20:12.779916 [NOTICE] switch_channel.c:602 New Channel dingaling/888 [2ed89b9b-bb8b-4606-a71d-2fe6dbd66d1d] 2009-07-14 10:20:12.779916 [NOTICE] switch_channel.c:600 Rename Channel dingaling/888->DingaLing/new [2ed89b9b-bb8b-4606-a71d-2fe6dbd66d1d] 2009-07-14 10:20:12.782051 [NOTICE] mod_dingaling.c:1084 Ring-Ready DingaLing/new! 2009-07-14 10:20:12.803916 [NOTICE] libdingaling.c:1309 SecSEND: ------------------------------------------------------------------------------- 2009-07-14 10:20:12.803916 [NOTICE] libdingaling.c:1309 SecSEND: ------------------------------------------------------------------------------- 2009-07-14 10:20:13.679871 [INFO] libdingaling.c:1307 SecRECV: ------------------------------------------------------------------------------- 2009-07-14 10:20:13.679871 [INFO] libdingaling.c:1307 SecRECV: ------------------------------------------------------------------------------- 2009-07-14 10:20:13.711870 [NOTICE] libdingaling.c:1309 SecSEND: ------------------------------------------------------------------------------- 2009-07-14 10:20:14.729818 [INFO] mod_dingaling.c:974 Stun Success 59.189.194.244:30578 2009-07-14 10:20:14.729818 [INFO] libdingaling.c:1307 SecRECV: ------------------------------------------------------------------------------- 2009-07-14 10:20:14.748836 [INFO] mod_dialplan_xml.c:252 Processing xxxxxx at gmail.com/Talk.v1046D90E88C->888 in context public 2009-07-14 10:20:14.748836 [NOTICE] switch_core_session.c:1391 Pre-Answer DingaLing/new! 2009-07-14 10:20:14.814814 [NOTICE] libdingaling.c:1309 SecSEND: ------------------------------------------------------------------------------- 2009-07-14 10:20:14.814814 [NOTICE] libdingaling.c:1309 SecSEND: ------------------------------------------------------------------------------- 2009-07-14 10:20:15.544777 [INFO] libdingaling.c:1307 SecRECV: ------------------------------------------------------------------------------- 2009-07-14 10:20:15.615820 [NOTICE] libdingaling.c:1309 SecSEND: ------------------------------------------------------------------------------- 2009-07-14 10:20:15.826763 [INFO] libdingaling.c:1307 SecRECV: ------------------------------------------------------------------------------- 2009-07-14 10:20:16.740726 [INFO] libdingaling.c:1307 SecRECV: ------------------------------------------------------------------------------- 2009-07-14 10:20:16.816760 [NOTICE] libdingaling.c:1309 SecSEND: ------------------------------------------------------------------------------- 2009-07-14 10:20:17.533727 [INFO] libdingaling.c:1307 SecRECV: ------------------------------------------------------------------------------- 2009-07-14 10:20:17.617673 [NOTICE] libdingaling.c:1309 SecSEND: ------------------------------------------------------------------------------- 2009-07-14 10:20:17.774664 [INFO] libdingaling.c:1307 SecRECV: ------------------------------------------------------------------------------- 2009-07-14 10:20:17.818663 [NOTICE] libdingaling.c:1309 SecSEND: ------------------------------------------------------------------------------- 2009-07-14 10:20:18.532660 [INFO] libdingaling.c:1307 SecRECV: ------------------------------------------------------------------------------- 2009-07-14 10:20:18.619622 [NOTICE] libdingaling.c:1309 SecSEND: ------------------------------------------------------------------------------- 2009-07-14 10:20:24.211338 [INFO] libdingaling.c:1307 SecRECV: ------------------------------------------------------------------------------- 2009-07-14 10:20:24.211338 [NOTICE] mod_dingaling.c:718 Hangup DingaLing/new [CS_EXECUTE] [NORMAL_CLEARING] 2009-07-14 10:20:24.216352 [NOTICE] switch_core_session.c:1085 Session 3 (DingaLing/new) Ended 2009-07-14 10:20:24.216352 [NOTICE] switch_core_session.c:1087 Close Channel DingaLing/new [CS_DESTROY] 2009-07-14 10:20:24.222338 [NOTICE] libdingaling.c:1309 SecSEND: ------------------------------------------------------------------------------- 2009-07-14 10:20:24.222338 [NOTICE] libdingaling.c:1309 SecSEND: ------------------------------------------------------------------------------- 2009-07-14 10:20:24.944314 [INFO] libdingaling.c:1307 SecRECV: ------------------------------------------------------------------------------- unknown session From dome at tel.co.th Mon Jul 13 21:30:56 2009 From: dome at tel.co.th (Dome Charoenyost) Date: Tue, 14 Jul 2009 11:30:56 +0700 Subject: [Freeswitch-users] Originate in Dial plan In-Reply-To: <87f2f3b90907131810q219c04f3ra385e0e1d252d31c@mail.gmail.com> References: <8ccbff060907120959ma7c0ey22adeae196f872c3@mail.gmail.com> <87f2f3b90907131810q219c04f3ra385e0e1d252d31c@mail.gmail.com> Message-ID: <8ccbff060907132130l329f7e1at87a16da0c2108e2@mail.gmail.com> 2009/7/14 Michael Collins : > What phone number do you call back? I mean, how do you know what the > customer's number is? Do you go by the caller id number? yes callback to caller id > > -MC > > On Sun, Jul 12, 2009 at 9:59 AM, Dome Charoenyost wrote: >> >> Dear sir, >> ? ? ? ? I want to user dialplan callback to customer. is posible to >> to this is dialplan XML ? >> Now i use javascript. >> my call flow. >> 1. customer call >> 2. FS rining and wait until customer hangup >> 3. callback to customer number >> >> >> Best Regards. >> >> Dome C. >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From velu.technical at gmail.com Mon Jul 13 23:37:50 2009 From: velu.technical at gmail.com (velusamy velu) Date: Tue, 14 Jul 2009 12:07:50 +0530 Subject: [Freeswitch-users] Problem in Adding another user in default directory Message-ID: <1452e2980907132337w45cc550ci587f811d9f3851f@mail.gmail.com> Dear All, How to create another user agent like 1000 to 1919 in internal profile. Please provide some steps to do it.. Thanks in Advance, Velusamy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090714/5f6c7ccc/attachment.html From jason at jasonjgw.net Tue Jul 14 00:10:02 2009 From: jason at jasonjgw.net (Jason White) Date: Tue, 14 Jul 2009 17:10:02 +1000 Subject: [Freeswitch-users] Problem in Adding another user in default directory In-Reply-To: <1452e2980907132337w45cc550ci587f811d9f3851f@mail.gmail.com> References: <1452e2980907132337w45cc550ci587f811d9f3851f@mail.gmail.com> Message-ID: <20090714071002.GA6485@jdc.jasonjgw.net> velusamy velu wrote: > How to create another user agent like 1000 to 1919 in internal > profile. Copy one of the existing files, edit it, and make all of the obvious changes. Then edit your dial plan so that the extension can be called. From msc at freeswitch.org Tue Jul 14 00:48:28 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 14 Jul 2009 00:48:28 -0700 Subject: [Freeswitch-users] Originate in Dial plan In-Reply-To: <8ccbff060907132130l329f7e1at87a16da0c2108e2@mail.gmail.com> References: <8ccbff060907120959ma7c0ey22adeae196f872c3@mail.gmail.com> <87f2f3b90907131810q219c04f3ra385e0e1d252d31c@mail.gmail.com> <8ccbff060907132130l329f7e1at87a16da0c2108e2@mail.gmail.com> Message-ID: <87f2f3b90907140048m7e1eea18x46e9418e43fb4007@mail.gmail.com> On Mon, Jul 13, 2009 at 9:30 PM, Dome Charoenyost wrote: > 2009/7/14 Michael Collins : > > What phone number do you call back? I mean, how do you know what the > > customer's number is? Do you go by the caller id number? > yes callback to caller id > Okay, here's a dialplan snippet that I used to successfully do the autocallback. In my case I used ext 1001 as the customer and portaudio as the "agent" if you get my meaning. Extension 1001 dials 9902, hangs up, and immediately the api_hangup_hook's originate command is executed. In this case it calls portaudio/auto_answer for the A-leg and user/1001 as the B-leg. I don't claim that it's the prettiest thing in the world but it definitely works. You'll need to adjust according to your specific situation. Let us know how it goes. BTW, what is the reason for this type of scenario? Just curious. -MC > > > > > > -MC > > > > On Sun, Jul 12, 2009 at 9:59 AM, Dome Charoenyost > wrote: > >> > >> Dear sir, > >> I want to user dialplan callback to customer. is posible to > >> to this is dialplan XML ? > >> Now i use javascript. > >> my call flow. > >> 1. customer call > >> 2. FS rining and wait until customer hangup > >> 3. callback to customer number > >> > >> > >> Best Regards. > >> > >> Dome C. > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090714/54ad43f0/attachment.html From andy at fabulous4.co.uk Tue Jul 14 02:07:09 2009 From: andy at fabulous4.co.uk (Andy) Date: Tue, 14 Jul 2009 10:07:09 +0100 Subject: [Freeswitch-users] Problems with Ping and re-registering brokengateways In-Reply-To: <191c3a030907130859y2665da03xf704f9db6070fb3c@mail.gmail.com> References: <0324DD608A074940AAC173B85A3978F2@D810> <191c3a030907130859y2665da03xf704f9db6070fb3c@mail.gmail.com> Message-ID: <78A03DDB5A5043C597807D0DFCF00DCA@D810> Hi Anthony, Thanks for your reply. The trace of the ping request looks like this. Any clues? send 674 bytes to udp/[77.240.48.94]:5060 at 09:07:25.479313: ------------------------------------------------------------------------ OPTIONS sip:voiptalk.org;transport=udp SIP/2.0 Via: SIP/2.0/UDP 77.86.49.249;rport;branch=z9hG4bK8rBQ4a33Ny02K Max-Forwards: 70 From: ;tag=a8U21NNZ23tBB To: Call-ID: 96da25e1-eaf8-122c-02ae-0019d15ec7e1 CSeq: 117662779 OPTIONS Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13850 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 ------------------------------------------------------------------------ recv 352 bytes from udp/[77.240.48.94]:5060 at 09:07:25.486124: ------------------------------------------------------------------------ SIP/2.0 484 Address Incomplete Via: SIP/2.0/UDP 77.86.49.249;rport=5060;branch=z9hG4bK8rBQ4a33Ny02K From: ;tag=a8U21NNZ23tBB To: ;tag=fd79486175647ed1617969929fdaad02.f21c Call-ID: 96da25e1-eaf8-122c-02ae-0019d15ec7e1 CSeq: 117662779 OPTIONS Server: OpenSIPS (1.5.1-notls (x86_64/linux)) Content-Length: 0 ------------------------------------------------------------------------ 2009-07-14 10:07:25.483507 [WARNING] sofia.c:2810 Ping failed voiptalk.org 2009-07-14 10:07:25.483507 [WARNING] sofia.c:2813 Unregister voiptalk.org Andy _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: 13 July 2009 16:59 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Problems with Ping and re-registering brokengateways Are they ignoring the options packet we send them or are they maybe getting lost behind NAT? we send an OPTIONS and even if we get a error back we consider that a successful reply. We did have a patch into SVN very recently to correct a problem with OPTIONS ping in a NAT situation. Maybe try latest trunk first then capture the console log with sip traffic in place if it still does not work so we can have a look. to capture the log use these 2 commands from the cli. sofia profile internal siptrace on console loglevel debug On Mon, Jul 13, 2009 at 7:59 AM, Andy wrote: Hi, I'm fairly sure my problem lies with my voip provider VoipTalk but wonder if you could help me understand a couple of things. My config is very simple, I'm using freeswitch to accept incoming calls via a voip gateway and record messages. Here's the problem: - When freeswitch starts the gateways are all created and register correctly - I have the ping parameter set to make sure the gateway stays alive. - The first time freeswitch pings the gateway it fails even though the registration appears intact as calls are still coming through to freeswitch - Freeswitch then tries to re-register the gateway but this fails. The SIP trace shows an Unauthorized message and the actual log entry is 'Registration Failed with status Operation has no matching challenge [904]' - eventually the registration times out with my provider and all is lost. - if I call 'sofia profile external restart' or restart the software this fixes the problem My questions are: 1) Why would the ping fail when the registration appears to be intact? 2) Whay would the auto re-register not work but a restart would? This ones driving me nuts so any help greatly appreciated. regards Andy _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference 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/20090714/2bf45931/attachment-0001.html From dome at tel.co.th Tue Jul 14 04:52:47 2009 From: dome at tel.co.th (Dome Charoenyost) Date: Tue, 14 Jul 2009 18:52:47 +0700 Subject: [Freeswitch-users] Originate in Dial plan In-Reply-To: <87f2f3b90907140048m7e1eea18x46e9418e43fb4007@mail.gmail.com> References: <8ccbff060907120959ma7c0ey22adeae196f872c3@mail.gmail.com> <87f2f3b90907131810q219c04f3ra385e0e1d252d31c@mail.gmail.com> <8ccbff060907132130l329f7e1at87a16da0c2108e2@mail.gmail.com> <87f2f3b90907140048m7e1eea18x46e9418e43fb4007@mail.gmail.com> Message-ID: <8ccbff060907140452s7b8bc06esd03b45424a0a545c@mail.gmail.com> Thanks it's work api_hangup_hook i'm looking for :) Dome C. 2009/7/14 Michael Collins : > > > On Mon, Jul 13, 2009 at 9:30 PM, Dome Charoenyost wrote: >> >> 2009/7/14 Michael Collins : >> > What phone number do you call back? I mean, how do you know what the >> > customer's number is? Do you go by the caller id number? >> yes callback to caller id > > Okay, here's a dialplan snippet that I used to successfully do the > autocallback. In my case I used ext 1001 as the customer and portaudio as > the "agent" if you get my meaning. Extension 1001 dials 9902, hangs up, and > immediately the api_hangup_hook's originate command is executed. In this > case it calls portaudio/auto_answer for the A-leg and user/1001 as the > B-leg. I don't claim that it's the prettiest thing in the world but it > definitely works. You'll need to adjust according to your specific > situation. > > ? > ??? > ??? > ????? > ????? > ????? > ????? > ????? > ????? > ??? > ? > > ? > ??? > ????? > ??? > ? > > > Let us know how it goes. BTW, what is the reason for this type of scenario? > Just curious. > -MC >> >> >> > >> > -MC >> > >> > On Sun, Jul 12, 2009 at 9:59 AM, Dome Charoenyost >> > wrote: >> >> >> >> Dear sir, >> >> ? ? ? ? I want to user dialplan callback to customer. is posible to >> >> to this is dialplan XML ? >> >> Now i use javascript. >> >> my call flow. >> >> 1. customer call >> >> 2. FS rining and wait until customer hangup >> >> 3. callback to customer number >> >> >> >> >> >> Best Regards. >> >> >> >> Dome C. >> >> >> >> _______________________________________________ >> >> Freeswitch-users mailing list >> >> Freeswitch-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> > >> > >> > _______________________________________________ >> > Freeswitch-users mailing list >> > Freeswitch-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From elihayun at gmail.com Tue Jul 14 05:02:18 2009 From: elihayun at gmail.com (Eli Hayun) Date: Tue, 14 Jul 2009 15:02:18 +0300 Subject: [Freeswitch-users] Get voicemail messages Message-ID: <4A5C73CA.40306@savion.huji.ac.il> Hi I am not using fixed xml files for the extension registration. I have LUA script to return an XML string to FS. Everything goes fine until I am trying to get the voice messages. When am entering my id, FS (or voicemail module) try to get the xml for that id, but it cant find it. My lua script did NOT recieved any xml request at that point. What should I do to solve the problem. Thanks Eli Hayun From yehavi.bourvine at gmail.com Tue Jul 14 05:12:27 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Tue, 14 Jul 2009 15:12:27 +0300 Subject: [Freeswitch-users] Originate in Dial plan In-Reply-To: <87f2f3b90907140048m7e1eea18x46e9418e43fb4007@mail.gmail.com> References: <8ccbff060907120959ma7c0ey22adeae196f872c3@mail.gmail.com> <87f2f3b90907131810q219c04f3ra385e0e1d252d31c@mail.gmail.com> <8ccbff060907132130l329f7e1at87a16da0c2108e2@mail.gmail.com> <87f2f3b90907140048m7e1eea18x46e9418e43fb4007@mail.gmail.com> Message-ID: 2009/7/14 Michael Collins > > > On Mon, Jul 13, 2009 at 9:30 PM, Dome Charoenyost wrote: > >> 2009/7/14 Michael Collins : >> > What phone number do you call back? I mean, how do you know what the >> > customer's number is? Do you go by the caller id number? >> yes callback to caller id >> > > Okay, here's a dialplan snippet that I used to successfully do the > autocallback. In my case I used ext 1001 as the customer and portaudio as > the "agent" if you get my meaning. Extension 1001 dials 9902, hangs up, and > immediately the api_hangup_hook's originate command is executed. In this > case it calls portaudio/auto_answer for the A-leg and user/1001 as the > B-leg. I don't claim that it's the prettiest thing in the world but it > definitely works. You'll need to adjust according to your specific > situation. > > > > > > > > > > > > > > > > > > > > > Let us know how it goes. BTW, what is the reason for this type of scenario? > Just curious. > -MC > >> >> >> > >> > -MC >> > >> > On Sun, Jul 12, 2009 at 9:59 AM, Dome Charoenyost >> wrote: >> >> >> >> Dear sir, >> >> I want to user dialplan callback to customer. is posible to >> >> to this is dialplan XML ? >> >> Now i use javascript. >> >> my call flow. >> >> 1. customer call >> >> 2. FS rining and wait until customer hangup >> >> 3. callback to customer number >> >> >> >> >> >> Best Regards. >> >> >> >> Dome C. >> >> >> >> _______________________________________________ >> >> Freeswitch-users mailing list >> >> Freeswitch-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> > >> > >> > _______________________________________________ >> > Freeswitch-users mailing list >> > Freeswitch-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090714/133488d3/attachment.html From elihayun at gmail.com Tue Jul 14 05:19:33 2009 From: elihayun at gmail.com (Eli Hayun) Date: Tue, 14 Jul 2009 15:19:33 +0300 Subject: [Freeswitch-users] Getting xml_request in LUA Message-ID: <4A5C77D5.6040105@savion.huji.ac.il> In the Perl example I found: > > > How to access request parameters and how to return data > > You have two hashes that are populated for you by freeswitch. Those > hashes are: > > * %XML_REQUEST > * %XML_DATA > > I want to use LUA to set the directory and dialplan xml. How do I get > the XML_REQUEST/XML_DATA from LUA? Is this the information you are looking for? http://wiki.freeswitch.org/wiki/Mod_lua#For_serving_configuration Hi Thanks fro your answer No, this is not the information i am looking for. What I need is the "section" value that FS pass when using HTTP to get the XML from. This information is not available (at least, I don't know how to get it) Thanks Eli From msc at freeswitch.org Tue Jul 14 06:00:10 2009 From: msc at freeswitch.org (Michael S Collins) Date: Tue, 14 Jul 2009 06:00:10 -0700 Subject: [Freeswitch-users] Get voicemail messages In-Reply-To: <4A5C73CA.40306@savion.huji.ac.il> References: <4A5C73CA.40306@savion.huji.ac.il> Message-ID: <4429558B-4DA5-43BF-9824-D5B407546CCB@freeswitch.org> On Jul 14, 2009, at 5:02 AM, Eli Hayun wrote: > Hi > I am not using fixed xml files for the extension registration. I have > LUA script to return an XML string to FS. > Everything goes fine until I am trying to get the voice messages. > When am entering my id, FS (or voicemail module) try to get the xml > for > that id, but it cant find it. My lua script did NOT recieved any xml > request at that point. > What should I do to solve the problem. > > Thanks > Eli Hayun Can you pastebin the log? Be sure to press F8 to turn up the debug level. -MC From brian at freeswitch.org Tue Jul 14 06:02:43 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 14 Jul 2009 08:02:43 -0500 Subject: [Freeswitch-users] Gafachi no passing caller number In-Reply-To: <20090714023411.151100@gmx.net> References: <20090714023411.151100@gmx.net> Message-ID: <9376D332-3935-4019-A440-CADD8B4DBD6E@freeswitch.org> You single quote them. {effective_caller_id_name='Paul Gascogne',effective_caller_id_number=16478343812}sofia/gateway/ sip.gafachi.com/164783486421 /b On Jul 13, 2009, at 9:34 PM, Klaus Teller wrote: > Hi, > > I tend to believe that we already had this working. Here is my > origination string: > > {effective_caller_id_name=Paul > Gascogne,effective_caller_id_number=16478343812}sofia/gateway/ > sip.gafachi.com/164783486421 > > The caller number is not being passed to the destination. Is there > something i'm missing? > > Thanks, > > Klaus. From brian at freeswitch.org Tue Jul 14 06:04:07 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 14 Jul 2009 08:04:07 -0500 Subject: [Freeswitch-users] Problems with Ping and re-registering brokengateways In-Reply-To: <78A03DDB5A5043C597807D0DFCF00DCA@D810> References: <0324DD608A074940AAC173B85A3978F2@D810> <191c3a030907130859y2665da03xf704f9db6070fb3c@mail.gmail.com> <78A03DDB5A5043C597807D0DFCF00DCA@D810> Message-ID: <6C973FA2-4148-46C5-B98C-13569CB7533E@freeswitch.org> They probably shouldn't be responding 484 to them... hrm /b On Jul 14, 2009, at 4:07 AM, Andy wrote: > Hi Anthony, > > Thanks for your reply. The trace of the ping request looks like > this. Any clues? > > send 674 bytes to udp/[77.240.48.94]:5060 at 09:07:25.479313: > > ------------------------------------------------------------------------ > OPTIONS sip:voiptalk.org;transport=udp SIP/2.0 > Via: SIP/2.0/UDP 77.86.49.249;rport;branch=z9hG4bK8rBQ4a33Ny02K > Max-Forwards: 70 > From: ;tag=a8U21NNZ23tBB > To: > Call-ID: 96da25e1-eaf8-122c-02ae-0019d15ec7e1 > CSeq: 117662779 OPTIONS > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13850 > 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 > > > ------------------------------------------------------------------------ > recv 352 bytes from udp/[77.240.48.94]:5060 at 09:07:25.486124: > > ------------------------------------------------------------------------ > SIP/2.0 484 Address Incomplete > Via: SIP/2.0/UDP > 77.86.49.249;rport=5060;branch=z9hG4bK8rBQ4a33Ny02K > From: ;tag=a8U21NNZ23tBB > To: ;tag=fd79486175647ed1617969929fdaad02.f21c > Call-ID: 96da25e1-eaf8-122c-02ae-0019d15ec7e1 > CSeq: 117662779 OPTIONS > Server: OpenSIPS (1.5.1-notls (x86_64/linux)) > Content-Length: 0 > > ------------------------------------------------------------------------ > 2009-07-14 10:07:25.483507 [WARNING] sofia.c:2810 Ping failed > voiptalk.org > 2009-07-14 10:07:25.483507 [WARNING] sofia.c:2813 Unregister > voiptalk.org > > Andy > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090714/6410520b/attachment-0001.html From yivzhenko at mksat.net Tue Jul 14 06:18:03 2009 From: yivzhenko at mksat.net (Yuriy Ivzhenko) Date: Tue, 14 Jul 2009 16:18:03 +0300 Subject: [Freeswitch-users] Freeswitch architecture In-Reply-To: <347F12AB-EA05-4090-A203-BDC011F02411@freeswitch.org> References: <89c9bbf80907080542y39f2b234tff9163ea34c0b968@mail.gmail.com> <347F12AB-EA05-4090-A203-BDC011F02411@freeswitch.org> Message-ID: <200907141618.03295.yivzhenko@mksat.net> On Wednesday 08 July 2009 16:29:50 Brian West wrote: > http://wiki.freeswitch.org i not found any essential information about architecture :-((((( .... may be bad looking? > > /b > > On Jul 8, 2009, at 7:42 AM, Maxim Tsvetov wrote: > > Hi All > > > > Where can I get information about internal Freeswitch architecture: > > 1) how modules interoperates with each other (maybe using corba or > > com > > objects or something else) > > 2) how core interoperates with other modules > > 3) how javascript function is translated to internal commands. > > > > In addition if you cand send me some schemas of Freeswitch > > architecture > > that will be great. > > > > Regards, > > Maxim Tsvetov > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090714/8df8bc10/attachment.html From brian at freeswitch.org Tue Jul 14 06:18:58 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 14 Jul 2009 08:18:58 -0500 Subject: [Freeswitch-users] Freeswitch architecture In-Reply-To: <200907141618.03295.yivzhenko@mksat.net> References: <89c9bbf80907080542y39f2b234tff9163ea34c0b968@mail.gmail.com> <347F12AB-EA05-4090-A203-BDC011F02411@freeswitch.org> <200907141618.03295.yivzhenko@mksat.net> Message-ID: <722CF8A4-9768-4F05-AAC9-3FE4305E6370@freeswitch.org> http://wiki.freeswitch.org/wiki/Main_Page#FreeSWITCH.E2.84.A2_Architecture /b On Jul 14, 2009, at 8:18 AM, Yuriy Ivzhenko wrote: > On Wednesday 08 July 2009 16:29:50 Brian West wrote: > > http://wiki.freeswitch.org > i not found any essential information about architecture :-((((( > .... may be bad looking? > > > > /b > > > > On Jul 8, 2009, at 7:42 AM, Maxim Tsvetov wrote: > > > Hi All > > > > > > Where can I get information about internal Freeswitch > architecture: > > > 1) how modules interoperates with each other (maybe using > corba or > > > com > > > objects or something else) > > > 2) how core interoperates with other modules > > > 3) how javascript function is translated to internal commands. > > > > > > In addition if you cand send me some schemas of Freeswitch > > > architecture > > > that will be great. > > > > > > Regards, > > > Maxim Tsvetov > > > > > > _______________________________________________ > > > Freeswitch-users mailing list > > > Freeswitch-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090714/031d19a8/attachment.html From elihayun at gmail.com Tue Jul 14 06:26:42 2009 From: elihayun at gmail.com (Eli Hayun) Date: Tue, 14 Jul 2009 16:26:42 +0300 Subject: [Freeswitch-users] Freeswitch-users Digest, Vol 37, Issue 75 In-Reply-To: References: Message-ID: <4A5C8792.1030807@savion.huji.ac.il> Message: 5 Date: Tue, 14 Jul 2009 06:00:10 -0700 From: Michael S Collins Subject: Re: [Freeswitch-users] Get voicemail messages To: "freeswitch-users at lists.freeswitch.org" Message-ID: <4429558B-4DA5-43BF-9824-D5B407546CCB at freeswitch.org> Content-Type: text/plain; charset=us-ascii; format=flowed; delsp=yes On Jul 14, 2009, at 5:02 AM, Eli Hayun wrote: > > Hi > > I am not using fixed xml files for the extension registration. I have > > LUA script to return an XML string to FS. > > Everything goes fine until I am trying to get the voice messages. > > When am entering my id, FS (or voicemail module) try to get the xml > > for > > that id, but it cant find it. My lua script did NOT recieved any xml > > request at that point. > > What should I do to solve the problem. > > > > Thanks > > Eli Hayun > Can you pastebin the log? Be sure to press F8 to turn up the debug level. Here is the log ( with the XML string for the requested id (80670) freeswitch at tst-sip-srv> API CALL [console(loglevel 7)] output: +OK console log level set to DEBUG freeswitch at tst-sip-srv> 2009-07-14 16:05:15.860230 [DEBUG] sofia.c:4554 IP 132.64.4.238 Rejected by acl "domains". Falling back to Digest auth. 2009-07-14 16:05:16.88367 [DEBUG] sofia.c:4554 IP 132.64.4.238 Rejected by acl "domains". Falling back to Digest auth. ->
<-- 2009-07-14 16:05:16.108310 [NOTICE] switch_channel.c:602 New Channel sofia/internal/80670 at 132.64.3.86 [f9716554-7076-11de-9237-7d312efadfc4] 2009-07-14 16:05:16.112125 [DEBUG] sofia.c:3215 Channel sofia/internal/80670 at 132.64.3.86 entering state [received][100] 2009-07-14 16:05:16.112125 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/80670 at 132.64.3.86) Running State Change CS_NEW 2009-07-14 16:05:16.112125 [DEBUG] sofia.c:3222 Remote SDP: v=0 o=root 1385664886 1385664886 IN IP4 132.64.4.238 s=call c=IN IP4 132.64.4.238 t=0 0 m=audio 57682 RTP/AVP 0 8 9 99 3 18 4 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:99 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:18 g729/8000 a=rtpmap:4 g723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:both a=ptime:20 2009-07-14 16:05:16.112125 [DEBUG] sofia_glue.c:3070 Audio Codec Compare [pcmu:0:8000:20]/[G7221:115:32000:20] 2009-07-14 16:05:16.112125 [DEBUG] sofia_glue.c:3070 Audio Codec Compare [pcmu:0:8000:20]/[G7221:107:16000:20] 2009-07-14 16:05:16.112125 [DEBUG] sofia_glue.c:3070 Audio Codec Compare [pcmu:0:8000:20]/[G722:9:8000:20] 2009-07-14 16:05:16.112125 [DEBUG] sofia_glue.c:3070 Audio Codec Compare [pcmu:0:8000:20]/[PCMU:0:8000:20] 2009-07-14 16:05:16.112125 [DEBUG] sofia_glue.c:2028 Set Codec sofia/internal/80670 at 132.64.3.86 PCMU/8000 20 ms 160 samples 2009-07-14 16:05:16.112125 [DEBUG] sofia_glue.c:3030 Set 2833 dtmf payload to 101 2009-07-14 16:05:16.112125 [DEBUG] switch_core_state_machine.c:403 (sofia/internal/80670 at 132.64.3.86) State NEW 2009-07-14 16:05:16.112125 [DEBUG] sofia.c:3381 (sofia/internal/80670 at 132.64.3.86) State Change CS_NEW -> CS_INIT 2009-07-14 16:05:16.112125 [DEBUG] switch_core_session.c:933 Send signal sofia/internal/80670 at 132.64.3.86 [BREAK] 2009-07-14 16:05:16.112125 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/80670 at 132.64.3.86) Running State Change CS_INIT 2009-07-14 16:05:16.116244 [DEBUG] switch_core_state_machine.c:480 (sofia/internal/80670 at 132.64.3.86) State INIT 2009-07-14 16:05:16.116244 [DEBUG] mod_sofia.c:83 sofia/internal/80670 at 132.64.3.86 SOFIA INIT 2009-07-14 16:05:16.116244 [DEBUG] mod_sofia.c:111 (sofia/internal/80670 at 132.64.3.86) State Change CS_INIT -> CS_ROUTING 2009-07-14 16:05:16.116244 [DEBUG] switch_core_session.c:933 Send signal sofia/internal/80670 at 132.64.3.86 [BREAK] 2009-07-14 16:05:16.116244 [DEBUG] switch_core_state_machine.c:480 (sofia/internal/80670 at 132.64.3.86) State INIT going to sleep 2009-07-14 16:05:16.116244 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/80670 at 132.64.3.86) Running State Change CS_ROUTING 2009-07-14 16:05:16.116244 [DEBUG] switch_core_state_machine.c:483 (sofia/internal/80670 at 132.64.3.86) State ROUTING 2009-07-14 16:05:16.116244 [DEBUG] mod_sofia.c:130 sofia/internal/80670 at 132.64.3.86 SOFIA ROUTING 2009-07-14 16:05:16.116244 [DEBUG] switch_core_state_machine.c:78 sofia/internal/80670 at 132.64.3.86 Standard ROUTING 2009-07-14 16:05:16.116244 [INFO] mod_dialplan_xml.c:252 Processing phone-1->*98 in context default Dialplan: sofia/internal/80670 at 132.64.3.86 parsing [default->unloop] continue=false Dialplan: sofia/internal/80670 at 132.64.3.86 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/internal/80670 at 132.64.3.86 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: sofia/internal/80670 at 132.64.3.86 parsing [default->tod_example] continue=true Dialplan: sofia/internal/80670 at 132.64.3.86 Regex (PASS) [tod_example] ${strftime(%w)}(2) =~ /^([1-5])$/ break=on-false Dialplan: sofia/internal/80670 at 132.64.3.86 Regex (PASS) [tod_example] ${strftime(%H%M)}(1605) =~ /^((09|1[0-7])[0-5][0-9]|1800)$/ break=on-false Dialplan: sofia/internal/80670 at 132.64.3.86 Action set(open=true) Dialplan: sofia/internal/80670 at 132.64.3.86 parsing [default->global-intercept] continue=false Dialplan: sofia/internal/80670 at 132.64.3.86 Regex (FAIL) [global-intercept] destination_number(*98) =~ /^886$/ break=on-false Dialplan: sofia/internal/80670 at 132.64.3.86 parsing [default->group-intercept] continue=false Dialplan: sofia/internal/80670 at 132.64.3.86 Regex (FAIL) [group-intercept] destination_number(*98) =~ /^\*8$/ break=on-false Dialplan: sofia/internal/80670 at 132.64.3.86 parsing [default->intercept-ext] continue=false Dialplan: sofia/internal/80670 at 132.64.3.86 Regex (FAIL) [intercept-ext] destination_number(*98) =~ /^\*\*(\d+)$/ break=on-false Dialplan: sofia/internal/80670 at 132.64.3.86 parsing [default->redial] continue=false Dialplan: sofia/internal/80670 at 132.64.3.86 Regex (FAIL) [redial] destination_number(*98) =~ /^870$/ break=on-false Dialplan: sofia/internal/80670 at 132.64.3.86 parsing [default->global] continue=true Dialplan: sofia/internal/80670 at 132.64.3.86 Regex (FAIL) [global] ${call_debug}(false) =~ /^true$/ break=never Dialplan: sofia/internal/80670 at 132.64.3.86 Regex (FAIL) [global] ${sip_has_crypto}() =~ /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=never Dialplan: sofia/internal/80670 at 132.64.3.86 Absolute Condition [global] Dialplan: sofia/internal/80670 at 132.64.3.86 Action hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) Dialplan: sofia/internal/80670 at 132.64.3.86 Action hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}) Dialplan: sofia/internal/80670 at 132.64.3.86 Action hash(insert/${domain_name}-last_dial/global/${uuid}) Dialplan: sofia/internal/80670 at 132.64.3.86 parsing [default->snom-demo-2] continue=false Dialplan: sofia/internal/80670 at 132.64.3.86 Regex (FAIL) [snom-demo-2] destination_number(*98) =~ /^9001$/ break=on-false Dialplan: sofia/internal/80670 at 132.64.3.86 parsing [default->snom-demo-1] continue=false Dialplan: sofia/internal/80670 at 132.64.3.86 Regex (FAIL) [snom-demo-1] destination_number(*98) =~ /^9000$/ break=on-false Dialplan: sofia/internal/80670 at 132.64.3.86 parsing [default->eavesdrop] continue=false Dialplan: sofia/internal/80670 at 132.64.3.86 Regex (FAIL) [eavesdrop] destination_number(*98) =~ /^88(.*)$|^\*0(.*)$/ break=on-false Dialplan: sofia/internal/80670 at 132.64.3.86 parsing [default->eavesdrop] continue=false Dialplan: sofia/internal/80670 at 132.64.3.86 Regex (FAIL) [eavesdrop] destination_number(*98) =~ /^779$/ break=on-false Dialplan: sofia/internal/80670 at 132.64.3.86 parsing [default->call_return] continue=false Dialplan: sofia/internal/80670 at 132.64.3.86 Regex (FAIL) [call_return] destination_number(*98) =~ /^\*69$|^869$|^lcr$/ break=on-false Dialplan: sofia/internal/80670 at 132.64.3.86 parsing [default->del-group] continue=false Dialplan: sofia/internal/80670 at 132.64.3.86 Regex (FAIL) [del-group] destination_number(*98) =~ /^80(\d{2})$/ break=on-false Dialplan: sofia/internal/80670 at 132.64.3.86 parsing [default->add-group] continue=false Dialplan: sofia/internal/80670 at 132.64.3.86 Regex (FAIL) [add-group] destination_number(*98) =~ /^81(\d{2})$/ break=on-false Dialplan: sofia/internal/80670 at 132.64.3.86 parsing [default->call-group-simo] continue=false Dialplan: sofia/internal/80670 at 132.64.3.86 Regex (FAIL) [call-group-simo] destination_number(*98) =~ /^82(\d{2})$/ break=on-false Dialplan: sofia/internal/80670 at 132.64.3.86 parsing [default->call-group-order] continue=false Dialplan: sofia/internal/80670 at 132.64.3.86 Regex (FAIL) [call-group-order] destination_number(*98) =~ /^83(\d{2})$/ break=on-false Dialplan: sofia/internal/80670 at 132.64.3.86 parsing [default->extension-intercom] continue=false Dialplan: sofia/internal/80670 at 132.64.3.86 Regex (FAIL) [extension-intercom] destination_number(*98) =~ /^(8888)$/ break=on-false Dialplan: sofia/internal/80670 at 132.64.3.86 parsing [default->Local_Extension] continue=false Dialplan: sofia/internal/80670 at 132.64.3.86 Regex (FAIL) [Local_Extension] destination_number(*98) =~ /^(80\d{3})$/ break=on-false Dialplan: sofia/internal/80670 at 132.64.3.86 parsing [default->group_dial_sales] continue=false Dialplan: sofia/internal/80670 at 132.64.3.86 Regex (FAIL) [group_dial_sales] destination_number(*98) =~ /^2000$/ break=on-false Dialplan: sofia/internal/80670 at 132.64.3.86 parsing [default->group_dial_support] continue=false Dialplan: sofia/internal/80670 at 132.64.3.86 Regex (FAIL) [group_dial_support] destination_number(*98) =~ /^2001$/ break=on-false Dialplan: sofia/internal/80670 at 132.64.3.86 parsing [default->group_dial_billing] continue=false Dialplan: sofia/internal/80670 at 132.64.3.86 Regex (FAIL) [group_dial_billing] destination_number(*98) =~ /^2002$/ break=on-false Dialplan: sofia/internal/80670 at 132.64.3.86 parsing [default->operator] continue=false Dialplan: sofia/internal/80670 at 132.64.3.86 Regex (FAIL) [operator] destination_number(*98) =~ /^(operator|0)$/ break=on-false Dialplan: sofia/internal/80670 at 132.64.3.86 parsing [default->vmain] continue=false Dialplan: sofia/internal/80670 at 132.64.3.86 Regex (PASS) [vmain] destination_number(*98) =~ /^vmain$|^86111$|^\*98$/ break=on-false Dialplan: sofia/internal/80670 at 132.64.3.86 Action answer() Dialplan: sofia/internal/80670 at 132.64.3.86 Action sleep(1000) Dialplan: sofia/internal/80670 at 132.64.3.86 Action voicemail(check default ${domain_name} ${caller_id_number}) 2009-07-14 16:05:16.120891 [DEBUG] switch_core_state_machine.c:114 (sofia/internal/80670 at 132.64.3.86) State Change CS_ROUTING -> CS_EXECUTE 2009-07-14 16:05:16.120891 [DEBUG] switch_core_session.c:933 Send signal sofia/internal/80670 at 132.64.3.86 [BREAK] 2009-07-14 16:05:16.120891 [DEBUG] switch_core_state_machine.c:483 (sofia/internal/80670 at 132.64.3.86) State ROUTING going to sleep 2009-07-14 16:05:16.120891 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/80670 at 132.64.3.86) Running State Change CS_EXECUTE 2009-07-14 16:05:16.120891 [DEBUG] switch_core_state_machine.c:490 (sofia/internal/80670 at 132.64.3.86) State EXECUTE 2009-07-14 16:05:16.120891 [DEBUG] mod_sofia.c:173 sofia/internal/80670 at 132.64.3.86 SOFIA EXECUTE 2009-07-14 16:05:16.120891 [DEBUG] switch_core_state_machine.c:151 sofia/internal/80670 at 132.64.3.86 Standard EXECUTE EXECUTE sofia/internal/80670 at 132.64.3.86 set(open=true) 2009-07-14 16:05:16.120891 [DEBUG] mod_dptools.c:748 sofia/internal/80670 at 132.64.3.86 SET [open]=[true] EXECUTE sofia/internal/80670 at 132.64.3.86 hash(insert/132.64.3.86-spymap/80670/f9716554-7076-11de-9237-7d312efadfc4) EXECUTE sofia/internal/80670 at 132.64.3.86 hash(insert/132.64.3.86-last_dial/80670/*98) EXECUTE sofia/internal/80670 at 132.64.3.86 hash(insert/132.64.3.86-last_dial/global/f9716554-7076-11de-9237-7d312efadfc4) EXECUTE sofia/internal/80670 at 132.64.3.86 answer() 2009-07-14 16:05:16.124221 [DEBUG] mod_dptools.c:649 sofia/internal/80670 at 132.64.3.86 receive message [ANSWER] 2009-07-14 16:05:16.124221 [DEBUG] sofia_glue.c:2262 AUDIO RTP [sofia/internal/80670 at 132.64.3.86] 132.64.3.86 port 31372 -> 132.64.4.238 port 57682 codec: 0 ms: 20 2009-07-14 16:05:16.124221 [DEBUG] switch_rtp.c:1138 Starting timer [soft] 160 bytes per 20ms 2009-07-14 16:05:16.138142 [DEBUG] mod_sofia.c:549 Local SDP sofia/internal/80670 at 132.64.3.86: v=0 o=FreeSWITCH 1247545344 1247545345 IN IP4 132.64.3.86 s=FreeSWITCH c=IN IP4 132.64.3.86 t=0 0 m=audio 31372 RTP/AVP 0 101 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 2009-07-14 16:05:16.138142 [DEBUG] switch_core_session.c:630 Send signal sofia/internal/80670 at 132.64.3.86 [BREAK] 2009-07-14 16:05:16.138142 [NOTICE] mod_dptools.c:649 Channel [sofia/internal/80670 at 132.64.3.86] has been answered 2009-07-14 16:05:16.138142 [DEBUG] switch_channel.c:182 sofia/internal/80670 at 132.64.3.86 receive message [AUDIO_SYNC] EXECUTE sofia/internal/80670 at 132.64.3.86 sleep(1000) 2009-07-14 16:05:16.138142 [DEBUG] switch_channel.c:182 sofia/internal/80670 at 132.64.3.86 receive message [AUDIO_SYNC] 2009-07-14 16:05:16.138142 [DEBUG] sofia.c:3215 Channel sofia/internal/80670 at 132.64.3.86 entering state [completed][200] 2009-07-14 16:05:16.216623 [DEBUG] sofia.c:3215 Channel sofia/internal/80670 at 132.64.3.86 entering state [ready][200] EXECUTE sofia/internal/80670 at 132.64.3.86 voicemail(check default 132.64.3.86 80670) 2009-07-14 16:05:17.148904 [DEBUG] mod_voicemail.c:776 [default] rwlock 2009-07-14 16:05:17.148904 [DEBUG] switch_ivr_play_say.c:118 No language specified - Using [en] 2009-07-14 16:05:17.152174 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[voicemail/vm-hello.wav] (en:en) 2009-07-14 16:05:17.152174 [DEBUG] switch_ivr_play_say.c:1097 Codec Activated L16 at 8000hz 1 channels 20ms 2009-07-14 16:05:17.152174 [DEBUG] switch_core_io.c:649 sofia/internal/80670 at 132.64.3.86 receive message [TRANSCODING_NECESSARY] 2009-07-14 16:05:18.368781 [DEBUG] switch_ivr_play_say.c:1391 done playing file 2009-07-14 16:05:18.488598 [DEBUG] switch_channel.c:182 sofia/internal/80670 at 132.64.3.86 receive message [AUDIO_SYNC] 2009-07-14 16:05:18.608382 [DEBUG] switch_channel.c:182 sofia/internal/80670 at 132.64.3.86 receive message [AUDIO_SYNC] 2009-07-14 16:05:18.708720 [WARNING] mod_voicemail.c:2072 Can't find user [80670 at 132.64.3.86] 2009-07-14 16:05:18.708720 [DEBUG] switch_ivr_play_say.c:118 No language specified - Using [en] 2009-07-14 16:05:18.712630 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[voicemail/vm-goodbye.wav] (en:en) 2009-07-14 16:05:18.712630 [DEBUG] switch_ivr_play_say.c:1097 Codec Activated L16 at 8000hz 1 channels 20ms 2009-07-14 16:05:18.712630 [DEBUG] switch_core_io.c:649 sofia/internal/80670 at 132.64.3.86 receive message [TRANSCODING_NECESSARY] 2009-07-14 16:05:19.188996 [DEBUG] switch_ivr_play_say.c:1391 done playing file 2009-07-14 16:05:19.308974 [NOTICE] switch_core_state_machine.c:179 Hangup sofia/internal/80670 at 132.64.3.86 [CS_EXECUTE] [NORMAL_CLEARING] 2009-07-14 16:05:19.308974 [DEBUG] switch_channel.c:1683 Send signal sofia/internal/80670 at 132.64.3.86 [KILL] 2009-07-14 16:05:19.308974 [DEBUG] switch_core_session.c:933 Send signal sofia/internal/80670 at 132.64.3.86 [BREAK] 2009-07-14 16:05:19.308974 [DEBUG] switch_core_state_machine.c:490 (sofia/internal/80670 at 132.64.3.86) State EXECUTE going to sleep 2009-07-14 16:05:19.308974 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/80670 at 132.64.3.86) Running State Change CS_HANGUP 2009-07-14 16:05:19.308974 [DEBUG] switch_core_state_machine.c:433 (sofia/internal/80670 at 132.64.3.86) State HANGUP 2009-07-14 16:05:19.308974 [DEBUG] mod_sofia.c:338 Channel sofia/internal/80670 at 132.64.3.86 hanging up, cause: NORMAL_CLEARING 2009-07-14 16:05:19.308974 [DEBUG] mod_sofia.c:393 Sending BYE to sofia/internal/80670 at 132.64.3.86 2009-07-14 16:05:19.312163 [DEBUG] switch_core_state_machine.c:46 sofia/internal/80670 at 132.64.3.86 Standard HANGUP, cause: NORMAL_CLEARING 2009-07-14 16:05:19.312163 [DEBUG] switch_core_state_machine.c:433 (sofia/internal/80670 at 132.64.3.86) State HANGUP going to sleep 2009-07-14 16:05:19.312163 [DEBUG] switch_core_state_machine.c:475 (sofia/internal/80670 at 132.64.3.86) State Change CS_HANGUP -> CS_REPORTING 2009-07-14 16:05:19.312163 [DEBUG] switch_core_session.c:933 Send signal sofia/internal/80670 at 132.64.3.86 [BREAK] 2009-07-14 16:05:19.312163 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/80670 at 132.64.3.86) Running State Change CS_REPORTING 2009-07-14 16:05:19.312163 [DEBUG] switch_core_state_machine.c:607 (sofia/internal/80670 at 132.64.3.86) State REPORTING 2009-07-14 16:05:19.312163 [DEBUG] switch_core_state_machine.c:53 sofia/internal/80670 at 132.64.3.86 Standard REPORTING, cause: NORMAL_CLEARING 2009-07-14 16:05:19.312163 [DEBUG] switch_core_state_machine.c:607 (sofia/internal/80670 at 132.64.3.86) State REPORTING going to sleep 2009-07-14 16:05:19.312163 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/80670 at 132.64.3.86) State Change CS_REPORTING -> CS_DESTROY 2009-07-14 16:05:19.312163 [DEBUG] switch_core_session.c:1067 Session 7 (sofia/internal/80670 at 132.64.3.86) Locked, Waiting on external entities 2009-07-14 16:05:19.312163 [NOTICE] switch_core_session.c:1085 Session 7 (sofia/internal/80670 at 132.64.3.86) Ended 2009-07-14 16:05:19.312163 [NOTICE] switch_core_session.c:1087 Close Channel sofia/internal/80670 at 132.64.3.86 [CS_DESTROY] 2009-07-14 16:05:19.312163 [DEBUG] switch_core_state_machine.c:559 (sofia/internal/80670 at 132.64.3.86) State DESTROY 2009-07-14 16:05:19.312163 [DEBUG] mod_sofia.c:255 sofia/internal/80670 at 132.64.3.86 SOFIA DESTROY 2009-07-14 16:05:19.312163 [DEBUG] switch_core_state_machine.c:60 sofia/internal/80670 at 132.64.3.86 Standard DESTROY 2009-07-14 16:05:19.312163 [DEBUG] switch_core_state_machine.c:559 (sofia/internal/80670 at 132.64.3.86) State DESTROY going to sleep - Thanks Eli Hayun From dftoro at yahoo.com Tue Jul 14 06:32:53 2009 From: dftoro at yahoo.com (Diego Toro) Date: Tue, 14 Jul 2009 06:32:53 -0700 (PDT) Subject: [Freeswitch-users] Dialogic cards Message-ID: <999132.13204.qm@web33505.mail.mud.yahoo.com> hi,?I think that may be... ? Analog cards:?D/41JCT-LS, D/120JCT-LS? (jct serie) Digital cards: D/600JCT-1E1 and DMV?serie. ? Diego --- On Mon, 7/13/09, Michael Collins wrote: From: Michael Collins Subject: Re: [Freeswitch-users] Dialogic cards To: freeswitch-users at lists.freeswitch.org Date: Monday, July 13, 2009, 4:45 PM What are the recommended cards to be used with freeswitch? Sangoma cards and Zaptel/DAHDI compatible cards work well. -MC -----Inline Attachment Follows----- _______________________________________________ Freeswitch-users mailing list Freeswitch-users at 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/20090714/f875361f/attachment.html From anthony.minessale at gmail.com Tue Jul 14 07:37:30 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 14 Jul 2009 09:37:30 -0500 Subject: [Freeswitch-users] Problems with Ping and re-registering brokengateways In-Reply-To: <78A03DDB5A5043C597807D0DFCF00DCA@D810> References: <0324DD608A074940AAC173B85A3978F2@D810> <191c3a030907130859y2665da03xf704f9db6070fb3c@mail.gmail.com> <78A03DDB5A5043C597807D0DFCF00DCA@D810> Message-ID: <191c3a030907140737s6c7cb091r91628802a38d6c3c@mail.gmail.com> yes this was plenty of information try r14242 On Tue, Jul 14, 2009 at 4:07 AM, Andy wrote: > Hi Anthony, > > Thanks for your reply. The trace of the ping request looks like this. Any > clues? > > send 674 bytes to udp/[77.240.48.94]:5060 at 09:07:25.479313: > ------------------------------------------------------------------------ > OPTIONS sip:voiptalk.org;transport=udp SIP/2.0 > Via: SIP/2.0/UDP 77.86.49.249;rport;branch=z9hG4bK8rBQ4a33Ny02K > Max-Forwards: 70 > From: ;tag=a8U21NNZ23tBB > To: > Call-ID: 96da25e1-eaf8-122c-02ae-0019d15ec7e1 > CSeq: 117662779 OPTIONS > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13850 > 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 > > ------------------------------------------------------------------------ > recv 352 bytes from udp/[77.240.48.94]:5060 at 09:07:25.486124: > ------------------------------------------------------------------------ > SIP/2.0 484 Address Incomplete > Via: SIP/2.0/UDP 77.86.49.249;rport=5060;branch=z9hG4bK8rBQ4a33Ny02K > From: ;tag=a8U21NNZ23tBB > To: ;tag=fd79486175647ed1617969929fdaad02.f21c > Call-ID: 96da25e1-eaf8-122c-02ae-0019d15ec7e1 > CSeq: 117662779 OPTIONS > Server: OpenSIPS (1.5.1-notls (x86_64/linux)) > Content-Length: 0 > ------------------------------------------------------------------------ > 2009-07-14 10:07:25.483507 [WARNING] sofia.c:2810 Ping failed voiptalk.org > 2009-07-14 10:07:25.483507 [WARNING] sofia.c:2813 Unregister voiptalk.org > > Andy > > ------------------------------ > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony > Minessale > *Sent:* 13 July 2009 16:59 > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Problems with Ping and re-registering > brokengateways > > Are they ignoring the options packet we send them or are they maybe getting > lost behind NAT? > we send an OPTIONS and even if we get a error back we consider that a > successful reply. > > We did have a patch into SVN very recently to correct a problem with > OPTIONS ping in a NAT situation. > > Maybe try latest trunk first then capture the console log with sip traffic > in place if it still does not work so we can have a look. > > to capture the log use these 2 commands from the cli. > > sofia profile internal siptrace on > console loglevel debug > > > > > On Mon, Jul 13, 2009 at 7:59 AM, Andy wrote: > >> Hi, >> >> I'm fairly sure my problem lies with my voip provider VoipTalk but wonder >> if you could help me understand a couple of things. My config is very >> simple, I'm using freeswitch to accept incoming calls via a voip gateway and >> record messages. Here's the problem: >> >> - When freeswitch starts the gateways are all created and register >> correctly >> >> - I have the ping parameter set to make sure the gateway stays alive. >> >> - The first time freeswitch pings the gateway it fails even though the >> registration appears intact as calls are still coming through to freeswitch >> >> - Freeswitch then tries to re-register the gateway but this fails. The SIP >> trace shows an Unauthorized message and the actual log entry is >> 'Registration Failed with status Operation has no matching challenge [904]' >> >> - eventually the registration times out with my provider and all is lost. >> >> - if I call 'sofia profile external restart' or restart the software this >> fixes the problem >> >> My questions are: >> >> 1) Why would the ping fail when the registration appears to be intact? >> 2) Whay would the auto re-register not work but a restart would? >> >> This ones driving me nuts so any help greatly appreciated. >> >> regards >> Andy >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > 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/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference 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/20090714/fe645332/attachment.html From anthony.minessale at gmail.com Tue Jul 14 08:04:32 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 14 Jul 2009 10:04:32 -0500 Subject: [Freeswitch-users] Get voicemail messages In-Reply-To: <4A5C73CA.40306@savion.huji.ac.il> References: <4A5C73CA.40306@savion.huji.ac.il> Message-ID: <191c3a030907140804y775562a3wca745dae4f67c1d5@mail.gmail.com> did you bind your lua script to directory lookups in addition to the dialplan? On Tue, Jul 14, 2009 at 7:02 AM, Eli Hayun wrote: > Hi > I am not using fixed xml files for the extension registration. I have > LUA script to return an XML string to FS. > Everything goes fine until I am trying to get the voice messages. > When am entering my id, FS (or voicemail module) try to get the xml for > that id, but it cant find it. My lua script did NOT recieved any xml > request at that point. > What should I do to solve the problem. > > Thanks > Eli Hayun > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference 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/20090714/6cc4a8d0/attachment-0001.html From klaus.teller at gmx.net Tue Jul 14 08:14:50 2009 From: klaus.teller at gmx.net (Klaus Teller) Date: Tue, 14 Jul 2009 17:14:50 +0200 Subject: [Freeswitch-users] Where is the country code? Message-ID: <20090714151450.79890@gmx.net> Hi, I'm playing with Freeswitch and Les.NET right now. It strikes me that the caller id as passed to javascript doesn't contain the country code. Anyone knows where teh issue lies? Thanks, Klaus. -- Jetzt kostenlos herunterladen: Internet Explorer 8 und Mozilla Firefox 3 - sicherer, schneller und einfacher! http://portal.gmx.net/de/go/chbrowser From pjintheusa at gmail.com Tue Jul 14 08:35:59 2009 From: pjintheusa at gmail.com (Phillip Jones) Date: Tue, 14 Jul 2009 11:35:59 -0400 Subject: [Freeswitch-users] leg_timeout Message-ID: <367751820907140835y322c576en1eb2ba931388d732@mail.gmail.com> Hi there, Here is my call flow: 1) leg A is bridged to leg B 2) when leg B is answered I play a confirm script - "please 1 to accept this call" I only want leg B to ring 20 seconds. BUT when the caller party answers, he should have as long as he needs to press 1. "leg_timeout" seems to be in play until the bridge is completed. I need it to reset when leg b is answered. I tried resetting the leg_timeout in the confirm script after leg b is answered. I also tried using leg_progress_timeout. Neither seemed to work. Any help or suggestions would be welcome. Phillip Jones -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090714/7be21ec1/attachment.html From svetikvoip at gmail.com Tue Jul 14 08:39:15 2009 From: svetikvoip at gmail.com (Svetik VOIP) Date: Tue, 14 Jul 2009 11:39:15 -0400 Subject: [Freeswitch-users] How to pass Message Waiting Ind. from VOIP provider directly to the phone Message-ID: <94790b850907140839m431f80cdu74d37dae17c5f7b@mail.gmail.com> Hi Guys, I have a question about configuring Message Waiting Indicator (MWI). I am running Freeswitch on Ubuntu 8.04 Desktop. I have Freeswitch connected to the external VOIP provider (voip.ms) and to the internal SIP box (Linksys RTP300) with one phone hooked to it. I let my VOIP provider to handle voicemail, and I would like Freeswitch to passthrough Message Waiting Indicator to the SIP box (and eventially to the phone), so it turns on when I have a new message in the VOIP provider voicemail . How to configure Freeswitch to achieve this? Before using Freeswitch I had SIP box hooked to my external provider directly, and I was getting Message Waiting Indicator on the phone no problem, but since I put FreeSwitch, it does not work anymore. I suspect it is because Freeswitch has its own voicemail system and it triggers MWI based on its state. I want to keep things simple and keep my voicemail at my VOIP provider. Thank you, Svetik -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090714/9f6af00d/attachment.html From larclap at yahoo.com Tue Jul 14 08:53:07 2009 From: larclap at yahoo.com (Lars Zeb) Date: Tue, 14 Jul 2009 08:53:07 -0700 Subject: [Freeswitch-users] Intercom error with SNOM Message-ID: <00a001ca049b$2e9e03b0$8bda0b10$@com> I am getting an error when I try to make an intercom call from a softphone to a SNOM 320. I get a "401 Unauthorized" in the siptrace and a "No Matching gateway found" in the log. I can successfully make an intercom call between my softphone and a Polycom 501, so it must be something with the SNOM. bkw suggested that the problem was in the challenge/response between FreeSWITCH and SNOM. In the SNOM's Setup/Advanced/Behavior page I have set "Challenge/Response" off, "Enable intercom" on and "Type of Intercom Answering" to "Handsfree". The error is still "401 Unauthorized". What more do I need to do? Thanks, Lars http://pastebin.freeswitch.org/9709 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090714/a77c8a3d/attachment.html From msc at freeswitch.org Tue Jul 14 09:16:38 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 14 Jul 2009 09:16:38 -0700 Subject: [Freeswitch-users] leg_timeout In-Reply-To: <367751820907140835y322c576en1eb2ba931388d732@mail.gmail.com> References: <367751820907140835y322c576en1eb2ba931388d732@mail.gmail.com> Message-ID: <87f2f3b90907140916o3f008a3br26bfca075a9e3d2e@mail.gmail.com> On Tue, Jul 14, 2009 at 8:35 AM, Phillip Jones wrote: > Hi there, > > Here is my call flow: > > 1) leg A is bridged to leg B > 2) when leg B is answered I play a confirm script - "please 1 to accept > this call" > > I only want leg B to ring 20 seconds. BUT when the caller party answers, he > should have as long as he needs to press 1. > > "leg_timeout" seems to be in play until the bridge is completed. I need it > to reset when leg b is answered. > Correct. Both "leg_timeout" and "leg_progress_timeout" are for controlling how long to wait prior to the B-leg answering. (leg_progress_timeout specifies how long to wait for any kind of progress, be it early media of some sort, ringing, or an answer.) > > I tried resetting the leg_timeout in the confirm script after leg b is > answered. I also tried using leg_progress_timeout. Neither seemed to work. > What exactly are you trying to do? The two variables you've mentioned shouldn't have any effect on the call after it has been established. > > Any help or suggestions would be welcome. > Could you pastebin your dialplan and a debug log of a call that does not work? See this page for some handy tips on using pastebin and collecting information for debugging purposes: http://wiki.freeswitch.org/wiki/Reporting_Bugs#Debugging_Steps -MC > > Phillip Jones > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090714/67646679/attachment.html From eweaver at meetingone.com Tue Jul 14 09:59:17 2009 From: eweaver at meetingone.com (Weaver, Eric) Date: Tue, 14 Jul 2009 09:59:17 -0700 Subject: [Freeswitch-users] Preventing disconnect on event_sockte close In-Reply-To: <87f2f3b90907131808l100afe7dwc95d0f624ff38d50@mail.gmail.com> References: <7DB369DA44D5CF42B6C23BD7D611A50FDA7E1D5FD6@VA3DIAXVS061.RED001.local> <87f2f3b90907131808l100afe7dwc95d0f624ff38d50@mail.gmail.com> Message-ID: <7DB369DA44D5CF42B6C23BD7D611A50FDA7E1D604D@VA3DIAXVS061.RED001.local> Tried it using FreeSWITCH Version 1.0.4pre9 (14036M) and the following steps Start netcat netcat -v -l -p 14000 place call, socket is connected via dial plan, enter the following. connect\n\n sendmsg call-command: execute execute-app-name: answer\n\n sendmsg call-command: execute execute-app-name: playback execute-app-arg: /home/eweaver/holdmusic.wav sendmsg call-command: execute execute-app-name: park Console window displays this message: 2009-07-14 10:42:29.245833 [ERR] switch_ivr.c:654 Cannot park channels that are under control already. at this point, ^C in the netcat window. Call is disconnected. Need to be able to park these calls so they can then be handled from an inbound event socket connection. eric From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Monday, July 13, 2009 7:08 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Preventing disconnect on event_sockte close I don't know if this will work for you but I just tested this scenario with uuid_park. After parking the call I disconnected the socket and the call continued. I did the same thing with uuid_transfer. After the transfer I disconnected the socket and the call continued. How are you handling the call and how is the socket getting disconnected? -MC On Mon, Jul 13, 2009 at 4:35 PM, Weaver, Eric > wrote: Using mod_event_socket in outbound mode, is there any to prevent a call from being disconnected when the outbound socket is closed ? I would like to handle the initial inbound call using outbound but after the disposition of the call is determined, close the socket and have that call managed using an inbound socket instead. Eric Weaver _______________________________________________ Freeswitch-users mailing list Freeswitch-users at 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/20090714/6dfdc4a1/attachment-0001.html From pjintheusa at gmail.com Tue Jul 14 10:02:37 2009 From: pjintheusa at gmail.com (Phillip Jones) Date: Tue, 14 Jul 2009 13:02:37 -0400 Subject: [Freeswitch-users] leg_timeout In-Reply-To: <87f2f3b90907140916o3f008a3br26bfca075a9e3d2e@mail.gmail.com> References: <367751820907140835y322c576en1eb2ba931388d732@mail.gmail.com> <87f2f3b90907140916o3f008a3br26bfca075a9e3d2e@mail.gmail.com> Message-ID: <367751820907141002g3a837a09meabdbc9743c3bee7@mail.gmail.com> Hi, Thanks for the reply. >> Both "leg_timeout" and "leg_progress_timeout" are for controlling how long to wait prior to the B-leg answering. I think this is my point. leg_timeout seems to control how long to wait prior to the bridge completeing, not the B-leg answering. In my situation I am using: Session.Execute("set", "group_confirm_key=exec"); Session.Execute("set", "group_confirm_file=javascript confirm.js"); my confirm.js prompts leg_b for 1 (to take the call). If leg_timeout is set to 10 you have 10 seconds to answer the call AND press 1. I just want call_timeout to be satisfied when the call is answered. Not when the called party presses 1 and the bridge is complete. I am new all this so I will work out how to use the pastebin etc. Thanks for your help. Phillip Jones On Tue, Jul 14, 2009 at 12:16 PM, Michael Collins wrote: > > > On Tue, Jul 14, 2009 at 8:35 AM, Phillip Jones wrote: > >> Hi there, >> >> Here is my call flow: >> >> 1) leg A is bridged to leg B >> 2) when leg B is answered I play a confirm script - "please 1 to accept >> this call" >> >> I only want leg B to ring 20 seconds. BUT when the caller party answers, >> he should have as long as he needs to press 1. >> >> "leg_timeout" seems to be in play until the bridge is completed. I need it >> to reset when leg b is answered. >> > > Correct. Both "leg_timeout" and "leg_progress_timeout" are for controlling > how long to wait prior to the B-leg answering. (leg_progress_timeout > specifies how long to wait for any kind of progress, be it early media of > some sort, ringing, or an answer.) > >> >> I tried resetting the leg_timeout in the confirm script after leg b is >> answered. I also tried using leg_progress_timeout. Neither seemed to work. >> > > What exactly are you trying to do? The two variables you've mentioned > shouldn't have any effect on the call after it has been established. > >> >> Any help or suggestions would be welcome. >> > > Could you pastebin your dialplan and a debug log of a call that does not > work? See this page for some handy tips on using pastebin and collecting > information for debugging purposes: > http://wiki.freeswitch.org/wiki/Reporting_Bugs#Debugging_Steps > > -MC > >> >> Phillip Jones >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090714/2a2904c6/attachment.html From msc at freeswitch.org Tue Jul 14 10:09:39 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 14 Jul 2009 10:09:39 -0700 Subject: [Freeswitch-users] Intercom error with SNOM In-Reply-To: <00a001ca049b$2e9e03b0$8bda0b10$@com> References: <00a001ca049b$2e9e03b0$8bda0b10$@com> Message-ID: <87f2f3b90907141009j28b6da38g4e1403a131fdcf6d@mail.gmail.com> On Tue, Jul 14, 2009 at 8:53 AM, Lars Zeb wrote: > I am getting an error when I try to make an intercom call from a > softphone to a SNOM 320. I get a ?401 Unauthorized? in the siptrace and a > ?No Matching gateway found? in the log. I can successfully make an intercom > call between my softphone and a Polycom 501, so it must be something with > the SNOM. > > > > bkw suggested that the problem was in the challenge/response between > FreeSWITCH and SNOM. In the SNOM?s Setup/Advanced/Behavior page I have set > ?Challenge/Response? off, ?Enable intercom? on and ?Type of Intercom > Answering? to ?Handsfree?. The error is still ?401 Unauthorized?. > > > > What more do I need to do? > Lars, It looks like FreeSWITCH is sending the challenge to the Snom. Note this line from the debug log: 2009-07-14 08:12:32.333079 [DEBUG] sofia.c:4587 IP 192.168.10.11 Rejected by acl "domains". Falling back to Digest auth. For testing purposes you can edit acl.conf.xml and add a new line right after: Add: Restart FS or issue this command at the CLI: reloadacl reloadxml Then try your call again. To learn more about how you can have your local users bypass the "domains" acl without editing acl.conf.xml then look in conf/directory/default/brian.xml. At the top of that file you will see a note about how adding a cidr= attribute to your user tag will let you bypass the domains ACL check. Enjoy! -MC > > > Thanks, Lars > > > > http://pastebin.freeswitch.org/9709 > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090714/cb8b333a/attachment.html From msc at freeswitch.org Tue Jul 14 10:11:42 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 14 Jul 2009 10:11:42 -0700 Subject: [Freeswitch-users] Preventing disconnect on event_sockte close In-Reply-To: <7DB369DA44D5CF42B6C23BD7D611A50FDA7E1D604D@VA3DIAXVS061.RED001.local> References: <7DB369DA44D5CF42B6C23BD7D611A50FDA7E1D5FD6@VA3DIAXVS061.RED001.local> <87f2f3b90907131808l100afe7dwc95d0f624ff38d50@mail.gmail.com> <7DB369DA44D5CF42B6C23BD7D611A50FDA7E1D604D@VA3DIAXVS061.RED001.local> Message-ID: <87f2f3b90907141011g46686396se27fa3cc0320963d@mail.gmail.com> On Tue, Jul 14, 2009 at 9:59 AM, Weaver, Eric wrote: > Tried it using FreeSWITCH Version 1.0.4pre9 (14036M) and the > following steps > > > > Start netcat > > > > netcat -v -l -p 14000 > > > > place call, socket is connected via dial plan, enter the following. > > > > connect\n\n > > > > sendmsg > > call-command: execute > > execute-app-name: answer\n\n > > > > > > sendmsg > > call-command: execute > > execute-app-name: playback > > execute-app-arg: /home/eweaver/holdmusic.wav > > > > > > > > > > sendmsg > > call-command: execute > > execute-app-name: park > > Try this: api uuid_park You'll need to capture the uuid at some point and store it. For testing I just manually copied and pasted it to/from the console screen. -MC > > > Console window displays this message: > > > > 2009-07-14 10:42:29.245833 [ERR] switch_ivr.c:654 Cannot park channels that > are under control already. > > > > at this point, ^C in the netcat window. Call is disconnected. > > > > > > Need to be able to park these calls so they can then be handled from an > inbound event socket connection. > > > > eric > > > > > > > > > > > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael > Collins > *Sent:* Monday, July 13, 2009 7:08 PM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Preventing disconnect on event_sockte > close > > > > I don't know if this will work for you but I just tested this scenario with > uuid_park. After parking the call I disconnected the socket and the call > continued. I did the same thing with uuid_transfer. After the transfer I > disconnected the socket and the call continued. > > How are you handling the call and how is the socket getting disconnected? > > -MC > > On Mon, Jul 13, 2009 at 4:35 PM, Weaver, Eric > wrote: > > Using mod_event_socket in outbound mode, is there any to prevent a call > from being disconnected when the outbound socket is closed ? I would like to > handle the initial inbound call using outbound but after the disposition of > the call is determined, close the socket and have that call managed using an > inbound socket instead. > > > > Eric Weaver > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090714/a11cca5c/attachment-0001.html From anthony.minessale at gmail.com Tue Jul 14 10:20:18 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 14 Jul 2009 12:20:18 -0500 Subject: [Freeswitch-users] leg_timeout In-Reply-To: <367751820907141002g3a837a09meabdbc9743c3bee7@mail.gmail.com> References: <367751820907140835y322c576en1eb2ba931388d732@mail.gmail.com> <87f2f3b90907140916o3f008a3br26bfca075a9e3d2e@mail.gmail.com> <367751820907141002g3a837a09meabdbc9743c3bee7@mail.gmail.com> Message-ID: <191c3a030907141020l48e8cb2cm6106ff4442ee1bed@mail.gmail.com> update to trunk and try setting group_confirm_cancel_timeout=true let me know if it works On Tue, Jul 14, 2009 at 12:02 PM, Phillip Jones wrote: > Hi, > > Thanks for the reply. > > >> Both "leg_timeout" and "leg_progress_timeout" are for controlling how > long to wait prior to the B-leg answering. > > I think this is my point. leg_timeout seems to control how long to wait > prior to the bridge completeing, not the B-leg answering. > > In my situation I am using: > > Session.Execute("set", "group_confirm_key=exec"); > Session.Execute("set", "group_confirm_file=javascript confirm.js"); > > my confirm.js prompts leg_b for 1 (to take the call). If leg_timeout is set > to 10 you have 10 seconds to answer the call AND press 1. > > I just want call_timeout to be satisfied when the call is answered. Not > when the called party presses 1 and the bridge is complete. > > I am new all this so I will work out how to use the pastebin etc. > > Thanks for your help. > > > Phillip Jones > > > > On Tue, Jul 14, 2009 at 12:16 PM, Michael Collins wrote: > >> >> >> On Tue, Jul 14, 2009 at 8:35 AM, Phillip Jones wrote: >> >>> Hi there, >>> >>> Here is my call flow: >>> >>> 1) leg A is bridged to leg B >>> 2) when leg B is answered I play a confirm script - "please 1 to accept >>> this call" >>> >>> I only want leg B to ring 20 seconds. BUT when the caller party answers, >>> he should have as long as he needs to press 1. >>> >>> "leg_timeout" seems to be in play until the bridge is completed. I need >>> it to reset when leg b is answered. >>> >> >> Correct. Both "leg_timeout" and "leg_progress_timeout" are for controlling >> how long to wait prior to the B-leg answering. (leg_progress_timeout >> specifies how long to wait for any kind of progress, be it early media of >> some sort, ringing, or an answer.) >> >>> >>> I tried resetting the leg_timeout in the confirm script after leg b is >>> answered. I also tried using leg_progress_timeout. Neither seemed to work. >>> >> >> What exactly are you trying to do? The two variables you've mentioned >> shouldn't have any effect on the call after it has been established. >> >>> >>> Any help or suggestions would be welcome. >>> >> >> Could you pastebin your dialplan and a debug log of a call that does not >> work? See this page for some handy tips on using pastebin and collecting >> information for debugging purposes: >> http://wiki.freeswitch.org/wiki/Reporting_Bugs#Debugging_Steps >> >> -MC >> >>> >>> Phillip Jones >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference 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/20090714/7c8a0205/attachment.html From anthony.minessale at gmail.com Tue Jul 14 10:24:37 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 14 Jul 2009 12:24:37 -0500 Subject: [Freeswitch-users] Preventing disconnect on event_sockte close In-Reply-To: <87f2f3b90907141011g46686396se27fa3cc0320963d@mail.gmail.com> References: <7DB369DA44D5CF42B6C23BD7D611A50FDA7E1D5FD6@VA3DIAXVS061.RED001.local> <87f2f3b90907131808l100afe7dwc95d0f624ff38d50@mail.gmail.com> <7DB369DA44D5CF42B6C23BD7D611A50FDA7E1D604D@VA3DIAXVS061.RED001.local> <87f2f3b90907141011g46686396se27fa3cc0320963d@mail.gmail.com> Message-ID: <191c3a030907141024u35c0509aqdc7112d82dba18cb@mail.gmail.com> or you can also do api uuid_transfer park inline or sendmsg call-command: execute execute-app-name: transfer execute-app-arg: park inline all of these will pull the call out of the control of your socket and into the care of the core. On Tue, Jul 14, 2009 at 12:11 PM, Michael Collins wrote: > > > On Tue, Jul 14, 2009 at 9:59 AM, Weaver, Eric wrote: > >> Tried it using FreeSWITCH Version 1.0.4pre9 (14036M) and the >> following steps >> >> >> >> Start netcat >> >> >> >> netcat -v -l -p 14000 >> >> >> >> place call, socket is connected via dial plan, enter the following. >> >> >> >> connect\n\n >> >> >> >> sendmsg >> >> call-command: execute >> >> execute-app-name: answer\n\n >> >> >> >> >> >> sendmsg >> >> call-command: execute >> >> execute-app-name: playback >> >> execute-app-arg: /home/eweaver/holdmusic.wav >> >> >> >> >> >> >> >> >> >> sendmsg >> >> call-command: execute >> >> execute-app-name: park >> >> > Try this: > api uuid_park > > You'll need to capture the uuid at some point and store it. For testing I > just manually copied and pasted it to/from the console screen. > -MC > >> >> >> Console window displays this message: >> >> >> >> 2009-07-14 10:42:29.245833 [ERR] switch_ivr.c:654 Cannot park channels >> that are under control already. >> >> >> >> at this point, ^C in the netcat window. Call is disconnected. >> >> >> >> >> >> Need to be able to park these calls so they can then be handled from an >> inbound event socket connection. >> >> >> >> eric >> >> >> >> >> >> >> >> >> >> >> >> >> >> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael >> Collins >> *Sent:* Monday, July 13, 2009 7:08 PM >> *To:* freeswitch-users at lists.freeswitch.org >> *Subject:* Re: [Freeswitch-users] Preventing disconnect on event_sockte >> close >> >> >> >> I don't know if this will work for you but I just tested this scenario >> with uuid_park. After parking the call I disconnected the socket and the >> call continued. I did the same thing with uuid_transfer. After the transfer >> I disconnected the socket and the call continued. >> >> How are you handling the call and how is the socket getting disconnected? >> >> -MC >> >> On Mon, Jul 13, 2009 at 4:35 PM, Weaver, Eric >> wrote: >> >> Using mod_event_socket in outbound mode, is there any to prevent a call >> from being disconnected when the outbound socket is closed ? I would like to >> handle the initial inbound call using outbound but after the disposition of >> the call is determined, close the socket and have that call managed using an >> inbound socket instead. >> >> >> >> Eric Weaver >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference 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/20090714/81383a23/attachment-0001.html From msc at freeswitch.org Tue Jul 14 10:37:46 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 14 Jul 2009 10:37:46 -0700 Subject: [Freeswitch-users] leg_timeout In-Reply-To: <191c3a030907141020l48e8cb2cm6106ff4442ee1bed@mail.gmail.com> References: <367751820907140835y322c576en1eb2ba931388d732@mail.gmail.com> <87f2f3b90907140916o3f008a3br26bfca075a9e3d2e@mail.gmail.com> <367751820907141002g3a837a09meabdbc9743c3bee7@mail.gmail.com> <191c3a030907141020l48e8cb2cm6106ff4442ee1bed@mail.gmail.com> Message-ID: <87f2f3b90907141037r260001d4i2c41dcc0e4c25c5@mail.gmail.com> FYI, This has been added to the wiki: http://wiki.freeswitch.org/wiki/Channel_Variables#group_confirm_cancel_timeout -MC On Tue, Jul 14, 2009 at 10:20 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > update to trunk and try setting > group_confirm_cancel_timeout=true > > let me know if it works > > > On Tue, Jul 14, 2009 at 12:02 PM, Phillip Jones wrote: > >> Hi, >> >> Thanks for the reply. >> >> >> Both "leg_timeout" and "leg_progress_timeout" are for controlling how >> long to wait prior to the B-leg answering. >> >> I think this is my point. leg_timeout seems to control how long to wait >> prior to the bridge completeing, not the B-leg answering. >> >> In my situation I am using: >> >> Session.Execute("set", "group_confirm_key=exec"); >> Session.Execute("set", "group_confirm_file=javascript confirm.js"); >> >> my confirm.js prompts leg_b for 1 (to take the call). If leg_timeout is >> set to 10 you have 10 seconds to answer the call AND press 1. >> >> I just want call_timeout to be satisfied when the call is answered. Not >> when the called party presses 1 and the bridge is complete. >> >> I am new all this so I will work out how to use the pastebin etc. >> >> Thanks for your help. >> >> >> Phillip Jones >> >> >> >> On Tue, Jul 14, 2009 at 12:16 PM, Michael Collins wrote: >> >>> >>> >>> On Tue, Jul 14, 2009 at 8:35 AM, Phillip Jones wrote: >>> >>>> Hi there, >>>> >>>> Here is my call flow: >>>> >>>> 1) leg A is bridged to leg B >>>> 2) when leg B is answered I play a confirm script - "please 1 to accept >>>> this call" >>>> >>>> I only want leg B to ring 20 seconds. BUT when the caller party answers, >>>> he should have as long as he needs to press 1. >>>> >>>> "leg_timeout" seems to be in play until the bridge is completed. I need >>>> it to reset when leg b is answered. >>>> >>> >>> Correct. Both "leg_timeout" and "leg_progress_timeout" are for >>> controlling how long to wait prior to the B-leg answering. >>> (leg_progress_timeout specifies how long to wait for any kind of progress, >>> be it early media of some sort, ringing, or an answer.) >>> >>>> >>>> I tried resetting the leg_timeout in the confirm script after leg b is >>>> answered. I also tried using leg_progress_timeout. Neither seemed to work. >>>> >>> >>> What exactly are you trying to do? The two variables you've mentioned >>> shouldn't have any effect on the call after it has been established. >>> >>>> >>>> Any help or suggestions would be welcome. >>>> >>> >>> Could you pastebin your dialplan and a debug log of a call that does not >>> work? See this page for some handy tips on using pastebin and collecting >>> information for debugging purposes: >>> http://wiki.freeswitch.org/wiki/Reporting_Bugs#Debugging_Steps >>> >>> -MC >>> >>>> >>>> Phillip Jones >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > 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/20090714/e06c16b6/attachment.html From larclap at yahoo.com Tue Jul 14 10:46:58 2009 From: larclap at yahoo.com (Lars Zeb) Date: Tue, 14 Jul 2009 10:46:58 -0700 Subject: [Freeswitch-users] Intercom error with SNOM In-Reply-To: <87f2f3b90907141009j28b6da38g4e1403a131fdcf6d@mail.gmail.com> References: <00a001ca049b$2e9e03b0$8bda0b10$@com> <87f2f3b90907141009j28b6da38g4e1403a131fdcf6d@mail.gmail.com> Message-ID: <00e301ca04ab$16337430$429a5c90$@com> Michael, I made the changes you suggested, but the result is the same. If it matters, I am at 14229. I stopped and restarted FreeSWITCH and then reran the intercom call. Lars 2009-07-14 10:41:40.553145 [DEBUG] sofia.c:4587 IP 192.168.10.11 Rejected by acl "domains". Falling back to Digest auth. 2009-07-14 10:41:40.562147 [DEBUG] sofia.c:4587 IP 192.168.10.11 Rejected by acl "domains". Falling back to Digest auth. 2009-07-14 10:41:40.565166 [NOTICE] switch_channel.c:602 New Channel sofia/internal/1009 at 192.168.10.29 [fac3f284-3710-4492-a9e7-667c7e816ea0] From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Tuesday, July 14, 2009 10:10 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Intercom error with SNOM On Tue, Jul 14, 2009 at 8:53 AM, Lars Zeb wrote: I am getting an error when I try to make an intercom call from a softphone to a SNOM 320. I get a "401 Unauthorized" in the siptrace and a "No Matching gateway found" in the log. I can successfully make an intercom call between my softphone and a Polycom 501, so it must be something with the SNOM. bkw suggested that the problem was in the challenge/response between FreeSWITCH and SNOM. In the SNOM's Setup/Advanced/Behavior page I have set "Challenge/Response" off, "Enable intercom" on and "Type of Intercom Answering" to "Handsfree". The error is still "401 Unauthorized". What more do I need to do? Lars, It looks like FreeSWITCH is sending the challenge to the Snom. Note this line from the debug log: 2009-07-14 08:12:32.333079 [DEBUG] sofia.c:4587 IP 192.168.10.11 Rejected by acl "domains". Falling back to Digest auth. For testing purposes you can edit acl.conf.xml and add a new line right after: Add: Restart FS or issue this command at the CLI: reloadacl reloadxml Then try your call again. To learn more about how you can have your local users bypass the "domains" acl without editing acl.conf.xml then look in conf/directory/default/brian.xml. At the top of that file you will see a note about how adding a cidr= attribute to your user tag will let you bypass the domains ACL check. Enjoy! -MC Thanks, Lars http://pastebin.freeswitch.org/9709 _______________________________________________ Freeswitch-users mailing list Freeswitch-users at 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/20090714/0864b1ae/attachment-0001.html From msc at freeswitch.org Tue Jul 14 10:58:09 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 14 Jul 2009 10:58:09 -0700 Subject: [Freeswitch-users] Intercom error with SNOM In-Reply-To: <00e301ca04ab$16337430$429a5c90$@com> References: <00a001ca049b$2e9e03b0$8bda0b10$@com> <87f2f3b90907141009j28b6da38g4e1403a131fdcf6d@mail.gmail.com> <00e301ca04ab$16337430$429a5c90$@com> Message-ID: <87f2f3b90907141058v4cf6d8bds62873fb0bb61553c@mail.gmail.com> Lars, Brian pointed out that the challenge is coming from the phone. Is 192.168.10.104 the Snom? -MC On Tue, Jul 14, 2009 at 10:46 AM, Lars Zeb wrote: > Michael, > > > > I made the changes you suggested, but the result is the same. If it > matters, I am at 14229. I stopped and restarted FreeSWITCH and then reran > the intercom call. > > > > Lars > > > > > > > > > > > > > > > > > > > > > > 2009-07-14 10:41:40.553145 [DEBUG] sofia.c:4587 IP 192.168.10.11 Rejected > by acl "domains". Falling back to Digest auth. > > 2009-07-14 10:41:40.562147 [DEBUG] sofia.c:4587 IP 192.168.10.11 Rejected > by acl "domains". Falling back to Digest auth. > > 2009-07-14 10:41:40.565166 [NOTICE] switch_channel.c:602 New Channel > sofia/internal/1009 at 192.168.10.29 [fac3f284-3710-4492-a9e7-667c7e816ea0] > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael > Collins > *Sent:* Tuesday, July 14, 2009 10:10 AM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Intercom error with SNOM > > > > > > On Tue, Jul 14, 2009 at 8:53 AM, Lars Zeb wrote: > > I am getting an error when I try to make an intercom call from a softphone > to a SNOM 320. I get a ?401 Unauthorized? in the siptrace and a ?No Matching > gateway found? in the log. I can successfully make an intercom call between > my softphone and a Polycom 501, so it must be something with the SNOM. > > > > bkw suggested that the problem was in the challenge/response between > FreeSWITCH and SNOM. In the SNOM?s Setup/Advanced/Behavior page I have set > ?Challenge/Response? off, ?Enable intercom? on and ?Type of Intercom > Answering? to ?Handsfree?. The error is still ?401 Unauthorized?. > > > > What more do I need to do? > > > Lars, > It looks like FreeSWITCH is sending the challenge to the Snom. Note this > line from the debug log: > 2009-07-14 08:12:32.333079 [DEBUG] sofia.c:4587 IP 192.168.10.11 Rejected > by acl "domains". Falling back to Digest auth. > > For testing purposes you can edit acl.conf.xml and add a new line right > after: > > > Add: > > > Restart FS or issue this command at the CLI: > reloadacl reloadxml > > Then try your call again. > > To learn more about how you can have your local users bypass the "domains" > acl without editing acl.conf.xml then look in > conf/directory/default/brian.xml. At the top of that file you will see a > note about how adding a cidr= attribute to your user tag will let you bypass > the domains ACL check. > > Enjoy! > -MC > > > > Thanks, Lars > > > > http://pastebin.freeswitch.org/9709 > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090714/d1049faf/attachment.html From peder at networkoblivion.com Tue Jul 14 11:05:37 2009 From: peder at networkoblivion.com (Peder) Date: Tue, 14 Jul 2009 13:05:37 -0500 Subject: [Freeswitch-users] Intercom error with SNOM In-Reply-To: <00e301ca04ab$16337430$429a5c90$@com> References: <00a001ca049b$2e9e03b0$8bda0b10$@com> <87f2f3b90907141009j28b6da38g4e1403a131fdcf6d@mail.gmail.com> <00e301ca04ab$16337430$429a5c90$@com> Message-ID: <06b101ca04ad$b0fc2b90$12f482b0$@com> I haven't followed the whole thread, but the acl listed is named "lan" and the rejected acl is named "domains". From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Lars Zeb Sent: Tuesday, July 14, 2009 12:47 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Intercom error with SNOM Michael, I made the changes you suggested, but the result is the same. If it matters, I am at 14229. I stopped and restarted FreeSWITCH and then reran the intercom call. Lars 2009-07-14 10:41:40.553145 [DEBUG] sofia.c:4587 IP 192.168.10.11 Rejected by acl "domains". Falling back to Digest auth. 2009-07-14 10:41:40.562147 [DEBUG] sofia.c:4587 IP 192.168.10.11 Rejected by acl "domains". Falling back to Digest auth. 2009-07-14 10:41:40.565166 [NOTICE] switch_channel.c:602 New Channel sofia/internal/1009 at 192.168.10.29 [fac3f284-3710-4492-a9e7-667c7e816ea0] From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Tuesday, July 14, 2009 10:10 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Intercom error with SNOM On Tue, Jul 14, 2009 at 8:53 AM, Lars Zeb wrote: I am getting an error when I try to make an intercom call from a softphone to a SNOM 320. I get a "401 Unauthorized" in the siptrace and a "No Matching gateway found" in the log. I can successfully make an intercom call between my softphone and a Polycom 501, so it must be something with the SNOM. bkw suggested that the problem was in the challenge/response between FreeSWITCH and SNOM. In the SNOM's Setup/Advanced/Behavior page I have set "Challenge/Response" off, "Enable intercom" on and "Type of Intercom Answering" to "Handsfree". The error is still "401 Unauthorized". What more do I need to do? Lars, It looks like FreeSWITCH is sending the challenge to the Snom. Note this line from the debug log: 2009-07-14 08:12:32.333079 [DEBUG] sofia.c:4587 IP 192.168.10.11 Rejected by acl "domains". Falling back to Digest auth. For testing purposes you can edit acl.conf.xml and add a new line right after: Add: Restart FS or issue this command at the CLI: reloadacl reloadxml Then try your call again. To learn more about how you can have your local users bypass the "domains" acl without editing acl.conf.xml then look in conf/directory/default/brian.xml. At the top of that file you will see a note about how adding a cidr= attribute to your user tag will let you bypass the domains ACL check. Enjoy! -MC Thanks, Lars http://pastebin.freeswitch.org/9709 _______________________________________________ Freeswitch-users mailing list Freeswitch-users at 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/20090714/3fd8f8e2/attachment-0001.html From lon at kickasspixels.com Tue Jul 14 11:10:54 2009 From: lon at kickasspixels.com (Lon Baker) Date: Tue, 14 Jul 2009 11:10:54 -0700 Subject: [Freeswitch-users] Even socket packets/chunks Message-ID: <5d3e0dc60907141110u4f4084ecyb8fdee94c5983a2d@mail.gmail.com> Hi, You confirm if FS ever sends partial or incomplete commands over the event socket? I have heard from other developers that the commands I am listening for may come across incomplete. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090714/cdc45973/attachment.html From pjintheusa at gmail.com Tue Jul 14 11:14:19 2009 From: pjintheusa at gmail.com (Phillip Jones) Date: Tue, 14 Jul 2009 14:14:19 -0400 Subject: [Freeswitch-users] leg_timeout In-Reply-To: <87f2f3b90907141037r260001d4i2c41dcc0e4c25c5@mail.gmail.com> References: <367751820907140835y322c576en1eb2ba931388d732@mail.gmail.com> <87f2f3b90907140916o3f008a3br26bfca075a9e3d2e@mail.gmail.com> <367751820907141002g3a837a09meabdbc9743c3bee7@mail.gmail.com> <191c3a030907141020l48e8cb2cm6106ff4442ee1bed@mail.gmail.com> <87f2f3b90907141037r260001d4i2c41dcc0e4c25c5@mail.gmail.com> Message-ID: <367751820907141114w56ae9adey4836238740a1ed43@mail.gmail.com> Thanks for your response. That does not seem to work. Here is my code: if(Session.Ready()) { Session.Execute("set", "ignore_early_media=true"); Session.Execute("set", "hangup_after_bridge=true"); Session.Execute("set", "ringback=${us-ring}"); Session.Answer(); string Caller_ID_Number = this.Session.GetVariable("caller_id_number"); Session.Execute("set", "group_confirm_key=exec"); *Session.Execute("set", "group_confirm_cancel_timeout=true"); * Session.Execute("set", "group_confirm_file=javascript confirm.js"); Session.Execute("bridge", "[leg_timeout=20,leg_confirm=y]sofia/gateway/broadvox/6095553828"); } Session.Hangup("USER_BUSY"); I also tried *group_confirm_cancel_leg_timeout* just in case. Am I missing something? On Tue, Jul 14, 2009 at 1:37 PM, Michael Collins wrote: > FYI, > This has been added to the wiki: > > http://wiki.freeswitch.org/wiki/Channel_Variables#group_confirm_cancel_timeout > > -MC > > > On Tue, Jul 14, 2009 at 10:20 AM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> update to trunk and try setting >> group_confirm_cancel_timeout=true >> >> let me know if it works >> >> >> On Tue, Jul 14, 2009 at 12:02 PM, Phillip Jones wrote: >> >>> Hi, >>> >>> Thanks for the reply. >>> >>> >> Both "leg_timeout" and "leg_progress_timeout" are for controlling how >>> long to wait prior to the B-leg answering. >>> >>> I think this is my point. leg_timeout seems to control how long to wait >>> prior to the bridge completeing, not the B-leg answering. >>> >>> In my situation I am using: >>> >>> Session.Execute("set", "group_confirm_key=exec"); >>> Session.Execute("set", "group_confirm_file=javascript confirm.js"); >>> >>> my confirm.js prompts leg_b for 1 (to take the call). If leg_timeout is >>> set to 10 you have 10 seconds to answer the call AND press 1. >>> >>> I just want call_timeout to be satisfied when the call is answered. Not >>> when the called party presses 1 and the bridge is complete. >>> >>> I am new all this so I will work out how to use the pastebin etc. >>> >>> Thanks for your help. >>> >>> >>> Phillip Jones >>> >>> >>> >>> On Tue, Jul 14, 2009 at 12:16 PM, Michael Collins wrote: >>> >>>> >>>> >>>> On Tue, Jul 14, 2009 at 8:35 AM, Phillip Jones wrote: >>>> >>>>> Hi there, >>>>> >>>>> Here is my call flow: >>>>> >>>>> 1) leg A is bridged to leg B >>>>> 2) when leg B is answered I play a confirm script - "please 1 to accept >>>>> this call" >>>>> >>>>> I only want leg B to ring 20 seconds. BUT when the caller party >>>>> answers, he should have as long as he needs to press 1. >>>>> >>>>> "leg_timeout" seems to be in play until the bridge is completed. I need >>>>> it to reset when leg b is answered. >>>>> >>>> >>>> Correct. Both "leg_timeout" and "leg_progress_timeout" are for >>>> controlling how long to wait prior to the B-leg answering. >>>> (leg_progress_timeout specifies how long to wait for any kind of progress, >>>> be it early media of some sort, ringing, or an answer.) >>>> >>>>> >>>>> I tried resetting the leg_timeout in the confirm script after leg b is >>>>> answered. I also tried using leg_progress_timeout. Neither seemed to work. >>>>> >>>> >>>> What exactly are you trying to do? The two variables you've mentioned >>>> shouldn't have any effect on the call after it has been established. >>>> >>>>> >>>>> Any help or suggestions would be welcome. >>>>> >>>> >>>> Could you pastebin your dialplan and a debug log of a call that does not >>>> work? See this page for some handy tips on using pastebin and collecting >>>> information for debugging purposes: >>>> http://wiki.freeswitch.org/wiki/Reporting_Bugs#Debugging_Steps >>>> >>>> -MC >>>> >>>>> >>>>> Phillip Jones >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> 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/20090714/8dab1cb2/attachment.html From msc at freeswitch.org Tue Jul 14 11:18:14 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 14 Jul 2009 11:18:14 -0700 Subject: [Freeswitch-users] leg_timeout In-Reply-To: <367751820907141114w56ae9adey4836238740a1ed43@mail.gmail.com> References: <367751820907140835y322c576en1eb2ba931388d732@mail.gmail.com> <87f2f3b90907140916o3f008a3br26bfca075a9e3d2e@mail.gmail.com> <367751820907141002g3a837a09meabdbc9743c3bee7@mail.gmail.com> <191c3a030907141020l48e8cb2cm6106ff4442ee1bed@mail.gmail.com> <87f2f3b90907141037r260001d4i2c41dcc0e4c25c5@mail.gmail.com> <367751820907141114w56ae9adey4836238740a1ed43@mail.gmail.com> Message-ID: <87f2f3b90907141118j55ec9960xaa0c329e499e1cdf@mail.gmail.com> Can you capture a debug output and put it in pastebin? That will help us track it down. -MC On Tue, Jul 14, 2009 at 11:14 AM, Phillip Jones wrote: > Thanks for your response. > > That does not seem to work. Here is my code: > > if(Session.Ready()) > { > Session.Execute("set", "ignore_early_media=true"); > Session.Execute("set", "hangup_after_bridge=true"); > Session.Execute("set", "ringback=${us-ring}"); > > Session.Answer(); > string Caller_ID_Number = this.Session.GetVariable("caller_id_number"); > Session.Execute("set", "group_confirm_key=exec"); > *Session.Execute("set", "group_confirm_cancel_timeout=true"); > * Session.Execute("set", "group_confirm_file=javascript confirm.js"); > Session.Execute("bridge", > "[leg_timeout=20,leg_confirm=y]sofia/gateway/broadvox/6095553828"); > } > Session.Hangup("USER_BUSY"); > > I also tried *group_confirm_cancel_leg_timeout* just in case. > > Am I missing something? > > > > > On Tue, Jul 14, 2009 at 1:37 PM, Michael Collins wrote: > >> FYI, >> This has been added to the wiki: >> >> http://wiki.freeswitch.org/wiki/Channel_Variables#group_confirm_cancel_timeout >> >> -MC >> >> >> On Tue, Jul 14, 2009 at 10:20 AM, Anthony Minessale < >> anthony.minessale at gmail.com> wrote: >> >>> update to trunk and try setting >>> group_confirm_cancel_timeout=true >>> >>> let me know if it works >>> >>> >>> On Tue, Jul 14, 2009 at 12:02 PM, Phillip Jones wrote: >>> >>>> Hi, >>>> >>>> Thanks for the reply. >>>> >>>> >> Both "leg_timeout" and "leg_progress_timeout" are for controlling how >>>> long to wait prior to the B-leg answering. >>>> >>>> I think this is my point. leg_timeout seems to control how long to wait >>>> prior to the bridge completeing, not the B-leg answering. >>>> >>>> In my situation I am using: >>>> >>>> Session.Execute("set", "group_confirm_key=exec"); >>>> Session.Execute("set", "group_confirm_file=javascript confirm.js"); >>>> >>>> my confirm.js prompts leg_b for 1 (to take the call). If leg_timeout is >>>> set to 10 you have 10 seconds to answer the call AND press 1. >>>> >>>> I just want call_timeout to be satisfied when the call is answered. Not >>>> when the called party presses 1 and the bridge is complete. >>>> >>>> I am new all this so I will work out how to use the pastebin etc. >>>> >>>> Thanks for your help. >>>> >>>> >>>> Phillip Jones >>>> >>>> >>>> >>>> On Tue, Jul 14, 2009 at 12:16 PM, Michael Collins wrote: >>>> >>>>> >>>>> >>>>> On Tue, Jul 14, 2009 at 8:35 AM, Phillip Jones wrote: >>>>> >>>>>> Hi there, >>>>>> >>>>>> Here is my call flow: >>>>>> >>>>>> 1) leg A is bridged to leg B >>>>>> 2) when leg B is answered I play a confirm script - "please 1 to >>>>>> accept this call" >>>>>> >>>>>> I only want leg B to ring 20 seconds. BUT when the caller party >>>>>> answers, he should have as long as he needs to press 1. >>>>>> >>>>>> "leg_timeout" seems to be in play until the bridge is completed. I >>>>>> need it to reset when leg b is answered. >>>>>> >>>>> >>>>> Correct. Both "leg_timeout" and "leg_progress_timeout" are for >>>>> controlling how long to wait prior to the B-leg answering. >>>>> (leg_progress_timeout specifies how long to wait for any kind of progress, >>>>> be it early media of some sort, ringing, or an answer.) >>>>> >>>>>> >>>>>> I tried resetting the leg_timeout in the confirm script after leg b is >>>>>> answered. I also tried using leg_progress_timeout. Neither seemed to work. >>>>>> >>>>> >>>>> What exactly are you trying to do? The two variables you've mentioned >>>>> shouldn't have any effect on the call after it has been established. >>>>> >>>>>> >>>>>> Any help or suggestions would be welcome. >>>>>> >>>>> >>>>> Could you pastebin your dialplan and a debug log of a call that does >>>>> not work? See this page for some handy tips on using pastebin and collecting >>>>> information for debugging purposes: >>>>> http://wiki.freeswitch.org/wiki/Reporting_Bugs#Debugging_Steps >>>>> >>>>> -MC >>>>> >>>>>> >>>>>> Phillip Jones >>>>>> >>>>>> _______________________________________________ >>>>>> Freeswitch-users mailing list >>>>>> Freeswitch-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> Anthony Minessale II >>> >>> FreeSWITCH http://www.freeswitch.org/ >>> ClueCon http://www.cluecon.com/ >>> Twitter: http://twitter.com/FreeSWITCH_wire >>> >>> AIM: anthm >>> MSN:anthony_minessale at hotmail.com >>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>> IRC: irc.freenode.net #freeswitch >>> >>> FreeSWITCH Developer Conference >>> 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090714/01847f87/attachment-0001.html From anthony.minessale at gmail.com Tue Jul 14 11:27:20 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 14 Jul 2009 13:27:20 -0500 Subject: [Freeswitch-users] leg_timeout In-Reply-To: <367751820907141114w56ae9adey4836238740a1ed43@mail.gmail.com> References: <367751820907140835y322c576en1eb2ba931388d732@mail.gmail.com> <87f2f3b90907140916o3f008a3br26bfca075a9e3d2e@mail.gmail.com> <367751820907141002g3a837a09meabdbc9743c3bee7@mail.gmail.com> <191c3a030907141020l48e8cb2cm6106ff4442ee1bed@mail.gmail.com> <87f2f3b90907141037r260001d4i2c41dcc0e4c25c5@mail.gmail.com> <367751820907141114w56ae9adey4836238740a1ed43@mail.gmail.com> Message-ID: <191c3a030907141127x255f21c5q7d80cf26ea8451a9@mail.gmail.com> You actually updated your code and recompiled it all too? This param was added about 30 seconds before I sent you the email. On Tue, Jul 14, 2009 at 1:14 PM, Phillip Jones wrote: > Thanks for your response. > > That does not seem to work. Here is my code: > > if(Session.Ready()) > { > Session.Execute("set", "ignore_early_media=true"); > Session.Execute("set", "hangup_after_bridge=true"); > Session.Execute("set", "ringback=${us-ring}"); > > Session.Answer(); > string Caller_ID_Number = this.Session.GetVariable("caller_id_number"); > Session.Execute("set", "group_confirm_key=exec"); > *Session.Execute("set", "group_confirm_cancel_timeout=true"); > * Session.Execute("set", "group_confirm_file=javascript confirm.js"); > Session.Execute("bridge", > "[leg_timeout=20,leg_confirm=y]sofia/gateway/broadvox/6095553828"); > } > Session.Hangup("USER_BUSY"); > > I also tried *group_confirm_cancel_leg_timeout* just in case. > > Am I missing something? > > > > > On Tue, Jul 14, 2009 at 1:37 PM, Michael Collins wrote: > >> FYI, >> This has been added to the wiki: >> >> http://wiki.freeswitch.org/wiki/Channel_Variables#group_confirm_cancel_timeout >> >> -MC >> >> >> On Tue, Jul 14, 2009 at 10:20 AM, Anthony Minessale < >> anthony.minessale at gmail.com> wrote: >> >>> update to trunk and try setting >>> group_confirm_cancel_timeout=true >>> >>> let me know if it works >>> >>> >>> On Tue, Jul 14, 2009 at 12:02 PM, Phillip Jones wrote: >>> >>>> Hi, >>>> >>>> Thanks for the reply. >>>> >>>> >> Both "leg_timeout" and "leg_progress_timeout" are for controlling how >>>> long to wait prior to the B-leg answering. >>>> >>>> I think this is my point. leg_timeout seems to control how long to wait >>>> prior to the bridge completeing, not the B-leg answering. >>>> >>>> In my situation I am using: >>>> >>>> Session.Execute("set", "group_confirm_key=exec"); >>>> Session.Execute("set", "group_confirm_file=javascript confirm.js"); >>>> >>>> my confirm.js prompts leg_b for 1 (to take the call). If leg_timeout is >>>> set to 10 you have 10 seconds to answer the call AND press 1. >>>> >>>> I just want call_timeout to be satisfied when the call is answered. Not >>>> when the called party presses 1 and the bridge is complete. >>>> >>>> I am new all this so I will work out how to use the pastebin etc. >>>> >>>> Thanks for your help. >>>> >>>> >>>> Phillip Jones >>>> >>>> >>>> >>>> On Tue, Jul 14, 2009 at 12:16 PM, Michael Collins wrote: >>>> >>>>> >>>>> >>>>> On Tue, Jul 14, 2009 at 8:35 AM, Phillip Jones wrote: >>>>> >>>>>> Hi there, >>>>>> >>>>>> Here is my call flow: >>>>>> >>>>>> 1) leg A is bridged to leg B >>>>>> 2) when leg B is answered I play a confirm script - "please 1 to >>>>>> accept this call" >>>>>> >>>>>> I only want leg B to ring 20 seconds. BUT when the caller party >>>>>> answers, he should have as long as he needs to press 1. >>>>>> >>>>>> "leg_timeout" seems to be in play until the bridge is completed. I >>>>>> need it to reset when leg b is answered. >>>>>> >>>>> >>>>> Correct. Both "leg_timeout" and "leg_progress_timeout" are for >>>>> controlling how long to wait prior to the B-leg answering. >>>>> (leg_progress_timeout specifies how long to wait for any kind of progress, >>>>> be it early media of some sort, ringing, or an answer.) >>>>> >>>>>> >>>>>> I tried resetting the leg_timeout in the confirm script after leg b is >>>>>> answered. I also tried using leg_progress_timeout. Neither seemed to work. >>>>>> >>>>> >>>>> What exactly are you trying to do? The two variables you've mentioned >>>>> shouldn't have any effect on the call after it has been established. >>>>> >>>>>> >>>>>> Any help or suggestions would be welcome. >>>>>> >>>>> >>>>> Could you pastebin your dialplan and a debug log of a call that does >>>>> not work? See this page for some handy tips on using pastebin and collecting >>>>> information for debugging purposes: >>>>> http://wiki.freeswitch.org/wiki/Reporting_Bugs#Debugging_Steps >>>>> >>>>> -MC >>>>> >>>>>> >>>>>> Phillip Jones >>>>>> >>>>>> _______________________________________________ >>>>>> Freeswitch-users mailing list >>>>>> Freeswitch-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> Anthony Minessale II >>> >>> FreeSWITCH http://www.freeswitch.org/ >>> ClueCon http://www.cluecon.com/ >>> Twitter: http://twitter.com/FreeSWITCH_wire >>> >>> AIM: anthm >>> MSN:anthony_minessale at hotmail.com >>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>> IRC: irc.freenode.net #freeswitch >>> >>> FreeSWITCH Developer Conference >>> 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 > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference 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/20090714/88ef3ad1/attachment.html From dave at 3c.co.uk Tue Jul 14 11:33:56 2009 From: dave at 3c.co.uk (David Knell) Date: Tue, 14 Jul 2009 15:33:56 -0300 Subject: [Freeswitch-users] Even socket packets/chunks In-Reply-To: <5d3e0dc60907141110u4f4084ecyb8fdee94c5983a2d@mail.gmail.com> References: <5d3e0dc60907141110u4f4084ecyb8fdee94c5983a2d@mail.gmail.com> Message-ID: <1247596436.4254.6.camel@dk-d820> Hi - You can't assume that 1 packet=1 command/event - it's true often enough to lull you in to a false sense of security, but false often enough that you'll end up with odd problems unless you do things properly. In any case, it's not hard to get it right - there's plenty of other instances where applications have to read from a socket until they hit \n\n, then possibly read content-length bytes: HTTP for a starter. --Dave > Hi, > > > You confirm if FS ever sends partial or incomplete commands over the > event socket? I have heard from other developers that the commands I > am listening for may come across incomplete. > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- David Knell, Director, 3C Limited T: +44 20 3298 2000 E: dave at 3c.co.uk W: http://www.3c.co.uk From pjintheusa at gmail.com Tue Jul 14 11:35:05 2009 From: pjintheusa at gmail.com (Phillip Jones) Date: Tue, 14 Jul 2009 14:35:05 -0400 Subject: [Freeswitch-users] leg_timeout In-Reply-To: <191c3a030907141127x255f21c5q7d80cf26ea8451a9@mail.gmail.com> References: <367751820907140835y322c576en1eb2ba931388d732@mail.gmail.com> <87f2f3b90907140916o3f008a3br26bfca075a9e3d2e@mail.gmail.com> <367751820907141002g3a837a09meabdbc9743c3bee7@mail.gmail.com> <191c3a030907141020l48e8cb2cm6106ff4442ee1bed@mail.gmail.com> <87f2f3b90907141037r260001d4i2c41dcc0e4c25c5@mail.gmail.com> <367751820907141114w56ae9adey4836238740a1ed43@mail.gmail.com> <191c3a030907141127x255f21c5q7d80cf26ea8451a9@mail.gmail.com> Message-ID: <367751820907141135t2c374338y54226c9721cfa739@mail.gmail.com> Ah - SVN Trunk - thought you meant DID trunk!!! My bad. Sorry - understand now! Will recompile and let you know. On Tue, Jul 14, 2009 at 2:27 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > You actually updated your code and recompiled it all too? > This param was added about 30 seconds before I sent you the email. > > > > On Tue, Jul 14, 2009 at 1:14 PM, Phillip Jones wrote: > >> Thanks for your response. >> >> That does not seem to work. Here is my code: >> >> if(Session.Ready()) >> { >> Session.Execute("set", "ignore_early_media=true"); >> Session.Execute("set", "hangup_after_bridge=true"); >> Session.Execute("set", "ringback=${us-ring}"); >> >> Session.Answer(); >> string Caller_ID_Number = >> this.Session.GetVariable("caller_id_number"); >> Session.Execute("set", "group_confirm_key=exec"); >> *Session.Execute("set", "group_confirm_cancel_timeout=true"); >> * Session.Execute("set", "group_confirm_file=javascript confirm.js"); >> Session.Execute("bridge", >> "[leg_timeout=20,leg_confirm=y]sofia/gateway/broadvox/6095553828"); >> } >> Session.Hangup("USER_BUSY"); >> >> I also tried *group_confirm_cancel_leg_timeout* just in case. >> >> Am I missing something? >> >> >> >> >> On Tue, Jul 14, 2009 at 1:37 PM, Michael Collins wrote: >> >>> FYI, >>> This has been added to the wiki: >>> >>> http://wiki.freeswitch.org/wiki/Channel_Variables#group_confirm_cancel_timeout >>> >>> -MC >>> >>> >>> On Tue, Jul 14, 2009 at 10:20 AM, Anthony Minessale < >>> anthony.minessale at gmail.com> wrote: >>> >>>> update to trunk and try setting >>>> group_confirm_cancel_timeout=true >>>> >>>> let me know if it works >>>> >>>> >>>> On Tue, Jul 14, 2009 at 12:02 PM, Phillip Jones wrote: >>>> >>>>> Hi, >>>>> >>>>> Thanks for the reply. >>>>> >>>>> >> Both "leg_timeout" and "leg_progress_timeout" are for controlling >>>>> how long to wait prior to the B-leg answering. >>>>> >>>>> I think this is my point. leg_timeout seems to control how long to wait >>>>> prior to the bridge completeing, not the B-leg answering. >>>>> >>>>> In my situation I am using: >>>>> >>>>> Session.Execute("set", "group_confirm_key=exec"); >>>>> Session.Execute("set", "group_confirm_file=javascript confirm.js"); >>>>> >>>>> my confirm.js prompts leg_b for 1 (to take the call). If leg_timeout is >>>>> set to 10 you have 10 seconds to answer the call AND press 1. >>>>> >>>>> I just want call_timeout to be satisfied when the call is answered. Not >>>>> when the called party presses 1 and the bridge is complete. >>>>> >>>>> I am new all this so I will work out how to use the pastebin etc. >>>>> >>>>> Thanks for your help. >>>>> >>>>> >>>>> Phillip Jones >>>>> >>>>> >>>>> >>>>> On Tue, Jul 14, 2009 at 12:16 PM, Michael Collins wrote: >>>>> >>>>>> >>>>>> >>>>>> On Tue, Jul 14, 2009 at 8:35 AM, Phillip Jones >>>>> > wrote: >>>>>> >>>>>>> Hi there, >>>>>>> >>>>>>> Here is my call flow: >>>>>>> >>>>>>> 1) leg A is bridged to leg B >>>>>>> 2) when leg B is answered I play a confirm script - "please 1 to >>>>>>> accept this call" >>>>>>> >>>>>>> I only want leg B to ring 20 seconds. BUT when the caller party >>>>>>> answers, he should have as long as he needs to press 1. >>>>>>> >>>>>>> "leg_timeout" seems to be in play until the bridge is completed. I >>>>>>> need it to reset when leg b is answered. >>>>>>> >>>>>> >>>>>> Correct. Both "leg_timeout" and "leg_progress_timeout" are for >>>>>> controlling how long to wait prior to the B-leg answering. >>>>>> (leg_progress_timeout specifies how long to wait for any kind of progress, >>>>>> be it early media of some sort, ringing, or an answer.) >>>>>> >>>>>>> >>>>>>> I tried resetting the leg_timeout in the confirm script after leg b >>>>>>> is answered. I also tried using leg_progress_timeout. Neither seemed to >>>>>>> work. >>>>>>> >>>>>> >>>>>> What exactly are you trying to do? The two variables you've mentioned >>>>>> shouldn't have any effect on the call after it has been established. >>>>>> >>>>>>> >>>>>>> Any help or suggestions would be welcome. >>>>>>> >>>>>> >>>>>> Could you pastebin your dialplan and a debug log of a call that does >>>>>> not work? See this page for some handy tips on using pastebin and collecting >>>>>> information for debugging purposes: >>>>>> http://wiki.freeswitch.org/wiki/Reporting_Bugs#Debugging_Steps >>>>>> >>>>>> -MC >>>>>> >>>>>>> >>>>>>> Phillip Jones >>>>>>> >>>>>>> _______________________________________________ >>>>>>> Freeswitch-users mailing list >>>>>>> Freeswitch-users at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE: >>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>>> >>>>>> >>>>>> _______________________________________________ >>>>>> Freeswitch-users mailing list >>>>>> Freeswitch-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> -- >>>> Anthony Minessale II >>>> >>>> FreeSWITCH http://www.freeswitch.org/ >>>> ClueCon http://www.cluecon.com/ >>>> Twitter: http://twitter.com/FreeSWITCH_wire >>>> >>>> AIM: anthm >>>> MSN:anthony_minessale at hotmail.com >>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>>> IRC: irc.freenode.net #freeswitch >>>> >>>> FreeSWITCH Developer Conference >>>> 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 >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > 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/20090714/38b4aa69/attachment-0001.html From anthony.minessale at gmail.com Tue Jul 14 11:56:29 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 14 Jul 2009 13:56:29 -0500 Subject: [Freeswitch-users] Even socket packets/chunks In-Reply-To: <1247596436.4254.6.camel@dk-d820> References: <5d3e0dc60907141110u4f4084ecyb8fdee94c5983a2d@mail.gmail.com> <1247596436.4254.6.camel@dk-d820> Message-ID: <191c3a030907141156y4614b28fy7feac44fe92b8532@mail.gmail.com> I took the time to write the ESL lib and release it BSD licensed. It does everything you need to talk to event socket from a series of languages from C all the way up... On Tue, Jul 14, 2009 at 1:33 PM, David Knell wrote: > Hi - > > You can't assume that 1 packet=1 command/event - it's true often enough > to lull you in to a false sense of security, but false often enough that > you'll end up with odd problems unless you do things properly. > > In any case, it's not hard to get it right - there's plenty of other > instances where applications have to read from a socket until they hit > \n\n, then possibly read content-length bytes: HTTP for a starter. > > --Dave > > > Hi, > > > > > > You confirm if FS ever sends partial or incomplete commands over the > > event socket? I have heard from other developers that the commands I > > am listening for may come across incomplete. > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > -- > David Knell, Director, 3C Limited > T: +44 20 3298 2000 > E: dave at 3c.co.uk > W: 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/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference 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/20090714/cca46e1f/attachment.html From larclap at yahoo.com Tue Jul 14 12:34:10 2009 From: larclap at yahoo.com (Lars Zeb) Date: Tue, 14 Jul 2009 12:34:10 -0700 Subject: [Freeswitch-users] Intercom error with SNOM In-Reply-To: <87f2f3b90907141058v4cf6d8bds62873fb0bb61553c@mail.gmail.com> References: <00a001ca049b$2e9e03b0$8bda0b10$@com> <87f2f3b90907141009j28b6da38g4e1403a131fdcf6d@mail.gmail.com> <00e301ca04ab$16337430$429a5c90$@com> <87f2f3b90907141058v4cf6d8bds62873fb0bb61553c@mail.gmail.com> Message-ID: <014501ca04ba$0faa07f0$2efe17d0$@com> Yes. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Tuesday, July 14, 2009 10:58 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Intercom error with SNOM Lars, Brian pointed out that the challenge is coming from the phone. Is 192.168.10.104 the Snom? -MC On Tue, Jul 14, 2009 at 10:46 AM, Lars Zeb wrote: Michael, I made the changes you suggested, but the result is the same. If it matters, I am at 14229. I stopped and restarted FreeSWITCH and then reran the intercom call. Lars 2009-07-14 10:41:40.553145 [DEBUG] sofia.c:4587 IP 192.168.10.11 Rejected by acl "domains". Falling back to Digest auth. 2009-07-14 10:41:40.562147 [DEBUG] sofia.c:4587 IP 192.168.10.11 Rejected by acl "domains". Falling back to Digest auth. 2009-07-14 10:41:40.565166 [NOTICE] switch_channel.c:602 New Channel sofia/internal/1009 at 192.168.10.29 [fac3f284-3710-4492-a9e7-667c7e816ea0] From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Tuesday, July 14, 2009 10:10 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Intercom error with SNOM On Tue, Jul 14, 2009 at 8:53 AM, Lars Zeb wrote: I am getting an error when I try to make an intercom call from a softphone to a SNOM 320. I get a "401 Unauthorized" in the siptrace and a "No Matching gateway found" in the log. I can successfully make an intercom call between my softphone and a Polycom 501, so it must be something with the SNOM. bkw suggested that the problem was in the challenge/response between FreeSWITCH and SNOM. In the SNOM's Setup/Advanced/Behavior page I have set "Challenge/Response" off, "Enable intercom" on and "Type of Intercom Answering" to "Handsfree". The error is still "401 Unauthorized". What more do I need to do? Lars, It looks like FreeSWITCH is sending the challenge to the Snom. Note this line from the debug log: 2009-07-14 08:12:32.333079 [DEBUG] sofia.c:4587 IP 192.168.10.11 Rejected by acl "domains". Falling back to Digest auth. For testing purposes you can edit acl.conf.xml and add a new line right after: Add: Restart FS or issue this command at the CLI: reloadacl reloadxml Then try your call again. To learn more about how you can have your local users bypass the "domains" acl without editing acl.conf.xml then look in conf/directory/default/brian.xml. At the top of that file you will see a note about how adding a cidr= attribute to your user tag will let you bypass the domains ACL check. Enjoy! -MC Thanks, Lars http://pastebin.freeswitch.org/9709 _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at 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/20090714/26816e66/attachment.html From andrew at hijacked.us Tue Jul 14 12:40:53 2009 From: andrew at hijacked.us (Andrew Thompson) Date: Tue, 14 Jul 2009 15:40:53 -0400 Subject: [Freeswitch-users] Intercom error with SNOM In-Reply-To: <00a001ca049b$2e9e03b0$8bda0b10$@com> References: <00a001ca049b$2e9e03b0$8bda0b10$@com> Message-ID: <20090714194053.GH28401@hijacked.us> On Tue, Jul 14, 2009 at 08:53:07AM -0700, Lars Zeb wrote: > I am getting an error when I try to make an intercom call from a softphone > to a SNOM 320. I get a "401 Unauthorized" in the siptrace and a "No Matching > gateway found" in the log. I can successfully make an intercom call between > my softphone and a Polycom 501, so it must be something with the SNOM. > > > > bkw suggested that the problem was in the challenge/response between > FreeSWITCH and SNOM. In the SNOM's Setup/Advanced/Behavior page I have set > "Challenge/Response" off, "Enable intercom" on and "Type of Intercom > Answering" to "Handsfree". The error is still "401 Unauthorized". > > > > What more do I need to do? > > Try disabling the intercom option and setting the 'answer after policy' to 'only in idle'. I think there's a bug in the snom firmware, see also: http://forum.snom.com/index.php?showtopic=1790&st=0&gopid=3688&#entry3688 Andrew From brian at freeswitch.org Tue Jul 14 12:41:21 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 14 Jul 2009 14:41:21 -0500 Subject: [Freeswitch-users] Intercom error with SNOM In-Reply-To: <014501ca04ba$0faa07f0$2efe17d0$@com> References: <00a001ca049b$2e9e03b0$8bda0b10$@com> <87f2f3b90907141009j28b6da38g4e1403a131fdcf6d@mail.gmail.com> <00e301ca04ab$16337430$429a5c90$@com> <87f2f3b90907141058v4cf6d8bds62873fb0bb61553c@mail.gmail.com> <014501ca04ba$0faa07f0$2efe17d0$@com> Message-ID: <81196D62-B915-4B03-A8E2-6E189FD7BC32@freeswitch.org> Make sure you 1. reboot. 2. make sure the setting is correct to not auth. 3. what firmware are you on? /b On Jul 14, 2009, at 2:34 PM, Lars Zeb wrote: > Yes. > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Michael Collins > Sent: Tuesday, July 14, 2009 10:58 AM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Intercom error with SNOM > > Lars, > > Brian pointed out that the challenge is coming from the phone. Is > 192.168.10.104 the Snom? > > -MC > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090714/a9a57b27/attachment-0001.html From larclap at yahoo.com Tue Jul 14 12:51:58 2009 From: larclap at yahoo.com (Lars Zeb) Date: Tue, 14 Jul 2009 12:51:58 -0700 Subject: [Freeswitch-users] Intercom error with SNOM In-Reply-To: <81196D62-B915-4B03-A8E2-6E189FD7BC32@freeswitch.org> References: <00a001ca049b$2e9e03b0$8bda0b10$@com> <87f2f3b90907141009j28b6da38g4e1403a131fdcf6d@mail.gmail.com> <00e301ca04ab$16337430$429a5c90$@com> <87f2f3b90907141058v4cf6d8bds62873fb0bb61553c@mail.gmail.com> <014501ca04ba$0faa07f0$2efe17d0$@com> <81196D62-B915-4B03-A8E2-6E189FD7BC32@freeswitch.org> Message-ID: <015e01ca04bc$8cab9be0$a602d3a0$@com> 2. Do you mean setting "Challenge Response on phone" on the SNOM? It is already set to no. 3. 7.3.14 From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Tuesday, July 14, 2009 12:41 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Intercom error with SNOM Make sure you 1. reboot. 2. make sure the setting is correct to not auth. 3. what firmware are you on? /b On Jul 14, 2009, at 2:34 PM, Lars Zeb wrote: Yes. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Tuesday, July 14, 2009 10:58 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Intercom error with SNOM Lars, Brian pointed out that the challenge is coming from the phone. Is 192.168.10.104 the Snom? -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090714/77fe412f/attachment.html From brian at freeswitch.org Tue Jul 14 12:59:15 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 14 Jul 2009 14:59:15 -0500 Subject: [Freeswitch-users] Intercom error with SNOM In-Reply-To: <015e01ca04bc$8cab9be0$a602d3a0$@com> References: <00a001ca049b$2e9e03b0$8bda0b10$@com> <87f2f3b90907141009j28b6da38g4e1403a131fdcf6d@mail.gmail.com> <00e301ca04ab$16337430$429a5c90$@com> <87f2f3b90907141058v4cf6d8bds62873fb0bb61553c@mail.gmail.com> <014501ca04ba$0faa07f0$2efe17d0$@com> <81196D62-B915-4B03-A8E2-6E189FD7BC32@freeswitch.org> <015e01ca04bc$8cab9be0$a602d3a0$@com> Message-ID: Do what Andrew said.. it has to be a bug :P (in the snom) /b On Jul 14, 2009, at 2:51 PM, Lars Zeb wrote: > 2. Do you mean setting ?Challenge Response on phone? on the SNOM? It > is already set to no. > 3. 7.3.14 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090714/db8fc1c7/attachment.html From larclap at yahoo.com Tue Jul 14 13:42:59 2009 From: larclap at yahoo.com (Lars Zeb) Date: Tue, 14 Jul 2009 13:42:59 -0700 Subject: [Freeswitch-users] Intercom error with SNOM In-Reply-To: References: <00a001ca049b$2e9e03b0$8bda0b10$@com> <87f2f3b90907141009j28b6da38g4e1403a131fdcf6d@mail.gmail.com> <00e301ca04ab$16337430$429a5c90$@com> <87f2f3b90907141058v4cf6d8bds62873fb0bb61553c@mail.gmail.com> <014501ca04ba$0faa07f0$2efe17d0$@com> <81196D62-B915-4B03-A8E2-6E189FD7BC32@freeswitch.org> <015e01ca04bc$8cab9be0$a602d3a0$@com> Message-ID: <017501ca04c3$ad21ae80$07650b80$@com> Thanks Michael and Brian and Andrew and Peder. I changed "Enable Intercom" to off and "'Answer After' Policy" to "only in idle" and it works. The next firmware will probably change the settings to mean what they should mean. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Tuesday, July 14, 2009 12:59 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Intercom error with SNOM Do what Andrew said.. it has to be a bug :P (in the snom) /b On Jul 14, 2009, at 2:51 PM, Lars Zeb wrote: 2. Do you mean setting "Challenge Response on phone" on the SNOM? It is already set to no. 3. 7.3.14 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090714/9dcfa426/attachment.html From eweaver at meetingone.com Tue Jul 14 14:14:56 2009 From: eweaver at meetingone.com (Weaver, Eric) Date: Tue, 14 Jul 2009 14:14:56 -0700 Subject: [Freeswitch-users] Preventing disconnect on event_sockte close In-Reply-To: <87f2f3b90907141011g46686396se27fa3cc0320963d@mail.gmail.com> References: <7DB369DA44D5CF42B6C23BD7D611A50FDA7E1D5FD6@VA3DIAXVS061.RED001.local> <87f2f3b90907131808l100afe7dwc95d0f624ff38d50@mail.gmail.com> <7DB369DA44D5CF42B6C23BD7D611A50FDA7E1D604D@VA3DIAXVS061.RED001.local> <87f2f3b90907141011g46686396se27fa3cc0320963d@mail.gmail.com> Message-ID: <7DB369DA44D5CF42B6C23BD7D611A50FDABBFA760F@VA3DIAXVS061.RED001.local> Using api uuid_park 81b65478-70b9-11de-bf26-b962186102f7 before terminating the NC session works, the call is not disconnected. Once that is done, I do not receive DTMF and cannot play prompts to the caller, they seem to be in limbo. I can uuid_kill the call but I need to get dtmf and play prompts to them. Perhaps Park is not where I need to put these calls ? From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Tuesday, July 14, 2009 11:12 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Preventing disconnect on event_sockte close On Tue, Jul 14, 2009 at 9:59 AM, Weaver, Eric > wrote: Tried it using FreeSWITCH Version 1.0.4pre9 (14036M) and the following steps Start netcat netcat -v -l -p 14000 place call, socket is connected via dial plan, enter the following. connect\n\n sendmsg call-command: execute execute-app-name: answer\n\n sendmsg call-command: execute execute-app-name: playback execute-app-arg: /home/eweaver/holdmusic.wav sendmsg call-command: execute execute-app-name: park Try this: api uuid_park You'll need to capture the uuid at some point and store it. For testing I just manually copied and pasted it to/from the console screen. -MC Console window displays this message: 2009-07-14 10:42:29.245833 [ERR] switch_ivr.c:654 Cannot park channels that are under control already. at this point, ^C in the netcat window. Call is disconnected. Need to be able to park these calls so they can then be handled from an inbound event socket connection. eric From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Monday, July 13, 2009 7:08 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Preventing disconnect on event_sockte close I don't know if this will work for you but I just tested this scenario with uuid_park. After parking the call I disconnected the socket and the call continued. I did the same thing with uuid_transfer. After the transfer I disconnected the socket and the call continued. How are you handling the call and how is the socket getting disconnected? -MC On Mon, Jul 13, 2009 at 4:35 PM, Weaver, Eric > wrote: Using mod_event_socket in outbound mode, is there any to prevent a call from being disconnected when the outbound socket is closed ? I would like to handle the initial inbound call using outbound but after the disposition of the call is determined, close the socket and have that call managed using an inbound socket instead. Eric Weaver _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at 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/20090714/43cafd72/attachment-0001.html From eweaver at meetingone.com Tue Jul 14 14:17:18 2009 From: eweaver at meetingone.com (Weaver, Eric) Date: Tue, 14 Jul 2009 14:17:18 -0700 Subject: [Freeswitch-users] Preventing disconnect on event_sockte close In-Reply-To: <87f2f3b90907141011g46686396se27fa3cc0320963d@mail.gmail.com> References: <7DB369DA44D5CF42B6C23BD7D611A50FDA7E1D5FD6@VA3DIAXVS061.RED001.local> <87f2f3b90907131808l100afe7dwc95d0f624ff38d50@mail.gmail.com> <7DB369DA44D5CF42B6C23BD7D611A50FDA7E1D604D@VA3DIAXVS061.RED001.local> <87f2f3b90907141011g46686396se27fa3cc0320963d@mail.gmail.com> Message-ID: <7DB369DA44D5CF42B6C23BD7D611A50FDABBFA7611@VA3DIAXVS061.RED001.local> Tried uuid_displace to play prompt, got this message on console 2009-07-14 15:16:17.189764 [ERR] switch_ivr_async.c:367 Can not displace session. Media not enabled on channel So how can the call be "parked" and still have media ? From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Tuesday, July 14, 2009 11:12 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Preventing disconnect on event_sockte close On Tue, Jul 14, 2009 at 9:59 AM, Weaver, Eric > wrote: Tried it using FreeSWITCH Version 1.0.4pre9 (14036M) and the following steps Start netcat netcat -v -l -p 14000 place call, socket is connected via dial plan, enter the following. connect\n\n sendmsg call-command: execute execute-app-name: answer\n\n sendmsg call-command: execute execute-app-name: playback execute-app-arg: /home/eweaver/holdmusic.wav sendmsg call-command: execute execute-app-name: park Try this: api uuid_park You'll need to capture the uuid at some point and store it. For testing I just manually copied and pasted it to/from the console screen. -MC Console window displays this message: 2009-07-14 10:42:29.245833 [ERR] switch_ivr.c:654 Cannot park channels that are under control already. at this point, ^C in the netcat window. Call is disconnected. Need to be able to park these calls so they can then be handled from an inbound event socket connection. eric From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Monday, July 13, 2009 7:08 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Preventing disconnect on event_sockte close I don't know if this will work for you but I just tested this scenario with uuid_park. After parking the call I disconnected the socket and the call continued. I did the same thing with uuid_transfer. After the transfer I disconnected the socket and the call continued. How are you handling the call and how is the socket getting disconnected? -MC On Mon, Jul 13, 2009 at 4:35 PM, Weaver, Eric > wrote: Using mod_event_socket in outbound mode, is there any to prevent a call from being disconnected when the outbound socket is closed ? I would like to handle the initial inbound call using outbound but after the disposition of the call is determined, close the socket and have that call managed using an inbound socket instead. Eric Weaver _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at 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/20090714/eeeb578c/attachment.html From pjintheusa at gmail.com Tue Jul 14 14:27:51 2009 From: pjintheusa at gmail.com (Phillip Jones) Date: Tue, 14 Jul 2009 17:27:51 -0400 Subject: [Freeswitch-users] Pastebin.freeswitch.org Message-ID: <367751820907141427o636fc28bsf36f8320f3b8783b@mail.gmail.com> Hi there, I am sure I am missing something. Can someone point out where to signup for a username / password to pastebin.freeswitch.org. I am pulling my hair out and feel kinda stupid asking this. Thanks Phillip Jones -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090714/8404b3aa/attachment.html From brian at freeswitch.org Tue Jul 14 14:33:41 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 14 Jul 2009 16:33:41 -0500 Subject: [Freeswitch-users] Pastebin.freeswitch.org In-Reply-To: <367751820907141427o636fc28bsf36f8320f3b8783b@mail.gmail.com> References: <367751820907141427o636fc28bsf36f8320f3b8783b@mail.gmail.com> Message-ID: Phillip, Well did you read the dialog box? :P It tells you the user/pass to use ... it keeps the spammers out! /b On Jul 14, 2009, at 4:27 PM, Phillip Jones wrote: > Hi there, > > I am sure I am missing something. Can someone point out where to > signup for a username / password to pastebin.freeswitch.org. > > I am pulling my hair out and feel kinda stupid asking this. > > Thanks > > > Phillip Jones > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090714/afb1b7b8/attachment.html From msc at freeswitch.org Tue Jul 14 14:40:02 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 14 Jul 2009 14:40:02 -0700 Subject: [Freeswitch-users] Preventing disconnect on event_sockte close In-Reply-To: <7DB369DA44D5CF42B6C23BD7D611A50FDABBFA760F@VA3DIAXVS061.RED001.local> References: <7DB369DA44D5CF42B6C23BD7D611A50FDA7E1D5FD6@VA3DIAXVS061.RED001.local> <87f2f3b90907131808l100afe7dwc95d0f624ff38d50@mail.gmail.com> <7DB369DA44D5CF42B6C23BD7D611A50FDA7E1D604D@VA3DIAXVS061.RED001.local> <87f2f3b90907141011g46686396se27fa3cc0320963d@mail.gmail.com> <7DB369DA44D5CF42B6C23BD7D611A50FDABBFA760F@VA3DIAXVS061.RED001.local> Message-ID: <87f2f3b90907141440l5235ab70h7d48ee961367f5e@mail.gmail.com> On Tue, Jul 14, 2009 at 2:14 PM, Weaver, Eric wrote: > Using api uuid_park 81b65478-70b9-11de-bf26-b962186102f7 before > terminating the NC session works, the call is not disconnected. > > > > Once that is done, I do not receive DTMF and cannot play prompts to the > caller, they seem to be in limbo. I can uuid_kill the call but I need to get > dtmf and play prompts to them. Perhaps Park is not where I need to put > these calls ? > > > To get a call out of park you need to bridge to it or transfer it to another extension. If you have an extension you can just uuid_transfer the parked call's uuid. If you have an existing call's uuid you can use uuid_bridge to bridge the two together. Could you remind me of the application you're building? Just curious what the big picture is. -MC > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael > Collins > *Sent:* Tuesday, July 14, 2009 11:12 AM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Preventing disconnect on event_sockte > close > > > > > > On Tue, Jul 14, 2009 at 9:59 AM, Weaver, Eric > wrote: > > Tried it using FreeSWITCH Version 1.0.4pre9 (14036M) and the following > steps > > > > Start netcat > > > > netcat -v -l -p 14000 > > > > place call, socket is connected via dial plan, enter the following. > > > > connect\n\n > > > > sendmsg > > call-command: execute > > execute-app-name: answer\n\n > > > > > > sendmsg > > call-command: execute > > execute-app-name: playback > > execute-app-arg: /home/eweaver/holdmusic.wav > > > > > > > > > > sendmsg > > call-command: execute > > execute-app-name: park > > > Try this: > api uuid_park > > You'll need to capture the uuid at some point and store it. For testing I > just manually copied and pasted it to/from the console screen. > -MC > > > > Console window displays this message: > > > > 2009-07-14 10:42:29.245833 [ERR] switch_ivr.c:654 Cannot park channels that > are under control already. > > > > at this point, ^C in the netcat window. Call is disconnected. > > > > > > Need to be able to park these calls so they can then be handled from an > inbound event socket connection. > > > > eric > > > > > > > > > > > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael > Collins > *Sent:* Monday, July 13, 2009 7:08 PM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Preventing disconnect on event_sockte > close > > > > I don't know if this will work for you but I just tested this scenario with > uuid_park. After parking the call I disconnected the socket and the call > continued. I did the same thing with uuid_transfer. After the transfer I > disconnected the socket and the call continued. > > How are you handling the call and how is the socket getting disconnected? > > -MC > > On Mon, Jul 13, 2009 at 4:35 PM, Weaver, Eric > wrote: > > Using mod_event_socket in outbound mode, is there any to prevent a call > from being disconnected when the outbound socket is closed ? I would like to > handle the initial inbound call using outbound but after the disposition of > the call is determined, close the socket and have that call managed using an > inbound socket instead. > > > > Eric Weaver > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090714/49dd88d8/attachment-0001.html From msc at freeswitch.org Tue Jul 14 14:43:55 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 14 Jul 2009 14:43:55 -0700 Subject: [Freeswitch-users] Pastebin.freeswitch.org In-Reply-To: <367751820907141427o636fc28bsf36f8320f3b8783b@mail.gmail.com> References: <367751820907141427o636fc28bsf36f8320f3b8783b@mail.gmail.com> Message-ID: <87f2f3b90907141443y3033c364n78f5dd33fd99c318@mail.gmail.com> On Tue, Jul 14, 2009 at 2:27 PM, Phillip Jones wrote: > Hi there, > > I am sure I am missing something. Can someone point out where to signup for > a username / password to pastebin.freeswitch.org. > > I am pulling my hair out and feel kinda stupid asking this. That's okay, you are neither the first nor the last person to get caught by this one. :) -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090714/795662e4/attachment.html From pjintheusa at gmail.com Tue Jul 14 15:01:43 2009 From: pjintheusa at gmail.com (Phillip Jones) Date: Tue, 14 Jul 2009 18:01:43 -0400 Subject: [Freeswitch-users] Pastebin.freeswitch.org In-Reply-To: <87f2f3b90907141443y3033c364n78f5dd33fd99c318@mail.gmail.com> References: <367751820907141427o636fc28bsf36f8320f3b8783b@mail.gmail.com> <87f2f3b90907141443y3033c364n78f5dd33fd99c318@mail.gmail.com> Message-ID: <367751820907141501v3a71fbe2v683393803147da5d@mail.gmail.com> Thank you! Not that obvious in IE actually - I should stick to FF. On Tue, Jul 14, 2009 at 5:43 PM, Michael Collins wrote: > > > On Tue, Jul 14, 2009 at 2:27 PM, Phillip Jones wrote: > >> Hi there, >> >> I am sure I am missing something. Can someone point out where to signup >> for a username / password to pastebin.freeswitch.org. >> >> I am pulling my hair out and feel kinda stupid asking this. > > > That's okay, you are neither the first nor the last person to get caught by > this one. :) > -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/20090714/1decbc49/attachment.html From eweaver at meetingone.com Tue Jul 14 15:03:11 2009 From: eweaver at meetingone.com (Weaver, Eric) Date: Tue, 14 Jul 2009 15:03:11 -0700 Subject: [Freeswitch-users] Preventing disconnect on event_sockte close In-Reply-To: <87f2f3b90907141440l5235ab70h7d48ee961367f5e@mail.gmail.com> References: <7DB369DA44D5CF42B6C23BD7D611A50FDA7E1D5FD6@VA3DIAXVS061.RED001.local> <87f2f3b90907131808l100afe7dwc95d0f624ff38d50@mail.gmail.com> <7DB369DA44D5CF42B6C23BD7D611A50FDA7E1D604D@VA3DIAXVS061.RED001.local> <87f2f3b90907141011g46686396se27fa3cc0320963d@mail.gmail.com> <7DB369DA44D5CF42B6C23BD7D611A50FDABBFA760F@VA3DIAXVS061.RED001.local> <87f2f3b90907141440l5235ab70h7d48ee961367f5e@mail.gmail.com> Message-ID: <7DB369DA44D5CF42B6C23BD7D611A50FDACD801F9A@VA3DIAXVS061.RED001.local> Looking a FS to use as Media mixer for conferencing platform. Not really doing call to call bridging. We really don't have extensions.... Conferences are created o the fly as needed. Already have the conf and call control app done and in production using a different audio mixer, I would like to put FS in place of it. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Tuesday, July 14, 2009 3:40 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Preventing disconnect on event_sockte close On Tue, Jul 14, 2009 at 2:14 PM, Weaver, Eric > wrote: Using api uuid_park 81b65478-70b9-11de-bf26-b962186102f7 before terminating the NC session works, the call is not disconnected. Once that is done, I do not receive DTMF and cannot play prompts to the caller, they seem to be in limbo. I can uuid_kill the call but I need to get dtmf and play prompts to them. Perhaps Park is not where I need to put these calls ? To get a call out of park you need to bridge to it or transfer it to another extension. If you have an extension you can just uuid_transfer the parked call's uuid. If you have an existing call's uuid you can use uuid_bridge to bridge the two together. Could you remind me of the application you're building? Just curious what the big picture is. -MC From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Tuesday, July 14, 2009 11:12 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Preventing disconnect on event_sockte close On Tue, Jul 14, 2009 at 9:59 AM, Weaver, Eric > wrote: Tried it using FreeSWITCH Version 1.0.4pre9 (14036M) and the following steps Start netcat netcat -v -l -p 14000 place call, socket is connected via dial plan, enter the following. connect\n\n sendmsg call-command: execute execute-app-name: answer\n\n sendmsg call-command: execute execute-app-name: playback execute-app-arg: /home/eweaver/holdmusic.wav sendmsg call-command: execute execute-app-name: park Try this: api uuid_park You'll need to capture the uuid at some point and store it. For testing I just manually copied and pasted it to/from the console screen. -MC Console window displays this message: 2009-07-14 10:42:29.245833 [ERR] switch_ivr.c:654 Cannot park channels that are under control already. at this point, ^C in the netcat window. Call is disconnected. Need to be able to park these calls so they can then be handled from an inbound event socket connection. eric From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Monday, July 13, 2009 7:08 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Preventing disconnect on event_sockte close I don't know if this will work for you but I just tested this scenario with uuid_park. After parking the call I disconnected the socket and the call continued. I did the same thing with uuid_transfer. After the transfer I disconnected the socket and the call continued. How are you handling the call and how is the socket getting disconnected? -MC On Mon, Jul 13, 2009 at 4:35 PM, Weaver, Eric > wrote: Using mod_event_socket in outbound mode, is there any to prevent a call from being disconnected when the outbound socket is closed ? I would like to handle the initial inbound call using outbound but after the disposition of the call is determined, close the socket and have that call managed using an inbound socket instead. Eric Weaver _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at 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/20090714/f5f4b6e5/attachment-0001.html From pjintheusa at gmail.com Tue Jul 14 15:22:42 2009 From: pjintheusa at gmail.com (Phillip Jones) Date: Tue, 14 Jul 2009 18:22:42 -0400 Subject: [Freeswitch-users] leg_timeout In-Reply-To: <367751820907141135t2c374338y54226c9721cfa739@mail.gmail.com> References: <367751820907140835y322c576en1eb2ba931388d732@mail.gmail.com> <87f2f3b90907140916o3f008a3br26bfca075a9e3d2e@mail.gmail.com> <367751820907141002g3a837a09meabdbc9743c3bee7@mail.gmail.com> <191c3a030907141020l48e8cb2cm6106ff4442ee1bed@mail.gmail.com> <87f2f3b90907141037r260001d4i2c41dcc0e4c25c5@mail.gmail.com> <367751820907141114w56ae9adey4836238740a1ed43@mail.gmail.com> <191c3a030907141127x255f21c5q7d80cf26ea8451a9@mail.gmail.com> <367751820907141135t2c374338y54226c9721cfa739@mail.gmail.com> Message-ID: <367751820907141522o491b40f4jbf79c25f3c053139@mail.gmail.com> Hi there, I downloaded the latest trunk, compiled and updated. Still no joy I am afraid. This is the log file in pastebin - http://pastebin.freeswitch.org/9712 Code in my managed DLL is at: http://pastebin.freeswitch.org/9715 Dialplan binds to above: confirm.js is at: http://pastebin.freeswitch.org/9713 Thanks again for your help on this. On Tue, Jul 14, 2009 at 2:35 PM, Phillip Jones wrote: > Ah - SVN Trunk - thought you meant DID trunk!!! My bad. > > Sorry - understand now! Will recompile and let you know. > > On Tue, Jul 14, 2009 at 2:27 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> You actually updated your code and recompiled it all too? >> This param was added about 30 seconds before I sent you the email. >> >> >> >> On Tue, Jul 14, 2009 at 1:14 PM, Phillip Jones wrote: >> >>> Thanks for your response. >>> >>> That does not seem to work. Here is my code: >>> >>> if(Session.Ready()) >>> { >>> Session.Execute("set", "ignore_early_media=true"); >>> Session.Execute("set", "hangup_after_bridge=true"); >>> Session.Execute("set", "ringback=${us-ring}"); >>> >>> Session.Answer(); >>> string Caller_ID_Number = >>> this.Session.GetVariable("caller_id_number"); >>> Session.Execute("set", "group_confirm_key=exec"); >>> *Session.Execute("set", "group_confirm_cancel_timeout=true"); >>> * Session.Execute("set", "group_confirm_file=javascript confirm.js"); >>> Session.Execute("bridge", >>> "[leg_timeout=20,leg_confirm=y]sofia/gateway/broadvox/6095553828"); >>> } >>> Session.Hangup("USER_BUSY"); >>> >>> I also tried *group_confirm_cancel_leg_timeout* just in case. >>> >>> Am I missing something? >>> >>> >>> >>> >>> On Tue, Jul 14, 2009 at 1:37 PM, Michael Collins wrote: >>> >>>> FYI, >>>> This has been added to the wiki: >>>> >>>> http://wiki.freeswitch.org/wiki/Channel_Variables#group_confirm_cancel_timeout >>>> >>>> -MC >>>> >>>> >>>> On Tue, Jul 14, 2009 at 10:20 AM, Anthony Minessale < >>>> anthony.minessale at gmail.com> wrote: >>>> >>>>> update to trunk and try setting >>>>> group_confirm_cancel_timeout=true >>>>> >>>>> let me know if it works >>>>> >>>>> >>>>> On Tue, Jul 14, 2009 at 12:02 PM, Phillip Jones wrote: >>>>> >>>>>> Hi, >>>>>> >>>>>> Thanks for the reply. >>>>>> >>>>>> >> Both "leg_timeout" and "leg_progress_timeout" are for controlling >>>>>> how long to wait prior to the B-leg answering. >>>>>> >>>>>> I think this is my point. leg_timeout seems to control how long to >>>>>> wait prior to the bridge completeing, not the B-leg answering. >>>>>> >>>>>> In my situation I am using: >>>>>> >>>>>> Session.Execute("set", "group_confirm_key=exec"); >>>>>> Session.Execute("set", "group_confirm_file=javascript confirm.js"); >>>>>> >>>>>> my confirm.js prompts leg_b for 1 (to take the call). If leg_timeout >>>>>> is set to 10 you have 10 seconds to answer the call AND press 1. >>>>>> >>>>>> I just want call_timeout to be satisfied when the call is answered. >>>>>> Not when the called party presses 1 and the bridge is complete. >>>>>> >>>>>> I am new all this so I will work out how to use the pastebin etc. >>>>>> >>>>>> Thanks for your help. >>>>>> >>>>>> >>>>>> Phillip Jones >>>>>> >>>>>> >>>>>> >>>>>> On Tue, Jul 14, 2009 at 12:16 PM, Michael Collins >>>>> > wrote: >>>>>> >>>>>>> >>>>>>> >>>>>>> On Tue, Jul 14, 2009 at 8:35 AM, Phillip Jones < >>>>>>> pjintheusa at gmail.com> wrote: >>>>>>> >>>>>>>> Hi there, >>>>>>>> >>>>>>>> Here is my call flow: >>>>>>>> >>>>>>>> 1) leg A is bridged to leg B >>>>>>>> 2) when leg B is answered I play a confirm script - "please 1 to >>>>>>>> accept this call" >>>>>>>> >>>>>>>> I only want leg B to ring 20 seconds. BUT when the caller party >>>>>>>> answers, he should have as long as he needs to press 1. >>>>>>>> >>>>>>>> "leg_timeout" seems to be in play until the bridge is completed. I >>>>>>>> need it to reset when leg b is answered. >>>>>>>> >>>>>>> >>>>>>> Correct. Both "leg_timeout" and "leg_progress_timeout" are for >>>>>>> controlling how long to wait prior to the B-leg answering. >>>>>>> (leg_progress_timeout specifies how long to wait for any kind of progress, >>>>>>> be it early media of some sort, ringing, or an answer.) >>>>>>> >>>>>>>> >>>>>>>> I tried resetting the leg_timeout in the confirm script after leg b >>>>>>>> is answered. I also tried using leg_progress_timeout. Neither seemed to >>>>>>>> work. >>>>>>>> >>>>>>> >>>>>>> What exactly are you trying to do? The two variables you've mentioned >>>>>>> shouldn't have any effect on the call after it has been established. >>>>>>> >>>>>>>> >>>>>>>> Any help or suggestions would be welcome. >>>>>>>> >>>>>>> >>>>>>> Could you pastebin your dialplan and a debug log of a call that does >>>>>>> not work? See this page for some handy tips on using pastebin and collecting >>>>>>> information for debugging purposes: >>>>>>> http://wiki.freeswitch.org/wiki/Reporting_Bugs#Debugging_Steps >>>>>>> >>>>>>> -MC >>>>>>> >>>>>>>> >>>>>>>> Phillip Jones >>>>>>>> >>>>>>>> _______________________________________________ >>>>>>>> Freeswitch-users mailing list >>>>>>>> Freeswitch-users at lists.freeswitch.org >>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>> UNSUBSCRIBE: >>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>> http://www.freeswitch.org >>>>>>>> >>>>>>>> >>>>>>> >>>>>>> _______________________________________________ >>>>>>> Freeswitch-users mailing list >>>>>>> Freeswitch-users at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE: >>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>>> >>>>>> >>>>>> _______________________________________________ >>>>>> Freeswitch-users mailing list >>>>>> Freeswitch-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>> >>>>> >>>>> -- >>>>> Anthony Minessale II >>>>> >>>>> FreeSWITCH http://www.freeswitch.org/ >>>>> ClueCon http://www.cluecon.com/ >>>>> Twitter: http://twitter.com/FreeSWITCH_wire >>>>> >>>>> AIM: anthm >>>>> MSN:anthony_minessale at hotmail.com >>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>>>> IRC: irc.freenode.net #freeswitch >>>>> >>>>> FreeSWITCH Developer Conference >>>>> 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 >>> >>> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> 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/20090714/3c69cb83/attachment.html From Kareem.Hamdy at trustvesta.com Tue Jul 14 17:04:05 2009 From: Kareem.Hamdy at trustvesta.com (Kareem Hamdy) Date: Tue, 14 Jul 2009 17:04:05 -0700 Subject: [Freeswitch-users] OpenZAP and FreeSWITCH w/ Sangoma In-Reply-To: References: Message-ID: <1134625859513549B3B943E0133490E202AC614768@TDCP-EXSTORE-01.ad.trustvesta.com> Hello: I would like to connect a Sangoma T1 card to FreeSWITCH. Most of the docs I see pertain to a PRI. When I leave out the d-chan notation, I get errors regarding not able to get the d-chan up and running in the CLI. Here's my info: [span wanpipe T1] trunk_type => t1 b-channel => 1:1-24 [span wanpipe T2] trunk_type => t1 b-channel => 2:1-24 ---- #================================================ # WANPIPE1 Configuration File #================================================ # # Date: Wed Dec 6 20:29:03 UTC 2006 # # Note: This file was generated automatically # by /usr/local/sbin/setup-sangoma program. # # If you want to edit this file, it is # recommended that you use wancfg program # to do so. #================================================ # Sangoma Technologies Inc. #================================================ [devices] wanpipe1 = WAN_AFT_TE1, Comment [interfaces] w1g1 = wanpipe1, , TDM_VOICE_API, Comment [wanpipe1] CARD_TYPE = AFT S514CPU = A CommPort = PRI AUTO_PCISLOT = NO PCISLOT = 1 PCIBUS = 6 FE_MEDIA = T1 FE_LCODE = B8ZS FE_FRAME = ESF FE_LINE = 1 TE_CLOCK = NORMAL TE_REF_CLOCK = 0 TE_HIGHIMPEDANCE = NO LBO = 0DB FE_TXTRISTATE = NO MTU = 1500 UDPPORT = 9000 TTL = 255 IGNORE_FRONT_END = NO TDMV_SPAN = 1 TDMV_DCHAN = 0 TDMV_HW_DTMF = YES TDMV_HW_FAX_DETECT = NO [w1g1] ACTIVE_CH = ALL TDMV_HWEC = YES MTU = 80 --- In freeSWITCH's openzap.conf.xml, I've tried pri_span as well as analog. I cannot find a straight up T1 wiki anywhere. Would someone please provide an example? Thanks, Kareem From msc at freeswitch.org Tue Jul 14 17:24:18 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 14 Jul 2009 17:24:18 -0700 Subject: [Freeswitch-users] OpenZAP and FreeSWITCH w/ Sangoma In-Reply-To: <1134625859513549B3B943E0133490E202AC614768@TDCP-EXSTORE-01.ad.trustvesta.com> References: <1134625859513549B3B943E0133490E202AC614768@TDCP-EXSTORE-01.ad.trustvesta.com> Message-ID: <87f2f3b90907141724q2735fac1jdacea3994db62782@mail.gmail.com> See inline comments On Tue, Jul 14, 2009 at 5:04 PM, Kareem Hamdy wrote: > Hello: > > I would like to connect a Sangoma T1 card to FreeSWITCH. Most of the docs > I see pertain to a PRI. When I leave out the d-chan notation, I get errors > regarding not able to get the d-chan up and running in the CLI. > > Here's my info: > > [span wanpipe T1] > trunk_type => t1 > b-channel => 1:1-24 b-channel => 1:1-23 d-channel => 1:24 > > > [span wanpipe T2] > trunk_type => t1 > b-channel => 2:1-24 > set up like span 1 example > > ---- > > #================================================ > # WANPIPE1 Configuration File > #================================================ > # > # Date: Wed Dec 6 20:29:03 UTC 2006 > # > # Note: This file was generated automatically > # by /usr/local/sbin/setup-sangoma program. > # > # If you want to edit this file, it is > # recommended that you use wancfg program > # to do so. > #================================================ > # Sangoma Technologies Inc. > #================================================ > > [devices] > wanpipe1 = WAN_AFT_TE1, Comment > > [interfaces] > w1g1 = wanpipe1, , TDM_VOICE_API, Comment > > [wanpipe1] > CARD_TYPE = AFT > S514CPU = A > CommPort = PRI > AUTO_PCISLOT = NO > PCISLOT = 1 > PCIBUS = 6 > FE_MEDIA = T1 > FE_LCODE = B8ZS > FE_FRAME = ESF > FE_LINE = 1 > TE_CLOCK = NORMAL > TE_REF_CLOCK = 0 > > TE_HIGHIMPEDANCE = NO > LBO = 0DB > FE_TXTRISTATE = NO > MTU = 1500 > UDPPORT = 9000 > TTL = 255 > IGNORE_FRONT_END = NO > TDMV_SPAN = 1 > TDMV_DCHAN = 0 TDMV_DCAHN = 24 > > TDMV_HW_DTMF = YES > TDMV_HW_FAX_DETECT = NO > > [w1g1] > ACTIVE_CH = ALL > TDMV_HWEC = YES > MTU = 80 > > > --- > > > > In freeSWITCH's openzap.conf.xml, I've tried pri_span as well as analog. > > I cannot find a straight up T1 wiki anywhere. Would someone please provide > an example? > > > Thanks, > Kareem > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090714/da16d89e/attachment.html From digilord at me.com Tue Jul 14 19:20:19 2009 From: digilord at me.com (DigiLord) Date: Tue, 14 Jul 2009 19:20:19 -0700 Subject: [Freeswitch-users] GXW4104 & FreeSwitch Message-ID: <8DC39E34-A395-42D5-B299-070605A2DCEE@me.com> Hello all, I am getting my feet wet with FreeSwitch by migrating my Asterisk box over. I have run into a few things that I am not sure how to accomplish. I have a Grandstream GXW4104 with one analog line connected. I have it connected and I am able to receive calls on my Polycom 501 (ext 2101) that is registered to the FreeSwitch server. The one problem is that CallerID is not the CallerID from the caller, it's the CallerID from the Grandstream device (ext 2100). On the same device there is HORRIBLE echo. I have set echo cancellation on the device to enabled and disabled to no avail. Under Asterisk there was no echo. I setup the device as a provider. Was that the right way to accomplish connecting this device to FS? Is there a way to enable sending an e-mail containing my voicemail messages like Asterisk does? Thanks in advance for any help you can give! Dan From brian at freeswitch.org Tue Jul 14 19:31:23 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 14 Jul 2009 21:31:23 -0500 Subject: [Freeswitch-users] GXW4104 & FreeSwitch In-Reply-To: <8DC39E34-A395-42D5-B299-070605A2DCEE@me.com> References: <8DC39E34-A395-42D5-B299-070605A2DCEE@me.com> Message-ID: <085EE9F4-A513-45FD-89E9-C66A0BE3715F@freeswitch.org> On Jul 14, 2009, at 9:20 PM, DigiLord wrote: > Hello all, > I am getting my feet wet with FreeSwitch by migrating my Asterisk box > over. I have run into a few things that I am not sure how to > accomplish. > > I have a Grandstream GXW4104 with one analog line connected. I have > it connected and I am able to receive calls on my Polycom 501 (ext > 2101) that is registered to the FreeSwitch server. The one problem is > that CallerID is not the CallerID from the caller, it's the CallerID > from the Grandstream device (ext 2100). How is the callerid passed on this device? > On the same device there is HORRIBLE echo. I have set echo > cancellation on the device to enebled and disabled to no avail. Under > Asterisk there was no echo. If it didn't have echo on asterisk it shouldn't have echo on FreeSWITCH, Can you describe the echo better? Are you using speaker phone? What codecs? > > > I setup the device as a provider. Was that the right way to > accomplish connecting this device to FS? > > Is there a way to enable sending an e-mail containing my voicemail > messages like Asterisk does? Yes check the mod_voicemail page on the wiki. /b > > > Thanks in advance for any help you can give! > > Dan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090714/eed9a9d7/attachment.html From shaheryarkh at googlemail.com Tue Jul 14 21:19:48 2009 From: shaheryarkh at googlemail.com (Muhammad Shahzad) Date: Wed, 15 Jul 2009 10:19:48 +0600 Subject: [Freeswitch-users] SIP Trace Option at Runtime Message-ID: Hi, Is there any CLI command to enable / disable SIP packet trace at runtime. I do know an option in SIP profile which enables / disable SIP trace but it to apply it i have reload mod_sofia, which at many times fail due to a running call. Thank you. -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090715/ca7ce7b2/attachment.html From jason at jasonjgw.net Tue Jul 14 21:32:25 2009 From: jason at jasonjgw.net (Jason White) Date: Wed, 15 Jul 2009 14:32:25 +1000 Subject: [Freeswitch-users] SIP Trace Option at Runtime In-Reply-To: References: Message-ID: <20090715043225.GA21117@jdc.jasonjgw.net> Muhammad Shahzad wrote: > Is there any CLI command to enable / disable SIP packet trace at runtime. sofia profile siptrace on sofia profile siptrace off sofia help would have answered your question. From elihayun at gmail.com Tue Jul 14 21:49:07 2009 From: elihayun at gmail.com (Eli Hayun) Date: Wed, 15 Jul 2009 07:49:07 +0300 Subject: [Freeswitch-users] Get voicemail messages In-Reply-To: References: Message-ID: <4A5D5FC3.4050701@savion.huji.ac.il> did you bind your lua script to directory lookups in addition to the dialplan? On Tue, Jul 14, 2009 at 7:02 AM, Eli Hayun wrote: > > Hi > > I am not using fixed xml files for the extension registration. I have > > LUA script to return an XML string to FS. > > Everything goes fine until I am trying to get the voice messages. > > When am entering my id, FS (or voicemail module) try to get the xml for > > that id, but it cant find it. My lua script did NOT recieved any xml > > request at that point. > > What should I do to solve the problem. > > > > Thanks > > Eli Hayun > > > Yes I did bind it: my lua.conf.xml is like this When an extension tried to register, I have no problem. But when I want to use VoiceMail to retrieve my messeges, I got a problem. Here is the partial log: 2009-07-15 07:44:49.373089 [INFO] mod_dialplan_xml.c:252 Processing Phone2->*98 in context default 2009-07-15 07:44:49.386466 [NOTICE] mod_dptools.c:649 Channel [sofia/internal/80671 at 132.64.3.86] has been answered 2009-07-15 07:44:51.933664 [WARNING] mod_voicemail.c:2072 Can't find user [80671 at 132.64.3.86] 2009-07-15 07:44:52.533435 [NOTICE] switch_core_state_machine.c:179 Hangup sofia/internal/80671 at 132.64.3.86 [CS_EXECUTE] [NORMAL_CLEARING] 2009-07-15 07:44:52.545698 [NOTICE] switch_core_session.c:1085 Session 3 (sofia/internal/80671 at 132.64.3.86) Ended 2009-07-15 07:44:52.545698 [NOTICE] switch_core_session.c:1087 Close Channel sofia/internal/80671 at 132.64.3.86 [CS_DESTROY] From brad.tuan at gmail.com Wed Jul 15 01:05:24 2009 From: brad.tuan at gmail.com (Brad Tuan) Date: Wed, 15 Jul 2009 16:05:24 +0800 Subject: [Freeswitch-users] How to set the IVR of VM menu?? Message-ID: How to set the date format , and the IVR flow ........?? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090715/c35eaf18/attachment-0001.html From tzury.by at reguluslabs.com Wed Jul 15 02:57:29 2009 From: tzury.by at reguluslabs.com (Tzury Bar Yochay) Date: Wed, 15 Jul 2009 12:57:29 +0300 Subject: [Freeswitch-users] SIP TLS (and SRTP) Message-ID: <10128ef10907150257u3ae1c20jb7772b66e8c1a738@mail.gmail.com> Hi all, I was following the instruction found at http://wiki.freeswitch.org/wiki/SIP_TLS When I got to Step 4 I saw that instruction of editing the dial-string. However, in my conf/dialplan/default.xml I did not found any matched entry . Version I am using: typing version at my FS CLI yields: FreeSWITCH 1.0.trunk (14144M) thanks, Tzury Bar Yochay From rupa at rupa.com Wed Jul 15 06:26:23 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Wed, 15 Jul 2009 08:26:23 -0500 Subject: [Freeswitch-users] SIP TLS (and SRTP) In-Reply-To: <10128ef10907150257u3ae1c20jb7772b66e8c1a738@mail.gmail.com> References: <10128ef10907150257u3ae1c20jb7772b66e8c1a738@mail.gmail.com> Message-ID: hmm... When I put that section in, I put the wrong filename. It should be directory/default.xml. I'll update the wiki. Also, it is a valid configuration to support tls but not srtp. I'll put a bit of a discussion in there talking about that. Setting sip_secure_media to true requires the endpoint do srtp. There is no way (that I know of) to say "do srtp if possible but if not fallback to clear". zrtp does fallback to clear if it can't negotiate keys. But zrtp is supported by far fewer endpoints and no hardphones (as of yet). On Wed, Jul 15, 2009 at 4:57 AM, Tzury Bar Yochay wrote: > Hi all, > > I was following the instruction found at > http://wiki.freeswitch.org/wiki/SIP_TLS > When I got to Step 4 I saw that instruction of editing the dial-string. > However, in my conf/dialplan/default.xml I did not found any matched entry > . > > Version I am using: > typing version at my FS CLI yields: FreeSWITCH 1.0.trunk (14144M) > > thanks, > Tzury Bar Yochay > > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090715/e465368c/attachment.html From brian at freeswitch.org Wed Jul 15 06:41:08 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 15 Jul 2009 08:41:08 -0500 Subject: [Freeswitch-users] SIP TLS (and SRTP) In-Reply-To: <10128ef10907150257u3ae1c20jb7772b66e8c1a738@mail.gmail.com> References: <10128ef10907150257u3ae1c20jb7772b66e8c1a738@mail.gmail.com> Message-ID: <8594764D-2F2A-431C-BA91-1C2D5A97C90D@freeswitch.org> It tells you to edit conf/directory/default.xml not dialplan/ default.xml and put as the dial-string. /b On Jul 15, 2009, at 4:57 AM, Tzury Bar Yochay wrote: > Hi all, > > I was following the instruction found at http://wiki.freeswitch.org/wiki/SIP_TLS > When I got to Step 4 I saw that instruction of editing the dial- > string. > However, in my conf/dialplan/default.xml I did not found any matched > entry . > > Version I am using: > typing version at my FS CLI yields: FreeSWITCH 1.0.trunk (14144M) > > thanks, > Tzury Bar Yochay From larclap at yahoo.com Wed Jul 15 07:29:28 2009 From: larclap at yahoo.com (Lars Zeb) Date: Wed, 15 Jul 2009 07:29:28 -0700 Subject: [Freeswitch-users] fs_cli - display variable values? Message-ID: <003401ca0558$a944a6b0$fbcdf410$@com> Is it possible to display the value of a variable in fs_cli? I tried "echo ${domain_name}", but it just echoed what I typed (${domain_name}), rather than its value. I do not know how to get help on an individual command from the help facility in fs_cli. I tried fs_cli itself and also the docs but could find nothing. Thanks, Lars -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090715/9bf80956/attachment.html From mrene_lists at avgs.ca Wed Jul 15 07:36:46 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Wed, 15 Jul 2009 10:36:46 -0400 Subject: [Freeswitch-users] fs_cli - display variable values? In-Reply-To: <003401ca0558$a944a6b0$fbcdf410$@com> References: <003401ca0558$a944a6b0$fbcdf410$@com> Message-ID: <3C946296-7580-44EE-B8E1-35417E26B182@avgs.ca> eval [expression] or eval uuid: [expression] or global_getvar varname or uuid_getvar uuid varname Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca Am 15-Jul-09 um 10:29 AM schrieb Lars Zeb: > Is it possible to display the value of a variable in fs_cli? I tried > ?echo ${domain_name}?, but it just echoed what I typed ($ > {domain_name}), rather than its value. > > I do not know how to get help on an individual command from the help > facility in fs_cli. I tried fs_cli itself and also the docs but > could find nothing. > > Thanks, Lars > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090715/8a9efc91/attachment.html From pjintheusa at gmail.com Wed Jul 15 07:38:24 2009 From: pjintheusa at gmail.com (Phillip Jones) Date: Wed, 15 Jul 2009 10:38:24 -0400 Subject: [Freeswitch-users] leg_timeout In-Reply-To: <367751820907141522o491b40f4jbf79c25f3c053139@mail.gmail.com> References: <367751820907140835y322c576en1eb2ba931388d732@mail.gmail.com> <87f2f3b90907140916o3f008a3br26bfca075a9e3d2e@mail.gmail.com> <367751820907141002g3a837a09meabdbc9743c3bee7@mail.gmail.com> <191c3a030907141020l48e8cb2cm6106ff4442ee1bed@mail.gmail.com> <87f2f3b90907141037r260001d4i2c41dcc0e4c25c5@mail.gmail.com> <367751820907141114w56ae9adey4836238740a1ed43@mail.gmail.com> <191c3a030907141127x255f21c5q7d80cf26ea8451a9@mail.gmail.com> <367751820907141135t2c374338y54226c9721cfa739@mail.gmail.com> <367751820907141522o491b40f4jbf79c25f3c053139@mail.gmail.com> Message-ID: <367751820907150738k4b0fae48u4ede9d87ff916324@mail.gmail.com> Hey Guys, I took a look at the source that Anthony updated. I see this: } else if (!strcasecmp((char *) hi->name, "group_confirm_file")) { ok = 1; } else if (!strcasecmp((char *) hi->name, "group_confirm_*clear*_timeout")) { ok = 1; } else if (!strcasecmp((char *) hi->name, "forked_dial")) { and: if (switch_true(switch_event_get_header(var_event, "group_confirm_*cancel*_timeout"))) { oglobals.cancel_timeout = -1; } I updated the *group_confirm_clear_timeout *to *group_confirm_cancel_timeout * and recompiled and this is now working just great. Thanks very much for incorporating this. It is much appreciated. Phillip Jones On Tue, Jul 14, 2009 at 6:22 PM, Phillip Jones wrote: > Hi there, > > I downloaded the latest trunk, compiled and updated. Still no joy I am > afraid. > > This is the log file in pastebin - http://pastebin.freeswitch.org/9712 > > Code in my managed DLL is at: http://pastebin.freeswitch.org/9715 > > Dialplan binds to above: > > > > > confirm.js is at: http://pastebin.freeswitch.org/9713 > > > Thanks again for your help on this. > > > > > > On Tue, Jul 14, 2009 at 2:35 PM, Phillip Jones wrote: > >> Ah - SVN Trunk - thought you meant DID trunk!!! My bad. >> >> Sorry - understand now! Will recompile and let you know. >> >> On Tue, Jul 14, 2009 at 2:27 PM, Anthony Minessale < >> anthony.minessale at gmail.com> wrote: >> >>> You actually updated your code and recompiled it all too? >>> This param was added about 30 seconds before I sent you the email. >>> >>> >>> >>> On Tue, Jul 14, 2009 at 1:14 PM, Phillip Jones wrote: >>> >>>> Thanks for your response. >>>> >>>> That does not seem to work. Here is my code: >>>> >>>> if(Session.Ready()) >>>> { >>>> Session.Execute("set", "ignore_early_media=true"); >>>> Session.Execute("set", "hangup_after_bridge=true"); >>>> Session.Execute("set", "ringback=${us-ring}"); >>>> >>>> Session.Answer(); >>>> string Caller_ID_Number = >>>> this.Session.GetVariable("caller_id_number"); >>>> Session.Execute("set", "group_confirm_key=exec"); >>>> *Session.Execute("set", "group_confirm_cancel_timeout=true"); >>>> * Session.Execute("set", "group_confirm_file=javascript >>>> confirm.js"); >>>> Session.Execute("bridge", >>>> "[leg_timeout=20,leg_confirm=y]sofia/gateway/broadvox/6095553828"); >>>> } >>>> Session.Hangup("USER_BUSY"); >>>> >>>> I also tried *group_confirm_cancel_leg_timeout* just in case. >>>> >>>> Am I missing something? >>>> >>>> >>>> >>>> >>>> On Tue, Jul 14, 2009 at 1:37 PM, Michael Collins wrote: >>>> >>>>> FYI, >>>>> This has been added to the wiki: >>>>> >>>>> http://wiki.freeswitch.org/wiki/Channel_Variables#group_confirm_cancel_timeout >>>>> >>>>> -MC >>>>> >>>>> >>>>> On Tue, Jul 14, 2009 at 10:20 AM, Anthony Minessale < >>>>> anthony.minessale at gmail.com> wrote: >>>>> >>>>>> update to trunk and try setting >>>>>> group_confirm_cancel_timeout=true >>>>>> >>>>>> let me know if it works >>>>>> >>>>>> >>>>>> On Tue, Jul 14, 2009 at 12:02 PM, Phillip Jones >>>>> > wrote: >>>>>> >>>>>>> Hi, >>>>>>> >>>>>>> Thanks for the reply. >>>>>>> >>>>>>> >> Both "leg_timeout" and "leg_progress_timeout" are for controlling >>>>>>> how long to wait prior to the B-leg answering. >>>>>>> >>>>>>> I think this is my point. leg_timeout seems to control how long to >>>>>>> wait prior to the bridge completeing, not the B-leg answering. >>>>>>> >>>>>>> In my situation I am using: >>>>>>> >>>>>>> Session.Execute("set", "group_confirm_key=exec"); >>>>>>> Session.Execute("set", "group_confirm_file=javascript confirm.js"); >>>>>>> >>>>>>> my confirm.js prompts leg_b for 1 (to take the call). If leg_timeout >>>>>>> is set to 10 you have 10 seconds to answer the call AND press 1. >>>>>>> >>>>>>> I just want call_timeout to be satisfied when the call is answered. >>>>>>> Not when the called party presses 1 and the bridge is complete. >>>>>>> >>>>>>> I am new all this so I will work out how to use the pastebin etc. >>>>>>> >>>>>>> Thanks for your help. >>>>>>> >>>>>>> >>>>>>> Phillip Jones >>>>>>> >>>>>>> >>>>>>> >>>>>>> On Tue, Jul 14, 2009 at 12:16 PM, Michael Collins < >>>>>>> msc at freeswitch.org> wrote: >>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> On Tue, Jul 14, 2009 at 8:35 AM, Phillip Jones < >>>>>>>> pjintheusa at gmail.com> wrote: >>>>>>>> >>>>>>>>> Hi there, >>>>>>>>> >>>>>>>>> Here is my call flow: >>>>>>>>> >>>>>>>>> 1) leg A is bridged to leg B >>>>>>>>> 2) when leg B is answered I play a confirm script - "please 1 to >>>>>>>>> accept this call" >>>>>>>>> >>>>>>>>> I only want leg B to ring 20 seconds. BUT when the caller party >>>>>>>>> answers, he should have as long as he needs to press 1. >>>>>>>>> >>>>>>>>> "leg_timeout" seems to be in play until the bridge is completed. I >>>>>>>>> need it to reset when leg b is answered. >>>>>>>>> >>>>>>>> >>>>>>>> Correct. Both "leg_timeout" and "leg_progress_timeout" are for >>>>>>>> controlling how long to wait prior to the B-leg answering. >>>>>>>> (leg_progress_timeout specifies how long to wait for any kind of progress, >>>>>>>> be it early media of some sort, ringing, or an answer.) >>>>>>>> >>>>>>>>> >>>>>>>>> I tried resetting the leg_timeout in the confirm script after leg b >>>>>>>>> is answered. I also tried using leg_progress_timeout. Neither seemed to >>>>>>>>> work. >>>>>>>>> >>>>>>>> >>>>>>>> What exactly are you trying to do? The two variables you've >>>>>>>> mentioned shouldn't have any effect on the call after it has been >>>>>>>> established. >>>>>>>> >>>>>>>>> >>>>>>>>> Any help or suggestions would be welcome. >>>>>>>>> >>>>>>>> >>>>>>>> Could you pastebin your dialplan and a debug log of a call that does >>>>>>>> not work? See this page for some handy tips on using pastebin and collecting >>>>>>>> information for debugging purposes: >>>>>>>> http://wiki.freeswitch.org/wiki/Reporting_Bugs#Debugging_Steps >>>>>>>> >>>>>>>> -MC >>>>>>>> >>>>>>>>> >>>>>>>>> Phillip Jones >>>>>>>>> >>>>>>>>> _______________________________________________ >>>>>>>>> Freeswitch-users mailing list >>>>>>>>> Freeswitch-users at lists.freeswitch.org >>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>> UNSUBSCRIBE: >>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>> http://www.freeswitch.org >>>>>>>>> >>>>>>>>> >>>>>>>> >>>>>>>> _______________________________________________ >>>>>>>> Freeswitch-users mailing list >>>>>>>> Freeswitch-users at lists.freeswitch.org >>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>> UNSUBSCRIBE: >>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>> http://www.freeswitch.org >>>>>>>> >>>>>>>> >>>>>>> >>>>>>> _______________________________________________ >>>>>>> Freeswitch-users mailing list >>>>>>> Freeswitch-users at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE: >>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> -- >>>>>> Anthony Minessale II >>>>>> >>>>>> FreeSWITCH http://www.freeswitch.org/ >>>>>> ClueCon http://www.cluecon.com/ >>>>>> Twitter: http://twitter.com/FreeSWITCH_wire >>>>>> >>>>>> AIM: anthm >>>>>> MSN:anthony_minessale at hotmail.com >>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>>>>> IRC: irc.freenode.net #freeswitch >>>>>> >>>>>> FreeSWITCH Developer Conference >>>>>> 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 >>>> >>>> >>> >>> >>> -- >>> Anthony Minessale II >>> >>> FreeSWITCH http://www.freeswitch.org/ >>> ClueCon http://www.cluecon.com/ >>> Twitter: http://twitter.com/FreeSWITCH_wire >>> >>> AIM: anthm >>> MSN:anthony_minessale at hotmail.com >>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>> IRC: irc.freenode.net #freeswitch >>> >>> FreeSWITCH Developer Conference >>> 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/20090715/440c585b/attachment-0001.html From brian at freeswitch.org Wed Jul 15 07:38:21 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 15 Jul 2009 09:38:21 -0500 Subject: [Freeswitch-users] fs_cli - display variable values? In-Reply-To: <003401ca0558$a944a6b0$fbcdf410$@com> References: <003401ca0558$a944a6b0$fbcdf410$@com> Message-ID: global_getvar will list all globals... "uuid_getvar uuid var" get the var off the uuid. /b On Jul 15, 2009, at 9:29 AM, Lars Zeb wrote: > Is it possible to display the value of a variable in fs_cli? I tried > ?echo ${domain_name}?, but it just echoed what I typed ($ > {domain_name}), rather than its value. > > I do not know how to get help on an individual command from the help > facility in fs_cli. I tried fs_cli itself and also the docs but > could find nothing. > > Thanks, Lars -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090715/c7e02bb7/attachment.html From anthony.minessale at gmail.com Wed Jul 15 07:51:51 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 15 Jul 2009 09:51:51 -0500 Subject: [Freeswitch-users] Preventing disconnect on event_sockte close In-Reply-To: <7DB369DA44D5CF42B6C23BD7D611A50FDACD801F9A@VA3DIAXVS061.RED001.local> References: <7DB369DA44D5CF42B6C23BD7D611A50FDA7E1D5FD6@VA3DIAXVS061.RED001.local> <87f2f3b90907131808l100afe7dwc95d0f624ff38d50@mail.gmail.com> <7DB369DA44D5CF42B6C23BD7D611A50FDA7E1D604D@VA3DIAXVS061.RED001.local> <87f2f3b90907141011g46686396se27fa3cc0320963d@mail.gmail.com> <7DB369DA44D5CF42B6C23BD7D611A50FDABBFA760F@VA3DIAXVS061.RED001.local> <87f2f3b90907141440l5235ab70h7d48ee961367f5e@mail.gmail.com> <7DB369DA44D5CF42B6C23BD7D611A50FDACD801F9A@VA3DIAXVS061.RED001.local> Message-ID: <191c3a030907150751u39c6b483w23f22b7667639d47@mail.gmail.com> You could dynamically transfer it to an empty conference. api uuid_transfer conference:foo inline You may want to consider joining irc and getting some realtime help to avoid a really long thread and report your solutions back here in a follow up email. On Tue, Jul 14, 2009 at 5:03 PM, Weaver, Eric wrote: > Looking a FS to use as Media mixer for conferencing platform. Not really > doing call to call bridging. We really don?t have extensions?. Conferences > are created o the fly as needed. Already have the conf and call control app > done and in production using a different audio mixer, I would like to put FS > in place of it. > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael > Collins > *Sent:* Tuesday, July 14, 2009 3:40 PM > > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Preventing disconnect on event_sockte > close > > > > > > On Tue, Jul 14, 2009 at 2:14 PM, Weaver, Eric > wrote: > > Using api uuid_park 81b65478-70b9-11de-bf26-b962186102f7 before terminating > the NC session works, the call is not disconnected. > > > > Once that is done, I do not receive DTMF and cannot play prompts to the > caller, they seem to be in limbo. I can uuid_kill the call but I need to get > dtmf and play prompts to them. Perhaps Park is not where I need to put > these calls ? > > > > To get a call out of park you need to bridge to it or transfer it to > another extension. If you have an extension you can just uuid_transfer the > parked call's uuid. If you have an existing call's uuid you can use > uuid_bridge to bridge the two together. > > Could you remind me of the application you're building? Just curious what > the big picture is. > -MC > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael > Collins > *Sent:* Tuesday, July 14, 2009 11:12 AM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Preventing disconnect on event_sockte > close > > > > > > On Tue, Jul 14, 2009 at 9:59 AM, Weaver, Eric > wrote: > > Tried it using FreeSWITCH Version 1.0.4pre9 (14036M) and the following > steps > > > > Start netcat > > > > netcat -v -l -p 14000 > > > > place call, socket is connected via dial plan, enter the following. > > > > connect\n\n > > > > sendmsg > > call-command: execute > > execute-app-name: answer\n\n > > > > > > sendmsg > > call-command: execute > > execute-app-name: playback > > execute-app-arg: /home/eweaver/holdmusic.wav > > > > > > > > > > sendmsg > > call-command: execute > > execute-app-name: park > > > Try this: > api uuid_park > > You'll need to capture the uuid at some point and store it. For testing I > just manually copied and pasted it to/from the console screen. > -MC > > > > Console window displays this message: > > > > 2009-07-14 10:42:29.245833 [ERR] switch_ivr.c:654 Cannot park channels that > are under control already. > > > > at this point, ^C in the netcat window. Call is disconnected. > > > > > > Need to be able to park these calls so they can then be handled from an > inbound event socket connection. > > > > eric > > > > > > > > > > > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael > Collins > *Sent:* Monday, July 13, 2009 7:08 PM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Preventing disconnect on event_sockte > close > > > > I don't know if this will work for you but I just tested this scenario with > uuid_park. After parking the call I disconnected the socket and the call > continued. I did the same thing with uuid_transfer. After the transfer I > disconnected the socket and the call continued. > > How are you handling the call and how is the socket getting disconnected? > > -MC > > On Mon, Jul 13, 2009 at 4:35 PM, Weaver, Eric > wrote: > > Using mod_event_socket in outbound mode, is there any to prevent a call > from being disconnected when the outbound socket is closed ? I would like to > handle the initial inbound call using outbound but after the disposition of > the call is determined, close the socket and have that call managed using an > inbound socket instead. > > > > Eric Weaver > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference 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/20090715/2e40fb09/attachment.html From anthony.minessale at gmail.com Wed Jul 15 08:25:03 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 15 Jul 2009 10:25:03 -0500 Subject: [Freeswitch-users] leg_timeout In-Reply-To: <367751820907150738k4b0fae48u4ede9d87ff916324@mail.gmail.com> References: <367751820907140835y322c576en1eb2ba931388d732@mail.gmail.com> <87f2f3b90907140916o3f008a3br26bfca075a9e3d2e@mail.gmail.com> <367751820907141002g3a837a09meabdbc9743c3bee7@mail.gmail.com> <191c3a030907141020l48e8cb2cm6106ff4442ee1bed@mail.gmail.com> <87f2f3b90907141037r260001d4i2c41dcc0e4c25c5@mail.gmail.com> <367751820907141114w56ae9adey4836238740a1ed43@mail.gmail.com> <191c3a030907141127x255f21c5q7d80cf26ea8451a9@mail.gmail.com> <367751820907141135t2c374338y54226c9721cfa739@mail.gmail.com> <367751820907141522o491b40f4jbf79c25f3c053139@mail.gmail.com> <367751820907150738k4b0fae48u4ede9d87ff916324@mail.gmail.com> Message-ID: <191c3a030907150825w283d45c6l3642342f6b11f228@mail.gmail.com> doh, i'm slipping. fixed in tree On Wed, Jul 15, 2009 at 9:38 AM, Phillip Jones wrote: > Hey Guys, > > I took a look at the source that Anthony updated. I see this: > > } else if (!strcasecmp((char *) hi->name, "group_confirm_file")) { > ok = 1; > } else if (!strcasecmp((char *) hi->name, "group_confirm_*clear*_timeout")) > { > ok = 1; > } else if (!strcasecmp((char *) hi->name, "forked_dial")) { > > and: > > if (switch_true(switch_event_get_header(var_event, "group_confirm_*cancel*_timeout"))) > { > oglobals.cancel_timeout = -1; > } > > I updated the *group_confirm_clear_timeout *to * > group_confirm_cancel_timeout* and recompiled and this is now working just > great. > > Thanks very much for incorporating this. It is much appreciated. > > > Phillip Jones > > > > > On Tue, Jul 14, 2009 at 6:22 PM, Phillip Jones wrote: > >> Hi there, >> >> I downloaded the latest trunk, compiled and updated. Still no joy I am >> afraid. >> >> This is the log file in pastebin - http://pastebin.freeswitch.org/9712 >> >> Code in my managed DLL is at: http://pastebin.freeswitch.org/9715 >> >> Dialplan binds to above: >> >> >> >> >> confirm.js is at: http://pastebin.freeswitch.org/9713 >> >> >> Thanks again for your help on this. >> >> >> >> >> >> On Tue, Jul 14, 2009 at 2:35 PM, Phillip Jones wrote: >> >>> Ah - SVN Trunk - thought you meant DID trunk!!! My bad. >>> >>> Sorry - understand now! Will recompile and let you know. >>> >>> On Tue, Jul 14, 2009 at 2:27 PM, Anthony Minessale < >>> anthony.minessale at gmail.com> wrote: >>> >>>> You actually updated your code and recompiled it all too? >>>> This param was added about 30 seconds before I sent you the email. >>>> >>>> >>>> >>>> On Tue, Jul 14, 2009 at 1:14 PM, Phillip Jones wrote: >>>> >>>>> Thanks for your response. >>>>> >>>>> That does not seem to work. Here is my code: >>>>> >>>>> if(Session.Ready()) >>>>> { >>>>> Session.Execute("set", "ignore_early_media=true"); >>>>> Session.Execute("set", "hangup_after_bridge=true"); >>>>> Session.Execute("set", "ringback=${us-ring}"); >>>>> >>>>> Session.Answer(); >>>>> string Caller_ID_Number = >>>>> this.Session.GetVariable("caller_id_number"); >>>>> Session.Execute("set", "group_confirm_key=exec"); >>>>> *Session.Execute("set", "group_confirm_cancel_timeout=true"); >>>>> * Session.Execute("set", "group_confirm_file=javascript >>>>> confirm.js"); >>>>> Session.Execute("bridge", >>>>> "[leg_timeout=20,leg_confirm=y]sofia/gateway/broadvox/6095553828"); >>>>> } >>>>> Session.Hangup("USER_BUSY"); >>>>> >>>>> I also tried *group_confirm_cancel_leg_timeout* just in case. >>>>> >>>>> Am I missing something? >>>>> >>>>> >>>>> >>>>> >>>>> On Tue, Jul 14, 2009 at 1:37 PM, Michael Collins wrote: >>>>> >>>>>> FYI, >>>>>> This has been added to the wiki: >>>>>> >>>>>> http://wiki.freeswitch.org/wiki/Channel_Variables#group_confirm_cancel_timeout >>>>>> >>>>>> -MC >>>>>> >>>>>> >>>>>> On Tue, Jul 14, 2009 at 10:20 AM, Anthony Minessale < >>>>>> anthony.minessale at gmail.com> wrote: >>>>>> >>>>>>> update to trunk and try setting >>>>>>> group_confirm_cancel_timeout=true >>>>>>> >>>>>>> let me know if it works >>>>>>> >>>>>>> >>>>>>> On Tue, Jul 14, 2009 at 12:02 PM, Phillip Jones < >>>>>>> pjintheusa at gmail.com> wrote: >>>>>>> >>>>>>>> Hi, >>>>>>>> >>>>>>>> Thanks for the reply. >>>>>>>> >>>>>>>> >> Both "leg_timeout" and "leg_progress_timeout" are for controlling >>>>>>>> how long to wait prior to the B-leg answering. >>>>>>>> >>>>>>>> I think this is my point. leg_timeout seems to control how long to >>>>>>>> wait prior to the bridge completeing, not the B-leg answering. >>>>>>>> >>>>>>>> In my situation I am using: >>>>>>>> >>>>>>>> Session.Execute("set", "group_confirm_key=exec"); >>>>>>>> Session.Execute("set", "group_confirm_file=javascript confirm.js"); >>>>>>>> >>>>>>>> my confirm.js prompts leg_b for 1 (to take the call). If leg_timeout >>>>>>>> is set to 10 you have 10 seconds to answer the call AND press 1. >>>>>>>> >>>>>>>> I just want call_timeout to be satisfied when the call is answered. >>>>>>>> Not when the called party presses 1 and the bridge is complete. >>>>>>>> >>>>>>>> I am new all this so I will work out how to use the pastebin etc. >>>>>>>> >>>>>>>> Thanks for your help. >>>>>>>> >>>>>>>> >>>>>>>> Phillip Jones >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> On Tue, Jul 14, 2009 at 12:16 PM, Michael Collins < >>>>>>>> msc at freeswitch.org> wrote: >>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> On Tue, Jul 14, 2009 at 8:35 AM, Phillip Jones < >>>>>>>>> pjintheusa at gmail.com> wrote: >>>>>>>>> >>>>>>>>>> Hi there, >>>>>>>>>> >>>>>>>>>> Here is my call flow: >>>>>>>>>> >>>>>>>>>> 1) leg A is bridged to leg B >>>>>>>>>> 2) when leg B is answered I play a confirm script - "please 1 to >>>>>>>>>> accept this call" >>>>>>>>>> >>>>>>>>>> I only want leg B to ring 20 seconds. BUT when the caller party >>>>>>>>>> answers, he should have as long as he needs to press 1. >>>>>>>>>> >>>>>>>>>> "leg_timeout" seems to be in play until the bridge is completed. I >>>>>>>>>> need it to reset when leg b is answered. >>>>>>>>>> >>>>>>>>> >>>>>>>>> Correct. Both "leg_timeout" and "leg_progress_timeout" are for >>>>>>>>> controlling how long to wait prior to the B-leg answering. >>>>>>>>> (leg_progress_timeout specifies how long to wait for any kind of progress, >>>>>>>>> be it early media of some sort, ringing, or an answer.) >>>>>>>>> >>>>>>>>>> >>>>>>>>>> I tried resetting the leg_timeout in the confirm script after leg >>>>>>>>>> b is answered. I also tried using leg_progress_timeout. Neither seemed to >>>>>>>>>> work. >>>>>>>>>> >>>>>>>>> >>>>>>>>> What exactly are you trying to do? The two variables you've >>>>>>>>> mentioned shouldn't have any effect on the call after it has been >>>>>>>>> established. >>>>>>>>> >>>>>>>>>> >>>>>>>>>> Any help or suggestions would be welcome. >>>>>>>>>> >>>>>>>>> >>>>>>>>> Could you pastebin your dialplan and a debug log of a call that >>>>>>>>> does not work? See this page for some handy tips on using pastebin and >>>>>>>>> collecting information for debugging purposes: >>>>>>>>> http://wiki.freeswitch.org/wiki/Reporting_Bugs#Debugging_Steps >>>>>>>>> >>>>>>>>> -MC >>>>>>>>> >>>>>>>>>> >>>>>>>>>> Phillip Jones >>>>>>>>>> >>>>>>>>>> _______________________________________________ >>>>>>>>>> Freeswitch-users mailing list >>>>>>>>>> Freeswitch-users at lists.freeswitch.org >>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>>> UNSUBSCRIBE: >>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>>> http://www.freeswitch.org >>>>>>>>>> >>>>>>>>>> >>>>>>>>> >>>>>>>>> _______________________________________________ >>>>>>>>> Freeswitch-users mailing list >>>>>>>>> Freeswitch-users at lists.freeswitch.org >>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>> UNSUBSCRIBE: >>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>> http://www.freeswitch.org >>>>>>>>> >>>>>>>>> >>>>>>>> >>>>>>>> _______________________________________________ >>>>>>>> Freeswitch-users mailing list >>>>>>>> Freeswitch-users at lists.freeswitch.org >>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>> UNSUBSCRIBE: >>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>> http://www.freeswitch.org >>>>>>>> >>>>>>>> >>>>>>> >>>>>>> >>>>>>> -- >>>>>>> Anthony Minessale II >>>>>>> >>>>>>> FreeSWITCH http://www.freeswitch.org/ >>>>>>> ClueCon http://www.cluecon.com/ >>>>>>> Twitter: http://twitter.com/FreeSWITCH_wire >>>>>>> >>>>>>> AIM: anthm >>>>>>> MSN:anthony_minessale at hotmail.com >>>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>>>>>> IRC: irc.freenode.net #freeswitch >>>>>>> >>>>>>> FreeSWITCH Developer Conference >>>>>>> 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 >>>>> >>>>> >>>> >>>> >>>> -- >>>> Anthony Minessale II >>>> >>>> FreeSWITCH http://www.freeswitch.org/ >>>> ClueCon http://www.cluecon.com/ >>>> Twitter: http://twitter.com/FreeSWITCH_wire >>>> >>>> AIM: anthm >>>> MSN:anthony_minessale at hotmail.com >>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>>> IRC: irc.freenode.net #freeswitch >>>> >>>> FreeSWITCH Developer Conference >>>> 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/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference 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/20090715/bacf2162/attachment-0001.html From Kareem.Hamdy at trustvesta.com Wed Jul 15 08:29:49 2009 From: Kareem.Hamdy at trustvesta.com (Kareem Hamdy) Date: Wed, 15 Jul 2009 08:29:49 -0700 Subject: [Freeswitch-users] Freeswitch-users Digest, Vol 37, Issue 90 In-Reply-To: References: Message-ID: <1134625859513549B3B943E0133490E202AC61485E@TDCP-EXSTORE-01.ad.trustvesta.com> Thanks Michael, but I'm setting up a T1, not a PRI. I should be able to use all 24 channels. ? -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of freeswitch-users-request at lists.freeswitch.org Sent: Wednesday, July 15, 2009 1:06 AM To: freeswitch-users at lists.freeswitch.org Subject: Freeswitch-users Digest, Vol 37, Issue 90 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: OpenZAP and FreeSWITCH w/ Sangoma (Michael Collins) 2. GXW4104 & FreeSwitch (DigiLord) 3. Re: GXW4104 & FreeSwitch (Brian West) 4. SIP Trace Option at Runtime (Muhammad Shahzad) 5. Re: SIP Trace Option at Runtime (Jason White) 6. Re: Get voicemail messages (Eli Hayun) 7. How to set the IVR of VM menu?? (Brad Tuan) ---------------------------------------------------------------------- Message: 1 Date: Tue, 14 Jul 2009 17:24:18 -0700 From: Michael Collins Subject: Re: [Freeswitch-users] OpenZAP and FreeSWITCH w/ Sangoma To: freeswitch-users at lists.freeswitch.org Message-ID: <87f2f3b90907141724q2735fac1jdacea3994db62782 at mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" See inline comments On Tue, Jul 14, 2009 at 5:04 PM, Kareem Hamdy wrote: > Hello: > > I would like to connect a Sangoma T1 card to FreeSWITCH. Most of the docs > I see pertain to a PRI. When I leave out the d-chan notation, I get errors > regarding not able to get the d-chan up and running in the CLI. > > Here's my info: > > [span wanpipe T1] > trunk_type => t1 > b-channel => 1:1-24 b-channel => 1:1-23 d-channel => 1:24 > > > [span wanpipe T2] > trunk_type => t1 > b-channel => 2:1-24 > set up like span 1 example > > ---- > > #================================================ > # WANPIPE1 Configuration File > #================================================ > # > # Date: Wed Dec 6 20:29:03 UTC 2006 > # > # Note: This file was generated automatically > # by /usr/local/sbin/setup-sangoma program. > # > # If you want to edit this file, it is > # recommended that you use wancfg program > # to do so. > #================================================ > # Sangoma Technologies Inc. > #================================================ > > [devices] > wanpipe1 = WAN_AFT_TE1, Comment > > [interfaces] > w1g1 = wanpipe1, , TDM_VOICE_API, Comment > > [wanpipe1] > CARD_TYPE = AFT > S514CPU = A > CommPort = PRI > AUTO_PCISLOT = NO > PCISLOT = 1 > PCIBUS = 6 > FE_MEDIA = T1 > FE_LCODE = B8ZS > FE_FRAME = ESF > FE_LINE = 1 > TE_CLOCK = NORMAL > TE_REF_CLOCK = 0 > > TE_HIGHIMPEDANCE = NO > LBO = 0DB > FE_TXTRISTATE = NO > MTU = 1500 > UDPPORT = 9000 > TTL = 255 > IGNORE_FRONT_END = NO > TDMV_SPAN = 1 > TDMV_DCHAN = 0 TDMV_DCAHN = 24 > > TDMV_HW_DTMF = YES > TDMV_HW_FAX_DETECT = NO > > [w1g1] > ACTIVE_CH = ALL > TDMV_HWEC = YES > MTU = 80 > > > --- > > > > In freeSWITCH's openzap.conf.xml, I've tried pri_span as well as analog. > > I cannot find a straight up T1 wiki anywhere. Would someone please provide > an example? > > > Thanks, > Kareem > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090714/da16d89e/attachment-0001.html ------------------------------ Message: 2 Date: Tue, 14 Jul 2009 19:20:19 -0700 From: DigiLord Subject: [Freeswitch-users] GXW4104 & FreeSwitch To: freeswitch-users at lists.freeswitch.org Message-ID: <8DC39E34-A395-42D5-B299-070605A2DCEE at me.com> Content-Type: text/plain; charset=us-ascii; format=flowed; delsp=yes Hello all, I am getting my feet wet with FreeSwitch by migrating my Asterisk box over. I have run into a few things that I am not sure how to accomplish. I have a Grandstream GXW4104 with one analog line connected. I have it connected and I am able to receive calls on my Polycom 501 (ext 2101) that is registered to the FreeSwitch server. The one problem is that CallerID is not the CallerID from the caller, it's the CallerID from the Grandstream device (ext 2100). On the same device there is HORRIBLE echo. I have set echo cancellation on the device to enabled and disabled to no avail. Under Asterisk there was no echo. I setup the device as a provider. Was that the right way to accomplish connecting this device to FS? Is there a way to enable sending an e-mail containing my voicemail messages like Asterisk does? Thanks in advance for any help you can give! Dan ------------------------------ Message: 3 Date: Tue, 14 Jul 2009 21:31:23 -0500 From: Brian West Subject: Re: [Freeswitch-users] GXW4104 & FreeSwitch To: freeswitch-users at lists.freeswitch.org Message-ID: <085EE9F4-A513-45FD-89E9-C66A0BE3715F at freeswitch.org> Content-Type: text/plain; charset="us-ascii" On Jul 14, 2009, at 9:20 PM, DigiLord wrote: > Hello all, > I am getting my feet wet with FreeSwitch by migrating my Asterisk box > over. I have run into a few things that I am not sure how to > accomplish. > > I have a Grandstream GXW4104 with one analog line connected. I have > it connected and I am able to receive calls on my Polycom 501 (ext > 2101) that is registered to the FreeSwitch server. The one problem is > that CallerID is not the CallerID from the caller, it's the CallerID > from the Grandstream device (ext 2100). How is the callerid passed on this device? > On the same device there is HORRIBLE echo. I have set echo > cancellation on the device to enebled and disabled to no avail. Under > Asterisk there was no echo. If it didn't have echo on asterisk it shouldn't have echo on FreeSWITCH, Can you describe the echo better? Are you using speaker phone? What codecs? > > > I setup the device as a provider. Was that the right way to > accomplish connecting this device to FS? > > Is there a way to enable sending an e-mail containing my voicemail > messages like Asterisk does? Yes check the mod_voicemail page on the wiki. /b > > > Thanks in advance for any help you can give! > > Dan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090714/eed9a9d7/attachment-0001.html ------------------------------ Message: 4 Date: Wed, 15 Jul 2009 10:19:48 +0600 From: Muhammad Shahzad Subject: [Freeswitch-users] SIP Trace Option at Runtime To: freeswitch-users at lists.freeswitch.org Message-ID: Content-Type: text/plain; charset="utf-8" Hi, Is there any CLI command to enable / disable SIP packet trace at runtime. I do know an option in SIP profile which enables / disable SIP trace but it to apply it i have reload mod_sofia, which at many times fail due to a running call. Thank you. -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090715/ca7ce7b2/attachment-0001.html ------------------------------ Message: 5 Date: Wed, 15 Jul 2009 14:32:25 +1000 From: Jason White Subject: Re: [Freeswitch-users] SIP Trace Option at Runtime To: freeswitch-users at lists.freeswitch.org Message-ID: <20090715043225.GA21117 at jdc.jasonjgw.net> Content-Type: text/plain; charset=us-ascii Muhammad Shahzad wrote: > Is there any CLI command to enable / disable SIP packet trace at runtime. sofia profile siptrace on sofia profile siptrace off sofia help would have answered your question. ------------------------------ Message: 6 Date: Wed, 15 Jul 2009 07:49:07 +0300 From: Eli Hayun Subject: Re: [Freeswitch-users] Get voicemail messages To: "freeswitch-users at lists.freeswitch.org" Message-ID: <4A5D5FC3.4050701 at savion.huji.ac.il> Content-Type: text/plain; charset=ISO-8859-1 did you bind your lua script to directory lookups in addition to the dialplan? On Tue, Jul 14, 2009 at 7:02 AM, Eli Hayun wrote: > > Hi > > I am not using fixed xml files for the extension registration. I have > > LUA script to return an XML string to FS. > > Everything goes fine until I am trying to get the voice messages. > > When am entering my id, FS (or voicemail module) try to get the xml for > > that id, but it cant find it. My lua script did NOT recieved any xml > > request at that point. > > What should I do to solve the problem. > > > > Thanks > > Eli Hayun > > > Yes I did bind it: my lua.conf.xml is like this When an extension tried to register, I have no problem. But when I want to use VoiceMail to retrieve my messeges, I got a problem. Here is the partial log: 2009-07-15 07:44:49.373089 [INFO] mod_dialplan_xml.c:252 Processing Phone2->*98 in context default 2009-07-15 07:44:49.386466 [NOTICE] mod_dptools.c:649 Channel [sofia/internal/80671 at 132.64.3.86] has been answered 2009-07-15 07:44:51.933664 [WARNING] mod_voicemail.c:2072 Can't find user [80671 at 132.64.3.86] 2009-07-15 07:44:52.533435 [NOTICE] switch_core_state_machine.c:179 Hangup sofia/internal/80671 at 132.64.3.86 [CS_EXECUTE] [NORMAL_CLEARING] 2009-07-15 07:44:52.545698 [NOTICE] switch_core_session.c:1085 Session 3 (sofia/internal/80671 at 132.64.3.86) Ended 2009-07-15 07:44:52.545698 [NOTICE] switch_core_session.c:1087 Close Channel sofia/internal/80671 at 132.64.3.86 [CS_DESTROY] ------------------------------ Message: 7 Date: Wed, 15 Jul 2009 16:05:24 +0800 From: Brad Tuan Subject: [Freeswitch-users] How to set the IVR of VM menu?? To: freeswitch-users Message-ID: Content-Type: text/plain; charset="iso-8859-1" How to set the date format , and the IVR flow ........?? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090715/c35eaf18/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 37, Issue 90 ************************************************ From brian at freeswitch.org Wed Jul 15 08:34:12 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 15 Jul 2009 10:34:12 -0500 Subject: [Freeswitch-users] Freeswitch-users Digest, Vol 37, Issue 90 In-Reply-To: <1134625859513549B3B943E0133490E202AC61485E@TDCP-EXSTORE-01.ad.trustvesta.com> References: <1134625859513549B3B943E0133490E202AC61485E@TDCP-EXSTORE-01.ad.trustvesta.com> Message-ID: <2E49E0A3-397F-4000-AADF-85796D18E301@freeswitch.org> Are you trying to do E&M? /b On Jul 15, 2009, at 10:29 AM, Kareem Hamdy wrote: > Thanks Michael, but I'm setting up a T1, not a PRI. I should be > able to use all 24 channels. From saeedahmad1981 at gmail.com Wed Jul 15 08:49:52 2009 From: saeedahmad1981 at gmail.com (Saeed Ahmad) Date: Wed, 15 Jul 2009 17:49:52 +0200 Subject: [Freeswitch-users] Originate in Dial plan In-Reply-To: <87f2f3b90907140048m7e1eea18x46e9418e43fb4007@mail.gmail.com> References: <8ccbff060907120959ma7c0ey22adeae196f872c3@mail.gmail.com> <87f2f3b90907131810q219c04f3ra385e0e1d252d31c@mail.gmail.com> <8ccbff060907132130l329f7e1at87a16da0c2108e2@mail.gmail.com> <87f2f3b90907140048m7e1eea18x46e9418e43fb4007@mail.gmail.com> Message-ID: Call back is quite cool where users are in areas where no callshops, internet and other calling facilities are available except mobile phones, users will pay both calls. There might be some other usages as well. - Saeed On Tue, Jul 14, 2009 at 9:48 AM, Michael Collins wrote: > > > On Mon, Jul 13, 2009 at 9:30 PM, Dome Charoenyost wrote: > >> 2009/7/14 Michael Collins : >> > What phone number do you call back? I mean, how do you know what the >> > customer's number is? Do you go by the caller id number? >> yes callback to caller id >> > > Okay, here's a dialplan snippet that I used to successfully do the > autocallback. In my case I used ext 1001 as the customer and portaudio as > the "agent" if you get my meaning. Extension 1001 dials 9902, hangs up, and > immediately the api_hangup_hook's originate command is executed. In this > case it calls portaudio/auto_answer for the A-leg and user/1001 as the > B-leg. I don't claim that it's the prettiest thing in the world but it > definitely works. You'll need to adjust according to your specific > situation. > > > > > > > > > > > > > > > > > > > > > Let us know how it goes. BTW, what is the reason for this type of scenario? > Just curious. > -MC > >> >> >> > >> > -MC >> > >> > On Sun, Jul 12, 2009 at 9:59 AM, Dome Charoenyost >> wrote: >> >> >> >> Dear sir, >> >> I want to user dialplan callback to customer. is posible to >> >> to this is dialplan XML ? >> >> Now i use javascript. >> >> my call flow. >> >> 1. customer call >> >> 2. FS rining and wait until customer hangup >> >> 3. callback to customer number >> >> >> >> >> >> Best Regards. >> >> >> >> Dome C. >> >> >> >> _______________________________________________ >> >> Freeswitch-users mailing list >> >> Freeswitch-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> > >> > >> > _______________________________________________ >> > Freeswitch-users mailing list >> > Freeswitch-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090715/aacfbd88/attachment.html From mike at jerris.com Wed Jul 15 08:56:39 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 15 Jul 2009 11:56:39 -0400 Subject: [Freeswitch-users] How to set the IVR of VM menu?? In-Reply-To: References: Message-ID: <930DA7E7-DD57-44D2-914B-9D301041EEC3@jerris.com> Please try looking on the wiki, this and many other questions should be answered for you there. Mike On Jul 15, 2009, at 4:05 AM, Brad Tuan wrote: > How to set the date format , and the IVR flow ........?? > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.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 Wed Jul 15 09:03:26 2009 From: saeedahmad1981 at gmail.com (Saeed Ahmad) Date: Wed, 15 Jul 2009 18:03:26 +0200 Subject: [Freeswitch-users] SIP Trace Option at Runtime In-Reply-To: <20090715043225.GA21117@jdc.jasonjgw.net> References: <20090715043225.GA21117@jdc.jasonjgw.net> Message-ID: and how can we do the same for console message? I noticed that if then even we press F8 on cli then it won't turn on the log message, is there anyway to enable them even loglevel is set to 'err' - Saeed On Wed, Jul 15, 2009 at 6:32 AM, Jason White wrote: > Muhammad Shahzad wrote: > > Is there any CLI command to enable / disable SIP packet trace at > runtime. > > sofia profile siptrace on > sofia profile siptrace off > > sofia help would have answered your question. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090715/93a0c247/attachment.html From brian at freeswitch.org Wed Jul 15 09:06:30 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 15 Jul 2009 11:06:30 -0500 Subject: [Freeswitch-users] SIP Trace Option at Runtime In-Reply-To: References: <20090715043225.GA21117@jdc.jasonjgw.net> Message-ID: where are you setting the loglevel to err at? /b On Jul 15, 2009, at 11:03 AM, Saeed Ahmad wrote: > and how can we do the same for console message? > > I noticed that if > > > > then even we press F8 on cli then it won't turn on the log message, > is there anyway to enable them even loglevel is set to 'err' > > - Saeed > From saeedahmad1981 at gmail.com Wed Jul 15 09:08:14 2009 From: saeedahmad1981 at gmail.com (Saeed Ahmad) Date: Wed, 15 Jul 2009 18:08:14 +0200 Subject: [Freeswitch-users] SIP Trace Option at Runtime In-Reply-To: References: <20090715043225.GA21117@jdc.jasonjgw.net> Message-ID: switch.conf.xml On Wed, Jul 15, 2009 at 6:06 PM, Brian West wrote: > where are you setting the loglevel to err at? > > /b > > On Jul 15, 2009, at 11:03 AM, Saeed Ahmad wrote: > > > and how can we do the same for console message? > > > > I noticed that if > > > > > > > > then even we press F8 on cli then it won't turn on the log message, > > is there anyway to enable them even loglevel is set to 'err' > > > > - Saeed > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090715/d1ed0522/attachment.html From brian at freeswitch.org Wed Jul 15 09:11:15 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 15 Jul 2009 11:11:15 -0500 Subject: [Freeswitch-users] SIP Trace Option at Runtime In-Reply-To: References: <20090715043225.GA21117@jdc.jasonjgw.net> Message-ID: <60FEB0EB-321A-431F-A8A2-1D92253D7040@freeswitch.org> you're lowering the core loglevel so you'll fsctl logelvel debug then console loglevel debug../. If you wish to start in err then edit console.conf.xml /b On Jul 15, 2009, at 11:08 AM, Saeed Ahmad wrote: > switch.conf.xml > From mrene_lists at avgs.ca Wed Jul 15 09:15:42 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Wed, 15 Jul 2009 12:15:42 -0400 Subject: [Freeswitch-users] SIP Trace Option at Runtime In-Reply-To: References: <20090715043225.GA21117@jdc.jasonjgw.net> Message-ID: Thats the global loglevel.... here's a summary of how logging works: When a log message is issued, FS checks the global level (fsctl loglevel xxx, or the one in switch.conf.xml) and discards anything less important than this loglevel. Once this is passed, the module which you use control FS will filter the logs it displays for you. This can be mod_console, mod_event_socket (for fs_cli), and even mod_logfile. It is clear that most messages wont make it to your screen if your global loglevel is at error. Change your switch.conf.xml level to debug and then you'll see them. Pressing F8 only control whatever you are using to connect to FS, not the global level. Let me know if that makes any sense to you, and if it does, a little documentation on the wiki could be of great use. Regards,, Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca Am 15-Jul-09 um 12:08 PM schrieb Saeed Ahmad: > switch.conf.xml > > > On Wed, Jul 15, 2009 at 6:06 PM, Brian West > wrote: > where are you setting the loglevel to err at? > > /b > > On Jul 15, 2009, at 11:03 AM, Saeed Ahmad wrote: > > > and how can we do the same for console message? > > > > I noticed that if > > > > > > > > then even we press F8 on cli then it won't turn on the log message, > > is there anyway to enable them even loglevel is set to 'err' > > > > - Saeed > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090715/09908ba1/attachment.html From mrene_lists at avgs.ca Wed Jul 15 09:16:57 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Wed, 15 Jul 2009 12:16:57 -0400 Subject: [Freeswitch-users] SIP Trace Option at Runtime In-Reply-To: References: <20090715043225.GA21117@jdc.jasonjgw.net> Message-ID: <45002ACD-36EC-4BBE-8D8F-9F74823938E7@avgs.ca> Oh and the default level for "siptrace" is CONSOLE, you can change it runtime (if you want to log it into log files) by doing sofia tracelevel [xxx] (where [xxx] is the loglevel) Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca Am 15-Jul-09 um 12:08 PM schrieb Saeed Ahmad: > switch.conf.xml > > > On Wed, Jul 15, 2009 at 6:06 PM, Brian West > wrote: > where are you setting the loglevel to err at? > > /b > > On Jul 15, 2009, at 11:03 AM, Saeed Ahmad wrote: > > > and how can we do the same for console message? > > > > I noticed that if > > > > > > > > then even we press F8 on cli then it won't turn on the log message, > > is there anyway to enable them even loglevel is set to 'err' > > > > - Saeed > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090715/b0f321ea/attachment.html From msc at freeswitch.org Wed Jul 15 09:25:56 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 15 Jul 2009 09:25:56 -0700 Subject: [Freeswitch-users] leg_timeout In-Reply-To: <367751820907150738k4b0fae48u4ede9d87ff916324@mail.gmail.com> References: <367751820907140835y322c576en1eb2ba931388d732@mail.gmail.com> <87f2f3b90907140916o3f008a3br26bfca075a9e3d2e@mail.gmail.com> <367751820907141002g3a837a09meabdbc9743c3bee7@mail.gmail.com> <191c3a030907141020l48e8cb2cm6106ff4442ee1bed@mail.gmail.com> <87f2f3b90907141037r260001d4i2c41dcc0e4c25c5@mail.gmail.com> <367751820907141114w56ae9adey4836238740a1ed43@mail.gmail.com> <191c3a030907141127x255f21c5q7d80cf26ea8451a9@mail.gmail.com> <367751820907141135t2c374338y54226c9721cfa739@mail.gmail.com> <367751820907141522o491b40f4jbf79c25f3c053139@mail.gmail.com> <367751820907150738k4b0fae48u4ede9d87ff916324@mail.gmail.com> Message-ID: <87f2f3b90907150925k448169c0iebf34d9def2aecd2@mail.gmail.com> On Wed, Jul 15, 2009 at 7:38 AM, Phillip Jones wrote: > Hey Guys, > > I took a look at the source that Anthony updated. I see this: > > } else if (!strcasecmp((char *) hi->name, "group_confirm_file")) { > ok = 1; > } else if (!strcasecmp((char *) hi->name, "group_confirm_*clear*_timeout")) > { > ok = 1; > } else if (!strcasecmp((char *) hi->name, "forked_dial")) { > > and: > > if (switch_true(switch_event_get_header(var_event, "group_confirm_*cancel*_timeout"))) > { > oglobals.cancel_timeout = -1; > } > > I updated the *group_confirm_clear_timeout *to * > group_confirm_cancel_timeout* and recompiled and this is now working just > great. > > Thanks very much for incorporating this. It is much appreciated. > > > Phillip Jones > > Phillip, Thank you very much for taking the time and initiative to dig a little. The devs definitely appreciate it when community members roll up their sleeves and do some investigative work. Karma++ for you! :) -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090715/f1618b2d/attachment.html From woof at iwoof.org Wed Jul 15 09:42:41 2009 From: woof at iwoof.org (Andy Spitzer) Date: Wed, 15 Jul 2009 12:42:41 -0400 Subject: [Freeswitch-users] copy and past "oops" in mod_event_socket.c Message-ID: Woof! Too simple to open a JIRA with a patch (and it actually works as written): 1804: } else if (!strncasecmp(cmd, "nolinger", 6)) { That should be an 8 as nolinger is 8 characters long. --Woof! From saeedahmad1981 at gmail.com Wed Jul 15 09:45:12 2009 From: saeedahmad1981 at gmail.com (Saeed Ahmad) Date: Wed, 15 Jul 2009 18:45:12 +0200 Subject: [Freeswitch-users] SIP Trace Option at Runtime In-Reply-To: <45002ACD-36EC-4BBE-8D8F-9F74823938E7@avgs.ca> References: <20090715043225.GA21117@jdc.jasonjgw.net> <45002ACD-36EC-4BBE-8D8F-9F74823938E7@avgs.ca> Message-ID: still a little confusion, what should be set in switch.conf.xml and console.conf.xml when system is in production and we only want to enable/disable logs on runtime. please also consider log files freeswitch.log & freeswitch.xml.fsxml, we don't want to log debug here. So main goal is without restarting FS we should be able to enable/disable logs on runtime (when logs are enabled then its ok to write logs in log files, but on disable it should stop writing) and everything on runtime. Thanks - Saeed On Wed, Jul 15, 2009 at 6:16 PM, Mathieu Rene wrote: > Oh and the default level for "siptrace" is CONSOLE, you can change it > runtime (if you want to log it into log files) by doing sofia tracelevel > [xxx] (where [xxx] is the loglevel) > Mathieu Rene > Avant-Garde Solutions Inc > Office: + 1 (514) 664-1044 x100 > Cell: +1 (514) 664-1044 x200 > mrene at avgs.ca > > > > Am 15-Jul-09 um 12:08 PM schrieb Saeed Ahmad: > > switch.conf.xml > > > On Wed, Jul 15, 2009 at 6:06 PM, Brian West wrote: > >> where are you setting the loglevel to err at? >> >> /b >> >> On Jul 15, 2009, at 11:03 AM, Saeed Ahmad wrote: >> >> > and how can we do the same for console message? >> > >> > I noticed that if >> > >> > >> > >> > then even we press F8 on cli then it won't turn on the log message, >> > is there anyway to enable them even loglevel is set to 'err' >> > >> > - Saeed >> > >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090715/0ff94499/attachment.html From mrene_lists at avgs.ca Wed Jul 15 09:45:48 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Wed, 15 Jul 2009 12:45:48 -0400 Subject: [Freeswitch-users] copy and past "oops" in mod_event_socket.c In-Reply-To: References: Message-ID: Thx Committed revision 14258. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca Am 15-Jul-09 um 12:42 PM schrieb Andy Spitzer: > nolinger From brian at freeswitch.org Wed Jul 15 09:51:42 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 15 Jul 2009 11:51:42 -0500 Subject: [Freeswitch-users] SIP Trace Option at Runtime In-Reply-To: References: <20090715043225.GA21117@jdc.jasonjgw.net> <45002ACD-36EC-4BBE-8D8F-9F74823938E7@avgs.ca> Message-ID: fsctl loglevel xxx - controls core loglevel console loglevel xxx - controls console loglevel If you fsctl loglevel 0 you basically turn it of and console loglevel 8 won't work anymore because you turned it off in the core. /b On Jul 15, 2009, at 11:45 AM, Saeed Ahmad wrote: > still a little confusion, > > what should be set in switch.conf.xml and console.conf.xml when > system is in production and we only want to enable/disable logs on > runtime. please also consider log files freeswitch.log & > freeswitch.xml.fsxml, we don't want to log debug here. > > So main goal is without restarting FS we should be able to enable/ > disable logs on runtime (when logs are enabled then its ok to write > logs in log files, but on disable it should stop writing) and > everything on runtime. > > Thanks > - Saeed > From larclap at yahoo.com Wed Jul 15 09:59:28 2009 From: larclap at yahoo.com (Lars Zeb) Date: Wed, 15 Jul 2009 09:59:28 -0700 Subject: [Freeswitch-users] contrib directory location Message-ID: <007301ca056d$9d6cd550$d8467ff0$@com> What is the address of the contrib directory? I would like to download it and its contents for study. Thanks, Lars -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090715/a53b9d5e/attachment.html From brian at freeswitch.org Wed Jul 15 10:05:46 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 15 Jul 2009 12:05:46 -0500 Subject: [Freeswitch-users] contrib directory location In-Reply-To: <007301ca056d$9d6cd550$d8467ff0$@com> References: <007301ca056d$9d6cd550$d8467ff0$@com> Message-ID: <3D133974-DCA3-40FB-9D83-03E3B4BAD4CB@freeswitch.org> If you have the freeswitch source tarball or svn check out just cd contrib /b On Jul 15, 2009, at 11:59 AM, Lars Zeb wrote: > What is the address of the contrib directory? I would like to > download it and its contents for study. > > Thanks, Lars > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090715/c2921277/attachment.html From saeedahmad1981 at gmail.com Wed Jul 15 10:28:42 2009 From: saeedahmad1981 at gmail.com (Saeed Ahmad) Date: Wed, 15 Jul 2009 19:28:42 +0200 Subject: [Freeswitch-users] SIP Trace Option at Runtime In-Reply-To: References: <20090715043225.GA21117@jdc.jasonjgw.net> <45002ACD-36EC-4BBE-8D8F-9F74823938E7@avgs.ca> Message-ID: fsctl loglevel 8 console loglevel now freeswitch.log is showing what i wanted but nothing on console, i am connected to ./fs_cli in vars.conf.xml i think its nothing to do with that! On Wed, Jul 15, 2009 at 6:51 PM, Brian West wrote: > fsctl loglevel xxx - controls core loglevel > console loglevel xxx - controls console loglevel > > If you fsctl loglevel 0 you basically turn it of and console loglevel > 8 won't work anymore because you turned it off in the core. > > /b > > > On Jul 15, 2009, at 11:45 AM, Saeed Ahmad wrote: > > > still a little confusion, > > > > what should be set in switch.conf.xml and console.conf.xml when > > system is in production and we only want to enable/disable logs on > > runtime. please also consider log files freeswitch.log & > > freeswitch.xml.fsxml, we don't want to log debug here. > > > > So main goal is without restarting FS we should be able to enable/ > > disable logs on runtime (when logs are enabled then its ok to write > > logs in log files, but on disable it should stop writing) and > > everything on runtime. > > > > Thanks > > - Saeed > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090715/88f5efe1/attachment.html From saeedahmad1981 at gmail.com Wed Jul 15 10:29:01 2009 From: saeedahmad1981 at gmail.com (Saeed Ahmad) Date: Wed, 15 Jul 2009 19:29:01 +0200 Subject: [Freeswitch-users] SIP Trace Option at Runtime In-Reply-To: References: <20090715043225.GA21117@jdc.jasonjgw.net> <45002ACD-36EC-4BBE-8D8F-9F74823938E7@avgs.ca> Message-ID: console loglevel 8 too On Wed, Jul 15, 2009 at 7:28 PM, Saeed Ahmad wrote: > fsctl loglevel 8 > console loglevel > now freeswitch.log is showing what i wanted but nothing on console, i am > connected to ./fs_cli > > in vars.conf.xml > > > > i think its nothing to do with that! > > On Wed, Jul 15, 2009 at 6:51 PM, Brian West wrote: > >> fsctl loglevel xxx - controls core loglevel >> console loglevel xxx - controls console loglevel >> >> If you fsctl loglevel 0 you basically turn it of and console loglevel >> 8 won't work anymore because you turned it off in the core. >> >> /b >> >> >> On Jul 15, 2009, at 11:45 AM, Saeed Ahmad wrote: >> >> > still a little confusion, >> > >> > what should be set in switch.conf.xml and console.conf.xml when >> > system is in production and we only want to enable/disable logs on >> > runtime. please also consider log files freeswitch.log & >> > freeswitch.xml.fsxml, we don't want to log debug here. >> > >> > So main goal is without restarting FS we should be able to enable/ >> > disable logs on runtime (when logs are enabled then its ok to write >> > logs in log files, but on disable it should stop writing) and >> > everything on runtime. >> > >> > Thanks >> > - Saeed >> > >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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/20090715/c69d84dc/attachment.html From saeedahmad1981 at gmail.com Wed Jul 15 10:41:39 2009 From: saeedahmad1981 at gmail.com (Saeed Ahmad) Date: Wed, 15 Jul 2009 19:41:39 +0200 Subject: [Freeswitch-users] SIP Trace Option at Runtime In-Reply-To: References: <20090715043225.GA21117@jdc.jasonjgw.net> <45002ACD-36EC-4BBE-8D8F-9F74823938E7@avgs.ca> Message-ID: it works! thanks On Wed, Jul 15, 2009 at 7:29 PM, Saeed Ahmad wrote: > console loglevel 8 > too > > On Wed, Jul 15, 2009 at 7:28 PM, Saeed Ahmad wrote: > >> fsctl loglevel 8 >> console loglevel >> now freeswitch.log is showing what i wanted but nothing on console, i am >> connected to ./fs_cli >> >> in vars.conf.xml >> >> >> >> i think its nothing to do with that! >> >> On Wed, Jul 15, 2009 at 6:51 PM, Brian West wrote: >> >>> fsctl loglevel xxx - controls core loglevel >>> console loglevel xxx - controls console loglevel >>> >>> If you fsctl loglevel 0 you basically turn it of and console loglevel >>> 8 won't work anymore because you turned it off in the core. >>> >>> /b >>> >>> >>> On Jul 15, 2009, at 11:45 AM, Saeed Ahmad wrote: >>> >>> > still a little confusion, >>> > >>> > what should be set in switch.conf.xml and console.conf.xml when >>> > system is in production and we only want to enable/disable logs on >>> > runtime. please also consider log files freeswitch.log & >>> > freeswitch.xml.fsxml, we don't want to log debug here. >>> > >>> > So main goal is without restarting FS we should be able to enable/ >>> > disable logs on runtime (when logs are enabled then its ok to write >>> > logs in log files, but on disable it should stop writing) and >>> > everything on runtime. >>> > >>> > Thanks >>> > - Saeed >>> > >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at 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/20090715/59c5a959/attachment.html From larclap at yahoo.com Wed Jul 15 11:33:33 2009 From: larclap at yahoo.com (Lars Zeb) Date: Wed, 15 Jul 2009 11:33:33 -0700 Subject: [Freeswitch-users] Dialplan with lua - error missing closing angle bracket? Message-ID: <00be01ca057a$c2572340$470569c0$@com> I copied the action element from http://wiki.freeswitch.org/wiki/Lua, "Sample Dialplan". When I try to reloadxml, the cli tells me that there is a missing right angle bracket. +OK [[error near line 3130]: missing >] Do the docs need updating or have I totally blown it? Thanks, Lars -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090715/d9a4ee37/attachment.html From sprice at gmail.com Wed Jul 15 11:47:37 2009 From: sprice at gmail.com (SP) Date: Wed, 15 Jul 2009 13:47:37 -0500 Subject: [Freeswitch-users] Dialplan with lua - error missing closing angle bracket? In-Reply-To: <00be01ca057a$c2572340$470569c0$@com> References: <00be01ca057a$c2572340$470569c0$@com> Message-ID: <7e2ac3270907151147r576eb502j4d9431673a7e8e3@mail.gmail.com> check your whitespace On Wed, Jul 15, 2009 at 13:33, Lars Zeb wrote: > I copied the action element from http://wiki.freeswitch.org/wiki/Lua, > ?Sample Dialplan?. > > > > When I try to reloadxml, the cli tells me that there is a missing right > angle bracket. > > +OK [[error near line 3130]: missing >] > > > > Do the docs need updating or have I totally blown it? > > > > Thanks, Lars > > > > > > > > > > > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090715/fabcc9d6/attachment.html From freeswitch-users at lists.freeswitch.org Wed Jul 15 17:40:18 2009 From: freeswitch-users at lists.freeswitch.org (freeswitch-users at lists.freeswitch.org) Date: Thu, 16 Jul 2009 08:40:18 +0800 Subject: [Freeswitch-users] How to set the IVR of VM menu?? Message-ID: Could you please just tell me where to set it?? >Please try looking on the wiki, this and many other questions should >be answered for you there. > >Mike > >On Jul 15, 2009, at 4:05 AM, Brad Tuan wrote: > >>* How to set the date format , and the IVR flow ........?? *>>* _______________________________________________ *>>* Freeswitch-users mailing list *>>* Freeswitch-users at 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/20090716/05832b1b/attachment.html From freeswitch-users at lists.freeswitch.org Wed Jul 15 22:07:21 2009 From: freeswitch-users at lists.freeswitch.org (freeswitch-users at lists.freeswitch.org) Date: Thu, 16 Jul 2009 01:07:21 -0400 Subject: [Freeswitch-users] Freeswitch architecture In-Reply-To: <200907141618.03295.yivzhenko@mksat.net> References: <89c9bbf80907080542y39f2b234tff9163ea34c0b968@mail.gmail.com> <347F12AB-EA05-4090-A203-BDC011F02411@freeswitch.org> <200907141618.03295.yivzhenko@mksat.net> Message-ID: <20090716050720.GJ28401@hijacked.us> On Tue, Jul 14, 2009 at 04:18:03PM +0300, Yuriy Ivzhenko wrote: > On Wednesday 08 July 2009 16:29:50 Brian West wrote: > > http://wiki.freeswitch.org > > i not found any essential information about architecture :-((((( > .... may be bad looking? I'd recommend just reading the code and looking at some of the simpler modules. Most of FreeSWITCH is remarkably readable (just steer clear of the XML parsing stuff :) ). Andrew From freeswitch-users at lists.freeswitch.org Thu Jul 16 00:47:17 2009 From: freeswitch-users at lists.freeswitch.org (freeswitch-users at lists.freeswitch.org) Date: Thu, 16 Jul 2009 10:47:17 +0300 Subject: [Freeswitch-users] SIP TLS (and SRTP) In-Reply-To: <8594764D-2F2A-431C-BA91-1C2D5A97C90D@freeswitch.org> References: <10128ef10907150257u3ae1c20jb7772b66e8c1a738@mail.gmail.com> <8594764D-2F2A-431C-BA91-1C2D5A97C90D@freeswitch.org> Message-ID: <10128ef10907160047o3ca4dc25o93b58ae97b94bb4c@mail.gmail.com> thanks allot, this was my mistake. /Tzury > It tells you to edit conf/directory/default.xml not dialplan/ > default.xml and put > > > as the dial-string. > > /b From freeswitch-users at lists.freeswitch.org Thu Jul 16 06:30:26 2009 From: freeswitch-users at lists.freeswitch.org (freeswitch-users at lists.freeswitch.org) Date: Thu, 16 Jul 2009 20:30:26 +0700 Subject: [Freeswitch-users] sip extermal profile for all IP Message-ID: <8ccbff060907160630r2188c8d8od0c5c677f81eb71b@mail.gmail.com> Dear All, How to set ${local_ip_v4} to 0.0.0.0 ? i want to set up external profile for All IP Best regards. Dome C. From freeswitch-users at lists.freeswitch.org Thu Jul 16 06:48:02 2009 From: freeswitch-users at lists.freeswitch.org (freeswitch-users at lists.freeswitch.org) Date: Thu, 16 Jul 2009 19:48:02 +0600 Subject: [Freeswitch-users] sip extermal profile for all IP In-Reply-To: <8ccbff060907160630r2188c8d8od0c5c677f81eb71b@mail.gmail.com> References: <8ccbff060907160630r2188c8d8od0c5c677f81eb71b@mail.gmail.com> Message-ID: I think by default it maps to all IPs unless you mention one here (by replacing ${local_ip_v4} to some ip address). Thank you. On Thu, Jul 16, 2009 at 7:30 PM, wrote: > Dear All, > > How to set ${local_ip_v4} to 0.0.0.0 ? i want to set up > external profile for All IP > > > Best regards. > > Dome C. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090716/32926e09/attachment.html From freeswitch-users at lists.freeswitch.org Thu Jul 16 07:02:46 2009 From: freeswitch-users at lists.freeswitch.org (freeswitch-users at lists.freeswitch.org) Date: Thu, 16 Jul 2009 17:02:46 +0300 Subject: [Freeswitch-users] "show channels" command with duration - patch included Message-ID: <4A5F3306.1000300@kinetix.gr> Hi, I usually find it very useful when I can retrieve a list of the currents calls along with durations. I noticed that the 'show channels' format does not include the duration (or the answered timestamp - so that one can extract it from there). So, I made a patch that includes the answered timestamp, the answered timestamp in epoch, and the duration in seconds. Of course these fields remain empty when the call hasn't been answered yet. I don't know if anyone else finds this functionality useful, so I am posting this patch here first (instead of JIRA) in order to get feedback from the users. If many of you (or the maintainers) find it interesting I can then proceed in posting it to JIRA. -- ------------------------------------------- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: regs at kinetix.gr ------------------------------------------- -------------- next part -------------- Index: src/mod/applications/mod_commands/mod_commands.c =================================================================== --- src/mod/applications/mod_commands/mod_commands.c (revision 14256) +++ src/mod/applications/mod_commands/mod_commands.c (working copy) @@ -2827,10 +2827,10 @@ } } if (strchr(argv[2], '%')) { - sprintf(sql, "select * from channels where uuid like '%s' or name like '%s' or cid_name like '%s' or cid_num like '%s' order by created_epoch", + sprintf(sql, "select *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration from channels where uuid like '%s' or name like '%s' or cid_name like '%s' or cid_num like '%s' order by created_epoch", argv[2], argv[2], argv[2], argv[2]); } else { - sprintf(sql, "select * from channels where uuid like '%%%s%%' or name like '%%%s%%' or cid_name like '%%%s%%' or cid_num like '%%%s%%' order by created_epoch", + sprintf(sql, "select *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration from channels where uuid like '%%%s%%' or name like '%%%s%%' or cid_name like '%%%s%%' or cid_num like '%%%s%%' order by created_epoch", argv[2], argv[2], argv[2], argv[2]); } @@ -2839,10 +2839,10 @@ as = argv[4]; } } else { - sprintf(sql, "select * from channels order by created_epoch"); + sprintf(sql, "select *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration from channels order by created_epoch"); } } else if (!strcasecmp(command, "channels")) { - sprintf(sql, "select * from channels order by created_epoch"); + sprintf(sql, "select *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration from channels order by created_epoch"); if (argv[1] && !strcasecmp(argv[1],"count")) { holder.justcount = 1; if (argv[3] && !strcasecmp(argv[2], "as")) { @@ -2850,7 +2850,7 @@ } } } else if (!strcasecmp(command, "distinct_channels")) { - sprintf(sql, "select * from channels left join calls on " + sprintf(sql, "select *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration from channels left join calls on " "channels.uuid=calls.caller_uuid where channels.uuid not in (select callee_uuid from calls) order by created_epoch"); if (argv[2] && !strcasecmp(argv[1], "as")) { as = argv[2]; Index: src/switch_core_sqldb.c =================================================================== --- src/switch_core_sqldb.c (revision 14256) +++ src/switch_core_sqldb.c (working copy) @@ -309,9 +309,21 @@ ); break; + case SWITCH_EVENT_CHANNEL_ANSWER: + { + + sql = switch_mprintf("update channels set answered='%s',answered_epoch='%ld' where uuid='%s'", + switch_event_get_header_nil(event, "event-date-local"), + (long)switch_epoch_time_now(NULL), + switch_event_get_header_nil(event, "unique-id") + ); + + } + break; case SWITCH_EVENT_CHANNEL_STATE: { char *state = switch_event_get_header_nil(event, "channel-state-number"); + switch_channel_state_t state_i = CS_DESTROY; if (!switch_strlen_zero(state)) { @@ -492,7 +504,9 @@ " read_rate VARCHAR(255),\n" " write_codec VARCHAR(255),\n" " write_rate VARCHAR(255),\n" - " secure VARCHAR(255)\n" + " secure VARCHAR(255),\n" + " answered VARCHAR(255),\n" + " answered_epoch INTEGER\n" ");\ncreate index uuindex on channels (uuid);\n"; char create_calls_sql[] = "CREATE TABLE calls (\n" From anthony.minessale at gmail.com Thu Jul 16 07:15:34 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 16 Jul 2009 09:15:34 -0500 Subject: [Freeswitch-users] "show channels" command with duration - patch included In-Reply-To: <4A5F3306.1000300@kinetix.gr> References: <4A5F3306.1000300@kinetix.gr> Message-ID: <191c3a030907160715k74e785c8j99b68a42dc019bb7@mail.gmail.com> I'm ok with the idea as long as it's thoroughly tested. If there is any more info you want to save from those events you should consider it now while we are modifying it. On Thu, Jul 16, 2009 at 9:02 AM, wrote: > Hi, > > I usually find it very useful when I can retrieve a list of the currents > calls along with durations. I noticed that the 'show channels' format does > not include the duration (or the answered timestamp - so that one can > extract it from there). So, I made a patch that includes the answered > timestamp, the answered timestamp in epoch, and the duration in seconds. Of > course these fields remain empty when the call hasn't been > answered yet. > > I don't know if anyone else finds this functionality useful, so I am > posting this patch here first (instead of JIRA) in order to get feedback > from the users. If many of you (or the maintainers) find it interesting I > can then proceed in posting it to JIRA. > > -- > ------------------------------------------- > Apostolos Pantsiopoulos > Kinetix Tele.com R & D > email: regs at kinetix.gr > ------------------------------------------- > > Index: src/mod/applications/mod_commands/mod_commands.c > =================================================================== > --- src/mod/applications/mod_commands/mod_commands.c (revision 14256) > +++ src/mod/applications/mod_commands/mod_commands.c (working copy) > @@ -2827,10 +2827,10 @@ > } > } > if (strchr(argv[2], '%')) { > - sprintf(sql, "select * from channels where > uuid like '%s' or name like '%s' or cid_name like '%s' or cid_num like '%s' > order by created_epoch", > + sprintf(sql, "select > *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration from channels > where uuid like '%s' or name like '%s' or cid_name like '%s' or cid_num like > '%s' order by created_epoch", > argv[2], argv[2], argv[2], > argv[2]); > } else { > - sprintf(sql, "select * from channels where > uuid like '%%%s%%' or name like '%%%s%%' or cid_name like '%%%s%%' or > cid_num like '%%%s%%' order by created_epoch", > + sprintf(sql, "select > *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration from channels > where uuid like '%%%s%%' or name like '%%%s%%' or cid_name like '%%%s%%' or > cid_num like '%%%s%%' order by created_epoch", > argv[2], argv[2], argv[2], > argv[2]); > > } > @@ -2839,10 +2839,10 @@ > as = argv[4]; > } > } else { > - sprintf(sql, "select * from channels order by > created_epoch"); > + sprintf(sql, "select > *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration from channels > order by created_epoch"); > } > } else if (!strcasecmp(command, "channels")) { > - sprintf(sql, "select * from channels order by > created_epoch"); > + sprintf(sql, "select > *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration from channels > order by created_epoch"); > if (argv[1] && !strcasecmp(argv[1],"count")) { > holder.justcount = 1; > if (argv[3] && !strcasecmp(argv[2], "as")) { > @@ -2850,7 +2850,7 @@ > } > } > } else if (!strcasecmp(command, "distinct_channels")) { > - sprintf(sql, "select * from channels left join calls on " > + sprintf(sql, "select > *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration from channels > left join calls on " > "channels.uuid=calls.caller_uuid where > channels.uuid not in (select callee_uuid from calls) order by > created_epoch"); > if (argv[2] && !strcasecmp(argv[1], "as")) { > as = argv[2]; > Index: src/switch_core_sqldb.c > =================================================================== > --- src/switch_core_sqldb.c (revision 14256) > +++ src/switch_core_sqldb.c (working copy) > @@ -309,9 +309,21 @@ > ); > > break; > + case SWITCH_EVENT_CHANNEL_ANSWER: > + { > + > + sql = switch_mprintf("update channels set > answered='%s',answered_epoch='%ld' where uuid='%s'", > + > switch_event_get_header_nil(event, "event-date-local"), > + > (long)switch_epoch_time_now(NULL), > + > switch_event_get_header_nil(event, "unique-id") > + ); > + > + } > + break; > case SWITCH_EVENT_CHANNEL_STATE: > { > char *state = switch_event_get_header_nil(event, > "channel-state-number"); > + > switch_channel_state_t state_i = CS_DESTROY; > > if (!switch_strlen_zero(state)) { > @@ -492,7 +504,9 @@ > " read_rate VARCHAR(255),\n" > " write_codec VARCHAR(255),\n" > " write_rate VARCHAR(255),\n" > - " secure VARCHAR(255)\n" > + " secure VARCHAR(255),\n" > + " answered VARCHAR(255),\n" > + " answered_epoch INTEGER\n" > ");\ncreate index uuindex on channels (uuid);\n"; > char create_calls_sql[] = > "CREATE TABLE calls (\n" > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference 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/20090716/ef6b146b/attachment.html From Prometheus001 at gmx.net Thu Jul 16 07:25:23 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Thu, 16 Jul 2009 16:25:23 +0200 Subject: [Freeswitch-users] "show channels" command with duration - patch included In-Reply-To: <4A5F3306.1000300@kinetix.gr> References: <4A5F3306.1000300@kinetix.gr> Message-ID: <4A5F3853.1030005@gmx.net> Well, that's very useful for us in order to have this info in our FS Operator panel. Best regards Peter freeswitch-users at lists.freeswitch.org schrieb: > Hi, > > I usually find it very useful when I can retrieve a list of the > currents calls along with durations. I noticed that the 'show > channels' format does not include the duration (or the answered > timestamp - so that one can extract it from there). So, I made a patch > that includes the answered timestamp, the answered timestamp in epoch, > and the duration in seconds. Of course these fields remain empty when > the call hasn't been > answered yet. > > I don't know if anyone else finds this functionality useful, so I am > posting this patch here first (instead of JIRA) in order to get > feedback from the users. If many of you (or the maintainers) find it > interesting I can then proceed in posting it to JIRA. > > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From fdhege at gmail.com Thu Jul 16 07:30:34 2009 From: fdhege at gmail.com (Dale) Date: Thu, 16 Jul 2009 10:30:34 -0400 Subject: [Freeswitch-users] Setting P-Asserted-ID to something other than the callerid Message-ID: <723C30E7-4AD0-418D-A4B4-10148367A9AA@gmail.com> Hello again, I wanted to first say thanks to Brain for helping me fix my from domain issue the other day. It helped quite a bit. Now with more testing and talking with the vendor (please don't shoot the messenger :) ) They want the caller id info in the from and the charge number/ screening number in the P-Asserted-ID. I have tested this and verified that this does work like they say it does by setting the callerid number to my charge number and setting the from user in the gateway config to the callerid I want displayed. But this solution doesn't scale very well. I know I can set the gateway option caller-id-in-from to get that part done. But is there a way to set the P-Asserted-ID to something other than the callerid? Any hints would be welcomed. Thanks, -Dale From brian at freeswitch.org Thu Jul 16 07:37:02 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 16 Jul 2009 09:37:02 -0500 Subject: [Freeswitch-users] Setting P-Asserted-ID to something other than the callerid In-Reply-To: <723C30E7-4AD0-418D-A4B4-10148367A9AA@gmail.com> References: <723C30E7-4AD0-418D-A4B4-10148367A9AA@gmail.com> Message-ID: <6BAA8BE5-BFA9-47C9-A1FC-651AAA303E6E@freeswitch.org> On Jul 16, 2009, at 9:30 AM, Dale wrote: > They want the caller id info in the from and the charge number/ > screening number in the P-Asserted-ID. > set the sip_h_P-Asserted-ID=contents of header its just a variable you need to set now. > > > -Dale From regs at kinetix.gr Thu Jul 16 07:37:13 2009 From: regs at kinetix.gr (Apostolos Pantsiopoulos) Date: Thu, 16 Jul 2009 17:37:13 +0300 Subject: [Freeswitch-users] "show channels" command with duration - patch included In-Reply-To: <191c3a030907160715k74e785c8j99b68a42dc019bb7@mail.gmail.com> References: <4A5F3306.1000300@kinetix.gr> <191c3a030907160715k74e785c8j99b68a42dc019bb7@mail.gmail.com> Message-ID: <4A5F3B19.8020507@kinetix.gr> Now that I come to think of it... It would be useful if we had the timestamp (and epoch) of the events PROGRESS and PROGRESS WITH MEDIA so that we can extract the PDD (Post Dial Delay) which is a very useful statistic. Adding the user's (for the incoming) and the gateway's (for the outbound) id would also be useful. In case these fields are empty the show channels command could ommit the (empty string). Are you planning to implement them yourselves or should I begin looking at the code? Anthony Minessale wrote: > I'm ok with the idea as long as it's thoroughly tested. > If there is any more info you want to save from those events you should > consider it now while we are modifying it. > > > On Thu, Jul 16, 2009 at 9:02 AM, > wrote: > > Hi, > > I usually find it very useful when I can retrieve a list of the > currents calls along with durations. I noticed that the 'show > channels' format does not include the duration (or the answered > timestamp - so that one can extract it from there). So, I made a > patch that includes the answered timestamp, the answered timestamp > in epoch, and the duration in seconds. Of course these fields remain > empty when the call hasn't been > answered yet. > > I don't know if anyone else finds this functionality useful, so I am > posting this patch here first (instead of JIRA) in order to get > feedback from the users. If many of you (or the maintainers) find it > interesting I can then proceed in posting it to JIRA. > > -- > ------------------------------------------- > Apostolos Pantsiopoulos > Kinetix Tele.com R & D > email: regs at kinetix.gr > ------------------------------------------- > > Index: src/mod/applications/mod_commands/mod_commands.c > =================================================================== > --- src/mod/applications/mod_commands/mod_commands.c (revision 14256) > +++ src/mod/applications/mod_commands/mod_commands.c (working copy) > @@ -2827,10 +2827,10 @@ > } > } > if (strchr(argv[2], '%')) { > - sprintf(sql, "select * from channels > where uuid like '%s' or name like '%s' or cid_name like '%s' or > cid_num like '%s' order by created_epoch", > + sprintf(sql, "select > *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration from > channels where uuid like '%s' or name like '%s' or cid_name like > '%s' or cid_num like '%s' order by created_epoch", > argv[2], argv[2], > argv[2], argv[2]); > } else { > - sprintf(sql, "select * from channels > where uuid like '%%%s%%' or name like '%%%s%%' or cid_name like > '%%%s%%' or cid_num like '%%%s%%' order by created_epoch", > + sprintf(sql, "select > *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration from > channels where uuid like '%%%s%%' or name like '%%%s%%' or cid_name > like '%%%s%%' or cid_num like '%%%s%%' order by created_epoch", > argv[2], argv[2], > argv[2], argv[2]); > > } > @@ -2839,10 +2839,10 @@ > as = argv[4]; > } > } else { > - sprintf(sql, "select * from channels order > by created_epoch"); > + sprintf(sql, "select > *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration from > channels order by created_epoch"); > } > } else if (!strcasecmp(command, "channels")) { > - sprintf(sql, "select * from channels order by > created_epoch"); > + sprintf(sql, "select > *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration from > channels order by created_epoch"); > if (argv[1] && !strcasecmp(argv[1],"count")) { > holder.justcount = 1; > if (argv[3] && !strcasecmp(argv[2], "as")) { > @@ -2850,7 +2850,7 @@ > } > } > } else if (!strcasecmp(command, "distinct_channels")) { > - sprintf(sql, "select * from channels left join calls > on " > + sprintf(sql, "select > *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration from > channels left join calls on " > "channels.uuid=calls.caller_uuid > where channels.uuid not in (select callee_uuid from calls) order by > created_epoch"); > if (argv[2] && !strcasecmp(argv[1], "as")) { > as = argv[2]; > Index: src/switch_core_sqldb.c > =================================================================== > --- src/switch_core_sqldb.c (revision 14256) > +++ src/switch_core_sqldb.c (working copy) > @@ -309,9 +309,21 @@ > ); > > break; > + case SWITCH_EVENT_CHANNEL_ANSWER: > + { > + > + sql = switch_mprintf("update channels set > answered='%s',answered_epoch='%ld' where uuid='%s'", > + > switch_event_get_header_nil(event, "event-date-local"), > + > (long)switch_epoch_time_now(NULL), > + > switch_event_get_header_nil(event, "unique-id") > + ); > + > + } > + break; > case SWITCH_EVENT_CHANNEL_STATE: > { > char *state = > switch_event_get_header_nil(event, "channel-state-number"); > + > switch_channel_state_t state_i = CS_DESTROY; > > if (!switch_strlen_zero(state)) { > @@ -492,7 +504,9 @@ > " read_rate VARCHAR(255),\n" > " write_codec VARCHAR(255),\n" > " write_rate VARCHAR(255),\n" > - " secure VARCHAR(255)\n" > + " secure VARCHAR(255),\n" > + " answered VARCHAR(255),\n" > + " answered_epoch INTEGER\n" > ");\ncreate index uuindex on channels (uuid);\n"; > char create_calls_sql[] = > "CREATE TABLE calls (\n" > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > 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 -- ------------------------------------------- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: regs at kinetix.gr ------------------------------------------- From krice at suspicious.org Thu Jul 16 07:41:25 2009 From: krice at suspicious.org (Ken Rice) Date: Thu, 16 Jul 2009 09:41:25 -0500 Subject: [Freeswitch-users] Setting P-Asserted-ID to something other than the callerid In-Reply-To: <723C30E7-4AD0-418D-A4B4-10148367A9AA@gmail.com> Message-ID: That's just right down screwing with the standards... PAID is the caller id... This particular definition is from the RFCs and 3GPP docs for IMS which is why we have standardized P- headers... Can your vendor not look at the P-Charging-Vector field? Also, From when used with PAID is more like an ANI not a CLID > From: Dale > Reply-To: > Date: Thu, 16 Jul 2009 10:30:34 -0400 > To: > Subject: [Freeswitch-users] Setting P-Asserted-ID to something other than the > callerid > > > Hello again, > > I wanted to first say thanks to Brain for helping me fix my from > domain issue the other day. It helped quite a bit. > > Now with more testing and talking with the vendor (please don't shoot > the messenger :) ) > > They want the caller id info in the from and the charge number/ > screening number in the P-Asserted-ID. > > I have tested this and verified that this does work like they say it > does by setting the callerid number to my charge number and setting > the from user in the gateway config to the callerid I want displayed. > But this solution doesn't scale very well. > > I know I can set the gateway option caller-id-in-from to get that part > done. But is there a way to set the P-Asserted-ID to something other > than the callerid? > > Any hints would be welcomed. > > Thanks, > > -Dale > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.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 Jul 16 07:43:45 2009 From: msc at freeswitch.org (Michael S Collins) Date: Thu, 16 Jul 2009 07:43:45 -0700 Subject: [Freeswitch-users] How to set the IVR of VM menu?? Message-ID: On Jul 15, 2009, at 5:40 PM, freeswitch-users at lists.freeswitch.org wrote: > Could you please just tell me where to set it?? The menu actions are defined in conf/autoload_configs/ivr.conf.xml The audio played for the menus is defined in conf/lang/en/demo/demo- ivr.xml -MC > > >Please try looking on the wiki, this and many other questions should > >be answered for you there. > > > >Mike > > > >On Jul 15, 2009, at 4:05 AM, Brad Tuan wrote: > > > >> How to set the date format , and the IVR flow ........?? > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20090716/50bf7815/attachment.html From anthony.minessale at gmail.com Thu Jul 16 07:46:29 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 16 Jul 2009 09:46:29 -0500 Subject: [Freeswitch-users] "show channels" command with duration - patch included In-Reply-To: <4A5F3B19.8020507@kinetix.gr> References: <4A5F3306.1000300@kinetix.gr> <191c3a030907160715k74e785c8j99b68a42dc019bb7@mail.gmail.com> <4A5F3B19.8020507@kinetix.gr> Message-ID: <191c3a030907160746p3ed81c31s58caf5fb39c2a47a@mail.gmail.com> I wasn't planning on implementing it but I was just mentioning that if you were going to do your patch, consider if there is any other info to store while the patient is on the operating table. The line we should not cross is to store all the info in the table since, really, you could be collecting those events in your application as well to store that info. But for the casual user, some more fields may be interesting. On Thu, Jul 16, 2009 at 9:37 AM, Apostolos Pantsiopoulos wrote: > Now that I come to think of it... > > It would be useful if we had the timestamp (and epoch) > of the events PROGRESS and PROGRESS WITH MEDIA so that we can extract > the PDD (Post Dial Delay) which is a very useful statistic. > > Adding the user's (for the incoming) and the gateway's (for the > outbound) id would also be useful. In case these fields are empty the > show channels command could ommit the (empty string). > > Are you planning to implement them yourselves or should I begin looking > at the code? > > > Anthony Minessale wrote: > > I'm ok with the idea as long as it's thoroughly tested. > > If there is any more info you want to save from those events you should > > consider it now while we are modifying it. > > > > > > On Thu, Jul 16, 2009 at 9:02 AM, > > wrote: > > > > Hi, > > > > I usually find it very useful when I can retrieve a list of the > > currents calls along with durations. I noticed that the 'show > > channels' format does not include the duration (or the answered > > timestamp - so that one can extract it from there). So, I made a > > patch that includes the answered timestamp, the answered timestamp > > in epoch, and the duration in seconds. Of course these fields remain > > empty when the call hasn't been > > answered yet. > > > > I don't know if anyone else finds this functionality useful, so I am > > posting this patch here first (instead of JIRA) in order to get > > feedback from the users. If many of you (or the maintainers) find it > > interesting I can then proceed in posting it to JIRA. > > > > -- > > ------------------------------------------- > > Apostolos Pantsiopoulos > > Kinetix Tele.com R & D > > email: regs at kinetix.gr > > ------------------------------------------- > > > > Index: src/mod/applications/mod_commands/mod_commands.c > > =================================================================== > > --- src/mod/applications/mod_commands/mod_commands.c (revision > 14256) > > +++ src/mod/applications/mod_commands/mod_commands.c (working > copy) > > @@ -2827,10 +2827,10 @@ > > } > > } > > if (strchr(argv[2], '%')) { > > - sprintf(sql, "select * from channels > > where uuid like '%s' or name like '%s' or cid_name like '%s' or > > cid_num like '%s' order by created_epoch", > > + sprintf(sql, "select > > *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration from > > channels where uuid like '%s' or name like '%s' or cid_name like > > '%s' or cid_num like '%s' order by created_epoch", > > argv[2], argv[2], > > argv[2], argv[2]); > > } else { > > - sprintf(sql, "select * from channels > > where uuid like '%%%s%%' or name like '%%%s%%' or cid_name like > > '%%%s%%' or cid_num like '%%%s%%' order by created_epoch", > > + sprintf(sql, "select > > *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration from > > channels where uuid like '%%%s%%' or name like '%%%s%%' or cid_name > > like '%%%s%%' or cid_num like '%%%s%%' order by created_epoch", > > argv[2], argv[2], > > argv[2], argv[2]); > > > > } > > @@ -2839,10 +2839,10 @@ > > as = argv[4]; > > } > > } else { > > - sprintf(sql, "select * from channels order > > by created_epoch"); > > + sprintf(sql, "select > > *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration from > > channels order by created_epoch"); > > } > > } else if (!strcasecmp(command, "channels")) { > > - sprintf(sql, "select * from channels order by > > created_epoch"); > > + sprintf(sql, "select > > *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration from > > channels order by created_epoch"); > > if (argv[1] && !strcasecmp(argv[1],"count")) { > > holder.justcount = 1; > > if (argv[3] && !strcasecmp(argv[2], "as")) { > > @@ -2850,7 +2850,7 @@ > > } > > } > > } else if (!strcasecmp(command, "distinct_channels")) { > > - sprintf(sql, "select * from channels left join calls > > on " > > + sprintf(sql, "select > > *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration from > > channels left join calls on " > > "channels.uuid=calls.caller_uuid > > where channels.uuid not in (select callee_uuid from calls) order by > > created_epoch"); > > if (argv[2] && !strcasecmp(argv[1], "as")) { > > as = argv[2]; > > Index: src/switch_core_sqldb.c > > =================================================================== > > --- src/switch_core_sqldb.c (revision 14256) > > +++ src/switch_core_sqldb.c (working copy) > > @@ -309,9 +309,21 @@ > > ); > > > > break; > > + case SWITCH_EVENT_CHANNEL_ANSWER: > > + { > > + > > + sql = switch_mprintf("update channels set > > answered='%s',answered_epoch='%ld' where uuid='%s'", > > + > > switch_event_get_header_nil(event, "event-date-local"), > > + > > (long)switch_epoch_time_now(NULL), > > + > > switch_event_get_header_nil(event, "unique-id") > > + ); > > + > > + } > > + break; > > case SWITCH_EVENT_CHANNEL_STATE: > > { > > char *state = > > switch_event_get_header_nil(event, "channel-state-number"); > > + > > switch_channel_state_t state_i = CS_DESTROY; > > > > if (!switch_strlen_zero(state)) { > > @@ -492,7 +504,9 @@ > > " read_rate VARCHAR(255),\n" > > " write_codec VARCHAR(255),\n" > > " write_rate VARCHAR(255),\n" > > - " secure VARCHAR(255)\n" > > + " secure VARCHAR(255),\n" > > + " answered VARCHAR(255),\n" > > + " answered_epoch INTEGER\n" > > ");\ncreate index uuindex on channels > (uuid);\n"; > > char create_calls_sql[] = > > "CREATE TABLE calls (\n" > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com > > > > > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > > > > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > 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 > > > -- > ------------------------------------------- > 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 > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference 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/20090716/9fdb77aa/attachment.html From fdhege at gmail.com Thu Jul 16 07:48:24 2009 From: fdhege at gmail.com (Dale) Date: Thu, 16 Jul 2009 10:48:24 -0400 Subject: [Freeswitch-users] Setting P-Asserted-ID to something other than the callerid In-Reply-To: <6BAA8BE5-BFA9-47C9-A1FC-651AAA303E6E@freeswitch.org> References: <723C30E7-4AD0-418D-A4B4-10148367A9AA@gmail.com> <6BAA8BE5-BFA9-47C9-A1FC-651AAA303E6E@freeswitch.org> Message-ID: <98E42E2F-642A-41AD-A6FC-7D92B412F1A1@gmail.com> That works perfectly. Thanks, -Dale On Jul 16, 2009, at 10:37 AM, Brian West wrote: > > On Jul 16, 2009, at 9:30 AM, Dale wrote: > >> They want the caller id info in the from and the charge number/ >> screening number in the P-Asserted-ID. >> > > set the sip_h_P-Asserted-ID=contents of header > > its just a variable you need to set now. >> >> >> -Dale > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.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 Jul 16 07:50:28 2009 From: msc at freeswitch.org (Michael S Collins) Date: Thu, 16 Jul 2009 07:50:28 -0700 Subject: [Freeswitch-users] "show channels" command with duration - patch included In-Reply-To: <4A5F3B19.8020507@kinetix.gr> References: <4A5F3306.1000300@kinetix.gr> <191c3a030907160715k74e785c8j99b68a42dc019bb7@mail.gmail.com> <4A5F3B19.8020507@kinetix.gr> Message-ID: <6079D960-F5EB-4774-93DA-87402CFC1157@freeswitch.org> I wonder if it would make sense to create a separate sub-command like "show channels stats" or something. That way we could put all sorts of nifty info there without breaking the existing command. Thoughts? -MC Sent from my iPhone On Jul 16, 2009, at 7:37 AM, Apostolos Pantsiopoulos wrote: > Now that I come to think of it... > > It would be useful if we had the timestamp (and epoch) > of the events PROGRESS and PROGRESS WITH MEDIA so that we can extract > the PDD (Post Dial Delay) which is a very useful statistic. > > Adding the user's (for the incoming) and the gateway's (for the > outbound) id would also be useful. In case these fields are empty the > show channels command could ommit the (empty string). > > Are you planning to implement them yourselves or should I begin > looking > at the code? > > > Anthony Minessale wrote: >> I'm ok with the idea as long as it's thoroughly tested. >> If there is any more info you want to save from those events you >> should >> consider it now while we are modifying it. >> >> >> On Thu, Jul 16, 2009 at 9:02 AM, > > wrote: >> >> Hi, >> >> I usually find it very useful when I can retrieve a list of the >> currents calls along with durations. I noticed that the 'show >> channels' format does not include the duration (or the answered >> timestamp - so that one can extract it from there). So, I made a >> patch that includes the answered timestamp, the answered timestamp >> in epoch, and the duration in seconds. Of course these fields >> remain >> empty when the call hasn't been >> answered yet. >> >> I don't know if anyone else finds this functionality useful, so >> I am >> posting this patch here first (instead of JIRA) in order to get >> feedback from the users. If many of you (or the maintainers) >> find it >> interesting I can then proceed in posting it to JIRA. >> >> -- >> ------------------------------------------- >> Apostolos Pantsiopoulos >> Kinetix Tele.com R & D >> email: regs at kinetix.gr >> ------------------------------------------- >> >> Index: src/mod/applications/mod_commands/mod_commands.c >> >> =================================================================== >> --- src/mod/applications/mod_commands/mod_commands.c >> (revision 14256) >> +++ src/mod/applications/mod_commands/mod_commands.c (working >> copy) >> @@ -2827,10 +2827,10 @@ >> } >> } >> if (strchr(argv[2], '%')) { >> - sprintf(sql, "select * from >> channels >> where uuid like '%s' or name like '%s' or cid_name like '%s' or >> cid_num like '%s' order by created_epoch", >> + sprintf(sql, "select >> *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration from >> channels where uuid like '%s' or name like '%s' or cid_name like >> '%s' or cid_num like '%s' order by created_epoch", >> argv[2], argv[2], >> argv[2], argv[2]); >> } else { >> - sprintf(sql, "select * from >> channels >> where uuid like '%%%s%%' or name like '%%%s%%' or cid_name like >> '%%%s%%' or cid_num like '%%%s%%' order by created_epoch", >> + sprintf(sql, "select >> *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration from >> channels where uuid like '%%%s%%' or name like '%%%s%%' or >> cid_name >> like '%%%s%%' or cid_num like '%%%s%%' order by created_epoch", >> argv[2], argv[2], >> argv[2], argv[2]); >> >> } >> @@ -2839,10 +2839,10 @@ >> as = argv[4]; >> } >> } else { >> - sprintf(sql, "select * from channels order >> by created_epoch"); >> + sprintf(sql, "select >> *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration from >> channels order by created_epoch"); >> } >> } else if (!strcasecmp(command, "channels")) { >> - sprintf(sql, "select * from channels order by >> created_epoch"); >> + sprintf(sql, "select >> *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration from >> channels order by created_epoch"); >> if (argv[1] && !strcasecmp(argv[1],"count")) { >> holder.justcount = 1; >> if (argv[3] && !strcasecmp(argv[2], "as")) { >> @@ -2850,7 +2850,7 @@ >> } >> } >> } else if (!strcasecmp(command, "distinct_channels")) { >> - sprintf(sql, "select * from channels left join >> calls >> on " >> + sprintf(sql, "select >> *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration from >> channels left join calls on " >> "channels.uuid=calls.caller_uuid >> where channels.uuid not in (select callee_uuid from calls) order >> by >> created_epoch"); >> if (argv[2] && !strcasecmp(argv[1], "as")) { >> as = argv[2]; >> Index: src/switch_core_sqldb.c >> >> =================================================================== >> --- src/switch_core_sqldb.c (revision 14256) >> +++ src/switch_core_sqldb.c (working copy) >> @@ -309,9 +309,21 @@ >> ); >> >> break; >> + case SWITCH_EVENT_CHANNEL_ANSWER: >> + { >> + >> + sql = switch_mprintf("update channels set >> answered='%s',answered_epoch='%ld' where uuid='%s'", >> + >> switch_event_get_header_nil(event, "event-date-local"), >> + >> (long)switch_epoch_time_now(NULL), >> + >> switch_event_get_header_nil(event, "unique-id") >> + ); >> + >> + } >> + break; >> case SWITCH_EVENT_CHANNEL_STATE: >> { >> char *state = >> switch_event_get_header_nil(event, "channel-state-number"); >> + >> switch_channel_state_t state_i = >> CS_DESTROY; >> >> if (!switch_strlen_zero(state)) { >> @@ -492,7 +504,9 @@ >> " read_rate VARCHAR(255),\n" >> " write_codec VARCHAR(255),\n" >> " write_rate VARCHAR(255),\n" >> - " secure VARCHAR(255)\n" >> + " secure VARCHAR(255),\n" >> + " answered VARCHAR(255),\n" >> + " answered_epoch INTEGER\n" >> ");\ncreate index uuindex on channels >> (uuid);\n"; >> char create_calls_sql[] = >> "CREATE TABLE calls (\n" >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> 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 > > > -- > ------------------------------------------- > 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 From dave at 3c.co.uk Thu Jul 16 07:51:18 2009 From: dave at 3c.co.uk (David Knell) Date: Thu, 16 Jul 2009 11:51:18 -0300 Subject: [Freeswitch-users] Setting P-Asserted-ID to something other than the callerid In-Reply-To: <723C30E7-4AD0-418D-A4B4-10148367A9AA@gmail.com> References: <723C30E7-4AD0-418D-A4B4-10148367A9AA@gmail.com> Message-ID: <1247755878.4289.64.camel@dk-d820> Hi Dale, You can set the header to anything you like by including something along the lines of in your dialplan. Cheers -- Dave > Hello again, > > I wanted to first say thanks to Brain for helping me fix my from > domain issue the other day. It helped quite a bit. > > Now with more testing and talking with the vendor (please don't shoot > the messenger :) ) > > They want the caller id info in the from and the charge number/ > screening number in the P-Asserted-ID. > > I have tested this and verified that this does work like they say it > does by setting the callerid number to my charge number and setting > the from user in the gateway config to the callerid I want displayed. > But this solution doesn't scale very well. > > I know I can set the gateway option caller-id-in-from to get that part > done. But is there a way to set the P-Asserted-ID to something other > than the callerid? > > Any hints would be welcomed. > > Thanks, > > -Dale > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- David Knell, Director, 3C Limited T: +44 20 3298 2000 E: dave at 3c.co.uk W: http://www.3c.co.uk From anthony.minessale at gmail.com Thu Jul 16 07:54:55 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 16 Jul 2009 09:54:55 -0500 Subject: [Freeswitch-users] "show channels" command with duration - patch included In-Reply-To: <6079D960-F5EB-4774-93DA-87402CFC1157@freeswitch.org> References: <4A5F3306.1000300@kinetix.gr> <191c3a030907160715k74e785c8j99b68a42dc019bb7@mail.gmail.com> <4A5F3B19.8020507@kinetix.gr> <6079D960-F5EB-4774-93DA-87402CFC1157@freeswitch.org> Message-ID: <191c3a030907160754g62aac895ge10a90476d5e890@mail.gmail.com> it doesn't really break anything to add more fields besides the ability to read it but it's already fairly wide as it is. Thats why we have "show channels as xml" On Thu, Jul 16, 2009 at 9:50 AM, Michael S Collins wrote: > I wonder if it would make sense to create a separate sub-command like > "show channels stats" or something. That way we could put all sorts of > nifty info there without breaking the existing command. > > Thoughts? > -MC > > Sent from my iPhone > > On Jul 16, 2009, at 7:37 AM, Apostolos Pantsiopoulos > wrote: > > > Now that I come to think of it... > > > > It would be useful if we had the timestamp (and epoch) > > of the events PROGRESS and PROGRESS WITH MEDIA so that we can extract > > the PDD (Post Dial Delay) which is a very useful statistic. > > > > Adding the user's (for the incoming) and the gateway's (for the > > outbound) id would also be useful. In case these fields are empty the > > show channels command could ommit the (empty string). > > > > Are you planning to implement them yourselves or should I begin > > looking > > at the code? > > > > > > Anthony Minessale wrote: > >> I'm ok with the idea as long as it's thoroughly tested. > >> If there is any more info you want to save from those events you > >> should > >> consider it now while we are modifying it. > >> > >> > >> On Thu, Jul 16, 2009 at 9:02 AM, >> > wrote: > >> > >> Hi, > >> > >> I usually find it very useful when I can retrieve a list of the > >> currents calls along with durations. I noticed that the 'show > >> channels' format does not include the duration (or the answered > >> timestamp - so that one can extract it from there). So, I made a > >> patch that includes the answered timestamp, the answered timestamp > >> in epoch, and the duration in seconds. Of course these fields > >> remain > >> empty when the call hasn't been > >> answered yet. > >> > >> I don't know if anyone else finds this functionality useful, so > >> I am > >> posting this patch here first (instead of JIRA) in order to get > >> feedback from the users. If many of you (or the maintainers) > >> find it > >> interesting I can then proceed in posting it to JIRA. > >> > >> -- > >> ------------------------------------------- > >> Apostolos Pantsiopoulos > >> Kinetix Tele.com R & D > >> email: regs at kinetix.gr > >> ------------------------------------------- > >> > >> Index: src/mod/applications/mod_commands/mod_commands.c > >> > >> =================================================================== > >> --- src/mod/applications/mod_commands/mod_commands.c > >> (revision 14256) > >> +++ src/mod/applications/mod_commands/mod_commands.c (working > >> copy) > >> @@ -2827,10 +2827,10 @@ > >> } > >> } > >> if (strchr(argv[2], '%')) { > >> - sprintf(sql, "select * from > >> channels > >> where uuid like '%s' or name like '%s' or cid_name like '%s' or > >> cid_num like '%s' order by created_epoch", > >> + sprintf(sql, "select > >> *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration from > >> channels where uuid like '%s' or name like '%s' or cid_name like > >> '%s' or cid_num like '%s' order by created_epoch", > >> argv[2], argv[2], > >> argv[2], argv[2]); > >> } else { > >> - sprintf(sql, "select * from > >> channels > >> where uuid like '%%%s%%' or name like '%%%s%%' or cid_name like > >> '%%%s%%' or cid_num like '%%%s%%' order by created_epoch", > >> + sprintf(sql, "select > >> *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration from > >> channels where uuid like '%%%s%%' or name like '%%%s%%' or > >> cid_name > >> like '%%%s%%' or cid_num like '%%%s%%' order by created_epoch", > >> argv[2], argv[2], > >> argv[2], argv[2]); > >> > >> } > >> @@ -2839,10 +2839,10 @@ > >> as = argv[4]; > >> } > >> } else { > >> - sprintf(sql, "select * from channels order > >> by created_epoch"); > >> + sprintf(sql, "select > >> *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration from > >> channels order by created_epoch"); > >> } > >> } else if (!strcasecmp(command, "channels")) { > >> - sprintf(sql, "select * from channels order by > >> created_epoch"); > >> + sprintf(sql, "select > >> *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration from > >> channels order by created_epoch"); > >> if (argv[1] && !strcasecmp(argv[1],"count")) { > >> holder.justcount = 1; > >> if (argv[3] && !strcasecmp(argv[2], "as")) { > >> @@ -2850,7 +2850,7 @@ > >> } > >> } > >> } else if (!strcasecmp(command, "distinct_channels")) { > >> - sprintf(sql, "select * from channels left join > >> calls > >> on " > >> + sprintf(sql, "select > >> *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration from > >> channels left join calls on " > >> "channels.uuid=calls.caller_uuid > >> where channels.uuid not in (select callee_uuid from calls) order > >> by > >> created_epoch"); > >> if (argv[2] && !strcasecmp(argv[1], "as")) { > >> as = argv[2]; > >> Index: src/switch_core_sqldb.c > >> > >> =================================================================== > >> --- src/switch_core_sqldb.c (revision 14256) > >> +++ src/switch_core_sqldb.c (working copy) > >> @@ -309,9 +309,21 @@ > >> ); > >> > >> break; > >> + case SWITCH_EVENT_CHANNEL_ANSWER: > >> + { > >> + > >> + sql = switch_mprintf("update channels set > >> answered='%s',answered_epoch='%ld' where uuid='%s'", > >> + > >> switch_event_get_header_nil(event, "event-date-local"), > >> + > >> (long)switch_epoch_time_now(NULL), > >> + > >> switch_event_get_header_nil(event, "unique-id") > >> + ); > >> + > >> + } > >> + break; > >> case SWITCH_EVENT_CHANNEL_STATE: > >> { > >> char *state = > >> switch_event_get_header_nil(event, "channel-state-number"); > >> + > >> switch_channel_state_t state_i = > >> CS_DESTROY; > >> > >> if (!switch_strlen_zero(state)) { > >> @@ -492,7 +504,9 @@ > >> " read_rate VARCHAR(255),\n" > >> " write_codec VARCHAR(255),\n" > >> " write_rate VARCHAR(255),\n" > >> - " secure VARCHAR(255)\n" > >> + " secure VARCHAR(255),\n" > >> + " answered VARCHAR(255),\n" > >> + " answered_epoch INTEGER\n" > >> ");\ncreate index uuindex on channels > >> (uuid);\n"; > >> char create_calls_sql[] = > >> "CREATE TABLE calls (\n" > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > >> > >> > >> > >> -- > >> Anthony Minessale II > >> > >> FreeSWITCH http://www.freeswitch.org/ > >> ClueCon http://www.cluecon.com/ > >> Twitter: http://twitter.com/FreeSWITCH_wire > >> > >> AIM: anthm > >> MSN:anthony_minessale at hotmail.com > >> > > > >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > >> > > > >> IRC: irc.freenode.net #freeswitch > >> > >> FreeSWITCH Developer Conference > >> 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 > > > > > > -- > > ------------------------------------------- > > 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/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference 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/20090716/781ef4d2/attachment.html From brian at freeswitch.org Thu Jul 16 07:58:16 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 16 Jul 2009 09:58:16 -0500 Subject: [Freeswitch-users] Setting P-Asserted-ID to something other than the callerid In-Reply-To: <1247755878.4289.64.camel@dk-d820> References: <723C30E7-4AD0-418D-A4B4-10148367A9AA@gmail.com> <1247755878.4289.64.camel@dk-d820> Message-ID: Kinda wrong there! Gotta use CDATA because it has < and > in the data you're setting. And you'll wanna export it I suspect. ]]> /b On Jul 16, 2009, at 9:51 AM, David Knell wrote: > Hi Dale, > > You can set the header to anything you like by including something > along > the lines of > > in your dialplan. > > Cheers -- -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090716/15b3fd87/attachment.html From regs at kinetix.gr Thu Jul 16 08:10:53 2009 From: regs at kinetix.gr (Apostolos Pantsiopoulos) Date: Thu, 16 Jul 2009 18:10:53 +0300 Subject: [Freeswitch-users] "show channels" command with duration - patch included In-Reply-To: <191c3a030907160754g62aac895ge10a90476d5e890@mail.gmail.com> References: <4A5F3306.1000300@kinetix.gr> <191c3a030907160715k74e785c8j99b68a42dc019bb7@mail.gmail.com> <4A5F3B19.8020507@kinetix.gr> <6079D960-F5EB-4774-93DA-87402CFC1157@freeswitch.org> <191c3a030907160754g62aac895ge10a90476d5e890@mail.gmail.com> Message-ID: <4A5F42FD.60504@kinetix.gr> OK I 'll start implementing the progress timestamp field. The only reason I mentioned the user/gateway id field is that FS admins gain a lot by looking at a row of "show channels" result and be able to see who is the caller and who is the callee. Anthony Minessale wrote: > it doesn't really break anything to add more fields besides the ability > to read it but it's already fairly wide as it is. > Thats why we have "show channels as xml" > > > On Thu, Jul 16, 2009 at 9:50 AM, Michael S Collins > wrote: > > I wonder if it would make sense to create a separate sub-command like > "show channels stats" or something. That way we could put all sorts of > nifty info there without breaking the existing command. > > Thoughts? > -MC > > Sent from my iPhone > > On Jul 16, 2009, at 7:37 AM, Apostolos Pantsiopoulos > > > wrote: > > > Now that I come to think of it... > > > > It would be useful if we had the timestamp (and epoch) > > of the events PROGRESS and PROGRESS WITH MEDIA so that we can extract > > the PDD (Post Dial Delay) which is a very useful statistic. > > > > Adding the user's (for the incoming) and the gateway's (for the > > outbound) id would also be useful. In case these fields are empty the > > show channels command could ommit the (empty string). > > > > Are you planning to implement them yourselves or should I begin > > looking > > at the code? > > > > > > Anthony Minessale wrote: > >> I'm ok with the idea as long as it's thoroughly tested. > >> If there is any more info you want to save from those events you > >> should > >> consider it now while we are modifying it. > >> > >> > >> On Thu, Jul 16, 2009 at 9:02 AM, > > >> >> wrote: > >> > >> Hi, > >> > >> I usually find it very useful when I can retrieve a list of the > >> currents calls along with durations. I noticed that the 'show > >> channels' format does not include the duration (or the answered > >> timestamp - so that one can extract it from there). So, I made a > >> patch that includes the answered timestamp, the answered > timestamp > >> in epoch, and the duration in seconds. Of course these fields > >> remain > >> empty when the call hasn't been > >> answered yet. > >> > >> I don't know if anyone else finds this functionality useful, so > >> I am > >> posting this patch here first (instead of JIRA) in order to get > >> feedback from the users. If many of you (or the maintainers) > >> find it > >> interesting I can then proceed in posting it to JIRA. > >> > >> -- > >> ------------------------------------------- > >> Apostolos Pantsiopoulos > >> Kinetix Tele.com R & D > >> email: regs at kinetix.gr > > > >> ------------------------------------------- > >> > >> Index: src/mod/applications/mod_commands/mod_commands.c > >> > >> =================================================================== > >> --- src/mod/applications/mod_commands/mod_commands.c > >> (revision 14256) > >> +++ src/mod/applications/mod_commands/mod_commands.c (working > >> copy) > >> @@ -2827,10 +2827,10 @@ > >> } > >> } > >> if (strchr(argv[2], '%')) { > >> - sprintf(sql, "select * from > >> channels > >> where uuid like '%s' or name like '%s' or cid_name like '%s' or > >> cid_num like '%s' order by created_epoch", > >> + sprintf(sql, "select > >> *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration from > >> channels where uuid like '%s' or name like '%s' or cid_name like > >> '%s' or cid_num like '%s' order by created_epoch", > >> argv[2], argv[2], > >> argv[2], argv[2]); > >> } else { > >> - sprintf(sql, "select * from > >> channels > >> where uuid like '%%%s%%' or name like '%%%s%%' or cid_name like > >> '%%%s%%' or cid_num like '%%%s%%' order by created_epoch", > >> + sprintf(sql, "select > >> *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration from > >> channels where uuid like '%%%s%%' or name like '%%%s%%' or > >> cid_name > >> like '%%%s%%' or cid_num like '%%%s%%' order by created_epoch", > >> argv[2], argv[2], > >> argv[2], argv[2]); > >> > >> } > >> @@ -2839,10 +2839,10 @@ > >> as = argv[4]; > >> } > >> } else { > >> - sprintf(sql, "select * from channels > order > >> by created_epoch"); > >> + sprintf(sql, "select > >> *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration from > >> channels order by created_epoch"); > >> } > >> } else if (!strcasecmp(command, "channels")) { > >> - sprintf(sql, "select * from channels order by > >> created_epoch"); > >> + sprintf(sql, "select > >> *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration from > >> channels order by created_epoch"); > >> if (argv[1] && !strcasecmp(argv[1],"count")) { > >> holder.justcount = 1; > >> if (argv[3] && !strcasecmp(argv[2], "as")) { > >> @@ -2850,7 +2850,7 @@ > >> } > >> } > >> } else if (!strcasecmp(command, "distinct_channels")) { > >> - sprintf(sql, "select * from channels left join > >> calls > >> on " > >> + sprintf(sql, "select > >> *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration from > >> channels left join calls on " > >> "channels.uuid=calls.caller_uuid > >> where channels.uuid not in (select callee_uuid from calls) order > >> by > >> created_epoch"); > >> if (argv[2] && !strcasecmp(argv[1], "as")) { > >> as = argv[2]; > >> Index: src/switch_core_sqldb.c > >> > >> =================================================================== > >> --- src/switch_core_sqldb.c (revision 14256) > >> +++ src/switch_core_sqldb.c (working copy) > >> @@ -309,9 +309,21 @@ > >> ); > >> > >> break; > >> + case SWITCH_EVENT_CHANNEL_ANSWER: > >> + { > >> + > >> + sql = switch_mprintf("update channels set > >> answered='%s',answered_epoch='%ld' where uuid='%s'", > >> + > >> switch_event_get_header_nil(event, "event-date-local"), > >> + > >> (long)switch_epoch_time_now(NULL), > >> + > >> switch_event_get_header_nil(event, "unique-id") > >> + ); > >> + > >> + } > >> + break; > >> case SWITCH_EVENT_CHANNEL_STATE: > >> { > >> char *state = > >> switch_event_get_header_nil(event, "channel-state-number"); > >> + > >> switch_channel_state_t state_i = > >> CS_DESTROY; > >> > >> if (!switch_strlen_zero(state)) { > >> @@ -492,7 +504,9 @@ > >> " read_rate VARCHAR(255),\n" > >> " write_codec VARCHAR(255),\n" > >> " write_rate VARCHAR(255),\n" > >> - " secure VARCHAR(255)\n" > >> + " secure VARCHAR(255),\n" > >> + " answered VARCHAR(255),\n" > >> + " answered_epoch INTEGER\n" > >> ");\ncreate index uuindex on channels > >> (uuid);\n"; > >> char create_calls_sql[] = > >> "CREATE TABLE calls (\n" > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > > >> > > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > >> > >> > >> > >> -- > >> Anthony Minessale II > >> > >> FreeSWITCH http://www.freeswitch.org/ > >> ClueCon http://www.cluecon.com/ > >> Twitter: http://twitter.com/FreeSWITCH_wire > >> > >> AIM: anthm > >> MSN:anthony_minessale at hotmail.com > > >> > > >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > >> > > >> IRC: irc.freenode.net > #freeswitch > >> > >> FreeSWITCH Developer Conference > >> 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 > > > > > > -- > > ------------------------------------------- > > 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/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > 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 -- ------------------------------------------- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: regs at kinetix.gr ------------------------------------------- From saeedahmad1981 at gmail.com Thu Jul 16 08:38:02 2009 From: saeedahmad1981 at gmail.com (Saeed Ahmed) Date: Thu, 16 Jul 2009 17:38:02 +0200 Subject: [Freeswitch-users] "show channels" command with duration - patch included In-Reply-To: <4A5F42FD.60504@kinetix.gr> References: <4A5F3306.1000300@kinetix.gr> <191c3a030907160715k74e785c8j99b68a42dc019bb7@mail.gmail.com> <4A5F3B19.8020507@kinetix.gr> <6079D960-F5EB-4774-93DA-87402CFC1157@freeswitch.org><191c3a030907160754g62aac895ge10a90476d5e890@mail.gmail.com> <4A5F42FD.60504@kinetix.gr> Message-ID: <9AEB5B242EBF4DD0A9B1544B7EAFC82C@saeedlaptop> Hi, its very useful feature for monitoring, I am doing it with XML RPC and getting the result on webpage. There is one issue which is nothing to do with that patch but in general: if we are using absolute_codec_string variable and codes are like G729,G723 then this *comma* between codec ruin the array. I think there should be other separator. - Saeed -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Apostolos Pantsiopoulos Sent: Thursday, July 16, 2009 5:11 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] "show channels" command with duration - patch included OK I 'll start implementing the progress timestamp field. The only reason I mentioned the user/gateway id field is that FS admins gain a lot by looking at a row of "show channels" result and be able to see who is the caller and who is the callee. Anthony Minessale wrote: > it doesn't really break anything to add more fields besides the ability > to read it but it's already fairly wide as it is. > Thats why we have "show channels as xml" > > > On Thu, Jul 16, 2009 at 9:50 AM, Michael S Collins > wrote: > > I wonder if it would make sense to create a separate sub-command like > "show channels stats" or something. That way we could put all sorts of > nifty info there without breaking the existing command. > > Thoughts? > -MC > > Sent from my iPhone > > On Jul 16, 2009, at 7:37 AM, Apostolos Pantsiopoulos > > > wrote: > > > Now that I come to think of it... > > > > It would be useful if we had the timestamp (and epoch) > > of the events PROGRESS and PROGRESS WITH MEDIA so that we can extract > > the PDD (Post Dial Delay) which is a very useful statistic. > > > > Adding the user's (for the incoming) and the gateway's (for the > > outbound) id would also be useful. In case these fields are empty the > > show channels command could ommit the (empty string). > > > > Are you planning to implement them yourselves or should I begin > > looking > > at the code? > > > > > > Anthony Minessale wrote: > >> I'm ok with the idea as long as it's thoroughly tested. > >> If there is any more info you want to save from those events you > >> should > >> consider it now while we are modifying it. > >> > >> > >> On Thu, Jul 16, 2009 at 9:02 AM, > > >> >> wrote: > >> > >> Hi, > >> > >> I usually find it very useful when I can retrieve a list of the > >> currents calls along with durations. I noticed that the 'show > >> channels' format does not include the duration (or the answered > >> timestamp - so that one can extract it from there). So, I made a > >> patch that includes the answered timestamp, the answered > timestamp > >> in epoch, and the duration in seconds. Of course these fields > >> remain > >> empty when the call hasn't been > >> answered yet. > >> > >> I don't know if anyone else finds this functionality useful, so > >> I am > >> posting this patch here first (instead of JIRA) in order to get > >> feedback from the users. If many of you (or the maintainers) > >> find it > >> interesting I can then proceed in posting it to JIRA. > >> > >> -- > >> ------------------------------------------- > >> Apostolos Pantsiopoulos > >> Kinetix Tele.com R & D > >> email: regs at kinetix.gr > > > >> ------------------------------------------- > >> > >> Index: src/mod/applications/mod_commands/mod_commands.c > >> > >> =================================================================== > >> --- src/mod/applications/mod_commands/mod_commands.c > >> (revision 14256) > >> +++ src/mod/applications/mod_commands/mod_commands.c (working > >> copy) > >> @@ -2827,10 +2827,10 @@ > >> } > >> } > >> if (strchr(argv[2], '%')) { > >> - sprintf(sql, "select * from > >> channels > >> where uuid like '%s' or name like '%s' or cid_name like '%s' or > >> cid_num like '%s' order by created_epoch", > >> + sprintf(sql, "select > >> *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration from > >> channels where uuid like '%s' or name like '%s' or cid_name like > >> '%s' or cid_num like '%s' order by created_epoch", > >> argv[2], argv[2], > >> argv[2], argv[2]); > >> } else { > >> - sprintf(sql, "select * from > >> channels > >> where uuid like '%%%s%%' or name like '%%%s%%' or cid_name like > >> '%%%s%%' or cid_num like '%%%s%%' order by created_epoch", > >> + sprintf(sql, "select > >> *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration from > >> channels where uuid like '%%%s%%' or name like '%%%s%%' or > >> cid_name > >> like '%%%s%%' or cid_num like '%%%s%%' order by created_epoch", > >> argv[2], argv[2], > >> argv[2], argv[2]); > >> > >> } > >> @@ -2839,10 +2839,10 @@ > >> as = argv[4]; > >> } > >> } else { > >> - sprintf(sql, "select * from channels > order > >> by created_epoch"); > >> + sprintf(sql, "select > >> *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration from > >> channels order by created_epoch"); > >> } > >> } else if (!strcasecmp(command, "channels")) { > >> - sprintf(sql, "select * from channels order by > >> created_epoch"); > >> + sprintf(sql, "select > >> *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration from > >> channels order by created_epoch"); > >> if (argv[1] && !strcasecmp(argv[1],"count")) { > >> holder.justcount = 1; > >> if (argv[3] && !strcasecmp(argv[2], "as")) { > >> @@ -2850,7 +2850,7 @@ > >> } > >> } > >> } else if (!strcasecmp(command, "distinct_channels")) { > >> - sprintf(sql, "select * from channels left join > >> calls > >> on " > >> + sprintf(sql, "select > >> *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration from > >> channels left join calls on " > >> "channels.uuid=calls.caller_uuid > >> where channels.uuid not in (select callee_uuid from calls) order > >> by > >> created_epoch"); > >> if (argv[2] && !strcasecmp(argv[1], "as")) { > >> as = argv[2]; > >> Index: src/switch_core_sqldb.c > >> > >> =================================================================== > >> --- src/switch_core_sqldb.c (revision 14256) > >> +++ src/switch_core_sqldb.c (working copy) > >> @@ -309,9 +309,21 @@ > >> ); > >> > >> break; > >> + case SWITCH_EVENT_CHANNEL_ANSWER: > >> + { > >> + > >> + sql = switch_mprintf("update channels set > >> answered='%s',answered_epoch='%ld' where uuid='%s'", > >> + > >> switch_event_get_header_nil(event, "event-date-local"), > >> + > >> (long)switch_epoch_time_now(NULL), > >> + > >> switch_event_get_header_nil(event, "unique-id") > >> + ); > >> + > >> + } > >> + break; > >> case SWITCH_EVENT_CHANNEL_STATE: > >> { > >> char *state = > >> switch_event_get_header_nil(event, "channel-state-number"); > >> + > >> switch_channel_state_t state_i = > >> CS_DESTROY; > >> > >> if (!switch_strlen_zero(state)) { > >> @@ -492,7 +504,9 @@ > >> " read_rate VARCHAR(255),\n" > >> " write_codec VARCHAR(255),\n" > >> " write_rate VARCHAR(255),\n" > >> - " secure VARCHAR(255)\n" > >> + " secure VARCHAR(255),\n" > >> + " answered VARCHAR(255),\n" > >> + " answered_epoch INTEGER\n" > >> ");\ncreate index uuindex on channels > >> (uuid);\n"; > >> char create_calls_sql[] = > >> "CREATE TABLE calls (\n" > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > > >> > > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > >> > >> > >> > >> -- > >> Anthony Minessale II > >> > >> FreeSWITCH http://www.freeswitch.org/ > >> ClueCon http://www.cluecon.com/ > >> Twitter: http://twitter.com/FreeSWITCH_wire > >> > >> AIM: anthm > >> MSN:anthony_minessale at hotmail.com > > >> > > >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > >> > > >> IRC: irc.freenode.net > #freeswitch > >> > >> FreeSWITCH Developer Conference > >> 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 > > > > > > -- > > ------------------------------------------- > > 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/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > 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 -- ------------------------------------------- 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 From dome at tel.co.th Thu Jul 16 08:38:37 2009 From: dome at tel.co.th (Dome Charoenyost) Date: Thu, 16 Jul 2009 22:38:37 +0700 Subject: [Freeswitch-users] sip extermal profile for all IP In-Reply-To: References: <8ccbff060907160630r2188c8d8od0c5c677f81eb71b@mail.gmail.com> Message-ID: <8ccbff060907160838y2d61e7a3me668250c18fb420@mail.gmail.com> I have eth0 and tap0 (vpn) FS bind only eth0 ip Dome C. 2009/7/16 : > I think by default it maps to all IPs unless you mention one here (by > replacing ${local_ip_v4} to some ip address). > > Thank you. > > > On Thu, Jul 16, 2009 at 7:30 PM, > wrote: >> >> Dear All, >> >> ? ? ? ? How to set ${local_ip_v4} to 0.0.0.0 ? i want to set up >> external profile for All IP >> >> >> Best regards. >> >> Dome C. >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > -- > Muhammad Shahzad > ----------------------------------- > CISCO Rich Media Communication Specialist (CRMCS) > CISCO Certified Network Associate (CCNA) > Cell: +92 334 422 40 88 > MSN: shari_786pk at hotmail.com > Email: shaheryarkh at googlemail.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 mrene_lists at avgs.ca Thu Jul 16 08:39:44 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Thu, 16 Jul 2009 11:39:44 -0400 Subject: [Freeswitch-users] "show channels" command with duration - patch included In-Reply-To: <9AEB5B242EBF4DD0A9B1544B7EAFC82C@saeedlaptop> References: <4A5F3306.1000300@kinetix.gr> <191c3a030907160715k74e785c8j99b68a42dc019bb7@mail.gmail.com> <4A5F3B19.8020507@kinetix.gr> <6079D960-F5EB-4774-93DA-87402CFC1157@freeswitch.org><191c3a030907160754g62aac895ge10a90476d5e890@mail.gmail.com> <4A5F42FD.60504@kinetix.gr> <9AEB5B242EBF4DD0A9B1544B7EAFC82C@saeedlaptop> Message-ID: If you want to use , within a { } block you can escape it with \, Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca Am 16-Jul-09 um 11:38 AM schrieb Saeed Ahmed: > Hi, > > its very useful feature for monitoring, I am doing it with XML RPC and > getting the result on webpage. > > There is one issue which is nothing to do with that patch but in > general: if > we are using absolute_codec_string variable and codes are like > G729,G723 > then this *comma* between codec ruin the array. I think there should > be > other separator. > > - Saeed > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Apostolos Pantsiopoulos > Sent: Thursday, July 16, 2009 5:11 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] "show channels" command with > duration - > patch included > > OK I 'll start implementing the progress timestamp field. > > The only reason I mentioned the user/gateway id field is that FS > admins > gain a lot by looking at a row of "show channels" result and be able > to > see who is the caller and who is the callee. > > Anthony Minessale wrote: >> it doesn't really break anything to add more fields besides the >> ability >> to read it but it's already fairly wide as it is. >> Thats why we have "show channels as xml" >> >> >> On Thu, Jul 16, 2009 at 9:50 AM, Michael S Collins >> > > wrote: >> >> I wonder if it would make sense to create a separate sub-command >> like >> "show channels stats" or something. That way we could put all >> sorts of >> nifty info there without breaking the existing command. >> >> Thoughts? >> -MC >> >> Sent from my iPhone >> >> On Jul 16, 2009, at 7:37 AM, Apostolos Pantsiopoulos >> > >> wrote: >> >>> Now that I come to think of it... >>> >>> It would be useful if we had the timestamp (and epoch) >>> of the events PROGRESS and PROGRESS WITH MEDIA so that we can > extract >>> the PDD (Post Dial Delay) which is a very useful statistic. >>> >>> Adding the user's (for the incoming) and the gateway's (for the >>> outbound) id would also be useful. In case these fields are empty > the >>> show channels command could ommit the (empty string). >>> >>> Are you planning to implement them yourselves or should I begin >>> looking >>> at the code? >>> >>> >>> Anthony Minessale wrote: >>>> I'm ok with the idea as long as it's thoroughly tested. >>>> If there is any more info you want to save from those events you >>>> should >>>> consider it now while we are modifying it. >>>> >>>> >>>> On Thu, Jul 16, 2009 at 9:02 AM, >> > >>>> > >> wrote: >>>> >>>> Hi, >>>> >>>> I usually find it very useful when I can retrieve a list of the >>>> currents calls along with durations. I noticed that the 'show >>>> channels' format does not include the duration (or the answered >>>> timestamp - so that one can extract it from there). So, I made > a >>>> patch that includes the answered timestamp, the answered >> timestamp >>>> in epoch, and the duration in seconds. Of course these fields >>>> remain >>>> empty when the call hasn't been >>>> answered yet. >>>> >>>> I don't know if anyone else finds this functionality useful, so >>>> I am >>>> posting this patch here first (instead of JIRA) in order to get >>>> feedback from the users. If many of you (or the maintainers) >>>> find it >>>> interesting I can then proceed in posting it to JIRA. >>>> >>>> -- >>>> ------------------------------------------- >>>> Apostolos Pantsiopoulos >>>> Kinetix Tele.com R & D >>>> email: regs at kinetix.gr >> > >>>> ------------------------------------------- >>>> >>>> Index: src/mod/applications/mod_commands/mod_commands.c >>>> >>>> > =================================================================== >>>> --- src/mod/applications/mod_commands/mod_commands.c >>>> (revision 14256) >>>> +++ src/mod/applications/mod_commands/mod_commands.c > (working >>>> copy) >>>> @@ -2827,10 +2827,10 @@ >>>> } >>>> } >>>> if (strchr(argv[2], '%')) { >>>> - sprintf(sql, "select * from >>>> channels >>>> where uuid like '%s' or name like '%s' or cid_name like '%s' or >>>> cid_num like '%s' order by created_epoch", >>>> + sprintf(sql, "select >>>> *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration > from >>>> channels where uuid like '%s' or name like '%s' or cid_name > like >>>> '%s' or cid_num like '%s' order by created_epoch", >>>> argv[2], > argv[2], >>>> argv[2], argv[2]); >>>> } else { >>>> - sprintf(sql, "select * from >>>> channels >>>> where uuid like '%%%s%%' or name like '%%%s%%' or cid_name like >>>> '%%%s%%' or cid_num like '%%%s%%' order by created_epoch", >>>> + sprintf(sql, "select >>>> *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration > from >>>> channels where uuid like '%%%s%%' or name like '%%%s%%' or >>>> cid_name >>>> like '%%%s%%' or cid_num like '%%%s%%' order by created_epoch", >>>> argv[2], > argv[2], >>>> argv[2], argv[2]); >>>> >>>> } >>>> @@ -2839,10 +2839,10 @@ >>>> as = argv[4]; >>>> } >>>> } else { >>>> - sprintf(sql, "select * from channels >> order >>>> by created_epoch"); >>>> + sprintf(sql, "select >>>> *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration > from >>>> channels order by created_epoch"); >>>> } >>>> } else if (!strcasecmp(command, "channels")) { >>>> - sprintf(sql, "select * from channels order by >>>> created_epoch"); >>>> + sprintf(sql, "select >>>> *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration > from >>>> channels order by created_epoch"); >>>> if (argv[1] && !strcasecmp(argv[1],"count")) { >>>> holder.justcount = 1; >>>> if (argv[3] && !strcasecmp(argv[2], "as")) { >>>> @@ -2850,7 +2850,7 @@ >>>> } >>>> } >>>> } else if (!strcasecmp(command, "distinct_channels")) { >>>> - sprintf(sql, "select * from channels left join >>>> calls >>>> on " >>>> + sprintf(sql, "select >>>> *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration > from >>>> channels left join calls on " >>>> "channels.uuid=calls.caller_uuid >>>> where channels.uuid not in (select callee_uuid from calls) > order >>>> by >>>> created_epoch"); >>>> if (argv[2] && !strcasecmp(argv[1], "as")) { >>>> as = argv[2]; >>>> Index: src/switch_core_sqldb.c >>>> >>>> > =================================================================== >>>> --- src/switch_core_sqldb.c (revision 14256) >>>> +++ src/switch_core_sqldb.c (working copy) >>>> @@ -309,9 +309,21 @@ >>>> ); >>>> >>>> break; >>>> + case SWITCH_EVENT_CHANNEL_ANSWER: >>>> + { >>>> + >>>> + sql = switch_mprintf("update channels > set >>>> answered='%s',answered_epoch='%ld' where uuid='%s'", >>>> + >>>> switch_event_get_header_nil(event, "event-date-local"), >>>> + >>>> (long)switch_epoch_time_now(NULL), >>>> + >>>> switch_event_get_header_nil(event, "unique-id") >>>> + ); >>>> + >>>> + } >>>> + break; >>>> case SWITCH_EVENT_CHANNEL_STATE: >>>> { >>>> char *state = >>>> switch_event_get_header_nil(event, "channel-state-number"); >>>> + >>>> switch_channel_state_t state_i = >>>> CS_DESTROY; >>>> >>>> if (!switch_strlen_zero(state)) { >>>> @@ -492,7 +504,9 @@ >>>> " read_rate VARCHAR(255),\n" >>>> " write_codec VARCHAR(255),\n" >>>> " write_rate VARCHAR(255),\n" >>>> - " secure VARCHAR(255)\n" >>>> + " secure VARCHAR(255),\n" >>>> + " answered VARCHAR(255),\n" >>>> + " answered_epoch INTEGER\n" >>>> ");\ncreate index uuindex on channels >>>> (uuid);\n"; >>>> char create_calls_sql[] = >>>> "CREATE TABLE calls (\n" >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >> >>>> > > >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> >>>> >>>> -- >>>> Anthony Minessale II >>>> >>>> FreeSWITCH http://www.freeswitch.org/ >>>> ClueCon http://www.cluecon.com/ >>>> Twitter: http://twitter.com/FreeSWITCH_wire >>>> >>>> AIM: anthm >>>> MSN:anthony_minessale at hotmail.com >> >>>> > > >>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> >>>> > > >>>> IRC: irc.freenode.net >> #freeswitch >>>> >>>> FreeSWITCH Developer Conference >>>> 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 >>> >>> >>> -- >>> ------------------------------------------- >>> 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/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> 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 > > > -- > ------------------------------------------- > 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 From regs at kinetix.gr Thu Jul 16 08:47:50 2009 From: regs at kinetix.gr (Apostolos Pantsiopoulos) Date: Thu, 16 Jul 2009 18:47:50 +0300 Subject: [Freeswitch-users] "show channels" command with duration - patch included In-Reply-To: <4A5F42FD.60504@kinetix.gr> References: <4A5F3306.1000300@kinetix.gr> <191c3a030907160715k74e785c8j99b68a42dc019bb7@mail.gmail.com> <4A5F3B19.8020507@kinetix.gr> <6079D960-F5EB-4774-93DA-87402CFC1157@freeswitch.org> <191c3a030907160754g62aac895ge10a90476d5e890@mail.gmail.com> <4A5F42FD.60504@kinetix.gr> Message-ID: <4A5F4BA6.50204@kinetix.gr> On a second thought... I am starting to agree with Michael. Nobody really collects stats about PDD from the 'show channels' command. This is usually done by using the cdrs. So, forget about the second request. The patch I sent earlier covers my needs (and I hope everybody else's too.) Apostolos Pantsiopoulos wrote: > OK I 'll start implementing the progress timestamp field. > > The only reason I mentioned the user/gateway id field is that FS admins > gain a lot by looking at a row of "show channels" result and be able to > see who is the caller and who is the callee. > > Anthony Minessale wrote: >> it doesn't really break anything to add more fields besides the ability >> to read it but it's already fairly wide as it is. >> Thats why we have "show channels as xml" >> >> >> On Thu, Jul 16, 2009 at 9:50 AM, Michael S Collins > > wrote: >> >> I wonder if it would make sense to create a separate sub-command like >> "show channels stats" or something. That way we could put all sorts of >> nifty info there without breaking the existing command. >> >> Thoughts? >> -MC >> >> Sent from my iPhone >> >> On Jul 16, 2009, at 7:37 AM, Apostolos Pantsiopoulos >> > >> wrote: >> >> > Now that I come to think of it... >> > >> > It would be useful if we had the timestamp (and epoch) >> > of the events PROGRESS and PROGRESS WITH MEDIA so that we can extract >> > the PDD (Post Dial Delay) which is a very useful statistic. >> > >> > Adding the user's (for the incoming) and the gateway's (for the >> > outbound) id would also be useful. In case these fields are empty the >> > show channels command could ommit the (empty string). >> > >> > Are you planning to implement them yourselves or should I begin >> > looking >> > at the code? >> > >> > >> > Anthony Minessale wrote: >> >> I'm ok with the idea as long as it's thoroughly tested. >> >> If there is any more info you want to save from those events you >> >> should >> >> consider it now while we are modifying it. >> >> >> >> >> >> On Thu, Jul 16, 2009 at 9:02 AM, >> > >> >> > >> wrote: >> >> >> >> Hi, >> >> >> >> I usually find it very useful when I can retrieve a list of the >> >> currents calls along with durations. I noticed that the 'show >> >> channels' format does not include the duration (or the answered >> >> timestamp - so that one can extract it from there). So, I made a >> >> patch that includes the answered timestamp, the answered >> timestamp >> >> in epoch, and the duration in seconds. Of course these fields >> >> remain >> >> empty when the call hasn't been >> >> answered yet. >> >> >> >> I don't know if anyone else finds this functionality useful, so >> >> I am >> >> posting this patch here first (instead of JIRA) in order to get >> >> feedback from the users. If many of you (or the maintainers) >> >> find it >> >> interesting I can then proceed in posting it to JIRA. >> >> >> >> -- >> >> ------------------------------------------- >> >> Apostolos Pantsiopoulos >> >> Kinetix Tele.com R & D >> >> email: regs at kinetix.gr >> > >> >> ------------------------------------------- >> >> >> >> Index: src/mod/applications/mod_commands/mod_commands.c >> >> >> >> =================================================================== >> >> --- src/mod/applications/mod_commands/mod_commands.c >> >> (revision 14256) >> >> +++ src/mod/applications/mod_commands/mod_commands.c (working >> >> copy) >> >> @@ -2827,10 +2827,10 @@ >> >> } >> >> } >> >> if (strchr(argv[2], '%')) { >> >> - sprintf(sql, "select * from >> >> channels >> >> where uuid like '%s' or name like '%s' or cid_name like '%s' or >> >> cid_num like '%s' order by created_epoch", >> >> + sprintf(sql, "select >> >> *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration from >> >> channels where uuid like '%s' or name like '%s' or cid_name like >> >> '%s' or cid_num like '%s' order by created_epoch", >> >> argv[2], argv[2], >> >> argv[2], argv[2]); >> >> } else { >> >> - sprintf(sql, "select * from >> >> channels >> >> where uuid like '%%%s%%' or name like '%%%s%%' or cid_name like >> >> '%%%s%%' or cid_num like '%%%s%%' order by created_epoch", >> >> + sprintf(sql, "select >> >> *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration from >> >> channels where uuid like '%%%s%%' or name like '%%%s%%' or >> >> cid_name >> >> like '%%%s%%' or cid_num like '%%%s%%' order by created_epoch", >> >> argv[2], argv[2], >> >> argv[2], argv[2]); >> >> >> >> } >> >> @@ -2839,10 +2839,10 @@ >> >> as = argv[4]; >> >> } >> >> } else { >> >> - sprintf(sql, "select * from channels >> order >> >> by created_epoch"); >> >> + sprintf(sql, "select >> >> *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration from >> >> channels order by created_epoch"); >> >> } >> >> } else if (!strcasecmp(command, "channels")) { >> >> - sprintf(sql, "select * from channels order by >> >> created_epoch"); >> >> + sprintf(sql, "select >> >> *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration from >> >> channels order by created_epoch"); >> >> if (argv[1] && !strcasecmp(argv[1],"count")) { >> >> holder.justcount = 1; >> >> if (argv[3] && !strcasecmp(argv[2], "as")) { >> >> @@ -2850,7 +2850,7 @@ >> >> } >> >> } >> >> } else if (!strcasecmp(command, "distinct_channels")) { >> >> - sprintf(sql, "select * from channels left join >> >> calls >> >> on " >> >> + sprintf(sql, "select >> >> *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration from >> >> channels left join calls on " >> >> "channels.uuid=calls.caller_uuid >> >> where channels.uuid not in (select callee_uuid from calls) order >> >> by >> >> created_epoch"); >> >> if (argv[2] && !strcasecmp(argv[1], "as")) { >> >> as = argv[2]; >> >> Index: src/switch_core_sqldb.c >> >> >> >> =================================================================== >> >> --- src/switch_core_sqldb.c (revision 14256) >> >> +++ src/switch_core_sqldb.c (working copy) >> >> @@ -309,9 +309,21 @@ >> >> ); >> >> >> >> break; >> >> + case SWITCH_EVENT_CHANNEL_ANSWER: >> >> + { >> >> + >> >> + sql = switch_mprintf("update channels set >> >> answered='%s',answered_epoch='%ld' where uuid='%s'", >> >> + >> >> switch_event_get_header_nil(event, "event-date-local"), >> >> + >> >> (long)switch_epoch_time_now(NULL), >> >> + >> >> switch_event_get_header_nil(event, "unique-id") >> >> + ); >> >> + >> >> + } >> >> + break; >> >> case SWITCH_EVENT_CHANNEL_STATE: >> >> { >> >> char *state = >> >> switch_event_get_header_nil(event, "channel-state-number"); >> >> + >> >> switch_channel_state_t state_i = >> >> CS_DESTROY; >> >> >> >> if (!switch_strlen_zero(state)) { >> >> @@ -492,7 +504,9 @@ >> >> " read_rate VARCHAR(255),\n" >> >> " write_codec VARCHAR(255),\n" >> >> " write_rate VARCHAR(255),\n" >> >> - " secure VARCHAR(255)\n" >> >> + " secure VARCHAR(255),\n" >> >> + " answered VARCHAR(255),\n" >> >> + " answered_epoch INTEGER\n" >> >> ");\ncreate index uuindex on channels >> >> (uuid);\n"; >> >> char create_calls_sql[] = >> >> "CREATE TABLE calls (\n" >> >> >> >> _______________________________________________ >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> >> > > >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> >> >> >> >> >> >> >> >> >> -- >> >> Anthony Minessale II >> >> >> >> FreeSWITCH http://www.freeswitch.org/ >> >> ClueCon http://www.cluecon.com/ >> >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> >> >> AIM: anthm >> >> MSN:anthony_minessale at hotmail.com >> >> >> > > >> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> >> >> > > >> >> IRC: irc.freenode.net >> #freeswitch >> >> >> >> FreeSWITCH Developer Conference >> >> 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 >> > >> > >> > -- >> > ------------------------------------------- >> > 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/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> 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 > > -- ------------------------------------------- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: regs at kinetix.gr ------------------------------------------- From fdhege at gmail.com Thu Jul 16 09:37:40 2009 From: fdhege at gmail.com (Dale) Date: Thu, 16 Jul 2009 12:37:40 -0400 Subject: [Freeswitch-users] Setting P-Asserted-ID to something other than the callerid In-Reply-To: References: Message-ID: <3953E116-DC6C-4324-B063-374CE37822F2@gmail.com> Who knows. I have asked why they are backwards when it comes to the RFC's. -Dale On Jul 16, 2009, at 10:41 AM, Ken Rice wrote: > That's just right down screwing with the standards... > > PAID is the caller id... This particular definition is from the RFCs > and > 3GPP docs for IMS which is why we have standardized P- headers... > > Can your vendor not look at the P-Charging-Vector field? > > Also, From when used with PAID is more like an ANI not a CLID > > >> From: Dale >> Reply-To: >> Date: Thu, 16 Jul 2009 10:30:34 -0400 >> To: >> Subject: [Freeswitch-users] Setting P-Asserted-ID to something >> other than the >> callerid >> >> >> Hello again, >> >> I wanted to first say thanks to Brain for helping me fix my from >> domain issue the other day. It helped quite a bit. >> >> Now with more testing and talking with the vendor (please don't shoot >> the messenger :) ) >> >> They want the caller id info in the from and the charge number/ >> screening number in the P-Asserted-ID. >> >> I have tested this and verified that this does work like they say it >> does by setting the callerid number to my charge number and setting >> the from user in the gateway config to the callerid I want displayed. >> But this solution doesn't scale very well. >> >> I know I can set the gateway option caller-id-in-from to get that >> part >> done. But is there a way to set the P-Asserted-ID to something other >> than the callerid? >> >> Any hints would be welcomed. >> >> Thanks, >> >> -Dale >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From fdhege at gmail.com Thu Jul 16 10:04:35 2009 From: fdhege at gmail.com (Dale) Date: Thu, 16 Jul 2009 13:04:35 -0400 Subject: [Freeswitch-users] Setting P-Asserted-ID to something other than the callerid In-Reply-To: References: Message-ID: Hey, you happen to have the RFC number and section they are violating? I'll dig it up. But figured I'd ask since you all seem to have them memorized by now. :) -Dale On Jul 16, 2009, at 10:41 AM, Ken Rice wrote: > That's just right down screwing with the standards... > > PAID is the caller id... This particular definition is from the RFCs > and > 3GPP docs for IMS which is why we have standardized P- headers... > > Can your vendor not look at the P-Charging-Vector field? > > Also, From when used with PAID is more like an ANI not a CLID > > >> From: Dale >> Reply-To: >> Date: Thu, 16 Jul 2009 10:30:34 -0400 >> To: >> Subject: [Freeswitch-users] Setting P-Asserted-ID to something >> other than the >> callerid >> >> >> Hello again, >> >> I wanted to first say thanks to Brain for helping me fix my from >> domain issue the other day. It helped quite a bit. >> >> Now with more testing and talking with the vendor (please don't shoot >> the messenger :) ) >> >> They want the caller id info in the from and the charge number/ >> screening number in the P-Asserted-ID. >> >> I have tested this and verified that this does work like they say it >> does by setting the callerid number to my charge number and setting >> the from user in the gateway config to the callerid I want displayed. >> But this solution doesn't scale very well. >> >> I know I can set the gateway option caller-id-in-from to get that >> part >> done. But is there a way to set the P-Asserted-ID to something other >> than the callerid? >> >> Any hints would be welcomed. >> >> Thanks, >> >> -Dale >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.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 Thu Jul 16 10:13:28 2009 From: sicfslist at gmail.com (Shelby Ramsey) Date: Thu, 16 Jul 2009 12:13:28 -0500 Subject: [Freeswitch-users] Setting P-Asserted-ID to something other than the callerid In-Reply-To: References: Message-ID: <35b355e90907161013j31fb4603r6d632fa0f4cc5abe@mail.gmail.com> PAI is defined @ http://www.ietf.org/rfc/rfc3325.txt For what they are trying to do Sonus suggested P-Charge-Info (which I think it is still in the draft stage) --> http://www.ietf.org/internet-drafts/draft-york-sipping-p-charge-info-06.txt -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090716/3f9eb95f/attachment.html From marketing at cluecon.com Thu Jul 16 11:29:38 2009 From: marketing at cluecon.com (Michael Collins) Date: Thu, 16 Jul 2009 11:29:38 -0700 Subject: [Freeswitch-users] ClueCon 2009 - News and Notes Message-ID: <87f2f3b90907161129m29921db0q611bb87b791ab522@mail.gmail.com> Only 19 more days until ClueCon! If you haven't registered then please do so right away by calling 877.742.CLUE. This year's event is going to be awesome - you don't want to miss it! Some news and notes: We have two more sponsors for our event this year: Nokia and Twilio! We are happy to welcome them both to the conference this year. Many of our sponsors are supplying items for each attendee as well as some nice prizes that will be raffled off at the end of events on Thursday afternoon. We also have a very special guest who will be speaking at ClueCon: Philip Zimmermann! Philip is the author of PGP and co-author of ZRTP. We look forward to his presentation on Wednesday. Another security expert, Dan York of Voxeo, will also be speaking on Wednesday. All of those who are interested in security will want to pay special attention to Wednesday's program. Thanks for your support and we look forward to seeing you in a few weeks! -Michael Collins http://www.cluecon.com 877.742.CLUE -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090716/dc1c2cbd/attachment.html From dave at 3c.co.uk Thu Jul 16 12:20:25 2009 From: dave at 3c.co.uk (David Knell) Date: Thu, 16 Jul 2009 16:20:25 -0300 Subject: [Freeswitch-users] Setting P-Asserted-ID to something other than the callerid In-Reply-To: References: <723C30E7-4AD0-418D-A4B4-10148367A9AA@gmail.com> <1247755878.4289.64.camel@dk-d820> Message-ID: <1247772025.4289.88.camel@dk-d820> Really? That line (+/- the IP address) came directly out of a working dialplan. To be fair, the box is running a faintly prehistoric FreeSWITCH - you crazy cats haven't been chewing on the tail of my cherished mouse of backwards compatibility again, have you?! What has been incorrect in this discussion is the name of the header: it's P-Asserted-Identity, not P-Asserted-ID. The fact that it's usually shortened to PAID doesn't help; nor does the fact that Remote-Party-ID (which is deprecated, but still widely used for the same job as P-Asserted-Identity) is about as well. --Dave > Kinda wrong there! > > > Gotta use CDATA because it has < and > in the data you're setting. > And you'll wanna export it I suspect. > > > ${caller_id_number}@1.2.3.4>]]> > > > /b > > > > > > On Jul 16, 2009, at 9:51 AM, David Knell wrote: > > > Hi Dale, > > > > You can set the header to anything you like by including something > > along > > the lines of > > > > in your dialplan. > > > > 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 -- David Knell, Director, 3C Limited T: +44 20 3298 2000 E: dave at 3c.co.uk W: http://www.3c.co.uk From brian at freeswitch.org Thu Jul 16 12:27:30 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 16 Jul 2009 14:27:30 -0500 Subject: [Freeswitch-users] Setting P-Asserted-ID to something other than the callerid In-Reply-To: <1247772025.4289.88.camel@dk-d820> References: <723C30E7-4AD0-418D-A4B4-10148367A9AA@gmail.com> <1247755878.4289.64.camel@dk-d820> <1247772025.4289.88.camel@dk-d820> Message-ID: Not sure how you were able to set the variable with < and > in it at all thats not been possible cuz the XML parser will barf on it usually. /b On Jul 16, 2009, at 2:20 PM, David Knell wrote: > Really? That line (+/- the IP address) came directly out of a working > dialplan. To be fair, the box is running a faintly prehistoric > FreeSWITCH - you crazy cats haven't been chewing on the tail of my > cherished mouse of backwards compatibility again, have you?! > > What has been incorrect in this discussion is the name of the header: > it's P-Asserted-Identity, not P-Asserted-ID. The fact that it's > usually > shortened to PAID doesn't help; nor does the fact that Remote-Party-ID > (which is deprecated, but still widely used for the same job as > P-Asserted-Identity) is about as well. > > --Dave From jens at vegeby.nu Thu Jul 16 12:36:35 2009 From: jens at vegeby.nu (Jens Vegeby) Date: Thu, 16 Jul 2009 21:36:35 +0200 Subject: [Freeswitch-users] ClueCon 2009 - News and Notes In-Reply-To: <87f2f3b90907161129m29921db0q611bb87b791ab522@mail.gmail.com> References: <87f2f3b90907161129m29921db0q611bb87b791ab522@mail.gmail.com> Message-ID: <30ee97110907161236n33bbfe39qcf1d8a009fded6b9@mail.gmail.com> Will there be video recordings availble online? On 7/16/09, Michael Collins wrote: > Only 19 more days until ClueCon! If you haven't registered then please do so > right away by calling 877.742.CLUE. This year's event is going to be awesome > - you don't want to miss it! > > Some news and notes: > We have two more sponsors for our event this year: Nokia and Twilio! We are > happy to welcome them both to the conference this year. Many of our sponsors > are supplying items for each attendee as well as some nice prizes that will > be raffled off at the end of events on Thursday afternoon. > > We also have a very special guest who will be speaking at ClueCon: Philip > Zimmermann! Philip is the author of PGP and co-author of ZRTP. We look > forward to his presentation on Wednesday. Another security expert, Dan York > of Voxeo, will also be speaking on Wednesday. All of those who are > interested in security will want to pay special attention to Wednesday's > program. > > Thanks for your support and we look forward to seeing you in a few weeks! > -Michael Collins > http://www.cluecon.com > 877.742.CLUE > -- Sent from my mobile device Mvh/Regards Jens From fdhege at gmail.com Thu Jul 16 12:42:59 2009 From: fdhege at gmail.com (Dale) Date: Thu, 16 Jul 2009 15:42:59 -0400 Subject: [Freeswitch-users] Setting P-Asserted-ID to something other than the callerid In-Reply-To: References: <723C30E7-4AD0-418D-A4B4-10148367A9AA@gmail.com> <1247755878.4289.64.camel@dk-d820> <1247772025.4289.88.camel@dk-d820> Message-ID: <53727858-AFDF-47EE-A0F8-5C69BB3DB204@gmail.com> Hmm, it seems to work on a copy of trunk from a few days ago. :) Both in the dialplan and the gateway config. But I could see how that might cause a problem with the xml. -Dale On Jul 16, 2009, at 3:27 PM, Brian West wrote: > Not sure how you were able to set the variable with < and > in it at > all thats not been possible cuz the XML parser will barf on it > usually. > > /b > > On Jul 16, 2009, at 2:20 PM, David Knell wrote: > >> Really? That line (+/- the IP address) came directly out of a >> working >> dialplan. To be fair, the box is running a faintly prehistoric >> FreeSWITCH - you crazy cats haven't been chewing on the tail of my >> cherished mouse of backwards compatibility again, have you?! >> >> What has been incorrect in this discussion is the name of the header: >> it's P-Asserted-Identity, not P-Asserted-ID. The fact that it's >> usually >> shortened to PAID doesn't help; nor does the fact that Remote-Party- >> ID >> (which is deprecated, but still widely used for the same job as >> P-Asserted-Identity) is about as well. >> >> --Dave > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.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 Jul 16 12:43:04 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 16 Jul 2009 14:43:04 -0500 Subject: [Freeswitch-users] ClueCon 2009 - News and Notes In-Reply-To: <30ee97110907161236n33bbfe39qcf1d8a009fded6b9@mail.gmail.com> References: <87f2f3b90907161129m29921db0q611bb87b791ab522@mail.gmail.com> <30ee97110907161236n33bbfe39qcf1d8a009fded6b9@mail.gmail.com> Message-ID: Yes their will be a couple of weeks after cluecon! /b On Jul 16, 2009, at 2:36 PM, Jens Vegeby wrote: > Will there be video recordings availble online? From msc at freeswitch.org Thu Jul 16 12:57:47 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 16 Jul 2009 12:57:47 -0700 Subject: [Freeswitch-users] Setting P-Asserted-ID to something other than the callerid In-Reply-To: <53727858-AFDF-47EE-A0F8-5C69BB3DB204@gmail.com> References: <723C30E7-4AD0-418D-A4B4-10148367A9AA@gmail.com> <1247755878.4289.64.camel@dk-d820> <1247772025.4289.88.camel@dk-d820> <53727858-AFDF-47EE-A0F8-5C69BB3DB204@gmail.com> Message-ID: <87f2f3b90907161257y65e39c07g595db93bd96b1fe3@mail.gmail.com> I think the parser is probably supposed to barf on consecutive < chars but it doesn't. I learned that the hard way when I left the closing tag off of an action in the dp. Consider this dp snippet: The above works but the 2nd log does not display. Just an FYI. -MC On Thu, Jul 16, 2009 at 12:42 PM, Dale wrote: > > Hmm, it seems to work on a copy of trunk from a few days ago. :) Both > in the dialplan and the gateway config. > > But I could see how that might cause a problem with the xml. > > -Dale > > On Jul 16, 2009, at 3:27 PM, Brian West wrote: > > > Not sure how you were able to set the variable with < and > in it at > > all thats not been possible cuz the XML parser will barf on it > > usually. > > > > /b > > > > On Jul 16, 2009, at 2:20 PM, David Knell wrote: > > > >> Really? That line (+/- the IP address) came directly out of a > >> working > >> dialplan. To be fair, the box is running a faintly prehistoric > >> FreeSWITCH - you crazy cats haven't been chewing on the tail of my > >> cherished mouse of backwards compatibility again, have you?! > >> > >> What has been incorrect in this discussion is the name of the header: > >> it's P-Asserted-Identity, not P-Asserted-ID. The fact that it's > >> usually > >> shortened to PAID doesn't help; nor does the fact that Remote-Party- > >> ID > >> (which is deprecated, but still widely used for the same job as > >> P-Asserted-Identity) is about as well. > >> > >> --Dave > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20090716/ac160dfc/attachment.html From tayeb.meftah at gmail.com Thu Jul 16 13:34:27 2009 From: tayeb.meftah at gmail.com (Meftah Tayeb) Date: Thu, 16 Jul 2009 20:34:27 +0000 Subject: [Freeswitch-users] Freeswitch ASR application example Message-ID: <4A5F8ED3.1060900@gmail.com> hello please cool anyone give me a Speech ASR script in LUA ? only recognise the speech and put it into a variable thanks! __________ Information from ESET NOD32 Antivirus, version of virus signature database 4251 (20090716) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com From jens at vegeby.nu Thu Jul 16 13:51:47 2009 From: jens at vegeby.nu (Jens Vegeby) Date: Thu, 16 Jul 2009 22:51:47 +0200 Subject: [Freeswitch-users] ClueCon 2009 - News and Notes In-Reply-To: References: <87f2f3b90907161129m29921db0q611bb87b791ab522@mail.gmail.com> <30ee97110907161236n33bbfe39qcf1d8a009fded6b9@mail.gmail.com> Message-ID: <30ee97110907161351p47f98e38xdb623f6ef897859e@mail.gmail.com> Great! On 7/16/09, Brian West wrote: > Yes their will be a couple of weeks after cluecon! > > /b > > On Jul 16, 2009, at 2:36 PM, Jens Vegeby wrote: > >> Will there be video recordings availble online? > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 Mvh/Regards Jens From krice at suspicious.org Thu Jul 16 13:55:00 2009 From: krice at suspicious.org (Ken Rice) Date: Thu, 16 Jul 2009 15:55:00 -0500 Subject: [Freeswitch-users] Setting P-Asserted-ID to something other than the callerid In-Reply-To: Message-ID: Hey Dale RFC4694 defines how to pass JIP LRN and CIC (and a few other things) as part of the tel URI... This has been adapted for use in SIP as the user part of the invite and from URIs... Theres also RFC3325 for PAID and if you check out the wikipedia page for IMS (IP Multimedia Subsystem) there are several other RFCs that are related for various "P headers" there. Most of these headers come out from the 3GPP/IMS working groups > From: Dale > Reply-To: > Date: Thu, 16 Jul 2009 13:04:35 -0400 > To: > Subject: Re: [Freeswitch-users] Setting P-Asserted-ID to something other than > the callerid > > > Hey, you happen to have the RFC number and section they are violating? > > I'll dig it up. But figured I'd ask since you all seem to have them > memorized by now. :) > > -Dale > > On Jul 16, 2009, at 10:41 AM, Ken Rice wrote: > >> That's just right down screwing with the standards... >> >> PAID is the caller id... This particular definition is from the RFCs >> and >> 3GPP docs for IMS which is why we have standardized P- headers... >> >> Can your vendor not look at the P-Charging-Vector field? >> >> Also, From when used with PAID is more like an ANI not a CLID >> >> >>> From: Dale >>> Reply-To: >>> Date: Thu, 16 Jul 2009 10:30:34 -0400 >>> To: >>> Subject: [Freeswitch-users] Setting P-Asserted-ID to something >>> other than the >>> callerid >>> >>> >>> Hello again, >>> >>> I wanted to first say thanks to Brain for helping me fix my from >>> domain issue the other day. It helped quite a bit. >>> >>> Now with more testing and talking with the vendor (please don't shoot >>> the messenger :) ) >>> >>> They want the caller id info in the from and the charge number/ >>> screening number in the P-Asserted-ID. >>> >>> I have tested this and verified that this does work like they say it >>> does by setting the callerid number to my charge number and setting >>> the from user in the gateway config to the callerid I want displayed. >>> But this solution doesn't scale very well. >>> >>> I know I can set the gateway option caller-id-in-from to get that >>> part >>> done. But is there a way to set the P-Asserted-ID to something other >>> than the callerid? >>> >>> Any hints would be welcomed. >>> >>> Thanks, >>> >>> -Dale >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From pjintheusa at gmail.com Thu Jul 16 14:07:32 2009 From: pjintheusa at gmail.com (Phillip Jones) Date: Thu, 16 Jul 2009 17:07:32 -0400 Subject: [Freeswitch-users] .NET demo / hangupHook() Message-ID: <367751820907161407x57c9f81cwe4465ad1c1e20bb6@mail.gmail.com> Hi there, I am looking at the FreeSWITCH.Demo.AppDemo class. Have it running nicely from a soft phone - all is well. However I am not seeing the hangupHook() method fired, when I hangup. Debug log for an example call is at: http://pastebin.freeswitch.org/9744 Reminder of demo code is here: http://pastebin.freeswitch.org/9745 dialplan.xml is simply: Are there changes to the demo required to get this method firing? Any help much appreciated. Phillip Jones -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090716/f42ef880/attachment.html From raul at etellicom.com Thu Jul 16 14:15:00 2009 From: raul at etellicom.com (Raul Fragoso) Date: Thu, 16 Jul 2009 18:15:00 -0300 Subject: [Freeswitch-users] sip extermal profile for all IP In-Reply-To: <8ccbff060907160630r2188c8d8od0c5c677f81eb71b@mail.gmail.com> References: <8ccbff060907160630r2188c8d8od0c5c677f81eb71b@mail.gmail.com> Message-ID: <1247778900.21971.8.camel@raul-laptop> You can not do that with a single profile. Each profile is bound to only one local IP, so if you need to bind to more than one you will have to create a new profile and set the specific sip-ip/rtp-ip params for them. Regards, RAul On Thu, 2009-07-16 at 20:30 +0700, freeswitch-users at lists.freeswitch.org wrote: > Dear All, > > How to set ${local_ip_v4} to 0.0.0.0 ? i want to set up > external profile for All IP > > > Best regards. > > Dome C. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From tayeb.meftah at gmail.com Thu Jul 16 14:17:05 2009 From: tayeb.meftah at gmail.com (Meftah Tayeb) Date: Thu, 16 Jul 2009 21:17:05 +0000 Subject: [Freeswitch-users] Freeswitch ASR application example Message-ID: <4A5F98D1.2060909@gmail.com> hello please cool anyone give me a Speech ASR script in LUA ? only recognise the speech and put it into a variable thanks! __________ Information from ESET NOD32 Antivirus, version of virus signature database 4251 (20090716) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com From mgg at giagnocavo.net Thu Jul 16 14:32:07 2009 From: mgg at giagnocavo.net (Michael Giagnocavo) Date: Thu, 16 Jul 2009 17:32:07 -0400 Subject: [Freeswitch-users] .NET demo / hangupHook() In-Reply-To: <367751820907161407x57c9f81cwe4465ad1c1e20bb6@mail.gmail.com> References: <367751820907161407x57c9f81cwe4465ad1c1e20bb6@mail.gmail.com> Message-ID: <6E8D2069C08AA84A83D336E996AE4C67027C6C33E7@mse17be1.mse17.exchange.ms> The debug log has this: 65.2009-07-16 16:48:41.432200 [DEBUG] switch_cpp.cpp:1124 AppFunction is in hangupCallback. 66.2009-07-16 16:48:41.432200 [WARNING] switch_cpp.cpp:1124 Thread will not be aborted because Hangup was called from the Run thread. The problem is the Demo doesn't have code to actually set hangupHook as the handler. Adding something like this to the app demo code: Session.HangupFunction = hangupHook; Should fix it. -Michael From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Phillip Jones Sent: Thursday, July 16, 2009 3:08 PM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] .NET demo / hangupHook() Hi there, I am looking at the FreeSWITCH.Demo.AppDemo class. Have it running nicely from a soft phone - all is well. However I am not seeing the hangupHook() method fired, when I hangup. Debug log for an example call is at: http://pastebin.freeswitch.org/9744 Reminder of demo code is here: http://pastebin.freeswitch.org/9745 dialplan.xml is simply: Are there changes to the demo required to get this method firing? Any help much appreciated. Phillip Jones -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090716/f64e98b5/attachment.html From larclap at yahoo.com Thu Jul 16 14:36:14 2009 From: larclap at yahoo.com (Lars Zeb) Date: Thu, 16 Jul 2009 14:36:14 -0700 Subject: [Freeswitch-users] Error in lua script with session:getVariable Message-ID: <001801ca065d$71f669e0$55e33da0$@com> I am getting an error in a lua script which I don't understand. Why is it returning nil in the script yet something in the cli? lua snippet: user_data = session:getVariable('user_data 1000 at 192.168.10.29 var callgroup'); freeswitch.console_log("INFO", " UserData group " .. user_data .. "\n") log: 2009-07-16 14:30:49.237343 [ERR] mod_lua.cpp:182 /usr/local/freeswitch/scripts/helloworld.lua:14: attempt to concatenate global 'user_data' (a nil value) stack traceback: /usr/local/freeswitch/scripts/helloworld.lua:14: in main chunk cli: freeswitch at internal> user_data 1000 at 192.168.10.29 var callgroup techsupport Thanks, Lars -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090716/fb3edfb5/attachment.html From mrene_lists at avgs.ca Thu Jul 16 14:42:19 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Thu, 16 Jul 2009 17:42:19 -0400 Subject: [Freeswitch-users] Error in lua script with session:getVariable In-Reply-To: <001801ca065d$71f669e0$55e33da0$@com> References: <001801ca065d$71f669e0$55e33da0$@com> Message-ID: <55B40657-61AE-46DE-952D-E5F496048B88@avgs.ca> You need to make an API call, not get a variable. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca Am 16-Jul-09 um 5:36 PM schrieb Lars Zeb: > I am getting an error in a lua script which I don?t understand. Why > is it returning nil in the script yet something in the cli? > > lua snippet: > user_data = session:getVariable('user_data 1000 at 192.168.10.29 var > callgroup'); > freeswitch.console_log("INFO", " UserData group " .. user_data .. > "\n") > > log: > 2009-07-16 14:30:49.237343 [ERR] mod_lua.cpp:182 /usr/local/ > freeswitch/scripts/helloworld.lua:14: attempt to concatenate global > 'user_data' (a nil value) > stack traceback: > /usr/local/freeswitch/scripts/helloworld.lua:14: in main chunk > > > cli: > freeswitch at internal> user_data 1000 at 192.168.10.29 var callgroup > techsupport > > Thanks, Lars > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20090716/68533c3d/attachment.html From mgg at giagnocavo.net Thu Jul 16 14:43:08 2009 From: mgg at giagnocavo.net (Michael Giagnocavo) Date: Thu, 16 Jul 2009 17:43:08 -0400 Subject: [Freeswitch-users] mod_managed users? Message-ID: <6E8D2069C08AA84A83D336E996AE4C67027C6C33F0@mse17be1.mse17.exchange.ms> Hey, if there are any mod_managed users on this list, I'd love it if you were able to let me know. I'd like to get feedback, positive or negative, on what worked, what didn't, and how mod_managed can improve for you. Feel free to write on list or directly to me: mgg at giagnocavo.net Thanks! -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090716/de66d029/attachment.html From dftoro at yahoo.com Thu Jul 16 14:54:30 2009 From: dftoro at yahoo.com (Diego Toro) Date: Thu, 16 Jul 2009 14:54:30 -0700 (PDT) Subject: [Freeswitch-users] mod_managed users? In-Reply-To: <6E8D2069C08AA84A83D336E996AE4C67027C6C33F0@mse17be1.mse17.exchange.ms> Message-ID: <36860.21084.qm@web33505.mail.mud.yahoo.com> Hey, I am here? :) ? I am working with mod_managed on Windows 2003 and Windows Vista with sucessfull.? I noted on user list the issue with LoadFile on Loader.cs when a assembly had reference to others assemblies, I change LoadFile by LoadFrom and the load is made fine. ? I use c# application and sqlserver 2005, using FS and mod_managed. ? Diego --- On Thu, 7/16/09, Michael Giagnocavo wrote: From: Michael Giagnocavo Subject: [Freeswitch-users] mod_managed users? To: "freeswitch-users at lists.freeswitch.org" Date: Thursday, July 16, 2009, 4:43 PM Hey, if there are any mod_managed users on this list, I?d love it if you were able to let me know. I?d like to get feedback, positive or negative, on what worked, what didn?t, and how mod_managed can improve for you. Feel free to write on list or directly to me: mgg at giagnocavo.net ? Thanks! -Michael -----Inline Attachment Follows----- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at 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/20090716/c759066b/attachment.html From tayeb.meftah at gmail.com Thu Jul 16 14:56:00 2009 From: tayeb.meftah at gmail.com (Meftah Tayeb) Date: Thu, 16 Jul 2009 21:56:00 +0000 Subject: [Freeswitch-users] mod_managed users? In-Reply-To: <6E8D2069C08AA84A83D336E996AE4C67027C6C33F0@mse17be1.mse17.exchange.ms> References: <6E8D2069C08AA84A83D336E996AE4C67027C6C33F0@mse17be1.mse17.exchange.ms> Message-ID: <4A5FA1F0.90202@gmail.com> hello, yes, i'm trying to develope modules using Mod_Managed is working very perfectly, except that i don't know how i can run it using MONO (no .Net framework) also i'm trying to develope a EndPoint using it thanks Michael Giagnocavo wrote: > > Hey, if there are any mod_managed users on this list, I'd love it if > you were able to let me know. I'd like to get feedback, positive or > negative, on what worked, what didn't, and how mod_managed can improve > for you. Feel free to write on list or directly to me: mgg at > giagnocavo.net > > > > Thanks! > > -Michael > > > > __________ Information from ESET NOD32 Antivirus, version of virus > signature database 4251 (20090716) __________ > > The message was checked by ESET NOD32 Antivirus. > > http://www.eset.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 > > > > __________ Information from ESET NOD32 Antivirus, version of virus signature database 4251 (20090716) __________ > > The message was checked by ESET NOD32 Antivirus. > > http://www.eset.com > > __________ Information from ESET NOD32 Antivirus, version of virus signature database 4251 (20090716) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090716/c19f14f4/attachment.html From pjintheusa at gmail.com Thu Jul 16 16:25:09 2009 From: pjintheusa at gmail.com (Phillip Jones) Date: Thu, 16 Jul 2009 19:25:09 -0400 Subject: [Freeswitch-users] .NET demo / hangupHook() In-Reply-To: <6E8D2069C08AA84A83D336E996AE4C67027C6C33E7@mse17be1.mse17.exchange.ms> References: <367751820907161407x57c9f81cwe4465ad1c1e20bb6@mail.gmail.com> <6E8D2069C08AA84A83D336E996AE4C67027C6C33E7@mse17be1.mse17.exchange.ms> Message-ID: <367751820907161625m1b040984ufe39e83d64bd16dc@mail.gmail.com> Perfect - thank you very much! On Thu, Jul 16, 2009 at 5:32 PM, Michael Giagnocavo wrote: > The debug log has this: > > 65.2009-07-16 16:48:41.432200 [DEBUG] switch_cpp.cpp:1124 AppFunction is in > hangupCallback. > > 66.2009-07-16 16:48:41.432200 [WARNING] switch_cpp.cpp:1124 Thread will not > be aborted because Hangup was called from the Run thread. > > > > The problem is the Demo doesn?t have code to actually set hangupHook as the > handler. Adding something like this to the app demo code: > > > > Session.HangupFunction = hangupHook; > > > > Should fix it. > > > > -Michael > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Phillip > Jones > *Sent:* Thursday, July 16, 2009 3:08 PM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* [Freeswitch-users] .NET demo / hangupHook() > > > > Hi there, > > I am looking at the FreeSWITCH.Demo.AppDemo class. Have it running nicely > from a soft phone - all is well. > > However I am not seeing the hangupHook() method fired, when I hangup. > > Debug log for an example call is at: http://pastebin.freeswitch.org/9744 > > Reminder of demo code is here: http://pastebin.freeswitch.org/9745 > > dialplan.xml is simply: > > > > > > > > Are there changes to the demo required to get this method firing? > > > Any help much appreciated. > > > Phillip Jones > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20090716/41e66cb9/attachment.html From larclap at yahoo.com Thu Jul 16 17:07:09 2009 From: larclap at yahoo.com (Lars Zeb) Date: Thu, 16 Jul 2009 17:07:09 -0700 Subject: [Freeswitch-users] Error in lua script with session:getVariable In-Reply-To: <55B40657-61AE-46DE-952D-E5F496048B88@avgs.ca> References: <001801ca065d$71f669e0$55e33da0$@com> <55B40657-61AE-46DE-952D-E5F496048B88@avgs.ca> Message-ID: <006501ca0672$87039dc0$950ad940$@com> Mathieu, Thanks for the reply. I'm very new with FreeSWITCH and not familiar with api calls. I tried: user_data = apiExecute("user_data", "1000 at 192.168.10.29 var callgroup"); But got a similar error: 2009-07-16 17:01:47.212904 [ERR] mod_lua.cpp:182 /usr/local/freeswitch/scripts/helloworld.lua:13: attempt to call global 'apiExecute' (a nil value) stack traceback: /usr/local/freeswitch/scripts/helloworld.lua:13: in main chunk I would appreciate an example or a link to the pertinent documentation. Thanks, Lars From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mathieu Rene Sent: Thursday, July 16, 2009 2:42 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Error in lua script with session:getVariable You need to make an API call, not get a variable. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca Am 16-Jul-09 um 5:36 PM schrieb Lars Zeb: I am getting an error in a lua script which I don't understand. Why is it returning nil in the script yet something in the cli? lua snippet: user_data = session:getVariable('user_data 1000 at 192.168.10.29 var callgroup'); freeswitch.console_log("INFO", " UserData group " .. user_data .. "\n") log: 2009-07-16 14:30:49.237343 [ERR] mod_lua.cpp:182 /usr/local/freeswitch/scripts/helloworld.lua:14: attempt to concatenate global 'user_data' (a nil value) stack traceback: /usr/local/freeswitch/scripts/helloworld.lua:14: in main chunk cli: freeswitch at internal> user_data 1000 at 192.168.10.29 var callgroup techsupport Thanks, Lars _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at 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/20090716/13ab0e0a/attachment.html From brian at freeswitch.org Thu Jul 16 17:11:28 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 16 Jul 2009 19:11:28 -0500 Subject: [Freeswitch-users] Error in lua script with session:getVariable In-Reply-To: <006501ca0672$87039dc0$950ad940$@com> References: <001801ca065d$71f669e0$55e33da0$@com> <55B40657-61AE-46DE-952D-E5F496048B88@avgs.ca> <006501ca0672$87039dc0$950ad940$@com> Message-ID: <0F52BC07-1CFE-4C0D-8A29-91D3C6F18D9B@freeswitch.org> http://wiki.freeswitch.org/wiki/Mod_lua#For_making_API_calls Please READ that page. These details are there! /b On Jul 16, 2009, at 7:07 PM, Lars Zeb wrote: > Mathieu, > > Thanks for the reply. I?m very new with FreeSWITCH and not familiar > with api calls. I tried: > > user_data = apiExecute("user_data", "1000 at 192.168.10.29 var > callgroup"); > > But got a similar error: > > 2009-07-16 17:01:47.212904 [ERR] mod_lua.cpp:182 /usr/local/ > freeswitch/scripts/helloworld.lua:13: attempt to call global > 'apiExecute' (a nil value) > stack traceback: > /usr/local/freeswitch/scripts/helloworld.lua:13: in main chunk > > I would appreciate an example or a link to the pertinent > documentation. > > Thanks, Lars > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Mathieu Rene > Sent: Thursday, July 16, 2009 2:42 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Error in lua script with > session:getVariable > > You need to make an API call, not get a variable. > > Mathieu Rene > Avant-Garde Solutions Inc > Office: + 1 (514) 664-1044 x100 > Cell: +1 (514) 664-1044 x200 > mrene at avgs.ca > > > > > Am 16-Jul-09 um 5:36 PM schrieb Lars Zeb: > > > I am getting an error in a lua script which I don?t understand. Why > is it returning nil in the script yet something in the cli? > > lua snippet: > user_data = session:getVariable('user_data 1000 at 192.168.10.29 var > callgroup'); > freeswitch.console_log("INFO", " UserData group " .. user_data .. > "\n") > > log: > 2009-07-16 14:30:49.237343 [ERR] mod_lua.cpp:182 /usr/local/ > freeswitch/scripts/helloworld.lua:14: attempt to concatenate global > 'user_data' (a nil value) > stack traceback: > /usr/local/freeswitch/scripts/helloworld.lua:14: in main chunk > > > cli: > freeswitch at internal> user_data 1000 at 192.168.10.29 var callgroup > techsupport > > Thanks, Lars > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20090716/f4c15437/attachment.html From dome at tel.co.th Thu Jul 16 18:19:35 2009 From: dome at tel.co.th (Dome Charoenyost) Date: Fri, 17 Jul 2009 08:19:35 +0700 Subject: [Freeswitch-users] sip extermal profile for all IP In-Reply-To: <1247778900.21971.8.camel@raul-laptop> References: <8ccbff060907160630r2188c8d8od0c5c677f81eb71b@mail.gmail.com> <1247778900.21971.8.camel@raul-laptop> Message-ID: <8ccbff060907161819o4b2fc44cp5087ec5553cdd2f8@mail.gmail.com> 2009/7/17 Raul Fragoso : > You can not do that with a single profile. Each profile is bound to only > one local IP, so if you need to bind to more than one you will have to > create a new profile and set the specific sip-ip/rtp-ip params for them. > Thanks. > Regards, > > RAul > > On Thu, 2009-07-16 at 20:30 +0700, freeswitch-users at lists.freeswitch.org > wrote: >> Dear All, >> >> ? ? ? ? ?How to set ${local_ip_v4} to 0.0.0.0 ? i want to set up >> external profile for All IP >> >> >> Best regards. >> >> Dome C. >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From shaheryarkh at googlemail.com Thu Jul 16 21:59:53 2009 From: shaheryarkh at googlemail.com (Muhammad Shahzad) Date: Fri, 17 Jul 2009 10:59:53 +0600 Subject: [Freeswitch-users] mod_managed users? In-Reply-To: <36860.21084.qm@web33505.mail.mud.yahoo.com> References: <6E8D2069C08AA84A83D336E996AE4C67027C6C33F0@mse17be1.mse17.exchange.ms> <36860.21084.qm@web33505.mail.mud.yahoo.com> Message-ID: I tried to install mod_managed on ubuntu-9.04, mono framework version 2.0. It gave me a lots of errors in Loader.cs, which seems to be SWIG related. Since i am not a expert in SWIG so i disabled this module. This happend long ago, i think FS svn revision 136xx. Let me try to compile it from latest FS revision and see if it works. I will let you know the results. Thank you. On Fri, Jul 17, 2009 at 3:54 AM, Diego Toro wrote: > Hey, I am here :) > > I am working with mod_managed on Windows 2003 and Windows Vista with > sucessfull. I noted on user list the issue with LoadFile on Loader.cs when > a assembly had reference to others assemblies, I change LoadFile by LoadFrom > and the load is made fine. > > I use c# application and sqlserver 2005, using FS and mod_managed. > > Diego > > --- On *Thu, 7/16/09, Michael Giagnocavo * wrote: > > > From: Michael Giagnocavo > Subject: [Freeswitch-users] mod_managed users? > To: "freeswitch-users at lists.freeswitch.org" < > freeswitch-users at lists.freeswitch.org> > Date: Thursday, July 16, 2009, 4:43 PM > > Hey, if there are any mod_managed users on this list, I?d love it if you > were able to let me know. I?d like to get feedback, positive or negative, on > what worked, what didn?t, and how mod_managed can improve for you. Feel free > to write on list or directly to me: mgg at giagnocavo.net > > > > Thanks! > > -Michael > -----Inline Attachment Follows----- > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090717/bab4453a/attachment.html From nicolas at medularis.com Fri Jul 17 13:35:34 2009 From: nicolas at medularis.com (Nicolas Brenner) Date: Fri, 17 Jul 2009 16:35:34 -0400 Subject: [Freeswitch-users] No credit = NETWORK_OUT_OF_ORDER ? Message-ID: <1b46b4e80907171335j729505c3oae8492b6582af8b8@mail.gmail.com> Hi, Today I ran out of credit in one of my voip providers. When this happened, all my outgoing calls started failing with hangup cause NETWORK_OUT_OF_ORDER. Once I got some more credit, the calls kept failing. I restarted freeswitch and then everything worked fine again. Unfortunately this is not something I'd like to reproduce, and the only thing I have is the logs (no SIP trace). But I was wodering if someone here has had a similar experience or could tell if this is plausible or even likely to happen. Another part of the platform I'm running, runs on Asterisk, using the same voip providers, nevertheless the calls originating there only failed during the no credit period, and began working again automatically as soon as credit was added to the account. Thanks, Nicolas -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090717/1d33b510/attachment.html From brian at freeswitch.org Fri Jul 17 13:58:47 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 17 Jul 2009 15:58:47 -0500 Subject: [Freeswitch-users] No credit = NETWORK_OUT_OF_ORDER ? In-Reply-To: <1b46b4e80907171335j729505c3oae8492b6582af8b8@mail.gmail.com> References: <1b46b4e80907171335j729505c3oae8492b6582af8b8@mail.gmail.com> Message-ID: <305E86FA-95BC-4D04-837D-AAAC291B34B1@freeswitch.org> Open a jira with everything you can provide.. I'll try my best to reproduce the issue. /b On Jul 17, 2009, at 3:35 PM, Nicolas Brenner wrote: > Hi, > > Today I ran out of credit in one of my voip providers. When this > happened, all my outgoing calls started failing with hangup cause > NETWORK_OUT_OF_ORDER. Once I got some more credit, the calls kept > failing. I restarted freeswitch and then everything worked fine again. > > Unfortunately this is not something I'd like to reproduce, and the > only thing I have is the logs (no SIP trace). But I was wodering if > someone here has had a similar experience or could tell if this is > plausible or even likely to happen. > > Another part of the platform I'm running, runs on Asterisk, using > the same voip providers, nevertheless the calls originating there > only failed during the no credit period, and began working again > automatically as soon as credit was added to the account. > > Thanks, > > Nicolas > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.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 Fri Jul 17 14:13:26 2009 From: nicolas at medularis.com (Nicolas Brenner) Date: Fri, 17 Jul 2009 17:13:26 -0400 Subject: [Freeswitch-users] No credit = NETWORK_OUT_OF_ORDER ? In-Reply-To: <1b46b4e80907171335j729505c3oae8492b6582af8b8@mail.gmail.com> References: <1b46b4e80907171335j729505c3oae8492b6582af8b8@mail.gmail.com> Message-ID: <1b46b4e80907171413i260e70d4u43d4b1d88fe9a492@mail.gmail.com> A little bit more info: When the calls failed, the following was recorded in the log: 2009-07-17 15:19:07.880175 [ERR] switch_ivr_originate.c:1495 Cannot create outgoing channel of type [sofia] cause: [NETWORK_OUT_OF_ORDER] 2009-07-17 15:19:07.880175 [DEBUG] switch_ivr_originate.c:2123 Originate Resulted in Error Cause: 38 [NETWORK_OUT_OF_ORDER] 2009-07-17 15:19:07.880175 [WARNING] mod_spidermonkey.c:3013 Cannot Create Outgoing Channel! [{ignore_early_media=true,originate_timeout=30,execute_on_answer='sched_hangup +30 ALLOTED_TIMEOUT'}sofia/gateway/mygateway/005698793046] 2009-07-17 15:19:07.880175 [NOTICE] new_energizer_async.js:15 *********** CAUSE: NETWORK_OUT_OF_ORDER *********** 2009-07-17 15:19:34.158980 [NOTICE] sofia_reg.c:319 Registering mygateway 2009-07-17 15:19:34.294518 [ERR] sofia_reg.c:1445 mygateway Registration Failed with status Operation has no matching challenge [904]. failure #37 2009-07-17 15:19:34.365065 [WARNING] sofia_reg.c:348 mygateway Failed Registration, setting retry to 190 seconds. I searched for the "Registration Failed with status Operation has no matching challenge" error on the list, and someone else had a similar issue, but apparently it had something to do with NAT, and in this case there's no NAT involved. Anyway, I'm running a rev 13973 so I'll update to the latest svn rev and hope it doesn't happen again. On Fri, Jul 17, 2009 at 4:35 PM, Nicolas Brenner wrote: > Hi, > > Today I ran out of credit in one of my voip providers. When this happened, > all my outgoing calls started failing with hangup cause > NETWORK_OUT_OF_ORDER. Once I got some more credit, the calls kept failing. I > restarted freeswitch and then everything worked fine again. > > Unfortunately this is not something I'd like to reproduce, and the only > thing I have is the logs (no SIP trace). But I was wodering if someone here > has had a similar experience or could tell if this is plausible or even > likely to happen. > > Another part of the platform I'm running, runs on Asterisk, using the same > voip providers, nevertheless the calls originating there only failed during > the no credit period, and began working again automatically as soon as > credit was added to the account. > > Thanks, > > Nicolas > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090717/6e043f87/attachment.html From lfurrea at gmail.com Fri Jul 17 16:58:12 2009 From: lfurrea at gmail.com (Luis F Urrea) Date: Fri, 17 Jul 2009 17:58:12 -0600 Subject: [Freeswitch-users] mod_conference: behavior when the conference is torn down before 31seconds Message-ID: Hi all, I am experiencing a behavior that I cannot clearly understand. Basically I "autocall" a few phones into a conference with the sip_auto_answer set to true, as follows: The conference establishes just fine and everyone can hear just fine. The "strange" behavior comes when the person calling to ext 773 hangs up before 31 seconds have passed, the rest of the phones stay up until they reach second 31 into the "conference". I am using snom phones and I see the BYE message arriving at the phones exactly at second 31 after the call establishes. The conference itself however does not exist after the person calling 773 hangs up (doing conference list on CLI shows NO active conferences). If the conference stays up more than 31 seconds, then when the person calling 773 hangs up, the rest of the phones hang up immediately as desired. Here's the log for a "page" that lasts less than 31 seconds: http://pastebin.freeswitch.org/9773 Here's the log of the phone for a "page" that lasts less than 31 seconds: http://pastebin.freeswitch.org/9774 Your inout is appreciated. Regards, Luis -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090717/f1ca8d62/attachment.html From jaybinks at gmail.com Fri Jul 17 20:49:27 2009 From: jaybinks at gmail.com (Jay Binks) Date: Sat, 18 Jul 2009 13:49:27 +1000 Subject: [Freeswitch-users] 302 redirects and continue_on_fail=true Message-ID: <0A05EBC4-8DC0-4D4E-9029-E1BA2D5E443F@gmail.com> I have an upstream provider that utilizes a load balancer that spits back 302 redirects with contact headers SIP/2.0 302 Moved Temporarily Via: SIP/2.0/UDP xxx.xxx.xxx.xxx: 5080;rport=5080;branch=z9hG4bKB6peFDvXZ5S2F;received=xxx.xxx.xxx.xxx From: "test" ;tag=ByFF2244HHvmj To: ;tag=1288540-274799759-385876096-3219999652 Call-ID: f6dc8d30-edd7-122c-c98e-000e7f301839 CSeq: 117820807 INVITE Contact: Server: MERA MVTS3G v.3.10.2-49-Release Content-Length: 0 in my dialplan I have multiple upstream suppliers in a failover setup so I setup some vars and sip headers then attempt the bridge. if it fails I then go on to do the same thing for a few other suppliers ( setup headers, attempt bridge ) so because of this I use continue_on_fail=true it appears Freeswitch sees the 302 as a temp failure and does not follow the redirect, and instead moves on to the next upstream and bridges there. ive read that I can selectively exclude temporary failures from continue_on_fail but im not sure thats exact enough for this situation. I do wish for continue_on_fail to ignore 302 moved temporarily but not ALL temporary failures ( for which there are probably many more causes ) any help would be greatly appreciated as Im not sure the best way to resolve this. Jay -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090718/007c8854/attachment.html From markmorreny at gmail.com Fri Jul 17 21:04:26 2009 From: markmorreny at gmail.com (mark morreny) Date: Sat, 18 Jul 2009 12:04:26 +0800 Subject: [Freeswitch-users] freeswitch on blackfin Message-ID: <20ad6b920907172104x3354cacal9567d91576b8ef75@mail.gmail.com> Hi, Have anyone tried getting freeswitch to work on uclinux/blackfin platform? Is there any info out there on how that can be done? Thanks for any info. Best Regards, Mark -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090718/328408bd/attachment.html From jmesquita at gmail.com Fri Jul 17 21:44:13 2009 From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Sat, 18 Jul 2009 01:44:13 -0300 Subject: [Freeswitch-users] FsGUI Message-ID: <5a8712120907172144u991d4c7o2b743116877ef0c0@mail.gmail.com> Dear folks, Even tho it might be premature, I would like to already spread the word. Check out FsGUI and feel free give feedback if this is a wanted tool and what direction it should take. Beware that the code is still contrib code and might now be yet mature for production use. http://wiki.freeswitch.org/wiki/Fsgui Thanks, Jo?o Mesquita -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090718/a6215ed7/attachment.html From steveu at coppice.org Fri Jul 17 22:19:12 2009 From: steveu at coppice.org (Steve Underwood) Date: Sat, 18 Jul 2009 13:19:12 +0800 Subject: [Freeswitch-users] freeswitch on blackfin In-Reply-To: <20ad6b920907172104x3354cacal9567d91576b8ef75@mail.gmail.com> References: <20ad6b920907172104x3354cacal9567d91576b8ef75@mail.gmail.com> Message-ID: <4A615B50.30703@coppice.org> mark morreny wrote: > Hi, > > Have anyone tried getting freeswitch to work on uclinux/blackfin > platform? > > Is there any info out there on how that can be done? > > Thanks for any info. > Look in the mailing list archive. This question comes up regularly. Steve From dome at tel.co.th Fri Jul 17 23:19:15 2009 From: dome at tel.co.th (Dome Charoenyost) Date: Sat, 18 Jul 2009 13:19:15 +0700 Subject: [Freeswitch-users] Nibble bill in B Leg Message-ID: <8ccbff060907172319s7cad6446o59887029a0c9dc42@mail.gmail.com> Dear sir, I found some problem when try to enable nibblebill in B-Leg My Dialplan problem is niblle do nothing until hanup call. i try to debug nibblebill and found some issue. nibblebill can't get billrate , billaccount from channel billrate = switch_channel_get_variable(channel, "nibble_rate"); billaccount = switch_channel_get_variable(channel, "nibble_account"); if (!billrate || !billaccount) { return SWITCH_STATUS_SUCCESS; } Dome C. From tayeb.meftah at gmail.com Sat Jul 18 02:25:24 2009 From: tayeb.meftah at gmail.com (Meftah Tayeb) Date: Sat, 18 Jul 2009 09:25:24 +0000 Subject: [Freeswitch-users] freeswitch on blackfin In-Reply-To: <20ad6b920907172104x3354cacal9567d91576b8ef75@mail.gmail.com> References: <20ad6b920907172104x3354cacal9567d91576b8ef75@mail.gmail.com> Message-ID: <4A619504.60503@gmail.com> hi mark, please go to the freeswitch web site (http://www.freeswitch.org and open Download / Install Guide. this contin a step by step guide to cross compil Freeswitch for Embedded system thanks; mark morreny wrote: > Hi, > > Have anyone tried getting freeswitch to work on uclinux/blackfin > platform? > > Is there any info out there on how that can be done? > > Thanks for any info. > > > Best Regards, > > 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 > > > > __________ Information from ESET NOD32 Antivirus, version of virus signature database 4256 (20090718) __________ > > The message was checked by ESET NOD32 Antivirus. > > http://www.eset.com > > __________ Information from ESET NOD32 Antivirus, version of virus signature database 4256 (20090718) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090718/c96c97d5/attachment.html From dome at tel.co.th Sat Jul 18 02:48:22 2009 From: dome at tel.co.th (Dome Charoenyost) Date: Sat, 18 Jul 2009 16:48:22 +0700 Subject: [Freeswitch-users] Nibble bill in B Leg In-Reply-To: <8ccbff060907172319s7cad6446o59887029a0c9dc42@mail.gmail.com> References: <8ccbff060907172319s7cad6446o59887029a0c9dc42@mail.gmail.com> Message-ID: <8ccbff060907180248l6d5a2dbfoa127d928b62e34@mail.gmail.com> I found enable_heartbeat_events variable can help this case. it's work fine. I update http://wiki.freeswitch.org/wiki/Mod_nibblebill already Dome C. 2009/7/18 Dome Charoenyost : > Dear sir, > > ? ? ?I found some problem when try to enable nibblebill in B-Leg > ? ? ?My Dialplan > > ? ? ? data="{nibble_rate=1,nibble_account=0838833133}sofia/external/191$1 at 203.xxx.xxx.xxx" > /> > > ? ? ?problem is niblle do nothing until hanup call. i try to debug > nibblebill and found some issue. nibblebill can't get billrate , > billaccount from channel > > ? ? ? ?billrate = switch_channel_get_variable(channel, "nibble_rate"); > ? ? ? ?billaccount = switch_channel_get_variable(channel, "nibble_account"); > ? ? ? ?if (!billrate || !billaccount) { > ? ? ? ? ? ? ? ?return SWITCH_STATUS_SUCCESS; > ? ? ? ?} > > > Dome C. > From hads at nice.net.nz Sat Jul 18 02:59:44 2009 From: hads at nice.net.nz (Hadley Rich) Date: Sat, 18 Jul 2009 21:59:44 +1200 Subject: [Freeswitch-users] freeswitch on blackfin In-Reply-To: <4A619504.60503@gmail.com> References: <20ad6b920907172104x3354cacal9567d91576b8ef75@mail.gmail.com> <4A619504.60503@gmail.com> Message-ID: <1247911184.1988.8.camel@sodium> On Sat, 2009-07-18 at 09:25 +0000, Meftah Tayeb wrote: > please go to the freeswitch web site (http://www.freeswitch.org and > open Download / Install Guide. > this contin a step by step guide to cross compil Freeswitch for > Embedded system That's not for blackfin. hads -- http://nicegear.co.nz New Zealand's Open Source Hardware Supplier From Prometheus001 at gmx.net Sat Jul 18 05:01:23 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Sat, 18 Jul 2009 14:01:23 +0200 Subject: [Freeswitch-users] mod_conference: behavior when the conference is torn down before 31seconds In-Reply-To: References: Message-ID: <4A61B993.6090402@gmx.net> Hello Luis, are you using encrypted TLS instead on SIP on this phone? I experienced a similar behaviour with 31 seocnds on TLS. Best regards Peter Luis F Urrea schrieb: > Hi all, > > I am experiencing a behavior that I cannot clearly understand. > Basically I "autocall" a few phones into a conference with the > sip_auto_answer set to true, as follows: > > > > data="conference_auto_outcall_prefix={sip_auto_answer=true}"/> > data="user/305"/> > data="user/303"/> > data="user/201"/> > > > > > > The conference establishes just fine and everyone can hear just fine. > > The "strange" behavior comes when the person calling to ext 773 hangs > up before 31 seconds have passed, the rest of the phones stay up until > they reach second 31 into the "conference". > > I am using snom phones and I see the BYE message arriving at the > phones exactly at second 31 after the call establishes. > > The conference itself however does not exist after the person calling > 773 hangs up (doing conference list on CLI shows NO active conferences). > > If the conference stays up more than 31 seconds, then when the person > calling 773 hangs up, the rest of the phones hang up immediately as > desired. > > Here's the log for a "page" that lasts less than 31 seconds: > > http://pastebin.freeswitch.org/9773 > > Here's the log of the phone for a "page" that lasts less than 31 seconds: > > http://pastebin.freeswitch.org/9774 > > Your inout is appreciated. > > Regards, > > 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 Prometheus001 at gmx.net Sat Jul 18 05:06:00 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Sat, 18 Jul 2009 14:06:00 +0200 Subject: [Freeswitch-users] FsGUI In-Reply-To: <5a8712120907172144u991d4c7o2b743116877ef0c0@mail.gmail.com> References: <5a8712120907172144u991d4c7o2b743116877ef0c0@mail.gmail.com> Message-ID: <4A61BAA8.7080403@gmx.net> Thanks, I have found the sources in contrib/jmesquita/fsgui Any recommendatioins how to compile it under Linux? Best regards Peter Jo?o Mesquita schrieb: > Dear folks, > > Even tho it might be premature, I would like to already spread the word. > > Check out FsGUI and feel free give feedback if this is a wanted tool > and what direction it should take. Beware that the code is still > contrib code and might now be yet mature for production use. > > http://wiki.freeswitch.org/wiki/Fsgui > > Thanks, > > Jo?o Mesquita > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From shaheryarkh at googlemail.com Sat Jul 18 05:57:37 2009 From: shaheryarkh at googlemail.com (Muhammad Shahzad) Date: Sat, 18 Jul 2009 18:57:37 +0600 Subject: [Freeswitch-users] mod_managed users? In-Reply-To: References: <6E8D2069C08AA84A83D336E996AE4C67027C6C33F0@mse17be1.mse17.exchange.ms> <36860.21084.qm@web33505.mail.mud.yahoo.com> Message-ID: Sorry for replying late. I have tried mod_managed again on same machine (Lenovo 3000 N200), same OS (Ubuntu-9.04) and Mono version 2.0.1 but with latest FS revision 14249. It compiles correctly this time but gives following error upon "make install", ===================================================================== making install mod_managed make[5]: *** No rule to make target `/usr/local/freeswitch/mod/mod_managed.so', needed by `local_install'. Stop. make[4]: *** [install] Error 1 make[3]: *** [mod_managed-install] Error 1 make[2]: *** [install-recursive] Error 1 ===================================================================== Here is compilation log when executing "make", if it could of any help. ===================================================================== making all mod_managed Compiling freeswitch_managed.cpp... g++ -I/usr/src/svn-src/freeswitch/src/include -I/usr/src/svn-src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE -D_REENTRANT -pthread -I/usr/include/mono-1.0 -I/usr/include/glib-2.0 -I/usr/lib/glib-2.0/include -DHAVE_CONFIG_H -c -o freeswitch_managed.o freeswitch_managed.cpp Compiling freeswitch_wrap.cpp... g++ -I/usr/src/svn-src/freeswitch/src/include -I/usr/src/svn-src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE -D_REENTRANT -pthread -I/usr/include/mono-1.0 -I/usr/include/glib-2.0 -I/usr/lib/glib-2.0/include -DHAVE_CONFIG_H -c -o freeswitch_wrap.o freeswitch_wrap.cpp Demo.cs(58,14): warning CS0169: The private method `FreeSWITCH.Demo.AppDemo.hangupHook()' is never used Compilation succeeded - 1 warning(s) Compiling mod_managed.cpp... /usr/src/svn-src/freeswitch/libtool --mode=compile --tag=CXX g++ -I/usr/src/svn-src/freeswitch/src/include -I/usr/src/svn-src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE -D_REENTRANT -pthread -I/usr/include/mono-1.0 -I/usr/include/glib-2.0 -I/usr/lib/glib-2.0/include -DHAVE_CONFIG_H -c -o mod_managed.lo mod_managed.cpp libtool: compile: g++ -I/usr/src/svn-src/freeswitch/src/include -I/usr/src/svn-src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE -D_REENTRANT -pthread -I/usr/include/mono-1.0 -I/usr/include/glib-2.0 -I/usr/lib/glib-2.0/include -DHAVE_CONFIG_H -c mod_managed.cpp -fPIC -DPIC -o .libs/mod_managed.o mod_managed.cpp: In function ?void InitManagedSession(ManagedSession*, char* (*)(void*, switch_input_type_t), void (*)())?: mod_managed.cpp:97: warning: deprecated conversion from string constant to ?char*? libtool: compile: g++ -I/usr/src/svn-src/freeswitch/src/include -I/usr/src/svn-src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE -D_REENTRANT -pthread -I/usr/include/mono-1.0 -I/usr/include/glib-2.0 -I/usr/lib/glib-2.0/include -DHAVE_CONFIG_H -c mod_managed.cpp -o mod_managed.o >/dev/null 2>&1 Creating mod_managed.la... cat: .libs/mod_managed.log: No such file or directory ===================================================================== Thank you. On Fri, Jul 17, 2009 at 10:59 AM, Muhammad Shahzad < shaheryarkh at googlemail.com> wrote: > I tried to install mod_managed on ubuntu-9.04, mono framework version 2.0. > It gave me a lots of errors in Loader.cs, which seems to be SWIG related. > Since i am not a expert in SWIG so i disabled this module. This happend long > ago, i think FS svn revision 136xx. > > Let me try to compile it from latest FS revision and see if it works. I > will let you know the results. > > Thank you. > > > > On Fri, Jul 17, 2009 at 3:54 AM, Diego Toro wrote: > >> Hey, I am here :) >> >> I am working with mod_managed on Windows 2003 and Windows Vista with >> sucessfull. I noted on user list the issue with LoadFile on Loader.cs when >> a assembly had reference to others assemblies, I change LoadFile by LoadFrom >> and the load is made fine. >> >> I use c# application and sqlserver 2005, using FS and mod_managed. >> >> Diego >> >> --- On *Thu, 7/16/09, Michael Giagnocavo * wrote: >> >> >> From: Michael Giagnocavo >> Subject: [Freeswitch-users] mod_managed users? >> To: "freeswitch-users at lists.freeswitch.org" < >> freeswitch-users at lists.freeswitch.org> >> Date: Thursday, July 16, 2009, 4:43 PM >> >> Hey, if there are any mod_managed users on this list, I?d love it if you >> were able to let me know. I?d like to get feedback, positive or negative, on >> what worked, what didn?t, and how mod_managed can improve for you. Feel free >> to write on list or directly to me: mgg at giagnocavo.net >> >> >> >> Thanks! >> >> -Michael >> -----Inline Attachment Follows----- >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Muhammad Shahzad > ----------------------------------- > CISCO Rich Media Communication Specialist (CRMCS) > CISCO Certified Network Associate (CCNA) > Cell: +92 334 422 40 88 > MSN: shari_786pk at hotmail.com > Email: shaheryarkh at googlemail.com > -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090718/abd4e22b/attachment.html From jmesquita at gmail.com Sat Jul 18 06:03:01 2009 From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Sat, 18 Jul 2009 10:03:01 -0300 Subject: [Freeswitch-users] FsGUI In-Reply-To: <4A61BAA8.7080403@gmx.net> References: <5a8712120907172144u991d4c7o2b743116877ef0c0@mail.gmail.com> <4A61BAA8.7080403@gmx.net> Message-ID: <5a8712120907180603g5a5053e6p9910e0d02f7cf18d@mail.gmail.com> Added to the wiki: http://wiki.freeswitch.org/wiki/Fsgui#Ubuntu jmesquita On Sat, Jul 18, 2009 at 9:06 AM, Peter P GMX wrote: > Thanks, > > I have found the sources in > contrib/jmesquita/fsgui > Any recommendatioins how to compile it under Linux? > > Best regards > Peter > > Jo?o Mesquita schrieb: > > Dear folks, > > > > Even tho it might be premature, I would like to already spread the word. > > > > Check out FsGUI and feel free give feedback if this is a wanted tool > > and what direction it should take. Beware that the code is still > > contrib code and might now be yet mature for production use. > > > > http://wiki.freeswitch.org/wiki/Fsgui > > > > Thanks, > > > > Jo?o Mesquita > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20090718/a43f23da/attachment.html From d at unwire.it Sat Jul 18 10:04:31 2009 From: d at unwire.it (Darin Weeks) Date: Sat, 18 Jul 2009 10:04:31 -0700 Subject: [Freeswitch-users] FsGUI In-Reply-To: <5a8712120907180603g5a5053e6p9910e0d02f7cf18d@mail.gmail.com> References: <5a8712120907172144u991d4c7o2b743116877ef0c0@mail.gmail.com> <4A61BAA8.7080403@gmx.net> <5a8712120907180603g5a5053e6p9910e0d02f7cf18d@mail.gmail.com> Message-ID: <989132e70907181004u2c15449n2e445ae7489960fc@mail.gmail.com> Thanks guys! Can't wait to check it out. Would you mind adding some screenshots to the wiki? -- Darin 2009/7/18 Jo?o Mesquita > Added to the wiki: > > http://wiki.freeswitch.org/wiki/Fsgui#Ubuntu > > jmesquita > > > On Sat, Jul 18, 2009 at 9:06 AM, Peter P GMX wrote: > >> Thanks, >> >> I have found the sources in >> contrib/jmesquita/fsgui >> Any recommendatioins how to compile it under Linux? >> >> Best regards >> Peter >> >> Jo?o Mesquita schrieb: >> > Dear folks, >> > >> > Even tho it might be premature, I would like to already spread the word. >> > >> > Check out FsGUI and feel free give feedback if this is a wanted tool >> > and what direction it should take. Beware that the code is still >> > contrib code and might now be yet mature for production use. >> > >> > http://wiki.freeswitch.org/wiki/Fsgui >> > >> > Thanks, >> > >> > Jo?o Mesquita >> > ------------------------------------------------------------------------ >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- ) ) ) ) ) ) ) ) ) UNWIRE.IT ( ( ( ( ( ( ( ( ( broadband internet for west hollywood d at unwire.it http://unwire.it -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090718/9e506d72/attachment.html From d at unwire.it Sat Jul 18 10:06:34 2009 From: d at unwire.it (Darin Weeks) Date: Sat, 18 Jul 2009 10:06:34 -0700 Subject: [Freeswitch-users] FsGUI In-Reply-To: <989132e70907181004u2c15449n2e445ae7489960fc@mail.gmail.com> References: <5a8712120907172144u991d4c7o2b743116877ef0c0@mail.gmail.com> <4A61BAA8.7080403@gmx.net> <5a8712120907180603g5a5053e6p9910e0d02f7cf18d@mail.gmail.com> <989132e70907181004u2c15449n2e445ae7489960fc@mail.gmail.com> Message-ID: <989132e70907181006h35a24c00h8f540096cbfc3f84@mail.gmail.com> DOH! Sorry... I just noticed the link at the bottom of the page to the graphic. Thanks! On Sat, Jul 18, 2009 at 10:04 AM, Darin Weeks wrote: > Thanks guys! Can't wait to check it out. Would you mind adding some > screenshots to the wiki? > > -- Darin > > 2009/7/18 Jo?o Mesquita > > Added to the wiki: >> >> http://wiki.freeswitch.org/wiki/Fsgui#Ubuntu >> >> jmesquita >> >> >> On Sat, Jul 18, 2009 at 9:06 AM, Peter P GMX wrote: >> >>> Thanks, >>> >>> I have found the sources in >>> contrib/jmesquita/fsgui >>> Any recommendatioins how to compile it under Linux? >>> >>> Best regards >>> Peter >>> >>> Jo?o Mesquita schrieb: >>> > Dear folks, >>> > >>> > Even tho it might be premature, I would like to already spread the >>> word. >>> > >>> > Check out FsGUI and feel free give feedback if this is a wanted tool >>> > and what direction it should take. Beware that the code is still >>> > contrib code and might now be yet mature for production use. >>> > >>> > http://wiki.freeswitch.org/wiki/Fsgui >>> > >>> > Thanks, >>> > >>> > Jo?o Mesquita >>> > >>> ------------------------------------------------------------------------ >>> > >>> > _______________________________________________ >>> > FreeSWITCH-users mailing list >>> > FreeSWITCH-users at lists.freeswitch.org >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> > UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> > http://www.freeswitch.org >>> > >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > > ) ) ) ) ) ) ) ) ) UNWIRE.IT ( ( ( ( ( ( ( ( ( > broadband internet for west hollywood > d at unwire.it http://unwire.it > -- ) ) ) ) ) ) ) ) ) UNWIRE.IT ( ( ( ( ( ( ( ( ( broadband internet for west hollywood d at unwire.it http://unwire.it -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090718/55ae3ccc/attachment.html From mgg at giagnocavo.net Sat Jul 18 11:47:36 2009 From: mgg at giagnocavo.net (Michael Giagnocavo) Date: Sat, 18 Jul 2009 14:47:36 -0400 Subject: [Freeswitch-users] mod_managed users? In-Reply-To: References: <6E8D2069C08AA84A83D336E996AE4C67027C6C33F0@mse17be1.mse17.exchange.ms> <36860.21084.qm@web33505.mail.mud.yahoo.com> Message-ID: <6E8D2069C08AA84A83D336E996AE4C67027C7F85D6@mse17be1.mse17.exchange.ms> Thanks for this report. I?ll look into the linux build shortly and perhaps be able to help get it working. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Muhammad Shahzad Sent: Saturday, July 18, 2009 6:58 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_managed users? Sorry for replying late. I have tried mod_managed again on same machine (Lenovo 3000 N200), same OS (Ubuntu-9.04) and Mono version 2.0.1 but with latest FS revision 14249. It compiles correctly this time but gives following error upon "make install", ===================================================================== making install mod_managed make[5]: *** No rule to make target `/usr/local/freeswitch/mod/mod_managed.so', needed by `local_install'. Stop. make[4]: *** [install] Error 1 make[3]: *** [mod_managed-install] Error 1 make[2]: *** [install-recursive] Error 1 ===================================================================== Here is compilation log when executing "make", if it could of any help. ===================================================================== making all mod_managed Compiling freeswitch_managed.cpp... g++ -I/usr/src/svn-src/freeswitch/src/include -I/usr/src/svn-src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE -D_REENTRANT -pthread -I/usr/include/mono-1.0 -I/usr/include/glib-2.0 -I/usr/lib/glib-2.0/include -DHAVE_CONFIG_H -c -o freeswitch_managed.o freeswitch_managed.cpp Compiling freeswitch_wrap.cpp... g++ -I/usr/src/svn-src/freeswitch/src/include -I/usr/src/svn-src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE -D_REENTRANT -pthread -I/usr/include/mono-1.0 -I/usr/include/glib-2.0 -I/usr/lib/glib-2.0/include -DHAVE_CONFIG_H -c -o freeswitch_wrap.o freeswitch_wrap.cpp Demo.cs(58,14): warning CS0169: The private method `FreeSWITCH.Demo.AppDemo.hangupHook()' is never used Compilation succeeded - 1 warning(s) Compiling mod_managed.cpp... /usr/src/svn-src/freeswitch/libtool --mode=compile --tag=CXX g++ -I/usr/src/svn-src/freeswitch/src/include -I/usr/src/svn-src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE -D_REENTRANT -pthread -I/usr/include/mono-1.0 -I/usr/include/glib-2.0 -I/usr/lib/glib-2.0/include -DHAVE_CONFIG_H -c -o mod_managed.lo mod_managed.cpp libtool: compile: g++ -I/usr/src/svn-src/freeswitch/src/include -I/usr/src/svn-src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE -D_REENTRANT -pthread -I/usr/include/mono-1.0 -I/usr/include/glib-2.0 -I/usr/lib/glib-2.0/include -DHAVE_CONFIG_H -c mod_managed.cpp -fPIC -DPIC -o .libs/mod_managed.o mod_managed.cpp: In function ?void InitManagedSession(ManagedSession*, char* (*)(void*, switch_input_type_t), void (*)())?: mod_managed.cpp:97: warning: deprecated conversion from string constant to ?char*? libtool: compile: g++ -I/usr/src/svn-src/freeswitch/src/include -I/usr/src/svn-src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE -D_REENTRANT -pthread -I/usr/include/mono-1.0 -I/usr/include/glib-2.0 -I/usr/lib/glib-2.0/include -DHAVE_CONFIG_H -c mod_managed.cpp -o mod_managed.o >/dev/null 2>&1 Creating mod_managed.la... cat: .libs/mod_managed.log: No such file or directory ===================================================================== Thank you. On Fri, Jul 17, 2009 at 10:59 AM, Muhammad Shahzad > wrote: I tried to install mod_managed on ubuntu-9.04, mono framework version 2.0. It gave me a lots of errors in Loader.cs, which seems to be SWIG related. Since i am not a expert in SWIG so i disabled this module. This happend long ago, i think FS svn revision 136xx. Let me try to compile it from latest FS revision and see if it works. I will let you know the results. Thank you. On Fri, Jul 17, 2009 at 3:54 AM, Diego Toro > wrote: Hey, I am here :) I am working with mod_managed on Windows 2003 and Windows Vista with sucessfull. I noted on user list the issue with LoadFile on Loader.cs when a assembly had reference to others assemblies, I change LoadFile by LoadFrom and the load is made fine. I use c# application and sqlserver 2005, using FS and mod_managed. Diego --- On Thu, 7/16/09, Michael Giagnocavo > wrote: From: Michael Giagnocavo > Subject: [Freeswitch-users] mod_managed users? To: "freeswitch-users at lists.freeswitch.org" > Date: Thursday, July 16, 2009, 4:43 PM Hey, if there are any mod_managed users on this list, I?d love it if you were able to let me know. I?d like to get feedback, positive or negative, on what worked, what didn?t, and how mod_managed can improve for you. Feel free to write on list or directly to me: mgg at giagnocavo.net Thanks! -Michael -----Inline Attachment Follows----- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090718/c98b8292/attachment.html From raul at etellicom.com Sat Jul 18 15:29:44 2009 From: raul at etellicom.com (Raul Fragoso) Date: Sat, 18 Jul 2009 19:29:44 -0300 Subject: [Freeswitch-users] 302 redirects and continue_on_fail=true In-Reply-To: <0A05EBC4-8DC0-4D4E-9029-E1BA2D5E443F@gmail.com> References: <0A05EBC4-8DC0-4D4E-9029-E1BA2D5E443F@gmail.com> Message-ID: <1247956185.30330.4.camel@raul-laptop> You can set continue_on_fail to the major (or all) causes that you want to handle and don't include 302 in those so it will continue to execute the extension, for example: continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION,CALL_REJECTED,USER_NOT_REGISTERED You can know more about the hangup causes here: http://wiki.freeswitch.org/wiki/Hangup_causes Regards, Raul On Sat, 2009-07-18 at 13:49 +1000, Jay Binks wrote: > I have an upstream provider that utilizes a load balancer that spits > back 302 redirects with contact headers > > > SIP/2.0 302 Moved Temporarily > Via: SIP/2.0/UDP > xxx.xxx.xxx.xxx:5080;rport=5080;branch=z9hG4bKB6peFDvXZ5S2F;received=xxx.xxx.xxx.xxx > From: "test" > ;tag=ByFF2244HHvmj > To: > ;tag=1288540-274799759-385876096-3219999652 > Call-ID: f6dc8d30-edd7-122c-c98e-000e7f301839 > CSeq: 117820807 INVITE > Contact: > Server: MERA MVTS3G v.3.10.2-49-Release > Content-Length: 0 > > > in my dialplan I have multiple upstream suppliers in a failover setup > so I setup some vars and sip headers then attempt the bridge. > if it fails I then go on to do the same thing for a few other > suppliers ( setup headers, attempt bridge ) so because of this I > use continue_on_fail=true > > > it appears Freeswitch sees the 302 as a temp failure and does not > follow the redirect, and instead moves on to the next upstream and > bridges there. > > > ive read that I can selectively exclude temporary failures from > continue_on_fail but im not sure thats exact enough for this > situation. > I do wish for continue_on_fail to ignore 302 moved temporarily but not > ALL temporary failures ( for which there are probably many more > causes ) > > > any help would be greatly appreciated as Im not sure the best way to > resolve this. > > > Jay > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.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 Sat Jul 18 16:55:09 2009 From: jaybinks at gmail.com (Jay Binks) Date: Sun, 19 Jul 2009 09:55:09 +1000 Subject: [Freeswitch-users] 302 redirects and continue_on_fail=true In-Reply-To: <1247956185.30330.4.camel@raul-laptop> References: <0A05EBC4-8DC0-4D4E-9029-E1BA2D5E443F@gmail.com> <1247956185.30330.4.camel@raul-laptop> Message-ID: Sure however my concern is that NORMAL_TEMPORARY_FAILURE is a generic failure and not only from 302 redirects. I'm not sure I'm game to do this, unless Im being completly paranoid which is possible. Can I get advice on this from a few Other users ? Jay On 19/07/2009, at 8:29 AM, Raul Fragoso wrote: > You can set continue_on_fail to the major (or all) causes that you > want > to handle and don't include 302 in those so it will continue to > execute > the extension, for example: > continue_on_fail= > NORMAL_TEMPORARY_FAILURE, > USER_BUSY, > NO_ANSWER, > TIMEOUT,NO_ROUTE_DESTINATION,CALL_REJECTED,USER_NOT_REGISTERED > > You can know more about the hangup causes here: > http://wiki.freeswitch.org/wiki/Hangup_causes > > Regards, > > Raul > > On Sat, 2009-07-18 at 13:49 +1000, Jay Binks wrote: >> I have an upstream provider that utilizes a load balancer that spits >> back 302 redirects with contact headers >> >> >> SIP/2.0 302 Moved Temporarily >> Via: SIP/2.0/UDP >> xxx.xxx.xxx.xxx: >> 5080;rport=5080;branch=z9hG4bKB6peFDvXZ5S2F;received=xxx.xxx.xxx.xxx >> From: "test" >> ;tag=ByFF2244HHvmj >> To: >> ;tag=1288540-274799759-385876096-3219999652 >> Call-ID: f6dc8d30-edd7-122c-c98e-000e7f301839 >> CSeq: 117820807 INVITE >> Contact: >> Server: MERA MVTS3G v.3.10.2-49-Release >> Content-Length: 0 >> >> >> in my dialplan I have multiple upstream suppliers in a failover setup >> so I setup some vars and sip headers then attempt the bridge. >> if it fails I then go on to do the same thing for a few other >> suppliers ( setup headers, attempt bridge ) so because of this I >> use continue_on_fail=true >> >> >> it appears Freeswitch sees the 302 as a temp failure and does not >> follow the redirect, and instead moves on to the next upstream and >> bridges there. >> >> >> ive read that I can selectively exclude temporary failures from >> continue_on_fail but im not sure thats exact enough for this >> situation. >> I do wish for continue_on_fail to ignore 302 moved temporarily but >> not >> ALL temporary failures ( for which there are probably many more >> causes ) >> >> >> any help would be greatly appreciated as Im not sure the best way to >> resolve this. >> >> >> Jay >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.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 Sun Jul 19 00:26:24 2009 From: woodydickson at gmail.com (Woody Dickson) Date: Sun, 19 Jul 2009 15:26:24 +0800 Subject: [Freeswitch-users] Can FreeSWITCH send and receive SIP MESSAGE Message-ID: Hi, I would like to use freeswitch as a gateway for sending and receiving short message. Does Freeswitch have the capability to send and recevie SIP MESSAGE? How can I set it up? I can't find any document on how to use Freeswitch for text message. Thanks, Woody -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090719/a938b903/attachment.html From mcampbellsmith at gmail.com Sun Jul 19 05:11:28 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Sun, 19 Jul 2009 22:11:28 +1000 Subject: [Freeswitch-users] NAT'd FS / publice softphone problems Message-ID: <33c87fa30907190511s1c4a9a9fi71f03d6491efeb95@mail.gmail.com> Hi All, I know this question has come up before but I couldn't find the answer that I could understand! Sorry in advance. My setup is: Freeswtch NAT'd (192.168.x.x) -> Router -> Internet -> Softphone with public IP I can easily get the softphones to register, but when I try to call from the softphone to voicemail (for example), I don't get any audio. I checked out this page: http://wiki.freeswitch.org/wiki/External_profile (section Switch with External SoftPhone) but I am not clear how I can get this to work. I have played around with the rtp-ip and external-rtp-ip but without success. Is it possible for someone to help me configure this so softphones that are outside the nat'd lan get audio correctly? Help appreciated! Thanks From mike at jerris.com Sun Jul 19 08:35:43 2009 From: mike at jerris.com (Michael Jerris) Date: Sun, 19 Jul 2009 11:35:43 -0400 Subject: [Freeswitch-users] Can FreeSWITCH send and receive SIP MESSAGE In-Reply-To: References: Message-ID: See the chat_send api command and it also should just work with presence peers. On Jul 19, 2009, at 3:26 AM, Woody Dickson wrote: > Hi, > > I would like to use freeswitch as a gateway for sending and > receiving short message. > > Does Freeswitch have the capability to send and recevie SIP MESSAGE? > > How can I set it up? I can't find any document on how to use > Freeswitch for text message. > > 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 From lzwierko at gmail.com Sun Jul 19 04:27:13 2009 From: lzwierko at gmail.com (=?UTF-8?B?xYF1a2FzeiBad2llcmtv?=) Date: Sun, 19 Jul 2009 13:27:13 +0200 Subject: [Freeswitch-users] Dial up from confernece issue In-Reply-To: References: <0A05EBC4-8DC0-4D4E-9029-E1BA2D5E443F@gmail.com> <1247956185.30330.4.camel@raul-laptop> Message-ID: <4A630311.9010003@gmail.com> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I want to use bgdial command to add a person to a already started conference (that is, call that person and when answered - add the channel to conference). The scenario is I have two sip clients registered in default context - 1000 and 1001. 1000 dials conference number (3001 in this case) and new conference is started. I want to dial out to second using bgdial, unfortunately mod_sofia drops the call with 'Cannot Blind Transfer 1 Legged calls' message. Should I use the bgdial command differently? Or perhaps I should do this totally differently? Logs attached below. Thanks for any help, Lukasz freeswitch at Zwierko-laptop> conference list API CALL [conference(list)] output: Conference 3001-192.168.0.1 (1 member) 3;sofia/internal/1000 at 192.168.0.1;25f125a5-3556-a449-85ec-e4001336f313;1000;1000;hear|speak|floor;0;0;300 freeswitch at Zwierko-laptop> conference 3001-192.168.0.1 bgdial sofia/default/1001 at 192.168.0.1 API CALL [conference(3001-192.168.0.1 bgdial sofia/default/1001 at 192.168.0.1)] output: OK freeswitch at Zwierko-laptop> 2009-07-19 13:19:44.451100 [NOTICE] switch_channel.c:602 New Channel sofia/internal/1001 at 192.168.0.1 [b9fada7f-9c1d-4949-af8a-a8220ce f9c5b] 2009-07-19 13:19:44.459100 [NOTICE] switch_channel.c:602 New Channel sofia/internal/48228882211 at 192.168.0.1 [4f6b26dd-a0cb-2846-ad17-5f517e60e2e7] 2009-07-19 13:19:44.488100 [INFO] mod_dialplan_xml.c:252 Processing TelkaSwitch->1001 in context public 2009-07-19 13:19:44.495100 [ERR] sofia.c:4174 Cannot Blind Transfer 1 Legged calls 2009-07-19 13:19:44.498100 [NOTICE] sofia.c:3775 Hangup sofia/internal/1001 at 192.168.0.1 [CS_CONSUME_MEDIA] [NO_USER_RESPONSE] 2009-07-19 13:19:44.501100 [ERR] mod_conference.c:4351 Cannot create outgoing channel, cause: NO_USER_RESPONSE 2009-07-19 13:19:44.512100 [NOTICE] switch_core_session.c:1085 Session 9 (sofia/internal/1001 at 192.168.0.1) Ended 2009-07-19 13:19:44.512100 [NOTICE] switch_core_session.c:1087 Close Channel sofia/internal/1001 at 192.168.0.1 [CS_DESTROY] 2009-07-19 13:19:44.628100 [NOTICE] switch_core_state_machine.c:179 Hangup sofia/internal/48228882211 at 192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING] 2009-07-19 13:19:44.708100 [NOTICE] switch_core_session.c:1085 Session 10 (sofia/internal/48228882211 at 192.168.0.1) Ended 2009-07-19 13:19:44.708100 [NOTICE] switch_core_session.c:1087 Close Channel sofia/internal/48228882211 at 192.168.0.1 [CS_DESTROY] -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.9 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iQEcBAEBAgAGBQJKYwMRAAoJED7LBosr0F2ullAH/j6AebaezM2/RQ4PVKeNbMEm yWYqC1bDmp5F56owRH6Vq7BRnXKB4roqV2NLFqLNRYwzq/S4bzc9p417/NckrACg DhmZ6tFd4ujLb6B1HvJMTKsDvnYCpn5EVCbENfKVIY4INDAcEYbncwUA21XxILI+ ztz+6qNPwOMOjY9aZaf1qpTcTcG2yn62mpvesmVeYS1vNpZFVpnQq4PrukDg+1xs N8EpJetP0FxhYzT/IiD9fS2wAzQSgJPgo0m7R4ezk/1NIF9f+o0irgc8zx+VgKw1 UhJ1FLhs8ObzhYclvwJxwTlG+ppI28uIVO8EItiPB4/ZhEjyPfNVHcqTvMG6wb4= =3Bm/ -----END PGP SIGNATURE----- From carlos.talbot at gmail.com Sun Jul 19 12:46:47 2009 From: carlos.talbot at gmail.com (Carlos Talbot) Date: Sun, 19 Jul 2009 14:46:47 -0500 Subject: [Freeswitch-users] FsGUI In-Reply-To: <5a8712120907180603g5a5053e6p9910e0d02f7cf18d@mail.gmail.com> References: <5a8712120907172144u991d4c7o2b743116877ef0c0@mail.gmail.com> <4A61BAA8.7080403@gmx.net> <5a8712120907180603g5a5053e6p9910e0d02f7cf18d@mail.gmail.com> Message-ID: <5800526b0907191246i5377b6acx37c5f9b2bffe9a4f@mail.gmail.com> FYI, for those interested I've built an FsGui MSI file compiled for Windows via VS 2008 & the QT SDK library. It includes 2 necessary QT dlls. Future builds of the MSI for Freeswitch will include this. Here's the temporary link http://pbxinaflash.com/tusc/fsgui.msi (until I can sync it up to files.freeswitch.org) Carlos 2009/7/18 Jo?o Mesquita > Added to the wiki: > > http://wiki.freeswitch.org/wiki/Fsgui#Ubuntu > > jmesquita > > > On Sat, Jul 18, 2009 at 9:06 AM, Peter P GMX wrote: > >> Thanks, >> >> I have found the sources in >> contrib/jmesquita/fsgui >> Any recommendatioins how to compile it under Linux? >> >> Best regards >> Peter >> >> Jo?o Mesquita schrieb: >> > Dear folks, >> > >> > Even tho it might be premature, I would like to already spread the word. >> > >> > Check out FsGUI and feel free give feedback if this is a wanted tool >> > and what direction it should take. Beware that the code is still >> > contrib code and might now be yet mature for production use. >> > >> > http://wiki.freeswitch.org/wiki/Fsgui >> > >> > Thanks, >> > >> > Jo?o Mesquita >> > ------------------------------------------------------------------------ >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20090719/b5985077/attachment.html From lzwierko at gmail.com Sun Jul 19 13:19:17 2009 From: lzwierko at gmail.com (=?UTF-8?B?xYF1a2FzeiBad2llcmtv?=) Date: Sun, 19 Jul 2009 22:19:17 +0200 Subject: [Freeswitch-users] Dial up from confernece issue Message-ID: <4A637FC5.6000301@gmail.com> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, sorry if you're getting this again, I'm not sure if this mail got deliverd to the mail-list (I didn't get a copy...) Anyway, I want to use bgdial command to add a person to a already started conference (that is, call that person and when answered - add the channel to conference). The scenario is I have two sip clients registered in default context - 1000 and 1001. 1000 dials conference number (3001 in this case) and new conference is started. I want to dial out to second using bgdial, unfortunately mod_sofia drops the call with 'Cannot Blind Transfer 1 Legged calls' message. Should I use the bgdial command differently? Or perhaps I should do this totally differently? Logs attached below. Thanks for any help, Lukasz freeswitch at Zwierko-laptop> conference list API CALL [conference(list)] output: Conference 3001-192.168.0.1 (1 member) 3;sofia/internal/1000 at 192.168.0.1;25f125a5-3556-a449-85ec-e4001336f313;1000;1000;hear|speak|floor;0;0;300 freeswitch at Zwierko-laptop> conference 3001-192.168.0.1 bgdial sofia/default/1001 at 192.168.0.1 API CALL [conference(3001-192.168.0.1 bgdial sofia/default/1001 at 192.168.0.1)] output: OK freeswitch at Zwierko-laptop> 2009-07-19 13:19:44.451100 [NOTICE] switch_channel.c:602 New Channel sofia/internal/1001 at 192.168.0.1 [b9fada7f-9c1d-4949-af8a-a8220ce f9c5b] 2009-07-19 13:19:44.459100 [NOTICE] switch_channel.c:602 New Channel sofia/internal/48228882211 at 192.168.0.1 [4f6b26dd-a0cb-2846-ad17-5f517e60e2e7] 2009-07-19 13:19:44.488100 [INFO] mod_dialplan_xml.c:252 Processing TelkaSwitch->1001 in context public 2009-07-19 13:19:44.495100 [ERR] sofia.c:4174 Cannot Blind Transfer 1 Legged calls 2009-07-19 13:19:44.498100 [NOTICE] sofia.c:3775 Hangup sofia/internal/1001 at 192.168.0.1 [CS_CONSUME_MEDIA] [NO_USER_RESPONSE] 2009-07-19 13:19:44.501100 [ERR] mod_conference.c:4351 Cannot create outgoing channel, cause: NO_USER_RESPONSE 2009-07-19 13:19:44.512100 [NOTICE] switch_core_session.c:1085 Session 9 (sofia/internal/1001 at 192.168.0.1) Ended 2009-07-19 13:19:44.512100 [NOTICE] switch_core_session.c:1087 Close Channel sofia/internal/1001 at 192.168.0.1 [CS_DESTROY] 2009-07-19 13:19:44.628100 [NOTICE] switch_core_state_machine.c:179 Hangup sofia/internal/48228882211 at 192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING] 2009-07-19 13:19:44.708100 [NOTICE] switch_core_session.c:1085 Session 10 (sofia/internal/48228882211 at 192.168.0.1) Ended 2009-07-19 13:19:44.708100 [NOTICE] switch_core_session.c:1087 Close Channel sofia/internal/48228882211 at 192.168.0.1 [CS_DESTROY] -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.9 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iQEcBAEBAgAGBQJKY3/FAAoJED7LBosr0F2uWo8H/iJEHblsAENCjRh/dsZvj9br Mq6txy7iafLE970XxvToaa0+FGBFxN+S6yQ6ampNPd8t+jl6WwC79Btwr+NLgXEc NcWpVQp65QxKxA+MgQOyqWIskcMcxdf4Uht3wuLPZtre0BpjcAFhykweYjOy1jFp AYAM61ShogHlpXtl9Z6upDvWPoOzdY4m13EM7f0NmpbC32Sg+OOULEtsxvSkZ8ah DBKDyDdXFo9iIcReDqjsu/kzAgrBsAZvOiEbSPoQTjZgzX+UrbgqIc+rhmP60vyt 8u8ufDgzh7MC/VQObHKHLe8e/Zbpaf+3JiGxZBtFyoUFyP3DjHoR5TYu5IsENPU= =QlRU -----END PGP SIGNATURE----- From brian at freeswitch.org Sun Jul 19 13:23:03 2009 From: brian at freeswitch.org (Brian West) Date: Sun, 19 Jul 2009 15:23:03 -0500 Subject: [Freeswitch-users] FsGUI In-Reply-To: <5800526b0907191246i5377b6acx37c5f9b2bffe9a4f@mail.gmail.com> References: <5a8712120907172144u991d4c7o2b743116877ef0c0@mail.gmail.com> <4A61BAA8.7080403@gmx.net> <5a8712120907180603g5a5053e6p9910e0d02f7cf18d@mail.gmail.com> <5800526b0907191246i5377b6acx37c5f9b2bffe9a4f@mail.gmail.com> Message-ID: <57FE57A0-24AE-462D-87B7-114D073C4F6D@freeswitch.org> Its sycned to files. now. /b On Jul 19, 2009, at 2:46 PM, Carlos Talbot wrote: > FYI, > > for those interested I've built an FsGui MSI file compiled for > Windows via VS 2008 & the QT SDK library. It includes 2 necessary QT > dlls. > > Future builds of the MSI for Freeswitch will include this. > > Here's the temporary link http://pbxinaflash.com/tusc/fsgui.msi > (until I can sync it up to files.freeswitch.org) > > Carlos > > 2009/7/18 Jo?o Mesquita > Added to the wiki: > > http://wiki.freeswitch.org/wiki/Fsgui#Ubuntu > > jmesquita > > > On Sat, Jul 18, 2009 at 9:06 AM, Peter P GMX > wrote: > Thanks, > > I have found the sources in > contrib/jmesquita/fsgui > Any recommendatioins how to compile it under Linux? > > Best regards > Peter > > Jo?o Mesquita schrieb: > > Dear folks, > > > > Even tho it might be premature, I would like to already spread the > word. > > > > Check out FsGUI and feel free give feedback if this is a wanted tool > > and what direction it should take. Beware that the code is still > > contrib code and might now be yet mature for production use. > > > > http://wiki.freeswitch.org/wiki/Fsgui > > > > Thanks, > > > > Jo?o Mesquita > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20090719/ba585ff9/attachment.html From brian at freeswitch.org Sun Jul 19 13:24:09 2009 From: brian at freeswitch.org (Brian West) Date: Sun, 19 Jul 2009 15:24:09 -0500 Subject: [Freeswitch-users] Dial up from confernece issue In-Reply-To: <4A637FC5.6000301@gmail.com> References: <4A637FC5.6000301@gmail.com> Message-ID: <41A88E3D-C420-4A1F-8B80-098EA18F9544@freeswitch.org> If a single leg call gets a 302 you can't really "transfer" it anywhere... What SVN rev are you on? /b On Jul 19, 2009, at 3:19 PM, ?ukasz Zwierko wrote: > Hi, > > sorry if you're getting this again, I'm not sure if this mail got > deliverd to the mail-list (I didn't get a copy...) > > Anyway, > > I want to use bgdial command to add a person to a already started > conference (that is, call that person and when answered - add the > channel to conference). > > The scenario is I have two sip clients registered in default context - > 1000 and 1001. 1000 dials conference number (3001 in this case) and > new > conference is started. I want to dial out to second using bgdial, > unfortunately mod_sofia drops the call with 'Cannot Blind Transfer 1 > Legged calls' message. > > Should I use the bgdial command differently? Or perhaps I should do > this > totally differently? Logs attached below. > > Thanks for any help, > > Lukasz From jmesquita at gmail.com Sun Jul 19 14:33:45 2009 From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Sun, 19 Jul 2009 18:33:45 -0300 Subject: [Freeswitch-users] FsGUI In-Reply-To: <57FE57A0-24AE-462D-87B7-114D073C4F6D@freeswitch.org> References: <5a8712120907172144u991d4c7o2b743116877ef0c0@mail.gmail.com> <4A61BAA8.7080403@gmx.net> <5a8712120907180603g5a5053e6p9910e0d02f7cf18d@mail.gmail.com> <5800526b0907191246i5377b6acx37c5f9b2bffe9a4f@mail.gmail.com> <57FE57A0-24AE-462D-87B7-114D073C4F6D@freeswitch.org> Message-ID: <5a8712120907191433t2239f8fkf97fce67639415ea@mail.gmail.com> Thank you very much for your support. Brian, how can I put MacOSX dmg and linux binaries on files.freeswitch.org? jmesquita On Sun, Jul 19, 2009 at 5:23 PM, Brian West wrote: > Its sycned to files. now. > /b > > On Jul 19, 2009, at 2:46 PM, Carlos Talbot wrote: > > FYI, > for those interested I've built an FsGui MSI file compiled for Windows via > VS 2008 & the QT SDK library. It includes 2 necessary QT dlls. > > Future builds of the MSI for Freeswitch will include this. > > Here's the temporary link http://pbxinaflash.com/tusc/fsgui.msi (until I > can sync it up to files.freeswitch.org) > > Carlos > > 2009/7/18 Jo?o Mesquita > >> Added to the wiki: >> >> http://wiki.freeswitch.org/wiki/Fsgui#Ubuntu >> >> jmesquita >> >> >> On Sat, Jul 18, 2009 at 9:06 AM, Peter P GMX wrote: >> >>> Thanks, >>> >>> I have found the sources in >>> contrib/jmesquita/fsgui >>> Any recommendatioins how to compile it under Linux? >>> >>> Best regards >>> Peter >>> >>> Jo?o Mesquita schrieb: >>> > Dear folks, >>> > >>> > Even tho it might be premature, I would like to already spread the >>> word. >>> > >>> > Check out FsGUI and feel free give feedback if this is a wanted tool >>> > and what direction it should take. Beware that the code is still >>> > contrib code and might now be yet mature for production use. >>> > >>> > http://wiki.freeswitch.org/wiki/Fsgui >>> > >>> > Thanks, >>> > >>> > Jo?o Mesquita >>> > >>> ------------------------------------------------------------------------ >>> > >>> > _______________________________________________ >>> > FreeSWITCH-users mailing list >>> > FreeSWITCH-users at lists.freeswitch.org >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> > UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> > http://www.freeswitch.org >>> > >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20090719/1bb160f8/attachment.html From msc at freeswitch.org Sun Jul 19 16:00:03 2009 From: msc at freeswitch.org (Michael S Collins) Date: Sun, 19 Jul 2009 16:00:03 -0700 Subject: [Freeswitch-users] NAT'd FS / publice softphone problems In-Reply-To: <33c87fa30907190511s1c4a9a9fi71f03d6491efeb95@mail.gmail.com> References: <33c87fa30907190511s1c4a9a9fi71f03d6491efeb95@mail.gmail.com> Message-ID: <4E741DC1-E045-4429-9CD4-E76941609FA3@freeswitch.org> What svn rev are you on? There have been some important changes recently. -MC Sent from my iPhone On Jul 19, 2009, at 5:11 AM, Mark Campbell-Smith wrote: > Hi All, > > I know this question has come up before but I couldn't find the answer > that I could understand! Sorry in advance. > > My setup is: > Freeswtch NAT'd (192.168.x.x) -> Router -> Internet -> Softphone > with public IP > > I can easily get the softphones to register, but when I try to call > from the softphone to voicemail (for example), I don't get any audio. > > I checked out this page: > http://wiki.freeswitch.org/wiki/External_profile (section Switch with > External SoftPhone) but I am not clear how I can get this to work. I > have played around with the rtp-ip and external-rtp-ip but without > success. > > Is it possible for someone to help me configure this so softphones > that are outside the nat'd lan get audio correctly? > > Help appreciated! > > 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 From yudha2008 at gmail.com Sun Jul 19 23:22:23 2009 From: yudha2008 at gmail.com (Baskar) Date: Mon, 20 Jul 2009 11:52:23 +0530 Subject: [Freeswitch-users] Freeswitch Windows Issues Message-ID: *Hi, I have installed freeswitch in windows, but when i star the freeswitch i get this error. Due to this i cant able to register my extension in softphone. how can i resolve this problem. 2009-07-20 10:55:05.390625 [CONSOLE] switch_core.c:1465 FreeSWITCH Version 1.0.trunk (13754M) Started. Crash Protection [Disabled] Max Sessions[1000] Session Rate[30] SQL [Enabled] freeswitch at Baskar> 2009-07-20 10:55:05.812500 [ERR] sofia.c:801 Error Creating SIP UA for profile: internal-ipv6 One more question in windows whether it is possible to connect the ODBC connection through JavaScript in freeswitch. I have configured inbound in Linux it is working fine but same script i tried in windows but i get this error. I have installed and configured MYSQL connector ODBC in window. But when is call the script i get this error. 2009-07-20 10:55:42.781250 [ERR] mod_spidermonkey.c:3393 Error loading ODBC 2009-07-20 10:55:42.781250 [ERR] Inbound1.js:5 ReferenceError: ODBC is not defined Can some one assist me to resolve this above error Thanks in advance. -- Thanks with Regards, N.Baskar * -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090720/43211174/attachment.html From velu.technical at gmail.com Sun Jul 19 23:30:50 2009 From: velu.technical at gmail.com (velusamy velu) Date: Mon, 20 Jul 2009 12:00:50 +0530 Subject: [Freeswitch-users] Creating a new User Agent Message-ID: <1452e2980907192330o14904fb8ke0aec88fd6afde0a@mail.gmail.com> Dear All, I want to create a new User Agent like sip configurations in Asterisk. I checked default user agents 1000 to 1001. But I have bit confused the relationship between default user agents and sip_profiles. I need some help from you all for the following questions, How to create new user agent ? How to relate the new user agent with sip internal profile ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090720/4c66dd72/attachment.html From helmut.kuper at ewetel.de Mon Jul 20 02:44:29 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Mon, 20 Jul 2009 11:44:29 +0200 Subject: [Freeswitch-users] Q931 TE State Timer In-Reply-To: <4A35F0F0.50406@ewetel.de> References: <4A3266E5.2000702@ewetel.de> <4A328E8E.6030607@freeswitch.org> <4A35F0F0.50406@ewetel.de> Message-ID: <4A643C7D.7010209@ewetel.de> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello community, to keep you informed about this. I got an early copy of the new openzap ISDN stack. I did some changes to get it running with my uplink provider. Most were dedicated to my uplink provider (AVAYA PBX, e.g. needed IEs in CALL_PROC, missed CON ACK, debug messages, ...) and a few were more general (like auto recover after a long term E1 link disconnect, not handled PROGRESS messages in cetrain call states, addind Timerhandlinf or call tear down) Since last friday I use it my prod environment (E1, Q931-TE, enBlock-Dialing). I have currently no problems. Timers work as expected and seems to keep the stack clean, although not all timers are implemented, yet ... So far Stefan and Tony did a very good job so far. Here is a snippet from my log showing the handling of my old originate problem of a missed RELEASE message from uplink after sending a DISCONNECT by the NEW stack: 2009-07-20 11:05:40.208698 [DEBUG] ozmod_isdn.c:1999 WRITE 9 - -------------------------------------------------------------------------------- [08 02 80 1e 45 08 02 81 90] 2009-07-20 11:05:40.208698 [NOTICE] Span:1 Starting timer 6 (timeout: 30000 ms) for call 30 [0x1e] 2009-07-20 11:05:40.208698 [DEBUG] Span:1 Call going from state 8 -> 11 2009-07-20 11:05:40.208698 [DEBUG] Span:1 Q931Rx43 return code: 0 [Waiting for RELEASE] 2009-07-20 11:06:10.279950 [NOTICE] Span:1 Stopping timer 6 for call 30 [0x1e] 2009-07-20 11:06:10.279950 [NOTICE] Span:1 Timer T305 timed out for call 30 2009-07-20 11:06:10.279950 [DEBUG] Span:1 Receiving message from Layer4 (size: 109, type: 77 [0x4d], CRV: 30, CRVFlag: 0) 2009-07-20 11:06:10.279950 [NOTICE] Span:1 Handling message from Layer4 in call state: 11 2009-07-20 11:06:10.279950 [DEBUG] Span:1 Processing RELEASE message from Local for CRV: 30 (0x1e) 2009-07-20 11:06:10.279950 [DEBUG] Span:1 Sending message to Q.921 (size: 109) 2009-07-20 11:06:10.279950 [DEBUG] Span:1 Creating Q.931 Message Header: ProtDisc 8 (0x8), CRV 30 (0x1e), CRVflag: 1 (0x1), MesType: 77 (0x4d) 2009-07-20 11:06:10.279950 [DEBUG] Span:1 XXX Adding IE 0x8 2009-07-20 11:06:10.279950 [DEBUG] Span:1 XXX Adding IE 0x28 2009-07-20 11:06:10.279950 [DEBUG] Span:1 XXX Adding IE 0x34 2009-07-20 11:06:10.279950 [DEBUG] Span:1 XXX Adding IE 0x1c 2009-07-20 11:06:10.279950 [NOTICE] Span:1 Unable to get reference for IE 28 (0x1c) 2009-07-20 11:06:10.279950 [DEBUG] Span:1 XXX Adding IE 0xd 2009-07-20 11:06:10.279950 [NOTICE] Span:1 Unable to get reference for IE 13 (0xd) 2009-07-20 11:06:10.279950 [DEBUG] ozmod_isdn.c:1999 WRITE 9 - -------------------------------------------------------------------------------- [08 02 80 1e 4d 08 02 81 e6] 2009-07-20 11:06:10.279950 [NOTICE] Span:1 Starting timer 9 (timeout: 4000 ms) for call 30 [0x1e] 2009-07-20 11:06:10.279950 [DEBUG] Span:1 Call going from state 11 -> 19 2009-07-20 11:06:10.279950 [DEBUG] Span:1 Q931Rx43 return code: 0 2009-07-20 11:06:10.419951 [DEBUG] ozmod_isdn.c:1181 READ 9 - -------------------------------------------------------------------------------- [08 02 00 1e 5a 08 02 81 d1] 2009-07-20 11:06:10.419951 [DEBUG] Span:1 Call is in state 19 [Release request] 2009-07-20 11:06:10.419951 [DEBUG] Span:1 Received message from Q.921 (ind 4, tei 0, size 13) MesType: 90, CRVFlag 0 (Originator), CRV 30 (Dialect: Q.931 TE) 2009-07-20 11:06:10.419951 [DEBUG] Span:1 Processing RELEASE COMPLETE message from Remote for CRV: 30 (0x1e) 2009-07-20 11:06:10.419951 [NOTICE] Span:1 Stopping timer 9 for call 30 [0x1e] 2009-07-20 11:06:10.419951 [DEBUG] Span:1 Sending message to Layer4 (size: 110) 2009-07-20 11:06:10.419951 [DEBUG] ozmod_isdn.c:696 Yay I got an event! Type:[5a] Size:[110] CRV: 30 (0x1e, CTX: Originator) 2009-07-20 11:06:10.419951 [DEBUG] ozmod_isdn.c:728 zchan 8124e08 (1:25) source isdn_data->channels_remote_crv[0x1e] 2009-07-20 11:06:10.419951 [DEBUG] ozmod_isdn.c:799 Received Release Complete message for channel 0 2009-07-20 11:06:10.419951 [DEBUG] ozmod_isdn.c:804 Changing state on 1:25 from HANGUP to DOWN :D A known bug is the handling of incomming calls aiming for non registered, but known targets. Currently caller isn't informed about that and so its call stays open. Will try to fix that. regards Helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) iD8DBQFKZDx94tZeNddg3dwRApjxAJ4tynu+xrzM3uMWlAhgnp2fdjmi0wCfZvgU 1wxVCgyskanESU0UxHQ4Ars= =4xqu -----END PGP SIGNATURE----- From tayeb.meftah at gmail.com Mon Jul 20 03:15:20 2009 From: tayeb.meftah at gmail.com (Meftah Tayeb) Date: Mon, 20 Jul 2009 10:15:20 +0000 Subject: [Freeswitch-users] Freeswitch Windows Issues In-Reply-To: References: Message-ID: <4A6443B8.5000106@gmail.com> hello, this is no an error this is only a problem if your Ipv6 is not installed go to your network connection (LAN) and install, select protocol and select Microsoft TCP/IP version 6 thanks Baskar wrote: > *Hi, > > I have installed freeswitch in windows, but when i star the > freeswitch i get this error. Due to this i cant able to register my > extension in softphone. how can i resolve this problem. > > 2009-07-20 10:55:05.390625 [CONSOLE] switch_core.c:1465 > FreeSWITCH Version 1.0.trunk (13754M) Started. > Crash Protection [Disabled] > Max Sessions[1000] > Session Rate[30] > SQL [Enabled] > > freeswitch at Baskar> 2009-07-20 10:55:05.812500 [ERR] sofia.c:801 Error > Creating SIP UA for profile: internal-ipv6 > > One more question in windows whether it is possible to connect the > ODBC connection through JavaScript in freeswitch. > > I have configured inbound in Linux it is working fine but same script > i tried in windows but i get this error. I have installed and > configured MYSQL connector ODBC in window. But when is call the script > i get this error. > > 2009-07-20 10:55:42.781250 [ERR] mod_spidermonkey.c:3393 Error loading > ODBC > 2009-07-20 10:55:42.781250 [ERR] Inbound1.js:5 ReferenceError: ODBC > is not defined > > Can some one assist me to resolve this above error > > Thanks in advance. > > -- > Thanks with 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 > > > > __________ Information from ESET NOD32 Antivirus, version of virus signature database 4260 (20090720) __________ > > The message was checked by ESET NOD32 Antivirus. > > http://www.eset.com > > __________ Information from ESET NOD32 Antivirus, version of virus signature database 4260 (20090720) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090720/6ff9bb19/attachment.html From yudha2008 at gmail.com Mon Jul 20 03:21:11 2009 From: yudha2008 at gmail.com (Baskar) Date: Mon, 20 Jul 2009 15:51:11 +0530 Subject: [Freeswitch-users] Freeswitch Windows Issues In-Reply-To: <4A6443B8.5000106@gmail.com> References: <4A6443B8.5000106@gmail.com> Message-ID: *Hi Meftah Tayeb**,* *One more question in windows whether it is possible to connect the ODBC connection through JavaScript in freeswitch. I have configured inbound in Linux it is working fine but same script i tried in windows but i get this error. I have installed and configured MYSQL connector ODBC in window. But when is call the script i get this error. 2009-07-20 10:55:42.781250 [ERR] mod_spidermonkey.c:3393 Error loading ODBC 2009-07-20 10:55:42.781250 [ERR] Inbound1.js:5 ReferenceError: ODBC is not defined Can some one assist me to resolve this above error Thanks in advance. -- Thanks with Regards, N.Baskar * -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090720/a5af8dbc/attachment.html From regs at kinetix.gr Mon Jul 20 04:01:58 2009 From: regs at kinetix.gr (Apostolos Pantsiopoulos) Date: Mon, 20 Jul 2009 14:01:58 +0300 Subject: [Freeswitch-users] Possible memory leak - need a second opinion Message-ID: <4A644EA6.70209@kinetix.gr> Hi I noticed that after a day of relatively moderate traffic (about 400 simultaneous channels average) the memory used by FS reached 1.3 GB of RAM. I tried to trace the leak (if any) with valgrind and got that : ==18894== 572 bytes in 1 blocks are definitely lost in loss record 148 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x7AFDE93: sofia_glue_do_invite (sofia_glue.c:1599) ==18894== by 0x7ACB191: sofia_on_init (mod_sofia.c:102) ==18894== by 0x405B124: switch_core_session_run (switch_core_state_machine.c:480) ==18894== by 0x4058204: switch_core_session_thread (switch_core_session.c:1064) ==18894== by 0x40CD6C5: dummy_worker (thread.c:138) ==18894== by 0xB5346A: start_thread (in /lib/libpthread-2.5.so) ==18894== by 0xAAADBD: clone (in /lib/libc-2.5.so) So, I assummed that this happens for every call. I tried testing it again by placing two calls before shutting down FS, but it only came up once. I wanted to get a second opinion before posting this to JIRA as an issue. I used revision 14269 of the SVN. I am attaching the valgrind output as well. I also noticed that only one of my CPU cores gets really busy when dealing with moderate traffic. From what I read in the mailing list users are encouraged to use 64bit multi core servers for FS because it scales up better. But this is not what I am seeing in practice. Could the single threaded architecture of libsofia be the cause of that behavior? -- ------------------------------------------- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: regs at kinetix.gr ------------------------------------------- -------------- next part -------------- ==18894== Memcheck, a memory error detector. ==18894== Copyright (C) 2002-2006, and GNU GPL'd, by Julian Seward et al. ==18894== Using LibVEX rev 1658, a library for dynamic binary translation. ==18894== Copyright (C) 2004-2006, and GNU GPL'd, by OpenWorks LLP. ==18894== Using valgrind-3.2.1, a dynamic binary instrumentation framework. ==18894== Copyright (C) 2000-2006, and GNU GPL'd, by Julian Seward et al. ==18894== For more details, rerun with: -v ==18894== ==18894== My PID = 18894, parent PID = 18741. Prog and args are: ==18894== /usr/local/bin/freeswitch ==18894== -vg ==18894== ==18894== Thread 13: ==18894== Syscall param epoll_ctl(event) points to uninitialised byte(s) ==18894== at 0xAAB3AE: epoll_ctl (in /lib/libc-2.5.so) ==18894== by 0x403E62C: switch_pollset_add (switch_apr.c:802) ==18894== by 0x403E72D: switch_socket_create_pollfd (switch_apr.c:842) ==18894== by 0x407A785: switch_rtp_set_local_address (switch_rtp.c:824) ==18894== by 0x407AD82: switch_rtp_new (switch_rtp.c:1242) ==18894== by 0x7AF8C74: sofia_glue_activate_rtp (sofia_glue.c:2322) ==18894== by 0x7ADECA5: sofia_event_callback (sofia.c:3772) ==18894== by 0x7B72DD8: nua_application_event (nua_stack.c:393) ==18894== by 0x7BB784C: su_base_port_execute_msgs (su_base_port.c:280) ==18894== by 0x7BB75F3: su_base_port_getmsgs (su_base_port.c:202) ==18894== by 0x7BB7B63: su_base_port_step (su_base_port.c:473) ==18894== by 0x7BC863A: su_port_step (su_port.h:340) ==18894== Address 0xCE52B58 is on thread 13's stack ==18894== ==18894== ERROR SUMMARY: 2 errors from 1 contexts (suppressed: 281 from 1) ==18894== malloc/free: in use at exit: 927,131 bytes in 606 blocks. ==18894== malloc/free: 360,322 allocs, 359,716 frees, 661,660,377 bytes allocated. ==18894== For counts of detected errors, rerun with: -v ==18894== searching for pointers to 606 not-freed blocks. ==18894== checked 22,451,028 bytes. ==18894== ==18894== Thread 1: ==18894== ==18894== 3 bytes in 1 blocks are still reachable in loss record 1 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0xA48DEF: strdup (in /lib/libc-2.5.so) ==18894== by 0x127B7848: PR_NewLogModule (prlog.c:363) ==18894== by 0x127BE34B: _PR_InitStuff (prinit.c:193) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== ==18894== ==18894== 3 bytes in 1 blocks are still reachable in loss record 2 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0xA48DEF: strdup (in /lib/libc-2.5.so) ==18894== by 0x127B7848: PR_NewLogModule (prlog.c:363) ==18894== by 0x127BE2BB: _PR_InitStuff (prinit.c:187) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== ==18894== ==18894== 3 bytes in 1 blocks are still reachable in loss record 3 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x127BB923: PR_Malloc (prmem.c:448) ==18894== by 0x127BDCFD: PR_CreateStack (pratom.c:386) ==18894== by 0x127B2AAB: _PR_InitFdCache (prfdcach.c:285) ==18894== by 0x127C6520: _PR_InitIO (ptio.c:1109) ==18894== by 0x127BE40E: _PR_InitStuff (prinit.c:236) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== ==18894== ==18894== 4 bytes in 1 blocks are still reachable in loss record 4 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0xA48DEF: strdup (in /lib/libc-2.5.so) ==18894== by 0x127B7848: PR_NewLogModule (prlog.c:363) ==18894== by 0x127BE363: _PR_InitStuff (prinit.c:194) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== ==18894== ==18894== 4 bytes in 1 blocks are still reachable in loss record 5 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0xA48DEF: strdup (in /lib/libc-2.5.so) ==18894== by 0x127B7848: PR_NewLogModule (prlog.c:363) ==18894== by 0x127BE2D3: _PR_InitStuff (prinit.c:188) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== ==18894== ==18894== 4 bytes in 1 blocks are still reachable in loss record 6 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x22AC7D: (within /lib/libcrypto.so.0.9.8b) ==18894== by 0x22B2FE: CRYPTO_malloc (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x26E4D6: (within /lib/libcrypto.so.0.9.8b) ==18894== by 0x26E51D: engine_cleanup_add_last (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x26EFC6: ENGINE_add (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x272711: ENGINE_load_dynamic (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x270BE6: ENGINE_load_builtin_engines (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x127FE136: Curl_ossl_init (in /usr/lib/libcurl.so.3.0.0) ==18894== by 0x1280F3AB: Curl_ssl_init (in /usr/lib/libcurl.so.3.0.0) ==18894== by 0x12807162: curl_global_init (in /usr/lib/libcurl.so.3.0.0) ==18894== by 0x1279FAB3: mod_spidermonkey_load (mod_spidermonkey.c:3751) ==18894== ==18894== ==18894== 5 bytes in 1 blocks are still reachable in loss record 7 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0xA48DEF: strdup (in /lib/libc-2.5.so) ==18894== by 0x127B7848: PR_NewLogModule (prlog.c:363) ==18894== by 0x127BE37B: _PR_InitStuff (prinit.c:195) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== ==18894== ==18894== 5 bytes in 1 blocks are still reachable in loss record 8 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0xA48DEF: strdup (in /lib/libc-2.5.so) ==18894== by 0x127B7848: PR_NewLogModule (prlog.c:363) ==18894== by 0x127BE303: _PR_InitStuff (prinit.c:190) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== ==18894== ==18894== 5 bytes in 1 blocks are still reachable in loss record 9 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0xA48DEF: strdup (in /lib/libc-2.5.so) ==18894== by 0x127B7848: PR_NewLogModule (prlog.c:363) ==18894== by 0x127BE2A3: _PR_InitStuff (prinit.c:186) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== ==18894== ==18894== 6 bytes in 1 blocks are still reachable in loss record 10 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0xA48DEF: strdup (in /lib/libc-2.5.so) ==18894== by 0x127B7848: PR_NewLogModule (prlog.c:363) ==18894== by 0x127BE31B: _PR_InitStuff (prinit.c:191) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== ==18894== ==18894== 6 bytes in 1 blocks are still reachable in loss record 11 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0xA48DEF: strdup (in /lib/libc-2.5.so) ==18894== by 0x127B7848: PR_NewLogModule (prlog.c:363) ==18894== by 0x127BE28B: _PR_InitStuff (prinit.c:185) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== ==18894== ==18894== 6 bytes in 1 blocks are still reachable in loss record 12 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0xA48DEF: strdup (in /lib/libc-2.5.so) ==18894== by 0xC23E31: _nc_setupterm (in /usr/lib/libncurses.so.5.5) ==18894== by 0xC24747: tgetent (in /usr/lib/libncurses.so.5.5) ==18894== by 0x4141441: term_set (term.c:925) ==18894== by 0x4141843: term_init (term.c:350) ==18894== by 0x4139E84: el_init (el.c:84) ==18894== by 0x4049449: switch_console_loop (switch_console.c:747) ==18894== by 0x4061CE7: switch_core_runtime_loop (switch_core.c:687) ==18894== by 0x804A453: main (switch.c:715) ==18894== ==18894== ==18894== 7 bytes in 1 blocks are still reachable in loss record 13 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0xA48DEF: strdup (in /lib/libc-2.5.so) ==18894== by 0x127B7848: PR_NewLogModule (prlog.c:363) ==18894== by 0x127BE333: _PR_InitStuff (prinit.c:192) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== ==18894== ==18894== 7 bytes in 1 blocks are still reachable in loss record 14 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0xA48DEF: strdup (in /lib/libc-2.5.so) ==18894== by 0x127B7848: PR_NewLogModule (prlog.c:363) ==18894== by 0x127BE2EB: _PR_InitStuff (prinit.c:189) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== ==18894== ==18894== 8 bytes in 1 blocks are still reachable in loss record 15 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0xA48DEF: strdup (in /lib/libc-2.5.so) ==18894== by 0xC2A532: (within /usr/lib/libncurses.so.5.5) ==18894== by 0xC2A837: _nc_trim_sgr0 (in /usr/lib/libncurses.so.5.5) ==18894== by 0xC248BE: tgetent (in /usr/lib/libncurses.so.5.5) ==18894== by 0x4141441: term_set (term.c:925) ==18894== by 0x4141843: term_init (term.c:350) ==18894== by 0x4139E84: el_init (el.c:84) ==18894== by 0x4049449: switch_console_loop (switch_console.c:747) ==18894== by 0x4061CE7: switch_core_runtime_loop (switch_core.c:687) ==18894== by 0x804A453: main (switch.c:715) ==18894== ==18894== ==18894== 8 bytes in 1 blocks are still reachable in loss record 16 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40DED94: sqlite3MallocX (sqliteInt.h:278) ==18894== by 0x40D3D03: sqlite3HashInsert (hash.c:366) ==18894== by 0x40531EE: switch_core_hash_insert (switch_core_hash.c:67) ==18894== by 0x7AA4EF1: mod_cdr_csv_load (mod_cdr_csv.c:314) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 8 bytes in 1 blocks are still reachable in loss record 17 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40DED94: sqlite3MallocX (sqliteInt.h:278) ==18894== by 0x40D3D03: sqlite3HashInsert (hash.c:366) ==18894== by 0x40531EE: switch_core_hash_insert (switch_core_hash.c:67) ==18894== by 0x400EC26: mod_logfile_load (mod_logfile.c:290) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 8 bytes in 1 blocks are still reachable in loss record 18 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x407176E: my_dup (switch_event.c:95) ==18894== by 0x40737A4: switch_event_bind_removable (switch_event.c:1176) ==18894== by 0x4073960: switch_event_bind (switch_event.c:1205) ==18894== by 0x4053B7A: switch_core_sqldb_start (switch_core_sqldb.c:584) ==18894== by 0x4063A91: switch_core_init (switch_core.c:1248) ==18894== by 0x406585C: switch_core_init_and_modload (switch_core.c:1422) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 9 bytes in 1 blocks are still reachable in loss record 19 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40DED94: sqlite3MallocX (sqliteInt.h:278) ==18894== by 0x40D3D03: sqlite3HashInsert (hash.c:366) ==18894== by 0x40531EE: switch_core_hash_insert (switch_core_hash.c:67) ==18894== by 0x4064603: switch_load_network_lists (switch_core.c:909) ==18894== by 0x40659E4: switch_core_init_and_modload (switch_core.c:1457) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 9 bytes in 1 blocks are still reachable in loss record 20 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40DED94: sqlite3MallocX (sqliteInt.h:278) ==18894== by 0x40D3D03: sqlite3HashInsert (hash.c:366) ==18894== by 0x40531EE: switch_core_hash_insert (switch_core_hash.c:67) ==18894== by 0x4064724: switch_load_network_lists (switch_core.c:921) ==18894== by 0x40659E4: switch_core_init_and_modload (switch_core.c:1457) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 12 bytes in 1 blocks are still reachable in loss record 21 of 193 ==18894== at 0x40046FF: calloc (vg_replace_malloc.c:279) ==18894== by 0x127BB985: PR_Calloc (prmem.c:455) ==18894== by 0x127B7834: PR_NewLogModule (prlog.c:361) ==18894== by 0x127BE28B: _PR_InitStuff (prinit.c:185) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== ==18894== ==18894== 12 bytes in 1 blocks are still reachable in loss record 22 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x22AC7D: (within /lib/libcrypto.so.0.9.8b) ==18894== by 0x22B2FE: CRYPTO_malloc (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x27E53F: lh_insert (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x280EDF: (within /lib/libcrypto.so.0.9.8b) ==18894== by 0x28141B: ERR_get_state (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x281ADA: ERR_clear_error (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x27271E: ENGINE_load_dynamic (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x270BE6: ENGINE_load_builtin_engines (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x127FE136: Curl_ossl_init (in /usr/lib/libcurl.so.3.0.0) ==18894== by 0x1280F3AB: Curl_ssl_init (in /usr/lib/libcurl.so.3.0.0) ==18894== by 0x12807162: curl_global_init (in /usr/lib/libcurl.so.3.0.0) ==18894== ==18894== ==18894== 12 bytes in 1 blocks are still reachable in loss record 23 of 193 ==18894== at 0x40046FF: calloc (vg_replace_malloc.c:279) ==18894== by 0x127BB985: PR_Calloc (prmem.c:455) ==18894== by 0x127B7834: PR_NewLogModule (prlog.c:361) ==18894== by 0x127BE2BB: _PR_InitStuff (prinit.c:187) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== ==18894== ==18894== 12 bytes in 1 blocks are still reachable in loss record 24 of 193 ==18894== at 0x40046FF: calloc (vg_replace_malloc.c:279) ==18894== by 0x127BB985: PR_Calloc (prmem.c:455) ==18894== by 0x127B2C12: _PR_InitMW (prmwait.c:243) ==18894== by 0x127BE41D: _PR_InitStuff (prinit.c:239) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== ==18894== ==18894== 12 bytes in 1 blocks are still reachable in loss record 25 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x127BB923: PR_Malloc (prmem.c:448) ==18894== by 0x127BDCBA: PR_CreateStack (pratom.c:382) ==18894== by 0x127B2AAB: _PR_InitFdCache (prfdcach.c:285) ==18894== by 0x127C6520: _PR_InitIO (ptio.c:1109) ==18894== by 0x127BE40E: _PR_InitStuff (prinit.c:236) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== ==18894== ==18894== 12 bytes in 1 blocks are still reachable in loss record 26 of 193 ==18894== at 0x40046FF: calloc (vg_replace_malloc.c:279) ==18894== by 0x127BB985: PR_Calloc (prmem.c:455) ==18894== by 0x127B7834: PR_NewLogModule (prlog.c:361) ==18894== by 0x127BE37B: _PR_InitStuff (prinit.c:195) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== ==18894== ==18894== 12 bytes in 1 blocks are still reachable in loss record 27 of 193 ==18894== at 0x40046FF: calloc (vg_replace_malloc.c:279) ==18894== by 0x127BB985: PR_Calloc (prmem.c:455) ==18894== by 0x127B7834: PR_NewLogModule (prlog.c:361) ==18894== by 0x127BE363: _PR_InitStuff (prinit.c:194) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== ==18894== ==18894== 12 bytes in 1 blocks are still reachable in loss record 28 of 193 ==18894== at 0x40046FF: calloc (vg_replace_malloc.c:279) ==18894== by 0x127BB985: PR_Calloc (prmem.c:455) ==18894== by 0x127B7834: PR_NewLogModule (prlog.c:361) ==18894== by 0x127BE34B: _PR_InitStuff (prinit.c:193) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== ==18894== ==18894== 12 bytes in 1 blocks are still reachable in loss record 29 of 193 ==18894== at 0x40046FF: calloc (vg_replace_malloc.c:279) ==18894== by 0x127BB985: PR_Calloc (prmem.c:455) ==18894== by 0x127B7834: PR_NewLogModule (prlog.c:361) ==18894== by 0x127BE333: _PR_InitStuff (prinit.c:192) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== ==18894== ==18894== 12 bytes in 1 blocks are still reachable in loss record 30 of 193 ==18894== at 0x40046FF: calloc (vg_replace_malloc.c:279) ==18894== by 0x127BB985: PR_Calloc (prmem.c:455) ==18894== by 0x127B7834: PR_NewLogModule (prlog.c:361) ==18894== by 0x127BE31B: _PR_InitStuff (prinit.c:191) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== ==18894== ==18894== 12 bytes in 1 blocks are still reachable in loss record 31 of 193 ==18894== at 0x40046FF: calloc (vg_replace_malloc.c:279) ==18894== by 0x127BB985: PR_Calloc (prmem.c:455) ==18894== by 0x127B7834: PR_NewLogModule (prlog.c:361) ==18894== by 0x127BE303: _PR_InitStuff (prinit.c:190) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== ==18894== ==18894== 12 bytes in 1 blocks are still reachable in loss record 32 of 193 ==18894== at 0x40046FF: calloc (vg_replace_malloc.c:279) ==18894== by 0x127BB985: PR_Calloc (prmem.c:455) ==18894== by 0x127B7834: PR_NewLogModule (prlog.c:361) ==18894== by 0x127BE2EB: _PR_InitStuff (prinit.c:189) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== ==18894== ==18894== 12 bytes in 1 blocks are still reachable in loss record 33 of 193 ==18894== at 0x40046FF: calloc (vg_replace_malloc.c:279) ==18894== by 0x127BB985: PR_Calloc (prmem.c:455) ==18894== by 0x127B7834: PR_NewLogModule (prlog.c:361) ==18894== by 0x127BE2D3: _PR_InitStuff (prinit.c:188) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== ==18894== ==18894== 12 bytes in 1 blocks are still reachable in loss record 34 of 193 ==18894== at 0x40046FF: calloc (vg_replace_malloc.c:279) ==18894== by 0x127BB985: PR_Calloc (prmem.c:455) ==18894== by 0x127B7834: PR_NewLogModule (prlog.c:361) ==18894== by 0x127BE2A3: _PR_InitStuff (prinit.c:186) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== ==18894== ==18894== 12 bytes in 2 blocks are still reachable in loss record 35 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40DED94: sqlite3MallocX (sqliteInt.h:278) ==18894== by 0x40D3D03: sqlite3HashInsert (hash.c:366) ==18894== by 0x40531EE: switch_core_hash_insert (switch_core_hash.c:67) ==18894== by 0x4064E08: switch_load_network_lists (switch_core.c:1077) ==18894== by 0x40659E4: switch_core_init_and_modload (switch_core.c:1457) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 12 bytes in 1 blocks are still reachable in loss record 36 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x22AC7D: (within /lib/libcrypto.so.0.9.8b) ==18894== by 0x22B2FE: CRYPTO_malloc (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x4C048E: (within /lib/libssl.so.0.9.8b) ==18894== by 0x4C06B6: SSL_COMP_get_compression_methods (in /lib/libssl.so.0.9.8b) ==18894== by 0x4C61F4: SSL_library_init (in /lib/libssl.so.0.9.8b) ==18894== by 0x127FE140: Curl_ossl_init (in /usr/lib/libcurl.so.3.0.0) ==18894== by 0x1280F3AB: Curl_ssl_init (in /usr/lib/libcurl.so.3.0.0) ==18894== by 0x12807162: curl_global_init (in /usr/lib/libcurl.so.3.0.0) ==18894== by 0x1279FAB3: mod_spidermonkey_load (mod_spidermonkey.c:3751) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== ==18894== ==18894== 12 bytes in 1 blocks are still reachable in loss record 37 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x9BE859: decompose_rpath (in /lib/ld-2.5.so) ==18894== by 0x9BF831: _dl_map_object (in /lib/ld-2.5.so) ==18894== by 0x9C32D5: openaux (in /lib/ld-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0x9C3894: _dl_map_object_deps (in /lib/ld-2.5.so) ==18894== by 0x9C8A8C: dl_open_worker (in /lib/ld-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0x9C8411: _dl_open (in /lib/ld-2.5.so) ==18894== by 0xB1FC4C: dlopen_doit (in /lib/libdl-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0xB202CB: _dlerror_run (in /lib/libdl-2.5.so) ==18894== ==18894== ==18894== 13 bytes in 1 blocks are definitely lost in loss record 38 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x412F754: crypto_alloc (alloc.c:97) ==18894== by 0x412FF9D: null_cipher_alloc (null_cipher.c:68) ==18894== by 0x412B53C: cipher_type_self_test (cipher.c:264) ==18894== by 0x412F0B5: crypto_kernel_load_cipher_type (crypto_kernel.c:310) ==18894== by 0x412F62C: crypto_kernel_init (crypto_kernel.c:151) ==18894== by 0x4129026: srtp_init (srtp.c:1081) ==18894== by 0x407AF62: switch_rtp_init (switch_rtp.c:611) ==18894== by 0x40639D6: switch_core_init (switch_core.c:1252) ==18894== by 0x406585C: switch_core_init_and_modload (switch_core.c:1422) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 13 bytes in 1 blocks are still reachable in loss record 39 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40DED94: sqlite3MallocX (sqliteInt.h:278) ==18894== by 0x40D3D03: sqlite3HashInsert (hash.c:366) ==18894== by 0x40531EE: switch_core_hash_insert (switch_core_hash.c:67) ==18894== by 0x4064511: switch_load_network_lists (switch_core.c:901) ==18894== by 0x40659E4: switch_core_init_and_modload (switch_core.c:1457) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 14 bytes in 1 blocks are still reachable in loss record 40 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40DED94: sqlite3MallocX (sqliteInt.h:278) ==18894== by 0x40D3D03: sqlite3HashInsert (hash.c:366) ==18894== by 0x40531EE: switch_core_hash_insert (switch_core_hash.c:67) ==18894== by 0x4064881: switch_load_network_lists (switch_core.c:937) ==18894== by 0x40659E4: switch_core_init_and_modload (switch_core.c:1457) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 14 bytes in 1 blocks are still reachable in loss record 41 of 193 ==18894== at 0x40054BB: realloc (vg_replace_malloc.c:306) ==18894== by 0xC21BDF: _nc_doalloc (in /usr/lib/libncurses.so.5.5) ==18894== by 0xC25667: tparm (in /usr/lib/libncurses.so.5.5) ==18894== by 0xC2A526: (within /usr/lib/libncurses.so.5.5) ==18894== by 0xC2A82B: _nc_trim_sgr0 (in /usr/lib/libncurses.so.5.5) ==18894== by 0xC248BE: tgetent (in /usr/lib/libncurses.so.5.5) ==18894== by 0x4141441: term_set (term.c:925) ==18894== by 0x4141843: term_init (term.c:350) ==18894== by 0x4139E84: el_init (el.c:84) ==18894== by 0x4049449: switch_console_loop (switch_console.c:747) ==18894== by 0x4061CE7: switch_core_runtime_loop (switch_core.c:687) ==18894== by 0x804A453: main (switch.c:715) ==18894== ==18894== ==18894== 14 bytes in 1 blocks are still reachable in loss record 42 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40DED94: sqlite3MallocX (sqliteInt.h:278) ==18894== by 0x40D3D03: sqlite3HashInsert (hash.c:366) ==18894== by 0x40531EE: switch_core_hash_insert (switch_core_hash.c:67) ==18894== by 0x40647CC: switch_load_network_lists (switch_core.c:927) ==18894== by 0x40659E4: switch_core_init_and_modload (switch_core.c:1457) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 16 bytes in 1 blocks are still reachable in loss record 43 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x22AC7D: (within /lib/libcrypto.so.0.9.8b) ==18894== by 0x22B2FE: CRYPTO_malloc (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x27DED9: sk_new (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x27DF4D: sk_new_null (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x26E3FA: (within /lib/libcrypto.so.0.9.8b) ==18894== by 0x26E50B: engine_cleanup_add_last (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x26EFC6: ENGINE_add (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x272711: ENGINE_load_dynamic (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x270BE6: ENGINE_load_builtin_engines (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x127FE136: Curl_ossl_init (in /usr/lib/libcurl.so.3.0.0) ==18894== by 0x1280F3AB: Curl_ssl_init (in /usr/lib/libcurl.so.3.0.0) ==18894== ==18894== ==18894== 16 bytes in 1 blocks are still reachable in loss record 44 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x22AC7D: (within /lib/libcrypto.so.0.9.8b) ==18894== by 0x22B2FE: CRYPTO_malloc (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x27DED9: sk_new (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x4C046C: (within /lib/libssl.so.0.9.8b) ==18894== by 0x4C06B6: SSL_COMP_get_compression_methods (in /lib/libssl.so.0.9.8b) ==18894== by 0x4C61F4: SSL_library_init (in /lib/libssl.so.0.9.8b) ==18894== by 0x127FE140: Curl_ossl_init (in /usr/lib/libcurl.so.3.0.0) ==18894== by 0x1280F3AB: Curl_ssl_init (in /usr/lib/libcurl.so.3.0.0) ==18894== by 0x12807162: curl_global_init (in /usr/lib/libcurl.so.3.0.0) ==18894== by 0x1279FAB3: mod_spidermonkey_load (mod_spidermonkey.c:3751) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== ==18894== ==18894== 18 bytes in 1 blocks are still reachable in loss record 45 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0xC21DCE: _nc_home_terminfo (in /usr/lib/libncurses.so.5.5) ==18894== by 0xC29449: _nc_next_db (in /usr/lib/libncurses.so.5.5) ==18894== by 0xC29F31: _nc_read_entry (in /usr/lib/libncurses.so.5.5) ==18894== by 0xC23FDC: _nc_setupterm (in /usr/lib/libncurses.so.5.5) ==18894== by 0xC24747: tgetent (in /usr/lib/libncurses.so.5.5) ==18894== by 0x4141441: term_set (term.c:925) ==18894== by 0x4141843: term_init (term.c:350) ==18894== by 0x4139E84: el_init (el.c:84) ==18894== by 0x4049449: switch_console_loop (switch_console.c:747) ==18894== by 0x4061CE7: switch_core_runtime_loop (switch_core.c:687) ==18894== by 0x804A453: main (switch.c:715) ==18894== ==18894== ==18894== 20 bytes in 1 blocks are still reachable in loss record 46 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x22AC7D: (within /lib/libcrypto.so.0.9.8b) ==18894== by 0x22B2FE: CRYPTO_malloc (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x27DEB6: sk_new (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x27DF4D: sk_new_null (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x26E3FA: (within /lib/libcrypto.so.0.9.8b) ==18894== by 0x26E50B: engine_cleanup_add_last (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x26EFC6: ENGINE_add (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x272711: ENGINE_load_dynamic (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x270BE6: ENGINE_load_builtin_engines (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x127FE136: Curl_ossl_init (in /usr/lib/libcurl.so.3.0.0) ==18894== by 0x1280F3AB: Curl_ssl_init (in /usr/lib/libcurl.so.3.0.0) ==18894== ==18894== ==18894== 20 bytes in 1 blocks are still reachable in loss record 47 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40DED94: sqlite3MallocX (sqliteInt.h:278) ==18894== by 0x40D3CD2: sqlite3HashInsert (hash.c:363) ==18894== by 0x40531EE: switch_core_hash_insert (switch_core_hash.c:67) ==18894== by 0x4064881: switch_load_network_lists (switch_core.c:937) ==18894== by 0x40659E4: switch_core_init_and_modload (switch_core.c:1457) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 20 bytes in 1 blocks are still reachable in loss record 48 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40DED94: sqlite3MallocX (sqliteInt.h:278) ==18894== by 0x40D3CD2: sqlite3HashInsert (hash.c:363) ==18894== by 0x40531EE: switch_core_hash_insert (switch_core_hash.c:67) ==18894== by 0x40647CC: switch_load_network_lists (switch_core.c:927) ==18894== by 0x40659E4: switch_core_init_and_modload (switch_core.c:1457) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 20 bytes in 1 blocks are still reachable in loss record 49 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40DED94: sqlite3MallocX (sqliteInt.h:278) ==18894== by 0x40D3CD2: sqlite3HashInsert (hash.c:363) ==18894== by 0x40531EE: switch_core_hash_insert (switch_core_hash.c:67) ==18894== by 0x4064724: switch_load_network_lists (switch_core.c:921) ==18894== by 0x40659E4: switch_core_init_and_modload (switch_core.c:1457) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 20 bytes in 1 blocks are still reachable in loss record 50 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40DED94: sqlite3MallocX (sqliteInt.h:278) ==18894== by 0x40D3CD2: sqlite3HashInsert (hash.c:363) ==18894== by 0x40531EE: switch_core_hash_insert (switch_core_hash.c:67) ==18894== by 0x7AA4EF1: mod_cdr_csv_load (mod_cdr_csv.c:314) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 20 bytes in 1 blocks are still reachable in loss record 51 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40DED94: sqlite3MallocX (sqliteInt.h:278) ==18894== by 0x40D3CD2: sqlite3HashInsert (hash.c:363) ==18894== by 0x40531EE: switch_core_hash_insert (switch_core_hash.c:67) ==18894== by 0x400EC26: mod_logfile_load (mod_logfile.c:290) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 20 bytes in 1 blocks are still reachable in loss record 52 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40DED94: sqlite3MallocX (sqliteInt.h:278) ==18894== by 0x40D3CD2: sqlite3HashInsert (hash.c:363) ==18894== by 0x40531EE: switch_core_hash_insert (switch_core_hash.c:67) ==18894== by 0x4064603: switch_load_network_lists (switch_core.c:909) ==18894== by 0x40659E4: switch_core_init_and_modload (switch_core.c:1457) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 20 bytes in 1 blocks are still reachable in loss record 53 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x400544A: realloc (vg_replace_malloc.c:306) ==18894== by 0x9C163F: _dl_lookup_symbol_x (in /lib/ld-2.5.so) ==18894== by 0x9C2334: _dl_relocate_object (in /lib/ld-2.5.so) ==18894== by 0x9C8B67: dl_open_worker (in /lib/ld-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0x9C8411: _dl_open (in /lib/ld-2.5.so) ==18894== by 0xB1FC4C: dlopen_doit (in /lib/libdl-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0xB202CB: _dlerror_run (in /lib/libdl-2.5.so) ==18894== by 0xB1FB83: dlopen@@GLIBC_2.1 (in /lib/libdl-2.5.so) ==18894== by 0x4067486: switch_dso_open (switch_dso.c:94) ==18894== ==18894== ==18894== 20 bytes in 1 blocks are still reachable in loss record 54 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x22AC7D: (within /lib/libcrypto.so.0.9.8b) ==18894== by 0x22B2FE: CRYPTO_malloc (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x27DEB6: sk_new (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x4C046C: (within /lib/libssl.so.0.9.8b) ==18894== by 0x4C06B6: SSL_COMP_get_compression_methods (in /lib/libssl.so.0.9.8b) ==18894== by 0x4C61F4: SSL_library_init (in /lib/libssl.so.0.9.8b) ==18894== by 0x127FE140: Curl_ossl_init (in /usr/lib/libcurl.so.3.0.0) ==18894== by 0x1280F3AB: Curl_ssl_init (in /usr/lib/libcurl.so.3.0.0) ==18894== by 0x12807162: curl_global_init (in /usr/lib/libcurl.so.3.0.0) ==18894== by 0x1279FAB3: mod_spidermonkey_load (mod_spidermonkey.c:3751) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== ==18894== ==18894== 20 bytes in 1 blocks are still reachable in loss record 55 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40DED94: sqlite3MallocX (sqliteInt.h:278) ==18894== by 0x40D3CD2: sqlite3HashInsert (hash.c:363) ==18894== by 0x40531EE: switch_core_hash_insert (switch_core_hash.c:67) ==18894== by 0x4064511: switch_load_network_lists (switch_core.c:901) ==18894== by 0x40659E4: switch_core_init_and_modload (switch_core.c:1457) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 20 bytes in 1 blocks are still reachable in loss record 56 of 193 ==18894== at 0x40046FF: calloc (vg_replace_malloc.c:279) ==18894== by 0xB2033B: _dlerror_run (in /lib/libdl-2.5.so) ==18894== by 0xB1FB83: dlopen@@GLIBC_2.1 (in /lib/libdl-2.5.so) ==18894== by 0x40674AF: switch_dso_open (switch_dso.c:96) ==18894== by 0x406C800: switch_loadable_module_load_module_ex (switch_loadable_module.c:795) ==18894== by 0x406D073: switch_loadable_module_init (switch_loadable_module.c:1159) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 20 bytes in 1 blocks are still reachable in loss record 57 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40DED94: sqlite3MallocX (sqliteInt.h:278) ==18894== by 0x40D3CD2: sqlite3HashInsert (hash.c:363) ==18894== by 0x40531EE: switch_core_hash_insert (switch_core_hash.c:67) ==18894== by 0x7AA4812: ??? (mod_cdr_csv.c:140) ==18894== by 0x7AA4B55: ??? (mod_cdr_csv.c:233) ==18894== by 0x405A631: switch_core_session_reporting_state (switch_core_state_machine.c:607) ==18894== by 0x405AC1A: switch_core_session_run (switch_core_state_machine.c:409) ==18894== by 0x4058204: switch_core_session_thread (switch_core_session.c:1064) ==18894== by 0x40CD6C5: dummy_worker (thread.c:138) ==18894== by 0xB5346A: start_thread (in /lib/libpthread-2.5.so) ==18894== by 0xAAADBD: clone (in /lib/libc-2.5.so) ==18894== ==18894== ==18894== 20 bytes in 1 blocks are still reachable in loss record 58 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40DED94: sqlite3MallocX (sqliteInt.h:278) ==18894== by 0x40D3CD2: sqlite3HashInsert (hash.c:363) ==18894== by 0x40531EE: switch_core_hash_insert (switch_core_hash.c:67) ==18894== by 0x7AA4812: ??? (mod_cdr_csv.c:140) ==18894== by 0x7AA4AA0: ??? (mod_cdr_csv.c:251) ==18894== by 0x405A631: switch_core_session_reporting_state (switch_core_state_machine.c:607) ==18894== by 0x405AC1A: switch_core_session_run (switch_core_state_machine.c:409) ==18894== by 0x4058204: switch_core_session_thread (switch_core_session.c:1064) ==18894== by 0x40CD6C5: dummy_worker (thread.c:138) ==18894== by 0xB5346A: start_thread (in /lib/libpthread-2.5.so) ==18894== by 0xAAADBD: clone (in /lib/libc-2.5.so) ==18894== ==18894== ==18894== 21 bytes in 1 blocks are definitely lost in loss record 59 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0xA48DEF: strdup (in /lib/libc-2.5.so) ==18894== by 0x40A9A52: switch_xml_config_parse_event (switch_xml_config.c:267) ==18894== by 0x11CAC65C: ??? (mod_voicemail.c:643) ==18894== by 0x11CAD21F: mod_voicemail_load (mod_voicemail.c:821) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 22 bytes in 1 blocks are still reachable in loss record 60 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x407176E: my_dup (switch_event.c:95) ==18894== by 0x40737A4: switch_event_bind_removable (switch_event.c:1176) ==18894== by 0x40ABD58: softtimer_load (switch_time.c:753) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D073: switch_loadable_module_init (switch_loadable_module.c:1159) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 24 bytes in 1 blocks are still reachable in loss record 61 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x127BB923: PR_Malloc (prmem.c:448) ==18894== by 0x127B26D2: _PR_Getfd (prfdcach.c:141) ==18894== by 0x127C8EBA: pt_SetMethods (ptio.c:3248) ==18894== by 0x127C65A0: _PR_InitIO (ptio.c:1113) ==18894== by 0x127BE40E: _PR_InitStuff (prinit.c:236) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== ==18894== ==18894== 24 bytes in 1 blocks are still reachable in loss record 62 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x127BB923: PR_Malloc (prmem.c:448) ==18894== by 0x127B26D2: _PR_Getfd (prfdcach.c:141) ==18894== by 0x127C8EBA: pt_SetMethods (ptio.c:3248) ==18894== by 0x127C6572: _PR_InitIO (ptio.c:1112) ==18894== by 0x127BE40E: _PR_InitStuff (prinit.c:236) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== ==18894== ==18894== 24 bytes in 1 blocks are still reachable in loss record 63 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x127BB923: PR_Malloc (prmem.c:448) ==18894== by 0x127B26D2: _PR_Getfd (prfdcach.c:141) ==18894== by 0x127C8EBA: pt_SetMethods (ptio.c:3248) ==18894== by 0x127C6544: _PR_InitIO (ptio.c:1111) ==18894== by 0x127BE40E: _PR_InitStuff (prinit.c:236) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== ==18894== ==18894== 24 bytes in 1 blocks are still reachable in loss record 64 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x407374D: switch_event_bind_removable (switch_event.c:1172) ==18894== by 0x40ABD58: softtimer_load (switch_time.c:753) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D073: switch_loadable_module_init (switch_loadable_module.c:1159) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 24 bytes in 1 blocks are still reachable in loss record 65 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x407374D: switch_event_bind_removable (switch_event.c:1172) ==18894== by 0x4073960: switch_event_bind (switch_event.c:1205) ==18894== by 0x4053B7A: switch_core_sqldb_start (switch_core_sqldb.c:584) ==18894== by 0x4063A91: switch_core_init (switch_core.c:1248) ==18894== by 0x406585C: switch_core_init_and_modload (switch_core.c:1422) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 28 bytes in 1 blocks are still reachable in loss record 66 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x9C3B6B: _dl_map_object_deps (in /lib/ld-2.5.so) ==18894== by 0x9C8A8C: dl_open_worker (in /lib/ld-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0x9C8411: _dl_open (in /lib/ld-2.5.so) ==18894== by 0xAE07F1: do_dlopen (in /lib/libc-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0xAE09A4: __libc_dlopen_mode (in /lib/libc-2.5.so) ==18894== by 0xB5BB06: pthread_cancel_init (in /lib/libpthread-2.5.so) ==18894== by 0xB5BC30: _Unwind_ForcedUnwind (in /lib/libpthread-2.5.so) ==18894== by 0xB59700: __pthread_unwind (in /lib/libpthread-2.5.so) ==18894== by 0xB543CF: pthread_exit (in /lib/libpthread-2.5.so) ==18894== ==18894== ==18894== 28 bytes in 1 blocks are still reachable in loss record 67 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x127BB923: PR_Malloc (prmem.c:448) ==18894== by 0x127B26E7: _PR_Getfd (prfdcach.c:144) ==18894== by 0x127C8EBA: pt_SetMethods (ptio.c:3248) ==18894== by 0x127C65A0: _PR_InitIO (ptio.c:1113) ==18894== by 0x127BE40E: _PR_InitStuff (prinit.c:236) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== ==18894== ==18894== 28 bytes in 1 blocks are still reachable in loss record 68 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x127BB923: PR_Malloc (prmem.c:448) ==18894== by 0x127B26E7: _PR_Getfd (prfdcach.c:144) ==18894== by 0x127C8EBA: pt_SetMethods (ptio.c:3248) ==18894== by 0x127C6572: _PR_InitIO (ptio.c:1112) ==18894== by 0x127BE40E: _PR_InitStuff (prinit.c:236) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== ==18894== ==18894== 28 bytes in 1 blocks are still reachable in loss record 69 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x127BB923: PR_Malloc (prmem.c:448) ==18894== by 0x127B26E7: _PR_Getfd (prfdcach.c:144) ==18894== by 0x127C8EBA: pt_SetMethods (ptio.c:3248) ==18894== by 0x127C6544: _PR_InitIO (ptio.c:1111) ==18894== by 0x127BE40E: _PR_InitStuff (prinit.c:236) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== ==18894== ==18894== 31 bytes in 1 blocks are still reachable in loss record 70 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0xA48DEF: strdup (in /lib/libc-2.5.so) ==18894== by 0x400EBEB: mod_logfile_load (mod_logfile.c:283) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 33 bytes in 1 blocks are still reachable in loss record 71 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40DED94: sqlite3MallocX (sqliteInt.h:278) ==18894== by 0x40D3D03: sqlite3HashInsert (hash.c:366) ==18894== by 0x40531EE: switch_core_hash_insert (switch_core_hash.c:67) ==18894== by 0x7AA4812: ??? (mod_cdr_csv.c:140) ==18894== by 0x7AA4B55: ??? (mod_cdr_csv.c:233) ==18894== by 0x405A631: switch_core_session_reporting_state (switch_core_state_machine.c:607) ==18894== by 0x405AC1A: switch_core_session_run (switch_core_state_machine.c:409) ==18894== by 0x4058204: switch_core_session_thread (switch_core_session.c:1064) ==18894== by 0x40CD6C5: dummy_worker (thread.c:138) ==18894== by 0xB5346A: start_thread (in /lib/libpthread-2.5.so) ==18894== by 0xAAADBD: clone (in /lib/libc-2.5.so) ==18894== ==18894== ==18894== 34 bytes in 5 blocks are still reachable in loss record 72 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40DED94: sqlite3MallocX (sqliteInt.h:278) ==18894== by 0x40D3D03: sqlite3HashInsert (hash.c:366) ==18894== by 0x40531EE: switch_core_hash_insert (switch_core_hash.c:67) ==18894== by 0x7AA518D: mod_cdr_csv_load (mod_cdr_csv.c:361) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 35 bytes in 1 blocks are still reachable in loss record 73 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40DED94: sqlite3MallocX (sqliteInt.h:278) ==18894== by 0x40D3D03: sqlite3HashInsert (hash.c:366) ==18894== by 0x40531EE: switch_core_hash_insert (switch_core_hash.c:67) ==18894== by 0x7AA4812: ??? (mod_cdr_csv.c:140) ==18894== by 0x7AA4AA0: ??? (mod_cdr_csv.c:251) ==18894== by 0x405A631: switch_core_session_reporting_state (switch_core_state_machine.c:607) ==18894== by 0x405AC1A: switch_core_session_run (switch_core_state_machine.c:409) ==18894== by 0x4058204: switch_core_session_thread (switch_core_session.c:1064) ==18894== by 0x40CD6C5: dummy_worker (thread.c:138) ==18894== by 0xB5346A: start_thread (in /lib/libpthread-2.5.so) ==18894== by 0xAAADBD: clone (in /lib/libc-2.5.so) ==18894== ==18894== ==18894== 40 bytes in 1 blocks are still reachable in loss record 74 of 193 ==18894== at 0x40046FF: calloc (vg_replace_malloc.c:279) ==18894== by 0x127BB985: PR_Calloc (prmem.c:455) ==18894== by 0x127CC5A2: _PR_InitThreads (ptthread.c:900) ==18894== by 0x127BE404: _PR_InitStuff (prinit.c:215) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== ==18894== ==18894== 40 bytes in 2 blocks are still reachable in loss record 75 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40DED94: sqlite3MallocX (sqliteInt.h:278) ==18894== by 0x40D3CD2: sqlite3HashInsert (hash.c:363) ==18894== by 0x40531EE: switch_core_hash_insert (switch_core_hash.c:67) ==18894== by 0x4064E08: switch_load_network_lists (switch_core.c:1077) ==18894== by 0x40659E4: switch_core_init_and_modload (switch_core.c:1457) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 44 bytes in 2 blocks are still reachable in loss record 76 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x9BF9B7: _dl_map_object (in /lib/ld-2.5.so) ==18894== by 0x9C32D5: openaux (in /lib/ld-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0x9C3894: _dl_map_object_deps (in /lib/ld-2.5.so) ==18894== by 0x9C8A8C: dl_open_worker (in /lib/ld-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0x9C8411: _dl_open (in /lib/ld-2.5.so) ==18894== by 0xB1FC4C: dlopen_doit (in /lib/libdl-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0xB202CB: _dlerror_run (in /lib/libdl-2.5.so) ==18894== by 0xB1FB83: dlopen@@GLIBC_2.1 (in /lib/libdl-2.5.so) ==18894== ==18894== ==18894== 45 bytes in 1 blocks are still reachable in loss record 77 of 193 ==18894== at 0x40054BB: realloc (vg_replace_malloc.c:306) ==18894== by 0xC21BDF: _nc_doalloc (in /usr/lib/libncurses.so.5.5) ==18894== by 0xC29ADC: _nc_read_file_entry (in /usr/lib/libncurses.so.5.5) ==18894== by 0xC29FA4: _nc_read_entry (in /usr/lib/libncurses.so.5.5) ==18894== by 0xC23FDC: _nc_setupterm (in /usr/lib/libncurses.so.5.5) ==18894== by 0xC24747: tgetent (in /usr/lib/libncurses.so.5.5) ==18894== by 0x4141441: term_set (term.c:925) ==18894== by 0x4141843: term_init (term.c:350) ==18894== by 0x4139E84: el_init (el.c:84) ==18894== by 0x4049449: switch_console_loop (switch_console.c:747) ==18894== by 0x4061CE7: switch_core_runtime_loop (switch_core.c:687) ==18894== by 0x804A453: main (switch.c:715) ==18894== ==18894== ==18894== 48 bytes in 1 blocks are still reachable in loss record 78 of 193 ==18894== at 0x40046FF: calloc (vg_replace_malloc.c:279) ==18894== by 0xC29C23: _nc_read_file_entry (in /usr/lib/libncurses.so.5.5) ==18894== by 0xC29FA4: _nc_read_entry (in /usr/lib/libncurses.so.5.5) ==18894== by 0xC23FDC: _nc_setupterm (in /usr/lib/libncurses.so.5.5) ==18894== by 0xC24747: tgetent (in /usr/lib/libncurses.so.5.5) ==18894== by 0x4141441: term_set (term.c:925) ==18894== by 0x4141843: term_init (term.c:350) ==18894== by 0x4139E84: el_init (el.c:84) ==18894== by 0x4049449: switch_console_loop (switch_console.c:747) ==18894== by 0x4061CE7: switch_core_runtime_loop (switch_core.c:687) ==18894== by 0x804A453: main (switch.c:715) ==18894== ==18894== ==18894== 55 bytes in 2 blocks are still reachable in loss record 79 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x9BCC9F: open_path (in /lib/ld-2.5.so) ==18894== by 0x9BF7C4: _dl_map_object (in /lib/ld-2.5.so) ==18894== by 0x9C32D5: openaux (in /lib/ld-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0x9C3894: _dl_map_object_deps (in /lib/ld-2.5.so) ==18894== by 0x9C8A8C: dl_open_worker (in /lib/ld-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0x9C8411: _dl_open (in /lib/ld-2.5.so) ==18894== by 0xB1FC4C: dlopen_doit (in /lib/libdl-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0xB202CB: _dlerror_run (in /lib/libdl-2.5.so) ==18894== ==18894== ==18894== 56 bytes in 1 blocks are still reachable in loss record 80 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x127BB923: PR_Malloc (prmem.c:448) ==18894== by 0x127C3F77: PR_NewCondVar (ptsynch.c:341) ==18894== by 0x12867B4E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 56 bytes in 1 blocks are still reachable in loss record 81 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x127BB923: PR_Malloc (prmem.c:448) ==18894== by 0x127C3F77: PR_NewCondVar (ptsynch.c:341) ==18894== by 0x12867B36: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 56 bytes in 1 blocks are still reachable in loss record 82 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x127BB923: PR_Malloc (prmem.c:448) ==18894== by 0x127C3F77: PR_NewCondVar (ptsynch.c:341) ==18894== by 0x12867B15: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 56 bytes in 1 blocks are still reachable in loss record 83 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x127BB923: PR_Malloc (prmem.c:448) ==18894== by 0x127C3F77: PR_NewCondVar (ptsynch.c:341) ==18894== by 0x12867AC7: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 56 bytes in 1 blocks are still reachable in loss record 84 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x127BB923: PR_Malloc (prmem.c:448) ==18894== by 0x127C3F77: PR_NewCondVar (ptsynch.c:341) ==18894== by 0x12867AAF: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 56 bytes in 1 blocks are still reachable in loss record 85 of 193 ==18894== at 0x40046FF: calloc (vg_replace_malloc.c:279) ==18894== by 0x127BB985: PR_Calloc (prmem.c:455) ==18894== by 0x127C43FF: PR_NewMonitor (ptsynch.c:457) ==18894== by 0x127CDA2D: _PR_UnixInit (unix.c:2877) ==18894== by 0x127BE422: _PR_InitStuff (prinit.c:240) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== ==18894== ==18894== 56 bytes in 1 blocks are still reachable in loss record 86 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x127BB923: PR_Malloc (prmem.c:448) ==18894== by 0x127C3F77: PR_NewCondVar (ptsynch.c:341) ==18894== by 0x127CC47A: _PR_InitThreads (ptthread.c:876) ==18894== by 0x127BE404: _PR_InitStuff (prinit.c:215) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== ==18894== ==18894== 56 bytes in 1 blocks are still reachable in loss record 87 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x127BB923: PR_Malloc (prmem.c:448) ==18894== by 0x127C3F77: PR_NewCondVar (ptsynch.c:341) ==18894== by 0x127C64B6: _PR_InitIO (ptio.c:1104) ==18894== by 0x127BE40E: _PR_InitStuff (prinit.c:236) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== ==18894== ==18894== 56 bytes in 1 blocks are still reachable in loss record 88 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x127BB923: PR_Malloc (prmem.c:448) ==18894== by 0x127C3F77: PR_NewCondVar (ptsynch.c:341) ==18894== by 0x127BEAEE: _PR_InitCallOnce (prinit.c:688) ==18894== by 0x127BE418: _PR_InitStuff (prinit.c:238) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== ==18894== ==18894== 63 bytes in 2 blocks are still reachable in loss record 89 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x9BE714: expand_dynamic_string_token (in /lib/ld-2.5.so) ==18894== by 0x9BF3A9: _dl_map_object (in /lib/ld-2.5.so) ==18894== by 0x9C8A30: dl_open_worker (in /lib/ld-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0x9C8411: _dl_open (in /lib/ld-2.5.so) ==18894== by 0xB1FC4C: dlopen_doit (in /lib/libdl-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0xB202CB: _dlerror_run (in /lib/libdl-2.5.so) ==18894== by 0xB1FB83: dlopen@@GLIBC_2.1 (in /lib/libdl-2.5.so) ==18894== by 0x4067486: switch_dso_open (switch_dso.c:94) ==18894== by 0x406C96B: switch_loadable_module_load_module_ex (switch_loadable_module.c:831) ==18894== ==18894== ==18894== 64 bytes in 1 blocks are still reachable in loss record 90 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x22AC7D: (within /lib/libcrypto.so.0.9.8b) ==18894== by 0x22B2FE: CRYPTO_malloc (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x27E5F6: lh_new (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x281272: (within /lib/libcrypto.so.0.9.8b) ==18894== by 0x280EA0: (within /lib/libcrypto.so.0.9.8b) ==18894== by 0x28141B: ERR_get_state (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x281ADA: ERR_clear_error (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x27271E: ENGINE_load_dynamic (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x270BE6: ENGINE_load_builtin_engines (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x127FE136: Curl_ossl_init (in /usr/lib/libcurl.so.3.0.0) ==18894== by 0x1280F3AB: Curl_ssl_init (in /usr/lib/libcurl.so.3.0.0) ==18894== ==18894== ==18894== 64 bytes in 1 blocks are still reachable in loss record 91 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40DED94: sqlite3MallocX (sqliteInt.h:278) ==18894== by 0x40D3A1C: rehash (hash.c:227) ==18894== by 0x40D3D42: sqlite3HashInsert (hash.c:378) ==18894== by 0x40531EE: switch_core_hash_insert (switch_core_hash.c:67) ==18894== by 0x7AA4EF1: mod_cdr_csv_load (mod_cdr_csv.c:314) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 64 bytes in 2 blocks are still reachable in loss record 92 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x9C8E78: dl_open_worker (in /lib/ld-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0x9C8411: _dl_open (in /lib/ld-2.5.so) ==18894== by 0xB1FC4C: dlopen_doit (in /lib/libdl-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0xB202CB: _dlerror_run (in /lib/libdl-2.5.so) ==18894== by 0xB1FB83: dlopen@@GLIBC_2.1 (in /lib/libdl-2.5.so) ==18894== by 0x4067486: switch_dso_open (switch_dso.c:94) ==18894== by 0x1279F92E: mod_spidermonkey_load (mod_spidermonkey.c:894) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== ==18894== ==18894== 64 bytes in 1 blocks are still reachable in loss record 93 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40DED94: sqlite3MallocX (sqliteInt.h:278) ==18894== by 0x40D3A1C: rehash (hash.c:227) ==18894== by 0x40D3D42: sqlite3HashInsert (hash.c:378) ==18894== by 0x40531EE: switch_core_hash_insert (switch_core_hash.c:67) ==18894== by 0x400EC26: mod_logfile_load (mod_logfile.c:290) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 64 bytes in 1 blocks are still reachable in loss record 94 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40DED94: sqlite3MallocX (sqliteInt.h:278) ==18894== by 0x40D3A1C: rehash (hash.c:227) ==18894== by 0x40D3D42: sqlite3HashInsert (hash.c:378) ==18894== by 0x40531EE: switch_core_hash_insert (switch_core_hash.c:67) ==18894== by 0x4064511: switch_load_network_lists (switch_core.c:901) ==18894== by 0x40659E4: switch_core_init_and_modload (switch_core.c:1457) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 64 bytes in 1 blocks are still reachable in loss record 95 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x128A3CBB: js_SetupLocks (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867AE5: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 64 bytes in 1 blocks are still reachable in loss record 96 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40DED94: sqlite3MallocX (sqliteInt.h:278) ==18894== by 0x40D3A1C: rehash (hash.c:227) ==18894== by 0x40D3D42: sqlite3HashInsert (hash.c:378) ==18894== by 0x40531EE: switch_core_hash_insert (switch_core_hash.c:67) ==18894== by 0x7AA4812: ??? (mod_cdr_csv.c:140) ==18894== by 0x7AA4B55: ??? (mod_cdr_csv.c:233) ==18894== by 0x405A631: switch_core_session_reporting_state (switch_core_state_machine.c:607) ==18894== by 0x405AC1A: switch_core_session_run (switch_core_state_machine.c:409) ==18894== by 0x4058204: switch_core_session_thread (switch_core_session.c:1064) ==18894== by 0x40CD6C5: dummy_worker (thread.c:138) ==18894== by 0xB5346A: start_thread (in /lib/libpthread-2.5.so) ==18894== ==18894== ==18894== 78 bytes in 1 blocks are still reachable in loss record 97 of 193 ==18894== at 0x40054BB: realloc (vg_replace_malloc.c:306) ==18894== by 0xC21BDF: _nc_doalloc (in /usr/lib/libncurses.so.5.5) ==18894== by 0xC29AFA: _nc_read_file_entry (in /usr/lib/libncurses.so.5.5) ==18894== by 0xC29FA4: _nc_read_entry (in /usr/lib/libncurses.so.5.5) ==18894== by 0xC23FDC: _nc_setupterm (in /usr/lib/libncurses.so.5.5) ==18894== by 0xC24747: tgetent (in /usr/lib/libncurses.so.5.5) ==18894== by 0x4141441: term_set (term.c:925) ==18894== by 0x4141843: term_init (term.c:350) ==18894== by 0x4139E84: el_init (el.c:84) ==18894== by 0x4049449: switch_console_loop (switch_console.c:747) ==18894== by 0x4061CE7: switch_core_runtime_loop (switch_core.c:687) ==18894== by 0x804A453: main (switch.c:715) ==18894== ==18894== ==18894== 81 bytes in 3 blocks are definitely lost in loss record 98 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0xA48DEF: strdup (in /lib/libc-2.5.so) ==18894== by 0x11CBE451: ??? (mod_limit.c:538) ==18894== by 0x4058785: switch_core_session_exec (switch_core_session.c:1474) ==18894== by 0x4058CD8: switch_core_session_execute_application (switch_core_session.c:1396) ==18894== by 0x405C794: switch_core_session_run (switch_core_state_machine.c:166) ==18894== by 0x4058204: switch_core_session_thread (switch_core_session.c:1064) ==18894== by 0x40CD6C5: dummy_worker (thread.c:138) ==18894== by 0xB5346A: start_thread (in /lib/libpthread-2.5.so) ==18894== by 0xAAADBD: clone (in /lib/libc-2.5.so) ==18894== ==18894== ==18894== 83 bytes in 1 blocks are still reachable in loss record 99 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0xC21BFC: _nc_doalloc (in /usr/lib/libncurses.so.5.5) ==18894== by 0xC2522E: _nc_tparm_analyze (in /usr/lib/libncurses.so.5.5) ==18894== by 0xC25391: tparm (in /usr/lib/libncurses.so.5.5) ==18894== by 0xC2A526: (within /usr/lib/libncurses.so.5.5) ==18894== by 0xC2A82B: _nc_trim_sgr0 (in /usr/lib/libncurses.so.5.5) ==18894== by 0xC248BE: tgetent (in /usr/lib/libncurses.so.5.5) ==18894== by 0x4141441: term_set (term.c:925) ==18894== by 0x4141843: term_init (term.c:350) ==18894== by 0x4139E84: el_init (el.c:84) ==18894== by 0x4049449: switch_console_loop (switch_console.c:747) ==18894== by 0x4061CE7: switch_core_runtime_loop (switch_core.c:687) ==18894== ==18894== ==18894== 88 bytes in 1 blocks are still reachable in loss record 100 of 193 ==18894== at 0x40046FF: calloc (vg_replace_malloc.c:279) ==18894== by 0x127BB985: PR_Calloc (prmem.c:455) ==18894== by 0x127C37E3: PR_NewLock (ptsynch.c:174) ==18894== by 0x12867B24: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 88 bytes in 1 blocks are still reachable in loss record 101 of 193 ==18894== at 0x40046FF: calloc (vg_replace_malloc.c:279) ==18894== by 0x127BB985: PR_Calloc (prmem.c:455) ==18894== by 0x127C37E3: PR_NewLock (ptsynch.c:174) ==18894== by 0x12867A9D: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 88 bytes in 1 blocks are still reachable in loss record 102 of 193 ==18894== at 0x40046FF: calloc (vg_replace_malloc.c:279) ==18894== by 0x127BB985: PR_Calloc (prmem.c:455) ==18894== by 0x127C37E3: PR_NewLock (ptsynch.c:174) ==18894== by 0x12868DFC: JS_InitArenaPool (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12893089: js_InitGC (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A94: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 88 bytes in 1 blocks are still reachable in loss record 103 of 193 ==18894== at 0x40046FF: calloc (vg_replace_malloc.c:279) ==18894== by 0x127BB985: PR_Calloc (prmem.c:455) ==18894== by 0x127C37E3: PR_NewLock (ptsynch.c:174) ==18894== by 0x127BEAB0: _PR_InitCallOnce (prinit.c:686) ==18894== by 0x127BE418: _PR_InitStuff (prinit.c:238) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== ==18894== ==18894== 88 bytes in 1 blocks are still reachable in loss record 104 of 193 ==18894== at 0x40046FF: calloc (vg_replace_malloc.c:279) ==18894== by 0x127BB985: PR_Calloc (prmem.c:455) ==18894== by 0x127C37E3: PR_NewLock (ptsynch.c:174) ==18894== by 0x12867AFD: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 88 bytes in 1 blocks are still reachable in loss record 105 of 193 ==18894== at 0x40046FF: calloc (vg_replace_malloc.c:279) ==18894== by 0x127BB985: PR_Calloc (prmem.c:455) ==18894== by 0x127C37E3: PR_NewLock (ptsynch.c:174) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 88 bytes in 1 blocks are still reachable in loss record 106 of 193 ==18894== at 0x40046FF: calloc (vg_replace_malloc.c:279) ==18894== by 0x127BB985: PR_Calloc (prmem.c:455) ==18894== by 0x127C37E3: PR_NewLock (ptsynch.c:174) ==18894== by 0x127CD9F8: _PR_UnixInit (unix.c:2875) ==18894== by 0x127BE422: _PR_InitStuff (prinit.c:240) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== ==18894== ==18894== 88 bytes in 1 blocks are still reachable in loss record 107 of 193 ==18894== at 0x40046FF: calloc (vg_replace_malloc.c:279) ==18894== by 0x127BB985: PR_Calloc (prmem.c:455) ==18894== by 0x127C37E3: PR_NewLock (ptsynch.c:174) ==18894== by 0x127B2BCE: _PR_InitMW (prmwait.c:241) ==18894== by 0x127BE41D: _PR_InitStuff (prinit.c:239) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== ==18894== ==18894== 88 bytes in 1 blocks are still reachable in loss record 108 of 193 ==18894== at 0x40046FF: calloc (vg_replace_malloc.c:279) ==18894== by 0x127BB985: PR_Calloc (prmem.c:455) ==18894== by 0x127C37E3: PR_NewLock (ptsynch.c:174) ==18894== by 0x127B73D5: _PR_InitLog (prlog.c:209) ==18894== by 0x127BE413: _PR_InitStuff (prinit.c:237) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== ==18894== ==18894== 88 bytes in 1 blocks are still reachable in loss record 109 of 193 ==18894== at 0x40046FF: calloc (vg_replace_malloc.c:279) ==18894== by 0x127BB985: PR_Calloc (prmem.c:455) ==18894== by 0x127C37E3: PR_NewLock (ptsynch.c:174) ==18894== by 0x127BDD50: PR_CreateStack (pratom.c:396) ==18894== by 0x127B2AAB: _PR_InitFdCache (prfdcach.c:285) ==18894== by 0x127C6520: _PR_InitIO (ptio.c:1109) ==18894== by 0x127BE40E: _PR_InitStuff (prinit.c:236) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== ==18894== ==18894== 88 bytes in 1 blocks are still reachable in loss record 110 of 193 ==18894== at 0x40046FF: calloc (vg_replace_malloc.c:279) ==18894== by 0x127BB985: PR_Calloc (prmem.c:455) ==18894== by 0x127C37E3: PR_NewLock (ptsynch.c:174) ==18894== by 0x127CC43C: _PR_InitThreads (ptthread.c:874) ==18894== by 0x127BE404: _PR_InitStuff (prinit.c:215) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== ==18894== ==18894== 88 bytes in 1 blocks are still reachable in loss record 111 of 193 ==18894== at 0x40046FF: calloc (vg_replace_malloc.c:279) ==18894== by 0x127BB985: PR_Calloc (prmem.c:455) ==18894== by 0x127C37E3: PR_NewLock (ptsynch.c:174) ==18894== by 0x127BE3B2: _PR_InitStuff (prinit.c:208) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== ==18894== ==18894== 88 bytes in 1 blocks are still reachable in loss record 112 of 193 ==18894== at 0x40046FF: calloc (vg_replace_malloc.c:279) ==18894== by 0x127BB985: PR_Calloc (prmem.c:455) ==18894== by 0x127C37E3: PR_NewLock (ptsynch.c:174) ==18894== by 0x127B72DB: _PR_InitLayerCache (prlayer.c:745) ==18894== by 0x127BE3A8: _PR_InitStuff (prinit.c:205) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== ==18894== ==18894== 88 bytes in 1 blocks are still reachable in loss record 113 of 193 ==18894== at 0x40046FF: calloc (vg_replace_malloc.c:279) ==18894== by 0x127BB985: PR_Calloc (prmem.c:455) ==18894== by 0x127C37E3: PR_NewLock (ptsynch.c:174) ==18894== by 0x127C64EB: _PR_InitIO (ptio.c:1106) ==18894== by 0x127BE40E: _PR_InitStuff (prinit.c:236) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== ==18894== ==18894== 88 bytes in 1 blocks are still reachable in loss record 114 of 193 ==18894== at 0x40046FF: calloc (vg_replace_malloc.c:279) ==18894== by 0x127BB985: PR_Calloc (prmem.c:455) ==18894== by 0x127C37E3: PR_NewLock (ptsynch.c:174) ==18894== by 0x127C6478: _PR_InitIO (ptio.c:1102) ==18894== by 0x127BE40E: _PR_InitStuff (prinit.c:236) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== ==18894== ==18894== 88 bytes in 1 blocks are still reachable in loss record 115 of 193 ==18894== at 0x40046FF: calloc (vg_replace_malloc.c:279) ==18894== by 0x127BB985: PR_Calloc (prmem.c:455) ==18894== by 0x127C37E3: PR_NewLock (ptsynch.c:174) ==18894== by 0x127B2A6D: _PR_InitFdCache (prfdcach.c:283) ==18894== by 0x127C6520: _PR_InitIO (ptio.c:1109) ==18894== by 0x127BE40E: _PR_InitStuff (prinit.c:236) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== ==18894== ==18894== 96 bytes in 1 blocks are still reachable in loss record 116 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x22AC7D: (within /lib/libcrypto.so.0.9.8b) ==18894== by 0x22B2FE: CRYPTO_malloc (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x27E5C7: lh_new (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x281272: (within /lib/libcrypto.so.0.9.8b) ==18894== by 0x280EA0: (within /lib/libcrypto.so.0.9.8b) ==18894== by 0x28141B: ERR_get_state (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x281ADA: ERR_clear_error (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x27271E: ENGINE_load_dynamic (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x270BE6: ENGINE_load_builtin_engines (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x127FE136: Curl_ossl_init (in /usr/lib/libcurl.so.3.0.0) ==18894== by 0x1280F3AB: Curl_ssl_init (in /usr/lib/libcurl.so.3.0.0) ==18894== ==18894== ==18894== 99 bytes in 4 blocks are still reachable in loss record 117 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x9C1937: _dl_new_object (in /lib/ld-2.5.so) ==18894== by 0x9BD020: _dl_map_object_from_fd (in /lib/ld-2.5.so) ==18894== by 0x9BF46B: _dl_map_object (in /lib/ld-2.5.so) ==18894== by 0x9C32D5: openaux (in /lib/ld-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0x9C3894: _dl_map_object_deps (in /lib/ld-2.5.so) ==18894== by 0x9C8A8C: dl_open_worker (in /lib/ld-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0x9C8411: _dl_open (in /lib/ld-2.5.so) ==18894== by 0xB1FC4C: dlopen_doit (in /lib/libdl-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== ==18894== ==18894== 100 bytes in 1 blocks are still reachable in loss record 118 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x22AC7D: (within /lib/libcrypto.so.0.9.8b) ==18894== by 0x22B2FE: CRYPTO_malloc (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x26E6F0: ENGINE_new (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x2737DB: ENGINE_load_padlock (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x270BEB: ENGINE_load_builtin_engines (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x127FE136: Curl_ossl_init (in /usr/lib/libcurl.so.3.0.0) ==18894== by 0x1280F3AB: Curl_ssl_init (in /usr/lib/libcurl.so.3.0.0) ==18894== by 0x12807162: curl_global_init (in /usr/lib/libcurl.so.3.0.0) ==18894== by 0x1279FAB3: mod_spidermonkey_load (mod_spidermonkey.c:3751) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== ==18894== ==18894== 100 bytes in 1 blocks are still reachable in loss record 119 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x22AC7D: (within /lib/libcrypto.so.0.9.8b) ==18894== by 0x22B2FE: CRYPTO_malloc (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x26E6F0: ENGINE_new (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x27264B: ENGINE_load_dynamic (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x270BE6: ENGINE_load_builtin_engines (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x127FE136: Curl_ossl_init (in /usr/lib/libcurl.so.3.0.0) ==18894== by 0x1280F3AB: Curl_ssl_init (in /usr/lib/libcurl.so.3.0.0) ==18894== by 0x12807162: curl_global_init (in /usr/lib/libcurl.so.3.0.0) ==18894== by 0x1279FAB3: mod_spidermonkey_load (mod_spidermonkey.c:3751) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== ==18894== ==18894== 100 bytes in 5 blocks are still reachable in loss record 120 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40DED94: sqlite3MallocX (sqliteInt.h:278) ==18894== by 0x40D3CD2: sqlite3HashInsert (hash.c:363) ==18894== by 0x40531EE: switch_core_hash_insert (switch_core_hash.c:67) ==18894== by 0x7AA518D: mod_cdr_csv_load (mod_cdr_csv.c:361) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 104 bytes in 1 blocks are still reachable in loss record 121 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40C7FFE: apr_allocator_create (apr_pools.c:91) ==18894== by 0x40502BF: switch_core_perform_new_memory_pool (switch_core_memory.c:353) ==18894== by 0x8036763: mod_loopback_load (mod_loopback.c:871) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 104 bytes in 1 blocks are still reachable in loss record 122 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40C7FFE: apr_allocator_create (apr_pools.c:91) ==18894== by 0x40502BF: switch_core_perform_new_memory_pool (switch_core_memory.c:353) ==18894== by 0x1279F75A: mod_spidermonkey_load (mod_spidermonkey.c:1007) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 104 bytes in 1 blocks are still reachable in loss record 123 of 193 ==18894== at 0x40046FF: calloc (vg_replace_malloc.c:279) ==18894== by 0x127BB985: PR_Calloc (prmem.c:455) ==18894== by 0x127C443C: PR_NewMonitor (ptsynch.c:463) ==18894== by 0x127CDA2D: _PR_UnixInit (unix.c:2877) ==18894== by 0x127BE422: _PR_InitStuff (prinit.c:240) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== ==18894== ==18894== 104 bytes in 1 blocks are still reachable in loss record 124 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40C7FFE: apr_allocator_create (apr_pools.c:91) ==18894== by 0x40502BF: switch_core_perform_new_memory_pool (switch_core_memory.c:353) ==18894== by 0x40643E4: switch_load_network_lists (switch_core.c:891) ==18894== by 0x40659E4: switch_core_init_and_modload (switch_core.c:1457) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 104 bytes in 1 blocks are still reachable in loss record 125 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40C7FFE: apr_allocator_create (apr_pools.c:91) ==18894== by 0x40502BF: switch_core_perform_new_memory_pool (switch_core_memory.c:353) ==18894== by 0x406C7CD: switch_loadable_module_load_module_ex (switch_loadable_module.c:785) ==18894== by 0x406D097: switch_loadable_module_init (switch_loadable_module.c:1160) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 104 bytes in 1 blocks are still reachable in loss record 126 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40C7FFE: apr_allocator_create (apr_pools.c:91) ==18894== by 0x40502BF: switch_core_perform_new_memory_pool (switch_core_memory.c:353) ==18894== by 0x11D2FDC6: mod_sndfile_load (mod_sndfile.c:400) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 104 bytes in 1 blocks are still reachable in loss record 127 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40C7FFE: apr_allocator_create (apr_pools.c:91) ==18894== by 0x40502BF: switch_core_perform_new_memory_pool (switch_core_memory.c:353) ==18894== by 0x406C7CD: switch_loadable_module_load_module_ex (switch_loadable_module.c:785) ==18894== by 0x406D073: switch_loadable_module_init (switch_loadable_module.c:1159) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 104 bytes in 1 blocks are still reachable in loss record 128 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40C7FFE: apr_allocator_create (apr_pools.c:91) ==18894== by 0x40502BF: switch_core_perform_new_memory_pool (switch_core_memory.c:353) ==18894== by 0x406CE99: switch_loadable_module_init (switch_loadable_module.c:1136) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 104 bytes in 1 blocks are still reachable in loss record 129 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40C7FFE: apr_allocator_create (apr_pools.c:91) ==18894== by 0x40502BF: switch_core_perform_new_memory_pool (switch_core_memory.c:353) ==18894== by 0x4065C3C: switch_scheduler_task_thread_start (switch_scheduler.c:295) ==18894== by 0x40639C5: switch_core_init (switch_core.c:1250) ==18894== by 0x406585C: switch_core_init_and_modload (switch_core.c:1422) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 104 bytes in 1 blocks are still reachable in loss record 130 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40C7FFE: apr_allocator_create (apr_pools.c:91) ==18894== by 0x40C88ED: apr_pool_initialize (apr_pools.c:522) ==18894== by 0x40C971E: apr_initialize (start.c:55) ==18894== by 0x8049C7B: main (switch.c:596) ==18894== ==18894== ==18894== 128 bytes in 1 blocks are still reachable in loss record 131 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0xC29E6D: _nc_read_file_entry (in /usr/lib/libncurses.so.5.5) ==18894== by 0xC29FA4: _nc_read_entry (in /usr/lib/libncurses.so.5.5) ==18894== by 0xC23FDC: _nc_setupterm (in /usr/lib/libncurses.so.5.5) ==18894== by 0xC24747: tgetent (in /usr/lib/libncurses.so.5.5) ==18894== by 0x4141441: term_set (term.c:925) ==18894== by 0x4141843: term_init (term.c:350) ==18894== by 0x4139E84: el_init (el.c:84) ==18894== by 0x4049449: switch_console_loop (switch_console.c:747) ==18894== by 0x4061CE7: switch_core_runtime_loop (switch_core.c:687) ==18894== by 0x804A453: main (switch.c:715) ==18894== ==18894== ==18894== 128 bytes in 1 blocks are still reachable in loss record 132 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x1287870C: JS_DHashAllocTable (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12878532: JS_DHashTableInit (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x128CA989: js_InitPropertyTree (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867B74: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 128 bytes in 1 blocks are still reachable in loss record 133 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x128A3D24: js_SetupLocks (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867AE5: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 132 bytes in 3 blocks are still reachable in loss record 134 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x9BE714: expand_dynamic_string_token (in /lib/ld-2.5.so) ==18894== by 0x9BF3A9: _dl_map_object (in /lib/ld-2.5.so) ==18894== by 0x9C8A30: dl_open_worker (in /lib/ld-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0x9C8411: _dl_open (in /lib/ld-2.5.so) ==18894== by 0xB1FC4C: dlopen_doit (in /lib/libdl-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0xB202CB: _dlerror_run (in /lib/libdl-2.5.so) ==18894== by 0xB1FB83: dlopen@@GLIBC_2.1 (in /lib/libdl-2.5.so) ==18894== by 0x4067486: switch_dso_open (switch_dso.c:94) ==18894== by 0x1279F92E: mod_spidermonkey_load (mod_spidermonkey.c:894) ==18894== ==18894== ==18894== 160 bytes in 1 blocks are possibly lost in loss record 135 of 193 ==18894== at 0x40046FF: calloc (vg_replace_malloc.c:279) ==18894== by 0x9C7CC9: _dl_allocate_tls (in /lib/ld-2.5.so) ==18894== by 0xB53B59: pthread_create@@GLIBC_2.1 (in /lib/libpthread-2.5.so) ==18894== by 0xB541D7: pthread_create at GLIBC_2.0 (in /lib/libpthread-2.5.so) ==18894== by 0x40CDA64: apr_thread_create (thread.c:176) ==18894== by 0x403ECF8: switch_thread_create (switch_apr.c:631) ==18894== by 0x4062738: switch_core_launch_thread (switch_core.c:400) ==18894== by 0x406D4D6: switch_loadable_module_init (switch_loadable_module.c:121) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 160 bytes in 1 blocks are still reachable in loss record 136 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x9C8881: add_to_global (in /lib/ld-2.5.so) ==18894== by 0x9C8D75: dl_open_worker (in /lib/ld-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0x9C8411: _dl_open (in /lib/ld-2.5.so) ==18894== by 0xB1FC4C: dlopen_doit (in /lib/libdl-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0xB202CB: _dlerror_run (in /lib/libdl-2.5.so) ==18894== by 0xB1FB83: dlopen@@GLIBC_2.1 (in /lib/libdl-2.5.so) ==18894== by 0x4067486: switch_dso_open (switch_dso.c:94) ==18894== by 0x406C96B: switch_loadable_module_load_module_ex (switch_loadable_module.c:831) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== ==18894== ==18894== 160 bytes in 1 blocks are possibly lost in loss record 137 of 193 ==18894== at 0x40046FF: calloc (vg_replace_malloc.c:279) ==18894== by 0x9C7CC9: _dl_allocate_tls (in /lib/ld-2.5.so) ==18894== by 0xB53B59: pthread_create@@GLIBC_2.1 (in /lib/libpthread-2.5.so) ==18894== by 0xB541D7: pthread_create at GLIBC_2.0 (in /lib/libpthread-2.5.so) ==18894== by 0x40CDA64: apr_thread_create (thread.c:176) ==18894== by 0x403ECF8: switch_thread_create (switch_apr.c:631) ==18894== by 0x11D90A0E: mod_local_stream_load (mod_local_stream.c:507) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 172 bytes in 1 blocks are still reachable in loss record 138 of 193 ==18894== at 0x40046FF: calloc (vg_replace_malloc.c:279) ==18894== by 0xC23C86: _nc_setupterm (in /usr/lib/libncurses.so.5.5) ==18894== by 0xC24747: tgetent (in /usr/lib/libncurses.so.5.5) ==18894== by 0x4141441: term_set (term.c:925) ==18894== by 0x4141843: term_init (term.c:350) ==18894== by 0x4139E84: el_init (el.c:84) ==18894== by 0x4049449: switch_console_loop (switch_console.c:747) ==18894== by 0x4061CE7: switch_core_runtime_loop (switch_core.c:687) ==18894== by 0x804A453: main (switch.c:715) ==18894== ==18894== ==18894== 172 bytes in 1 blocks are still reachable in loss record 139 of 193 ==18894== at 0x40046FF: calloc (vg_replace_malloc.c:279) ==18894== by 0x127BB985: PR_Calloc (prmem.c:455) ==18894== by 0x127CC4BE: _PR_InitThreads (ptthread.c:878) ==18894== by 0x127BE404: _PR_InitStuff (prinit.c:215) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== ==18894== ==18894== 176 bytes in 4 blocks are still reachable in loss record 140 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x9C3905: _dl_map_object_deps (in /lib/ld-2.5.so) ==18894== by 0x9C8A8C: dl_open_worker (in /lib/ld-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0x9C8411: _dl_open (in /lib/ld-2.5.so) ==18894== by 0xB1FC4C: dlopen_doit (in /lib/libdl-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0xB202CB: _dlerror_run (in /lib/libdl-2.5.so) ==18894== by 0xB1FB83: dlopen@@GLIBC_2.1 (in /lib/libdl-2.5.so) ==18894== by 0x4067486: switch_dso_open (switch_dso.c:94) ==18894== by 0x406C96B: switch_loadable_module_load_module_ex (switch_loadable_module.c:831) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== ==18894== ==18894== 192 bytes in 3 blocks are still reachable in loss record 141 of 193 ==18894== at 0x40046FF: calloc (vg_replace_malloc.c:279) ==18894== by 0x9C6573: _dl_check_map_versions (in /lib/ld-2.5.so) ==18894== by 0x9C8D68: dl_open_worker (in /lib/ld-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0x9C8411: _dl_open (in /lib/ld-2.5.so) ==18894== by 0xB1FC4C: dlopen_doit (in /lib/libdl-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0xB202CB: _dlerror_run (in /lib/libdl-2.5.so) ==18894== by 0xB1FB83: dlopen@@GLIBC_2.1 (in /lib/libdl-2.5.so) ==18894== by 0x4067486: switch_dso_open (switch_dso.c:94) ==18894== by 0x1279F92E: mod_spidermonkey_load (mod_spidermonkey.c:894) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== ==18894== ==18894== 195 bytes in 5 blocks are still reachable in loss record 142 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x9C1937: _dl_new_object (in /lib/ld-2.5.so) ==18894== by 0x9BD020: _dl_map_object_from_fd (in /lib/ld-2.5.so) ==18894== by 0x9BF46B: _dl_map_object (in /lib/ld-2.5.so) ==18894== by 0x9C8A30: dl_open_worker (in /lib/ld-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0x9C8411: _dl_open (in /lib/ld-2.5.so) ==18894== by 0xB1FC4C: dlopen_doit (in /lib/libdl-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0xB202CB: _dlerror_run (in /lib/libdl-2.5.so) ==18894== by 0xB1FB83: dlopen@@GLIBC_2.1 (in /lib/libdl-2.5.so) ==18894== by 0x4067486: switch_dso_open (switch_dso.c:94) ==18894== ==18894== ==18894== 220 bytes in 1 blocks are definitely lost in loss record 143 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x412F754: crypto_alloc (alloc.c:97) ==18894== by 0x41320C7: aes_cbc_alloc (aes_cbc.c:71) ==18894== by 0x412B53C: cipher_type_self_test (cipher.c:264) ==18894== by 0x412F0B5: crypto_kernel_load_cipher_type (crypto_kernel.c:310) ==18894== by 0x412F668: crypto_kernel_init (crypto_kernel.c:157) ==18894== by 0x4129026: srtp_init (srtp.c:1081) ==18894== by 0x407AF62: switch_rtp_init (switch_rtp.c:611) ==18894== by 0x40639D6: switch_core_init (switch_core.c:1252) ==18894== by 0x406585C: switch_core_init_and_modload (switch_core.c:1422) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 240 bytes in 1 blocks are definitely lost in loss record 144 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x412F754: crypto_alloc (alloc.c:97) ==18894== by 0x412CF0B: aes_icm_alloc_ismacryp (aes_icm.c:115) ==18894== by 0x412CF9B: aes_icm_alloc (aes_icm.c:134) ==18894== by 0x412B53C: cipher_type_self_test (cipher.c:264) ==18894== by 0x412F0B5: crypto_kernel_load_cipher_type (crypto_kernel.c:310) ==18894== by 0x412F64A: crypto_kernel_init (crypto_kernel.c:154) ==18894== by 0x4129026: srtp_init (srtp.c:1081) ==18894== by 0x407AF62: switch_rtp_init (switch_rtp.c:611) ==18894== by 0x40639D6: switch_core_init (switch_core.c:1252) ==18894== by 0x406585C: switch_core_init_and_modload (switch_core.c:1422) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 396 bytes in 1 blocks are still reachable in loss record 145 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x22AC7D: (within /lib/libcrypto.so.0.9.8b) ==18894== by 0x22B2FE: CRYPTO_malloc (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x2813CD: ERR_get_state (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x281ADA: ERR_clear_error (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x27271E: ENGINE_load_dynamic (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x270BE6: ENGINE_load_builtin_engines (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x127FE136: Curl_ossl_init (in /usr/lib/libcurl.so.3.0.0) ==18894== by 0x1280F3AB: Curl_ssl_init (in /usr/lib/libcurl.so.3.0.0) ==18894== by 0x12807162: curl_global_init (in /usr/lib/libcurl.so.3.0.0) ==18894== by 0x1279FAB3: mod_spidermonkey_load (mod_spidermonkey.c:3751) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== ==18894== ==18894== 440 bytes in 2 blocks are still reachable in loss record 146 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x9C3B6B: _dl_map_object_deps (in /lib/ld-2.5.so) ==18894== by 0x9C8A8C: dl_open_worker (in /lib/ld-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0x9C8411: _dl_open (in /lib/ld-2.5.so) ==18894== by 0xB1FC4C: dlopen_doit (in /lib/libdl-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0xB202CB: _dlerror_run (in /lib/libdl-2.5.so) ==18894== by 0xB1FB83: dlopen@@GLIBC_2.1 (in /lib/libdl-2.5.so) ==18894== by 0x4067486: switch_dso_open (switch_dso.c:94) ==18894== by 0x406C96B: switch_loadable_module_load_module_ex (switch_loadable_module.c:831) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== ==18894== ==18894== 512 bytes in 1 blocks are still reachable in loss record 147 of 193 ==18894== at 0x40046FF: calloc (vg_replace_malloc.c:279) ==18894== by 0x127BB985: PR_Calloc (prmem.c:455) ==18894== by 0x127BA8F9: _PR_InitTPD (prtpd.c:96) ==18894== by 0x127BE3A3: _PR_InitStuff (prinit.c:204) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== ==18894== ==18894== 572 bytes in 1 blocks are definitely lost in loss record 148 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x7AFDE93: sofia_glue_do_invite (sofia_glue.c:1599) ==18894== by 0x7ACB191: sofia_on_init (mod_sofia.c:102) ==18894== by 0x405B124: switch_core_session_run (switch_core_state_machine.c:480) ==18894== by 0x4058204: switch_core_session_thread (switch_core_session.c:1064) ==18894== by 0x40CD6C5: dummy_worker (thread.c:138) ==18894== by 0xB5346A: start_thread (in /lib/libpthread-2.5.so) ==18894== by 0xAAADBD: clone (in /lib/libc-2.5.so) ==18894== ==18894== ==18894== 660 bytes in 3 blocks are still reachable in loss record 149 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x9C3B6B: _dl_map_object_deps (in /lib/ld-2.5.so) ==18894== by 0x9C8A8C: dl_open_worker (in /lib/ld-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0x9C8411: _dl_open (in /lib/ld-2.5.so) ==18894== by 0xB1FC4C: dlopen_doit (in /lib/libdl-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0xB202CB: _dlerror_run (in /lib/libdl-2.5.so) ==18894== by 0xB1FB83: dlopen@@GLIBC_2.1 (in /lib/libdl-2.5.so) ==18894== by 0x4067486: switch_dso_open (switch_dso.c:94) ==18894== by 0x1279F92E: mod_spidermonkey_load (mod_spidermonkey.c:894) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== ==18894== ==18894== 944 bytes in 6 blocks are still reachable in loss record 150 of 193 ==18894== at 0x40046FF: calloc (vg_replace_malloc.c:279) ==18894== by 0x9C6573: _dl_check_map_versions (in /lib/ld-2.5.so) ==18894== by 0x9C8D68: dl_open_worker (in /lib/ld-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0x9C8411: _dl_open (in /lib/ld-2.5.so) ==18894== by 0xB1FC4C: dlopen_doit (in /lib/libdl-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0xB202CB: _dlerror_run (in /lib/libdl-2.5.so) ==18894== by 0xB1FB83: dlopen@@GLIBC_2.1 (in /lib/libdl-2.5.so) ==18894== by 0x4067486: switch_dso_open (switch_dso.c:94) ==18894== by 0x406C96B: switch_loadable_module_load_module_ex (switch_loadable_module.c:831) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== ==18894== ==18894== 1,024 bytes in 1 blocks are still reachable in loss record 151 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x4062333: switch_core_set_globals (switch_core.c:449) ==18894== by 0x804A06D: main (switch.c:667) ==18894== ==18894== ==18894== 1,061 bytes in 34 blocks are still reachable in loss record 152 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x9BE714: expand_dynamic_string_token (in /lib/ld-2.5.so) ==18894== by 0x9BF3A9: _dl_map_object (in /lib/ld-2.5.so) ==18894== by 0x9C8A30: dl_open_worker (in /lib/ld-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0x9C8411: _dl_open (in /lib/ld-2.5.so) ==18894== by 0xB1FC4C: dlopen_doit (in /lib/libdl-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0xB202CB: _dlerror_run (in /lib/libdl-2.5.so) ==18894== by 0xB1FB83: dlopen@@GLIBC_2.1 (in /lib/libdl-2.5.so) ==18894== by 0x40674AF: switch_dso_open (switch_dso.c:96) ==18894== by 0x406CC16: switch_loadable_module_load_module_ex (switch_loadable_module.c:804) ==18894== ==18894== ==18894== 1,061 bytes in 34 blocks are still reachable in loss record 153 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x9C1937: _dl_new_object (in /lib/ld-2.5.so) ==18894== by 0x9BD020: _dl_map_object_from_fd (in /lib/ld-2.5.so) ==18894== by 0x9BF46B: _dl_map_object (in /lib/ld-2.5.so) ==18894== by 0x9C8A30: dl_open_worker (in /lib/ld-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0x9C8411: _dl_open (in /lib/ld-2.5.so) ==18894== by 0xB1FC4C: dlopen_doit (in /lib/libdl-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0xB202CB: _dlerror_run (in /lib/libdl-2.5.so) ==18894== by 0xB1FB83: dlopen@@GLIBC_2.1 (in /lib/libdl-2.5.so) ==18894== by 0x40674AF: switch_dso_open (switch_dso.c:96) ==18894== ==18894== ==18894== 1,344 bytes in 1 blocks are still reachable in loss record 154 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x7B17A2E: msg_mclass_clone (msg_mclass.c:112) ==18894== by 0x7B94268: sip_extend_mclass (sip_parser.c:132) ==18894== by 0x7AE4887: config_sofia (sofia.c:2007) ==18894== by 0x7AC85C9: mod_sofia_load (mod_sofia.c:3264) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 1,381 bytes in 1 blocks are still reachable in loss record 155 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0xC29666: _nc_read_file_entry (in /usr/lib/libncurses.so.5.5) ==18894== by 0xC29FA4: _nc_read_entry (in /usr/lib/libncurses.so.5.5) ==18894== by 0xC23FDC: _nc_setupterm (in /usr/lib/libncurses.so.5.5) ==18894== by 0xC24747: tgetent (in /usr/lib/libncurses.so.5.5) ==18894== by 0x4141441: term_set (term.c:925) ==18894== by 0x4141843: term_init (term.c:350) ==18894== by 0x4139E84: el_init (el.c:84) ==18894== by 0x4049449: switch_console_loop (switch_console.c:747) ==18894== by 0x4061CE7: switch_core_runtime_loop (switch_core.c:687) ==18894== by 0x804A453: main (switch.c:715) ==18894== ==18894== ==18894== 1,408 bytes in 16 blocks are still reachable in loss record 156 of 193 ==18894== at 0x40046FF: calloc (vg_replace_malloc.c:279) ==18894== by 0x127BB985: PR_Calloc (prmem.c:455) ==18894== by 0x127C37E3: PR_NewLock (ptsynch.c:174) ==18894== by 0x128A3CF4: js_SetupLocks (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867AE5: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 1,600 (44 direct, 1,556 indirect) bytes in 1 blocks are definitely lost in loss record 157 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40724DA: switch_event_create_subclass_detailed (switch_event.c:628) ==18894== by 0x40A902E: switch_event_import_xml (switch_xml_config.c:55) ==18894== by 0x11CAC4EE: ??? (mod_voicemail.c:619) ==18894== by 0x11CAD21F: mod_voicemail_load (mod_voicemail.c:821) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 64 bytes in 4 blocks are indirectly lost in loss record 158 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40720BF: switch_event_base_add_header (switch_event.c:737) ==18894== by 0x40721A5: switch_event_add_header_string (switch_event.c:788) ==18894== by 0x11CAC625: ??? (mod_voicemail.c:637) ==18894== by 0x11CAD21F: mod_voicemail_load (mod_voicemail.c:821) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 79 bytes in 4 blocks are indirectly lost in loss record 159 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x407176E: my_dup (switch_event.c:95) ==18894== by 0x40720E8: switch_event_base_add_header (switch_event.c:745) ==18894== by 0x40721A5: switch_event_add_header_string (switch_event.c:788) ==18894== by 0x11CAC625: ??? (mod_voicemail.c:637) ==18894== by 0x11CAD21F: mod_voicemail_load (mod_voicemail.c:821) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 99 bytes in 4 blocks are indirectly lost in loss record 160 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x407176E: my_dup (switch_event.c:95) ==18894== by 0x4072188: switch_event_add_header_string (switch_event.c:788) ==18894== by 0x11CAC625: ??? (mod_voicemail.c:637) ==18894== by 0x11CAD21F: mod_voicemail_load (mod_voicemail.c:821) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 150 bytes in 37 blocks are indirectly lost in loss record 161 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x407176E: my_dup (switch_event.c:95) ==18894== by 0x4072188: switch_event_add_header_string (switch_event.c:788) ==18894== by 0x40A8FE4: switch_event_import_xml (switch_xml_config.c:63) ==18894== by 0x11CAC4EE: ??? (mod_voicemail.c:619) ==18894== by 0x11CAD21F: mod_voicemail_load (mod_voicemail.c:821) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 572 bytes in 37 blocks are indirectly lost in loss record 162 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x407176E: my_dup (switch_event.c:95) ==18894== by 0x40720E8: switch_event_base_add_header (switch_event.c:745) ==18894== by 0x40721A5: switch_event_add_header_string (switch_event.c:788) ==18894== by 0x40A8FE4: switch_event_import_xml (switch_xml_config.c:63) ==18894== by 0x11CAC4EE: ??? (mod_voicemail.c:619) ==18894== by 0x11CAD21F: mod_voicemail_load (mod_voicemail.c:821) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 592 bytes in 37 blocks are indirectly lost in loss record 163 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40720BF: switch_event_base_add_header (switch_event.c:737) ==18894== by 0x40721A5: switch_event_add_header_string (switch_event.c:788) ==18894== by 0x40A8FE4: switch_event_import_xml (switch_xml_config.c:63) ==18894== by 0x11CAC4EE: ??? (mod_voicemail.c:619) ==18894== by 0x11CAD21F: mod_voicemail_load (mod_voicemail.c:821) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 1,700 bytes in 1 blocks are still reachable in loss record 164 of 193 ==18894== at 0x40054BB: realloc (vg_replace_malloc.c:306) ==18894== by 0xC21BDF: _nc_doalloc (in /usr/lib/libncurses.so.5.5) ==18894== by 0xC29B19: _nc_read_file_entry (in /usr/lib/libncurses.so.5.5) ==18894== by 0xC29FA4: _nc_read_entry (in /usr/lib/libncurses.so.5.5) ==18894== by 0xC23FDC: _nc_setupterm (in /usr/lib/libncurses.so.5.5) ==18894== by 0xC24747: tgetent (in /usr/lib/libncurses.so.5.5) ==18894== by 0x4141441: term_set (term.c:925) ==18894== by 0x4141843: term_init (term.c:350) ==18894== by 0x4139E84: el_init (el.c:84) ==18894== by 0x4049449: switch_console_loop (switch_console.c:747) ==18894== by 0x4061CE7: switch_core_runtime_loop (switch_core.c:687) ==18894== by 0x804A453: main (switch.c:715) ==18894== ==18894== ==18894== 2,481 bytes in 4 blocks are still reachable in loss record 165 of 193 ==18894== at 0x40046FF: calloc (vg_replace_malloc.c:279) ==18894== by 0x9C16CA: _dl_new_object (in /lib/ld-2.5.so) ==18894== by 0x9BD020: _dl_map_object_from_fd (in /lib/ld-2.5.so) ==18894== by 0x9BF46B: _dl_map_object (in /lib/ld-2.5.so) ==18894== by 0x9C32D5: openaux (in /lib/ld-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0x9C3894: _dl_map_object_deps (in /lib/ld-2.5.so) ==18894== by 0x9C8A8C: dl_open_worker (in /lib/ld-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0x9C8411: _dl_open (in /lib/ld-2.5.so) ==18894== by 0xB1FC4C: dlopen_doit (in /lib/libdl-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== ==18894== ==18894== 2,544 bytes in 34 blocks are still reachable in loss record 166 of 193 ==18894== at 0x40046FF: calloc (vg_replace_malloc.c:279) ==18894== by 0x9C6573: _dl_check_map_versions (in /lib/ld-2.5.so) ==18894== by 0x9C8D68: dl_open_worker (in /lib/ld-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0x9C8411: _dl_open (in /lib/ld-2.5.so) ==18894== by 0xB1FC4C: dlopen_doit (in /lib/libdl-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0xB202CB: _dlerror_run (in /lib/libdl-2.5.so) ==18894== by 0xB1FB83: dlopen@@GLIBC_2.1 (in /lib/libdl-2.5.so) ==18894== by 0x40674AF: switch_dso_open (switch_dso.c:96) ==18894== by 0x406CC16: switch_loadable_module_load_module_ex (switch_loadable_module.c:804) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== ==18894== ==18894== 3,072 bytes in 1 blocks are still reachable in loss record 167 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x1287870C: JS_DHashAllocTable (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12878532: JS_DHashTableInit (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x128930BF: js_InitGC (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A94: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 3,235 bytes in 5 blocks are still reachable in loss record 168 of 193 ==18894== at 0x40046FF: calloc (vg_replace_malloc.c:279) ==18894== by 0x9C16CA: _dl_new_object (in /lib/ld-2.5.so) ==18894== by 0x9BD020: _dl_map_object_from_fd (in /lib/ld-2.5.so) ==18894== by 0x9BF46B: _dl_map_object (in /lib/ld-2.5.so) ==18894== by 0x9C8A30: dl_open_worker (in /lib/ld-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0x9C8411: _dl_open (in /lib/ld-2.5.so) ==18894== by 0xB1FC4C: dlopen_doit (in /lib/libdl-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0xB202CB: _dlerror_run (in /lib/libdl-2.5.so) ==18894== by 0xB1FB83: dlopen@@GLIBC_2.1 (in /lib/libdl-2.5.so) ==18894== by 0x4067486: switch_dso_open (switch_dso.c:94) ==18894== ==18894== ==18894== 3,744 bytes in 36 blocks are still reachable in loss record 169 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40C7FFE: apr_allocator_create (apr_pools.c:91) ==18894== by 0x40502BF: switch_core_perform_new_memory_pool (switch_core_memory.c:353) ==18894== by 0x406C7CD: switch_loadable_module_load_module_ex (switch_loadable_module.c:785) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 6,920 bytes in 34 blocks are still reachable in loss record 170 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x9C3B6B: _dl_map_object_deps (in /lib/ld-2.5.so) ==18894== by 0x9C8A8C: dl_open_worker (in /lib/ld-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0x9C8411: _dl_open (in /lib/ld-2.5.so) ==18894== by 0xB1FC4C: dlopen_doit (in /lib/libdl-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0xB202CB: _dlerror_run (in /lib/libdl-2.5.so) ==18894== by 0xB1FB83: dlopen@@GLIBC_2.1 (in /lib/libdl-2.5.so) ==18894== by 0x40674AF: switch_dso_open (switch_dso.c:96) ==18894== by 0x406CC16: switch_loadable_module_load_module_ex (switch_loadable_module.c:804) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== ==18894== ==18894== 8,192 bytes in 1 blocks are still reachable in loss record 171 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40C8710: apr_pool_create_ex (apr_pools.c:300) ==18894== by 0x40502E6: switch_core_perform_new_memory_pool (switch_core_memory.c:357) ==18894== by 0x40643E4: switch_load_network_lists (switch_core.c:891) ==18894== by 0x40659E4: switch_core_init_and_modload (switch_core.c:1457) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 8,192 bytes in 1 blocks are still reachable in loss record 172 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40C8710: apr_pool_create_ex (apr_pools.c:300) ==18894== by 0x40502E6: switch_core_perform_new_memory_pool (switch_core_memory.c:357) ==18894== by 0x1279F75A: mod_spidermonkey_load (mod_spidermonkey.c:1007) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 8,192 bytes in 1 blocks are still reachable in loss record 173 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x4144CD8: libedit_fgetln (fgetln.c:60) ==18894== by 0x4145E8F: history (history.c:685) ==18894== by 0x4049C9E: switch_console_loop (switch_console.c:804) ==18894== by 0x4061CE7: switch_core_runtime_loop (switch_core.c:687) ==18894== by 0x804A453: main (switch.c:715) ==18894== ==18894== ==18894== 8,192 bytes in 1 blocks are still reachable in loss record 174 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40C8B7C: apr_palloc (apr_pools.c:300) ==18894== by 0x404FC7F: switch_core_perform_alloc (switch_core_memory.c:442) ==18894== by 0x40677B3: switch_loadable_module_create_interface (switch_loadable_module.c:1632) ==18894== by 0x10D88DC9: mod_dptools_load (mod_dptools.c:2754) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 8,192 bytes in 1 blocks are still reachable in loss record 175 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40C8710: apr_pool_create_ex (apr_pools.c:300) ==18894== by 0x40CDA36: apr_thread_create (thread.c:171) ==18894== by 0x403ECF8: switch_thread_create (switch_apr.c:631) ==18894== by 0x7AFF689: sofia_presence_event_thread_start (sofia_presence.c:755) ==18894== by 0x7AFF791: sofia_presence_event_handler (sofia_presence.c:768) ==18894== by 0x4073C1B: switch_event_deliver (switch_event.c:342) ==18894== by 0x4073DAA: switch_event_dispatch_thread (switch_event.c:255) ==18894== by 0x40CD6C5: dummy_worker (thread.c:138) ==18894== by 0xB5346A: start_thread (in /lib/libpthread-2.5.so) ==18894== by 0xAAADBD: clone (in /lib/libc-2.5.so) ==18894== ==18894== ==18894== 8,192 bytes in 1 blocks are still reachable in loss record 176 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40C8710: apr_pool_create_ex (apr_pools.c:300) ==18894== by 0x40502E6: switch_core_perform_new_memory_pool (switch_core_memory.c:357) ==18894== by 0x11D2FDC6: mod_sndfile_load (mod_sndfile.c:400) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 8,192 bytes in 1 blocks are still reachable in loss record 177 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40C8B7C: apr_palloc (apr_pools.c:300) ==18894== by 0x404FC7F: switch_core_perform_alloc (switch_core_memory.c:442) ==18894== by 0x11CD6624: ??? (switch_loadable_module.h:396) ==18894== by 0x11CD6E91: mod_voipcodecs_load (mod_voipcodecs.c:712) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 8,192 bytes in 1 blocks are still reachable in loss record 178 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40C8710: apr_pool_create_ex (apr_pools.c:300) ==18894== by 0x40502E6: switch_core_perform_new_memory_pool (switch_core_memory.c:357) ==18894== by 0x8036763: mod_loopback_load (mod_loopback.c:871) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 8,192 bytes in 1 blocks are still reachable in loss record 179 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40C8710: apr_pool_create_ex (apr_pools.c:300) ==18894== by 0x40502E6: switch_core_perform_new_memory_pool (switch_core_memory.c:357) ==18894== by 0x406C7CD: switch_loadable_module_load_module_ex (switch_loadable_module.c:785) ==18894== by 0x406D073: switch_loadable_module_init (switch_loadable_module.c:1159) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 8,192 bytes in 1 blocks are still reachable in loss record 180 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40C8710: apr_pool_create_ex (apr_pools.c:300) ==18894== by 0x40502E6: switch_core_perform_new_memory_pool (switch_core_memory.c:357) ==18894== by 0x406CE99: switch_loadable_module_init (switch_loadable_module.c:1136) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 8,192 bytes in 1 blocks are still reachable in loss record 181 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40C8710: apr_pool_create_ex (apr_pools.c:300) ==18894== by 0x40502E6: switch_core_perform_new_memory_pool (switch_core_memory.c:357) ==18894== by 0x4065C3C: switch_scheduler_task_thread_start (switch_scheduler.c:295) ==18894== by 0x40639C5: switch_core_init (switch_core.c:1250) ==18894== by 0x406585C: switch_core_init_and_modload (switch_core.c:1422) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 8,192 bytes in 1 blocks are still reachable in loss record 182 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40C8710: apr_pool_create_ex (apr_pools.c:300) ==18894== by 0x40502E6: switch_core_perform_new_memory_pool (switch_core_memory.c:357) ==18894== by 0x406C7CD: switch_loadable_module_load_module_ex (switch_loadable_module.c:785) ==18894== by 0x406D097: switch_loadable_module_init (switch_loadable_module.c:1160) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 8,192 bytes in 1 blocks are still reachable in loss record 183 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40C8710: apr_pool_create_ex (apr_pools.c:300) ==18894== by 0x40CDA36: apr_thread_create (thread.c:171) ==18894== by 0x403ECF8: switch_thread_create (switch_apr.c:631) ==18894== by 0x4065CA6: switch_scheduler_task_thread_start (switch_scheduler.c:300) ==18894== by 0x40639C5: switch_core_init (switch_core.c:1250) ==18894== by 0x406585C: switch_core_init_and_modload (switch_core.c:1422) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 8,192 bytes in 1 blocks are still reachable in loss record 184 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40C8B7C: apr_palloc (apr_pools.c:300) ==18894== by 0x404FC7F: switch_core_perform_alloc (switch_core_memory.c:442) ==18894== by 0x4067843: switch_loadable_module_create_interface (switch_loadable_module.c:1635) ==18894== by 0x1035CCB3: mod_commands_load (mod_commands.c:3565) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 8,192 bytes in 1 blocks are still reachable in loss record 185 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40C8710: apr_pool_create_ex (apr_pools.c:300) ==18894== by 0x804A15D: main (switch.c:676) ==18894== ==18894== ==18894== 8,192 bytes in 1 blocks are still reachable in loss record 186 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40C8710: apr_pool_create_ex (apr_pools.c:300) ==18894== by 0x40C9747: apr_initialize (start.c:58) ==18894== by 0x8049C7B: main (switch.c:596) ==18894== ==18894== ==18894== 8,192 bytes in 1 blocks are still reachable in loss record 187 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40C8710: apr_pool_create_ex (apr_pools.c:300) ==18894== by 0x40C892E: apr_pool_initialize (apr_pools.c:527) ==18894== by 0x40C971E: apr_initialize (start.c:55) ==18894== by 0x8049C7B: main (switch.c:596) ==18894== ==18894== ==18894== 9,388 bytes in 1 blocks are still reachable in loss record 188 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x12867A2F: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 16,384 bytes in 2 blocks are still reachable in loss record 189 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40C8710: apr_pool_create_ex (apr_pools.c:300) ==18894== by 0x40CDA36: apr_thread_create (thread.c:171) ==18894== by 0x403ECF8: switch_thread_create (switch_apr.c:631) ==18894== by 0x4062738: switch_core_launch_thread (switch_core.c:400) ==18894== by 0x406D4D6: switch_loadable_module_init (switch_loadable_module.c:121) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 21,733 bytes in 34 blocks are still reachable in loss record 190 of 193 ==18894== at 0x40046FF: calloc (vg_replace_malloc.c:279) ==18894== by 0x9C16CA: _dl_new_object (in /lib/ld-2.5.so) ==18894== by 0x9BD020: _dl_map_object_from_fd (in /lib/ld-2.5.so) ==18894== by 0x9BF46B: _dl_map_object (in /lib/ld-2.5.so) ==18894== by 0x9C8A30: dl_open_worker (in /lib/ld-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0x9C8411: _dl_open (in /lib/ld-2.5.so) ==18894== by 0xB1FC4C: dlopen_doit (in /lib/libdl-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0xB202CB: _dlerror_run (in /lib/libdl-2.5.so) ==18894== by 0xB1FB83: dlopen@@GLIBC_2.1 (in /lib/libdl-2.5.so) ==18894== by 0x40674AF: switch_dso_open (switch_dso.c:96) ==18894== ==18894== ==18894== 200,704 bytes in 1 blocks are still reachable in loss record 191 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40C8B7C: apr_palloc (apr_pools.c:300) ==18894== by 0x40BEA4A: apr_queue_create (apr_queue.c:129) ==18894== by 0x403E4AA: switch_queue_create (switch_apr.c:892) ==18894== by 0x7AC85B5: mod_sofia_load (mod_sofia.c:3262) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 200,704 bytes in 1 blocks are still reachable in loss record 192 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40C8B7C: apr_palloc (apr_pools.c:300) ==18894== by 0x40BEA4A: apr_queue_create (apr_queue.c:129) ==18894== by 0x403E4AA: switch_queue_create (switch_apr.c:892) ==18894== by 0x7AC8595: mod_sofia_load (mod_sofia.c:3261) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 294,912 bytes in 36 blocks are still reachable in loss record 193 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40C8710: apr_pool_create_ex (apr_pools.c:300) ==18894== by 0x40502E6: switch_core_perform_new_memory_pool (switch_core_memory.c:357) ==18894== by 0x406C7CD: switch_loadable_module_load_module_ex (switch_loadable_module.c:785) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== LEAK SUMMARY: ==18894== definitely lost: 1,191 bytes in 9 blocks. ==18894== indirectly lost: 1,556 bytes in 123 blocks. ==18894== possibly lost: 320 bytes in 2 blocks. ==18894== still reachable: 924,064 bytes in 472 blocks. ==18894== suppressed: 0 bytes in 0 blocks. From talk2ram at gmail.com Mon Jul 20 04:19:55 2009 From: talk2ram at gmail.com (ram) Date: Mon, 20 Jul 2009 16:49:55 +0530 Subject: [Freeswitch-users] Creating a new User Agent In-Reply-To: <1452e2980907192330o14904fb8ke0aec88fd6afde0a@mail.gmail.com> References: <1452e2980907192330o14904fb8ke0aec88fd6afde0a@mail.gmail.com> Message-ID: On Mon, Jul 20, 2009 at 12:00 PM, velusamy velu wrote: > Dear All, > I want to create a new User Agent like sip configurations in > Asterisk. I checked default user agents 1000 to 1001. But I have bit > confused the relationship between default user agents and sip_profiles. > > I need some help from you all for the following questions, > How to create new user agent ? > How to relate the new user agent with sip internal > profile ? > Hi have you looked at this link http://wiki.freeswitch.org/wiki/Getting_Started_Guide#What_SIP_Profiles_Do ram -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090720/3492ea21/attachment.html From technical at ttnc.co.uk Mon Jul 20 04:22:53 2009 From: technical at ttnc.co.uk (TTNC - Adnan Barakat) Date: Mon, 20 Jul 2009 12:22:53 +0100 Subject: [Freeswitch-users] call fails when using leg_timeout Message-ID: <4A64538D.6070301@ttnc.co.uk> Hi, I seem to have come across a strange problem; Basically I'm trying to dial 3 destinations one after another, until the destination dialled is answered, and I only want the destination to ring for 20 seconds. If I do this from the console what I'm trying to achieve works fine; originate {leg_timeout=20,ignore_early_media=true}sofia/internal/123 at 1.2.3.4|sofia/internal/456 at 1.2.3.4|sofia/internal/789 at 1.2.3.4 &park() But if I do it from dialplan it doesn't; Doing this below dials the first destination for 20 seconds, then the second destination for 5 seconds or so, then the call terminates with ORIGINATOR_CANCEL Any ideas what might be causing this and any solutions would be appreciated. Many thanks Adnan From technical at ttnc.co.uk Mon Jul 20 04:47:46 2009 From: technical at ttnc.co.uk (TTNC - Adnan Barakat) Date: Mon, 20 Jul 2009 12:47:46 +0100 Subject: [Freeswitch-users] caller_id 0000000000 Message-ID: <4A645962.6080507@ttnc.co.uk> Hi We have a FreeSWITCH server receiving calls from a provider, we process the call (play greeting messages etc...) then pass the call out again to an end destination via the same provider. But if the caller is witholding their cli our provider send the call to us with the sip_from_user variable to 'nobody'. Is there anyway to remove the caller_id when we don't receive it rather than override it with 0000000000? or send the call in the same way we receive it ie. from 'nobody'? Thanks Adnan From tayeb.meftah at gmail.com Mon Jul 20 06:36:17 2009 From: tayeb.meftah at gmail.com (Meftah Tayeb) Date: Mon, 20 Jul 2009 13:36:17 +0000 Subject: [Freeswitch-users] Freeswitch Windows Issues In-Reply-To: References: <4A6443B8.5000106@gmail.com> Message-ID: <4A6472D1.6030603@gmail.com> hello baskar, i think Freeswitch ODBC Support is not enabled for Windows you must compile it with ODBC Support enabled thanks Baskar wrote: > *Hi Meftah Tayeb**,* > > *One more question in windows whether it is possible to connect the > ODBC connection through JavaScript in freeswitch. > > I have configured inbound in Linux it is working fine but same script > i tried in windows but i get this error. I have installed and > configured MYSQL connector ODBC in window. But when is call the script > i get this error. > > 2009-07-20 10:55:42.781250 [ERR] mod_spidermonkey.c:3393 Error loading > ODBC > 2009-07-20 10:55:42.781250 [ERR] Inbound1.js:5 ReferenceError: ODBC > is not defined > > Can some one assist me to resolve this above error > > Thanks in advance. > > -- > Thanks with 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 > > > > __________ Information from ESET NOD32 Antivirus, version of virus signature database 4260 (20090720) __________ > > The message was checked by ESET NOD32 Antivirus. > > http://www.eset.com > > __________ Information from ESET NOD32 Antivirus, version of virus signature database 4260 (20090720) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com From mike at jerris.com Mon Jul 20 06:58:49 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 20 Jul 2009 09:58:49 -0400 Subject: [Freeswitch-users] Freeswitch Windows Issues In-Reply-To: <4A6472D1.6030603@gmail.com> References: <4A6443B8.5000106@gmail.com> <4A6472D1.6030603@gmail.com> Message-ID: <5E024C30-C98D-4EC0-9E9F-F126FAED0BE3@jerris.com> This is not correct. Do you have the odbc mod for spidermonkey loaded? On Jul 20, 2009, at 9:36 AM, Meftah Tayeb wrote: > hello baskar, > i think Freeswitch ODBC Support is not enabled for Windows > you must compile it with ODBC Support enabled > thanks > Baskar wrote: >> *Hi Meftah Tayeb**,* >> >> *One more question in windows whether it is possible to connect the >> ODBC connection through JavaScript in freeswitch. >> >> I have configured inbound in Linux it is working fine but same script >> i tried in windows but i get this error. I have installed and >> configured MYSQL connector ODBC in window. But when is call the >> script >> i get this error. >> >> 2009-07-20 10:55:42.781250 [ERR] mod_spidermonkey.c:3393 Error >> loading >> ODBC >> 2009-07-20 10:55:42.781250 [ERR] Inbound1.js:5 ReferenceError: ODBC >> is not defined >> >> Can some one assist me to resolve this above error >> >> Thanks in advance. >> >> -- >> Thanks with 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 >> >> >> >> __________ Information from ESET NOD32 Antivirus, version of virus >> signature database 4260 (20090720) __________ >> >> The message was checked by ESET NOD32 Antivirus. >> >> http://www.eset.com >> >> > > > > __________ Information from ESET NOD32 Antivirus, version of virus > signature database 4260 (20090720) __________ > > The message was checked by ESET NOD32 Antivirus. > > http://www.eset.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 zolotov at altron.ua Mon Jul 20 03:12:07 2009 From: zolotov at altron.ua (Evgeniy Zolotov) Date: Mon, 20 Jul 2009 13:12:07 +0300 Subject: [Freeswitch-users] mod_fax problems Message-ID: <3210356507.20090720131207@altron.ua> Hello! I have some problems with receiving fax messages. My FreeSWITCH cann't decode some faxes. As I understood this problem has occurred with Panasonic and Panasonic-like fax-machines. I have recorded wav-file, which corresponds to the receiving fax (see fax.wav.tgz) and log-file of this session (fax.log.tgz). I tried decode this fax with the help of transferring them from one FreeSWITCH to another, but with no luck (I used for this SIP and openzap - result the same). Transferring from one FreeSWITCH to another with the help of 'txfax' works perfect. OS: Centos 5.2 FreeSWITCH: 1.0.2 (11053) Configuration file for mod_fax - original. So, please, help me to understand this situation. Thanks, Evgeniy. mailto:zolotov at altron.ua -------------- next part -------------- A non-text attachment was scrubbed... Name: fax.log.tgz Type: application/x-compressed Size: 2672 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090720/e164ae4e/attachment.bin -------------- next part -------------- A non-text attachment was scrubbed... Name: fax.wav.tgz Type: application/x-compressed Size: 595910 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090720/e164ae4e/attachment-0001.bin From brian at freeswitch.org Mon Jul 20 07:23:22 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 20 Jul 2009 09:23:22 -0500 Subject: [Freeswitch-users] mod_fax problems In-Reply-To: <3210356507.20090720131207@altron.ua> References: <3210356507.20090720131207@altron.ua> Message-ID: I would highly recommend you update to SVN trunk. /b On Jul 20, 2009, at 5:12 AM, Evgeniy Zolotov wrote: > Hello! > > I have some problems with receiving fax messages. My FreeSWITCH > cann't decode some faxes. As I understood this problem has occurred > with Panasonic and Panasonic-like fax-machines. > I have recorded wav-file, which corresponds to the receiving fax > (see fax.wav.tgz) and log-file of this session (fax.log.tgz). > I tried decode this fax with the help of transferring them from one > FreeSWITCH to another, but with no luck (I used for this SIP and > openzap - result the same). > Transferring from one FreeSWITCH to another with the help of 'txfax' > works perfect. > > OS: Centos 5.2 > FreeSWITCH: 1.0.2 (11053) > > Configuration file for mod_fax - original. > > So, please, help me to understand this situation. > > Thanks, Evgeniy. From anthony.minessale at gmail.com Mon Jul 20 07:29:28 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 20 Jul 2009 09:29:28 -0500 Subject: [Freeswitch-users] mod_conference: behavior when the conference is torn down before 31seconds In-Reply-To: <4A61B993.6090402@gmx.net> References: <4A61B993.6090402@gmx.net> Message-ID: <191c3a030907200729l5783d877m3ef5159dba25624e@mail.gmail.com> the problem stems from the fact that you did an outcall to those other addresses. The 30 sec is the timeout waiting for those calls to establish. The outcome of those outbound calls must be determined before the conf will end. On Sat, Jul 18, 2009 at 7:01 AM, Peter P GMX wrote: > Hello Luis, > > are you using encrypted TLS instead on SIP on this phone? I experienced > a similar behaviour with 31 seocnds on TLS. > > Best regards > Peter > > Luis F Urrea schrieb: > > Hi all, > > > > I am experiencing a behavior that I cannot clearly understand. > > Basically I "autocall" a few phones into a conference with the > > sip_auto_answer set to true, as follows: > > > > > > > > > data="conference_auto_outcall_prefix={sip_auto_answer=true}"/> > > > data="user/305"/> > > > data="user/303"/> > > > data="user/201"/> > > > > > > > > > > > > The conference establishes just fine and everyone can hear just fine. > > > > The "strange" behavior comes when the person calling to ext 773 hangs > > up before 31 seconds have passed, the rest of the phones stay up until > > they reach second 31 into the "conference". > > > > I am using snom phones and I see the BYE message arriving at the > > phones exactly at second 31 after the call establishes. > > > > The conference itself however does not exist after the person calling > > 773 hangs up (doing conference list on CLI shows NO active conferences). > > > > If the conference stays up more than 31 seconds, then when the person > > calling 773 hangs up, the rest of the phones hang up immediately as > > desired. > > > > Here's the log for a "page" that lasts less than 31 seconds: > > > > http://pastebin.freeswitch.org/9773 > > > > Here's the log of the phone for a "page" that lasts less than 31 seconds: > > > > http://pastebin.freeswitch.org/9774 > > > > Your inout is appreciated. > > > > Regards, > > > > 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 > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference 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/20090720/77f5af7c/attachment.html From anthony.minessale at gmail.com Mon Jul 20 07:58:57 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 20 Jul 2009 09:58:57 -0500 Subject: [Freeswitch-users] call fails when using leg_timeout In-Reply-To: <4A64538D.6070301@ttnc.co.uk> References: <4A64538D.6070301@ttnc.co.uk> Message-ID: <191c3a030907200758p24978436kcdbe5259a2e2d865@mail.gmail.com> in a bridge situation you also have to set originate_timeout to the total time you are willing to wait for the combined leg timeouts i.e. 60 for 20x20x20 On Mon, Jul 20, 2009 at 6:22 AM, TTNC - Adnan Barakat wrote: > Hi, > > I seem to have come across a strange problem; Basically I'm trying to > dial 3 destinations one after another, until the destination dialled is > answered, and I only want the destination to ring for 20 seconds. > > If I do this from the console what I'm trying to achieve works fine; > > originate > {leg_timeout=20,ignore_early_media=true}sofia/internal/123 at 1.2.3.4 > |sofia/internal/456 at 1.2.3.4|sofia/internal/789 at 1.2.3.4 > &park() > > But if I do it from dialplan it doesn't; Doing this below dials the > first destination for 20 seconds, then the second destination for 5 > seconds or so, then the call terminates with ORIGINATOR_CANCEL > > data="{leg_timeout=20,ignore_early_media=true}sofia/internal/123 at 1.2.3.4 > |sofia/internal/456 at 1.2.3.4|sofia/internal/789 at 1.2.3.4" > /> > > Any ideas what might be causing this and any solutions would be > appreciated. > > > Many thanks > > Adnan > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference 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/20090720/632fdf0c/attachment.html From zolotov at altron.ua Mon Jul 20 08:06:25 2009 From: zolotov at altron.ua (Evgeniy Zolotov) Date: Mon, 20 Jul 2009 18:06:25 +0300 Subject: [Freeswitch-users] mod_fax problems In-Reply-To: References: <3210356507.20090720131207@altron.ua> Message-ID: <1445760366.20090720180625@altron.ua> Hello, Brian. I understand that this is the first thing that I should make. But we have many our own changes in modules and after update of all project we'll need to transfer them too. But this is the great peace of job. So probably update of mod_fax and spandsp would be enough? ?? ?????? 20 ???? 2009 ?., 17:23:22: > I would highly recommend you update to SVN trunk. > /b > On Jul 20, 2009, at 5:12 AM, Evgeniy Zolotov wrote: >> Hello! >> >> I have some problems with receiving fax messages. My FreeSWITCH >> cann't decode some faxes. As I understood this problem has occurred >> with Panasonic and Panasonic-like fax-machines. >> I have recorded wav-file, which corresponds to the receiving fax >> (see fax.wav.tgz) and log-file of this session (fax.log.tgz). >> I tried decode this fax with the help of transferring them from one >> FreeSWITCH to another, but with no luck (I used for this SIP and >> openzap - result the same). >> Transferring from one FreeSWITCH to another with the help of 'txfax' >> works perfect. >> >> OS: Centos 5.2 >> FreeSWITCH: 1.0.2 (11053) >> >> Configuration file for mod_fax - original. >> >> So, please, help me to understand this situation. >> >> 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 -- Thanks, Evgeniy. mailto:zolotov at altron.ua From brian at freeswitch.org Mon Jul 20 08:09:37 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 20 Jul 2009 10:09:37 -0500 Subject: [Freeswitch-users] mod_fax problems In-Reply-To: <1445760366.20090720180625@altron.ua> References: <3210356507.20090720131207@altron.ua> <1445760366.20090720180625@altron.ua> Message-ID: <0C683BCA-1F4D-42CA-8ADA-4F1A5B744EAB@freeswitch.org> If you have made changes to the included modules you'll need to report your changes to jira http://jira.freeswitch.org Thanks, /b On Jul 20, 2009, at 10:06 AM, Evgeniy Zolotov wrote: > Hello, Brian. > > I understand that this is the first thing that I should make. > But we have many our own changes in modules and after update of all > project we'll need to transfer them too. But this is the great peace > of job. > So probably update of mod_fax and spandsp would be enough? From zolotov at altron.ua Mon Jul 20 08:30:31 2009 From: zolotov at altron.ua (Evgeniy Zolotov) Date: Mon, 20 Jul 2009 18:30:31 +0300 Subject: [Freeswitch-users] mod_fax problems In-Reply-To: <0C683BCA-1F4D-42CA-8ADA-4F1A5B744EAB@freeswitch.org> References: <3210356507.20090720131207@altron.ua> <1445760366.20090720180625@altron.ua> <0C683BCA-1F4D-42CA-8ADA-4F1A5B744EAB@freeswitch.org> Message-ID: <151636129.20090720183031@altron.ua> Hello, Brian. Ok, I understand. So I'll try to update to SVN trunk and then post log, if any problems still exists. ?? ?????? 20 ???? 2009 ?., 18:09:37: > If you have made changes to the included modules you'll need to report > your changes to jira http://jira.freeswitch.org > Thanks, > /b > On Jul 20, 2009, at 10:06 AM, Evgeniy Zolotov wrote: >> Hello, Brian. >> >> I understand that this is the first thing that I should make. >> But we have many our own changes in modules and after update of all >> project we'll need to transfer them too. But this is the great peace >> of job. >> So probably update of mod_fax and spandsp would be enough? > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Thanks, Evgeniy. mailto:zolotov at altron.ua From zolotov at altron.ua Mon Jul 20 08:40:15 2009 From: zolotov at altron.ua (Evgeniy Zolotov) Date: Mon, 20 Jul 2009 18:40:15 +0300 Subject: [Freeswitch-users] mod_fax problems Message-ID: <1810546946.20090720184015@altron.ua> Hello, Brian. Ok, I understand. So I'll try to update to SVN trunk and then post log, if any problems still exists. ?? ?????? 20 ???? 2009 ?., 18:09:37: > If you have made changes to the included modules you'll need to report > your changes to jira http://jira.freeswitch.org > Thanks, > /b > On Jul 20, 2009, at 10:06 AM, Evgeniy Zolotov wrote: >> Hello, Brian. >> >> I understand that this is the first thing that I should make. >> But we have many our own changes in modules and after update of all >> project we'll need to transfer them too. But this is the great peace >> of job. >> So probably update of mod_fax and spandsp would be enough? > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Thanks, Evgeniy. mailto:zolotov at altron.ua From msc at freeswitch.org Mon Jul 20 09:40:51 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 20 Jul 2009 09:40:51 -0700 Subject: [Freeswitch-users] Creating a new User Agent In-Reply-To: <1452e2980907192330o14904fb8ke0aec88fd6afde0a@mail.gmail.com> References: <1452e2980907192330o14904fb8ke0aec88fd6afde0a@mail.gmail.com> Message-ID: <87f2f3b90907200940tdfa995bk6bf465c2f0880788@mail.gmail.com> On Sun, Jul 19, 2009 at 11:30 PM, velusamy velu wrote: > Dear All, > I want to create a new User Agent like sip configurations in > Asterisk. I checked default user agents 1000 to 1001. But I have bit > confused the relationship between default user agents and sip_profiles. > > I need some help from you all for the following questions, > How to create new user agent ? > How to relate the new user agent with sip internal > profile ? I believe you might be mixing terminology. There is a difference between a "user" and a "user agent." In FreeSWITCH, a SIP profile *is* a user agent. In the default configuration, in conf/directory/default/ there are 20 pre-configured users. 1000, 1001, etc. are simply SIP users that are ready for use. Point a SIP phone at your FreeSWITCH IP address and set the auth user name to "1000" and the password to "1234" and it should register just fine. To learn more about SIP profiles I do recommend the link that ram posted. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090720/99cfb45b/attachment.html From technical at ttnc.co.uk Mon Jul 20 09:48:41 2009 From: technical at ttnc.co.uk (TTNC - Adnan Barakat) Date: Mon, 20 Jul 2009 17:48:41 +0100 Subject: [Freeswitch-users] call fails when using leg_timeout In-Reply-To: <191c3a030907200758p24978436kcdbe5259a2e2d865@mail.gmail.com> References: <4A64538D.6070301@ttnc.co.uk> <191c3a030907200758p24978436kcdbe5259a2e2d865@mail.gmail.com> Message-ID: <4A649FE9.3060104@ttnc.co.uk> Anthony Minessale wrote: > in a bridge situation you also have to set originate_timeout to the > total time you are willing to wait for the combined leg timeouts > > i.e. 60 for 20x20x20 I added originate_timeout=60 but now only the first destination rings for 30 seconds, then the call is still terminated with ORIGINATOR_CANCEL Any other ideas? Thanks Adnan From dujinfang at gmail.com Mon Jul 20 10:39:46 2009 From: dujinfang at gmail.com (Seven Du) Date: Tue, 21 Jul 2009 01:39:46 +0800 Subject: [Freeswitch-users] FreeSWITCH-Air, Another GUI? Message-ID: ALL, I know you guys more prefer a CLI version of softphone to a GUI version. But I still would like to share this: http://wiki.freeswitch.org/wiki/FsAir And feel free to give me feedbacks. I'v only played a few days of ActionScript, it's highly appreciated if someone can give me help on the following problems. 1) Bounce the icon on Mac on incoming call. 2) Show a small window on incoming call. 3) Is it possible to block read/write a socket? I want to implement sendRecv() in ActionScript like in the C version of ESL. Thanks. From zolotov at altron.ua Mon Jul 20 11:02:25 2009 From: zolotov at altron.ua (Evgeniy Zolotov) Date: Mon, 20 Jul 2009 21:02:25 +0300 Subject: [Freeswitch-users] mod_fax problems In-Reply-To: <1810546946.20090720184015@altron.ua> References: <1810546946.20090720184015@altron.ua> Message-ID: <1717692390.20090720210225@altron.ua> I've updated to current trunk: freeswitch at test> version FreeSWITCH Version 1.0.trunk (14299) But problem still exists. I've attached a new log. So if anybody has an idea it would be greatly appreciated. ?? ?????? 20 ???? 2009 ?., 18:40:15: > Hello, Brian. > Ok, I understand. So I'll try to update to SVN trunk and then post > log, if any problems still exists. > ?? ?????? 20 ???? 2009 ?., 18:09:37: >> If you have made changes to the included modules you'll need to report >> your changes to jira http://jira.freeswitch.org >> Thanks, >> /b >> On Jul 20, 2009, at 10:06 AM, Evgeniy Zolotov wrote: >>> Hello, Brian. >>> >>> I understand that this is the first thing that I should make. >>> But we have many our own changes in modules and after update of all >>> project we'll need to transfer them too. But this is the great peace >>> of job. >>> So probably update of mod_fax and spandsp would be enough? >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > Thanks, Evgeniy. > mailto:zolotov at altron.ua > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Thanks, Evgeniy. mailto:zolotov at altron.ua -------------- next part -------------- A non-text attachment was scrubbed... Name: fax.log.tgz Type: application/x-compressed Size: 2746 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090720/9f80c6a0/attachment.bin From marketing at cluecon.com Mon Jul 20 10:58:59 2009 From: marketing at cluecon.com (Michael Collins) Date: Mon, 20 Jul 2009 10:58:59 -0700 Subject: [Freeswitch-users] ClueCon 2009 - Last Chance To Book At The Wyndham Message-ID: <87f2f3b90907201058i33465e82k7a3608436064fe06@mail.gmail.com> Greetings! We would like to let everyone know that the hotel where ClueCon 2009 is being held - The Wyndham Chicago - is completely booked up outside of the block of rooms that we have reserved. There are still rooms available. However, after noon CST tomorrow, July 21, those rooms will no longer be reserved for ClueCon attendees so you have until then to get your reservations in. After noon tomorrow we will no longer be able to guarantee that you have an chance to register with the hotel, so please act fast! See you in Chicago. Thanks for supporting ClueCon! -Michael http://www.cluecon.com 877.742.CLUE -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090720/536fa5be/attachment.html From brian at freeswitch.org Mon Jul 20 11:29:05 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 20 Jul 2009 13:29:05 -0500 Subject: [Freeswitch-users] mod_fax problems In-Reply-To: <1717692390.20090720210225@altron.ua> References: <1810546946.20090720184015@altron.ua> <1717692390.20090720210225@altron.ua> Message-ID: <3B4D48C6-4556-4B76-98F7-3A0BC34C133F@freeswitch.org> Please follow the bug reporting guidelines here http://wiki.freeswitch.org/wiki/Reporting_Bugs If you can open a jira and attach all the info for the issue you're having. Thanks, Brian On Jul 20, 2009, at 1:02 PM, Evgeniy Zolotov wrote: > I've updated to current trunk: > > freeswitch at test> version > FreeSWITCH Version 1.0.trunk (14299) > > But problem still exists. I've attached a new log. > > So if anybody has an idea it would be greatly appreciated. From tayeb.meftah at gmail.com Mon Jul 20 11:45:43 2009 From: tayeb.meftah at gmail.com (Meftah Tayeb) Date: Mon, 20 Jul 2009 18:45:43 +0000 Subject: [Freeswitch-users] FreeSWITCH-Air, Another GUI? In-Reply-To: References: Message-ID: <4A64BB57.50206@gmail.com> hellok, i can help about Action Script by providing some EBooks (CHM/PDF) to you DelphiWorld in #Freeswitch thanks Seven Du wrote: > ALL, > > I know you guys more prefer a CLI version of softphone to a GUI > version. But I still would like to share this: > > http://wiki.freeswitch.org/wiki/FsAir > > And feel free to give me feedbacks. > > I'v only played a few days of ActionScript, it's highly appreciated if > someone can give me help on the following problems. > > > 1) Bounce the icon on Mac on incoming call. > 2) Show a small window on incoming call. > 3) Is it possible to block read/write a socket? I want to implement > sendRecv() in ActionScript like in the C version of ESL. > > 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 > > > __________ Information from ESET NOD32 Antivirus, version of virus signature database 4261 (20090720) __________ > > The message was checked by ESET NOD32 Antivirus. > > http://www.eset.com > > > __________ Information from ESET NOD32 Antivirus, version of virus signature database 4261 (20090720) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com From gabe at gundy.org Mon Jul 20 13:09:27 2009 From: gabe at gundy.org (Gabriel Gunderson) Date: Mon, 20 Jul 2009 14:09:27 -0600 Subject: [Freeswitch-users] Sonus - what's the latest? Message-ID: <903da5680907201309m45d84647ucc2041f65b655f7@mail.gmail.com> All, Anyone setup FS with Sonus lately? I've just ordered service for a customer and it should be hooked-up pretty soon. Since ordering, I was made aware the they're using Sonus on the back-end. Well, I've been reading up on it and I'm getting a bit worried :( Anyone know if this is still the state of things? http://wiki.freeswitch.org/wiki/RTP_Issues#Sonus Thanks, Gabe From kristian.kielhofner at gmail.com Mon Jul 20 14:06:14 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Mon, 20 Jul 2009 17:06:14 -0400 Subject: [Freeswitch-users] Sonus - what's the latest? In-Reply-To: <903da5680907201309m45d84647ucc2041f65b655f7@mail.gmail.com> References: <903da5680907201309m45d84647ucc2041f65b655f7@mail.gmail.com> Message-ID: <2d9149cd0907201406h3a550817ya2b29ec40661662@mail.gmail.com> On Mon, Jul 20, 2009 at 4:09 PM, Gabriel Gunderson wrote: > All, > > Anyone setup FS with Sonus lately? ?I've just ordered service for a > customer and it should be hooked-up pretty soon. ?Since ordering, I > was made aware the they're using Sonus on the back-end. ?Well, I've > been reading up on it and I'm getting a bit worried :( > > Anyone know if this is still the state of things? > http://wiki.freeswitch.org/wiki/RTP_Issues#Sonus > > Thanks, > Gabe > Gabe, Every day... I don't know of any new issues with Sonus as used by the carriers I deal with but YMMV. -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From msc at freeswitch.org Mon Jul 20 15:20:42 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 20 Jul 2009 15:20:42 -0700 Subject: [Freeswitch-users] call fails when using leg_timeout In-Reply-To: <4A649FE9.3060104@ttnc.co.uk> References: <4A64538D.6070301@ttnc.co.uk> <191c3a030907200758p24978436kcdbe5259a2e2d865@mail.gmail.com> <4A649FE9.3060104@ttnc.co.uk> Message-ID: <87f2f3b90907201520t210b6f6chd743fa6550377013@mail.gmail.com> On Mon, Jul 20, 2009 at 9:48 AM, TTNC - Adnan Barakat wrote: > Anthony Minessale wrote: > > in a bridge situation you also have to set originate_timeout to the > > total time you are willing to wait for the combined leg timeouts > > > > i.e. 60 for 20x20x20 > I added originate_timeout=60 but now only the first destination rings > for 30 seconds, then the call is still terminated with ORIGINATOR_CANCEL > > Any other ideas? > I just did a test with this syntax and it worked for me. Please try it and report back. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090720/a1523e2a/attachment.html From steveu at coppice.org Mon Jul 20 17:22:48 2009 From: steveu at coppice.org (Steve Underwood) Date: Tue, 21 Jul 2009 08:22:48 +0800 Subject: [Freeswitch-users] mod_fax problems In-Reply-To: <3210356507.20090720131207@altron.ua> References: <3210356507.20090720131207@altron.ua> Message-ID: <4A650A58.3000607@coppice.org> Hi Evgeniy, Evgeniy Zolotov wrote: > Hello! > > I have some problems with receiving fax messages. My FreeSWITCH > cann't decode some faxes. As I understood this problem has occurred > with Panasonic and Panasonic-like fax-machines. > I have recorded wav-file, which corresponds to the receiving fax > (see fax.wav.tgz) and log-file of this session (fax.log.tgz). > I tried decode this fax with the help of transferring them from one > FreeSWITCH to another, but with no luck (I used for this SIP and > openzap - result the same). > Transferring from one FreeSWITCH to another with the help of 'txfax' > works perfect. > > OS: Centos 5.2 > FreeSWITCH: 1.0.2 (11053) > > Configuration file for mod_fax - original. > > So, please, help me to understand this situation. > > Thanks, Evgeniy The audio you posted doesn't seem to come from the call in your log file. The audio is very odd, with sections of a FAX called joined together in a strange order. The log file is very simple, and tells you exactly what went wrong. It says: Fax processing not successful - result (10) Far end is not able to transmit. and the reason it says this is because the DIS message from the far end says: .... ...0= Ready to transmit a fax document (polling): Not set Regards, Steve From yudha2008 at gmail.com Mon Jul 20 21:25:49 2009 From: yudha2008 at gmail.com (Baskar) Date: Tue, 21 Jul 2009 09:55:49 +0530 Subject: [Freeswitch-users] Freeswitch Windows Issues In-Reply-To: <5E024C30-C98D-4EC0-9E9F-F126FAED0BE3@jerris.com> References: <4A6443B8.5000106@gmail.com> <4A6472D1.6030603@gmail.com> <5E024C30-C98D-4EC0-9E9F-F126FAED0BE3@jerris.com> Message-ID: *Hi Michael Jerris, In linux if ODBC modules to be load in freeswitch i can load by this command make mod_spidermonkey_odbc-install make install But in windows how can i enable the modules for mod_spidermonkey? I checked whether mod spidermonkey is loaded by this command load mod_spidermonkey . freeswitch at Baskar>load mod_spidermonkey 2009-07-21 09:42:14.859375 [WARNING] switch_loadable_module.c:939 Module mod_spidermonkey Already **Loaded! ** API CALL [load(mod_spidermonkey)] output:-ERR [Module already loaded] **But output say is already loaded but still i get this error**: **2009-07-20 10:55:42.781250 [ERR] mod_spidermonkey.c:3393 Error loading ODBC 2009-07-20 10:55:42.781250 [ERR] Inbound1.js:5 ReferenceError: ODBC is not defined* * Can some one assist me to resolve this above error. Thanks for reply from Meftah Tayeb,** Michael Jerris*. * **-- Thanks with Regards, N.Baskar * -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090721/fabee107/attachment.html From yehavi.bourvine at gmail.com Mon Jul 20 21:41:45 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Tue, 21 Jul 2009 07:41:45 +0300 Subject: [Freeswitch-users] BLF & Directed call pickup on Polycom phones Message-ID: Hello, I am trying to integrate Polycom phones with a FrewSwitch server, and have some problems with BLF and directed pickup. I've defined a buddy list with BW (buddy watch) on. One of the phone's line buttons (one fo the 3 ones on a Polycom-501 model) is assigned to this buddy and indeed shows its status. I would like to pickup a call to this buddy by pressing its button when his phone rings; however, this generates a second call to him... Using a SNOM phones this works ok. Has anyone managed to make it working with Polycom? Thanks! __Yehavi: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090721/1cb95215/attachment.html From technical at ttnc.co.uk Mon Jul 20 21:55:13 2009 From: technical at ttnc.co.uk (TTNC - Adnan Barakat) Date: Tue, 21 Jul 2009 05:55:13 +0100 Subject: [Freeswitch-users] call fails when using leg_timeout In-Reply-To: <87f2f3b90907201520t210b6f6chd743fa6550377013@mail.gmail.com> References: <4A64538D.6070301@ttnc.co.uk> <191c3a030907200758p24978436kcdbe5259a2e2d865@mail.gmail.com> <4A649FE9.3060104@ttnc.co.uk> <87f2f3b90907201520t210b6f6chd743fa6550377013@mail.gmail.com> Message-ID: <4A654A31.4040508@ttnc.co.uk> Michael Collins wrote: > I just did a test with this syntax and it worked for me. Please try it > and report back. > > data="{ignore_early_media=true}[leg_timeout=20]sofia/internal/123 at 1.2.3.4 > |[leg_timeout=20]sofia/internal/456 at 1.2.3.4 > |[leg_timeout=20]sofia/internal/789 at 1.2.3.4 > "/> Thanks Michael, just tried that, but unfortunately still doesn't work. It seems that there is a hard limit somewhere of 30 sec, I've just tried different timeout values, and it's terminating everytime at 30 sec. Any other ideas? Thanks Adnan From mrene_lists at avgs.ca Mon Jul 20 21:59:35 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Tue, 21 Jul 2009 00:59:35 -0400 Subject: [Freeswitch-users] call fails when using leg_timeout In-Reply-To: <4A654A31.4040508@ttnc.co.uk> References: <4A64538D.6070301@ttnc.co.uk> <191c3a030907200758p24978436kcdbe5259a2e2d865@mail.gmail.com> <4A649FE9.3060104@ttnc.co.uk> <87f2f3b90907201520t210b6f6chd743fa6550377013@mail.gmail.com> <4A654A31.4040508@ttnc.co.uk> Message-ID: It is possible that your inbound carrier applies some timeout rules. Try the following before your bridge: Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca Am 21-Jul-09 um 12:55 AM schrieb TTNC - Adnan Barakat: > Michael Collins wrote: >> I just did a test with this syntax and it worked for me. Please try >> it >> and report back. >> >> > data="{ignore_early_media=true}[leg_timeout=20]sofia/internal/123 at 1.2.3.4 >> |[leg_timeout=20]sofia/internal/456 at 1.2.3.4 >> |[leg_timeout=20]sofia/internal/789 at 1.2.3.4 >> "/> > Thanks Michael, just tried that, but unfortunately still doesn't work. > It seems that there is a hard limit somewhere of 30 sec, I've just > tried > different timeout values, and it's terminating everytime at 30 sec. > > Any other ideas? > > > Thanks > > Adnan > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From rupa at rupa.com Mon Jul 20 22:04:03 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Tue, 21 Jul 2009 00:04:03 -0500 Subject: [Freeswitch-users] Freeswitch Windows Issues In-Reply-To: References: <4A6443B8.5000106@gmail.com> <4A6472D1.6030603@gmail.com> <5E024C30-C98D-4EC0-9E9F-F126FAED0BE3@jerris.com> Message-ID: mod_spidermonkey_odbc is a mod for spidermonkey, not freeswitch. So, you need to enable this mod in the spidermonkey config file, not freeswitch's. This is in conf/spidermonkey.conf.xml. The default has mod_spidermonkey_odbc commented out. On Mon, Jul 20, 2009 at 11:25 PM, Baskar wrote: > *Hi Michael Jerris, > > In linux if ODBC modules to be load in freeswitch i can load by this > command > > make mod_spidermonkey_odbc-install > make install > > But in windows how can i enable the modules for mod_spidermonkey? > > I checked whether mod spidermonkey is loaded by this command load > mod_spidermonkey . > > freeswitch at Baskar>load > mod_spidermonkey > 2009-07-21 09:42:14.859375 [WARNING] switch_loadable_module.c:939 Module > mod_spidermonkey Already **Loaded! ** > > API CALL [load(mod_spidermonkey)] > output:-ERR [Module already loaded] > > **But output say is already loaded but still i get this error**: > > **2009-07-20 10:55:42.781250 [ERR] mod_spidermonkey.c:3393 Error loading > ODBC > 2009-07-20 10:55:42.781250 [ERR] Inbound1.js:5 ReferenceError: ODBC is not > defined* > * > Can some one assist me to resolve this above error. > > Thanks for reply from Meftah Tayeb,** Michael Jerris*. > > * > **-- > Thanks with 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 > > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090721/5edaf5f5/attachment.html From technical at ttnc.co.uk Mon Jul 20 23:06:17 2009 From: technical at ttnc.co.uk (TTNC - Adnan Barakat) Date: Tue, 21 Jul 2009 07:06:17 +0100 Subject: [Freeswitch-users] call fails when using leg_timeout In-Reply-To: References: <4A64538D.6070301@ttnc.co.uk> <191c3a030907200758p24978436kcdbe5259a2e2d865@mail.gmail.com> <4A649FE9.3060104@ttnc.co.uk> <87f2f3b90907201520t210b6f6chd743fa6550377013@mail.gmail.com> <4A654A31.4040508@ttnc.co.uk> Message-ID: <4A655AD9.4050006@ttnc.co.uk> Mathieu Rene wrote: > It is possible that your inbound carrier applies some timeout rules. > Try the following before your bridge: > > Not that I know of, I just tried with ring_ready, and it still doesn't work. Thanks, Adnan From technical at ttnc.co.uk Mon Jul 20 23:25:29 2009 From: technical at ttnc.co.uk (TTNC - Adnan Barakat) Date: Tue, 21 Jul 2009 07:25:29 +0100 Subject: [Freeswitch-users] call fails when using leg_timeout In-Reply-To: <4A655AD9.4050006@ttnc.co.uk> References: <4A64538D.6070301@ttnc.co.uk> <191c3a030907200758p24978436kcdbe5259a2e2d865@mail.gmail.com> <4A649FE9.3060104@ttnc.co.uk> <87f2f3b90907201520t210b6f6chd743fa6550377013@mail.gmail.com> <4A654A31.4040508@ttnc.co.uk> <4A655AD9.4050006@ttnc.co.uk> Message-ID: <4A655F59.6090604@ttnc.co.uk> TTNC - Adnan Barakat wrote: > Mathieu Rene wrote: >> It is possible that your inbound carrier applies some timeout rules. >> Try the following before your bridge: >> >> > Not that I know of, I just tried with ring_ready, and it still doesn't work. Sorry guys, turns out it's a timeout on the VoIP phone I'm using. Thanks Adnan From elihayun at gmail.com Mon Jul 20 23:51:54 2009 From: elihayun at gmail.com (Eli Hayun) Date: Tue, 21 Jul 2009 09:51:54 +0300 Subject: [Freeswitch-users] How to detect that a party is in a conversation? Message-ID: <4A65658A.10009@savion.huji.ac.il> I want to know if the party that I am calling to is in a middle of a conversation. I did not get a busy line because he had more then one line defined. I tried to set max_calls=1 with no luck. any suggestions? Thanx Eli From elihayun at gmail.com Tue Jul 21 01:12:25 2009 From: elihayun at gmail.com (Eli Hayun) Date: Tue, 21 Jul 2009 11:12:25 +0300 Subject: [Freeswitch-users] Getting error without continuation Message-ID: <4A657869.5010202@savion.huji.ac.il> Hi I set continue_on_fail=true but I keep getting error : 2009-07-21 11:00:53.284148 [INFO] mod_dialplan_xml.c:252 Processing phone-1->limit_exceeded in context default 2 instead of continue to the line after the "bridge" What am I doing wrong? Eli From tayeb.meftah at gmail.com Tue Jul 21 01:50:17 2009 From: tayeb.meftah at gmail.com (Meftah Tayeb) Date: Tue, 21 Jul 2009 08:50:17 +0000 Subject: [Freeswitch-users] Freeswitch Windows Issues In-Reply-To: References: <4A6443B8.5000106@gmail.com> <4A6472D1.6030603@gmail.com> <5E024C30-C98D-4EC0-9E9F-F126FAED0BE3@jerris.com> Message-ID: <4A658149.9070407@gmail.com> hello baskar please open: c:\program files\freeswitch\conf\autoload_configs\modules.conf.xml and try to see if the module mod_spidermonkey_odbc is loaded are you using the binary installer (MSI) or you are compiling it? i think if you use the .MSI file the odbc module is not installed and loaded, i don't see it in my modules.conf.xml thanks, Meftah Tayeb DelphiWorld@#Freeswitch Global Voice Communication Baskar wrote: > *Hi Michael Jerris, > > In linux if ODBC modules to be load in freeswitch i can load by this > command > > make mod_spidermonkey_odbc-install > make install > > But in windows how can i enable the modules for mod_spidermonkey? > > I checked whether mod spidermonkey is loaded by this command load > mod_spidermonkey . > > freeswitch at Baskar>load > mod_spidermonkey > > 2009-07-21 09:42:14.859375 [WARNING] switch_loadable_module.c:939 > Module mod_spidermonkey Already > **Loaded! ** > API CALL [load(mod_spidermonkey)] > output:-ERR [Module already loaded] > > **But output say is already loaded but still i get this error**: > > **2009-07-20 10:55:42.781250 [ERR] mod_spidermonkey.c:3393 Error > loading ODBC > 2009-07-20 10:55:42.781250 [ERR] Inbound1.js:5 ReferenceError: ODBC > is not defined* > * > Can some one assist me to resolve this above error. > > Thanks for reply from Meftah Tayeb,** Michael Jerris*. > * > **-- > Thanks with 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 > > > > __________ Information from ESET NOD32 Antivirus, version of virus signature database 4262 (20090720) __________ > > The message was checked by ESET NOD32 Antivirus. > > http://www.eset.com > > __________ Information from ESET NOD32 Antivirus, version of virus signature database 4262 (20090720) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090721/776af1a2/attachment.html From tayeb.meftah at gmail.com Tue Jul 21 02:00:31 2009 From: tayeb.meftah at gmail.com (Meftah Tayeb) Date: Tue, 21 Jul 2009 09:00:31 +0000 Subject: [Freeswitch-users] Freeswitch Windows Issues In-Reply-To: References: <4A6443B8.5000106@gmail.com> <4A6472D1.6030603@gmail.com> <5E024C30-C98D-4EC0-9E9F-F126FAED0BE3@jerris.com> Message-ID: <4A6583AF.5020504@gmail.com> hello baskar, sory, i see it now in spidermonkey.conf.xml line: encommant it and restart your freeswitch thanks Baskar wrote: > *Hi Michael Jerris, > > In linux if ODBC modules to be load in freeswitch i can load by this > command > > make mod_spidermonkey_odbc-install > make install > > But in windows how can i enable the modules for mod_spidermonkey? > > I checked whether mod spidermonkey is loaded by this command load > mod_spidermonkey . > > freeswitch at Baskar>load > mod_spidermonkey > > 2009-07-21 09:42:14.859375 [WARNING] switch_loadable_module.c:939 > Module mod_spidermonkey Already > **Loaded! ** > API CALL [load(mod_spidermonkey)] > output:-ERR [Module already loaded] > > **But output say is already loaded but still i get this error**: > > **2009-07-20 10:55:42.781250 [ERR] mod_spidermonkey.c:3393 Error > loading ODBC > 2009-07-20 10:55:42.781250 [ERR] Inbound1.js:5 ReferenceError: ODBC > is not defined* > * > Can some one assist me to resolve this above error. > > Thanks for reply from Meftah Tayeb,** Michael Jerris*. > * > **-- > Thanks with 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 > > > > __________ Information from ESET NOD32 Antivirus, version of virus signature database 4262 (20090720) __________ > > The message was checked by ESET NOD32 Antivirus. > > http://www.eset.com > > __________ Information from ESET NOD32 Antivirus, version of virus signature database 4262 (20090720) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090721/835f86ce/attachment.html From helmut.kuper at ewetel.de Tue Jul 21 02:52:26 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Tue, 21 Jul 2009 11:52:26 +0200 Subject: [Freeswitch-users] Q931 TE State Timer In-Reply-To: <4A643C7D.7010209@ewetel.de> References: <4A3266E5.2000702@ewetel.de> <4A328E8E.6030607@freeswitch.org> <4A35F0F0.50406@ewetel.de> <4A643C7D.7010209@ewetel.de> Message-ID: <4A658FDA.8080908@ewetel.de> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, I fixed the bug that openzap doesn't send a RELEASE on incomming calls for not registered but existing extensions. Day 3 running the new Q931-TE stack is still successful :) It seems I have still no open calls left after 1000+ of total calls ... regards helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) iD8DBQFKZY/a4tZeNddg3dwRAjYRAJ9fV1BJiJyyyrG2A5BWEUbhJZQ1bgCgkazo G8lw40GrRvfynDmYDZrLUU0= =pZ2r -----END PGP SIGNATURE----- From elihayun at gmail.com Tue Jul 21 03:49:20 2009 From: elihayun at gmail.com (Eli Hayun) Date: Tue, 21 Jul 2009 13:49:20 +0300 Subject: [Freeswitch-users] How to initiate a call without dialing Message-ID: <4A659D30.5020600@savion.huji.ac.il> Is there is a way to initiate a call without making any dial manually? Suppose that I want to initiate a call let say every day at 17:00, is there is a way to do it? Eli From yehavi.bourvine at gmail.com Tue Jul 21 05:22:09 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Tue, 21 Jul 2009 15:22:09 +0300 Subject: [Freeswitch-users] BLF & Directed call pickup on Polycom phones In-Reply-To: References: Message-ID: After playing a little with SNOM phones I see that for doing BLF the SNOM subscribes for number XXXX (the real number), but when I want to pickup a ringing extension it dials **XXXX which is catched by FreeSwitch and handled by the pickup code (probably the intercept function). I would like to mimic this on Polycom phones. Thus, I want the phone to subscribe for *ZXXXX and catch the *Z prefix: - If it is a subscribe command, then strip *Z and subscribe to it. - If this is INVITE and the destination is ringing - strip *Z and and call intercept. - If this is INVITE and the destination is free - ring it. I know roughly how to do the last two items, but how can I catch the SUBSCRIBE, modify the destination number and then call the actual function? Thanks! __Yehavi: 2009/7/21 Yehavi Bourvine > Hello, > > I am trying to integrate Polycom phones with a FrewSwitch server, and > have some problems with BLF and directed pickup. > > I've defined a buddy list with BW (buddy watch) on. One of > the phone's line buttons (one fo the 3 ones on a Polycom-501 model) is > assigned to this buddy and indeed shows its status. I would like to pickup a > call to this buddy by pressing its button when his phone rings; however, > this generates a second call to him... > > Using a SNOM phones this works ok. Has anyone managed to make it working > with Polycom? > > Thanks! __Yehavi: > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090721/3e5ac6bb/attachment.html From javieraristizabal at gmail.com Tue Jul 21 06:19:09 2009 From: javieraristizabal at gmail.com (=?ISO-8859-1?Q?Javier_Aristiz=E1bal?=) Date: Tue, 21 Jul 2009 08:19:09 -0500 Subject: [Freeswitch-users] BLF & Directed call pickup on Polycom phones In-Reply-To: References: Message-ID: I dunno about BLF on Polycom phones. But for the call pickup, check the phone dialplan if permit **XXXX. Javier -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090721/6ed7f48a/attachment.html From regs at kinetix.gr Tue Jul 21 06:29:15 2009 From: regs at kinetix.gr (Apostolos Pantsiopoulos) Date: Tue, 21 Jul 2009 16:29:15 +0300 Subject: [Freeswitch-users] Possible memory leak - need a second opinion In-Reply-To: <4A644EA6.70209@kinetix.gr> References: <4A644EA6.70209@kinetix.gr> Message-ID: <4A65C2AB.5020001@kinetix.gr> Since nobody replied I am posting it to JIRA. Apostolos Pantsiopoulos wrote: > Hi I noticed that after a day of relatively moderate traffic (about 400 > simultaneous channels average) the memory used by FS reached 1.3 GB of > RAM. I tried to trace the leak (if any) with valgrind and got that : > > ==18894== 572 bytes in 1 blocks are definitely lost in loss record 148 > of 193 > ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) > ==18894== by 0x7AFDE93: sofia_glue_do_invite (sofia_glue.c:1599) > ==18894== by 0x7ACB191: sofia_on_init (mod_sofia.c:102) > ==18894== by 0x405B124: switch_core_session_run > (switch_core_state_machine.c:480) > ==18894== by 0x4058204: switch_core_session_thread > (switch_core_session.c:1064) > ==18894== by 0x40CD6C5: dummy_worker (thread.c:138) > ==18894== by 0xB5346A: start_thread (in /lib/libpthread-2.5.so) > ==18894== by 0xAAADBD: clone (in /lib/libc-2.5.so) > > > So, I assummed that this happens for every call. I tried testing it > again by placing two calls before shutting down FS, but it only came up > once. I wanted to get a second opinion before posting this to JIRA as an > issue. > > I used revision 14269 of the SVN. I am attaching the valgrind output as > well. > > I also noticed that only one of my CPU cores gets really busy when > dealing with moderate traffic. From what I read in the mailing list > users are encouraged to use 64bit multi core servers for FS because it > scales up better. But this is not what I am seeing in practice. Could > the single threaded architecture of libsofia be the cause of that behavior? > > > > > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 ------------------------------------------- From yudha2008 at gmail.com Tue Jul 21 06:47:59 2009 From: yudha2008 at gmail.com (Baskar) Date: Tue, 21 Jul 2009 19:17:59 +0530 Subject: [Freeswitch-users] Freeswitch Windows Issues In-Reply-To: <4A6583AF.5020504@gmail.com> References: <4A6443B8.5000106@gmail.com> <4A6472D1.6030603@gmail.com> <5E024C30-C98D-4EC0-9E9F-F126FAED0BE3@jerris.com> <4A6583AF.5020504@gmail.com> Message-ID: *Hi, Problem has been resolved. **Thanks for reply from Meftah Tayeb,** Michael Jerris, Rupa Schomaker and freeswitch-users. -- Thanks with Regards, N.Baskar * -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090721/fcae9e60/attachment.html From brian at freeswitch.org Tue Jul 21 06:55:15 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 21 Jul 2009 08:55:15 -0500 Subject: [Freeswitch-users] Possible memory leak - need a second opinion In-Reply-To: <4A65C2AB.5020001@kinetix.gr> References: <4A644EA6.70209@kinetix.gr> <4A65C2AB.5020001@kinetix.gr> Message-ID: <0238ECAC-FC98-42AA-8824-B90FB6232520@freeswitch.org> If you're gonna open a jira we'll need a sipp scenario file that will reproduce the issue along with sip traces that can show what is going on. /b On Jul 21, 2009, at 8:29 AM, Apostolos Pantsiopoulos wrote: > Since nobody replied I am posting it to JIRA. > > Apostolos Pantsiopoulos wrote: >> Hi I noticed that after a day of relatively moderate traffic (about >> 400 >> simultaneous channels average) the memory used by FS reached 1.3 GB >> of >> RAM. I tried to trace the leak (if any) with valgrind and got that : >> >> ==18894== 572 bytes in 1 blocks are definitely lost in loss record >> 148 >> of 193 >> ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) >> ==18894== by 0x7AFDE93: sofia_glue_do_invite (sofia_glue.c:1599) >> ==18894== by 0x7ACB191: sofia_on_init (mod_sofia.c:102) >> ==18894== by 0x405B124: switch_core_session_run >> (switch_core_state_machine.c:480) >> ==18894== by 0x4058204: switch_core_session_thread >> (switch_core_session.c:1064) >> ==18894== by 0x40CD6C5: dummy_worker (thread.c:138) >> ==18894== by 0xB5346A: start_thread (in /lib/libpthread-2.5.so) >> ==18894== by 0xAAADBD: clone (in /lib/libc-2.5.so) >> >> >> So, I assummed that this happens for every call. I tried testing it >> again by placing two calls before shutting down FS, but it only >> came up >> once. I wanted to get a second opinion before posting this to JIRA >> as an >> issue. >> >> I used revision 14269 of the SVN. I am attaching the valgrind >> output as >> well. >> >> I also noticed that only one of my CPU cores gets really busy when >> dealing with moderate traffic. From what I read in the mailing list >> users are encouraged to use 64bit multi core servers for FS because >> it >> scales up better. But this is not what I am seeing in practice. Could >> the single threaded architecture of libsofia be the cause of that >> behavior? >> >> >> >> >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 From intralanman at freeswitch.org Tue Jul 21 06:56:33 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Tue, 21 Jul 2009 09:56:33 -0400 Subject: [Freeswitch-users] How to initiate a call without dialing In-Reply-To: <4A659D30.5020600@savion.huji.ac.il> References: <4A659D30.5020600@savion.huji.ac.il> Message-ID: <4A65C911.50707@freeswitch.org> Eli Hayun wrote: > Is there is a way to initiate a call without making any dial manually? > i think the api command "originate" is what you're looking for -Ray From helmut.kuper at ewetel.de Tue Jul 21 07:01:54 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Tue, 21 Jul 2009 16:01:54 +0200 Subject: [Freeswitch-users] Question: Capturing VoiceData received from Sagnoma E1 card Message-ID: <4A65CA52.90002@ewetel.de> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, For outgoing calls I'm hunting the cause for missing some 100ms of voice data send from remote right after pickup the remote phone (e.g. initial "Hello?" sound like "o?" or even nothing) On FreeSwitch server I captured the VoIP data to the called VoIP-Phone on the sofia interface. Using wireshark it also shows that the voice data from remote is missed. Using Mobil phones or ISDN phones calling the same remote party there is never a bit missed. This problem occurs rare - once or twice per day and per local voip phone, but it's quite anoying. So is there a way to capture the correspondig ISDN voice data FS receives before it is transmitted via RTP or just droped? I want to c whether FS drops the early RTP packets or whether FS never got the data from ISDN. Sofia Profile is using The dialplan portion is: Any ideas to refine my debugging? regard helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) iD8DBQFKZcpS4tZeNddg3dwRAnVDAKCxXXkdbf0RKeeSMFYucCIno3tA9gCfUzbD 148BfuKavTtBoJNScRQDmSk= =JbtY -----END PGP SIGNATURE----- From rupa at rupa.com Tue Jul 21 07:13:34 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Tue, 21 Jul 2009 09:13:34 -0500 Subject: [Freeswitch-users] Getting error without continuation In-Reply-To: <4A657869.5010202@savion.huji.ac.il> References: <4A657869.5010202@savion.huji.ac.il> Message-ID: Didn't you just set max_calls=1? Maybe takes that out so you can handle more than 1 call.... On Tue, Jul 21, 2009 at 3:12 AM, Eli Hayun wrote: > Hi > I set continue_on_fail=true but I keep getting error : > 2009-07-21 11:00:53.284148 [INFO] mod_dialplan_xml.c:252 Processing > phone-1->limit_exceeded in context default > 2 > instead of continue to the line after the "bridge" > What am I doing wrong? > > Eli > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090721/01cfdaf1/attachment.html From vkozak at abisoft.spb.ru Tue Jul 21 01:46:03 2009 From: vkozak at abisoft.spb.ru (Kozak Vladimir) Date: Tue, 21 Jul 2009 12:46:03 +0400 Subject: [Freeswitch-users] Bridge command with domain, using FS outbound connection API not work [NORMAL_TEMPORARY_FAILURE]. Message-ID: Hello, Problem with bridge command using FS outbound connection API. Configuration: 1. X-lite: auth/disp/user name: 1001; domain: rantipin.starpoundtech.net; Register with domain and receive incomming calls: true; proxy: 172.26.200.250:5060 2 eyeBeam: auth/disp/user name: 1000; domain: master.agent.rantipin.starpoundtech.net; Register with domain and receive incomming calls: true; proxy: 172.26.200.250:5060 I start nc -v -l 127.0.0.1 -p 8084 Make call from eyeBeam, recive connected event. I execute commands at nc: 1. sendmsg call-command: execute execute-app-name: bridge execute-app-arg: sofia/internal/1001 at 172.26.200.250 - OK, work. 2. sendmsg call-command: execute execute-app-name: bridge execute-app-arg: sofia/internal/1001 at rantipin.starpoundtech.net - Not work. nc response: Content-Type: command/reply Reply-Text: +OK Content-Type: text/disconnect-notice Controlled-Session-UUID: a7e94e62-2c4b-48ba-81c2-019300b420d6 Content-Disposition: disconnect FS log: 2009-07-21 19:12:24 [DEBUG] switch_core_session.c:706 switch_core_session_queue_private_event() Send signal sofia/internal/1000 at master.agent.rantipin.starpoundtech.net [BREAK] 2009-07-21 19:12:24 [DEBUG] switch_ivr.c:540 switch_ivr_parse_event() sofia/internal/1000 at master.agent.rantipin.starpoundtech.net Command Execute bridge(sofia/internal/1001 at rantipin.starpoundtech.net) 2009-07-21 19:12:24 [NOTICE] switch_channel.c:567 switch_channel_set_name() New Channel sofia/internal/1001 at rantipin.starpoundtech.net [84861d3f-88a5-49c7-82cc-52f7580c44a8] 2009-07-21 19:12:24 [DEBUG] mod_sofia.c:2535 sofia_outgoing_channel() (sofia/internal/1001 at rantipin.starpoundtech.net) State Change CS_NEW -> CS_INIT 2009-07-21 19:12:24 [DEBUG] switch_core_session.c:820 switch_core_session_signal_state_change() Send signal sofia/internal/1001 at rantipin.starpoundtech.net [BREAK] 2009-07-21 19:12:24 [DEBUG] switch_core_state_machine.c:383 switch_core_session_run() (sofia/internal/1001 at rantipin.starpoundtech.net) Running State Change CS_INIT 2009-07-21 19:12:24 [DEBUG] switch_core_state_machine.c:454 switch_core_session_run() (sofia/internal/1001 at rantipin.starpoundtech.net) State INIT 2009-07-21 19:12:24 [DEBUG] mod_sofia.c:83 sofia_on_init() sofia/internal/1001 at rantipin.starpoundtech.net SOFIA INIT 2009-07-21 19:12:24 [DEBUG] mod_sofia.c:111 sofia_on_init() (sofia/internal/1001 at rantipin.starpoundtech.net) State Change CS_INIT -> CS_ROUTING 2009-07-21 19:12:24 [DEBUG] switch_core_session.c:820 switch_core_session_signal_state_change() Send signal sofia/internal/1001 at rantipin.starpoundtech.net [BREAK] 2009-07-21 19:12:24 [DEBUG] switch_core_state_machine.c:454 switch_core_session_run() (sofia/internal/1001 at rantipin.starpoundtech.net) State INIT going to sleep 2009-07-21 19:12:24 [DEBUG] switch_core_state_machine.c:383 switch_core_session_run() (sofia/internal/1001 at rantipin.starpoundtech.net) Running State Change CS_ROUTING 2009-07-21 19:12:24 [DEBUG] switch_core_state_machine.c:457 switch_core_session_run() (sofia/internal/1001 at rantipin.starpoundtech.net) State ROUTING 2009-07-21 19:12:24 [DEBUG] mod_sofia.c:130 sofia_on_routing() sofia/internal/1001 at rantipin.starpoundtech.net SOFIA ROUTING 2009-07-21 19:12:24 [DEBUG] switch_ivr_originate.c:63 originate_on_routing() (sofia/internal/1001 at rantipin.starpoundtech.net) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2009-07-21 19:12:24 [DEBUG] switch_core_session.c:820 switch_core_session_signal_state_change() Send signal sofia/internal/1001 at rantipin.starpoundtech.net [BREAK] 2009-07-21 19:12:24 [DEBUG] switch_core_state_machine.c:457 switch_core_session_run() (sofia/internal/1001 at rantipin.starpoundtech.net) State ROUTING going to sleep 2009-07-21 19:12:24 [DEBUG] switch_core_state_machine.c:383 switch_core_session_run() (sofia/internal/1001 at rantipin.starpoundtech.net) Running State Change CS_CONSUME_MEDIA 2009-07-21 19:12:24 [DEBUG] switch_core_state_machine.c:476 switch_core_session_run() (sofia/internal/1001 at rantipin.starpoundtech.net) State CONSUME_MEDIA 2009-07-21 19:12:24 [DEBUG] sofia.c:2728 sofia_handle_sip_i_state() Channel sofia/internal/1001 at rantipin.starpoundtech.net entering state [calling] 2009-07-21 19:12:24 [DEBUG] sofia.c:2728 sofia_handle_sip_i_state() Channel sofia/internal/1001 at rantipin.starpoundtech.net entering state [terminated] 2009-07-21 19:12:24 [NOTICE] sofia.c:3279 sofia_handle_sip_i_state() Hangup sofia/internal/1001 at rantipin.starpoundtech.net [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE] 2009-07-21 19:12:24 [DEBUG] switch_channel.c:1566 switch_channel_perform_hangup() Send signal sofia/internal/1001 at rantipin.starpoundtech.net [KILL] 2009-07-21 19:12:24 [DEBUG] switch_core_session.c:820 switch_core_session_signal_state_change() Send signal sofia/internal/1001 at rantipin.starpoundtech.net [BREAK] 2009-07-21 19:12:24 [DEBUG] switch_core_state_machine.c:476 switch_core_session_run() (sofia/internal/1001 at rantipin.starpoundtech.net) State CONSUME_MEDIA going to sleep 2009-07-21 19:12:24 [DEBUG] switch_core_state_machine.c:383 switch_core_session_run() (sofia/internal/1001 at rantipin.starpoundtech.net) Running State Change CS_HANGUP 2009-07-21 19:12:24 [DEBUG] switch_core_state_machine.c:414 switch_core_session_run() (sofia/internal/1001 at rantipin.starpoundtech.net) State HANGUP 2009-07-21 19:12:24 [DEBUG] mod_sofia.c:253 sofia_on_hangup() sofia/internal/1001 at rantipin.starpoundtech.net Overriding SIP cause 503 with 503 from the other leg 2009-07-21 19:12:24 [DEBUG] mod_sofia.c:287 sofia_on_hangup() Channel sofia/internal/1001 at rantipin.starpoundtech.net hanging up, cause: NORMAL_TEMPORARY_FAILURE 2009-07-21 19:12:24 [DEBUG] switch_core_state_machine.c:46 switch_core_standard_on_hangup() sofia/internal/1001 at rantipin.starpoundtech.net Standard HANGUP, cause: NORMAL_TEMPORARY_FAILURE 2009-07-21 19:12:24 [DEBUG] switch_core_state_machine.c:414 switch_core_session_run() (sofia/internal/1001 at rantipin.starpoundtech.net) State HANGUP going to sleep 2009-07-21 19:12:24 [DEBUG] switch_core_session.c:952 switch_core_session_thread() Session 2 (sofia/internal/1001 at rantipin.starpoundtech.net) Locked, Waiting on external entities 2009-07-21 19:12:24 [DEBUG] switch_ivr_originate.c:2014 switch_ivr_originate() Originate Resulted in Error Cause: 41 [NORMAL_TEMPORARY_FAILURE] 2009-07-21 19:12:24 [INFO] mod_dptools.c:1998 audio_bridge_function() Originate Failed. Cause: NORMAL_TEMPORARY_FAILURE 2009-07-21 19:12:24 [NOTICE] mod_dptools.c:2030 audio_bridge_function() Hangup sofia/internal/1000 at master.agent.rantipin.starpoundtech.net [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE] 2009-07-21 19:12:24 [DEBUG] switch_channel.c:1566 switch_channel_perform_hangup() Send signal sofia/internal/1000 at master.agent.rantipin.starpoundtech.net [KILL] 2009-07-21 19:12:24 [DEBUG] switch_core_session.c:820 switch_core_session_signal_state_change() Send signal sofia/internal/1000 at master.agent.rantipin.starpoundtech.net [BREAK] 2009-07-21 19:12:24 [DEBUG] switch_core_state_machine.c:464 switch_core_session_run() (sofia/internal/1000 at master.agent.rantipin.starpoundtech.net) State EXECUTE going to sleep 2009-07-21 19:12:24 [DEBUG] switch_core_state_machine.c:383 switch_core_session_run() (sofia/internal/1000 at master.agent.rantipin.starpoundtech.net) Running State Change CS_HANGUP 2009-07-21 19:12:24 [DEBUG] switch_core_state_machine.c:414 switch_core_session_run() (sofia/internal/1000 at master.agent.rantipin.starpoundtech.net) State HANGUP 2009-07-21 19:12:24 [DEBUG] mod_sofia.c:253 sofia_on_hangup() sofia/internal/1000 at master.agent.rantipin.starpoundtech.net Overriding SIP cause 503 with 503 from the other leg 2009-07-21 19:12:24 [DEBUG] mod_sofia.c:287 sofia_on_hangup() Channel sofia/internal/1000 at master.agent.rantipin.starpoundtech.net hanging up, cause: NORMAL_TEMPORARY_FAILURE 2009-07-21 19:12:24 [DEBUG] mod_sofia.c:361 sofia_on_hangup() Responding to INVITE with: 503 2009-07-21 19:12:24 [DEBUG] switch_core_state_machine.c:46 switch_core_standard_on_hangup() sofia/internal/1000 at master.agent.rantipin.starpoundtech.net Standard HANGUP, cause: NORMAL_TEMPORARY_FAILURE 2009-07-21 19:12:24 [DEBUG] switch_core_state_machine.c:414 switch_core_session_run() (sofia/internal/1000 at master.agent.rantipin.starpoundtech.net) State HANGUP going to sleep 2009-07-21 19:12:24 [DEBUG] switch_core_session.c:952 switch_core_session_thread() Session 1 (sofia/internal/1000 at master.agent.rantipin.starpoundtech.net) Locked, Waiting on external entities 2009-07-21 19:12:24 [NOTICE] switch_core_session.c:970 switch_core_session_thread() Session 2 (sofia/internal/1001 at rantipin.starpoundtech.net) Ended 2009-07-21 19:12:24 [NOTICE] switch_core_session.c:972 switch_core_session_thread() Close Channel sofia/internal/1001 at rantipin.starpoundtech.net [CS_HANGUP] 2009-07-21 19:12:24 [DEBUG] mod_event_socket.c:1979 listener_run() Session complete, waiting for children 2009-07-21 19:12:24 [DEBUG] mod_event_socket.c:2012 listener_run() Connection Closed 2009-07-21 19:12:24 [NOTICE] switch_core_session.c:970 switch_core_session_thread() Session 1 (sofia/internal/1000 at master.agent.rantipin.starpoundtech.net) Ended 2009-07-21 19:12:24 [NOTICE] switch_core_session.c:972 switch_core_session_thread() Close Channel sofia/internal/1000 at master.agent.rantipin.starpoundtech.net [CS_HANGUP] What can be not correct? Thank you. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090721/60728d9c/attachment.html From brian at freeswitch.org Tue Jul 21 08:03:54 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 21 Jul 2009 10:03:54 -0500 Subject: [Freeswitch-users] Bridge command with domain, using FS outbound connection API not work [NORMAL_TEMPORARY_FAILURE]. In-Reply-To: References: Message-ID: <74240E91-6797-4D03-80D4-A5F9EBF77853@freeswitch.org> http://wiki.freeswitch.org/wiki/Dialplan_XML#SIP-Specific_Dialstrings "Dialing A Registered User" is what you should refer to /b On Jul 21, 2009, at 3:46 AM, Kozak Vladimir wrote: > Hello, > > Problem with bridge command using FS outbound connection API. > Configuration: > 1. X-lite: auth/disp/user name: 1001; domain: > rantipin.starpoundtech.net; Register with domain and receive > incomming calls: true; proxy: 172.26.200.250:5060 > 2 eyeBeam: auth/disp/user name: 1000; domain: > master.agent.rantipin.starpoundtech.net; Register with domain and > receive incomming calls: true; proxy: 172.26.200.250:5060 > > I start nc -v -l 127.0.0.1 -p 8084 > Make call from eyeBeam, recive connected event. > > I execute commands at nc: > 1. > sendmsg > call-command: execute > execute-app-name: bridge > execute-app-arg: sofia/internal/1001 at 172.26.200.250 > > - OK, work. > > 2. > sendmsg > call-command: execute > execute-app-name: bridge > execute-app-arg: sofia/internal/1001 at rantipin.starpoundtech.net > > - Not work. > > nc response: > > Content-Type: command/reply > Reply-Text: +OK > > Content-Type: text/disconnect-notice > Controlled-Session-UUID: a7e94e62-2c4b-48ba-81c2-019300b420d6 > Content-Disposition: disconnect > > FS log: > > 2009-07-21 19:12:24 [DEBUG] switch_core_session.c:706 > switch_core_session_queue_private_event() Send signal sofia/internal/1000 at master.agent.rantipin.starpoundtech.net > [BREAK] > 2009-07-21 19:12:24 [DEBUG] switch_ivr.c:540 > switch_ivr_parse_event() sofia/internal/1000 at master.agent.rantipin.starpoundtech.net > Command Execute bridge(sofia/internal/ > 1001 at rantipin.starpoundtech.net) > 2009-07-21 19:12:24 [NOTICE] switch_channel.c:567 > switch_channel_set_name() New Channel sofia/internal/1001 at rantipin.starpoundtech.net > [84861d3f-88a5-49c7-82cc-52f7580c44a8] > 2009-07-21 19:12:24 [DEBUG] mod_sofia.c:2535 > sofia_outgoing_channel() (sofia/internal/1001 at rantipin.starpoundtech.net > ) State Change CS_NEW -> CS_INIT > 2009-07-21 19:12:24 [DEBUG] switch_core_session.c:820 > switch_core_session_signal_state_change() Send signal sofia/internal/1001 at rantipin.starpoundtech.net > [BREAK] > 2009-07-21 19:12:24 [DEBUG] switch_core_state_machine.c:383 > switch_core_session_run() (sofia/internal/1001 at rantipin.starpoundtech.net > ) Running State Change CS_INIT > 2009-07-21 19:12:24 [DEBUG] switch_core_state_machine.c:454 > switch_core_session_run() (sofia/internal/1001 at rantipin.starpoundtech.net > ) State INIT > 2009-07-21 19:12:24 [DEBUG] mod_sofia.c:83 sofia_on_init() sofia/internal/1001 at rantipin.starpoundtech.net > SOFIA INIT > 2009-07-21 19:12:24 [DEBUG] mod_sofia.c:111 sofia_on_init() (sofia/internal/1001 at rantipin.starpoundtech.net > ) State Change CS_INIT -> CS_ROUTING > 2009-07-21 19:12:24 [DEBUG] switch_core_session.c:820 > switch_core_session_signal_state_change() Send signal sofia/internal/1001 at rantipin.starpoundtech.net > [BREAK] > 2009-07-21 19:12:24 [DEBUG] switch_core_state_machine.c:454 > switch_core_session_run() (sofia/internal/1001 at rantipin.starpoundtech.net > ) State INIT going to sleep > 2009-07-21 19:12:24 [DEBUG] switch_core_state_machine.c:383 > switch_core_session_run() (sofia/internal/1001 at rantipin.starpoundtech.net > ) Running State Change CS_ROUTING > 2009-07-21 19:12:24 [DEBUG] switch_core_state_machine.c:457 > switch_core_session_run() (sofia/internal/1001 at rantipin.starpoundtech.net > ) State ROUTING > 2009-07-21 19:12:24 [DEBUG] mod_sofia.c:130 sofia_on_routing() sofia/internal/1001 at rantipin.starpoundtech.net > SOFIA ROUTING > 2009-07-21 19:12:24 [DEBUG] switch_ivr_originate.c:63 > originate_on_routing() (sofia/internal/ > 1001 at rantipin.starpoundtech.net) State Change CS_ROUTING -> > CS_CONSUME_MEDIA > 2009-07-21 19:12:24 [DEBUG] switch_core_session.c:820 > switch_core_session_signal_state_change() Send signal sofia/internal/1001 at rantipin.starpoundtech.net > [BREAK] > 2009-07-21 19:12:24 [DEBUG] switch_core_state_machine.c:457 > switch_core_session_run() (sofia/internal/1001 at rantipin.starpoundtech.net > ) State ROUTING going to sleep > 2009-07-21 19:12:24 [DEBUG] switch_core_state_machine.c:383 > switch_core_session_run() (sofia/internal/1001 at rantipin.starpoundtech.net > ) Running State Change CS_CONSUME_MEDIA > 2009-07-21 19:12:24 [DEBUG] switch_core_state_machine.c:476 > switch_core_session_run() (sofia/internal/1001 at rantipin.starpoundtech.net > ) State CONSUME_MEDIA > 2009-07-21 19:12:24 [DEBUG] sofia.c:2728 sofia_handle_sip_i_state() > Channel sofia/internal/1001 at rantipin.starpoundtech.net entering > state [calling] > 2009-07-21 19:12:24 [DEBUG] sofia.c:2728 sofia_handle_sip_i_state() > Channel sofia/internal/1001 at rantipin.starpoundtech.net entering > state [terminated] > 2009-07-21 19:12:24 [NOTICE] sofia.c:3279 sofia_handle_sip_i_state() > Hangup sofia/internal/1001 at rantipin.starpoundtech.net > [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE] > 2009-07-21 19:12:24 [DEBUG] switch_channel.c:1566 > switch_channel_perform_hangup() Send signal sofia/internal/1001 at rantipin.starpoundtech.net > [KILL] > 2009-07-21 19:12:24 [DEBUG] switch_core_session.c:820 > switch_core_session_signal_state_change() Send signal sofia/internal/1001 at rantipin.starpoundtech.net > [BREAK] > 2009-07-21 19:12:24 [DEBUG] switch_core_state_machine.c:476 > switch_core_session_run() (sofia/internal/1001 at rantipin.starpoundtech.net > ) State CONSUME_MEDIA going to sleep > 2009-07-21 19:12:24 [DEBUG] switch_core_state_machine.c:383 > switch_core_session_run() (sofia/internal/1001 at rantipin.starpoundtech.net > ) Running State Change CS_HANGUP > 2009-07-21 19:12:24 [DEBUG] switch_core_state_machine.c:414 > switch_core_session_run() (sofia/internal/1001 at rantipin.starpoundtech.net > ) State HANGUP > 2009-07-21 19:12:24 [DEBUG] mod_sofia.c:253 sofia_on_hangup() sofia/internal/1001 at rantipin.starpoundtech.net > Overriding SIP cause 503 with 503 from the other leg > 2009-07-21 19:12:24 [DEBUG] mod_sofia.c:287 sofia_on_hangup() > Channel sofia/internal/1001 at rantipin.starpoundtech.net hanging up, > cause: NORMAL_TEMPORARY_FAILURE > 2009-07-21 19:12:24 [DEBUG] switch_core_state_machine.c:46 > switch_core_standard_on_hangup() sofia/internal/1001 at rantipin.starpoundtech.net > Standard HANGUP, cause: NORMAL_TEMPORARY_FAILURE > 2009-07-21 19:12:24 [DEBUG] switch_core_state_machine.c:414 > switch_core_session_run() (sofia/internal/1001 at rantipin.starpoundtech.net > ) State HANGUP going to sleep > 2009-07-21 19:12:24 [DEBUG] switch_core_session.c:952 > switch_core_session_thread() Session 2 (sofia/internal/1001 at rantipin.starpoundtech.net > ) Locked, Waiting on external entities > 2009-07-21 19:12:24 [DEBUG] switch_ivr_originate.c:2014 > switch_ivr_originate() Originate Resulted in Error Cause: 41 > [NORMAL_TEMPORARY_FAILURE] > 2009-07-21 19:12:24 [INFO] mod_dptools.c:1998 > audio_bridge_function() Originate Failed. Cause: > NORMAL_TEMPORARY_FAILURE > 2009-07-21 19:12:24 [NOTICE] mod_dptools.c:2030 > audio_bridge_function() Hangup sofia/internal/1000 at master.agent.rantipin.starpoundtech.net > [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE] > 2009-07-21 19:12:24 [DEBUG] switch_channel.c:1566 > switch_channel_perform_hangup() Send signal sofia/internal/1000 at master.agent.rantipin.starpoundtech.net > [KILL] > 2009-07-21 19:12:24 [DEBUG] switch_core_session.c:820 > switch_core_session_signal_state_change() Send signal sofia/internal/1000 at master.agent.rantipin.starpoundtech.net > [BREAK] > 2009-07-21 19:12:24 [DEBUG] switch_core_state_machine.c:464 > switch_core_session_run() (sofia/internal/1000 at master.agent.rantipin.starpoundtech.net > ) State EXECUTE going to sleep > 2009-07-21 19:12:24 [DEBUG] switch_core_state_machine.c:383 > switch_core_session_run() (sofia/internal/1000 at master.agent.rantipin.starpoundtech.net > ) Running State Change CS_HANGUP > 2009-07-21 19:12:24 [DEBUG] switch_core_state_machine.c:414 > switch_core_session_run() (sofia/internal/1000 at master.agent.rantipin.starpoundtech.net > ) State HANGUP > 2009-07-21 19:12:24 [DEBUG] mod_sofia.c:253 sofia_on_hangup() sofia/internal/1000 at master.agent.rantipin.starpoundtech.net > Overriding SIP cause 503 with 503 from the other leg > 2009-07-21 19:12:24 [DEBUG] mod_sofia.c:287 sofia_on_hangup() > Channel sofia/internal/1000 at master.agent.rantipin.starpoundtech.net > hanging up, cause: NORMAL_TEMPORARY_FAILURE > 2009-07-21 19:12:24 [DEBUG] mod_sofia.c:361 sofia_on_hangup() > Responding to INVITE with: 503 > 2009-07-21 19:12:24 [DEBUG] switch_core_state_machine.c:46 > switch_core_standard_on_hangup() sofia/internal/1000 at master.agent.rantipin.starpoundtech.net > Standard HANGUP, cause: NORMAL_TEMPORARY_FAILURE > 2009-07-21 19:12:24 [DEBUG] switch_core_state_machine.c:414 > switch_core_session_run() (sofia/internal/1000 at master.agent.rantipin.starpoundtech.net > ) State HANGUP going to sleep > 2009-07-21 19:12:24 [DEBUG] switch_core_session.c:952 > switch_core_session_thread() Session 1 (sofia/internal/1000 at master.agent.rantipin.starpoundtech.net > ) Locked, Waiting on external entities > 2009-07-21 19:12:24 [NOTICE] switch_core_session.c:970 > switch_core_session_thread() Session 2 (sofia/internal/1001 at rantipin.starpoundtech.net > ) Ended > 2009-07-21 19:12:24 [NOTICE] switch_core_session.c:972 > switch_core_session_thread() Close Channel sofia/internal/1001 at rantipin.starpoundtech.net > [CS_HANGUP] > 2009-07-21 19:12:24 [DEBUG] mod_event_socket.c:1979 listener_run() > Session complete, waiting for children > 2009-07-21 19:12:24 [DEBUG] mod_event_socket.c:2012 listener_run() > Connection Closed > 2009-07-21 19:12:24 [NOTICE] switch_core_session.c:970 > switch_core_session_thread() Session 1 (sofia/internal/1000 at master.agent.rantipin.starpoundtech.net > ) Ended > 2009-07-21 19:12:24 [NOTICE] switch_core_session.c:972 > switch_core_session_thread() Close Channel sofia/internal/1000 at master.agent.rantipin.starpoundtech.net > [CS_HANGUP] > > What can be not correct? > 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 msc at freeswitch.org Tue Jul 21 08:29:42 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 21 Jul 2009 08:29:42 -0700 Subject: [Freeswitch-users] Q931 TE State Timer In-Reply-To: <4A658FDA.8080908@ewetel.de> References: <4A3266E5.2000702@ewetel.de> <4A328E8E.6030607@freeswitch.org> <4A35F0F0.50406@ewetel.de> <4A643C7D.7010209@ewetel.de> <4A658FDA.8080908@ewetel.de> Message-ID: <87f2f3b90907210829o38f18c23xf893f88fd4690ba5@mail.gmail.com> Helmut, This is wonderful news. Please keep up the good work. -MC On Tue, Jul 21, 2009 at 2:52 AM, Helmut Kuper wrote: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > Hello, > > I fixed the bug that openzap doesn't send a RELEASE on incomming calls > for not registered but existing extensions. > > Day 3 running the new Q931-TE stack is still successful :) It seems I > have still no open calls left after 1000+ of total calls ... > > regards > helmut > > -----BEGIN PGP SIGNATURE----- > Version: GnuPG v1.4.7 (MingW32) > > iD8DBQFKZY/a4tZeNddg3dwRAjYRAJ9fV1BJiJyyyrG2A5BWEUbhJZQ1bgCgkazo > G8lw40GrRvfynDmYDZrLUU0= > =pZ2r > -----END PGP SIGNATURE----- > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20090721/3959200c/attachment.html From nicolas at medularis.com Tue Jul 21 08:35:00 2009 From: nicolas at medularis.com (Nicolas Brenner) Date: Tue, 21 Jul 2009 11:35:00 -0400 Subject: [Freeswitch-users] Best way to bridge 2 calls with LCR? Message-ID: <1b46b4e80907210835s6e516e2ft84f3d7329eee1535@mail.gmail.com> I would like to originate 2 calls from FS and then bridge them. There's no incoming call so I think there's no dialplan involved. What I'd like to do now is apply lcr rules to these calls. I've come up with 2 options so far: 1) call lcr through the socket twice (once for each phonenumber) and then originate the calls through the socket too 2) have a javascript file which runs the actions above, run the script through the socket with 'jsrun' How would you do it? For what I've read on the list, usually the recommended way is to stay away from javascript as much as possible because it is not as efficient as doing everything from the dialplan. Does this mean the first option is the best? or is there a "dialplan way" of doing it? Thank you very much for your help! Nicolas -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090721/793f2c18/attachment.html From rupa at rupa.com Tue Jul 21 08:43:26 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Tue, 21 Jul 2009 10:43:26 -0500 Subject: [Freeswitch-users] Best way to bridge 2 calls with LCR? In-Reply-To: <1b46b4e80907210835s6e516e2ft84f3d7329eee1535@mail.gmail.com> References: <1b46b4e80907210835s6e516e2ft84f3d7329eee1535@mail.gmail.com> Message-ID: lcr api command doesn't really return a usable dialstring (it was originally done for debug purposes). I could add an "as xml" option if needed... Anyway, to do this from the dialplan: remember that originate's usage is: -USAGE |&() [] [] [] [] [] so, the first argument is the call url and the second would be an extension. so: 1) execute lcr for the first leg of the call 2) execute originate with: originate ${lcr_auto_route} extension extension just needs to match something in your dialplan. In extension, you'd do another lcr lookup and then bridge to that leg's ${lcr_auto_route} value. On Tue, Jul 21, 2009 at 10:35 AM, Nicolas Brenner wrote: > I would like to originate 2 calls from FS and then bridge them. There's no > incoming call so I think there's no dialplan involved. > What I'd like to do now is apply lcr rules to these calls. I've come up > with 2 options so far: > > 1) call lcr through the socket twice (once for each phonenumber) and then > originate the calls through the socket too > 2) have a javascript file which runs the actions above, run the script > through the socket with 'jsrun' > > How would you do it? > > For what I've read on the list, usually the recommended way is to stay away > from javascript as much as possible because it is not as efficient as doing > everything from the dialplan. Does this mean the first option is the best? > or is there a "dialplan way" of doing it? > > Thank you very much for your help! > > Nicolas > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090721/2ade0ec8/attachment.html From marketing at cluecon.com Tue Jul 21 09:15:56 2009 From: marketing at cluecon.com (Michael Collins) Date: Tue, 21 Jul 2009 09:15:56 -0700 Subject: [Freeswitch-users] ClueCon 2009 - GREAT NEWS - Hotel and early registration extended through July 27! Message-ID: <87f2f3b90907210915i2c98d072nb784c9f1676d04b3@mail.gmail.com> Hello everyone! We are happy to announce that the ClueCon team has been able to extend the early bird registration price AND the hotel room reservations through Monday July 27th. This is important because, outside of our ClueCon block of rooms, the Wyndham is completely filled up. Please make your hotel reservations right away. If you haven't secured your ClueCon registration yet then please call 877.742.CLUE and we will get you set up with the early bird special of $499. Don't delay! ClueCon 2009 starts two weeks from today. It will be here before you know it. See you all in Chicago! -Michael http://www.cluecon.com 877.742.CLUE -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090721/5ba1b846/attachment.html From nicolas at medularis.com Tue Jul 21 09:27:07 2009 From: nicolas at medularis.com (Nicolas Brenner) Date: Tue, 21 Jul 2009 12:27:07 -0400 Subject: [Freeswitch-users] Best way to bridge 2 calls with LCR? In-Reply-To: References: <1b46b4e80907210835s6e516e2ft84f3d7329eee1535@mail.gmail.com> Message-ID: <1b46b4e80907210927n4885b4c5g180a06e32bedc7c4@mail.gmail.com> That looks like a good way to go about it. How can I access channel variables through the socket using the api? I mean, how do I recover the value of ${lcr_auto_route}? I would need to add some other variables, like ignore_early_media=true and a uuid that 'links' the two calls so I can track it listening for events. Thanks! Nicolas On Tue, Jul 21, 2009 at 11:43 AM, Rupa Schomaker wrote: > lcr api command doesn't really return a usable dialstring (it was > originally done for debug purposes). I could add an "as xml" option if > needed... > > Anyway, to do this from the dialplan: > > remember that originate's usage is: > > -USAGE |&() [] > [] [] [] [] > > so, the first argument is the call url and the second would be an > extension. so: > > 1) execute lcr for the first leg of the call > 2) execute originate with: > > originate ${lcr_auto_route} extension > > extension just needs to match something in your dialplan. > > In extension, you'd do another lcr lookup and then bridge to that leg's > ${lcr_auto_route} value. > > > > On Tue, Jul 21, 2009 at 10:35 AM, Nicolas Brenner wrote: > >> I would like to originate 2 calls from FS and then bridge them. There's no >> incoming call so I think there's no dialplan involved. >> What I'd like to do now is apply lcr rules to these calls. I've come up >> with 2 options so far: >> >> 1) call lcr through the socket twice (once for each phonenumber) and then >> originate the calls through the socket too >> 2) have a javascript file which runs the actions above, run the script >> through the socket with 'jsrun' >> >> How would you do it? >> >> For what I've read on the list, usually the recommended way is to stay >> away from javascript as much as possible because it is not as efficient as >> doing everything from the dialplan. Does this mean the first option is the >> best? or is there a "dialplan way" of doing it? >> >> Thank you very much for your help! >> >> Nicolas >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > -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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090721/191199b9/attachment.html From larclap at yahoo.com Tue Jul 21 09:40:06 2009 From: larclap at yahoo.com (Lars Zeb) Date: Tue, 21 Jul 2009 09:40:06 -0700 Subject: [Freeswitch-users] Inbound call routing help In-Reply-To: References: <004a01c9da6f$c1e40120$45ac0360$@com> Message-ID: <00a501ca0a21$e77e67e0$b67b37a0$@com> Brian, Pressing * no longer works. I don't know if my dialplan is causing the error or something in FreeSWITCH has changed. When pressing * during the greeting, the call immediately hangs up. Any ideas? http://pastebin.freeswitch.org/9803 Thanks, Lars From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Thursday, May 21, 2009 5:01 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Inbound call routing help Try pressing * during the greeting and make sure you have the vmain extension so you can login. /b On May 21, 2009, at 6:56 PM, Lars Zeb wrote: I want to setup a dialplan for a single DID. I would like it to go to a specific extension, and if not picked up in 15 seconds, go to voicemail. I have set this scenario up and it works. But I would also like this person to be able to call this DID from outside FS via a phone and be able to retrieve their voicemail. I've seen the example of how to pick up an extension's voicemail while inside FS by checking to see if the destination_number is the same as the caller_id_number, and if so, listen to voicemail, otherwise leave the message with voicemail. But I don't have a clue how to accomplish this from outside, other than dedicating another DID to solely retrieving voicemail from outside. Any ideas? Thanks, Lars ______________ Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090721/3bdc38dc/attachment.html From rupa at rupa.com Tue Jul 21 09:54:22 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Tue, 21 Jul 2009 11:54:22 -0500 Subject: [Freeswitch-users] Best way to bridge 2 calls with LCR? In-Reply-To: <1b46b4e80907210927n4885b4c5g180a06e32bedc7c4@mail.gmail.com> References: <1b46b4e80907210835s6e516e2ft84f3d7329eee1535@mail.gmail.com> <1b46b4e80907210927n4885b4c5g180a06e32bedc7c4@mail.gmail.com> Message-ID: Ok, if you want to do it from the socket api, then I need to make a 'as xml' option to mod_lcr and give you lcr_auto_route as one of the nodes in the returned xml. Then you can do your own substitution in the originate line... In that case, you'd call lcr twice and do: originate lcr_auto_route1 &bridge(lcr_auto_route2) How soon do you need this? On Tue, Jul 21, 2009 at 11:27 AM, Nicolas Brenner wrote: > That looks like a good way to go about it. > > How can I access channel variables through the socket using the api? I > mean, how do I recover the value of ${lcr_auto_route}? I would need to add > some other variables, like ignore_early_media=true and a uuid that 'links' > the two calls so I can track it listening for events. > > Thanks! > > Nicolas > > > On Tue, Jul 21, 2009 at 11:43 AM, Rupa Schomaker wrote: > >> lcr api command doesn't really return a usable dialstring (it was >> originally done for debug purposes). I could add an "as xml" option if >> needed... >> >> Anyway, to do this from the dialplan: >> >> remember that originate's usage is: >> >> -USAGE |&() [] >> [] [] [] [] >> >> so, the first argument is the call url and the second would be an >> extension. so: >> >> 1) execute lcr for the first leg of the call >> 2) execute originate with: >> >> originate ${lcr_auto_route} extension >> >> extension just needs to match something in your dialplan. >> >> In extension, you'd do another lcr lookup and then bridge to that leg's >> ${lcr_auto_route} value. >> >> >> >> On Tue, Jul 21, 2009 at 10:35 AM, Nicolas Brenner wrote: >> >>> I would like to originate 2 calls from FS and then bridge them. There's >>> no incoming call so I think there's no dialplan involved. >>> What I'd like to do now is apply lcr rules to these calls. I've come up >>> with 2 options so far: >>> >>> 1) call lcr through the socket twice (once for each phonenumber) and then >>> originate the calls through the socket too >>> 2) have a javascript file which runs the actions above, run the script >>> through the socket with 'jsrun' >>> >>> How would you do it? >>> >>> For what I've read on the list, usually the recommended way is to stay >>> away from javascript as much as possible because it is not as efficient as >>> doing everything from the dialplan. Does this mean the first option is the >>> best? or is there a "dialplan way" of doing it? >>> >>> Thank you very much for your help! >>> >>> Nicolas >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> -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 >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090721/d731caa8/attachment.html From nicolas at medularis.com Tue Jul 21 11:00:41 2009 From: nicolas at medularis.com (Nicolas Brenner) Date: Tue, 21 Jul 2009 14:00:41 -0400 Subject: [Freeswitch-users] Best way to bridge 2 calls with LCR? In-Reply-To: References: <1b46b4e80907210835s6e516e2ft84f3d7329eee1535@mail.gmail.com> <1b46b4e80907210927n4885b4c5g180a06e32bedc7c4@mail.gmail.com> Message-ID: <1b46b4e80907211100r4f9967fesb1b2c5fba4785a66@mail.gmail.com> Thank you very much for the offer, but I don't want to bother you with this. I can just parse the string returned by lcr and get the gateway, that's all I really need to create my complete originate command. I am using the socket api because it is easier for me to understand how to do it, nevertheless I'd really like to know how to do it with the dialplan. What I don't understand very well about using the dialplan for this, is how to do the first originate command (which I need to do using the socket api). What puzzles me is that according to the originate syntax, I need to use an extension or call an application, yet for the first call I would have to use a dummy extension as I only need to hit the dialplan section that calls lcr once to originate the first call with an extension that hits the section of the dialplan where lcr gets called again and the calls get bridged. I'm thinking something like this: 1) call originate from socket api to hit dialplan section that does all the work (this originate command is what I don't understand, is there another way of "hitting the dialplan" besides calling originate?) 2) hit dialplan section which calls lcr for first number and bridges to an extension 3) the extension calls lcr fir the second number and originates the second call On steps 2 and 3 I could just use set data to set the additional variables I need. The first step is what troubles me. Thank you! Nicolas On Tue, Jul 21, 2009 at 12:54 PM, Rupa Schomaker wrote: > Ok, if you want to do it from the socket api, then I need to make a 'as > xml' option to mod_lcr and give you lcr_auto_route as one of the nodes in > the returned xml. Then you can do your own substitution in the originate > line... In that case, you'd call lcr twice and do: > > originate lcr_auto_route1 &bridge(lcr_auto_route2) > > How soon do you need this? > > > On Tue, Jul 21, 2009 at 11:27 AM, Nicolas Brenner wrote: > >> That looks like a good way to go about it. >> >> How can I access channel variables through the socket using the api? I >> mean, how do I recover the value of ${lcr_auto_route}? I would need to add >> some other variables, like ignore_early_media=true and a uuid that 'links' >> the two calls so I can track it listening for events. >> >> Thanks! >> >> Nicolas >> >> >> On Tue, Jul 21, 2009 at 11:43 AM, Rupa Schomaker wrote: >> >>> lcr api command doesn't really return a usable dialstring (it was >>> originally done for debug purposes). I could add an "as xml" option if >>> needed... >>> >>> Anyway, to do this from the dialplan: >>> >>> remember that originate's usage is: >>> >>> -USAGE |&() [] >>> [] [] [] [] >>> >>> so, the first argument is the call url and the second would be an >>> extension. so: >>> >>> 1) execute lcr for the first leg of the call >>> 2) execute originate with: >>> >>> originate ${lcr_auto_route} extension >>> >>> extension just needs to match something in your dialplan. >>> >>> In extension, you'd do another lcr lookup and then bridge to that leg's >>> ${lcr_auto_route} value. >>> >>> >>> >>> On Tue, Jul 21, 2009 at 10:35 AM, Nicolas Brenner >> > wrote: >>> >>>> I would like to originate 2 calls from FS and then bridge them. There's >>>> no incoming call so I think there's no dialplan involved. >>>> What I'd like to do now is apply lcr rules to these calls. I've come up >>>> with 2 options so far: >>>> >>>> 1) call lcr through the socket twice (once for each phonenumber) and >>>> then originate the calls through the socket too >>>> 2) have a javascript file which runs the actions above, run the script >>>> through the socket with 'jsrun' >>>> >>>> How would you do it? >>>> >>>> For what I've read on the list, usually the recommended way is to stay >>>> away from javascript as much as possible because it is not as efficient as >>>> doing everything from the dialplan. Does this mean the first option is the >>>> best? or is there a "dialplan way" of doing it? >>>> >>>> Thank you very much for your help! >>>> >>>> Nicolas >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> -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 >>> >>> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > -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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090721/93726fc6/attachment.html From rupa at rupa.com Tue Jul 21 11:21:00 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Tue, 21 Jul 2009 13:21:00 -0500 Subject: [Freeswitch-users] Best way to bridge 2 calls with LCR? In-Reply-To: <1b46b4e80907211100r4f9967fesb1b2c5fba4785a66@mail.gmail.com> References: <1b46b4e80907210835s6e516e2ft84f3d7329eee1535@mail.gmail.com> <1b46b4e80907210927n4885b4c5g180a06e32bedc7c4@mail.gmail.com> <1b46b4e80907211100r4f9967fesb1b2c5fba4785a66@mail.gmail.com> Message-ID: Well, the "as xml" is something I've been meaning to do, so I'm gonna get that checked in today sometime anyway. If you want to do any programmatic processing of the lcr data, the as xml is the way to go rather than parsing the strings. As for originate + lcr.... You can use the loopback endpoint and do it all in the dialplan: originate loopback/firstnumber secondnumber This will hit your dialplan with firstnumber first which you can lcr route. Then when that call establishes, it'll hit the dialplan with the second number which will also be routed through lcr. Is that more what you are looking for? This way all the 'routing' logic can be done via the dialplan. On Tue, Jul 21, 2009 at 1:00 PM, Nicolas Brenner wrote: > Thank you very much for the offer, but I don't want to bother you with > this. > > I can just parse the string returned by lcr and get the gateway, that's all > I really need to create my complete originate command. > > I am using the socket api because it is easier for me to understand how to > do it, nevertheless I'd really like to know how to do it with the dialplan. > > What I don't understand very well about using the dialplan for this, is how > to do the first originate command (which I need to do using the socket api). > What puzzles me is that according to the originate syntax, I need to use an > extension or call an application, yet for the first call I would have to use > a dummy extension as I only need to hit the dialplan section that calls lcr > once to originate the first call with an extension that hits the section of > the dialplan where lcr gets called again and the calls get bridged. > > I'm thinking something like this: > > 1) call originate from socket api to hit dialplan section that does all the > work (this originate command is what I don't understand, is there another > way of "hitting the dialplan" besides calling originate?) > > 2) hit dialplan section which calls lcr for first number and bridges to an > extension > > 3) the extension calls lcr fir the second number and originates the second > call > > On steps 2 and 3 I could just use set data to set the additional variables > I need. The first step is what troubles me. > > > Thank you! > > > Nicolas > > > On Tue, Jul 21, 2009 at 12:54 PM, Rupa Schomaker wrote: > >> Ok, if you want to do it from the socket api, then I need to make a 'as >> xml' option to mod_lcr and give you lcr_auto_route as one of the nodes in >> the returned xml. Then you can do your own substitution in the originate >> line... In that case, you'd call lcr twice and do: >> >> originate lcr_auto_route1 &bridge(lcr_auto_route2) >> >> How soon do you need this? >> >> >> On Tue, Jul 21, 2009 at 11:27 AM, Nicolas Brenner wrote: >> >>> That looks like a good way to go about it. >>> >>> How can I access channel variables through the socket using the api? I >>> mean, how do I recover the value of ${lcr_auto_route}? I would need to add >>> some other variables, like ignore_early_media=true and a uuid that 'links' >>> the two calls so I can track it listening for events. >>> >>> Thanks! >>> >>> Nicolas >>> >>> >>> On Tue, Jul 21, 2009 at 11:43 AM, Rupa Schomaker wrote: >>> >>>> lcr api command doesn't really return a usable dialstring (it was >>>> originally done for debug purposes). I could add an "as xml" option if >>>> needed... >>>> >>>> Anyway, to do this from the dialplan: >>>> >>>> remember that originate's usage is: >>>> >>>> -USAGE |&() [] >>>> [] [] [] [] >>>> >>>> so, the first argument is the call url and the second would be an >>>> extension. so: >>>> >>>> 1) execute lcr for the first leg of the call >>>> 2) execute originate with: >>>> >>>> originate ${lcr_auto_route} extension >>>> >>>> extension just needs to match something in your dialplan. >>>> >>>> In extension, you'd do another lcr lookup and then bridge to that leg's >>>> ${lcr_auto_route} value. >>>> >>>> >>>> >>>> On Tue, Jul 21, 2009 at 10:35 AM, Nicolas Brenner < >>>> nicolas at medularis.com> wrote: >>>> >>>>> I would like to originate 2 calls from FS and then bridge them. There's >>>>> no incoming call so I think there's no dialplan involved. >>>>> What I'd like to do now is apply lcr rules to these calls. I've come up >>>>> with 2 options so far: >>>>> >>>>> 1) call lcr through the socket twice (once for each phonenumber) and >>>>> then originate the calls through the socket too >>>>> 2) have a javascript file which runs the actions above, run the script >>>>> through the socket with 'jsrun' >>>>> >>>>> How would you do it? >>>>> >>>>> For what I've read on the list, usually the recommended way is to stay >>>>> away from javascript as much as possible because it is not as efficient as >>>>> doing everything from the dialplan. Does this mean the first option is the >>>>> best? or is there a "dialplan way" of doing it? >>>>> >>>>> Thank you very much for your help! >>>>> >>>>> Nicolas >>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> -- >>>> -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 >>>> >>>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> -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 >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090721/96f270b0/attachment.html From rupa at rupa.com Tue Jul 21 11:51:54 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Tue, 21 Jul 2009 13:51:54 -0500 Subject: [Freeswitch-users] Best way to bridge 2 calls with LCR? In-Reply-To: References: <1b46b4e80907210835s6e516e2ft84f3d7329eee1535@mail.gmail.com> <1b46b4e80907210927n4885b4c5g180a06e32bedc7c4@mail.gmail.com> <1b46b4e80907211100r4f9967fesb1b2c5fba4785a66@mail.gmail.com> Message-ID: Just a note that the "as xml" syntax has been added to current trunk. On Tue, Jul 21, 2009 at 1:21 PM, Rupa Schomaker wrote: > Well, the "as xml" is something I've been meaning to do, so I'm gonna get > that checked in today sometime anyway. If you want to do any programmatic > processing of the lcr data, the as xml is the way to go rather than parsing > the strings. > > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090721/3c03681d/attachment.html From lon at kickasspixels.com Tue Jul 21 12:01:56 2009 From: lon at kickasspixels.com (Lon Baker) Date: Tue, 21 Jul 2009 12:01:56 -0700 Subject: [Freeswitch-users] Call confirm ivr Message-ID: <5d3e0dc60907211201t2be7ea8dq2b2931f854bdc455@mail.gmail.com> Hi there, I am putting together an ivr to allow the recipient of a call to accept, route to voice mail or eavesdrop on voicemail. The current path is to answer the inbound call, park it, using the bgapi call to the recipient and play the IVR. Basically: 1. Answer 2. Playback greeting 3. UUID_PARK 4. Set filter for BACKGROUND_JOB 5. BGAPI Originate to the recipient with customer variables for processing 6. Do new IVR for recipient and process their input to route call. I am not sure is this the right path. Is there a better way? Lon -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090721/f1981cba/attachment.html From lzwierko at gmail.com Tue Jul 21 12:38:01 2009 From: lzwierko at gmail.com (=?UTF-8?B?xYF1a2FzeiBad2llcmtv?=) Date: Tue, 21 Jul 2009 21:38:01 +0200 Subject: [Freeswitch-users] Dial up from confernece issue In-Reply-To: <41A88E3D-C420-4A1F-8B80-098EA18F9544@freeswitch.org> References: <4A637FC5.6000301@gmail.com> <41A88E3D-C420-4A1F-8B80-098EA18F9544@freeswitch.org> Message-ID: <4A661919.8030500@gmail.com> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Brian, I've just updated to 14310 and it's the same. The thing seems that sofia module rejects the call in quite early stage, so there is no 302 answer from remote SIP peer (as no INVITE was sent). Again, I'm exercising a very simple scenario with default FS configuration (just downloaded from svn), so I don't really know what's wrong here... Perhaps there is a different way to attach a call to a existing conference? Perhaps I should just originate new call (with the 'originate' command), and when received, pass it it conference application with the conference-id of the conference that I want to attach it to? Does that make any sense? Thanks ? freeswitch at Zwierko-laptop> conference list API CALL [conference(list)] output: Conference 3001-192.168.0.1 (1 member) 3;sofia/internal/1000 at 192.168.0.1;25f125a5-3556-a449-85ec-e4001336f313;1000;1000;hear|speak|floor;0;0;300 freeswitch at Zwierko-laptop> conference 3001-192.168.0.1 bgdial sofia/default/1001 at 192.168.0.1 API CALL [conference(3001-192.168.0.1 bgdial sofia/default/1001 at 192.168.0.1)] output: OK freeswitch at Zwierko-laptop> 2009-07-19 13:19:44.451100 [NOTICE] switch_channel.c:602 New Channel sofia/internal/1001 at 192.168.0.1 [b9fada7f-9c1d-4949-af8a-a8220ce f9c5b] 2009-07-19 13:19:44.459100 [NOTICE] switch_channel.c:602 New Channel sofia/internal/48228882211 at 192.168.0.1 [4f6b26dd-a0cb-2846-ad17-5f517e60e2e7] 2009-07-19 13:19:44.488100 [INFO] mod_dialplan_xml.c:252 Processing TelkaSwitch->1001 in context public 2009-07-19 13:19:44.495100 [ERR] sofia.c:4174 Cannot Blind Transfer 1 Legged calls 2009-07-19 13:19:44.498100 [NOTICE] sofia.c:3775 Hangup sofia/internal/1001 at 192.168.0.1 [CS_CONSUME_MEDIA] [NO_USER_RESPONSE] 2009-07-19 13:19:44.501100 [ERR] mod_conference.c:4351 Cannot create outgoing channel, cause: NO_USER_RESPONSE 2009-07-19 13:19:44.512100 [NOTICE] switch_core_session.c:1085 Session 9 (sofia/internal/1001 at 192.168.0.1) Ended 2009-07-19 13:19:44.512100 [NOTICE] switch_core_session.c:1087 Close Channel sofia/internal/1001 at 192.168.0.1 [CS_DESTROY] 2009-07-19 13:19:44.628100 [NOTICE] switch_core_state_machine.c:179 Hangup sofia/internal/48228882211 at 192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING] 2009-07-19 13:19:44.708100 [NOTICE] switch_core_session.c:1085 Session 10 (sofia/internal/48228882211 at 192.168.0.1) Ended 2009-07-19 13:19:44.708100 [NOTICE] switch_core_session.c:1087 Close Channel sofia/internal/48228882211 at 192.168.0.1 [CS_DESTROY] Brian West wrote: > If a single leg call gets a 302 you can't really "transfer" it > anywhere... What SVN rev are you on? > > /b > > On Jul 19, 2009, at 3:19 PM, ?ukasz Zwierko wrote: > >> Hi, >> >> sorry if you're getting this again, I'm not sure if this mail got >> deliverd to the mail-list (I didn't get a copy...) >> >> Anyway, >> >> I want to use bgdial command to add a person to a already started >> conference (that is, call that person and when answered - add the >> channel to conference). >> >> The scenario is I have two sip clients registered in default context - >> 1000 and 1001. 1000 dials conference number (3001 in this case) and >> new >> conference is started. I want to dial out to second using bgdial, >> unfortunately mod_sofia drops the call with 'Cannot Blind Transfer 1 >> Legged calls' message. >> >> Should I use the bgdial command differently? Or perhaps I should do >> this >> totally differently? Logs attached below. >> >> Thanks for any help, >> >> Lukasz > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.9 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iQEcBAEBAgAGBQJKZhkZAAoJED7LBosr0F2uE0wIAILk4StkVFGr4QVNsn7dob3d C1UBQnHOPezxlRmyT/lZjeN0Ddw+LZdvC5/Z14V8qjItsar2BDxT65AtVdryaKZq 9wlaEpGCoE377YGKM/k+hi8FYzvTkL1/Oz7aFGW/wpe2gbxKk1YWFSeU13iGpsZV 2byaY0qLdsGrs3CL3XMs69tKHmnnPcdM5p6xSYlOpKeE8/jUNJ+W7cOo0CcmVFf8 Mybwlhq7S7g6cKOD3WqgmBzMJi0pZRBgdz6x6uinAGmiSmTJIWO6+8BNjSIN373U OS7ivn8Gu4Tub50NBhkjhIEM3Kf+2JLQBRkwT0Mr4heIle9ZFe5UWbMFy0g8GbE= =xG/h -----END PGP SIGNATURE----- From brian at freeswitch.org Tue Jul 21 12:42:07 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 21 Jul 2009 14:42:07 -0500 Subject: [Freeswitch-users] Dial up from confernece issue In-Reply-To: <4A661919.8030500@gmail.com> References: <4A637FC5.6000301@gmail.com> <41A88E3D-C420-4A1F-8B80-098EA18F9544@freeswitch.org> <4A661919.8030500@gmail.com> Message-ID: Its a 302 on a single leg call right? /b On Jul 21, 2009, at 2:38 PM, ?ukasz Zwierko wrote: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > Brian, > > I've just updated to 14310 and it's the same. The thing seems that > sofia > module rejects the call in quite early stage, so there is no 302 > answer > from remote SIP peer (as no INVITE was sent). > Again, I'm exercising a very simple scenario with default FS > configuration (just downloaded from svn), so I don't really know > what's > wrong here... > Perhaps there is a different way to attach a call to a existing > conference? > Perhaps I should just originate new call (with the 'originate' > command), > and when received, pass it it conference application with the > conference-id of the conference that I want to attach it to? Does that > make any sense? > > Thanks > > ? From rupa at rupa.com Tue Jul 21 13:01:03 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Tue, 21 Jul 2009 15:01:03 -0500 Subject: [Freeswitch-users] Call confirm ivr In-Reply-To: <5d3e0dc60907211201t2be7ea8dq2b2931f854bdc455@mail.gmail.com> References: <5d3e0dc60907211201t2be7ea8dq2b2931f854bdc455@mail.gmail.com> Message-ID: Maybe look at the group_confirm_* stuff. http://wiki.freeswitch.org/wiki/Freeswitch_IVR_Originate#Answer_confirmation There is a way to get it to execute a script as well which is probably waht you want. This would be simpler than doing the work you are saying below. On Tue, Jul 21, 2009 at 2:01 PM, Lon Baker wrote: > Hi there, > I am putting together an ivr to allow the recipient of a call to accept, > route to voice mail or eavesdrop on voicemail. > > The current path is to answer the inbound call, park it, using the bgapi > call to the recipient and play the IVR. > > Basically: > > 1. Answer > 2. Playback greeting > 3. UUID_PARK > 4. Set filter for BACKGROUND_JOB > 5. BGAPI Originate to the recipient with customer variables for > processing > 6. Do new IVR for recipient and process their input to route call. > > I am not sure is this the right path. > > Is there a better way? > > Lon > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090721/07027e0f/attachment.html From nicolas at medularis.com Tue Jul 21 13:05:22 2009 From: nicolas at medularis.com (Nicolas Brenner) Date: Tue, 21 Jul 2009 16:05:22 -0400 Subject: [Freeswitch-users] Best way to bridge 2 calls with LCR? In-Reply-To: References: <1b46b4e80907210835s6e516e2ft84f3d7329eee1535@mail.gmail.com> <1b46b4e80907210927n4885b4c5g180a06e32bedc7c4@mail.gmail.com> <1b46b4e80907211100r4f9967fesb1b2c5fba4785a66@mail.gmail.com> Message-ID: <1b46b4e80907211305g17b838b1s9e5acd72da7f1792@mail.gmail.com> Now I understand! thank you very much for your explanation, very clear! On Tue, Jul 21, 2009 at 2:21 PM, Rupa Schomaker wrote: > Well, the "as xml" is something I've been meaning to do, so I'm gonna get > that checked in today sometime anyway. If you want to do any programmatic > processing of the lcr data, the as xml is the way to go rather than parsing > the strings. > > As for originate + lcr.... You can use the loopback endpoint and do it all > in the dialplan: > > originate loopback/firstnumber secondnumber > > This will hit your dialplan with firstnumber first which you can lcr > route. Then when that call establishes, it'll hit the dialplan with the > second number which will also be routed through lcr. > > Is that more what you are looking for? > > This way all the 'routing' logic can be done via the dialplan. > > > On Tue, Jul 21, 2009 at 1:00 PM, Nicolas Brenner wrote: > >> Thank you very much for the offer, but I don't want to bother you with >> this. >> >> I can just parse the string returned by lcr and get the gateway, that's >> all I really need to create my complete originate command. >> >> I am using the socket api because it is easier for me to understand how to >> do it, nevertheless I'd really like to know how to do it with the dialplan. >> >> What I don't understand very well about using the dialplan for this, is >> how to do the first originate command (which I need to do using the socket >> api). What puzzles me is that according to the originate syntax, I need to >> use an extension or call an application, yet for the first call I would have >> to use a dummy extension as I only need to hit the dialplan section that >> calls lcr once to originate the first call with an extension that hits the >> section of the dialplan where lcr gets called again and the calls get >> bridged. >> >> I'm thinking something like this: >> >> 1) call originate from socket api to hit dialplan section that does all >> the work (this originate command is what I don't understand, is there >> another way of "hitting the dialplan" besides calling originate?) >> >> 2) hit dialplan section which calls lcr for first number and bridges to an >> extension >> >> 3) the extension calls lcr fir the second number and originates the second >> call >> >> On steps 2 and 3 I could just use set data to set the additional variables >> I need. The first step is what troubles me. >> >> >> Thank you! >> >> >> Nicolas >> >> >> On Tue, Jul 21, 2009 at 12:54 PM, Rupa Schomaker wrote: >> >>> Ok, if you want to do it from the socket api, then I need to make a 'as >>> xml' option to mod_lcr and give you lcr_auto_route as one of the nodes in >>> the returned xml. Then you can do your own substitution in the originate >>> line... In that case, you'd call lcr twice and do: >>> >>> originate lcr_auto_route1 &bridge(lcr_auto_route2) >>> >>> How soon do you need this? >>> >>> >>> On Tue, Jul 21, 2009 at 11:27 AM, Nicolas Brenner >> > wrote: >>> >>>> That looks like a good way to go about it. >>>> >>>> How can I access channel variables through the socket using the api? I >>>> mean, how do I recover the value of ${lcr_auto_route}? I would need to add >>>> some other variables, like ignore_early_media=true and a uuid that 'links' >>>> the two calls so I can track it listening for events. >>>> >>>> Thanks! >>>> >>>> Nicolas >>>> >>>> >>>> On Tue, Jul 21, 2009 at 11:43 AM, Rupa Schomaker wrote: >>>> >>>>> lcr api command doesn't really return a usable dialstring (it was >>>>> originally done for debug purposes). I could add an "as xml" option if >>>>> needed... >>>>> >>>>> Anyway, to do this from the dialplan: >>>>> >>>>> remember that originate's usage is: >>>>> >>>>> -USAGE |&() [] >>>>> [] [] [] [] >>>>> >>>>> so, the first argument is the call url and the second would be an >>>>> extension. so: >>>>> >>>>> 1) execute lcr for the first leg of the call >>>>> 2) execute originate with: >>>>> >>>>> originate ${lcr_auto_route} extension >>>>> >>>>> extension just needs to match something in your dialplan. >>>>> >>>>> In extension, you'd do another lcr lookup and then bridge to that leg's >>>>> ${lcr_auto_route} value. >>>>> >>>>> >>>>> >>>>> On Tue, Jul 21, 2009 at 10:35 AM, Nicolas Brenner < >>>>> nicolas at medularis.com> wrote: >>>>> >>>>>> I would like to originate 2 calls from FS and then bridge them. >>>>>> There's no incoming call so I think there's no dialplan involved. >>>>>> What I'd like to do now is apply lcr rules to these calls. I've come >>>>>> up with 2 options so far: >>>>>> >>>>>> 1) call lcr through the socket twice (once for each phonenumber) and >>>>>> then originate the calls through the socket too >>>>>> 2) have a javascript file which runs the actions above, run the script >>>>>> through the socket with 'jsrun' >>>>>> >>>>>> How would you do it? >>>>>> >>>>>> For what I've read on the list, usually the recommended way is to stay >>>>>> away from javascript as much as possible because it is not as efficient as >>>>>> doing everything from the dialplan. Does this mean the first option is the >>>>>> best? or is there a "dialplan way" of doing it? >>>>>> >>>>>> Thank you very much for your help! >>>>>> >>>>>> Nicolas >>>>>> >>>>>> _______________________________________________ >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>> >>>>> >>>>> -- >>>>> -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 >>>>> >>>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> -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 >>> >>> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > -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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090721/476cfe9b/attachment.html From nicolas at medularis.com Tue Jul 21 13:05:37 2009 From: nicolas at medularis.com (Nicolas Brenner) Date: Tue, 21 Jul 2009 16:05:37 -0400 Subject: [Freeswitch-users] Best way to bridge 2 calls with LCR? In-Reply-To: References: <1b46b4e80907210835s6e516e2ft84f3d7329eee1535@mail.gmail.com> <1b46b4e80907210927n4885b4c5g180a06e32bedc7c4@mail.gmail.com> <1b46b4e80907211100r4f9967fesb1b2c5fba4785a66@mail.gmail.com> Message-ID: <1b46b4e80907211305m2b21f576n757245b78200939c@mail.gmail.com> Great! Thanks! On Tue, Jul 21, 2009 at 2:51 PM, Rupa Schomaker wrote: > Just a note that the "as xml" syntax has been added to current trunk. > > On Tue, Jul 21, 2009 at 1:21 PM, Rupa Schomaker wrote: > >> Well, the "as xml" is something I've been meaning to do, so I'm gonna get >> that checked in today sometime anyway. If you want to do any programmatic >> processing of the lcr data, the as xml is the way to go rather than parsing >> the strings. >> >> > -- > -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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090721/19812cfc/attachment.html From lon at kickasspixels.com Tue Jul 21 13:29:40 2009 From: lon at kickasspixels.com (Lon Baker) Date: Tue, 21 Jul 2009 13:29:40 -0700 Subject: [Freeswitch-users] Call confirm ivr In-Reply-To: References: <5d3e0dc60907211201t2be7ea8dq2b2931f854bdc455@mail.gmail.com> Message-ID: <576617A9-AA8E-47BB-96FA-4D882592E50A@kickasspixels.com> Thanks. I have looked at that, but everything has to run over the event_socket to the application logic we are building. I didn't see a way to exec a remote script/URL. Lon On Jul 21, 2009, at 1:01 PM, Rupa Schomaker wrote: > Maybe look at the group_confirm_* stuff. > > http://wiki.freeswitch.org/wiki/Freeswitch_IVR_Originate#Answer_confirmation > > There is a way to get it to execute a script as well which is > probably waht you want. This would be simpler than doing the work > you are saying below. > > On Tue, Jul 21, 2009 at 2:01 PM, Lon Baker > wrote: > Hi there, > > I am putting together an ivr to allow the recipient of a call to > accept, route to voice mail or eavesdrop on voicemail. > > The current path is to answer the inbound call, park it, using the > bgapi call to the recipient and play the IVR. > > Basically: > Answer > Playback greeting > UUID_PARK > Set filter for BACKGROUND_JOB > BGAPI Originate to the recipient with customer variables for > processing > Do new IVR for recipient and process their input to route call. > I am not sure is this the right path. > > Is there a better way? > > Lon > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org > > > > > -- > -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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090721/3228aeaa/attachment.html From msc at freeswitch.org Tue Jul 21 14:58:30 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 21 Jul 2009 14:58:30 -0700 Subject: [Freeswitch-users] ClueCon 2009 - Check Out The 15" MacBook Pro That We're Giving Away Message-ID: <87f2f3b90907211458x109d7611m2d64a5dfa0e5fcab@mail.gmail.com> This just in: We have a picture of the incredible MacBook Pro that we will be giving away this year: http://cluecon.com/node/38 Remember, all paid attendees are eligible to win this beautiful unit, so register today! Call 877.742.CLUE and get signed up. -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090721/fc3cf52b/attachment.html From pete at privateconnect.com Tue Jul 21 21:35:12 2009 From: pete at privateconnect.com (Pete Mueller) Date: Tue, 21 Jul 2009 21:35:12 -0700 Subject: [Freeswitch-users] Confusing handling of incoming calls Message-ID: <20090721213512.2ad02225396a31c9de30536f2e338977.fae79d6455.wbe@email04.secureserver.net> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090721/7160996a/attachment.html From lzwierko at gmail.com Tue Jul 21 22:45:54 2009 From: lzwierko at gmail.com (=?ISO-8859-2?Q?=A3ukasz_Zwierko?=) Date: Wed, 22 Jul 2009 07:45:54 +0200 Subject: [Freeswitch-users] Dial up from confernece issue In-Reply-To: References: <4A637FC5.6000301@gmail.com> <41A88E3D-C420-4A1F-8B80-098EA18F9544@freeswitch.org> <4A661919.8030500@gmail.com> Message-ID: I'm not sure how this exactly works, but I suppose that it is a single leg call, which upon answer would be attached to the conference (?) somehow. But again, this call does not originate outside FS so what would be the cause for 302? 2009/7/21 Brian West : > Its a 302 on a single leg call right? > > /b > > On Jul 21, 2009, at 2:38 PM, ?ukasz Zwierko wrote: > >> -----BEGIN PGP SIGNED MESSAGE----- >> Hash: SHA1 >> >> Brian, >> >> I've just updated to 14310 and it's the same. The thing seems that >> sofia >> module rejects the call in quite early stage, so there is no 302 >> answer >> from remote SIP peer (as no INVITE was sent). >> Again, I'm exercising a very simple scenario with default FS >> configuration (just downloaded from svn), so I don't really know >> what's >> wrong here... >> Perhaps there is a different way to attach a call to a existing >> conference? >> Perhaps I should just originate new call (with the 'originate' >> command), >> and when received, pass it it conference application with the >> conference-id of the conference that I want to attach it to? Does that >> make any sense? >> >> 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 > From elihayun at gmail.com Wed Jul 22 00:19:43 2009 From: elihayun at gmail.com (Eli Hayun) Date: Wed, 22 Jul 2009 10:19:43 +0300 Subject: [Freeswitch-users] How to initiate a call without dialing In-Reply-To: <4A65C911.50707@freeswitch.org> References: <4A659D30.5020600@savion.huji.ac.il> <4A65C911.50707@freeswitch.org> Message-ID: <4A66BD8F.8050108@savion.huji.ac.il> Raymond Chandler wrote: > Eli Hayun wrote: > >> Is there is a way to initiate a call without making any dial manually? >> >> > i think the api command "originate" is what you're looking for > > -Ray > > _______________________________________________ > Thanks, I figure that out, but now I have another problem. When I do that, the name display as "FreeSwitch" and the number is display as "00000000000" I tried to set "outbound_caller_name" with no success. How should I solve that? Thanks Eli From solko at gcdf.pl Wed Jul 22 01:06:01 2009 From: solko at gcdf.pl (Szymon Olko) Date: Wed, 22 Jul 2009 10:06:01 +0200 Subject: [Freeswitch-users] How to initiate a call without dialing In-Reply-To: <4A66BD8F.8050108@savion.huji.ac.il> References: <4A659D30.5020600@savion.huji.ac.il> <4A65C911.50707@freeswitch.org> <4A66BD8F.8050108@savion.huji.ac.il> Message-ID: <4A66C869.3070806@gcdf.pl> Eli Hayun pisze: > Raymond Chandler wrote: >> Eli Hayun wrote: >> >>> Is there is a way to initiate a call without making any dial manually? >>> >>> >> i think the api command "originate" is what you're looking for >> >> -Ray >> >> _______________________________________________ >> > Thanks, I figure that out, but now I have another problem. When I do > that, the name display as "FreeSwitch" and the number is display as > "00000000000" > I tried to set "outbound_caller_name" with no success. > How should I solve that? > > Thanks > Eli > Read wiki, it explains a lot. http://wiki.freeswitch.org/wiki/Mod_commands#originate use it like that: originate sofia/internal/1001%192.168.1.1 3001 XML default Name 1213232 OR originate sofia/internal/1001%192.168.1.1 &conference(test) '' '' Name 1213232 From elihayun at gmail.com Wed Jul 22 01:44:52 2009 From: elihayun at gmail.com (Eli Hayun) Date: Wed, 22 Jul 2009 11:44:52 +0300 Subject: [Freeswitch-users] How to initiate a call without dialing In-Reply-To: <4A66C869.3070806@gcdf.pl> References: <4A659D30.5020600@savion.huji.ac.il> <4A65C911.50707@freeswitch.org> <4A66BD8F.8050108@savion.huji.ac.il> <4A66C869.3070806@gcdf.pl> Message-ID: <4A66D184.3090201@savion.huji.ac.il> Szymon Olko wrote: > Eli Hayun pisze: > >> Raymond Chandler wrote: >> >>> Eli Hayun wrote: >>> >>> >>>> Is there is a way to initiate a call without making any dial manually? >>>> >>>> >>>> >>> i think the api command "originate" is what you're looking for >>> >>> -Ray >>> >>> _______________________________________________ >>> >>> >> Thanks, I figure that out, but now I have another problem. When I do >> that, the name display as "FreeSwitch" and the number is display as >> "00000000000" >> I tried to set "outbound_caller_name" with no success. >> How should I solve that? >> >> Thanks >> Eli >> >> > Read wiki, it explains a lot. > > http://wiki.freeswitch.org/wiki/Mod_commands#originate > > use it like that: > originate sofia/internal/1001%192.168.1.1 3001 XML default Name 1213232 > > OR > > originate sofia/internal/1001%192.168.1.1 &conference(test) '' '' Name 1213232 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > Thanks alot. Its working now. Eli From rupa at rupa.com Wed Jul 22 02:12:30 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Wed, 22 Jul 2009 04:12:30 -0500 Subject: [Freeswitch-users] Confusing handling of incoming calls In-Reply-To: <20090721213512.2ad02225396a31c9de30536f2e338977.fae79d6455.wbe@email04.secureserver.net> References: <20090721213512.2ad02225396a31c9de30536f2e338977.fae79d6455.wbe@email04.secureserver.net> Message-ID: On Tue, Jul 21, 2009 at 11:35 PM, Pete Mueller wrote: > My goal is: > 0) figure out why the bandwidth gateway is being processed as "internal" > (this is more of a security thing) > they are probably terminating traffic on port 5060 rather than 5080. 5060 is internal, 5080 is external. > > 1) have both gateways enter at the same point in the dialplan (this seems > to be the purpose of the "Extension" param) > I'd drop the extension param and instead match on the destination_number (the DID used to reach you). > 2) be able to identify which gateway the call came in on. I was hoping to > set a param in the gateway configuration that would be passed through onto > the channel, but have not found one. Worst case, I could have each gateway > enter at a different extension in the dialplan, however, that doesn't seem > to be working if the channel comes in the "internal" profile. > Not sure here... gateways are an outbound thing. Inbound calls just hit your dialplan and you process from there. A sledgehammer approach would be to have a different sip_profile for each gateway. But that is just silly. Flowroute at least puts their name in the sdp.... > Thanks for your help. I've provided INFO dumps from both gateways if they > help... > -pete > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090722/61a4b30b/attachment.html From pete at privateconnect.com Wed Jul 22 03:11:26 2009 From: pete at privateconnect.com (Pete Mueller) Date: Wed, 22 Jul 2009 03:11:26 -0700 Subject: [Freeswitch-users] Confusing handling of incoming calls Message-ID: <20090722031126.2ad02225396a31c9de30536f2e338977.ca479dcfc2.wbe@email04.secureserver.net> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090722/11590c21/attachment.html From rdenert at tng.de Wed Jul 22 03:23:02 2009 From: rdenert at tng.de (Rudolf Denert) Date: Wed, 22 Jul 2009 12:23:02 +0200 (CEST) Subject: [Freeswitch-users] Playing sound files in a conference In-Reply-To: <25598690.126581248258109020.JavaMail.root@zimbra.tng.de> Message-ID: <4286351.126631248258182227.JavaMail.root@zimbra.tng.de> Hallo everybody! I would like to play soundfiles in a existing conference. The procedure is this: Someone calls the number of the conference. Then this person types the pin in to his phone. The next step is that he has to say the name for example "John". This file is saved in a special folder. When he enters the conference room everybody in this existing conference should here: "John" "has enterd the room". If he leaves then everybody should hear: "John" "has left the room". Of course there are two soundfiles. "John" is what the caller has spoken into his phone and the second one a generated file from me. They should be played in succession. Is it possible to implement this with lua? If yes, how can I do that. Thanks for your help. Greetz From brian at freeswitch.org Wed Jul 22 03:43:37 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 22 Jul 2009 05:43:37 -0500 Subject: [Freeswitch-users] Confusing handling of incoming calls In-Reply-To: <20090722031126.2ad02225396a31c9de30536f2e338977.ca479dcfc2.wbe@email04.secureserver.net> References: <20090722031126.2ad02225396a31c9de30536f2e338977.ca479dcfc2.wbe@email04.secureserver.net> Message-ID: On Jul 22, 2009, at 5:11 AM, Pete Mueller wrote: > 0) Rupa, you are absolutely right, I forgot that. ports was never > an issue because previous gateways all REGISTERed. I will have to > swap my ports around as bandwidth is not flexible. What do you mean here? > > 1) I thought of this, but I have hundreds of DID, (around 600 at the > moment) and maintaining that mapping in the dialplan would be a > mess. AFTER I know what gateway the call arrived on, I have a > database for each gateway that helps me process from there. XML_CURL? > > 2) Yes, separate profiles would work, but does sound gross. I'm > going to swap my ports around and see if that clears things up... > > -pete -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090722/ba51e974/attachment.html From brian at freeswitch.org Wed Jul 22 03:50:02 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 22 Jul 2009 05:50:02 -0500 Subject: [Freeswitch-users] Dial up from confernece issue In-Reply-To: References: <4A637FC5.6000301@gmail.com> <41A88E3D-C420-4A1F-8B80-098EA18F9544@freeswitch.org> <4A661919.8030500@gmail.com> Message-ID: <122E4883-9AF7-4F7A-AAC0-2562ECBDBFDF@freeswitch.org> The far end you're calling is sending a 302 can you check the sip traffic please. /b On Jul 22, 2009, at 12:45 AM, ?ukasz Zwierko wrote: > I'm not sure how this exactly works, but I suppose that it is a single > leg call, which upon answer would be attached to the conference (?) > somehow. But again, this call does not originate outside FS so what > would be the cause for 302? From elihayun at gmail.com Wed Jul 22 05:45:08 2009 From: elihayun at gmail.com (Eli Hayun) Date: Wed, 22 Jul 2009 15:45:08 +0300 Subject: [Freeswitch-users] Limit is not working when originate a call Message-ID: <4A6709D4.6000704@savion.huji.ac.il> Hi I set the limit to 1 on the extension like that When I am trying to make a call the that destination i transfered to limit_exceeded dialplan, just like I want The problem is, that when I am trying to make a call using "originate" I am not getting the limitation. Why is that? From rupa at rupa.com Wed Jul 22 06:04:35 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Wed, 22 Jul 2009 08:04:35 -0500 Subject: [Freeswitch-users] Confusing handling of incoming calls In-Reply-To: <20090722031126.2ad02225396a31c9de30536f2e338977.ca479dcfc2.wbe@email04.secureserver.net> References: <20090722031126.2ad02225396a31c9de30536f2e338977.ca479dcfc2.wbe@email04.secureserver.net> Message-ID: On Wed, Jul 22, 2009 at 5:11 AM, Pete Mueller wrote: > 0) Rupa, you are absolutely right, I forgot that. ports was never an issue > because previous gateways all REGISTERed. I will have to swap my ports > around as bandwidth is not flexible. > You can't tell bandwidth.com to use port 5080? > > 1) I thought of this, but I have hundreds of DID, (around 600 at the > moment) and maintaining that mapping in the dialplan would be a mess. AFTER > I know what gateway the call arrived on, I have a database for each gateway > that helps me process from there. > You have cases where the same DID maps differently for one gateway or another? If not, why is the gateway part of the database query? > > 2) Yes, separate profiles would work, but does sound gross. I'm going to > swap my ports around and see if that clears things up... > > -pete > > -------- Original Message -------- > Subject: Re: [Freeswitch-users] Confusing handling of incoming calls > From: Rupa Schomaker > Date: Wed, July 22, 2009 2:12 am > To: freeswitch-users at lists.freeswitch.org > > > > On Tue, Jul 21, 2009 at 11:35 PM, Pete Mueller wrote: > >> My goal is: >> 0) figure out why the bandwidth gateway is being processed as "internal" >> (this is more of a security thing) >> > > they are probably terminating traffic on port 5060 rather than 5080. 5060 > is internal, 5080 is external. > > >> >> 1) have both gateways enter at the same point in the dialplan (this seems >> to be the purpose of the "Extension" param) >> > > I'd drop the extension param and instead match on the destination_number > (the DID used to reach you). > > >> 2) be able to identify which gateway the call came in on. I was hoping to >> set a param in the gateway configuration that would be passed through onto >> the channel, but have not found one. Worst case, I could have each gateway >> enter at a different extension in the dialplan, however, that doesn't seem >> to be working if the channel comes in the "internal" profile. >> > > Not sure here... gateways are an outbound thing. Inbound calls just hit > your dialplan and you process from there. A sledgehammer approach would be > to have a different sip_profile for each gateway. But that is just silly. > Flowroute at least puts their name in the sdp.... > > >> Thanks for your help. I've provided INFO dumps from both gateways if they >> help... >> -pete >> > > > -- > -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 > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090722/0e3267c6/attachment.html From brian at freeswitch.org Wed Jul 22 06:10:03 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 22 Jul 2009 08:10:03 -0500 Subject: [Freeswitch-users] Confusing handling of incoming calls In-Reply-To: References: <20090722031126.2ad02225396a31c9de30536f2e338977.ca479dcfc2.wbe@email04.secureserver.net> Message-ID: <140A2687-B8F6-407D-9B67-A91DB5496AC4@freeswitch.org> On Jul 22, 2009, at 8:04 AM, Rupa Schomaker wrote: > > > On Wed, Jul 22, 2009 at 5:11 AM, Pete Mueller > wrote: > 0) Rupa, you are absolutely right, I forgot that. ports was never > an issue because previous gateways all REGISTERed. I will have to > swap my ports around as bandwidth is not flexible. > > You can't tell bandwidth.com to use port 5080? Yes you can... I do it all the time. > > 1) I thought of this, but I have hundreds of DID, (around 600 at the > moment) and maintaining that mapping in the dialplan would be a > mess. AFTER I know what gateway the call arrived on, I have a > database for each gateway that helps me process from there. > > You have cases where the same DID maps differently for one gateway > or another? If not, why is the gateway part of the database query? > > 2) Yes, separate profiles would work, but does sound gross. I'm > going to swap my ports around and see if that clears things up... > > -pete -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090722/393f04e9/attachment.html From intralanman at freeswitch.org Wed Jul 22 06:47:58 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Wed, 22 Jul 2009 09:47:58 -0400 Subject: [Freeswitch-users] Playing sound files in a conference In-Reply-To: <4286351.126631248258182227.JavaMail.root@zimbra.tng.de> References: <4286351.126631248258182227.JavaMail.root@zimbra.tng.de> Message-ID: <4A67188E.5060802@freeswitch.org> you could hang on the event socket and catch the conference events, then play the sounds via the "conference" api commands -Ray Rudolf Denert wrote: > Hallo everybody! > > I would like to play soundfiles in a existing conference. > > The procedure is this: > > Someone calls the number of the conference. Then this person types the pin in to his phone. The next step is that he has to say the name for example "John". This file is saved in a special folder. When he enters the conference room everybody in this existing conference should here: "John" "has enterd the room". If he leaves then everybody should hear: "John" "has left the room". Of course there are two soundfiles. "John" is what the caller has spoken into his phone and the second one a generated file from me. They should be played in succession. > > Is it possible to implement this with lua? If yes, how can I do that. > > Thanks for your help. > > Greetz > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.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 Wed Jul 22 07:46:06 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 22 Jul 2009 10:46:06 -0400 Subject: [Freeswitch-users] Limit is not working when originate a call In-Reply-To: <4A6709D4.6000704@savion.huji.ac.il> References: <4A6709D4.6000704@savion.huji.ac.il> Message-ID: <2C85EB6C-1D80-4DC7-8D14-06B17CD1046D@jerris.com> because your not running limit at all when you are doing an originate directly. You can use loopback to originate through a dialplan extension. Mike On Jul 22, 2009, at 8:45 AM, Eli Hayun wrote: > Hi > I set the limit to 1 on the extension like that > > > > When I am trying to make a call the that destination i transfered to > limit_exceeded dialplan, just like I want > > The problem is, that when I am trying to make a call using > "originate" I > am not getting the limitation. > Why is 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 msc at freeswitch.org Wed Jul 22 10:30:16 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 22 Jul 2009 10:30:16 -0700 Subject: [Freeswitch-users] Good Information On How To Submit Bug Reports Message-ID: <87f2f3b90907221030y3f2cc840y504cdbb3ac6194ec@mail.gmail.com> FYI, Brian West called to my attention that one of our community members, John Wehle, has been very good at submitting useful bug reports, in many cases with patches. His style of reporting is worthy of imitation, so I've added a few links to the JIRA section of the Reporting Bugs wiki page: http://wiki.freeswitch.org/wiki/Reporting_Bugs#Examples_Of_Well-Written_JIRA_Submissions Please feel free to check it out. If you have any questions on what a bug report should look like then definitely read some of John's submissions and emulate his style. By submitting useful bug reports you will save the FreeSWITCH developers countless hours and headaches, not to mention the warm fuzzies you'll feel inside. :) -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090722/01374541/attachment.html From pete at privateconnect.com Wed Jul 22 11:30:11 2009 From: pete at privateconnect.com (Pete Mueller) Date: Wed, 22 Jul 2009 11:30:11 -0700 Subject: [Freeswitch-users] Confusing handling of incoming calls Message-ID: <20090722113011.2ad02225396a31c9de30536f2e338977.3d5b2c362e.wbe@email04.secureserver.net> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090722/ea917137/attachment-0001.html From larclap at yahoo.com Wed Jul 22 13:13:56 2009 From: larclap at yahoo.com (Lars Zeb) Date: Wed, 22 Jul 2009 13:13:56 -0700 Subject: [Freeswitch-users] Asterisk key during message hangs up call In-Reply-To: References: <004a01c9da6f$c1e40120$45ac0360$@com> Message-ID: <00b501ca0b08$f178a520$d469ef60$@com> Brian, When calling into FreeSWITCH and pressing * during the greeting, the call immediately hangs up. It used to ask for the mailbox number to retrieve messages. It no longer works. I don't know if my dialplan is causing the error or something in FreeSWITCH has changed. Any ideas? http://pastebin.freeswitch.org/9803 Thanks, Lars -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090722/37516c17/attachment.html From lfurrea at gmail.com Wed Jul 22 16:21:07 2009 From: lfurrea at gmail.com (Luis F Urrea) Date: Wed, 22 Jul 2009 17:21:07 -0600 Subject: [Freeswitch-users] Asterisk key during message hangs up call In-Reply-To: <00b501ca0b08$f178a520$d469ef60$@com> References: <004a01c9da6f$c1e40120$45ac0360$@com> <00b501ca0b08$f178a520$d469ef60$@com> Message-ID: I don't know if this may be related but in voicemail.conf.xml by default the two params that follow are defined: And pressing 9 during the greeting does not send me to the operator. I am on trunk rev 14123M On Wed, Jul 22, 2009 at 2:13 PM, Lars Zeb wrote: > Brian, > > > > When calling into FreeSWITCH and pressing * during the greeting, the call > immediately hangs up. > > > > It used to ask for the mailbox number to retrieve messages. It no longer > works. I don?t know if my dialplan is causing the error or something in > FreeSWITCH has changed. > > > > Any ideas? > > > > http://pastebin.freeswitch.org/9803 > > > > Thanks, Lars > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20090722/cdd6efc0/attachment.html From anthony.minessale at gmail.com Wed Jul 22 16:33:08 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 22 Jul 2009 18:33:08 -0500 Subject: [Freeswitch-users] Asterisk key during message hangs up call In-Reply-To: <00b501ca0b08$f178a520$d469ef60$@com> References: <004a01c9da6f$c1e40120$45ac0360$@com> <00b501ca0b08$f178a520$d469ef60$@com> Message-ID: <191c3a030907221633k6594901dj22b9c7ded8af19e4@mail.gmail.com> you are using a channel created with a script and you did not set js session.autoHangup(0) lua session:autoHangup(0) so when the * makes the call transfer the script kills the channel. On Wed, Jul 22, 2009 at 3:13 PM, Lars Zeb wrote: > Brian, > > > > When calling into FreeSWITCH and pressing * during the greeting, the call > immediately hangs up. > > > > It used to ask for the mailbox number to retrieve messages. It no longer > works. I don?t know if my dialplan is causing the error or something in > FreeSWITCH has changed. > > > > Any ideas? > > > > http://pastebin.freeswitch.org/9803 > > > > Thanks, Lars > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference 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/20090722/9bbc547a/attachment.html From mike at jerris.com Wed Jul 22 16:46:30 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 22 Jul 2009 19:46:30 -0400 Subject: [Freeswitch-users] Asterisk key during message hangs up call In-Reply-To: References: <004a01c9da6f$c1e40120$45ac0360$@com> <00b501ca0b08$f178a520$d469ef60$@com> Message-ID: <01355490-8C46-4E90-A4C1-51F4921AEFBF@jerris.com> Do you have anything on that extension? On Jul 22, 2009, at 7:21 PM, Luis F Urrea wrote: > I don't know if this may be related but in voicemail.conf.xml by > default the two params that follow are defined: > > > > > And pressing 9 during the greeting does not send me to the operator. > > I am on trunk rev 14123M > > On Wed, Jul 22, 2009 at 2:13 PM, Lars Zeb wrote: > Brian, > > > > When calling into FreeSWITCH and pressing * during the greeting, the > call immediately hangs up. > > > > It used to ask for the mailbox number to retrieve messages. It no > longer works. I don?t know if my dialplan is causing the error or so > mething in FreeSWITCH has changed. > > > > Any ideas? > > > > http://pastebin.freeswitch.org/9803 > > > > Thanks, Lars > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20090722/c99dda3d/attachment.html From elihayun at gmail.com Wed Jul 22 22:04:07 2009 From: elihayun at gmail.com (Eli Hayun) Date: Thu, 23 Jul 2009 08:04:07 +0300 Subject: [Freeswitch-users] Limit is not working when originate a call In-Reply-To: <2C85EB6C-1D80-4DC7-8D14-06B17CD1046D@jerris.com> References: <4A6709D4.6000704@savion.huji.ac.il> <2C85EB6C-1D80-4DC7-8D14-06B17CD1046D@jerris.com> Message-ID: <4A67EF47.6000402@savion.huji.ac.il> Michael Jerris wrote: > because your not running limit at all when you are doing an originate > directly. You can use loopback to originate through a dialplan > extension. > > Mike > > On Jul 22, 2009, at 8:45 AM, Eli Hayun wrote: > > >> Hi >> I set the limit to 1 on the extension like that >> >> >> >> When I am trying to make a call the that destination i transfered to >> limit_exceeded dialplan, just like I want >> >> The problem is, that when I am trying to make a call using >> "originate" I >> am not getting the limitation. >> Why is 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 >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > Thanks for answer. I am calling "Originate" from JS. I tried to call "limit_hash" from JS but with no success. I did it like that: lmt = apiExecute("limit_hash", dialed_ext + " " + dialed_ext + " 1"); I could't find any documentation on that. can u help ? Thanks Eli From anthony.minessale at gmail.com Thu Jul 23 05:39:47 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 23 Jul 2009 07:39:47 -0500 Subject: [Freeswitch-users] Limit is not working when originate a call In-Reply-To: <4A67EF47.6000402@savion.huji.ac.il> References: <4A6709D4.6000704@savion.huji.ac.il> <2C85EB6C-1D80-4DC7-8D14-06B17CD1046D@jerris.com> <4A67EF47.6000402@savion.huji.ac.il> Message-ID: <191c3a030907230539i5a640e2do4377d8797ff8f279@mail.gmail.com> limit is for inbound calls you cannot call it after you already made the call. The correct approach would be to not make the call at all. you could maybe use the limit FSAPI interface with apiExecute to check if the limit was exceeded and then not bother to place the call to begin with. otherwise it's sort of like putting a prisoner in the electric chair then giving him his trial. On Thu, Jul 23, 2009 at 12:04 AM, Eli Hayun wrote: > Michael Jerris wrote: > > because your not running limit at all when you are doing an originate > > directly. You can use loopback to originate through a dialplan > > extension. > > > > Mike > > > > On Jul 22, 2009, at 8:45 AM, Eli Hayun wrote: > > > > > >> Hi > >> I set the limit to 1 on the extension like that > >> > >> > >> > >> When I am trying to make a call the that destination i transfered to > >> limit_exceeded dialplan, just like I want > >> > >> The problem is, that when I am trying to make a call using > >> "originate" I > >> am not getting the limitation. > >> Why is 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 > >> > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > Thanks for answer. > I am calling "Originate" from JS. I tried to call "limit_hash" from JS > but with no success. I did it like that: > > lmt = apiExecute("limit_hash", dialed_ext + " " + dialed_ext + " 1"); > > I could't find any documentation on that. > can u help ? > > Thanks > Eli > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference 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/20090723/e19fe0fa/attachment.html From larclap at yahoo.com Thu Jul 23 07:23:11 2009 From: larclap at yahoo.com (Lars Zeb) Date: Thu, 23 Jul 2009 07:23:11 -0700 Subject: [Freeswitch-users] Asterisk key during message hangs up call In-Reply-To: <191c3a030907221633k6594901dj22b9c7ded8af19e4@mail.gmail.com> References: <004a01c9da6f$c1e40120$45ac0360$@com> <00b501ca0b08$f178a520$d469ef60$@com> <191c3a030907221633k6594901dj22b9c7ded8af19e4@mail.gmail.com> Message-ID: <004c01ca0ba1$1bb6de90$53249bb0$@com> Thanks for the reply. This is my first attempt at using a script. I tried: session:autoHangup(0) or session:autoHangup(false) but got an error: 2009-07-23 07:19:09.652238 [ERR] mod_lua.cpp:182 /usr/local/freeswitch/scripts/inbound_calls.lua:29: attempt to call method 'autoHangup' (a nil value) stack traceback: /usr/local/freeswitch/scripts/inbound_calls.lua:29: in main chunk I looked at the documentation and tried: session:setAutoHangup(false) and the script proceeded without error. However, looking at the log, I do not see the setAutoHangup being called. Also, when pressing *, I get a fast, busy signal. I have pasted the script and log at http://pastebin.freeswitch.org/9836 Thanks again, Lars From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Wednesday, July 22, 2009 4:33 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Asterisk key during message hangs up call you are using a channel created with a script and you did not set js session.autoHangup(0) lua session:autoHangup(0) so when the * makes the call transfer the script kills the channel. On Wed, Jul 22, 2009 at 3:13 PM, Lars Zeb wrote: Brian, When calling into FreeSWITCH and pressing * during the greeting, the call immediately hangs up. It used to ask for the mailbox number to retrieve messages. It no longer works. I don't know if my dialplan is causing the error or something in FreeSWITCH has changed. Any ideas? http://pastebin.freeswitch.org/9803 Thanks, Lars _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference 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/20090723/bea25341/attachment.html From anthony.minessale at gmail.com Thu Jul 23 07:41:28 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 23 Jul 2009 09:41:28 -0500 Subject: [Freeswitch-users] Asterisk key during message hangs up call In-Reply-To: <004c01ca0ba1$1bb6de90$53249bb0$@com> References: <004a01c9da6f$c1e40120$45ac0360$@com> <00b501ca0b08$f178a520$d469ef60$@com> <191c3a030907221633k6594901dj22b9c7ded8af19e4@mail.gmail.com> <004c01ca0ba1$1bb6de90$53249bb0$@com> Message-ID: <191c3a030907230741uc04b0cdoafcee3f79be55850@mail.gmail.com> I meant to pick one based on whichever lang you were using not to literally write what i said. anyway, yes so now you solved your autoHangup make a new debug trace like the one you looked at before now which should be different. On Thu, Jul 23, 2009 at 9:23 AM, Lars Zeb wrote: > Thanks for the reply. This is my first attempt at using a script. > > > > I tried: > > > > session:autoHangup(0) or session:autoHangup(false) > > > > but got an error: > > > > 2009-07-23 07:19:09.652238 [ERR] mod_lua.cpp:182 > /usr/local/freeswitch/scripts/inbound_calls.lua:29: attempt to call method > 'autoHangup' (a nil value) > > stack traceback: > > /usr/local/freeswitch/scripts/inbound_calls.lua:29: in main chunk > > > > I looked at the documentation and tried: > > > > session:setAutoHangup(false) > > > > and the script proceeded without error. However, looking at the log, I do > not see the setAutoHangup being called. Also, when pressing *, I get a fast, > busy signal. > > > > I have pasted the script and log at http://pastebin.freeswitch.org/9836 > > > > Thanks again, Lars > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony > Minessale > *Sent:* Wednesday, July 22, 2009 4:33 PM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Asterisk key during message hangs up > call > > > > you are using a channel created with a script and you did not set > > js > session.autoHangup(0) > > lua > session:autoHangup(0) > > so when the * makes the call transfer the script kills the channel. > > On Wed, Jul 22, 2009 at 3:13 PM, Lars Zeb wrote: > > Brian, > > > > When calling into FreeSWITCH and pressing * during the greeting, the call > immediately hangs up. > > > > It used to ask for the mailbox number to retrieve messages. It no longer > works. I don?t know if my dialplan is causing the error or something in > FreeSWITCH has changed. > > > > Any ideas? > > > > http://pastebin.freeswitch.org/9803 > > > > Thanks, Lars > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > 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/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference 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/20090723/74a679fb/attachment-0001.html From rdenert at tng.de Thu Jul 23 08:59:42 2009 From: rdenert at tng.de (Rudolf Denert) Date: Thu, 23 Jul 2009 17:59:42 +0200 (CEST) Subject: [Freeswitch-users] Problem with Caller controls Message-ID: <13405311.137361248364782823.JavaMail.root@zimbra.tng.de> Hello, I?m getting always the message: [ERR] mod_conference.c:5463 conference_new() Unable to install caller controls group 'test' I made a new caller-controls in the conference.conf.xml. It has the name "test". I also implemented the line: param name="caller-controls" value="test" in my conference-profil. Did I forget something? Thanks for your help! Greetz From larclap at yahoo.com Thu Jul 23 11:12:07 2009 From: larclap at yahoo.com (Lars Zeb) Date: Thu, 23 Jul 2009 11:12:07 -0700 Subject: [Freeswitch-users] Asterisk key during message hangs up call In-Reply-To: <191c3a030907230741uc04b0cdoafcee3f79be55850@mail.gmail.com> References: <004a01c9da6f$c1e40120$45ac0360$@com> <00b501ca0b08$f178a520$d469ef60$@com> <191c3a030907221633k6594901dj22b9c7ded8af19e4@mail.gmail.com> <004c01ca0ba1$1bb6de90$53249bb0$@com> <191c3a030907230741uc04b0cdoafcee3f79be55850@mail.gmail.com> Message-ID: <00a501ca0bc1$16f180c0$44d48240$@com> Your message made me look at the documentation, which was helpful. http://pastebin.freeswitch.org/9838 When I press *, I get a busy signal. Please disregard the USER_NOT_REGISTERED error in the log; one of the endpoints I bridged to is off-line. Thanks, Lars From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Thursday, July 23, 2009 7:41 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Asterisk key during message hangs up call I meant to pick one based on whichever lang you were using not to literally write what i said. anyway, yes so now you solved your autoHangup make a new debug trace like the one you looked at before now which should be different. On Thu, Jul 23, 2009 at 9:23 AM, Lars Zeb wrote: Thanks for the reply. This is my first attempt at using a script. I tried: session:autoHangup(0) or session:autoHangup(false) but got an error: 2009-07-23 07:19:09.652238 [ERR] mod_lua.cpp:182 /usr/local/freeswitch/scripts/inbound_calls.lua:29: attempt to call method 'autoHangup' (a nil value) stack traceback: /usr/local/freeswitch/scripts/inbound_calls.lua:29: in main chunk I looked at the documentation and tried: session:setAutoHangup(false) and the script proceeded without error. However, looking at the log, I do not see the setAutoHangup being called. Also, when pressing *, I get a fast, busy signal. I have pasted the script and log at http://pastebin.freeswitch.org/9836 Thanks again, Lars From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Wednesday, July 22, 2009 4:33 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Asterisk key during message hangs up call you are using a channel created with a script and you did not set js session.autoHangup(0) lua session:autoHangup(0) so when the * makes the call transfer the script kills the channel. On Wed, Jul 22, 2009 at 3:13 PM, Lars Zeb wrote: Brian, When calling into FreeSWITCH and pressing * during the greeting, the call immediately hangs up. It used to ask for the mailbox number to retrieve messages. It no longer works. I don't know if my dialplan is causing the error or something in FreeSWITCH has changed. Any ideas? http://pastebin.freeswitch.org/9803 Thanks, Lars _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference 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/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference 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/20090723/48cd1586/attachment.html From pjintheusa at gmail.com Thu Jul 23 11:13:52 2009 From: pjintheusa at gmail.com (Phillip Jones) Date: Thu, 23 Jul 2009 14:13:52 -0400 Subject: [Freeswitch-users] Barge on on prompts Message-ID: <367751820907231113p1a5d772eta359d9e25146dec1@mail.gmail.com> Hi there, Very simple scenario: Session.DtmfReceivedFunction = (d, t) => { Log.WriteLine(LogLevel.Info, "Received {0} for {1}.", d, t); CollectedDigits = d.ToString().Trim(); return ""; }; Session.flushDigits(); Session.StreamFile(VoicemailPromptsDirectory + "abigfile.wav", 0); Question is, it there a way to kill the streaming when the a digit is pressed? I would use the Session.PlayAndGetDigits() but that does not help when want to string things together like: Session.StreamFile(VoicemailPromptsDirectory + "vm-to_delete_the_message.wav", 0); Session.StreamFile(VoicemailPromptsDirectory + "vm-press.wav", 0); Session.Say("7", "en", "number", "pronounced"); Any help would be appreciated. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090723/6435af85/attachment.html From msc at freeswitch.org Thu Jul 23 11:40:07 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 23 Jul 2009 11:40:07 -0700 Subject: [Freeswitch-users] Barge on on prompts In-Reply-To: <367751820907231113p1a5d772eta359d9e25146dec1@mail.gmail.com> References: <367751820907231113p1a5d772eta359d9e25146dec1@mail.gmail.com> Message-ID: <87f2f3b90907231140j24c6d2efhdf3316bee4d87d73@mail.gmail.com> I think you might want to check out phrase macros... http://wiki.freeswitch.org/wiki/Speech_Phrase_Management -MC On Thu, Jul 23, 2009 at 11:13 AM, Phillip Jones wrote: > Hi there, > > Very simple scenario: > > Session.DtmfReceivedFunction = (d, t) => > { > Log.WriteLine(LogLevel.Info, "Received {0} for {1}.", d, t); > CollectedDigits = d.ToString().Trim(); > return ""; > }; > > > Session.flushDigits(); > Session.StreamFile(VoicemailPromptsDirectory + "abigfile.wav", 0); > > Question is, it there a way to kill the streaming when the a digit is > pressed? > > I would use the Session.PlayAndGetDigits() > > but that does not help when want to string things together like: > > Session.StreamFile(VoicemailPromptsDirectory + > "vm-to_delete_the_message.wav", 0); > Session.StreamFile(VoicemailPromptsDirectory + "vm-press.wav", 0); > Session.Say("7", "en", "number", "pronounced"); > > Any help would be 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090723/0cec4539/attachment-0001.html From pjintheusa at gmail.com Thu Jul 23 12:39:02 2009 From: pjintheusa at gmail.com (Phillip Jones) Date: Thu, 23 Jul 2009 15:39:02 -0400 Subject: [Freeswitch-users] Barge on on prompts In-Reply-To: <87f2f3b90907231140j24c6d2efhdf3316bee4d87d73@mail.gmail.com> References: <367751820907231113p1a5d772eta359d9e25146dec1@mail.gmail.com> <87f2f3b90907231140j24c6d2efhdf3316bee4d87d73@mail.gmail.com> Message-ID: <367751820907231239m308a3ee2s4be1cefdadf8e192@mail.gmail.com> Hi there, Thanks for the reply. That information is extremely useful. Given the code below though - when if I press '1' when the phrase is playing - playing does not stop. It continues. I am looking for a method to barge in and collect & react to digits immediately. Session.DtmfReceivedFunction = (d, t) => { Log.WriteLine(LogLevel.Info, "Received {0} for {1}.", d, t); Session.StreamFile("", 0); CollectedDigits = d.ToString().Trim(); return ""; }; Session.SayPhrase("msgcount", "187346", "en"); Any ideas? I am sure I must be missing something simple. Thanks a lot. Phillip Jones On Thu, Jul 23, 2009 at 2:40 PM, Michael Collins wrote: > I think you might want to check out phrase macros... > http://wiki.freeswitch.org/wiki/Speech_Phrase_Management > -MC > > On Thu, Jul 23, 2009 at 11:13 AM, Phillip Jones wrote: > >> Hi there, >> >> Very simple scenario: >> >> Session.DtmfReceivedFunction = (d, t) => >> { >> Log.WriteLine(LogLevel.Info, "Received {0} for {1}.", d, t); >> CollectedDigits = d.ToString().Trim(); >> return ""; >> }; >> >> >> Session.flushDigits(); >> Session.StreamFile(VoicemailPromptsDirectory + "abigfile.wav", 0); >> >> Question is, it there a way to kill the streaming when the a digit is >> pressed? >> >> I would use the Session.PlayAndGetDigits() >> >> but that does not help when want to string things together like: >> >> Session.StreamFile(VoicemailPromptsDirectory + >> "vm-to_delete_the_message.wav", 0); >> Session.StreamFile(VoicemailPromptsDirectory + "vm-press.wav", 0); >> Session.Say("7", "en", "number", "pronounced"); >> >> Any help would be 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090723/fb2f57f6/attachment.html From dave at 3c.co.uk Thu Jul 23 13:17:14 2009 From: dave at 3c.co.uk (David Knell) Date: Thu, 23 Jul 2009 17:17:14 -0300 Subject: [Freeswitch-users] Barge on on prompts In-Reply-To: <367751820907231239m308a3ee2s4be1cefdadf8e192@mail.gmail.com> References: <367751820907231113p1a5d772eta359d9e25146dec1@mail.gmail.com> <87f2f3b90907231140j24c6d2efhdf3316bee4d87d73@mail.gmail.com> <367751820907231239m308a3ee2s4be1cefdadf8e192@mail.gmail.com> Message-ID: <1248380234.16040.17.camel@dk-d820> Hi Phillip, You need to call FreeSWITCH's break function - I'd guess Session.Break(); might do it for you, but no guarantees. --Dave > Hi there, > > Thanks for the reply. That information is extremely useful. > > Given the code below though - when if I press '1' when the phrase is > playing - playing does not stop. It continues. I am looking for a > method to barge in and collect & react to digits immediately. > > > Session.DtmfReceivedFunction = (d, t) => > { > Log.WriteLine(LogLevel.Info, "Received {0} for > {1}.", d, t); > Session.StreamFile("", 0); > CollectedDigits = d.ToString().Trim(); > return ""; > > }; > > Session.SayPhrase("msgcount", "187346", "en"); > > > Any ideas? I am sure I must be missing something simple. > > Thanks a lot. > > > Phillip Jones > > > > > > On Thu, Jul 23, 2009 at 2:40 PM, Michael Collins > wrote: > I think you might want to check out phrase macros... > http://wiki.freeswitch.org/wiki/Speech_Phrase_Management > -MC > > > On Thu, Jul 23, 2009 at 11:13 AM, Phillip Jones > wrote: > > > Hi there, > > Very simple scenario: > > Session.DtmfReceivedFunction = (d, t) => > { > Log.WriteLine(LogLevel.Info, "Received {0} for > {1}.", d, t); > CollectedDigits = d.ToString().Trim(); > return ""; > }; > > > Session.flushDigits(); > Session.StreamFile(VoicemailPromptsDirectory + > "abigfile.wav", 0); > > Question is, it there a way to kill the streaming when > the a digit is pressed? > > I would use the Session.PlayAndGetDigits() > > but that does not help when want to string things > together like: > > Session.StreamFile(VoicemailPromptsDirectory + > "vm-to_delete_the_message.wav", 0); > Session.StreamFile(VoicemailPromptsDirectory + > "vm-press.wav", 0); > Session.Say("7", "en", "number", "pronounced"); > > Any help would be 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 > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- David Knell, Director, 3C Limited T: +44 20 3298 2000 E: dave at 3c.co.uk W: http://www.3c.co.uk From pjintheusa at gmail.com Thu Jul 23 14:12:24 2009 From: pjintheusa at gmail.com (Phillip Jones) Date: Thu, 23 Jul 2009 17:12:24 -0400 Subject: [Freeswitch-users] Barge on on prompts In-Reply-To: <1248380234.16040.17.camel@dk-d820> References: <367751820907231113p1a5d772eta359d9e25146dec1@mail.gmail.com> <87f2f3b90907231140j24c6d2efhdf3316bee4d87d73@mail.gmail.com> <367751820907231239m308a3ee2s4be1cefdadf8e192@mail.gmail.com> <1248380234.16040.17.camel@dk-d820> Message-ID: <367751820907231412u70560cbs92c1ad316bb5b552@mail.gmail.com> Ah! That you very much. Not Session.Break() but: Session.DtmfReceivedFunction = (d, t) => { Log.WriteLine(LogLevel.Info, "Received {0} for {1}.", d, t); CollectedDigits = d.ToString().Trim(); return "break"; } Thanks to you both for your help on this. On Thu, Jul 23, 2009 at 4:17 PM, David Knell wrote: > Hi Phillip, > > You need to call FreeSWITCH's break function - I'd guess > Session.Break(); might do it for you, but no guarantees. > > --Dave > > > Hi there, > > > > Thanks for the reply. That information is extremely useful. > > > > Given the code below though - when if I press '1' when the phrase is > > playing - playing does not stop. It continues. I am looking for a > > method to barge in and collect & react to digits immediately. > > > > > > Session.DtmfReceivedFunction = (d, t) => > > { > > Log.WriteLine(LogLevel.Info, "Received {0} for > > {1}.", d, t); > > Session.StreamFile("", 0); > > CollectedDigits = d.ToString().Trim(); > > return ""; > > > > }; > > > > Session.SayPhrase("msgcount", "187346", "en"); > > > > > > Any ideas? I am sure I must be missing something simple. > > > > Thanks a lot. > > > > > > Phillip Jones > > > > > > > > > > > > On Thu, Jul 23, 2009 at 2:40 PM, Michael Collins > > wrote: > > I think you might want to check out phrase macros... > > http://wiki.freeswitch.org/wiki/Speech_Phrase_Management > > -MC > > > > > > On Thu, Jul 23, 2009 at 11:13 AM, Phillip Jones > > wrote: > > > > > > Hi there, > > > > Very simple scenario: > > > > Session.DtmfReceivedFunction = (d, t) => > > { > > Log.WriteLine(LogLevel.Info, "Received {0} for > > {1}.", d, t); > > CollectedDigits = d.ToString().Trim(); > > return ""; > > }; > > > > > > Session.flushDigits(); > > Session.StreamFile(VoicemailPromptsDirectory + > > "abigfile.wav", 0); > > > > Question is, it there a way to kill the streaming when > > the a digit is pressed? > > > > I would use the Session.PlayAndGetDigits() > > > > but that does not help when want to string things > > together like: > > > > Session.StreamFile(VoicemailPromptsDirectory + > > "vm-to_delete_the_message.wav", 0); > > Session.StreamFile(VoicemailPromptsDirectory + > > "vm-press.wav", 0); > > Session.Say("7", "en", "number", "pronounced"); > > > > Any help would be 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 > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > -- > David Knell, Director, 3C Limited > T: +44 20 3298 2000 > E: dave at 3c.co.uk > W: 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090723/a7851342/attachment.html From velu.technical at gmail.com Thu Jul 23 23:22:48 2009 From: velu.technical at gmail.com (velusamy velu) Date: Fri, 24 Jul 2009 11:52:48 +0530 Subject: [Freeswitch-users] A stun server lookup Message-ID: <1452e2980907232322h2229bd8bp25d50b73ed59fb7b@mail.gmail.com> Dear All, When I start the freeSWITCH, I am receiving the following errors, 2009-07-24 16:56:23 [ERR] sofia_glue.c:566 sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478[Remote Address Error!] 2009-07-24 16:56:23 [ERR] sofia.c:1972 config_sofia() Failed to get external ip. I commented the stun configurations in vars.xml.conf file eventhough I am receiving the same error. Pleas any one give solution to solve this error.... Regards, Velusamy. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090724/96b49d8c/attachment-0001.html From jason at jasonjgw.net Fri Jul 24 00:12:41 2009 From: jason at jasonjgw.net (Jason White) Date: Fri, 24 Jul 2009 17:12:41 +1000 Subject: [Freeswitch-users] A stun server lookup In-Reply-To: <1452e2980907232322h2229bd8bp25d50b73ed59fb7b@mail.gmail.com> References: <1452e2980907232322h2229bd8bp25d50b73ed59fb7b@mail.gmail.com> Message-ID: <20090724071241.GA31649@jdc.jasonjgw.net> velusamy velu wrote: > I commented the stun configurations in vars.xml.conf file eventhough I > am receiving the same error. > > Pleas any one give solution to solve this error.... Edit vars.xml, change the variables that use Stun to be wahtever you want your ext-sip-ip and ext-rtp-ip addresses to be, then restart the external profile sofia profile external restart reloadxml or restart FreeSWITCH. From lzwierko at gmail.com Fri Jul 24 00:17:14 2009 From: lzwierko at gmail.com (=?ISO-8859-2?Q?=A3ukasz_Zwierko?=) Date: Fri, 24 Jul 2009 09:17:14 +0200 Subject: [Freeswitch-users] Dial up from confernece issue In-Reply-To: <122E4883-9AF7-4F7A-AAC0-2562ECBDBFDF@freeswitch.org> References: <4A637FC5.6000301@gmail.com> <41A88E3D-C420-4A1F-8B80-098EA18F9544@freeswitch.org> <4A661919.8030500@gmail.com> <122E4883-9AF7-4F7A-AAC0-2562ECBDBFDF@freeswitch.org> Message-ID: Ok Brian, you were right after all - I've had my X-lite incorrectly configured, sorry for wasting your time. thanks, LZ W dniu 22 lipca 2009 12:50 u?ytkownik Brian West napisa?: > The far end you're calling is sending a 302 can you check the sip > traffic please. > > /b > > On Jul 22, 2009, at 12:45 AM, ?ukasz Zwierko wrote: > >> I'm not sure how this exactly works, but I suppose that it is a single >> leg call, which upon answer would be attached to the conference (?) >> somehow. But again, this call does not originate outside FS so what >> would be the cause for 302? > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From hoaianh at gmx.de Fri Jul 24 06:45:29 2009 From: hoaianh at gmx.de (Ngo-Vi Hoai-Anh) Date: Fri, 24 Jul 2009 15:45:29 +0200 Subject: [Freeswitch-users] newbie question Message-ID: <4A69BAF9.2000408@gmx.de> Hi folk, I'm very new to FreeSwitch. I've read all the FAQs and traced the mailing list back to 12.2008 but still not found the answers for my questions. Please help! 1. Is it possible to make unauthenticated call to FS in the manner 1000@? 2. Is there already a java implementation for FS like http://asterisk-java.org/ for Asterisk? Thank you . Hoai-Anh From thangappan143 at gmail.com Thu Jul 23 22:42:17 2009 From: thangappan143 at gmail.com (Thangappan.M) Date: Fri, 24 Jul 2009 11:12:17 +0530 Subject: [Freeswitch-users] Problem in mod_perl Message-ID: <7aa29e790907232242v5f6db729p7b1158114fcfeb9@mail.gmail.com> I am new to Freeswitch, I started to write a dial plan using perl instead of xml in the case of IVR. I used the following statement in the dialplan/default.xml file I am using Twinkle Soft phone.When I am calling to 5000 in the freeswitch console it tells the following error. Invalid application perl. >From that I understood there is no Perl module has been installed.Then I uncommented the line from modules.conf.xml. Again I checked with my Perl version it also supports usemultiplicity. Where I made a mistake? Can anyone please solve my problem? I want to execute the Perl script in the dial plan. -- Regards, Thangappan.M -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090724/dedac132/attachment.html From niall.crosby at gmail.com Fri Jul 24 07:29:51 2009 From: niall.crosby at gmail.com (Niall Crosby) Date: Fri, 24 Jul 2009 15:29:51 +0100 Subject: [Freeswitch-users] newbie question In-Reply-To: <4A69BAF9.2000408@gmx.de> References: <4A69BAF9.2000408@gmx.de> Message-ID: <4aec92830907240729l1892125fy127f17a42cd4bb9d@mail.gmail.com> Hi Hoai-Anh, a) Disable forced registration: In sip_profiles\internal.xml set auth-calls = false b) Enable calls from any IP: In sip_profiles\zinternal take out ) This is what I had to do to get SIPP working without registering first. I also program Java, the best I could find is the socket event interface. Hope this helps, Niall. 2009/7/24 Ngo-Vi Hoai-Anh : > Hi folk, > > I'm very new to FreeSwitch. I've read all the FAQs and traced the > mailing list back to 12.2008 but still not found the answers for my > questions. Please help! > > 1. Is it possible to make unauthenticated call to FS in the manner > 1000@? > 2. Is there already a java implementation for FS like > http://asterisk-java.org/ for Asterisk? > > Thank you . > Hoai-Anh > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- -- Sremium Ltd. Reg Number: 451937 Mobile: +353 (0)87 2393174 Web: www.sremium.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. Statements and opinions expressed in this e-mail may not represent those of Sremium. 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 immediately and delete the material from any computer. From hoaianh at gmx.de Fri Jul 24 08:02:41 2009 From: hoaianh at gmx.de (Ngo-Vi Hoai-Anh) Date: Fri, 24 Jul 2009 17:02:41 +0200 Subject: [Freeswitch-users] newbie question In-Reply-To: <4aec92830907240729l1892125fy127f17a42cd4bb9d@mail.gmail.com> References: <4A69BAF9.2000408@gmx.de> <4aec92830907240729l1892125fy127f17a42cd4bb9d@mail.gmail.com> Message-ID: <4A69CD11.5040905@gmx.de> Thank you Niall :-) Niall Crosby schrieb: > Hi Hoai-Anh, > > a) Disable forced registration: > In sip_profiles\internal.xml set auth-calls = false > > b) Enable calls from any IP: > In sip_profiles\zinternal take out value="domains"/>) > > This is what I had to do to get SIPP working without registering first. > > I also program Java, the best I could find is the socket event interface. > > Hope this helps, > Niall. > > > 2009/7/24 Ngo-Vi Hoai-Anh : > >> Hi folk, >> >> I'm very new to FreeSwitch. I've read all the FAQs and traced the >> mailing list back to 12.2008 but still not found the answers for my >> questions. Please help! >> >> 1. Is it possible to make unauthenticated call to FS in the manner >> 1000@? >> 2. Is there already a java implementation for FS like >> http://asterisk-java.org/ for Asterisk? >> >> Thank you . >> Hoai-Anh >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > > From shawn at sboyle.com Fri Jul 24 08:36:31 2009 From: shawn at sboyle.com (Shawn Boyle) Date: Fri, 24 Jul 2009 11:36:31 -0400 Subject: [Freeswitch-users] Problem in mod_perl In-Reply-To: <7aa29e790907232242v5f6db729p7b1158114fcfeb9@mail.gmail.com> Message-ID: Did you also uncomment the line: languages/mod_perl in modules.conf when you compiled FS? I believe it's commented out by default. [Something I personally disagree with...but I would bear Larry Wall's children if I could manage it physiologically.] -Shawn ________________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Thangappan.M Sent: Friday, July 24, 2009 1:42 AM To: FreeSWITCH-users at lists.freeswitch.org Subject: [Freeswitch-users] Problem in mod_perl Where I made a mistake? Can anyone please solve my problem? I want to execute the Perl script? in the dial plan. -- Regards, Thangappan.M From gshfreesw at gmail.com Fri Jul 24 08:45:15 2009 From: gshfreesw at gmail.com (Gu Sh) Date: Fri, 24 Jul 2009 11:45:15 -0400 Subject: [Freeswitch-users] IAX Transfer support Message-ID: <5070fcbd0907240845q246b80d6i5837c0d4f4451634@mail.gmail.com> I have been using freeswitch for over a year and I love all of the features, extensibility etc. Recently one of the clients wanted to use a IAX client and call from the IAX client works fine but there was one feature requested by my client that did not work. The feature is the "IAX Transfer" and I see the Transfer message come through by turning up debugging in the iax.conf file but freeswitch does not do anything with it. What is the current status of IAX support on freeswitch? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090724/57e2294b/attachment.html From anthony.minessale at gmail.com Fri Jul 24 08:51:56 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 24 Jul 2009 10:51:56 -0500 Subject: [Freeswitch-users] Problem in mod_perl In-Reply-To: <7aa29e790907232242v5f6db729p7b1158114fcfeb9@mail.gmail.com> References: <7aa29e790907232242v5f6db729p7b1158114fcfeb9@mail.gmail.com> Message-ID: <191c3a030907240851t3bd46f9n65d86063cdc460a1@mail.gmail.com> edit modules.conf in th build root for FS uncomment the line that builds mod_perl issue "make mod_perl-install" from the shell edit /usr/local/freeswitch/conf/autoload_configs/modules.conf.xml and uncomment the mod_perl line. On Fri, Jul 24, 2009 at 12:42 AM, Thangappan.M wrote: > I am new to Freeswitch, I started to write a dial plan using perl instead > of xml in the case of IVR. > I used the following statement in the dialplan/default.xml file > > > > > > > I am using Twinkle Soft phone.When I am calling to 5000 in the freeswitch > console it tells the following error. > Invalid application perl. > > From that I understood there is no Perl module has been installed.Then I > uncommented the line from modules.conf.xml. Again > I checked with my Perl version it also supports > usemultiplicity. > > Where I made a mistake? > Can anyone please solve my problem? > I want to execute the Perl script in the dial plan. > > > -- > Regards, > Thangappan.M > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference 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/20090724/86356eaf/attachment-0001.html From anthony.minessale at gmail.com Fri Jul 24 08:58:58 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 24 Jul 2009 10:58:58 -0500 Subject: [Freeswitch-users] IAX Transfer support In-Reply-To: <5070fcbd0907240845q246b80d6i5837c0d4f4451634@mail.gmail.com> References: <5070fcbd0907240845q246b80d6i5837c0d4f4451634@mail.gmail.com> Message-ID: <191c3a030907240858g308f5e20x3d9173c1e9dd359f@mail.gmail.com> mod_iax was one of the first endpoint modules made in FS and it has seen little attention since it was first added. The primary purpose of the module was to have alternate endpoints to help work on the abstraction concepts in the core. Since it actually works, we left it in. But we really don't have any plans to do much else with it. The IAX lib we use in mod_iax is a heavily modified version of the iax client lib used in most iax softphones, its was really only designed for one or 2 calls max. An open source and liberally licensed (BSD/MIT) IAX library that is scalable and meets all of the programmatic challenges presented by the iax spec is really needed to advance the protocol any further. In addition a few enhancements should be made to the protocol to improve it's scalability in general. On Fri, Jul 24, 2009 at 10:45 AM, Gu Sh wrote: > I have been using freeswitch for over a year and I love all of the > features, extensibility etc. Recently one of the clients wanted to use a IAX > client and call from the IAX client works fine but there was one feature > requested by my client that did not work. The feature is the "IAX Transfer" > and I see the Transfer message come through by turning up debugging in the > iax.conf file but freeswitch does not do anything with it. > > What is the current status of IAX support on freeswitch? > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference 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/20090724/afd8c47e/attachment.html From gregt at cgicommunications.com Fri Jul 24 08:59:41 2009 From: gregt at cgicommunications.com (Greg Thoen) Date: Fri, 24 Jul 2009 11:59:41 -0400 Subject: [Freeswitch-users] setInputCallback not working with Javascript? In-Reply-To: <554DE828-B96A-4E17-A974-151CAC50A0E5@freeswitch.org> References: <8CB7A629F5159D9-17A0-AA5@webmail-de13.sysops.aol.com> <554DE828-B96A-4E17-A974-151CAC50A0E5@freeswitch.org> Message-ID: Regarding the second part of his question, I am having a hard time stripping SpeechTools.jm into a very simple speech recognition example. I also cannot get collectInput to receive the type of "event", only "dtmf" -- Greg Thoen, Vice President CGI Communications, Inc. 1-585-427-0020 x260 On Mar 24, 2009, at 9:47 AM, Brian West wrote: > Javascript doesn't use the Core Session constructor. Its not the > same as the other languages. > > /b > > On Mar 24, 2009, at 1:40 AM, mszlazak at aol.com wrote: > >> I'm getting in build 12653M: >> >> [ERR] notify.js:130 mod_spidermonkey() TypeError: >> session.setInputCallback is not a function >> >> The wiki says this function should work in Javascript. >> >> http://wiki.freeswitch.org/wiki/ >> CoreSession_Constructor#session:setInputCallback >> >> Also, has there been changes to session.collectInput with >> type="event"? I get dtmf type events with my callback function but >> can't seem to get type="event" with speech events. >> >> 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090724/4c25d099/attachment.html From intralanman at freeswitch.org Fri Jul 24 09:02:06 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Fri, 24 Jul 2009 12:02:06 -0400 Subject: [Freeswitch-users] IAX Transfer support In-Reply-To: <5070fcbd0907240845q246b80d6i5837c0d4f4451634@mail.gmail.com> References: <5070fcbd0907240845q246b80d6i5837c0d4f4451634@mail.gmail.com> Message-ID: <4A69DAFE.2050709@freeswitch.org> Gu Sh wrote: > > What is the current status of IAX support on freeswitch? Basically unsupported, the module still builds and sort of works, but there isn't much interest in maintaining it. It would definitely be a lot better for IAX lovers if someone wanted to take on maintaining it, or fund its maintenance. -Ray From mike at jerris.com Fri Jul 24 09:03:55 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 24 Jul 2009 12:03:55 -0400 Subject: [Freeswitch-users] Problem in mod_perl In-Reply-To: References: Message-ID: <2079FE7F-23F1-44A9-B2EA-0B69238E89EC@jerris.com> It is not built by default because it requires manual intervention to make sure you have a proper threadsafe perl and all its dev libs installed first. We work hard to make sure all default modules build out of the box with minimal external dependencies. Also, this module still does not work 100% on some platforms (solaris?) Mike On Jul 24, 2009, at 11:36 AM, Shawn Boyle wrote: > Did you also uncomment the line: > > languages/mod_perl > > in modules.conf when you compiled FS? I believe it's commented out > by default. [Something I personally disagree with...but I would bear > Larry Wall's children if I could manage it physiologically.] > > -Shawn From mike at jerris.com Fri Jul 24 09:05:16 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 24 Jul 2009 12:05:16 -0400 Subject: [Freeswitch-users] IAX Transfer support In-Reply-To: <5070fcbd0907240845q246b80d6i5837c0d4f4451634@mail.gmail.com> References: <5070fcbd0907240845q246b80d6i5837c0d4f4451634@mail.gmail.com> Message-ID: On Jul 24, 2009, at 11:45 AM, Gu Sh wrote: > I have been using freeswitch for over a year and I love all of the > features, extensibility etc. Recently one of the clients wanted to > use a IAX client and call from the IAX client works fine but there > was one feature requested by my client that did not work. The > feature is the "IAX Transfer" and I see the Transfer message come > through by turning up debugging in the iax.conf file but freeswitch > does not do anything with it. > > What is the current status of IAX support on freeswitch? > _______________________________________________ IAX transfer is not supported and no one has really touched that module in a couple years. If anyone is interested in enhancing that module we would be glad to accept patches to improve it but we have no roadmap plans to add anything to it. Mike From mattdfong at gmail.com Fri Jul 24 10:58:48 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Fri, 24 Jul 2009 10:58:48 -0700 Subject: [Freeswitch-users] Application to Record Calls - Out of Band Message-ID: <4256bf830907241058h67d1c798r9ddb56c8d2845611@mail.gmail.com> Hi, I'm trying to build an application that provides statistics of calls and call recording. Someone told me this could be done out of band with a SPAN (?) port that would replicate SIP and media packets to a separate NIC without having to actually pass the real-calls thru FreeSWITCH. It was explained that this SPAN port would in the SBC would replicate data received. If this is done, is there a way I can utilize FreeSWITCH to interpret these packets without actually having any control of the calls? If so how? Sorry, I'm new to telco, so hopefully this post makes sense to someone. --matt -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090724/ffbb04d2/attachment.html From dome at tel.co.th Fri Jul 24 12:25:02 2009 From: dome at tel.co.th (Dome Charoenyost) Date: Sat, 25 Jul 2009 02:25:02 +0700 Subject: [Freeswitch-users] freeswitch-1.0.4pre10 ? Message-ID: <8ccbff060907241225m476078xb87dcbfa7c51e5f7@mail.gmail.com> I found freeswitch-1.0.4pre10 in http://files.freeswitch.org/freeswitch-1.0.4pre10.tar.bz2 But no news about this version Dome C. From dave at 3c.co.uk Fri Jul 24 14:56:09 2009 From: dave at 3c.co.uk (David Knell) Date: Fri, 24 Jul 2009 18:56:09 -0300 Subject: [Freeswitch-users] Application to Record Calls - Out of Band In-Reply-To: <4256bf830907241058h67d1c798r9ddb56c8d2845611@mail.gmail.com> References: <4256bf830907241058h67d1c798r9ddb56c8d2845611@mail.gmail.com> Message-ID: <1248472569.4360.6.camel@dk-d820> Hi Matt, FreeSWITCH probably isn't what you want. A quick Google for 'sip call sniffer' found this: http://www.enderunix.org/voipong/ which might well be a more appropriate starting point. A SPAN port is just a port on a network switch which has the traffic going to/from another port (or ports) replicated to it. Cheers -- Dave > Hi, > > > I'm trying to build an application that provides statistics of calls > and call recording. Someone told me this could be done out of band > with a SPAN (?) port that would replicate SIP and media packets to a > separate NIC without having to actually pass the real-calls thru > FreeSWITCH. It was explained that this SPAN port would in the SBC > would replicate data received. > > > If this is done, is there a way I can utilize FreeSWITCH to interpret > these packets without actually having any control of the calls? If so > how? Sorry, I'm new to telco, so hopefully this post makes sense to > someone. > > > --matt > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- David Knell, Director, 3C Limited T: +44 20 3298 2000 E: dave at 3c.co.uk W: http://www.3c.co.uk From thangappan143 at gmail.com Fri Jul 24 22:50:44 2009 From: thangappan143 at gmail.com (Thangappan.M) Date: Sat, 25 Jul 2009 11:20:44 +0530 Subject: [Freeswitch-users] Regarding IVR Message-ID: <7aa29e790907242250o152c539x16a4b25c1e2e53c0@mail.gmail.com> I am learning freeswitch for implementing IVR in this software. In our organization we are using Perl language.So I decided to implement a IVR in Perl on Freeswitch. What are the steps I need to do here. I am also new to IVR.Can you little bit explain about IVR (not basic) how to define a menu like that.. -- Regards, Thangappan.M -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090725/6e3e8b6b/attachment.html From testeador01 at gmail.com Sat Jul 25 07:20:26 2009 From: testeador01 at gmail.com (Milena) Date: Sat, 25 Jul 2009 09:20:26 -0500 Subject: [Freeswitch-users] Command to get FreeSWITCH's internal time Message-ID: Hello everyone, I'm using the inbound event socket to receive some information about the status of my FreeSWITCH system and i wanted to know if there is an api command that can be used to get the FreeSWITCH time, I tried searching around in the docs and in google but i couldn't find an answer. Thanks for your attention and thanks in advance if anyone can assist me with this. Have a nice time and lots of cookies :) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090725/c1150850/attachment.html From asannucci at gmail.com Sat Jul 25 07:40:23 2009 From: asannucci at gmail.com (bakko) Date: Sat, 25 Jul 2009 16:40:23 +0200 Subject: [Freeswitch-users] FS and Nokia E71 Message-ID: <3E9424AC46E1484E9AEEBA20FEA8C46A@voztovoice> Hi all, i have some problems tu make calls from a nokia E71 connect to FS. This is the scenario: Nokia E71 -> NAT - Internet - FS (public IP) I'm using the 5059 UDP port in FS. A can receive calls from others phones to nokia E71 but i can't make calls from nokia E71. If a change the UDP port to 5060 in vars.xml all work fine. I think the problem is the VoIP cllient in the nokia E71 but i dont?t know how resolve the problem. Thank's in advance. BR From talk2ram at gmail.com Sat Jul 25 08:12:35 2009 From: talk2ram at gmail.com (ram) Date: Sat, 25 Jul 2009 20:42:35 +0530 Subject: [Freeswitch-users] Regarding IVR In-Reply-To: <7aa29e790907242250o152c539x16a4b25c1e2e53c0@mail.gmail.com> References: <7aa29e790907242250o152c539x16a4b25c1e2e53c0@mail.gmail.com> Message-ID: On Sat, Jul 25, 2009 at 11:20 AM, Thangappan.M wrote: > I am learning freeswitch for implementing IVR in this software. > In our organization we are using Perl language.So I decided to implement a > IVR in Perl on Freeswitch. > What are the steps I need to do here. > > I am also new to IVR.Can you little bit explain about IVR (not basic) how > to define a menu like that.. > > Hi have you looked this examples http://wiki.freeswitch.org/wiki/SOHO_PBX_Example Ram -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090725/3aaade76/attachment.html From msc at freeswitch.org Sat Jul 25 18:45:36 2009 From: msc at freeswitch.org (Michael S Collins) Date: Sat, 25 Jul 2009 18:45:36 -0700 Subject: [Freeswitch-users] Command to get FreeSWITCH's internal time In-Reply-To: References: Message-ID: How about this: strftime Try it at the CLI -MC Sent from my iPhone On Jul 25, 2009, at 7:20 AM, Milena wrote: > Hello everyone, > > I'm using the inbound event socket to receive some information about > the status of my FreeSWITCH system and i wanted to know if there is > an api command that can be used to get the FreeSWITCH time, I tried > searching around in the docs and in google but i couldn't find an > answer. Thanks for your attention and thanks in advance if anyone > can assist me with this. > > Have a nice time and lots of cookies :) > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mgg at giagnocavo.net Sun Jul 26 00:18:37 2009 From: mgg at giagnocavo.net (Michael Giagnocavo) Date: Sun, 26 Jul 2009 03:18:37 -0400 Subject: [Freeswitch-users] mod_managed users? In-Reply-To: References: <6E8D2069C08AA84A83D336E996AE4C67027C6C33F0@mse17be1.mse17.exchange.ms> <36860.21084.qm@web33505.mail.mud.yahoo.com> Message-ID: <6E8D2069C08AA84A83D336E996AE4C6702C479459E@mse17be1.mse17.exchange.ms> Hello, I just checked in a new mod_managed. It breaks backwards compatibility, but adds scripting and reloading support. I tested it on CentOS 5.3 x64 with Mono 2.4.2.2. Just make & make install seemed to take care of everything. Let me know if you have better luck with this version. Thanks, Michael From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Muhammad Shahzad Sent: Saturday, July 18, 2009 6:58 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_managed users? Sorry for replying late. I have tried mod_managed again on same machine (Lenovo 3000 N200), same OS (Ubuntu-9.04) and Mono version 2.0.1 but with latest FS revision 14249. It compiles correctly this time but gives following error upon "make install", ===================================================================== making install mod_managed make[5]: *** No rule to make target `/usr/local/freeswitch/mod/mod_managed.so', needed by `local_install'. Stop. make[4]: *** [install] Error 1 make[3]: *** [mod_managed-install] Error 1 make[2]: *** [install-recursive] Error 1 ===================================================================== Here is compilation log when executing "make", if it could of any help. ===================================================================== making all mod_managed Compiling freeswitch_managed.cpp... g++ -I/usr/src/svn-src/freeswitch/src/include -I/usr/src/svn-src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE -D_REENTRANT -pthread -I/usr/include/mono-1.0 -I/usr/include/glib-2.0 -I/usr/lib/glib-2.0/include -DHAVE_CONFIG_H -c -o freeswitch_managed.o freeswitch_managed.cpp Compiling freeswitch_wrap.cpp... g++ -I/usr/src/svn-src/freeswitch/src/include -I/usr/src/svn-src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE -D_REENTRANT -pthread -I/usr/include/mono-1.0 -I/usr/include/glib-2.0 -I/usr/lib/glib-2.0/include -DHAVE_CONFIG_H -c -o freeswitch_wrap.o freeswitch_wrap.cpp Demo.cs(58,14): warning CS0169: The private method `FreeSWITCH.Demo.AppDemo.hangupHook()' is never used Compilation succeeded - 1 warning(s) Compiling mod_managed.cpp... /usr/src/svn-src/freeswitch/libtool --mode=compile --tag=CXX g++ -I/usr/src/svn-src/freeswitch/src/include -I/usr/src/svn-src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE -D_REENTRANT -pthread -I/usr/include/mono-1.0 -I/usr/include/glib-2.0 -I/usr/lib/glib-2.0/include -DHAVE_CONFIG_H -c -o mod_managed.lo mod_managed.cpp libtool: compile: g++ -I/usr/src/svn-src/freeswitch/src/include -I/usr/src/svn-src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE -D_REENTRANT -pthread -I/usr/include/mono-1.0 -I/usr/include/glib-2.0 -I/usr/lib/glib-2.0/include -DHAVE_CONFIG_H -c mod_managed.cpp -fPIC -DPIC -o .libs/mod_managed.o mod_managed.cpp: In function ?void InitManagedSession(ManagedSession*, char* (*)(void*, switch_input_type_t), void (*)())?: mod_managed.cpp:97: warning: deprecated conversion from string constant to ?char*? libtool: compile: g++ -I/usr/src/svn-src/freeswitch/src/include -I/usr/src/svn-src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE -D_REENTRANT -pthread -I/usr/include/mono-1.0 -I/usr/include/glib-2.0 -I/usr/lib/glib-2.0/include -DHAVE_CONFIG_H -c mod_managed.cpp -o mod_managed.o >/dev/null 2>&1 Creating mod_managed.la... cat: .libs/mod_managed.log: No such file or directory ===================================================================== Thank you. On Fri, Jul 17, 2009 at 10:59 AM, Muhammad Shahzad > wrote: I tried to install mod_managed on ubuntu-9.04, mono framework version 2.0. It gave me a lots of errors in Loader.cs, which seems to be SWIG related. Since i am not a expert in SWIG so i disabled this module. This happend long ago, i think FS svn revision 136xx. Let me try to compile it from latest FS revision and see if it works. I will let you know the results. Thank you. On Fri, Jul 17, 2009 at 3:54 AM, Diego Toro > wrote: Hey, I am here :) I am working with mod_managed on Windows 2003 and Windows Vista with sucessfull. I noted on user list the issue with LoadFile on Loader.cs when a assembly had reference to others assemblies, I change LoadFile by LoadFrom and the load is made fine. I use c# application and sqlserver 2005, using FS and mod_managed. Diego --- On Thu, 7/16/09, Michael Giagnocavo > wrote: From: Michael Giagnocavo > Subject: [Freeswitch-users] mod_managed users? To: "freeswitch-users at lists.freeswitch.org" > Date: Thursday, July 16, 2009, 4:43 PM Hey, if there are any mod_managed users on this list, I?d love it if you were able to let me know. I?d like to get feedback, positive or negative, on what worked, what didn?t, and how mod_managed can improve for you. Feel free to write on list or directly to me: mgg at giagnocavo.net Thanks! -Michael -----Inline Attachment Follows----- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090726/593d5a26/attachment-0001.html From elihayun at gmail.com Sun Jul 26 00:29:58 2009 From: elihayun at gmail.com (Eli Hayun) Date: Sun, 26 Jul 2009 10:29:58 +0300 Subject: [Freeswitch-users] Limit is not working when originate a call In-Reply-To: <191c3a030907230539i5a640e2do4377d8797ff8f279@mail.gmail.com> References: <4A6709D4.6000704@savion.huji.ac.il> <2C85EB6C-1D80-4DC7-8D14-06B17CD1046D@jerris.com> <4A67EF47.6000402@savion.huji.ac.il> <191c3a030907230539i5a640e2do4377d8797ff8f279@mail.gmail.com> Message-ID: <4A6C05F6.4000501@savion.huji.ac.il> Anthony Minessale wrote: > limit is for inbound calls > you cannot call it after you already made the call. > The correct approach would be to not make the call at all. > > you could maybe use the limit FSAPI interface with apiExecute to check > if the limit was exceeded and > then not bother to place the call to begin with. > > otherwise it's sort of like putting a prisoner in the electric chair > then giving him his trial. > > > On Thu, Jul 23, 2009 at 12:04 AM, Eli Hayun > wrote: > > Michael Jerris wrote: > > because your not running limit at all when you are doing an > originate > > directly. You can use loopback to originate through a dialplan > > extension. > > > > Mike > > > > On Jul 22, 2009, at 8:45 AM, Eli Hayun wrote: > > > > > >> Hi > >> I set the limit to 1 on the extension like that > >> > >> > >> > >> When I am trying to make a call the that destination i > transfered to > >> limit_exceeded dialplan, just like I want > >> > >> The problem is, that when I am trying to make a call using > >> "originate" I > >> am not getting the limitation. > >> Why is 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 > >> > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > Thanks for answer. > I am calling "Originate" from JS. I tried to call "limit_hash" from JS > but with no success. I did it like that: > > lmt = apiExecute("limit_hash", dialed_ext + " " + dialed_ext + " 1"); > > I could't find any documentation on that. > can u help ? > > Thanks > Eli > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 Thanks for replay, but how do I do that? I tried to use : lmt = apiExecute("limit_hash", extno + " " + extno + " 1"); console_log("info","*** Limit ***" + lmt + "\n"); But it gave me "Invalid command". What is the exact way to do that. The documentation on that, is missing. From saeedahmad1981 at gmail.com Sun Jul 26 02:27:54 2009 From: saeedahmad1981 at gmail.com (Saeed Ahmad) Date: Sun, 26 Jul 2009 11:27:54 +0200 Subject: [Freeswitch-users] Mod_Limit - Limiting different destinations on same IP Message-ID: Dear All, Is it possible to apply different limits on a same IP by different destinations. Example: IP: 1.2.3.4 Destination: Germany Mobile (491) => max-channels=10 Destination: Germany (49) => max-channel=20 Sometimes the supplier provides limited capacity on different destinations; so in this case its necessary to apply the limit so that after limit exceeds the call can go to next endpoint. Thanks Saeed. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090726/f3ddd1b6/attachment.html From krice at freeswitch.org Sun Jul 26 02:47:37 2009 From: krice at freeswitch.org (Ken Rice) Date: Sun, 26 Jul 2009 04:47:37 -0500 Subject: [Freeswitch-users] Mod_Limit - Limiting different destinations on same IP In-Reply-To: Message-ID: They don?t just 503 the call once you hit the limit? And look at the options for using limit... Be a little creative and you can do just want you want to do with some regex From: Saeed Ahmad Reply-To: Date: Sun, 26 Jul 2009 11:27:54 +0200 To: Subject: [Freeswitch-users] Mod_Limit - Limiting different destinations on same IP Dear All, Is it possible to apply different limits on a same IP by different destinations. Example: IP: 1.2.3.4 Destination: Germany??Mobile (491) =>?max-channels=10 Destination: Germany (49) => max-channel=20 ? Sometimes the supplier provides limited capacity on different destinations; so in this case its necessary to apply the limit so that after limit exceeds the call can go to next endpoint. ? Thanks Saeed. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at 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/20090726/a0ecb6d5/attachment.html From saeedahmad1981 at gmail.com Sun Jul 26 03:33:06 2009 From: saeedahmad1981 at gmail.com (Saeed Ahmad) Date: Sun, 26 Jul 2009 12:33:06 +0200 Subject: [Freeswitch-users] Mod_Limit - Limiting different destinations on same IP In-Reply-To: References: Message-ID: Can mod limit also be used to apply limit on outbound supplier ip? i was unable to find a way to do that... - Saeed On Sun, Jul 26, 2009 at 11:47 AM, Ken Rice wrote: > They don?t just 503 the call once you hit the limit? > > And look at the options for using limit... Be a little creative and you can > do just want you want to do with some regex > > ------------------------------ > *From: *Saeed Ahmad > *Reply-To: * > *Date: *Sun, 26 Jul 2009 11:27:54 +0200 > *To: * > *Subject: *[Freeswitch-users] Mod_Limit - Limiting different destinations > on same IP > > > Dear All, > > Is it possible to apply different limits on a same IP by different > destinations. > > Example: > > IP: 1.2.3.4 > Destination: Germany Mobile (491) => max-channels=10 > Destination: Germany (49) => max-channel=20 > > Sometimes the supplier provides limited capacity on different destinations; > so in this case its necessary to apply the limit so that after limit exceeds > the call can go to next endpoint. > > Thanks > Saeed. > > ------------------------------ > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20090726/b2ebc6fe/attachment.html From shaheryarkh at googlemail.com Sun Jul 26 05:19:43 2009 From: shaheryarkh at googlemail.com (Muhammad Shahzad) Date: Sun, 26 Jul 2009 17:19:43 +0500 Subject: [Freeswitch-users] A possible bug in FS causing Linux Kernel crash Message-ID: Hi, I am having random Linux Kernel crash problems while running FreeSWITCH as Skype to/from SIP gateway on one of our production servers. This machine is running CentOS 5.2, Kernel version 2.6.18-92.1.22.el5.centos.plusPAE with FS svn revision number 13754. At time of Kernel crash i could find following crash messages which point to some source code file in FS source tree. --------------------- Kernel Begin ------------------------ 3 Time(s): ======================= 3 Time(s): [] syscall_call+0x7/0xb 3 Time(s): [] sys_delete_module+0x192/0x1b8 3 Time(s): [] audit_syscall_entry+0x14b/0x17d 3 Time(s): [] remove_proc_entry+0x139/0x18c 3 Time(s): [] alsa_sound_exit+0xa/0x30 [snd] 3 Time(s): [] snd_info_done+0x46/0x49 [snd] 3 Time(s): BUG: warning at fs/proc/generic.c:732/remove_proc_entry() (Not tainted) 1 Time(s): snd-malloc: Memory leak? pages not freed = 1 ---------------------- Kernel End ------------------------- While the problem seems to arise from ALSA kernel module but it blames FS file fs/proc/generic.c:732 for this. The only FS module that is using ALSA is mod_skypiax but as far as i remember that module is using FS internal routines to allocate and de-allocate sound driver services for Skype client. Please suggest a solution. Thank you. -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090726/b0e5d401/attachment.html From gmaruzz at celliax.org Sun Jul 26 05:37:39 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Sun, 26 Jul 2009 14:37:39 +0200 Subject: [Freeswitch-users] A possible bug in FS causing Linux Kernel crash In-Reply-To: References: Message-ID: <7b197bef0907260537n5b88a583g3a37139a64aa452f@mail.gmail.com> Ciao Muhammad, I've got many problems with ALSA drivers, including various kind of crashes. To make a looong story short, use the alsa_drivers version 1.0.20, they have not yet crashed on me. Also, if you want to test it, you can compile the customized snd-dummy driver you find in the svn code, it is a try to have much more efficiency bot in softirqs and context switches, allows for 64 Skype instances (128 subdevices), etc. it is to be compiled with alsa_drivers 1.0.20 too. Is my feeling (I mean, almost sure) they got spin_locking wrong in previous versions, and it crashes the kernel when you "really" use it (Skype clients have a demented usage of alsa). BTW, I'm in the process of revamp the code, fix the bugs and apply patches. Please, have a look at the new wiki page with lots of new content, I'll send a mail to the ML tomorrow :-) 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 Sun, Jul 26, 2009 at 2:19 PM, Muhammad Shahzad wrote: > Hi, > > I am having random Linux Kernel crash problems while running FreeSWITCH as > Skype to/from SIP gateway on one of our production servers. This machine is > running CentOS 5.2, Kernel version 2.6.18-92.1.22.el5.centos.plusPAE with FS > svn revision number 13754. > > At time of Kernel crash i could find following crash messages which point to > some source code file in FS source tree. > > ?--------------------- Kernel Begin ------------------------ > > > ?3 Time(s):? ======================= > ?3 Time(s):? [] syscall_call+0x7/0xb > ?3 Time(s):? [] sys_delete_module+0x192/0x1b8 > ?3 Time(s):? [] audit_syscall_entry+0x14b/0x17d > ?3 Time(s):? [] remove_proc_entry+0x139/0x18c > ?3 Time(s):? [] alsa_sound_exit+0xa/0x30 [snd] > ?3 Time(s):? [] snd_info_done+0x46/0x49 [snd] > ?3 Time(s): BUG: warning at fs/proc/generic.c:732/remove_proc_entry() (Not > tainted) > ?1 Time(s): snd-malloc: Memory leak?? pages not freed = 1 > > ?---------------------- Kernel End ------------------------- > > While the problem seems to arise from ALSA kernel module but it blames FS > file fs/proc/generic.c:732 for this. The only FS module that is using ALSA > is mod_skypiax but as far as i remember that module is using FS internal > routines to allocate and de-allocate sound driver services for Skype client. > > Please suggest a solution. > > Thank you. > > > -- > Muhammad Shahzad > ----------------------------------- > CISCO Rich Media Communication Specialist (CRMCS) > CISCO Certified Network Associate (CCNA) > Cell: +92 334 422 40 88 > MSN: shari_786pk at hotmail.com > Email: shaheryarkh at googlemail.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 gmaruzz at celliax.org Sun Jul 26 05:40:11 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Sun, 26 Jul 2009 14:40:11 +0200 Subject: [Freeswitch-users] A possible bug in FS causing Linux Kernel crash In-Reply-To: <7b197bef0907260537n5b88a583g3a37139a64aa452f@mail.gmail.com> References: <7b197bef0907260537n5b88a583g3a37139a64aa452f@mail.gmail.com> Message-ID: <7b197bef0907260540v434463ck5fc2943074a0e5bc@mail.gmail.com> Performance problems and other issues (eg crashes on ALSA drivers) has been reported for Skypiax on CentOS, albeit various users got good success on same CentOS. The section down below, "Extreme" Performances on Linux solves all problems for the user that got issues on CentOS. http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk#.22Extreme.22_Performances_on_Linux On Sun, Jul 26, 2009 at 2:37 PM, Giovanni Maruzzelli wrote: > Ciao Muhammad, > > I've got many problems with ALSA drivers, including various kind of crashes. > > To make a looong story short, use the alsa_drivers version 1.0.20, > they have not yet crashed on me. > > Also, if you want to test it, you can compile the customized snd-dummy > driver you find in the svn code, it is a try to have much more > efficiency bot in softirqs and context switches, allows for 64 Skype > instances (128 subdevices), etc. it is to be compiled with > alsa_drivers 1.0.20 too. > > Is my feeling (I mean, almost sure) they got spin_locking wrong in > previous versions, and it crashes the kernel when you "really" use it > (Skype clients have a demented usage of alsa). > > BTW, I'm in the process of revamp the code, fix the bugs and apply > patches. Please, have a look at the new wiki page with lots of new > content, I'll send a mail to the ML tomorrow :-) > > > 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 Sun, Jul 26, 2009 at 2:19 PM, Muhammad > Shahzad wrote: >> Hi, >> >> I am having random Linux Kernel crash problems while running FreeSWITCH as >> Skype to/from SIP gateway on one of our production servers. This machine is >> running CentOS 5.2, Kernel version 2.6.18-92.1.22.el5.centos.plusPAE with FS >> svn revision number 13754. >> >> At time of Kernel crash i could find following crash messages which point to >> some source code file in FS source tree. >> >> ?--------------------- Kernel Begin ------------------------ >> >> >> ?3 Time(s):? ======================= >> ?3 Time(s):? [] syscall_call+0x7/0xb >> ?3 Time(s):? [] sys_delete_module+0x192/0x1b8 >> ?3 Time(s):? [] audit_syscall_entry+0x14b/0x17d >> ?3 Time(s):? [] remove_proc_entry+0x139/0x18c >> ?3 Time(s):? [] alsa_sound_exit+0xa/0x30 [snd] >> ?3 Time(s):? [] snd_info_done+0x46/0x49 [snd] >> ?3 Time(s): BUG: warning at fs/proc/generic.c:732/remove_proc_entry() (Not >> tainted) >> ?1 Time(s): snd-malloc: Memory leak?? pages not freed = 1 >> >> ?---------------------- Kernel End ------------------------- >> >> While the problem seems to arise from ALSA kernel module but it blames FS >> file fs/proc/generic.c:732 for this. The only FS module that is using ALSA >> is mod_skypiax but as far as i remember that module is using FS internal >> routines to allocate and de-allocate sound driver services for Skype client. >> >> Please suggest a solution. >> >> Thank you. >> >> >> -- >> Muhammad Shahzad >> ----------------------------------- >> CISCO Rich Media Communication Specialist (CRMCS) >> CISCO Certified Network Associate (CCNA) >> Cell: +92 334 422 40 88 >> MSN: shari_786pk at hotmail.com >> Email: shaheryarkh at googlemail.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 shaheryarkh at googlemail.com Sun Jul 26 05:44:02 2009 From: shaheryarkh at googlemail.com (Muhammad Shahzad) Date: Sun, 26 Jul 2009 17:44:02 +0500 Subject: [Freeswitch-users] A possible bug in FS causing Linux Kernel crash In-Reply-To: <7b197bef0907260540v434463ck5fc2943074a0e5bc@mail.gmail.com> References: <7b197bef0907260537n5b88a583g3a37139a64aa452f@mail.gmail.com> <7b197bef0907260540v434463ck5fc2943074a0e5bc@mail.gmail.com> Message-ID: Thanks. Let me try it and let you know the results. Thank you. On Sun, Jul 26, 2009 at 5:40 PM, Giovanni Maruzzelli wrote: > Performance problems and other issues (eg crashes on ALSA drivers) has > been reported for Skypiax on CentOS, albeit various users got good > success on same CentOS. The section down below, "Extreme" Performances > on Linux solves all problems for the user that got issues on CentOS. > > > http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk#.22Extreme.22_Performances_on_Linux > > > > On Sun, Jul 26, 2009 at 2:37 PM, Giovanni Maruzzelli > wrote: > > Ciao Muhammad, > > > > I've got many problems with ALSA drivers, including various kind of > crashes. > > > > To make a looong story short, use the alsa_drivers version 1.0.20, > > they have not yet crashed on me. > > > > Also, if you want to test it, you can compile the customized snd-dummy > > driver you find in the svn code, it is a try to have much more > > efficiency bot in softirqs and context switches, allows for 64 Skype > > instances (128 subdevices), etc. it is to be compiled with > > alsa_drivers 1.0.20 too. > > > > Is my feeling (I mean, almost sure) they got spin_locking wrong in > > previous versions, and it crashes the kernel when you "really" use it > > (Skype clients have a demented usage of alsa). > > > > BTW, I'm in the process of revamp the code, fix the bugs and apply > > patches. Please, have a look at the new wiki page with lots of new > > content, I'll send a mail to the ML tomorrow :-) > > > > > > 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 Sun, Jul 26, 2009 at 2:19 PM, Muhammad > > Shahzad wrote: > >> Hi, > >> > >> I am having random Linux Kernel crash problems while running FreeSWITCH > as > >> Skype to/from SIP gateway on one of our production servers. This machine > is > >> running CentOS 5.2, Kernel version 2.6.18-92.1.22.el5.centos.plusPAE > with FS > >> svn revision number 13754. > >> > >> At time of Kernel crash i could find following crash messages which > point to > >> some source code file in FS source tree. > >> > >> --------------------- Kernel Begin ------------------------ > >> > >> > >> 3 Time(s): ======================= > >> 3 Time(s): [] syscall_call+0x7/0xb > >> 3 Time(s): [] sys_delete_module+0x192/0x1b8 > >> 3 Time(s): [] audit_syscall_entry+0x14b/0x17d > >> 3 Time(s): [] remove_proc_entry+0x139/0x18c > >> 3 Time(s): [] alsa_sound_exit+0xa/0x30 [snd] > >> 3 Time(s): [] snd_info_done+0x46/0x49 [snd] > >> 3 Time(s): BUG: warning at fs/proc/generic.c:732/remove_proc_entry() > (Not > >> tainted) > >> 1 Time(s): snd-malloc: Memory leak? pages not freed = 1 > >> > >> ---------------------- Kernel End ------------------------- > >> > >> While the problem seems to arise from ALSA kernel module but it blames > FS > >> file fs/proc/generic.c:732 for this. The only FS module that is using > ALSA > >> is mod_skypiax but as far as i remember that module is using FS internal > >> routines to allocate and de-allocate sound driver services for Skype > client. > >> > >> Please suggest a solution. > >> > >> Thank you. > >> > >> > >> -- > >> Muhammad Shahzad > >> ----------------------------------- > >> CISCO Rich Media Communication Specialist (CRMCS) > >> CISCO Certified Network Associate (CCNA) > >> Cell: +92 334 422 40 88 > >> MSN: shari_786pk at hotmail.com > >> Email: shaheryarkh at googlemail.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 > -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090726/22a45ab5/attachment.html From gmaruzz at celliax.org Sun Jul 26 06:26:51 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Sun, 26 Jul 2009 15:26:51 +0200 Subject: [Freeswitch-users] A possible bug in FS causing Linux Kernel crash In-Reply-To: References: Message-ID: <7b197bef0907260626u5e273b6bj235a7edb3fdfd949@mail.gmail.com> On Sun, Jul 26, 2009 at 2:19 PM, Muhammad Shahzad wrote: > Hi, > > I am having random Linux Kernel crash problems while running FreeSWITCH as > Skype to/from SIP gateway on one of our production servers. This machine is > running CentOS 5.2, Kernel version 2.6.18-92.1.22.el5.centos.plusPAE with FS > svn revision number 13754. > > While the problem seems to arise from ALSA kernel module but it blames FS > file fs/proc/generic.c:732 for this. The only FS module that is using ALSA > is mod_skypiax but as far as i remember that module is using FS internal > routines to allocate and de-allocate sound driver services for Skype client. Also, please note that neither mod_skypiax nor FreeSWITCH have nothing to do with ALSA (eg: no ALSA code at all in mod_skypiax or FreeSWITCH). Is the Skype client instance that uses the sound driver, just like on a desktop Skype client usage The Skype client instances are started by a shell script, but you could as well start them from the command line, and are completely autonomous from FreeSWITCH (FS do not allocate or deallocate sound driver services for them). Summary: it's just the ALSA drivers that are to blame :-) -giovanni From dftoro at yahoo.com Sun Jul 26 07:47:28 2009 From: dftoro at yahoo.com (Diego Toro) Date: Sun, 26 Jul 2009 07:47:28 -0700 (PDT) Subject: [Freeswitch-users] mod_managed users? In-Reply-To: <6E8D2069C08AA84A83D336E996AE4C6702C479459E@mse17be1.mse17.exchange.ms> Message-ID: <250325.39922.qm@web33504.mail.mud.yahoo.com> Hi Michael, ? Thank you for your job with mod_managed, I get lastest version with mod_managed but the files PluginInterfaces.cs, PluginManager.cs and ScriptPluginManager.cs were not downloaded. ? Diego --- On Sun, 7/26/09, Michael Giagnocavo wrote: From: Michael Giagnocavo Subject: Re: [Freeswitch-users] mod_managed users? To: "freeswitch-users at lists.freeswitch.org" Date: Sunday, July 26, 2009, 2:18 AM Hello, ? ??????????????? I just checked in a new mod_managed. It breaks backwards compatibility, but adds scripting and reloading support. ? ??????????????? I tested it on CentOS 5.3 x64 with Mono 2.4.2.2. Just make & make install seemed to take care of everything. ? ??????????????? Let me know if you have better luck with this version. ? Thanks, Michael ? From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Muhammad Shahzad Sent: Saturday, July 18, 2009 6:58 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_managed users? ? Sorry for replying late. I have tried mod_managed again on same machine (Lenovo 3000 N200), same OS (Ubuntu-9.04) and Mono version 2.0.1 but with latest FS revision 14249. It compiles correctly this time but gives following error upon "make install", ===================================================================== making install mod_managed make[5]: *** No rule to make target `/usr/local/freeswitch/mod/mod_managed.so', needed by `local_install'.? Stop. make[4]: *** [install] Error 1 make[3]: *** [mod_managed-install] Error 1 make[2]: *** [install-recursive] Error 1 ===================================================================== Here is compilation log when executing "make", if it could of any help. ===================================================================== making all mod_managed Compiling freeswitch_managed.cpp... g++ -I/usr/src/svn-src/freeswitch/src/include -I/usr/src/svn-src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE -D_REENTRANT -pthread -I/usr/include/mono-1.0 -I/usr/include/glib-2.0 -I/usr/lib/glib-2.0/include -DHAVE_CONFIG_H -c -o freeswitch_managed.o freeswitch_managed.cpp Compiling freeswitch_wrap.cpp... g++ -I/usr/src/svn-src/freeswitch/src/include -I/usr/src/svn-src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE -D_REENTRANT -pthread -I/usr/include/mono-1.0 -I/usr/include/glib-2.0 -I/usr/lib/glib-2.0/include -DHAVE_CONFIG_H -c -o freeswitch_wrap.o freeswitch_wrap.cpp Demo.cs(58,14): warning CS0169: The private method `FreeSWITCH.Demo.AppDemo.hangupHook()' is never used Compilation succeeded - 1 warning(s) Compiling mod_managed.cpp... /usr/src/svn-src/freeswitch/libtool --mode=compile --tag=CXX g++ -I/usr/src/svn-src/freeswitch/src/include -I/usr/src/svn-src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE -D_REENTRANT -pthread -I/usr/include/mono-1.0 -I/usr/include/glib-2.0 -I/usr/lib/glib-2.0/include -DHAVE_CONFIG_H -c -o mod_managed.lo mod_managed.cpp libtool: compile:? g++ -I/usr/src/svn-src/freeswitch/src/include -I/usr/src/svn-src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE -D_REENTRANT -pthread -I/usr/include/mono-1.0 -I/usr/include/glib-2.0 -I/usr/lib/glib-2.0/include -DHAVE_CONFIG_H -c mod_managed.cpp? -fPIC -DPIC -o .libs/mod_managed.o mod_managed.cpp: In function ?void InitManagedSession(ManagedSession*, char* (*)(void*, switch_input_type_t), void (*)())?: mod_managed.cpp:97: warning: deprecated conversion from string constant to ?char*? libtool: compile:? g++ -I/usr/src/svn-src/freeswitch/src/include -I/usr/src/svn-src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE -D_REENTRANT -pthread -I/usr/include/mono-1.0 -I/usr/include/glib-2.0 -I/usr/lib/glib-2.0/include -DHAVE_CONFIG_H -c mod_managed.cpp -o mod_managed.o >/dev/null 2>&1 Creating mod_managed.la... cat: .libs/mod_managed.log: No such file or directory ===================================================================== Thank you. On Fri, Jul 17, 2009 at 10:59 AM, Muhammad Shahzad wrote: I tried to install mod_managed on ubuntu-9.04, mono framework version 2.0. It gave me a lots of errors in Loader.cs, which seems to be SWIG related. Since i am not a expert in SWIG so i disabled this module. This happend long ago, i think FS svn revision 136xx. Let me try to compile it from latest FS revision and see if it works. I will let you know the results. Thank you. On Fri, Jul 17, 2009 at 3:54 AM, Diego Toro wrote: Hey, I am here? :) ? I am working with mod_managed on Windows 2003 and Windows Vista with sucessfull.? I noted on user list the issue with LoadFile on Loader.cs when a assembly had reference to others assemblies, I change LoadFile by LoadFrom and the load is made fine. ? I use c# application and sqlserver 2005, using FS and mod_managed. ? Diego --- On Thu, 7/16/09, Michael Giagnocavo wrote: From: Michael Giagnocavo Subject: [Freeswitch-users] mod_managed users? To: "freeswitch-users at lists.freeswitch.org" Date: Thursday, July 16, 2009, 4:43 PM ? Hey, if there are any mod_managed users on this list, I?d love it if you were able to let me know. I?d like to get feedback, positive or negative, on what worked, what didn?t, and how mod_managed can improve for you. Feel free to write on list or directly to me: mgg at giagnocavo.net ? Thanks! -Michael -----Inline Attachment Follows----- ? _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -----Inline Attachment Follows----- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at 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/20090726/92aee1cf/attachment-0001.html From raul at etellicom.com Sun Jul 26 09:25:01 2009 From: raul at etellicom.com (Raul Fragoso) Date: Sun, 26 Jul 2009 13:25:01 -0300 Subject: [Freeswitch-users] Command to get FreeSWITCH's internal time In-Reply-To: References: Message-ID: <1248625501.5655.79.camel@raul-laptop> Every event has a header 'Event-Date-Local', which has the local date and time. If want to actively retrieve the date and time, you can send this API request to the server: api strftime Regards, Raul On Sat, 2009-07-25 at 09:20 -0500, Milena wrote: > Hello everyone, > > I'm using the inbound event socket to receive some information about > the status of my FreeSWITCH system and i wanted to know if there is an > api command that can be used to get the FreeSWITCH time, I tried > searching around in the docs and in google but i couldn't find an > answer. Thanks for your attention and thanks in advance if anyone can > assist me with this. > > Have a nice time and lots of cookies :) > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mgg at giagnocavo.net Sun Jul 26 09:37:05 2009 From: mgg at giagnocavo.net (Michael Giagnocavo) Date: Sun, 26 Jul 2009 12:37:05 -0400 Subject: [Freeswitch-users] mod_managed users? In-Reply-To: <250325.39922.qm@web33504.mail.mud.yahoo.com> References: <6E8D2069C08AA84A83D336E996AE4C6702C479459E@mse17be1.mse17.exchange.ms> <250325.39922.qm@web33504.mail.mud.yahoo.com> Message-ID: <6E8D2069C08AA84A83D336E996AE4C6702C47945B3@mse17be1.mse17.exchange.ms> Ah, that?s embarrassing. I added them and tried building FreeSWITCH.Managed from svn and it worked fine now. (I?ll kick off a new complete build in a minute.) -Michael From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Diego Toro Sent: Sunday, July 26, 2009 8:47 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_managed users? Hi Michael, Thank you for your job with mod_managed, I get lastest version with mod_managed but the files PluginInterfaces.cs, PluginManager.cs and ScriptPluginManager.cs were not downloaded. Diego --- On Sun, 7/26/09, Michael Giagnocavo wrote: From: Michael Giagnocavo Subject: Re: [Freeswitch-users] mod_managed users? To: "freeswitch-users at lists.freeswitch.org" Date: Sunday, July 26, 2009, 2:18 AM Hello, I just checked in a new mod_managed. It breaks backwards compatibility, but adds scripting and reloading support. I tested it on CentOS 5.3 x64 with Mono 2.4.2.2. Just make & make install seemed to take care of everything. Let me know if you have better luck with this version. Thanks, Michael From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Muhammad Shahzad Sent: Saturday, July 18, 2009 6:58 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_managed users? Sorry for replying late. I have tried mod_managed again on same machine (Lenovo 3000 N200), same OS (Ubuntu-9.04) and Mono version 2.0.1 but with latest FS revision 14249. It compiles correctly this time but gives following error upon "make install", ===================================================================== making install mod_managed make[5]: *** No rule to make target `/usr/local/freeswitch/mod/mod_managed.so', needed by `local_install'. Stop. make[4]: *** [install] Error 1 make[3]: *** [mod_managed-install] Error 1 make[2]: *** [install-recursive] Error 1 ===================================================================== Here is compilation log when executing "make", if it could of any help. ===================================================================== making all mod_managed Compiling freeswitch_managed.cpp... g++ -I/usr/src/svn-src/freeswitch/src/include -I/usr/src/svn-src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE -D_REENTRANT -pthread -I/usr/include/mono-1.0 -I/usr/include/glib-2.0 -I/usr/lib/glib-2.0/include -DHAVE_CONFIG_H -c -o freeswitch_managed.o freeswitch_managed.cpp Compiling freeswitch_wrap.cpp... g++ -I/usr/src/svn-src/freeswitch/src/include -I/usr/src/svn-src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE -D_REENTRANT -pthread -I/usr/include/mono-1.0 -I/usr/include/glib-2.0 -I/usr/lib/glib-2.0/include -DHAVE_CONFIG_H -c -o freeswitch_wrap.o freeswitch_wrap.cpp Demo.cs(58,14): warning CS0169: The private method `FreeSWITCH.Demo.AppDemo.hangupHook()' is never used Compilation succeeded - 1 warning(s) Compiling mod_managed.cpp... /usr/src/svn-src/freeswitch/libtool --mode=compile --tag=CXX g++ -I/usr/src/svn-src/freeswitch/src/include -I/usr/src/svn-src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE -D_REENTRANT -pthread -I/usr/include/mono-1.0 -I/usr/include/glib-2.0 -I/usr/lib/glib-2.0/include -DHAVE_CONFIG_H -c -o mod_managed.lo mod_managed.cpp libtool: compile: g++ -I/usr/src/svn-src/freeswitch/src/include -I/usr/src/svn-src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE -D_REENTRANT -pthread -I/usr/include/mono-1.0 -I/usr/include/glib-2.0 -I/usr/lib/glib-2.0/include -DHAVE_CONFIG_H -c mod_managed.cpp -fPIC -DPIC -o .libs/mod_managed.o mod_managed.cpp: In function ?void InitManagedSession(ManagedSession*, char* (*)(void*, switch_input_type_t), void (*)())?: mod_managed.cpp:97: warning: deprecated conversion from string constant to ?char*? libtool: compile: g++ -I/usr/src/svn-src/freeswitch/src/include -I/usr/src/svn-src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE -D_REENTRANT -pthread -I/usr/include/mono-1.0 -I/usr/include/glib-2.0 -I/usr/lib/glib-2.0/include -DHAVE_CONFIG_H -c mod_managed.cpp -o mod_managed.o >/dev/null 2>&1 Creating mod_managed.la... cat: .libs/mod_managed.log: No such file or directory ===================================================================== Thank you. On Fri, Jul 17, 2009 at 10:59 AM, Muhammad Shahzad > wrote: I tried to install mod_managed on ubuntu-9.04, mono framework version 2.0. It gave me a lots of errors in Loader.cs, which seems to be SWIG related. Since i am not a expert in SWIG so i disabled this module. This happend long ago, i think FS svn revision 136xx. Let me try to compile it from latest FS revision and see if it works. I will let you know the results. Thank you. On Fri, Jul 17, 2009 at 3:54 AM, Diego Toro > wrote: Hey, I am here :) I am working with mod_managed on Windows 2003 and Windows Vista with sucessfull. I noted on user list the issue with LoadFile on Loader.cs when a assembly had reference to others assemblies, I change LoadFile by LoadFrom and the load is made fine. I use c# application and sqlserver 2005, using FS and mod_managed. Diego --- On Thu, 7/16/09, Michael Giagnocavo > wrote: From: Michael Giagnocavo > Subject: [Freeswitch-users] mod_managed users? To: "freeswitch-users at lists.freeswitch.org" > Date: Thursday, July 16, 2009, 4:43 PM Hey, if there are any mod_managed users on this list, I?d love it if you were able to let me know. I?d like to get feedback, positive or negative, on what worked, what didn?t, and how mod_managed can improve for you. Feel free to write on list or directly to me: mgg at giagnocavo.net Thanks! -Michael -----Inline Attachment Follows----- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -----Inline Attachment Follows----- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at 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/20090726/1f1e9e80/attachment-0001.html From a.afzali2003 at gmail.com Sun Jul 26 09:00:21 2009 From: a.afzali2003 at gmail.com (afshin afzali) Date: Sun, 26 Jul 2009 20:30:21 +0430 Subject: [Freeswitch-users] Sofia SIP Subscription For External Events Message-ID: Hi Guys, I'm going to use OpenSER as SIP Platform (Registrar, Proxy, Presence) and FreeSWITCH in my application. So I need to subscribe the FreeSWITCH for presence information of users who involve in the application. After some looking of Sofia.conf.xml , it seems there is not support to doing so, Is it right ? appreciate all comments, -- afshin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090726/fc91d5f1/attachment.html From thangappan143 at gmail.com Sun Jul 26 22:07:10 2009 From: thangappan143 at gmail.com (Thangappan.M) Date: Mon, 27 Jul 2009 10:37:10 +0530 Subject: [Freeswitch-users] Which method Can I use in IVR Message-ID: <7aa29e790907262207n5d50c96cl2f19e2d152399db6@mail.gmail.com> Dear all, I am learning how to implement a IVR in Freeswitch.In our organization we are using Perl scripting language for doing this.So In freeswitch also I need to use Perl. So far I heard two methods for executing IVR. One is in dial plan using perl application.( In perl I create IVR menu and play the voice files) Another one is using event socket.In dial plan I specified socket application and write a Perl script which is listening that particular port and get the session Id. Have I understood correctly?.If it is correct means tell which method can I use?. Other make me understand well. I have seen downloaded perl IVR menu from freeswitch site.In that they called some internal functions like playandGetDigits,StreamFile,ready ...etc. These functions is been called by using $session variable.Where these functions are defined.? -- Regards, Thangappan.M -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090727/bf394aa0/attachment.html From velu.technical at gmail.com Sun Jul 26 22:23:19 2009 From: velu.technical at gmail.com (velusamy velu) Date: Mon, 27 Jul 2009 10:53:19 +0530 Subject: [Freeswitch-users] IAX configurations Message-ID: <1452e2980907262223u677eef21o282f39e2a56209cc@mail.gmail.com> Dear All, I have tried to call a Asterisk extension from FreeSWITCH. I have done the following configurations, * I have enabled mod_iax module in modules.conf.xml file. * Next I have configure following extension in dialplan. * Next I have configured a 222 user in sip.conf file at Asterisk machine. * I wrote dialplan for that extension in extension.conf file. When I tried to call 222 from FreeSWITCH, I have received following error in Console. "[ERR] switch_core_session.c:255 switch_core_session_outgoing_channel() Could not locate channel type iax" What would be the problem? Is there any configuration I missed? Please help me ..... Regards, K.Velusamy. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090727/9d12c4ca/attachment.html From mike at jerris.com Sun Jul 26 22:41:58 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 27 Jul 2009 01:41:58 -0400 Subject: [Freeswitch-users] IAX configurations In-Reply-To: <1452e2980907262223u677eef21o282f39e2a56209cc@mail.gmail.com> References: <1452e2980907262223u677eef21o282f39e2a56209cc@mail.gmail.com> Message-ID: <98E444F7-ECF8-40DF-82D2-89A48DC11EAC@jerris.com> mod_iax isn't loaded. I suggest using sip anyways. Mike On Jul 27, 2009, at 1:23 AM, velusamy velu wrote: > Dear All, > I have tried to call a Asterisk extension from FreeSWITCH. I > have done the following configurations, > * I have enabled mod_iax module in > modules.conf.xml file. > * Next I have configure following > extension in dialplan. > > field="destination_number" expression="^(222)$"> > application="bridge" data="iax/222:222 at 192.168.6.94/$1"/> > > > * Next I have configured a 222 user in > sip.conf file at Asterisk machine. > * I wrote dialplan for that extension in > extension.conf file. > > When I tried to call 222 from FreeSWITCH, I have received > following error in Console. > "[ERR] switch_core_session.c:255 > switch_core_session_outgoing_channel() Could not locate channel type > iax" > > What would be the problem? Is there any configuration I > missed? Please help me ..... > > Regards, > K.Velusamy. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20090727/e83ccb54/attachment.html From velu.technical at gmail.com Sun Jul 26 22:59:47 2009 From: velu.technical at gmail.com (velusamy velu) Date: Mon, 27 Jul 2009 11:29:47 +0530 Subject: [Freeswitch-users] IAX configurations In-Reply-To: <98E444F7-ECF8-40DF-82D2-89A48DC11EAC@jerris.com> References: <1452e2980907262223u677eef21o282f39e2a56209cc@mail.gmail.com> <98E444F7-ECF8-40DF-82D2-89A48DC11EAC@jerris.com> Message-ID: <1452e2980907262259k30713f4ft363592ac77ee3e33@mail.gmail.com> I have loaded mod_iax now that error didn't come. But, When I call I have received following message in the console. "[INFO] mod_dptools.c:1998 audio_bridge_function() Originate Failed. Cause: FACILITY_REJECTED" What configuration I missed? How to use sip to connect the Asterisk? Please give solutions above questions... __ Velusamy On Mon, Jul 27, 2009 at 11:11 AM, Michael Jerris wrote: > mod_iax isn't loaded. I suggest using sip anyways. > Mike > > On Jul 27, 2009, at 1:23 AM, velusamy velu wrote: > > Dear All, > I have tried to call a Asterisk extension from FreeSWITCH. I have done > the following configurations, > * I have enabled mod_iax module in > modules.conf.xml file. > * Next I have configure following extension in > dialplan. > > field="destination_number" expression="^(222)$"> > data="iax/222:222 at 192.168.6.94/$1"/> > > > * Next I have configured a 222 user in sip.conf > file at Asterisk machine. > * I wrote dialplan for that extension in > extension.conf file. > > When I tried to call 222 from FreeSWITCH, I have received following > error in Console. > "[ERR] switch_core_session.c:255 > switch_core_session_outgoing_channel() Could not locate channel type iax" > > What would be the problem? Is there any configuration I missed? > Please help me ..... > > Regards, > K.Velusamy. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20090727/a7d0e128/attachment.html From yudha2008 at gmail.com Sun Jul 26 23:40:42 2009 From: yudha2008 at gmail.com (Baskar) Date: Mon, 27 Jul 2009 12:10:42 +0530 Subject: [Freeswitch-users] core dump Message-ID: Hi, I get core dump segmentation fault in freeswitch machine frequently. can any one assist me what is error in the freeswitch. i have pasted the logs in freeswitch pastebin. This is the link http://pastebin.freeswitch.org/9851 -- Thanks with Regards, N.Baskar -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090727/5e4bfde5/attachment.html From darklion11 at yahoo.com Sun Jul 26 23:41:29 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Sun, 26 Jul 2009 23:41:29 -0700 (PDT) Subject: [Freeswitch-users] Multiple gateways: Priority the first bridge with prefix Message-ID: <24674260.post@talk.nabble.com> Hi FS Users, I just want to try multiple gateways. It works actually like this... But I test call like 5133333 at 222.333.444.555, it also calls the second bridge 111.222.333.333. It there any way to determine which prefix will call to a bridge specified. E.g. for bridge 1: with prefix of 51 the call with run to 222.333.444.555 not at the second bridge and vice versa. Please help.. -- View this message in context: http://www.nabble.com/Multiple-gateways%3A-Priority-the-first-bridge-with-prefix-tp24674260p24674260.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From darklion11 at yahoo.com Sun Jul 26 23:42:41 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Sun, 26 Jul 2009 23:42:41 -0700 (PDT) Subject: [Freeswitch-users] Multiple gateways: Priority the first bridge with prefix Message-ID: <24674279.post@talk.nabble.com> Hi FS Users, I just want to try multiple gateways. It works actually like this... But I test call like 5133333 at 222.333.444.555, it also calls the second bridge 111.222.333.333. It there any way to determine which prefix will call to a bridge specified. E.g. for bridge 1: with prefix of 51 the call with run to 222.333.444.555 not at the second bridge and vice versa. Please help.. -- View this message in context: http://www.nabble.com/Multiple-gateways%3A-Priority-the-first-bridge-with-prefix-tp24674279p24674279.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From darklion11 at yahoo.com Sun Jul 26 23:42:50 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Sun, 26 Jul 2009 23:42:50 -0700 (PDT) Subject: [Freeswitch-users] Multiple gateways: Priority the first bridge with prefix Message-ID: <24674280.post@talk.nabble.com> Hi FS Users, I just want to try multiple gateways. It works actually like this... But I test call like 5133333 at 222.333.444.555, it also calls the second bridge 111.222.333.333. It there any way to determine which prefix will call to a bridge specified. E.g. for bridge 1: with prefix of 51 the call with run to 222.333.444.555 not at the second bridge and vice versa. Please help.. -- View this message in context: http://www.nabble.com/Multiple-gateways%3A-Priority-the-first-bridge-with-prefix-tp24674280p24674280.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From darklion11 at yahoo.com Sun Jul 26 23:52:32 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Sun, 26 Jul 2009 23:52:32 -0700 (PDT) Subject: [Freeswitch-users] Multiple gateways: Priority the first bridge with prefix Message-ID: <24674381.post@talk.nabble.com> Hi FS Users, I just want to try multiple gateways. It works actually like this... But I test call like 5133333 at 222.333.444.555, it also calls the second bridge 111.222.333.333. It there any way to determine which prefix will call to a bridge specified. E.g. for bridge 1: with prefix of 51 the call with run to 222.333.444.555 not at the second bridge and vice versa. Please help.. -- View this message in context: http://www.nabble.com/Multiple-gateways%3A-Priority-the-first-bridge-with-prefix-tp24674381p24674381.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From gcd at i.ph Mon Jul 27 00:04:34 2009 From: gcd at i.ph (Nandy Dagondon) Date: Mon, 27 Jul 2009 15:04:34 +0800 Subject: [Freeswitch-users] Multiple gateways: Priority the first bridge with prefix In-Reply-To: <24674280.post@talk.nabble.com> References: <24674280.post@talk.nabble.com> Message-ID: <7d0bfd8c0907270004s26db6c05j4d760a5a6ab5fcf3@mail.gmail.com> in my implementation, i would use 2 separate conditions that looks like this: On Mon, Jul 27, 2009 at 2:42 PM, Edmar Cruz wrote: > > Hi FS Users, > > I just want to try multiple gateways. It works actually like this... > > > > But I test call like 5133333 at 222.333.444.555, it also calls the > second bridge 111.222.333.333. > > It there any way to determine which prefix will call to a bridge > specified. > > E.g. > > for bridge 1: with prefix of 51 the call with run to 222.333.444.555 not > at the second bridge and vice versa. Please help.. > -- > View this message in context: > http://www.nabble.com/Multiple-gateways%3A-Priority-the-first-bridge-with-prefix-tp24674280p24674280.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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090727/c8a7d177/attachment.html From jason at jasonjgw.net Mon Jul 27 00:07:56 2009 From: jason at jasonjgw.net (Jason White) Date: Mon, 27 Jul 2009 17:07:56 +1000 Subject: [Freeswitch-users] Dial plan contexts Message-ID: <20090727070756.GA22463@jdc.jasonjgw.net> Has anything changed in the handling of dial plan contexts recently? As of rev. 14363, the context setting in the Sofia profile seems to be overriding the context setting in the user's definition in the directory. As per the default configuration, I have user-context set to public in my internal profile, my user has its context set to "default", but calls made from the phone registered to that user ID end up in public context when they reach the dial plan. Either something has changed or there's something wierd in my configuration that I haven't tracked down. I haven't made any changes to any of the profiles or users recently, though, and it was working under an older revision. From darklion11 at yahoo.com Mon Jul 27 00:47:40 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Mon, 27 Jul 2009 00:47:40 -0700 (PDT) Subject: [Freeswitch-users] Multiple gateways: Priority the first bridge with prefix In-Reply-To: <7d0bfd8c0907270004s26db6c05j4d760a5a6ab5fcf3@mail.gmail.com> References: <24674260.post@talk.nabble.com> <7d0bfd8c0907270004s26db6c05j4d760a5a6ab5fcf3@mail.gmail.com> Message-ID: <24675073.post@talk.nabble.com> Not working just the same both of them are running Nandy Dagondon wrote: > > in my implementation, i would use 2 separate conditions that looks like > this: > > > > > > > > > On Mon, Jul 27, 2009 at 2:42 PM, Edmar Cruz wrote: > >> >> Hi FS Users, >> >> I just want to try multiple gateways. It works actually like this... >> >> >> >> But I test call like 5133333 at 222.333.444.555, it also calls the >> second bridge 111.222.333.333. >> >> It there any way to determine which prefix will call to a bridge >> specified. >> >> E.g. >> >> for bridge 1: with prefix of 51 the call with run to 222.333.444.555 >> not >> at the second bridge and vice versa. Please help.. >> -- >> View this message in context: >> http://www.nabble.com/Multiple-gateways%3A-Priority-the-first-bridge-with-prefix-tp24674280p24674280.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/Multiple-gateways%3A-Priority-the-first-bridge-with-prefix-tp24674260p24675073.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From elihayun at gmail.com Mon Jul 27 00:59:52 2009 From: elihayun at gmail.com (Eli Hayun) Date: Mon, 27 Jul 2009 10:59:52 +0300 Subject: [Freeswitch-users] Limit is not working when originate a call In-Reply-To: <4A6C05F6.4000501@savion.huji.ac.il> References: <4A6709D4.6000704@savion.huji.ac.il> <2C85EB6C-1D80-4DC7-8D14-06B17CD1046D@jerris.com> <4A67EF47.6000402@savion.huji.ac.il> <191c3a030907230539i5a640e2do4377d8797ff8f279@mail.gmail.com> <4A6C05F6.4000501@savion.huji.ac.il> Message-ID: <4A6D5E78.1060307@savion.huji.ac.il> Anthony Minessale wrote: >> limit is for inbound calls >> you cannot call it after you already made the call. >> The correct approach would be to not make the call at all. >> >> you could maybe use the limit FSAPI interface with apiExecute to check >> if the limit was exceeded and >> then not bother to place the call to begin with. >> >> otherwise it's sort of like putting a prisoner in the electric chair >> then giving him his trial. >> >> >> Can you tell me how to do that? I set the limit as: Now, how do I know what is the current limit of ${destination_number} Can you give me a JS (or lua) example? Thanks Eli From jason at jasonjgw.net Mon Jul 27 01:16:45 2009 From: jason at jasonjgw.net (Jason White) Date: Mon, 27 Jul 2009 18:16:45 +1000 Subject: [Freeswitch-users] Multiple gateways: Priority the first bridge with prefix In-Reply-To: <24675073.post@talk.nabble.com> References: <24674260.post@talk.nabble.com> <7d0bfd8c0907270004s26db6c05j4d760a5a6ab5fcf3@mail.gmail.com> <24675073.post@talk.nabble.com> Message-ID: <20090727081645.GA31511@jdc.jasonjgw.net> Edmar Cruz wrote: > > Not working just the same both of them are running Do you have them as separate extensions in the dial plan? From gcd at i.ph Mon Jul 27 01:44:15 2009 From: gcd at i.ph (Nandy Dagondon) Date: Mon, 27 Jul 2009 16:44:15 +0800 Subject: [Freeswitch-users] Multiple gateways: Priority the first bridge with prefix In-Reply-To: <20090727081645.GA31511@jdc.jasonjgw.net> References: <24674260.post@talk.nabble.com> <7d0bfd8c0907270004s26db6c05j4d760a5a6ab5fcf3@mail.gmail.com> <24675073.post@talk.nabble.com> <20090727081645.GA31511@jdc.jasonjgw.net> Message-ID: <7d0bfd8c0907270144q74e4d0td2b810a49302ccae@mail.gmail.com> ed, i mean you use separate extension names: btw, you should also use separate gateway names "sip1" and "sip2". so differentiate them in the bridge application. On Mon, Jul 27, 2009 at 4:16 PM, Jason White wrote: > Edmar Cruz wrote: > > > > Not working just the same both of them are running > > Do you have them as separate extensions in the dial plan? > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20090727/64739dc6/attachment-0001.html From helmut.kuper at ewetel.de Mon Jul 27 02:34:49 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Mon, 27 Jul 2009 11:34:49 +0200 Subject: [Freeswitch-users] Question: Capturing VoiceData received from Sagnoma E1 card In-Reply-To: <4A65CA52.90002@ewetel.de> References: <4A65CA52.90002@ewetel.de> Message-ID: <4A6D74B9.3040306@ewetel.de> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, any ideas? regards Helmut On 21.07.2009 16:01, Helmut Kuper wrote: > Hello, > > > For outgoing calls I'm hunting the cause for missing some 100ms of voice > data send from remote right after pickup the remote phone (e.g. initial > "Hello?" sound like "o?" or even nothing) > > On FreeSwitch server I captured the VoIP data to the called VoIP-Phone > on the sofia interface. Using wireshark it also shows that the voice > data from remote is missed. Using Mobil phones or ISDN phones calling > the same remote party there is never a bit missed. > > This problem occurs rare - once or twice per day and per local voip > phone, but it's quite anoying. > > So is there a way to capture the correspondig ISDN voice data FS > receives before it is transmitted via RTP or just droped? I want to c > whether FS drops the early RTP packets or whether FS never got the data > from ISDN. > > > > Sofia Profile is using > > > The dialplan portion is: > > expression="^94([0-9]+)$" break="never"> > > data="effective_caller_id_name=anonymous"/> > data="effective_caller_id_number=anonymous"/> > > expression="^([0-9]+)$"> > data="ignore_early_media=true"/> > data="absolute_codec_string=PCMA"/> > data="continue_on_fail=true"/> > > data="${destination_number} XML et_internal_error"/> > > > Any ideas to refine my debugging? > > regard > 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 -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) iD8DBQFKbXS54tZeNddg3dwRAp07AJ9e9gNY/MR4byUvpeR6so9Ap3cx8ACaA9SP EodxfZrtLAZiYtzYtQsBldY= =ZfJl -----END PGP SIGNATURE----- From rupa at rupa.com Mon Jul 27 05:38:14 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Mon, 27 Jul 2009 07:38:14 -0500 Subject: [Freeswitch-users] core dump In-Reply-To: References: Message-ID: That backtrace is not useful because gdb was unable to locate the freeswitch binary. Since you are running it from /opt/freeswitch/bin directory, don't use 'bin/freeswitch' to point gdb to the binary: gdb freeswitch corefile On Mon, Jul 27, 2009 at 1:40 AM, Baskar wrote: > Hi, > > I get core dump segmentation fault in freeswitch machine frequently. can > any one assist me what is error in the freeswitch. i have pasted the logs in > freeswitch pastebin. > > This is the link http://pastebin.freeswitch.org/9851 > > > > -- > Thanks with 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 > > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090727/ca477a25/attachment.html From yudha2008 at gmail.com Mon Jul 27 06:00:32 2009 From: yudha2008 at gmail.com (Baskar) Date: Mon, 27 Jul 2009 06:00:32 -0700 Subject: [Freeswitch-users] core dump In-Reply-To: References: Message-ID: *Hi Rupa, I get core dump segmentation fault in freeswitch machine frequently. can any one assist me what is error in the freeswitch. i have pasted the logs in freeswitch pastebin. This is the link http://pastebin.freeswitch.org/9854 Can some one assist me what is error in freeswitch to hit core dump. -- Thanks with Regards, N.Baskar * -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090727/a7e16fb6/attachment.html From rupa at rupa.com Mon Jul 27 06:10:33 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Mon, 27 Jul 2009 08:10:33 -0500 Subject: [Freeswitch-users] core dump In-Reply-To: References: Message-ID: That backtrace is useless since you are not giving the right path to freeswitch. Look at line 9 of the pastebin. Until you fix that we can't look at the backtrace. On Mon, Jul 27, 2009 at 8:00 AM, Baskar wrote: > *Hi Rupa, > > I get core dump segmentation fault in freeswitch machine frequently. can > any one assist me what is error in the freeswitch. i have pasted the logs in > freeswitch pastebin. > > This is the link http://pastebin.freeswitch.org/9854 > > Can some one assist me what is error in freeswitch to hit core dump. > > > > -- > Thanks with 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 > > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090727/00c52c72/attachment.html From mike at jerris.com Mon Jul 27 06:17:28 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 27 Jul 2009 09:17:28 -0400 Subject: [Freeswitch-users] core dump In-Reply-To: References: Message-ID: Did you read his response to you? Please generate a usable backtrace as rupa explained and post a bug to jira freeswitch.org On Jul 27, 2009, at 9:00 AM, Baskar wrote: > Hi Rupa, > > I get core dump segmentation fault in freeswitch machine > frequently. can any one assist me what is error in the freeswitch. > i have pasted the logs in freeswitch pastebin. > > This is the link http://pastebin.freeswitch.org/9854 > > Can some one assist me what is error in freeswitch to hit core dump. > > > > -- > Thanks with 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090727/0731f784/attachment.html From gmaruzz at celliax.org Mon Jul 27 07:08:55 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Mon, 27 Jul 2009 16:08:55 +0200 Subject: [Freeswitch-users] Skypiax, Skype endpoint and trunk, revamped Message-ID: <7b197bef0907270708h1dd55de2m92b5de869da6404d@mail.gmail.com> Ciao FreeSWITCHers, please have a look at the much changed wiki page: http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk and checkout and test the code in svn. Much has happened, various bug fixes and features added. Most notable: - multiple instances of the same Skype username on Linux (eg: running 20 concurrent channels as "Bob" Skype user) - adding and removing interfaces on the fly (patch sent by Muhammad Shahzad) - easier creation of Skype clients configuration directory - reduced latency - better robustness - running as Windows Service - customized ALSA driver for more devices with less IRQs and context switches - custom kernel tickless and 100HZ (eg. solves high load problems in CentOS and in virtualization) - interactive command line client for prototyping Also, please note that ALSA drivers version 1.0.20 seems to be much more stable in our kind of usage (snd-dummy). Various other enhancements will come, but in the mean time please give feedback on the current svn code (we want to be robust for the 1.0.4 Release :-) ) See you all at www.cluecon.com, talk on Skypiax August 4th at 4.30 pm ! -giovanni 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 Mon Jul 27 07:09:43 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 27 Jul 2009 09:09:43 -0500 Subject: [Freeswitch-users] core dump In-Reply-To: References: Message-ID: <191c3a030907270709pa530cey4dc70af085853e6a@mail.gmail.com> Again, I like I have to say weekly, Please do not report bugs on the mailing list. http://jira.freeswitch.org Also, please completely re-checkout and rebuild latest trunk and erase your prior freeswitch install before filing the jira. We only accept bug reports of this nature confirmed on a fresh build of latest SVN. On Mon, Jul 27, 2009 at 8:17 AM, Michael Jerris wrote: > Did you read his response to you? Please generate a usable backtrace as > rupa explained and post a bug to jira freeswitch.org > > > On Jul 27, 2009, at 9:00 AM, Baskar wrote: > > *Hi Rupa, > > I get core dump segmentation fault in freeswitch machine frequently. can > any one assist me what is error in the freeswitch. i have pasted the logs in > freeswitch pastebin. > > This is the link > http://pastebin.freeswitch.org/9854 > > Can some one assist me what is error in freeswitch to hit core dump. > > > > -- > Thanks with 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 > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference 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/20090727/07d74b71/attachment-0001.html From brian at freeswitch.org Mon Jul 27 07:20:38 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 27 Jul 2009 09:20:38 -0500 Subject: [Freeswitch-users] Dial plan contexts In-Reply-To: <20090727070756.GA22463@jdc.jasonjgw.net> References: <20090727070756.GA22463@jdc.jasonjgw.net> Message-ID: Jason, You need to set the context to on the profile and the user_context variable on the user to default. There is no such thing as a user- context param on the profile. There is a user_context variable on the user. /b On Jul 27, 2009, at 2:07 AM, Jason White wrote: > As per the default configuration, I have user-context set to public > in my > internal profile, my user has its context set to "default", but > calls made > from the phone registered to that user ID end up in public context > when they > reach the dial plan. From anthony.minessale at gmail.com Mon Jul 27 07:36:26 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 27 Jul 2009 09:36:26 -0500 Subject: [Freeswitch-users] Limit is not working when originate a call In-Reply-To: <4A6D5E78.1060307@savion.huji.ac.il> References: <4A6709D4.6000704@savion.huji.ac.il> <2C85EB6C-1D80-4DC7-8D14-06B17CD1046D@jerris.com> <4A67EF47.6000402@savion.huji.ac.il> <191c3a030907230539i5a640e2do4377d8797ff8f279@mail.gmail.com> <4A6C05F6.4000501@savion.huji.ac.il> <4A6D5E78.1060307@savion.huji.ac.il> Message-ID: <191c3a030907270736r303abfadraadc0527b1d45e34@mail.gmail.com> I suggest you study FS more because if you can't tell what to do with the info provided you have some fundamentals to review before proceeding. On Mon, Jul 27, 2009 at 2:59 AM, Eli Hayun wrote: > Anthony Minessale wrote: > >> limit is for inbound calls > >> you cannot call it after you already made the call. > >> The correct approach would be to not make the call at all. > >> > >> you could maybe use the limit FSAPI interface with apiExecute to check > >> if the limit was exceeded and > >> then not bother to place the call to begin with. > >> > >> otherwise it's sort of like putting a prisoner in the electric chair > >> then giving him his trial. > >> > >> > >> > Can you tell me how to do that? > I set the limit as: > > Now, how do I know what is the current limit of ${destination_number} > Can you give me a JS (or lua) example? > Thanks > Eli > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference 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/20090727/22895d3e/attachment.html From xengelpublicx at gmail.com Mon Jul 27 07:42:11 2009 From: xengelpublicx at gmail.com (Vladimir Elizarov) Date: Mon, 27 Jul 2009 18:42:11 +0400 Subject: [Freeswitch-users] linksys spa962+spa932 blf, hold and intercept of ringing extensions Message-ID: <4A6DBCC3.8010107@gmail.com> Hello. I am trying to configure the linksys spa-932 (at http://wiki.freeswitch.org/wiki/Interop_List#Linksys_SPA932 stated that he works with freeswitch). I said "Server-Type option set to" RFC3265_4235 "" added to the unit 1 key 1 string: "fnc = blf + sd; sub = 1002 at pbx0.test.lan; nme = test". The button blinks orange. if call on 1000 (spa962). This subscription runs spa932 and starts to show the status of the phone 1002. Thanks. -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 261 bytes Desc: OpenPGP digital signature Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090727/2fdbdbe7/attachment.bin From jgonzalez at sqli.com Mon Jul 27 08:58:29 2009 From: jgonzalez at sqli.com (julien) Date: Mon, 27 Jul 2009 17:58:29 +0200 Subject: [Freeswitch-users] SIP trunk (PBX - FreeSwitch) routing problem. Message-ID: <4A6DCEA5.2090502@sqli.com> I'm currently trying to connect FreeSwitch to a PBX (Alcatel-Lucent), thanks to a SIP trunk. SIP trunks are available and working on the PBX thanks to a recent update. My problem is that I can't call phones linked to the PBX. When I try to call 300, I've got this message in freeswitch console : 2009-07-27 17:38:48.514105 [NOTICE] switch_channel.c:602 New Channel sofia/internal/[EMAIL PROTECTED] [93d5a10e-7ac3-11de-b456-e5e56113066d] 2009-07-27 17:38:48.516907 [INFO] mod_dialplan_xml.c:252 Processing jgonzalez jgonzalez->300 in context default 2009-07-27 17:38:48.521084 [NOTICE] switch_channel.c:602 New Channel sofia/external/[EMAIL PROTECTED] [93d69816-7ac3-11de-b456-e5e56113066d] 2009-07-27 17:38:48.636073 [NOTICE] sofia.c:3775 Hangup sofia/external/[EMAIL PROTECTED] [CS_CONSUME_MEDIA] [NO_ROUTE_DESTINATION] 2009-07-27 17:38:48.636073 [INFO] mod_dptools.c:2091 Originate Failed. Cause: NO_ROUTE_DESTINATION 2009-07-27 17:38:48.637788 [NOTICE] mod_dptools.c:633 Hangup sofia/internal/[EMAIL PROTECTED] [CS_EXECUTE] [NORMAL_CLEARING] 2009-07-27 17:38:48.639047 [NOTICE] switch_core_session.c:1085 Session 15 (sofia/internal/[EMAIL PROTECTED]) Ended 2009-07-27 17:38:48.639047 [NOTICE] switch_core_session.c:1087 Close Channel sofia/internal/[EMAIL PROTECTED] [CS_DESTROY] 2009-07-27 17:38:48.643042 [NOTICE] switch_core_session.c:1085 Session 16 (sofia/external/[EMAIL PROTECTED]) Ended 2009-07-27 17:38:48.643042 [NOTICE] switch_core_session.c:1087 Close Channel sofia/external/[EMAIL PROTECTED] [CS_DESTROY] I've defined, in sip_profiles/external, a gateway to the PBX this way : And in the dialplan default.xml : (for the moment, I'm trying only with the number 300 which is a correct number of the phone system). As you can see, I'm far from being an expert of FreeSwitch, SIP or even VoIP in general. I'm learning. I hope you can help me. Regards, Julien Gonzalez. From brian at freeswitch.org Mon Jul 27 09:09:54 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 27 Jul 2009 11:09:54 -0500 Subject: [Freeswitch-users] SIP trunk (PBX - FreeSwitch) routing problem. In-Reply-To: <4A6DCEA5.2090502@sqli.com> References: <4A6DCEA5.2090502@sqli.com> Message-ID: <60EC1D48-40FD-4BF9-B5D2-D8FAE1EBEEEC@freeswitch.org> I have to guess that you put this at the bottom of the default.xml? /b On Jul 27, 2009, at 10:58 AM, julien wrote: > And in the dialplan default.xml : > > > > > > > > > > From jgonzalez at sqli.com Mon Jul 27 09:16:20 2009 From: jgonzalez at sqli.com (julien) Date: Mon, 27 Jul 2009 18:16:20 +0200 Subject: [Freeswitch-users] SIP trunk (PBX - FreeSwitch) routing problem. In-Reply-To: <60EC1D48-40FD-4BF9-B5D2-D8FAE1EBEEEC@freeswitch.org> References: <4A6DCEA5.2090502@sqli.com> <60EC1D48-40FD-4BF9-B5D2-D8FAE1EBEEEC@freeswitch.org> Message-ID: <4A6DD2D4.8040800@sqli.com> No totally at the bottom. Before : Brian West a ?crit : > I have to guess that you put this at the bottom of the default.xml? > > /b > > On Jul 27, 2009, at 10:58 AM, julien wrote: > > >> And in the dialplan default.xml : >> >> >> >> >> >> >> >> >> >> >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From jesse.peterson at exbiblio.com Mon Jul 27 09:25:12 2009 From: jesse.peterson at exbiblio.com (Jesse Peterson) Date: Mon, 27 Jul 2009 09:25:12 -0700 Subject: [Freeswitch-users] Operation has no matching challenge Message-ID: <1B7E58CE-5BF3-4C45-9A27-02E9B84F4BA4@exbiblio.com> Hello, I am getting some SIP registration problems with FreeSWITCH 1.0.4pre10 and my ITSP (Vitelity). The error in the logs is such: 2009-07-27 08:54:48.217815 [ERR] sofia_reg.c:1460 vitelity-out Registration Failed with status Operation has no matching challenge [904]. failure #56 While these errors are happening the gateway state (via "sofia status") is FAIL_WAIT. With the ever-increasing back-off wait (60, 90, 120, 150, ..., seconds) the registration never resumes. Now one might suspect that there is something wrong with the configuration/ authorization but this problem is intermittent: a simple "sofia profile external restart" restores the registration and all is well (state turns to REGED) and of course the initial registration succeeds just fine, too. Quite an annoying problem as you never quite know when your gateway is registered when you pick up the receiver of your phone giving the impression of unreliable service! I suspect this to be the same problem, but with a different error message, that has been reported before[1][2]. Thoughts? Anything I should try? Thanks, - Jesse [1] http://lists.freeswitch.org/pipermail/freeswitch-users/2009-February/011269.html [2] http://lists.freeswitch.org/pipermail/freeswitch-users/2008-September/006272.html From brian at freeswitch.org Mon Jul 27 09:30:00 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 27 Jul 2009 11:30:00 -0500 Subject: [Freeswitch-users] Operation has no matching challenge In-Reply-To: <1B7E58CE-5BF3-4C45-9A27-02E9B84F4BA4@exbiblio.com> References: <1B7E58CE-5BF3-4C45-9A27-02E9B84F4BA4@exbiblio.com> Message-ID: Can you get a sofia loglevel all 9 and a sip trace? /b On Jul 27, 2009, at 11:25 AM, Jesse Peterson wrote: > Hello, > > I am getting some SIP registration problems with FreeSWITCH 1.0.4pre10 > and my ITSP (Vitelity). The error in the logs is such: > > 2009-07-27 08:54:48.217815 [ERR] sofia_reg.c:1460 vitelity-out > Registration Failed with status Operation has no matching challenge > [904]. failure #56 > > While these errors are happening the gateway state (via "sofia > status") is FAIL_WAIT. With the ever-increasing back-off wait (60, 90, > 120, 150, ..., seconds) the registration never resumes. Now one might > suspect that there is something wrong with the configuration/ > authorization but this problem is intermittent: a simple "sofia > profile external restart" restores the registration and all is well > (state turns to REGED) and of course the initial registration succeeds > just fine, too. Quite an annoying problem as you never quite know when > your gateway is registered when you pick up the receiver of your phone > giving the impression of unreliable service! > > I suspect this to be the same problem, but with a different error > message, that has been reported before[1][2]. > > Thoughts? Anything I should try? Thanks, > - Jesse From miconda at gmail.com Mon Jul 27 09:45:35 2009 From: miconda at gmail.com (Daniel-Constantin Mierla) Date: Mon, 27 Jul 2009 18:45:35 +0200 Subject: [Freeswitch-users] freeswitch and siremis v0.9.3 Message-ID: <4A6DD9AF.2080406@gmail.com> Hello, recently released siremis v0.9.3 adds support for communication with freeswitch via event socket. siremis is an open source web management interface targeting the SIP routing engines kamailio (openser) and sip-router.org. freeswitch fits perfectly in the picture since it completes the routing engines with rich media services. The new release includes php code to communicate with freeswitch via tcp/event socket and a panel to send commands/display response. Code is grouped like a library, new features being straightforward to develop. For some commands, the output is pretty formatted - screenshot: http://www.asipto.com/gallery/v/siremis/siremis_20.jpg.html?g2_imageViewsIndex=1 More is planned for the future (e.g., display active calls of a certain user, click to end an active call). Feedback and contributions are welcome, visit: http://siremis.asipto.com Cheers, Daniel -- Daniel-Constantin Mierla * SIP Router Bootcamp * Kamailio (OpenSER) and Asterisk Training * Berlin, Germany, Sep 1-4, 2009 * http://www.asipto.com/index.php/sip-router-bootcamp/ From msc at freeswitch.org Mon Jul 27 10:00:42 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 27 Jul 2009 10:00:42 -0700 Subject: [Freeswitch-users] Which method Can I use in IVR In-Reply-To: <7aa29e790907262207n5d50c96cl2f19e2d152399db6@mail.gmail.com> References: <7aa29e790907262207n5d50c96cl2f19e2d152399db6@mail.gmail.com> Message-ID: <87f2f3b90907271000h557e3c05jbd3839b9de595292@mail.gmail.com> See comments inline... On Sun, Jul 26, 2009 at 10:07 PM, Thangappan.M wrote: > Dear all, > > I am learning how to implement a IVR in Freeswitch.In our organization we > are using Perl scripting language for doing this.So In freeswitch also I > need to use Perl. Tony, Brian, and I all like Perl. :) > > > So far I heard two methods for executing IVR. > One is in dial plan using perl application.( In perl I create IVR > menu and play the voice files) > Another one is using event socket.In dial plan I specified socket > application and write a Perl script which is listening that particular port > and get the session Id. > Yes, you can call a script from the dialplan using syntax like this: OR You can call an outbound socket connection like this: > > Have I understood correctly?.If it is correct means tell which method can I > use?. Other make me understand well. You're on the right track. As to which method to use, that depends on your circumstances. How much does it need to scale? Do you want the IVR "brain" to reside physically on a different server than the FS server? Think about those things. > > I have seen downloaded perl IVR menu from freeswitch site.In that they > called some internal functions like playandGetDigits,StreamFile,ready > ...etc. > > These functions is been called by using $session variable.Where these > functions are defined.? > When you call a Perl script from the dialplan the script automatically has access to a variable called $session. Check this for more information: http://wiki.freeswitch.org/wiki/Mod_perl#Programming_with_mod_perl Of course, when using the outbound event socket you will not have this magic $session variable. Your best bet to learn more about the socket interface is to look at the sample scripts in src/libs/esl/perl/. (server.pl, server2.pl, and server3.pl) If you are building an IVR with Perl and the event socket be sure to check out src/libs/esl/perl/ESL/IVR.pm which is a small Perl module with some simple abstractions to make IVR programming a bit more convenient. I recommend that you try and create a simple IVR using each method and get a feel for how each one works. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090727/6902070a/attachment.html From miles.chet at gmail.com Mon Jul 27 10:35:48 2009 From: miles.chet at gmail.com (roberto) Date: Mon, 27 Jul 2009 14:35:48 -0300 Subject: [Freeswitch-users] Which method Can I use in IVR In-Reply-To: <87f2f3b90907271000h557e3c05jbd3839b9de595292@mail.gmail.com> References: <7aa29e790907262207n5d50c96cl2f19e2d152399db6@mail.gmail.com> <87f2f3b90907271000h557e3c05jbd3839b9de595292@mail.gmail.com> Message-ID: Michael, For scale reasons is the best choice event socket? thanks, On Mon, Jul 27, 2009 at 2:00 PM, Michael Collins wrote: > See comments inline... > > On Sun, Jul 26, 2009 at 10:07 PM, Thangappan.M > wrote: >> >> Dear all, >> >> ?I am learning how to implement a IVR in Freeswitch.In our organization we >> are using Perl scripting language for doing this.So In freeswitch also I >> need to use Perl. > > Tony, Brian, and I all like Perl. :) > >> >> >> ?So far I heard two methods for executing IVR. >> ??????? One is in dial plan using perl application.( In perl I create IVR >> menu and play the voice files) >> ??????? Another one is using event socket.In dial plan I specified socket >> application and write a Perl script which is listening that particular port >> and get the session Id. > > Yes, you can call a script from the dialplan using syntax like this: > > > OR > > You can call an outbound socket connection like this: > > >> >> Have I understood correctly?.If it is correct means tell which method can >> I use?. Other make me understand well. > > You're on the right track. As to which method to use, that depends on your > circumstances. How much does it need to scale? Do you want the IVR "brain" > to reside physically on a different server than the FS server? Think about > those things. > >> >> >> I have seen downloaded perl IVR menu from freeswitch site.In that they >> called some internal functions like playandGetDigits,StreamFile,ready >> ...etc. >> >> These functions is been called by using $session variable.Where these >> functions are defined.? > > When you call a Perl script from the dialplan the script automatically has > access to a variable called $session.? Check this for more information: > http://wiki.freeswitch.org/wiki/Mod_perl#Programming_with_mod_perl > > Of course, when using the outbound event socket you will not have this magic > $session variable. Your best bet to learn more about the socket interface is > to look at the sample scripts in src/libs/esl/perl/. (server.pl, server2.pl, > and server3.pl) If you are building an IVR with Perl and the event socket be > sure to check out src/libs/esl/perl/ESL/IVR.pm which is a small Perl module > with some simple abstractions to make IVR programming a bit more convenient. > > I recommend that you try and create a simple IVR using each method and get a > feel for how each one works. > > -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 marketing at cluecon.com Mon Jul 27 11:50:01 2009 From: marketing at cluecon.com (Michael Collins) Date: Mon, 27 Jul 2009 11:50:01 -0700 Subject: [Freeswitch-users] ClueCon 2009 - Last Chance For Early Bird Special! Message-ID: <87f2f3b90907271150y2b5c5db1pb7466590a87c0a48@mail.gmail.com> ClueCon is next week! We're all gearing up for a great event. Here is some important information for those who've not already paid: Today is the last day to receive the $499 early bird special. After today, the price will go up to $699. If you have registered at the ClueCon website but you have not yet paid then please call 877.742.CLUE immediately! We want to make sure that you get the early bird rate. All registrations after today (Monday July 27) will be $699. Thank you for your support of ClueCon 2009! We are looking forward to seeing everyone in person in Chicago. -Michael Collins http://www.cluecon.com 877.742.CLUE -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090727/ad6aaa33/attachment.html From msc at freeswitch.org Mon Jul 27 11:57:46 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 27 Jul 2009 11:57:46 -0700 Subject: [Freeswitch-users] Which method Can I use in IVR In-Reply-To: References: <7aa29e790907262207n5d50c96cl2f19e2d152399db6@mail.gmail.com> <87f2f3b90907271000h557e3c05jbd3839b9de595292@mail.gmail.com> Message-ID: <87f2f3b90907271157h2aefb8ceu581c0fd2b6b4a89d@mail.gmail.com> On Mon, Jul 27, 2009 at 10:35 AM, roberto wrote: > Michael, > > For scale reasons is the best choice event socket? > Yes. You can have the IVR stuff running on a separate server altogether. It also gives you great flexibility in designing a setup where you can have a db backend and/or a backup IVR server. The socket method requires a little more effort up front but it pays off in power and flexibility. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090727/2ed51b57/attachment.html From gmaruzz at celliax.org Mon Jul 27 12:31:01 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Mon, 27 Jul 2009 21:31:01 +0200 Subject: [Freeswitch-users] Slashdot: How-to-Help-With-a-University-ICT-Strategy Message-ID: <7b197bef0907271231y262c7b06v3a83ebcef86f7f54@mail.gmail.com> http://ask.slashdot.org/story/09/07/27/1652247/How-to-Help-With-a-University-ICT-Strategy " An anonymous reader writes "I have been asked to contribute to my university's revised ICT (Information and Communication Technology) strategy and I am curious what fellow Slashdot members consider to be the main advice in this context. What are the major mistakes that organizations like universities make? Given the complexity of the different participants in a university, how does one have a coherent strategy that fulfills the needs of such a wide audience? How does one promote open source in a managerial culture? How does one deal with the curse of the virtual learning environment?"" http://ask.slashdot.org/comments.pl?sid=1316571&cid=28842157 From testeador01 at gmail.com Mon Jul 27 12:34:54 2009 From: testeador01 at gmail.com (Milena) Date: Mon, 27 Jul 2009 14:34:54 -0500 Subject: [Freeswitch-users] Command to get FreeSWITCH's internal time In-Reply-To: <1248625501.5655.79.camel@raul-laptop> References: <1248625501.5655.79.camel@raul-laptop> Message-ID: *Hello everyone, strftime *does what i want (now I can't figure out why didn't I see it before in the wiki page), thank you very much for your replies, have a nice time :) 2009/7/26 Raul Fragoso > Every event has a header 'Event-Date-Local', which has the local date > and time. If want to actively retrieve the date and time, you can send > this API request to the server: api strftime > > Regards, > > Raul > > On Sat, 2009-07-25 at 09:20 -0500, Milena wrote: > > Hello everyone, > > > > I'm using the inbound event socket to receive some information about > > the status of my FreeSWITCH system and i wanted to know if there is an > > api command that can be used to get the FreeSWITCH time, I tried > > searching around in the docs and in google but i couldn't find an > > answer. Thanks for your attention and thanks in advance if anyone can > > assist me with this. > > > > Have a nice time and lots of cookies :) > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20090727/d887db3b/attachment.html From william.suffill at gmail.com Mon Jul 27 13:17:41 2009 From: william.suffill at gmail.com (William Suffill) Date: Mon, 27 Jul 2009 16:17:41 -0400 Subject: [Freeswitch-users] IAX configurations In-Reply-To: <1452e2980907262259k30713f4ft363592ac77ee3e33@mail.gmail.com> References: <1452e2980907262223u677eef21o282f39e2a56209cc@mail.gmail.com> <98E444F7-ECF8-40DF-82D2-89A48DC11EAC@jerris.com> <1452e2980907262259k30713f4ft363592ac77ee3e33@mail.gmail.com> Message-ID: <6b65470d0907271317q7476ed6x7b85ef5eb0c478d4@mail.gmail.com> http://wiki.freeswitch.org/wiki/Connecting_Freeswitch_And_Asterisk Goes into some detail with connecting to Asterisk via SIP From oseslija at gmail.com Mon Jul 27 13:45:06 2009 From: oseslija at gmail.com (Ognjen Seslija) Date: Mon, 27 Jul 2009 22:45:06 +0200 Subject: [Freeswitch-users] linksys spa962+spa932 blf, hold and intercept of ringing extensions In-Reply-To: <4A6DBCC3.8010107@gmail.com> References: <4A6DBCC3.8010107@gmail.com> Message-ID: <4468a6770907271345j5062e625pffc77fb26ed72ebb@mail.gmail.com> Hello, I authored that wiki article. The following key will work: fnc=blf+sd+cp;sub=1002@$PROXY You need to make sure that presence is not off in the profile. Also "cp" in the key will enable you to do the intercept of "ringing" call to watched extension. For further help please join #freeswitch IRC channel. Regards, Ognjen fnc=blf+sd+cp;sub=4601@$PROXY On Mon, Jul 27, 2009 at 4:42 PM, Vladimir Elizarov wrote: > Hello. > > I am trying to configure the linksys spa-932 (at > http://wiki.freeswitch.org/wiki/Interop_List#Linksys_SPA932 stated that > he works with freeswitch). > I said "Server-Type option set to" RFC3265_4235 "" added to the unit 1 > key 1 string: "fnc = blf + sd; sub = 1002 at pbx0.test.lan; nme = test". > > The button blinks orange. if call on 1000 (spa962). This subscription > runs spa932 and starts to show the status of the phone 1002. > > 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/20090727/44498024/attachment-0001.html From keithl at voxtelecom.co.za Mon Jul 27 14:23:59 2009 From: keithl at voxtelecom.co.za (Keith Laaks) Date: Mon, 27 Jul 2009 23:23:59 +0200 Subject: [Freeswitch-users] Distortion on approx first 200ms of G722 prompts on DECT based CPE Message-ID: <1B99233662E2104983E3550185D3ED73628FA0@xena.internal.datapro.co.za> Hi All, I am testing a range of G722 capable DECT based CPE. With one range, I have noticed that the first 200ms or so of each separate prompt file being played back is played out distorted from the DECT handset. When having a normal conversation, the quality is excellent, but when accessing your vmail, all the individual audio files making up the menu choices exhibit the distortion, which is pretty annoying. The same unit using G729, alaw or ulaw works 100%. I wonder if anybody else has uncounted this issue? My guess at this point - There may be a short break in the RTP between the separate files being played out by FS that makes up any menu. During this time the DECT handset's AGC probably goes to MAX amplification (as its not receiving any input during the short break in RTP). Then, when the RTP returns at the start of the next file, the AGC boosts the audio into clipping zone and takes 200ms to dampen down back to normal good levels. Looks like in these devices the G722 encode/decode is actually done in the DECT handset and not the voip-base unit. Is there any parameter that can be set in FS to ensure that the RTP keeps flowing, sending 'silence' between prompts ? Would be interesting to validate the above 'guess'. Best Regards Keith -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090727/2ece9638/attachment.html From jesse.peterson at exbiblio.com Mon Jul 27 16:34:21 2009 From: jesse.peterson at exbiblio.com (Jesse Peterson) Date: Mon, 27 Jul 2009 16:34:21 -0700 Subject: [Freeswitch-users] Operation has no matching challenge In-Reply-To: References: <1B7E58CE-5BF3-4C45-9A27-02E9B84F4BA4@exbiblio.com> Message-ID: <5F25DBDD-B65C-444C-9189-09416D66042E@exbiblio.com> Just to keep those interested informed this thread is being tracked as: http://jira.freeswitch.org/browse/SFSIP-169 On Jul 27, 2009, at 9:30 AM, Brian West wrote: > Can you get a sofia loglevel all 9 > and a sip trace? > > /b > > On Jul 27, 2009, at 11:25 AM, Jesse Peterson wrote: > >> Hello, >> >> I am getting some SIP registration problems with FreeSWITCH >> 1.0.4pre10 >> and my ITSP (Vitelity). The error in the logs is such: >> >> 2009-07-27 08:54:48.217815 [ERR] sofia_reg.c:1460 vitelity-out >> Registration Failed with status Operation has no matching challenge >> [904]. failure #56 Thanks, - Jesse From jason at jasonjgw.net Mon Jul 27 17:01:26 2009 From: jason at jasonjgw.net (Jason White) Date: Tue, 28 Jul 2009 10:01:26 +1000 Subject: [Freeswitch-users] Dial plan contexts In-Reply-To: References: <20090727070756.GA22463@jdc.jasonjgw.net> Message-ID: <20090728000126.GA9387@jdc.jasonjgw.net> >From the profile: >From the user's entry in the directory: but under rev. 14363 when the phone registered to that user makes a call, the dial plan is searched in public context. I hope this helps to clarify. I tried resetting my configuration using Git to a known good state, but with no change to the above behaviour. I'm going to rebuild with the latest from svn soon. From anthony.minessale at gmail.com Mon Jul 27 17:06:00 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 27 Jul 2009 19:06:00 -0500 Subject: [Freeswitch-users] Distortion on approx first 200ms of G722 prompts on DECT based CPE In-Reply-To: <1B99233662E2104983E3550185D3ED73628FA0@xena.internal.datapro.co.za> References: <1B99233662E2104983E3550185D3ED73628FA0@xena.internal.datapro.co.za> Message-ID: <191c3a030907271706yb3d55e4m1da94a18b6f9bb0b@mail.gmail.com> you can set the global var send_silence_when_idle=true in vars.xml On Mon, Jul 27, 2009 at 4:23 PM, Keith Laaks wrote: > Hi All, > > > > I am testing a range of G722 capable DECT based CPE. > > With one range, I have noticed that the first 200ms or so of each separate > prompt file being played back is played out distorted from the DECT handset. > > When having a normal conversation, the quality is excellent, but when > accessing your vmail, all the individual audio files making up the menu > choices exhibit the distortion, which is pretty annoying. > > The same unit using G729, alaw or ulaw works 100%. > > > > I wonder if anybody else has uncounted this issue? > > > > My guess at this point ? > > There may be a short break in the RTP between the separate files being > played out by FS that makes up any menu. > > During this time the DECT handset?s AGC probably goes to MAX amplification > (as its not receiving any input during the short break in RTP). > > Then, when the RTP returns at the start of the next file, the AGC boosts > the audio into clipping zone and takes 200ms to dampen down back to normal > good levels. > > > > Looks like in these devices the G722 encode/decode is actually done in the > DECT handset and not the voip-base unit. > > > > Is there any parameter that can be set in FS to ensure that the RTP keeps > flowing, sending ?silence? between prompts ? Would be interesting to > validate the above ?guess?. > > > > > > Best Regards > > > > Keith > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference 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/20090727/3517e201/attachment.html From jason at jasonjgw.net Mon Jul 27 18:13:42 2009 From: jason at jasonjgw.net (Jason White) Date: Tue, 28 Jul 2009 11:13:42 +1000 Subject: [Freeswitch-users] Dial plan contexts In-Reply-To: <20090728000126.GA9387@jdc.jasonjgw.net> References: <20090727070756.GA22463@jdc.jasonjgw.net> <20090728000126.GA9387@jdc.jasonjgw.net> Message-ID: <20090728011342.GA4273@jdc.jasonjgw.net> With apologies to all, it was something that sneaked into my configuration that I'm still tracking down. From darklion11 at yahoo.com Mon Jul 27 18:23:10 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Mon, 27 Jul 2009 18:23:10 -0700 (PDT) Subject: [Freeswitch-users] Multiple gateways: Priority the first bridge with prefix In-Reply-To: <7d0bfd8c0907270144q74e4d0td2b810a49302ccae@mail.gmail.com> References: <24674260.post@talk.nabble.com> <7d0bfd8c0907270004s26db6c05j4d760a5a6ab5fcf3@mail.gmail.com> <24675073.post@talk.nabble.com> <20090727081645.GA31511@jdc.jasonjgw.net> <7d0bfd8c0907270144q74e4d0td2b810a49302ccae@mail.gmail.com> Message-ID: <24691020.post@talk.nabble.com> Yes, I actually just want to not be able to communicate with the other bridges. I have already this extension name = "sample-1". Freeswitch gets the first extension the 2nd also trigger it. When the calls finds the match it suits perfectly but I just want that I do not want to view the bridges with CS_DESTROY or hangup_after_false if not found. Nandy Dagondon wrote: > > ed, > > i mean you use separate extension names: > > > > > > > > > > > > btw, you should also use separate gateway names "sip1" and "sip2". so > differentiate them in the bridge application. > > On Mon, Jul 27, 2009 at 4:16 PM, Jason White wrote: > >> Edmar Cruz wrote: >> > >> > Not working just the same both of them are running >> >> Do you have them as separate extensions in the dial plan? >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/Multiple-gateways%3A-Priority-the-first-bridge-with-prefix-tp24674260p24691020.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From dujinfang at gmail.com Mon Jul 27 21:37:11 2009 From: dujinfang at gmail.com (Seven Du) Date: Tue, 28 Jul 2009 12:37:11 +0800 Subject: [Freeswitch-users] Skypiax, Skype endpoint and trunk, revamped In-Reply-To: <7b197bef0907270708h1dd55de2m92b5de869da6404d@mail.gmail.com> References: <7b197bef0907270708h1dd55de2m92b5de869da6404d@mail.gmail.com> Message-ID: <23f91030907272137r1c4879bp22dcb23cda7cc5fd@mail.gmail.com> Thanks for the great work. Just want you know that 20 channels with the same username works well on my server. And echo() works without any problem. An updated version of Round Robin hunt and a minor bug posted on jira. Thanks again. 2009/7/27 Giovanni Maruzzelli > Ciao FreeSWITCHers, > > please have a look at the much changed wiki page: > http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk and > checkout and test the code in svn. > > Much has happened, various bug fixes and features added. > > Most notable: > - multiple instances of the same Skype username on Linux (eg: running > 20 concurrent channels as "Bob" Skype user) > - adding and removing interfaces on the fly (patch sent by Muhammad > Shahzad) > - easier creation of Skype clients configuration directory > - reduced latency > - better robustness > - running as Windows Service > - customized ALSA driver for more devices with less IRQs and context > switches > - custom kernel tickless and 100HZ (eg. solves high load problems in > CentOS and in virtualization) > - interactive command line client for prototyping > > Also, please note that ALSA drivers version 1.0.20 seems to be much > more stable in our kind of usage (snd-dummy). > > Various other enhancements will come, but in the mean time please give > feedback on the current svn code (we want to be robust for the 1.0.4 > Release :-) ) > > See you all at www.cluecon.com, talk on Skypiax August 4th at 4.30 pm ! > > -giovanni > > > > > > 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 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20090728/5c561c02/attachment-0001.html From jason at jasonjgw.net Mon Jul 27 22:25:18 2009 From: jason at jasonjgw.net (Jason White) Date: Tue, 28 Jul 2009 15:25:18 +1000 Subject: [Freeswitch-users] Multiple gateways: Priority the first bridge with prefix In-Reply-To: <24691020.post@talk.nabble.com> References: <24674260.post@talk.nabble.com> <7d0bfd8c0907270004s26db6c05j4d760a5a6ab5fcf3@mail.gmail.com> <24675073.post@talk.nabble.com> <20090727081645.GA31511@jdc.jasonjgw.net> <7d0bfd8c0907270144q74e4d0td2b810a49302ccae@mail.gmail.com> <24691020.post@talk.nabble.com> Message-ID: <20090728052518.GA10571@jdc.jasonjgw.net> Edmar Cruz wrote: > > Yes, I actually just want to not be able to communicate with the other > bridges. I have already this extension name = "sample-1". Freeswitch gets > the first extension the 2nd also trigger it. When the calls finds the match > it suits perfectly but I just want that I do not want to view the bridges > with CS_DESTROY or hangup_after_false if not found. The above text is absolutely incoherent and incomprehensible, so I don't understand what you are trying to say. Try setting on the first extension and see whether that does what you want. I hope this help. From velu.technical at gmail.com Mon Jul 27 23:10:00 2009 From: velu.technical at gmail.com (velusamy velu) Date: Tue, 28 Jul 2009 11:40:00 +0530 Subject: [Freeswitch-users] Connecting FreeSWITCH with Asterisk Message-ID: <1452e2980907272310y49bdab1fg39b490ba83c845@mail.gmail.com> Dear All, I have tried to connect the FreeSWITCH with Asterisk I have followed steps which is provided in the following link, http://wiki.freeswitch.org/wiki/Connecting_Freeswitch_And_Asterisk I have tried to call "2000" from FreeSWITCH, but I have received the following message in Asterisk console "NOTICE[6346]: chan_sip.c:10543 handle_request_invite: Failed to authenticate user "Velusamy" >;tag=69Q648H9NjrSK" I have read "Using Authentication" topic in the link, But I did understand that topic.. They have mentioned "HOSTNAME.DOMAIN.COM" in that topic. Which hostname I have to specify here? Please help me.... Regards, Velusamy.K -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090728/75c1324c/attachment.html From thangappan143 at gmail.com Mon Jul 27 23:16:43 2009 From: thangappan143 at gmail.com (Thangappan.M) Date: Tue, 28 Jul 2009 11:46:43 +0530 Subject: [Freeswitch-users] ESL problem Message-ID: <7aa29e790907272316m43671f92jf9f5442f6c7caf22@mail.gmail.com> Dear all, In the previous post, I got the information that using event outbound socket we can implement the IVR and also see the example in libs/esl/perl/server2.pl. I have seen it and understood the flow of the script.But when I was running that script it tells the following error. Can't locate loadable object for module ESL in @INC (@INC contains: /etc/perl /usr/local/lib/perl/5.8.8 /usr/local/share/perl/5.8.8 /usr/lib/perl5 /usr/share/perl5 /usr/lib/perl/5.8 /usr/share/perl/5.8 /usr/local/lib/site_perl .) at ESL.pm line 11 Compilation failed in require at server2.pl line 1. Then I checked the 11th line in the ESL.pm that line is (bootstrap ESL;) But in perl directory there is no directory called ESL. What would be the issue?. Is ESL necessary is necessary for implementing IVR using event outbound socket? -- Regards, Thangappan.M -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090728/052ac2b0/attachment.html From msc at freeswitch.org Mon Jul 27 23:53:47 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 27 Jul 2009 23:53:47 -0700 Subject: [Freeswitch-users] ESL problem In-Reply-To: <7aa29e790907272316m43671f92jf9f5442f6c7caf22@mail.gmail.com> References: <7aa29e790907272316m43671f92jf9f5442f6c7caf22@mail.gmail.com> Message-ID: <87f2f3b90907272353n142f268atb804509038147f0b@mail.gmail.com> Make certain that you've built both libesl and the Perl mod. Change directory into /usr/src/freeswitch.trunk/libs/esl (or whatever your path is to libs/esl) and do these commands: make make perlmod Then give it another shot. -MC On Mon, Jul 27, 2009 at 11:16 PM, Thangappan.M wrote: > Dear all, > > In the previous post, I got the information that using event outbound > socket we can implement the IVR and also see the example in > libs/esl/perl/server2.pl. > > I have seen it and understood the flow of the script.But when I was > running that script it tells the following error. > > Can't locate loadable object for module ESL in @INC (@INC contains: > /etc/perl /usr/local/lib/perl/5.8.8 /usr/local/share/perl/5.8.8 > /usr/lib/perl5 /usr/share/perl5 /usr/lib/perl/5.8 /usr/share/perl/5.8 > /usr/local/lib/site_perl .) at ESL.pm line 11 > Compilation failed in require at server2.pl line 1. > > Then I checked the 11th line in the ESL.pm that line is (bootstrap ESL;) > But in perl directory there is no directory called ESL. > > What would be the issue?. > Is ESL necessary is necessary for implementing IVR using event outbound > socket? > > > -- > Regards, > Thangappan.M > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20090727/4184f352/attachment.html From msc at freeswitch.org Tue Jul 28 00:02:06 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 28 Jul 2009 00:02:06 -0700 Subject: [Freeswitch-users] Connecting FreeSWITCH with Asterisk In-Reply-To: <1452e2980907272310y49bdab1fg39b490ba83c845@mail.gmail.com> References: <1452e2980907272310y49bdab1fg39b490ba83c845@mail.gmail.com> Message-ID: <87f2f3b90907280002v442c1071vb8bd0525dce290c8@mail.gmail.com> Before you go any further, could you let us know what you are trying to accomplish? What kind of calls need to go to/from Asterisk/FreeSWITCH? Do you require some sort of authentication? Are the FS and Ast machines on the same LAN? It might help for you to pastebin the output from the FS CLI when you make a test call to the Asterisk box as that might give you some clue as to what isn't working. -MC On Mon, Jul 27, 2009 at 11:10 PM, velusamy velu wrote: > Dear All, > > I have tried to connect the FreeSWITCH with Asterisk > > I have followed steps which is provided in the following link, > http://wiki.freeswitch.org/wiki/Connecting_Freeswitch_And_Asterisk > > I have tried to call "2000" from FreeSWITCH, but I have received the > following message in Asterisk console > > "NOTICE[6346]: chan_sip.c:10543 handle_request_invite: Failed to > authenticate user "Velusamy" > >;tag=69Q648H9NjrSK" > > I have read "Using Authentication" topic in the link, But I did understand > that topic.. > They have mentioned "HOSTNAME.DOMAIN.COM" in that topic. Which hostname I > have to specify here? > > Please help me.... > > Regards, > Velusamy.K > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20090728/498b4a08/attachment.html From velu.technical at gmail.com Tue Jul 28 00:32:22 2009 From: velu.technical at gmail.com (velusamy velu) Date: Tue, 28 Jul 2009 13:02:22 +0530 Subject: [Freeswitch-users] Connecting FreeSWITCH with Asterisk In-Reply-To: <87f2f3b90907280002v442c1071vb8bd0525dce290c8@mail.gmail.com> References: <1452e2980907272310y49bdab1fg39b490ba83c845@mail.gmail.com> <87f2f3b90907280002v442c1071vb8bd0525dce290c8@mail.gmail.com> Message-ID: <1452e2980907280032r46a0760ekdef3cb6c29242b84@mail.gmail.com> Dear, I am just testing that how to connect FreeSWITCH with Asterisk. I don't want any sort of authentication. Yes, the FS and Asterisk are on the same LAN.. My intention is that When I call an extension from FS, the dial plan should bridge a user in Asterisk.. Please give some suggestions... Thanks in Advance. Regards, Velusamy. On Tue, Jul 28, 2009 at 12:32 PM, Michael Collins wrote: > Before you go any further, could you let us know what you are trying to > accomplish? What kind of calls need to go to/from Asterisk/FreeSWITCH? Do > you require some sort of authentication? Are the FS and Ast machines on the > same LAN? > > It might help for you to pastebin the output from the FS CLI when you make > a test call to the Asterisk box as that might give you some clue as to what > isn't working. > > -MC > > On Mon, Jul 27, 2009 at 11:10 PM, velusamy velu wrote: > >> Dear All, >> >> I have tried to connect the FreeSWITCH with Asterisk >> >> I have followed steps which is provided in the following link, >> http://wiki.freeswitch.org/wiki/Connecting_Freeswitch_And_Asterisk >> >> I have tried to call "2000" from FreeSWITCH, but I have received the >> following message in Asterisk console >> >> "NOTICE[6346]: chan_sip.c:10543 handle_request_invite: Failed to >> authenticate user "Velusamy" >> >;tag=69Q648H9NjrSK" >> >> I have read "Using Authentication" topic in the link, But I did understand >> that topic.. >> They have mentioned "HOSTNAME.DOMAIN.COM" in that topic. Which hostname >> I have to specify here? >> >> Please help me.... >> >> Regards, >> Velusamy.K >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20090728/de45ac70/attachment-0001.html From brian at freeswitch.org Tue Jul 28 01:26:05 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 28 Jul 2009 03:26:05 -0500 Subject: [Freeswitch-users] ESL problem In-Reply-To: <87f2f3b90907272353n142f268atb804509038147f0b@mail.gmail.com> References: <7aa29e790907272316m43671f92jf9f5442f6c7caf22@mail.gmail.com> <87f2f3b90907272353n142f268atb804509038147f0b@mail.gmail.com> Message-ID: Don't forget you need to install libs/esl/perl/ESL.so and libs/esl/ perl/ESL.pm into your system perl library path. /b On Jul 28, 2009, at 1:53 AM, Michael Collins wrote: > Make certain that you've built both libesl and the Perl mod. Change > directory into /usr/src/freeswitch.trunk/libs/esl (or whatever your > path is to libs/esl) and do these commands: > make > make perlmod > > Then give it another shot. > -MC From jgonzalez at sqli.com Tue Jul 28 02:32:23 2009 From: jgonzalez at sqli.com (julien) Date: Tue, 28 Jul 2009 11:32:23 +0200 Subject: [Freeswitch-users] SIP trunk (PBX - FreeSwitch) routing problem. In-Reply-To: <60EC1D48-40FD-4BF9-B5D2-D8FAE1EBEEEC@freeswitch.org> References: <4A6DCEA5.2090502@sqli.com> <60EC1D48-40FD-4BF9-B5D2-D8FAE1EBEEEC@freeswitch.org> Message-ID: <4A6EC5A7.60802@sqli.com> Hello brian, It was not exactly at the bottom but before I tried to put it higher in the dialplan but it still doesn't work (with the same error). Thanks for your help. Brian West a ?crit : > I have to guess that you put this at the bottom of the default.xml? > > /b > > On Jul 27, 2009, at 10:58 AM, julien wrote: > > >> And in the dialplan default.xml : >> >> >> >> >> >> >> >> >> >> >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From gregory.charles at sogeti.com Tue Jul 28 02:43:42 2009 From: gregory.charles at sogeti.com (Gregory Charles) Date: Tue, 28 Jul 2009 11:43:42 +0200 Subject: [Freeswitch-users] SIP instant messaging presence signaling doesn't work. Message-ID: <20090728114342.7ej52d57dick0kss@mail.sogeti.com> Hi everybody, ? ?I intend to use Freeswitch with two Ekiga Softphones. SIP Instant? messaging works between the two softphones but SIP presence signaling? is not managed by the softphones. I try to use other softphones? (QuteCom and SIPCommunicator) and it is the same. I have the following? error in my FreeSwitch console: ? ?[WARNING] sofia.c:75 sofia_handle_sip_r_notify() delete? subscriptions for failed notify ? ?Is there any special configuration for SIP instant messaging presence? ? ?Thanks. ? ?G.C. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090728/07ced992/attachment.html -------------- next part -------------- Hi everybody, I intend to use Freeswitch with two Ekiga Softphones. SIP Instant messaging works between the two softphones but SIP presence signaling is not managed by the softphones. I try to use other softphones (QuteCom and SIPCommunicator) and it is the same. I have the following error in my FreeSwitch console: [WARNING] sofia.c:75 sofia_handle_sip_r_notify() delete subscriptions for failed notify Is there any special configuration for SIP instant messaging presence? Thanks. G.C. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090728/07ced992/attachment-0001.html From brian at freeswitch.org Tue Jul 28 02:46:29 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 28 Jul 2009 04:46:29 -0500 Subject: [Freeswitch-users] SIP trunk (PBX - FreeSwitch) routing problem. In-Reply-To: <4A6EC5A7.60802@sqli.com> References: <4A6DCEA5.2090502@sqli.com> <60EC1D48-40FD-4BF9-B5D2-D8FAE1EBEEEC@freeswitch.org> <4A6EC5A7.60802@sqli.com> Message-ID: Well press F8 and increase the debug level.. then try again you'll prob. see that its not finding it NOR matching it anywhere in your dialplan. /b On Jul 28, 2009, at 4:32 AM, julien wrote: > Hello brian, > It was not exactly at the bottom but before > > > > I tried to put it higher in the dialplan but it still doesn't work > (with > the same error). > > Thanks for your help. From jason at jasonjgw.net Tue Jul 28 02:58:18 2009 From: jason at jasonjgw.net (Jason White) Date: Tue, 28 Jul 2009 19:58:18 +1000 Subject: [Freeswitch-users] SIP trunk (PBX - FreeSwitch) routing problem. In-Reply-To: <4A6EC5A7.60802@sqli.com> References: <4A6DCEA5.2090502@sqli.com> <60EC1D48-40FD-4BF9-B5D2-D8FAE1EBEEEC@freeswitch.org> <4A6EC5A7.60802@sqli.com> Message-ID: <20090728095818.GA23876@jdc.jasonjgw.net> julien wrote: > It was not exactly at the bottom but before > > Why not put it in the default directory, from which it will be included by the above line? If necessary, you could comment out any entries in default.xml that might be matched first. I've debugged this kind of problem before, and the best solution has always been to read the logs carefully to see which extensions matched (or didn't match). Also, if necessary, check out freeswitch/log/freeswitch.xml.fsxml to see where your extension ends up in the final dial plan. From mike at jerris.com Tue Jul 28 05:37:25 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 28 Jul 2009 08:37:25 -0400 Subject: [Freeswitch-users] SIP instant messaging presence signaling doesn't work. In-Reply-To: <20090728114342.7ej52d57dick0kss@mail.sogeti.com> References: <20090728114342.7ej52d57dick0kss@mail.sogeti.com> Message-ID: You must turn on the option to manage presence in the sip profile. Mike On Jul 28, 2009, at 5:43 AM, Gregory Charles wrote: > Hi everybody, > > I intend to use Freeswitch with two Ekiga Softphones. SIP Instant > messaging works between the two softphones but SIP presence signaling > is not managed by the softphones. I try to use other softphones > (QuteCom and SIPCommunicator) and it is the same. I have the following > error in my FreeSwitch console: > > [WARNING] sofia.c:75 sofia_handle_sip_r_notify() delete > subscriptions for failed notify > > Is there any special configuration for SIP instant messaging > presence? > > Thanks. > > G.C. > Hi everybody, > > I intend to use Freeswitch with two Ekiga Softphones. SIP Instant > messaging works between the two softphones but SIP presence > signaling is not managed by the softphones. I try to use other > softphones (QuteCom and SIPCommunicator) and it is the same. I have > the following error in my FreeSwitch console: > > [WARNING] sofia.c:75 sofia_handle_sip_r_notify() delete > subscriptions for failed notify > > Is there any special configuration for SIP instant messaging presence? > > Thanks. > > G.C. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From jgonzalez at sqli.com Tue Jul 28 08:00:58 2009 From: jgonzalez at sqli.com (julien) Date: Tue, 28 Jul 2009 17:00:58 +0200 Subject: [Freeswitch-users] SIP trunk (PBX - FreeSwitch) routing problem. In-Reply-To: References: <4A6DCEA5.2090502@sqli.com> <60EC1D48-40FD-4BF9-B5D2-D8FAE1EBEEEC@freeswitch.org> <4A6EC5A7.60802@sqli.com> Message-ID: <4A6F12AA.1030605@sqli.com> Thanks for the tip Brian. It seems that the extension matches successfully in the dialplan (PASS, instead of FAIL for all other entries of the dialplan) : Dialplan: sofia/internal/[EMAIL PROTECTED] parsing [default->pbxlyon] continue=false Dialplan: sofia/internal/[EMAIL PROTECTED] Regex (PASS) [pbxlyon] destination_number(300) =~ /300/ break=on-false But it leads nowhere. After the match the connection to the PBX fails : 2009-07-28 16:16:43.963836 [NOTICE] switch_channel.c:602 New Channel sofia/external/300 [46fca878-7b81-11de-a9c2-0f49fee5280a] 2009-07-28 16:16:43.963836 [DEBUG] mod_sofia.c:2751 (sofia/external/300) State Change CS_NEW -> CS_INIT 2009-07-28 16:16:43.963836 [DEBUG] switch_core_session.c:933 Send signal sofia/external/300 [BREAK] 2009-07-28 16:16:43.973759 [DEBUG] switch_core_state_machine.c:397 (sofia/external/300) Running State Change CS_INIT 2009-07-28 16:16:43.973759 [DEBUG] switch_core_state_machine.c:480 (sofia/external/300) State INIT 2009-07-28 16:16:43.973759 [DEBUG] mod_sofia.c:83 sofia/external/300 SOFIA INIT 2009-07-28 16:16:43.975221 [DEBUG] mod_sofia.c:111 (sofia/external/300) State Change CS_INIT -> CS_ROUTING 2009-07-28 16:16:43.975221 [DEBUG] switch_core_session.c:933 Send signal sofia/external/300 [BREAK] 2009-07-28 16:16:43.975221 [DEBUG] sofia.c:3215 Channel sofia/external/300 entering state [calling][0] 2009-07-28 16:16:43.975221 [DEBUG] switch_core_state_machine.c:480 (sofia/external/300) State INIT going to sleep 2009-07-28 16:16:43.975221 [DEBUG] switch_core_state_machine.c:397 (sofia/external/300) Running State Change CS_ROUTING 2009-07-28 16:16:43.975221 [DEBUG] switch_core_state_machine.c:483 (sofia/external/300) State ROUTING 2009-07-28 16:16:43.975221 [DEBUG] mod_sofia.c:130 sofia/external/300 SOFIA ROUTING 2009-07-28 16:16:43.975221 [DEBUG] switch_ivr_originate.c:63 (sofia/external/300) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2009-07-28 16:16:43.975221 [DEBUG] switch_core_session.c:933 Send signal sofia/external/300 [BREAK] 2009-07-28 16:16:43.976763 [DEBUG] switch_core_state_machine.c:483 (sofia/external/300) State ROUTING going to sleep 2009-07-28 16:16:43.976763 [DEBUG] switch_core_state_machine.c:397 (sofia/external/300) Running State Change CS_CONSUME_MEDIA 2009-07-28 16:16:43.976763 [DEBUG] switch_core_state_machine.c:502 (sofia/external/300) State CONSUME_MEDIA 2009-07-28 16:16:44.82786 [DEBUG] sofia.c:3215 Channel sofia/external/300 entering state [terminated][404] 2009-07-28 16:16:44.82786 [NOTICE] sofia.c:3775 Hangup sofia/external/300 [CS_CONSUME_MEDIA] [NO_ROUTE_DESTINATION] It looks to me that it's more a problem from the gateway than from the dialplan? Don't you think so? Do you think the way I defined my gateway is good for a connexion to a PBX ? Thanks for your replies Brian and Jason. Brian West a ?crit : > Well press F8 and increase the debug level.. then try again you'll > prob. see that its not finding it NOR matching it anywhere in your > dialplan. > > /b > > On Jul 28, 2009, at 4:32 AM, julien wrote: > > >> Hello brian, >> It was not exactly at the bottom but before >> >> >> >> I tried to put it higher in the dialplan but it still doesn't work >> (with >> the same error). >> >> Thanks for your help. >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.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 Jul 28 08:11:17 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 28 Jul 2009 10:11:17 -0500 Subject: [Freeswitch-users] SIP trunk (PBX - FreeSwitch) routing problem. In-Reply-To: <4A6F12AA.1030605@sqli.com> References: <4A6DCEA5.2090502@sqli.com> <60EC1D48-40FD-4BF9-B5D2-D8FAE1EBEEEC@freeswitch.org> <4A6EC5A7.60802@sqli.com> <4A6F12AA.1030605@sqli.com> Message-ID: The remote end said 404 /b On Jul 28, 2009, at 10:00 AM, julien wrote: > 2009-07-28 16:16:44.82786 [DEBUG] sofia.c:3215 Channel > sofia/external/300 entering state [terminated][404] From vkozak at abisoft.spb.ru Tue Jul 28 08:16:45 2009 From: vkozak at abisoft.spb.ru (Kozak Vladimir) Date: Tue, 28 Jul 2009 19:16:45 +0400 Subject: [Freeswitch-users] originate in dialplan Message-ID: <793928086CB844E0852A49DE57CADEC8@abisoft.biz> Hello, Please tell me, how can I execute originate new call and uuid_bridge in dial plan. I tried to make like thise: result: [ERR] switch_core_session.c:1239 switch_core_session_execute_application() Invalid Application originate [ERR] switch_core_session.c:1239 switch_core_session_execute_application() Invalid Application uuid_bridge Best regards. vkozak -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090728/49be300c/attachment-0001.html From msc at freeswitch.org Tue Jul 28 09:07:20 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 28 Jul 2009 09:07:20 -0700 Subject: [Freeswitch-users] originate in dialplan In-Reply-To: <793928086CB844E0852A49DE57CADEC8@abisoft.biz> References: <793928086CB844E0852A49DE57CADEC8@abisoft.biz> Message-ID: <87f2f3b90907280907r1bcf4705i695dcc9f3ed4091a@mail.gmail.com> What exactly are you trying to accomplish with this dialplan entry? That will help us answer your question. -MC 2009/7/28 Kozak Vladimir > Hello, > > Please tell me, how can I execute originate new call and uuid_bridge in > dial plan. > I tried to make like thise: > data="user/$${destination_end_point} &playback(${hold_music})"/> > data="user/$${destination_end_point}, &playback($${hold_music})"/> > > > result: > [ERR] switch_core_session.c:1239 > switch_core_session_execute_application() Invalid Application originate > [ERR] switch_core_session.c:1239 > switch_core_session_execute_application() Invalid Application uuid_bridge > Best regards. > vkozak > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20090728/6012eb5e/attachment.html From helmut.kuper at ewetel.de Tue Jul 28 09:14:03 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Tue, 28 Jul 2009 18:14:03 +0200 Subject: [Freeswitch-users] CELT codec code number Message-ID: <4A6F23CB.6090405@ewetel.de> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, today I finylly got a working Ekiga Softphone version which is able to use high quality celt codec with FS :) On my way to get it work with FS I found that Ekiga currently uses codec code 95 in SDP while FS uses 114. Changing FS to 95 made it works so I'm now able to listen to my PBX-based MP3-Player on Windows Desktop instead of using Ubuntu. veeeeeerrry cool work of FS team !!!!! Concerning the codec code 95, 114 or whatever I found the link below, which states that every codec code between 96 and 127 is OK but it seems they prefer 97 ... http://tools.ietf.org/html/draft-valin-celt-rtp-profile-02#section-5 You can get windows Ekiga 3.2.5 (with celt 0.5.1) here: http://wwwuser.gwdg.de/~mrickma/ekiga/ekiga-setup-3.2.5-release.exe regards Helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) iD8DBQFKbyPL4tZeNddg3dwRAiDMAKChqgWeirYklgra5nN7NGwZSpK6wQCgjYox Q/okubHauhgjtoiogzFM9mI= =Ml4q -----END PGP SIGNATURE----- From zolotov at altron.ua Tue Jul 28 09:24:38 2009 From: zolotov at altron.ua (Evgeniy Zolotov) Date: Tue, 28 Jul 2009 19:24:38 +0300 Subject: [Freeswitch-users] originate in dialplan In-Reply-To: <793928086CB844E0852A49DE57CADEC8@abisoft.biz> References: <793928086CB844E0852A49DE57CADEC8@abisoft.biz> Message-ID: <1783498449.20090728192438@altron.ua> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090728/579a61a4/attachment.html From brian at freeswitch.org Tue Jul 28 09:23:36 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 28 Jul 2009 11:23:36 -0500 Subject: [Freeswitch-users] CELT codec code number In-Reply-To: <4A6F23CB.6090405@ewetel.de> References: <4A6F23CB.6090405@ewetel.de> Message-ID: <8AD48BD2-8752-4F99-9CF7-96955C485E98@freeswitch.org> On Jul 28, 2009, at 11:14 AM, Helmut Kuper wrote: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > Hello, > > today I finylly got a working Ekiga Softphone version which is able to > use high quality celt codec with FS :) > > > On my way to get it work with FS I found that Ekiga currently uses > codec > code 95 in SDP while FS uses 114. Changing FS to 95 made it works so > I'm > now able to listen to my PBX-based MP3-Player on Windows Desktop > instead > of using Ubuntu. It should work if they use different codec numbers.... I suspect we are sending on 114 and receiving on 95 which is what should take place. This is one of those areas most people fail to implement properly. We send the remote our RTP map they send us theirs... Can you get a packet capture of this taking place so I can verify who is at fault? /b > veeeeeerrry cool work of FS team !!!!! > > Concerning the codec code 95, 114 or whatever I found the link below, > which states that every codec code between 96 and 127 is OK but it > seems > they prefer 97 ... > > http://tools.ietf.org/html/draft-valin-celt-rtp-profile-02#section-5 > > > You can get windows Ekiga 3.2.5 (with celt 0.5.1) here: > http://wwwuser.gwdg.de/~mrickma/ekiga/ekiga-setup-3.2.5-release.exe > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090728/e946c19f/attachment.html From mike at jerris.com Tue Jul 28 10:06:21 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 28 Jul 2009 13:06:21 -0400 Subject: [Freeswitch-users] CELT codec code number In-Reply-To: <8AD48BD2-8752-4F99-9CF7-96955C485E98@freeswitch.org> References: <4A6F23CB.6090405@ewetel.de> <8AD48BD2-8752-4F99-9CF7-96955C485E98@freeswitch.org> Message-ID: <206116BA-A08A-45A4-B72F-990E5CC680FE@jerris.com> using 95 is wrong. That is not part of the dynamic range for unassigned codecs. This needs to be fixed on their side. MIke On Jul 28, 2009, at 12:23 PM, Brian West wrote: > > On Jul 28, 2009, at 11:14 AM, Helmut Kuper wrote: > >> -----BEGIN PGP SIGNED MESSAGE----- >> Hash: SHA1 >> >> Hello, >> >> today I finylly got a working Ekiga Softphone version which is able >> to >> use high quality celt codec with FS :) >> >> >> On my way to get it work with FS I found that Ekiga currently uses >> codec >> code 95 in SDP while FS uses 114. Changing FS to 95 made it works >> so I'm >> now able to listen to my PBX-based MP3-Player on Windows Desktop >> instead >> of using Ubuntu. > > It should work if they use different codec numbers.... I suspect we > are sending on 114 and receiving on 95 which is what should take > place. This is one of those areas most people fail to implement > properly. We send the remote our RTP map they send us theirs... Can > you get a packet capture of this taking place so I can verify who is > at fault? > > /b > > > >> veeeeeerrry cool work of FS team !!!!! >> >> Concerning the codec code 95, 114 or whatever I found the link below, >> which states that every codec code between 96 and 127 is OK but it >> seems >> they prefer 97 ... >> >> http://tools.ietf.org/html/draft-valin-celt-rtp-profile-02#section-5 >> >> >> You can get windows Ekiga 3.2.5 (with celt 0.5.1) here: >> http://wwwuser.gwdg.de/~mrickma/ekiga/ekiga-setup-3.2.5-release.exe >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20090728/358c4680/attachment.html From brian at freeswitch.org Tue Jul 28 10:13:31 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 28 Jul 2009 12:13:31 -0500 Subject: [Freeswitch-users] CELT codec code number In-Reply-To: <206116BA-A08A-45A4-B72F-990E5CC680FE@jerris.com> References: <4A6F23CB.6090405@ewetel.de> <8AD48BD2-8752-4F99-9CF7-96955C485E98@freeswitch.org> <206116BA-A08A-45A4-B72F-990E5CC680FE@jerris.com> Message-ID: I totally missed this at first... but 95 wouldn't dynamically work because its not 96-127 /b On Jul 28, 2009, at 12:06 PM, Michael Jerris wrote: > using 95 is wrong. That is not part of the dynamic range for > unassigned codecs. This needs to be fixed on their side. > > MIke From kristian.kielhofner at gmail.com Tue Jul 28 10:28:03 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Tue, 28 Jul 2009 13:28:03 -0400 Subject: [Freeswitch-users] AbsoluteTimeout: The FreeSWITCH way Message-ID: <2d9149cd0907281028u7bb8ed52ufe72370aaae9311c@mail.gmail.com> Hello everyone, I need to set a maximum call duration. What is the current recommended way to implement this in FreeSWITCH? I'm looking for something similar to AbsoluteTimeout() in Asterisk. Thanks! -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From saeedahmad1981 at gmail.com Tue Jul 28 10:48:11 2009 From: saeedahmad1981 at gmail.com (Saeed Ahmad) Date: Tue, 28 Jul 2009 19:48:11 +0200 Subject: [Freeswitch-users] AbsoluteTimeout: The FreeSWITCH way In-Reply-To: <2d9149cd0907281028u7bb8ed52ufe72370aaae9311c@mail.gmail.com> References: <2d9149cd0907281028u7bb8ed52ufe72370aaae9311c@mail.gmail.com> Message-ID: On Tue, Jul 28, 2009 at 7:28 PM, Kristian Kielhofner < kristian.kielhofner at gmail.com> wrote: > Hello everyone, > > I need to set a maximum call duration. What is the current > recommended way to implement this in FreeSWITCH? I'm looking for > something similar to AbsoluteTimeout() in Asterisk. > > Thanks! > > -- > Kristian Kielhofner > http://www.astlinux.org > http://blog.krisk.org > http://www.star2star.com > http://www.submityoursip.com > http://www.voalte.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/20090728/acb9d27e/attachment.html From msc at freeswitch.org Tue Jul 28 10:50:49 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 28 Jul 2009 10:50:49 -0700 Subject: [Freeswitch-users] AbsoluteTimeout: The FreeSWITCH way In-Reply-To: <2d9149cd0907281028u7bb8ed52ufe72370aaae9311c@mail.gmail.com> References: <2d9149cd0907281028u7bb8ed52ufe72370aaae9311c@mail.gmail.com> Message-ID: <87f2f3b90907281050g14cd8c74g3bfa3e5481fa2ecf@mail.gmail.com> What needs to happen at the end of the timeout? In any case you can use the sched_XXX APIs: sched_api sched_transfer sched_hangup You can get fancy or just hangup up on the call after X number of seconds... :) -MC On Tue, Jul 28, 2009 at 10:28 AM, Kristian Kielhofner < kristian.kielhofner at gmail.com> wrote: > Hello everyone, > > I need to set a maximum call duration. What is the current > recommended way to implement this in FreeSWITCH? I'm looking for > something similar to AbsoluteTimeout() in Asterisk. > > Thanks! > > -- > Kristian Kielhofner > http://www.astlinux.org > http://blog.krisk.org > http://www.star2star.com > http://www.submityoursip.com > http://www.voalte.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/20090728/39c90253/attachment.html From mrene_lists at avgs.ca Tue Jul 28 10:54:58 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Tue, 28 Jul 2009 13:54:58 -0400 Subject: [Freeswitch-users] AbsoluteTimeout: The FreeSWITCH way In-Reply-To: <87f2f3b90907281050g14cd8c74g3bfa3e5481fa2ecf@mail.gmail.com> References: <2d9149cd0907281028u7bb8ed52ufe72370aaae9311c@mail.gmail.com> <87f2f3b90907281050g14cd8c74g3bfa3e5481fa2ecf@mail.gmail.com> Message-ID: You can also schedule a playback then a hangup, what comes after the ! is the hangup cause. sched_broadcast,Schedule a broadcast in the future,[+] > Actually am connecting Freeswitch and Asterisks > > Do you think the issue is the codec? No. I've explained this already: the issue in the log that you provided is a failure of FreeSWITCH to match your dial plan extension. From norm at goes.com Sat Jul 4 04:42:08 2009 From: norm at goes.com (Norman Brandinger) Date: Sat, 4 Jul 2009 07:42:08 -0400 (EDT) Subject: [Freeswitch-users] Baby Update! Message-ID: <46295.75.97.76.77.1246707728.squirrel@mail.goes.com> Congrats, Hope mom and baby are doing well. Let's see some photos. Norm > Kaiden Anthony Chandler arrived Friday July 3rd at 1411 EDT 7lbs 10oz > > YAY... Congrats mr Lanman! > > /b > > On Jul 3, 2009, at 8:58 AM, David Knell wrote: > >> On Fri, 2009-07-03 at 17:43 +0600, Muhammad Shahzad wrote: >>> Congratulations to Ray and Samantha. Lets see what new features and >>> bug fixes we will get in their "new version"..! ;-) >> >> Bug fixes..?! I'd refer you to Philip Larkin (went to my school, a >> bit >> before my time, poet, deceased, recently voted "Britain's favourite >> poet") whose "This Be The Verse" suggests otherwise: >> http://www.artofeurope.com/larkin/lar2.htm >> >> [as a recent father myself, I'm trying to prove him wrong..] >> >> --Dave >> >> -- >> David Knell, Director, 3C Limited >> T: +44 20 3298 2000 >> E: dave at 3c.co.uk >> W: 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 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > Global Online (GOES) 271 Main St., Suite C Hackettstown, NJ 07840-2032 (908) 813-0600 x8105 From codecomplete at free.fr Sat Jul 4 13:05:12 2009 From: codecomplete at free.fr (Fred-145) Date: Sat, 4 Jul 2009 13:05:12 -0700 (PDT) Subject: [Freeswitch-users] Cheapest, most compact FreeSwitch appliance? Message-ID: <24337599.post@talk.nabble.com> Hello Atcom's IP appliance is nice but it uses a BlackFish CPU, which is not supported by FreeSwitch yet. What FreeSwitch-appliance do you is the cheapest, most compact out there? If possible, I'd rather a device that has room for an SDD or 2.5" HD so as to be able to use a standard *nix distro. Thank you. -- View this message in context: http://www.nabble.com/Cheapest%2C-most-compact-FreeSwitch-appliance--tp24337599p24337599.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From brian at freeswitch.org Sat Jul 4 13:18:53 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 4 Jul 2009 15:18:53 -0500 Subject: [Freeswitch-users] Cheapest, most compact FreeSwitch appliance? In-Reply-To: <24337599.post@talk.nabble.com> References: <24337599.post@talk.nabble.com> Message-ID: I use one of the intel atom boxes at home. /b On Jul 4, 2009, at 3:05 PM, Fred-145 wrote: > What FreeSwitch-appliance do you is the cheapest, most compact out > there? If > possible, I'd rather a device that has room for an SDD or 2.5" HD so > as to > be able to use a standard *nix distro. From jay.fenton at howlertech.com Sat Jul 4 13:59:26 2009 From: jay.fenton at howlertech.com (Jay Fenton) Date: Sat, 4 Jul 2009 22:59:26 +0200 Subject: [Freeswitch-users] Cheapest, most compact FreeSwitch appliance? In-Reply-To: References: <24337599.post@talk.nabble.com> Message-ID: > I use one of the intel atom boxes at home. A BeagleBoard (http://beagleboard.org/) would make for a decent FreeSWITCH appliance (tiny and only 2 watts). -- Regards, Jay Fenton, CTO Howler Technologies Ltd, 11 Charlotte Mews, London, W1T 4EQ tel: +44 207 099 7095 fax: +44 207 099 7098 http://www.howlertech.com/ http://www.linkedin.com/in/jfenton Registered in England & Wales, Company No. 06285634 From brian at freeswitch.org Sat Jul 4 14:06:50 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 4 Jul 2009 16:06:50 -0500 Subject: [Freeswitch-users] Cheapest, most compact FreeSwitch appliance? In-Reply-To: References: <24337599.post@talk.nabble.com> Message-ID: The problem is what case options do you have for such a device? /b On Jul 4, 2009, at 3:59 PM, Jay Fenton wrote: > >> I use one of the intel atom boxes at home. > > A BeagleBoard (http://beagleboard.org/) would make for a decent > FreeSWITCH appliance (tiny and only 2 watts). > > -- > Regards, > > Jay Fenton, CTO > Howler Technologies Ltd, 11 Charlotte Mews, London, W1T 4EQ > tel: +44 207 099 7095 fax: +44 207 099 7098 > http://www.howlertech.com/ > http://www.linkedin.com/in/jfenton > > Registered in England & Wales, Company No. 06285634 > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.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 Jul 4 14:07:08 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 4 Jul 2009 16:07:08 -0500 Subject: [Freeswitch-users] Cheapest, most compact FreeSwitch appliance? In-Reply-To: <24337599.post@talk.nabble.com> References: <24337599.post@talk.nabble.com> Message-ID: <09DDA996-8374-4CC7-BC79-89B9F7BDCE07@freeswitch.org> http://www.marvell.com/products/embedded_processors/developer/kirkwood/sheevaplug.jsp That would work better. /b On Jul 4, 2009, at 3:05 PM, Fred-145 wrote: > > Hello > > Atcom's IP appliance is nice but it uses a BlackFish CPU, which is not > supported by FreeSwitch yet. > > What FreeSwitch-appliance do you is the cheapest, most compact out > there? If > possible, I'd rather a device that has room for an SDD or 2.5" HD so > as to > be able to use a standard *nix distro. > > Thank you. > -- > View this message in context: http://www.nabble.com/Cheapest%2C-most-compact-FreeSwitch-appliance--tp24337599p24337599.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 william.suffill at gmail.com Sat Jul 4 15:21:10 2009 From: william.suffill at gmail.com (William Suffill) Date: Sat, 4 Jul 2009 18:21:10 -0400 Subject: [Freeswitch-users] Cheapest, most compact FreeSwitch appliance? In-Reply-To: <09DDA996-8374-4CC7-BC79-89B9F7BDCE07@freeswitch.org> References: <24337599.post@talk.nabble.com> <09DDA996-8374-4CC7-BC79-89B9F7BDCE07@freeswitch.org> Message-ID: <6b65470d0907041521t2f79d079x481e2759bfd6a16a@mail.gmail.com> Ya I have a SheevaPlug but yet to have anything interesting to report about making it do anything. The potential is there tho. -- W From gcd at i.ph Sat Jul 4 19:39:16 2009 From: gcd at i.ph (Nandy Dagondon) Date: Sun, 5 Jul 2009 10:39:16 +0800 Subject: [Freeswitch-users] Cheapest, most compact FreeSwitch appliance? In-Reply-To: <6b65470d0907041521t2f79d079x481e2759bfd6a16a@mail.gmail.com> References: <24337599.post@talk.nabble.com> <09DDA996-8374-4CC7-BC79-89B9F7BDCE07@freeswitch.org> <6b65470d0907041521t2f79d079x481e2759bfd6a16a@mail.gmail.com> Message-ID: <7d0bfd8c0907041939h5d49729dwab8f937b3862b3f5@mail.gmail.com> we have a forum on compact,fanless last may. On Sun, Jul 5, 2009 at 6:21 AM, William Suffill wrote: > Ya I have a SheevaPlug but yet to have anything interesting to report > about making it do anything. The potential is there tho. > > -- W > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090705/555d3afe/attachment-0002.html From gcd at i.ph Sat Jul 4 19:41:18 2009 From: gcd at i.ph (Nandy Dagondon) Date: Sun, 5 Jul 2009 10:41:18 +0800 Subject: [Freeswitch-users] Compact, fanless appliance? In-Reply-To: <23430873.post@talk.nabble.com> References: <23193738.post@talk.nabble.com> <1241015506.11362.1.camel@portable-evil> <23317579.post@talk.nabble.com> <7d0bfd8c0904301908o7bca18b5gfe8a830f1f54b41e@mail.gmail.com> <23364535.post@talk.nabble.com> <49FEA513.8020109@mctelefonia.com> <23366596.post@talk.nabble.com> <49FEDA25.2050703@mctelefonia.com> <23430873.post@talk.nabble.com> Message-ID: <7d0bfd8c0907041941s75e541dex982fae2195858ea4@mail.gmail.com> just bumping this topic. -nandy On Fri, May 8, 2009 at 12:44 AM, Fred-145 wrote: > > > Antonio Gallo wrote: > > Alix cases are like 6/9 ? from their shop site. I think its easy to find > > someone who work with aluminium that can make for you custom boxes for > > like like 6/20 ? at pcs > > Unfortunately, none of the PCEngines cases ( > www.pcengines.ch/order1.php?c=2) > allow for a PCI slot, either on top of the mobo, or away from it :-/ > > I'll see if I can get those from Soekris (http://soekris.eu/shop/cases_en/ > ) > allow this, and if I can get a good price for a case + PSU. > > Thank you. > -- > View this message in context: > http://www.nabble.com/Compact%2C-fanless-appliance--tp23193738p23430873.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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090705/1bb0745b/attachment-0002.html From krice at suspicious.org Sat Jul 4 19:49:20 2009 From: krice at suspicious.org (Ken Rice) Date: Sat, 04 Jul 2009 21:49:20 -0500 Subject: [Freeswitch-users] Compact, fanless appliance? In-Reply-To: <7d0bfd8c0907041941s75e541dex982fae2195858ea4@mail.gmail.com> Message-ID: No need to bump these things as this is a mailing list and it annoys quite a few people when you do that From: Nandy Dagondon Reply-To: Date: Sun, 5 Jul 2009 10:41:18 +0800 To: Subject: Re: [Freeswitch-users] Compact, fanless appliance? just bumping this topic. -nandy On Fri, May 8, 2009 at 12:44 AM, Fred-145 wrote: > > > Antonio Gallo wrote: >> > Alix cases are like 6/9 ? from their shop site. I think its easy to find >> > someone who work with aluminium that can make for you custom boxes for >> > like like 6/20 ? at pcs > > Unfortunately, none of the PCEngines cases (www.pcengines.ch/order1.php?c=2 > ) > allow for a PCI slot, either on top of the mobo, or away from it :-/ > > I'll see if I can get those from Soekris (http://soekris.eu/shop/cases_en/) > allow this, and if I can get a good price for a case + PSU. > > Thank you. > -- > View this message in context: > http://www.nabble.com/Compact%2C-fanless-appliance--tp23193738p23430873.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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090704/c5c69221/attachment-0002.html From jay.fenton at howlertech.com Sun Jul 5 03:20:37 2009 From: jay.fenton at howlertech.com (Jay Fenton) Date: Sun, 5 Jul 2009 12:20:37 +0200 Subject: [Freeswitch-users] Cheapest, most compact FreeSwitch appliance? In-Reply-To: References: <24337599.post@talk.nabble.com> Message-ID: <7C0A0FFD-EC03-46DE-A277-AE793017D617@howlertech.com> > The problem is what case options do you have for such a device? https://specialcomp.com/beagleboard/order.htm If you look on there there's a clear acrylic case for the BeagleBoard - I haven't seen any others available. -- Regards, Jay Fenton, CTO Howler Technologies Ltd, 11 Charlotte Mews, London, W1T 4EQ tel: +44 207 099 7095 fax: +44 207 099 7098 http://www.howlertech.com/ http://www.linkedin.com/in/jfenton Registered in England & Wales, Company No. 06285634 From gustavodartagnan at yahoo.com Sun Jul 5 10:43:43 2009 From: gustavodartagnan at yahoo.com (Gustavo Dartagnan Xavier) Date: Sun, 5 Jul 2009 10:43:43 -0700 (PDT) Subject: [Freeswitch-users] Trying to confirm answer directly on Originate command Message-ID: <537628.53680.qm@web57108.mail.re3.yahoo.com> I'm trying to build a click to call app using the FreeSWITCH webapi. Maybe I'm trying the wrong way, but, I couldn't understand why it's playing the confirmation audio only on the second leg, and not only on the first one. Is this originate command correct? http://10.141.1.137:8080/webapi/originate?{forked_dial=false,ignore_early_media=true,origination_caller_id_name=4140637770,origination_caller_id_number=4140637770,originate_timeout=30}[group_confirm_key=5,group_confirm_file=playback/usr/share/sounds/alsa/Noise.wav]sofia/gateway/cs2k/141130252530 at 10.140.131.208%20&bridge({ignore_early_media=false,origination_caller_id_name=4140637770,origination_caller_id_number=4140637770,call_timeout=60}sofia/gateway/cs2k/141140636805 at 10.140.131.208) Is there an easy way to do it? Thanks in advance, Gustavo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090705/0472d112/attachment-0002.html From codecomplete at free.fr Sun Jul 5 12:11:20 2009 From: codecomplete at free.fr (Fred-145) Date: Sun, 5 Jul 2009 12:11:20 -0700 (PDT) Subject: [Freeswitch-users] Cheapest, most compact FreeSwitch appliance? In-Reply-To: <7C0A0FFD-EC03-46DE-A277-AE793017D617@howlertech.com> References: <24337599.post@talk.nabble.com> <7C0A0FFD-EC03-46DE-A277-AE793017D617@howlertech.com> Message-ID: <24345928.post@talk.nabble.com> jfenton wrote: > If you look on there there's a clear acrylic case for the BeagleBoard > - I haven't seen any others available. That's often the problem with appliances: Once the CPU/modo, RAM, SDD/HD, PSU, case, and a PCI FXO board are computed, we end up with something as expensive as a regular PC. Is acrylic easy to work with, eg. using a Dremel? If it is, I could save money on the case, and stick a PCI card beneath the mobo and have a stand-alone, compact thingy while still using a standard Linux distro. -- View this message in context: http://www.nabble.com/Cheapest%2C-most-compact-FreeSwitch-appliance--tp24337599p24345928.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From tzury.by at reguluslabs.com Sun Jul 5 00:04:47 2009 From: tzury.by at reguluslabs.com (Tzury Bar Yochay) Date: Sun, 5 Jul 2009 10:04:47 +0300 Subject: [Freeswitch-users] Full Redundant Environment For FS In-Reply-To: <10128ef10907042256o7ff8887euf1046915aba3017d@mail.gmail.com> References: <10128ef10907042256o7ff8887euf1046915aba3017d@mail.gmail.com> Message-ID: <10128ef10907050004i721bb406r38e4659e698fe46e@mail.gmail.com> Hi all, I was wondering whether this is possible to achieve with FS of not. And if so what are the best practices for this. We wish to have our environment fully redundant, that is, live session should continue uninterrupted when an FS server goes down. Is there a document somewhere describe how to do it? From mgg at giagnocavo.net Sun Jul 5 14:06:48 2009 From: mgg at giagnocavo.net (Michael Giagnocavo) Date: Sun, 5 Jul 2009 17:06:48 -0400 Subject: [Freeswitch-users] Full Redundant Environment For FS In-Reply-To: <10128ef10907050004i721bb406r38e4659e698fe46e@mail.gmail.com> References: <10128ef10907042256o7ff8887euf1046915aba3017d@mail.gmail.com> <10128ef10907050004i721bb406r38e4659e698fe46e@mail.gmail.com> Message-ID: <6E8D2069C08AA84A83D336E996AE4C67027C3AD362@mse17be1.mse17.exchange.ms> In case of FS or server failure, this is not available yet. Search archives for more info, but in summary, it's a lot of work. However, you can do things like virtualization to have planned live migration. -Michael -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Tzury Bar Yochay Sent: Sunday, July 05, 2009 1:05 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Full Redundant Environment For FS Hi all, I was wondering whether this is possible to achieve with FS of not. And if so what are the best practices for this. We wish to have our environment fully redundant, that is, live session should continue uninterrupted when an FS server goes down. Is there a document somewhere describe how to do 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 geoffreymina at gmail.com Sun Jul 5 15:29:05 2009 From: geoffreymina at gmail.com (geoffreymina at gmail.com) Date: Sun, 05 Jul 2009 22:29:05 +0000 Subject: [Freeswitch-users] Contemplating a jump from Asterisk to Freeswitch Message-ID: <000e0cd4043ed3da65046dfcea6d@google.com> Hello, I have been reading through the on-line info as well as some reviews of the FreeSwitch platform. I am fairly convinced at this point that FreeSwitch is at least something I need to carefully look into. Our company utilizes asterisk to support our SaaS ACD/VPD/IVR platform. We currently support many thousands of concurrent agents (inbound and outbound). I have spent a lot of time trouble shooting bugs and working through 'issues' with asterisk. While I have tamed the beast, I am still not thrilled with the performance, nor am I very excited about the direction the project appears to be heading. It seems like every time a 'fix' is committed to SVN, it breaks something else. It's kind of like the wild-wild-west over there... and it certainly doesn't give me the warm/fuzzies when thinking about the future of my company. One of the benefits of our architecture is that our business logic is completely abstracted from the asterisk system. We use a combination of FastAGI and AMI to control channels on the asterisk server. We have a Java based server which interfaces with the higher level call routing engines. It looks to me like the Mod_event_socket would probably satisfy my requirements for controlling the calls via an external process, although it doesn't look as cut/dry as the FastAGI model. I haven't seen anything which would let me know the equivalent of the FastAGI 'script' being requested. The other thing I haven't seen is how to dynamically create conferences on the fly and redirect channels into them. We use app_conference on asterisk to avoid the ztdummy issue. Once the higher level intelligence engine determines two channels need to speak with each other, they are both redirected via AMI Redirect into a dynamic Conference created just for that particular call. Also - what is the status of call progress on FreeSwitch? Some things that are important to me are answering machine detection as well as detecting SIT intercept tones in the early media stream... any love here? I have a ton more questions, but this seems like a good start. Thanks! Geoff -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090705/de712507/attachment-0002.html From edpimentl at gmail.com Sun Jul 5 16:17:16 2009 From: edpimentl at gmail.com (EdPimentl) Date: Sun, 5 Jul 2009 19:17:16 -0400 Subject: [Freeswitch-users] Cheapest, most compact FreeSwitch appliance? In-Reply-To: <24345928.post@talk.nabble.com> References: <24337599.post@talk.nabble.com> <7C0A0FFD-EC03-46DE-A277-AE793017D617@howlertech.com> <24345928.post@talk.nabble.com> Message-ID: <9dc4a1670907051617k3c112eeal2356d3df55283f89@mail.gmail.com> Back in April I posted these links on the list, in regards to a similar question http://www.logicsupply.com/products/dex4501 http://www.logicsupply.com/products/de945_fl http://en.wikipedia.org/wiki/NSLU2 http://en.wikipedia.org/wiki/Gumstix http://www.cheaprouter.us http://www.avm.de/en/Produkte/FRITZBox/FRITZ_Box_Fon_WLAN_7270/index.php http://www.plugcomputer.org/ http://forum.openwrt.org/viewtopic.php?pid=83701#p83701 http://www.pikatechnologies.com/ http://www.pikatechnologies.com/english/View.asp?x=608 http://fit-pc2.com http://www.cappuccinopc.com/solutions/fanless.asp http://www.wdlsystems.com/ebox/ebox.shtml http://www.linuxdevices.com/articles/AT2016997232.html -E http://Gpro.ws http://WatchNtweet.Me (Watch and Chat/Tweet) SocialTV http://TwebEX.com (Twitter Based Online Web Conference Platform) http://DatR.Ws (Cloud Computing Media Sharing, Access and Publish) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090705/a513cc8a/attachment-0002.html From gcd at i.ph Sun Jul 5 16:21:03 2009 From: gcd at i.ph (Nandy Dagondon) Date: Mon, 6 Jul 2009 07:21:03 +0800 Subject: [Freeswitch-users] Compact, fanless appliance? In-Reply-To: References: <7d0bfd8c0907041941s75e541dex982fae2195858ea4@mail.gmail.com> Message-ID: <7d0bfd8c0907051621u1c3f553fh96a9df8557952e47@mail.gmail.com> ok. w/ my apologies. - nandy On Sun, Jul 5, 2009 at 10:49 AM, Ken Rice wrote: > No need to bump these things as this is a mailing list and it annoys > quite a few people when you do that > > > ------------------------------ > *From: *Nandy Dagondon > *Reply-To: * > *Date: *Sun, 5 Jul 2009 10:41:18 +0800 > *To: * > *Subject: *Re: [Freeswitch-users] Compact, fanless appliance? > > just bumping this topic. > -nandy > > On Fri, May 8, 2009 at 12:44 AM, Fred-145 wrote: > > > > Antonio Gallo wrote: > > Alix cases are like 6/9 ? from their shop site. I think its easy to find > > someone who work with aluminium that can make for you custom boxes for > > like like 6/20 ? at pcs > > Unfortunately, none of the PCEngines cases ( > www.pcengines.ch/order1.php?c=2 ) > allow for a PCI slot, either on top of the mobo, or away from it :-/ > > I'll see if I can get those from Soekris ( > http://soekris.eu/shop/cases_en/) > allow this, and if I can get a good price for a case + PSU. > > Thank you. > -- > View this message in context: > http://www.nabble.com/Compact%2C-fanless-appliance--tp23193738p23430873.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 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090706/d9824e10/attachment-0002.html From dave at 3c.co.uk Sun Jul 5 16:22:49 2009 From: dave at 3c.co.uk (David Knell) Date: Sun, 05 Jul 2009 20:22:49 -0300 Subject: [Freeswitch-users] Contemplating a jump from Asterisk to Freeswitch In-Reply-To: <000e0cd4043ed3da65046dfcea6d@google.com> References: <000e0cd4043ed3da65046dfcea6d@google.com> Message-ID: <1246836169.31637.6.camel@dk-d820> Hi Geoff, > One of the benefits of our architecture is that our business logic is > completely abstracted from the asterisk system. We use a combination > of FastAGI and AMI to control channels on the asterisk server. We have > a Java based server which interfaces with the higher level call > routing engines. It looks to me like the Mod_event_socket would > probably satisfy my requirements for controlling the calls via an > external process, although it doesn't look as cut/dry as the FastAGI > model. I haven't seen anything which would let me know the equivalent > of the FastAGI 'script' being requested. Three possibilities spring to mind:- * have each distinct 'script' listen on a different socket; * set a variable in the dialplan to a script name or other identifier before making the outbound socket connection; * have your event socket handler work out what to do itself based on the dialled number, or whatever other criteria you'd use. > The other thing I haven't seen is how to dynamically create > conferences on the fly and redirect channels into them. We use > app_conference on asterisk to avoid the ztdummy issue. Once the higher > level intelligence engine determines two channels need to speak with > each other, they are both redirected via AMI Redirect into a dynamic > Conference created just for that particular call. Choose a (unique) conference ID, and execute conference on each of the channels. > Also - what is the status of call progress on FreeSwitch? Some things > that are important to me are answering machine detection as well as > detecting SIT intercept tones in the early media stream... any love > here? Not sure on these, but I'm *am* sure that someone else will be ;-) Cheers -- Dave -- David Knell, Director, 3C Limited T: +44 20 3298 2000 E: dave at 3c.co.uk W: http://www.3c.co.uk From codecomplete at free.fr Sun Jul 5 18:22:26 2009 From: codecomplete at free.fr (Fred-145) Date: Sun, 5 Jul 2009 18:22:26 -0700 (PDT) Subject: [Freeswitch-users] Cheapest, most compact FreeSwitch appliance? In-Reply-To: <9dc4a1670907051617k3c112eeal2356d3df55283f89@mail.gmail.com> References: <24337599.post@talk.nabble.com> <7C0A0FFD-EC03-46DE-A277-AE793017D617@howlertech.com> <24345928.post@talk.nabble.com> <9dc4a1670907051617k3c112eeal2356d3df55283f89@mail.gmail.com> Message-ID: <24348506.post@talk.nabble.com> EdPimentl wrote: > Back in April I posted these links on the list, in regards to a similar > question Thanks Ed, but the problem with all those, is: - they typically have so little RAM/Flash RAM that they can't run a regular Linux distro, which means that we're stuck with whatever software is available with the customized distro for the appliance - they don't have room for a PCI card, which means that we have to have an external VoIP gateway to connect the appliance to the POTS - they're as expensive or more expensive than a regular PC At this point, there doesn't seem to be any appliance that can run FreeSwitch and handle a POTS line in a compact, sub-$200 price-range. -- View this message in context: http://www.nabble.com/Cheapest%2C-most-compact-FreeSwitch-appliance--tp24337599p24348506.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From kristian.kielhofner at gmail.com Sun Jul 5 18:28:50 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Sun, 5 Jul 2009 21:28:50 -0400 Subject: [Freeswitch-users] Contemplating a jump from Asterisk to Freeswitch In-Reply-To: <000e0cd4043ed3da65046dfcea6d@google.com> References: <000e0cd4043ed3da65046dfcea6d@google.com> Message-ID: <2d9149cd0907051828g66360c5r52c63bca41aaff21@mail.gmail.com> On Sun, Jul 5, 2009 at 6:29 PM, wrote: > > Also - what is the status of call progress on FreeSwitch? Some things that > are important to me are answering machine detection as well as detecting SIT > intercept tones in the early media stream... any love here? > Not my specialty but I'll try... Answering machine detection can be done with mod_vmd: http://wiki.freeswitch.org/wiki/Mod_vmd Tone detection: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_tone_detect -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From msc at freeswitch.org Sun Jul 5 22:43:55 2009 From: msc at freeswitch.org (Michael S Collins) Date: Sun, 5 Jul 2009 22:43:55 -0700 Subject: [Freeswitch-users] Contemplating a jump from Asterisk to Freeswitch In-Reply-To: <000e0cd4043ed3da65046dfcea6d@google.com> References: <000e0cd4043ed3da65046dfcea6d@google.com> Message-ID: A few questions for you if I may: FreeSWITCH doesn't yet have a GUI -are you okay with XML config files? Do you have TDM circuits for your outbound traffic or are you using a SIP provider? BTW, mod_vmd is used to detect an answering machine beep, but it does not detect human vs. machine. For that you'll need mod_amd which isn't free but is available at a reasonable price. (email consulting at FreeSWITCH.org ) FYI, detecting SIT tones is always a challenge if you telco forces you to listen inband. You'll need a little processing power and the tone_detect app. I've done it on a PRI and cheap Tormenta 2 clone and it actually works pretty well. -MC Sent from my iPhone On Jul 5, 2009, at 3:29 PM, geoffreymina at gmail.com wrote: > Hello, > I have been reading through the on-line info as well as some reviews > of the FreeSwitch platform. I am fairly convinced at this point that > FreeSwitch is at least something I need to carefully look into. > > Our company utilizes asterisk to support our SaaS ACD/VPD/IVR > platform. We currently support many thousands of concurrent agents > (inbound and outbound). I have spent a lot of time trouble shooting > bugs and working through 'issues' with asterisk. While I have tamed > the beast, I am still not thrilled with the performance, nor am I > very excited about the direction the project appears to be heading. > It seems like every time a 'fix' is committed to SVN, it breaks > something else. It's kind of like the wild-wild-west over there... > and it certainly doesn't give me the warm/fuzzies when thinking > about the future of my company. > > One of the benefits of our architecture is that our business logic > is completely abstracted from the asterisk system. We use a > combination of FastAGI and AMI to control channels on the asterisk > server. We have a Java based server which interfaces with the higher > level call routing engines. It looks to me like the Mod_event_socket > would probably satisfy my requirements for controlling the calls via > an external process, although it doesn't look as cut/dry as the > FastAGI model. I haven't seen anything which would let me know the > equivalent of the FastAGI 'script' being requested. > > The other thing I haven't seen is how to dynamically create > conferences on the fly and redirect channels into them. We use > app_conference on asterisk to avoid the ztdummy issue. Once the > higher level intelligence engine determines two channels need to > speak with each other, they are both redirected via AMI Redirect > into a dynamic Conference created just for that particular call. > > Also - what is the status of call progress on FreeSwitch? Some > things that are important to me are answering machine detection as > well as detecting SIT intercept tones in the early media stream... > any love here? > > I have a ton more questions, but this seems like a good start. > > Thanks! > Geoff > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From talk2ram at gmail.com Mon Jul 6 02:35:13 2009 From: talk2ram at gmail.com (ram) Date: Mon, 6 Jul 2009 15:05:13 +0530 Subject: [Freeswitch-users] Real time Integration with Opensips Message-ID: Hi I am using Opensips as registrar and proxy * boxes as PSTN and VoIP Termination I would like to try replacing the * boxes with FS box in my lab so how can i make realtime Users integration with FS ( i use to do with Views in Mysql for Opensips and * boxes) Any URL and example available success people . just suggest me. Ram -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090706/0ade903b/attachment-0002.html From jaybinks at gmail.com Mon Jul 6 03:23:51 2009 From: jaybinks at gmail.com (Jay Binks) Date: Mon, 06 Jul 2009 20:23:51 +1000 Subject: [Freeswitch-users] Real time Integration with Opensips In-Reply-To: References: Message-ID: <1246875831.5330.48.camel@jay-desktop.home.gateway> sounds like the simplest way would be to use a web application ( PHP or something similar ) that handles the users Directory.. that way you can keep your DB exactly the same and just pull the required fields. Jay On Mon, 2009-07-06 at 15:05 +0530, ram wrote: > Hi > > I am using Opensips as registrar and proxy > * boxes as PSTN and VoIP Termination > > I would like to try replacing the * boxes with FS box in my lab > > so how can i make realtime Users integration with FS ( i use to do > with Views in Mysql for Opensips and * boxes) > > Any URL and example available > > success people . just suggest me. > > Ram > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090706/87f3b701/attachment-0002.html From talk2ram at gmail.com Mon Jul 6 04:03:46 2009 From: talk2ram at gmail.com (ram) Date: Mon, 6 Jul 2009 16:33:46 +0530 Subject: [Freeswitch-users] Real time Integration with Opensips In-Reply-To: <1246875831.5330.48.camel@jay-desktop.home.gateway> References: <1246875831.5330.48.camel@jay-desktop.home.gateway> Message-ID: On Mon, Jul 6, 2009 at 3:53 PM, Jay Binks wrote: > sounds like the simplest way would be to use a web application ( PHP or > something similar ) > that handles the users Directory.. that way you can keep your DB exactly > the same and just pull the required fields. > thanks for quick reply but if it grows 10K+ users maintaining folders will be tough. can some one point me mysql Internal and external schema for mysql iam trying to search not found Ram -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090706/3572d052/attachment-0002.html From jaybinks at gmail.com Mon Jul 6 04:28:51 2009 From: jaybinks at gmail.com (Jay Binks) Date: Mon, 06 Jul 2009 21:28:51 +1000 Subject: [Freeswitch-users] Real time Integration with Opensips In-Reply-To: References: <1246875831.5330.48.camel@jay-desktop.home.gateway> Message-ID: <1246879731.5330.51.camel@jay-desktop.home.gateway> hmmm ok... was concerned my terminology may confuse. in Freeswitch sip_users are stored in a "User Directory" ( nothing to do with the filesystem ) it is a directory of all users ... ( like yellow pages is a directory .. ) look here .. http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide Freeswitch can get this user directory ( list of users ) in many ways.. one way is using CURL, to ask a web server to generate an XML file . this XML file can be created with PHP from your existing DB Structure this is what I was suggesting.. another way is to look at using OBDC to connect to mysql, however Im not a fan of OBDC. Jay On Mon, 2009-07-06 at 16:33 +0530, ram wrote: > > > > On Mon, Jul 6, 2009 at 3:53 PM, Jay Binks wrote: > > sounds like the simplest way would be to use a web application > ( PHP or something similar ) > that handles the users Directory.. that way you can keep your > DB exactly the same and just pull the required fields. > > thanks for quick reply > > but if it grows 10K+ users maintaining folders will be tough. > > can some one point me mysql Internal and external schema for mysql > > iam trying to search not found > > Ram > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090706/af57855a/attachment-0002.html From geoffreymina at gmail.com Mon Jul 6 05:14:36 2009 From: geoffreymina at gmail.com (Geoffrey Mina) Date: Mon, 6 Jul 2009 08:14:36 -0400 Subject: [Freeswitch-users] Contemplating a jump from Asterisk to Freeswitch In-Reply-To: References: <000e0cd4043ed3da65046dfcea6d@google.com> Message-ID: <9f9c89f50907060514l417e1497x4f81a534c4306d13@mail.gmail.com> I love the fact that there is no GUI. I have never used any GUI for asterisk, so that is certainly not a problem. XML is fine with me. We are a pure VoIP environment. I have many wholesale SIP providers whom I interface with. AMD and SIT detection are very important to me. Because of that, I am exploring a relationship with Sangoma for their SIP based CPD product to satisfy those requirements. There are a couple things which I don't like... namely that it only runs on windows, but I may be able to ignore that for the time being. thanks. On Mon, Jul 6, 2009 at 1:43 AM, Michael S Collins wrote: > A few questions for you if I may: > FreeSWITCH doesn't yet have a GUI -are you okay with XML config files? > > Do you have TDM circuits for your outbound traffic or are you using a > SIP provider? > > BTW, mod_vmd is used to detect an answering machine beep, but it does > not detect human vs. machine. For that you'll need mod_amd which isn't > free but is available at a reasonable price. (email consulting at FreeSWITCH.org > ) > > FYI, detecting SIT tones is always a challenge if you telco forces you > to listen inband. You'll need a little processing power and the > tone_detect app. I've done it on a PRI and cheap Tormenta 2 clone and > it actually works pretty well. > > -MC > > Sent from my iPhone > > On Jul 5, 2009, at 3:29 PM, geoffreymina at gmail.com wrote: > >> Hello, >> I have been reading through the on-line info as well as some reviews >> of the FreeSwitch platform. I am fairly convinced at this point that >> FreeSwitch is at least something I need to carefully look into. >> >> Our company utilizes asterisk to support our SaaS ACD/VPD/IVR >> platform. We currently support many thousands of concurrent agents >> (inbound and outbound). I have spent a lot of time trouble shooting >> bugs and working through 'issues' with asterisk. While I have tamed >> the beast, I am still not thrilled with the performance, nor am I >> very excited about the direction the project appears to be heading. >> It seems like every time a 'fix' is committed to SVN, it breaks >> something else. It's kind of like the wild-wild-west over there... >> and it certainly doesn't give me the warm/fuzzies when thinking >> about the future of my company. >> >> One of the benefits of our architecture is that our business logic >> is completely abstracted from the asterisk system. We use a >> combination of FastAGI and AMI to control channels on the asterisk >> server. We have a Java based server which interfaces with the higher >> level call routing engines. It looks to me like the Mod_event_socket >> would probably satisfy my requirements for controlling the calls via >> an external process, although it doesn't look as cut/dry as the >> FastAGI model. I haven't seen anything which would let me know the >> equivalent of the FastAGI 'script' being requested. >> >> The other thing I haven't seen is how to dynamically create >> conferences on the fly and redirect channels into them. We use >> app_conference on asterisk to avoid the ztdummy issue. ?Once the >> higher level intelligence engine determines two channels need to >> speak with each other, they are both redirected via AMI Redirect >> into a dynamic Conference created just for that particular call. >> >> Also - what is the status of call progress on FreeSwitch? Some >> things that are important to me are answering machine detection as >> well as detecting SIT intercept tones in the early media stream... >> any love here? >> >> I have a ton more questions, but this seems like a good start. >> >> Thanks! >> Geoff >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From david.goodenough at linkchoose.co.uk Mon Jul 6 02:23:21 2009 From: david.goodenough at linkchoose.co.uk (David Goodenough) Date: Mon, 6 Jul 2009 10:23:21 +0100 Subject: [Freeswitch-users] Cheapest, most compact FreeSwitch appliance? In-Reply-To: <24348506.post@talk.nabble.com> References: <24337599.post@talk.nabble.com> <9dc4a1670907051617k3c112eeal2356d3df55283f89@mail.gmail.com> <24348506.post@talk.nabble.com> Message-ID: <200907061023.21388.david.goodenough@linkchoose.co.uk> On Monday 06 July 2009, Fred-145 wrote: > EdPimentl wrote: > > Back in April I posted these links on the list, in regards to a similar > > question > > Thanks Ed, but the problem with all those, is: > - they typically have so little RAM/Flash RAM that they can't run a regular > Linux distro, which means that we're stuck with whatever software is > available with the customized distro for the appliance > - they don't have room for a PCI card, which means that we have to have an > external VoIP gateway to connect the appliance to the POTS > - they're as expensive or more expensive than a regular PC > > At this point, there doesn't seem to be any appliance that can run > FreeSwitch and handle a POTS line in a compact, sub-$200 price-range. I though that I had read somewhere about someone using the Marvel ShevaPlug and a 2 line USB POTS adapter. That should be under $200. David From anthony.minessale at gmail.com Mon Jul 6 08:43:57 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 6 Jul 2009 10:43:57 -0500 Subject: [Freeswitch-users] Contemplating a jump from Asterisk to Freeswitch In-Reply-To: <000e0cd4043ed3da65046dfcea6d@google.com> References: <000e0cd4043ed3da65046dfcea6d@google.com> Message-ID: <191c3a030907060843k30bd1b79q5dcf3056a71b9890@mail.gmail.com> The best way to describe event socket to someone familiar with asterisk is that its a combination of AGI and AMI which can be used bidirectional. You can: connect one inbound socket from a client and control every call at once using events. connect one inbound socket then latch on to an existing single call and control it. connect one outbound socket to your application per call and control it. In all cases you have the option for full control which allows you to gain access to log, event, and FSAPI commands (the equiv of cli commands in asterisk) You can have your script listen on a dedicated port or use the ivrd example which is a daemon written in C that gets the desired script name from a channel variable and executes it on the remote end of the socket using STDIN/STDOUT as the socket. The other big difference besides that the single protocol does all these things is that we have a BSD licensed client library in our source tree called ESL. its in the libs/esl directory. This can be use to write clients in C or several other higher level languages using swig. fs_cli that is built with FS is written using ESL. Perl, Ruby, Python, Lua, PHP are all working and there is the beginning of a JAVA one which is stubbed out but just needs a little bit of work to finish it off and you could have that too. On Sun, Jul 5, 2009 at 5:29 PM, wrote: > Hello, > I have been reading through the on-line info as well as some reviews of the > FreeSwitch platform. I am fairly convinced at this point that FreeSwitch is > at least something I need to carefully look into. > > Our company utilizes asterisk to support our SaaS ACD/VPD/IVR platform. We > currently support many thousands of concurrent agents (inbound and > outbound). I have spent a lot of time trouble shooting bugs and working > through 'issues' with asterisk. While I have tamed the beast, I am still not > thrilled with the performance, nor am I very excited about the direction the > project appears to be heading. It seems like every time a 'fix' is committed > to SVN, it breaks something else. It's kind of like the wild-wild-west over > there... and it certainly doesn't give me the warm/fuzzies when thinking > about the future of my company. > > One of the benefits of our architecture is that our business logic is > completely abstracted from the asterisk system. We use a combination of > FastAGI and AMI to control channels on the asterisk server. We have a Java > based server which interfaces with the higher level call routing engines. It > looks to me like the Mod_event_socket would probably satisfy my requirements > for controlling the calls via an external process, although it doesn't look > as cut/dry as the FastAGI model. I haven't seen anything which would let me > know the equivalent of the FastAGI 'script' being requested. > > The other thing I haven't seen is how to dynamically create conferences on > the fly and redirect channels into them. We use app_conference on asterisk > to avoid the ztdummy issue. Once the higher level intelligence engine > determines two channels need to speak with each other, they are both > redirected via AMI Redirect into a dynamic Conference created just for that > particular call. > > Also - what is the status of call progress on FreeSwitch? Some things that > are important to me are answering machine detection as well as detecting SIT > intercept tones in the early media stream... any love here? > > I have a ton more questions, but this seems like a good start. > > Thanks! > Geoff > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference 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/20090706/c9aacd0c/attachment-0002.html From lordwizard007 at gmail.com Mon Jul 6 09:25:30 2009 From: lordwizard007 at gmail.com (lw) Date: Mon, 6 Jul 2009 12:25:30 -0400 Subject: [Freeswitch-users] Baby Update! In-Reply-To: <86a32abc0907031746x61996750k2a423d3ef0db70da@mail.gmail.com> References: <3EC5EF44-BF81-4C23-AD0B-B4AAB5F7DFCE@freeswitch.org> <1246629538.4185.9.camel@dk-d820> <41646A56-C830-489D-99D7-CE5C1218E194@freeswitch.org> <7b197bef0907031627o4e061c4by745a4492f2de0608@mail.gmail.com> <23FF9919-E96F-4067-832F-6E19A83A4F1F@freeswitch.org> <86a32abc0907031746x61996750k2a423d3ef0db70da@mail.gmail.com> Message-ID: Congrats! On Fri, Jul 3, 2009 at 8:46 PM, Diego Viola wrote: > Congrats! > > > On Fri, Jul 3, 2009 at 7:35 PM, Brian West wrote: > >> Remember send him a little something to help out with the last minute >> expenses! ;) >> >> Btw lanboy will be at ClueCon ;) As will lanwife! >> >> /b >> >> On Jul 3, 2009, at 6:27 PM, Giovanni Maruzzelli wrote: >> >> > Yeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeh! >> > >> > >> > On Fri, Jul 3, 2009 at 10:31 PM, Brian West >> > wrote: >> >> Kaiden Anthony Chandler arrived Friday July 3rd at 1411 EDT 7lbs 10oz >> >> YAY... Congrats mr Lanman! >> >> /b >> >> On Jul 3, 2009, at 8:58 AM, David Knell wrote: >> >> >> >> On Fri, 2009-07-03 at 17:43 +0600, Muhammad Shahzad wrote: >> >> >> >> Congratulations to Ray and Samantha. Lets see what new features and >> >> >> >> bug fixes we will get in their "new version"..! ;-) >> >> >> >> Bug fixes..?! I'd refer you to Philip Larkin (went to my school, a >> >> bit >> >> before my time, poet, deceased, recently voted "Britain's favourite >> >> poet") whose "This Be The Verse" suggests otherwise: >> >> http://www.artofeurope.com/larkin/lar2.htm >> >> >> >> [as a recent father myself, I'm trying to prove him wrong..] >> >> >> >> --Dave >> >> >> >> -- >> >> David Knell, Director, 3C Limited >> >> T: +44 20 3298 2000 >> >> E: dave at 3c.co.uk >> >> W: 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 >> >> >> >> >> >> _______________________________________________ >> >> Freeswitch-users mailing list >> >> Freeswitch-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> >> >> >> >> > >> > _______________________________________________ >> > Freeswitch-users mailing list >> > Freeswitch-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090706/30884091/attachment-0002.html From jgarland at gmail.com Mon Jul 6 12:45:39 2009 From: jgarland at gmail.com (Jason Garland) Date: Mon, 06 Jul 2009 15:45:39 -0400 Subject: [Freeswitch-users] Cluecon 2009 hotel deals get a free $50 prepaid mastercard, and $10 off each night with a coupon code Message-ID: <4A525463.6000109@gmail.com> Use the following link to get a free $50 prepaid mastercard via Expedia: Expedia July Sale: Book a qualified 3+ night hotel stay & get a $50 Prepaid MasterCard? card for gas! - Expires 7/31/09 Then enter coupon code: 09JUL10 I think this coupon code expires on July 10th. If anyone manages to find some better coupon codes please post them. This is the best I could find. Expedia seems to have the best rates, and they count towards the minimum hotel room requirements imposed by the hotel for the fine folks running Cluecon. My total price after all the taxes and discounts for 3 nights ended up at $511.20 And that doesn't include the $50 prepaid mastercard. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090706/40dc032d/attachment-0002.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image-3370729-10655845 Type: image/gif Size: 50 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090706/40dc032d/attachment-0002.gif From ronmccar at gmail.com Mon Jul 6 15:53:04 2009 From: ronmccar at gmail.com (Ron McCarthy) Date: Mon, 6 Jul 2009 15:53:04 -0700 Subject: [Freeswitch-users] Best OS for FreeSwitch? Message-ID: <3885f4fe0907061553s39eb723as4782bf87097a041@mail.gmail.com> Hey list, We are running FS on FreeBSD 7.2 right now and cannot get it to push over 48 CPS on a Dual core Xeon (2.4 ghz), we start running into issues, max concurrent calls is around 500 as well. Asterisk can do this so I would FS could out perform this! We run into major PDD issues more then anything, where FS takes 10+ seconds to respond to a invite, it's very weird and bad. I have seen some issues with FreeBSD and FS so id like to try a different OS and see what our results are. What does everyone recommend for just raw performance and speed? I use to always run Slackware until we moved everything over to FreeBSD, would that be a good choice again? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090706/da2c4b12/attachment-0002.html From brian at freeswitch.org Mon Jul 6 16:03:25 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 6 Jul 2009 18:03:25 -0500 Subject: [Freeswitch-users] Best OS for FreeSwitch? In-Reply-To: <3885f4fe0907061553s39eb723as4782bf87097a041@mail.gmail.com> References: <3885f4fe0907061553s39eb723as4782bf87097a041@mail.gmail.com> Message-ID: CentOS 5.3 /b On Jul 6, 2009, at 5:53 PM, Ron McCarthy wrote: > Hey list, > > We are running FS on FreeBSD 7.2 right now and cannot get it to push > over 48 CPS on a Dual core Xeon (2.4 ghz), we start running into > issues, max concurrent calls is around 500 as well. Asterisk can do > this so I would FS could out perform this! We run into major PDD > issues more then anything, where FS takes 10+ seconds to respond to > a invite, it's very weird and bad. > > I have seen some issues with FreeBSD and FS so id like to try a > different OS and see what our results are. What does everyone > recommend for just raw performance and speed? I use to always run > Slackware until we moved everything over to FreeBSD, would that be a > good choice again? > > Thanks From ronmccar at gmail.com Mon Jul 6 16:04:20 2009 From: ronmccar at gmail.com (Ron McCarthy) Date: Mon, 6 Jul 2009 16:04:20 -0700 Subject: [Freeswitch-users] Best OS for FreeSwitch? In-Reply-To: References: <3885f4fe0907061553s39eb723as4782bf87097a041@mail.gmail.com> Message-ID: <3885f4fe0907061604s154e71a9jb06ce5a4d970f8c@mail.gmail.com> 64bit is better as well correct? On Mon, Jul 6, 2009 at 4:03 PM, Brian West wrote: > CentOS 5.3 > > /b > > On Jul 6, 2009, at 5:53 PM, Ron McCarthy wrote: > > > Hey list, > > > > We are running FS on FreeBSD 7.2 right now and cannot get it to push > > over 48 CPS on a Dual core Xeon (2.4 ghz), we start running into > > issues, max concurrent calls is around 500 as well. Asterisk can do > > this so I would FS could out perform this! We run into major PDD > > issues more then anything, where FS takes 10+ seconds to respond to > > a invite, it's very weird and bad. > > > > I have seen some issues with FreeBSD and FS so id like to try a > > different OS and see what our results are. What does everyone > > recommend for just raw performance and speed? I use to always run > > Slackware until we moved everything over to FreeBSD, would that be a > > good choice again? > > > > 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/20090706/a219fd90/attachment-0002.html From brian at freeswitch.org Mon Jul 6 16:06:43 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 6 Jul 2009 18:06:43 -0500 Subject: [Freeswitch-users] Best OS for FreeSwitch? In-Reply-To: <3885f4fe0907061604s154e71a9jb06ce5a4d970f8c@mail.gmail.com> References: <3885f4fe0907061553s39eb723as4782bf87097a041@mail.gmail.com> <3885f4fe0907061604s154e71a9jb06ce5a4d970f8c@mail.gmail.com> Message-ID: Yes, 32bit can only serve as a boat anchor in my opinion.... also don't run a 32bit OS on a 64bit CPU, you might as well just paypal me half the money you spent on the CPU's in the first place and snagged a couple of P4's :P /b On Jul 6, 2009, at 6:04 PM, Ron McCarthy wrote: > 64bit is better as well correct? > > > On Mon, Jul 6, 2009 at 4:03 PM, Brian West > wrote: > CentOS 5.3 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090706/b3f940a7/attachment-0002.html From hyppolite72 at yahoo.com Mon Jul 6 16:20:05 2009 From: hyppolite72 at yahoo.com (Jean-Marc Hyppolite) Date: Mon, 6 Jul 2009 16:20:05 -0700 (PDT) Subject: [Freeswitch-users] Controlling MOH from a java application Message-ID: <219371.37062.qm@web35605.mail.mud.yahoo.com> Hello, ? First of all, I would like to thank Anthony, Brian and all the developers for this wonderful piece of software. Very good job. ? I would like to know how I can start and stop Music On Hold from a JAVA script (using mod_java) similar to the StartMusicOnHold and StopMusicHold functions found in AGI (Asterisk-Java). ? I am using FreeSWITCH as an IVR server. I would like to be able to put the caller on hold while doing some other stuff. ? Thanks in advance. ? Jean-Marc. __________________________________________________________________ Yahoo! Canada Toolbar: Search from anywhere on the web, and bookmark your favourite sites. Download it now http://ca.toolbar.yahoo.com. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090706/27778bcd/attachment-0002.html From max.bridgewater at gmail.com Mon Jul 6 16:24:06 2009 From: max.bridgewater at gmail.com (Max Bridgewater) Date: Mon, 6 Jul 2009 19:24:06 -0400 Subject: [Freeswitch-users] Start Freeswitch as Daemon on CentOS 5 Message-ID: Hi, I have a server that i log into using SSH. Then in my local SSH terminal, i start Freeswitch with: /usr/local/freeswitch/bin/freeswitch -nonat & Yet, when i close the terminal window, Freeswitch also dies. I was hoping that the ampersand would make it run as a dameon process that would live pass the lifetime of the terminal. Any trick? Thanks, Max. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090706/679d400f/attachment-0002.html From brian at freeswitch.org Mon Jul 6 16:27:22 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 6 Jul 2009 18:27:22 -0500 Subject: [Freeswitch-users] Controlling MOH from a java application In-Reply-To: <219371.37062.qm@web35605.mail.mud.yahoo.com> References: <219371.37062.qm@web35605.mail.mud.yahoo.com> Message-ID: uuid_hold uuid_hold off These two api's will do it. /b On Jul 6, 2009, at 6:20 PM, Jean-Marc Hyppolite wrote: > Hello, > > First of all, I would like to thank Anthony, Brian and all the > developers for this wonderful piece of software. Very good job. > > I would like to know how I can start and stop Music On Hold from a > JAVA script (using mod_java) similar to the StartMusicOnHold and > StopMusicHold functions found in AGI (Asterisk-Java). > > I am using FreeSWITCH as an IVR server. I would like to be able to > put the caller on hold while doing some other stuff. > > Thanks in advance. > > Jean-Marc. > > > Yahoo! Canada Toolbar : Search from anywhere on the web and > bookmark your favourite sites. Download it now! > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090706/3fc83eb3/attachment-0002.html From brian at freeswitch.org Mon Jul 6 16:27:39 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 6 Jul 2009 18:27:39 -0500 Subject: [Freeswitch-users] Start Freeswitch as Daemon on CentOS 5 In-Reply-To: References: Message-ID: add a -nc in there too. > /usr/local/freeswitch/bin/freeswitch -nonat -nc & /b On Jul 6, 2009, at 6:24 PM, Max Bridgewater wrote: > Hi, > > I have a server that i log into using SSH. Then in my local SSH > terminal, i start Freeswitch with: > > /usr/local/freeswitch/bin/freeswitch -nonat & > > > Yet, when i close the terminal window, Freeswitch also dies. I was > hoping that the ampersand would make it run as a dameon process that > would live pass the lifetime of the terminal. > > Any trick? > > Thanks, > > Max. From sicfslist at gmail.com Mon Jul 6 16:30:03 2009 From: sicfslist at gmail.com (Shelby Ramsey) Date: Mon, 6 Jul 2009 18:30:03 -0500 Subject: [Freeswitch-users] Start Freeswitch as Daemon on CentOS 5 In-Reply-To: References: Message-ID: <35b355e90907061630l4b4c738codd12a998bbb39333@mail.gmail.com> use the -nc flag ... that will do the trick -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090706/8622502f/attachment-0002.html From max.bridgewater at gmail.com Mon Jul 6 16:31:28 2009 From: max.bridgewater at gmail.com (Max Bridgewater) Date: Mon, 6 Jul 2009 19:31:28 -0400 Subject: [Freeswitch-users] Start Freeswitch as Daemon on CentOS 5 In-Reply-To: References: Message-ID: Cool! Thanks. On Mon, Jul 6, 2009 at 7:27 PM, Brian West wrote: > add a -nc in there too. > > > /usr/local/freeswitch/bin/freeswitch -nonat -nc & > > > /b > > On Jul 6, 2009, at 6:24 PM, Max Bridgewater wrote: > > > Hi, > > > > I have a server that i log into using SSH. Then in my local SSH > > terminal, i start Freeswitch with: > > > > /usr/local/freeswitch/bin/freeswitch -nonat & > > > > > > Yet, when i close the terminal window, Freeswitch also dies. I was > > hoping that the ampersand would make it run as a dameon process that > > would live pass the lifetime of the terminal. > > > > Any trick? > > > > Thanks, > > > > Max. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090706/f2625f06/attachment-0002.html From jens at vegeby.nu Mon Jul 6 16:35:24 2009 From: jens at vegeby.nu (Jens Vegeby) Date: Tue, 7 Jul 2009 01:35:24 +0200 Subject: [Freeswitch-users] Start Freeswitch as Daemon on CentOS 5 In-Reply-To: References: Message-ID: <30ee97110907061635pdb8978bvb3f3effb114c7fc8@mail.gmail.com> Use the -nc (no console) command line parameter. There is a centos init script somewhere in the svn source tree. /Jens On 7/7/09, Max Bridgewater wrote: > Hi, > > I have a server that i log into using SSH. Then in my local SSH terminal, i > start Freeswitch with: > > /usr/local/freeswitch/bin/freeswitch -nonat & > > > Yet, when i close the terminal window, Freeswitch also dies. I was hoping > that the ampersand would make it run as a dameon process that would live > pass the lifetime of the terminal. > > Any trick? > > Thanks, > > Max. > -- Sent from my mobile device Mvh/Regards Jens From brian at freeswitch.org Mon Jul 6 16:38:15 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 6 Jul 2009 18:38:15 -0500 Subject: [Freeswitch-users] Start Freeswitch as Daemon on CentOS 5 In-Reply-To: <30ee97110907061635pdb8978bvb3f3effb114c7fc8@mail.gmail.com> References: <30ee97110907061635pdb8978bvb3f3effb114c7fc8@mail.gmail.com> Message-ID: <54E2126D-86BD-49D6-8C18-D34F8F8D60E1@freeswitch.org> build/freeswitch.init.redhat /b On Jul 6, 2009, at 6:35 PM, Jens Vegeby wrote: > Use the -nc (no console) command line parameter. > > There is a centos init script somewhere in the svn source tree. > > /Jens > > On 7/7/09, Max Bridgewater wrote: >> Hi, >> >> I have a server that i log into using SSH. Then in my local SSH >> terminal, i >> start Freeswitch with: >> >> /usr/local/freeswitch/bin/freeswitch -nonat & >> >> >> Yet, when i close the terminal window, Freeswitch also dies. I was >> hoping >> that the ampersand would make it run as a dameon process that would >> live >> pass the lifetime of the terminal. >> >> Any trick? >> >> Thanks, >> >> Max. >> > > -- > Sent from my mobile device > > Mvh/Regards Jens > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brad.tuan at gmail.com Mon Jul 6 19:01:49 2009 From: brad.tuan at gmail.com (Brad Tuan) Date: Tue, 7 Jul 2009 10:01:49 +0800 Subject: [Freeswitch-users] How to modify my INVITE msg?? Message-ID: For example, send a "INVITE 1001123 at xxx.xxx.xxx.xxx" to my FS user 1001 How to do this?? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090707/94277ed9/attachment-0002.html From brian at freeswitch.org Mon Jul 6 19:23:21 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 6 Jul 2009 21:23:21 -0500 Subject: [Freeswitch-users] How to modify my INVITE msg?? In-Reply-To: References: Message-ID: Try this: /b On Jul 6, 2009, at 9:01 PM, Brad Tuan wrote: > For example, send a "INVITE 1001123 at xxx.xxx.xxx.xxx" to my FS user > 1001 > > How to do this?? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090706/4ff987d1/attachment-0002.html From brad.tuan at gmail.com Mon Jul 6 19:48:04 2009 From: brad.tuan at gmail.com (Brad Tuan) Date: Tue, 7 Jul 2009 10:48:04 +0800 Subject: [Freeswitch-users] How to modify my INVITE msg?? Message-ID: Useless , the dialplan was changed like this: but when 1003 call 1001 ,the request is still "Request-Line: INVITE sip:1001 at 192.168.141.182 SIP/2.0" Dialplan: sofia/internal/1003 at 192.168.141.182 parsing [default->transfer_to_516003] continue=false Dialplan: sofia/internal/1003 at 192.168.141.182 Regex (FAIL) [transfer_to_516003] destination_number(1001) =~ /^516001$/ break=on-false Dialplan: sofia/internal/1003 at 192.168.141.182 parsing [default->transfer_to_516003] continue=false Dialplan: sofia/internal/1003 at 192.168.141.182 Regex (FAIL) [transfer_to_516003] destination_number(1001) =~ /^516002$/ break=on-false Dialplan: sofia/internal/1003 at 192.168.141.182 parsing [default->skype_to_1001] continue=false Dialplan: sofia/internal/1003 at 192.168.141.182 Regex (PASS) [skype_to_1001] destination_number(1001) =~ /^1001$/ break=on-false Dialplan: sofia/internal/1003 at 192.168.141.182 Action bridge(sofia/profile/1001123${regex(${sofia_contact( 1001@${domain})}|^[^\@]+(.*)|%1)}) 2009-07-07 10:38:46 [DEBUG] switch_core_state_machine.c:114 switch_core_standard_on_routing() (sofia/internal/1003 at 192.168.141.182) State Change CS_ROUTING -> CS_EXECUTE 2009-07-07 10:38:46 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/internal/1003 at 192.168.141.182 [BREAK] 2009-07-07 10:38:46 [DEBUG] switch_core_state_machine.c:483 switch_core_session_run() (sofia/internal/1003 at 192.168.141.182) State ROUTING going to sleep 2009-07-07 10:38:46 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/internal/1003 at 192.168.141.182) Running State Change CS_EXECUTE 2009-07-07 10:38:46 [DEBUG] switch_core_state_machine.c:490 switch_core_session_run() (sofia/internal/1003 at 192.168.141.182) State EXECUTE 2009-07-07 10:38:46 [DEBUG] mod_sofia.c:173 sofia_on_execute() sofia/internal/1003 at 192.168.141.182 SOFIA EXECUTE 2009-07-07 10:38:46 [DEBUG] switch_core_state_machine.c:151 switch_core_standard_on_execute() sofia/internal/1003 at 192.168.141.182Standard EXECUTE EXECUTE sofia/internal/1003 at 192.168.141.182 bridge( sofia/profile/1001123 at 192.168.141.182:29084;rinstance=b4a8ae8884b9ed6b) 2009-07-07 10:38:46 [ERR] mod_sofia.c:2681 sofia_outgoing_channel() Invalid Profile FS return it is a Invalid Profile....Why?? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090707/8775fcaf/attachment-0002.html From brad.tuan at gmail.com Mon Jul 6 19:56:30 2009 From: brad.tuan at gmail.com (Brad Tuan) Date: Tue, 7 Jul 2009 10:56:30 +0800 Subject: [Freeswitch-users] How to modify my INVITE msg?? Message-ID: Useless , the dialplan was changed like this: but when 1003 call 1001 ,the request is still "Request-Line: INVITE sip:1001 at 192.168.141.182 SIP/2.0" Dialplan: sofia/internal/1003 at 192.168.141.182 parsing [default->transfer_to_516003] continue=false Dialplan: sofia/internal/1003 at 192.168.141.182 Regex (FAIL) [transfer_to_516003] destination_number(1001) =~ /^516001$/ break=on-false Dialplan: sofia/internal/1003 at 192.168.141.182 parsing [default->transfer_to_516003] continue=false Dialplan: sofia/internal/1003 at 192.168.141.182 Regex (FAIL) [transfer_to_516003] destination_number(1001) =~ /^516002$/ break=on-false Dialplan: sofia/internal/1003 at 192.168.141.182 parsing [default->skype_to_1001] continue=false Dialplan: sofia/internal/1003 at 192.168.141.182 Regex (PASS) [skype_to_1001] destination_number(1001) =~ /^1001$/ break=on-false Dialplan: sofia/internal/1003 at 192.168.141.182 Action bridge(sofia/profile/1001123${regex(${sofia_contact( 1001@${domain})}|^[^\@]+(.*)|%1)}) 2009-07-07 10:38:46 [DEBUG] switch_core_state_machine.c:114 switch_core_standard_on_routing() (sofia/internal/1003 at 192.168.141.182) State Change CS_ROUTING -> CS_EXECUTE 2009-07-07 10:38:46 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/internal/1003 at 192.168.141.182 [BREAK] 2009-07-07 10:38:46 [DEBUG] switch_core_state_machine.c:483 switch_core_session_run() (sofia/internal/1003 at 192.168.141.182) State ROUTING going to sleep 2009-07-07 10:38:46 [DEBUG] switch_core_state_machine.c:397 switch_core_session_run() (sofia/internal/1003 at 192.168.141.182) Running State Change CS_EXECUTE 2009-07-07 10:38:46 [DEBUG] switch_core_state_machine.c:490 switch_core_session_run() (sofia/internal/1003 at 192.168.141.182) State EXECUTE 2009-07-07 10:38:46 [DEBUG] mod_sofia.c:173 sofia_on_execute() sofia/internal/1003 at 192.168.141.182 SOFIA EXECUTE 2009-07-07 10:38:46 [DEBUG] switch_core_state_machine.c:151 switch_core_standard_on_execute() sofia/internal/1003 at 192.168.141.182Standard EXECUTE EXECUTE sofia/internal/1003 at 192.168.141.182 bridge( sofia/profile/1001123 at 192.168.141.182:29084;rinstance=b4a8ae8884b9ed6b) 2009-07-07 10:38:46 [ERR] mod_sofia.c:2681 sofia_outgoing_channel() Invalid Profile FS return it is a Invalid Profile....Why?? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090707/66412b35/attachment-0002.html From hyppolite72 at yahoo.com Mon Jul 6 20:21:31 2009 From: hyppolite72 at yahoo.com (Jean-Marc Hyppolite) Date: Mon, 6 Jul 2009 20:21:31 -0700 (PDT) Subject: [Freeswitch-users] Controlling MOH from a java application Message-ID: <443906.62890.qm@web35604.mail.mud.yahoo.com> Hello Brian, ? Thank you for your quick answer. I tried the two API functions but with no result. The caller is not able to hear any music. But, when I use two extensions (one calling the other), MOH does work. ? My code on the JAVA side (for test purposes) ? session.answer(); session.sleep(500); ? session.execute("eval", "uuid_hold " + session.get_uuid()); ? java_function(); // lasts 30 to 40 seconds ? session.execute("eval", "uuid_hold off " + session.get_uuid()); ? session.sleep(500); session.hangup(); ? Thanks for the help. ? Jean-Marc. ? ? ? --- On Mon, 7/6/09, Brian West wrote: From: Brian West Subject: Re: [Freeswitch-users] Controlling MOH from a java application To: freeswitch-users at lists.freeswitch.org Received: Monday, July 6, 2009, 7:27 PM uuid_hold uuid_hold off These two api's will do it. /b On Jul 6, 2009, at 6:20 PM, Jean-Marc Hyppolite wrote: Hello, ? First of all, I would like to thank Anthony, Brian and all the developers for this wonderful piece of software. Very good job. ? I would like to know how I can start and stop Music On Hold from a JAVA script (using mod_java) similar to the StartMusicOnHold and StopMusicHold functions found in AGI (Asterisk-Java). ? I am using FreeSWITCH as an IVR server. I would like to be able to put the caller on hold while doing some other stuff. ? Thanks in advance. ? Jean-Marc. Yahoo! Canada Toolbar : Search from anywhere on the web and bookmark your favourite sites. Download it now! _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -----Inline Attachment Follows----- _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org __________________________________________________________________ Yahoo! Canada Toolbar: Search from anywhere on the web, and bookmark your favourite sites. Download it now http://ca.toolbar.yahoo.com. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090706/fb0e8db9/attachment-0002.html From elihayun at gmail.com Mon Jul 6 21:53:37 2009 From: elihayun at gmail.com (Eli Hayun) Date: Tue, 07 Jul 2009 07:53:37 +0300 Subject: [Freeswitch-users] How to get the hook state? Message-ID: <4A52D4D1.1010805@gmail.com> Hi I am a newbie in FreeSwitch and my question is: When I am calling to an extension, how should I know in advance what is the hook status. I tried to find out a variable that can get me this information but with no success. any help? From brad.tuan at gmail.com Mon Jul 6 23:15:27 2009 From: brad.tuan at gmail.com (Brad Tuan) Date: Tue, 7 Jul 2009 14:15:27 +0800 Subject: [Freeswitch-users] How to modify the Subject and Body when sending voicemail?? Message-ID: As title, How to custom the Subject and Body and ... of the mail ?? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090707/0af57d8a/attachment-0002.html From jason at jasonjgw.net Mon Jul 6 23:42:15 2009 From: jason at jasonjgw.net (Jason White) Date: Tue, 7 Jul 2009 16:42:15 +1000 Subject: [Freeswitch-users] How to modify the Subject and Body when sending voicemail?? In-Reply-To: References: Message-ID: <20090707064215.GA21128@jdc.jasonjgw.net> Brad Tuan wrote: > As title, How to custom the Subject and Body and ... of the mail ?? Have a look at the notify-voicemail.tpl and voicemail.tpl files, and the template parameters in voicemail.conf.xml in the default FreeSWITCH configuration to see how it all works and to decide what to edit. From dujinfang at gmail.com Tue Jul 7 00:06:16 2009 From: dujinfang at gmail.com (seven) Date: Tue, 7 Jul 2009 15:06:16 +0800 Subject: [Freeswitch-users] dp tools record to 1 channel but uuid_record record to 2 channels which cannot been playback Message-ID: <0DA560FF-D5F1-4E86-B8CD-6DBE0B9C9B47@gmail.com> # play test.wav Input File : 'test.wav' Sample Size : 16-bit (2 bytes) Sample Encoding: signed (2's complement) Channels : 2 Sample Rate : 16000 1) is it the default behavior that uuid_record record with 2 channels 2) is it reasonable that FS can record to 2 channels but cannot playback? 3) do I need to set RECORD_STEREO=false before uuid_record? Thanks for help. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090707/6d1a41cd/attachment-0002.html From jason at jasonjgw.net Tue Jul 7 00:16:32 2009 From: jason at jasonjgw.net (Jason White) Date: Tue, 7 Jul 2009 17:16:32 +1000 Subject: [Freeswitch-users] dp tools record to 1 channel but uuid_record record to 2 channels which cannot been playback In-Reply-To: <0DA560FF-D5F1-4E86-B8CD-6DBE0B9C9B47@gmail.com> References: <0DA560FF-D5F1-4E86-B8CD-6DBE0B9C9B47@gmail.com> Message-ID: <20090707071632.GA22797@jdc.jasonjgw.net> seven wrote: > 1) is it the default behavior that uuid_record record with 2 channels Yes. > 2) is it reasonable that FS can record to 2 channels but cannot > playback? Could you explain what happens when you play back the files? > 3) do I need to set RECORD_STEREO=false before uuid_record? That depends on whether you want two-channel output files or not. From dujinfang at gmail.com Tue Jul 7 00:30:27 2009 From: dujinfang at gmail.com (seven) Date: Tue, 7 Jul 2009 15:30:27 +0800 Subject: [Freeswitch-users] dp tools record to 1 channel but uuid_record record to 2 channels which cannot been playback In-Reply-To: <20090707071632.GA22797@jdc.jasonjgw.net> References: <0DA560FF-D5F1-4E86-B8CD-6DBE0B9C9B47@gmail.com> <20090707071632.GA22797@jdc.jasonjgw.net> Message-ID: <05D41406-17C3-43DF-BA94-3753B191B7BC@gmail.com> > >> 2) is it reasonable that FS can record to 2 channels but cannot >> playback? > > Could you explain what happens when you play back the files? yes, it's here: http://pastebin.freeswitch.org/9641 > >> 3) do I need to set RECORD_STEREO=false before uuid_record? > > That depends on whether you want two-channel output files or not. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.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 Tue Jul 7 00:56:33 2009 From: jason at jasonjgw.net (Jason White) Date: Tue, 7 Jul 2009 17:56:33 +1000 Subject: [Freeswitch-users] dp tools record to 1 channel but uuid_record record to 2 channels which cannot been playback In-Reply-To: <05D41406-17C3-43DF-BA94-3753B191B7BC@gmail.com> References: <0DA560FF-D5F1-4E86-B8CD-6DBE0B9C9B47@gmail.com> <20090707071632.GA22797@jdc.jasonjgw.net> <05D41406-17C3-43DF-BA94-3753B191B7BC@gmail.com> Message-ID: <20090707075633.GA25395@jdc.jasonjgw.net> seven wrote: > yes, it's here: http://pastebin.freeswitch.org/9641 Judging by the error message, it's a known limitation. You are welcome to work on a fix, or pay the develoeprs to fix it, or offer a bounty that might encourage someone to work on it, or wait until it gets fixed. Meanwhile, convert the file to mono and try again. Sox should be able to do this, for example. From dujinfang at gmail.com Tue Jul 7 01:11:00 2009 From: dujinfang at gmail.com (seven) Date: Tue, 7 Jul 2009 16:11:00 +0800 Subject: [Freeswitch-users] dp tools record to 1 channel but uuid_record record to 2 channels which cannot been playback In-Reply-To: <20090707075633.GA25395@jdc.jasonjgw.net> References: <0DA560FF-D5F1-4E86-B8CD-6DBE0B9C9B47@gmail.com> <20090707071632.GA22797@jdc.jasonjgw.net> <05D41406-17C3-43DF-BA94-3753B191B7BC@gmail.com> <20090707075633.GA25395@jdc.jasonjgw.net> Message-ID: <2E6E3DB3-471A-4D87-8BD6-7FA61B3ED9F9@gmail.com> On Jul 7, 2009, at 3:56 PM, Jason White wrote: > seven wrote: > >> yes, it's here: http://pastebin.freeswitch.org/9641 > > Judging by the error message, it's a known limitation. You are > welcome to work > on a fix, or pay the develoeprs to fix it, or offer a bounty that > might > encourage someone to work on it, or wait until it gets fixed. > > Meanwhile, convert the file to mono and try again. > > Sox should be able to do this, for example. > > Thanks, I'm using sox. > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mariusz_kolo at wp.pl Tue Jul 7 03:35:02 2009 From: mariusz_kolo at wp.pl (=?ISO-8859-2?Q?Mariusz_Ko=B3odziejczyk?=) Date: Tue, 07 Jul 2009 12:35:02 +0200 Subject: [Freeswitch-users] Record_session cutting wav files Message-ID: <4A5324D6.3070600@wp.pl> Hello I saw a strange behavior when i'm using record_session for outbound call. Recorded file is 24:20 time length, but in logs should have about 25:18. Here ma log: start recording about 2009-07-07 10:56:15 2009-07-07 10:56:15.108447 [DEBUG] mod_dptools.c:748 sofia/internal/1062 SET [czas]=[2009-07-07-10-56-15] - variable $czas = "2009-07-07-10-56-15" i use it in filename below EXECUTE sofia/internal/1062 record_session(/records/2009-07-07/069630d2-6ad4-11de-aa6e-e31e1f9431c0.10XX.07135XXXXX.2009-07-07-10-56-15.out.ISDN.wav) ..... stop recording 2009-07-07 11:21:33.291177 [DEBUG] switch_ivr_async.c:444 Stop recording file /records/2009-07-07/069630d2-6ad4-11de-aa6e-e31e1f9431c0.10XX.07135XXXXX.2009-07-07-10-56-15.out.ISDN.wav file should have about: 25:18 time length when i listen a file it's really cut My piece of dialplan: freeswitch version: FreeSWITCH Version 1.0.trunk (14013) linux: Linux pbx2 2.6.26-1-686 #1 SMP Fri Mar 13 18:08:45 UTC 2009 i686 GNU/Linux Thanks From jalsot at gmail.com Tue Jul 7 03:48:20 2009 From: jalsot at gmail.com (Tamas) Date: Tue, 07 Jul 2009 12:48:20 +0200 Subject: [Freeswitch-users] Record_session cutting wav files In-Reply-To: <4A5324D6.3070600@wp.pl> References: <4A5324D6.3070600@wp.pl> Message-ID: <4A5327F4.1040206@gmail.com> Hello, please try out FS >= r14143 as there were some fixes around call recording and media bugs. Please let us know the results. Regards, Tamas Mariusz Ko?odziejczyk ?rta: > Hello > > I saw a strange behavior when i'm using record_session for outbound > call. Recorded file is 24:20 time length, but in logs should have about > 25:18. > > Here ma log: > > start recording about 2009-07-07 10:56:15 > 2009-07-07 10:56:15.108447 [DEBUG] mod_dptools.c:748 sofia/internal/1062 > SET [czas]=[2009-07-07-10-56-15] - variable $czas = > "2009-07-07-10-56-15" i use it in filename below > EXECUTE sofia/internal/1062 > record_session(/records/2009-07-07/069630d2-6ad4-11de-aa6e-e31e1f9431c0.10XX.07135XXXXX.2009-07-07-10-56-15.out.ISDN.wav) > ..... > stop recording > 2009-07-07 11:21:33.291177 [DEBUG] switch_ivr_async.c:444 Stop recording > file > /records/2009-07-07/069630d2-6ad4-11de-aa6e-e31e1f9431c0.10XX.07135XXXXX.2009-07-07-10-56-15.out.ISDN.wav > > file should have about: 25:18 time length > when i listen a file it's really cut > > My piece of dialplan: > > > > > > > > > > > > > > > > > > > data="/records/${dir}/${uuid}.${caller_id_number}.$1.${czas}.out.ISDN.wav"/> > > > data="{origination_caller_id_number=${cti_gateway_number},effective_caller_id_number=${cti_gateway_number}}openzap/1/A/$1"/> > > > freeswitch version: FreeSWITCH Version 1.0.trunk (14013) > linux: Linux pbx2 2.6.26-1-686 #1 SMP Fri Mar 13 18:08:45 UTC 2009 i686 > GNU/Linux > > 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 > From jgarland at gmail.com Tue Jul 7 04:44:35 2009 From: jgarland at gmail.com (Jason Garland) Date: Tue, 7 Jul 2009 07:44:35 -0400 Subject: [Freeswitch-users] Cheapest, most compact FreeSwitch appliance? In-Reply-To: <200907061023.21388.david.goodenough@linkchoose.co.uk> References: <24337599.post@talk.nabble.com> <9dc4a1670907051617k3c112eeal2356d3df55283f89@mail.gmail.com> <24348506.post@talk.nabble.com> <200907061023.21388.david.goodenough@linkchoose.co.uk> Message-ID: <1CCA942A-B45E-4EC3-AF26-89FF832B9D8C@gmail.com> I have FreeSwitch running on a $50 Linksys NSLU2 Sent from my iPhone On Jul 6, 2009, at 5:23 AM, David Goodenough wrote: > On Monday 06 July 2009, Fred-145 wrote: >> EdPimentl wrote: >>> Back in April I posted these links on the list, in regards to a >>> similar >>> question >> >> Thanks Ed, but the problem with all those, is: >> - they typically have so little RAM/Flash RAM that they can't run a >> regular >> Linux distro, which means that we're stuck with whatever software is >> available with the customized distro for the appliance >> - they don't have room for a PCI card, which means that we have to >> have an >> external VoIP gateway to connect the appliance to the POTS >> - they're as expensive or more expensive than a regular PC >> >> At this point, there doesn't seem to be any appliance that can run >> FreeSwitch and handle a POTS line in a compact, sub-$200 price- >> range. > > I though that I had read somewhere about someone using the Marvel > ShevaPlug and a 2 line USB POTS adapter. That should be under $200. > > David > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.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 Jul 7 06:18:58 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 7 Jul 2009 08:18:58 -0500 Subject: [Freeswitch-users] How to modify my INVITE msg?? In-Reply-To: References: Message-ID: <4D91CAF3-8BA4-4EBB-96EC-112D612B786B@freeswitch.org> Well put the right profile name in there... instead of just "profile" /b On Jul 6, 2009, at 9:48 PM, Brad Tuan wrote: > 2009-07-07 10:38:46 [ERR] mod_sofia.c:2681 sofia_outgoing_channel() > Invalid Profile > > FS return it is a Invalid Profile....Why?? > __________________________________ From brian at freeswitch.org Tue Jul 7 06:23:50 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 7 Jul 2009 08:23:50 -0500 Subject: [Freeswitch-users] dp tools record to 1 channel but uuid_record record to 2 channels which cannot been playback In-Reply-To: <20090707075633.GA25395@jdc.jasonjgw.net> References: <0DA560FF-D5F1-4E86-B8CD-6DBE0B9C9B47@gmail.com> <20090707071632.GA22797@jdc.jasonjgw.net> <05D41406-17C3-43DF-BA94-3753B191B7BC@gmail.com> <20090707075633.GA25395@jdc.jasonjgw.net> Message-ID: <2D58177C-7A5B-49BA-A9F1-422F1E120BB1@freeswitch.org> OK let me comment ion this. Your voip connection is a single channel mono. The recording is two channel stereo.. the caller is in the left and the callee is in the right. This is a very helpful tool for call centers when your agent gets into a fight with the caller. FreeSWITCH can not shove a stereo signal down a mono line. Its rather obvious that it has to mux it into one channel... which it does... the other alternative is to just hang up the call and say WOOPS can't do it. /b On Jul 7, 2009, at 2:56 AM, Jason White wrote: > seven wrote: > >> yes, it's here: http://pastebin.freeswitch.org/9641 > > Judging by the error message, it's a known limitation. You are > welcome to work > on a fix, or pay the develoeprs to fix it, or offer a bounty that > might > encourage someone to work on it, or wait until it gets fixed. > > Meanwhile, convert the file to mono and try again. > > Sox should be able to do this, for example. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090707/e81454d1/attachment-0002.html From brian at freeswitch.org Tue Jul 7 06:41:35 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 7 Jul 2009 08:41:35 -0500 Subject: [Freeswitch-users] How to get the hook state? In-Reply-To: <4A52D4D1.1010805@gmail.com> References: <4A52D4D1.1010805@gmail.com> Message-ID: What are you trying to accomplish? /b On Jul 6, 2009, at 11:53 PM, Eli Hayun wrote: > Hi > I am a newbie in FreeSwitch and my question is: > When I am calling to an extension, how should I know in advance what > is > the hook status. I tried to find out a variable that can get me this > information but with no success. > any help? From lubimov at neolant.ru Tue Jul 7 06:53:02 2009 From: lubimov at neolant.ru (Alexey Lubimov) Date: Tue, 07 Jul 2009 17:53:02 +0400 Subject: [Freeswitch-users] freeswitch & sipnet.ru & caller id Message-ID: <4A53533E.9050305@neolant.ru> Good day. I have external sip gateway on sipnet.ru Logs from sipnet.ru contain my ip address (193.112.5.111) instead actual number (111 at 193.112.5.111) in field "caller id". Is it possible to set caller id in actual number instead ip address? 07/07/09 14:08 74959345610 193.112.5.111 Russia Moscow 00381982 0:14 0.01750 0.00408 From brian at freeswitch.org Tue Jul 7 07:01:55 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 7 Jul 2009 09:01:55 -0500 Subject: [Freeswitch-users] freeswitch & sipnet.ru & caller id In-Reply-To: <4A53533E.9050305@neolant.ru> References: <4A53533E.9050305@neolant.ru> Message-ID: I would need to see the sip packets to know why its doing that. /b On Jul 7, 2009, at 8:53 AM, Alexey Lubimov wrote: > Good day. > > I have external sip gateway on sipnet.ru > > Logs from sipnet.ru contain my ip address (193.112.5.111) instead > actual number (111 at 193.112.5.111) in field "caller id". > > Is it possible to set caller id in actual number instead ip address? > > > > 07/07/09 14:08 74959345610 193.112.5.111 Russia Moscow 00381982 0:14 > 0.01750 0.00408 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090707/5528f9c5/attachment-0002.html From mcampbellsmith at gmail.com Tue Jul 7 07:11:38 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Tue, 7 Jul 2009 22:11:38 +0800 Subject: [Freeswitch-users] 2 voicemail questions Message-ID: <33c87fa30907070711gca33ba5vd251d6ae1e89b5a2@mail.gmail.com> Hi! I have 2 questions regarding voicemail ... 1. Can I email the voicemail message to multiple email addresses? If so, what format is this in? 2. How can I make Freeswitch dial a number AFTER a voicemail is left? Thanks! From lfurrea at gmail.com Tue Jul 7 08:25:38 2009 From: lfurrea at gmail.com (Luis F Urrea) Date: Tue, 7 Jul 2009 09:25:38 -0600 Subject: [Freeswitch-users] mod_fifo: Selection of consumers in a certain order. Message-ID: Hi all, We are in the need of a certain application using mod fifo. Basically we are doing the following as described in the wiki: {call_timeout=30,fifo_member_wait=nowait}user/1009@$${domain} {call_timeout=30,fifo_member_wait=nowait}user/1008@$${domain} Things work fine, but we have noticed that the consumers are selected in kind of a load balancing basis, so that if member 1009 answered the last call then the next call goes to 1008 even if 1009 is available. We would like to know if there is a way to select the next consumer available in a different fashion such as in strict order setting a preference for each member. Say member 1 preference=1 member 2 preference=2 so that only if member 1 is on the phone the call rolls to member 2. Hope it makes sense. All input on how to achieve this is appreciated. Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090707/0d3f2c3e/attachment-0002.html From brian at freeswitch.org Tue Jul 7 08:28:35 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 7 Jul 2009 10:28:35 -0500 Subject: [Freeswitch-users] Fwd: [UniMRCP] Flite Plugin Available References: <892298.53232.qm@web111312.mail.gq1.yahoo.com> Message-ID: <1442D19E-110B-4757-B061-898C960272C2@freeswitch.org> Arsen just keeps em coming... Thank you! /b Begin forwarded message: > From: Arsen Chaloyan > Date: July 7, 2009 10:20:13 AM CDT > To: Brian West > Subject: Fw: [UniMRCP] Flite Plugin Available > > > > ----- Forwarded Message ---- > From: Arsen Chaloyan > To: UniMRCP > Cc: unimrcp-announcements at googlegroups.com > Sent: Tuesday, July 7, 2009 8:16:16 PM > Subject: [UniMRCP] Flite Plugin Available > > I would like to announce the availability of Flite TTS plugin for > UniMRCP server. > Special thanks goes to Garmt, who initially contributed and helped > develop the plugin. > > Currently supported TTS features are as follows: > > English voices: > awb > kal > rms > slt > Methods: > SPEAK > STOP > PAUSE > RESUME > BARGE-IN-OCCURRED > Events: > SPEAK-COMPLETE > Synthesizer Speech Data: > text/plain > > For the instructions on how to build and configure Flite with > UniMRCP refer to > http://code.google.com/p/unimrcp/wiki/FlitePlugin > > Please note, everything is working now, nevertheless this is basic > availability only. > I have mostly tested the integrated solution in the following setup > SIPPhone -> FreeSWITCH/UniMRCPClient -> UniMRCPServer/Flite > > > Feedback is welcome. > Thanks, > > -- > Arsen Chaloyan > The author of UniMRCP > http://www.unimrcp.org > > --~--~---------~--~----~------------~-------~--~----~ > You received this message because you are subscribed to the Google > Groups "UniMRCP" group. > To post to this group, send email to unimrcp at googlegroups.com > To unsubscribe from this group, send email to unimrcp+unsubscribe at googlegroups.com > For more options, visit this group at http://groups.google.com/group/unimrcp?hl=en > -~----------~----~----~----~------~----~------~--~--- > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090707/f00595ab/attachment-0002.html From brian at freeswitch.org Tue Jul 7 08:30:39 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 7 Jul 2009 10:30:39 -0500 Subject: [Freeswitch-users] 2 voicemail questions In-Reply-To: <33c87fa30907070711gca33ba5vd251d6ae1e89b5a2@mail.gmail.com> References: <33c87fa30907070711gca33ba5vd251d6ae1e89b5a2@mail.gmail.com> Message-ID: <839078F3-48A6-4EF5-BC9F-22475CA54644@freeswitch.org> On Jul 7, 2009, at 9:11 AM, Mark Campbell-Smith wrote: > Hi! > > I have 2 questions regarding voicemail ... > > 1. Can I email the voicemail message to multiple email addresses? If > so, what format is this in? > Try a comma sep. list. Not sure if it will work. > > 2. How can I make Freeswitch dial a number AFTER a voicemail is left? api Hangup hook? > > Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090707/87004e3a/attachment-0002.html From dujinfang at gmail.com Tue Jul 7 08:49:08 2009 From: dujinfang at gmail.com (Seven Du) Date: Tue, 7 Jul 2009 23:49:08 +0800 Subject: [Freeswitch-users] mod_fifo: Selection of consumers in a certain order. In-Reply-To: References: Message-ID: <696807A5-33BE-4E48-8AEF-584D0A4F5FF5@gmail.com> On Jul 7, 2009, at 11:25 PM, Luis F Urrea wrote: > Hi all, > > We are in the need of a certain application using mod fifo. > Basically we are doing the following as described in the wiki: > > > lag="5">{call_timeout=30,fifo_member_wait=nowait}user/1009@$$ > {domain} > lag="5">{call_timeout=30,fifo_member_wait=nowait}user/1008@$$ > {domain} > > > > > > > > > > > > > > > > > > > Things work fine, but we have noticed that the consumers are > selected in kind of a load balancing basis, so that if member 1009 > answered the last call then the next call goes to 1008 even if 1009 > is available. > > > We would like to know if there is a way to select the next consumer > available in a different fashion such as in strict order setting a > preference > for each member. No. Maybe you'd like to patch it or add a wishlist to jira or add a bounty. But, depending on how many agents you have, a '|' separated dialstring might do the trick: > lag="5">{leg_timeout=30,fifo_member_wait=nowait}user/1009|user/1008| > user/1007... > > > Say > member 1 preference=1 > member 2 preference=2 > > > so that only if member 1 is on the phone the call rolls to member 2. > > Hope it makes sense. > > All input on how to achieve this is appreciated. > > 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 mctch at yahoo.com Tue Jul 7 10:42:16 2009 From: mctch at yahoo.com (Mark Crane) Date: Tue, 7 Jul 2009 10:42:16 -0700 (PDT) Subject: [Freeswitch-users] 2 voicemail questions Message-ID: <348845.78768.qm@web56402.mail.re3.yahoo.com> 1. Can I email the voicemail message to multiple email addresses?? If so, what format is this in? ? ? ? I've been doing this successfully for quite some time using the mailer script that I wrote I just updated the wiki so that it would show the version I have been using that allows you to send multiple emails: You can use a comma or semi-colon between emails and send as many as you want. example: or http://wiki.freeswitch.org/wiki/PHP_email#mailer_app.php The script can send to a mail server to send over plain smtp, smtp authentication or even smtp tls which works with gmail. Mark J Crane mctch at yahoo.com --- On Tue, 7/7/09, Mark Campbell-Smith wrote: From: Mark Campbell-Smith Subject: [Freeswitch-users] 2 voicemail questions To: freeswitch-users at lists.freeswitch.org Date: Tuesday, July 7, 2009, 8:11 AM Hi! I have 2 questions regarding voicemail ... 1. Can I email the voicemail message to multiple email addresses?? If so, what format is this in? ? ? ? 2. How can I make Freeswitch dial a number AFTER a voicemail is left? 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/20090707/399ec050/attachment-0002.html From max.bridgewater at gmail.com Tue Jul 7 10:47:58 2009 From: max.bridgewater at gmail.com (Max Bridgewater) Date: Tue, 7 Jul 2009 13:47:58 -0400 Subject: [Freeswitch-users] Failed DTMF Sanity check Message-ID: Hi Guys, I keep getting this message printed in red on my consolde; and whenevr i get it, DTMF will stop being trasmitting in that session. [ERR] switch_rtp.c:2013 Failed DTMF sanity check. What does that mean and how can i prevent this from occurring? I'm using the socket API to send DTMF signals to freeswitch. Thanks, Max. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090707/f9596ba1/attachment-0002.html From brian at freeswitch.org Tue Jul 7 10:55:28 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 7 Jul 2009 12:55:28 -0500 Subject: [Freeswitch-users] Failed DTMF Sanity check In-Reply-To: References: Message-ID: <7A3C7717-10DD-4436-8A21-403E34E6FB72@freeswitch.org> That usually means your device you're using is broken for sending rfc2833.... can you tell me what device are you using? /b On Jul 7, 2009, at 12:47 PM, Max Bridgewater wrote: > Hi Guys, > > I keep getting this message printed in red on my consolde; and > whenevr i get it, DTMF will stop being trasmitting in that session. > > [ERR] switch_rtp.c:2013 Failed DTMF sanity check. > > What does that mean and how can i prevent this from occurring? I'm > using the socket API to send DTMF signals to freeswitch. > > Thanks, > Max. From msc at freeswitch.org Tue Jul 7 10:59:00 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 7 Jul 2009 10:59:00 -0700 Subject: [Freeswitch-users] Follow FreeSWITCH_wire on Twitter! Message-ID: <87f2f3b90907071059j3a45a7d0x1d727e0f51c35b39@mail.gmail.com> Okay everyone, spread the word: we have a Twitter channel for everyone to follow. It's called "FreeSWITCH_wire" and it's were you'll see up-to-the minute updates on FreeSWITCH development, new releases, important news and the like. Everyone go follow FreeSWITCH_wire right now, and don't forget to tell everyone in the VoIP world that they need to follow it as well! -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090707/a8c3d823/attachment-0002.html From max.bridgewater at gmail.com Tue Jul 7 11:07:54 2009 From: max.bridgewater at gmail.com (Max Bridgewater) Date: Tue, 7 Jul 2009 14:07:54 -0400 Subject: [Freeswitch-users] Failed DTMF Sanity check In-Reply-To: <7A3C7717-10DD-4436-8A21-403E34E6FB72@freeswitch.org> References: <7A3C7717-10DD-4436-8A21-403E34E6FB72@freeswitch.org> Message-ID: Actually I'm using Voxeo to generate DTMFs. They have the following construct that allows me to play DTMF: I think it's not standard VXML. How can i track this easily or at least capture the RTP stream so i can send it to them? Max. On Tue, Jul 7, 2009 at 1:55 PM, Brian West wrote: > That usually means your device you're using is broken for sending > rfc2833.... can you tell me what device are you using? > > /b > > On Jul 7, 2009, at 12:47 PM, Max Bridgewater wrote: > > > Hi Guys, > > > > I keep getting this message printed in red on my consolde; and > > whenevr i get it, DTMF will stop being trasmitting in that session. > > > > [ERR] switch_rtp.c:2013 Failed DTMF sanity check. > > > > What does that mean and how can i prevent this from occurring? I'm > > using the socket API to send DTMF signals to freeswitch. > > > > Thanks, > > Max. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090707/133e5510/attachment-0002.html From brian at freeswitch.org Tue Jul 7 11:12:44 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 7 Jul 2009 13:12:44 -0500 Subject: [Freeswitch-users] Failed DTMF Sanity check In-Reply-To: References: <7A3C7717-10DD-4436-8A21-403E34E6FB72@freeswitch.org> Message-ID: Can you get me an RTP trace bet they might be doing it wrong... seems to be common. /b On Jul 7, 2009, at 1:07 PM, Max Bridgewater wrote: > Actually I'm using Voxeo to generate DTMFs. They have the following > construct that allows me to play DTMF: > > > > I think it's not standard VXML. How can i track this easily or at > least capture the RTP stream so i can send it to them? > > Max. From msc at freeswitch.org Tue Jul 7 11:15:03 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 7 Jul 2009 11:15:03 -0700 Subject: [Freeswitch-users] Failed DTMF Sanity check In-Reply-To: References: <7A3C7717-10DD-4436-8A21-403E34E6FB72@freeswitch.org> Message-ID: <87f2f3b90907071115n1664d204tcb50cfbe7e5a9195@mail.gmail.com> There are some troubleshooting tips here: http://wiki.freeswitch.org/wiki/Reporting_Bugs#Debugging_Specific_Technologies There are several ways of capturing packets on your system and the above link explains how to set them up. -MC On Tue, Jul 7, 2009 at 11:07 AM, Max Bridgewater wrote: > Actually I'm using Voxeo to generate DTMFs. They have the following > construct that allows me to play DTMF: > > > > I think it's not standard VXML. How can i track this easily or at least > capture the RTP stream so i can send it to them? > > Max. > > > > On Tue, Jul 7, 2009 at 1:55 PM, Brian West wrote: > >> That usually means your device you're using is broken for sending >> rfc2833.... can you tell me what device are you using? >> >> /b >> >> On Jul 7, 2009, at 12:47 PM, Max Bridgewater wrote: >> >> > Hi Guys, >> > >> > I keep getting this message printed in red on my consolde; and >> > whenevr i get it, DTMF will stop being trasmitting in that session. >> > >> > [ERR] switch_rtp.c:2013 Failed DTMF sanity check. >> > >> > What does that mean and how can i prevent this from occurring? I'm >> > using the socket API to send DTMF signals to freeswitch. >> > >> > Thanks, >> > Max. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090707/3dd29d8c/attachment-0002.html From max.bridgewater at gmail.com Tue Jul 7 11:19:31 2009 From: max.bridgewater at gmail.com (Max Bridgewater) Date: Tue, 7 Jul 2009 14:19:31 -0400 Subject: [Freeswitch-users] Failed DTMF Sanity check In-Reply-To: References: <7A3C7717-10DD-4436-8A21-403E34E6FB72@freeswitch.org> Message-ID: > Can you get me an RTP trace bet they might be doing it wrong... seems > to be common. > Hmm Sorry. Can i activate RTP traces in Freeswitch somehow or do i need to run Something like Wireshark? Pascal. > > /b > > On Jul 7, 2009, at 1:07 PM, Max Bridgewater wrote: > > > Actually I'm using Voxeo to generate DTMFs. They have the following > > construct that allows me to play DTMF: > > > > > > > > I think it's not standard VXML. How can i track this easily or at > > least capture the RTP stream so i can send it to them? > > > > Max. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090707/1c556506/attachment-0002.html From msc at freeswitch.org Tue Jul 7 11:40:54 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 7 Jul 2009 11:40:54 -0700 Subject: [Freeswitch-users] Failed DTMF Sanity check In-Reply-To: References: <7A3C7717-10DD-4436-8A21-403E34E6FB72@freeswitch.org> Message-ID: <87f2f3b90907071140l3cde3a79se5a0b4adc1eeab81@mail.gmail.com> You'll need Wireshark or similar. Some tips can be found here: http://wiki.freeswitch.org/wiki/Reporting_Bugs#Debugging_Specific_Technologies -MC On Tue, Jul 7, 2009 at 11:19 AM, Max Bridgewater wrote: > > Can you get me an RTP trace bet they might be doing it wrong... seems >> to be common. >> > > > Hmm Sorry. Can i activate RTP traces in Freeswitch somehow or do i need to > run Something like Wireshark? > > Pascal. > >> >> /b >> >> On Jul 7, 2009, at 1:07 PM, Max Bridgewater wrote: >> >> > Actually I'm using Voxeo to generate DTMFs. They have the following >> > construct that allows me to play DTMF: >> > >> > >> > >> > I think it's not standard VXML. How can i track this easily or at >> > least capture the RTP stream so i can send it to them? >> > >> > Max. >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090707/3b0306c4/attachment-0002.html From max.bridgewater at gmail.com Tue Jul 7 11:48:24 2009 From: max.bridgewater at gmail.com (Max Bridgewater) Date: Tue, 7 Jul 2009 14:48:24 -0400 Subject: [Freeswitch-users] Failed DTMF Sanity check In-Reply-To: <87f2f3b90907071115n1664d204tcb50cfbe7e5a9195@mail.gmail.com> References: <7A3C7717-10DD-4436-8A21-403E34E6FB72@freeswitch.org> <87f2f3b90907071115n1664d204tcb50cfbe7e5a9195@mail.gmail.com> Message-ID: Thanks! What is the best way to send you the 4M pcap file? On Tue, Jul 7, 2009 at 2:15 PM, Michael Collins wrote: > There are some troubleshooting tips here: > > http://wiki.freeswitch.org/wiki/Reporting_Bugs#Debugging_Specific_Technologies > > There are several ways of capturing packets on your system and the above > link explains how to set them up. > -MC > > > On Tue, Jul 7, 2009 at 11:07 AM, Max Bridgewater < > max.bridgewater at gmail.com> wrote: > >> Actually I'm using Voxeo to generate DTMFs. They have the following >> construct that allows me to play DTMF: >> >> >> >> I think it's not standard VXML. How can i track this easily or at least >> capture the RTP stream so i can send it to them? >> >> Max. >> >> >> >> On Tue, Jul 7, 2009 at 1:55 PM, Brian West wrote: >> >>> That usually means your device you're using is broken for sending >>> rfc2833.... can you tell me what device are you using? >>> >>> /b >>> >>> On Jul 7, 2009, at 12:47 PM, Max Bridgewater wrote: >>> >>> > Hi Guys, >>> > >>> > I keep getting this message printed in red on my consolde; and >>> > whenevr i get it, DTMF will stop being trasmitting in that session. >>> > >>> > [ERR] switch_rtp.c:2013 Failed DTMF sanity check. >>> > >>> > What does that mean and how can i prevent this from occurring? I'm >>> > using the socket API to send DTMF signals to freeswitch. >>> > >>> > Thanks, >>> > Max. >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090707/b757abdd/attachment-0002.html From brian at freeswitch.org Tue Jul 7 11:52:26 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 7 Jul 2009 13:52:26 -0500 Subject: [Freeswitch-users] Failed DTMF Sanity check In-Reply-To: References: <7A3C7717-10DD-4436-8A21-403E34E6FB72@freeswitch.org> <87f2f3b90907071115n1664d204tcb50cfbe7e5a9195@mail.gmail.com> Message-ID: <64908434-88C2-4B98-8C1C-15CD2585EB53@freeswitch.org> email it directly to me off list please. /b On Jul 7, 2009, at 1:48 PM, Max Bridgewater wrote: > Thanks! What is the best way to send you the 4M pcap file? From msc at freeswitch.org Tue Jul 7 11:53:43 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 7 Jul 2009 11:53:43 -0700 Subject: [Freeswitch-users] Failed DTMF Sanity check In-Reply-To: References: <7A3C7717-10DD-4436-8A21-403E34E6FB72@freeswitch.org> <87f2f3b90907071115n1664d204tcb50cfbe7e5a9195@mail.gmail.com> Message-ID: <87f2f3b90907071153u3212d5c0vcb49270d1031b0fe@mail.gmail.com> Put it out on a webserver where one of the devs can grab it with a browser. -MC On Tue, Jul 7, 2009 at 11:48 AM, Max Bridgewater wrote: > Thanks! What is the best way to send you the 4M pcap file? > > > On Tue, Jul 7, 2009 at 2:15 PM, Michael Collins wrote: > >> There are some troubleshooting tips here: >> >> http://wiki.freeswitch.org/wiki/Reporting_Bugs#Debugging_Specific_Technologies >> >> There are several ways of capturing packets on your system and the above >> link explains how to set them up. >> -MC >> >> >> On Tue, Jul 7, 2009 at 11:07 AM, Max Bridgewater < >> max.bridgewater at gmail.com> wrote: >> >>> Actually I'm using Voxeo to generate DTMFs. They have the following >>> construct that allows me to play DTMF: >>> >>> >>> >>> I think it's not standard VXML. How can i track this easily or at least >>> capture the RTP stream so i can send it to them? >>> >>> Max. >>> >>> >>> >>> On Tue, Jul 7, 2009 at 1:55 PM, Brian West wrote: >>> >>>> That usually means your device you're using is broken for sending >>>> rfc2833.... can you tell me what device are you using? >>>> >>>> /b >>>> >>>> On Jul 7, 2009, at 12:47 PM, Max Bridgewater wrote: >>>> >>>> > Hi Guys, >>>> > >>>> > I keep getting this message printed in red on my consolde; and >>>> > whenevr i get it, DTMF will stop being trasmitting in that session. >>>> > >>>> > [ERR] switch_rtp.c:2013 Failed DTMF sanity check. >>>> > >>>> > What does that mean and how can i prevent this from occurring? I'm >>>> > using the socket API to send DTMF signals to freeswitch. >>>> > >>>> > Thanks, >>>> > Max. >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090707/fcf82335/attachment-0002.html From jalsot at gmail.com Tue Jul 7 12:01:28 2009 From: jalsot at gmail.com (Tamas) Date: Tue, 07 Jul 2009 21:01:28 +0200 Subject: [Freeswitch-users] Failed DTMF Sanity check In-Reply-To: <87f2f3b90907071153u3212d5c0vcb49270d1031b0fe@mail.gmail.com> References: <7A3C7717-10DD-4436-8A21-403E34E6FB72@freeswitch.org> <87f2f3b90907071115n1664d204tcb50cfbe7e5a9195@mail.gmail.com> <87f2f3b90907071153u3212d5c0vcb49270d1031b0fe@mail.gmail.com> Message-ID: <4A539B88.6070407@gmail.com> http://filebin.ca/ (up to 50MB) Tamas Michael Collins ?rta: > Put it out on a webserver where one of the devs can grab it with a > browser. > -MC > > On Tue, Jul 7, 2009 at 11:48 AM, Max Bridgewater > > wrote: > > Thanks! What is the best way to send you the 4M pcap file? > > > On Tue, Jul 7, 2009 at 2:15 PM, Michael Collins > > wrote: > > There are some troubleshooting tips here: > > http://wiki.freeswitch.org/wiki/Reporting_Bugs#Debugging_Specific_Technologies > > There are several ways of capturing packets on your system and > the above link explains how to set them up. > -MC > > > On Tue, Jul 7, 2009 at 11:07 AM, Max Bridgewater > > > wrote: > > Actually I'm using Voxeo to generate DTMFs. They have the > following construct that allows me to play DTMF: > > > > I think it's not standard VXML. How can i track this > easily or at least capture the RTP stream so i can send it > to them? > > Max. > > > > On Tue, Jul 7, 2009 at 1:55 PM, Brian West > > wrote: > > That usually means your device you're using is broken > for sending > rfc2833.... can you tell me what device are you using? > > /b > > On Jul 7, 2009, at 12:47 PM, Max Bridgewater wrote: > > > Hi Guys, > > > > I keep getting this message printed in red on my > consolde; and > > whenevr i get it, DTMF will stop being trasmitting > in that session. > > > > [ERR] switch_rtp.c:2013 Failed DTMF sanity check. > > > > What does that mean and how can i prevent this from > occurring? I'm > > using the socket API to send DTMF signals to freeswitch. > > > > Thanks, > > Max. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From marketing at cluecon.com Tue Jul 7 12:14:04 2009 From: marketing at cluecon.com (Michael Collins) Date: Tue, 7 Jul 2009 12:14:04 -0700 Subject: [Freeswitch-users] ClueCon 2009 Updates - July 7, 2009 Message-ID: <87f2f3b90907071214w60f0d1b0x21d39b7e5ad661ac@mail.gmail.com> Hello folks! We have a few more updates. First of all, if you haven't already heard, we've extended the early bird sign up to go through July 21. That's only two weeks away, so if you haven't already registered then please call us at 877.742.CLUE and we'll get you set up. Secondly, there are some updates on the ClueCon blog: http://cluecon.com/blog/1 The breakfast and lunch menus have been posted. (Subject to change, of course. :) Also, we have a synopsis for Irv Shapiro's talk, entitled "Cloud Telephony" on the latest blog post. Can't wait to see you all in Chicago! -Michael Collins http://www.cluecon.com 877.742.CLUE -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090707/a8639017/attachment-0002.html From anthony.minessale at gmail.com Tue Jul 7 13:37:54 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 7 Jul 2009 15:37:54 -0500 Subject: [Freeswitch-users] Controlling MOH from a java application In-Reply-To: <443906.62890.qm@web35604.mail.mud.yahoo.com> References: <443906.62890.qm@web35604.mail.mud.yahoo.com> Message-ID: <191c3a030907071337i6eb2ce09i387d9912a05d36bd@mail.gmail.com> FSAPI commands are accessed via the API obj api = new API(); api.execute("uuid_hold", session.get_uuid()); ... api.execute("uuid_hold", "off " + session.get_uuid()); On Mon, Jul 6, 2009 at 10:21 PM, Jean-Marc Hyppolite wrote: > Hello Brian, > > Thank you for your quick answer. I tried the two API functions but with no > result. The caller is not able to hear any music. But, when I use two > extensions (one calling the other), MOH does work. > > My code on the JAVA side (for test purposes) > > session.answer(); > session.sleep(500); > > session.execute("eval", "uuid_hold " + session.get_uuid()); > > java_function(); // lasts 30 to 40 seconds > > session.execute("eval", "uuid_hold off " + session.get_uuid()); > > session.sleep(500); > session.hangup(); > > Thanks for the help. > > Jean-Marc. > > > > > --- On *Mon, 7/6/09, Brian West * wrote: > > > From: Brian West > Subject: Re: [Freeswitch-users] Controlling MOH from a java application > To: freeswitch-users at lists.freeswitch.org > Received: Monday, July 6, 2009, 7:27 PM > > > uuid_hold > uuid_hold off > > These two api's will do it. > > /b > > > > On Jul 6, 2009, at 6:20 PM, Jean-Marc Hyppolite wrote: > > Hello, > > First of all, I would like to thank Anthony, Brian and all the developers > for this wonderful piece of software. Very good job. > > I would like to know how I can start and stop Music On Hold from a JAVA > script (using mod_java) similar to the StartMusicOnHold and StopMusicHold > functions found in AGI (Asterisk-Java). > > I am using FreeSWITCH as an IVR server. I would like to be able to put the > caller on hold while doing some other stuff. > > Thanks in advance. > > Jean-Marc. > > > ------------------------------ > > *Yahoo! Canada Toolbar :* Search from anywhere on the web and bookmark > your favourite sites. Download it now! > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -----Inline Attachment Follows----- > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > ------------------------------ > *All new Yahoo! Mail - * Get > a sneak peak at messages with a handy reading pane. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference 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/20090707/db4c86d2/attachment-0002.html From kees at mroffice.org Tue Jul 7 14:26:38 2009 From: kees at mroffice.org (Kees Varekamp) Date: Wed, 8 Jul 2009 09:26:38 +1200 Subject: [Freeswitch-users] Leaking stream handle Message-ID: <98d38dcf0907071426m581d07bcr27702c5e5bd3b574@mail.gmail.com> I am testing Freeswitch as an alternative to Asterisk. So far, so good, except for the following: - I have a lua channel listening to: - session:streamFile('local_stream://moh') - I have a socket bridging this channel to a sip gateway: - SendMsg 1e0cc726-6b33-11de-bae1-5fd843059ad5 - call-command: execute - execute-app-name: bridge - execute-app-arg: sofia/gateway// - This all works well, but the console says: - [CRIT] mod_local_stream.c:234 Leaking stream handle! [switch_ivr_play_file() src/switch_ivr_play_say.c:1026] Is this something I should be worried about? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090708/6c8b1e9e/attachment-0002.html From vladislaus at gmail.com Tue Jul 7 14:29:01 2009 From: vladislaus at gmail.com (Andres Gomez) Date: Tue, 7 Jul 2009 16:29:01 -0500 Subject: [Freeswitch-users] Clustering Freeswitch Message-ID: <31da44070907071429g3d31687bj714b827b0ec69f02@mail.gmail.com> Hello to all Exist any solution to clustering. Any load balancing appliance o heartbeat test?. Regards Andres Gomez. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090707/1fd6c9b7/attachment-0002.html From sprice at gmail.com Tue Jul 7 14:39:06 2009 From: sprice at gmail.com (SP) Date: Tue, 7 Jul 2009 16:39:06 -0500 Subject: [Freeswitch-users] Clustering Freeswitch In-Reply-To: <31da44070907071429g3d31687bj714b827b0ec69f02@mail.gmail.com> References: <31da44070907071429g3d31687bj714b827b0ec69f02@mail.gmail.com> Message-ID: <7e2ac3270907071439o7bf90e45g6c6017b48f6dfa64@mail.gmail.com> openser/kamailio/ser/opensips you pick a name DNS SRV On Tue, Jul 7, 2009 at 16:29, Andres Gomez wrote: > Hello to all > > Exist any solution to clustering. Any load balancing appliance o heartbeat > test?. > > Regards > > Andres Gomez. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090707/13feee7b/attachment-0002.html From sicfslist at gmail.com Tue Jul 7 14:39:59 2009 From: sicfslist at gmail.com (Shelby Ramsey) Date: Tue, 7 Jul 2009 16:39:59 -0500 Subject: [Freeswitch-users] Clustering Freeswitch In-Reply-To: <31da44070907071429g3d31687bj714b827b0ec69f02@mail.gmail.com> References: <31da44070907071429g3d31687bj714b827b0ec69f02@mail.gmail.com> Message-ID: <35b355e90907071439r698907b8o3b2dae2ef1b22620@mail.gmail.com> Andres, OpenSIP's works very well as a load balancer. You could also use DNS SRV (if the clients support it), round robin DNS .... Really just depends on what you are trying to accomplish. SDR -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090707/289a80b5/attachment-0002.html From anthony.minessale at gmail.com Tue Jul 7 14:53:13 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 7 Jul 2009 16:53:13 -0500 Subject: [Freeswitch-users] Leaking stream handle In-Reply-To: <98d38dcf0907071426m581d07bcr27702c5e5bd3b574@mail.gmail.com> References: <98d38dcf0907071426m581d07bcr27702c5e5bd3b574@mail.gmail.com> Message-ID: <191c3a030907071453g6a3c5eb3ha4ed77d721403b1a@mail.gmail.com> you should stop playing the file first. or transfer the call to the bridge app instead of executing it direct. send this over event_socket (replacing uuid of course) api uuid_transfer 1e0cc726-6b33-11de-bae1-5fd843059ad5 bridge:sofia/gateway// inline On Tue, Jul 7, 2009 at 4:26 PM, Kees Varekamp wrote: > I am testing Freeswitch as an alternative to Asterisk. So far, so good, > except for the following: > > > - I have a lua channel listening to: > - session:streamFile('local_stream://moh') > - I have a socket bridging this channel to a sip gateway: > - SendMsg 1e0cc726-6b33-11de-bae1-5fd843059ad5 > - call-command: execute > - execute-app-name: bridge > - execute-app-arg: sofia/gateway// > - This all works well, but the console says: > - [CRIT] mod_local_stream.c:234 Leaking stream handle! > [switch_ivr_play_file() src/switch_ivr_play_say.c:1026] > > > Is this something I should be worried about? 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/20090707/e44e6c53/attachment-0002.html From raul at etellicom.com Tue Jul 7 18:14:40 2009 From: raul at etellicom.com (Raul Fragoso) Date: Tue, 07 Jul 2009 22:14:40 -0300 Subject: [Freeswitch-users] Cheapest, most compact FreeSwitch appliance? In-Reply-To: <24348506.post@talk.nabble.com> References: <24337599.post@talk.nabble.com> <7C0A0FFD-EC03-46DE-A277-AE793017D617@howlertech.com> <24345928.post@talk.nabble.com> <9dc4a1670907051617k3c112eeal2356d3df55283f89@mail.gmail.com> <24348506.post@talk.nabble.com> Message-ID: <1247015680.4515.15.camel@raul-laptop> I'm about to order one of these eBox systems: http://www.wdlsystems.com/modperl/view_services.cgi?r=detail&prod_num=1EBX33J&aisle_id=1073 They sell for under $200, have decent specs and some models come with a mini-PCI slot, which can be used to attach a POTS card. Regards, Raul On Sun, 2009-07-05 at 18:22 -0700, Fred-145 wrote: > > EdPimentl wrote: > > Back in April I posted these links on the list, in regards to a similar > > question > > Thanks Ed, but the problem with all those, is: > - they typically have so little RAM/Flash RAM that they can't run a regular > Linux distro, which means that we're stuck with whatever software is > available with the customized distro for the appliance > - they don't have room for a PCI card, which means that we have to have an > external VoIP gateway to connect the appliance to the POTS > - they're as expensive or more expensive than a regular PC > > At this point, there doesn't seem to be any appliance that can run > FreeSwitch and handle a POTS line in a compact, sub-$200 price-range. From brian at freeswitch.org Tue Jul 7 18:26:54 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 7 Jul 2009 20:26:54 -0500 Subject: [Freeswitch-users] Cheapest, most compact FreeSwitch appliance? In-Reply-To: <1247015680.4515.15.camel@raul-laptop> References: <24337599.post@talk.nabble.com> <7C0A0FFD-EC03-46DE-A277-AE793017D617@howlertech.com> <24345928.post@talk.nabble.com> <9dc4a1670907051617k3c112eeal2356d3df55283f89@mail.gmail.com> <24348506.post@talk.nabble.com> <1247015680.4515.15.camel@raul-laptop> Message-ID: <318EF937-FF6E-4D40-A432-1BFA797C2040@freeswitch.org> I'll take two please! ;) /b On Jul 7, 2009, at 8:14 PM, Raul Fragoso wrote: > I'm about to order one of these eBox systems: > http://www.wdlsystems.com/modperl/view_services.cgi?r=detail&prod_num=1EBX33J&aisle_id=1073 > > They sell for under $200, have decent specs and some models come > with a > mini-PCI slot, which can be used to attach a POTS card. > > Regards, > > Raul > > On Sun, 2009-07-05 at 18:22 -0700, Fred-145 wrote: >> >> EdPimentl wrote: >>> Back in April I posted these links on the list, in regards to a >>> similar >>> question >> >> Thanks Ed, but the problem with all those, is: >> - they typically have so little RAM/Flash RAM that they can't run a >> regular >> Linux distro, which means that we're stuck with whatever software is >> available with the customized distro for the appliance >> - they don't have room for a PCI card, which means that we have to >> have an >> external VoIP gateway to connect the appliance to the POTS >> - they're as expensive or more expensive than a regular PC >> >> At this point, there doesn't seem to be any appliance that can run >> FreeSwitch and handle a POTS line in a compact, sub-$200 price- >> range. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From intralanman at freeswitch.org Tue Jul 7 18:28:56 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Tue, 07 Jul 2009 21:28:56 -0400 Subject: [Freeswitch-users] Baby Update! In-Reply-To: <41646A56-C830-489D-99D7-CE5C1218E194@freeswitch.org> References: <3EC5EF44-BF81-4C23-AD0B-B4AAB5F7DFCE@freeswitch.org> <1246629538.4185.9.camel@dk-d820> <41646A56-C830-489D-99D7-CE5C1218E194@freeswitch.org> Message-ID: <4A53F658.2030400@freeswitch.org> Brian West wrote: > Kaiden Anthony Chandler arrived Friday July 3rd at 1411 EDT 7lbs 10oz > > YAY... Congrats mr Lanman! > > /b THANKS BRIAN!!!! And the few others that sent money. The hospital is a little over an hour drive, so the money definitely helped out with gas, etc... You really are a great community... -Ray -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090707/9f1c891c/attachment-0002.html From hads at nice.net.nz Tue Jul 7 18:52:59 2009 From: hads at nice.net.nz (Hadley Rich) Date: Wed, 08 Jul 2009 13:52:59 +1200 Subject: [Freeswitch-users] Cheapest, most compact FreeSwitch appliance? In-Reply-To: References: <24337599.post@talk.nabble.com> Message-ID: <1247017979.30254.11.camel@lithium.nice.net.nz> On Sat, 2009-07-04 at 15:18 -0500, Brian West wrote: > I use one of the intel atom boxes at home. These guys have a case/PCI-riser for an Intel Atom board which would make a nice little appliance. http://www.mini-box.com/I-O-shield-and-riser-card-for-D945GSEJT hads -- http://nicegear.co.nz New Zealand's Open Source Hardware Supplier From brian at freeswitch.org Tue Jul 7 19:02:11 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 7 Jul 2009 21:02:11 -0500 Subject: [Freeswitch-users] Cheapest, most compact FreeSwitch appliance? In-Reply-To: <1247017979.30254.11.camel@lithium.nice.net.nz> References: <24337599.post@talk.nabble.com> <1247017979.30254.11.camel@lithium.nice.net.nz> Message-ID: <3ACD5095-3594-4990-8360-3131EF374D32@freeswitch.org> I'll take two of those too! :) /b On Jul 7, 2009, at 8:52 PM, Hadley Rich wrote: > On Sat, 2009-07-04 at 15:18 -0500, Brian West wrote: >> I use one of the intel atom boxes at home. > > These guys have a case/PCI-riser for an Intel Atom board which would > make a nice little appliance. > > http://www.mini-box.com/I-O-shield-and-riser-card-for-D945GSEJT > > hads > -- > http://nicegear.co.nz > New Zealand's Open Source Hardware Supplier > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090707/98ecd69e/attachment-0002.html From craig at overthewire.com.au Tue Jul 7 19:11:16 2009 From: craig at overthewire.com.au (Craig Askings) Date: Wed, 8 Jul 2009 12:11:16 +1000 Subject: [Freeswitch-users] Cheapest, most compact FreeSwitch appliance? In-Reply-To: <1247015680.4515.15.camel@raul-laptop> References: <24337599.post@talk.nabble.com> <7C0A0FFD-EC03-46DE-A277-AE793017D617@howlertech.com> <24345928.post@talk.nabble.com> <9dc4a1670907051617k3c112eeal2356d3df55283f89@mail.gmail.com> <24348506.post@talk.nabble.com> <1247015680.4515.15.camel@raul-laptop> Message-ID: <8cc991dd0907071911v1c26861ck43f54d43c2f055ce@mail.gmail.com> Does the mini-pci slot have external access to route the POTS cable? 2009/7/8 Raul Fragoso : > I'm about to order one of these eBox systems: > http://www.wdlsystems.com/modperl/view_services.cgi?r=detail&prod_num=1EBX33J&aisle_id=1073 > > They sell for under $200, have decent specs and some models come with a > mini-PCI slot, which can be used to attach a POTS card. > > Regards, > > Raul -- Craig Askings Network Engineer | Over the Wire Pty Ltd craig at overthewire.com.au | www.overthewire.com.au Phone: 07 3847 9292 | Fax: 07 3847 9696 | Mobile: 0404 019 365 From hyppolite72 at yahoo.com Tue Jul 7 19:22:10 2009 From: hyppolite72 at yahoo.com (Jean-Marc Hyppolite) Date: Tue, 7 Jul 2009 19:22:10 -0700 (PDT) Subject: [Freeswitch-users] Controlling MOH from a java application Message-ID: <602658.55818.qm@web35601.mail.mud.yahoo.com> Hello, ? Thank you Anthony. ? My problem now is when the call is put on hold, no music is heard from the caller. ? Thank you again. ? Jean-Marc. --- On Tue, 7/7/09, Anthony Minessale wrote: From: Anthony Minessale Subject: Re: [Freeswitch-users] Controlling MOH from a java application To: freeswitch-users at lists.freeswitch.org Received: Tuesday, July 7, 2009, 4:37 PM FSAPI commands are accessed via the API obj api = new API(); api.execute("uuid_hold", session.get_uuid()); ... api.execute("uuid_hold", "off " + session.get_uuid()); On Mon, Jul 6, 2009 at 10:21 PM, Jean-Marc Hyppolite wrote: Hello Brian, ? Thank you for your quick answer. I tried the two API functions but with no result. The caller is not able to hear any music. But, when I use two extensions (one calling the other), MOH does work. ? My code on the JAVA side (for test purposes) ? session.answer(); session.sleep(500); ? session.execute("eval", "uuid_hold " + session.get_uuid()); ? java_function(); // lasts 30 to 40 seconds ? session.execute("eval", "uuid_hold off " + session.get_uuid()); ? session.sleep(500); session.hangup(); ? Thanks for the help. ? Jean-Marc. ? ? ? --- On Mon, 7/6/09, Brian West wrote: From: Brian West Subject: Re: [Freeswitch-users] Controlling MOH from a java application To: freeswitch-users at lists.freeswitch.org Received: Monday, July 6, 2009, 7:27 PM uuid_hold uuid_hold off These two api's will do it. /b On Jul 6, 2009, at 6:20 PM, Jean-Marc Hyppolite wrote: Hello, ? First of all, I would like to thank Anthony, Brian and all the developers for this wonderful piece of software. Very good job. ? I would like to know how I can start and stop Music On Hold from a JAVA script (using mod_java) similar to the StartMusicOnHold and StopMusicHold functions found in AGI (Asterisk-Java). ? I am using FreeSWITCH as an IVR server. I would like to be able to put the caller on hold while doing some other stuff. ? Thanks in advance. ? Jean-Marc. Yahoo! Canada Toolbar : Search from anywhere on the web and bookmark your favourite sites. Download it now! _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -----Inline Attachment Follows----- _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org All new Yahoo! Mail - Get a sneak peak at messages with a handy reading pane. _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -----Inline Attachment Follows----- _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org __________________________________________________________________ Yahoo! Canada Toolbar: Search from anywhere on the web, and bookmark your favourite sites. Download it now http://ca.toolbar.yahoo.com. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090707/fdf9ed2e/attachment-0002.html From brian at freeswitch.org Tue Jul 7 19:28:08 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 7 Jul 2009 21:28:08 -0500 Subject: [Freeswitch-users] Controlling MOH from a java application In-Reply-To: <602658.55818.qm@web35601.mail.mud.yahoo.com> References: <602658.55818.qm@web35601.mail.mud.yahoo.com> Message-ID: <33BE7599-9BA8-4743-A725-C260E89BE8C8@freeswitch.org> Can you open a jira please. /b On Jul 7, 2009, at 9:22 PM, Jean-Marc Hyppolite wrote: > Hello, > > Thank you Anthony. > > My problem now is when the call is put on hold, no music is heard > from the caller. > > Thank you again. > > Jean-Marc. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090707/2c3f3de6/attachment-0002.html From Nick.Lemberger at lkfd.net Tue Jul 7 19:46:15 2009 From: Nick.Lemberger at lkfd.net (Nick Lemberger) Date: Tue, 07 Jul 2009 21:46:15 -0500 Subject: [Freeswitch-users] Force SUBSCRIBE or sendevent NOTIFY without subscription Message-ID: <4A53C228020000FE00009B5D@application-tr-fa-1.lakefield.telco> Is it possible to force a sofia profile to subscribe to an event or use sendevent to force send a NOTIFY to a SIP endpoint? I'm trying to use FreeSwitch as a voicemail server but the sending switch doesn't send SIP SUBSCRIBE messages. I'd like to send unsolicited SIP notifies to turn on MWI indicators as that's what the switch expects. Any ideas, or is this even possible with FreeSwitch? Thanks, Nick From hyppolite72 at yahoo.com Tue Jul 7 19:54:46 2009 From: hyppolite72 at yahoo.com (Jean-Marc Hyppolite) Date: Tue, 7 Jul 2009 19:54:46 -0700 (PDT) Subject: [Freeswitch-users] Controlling MOH from a java application Message-ID: <141988.11413.qm@web35607.mail.mud.yahoo.com> Thank you Brian. --- On Tue, 7/7/09, Brian West wrote: From: Brian West Subject: Re: [Freeswitch-users] Controlling MOH from a java application To: freeswitch-users at lists.freeswitch.org Received: Tuesday, July 7, 2009, 10:28 PM Can you open a jira please. /b On Jul 7, 2009, at 9:22 PM, Jean-Marc Hyppolite wrote: Hello, ? Thank you Anthony. ? My problem now is when the call is put on hold, no music is heard from the caller. ? Thank you again. ? Jean-Marc. -----Inline Attachment Follows----- _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org __________________________________________________________________ Make your browsing faster, safer, and easier with the new Internet Explorer? 8. Optimized for Yahoo! Get it Now for Free! at http://downloads.yahoo.com/ca/internetexplorer/ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090707/86d8fbeb/attachment-0002.html From brian at freeswitch.org Tue Jul 7 19:59:09 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 7 Jul 2009 21:59:09 -0500 Subject: [Freeswitch-users] Force SUBSCRIBE or sendevent NOTIFY without subscription In-Reply-To: <4A53C228020000FE00009B5D@application-tr-fa-1.lakefield.telco> References: <4A53C228020000FE00009B5D@application-tr-fa-1.lakefield.telco> Message-ID: <3B3DB3A0-5EB9-475F-A456-1B34952DFA13@freeswitch.org> Yes you can sendevent NOTIFY Here is the headers you'll need.. some are optional const char *profile_name = switch_event_get_header(event, "profile"); const char *ct = switch_event_get_header(event, "content-type"); const char *es = switch_event_get_header(event, "event-string"); const char *user = switch_event_get_header(event, "user"); const char *host = switch_event_get_header(event, "host"); const char *call_id = switch_event_get_header(event, "call-id"); const char *uuid = switch_event_get_header(event, "uuid"); const char *body = switch_event_get_body(event); const char *to_uri = switch_event_get_header(event, "to-uri"); const char *from_uri = switch_event_get_header(event, "from-uri"); See mod_sofia.c line 2887 /b PS: you also have SEND_MESSAGE as an event you can send below that in mod_sofia.c On Jul 7, 2009, at 9:46 PM, Nick Lemberger wrote: > Is it possible to force a sofia profile to subscribe to an event or > use sendevent to force send a NOTIFY to a SIP endpoint? > > I'm trying to use FreeSwitch as a voicemail server but the sending > switch doesn't send SIP SUBSCRIBE messages. I'd like to send > unsolicited SIP notifies to turn on MWI indicators as that's what > the switch expects. > > Any ideas, or is this even possible with FreeSwitch? > > Thanks, > Nick From yehavi.bourvine at gmail.com Tue Jul 7 21:40:36 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Wed, 8 Jul 2009 07:40:36 +0300 Subject: [Freeswitch-users] How to get the hook state? In-Reply-To: References: <4A52D4D1.1010805@gmail.com> Message-ID: Hello, The problem we are trying to solve here is handling a busy state according to the user's prefference (some want a busy to be heard, some want the call to go to voicemail, and some want to get the second call). The first step is finding that an extension is busy. It would be nice in the future to know also other states of an extension (like - not registered, etc.). Thanks, __Yehavi: 2009/7/7 Brian West > What are you trying to accomplish? > > /b > > On Jul 6, 2009, at 11:53 PM, Eli Hayun wrote: > > > Hi > > I am a newbie in FreeSwitch and my question is: > > When I am calling to an extension, how should I know in advance what > > is > > the hook status. I tried to find out a variable that can get me this > > information but with no success. > > any help? > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090708/66f98d37/attachment-0002.html From shiyanov at gmail.com Tue Jul 7 23:39:15 2009 From: shiyanov at gmail.com (Artem Shiyanov) Date: Wed, 8 Jul 2009 10:39:15 +0400 Subject: [Freeswitch-users] Can't mute SIP channel with "receiveonly" in SDP Message-ID: Hy all! With Asterisk I can mute SIP channel using re-INVITE with "a=receiveonly" in media description. But this feature doesn't work with Freeswitch. For sure, there is old good method: transfer both legs to the conference room where one leg is able to listen/talk, the other one - only to listen, but this is unwanted workaround for me.. So I wonder: is there any other (preferably through the SIP) way to "mute" given SIP channel with Freeswitch? Thanks for all, Artem Shiyanov -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090708/32e2a3f9/attachment-0002.html From Claudio.Cavalera at italtel.it Wed Jul 8 01:53:39 2009 From: Claudio.Cavalera at italtel.it (Cavalera Claudio Luigi) Date: Wed, 8 Jul 2009 10:53:39 +0200 Subject: [Freeswitch-users] ClueCon 2009 Updates - July 7, 2009 In-Reply-To: <87f2f3b90907071214w60f0d1b0x21d39b7e5ad661ac@mail.gmail.com> Message-ID: I'm really missing this event! Can't we organize a ClueCon Winter edition in Europe too? :-) BRs, Claudio ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Tuesday, July 07, 2009 9:14 PM To: marketing at cluecon.com Subject: [Freeswitch-users] ClueCon 2009 Updates - July 7, 2009 Hello folks! We have a few more updates. First of all, if you haven't already heard, we've extended the early bird sign up to go through July 21. That's only two weeks away, so if you haven't already registered then please call us at 877.742.CLUE and we'll get you set up. Secondly, there are some updates on the ClueCon blog: http://cluecon.com/blog/1 The breakfast and lunch menus have been posted. (Subject to change, of course. :) Also, we have a synopsis for Irv Shapiro's talk, entitled "Cloud Telephony" on the latest blog post. Can't wait to see you all in Chicago! -Michael Collins http://www.cluecon.com 877.742.CLUE 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. ----------------------------------------------------------------------------------------------------- -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090708/82ee3da4/attachment-0002.html From michal.bielicki at halo2.pl Wed Jul 8 02:13:59 2009 From: michal.bielicki at halo2.pl (Michal Bielicki) Date: Wed, 8 Jul 2009 11:13:59 +0200 Subject: [Freeswitch-users] ClueCon 2009 Updates - July 7, 2009 In-Reply-To: References: Message-ID: <04909F90-2B3C-4796-BB85-7DB13D5D6493@halo2.pl> The GUUG will be doing something along those lines in fall 2010 in Germany. Am 08.07.2009 um 10:53 schrieb Cavalera Claudio Luigi: > I'm really missing this event! > Can't we organize a ClueCon Winter edition in Europe too? :-) > BRs, > Claudio > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Michael Collins > Sent: Tuesday, July 07, 2009 9:14 PM > To: marketing at cluecon.com > Subject: [Freeswitch-users] ClueCon 2009 Updates - July 7, 2009 > > Hello folks! > > We have a few more updates. First of all, if you haven't already > heard, we've extended the early bird sign up to go through July 21. > That's only two weeks away, so if you haven't already registered > then please call us at 877.742.CLUE and we'll get you set up. > Secondly, there are some updates on the ClueCon blog: > http://cluecon.com/blog/1 > > The breakfast and lunch menus have been posted. (Subject to change, > of course. :) Also, we have a synopsis for Irv Shapiro's talk, > entitled "Cloud Telephony" on the latest blog post. > > Can't wait to see you all in Chicago! > -Michael Collins > http://www.cluecon.com > 877.742.CLUE > > 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 Michal Bielicki Leiter der Niederlassung HaloKwadrat Sp. z o.o. Niederlassung Kleinmachnow Eingetragen im Handelsregister beim Amtsgericht Potsdam, HRB21422P Ust.Id.: DE261885536 Geschaeftsfuehrer: Aleksander Wiercinski Meiereifeld 2b, 14532 Kleinmachnow t. +49 33203 263220 f. +49 33203 263229 sip. info at halokwadrat.de e. michal.bielicki at halokwadrat.de | w. www.halokwadrat.de Hauptgesch?ftsstelle: Halo Kwadrat Sp. z o.o. ul. Polna 46/14 00-644 Warszawa, Polen EIngetragen im HRB Warszawa, KRS 0000153539 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090708/95b1a406/attachment-0002.html -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/pkcs7-signature Size: 2453 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090708/95b1a406/attachment-0002.bin From mariusz_kolo at wp.pl Wed Jul 8 04:23:50 2009 From: mariusz_kolo at wp.pl (=?ISO-8859-2?Q?Mariusz_Ko=B3odziejczyk?=) Date: Wed, 08 Jul 2009 13:23:50 +0200 Subject: [Freeswitch-users] Record_session cutting wav files In-Reply-To: <4A5324D6.3070600@wp.pl> References: <4A5324D6.3070600@wp.pl> Message-ID: <4A5481C6.2060207@wp.pl> Hello After upgrade to latest version wavs recording is OK Thanks a lot >Hello, >please try out FS >= r14143 as there were some fixes around call >recording and media bugs. >Please let us know the results. >Regards, > Tamas Mariusz Ko?odziejczyk pisze: > Hello > > I saw a strange behavior when i'm using record_session for outbound > call. Recorded file is 24:20 time length, but in logs should have > about 25:18. > > Here ma log: > > start recording about 2009-07-07 10:56:15 > 2009-07-07 10:56:15.108447 [DEBUG] mod_dptools.c:748 > sofia/internal/1062 SET [czas]=[2009-07-07-10-56-15] - variable $czas > = "2009-07-07-10-56-15" i use it in filename below > EXECUTE sofia/internal/1062 > record_session(/records/2009-07-07/069630d2-6ad4-11de-aa6e-e31e1f9431c0.10XX.07135XXXXX.2009-07-07-10-56-15.out.ISDN.wav) > > ..... > stop recording > 2009-07-07 11:21:33.291177 [DEBUG] switch_ivr_async.c:444 Stop > recording file > /records/2009-07-07/069630d2-6ad4-11de-aa6e-e31e1f9431c0.10XX.07135XXXXX.2009-07-07-10-56-15.out.ISDN.wav > > > file should have about: 25:18 time length > when i listen a file it's really cut > > My piece of dialplan: > > > > > > > > > > > > > > > data="czas=${strftime(%Y-%m-%d-%H-%M-%S)}"/> > > > data="/records/${dir}/${uuid}.${caller_id_number}.$1.${czas}.out.ISDN.wav"/> > > > > data="{origination_caller_id_number=${cti_gateway_number},effective_caller_id_number=${cti_gateway_number}}openzap/1/A/$1"/> > > > > freeswitch version: FreeSWITCH Version 1.0.trunk (14013) > linux: Linux pbx2 2.6.26-1-686 #1 SMP Fri Mar 13 18:08:45 UTC 2009 > i686 GNU/Linux > > Thanks > From mariusz_kolo at wp.pl Wed Jul 8 04:24:08 2009 From: mariusz_kolo at wp.pl (=?ISO-8859-2?Q?Mariusz_Ko=B3odziejczyk?=) Date: Wed, 08 Jul 2009 13:24:08 +0200 Subject: [Freeswitch-users] Record_session cutting wav files In-Reply-To: <4A5324D6.3070600@wp.pl> References: <4A5324D6.3070600@wp.pl> Message-ID: <4A5481D8.3090204@wp.pl> Hello After upgrade to latest version wavs recording is OK Thanks a lot >Hello, >please try out FS >= r14143 as there were some fixes around call >recording and media bugs. >Please let us know the results. >Regards, > Tamas Mariusz Ko?odziejczyk pisze: > Hello > > I saw a strange behavior when i'm using record_session for outbound > call. Recorded file is 24:20 time length, but in logs should have > about 25:18. > > Here ma log: > > start recording about 2009-07-07 10:56:15 > 2009-07-07 10:56:15.108447 [DEBUG] mod_dptools.c:748 > sofia/internal/1062 SET [czas]=[2009-07-07-10-56-15] - variable $czas > = "2009-07-07-10-56-15" i use it in filename below > EXECUTE sofia/internal/1062 > record_session(/records/2009-07-07/069630d2-6ad4-11de-aa6e-e31e1f9431c0.10XX.07135XXXXX.2009-07-07-10-56-15.out.ISDN.wav) > > ..... > stop recording > 2009-07-07 11:21:33.291177 [DEBUG] switch_ivr_async.c:444 Stop > recording file > /records/2009-07-07/069630d2-6ad4-11de-aa6e-e31e1f9431c0.10XX.07135XXXXX.2009-07-07-10-56-15.out.ISDN.wav > > > file should have about: 25:18 time length > when i listen a file it's really cut > > My piece of dialplan: > > > > > > > > > > > > > > > data="czas=${strftime(%Y-%m-%d-%H-%M-%S)}"/> > > > data="/records/${dir}/${uuid}.${caller_id_number}.$1.${czas}.out.ISDN.wav"/> > > > > data="{origination_caller_id_number=${cti_gateway_number},effective_caller_id_number=${cti_gateway_number}}openzap/1/A/$1"/> > > > > freeswitch version: FreeSWITCH Version 1.0.trunk (14013) > linux: Linux pbx2 2.6.26-1-686 #1 SMP Fri Mar 13 18:08:45 UTC 2009 > i686 GNU/Linux > > Thanks > From rupa at rupa.com Wed Jul 8 05:14:52 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Wed, 8 Jul 2009 07:14:52 -0500 Subject: [Freeswitch-users] How to get the hook state? In-Reply-To: References: <4A52D4D1.1010805@gmail.com> Message-ID: I think you can (mostly) solved this using mod_limit and some dialplan work. Something like: if you want the second call to ring, limit should be set to 2 or higher. for either VM or busy, limit should be set to 1 if limit was reached and the user wants busy, send back busy if limit was reached and the user wants VM, transfer to voicemail Remember that you have to check the limit both on outbound calls and on inbound calls so that you get the desired behavior. Also, you'll have to special handle outbound calls so they don't fail if the limit is reached (think transfer). On Tue, Jul 7, 2009 at 11:40 PM, Yehavi Bourvine wrote: > Hello, > > The problem we are trying to solve here is handling a busy state > according to the user's prefference (some want a busy to be heard, some want > the call to go to voicemail, and some want to get the second call). > > The first step is finding that an extension is busy. It would be nice in > the future to know also other states of an extension (like - not registered, > etc.). > > Thanks, __Yehavi: > > 2009/7/7 Brian West > > What are you trying to accomplish? >> >> /b >> >> On Jul 6, 2009, at 11:53 PM, Eli Hayun wrote: >> >> > Hi >> > I am a newbie in FreeSwitch and my question is: >> > When I am calling to an extension, how should I know in advance what >> > is >> > the hook status. I tried to find out a variable that can get me this >> > information but with no success. >> > any help? >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090708/306aa6bb/attachment-0002.html From brian at freeswitch.org Wed Jul 8 05:15:18 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 8 Jul 2009 07:15:18 -0500 Subject: [Freeswitch-users] Can't mute SIP channel with "receiveonly" in SDP In-Reply-To: References: Message-ID: <9BD68E4E-F7C7-4E48-AF0E-D7E46F8F6194@freeswitch.org> You could put a feature request bounty on jira. /b On Jul 8, 2009, at 1:39 AM, Artem Shiyanov wrote: > Hy all! > > With Asterisk I can mute SIP channel using re-INVITE with > "a=receiveonly" in media description. But this feature doesn't work > with Freeswitch. For sure, there is old good method: transfer both > legs to the conference room where one leg is able to listen/talk, > the other one - only to listen, but this is unwanted workaround for > me.. > So I wonder: is there any other (preferably through the SIP) way to > "mute" given SIP channel with Freeswitch? > > Thanks for all, > Artem Shiyanov From brian at freeswitch.org Wed Jul 8 05:20:08 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 8 Jul 2009 07:20:08 -0500 Subject: [Freeswitch-users] How to get the hook state? In-Reply-To: References: <4A52D4D1.1010805@gmail.com> Message-ID: continue_on_fail=user_busy (not 100% reliable because some devices won't say that the user is busy and ring the second line) facility_not_subscribed is what you'll get if they aren't registered. /b On Jul 7, 2009, at 11:40 PM, Yehavi Bourvine wrote: > Hello, > > The problem we are trying to solve here is handling a busy state > according to the user's prefference (some want a busy to be heard, > some want the call to go to voicemail, and some want to get the > second call). > > The first step is finding that an extension is busy. It would be > nice in the future to know also other states of an extension (like - > not registered, etc.). > > Thanks, __Yehavi: From maxim.tsvetov at gmail.com Wed Jul 8 05:42:03 2009 From: maxim.tsvetov at gmail.com (Maxim Tsvetov) Date: Wed, 8 Jul 2009 16:42:03 +0400 Subject: [Freeswitch-users] Freeswitch architecture Message-ID: <89c9bbf80907080542y39f2b234tff9163ea34c0b968@mail.gmail.com> Hi All Where can I get information about internal Freeswitch architecture: 1) how modules interoperates with each other (maybe using corba or com objects or something else) 2) how core interoperates with other modules 3) how javascript function is translated to internal commands. In addition if you cand send me some schemas of Freeswitch architecture that will be great. Regards, Maxim Tsvetov -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090708/7488203c/attachment-0002.html From maxim.tsvetov at gmail.com Wed Jul 8 05:53:43 2009 From: maxim.tsvetov at gmail.com (Maxim Tsvetov) Date: Wed, 8 Jul 2009 16:53:43 +0400 Subject: [Freeswitch-users] CallID Message-ID: <89c9bbf80907080553jf7a7fbbga6c963599f4dabb2@mail.gmail.com> Hi All Can somebody explain me algorithm of assigning CallID to new calls in Freeswitch. Are they unique? If not - how frequent they become duplicated. Regards, Maxim Tsvetov -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090708/9a9f3a1b/attachment-0002.html From brian at freeswitch.org Wed Jul 8 06:29:28 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 8 Jul 2009 08:29:28 -0500 Subject: [Freeswitch-users] CallID In-Reply-To: <89c9bbf80907080553jf7a7fbbga6c963599f4dabb2@mail.gmail.com> References: <89c9bbf80907080553jf7a7fbbga6c963599f4dabb2@mail.gmail.com> Message-ID: They are uuid's and yes they are unique. /b On Jul 8, 2009, at 7:53 AM, Maxim Tsvetov wrote: > Hi All > Can somebody explain me algorithm of assigning CallID to new calls > in Freeswitch. Are they unique? If not - how frequent they become > duplicated. > Regards, > Maxim Tsvetov From brian at freeswitch.org Wed Jul 8 06:29:50 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 8 Jul 2009 08:29:50 -0500 Subject: [Freeswitch-users] Freeswitch architecture In-Reply-To: <89c9bbf80907080542y39f2b234tff9163ea34c0b968@mail.gmail.com> References: <89c9bbf80907080542y39f2b234tff9163ea34c0b968@mail.gmail.com> Message-ID: <347F12AB-EA05-4090-A203-BDC011F02411@freeswitch.org> http://wiki.freeswitch.org /b On Jul 8, 2009, at 7:42 AM, Maxim Tsvetov wrote: > Hi All > > Where can I get information about internal Freeswitch architecture: > 1) how modules interoperates with each other (maybe using corba or > com > objects or something else) > 2) how core interoperates with other modules > 3) how javascript function is translated to internal commands. > > In addition if you cand send me some schemas of Freeswitch > architecture > that will be great. > > Regards, > Maxim Tsvetov > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.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 Jul 8 07:01:24 2009 From: Claudio.Cavalera at italtel.it (Cavalera Claudio Luigi) Date: Wed, 8 Jul 2009 16:01:24 +0200 Subject: [Freeswitch-users] ClueCon 2009 Updates - July 7, 2009 In-Reply-To: <04909F90-2B3C-4796-BB85-7DB13D5D6493@halo2.pl> Message-ID: Advertise it a bit when it will happen! :-) ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michal Bielicki Sent: Wednesday, July 08, 2009 11:14 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] ClueCon 2009 Updates - July 7, 2009 The GUUG will be doing something along those lines in fall 2010 in Germany. Am 08.07.2009 um 10:53 schrieb Cavalera Claudio Luigi: I'm really missing this event! Can't we organize a ClueCon Winter edition in Europe too? :-) BRs, Claudio ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Tuesday, July 07, 2009 9:14 PM To: marketing at cluecon.com Subject: [Freeswitch-users] ClueCon 2009 Updates - July 7, 2009 Hello folks! We have a few more updates. First of all, if you haven't already heard, we've extended the early bird sign up to go through July 21. That's only two weeks away, so if you haven't already registered then please call us at 877.742.CLUE and we'll get you set up. Secondly, there are some updates on the ClueCon blog: http://cluecon.com/blog/1 The breakfast and lunch menus have been posted. (Subject to change, of course. :) Also, we have a synopsis for Irv Shapiro's talk, entitled "Cloud Telephony" on the latest blog post. Can't wait to see you all in Chicago! -Michael Collins http://www.cluecon.com 877.742.CLUE 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 Michal Bielicki Leiter der Niederlassung HaloKwadrat Sp. z o.o. Niederlassung Kleinmachnow Eingetragen im Handelsregister beim Amtsgericht Potsdam, HRB21422P Ust.Id.: DE261885536 Geschaeftsfuehrer: Aleksander Wiercinski Meiereifeld 2b, 14532 Kleinmachnow t. +49 33203 263220 f. +49 33203 263229 sip. info at halokwadrat.de e. michal.bielicki at halokwadrat.de | w. www.halokwadrat.de Hauptgesch?ftsstelle: Halo Kwadrat Sp. z o.o. ul. Polna 46/14 00-644 Warszawa, Polen EIngetragen im HRB Warszawa, KRS 0000153539 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090708/ad9b346a/attachment-0002.html From sprice at gmail.com Wed Jul 8 07:11:02 2009 From: sprice at gmail.com (SP) Date: Wed, 8 Jul 2009 09:11:02 -0500 Subject: [Freeswitch-users] Freeswitch architecture In-Reply-To: <89c9bbf80907080542y39f2b234tff9163ea34c0b968@mail.gmail.com> References: <89c9bbf80907080542y39f2b234tff9163ea34c0b968@mail.gmail.com> Message-ID: <7e2ac3270907080711i48ebaf9coed950556e712c8aa@mail.gmail.com> http://cluecon.com/node/3 On Wed, Jul 8, 2009 at 07:42, Maxim Tsvetov wrote: > Hi All > > Where can I get information about internal Freeswitch architecture: > 1) how modules interoperates with each other (maybe using corba or com > objects or something else) > 2) how core interoperates with other modules > 3) how javascript function is translated to internal commands. > > In addition if you cand send me some schemas of Freeswitch architecture > that will be great. > > Regards, > Maxim Tsvetov > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090708/2ec4f18e/attachment-0002.html From Prometheus001 at gmx.net Wed Jul 8 08:50:36 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Wed, 08 Jul 2009 17:50:36 +0200 Subject: [Freeswitch-users] Sangoma 108 and libpri problems - only distortion sound Message-ID: <4A54C04C.7080804@gmx.net> Hello, I installed a Sangoma A108 with openzap and libpri. Signalling (E1) works sometimes (inbound and outbound calls are connected) but not always. Sound is just distortion but connection is stable. 2 questions: 1) What is the best way to go with Sangoma? OpenZAP with libpri or without libpri? (I remember there are some timer problems in openzap when not using libpri but this might have changed) However Sangoma recommends OpenZAP on their wiki. 2) What might cause the distortion? I crosschecked the config files and had a look at the interrupts (<2k/sec). Seems to be ok. ACPI and APIC is turned on. Freeswitch starts successfully with all the channels enabled. "oz dump" shows D-Channel up. "oz libpri debug 1 all" shows debugging messages with no special warnings. Best regards Peter Some confs for 1 Channel: Wanpipe1.conf [devices] wanpipe1 = WAN_AFT_TE1, Comment [interfaces] w1g1 = wanpipe1, , TDM_VOICE_API, Comment [wanpipe1] CARD_TYPE = AFT S514CPU = A CommPort = PRI AUTO_PCISLOT = NO PCISLOT = 4 PCIBUS = 11 FE_MEDIA = E1 FE_LCODE = HDB3 FE_FRAME = CRC4 FE_LINE = 1 TE_CLOCK = NORMAL TE_REF_CLOCK = 0 TE_SIG_MODE = CCS TE_HIGHIMPEDANCE = NO LBO = 120OH FE_TXTRISTATE = NO MTU = 1500 UDPPORT = 9000 TTL = 255 IGNORE_FRONT_END = NO TDMV_SPAN = 1 TDMV_DCHAN = 16 TDMV_HW_DTMF = YES TDMV_HW_FAX_DETECT = YES [w1g1] ACTIVE_CH = ALL TDMV_HWEC = YES MTU = 80 openzap.conf [span wanpipe PRI_1] name => OpenZAP number => 1 trunk_type => e1 b-channel => 1:1-15 d-channel => 1:16 b-channel => 1:17-31 openzap.conf.xml From paparoga at mailinator.com Wed Jul 8 07:41:40 2009 From: paparoga at mailinator.com (paparoga at mailinator.com) Date: Wed, 8 Jul 2009 16:41:40 +0200 Subject: [Freeswitch-users] simple originate / bridge js Message-ID: <200907081641.40381.paparoga@mailinator.com> Hi all, I'm attempting to setup a simple alarm handling machine. It should be triggered by an external event, dial a phone number (depending on the alarm type), and play a few wav files indicating the failure happened. Using Free, up to now, I've created a simple IVR and connected it to my EXT. 118. Connecting a softphone to the ext. 1001 and dialing the ext 118 the IVR is ok. Also using the following Free command from the console all is ok: originate sofia/zz.xxx.200.29/1001 118 My softphone at ext 1001 get ringed and then connected to the IVR at ext. 118. I cannot get the same from a simple js. I tried: ===================================== session = new Session("sofia/zz.xxx.200.29/1001"); //session = new Session(); //session.originate(session, "sofia/zz.yyy.200.29/118"); session.execute("bridge", "sofia/default/118"); ===================================== and almost all possibles variations, but I'get this result: 2009-07-08 16:23:00 [NOTICE] sofia.c:3220 sofia_handle_sip_i_state() Channel [sofia/internal/1001] has been answered 2009-07-08 16:23:00 [WARNING] mod_sofia.c:2495 sofia_outgoing_channel() Cannot locate registered user 118 at default 2009-07-08 16:23:00 [NOTICE] mod_sofia.c:2591 sofia_outgoing_channel() Close Channel N/A [CS_NEW] How can I tell to the script that the EXT. 118 is an IVR and not a registered USER? By the way, I attempted also the 'transfer' function, but I get the following: 2009-07-08 16:35:43 [NOTICE] sofia.c:2782 sofia_handle_sip_i_state() Ring-Ready sofia/internal/1001! 2009-07-08 16:35:50 [NOTICE] sofia.c:3220 sofia_handle_sip_i_state() Channel [sofia/internal/1001] has been answered 2009-07-08 16:35:50 [NOTICE] switch_ivr.c:1343 switch_ivr_session_transfer() Transfer sofia/internal/1001 to XML[sofia/default/118 at default] 2009-07-08 16:35:50 [NOTICE] mod_spidermonkey.c:2994 session_destroy() Hangup sofia/internal/1001 [CS_ROUTING] [NORMAL_CLEARING] 2009-07-08 16:35:50 [NOTICE] switch_core_session.c:970 switch_core_session_thread() Session 8 (sofia/internal/1001) Ended 2009-07-08 16:35:50 [NOTICE] switch_core_session.c:972 switch_core_session_thread() Close Channel sofia/internal/1001 [CS_HANGUP] Any suggestion? Regards Kowalsky From larclap at yahoo.com Wed Jul 8 09:33:34 2009 From: larclap at yahoo.com (Lars Zeb) Date: Wed, 8 Jul 2009 09:33:34 -0700 Subject: [Freeswitch-users] Can't understand documentation Message-ID: <009301c9ffe9$d6424560$82c6d020$@com> On http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session, at the bottom of "Activating via DTMF", it states: The other party doesn't hear the DTMFs but maybe its comfort noisy is disappearing for a very short time. When that sip client *starts* a call the above dialplan forbids activating recording. Can someone explain what these two sentences mean? Thanks, Lars -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090708/243a181c/attachment-0002.html From anthony.minessale at gmail.com Wed Jul 8 09:40:11 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 8 Jul 2009 11:40:11 -0500 Subject: [Freeswitch-users] simple originate / bridge js In-Reply-To: <200907081641.40381.paparoga@mailinator.com> References: <200907081641.40381.paparoga@mailinator.com> Message-ID: <191c3a030907080940x32dd2469w7ba5e32601bb1e9c@mail.gmail.com> in the first example you are transferring to 118 in the script you are calling a sip address but you are not supplying the domain try var my_domain = "1.2.3.4" session.execute("bridge", "sofia/default/118%" + my_domain); where you set my_domain to whatever domain your phone registered with. On Wed, Jul 8, 2009 at 9:41 AM, wrote: > Hi all, > > I'm attempting to setup a simple alarm handling machine. > > It should be triggered by an external event, dial a phone number (depending > on > the alarm type), and play a few wav files indicating the failure happened. > > Using Free, up to now, I've created a simple IVR and connected it to my > EXT. > 118. > > Connecting a softphone to the ext. 1001 and dialing the ext 118 the IVR is > ok. > > Also using the following Free command from the console all is ok: > > originate sofia/zz.xxx.200.29/1001 118 > > My softphone at ext 1001 get ringed and then connected to the IVR at ext. > 118. > > I cannot get the same from a simple js. > > I tried: > ===================================== > session = new Session("sofia/zz.xxx.200.29/1001"); > //session = new Session(); > //session.originate(session, "sofia/zz.yyy.200.29/118"); > session.execute("bridge", "sofia/default/118"); > ===================================== > and almost all possibles variations, but I'get this result: > > 2009-07-08 16:23:00 [NOTICE] sofia.c:3220 sofia_handle_sip_i_state() > Channel > [sofia/internal/1001] has been answered > 2009-07-08 16:23:00 [WARNING] mod_sofia.c:2495 sofia_outgoing_channel() > Cannot > locate registered user 118 at default > 2009-07-08 16:23:00 [NOTICE] mod_sofia.c:2591 sofia_outgoing_channel() > Close > Channel N/A [CS_NEW] > > How can I tell to the script that the EXT. 118 is an IVR and not a > registered > USER? > > By the way, I attempted also the 'transfer' function, but I get the > following: > > 2009-07-08 16:35:43 [NOTICE] sofia.c:2782 sofia_handle_sip_i_state() > Ring-Ready sofia/internal/1001! > 2009-07-08 16:35:50 [NOTICE] sofia.c:3220 sofia_handle_sip_i_state() > Channel > [sofia/internal/1001] has been answered > 2009-07-08 16:35:50 [NOTICE] switch_ivr.c:1343 > switch_ivr_session_transfer() > Transfer sofia/internal/1001 to XML[sofia/default/118 at default] > 2009-07-08 16:35:50 [NOTICE] mod_spidermonkey.c:2994 session_destroy() > Hangup > sofia/internal/1001 [CS_ROUTING] [NORMAL_CLEARING] > 2009-07-08 16:35:50 [NOTICE] switch_core_session.c:970 > switch_core_session_thread() Session 8 (sofia/internal/1001) Ended > 2009-07-08 16:35:50 [NOTICE] switch_core_session.c:972 > switch_core_session_thread() Close Channel sofia/internal/1001 [CS_HANGUP] > > > Any suggestion? > > Regards > > Kowalsky > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference 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/20090708/7dc0ff01/attachment-0002.html From msc at freeswitch.org Wed Jul 8 09:48:08 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 8 Jul 2009 09:48:08 -0700 Subject: [Freeswitch-users] simple originate / bridge js In-Reply-To: <200907081641.40381.paparoga@mailinator.com> References: <200907081641.40381.paparoga@mailinator.com> Message-ID: <87f2f3b90907080948v27f7c1d6r7784da1d2353a2f4@mail.gmail.com> On Wed, Jul 8, 2009 at 7:41 AM, wrote: > Hi all, > > I'm attempting to setup a simple alarm handling machine. > > It should be triggered by an external event, dial a phone number (depending > on > the alarm type), and play a few wav files indicating the failure happened. > > Using Free, up to now, I've created a simple IVR and connected it to my > EXT. > 118. > > Connecting a softphone to the ext. 1001 and dialing the ext 118 the IVR is > ok. > > Also using the following Free command from the console all is ok: > > originate sofia/zz.xxx.200.29/1001 118 > > My softphone at ext 1001 get ringed and then connected to the IVR at ext. > 118. > > I cannot get the same from a simple js. > > I tried: > ===================================== > session = new Session("sofia/zz.xxx.200.29/1001"); > //session = new Session(); > //session.originate(session, "sofia/zz.yyy.200.29/118"); > session.execute("bridge", "sofia/default/118"); > ===================================== > and almost all possibles variations, but I'get this result: > > 2009-07-08 16:23:00 [NOTICE] sofia.c:3220 sofia_handle_sip_i_state() > Channel > [sofia/internal/1001] has been answered > 2009-07-08 16:23:00 [WARNING] mod_sofia.c:2495 sofia_outgoing_channel() > Cannot > locate registered user 118 at default > 2009-07-08 16:23:00 [NOTICE] mod_sofia.c:2591 sofia_outgoing_channel() > Close > Channel N/A [CS_NEW] > > How can I tell to the script that the EXT. 118 is an IVR and not a > registered > USER? It looks like 118 is defined in your dialplan so use transfer instead of bridge: session.execute("transfer", "118 XML default"); -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090708/eb3464bb/attachment-0002.html From msc at freeswitch.org Wed Jul 8 10:07:11 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 8 Jul 2009 10:07:11 -0700 Subject: [Freeswitch-users] Can't understand documentation In-Reply-To: <009301c9ffe9$d6424560$82c6d020$@com> References: <009301c9ffe9$d6424560$82c6d020$@com> Message-ID: <87f2f3b90907081007h73cc07b5s63eecd0f26710bf3@mail.gmail.com> We'll clean it up a bit, but for reference here's what they mean: #1 - the other party won't hear the DTMFs, but the DTMFs might disrupt the comfort noise generation (CNG). Perhaps instead of comfort noise the part might hear a short period of complete silence #2 - this sentence is awkwardly written but I believe it is referring to the bind_meta_app settings in the example dialplan, which are on the B leg only; if the SIP client in question is the A leg then the bind_meta_app would not be available. -MC On Wed, Jul 8, 2009 at 9:33 AM, Lars Zeb wrote: > On http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session, > at the bottom of ?Activating via DTMF?, it states: > > > > The other party doesn't hear the DTMFs but maybe its comfort noisy is > disappearing for a very short time. When that sip client *starts* a call the > above dialplan forbids activating recording. > > > > Can someone explain what these two sentences mean? > > > > Thanks, Lars > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090708/11a194f4/attachment-0002.html From Nick.Lemberger at lkfd.net Wed Jul 8 11:24:41 2009 From: Nick.Lemberger at lkfd.net (Nick Lemberger) Date: Wed, 08 Jul 2009 13:24:41 -0500 Subject: [Freeswitch-users] LUA Event generation Message-ID: <4A549E23.2C9A.00FE.0@lkfd.net> When I create an even with mod_lua, I need to add the even variable name as the 1st argument too all of the event statements - this is not how it's listed in the examples in the Wiki, is this new and should I update the WIKI or am I doing something wrong? The way I needed to write it to get it to work looks like this: ie: http://pastebin.freeswitch.org/9654 The examples in the wiki are just missing the first argument for addHeader(), addBody() and fire(). ie: http://wiki.freeswitch.org/wiki/Mod_lua#Events -Nick From paparoga at mailinator.com Wed Jul 8 13:15:57 2009 From: paparoga at mailinator.com (paparoga at mailinator.com) Date: Wed, 8 Jul 2009 22:15:57 +0200 Subject: [Freeswitch-users] simple originate / bridge js Message-ID: <200907082215.57328.paparoga@mailinator.com> First of all I apologise for my long post. In the meantime I reworked the simple js as suggested (adding the domain or using transfer instead of bridge) but yet the script doesn't work. Let me add some other info. I just cloned the '5000' demo_ivr and reworked a little to reach my target. Next I added: into the default.xml dialplan. Now all is ok if I connect Ekiga to FreeSwitch as user 1001 (for example) and than I dial '118'. The IVR works fine. Next from the console: originate sofia/my.freeswitch.address/1001 118 I get my Ekiga Phone ringing and connecter to the ext. 118 (the ivr) That is the log: ========================================================= originate sofia/10.0.0.33/1001 118 2009-07-08 21:56:57 [NOTICE] switch_channel.c:567 switch_channel_set_name() New Channel sofia/internal/1001 [7e660b20-6bf9-11de-b2fa-f3963e050c84] 2009-07-08 21:56:57 [NOTICE] sofia.c:2782 sofia_handle_sip_i_state() Ring-Ready sofia/internal/1001! 2009-07-08 21:57:00 [NOTICE] sofia.c:3220 sofia_handle_sip_i_state() Channel [sofia/internal/1001] has been answered 2009-07-08 21:57:00 [NOTICE] switch_ivr.c:1343 switch_ivr_session_transfer() Transfer sofia/internal/1001 to XML[118 at default] API CALL [originate(sofia/10.0.0.33/1001 118)] output: +OK 7e660b20-6bf9-11de-b2fa-f3963e050c84 freeswitch at Linux61> 2009-07-08 21:57:00 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing FreeSWITCH->118 in context default 2009-07-08 21:57:02 [WARNING] switch_core_file.c:119 switch_core_perform_file_open() Sample rate doesn't match 2009-07-08 21:57:15 [WARNING] switch_core_file.c:119 switch_core_perform_file_open() Sample rate doesn't match 2009-07-08 21:57:21 [NOTICE] sofia.c:3279 sofia_handle_sip_i_state() Hangup sofia/internal/1001 [CS_EXECUTE] [NORMAL_CLEARING] 2009-07-08 21:57:25 [NOTICE] switch_core_session.c:970 switch_core_session_thread() Session 33 (sofia/internal/1001) Ended 2009-07-08 21:57:25 [NOTICE] switch_core_session.c:972 switch_core_session_thread() Close Channel sofia/internal/1001 [CS_HANGUP] ==================================================== Using the js instead it looks like if the bridge commands looks for a REGISTERED user at ext. 118, and so it fails. Using the transfer option, as suggested, the Ekiga user (1001) get to be connected to the ext 118, bur the connection drops immediately after. ==================================================== freeswitch at Linux61> jsrun alarm.js API CALL [jsrun(alarm.js)] output: OK freeswitch at Linux61> 2009-07-08 22:03:47 [NOTICE] switch_channel.c:567 switch_channel_set_name() New Channel sofia/internal/1001 [7267f936-6bfa-11de-b2fa-f3963e050c84] 2009-07-08 22:03:47 [NOTICE] sofia.c:2782 sofia_handle_sip_i_state() Ring-Ready sofia/internal/1001! 2009-07-08 22:03:51 [NOTICE] sofia.c:3220 sofia_handle_sip_i_state() Channel [sofia/internal/1001] has been answered 2009-07-08 22:03:51 [NOTICE] switch_ivr.c:1343 switch_ivr_session_transfer() Transfer sofia/internal/1001 to XML[118 at default] 2009-07-08 22:03:51 [NOTICE] mod_spidermonkey.c:2994 session_destroy() Hangup sofia/internal/1001 [CS_ROUTING] [NORMAL_CLEARING] 2009-07-08 22:03:51 [NOTICE] switch_core_session.c:970 switch_core_session_thread() Session 35 (sofia/internal/1001) Ended 2009-07-08 22:03:51 [NOTICE] switch_core_session.c:972 switch_core_session_thread() Close Channel sofia/internal/1001 [CS_HANGUP] ================================================== May be I'm on the wrong way, but this is my target: 1) From the external Perl freeswitch interface call a simple js 2) Make this script dial a sip/pstn phone number at the assistance location 3) Connect the just dialled assistance location to the IVR at ext 118 and let the support people hear some info about the raising fault. Thanks in advance for any suggestion. Roberto From anthony.minessale at gmail.com Wed Jul 8 13:29:46 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 8 Jul 2009 15:29:46 -0500 Subject: [Freeswitch-users] Less Than a Month Until ClueCon Message-ID: <191c3a030907081329q53ec3609i37b9ee87a31c9247@mail.gmail.com> Hi everyone, We are less than a month away from ClueCon 2009 and I would like to urge anyone who is considering attending to sign up ASAP to make sure you are properly counted in the food totals and the early bird pricing. Bring your laptops and all your gizmos and get ready to dive into telephony for 3 fun-filled days. We will be having FULL BREAKFAST on the first 2 mornings (continental on day 3) FULL LUNCH, and OPEN BAR for 2 hours the first 2 nights for ALL atendees included in your attendance fee. We also will be giving away several prizes provided by our various sponsors. Every paid registration gives you a chance to win one of many goodies such as phones/tdm cards etc. REGISTER NOW http://www.cluecon.com or CALL (877) 742 -CLUE or INSTALL FreeSWITCH and dial 5000 and choose the "register for cluecon" option on the ivr. or E-MAIL marketing at cluecon.com to discuss sponsoring ot participating in helping out with logistics etc for a reduced attendance fee. -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference 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/20090708/88206d6e/attachment-0002.html From diego.viola at gmail.com Wed Jul 8 13:32:34 2009 From: diego.viola at gmail.com (Diego Viola) Date: Wed, 8 Jul 2009 16:32:34 -0400 Subject: [Freeswitch-users] FreeSWITCH article for OSnews Message-ID: <86a32abc0907081332m3e25c1c7x73d8753e8b1cc616@mail.gmail.com> Hey guys. I wrote the OSnews staff about the possibility to post some FreeSWITCH articles in the OSnews site, as they have published some Asterisk articles before, I thought that it would be nice to post something about FreeSWITCH as well, since it deserves more attention. This is what they said: "I don?t think any of the staff have any knowledge on this topic to write an article, so the best we could do is a page 2 item. As I see it, there?s two things you can do here: 1. Send the item to us using the "Submit News" link at the top of the page, it will appear in the back end for all of us to see and one of the staff may pick it up. 2. Write an article about FreeSWITCH, what it is, and how it differs from Asterisk using your knowledge an submit it in the usual manner, you?re much more likely to get on the front page then. Kind regards, Kroc." So if you guys are interested and want to help me to write a FreeSWITCH article, maybe we could send them so they publish it there. Diego -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090708/3c5073d6/attachment-0002.html From anthony.minessale at gmail.com Wed Jul 8 13:42:57 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 8 Jul 2009 15:42:57 -0500 Subject: [Freeswitch-users] simple originate / bridge js In-Reply-To: <200907082215.57328.paparoga@mailinator.com> References: <200907082215.57328.paparoga@mailinator.com> Message-ID: <191c3a030907081342l3275021k1025900592221e20@mail.gmail.com> you also have to set session.setAutoHangup(0); or it will hangup as soon as it exits the script. On Wed, Jul 8, 2009 at 3:15 PM, wrote: > First of all I apologise for my long post. > > In the meantime I reworked the simple js as suggested (adding the domain or > using transfer instead of bridge) but yet the script doesn't work. > > Let me add some other info. > > I just cloned the '5000' demo_ivr and reworked a little to reach my target. > > Next I added: > > > > > > > > > > > into the default.xml dialplan. > > Now all is ok if I connect Ekiga to FreeSwitch as user 1001 (for example) > and > than I dial '118'. > > The IVR works fine. > > Next from the console: > > originate sofia/my.freeswitch.address/1001 118 > > I get my Ekiga Phone ringing and connecter to the ext. 118 (the ivr) > > That is the log: > ========================================================= > originate sofia/10.0.0.33/1001 118 > > 2009-07-08 21:56:57 [NOTICE] switch_channel.c:567 switch_channel_set_name() > New Channel sofia/internal/1001 [7e660b20-6bf9-11de-b2fa-f3963e050c84] > 2009-07-08 21:56:57 [NOTICE] sofia.c:2782 sofia_handle_sip_i_state() > Ring-Ready sofia/internal/1001! > 2009-07-08 21:57:00 [NOTICE] sofia.c:3220 sofia_handle_sip_i_state() > Channel > [sofia/internal/1001] has been answered > 2009-07-08 21:57:00 [NOTICE] switch_ivr.c:1343 > switch_ivr_session_transfer() > Transfer sofia/internal/1001 to XML[118 at default] > API CALL [originate(sofia/10.0.0.33/1001 118)] output: > +OK 7e660b20-6bf9-11de-b2fa-f3963e050c84 > > freeswitch at Linux61> 2009-07-08 21:57:00 [INFO] mod_dialplan_xml.c:233 > dialplan_hunt() Processing FreeSWITCH->118 in context default > 2009-07-08 21:57:02 [WARNING] switch_core_file.c:119 > switch_core_perform_file_open() Sample rate doesn't match > 2009-07-08 21:57:15 [WARNING] switch_core_file.c:119 > switch_core_perform_file_open() Sample rate doesn't match > 2009-07-08 21:57:21 [NOTICE] sofia.c:3279 sofia_handle_sip_i_state() Hangup > sofia/internal/1001 [CS_EXECUTE] [NORMAL_CLEARING] > 2009-07-08 21:57:25 [NOTICE] switch_core_session.c:970 > switch_core_session_thread() Session 33 (sofia/internal/1001) Ended > 2009-07-08 21:57:25 [NOTICE] switch_core_session.c:972 > switch_core_session_thread() Close Channel sofia/internal/1001 [CS_HANGUP] > ==================================================== > > Using the js instead it looks like if the bridge commands looks for a > REGISTERED user at ext. 118, and so it fails. > > Using the transfer option, as suggested, the Ekiga user (1001) get to be > connected to the ext 118, bur the connection drops immediately after. > > ==================================================== > freeswitch at Linux61> jsrun alarm.js > API CALL [jsrun(alarm.js)] output: > OK > > freeswitch at Linux61> 2009-07-08 22:03:47 [NOTICE] switch_channel.c:567 > switch_channel_set_name() New Channel sofia/internal/1001 > [7267f936-6bfa-11de-b2fa-f3963e050c84] > 2009-07-08 22:03:47 [NOTICE] sofia.c:2782 sofia_handle_sip_i_state() > Ring-Ready sofia/internal/1001! > 2009-07-08 22:03:51 [NOTICE] sofia.c:3220 sofia_handle_sip_i_state() > Channel > [sofia/internal/1001] has been answered > 2009-07-08 22:03:51 [NOTICE] switch_ivr.c:1343 > switch_ivr_session_transfer() > Transfer sofia/internal/1001 to XML[118 at default] > 2009-07-08 22:03:51 [NOTICE] mod_spidermonkey.c:2994 session_destroy() > Hangup > sofia/internal/1001 [CS_ROUTING] [NORMAL_CLEARING] > 2009-07-08 22:03:51 [NOTICE] switch_core_session.c:970 > switch_core_session_thread() Session 35 (sofia/internal/1001) Ended > 2009-07-08 22:03:51 [NOTICE] switch_core_session.c:972 > switch_core_session_thread() Close Channel sofia/internal/1001 [CS_HANGUP] > ================================================== > > May be I'm on the wrong way, but this is my target: > > 1) From the external Perl freeswitch interface call a simple js > 2) Make this script dial a sip/pstn phone number at the assistance location > 3) Connect the just dialled assistance location to the IVR at ext 118 and > let > the support people hear some info about the raising fault. > > Thanks in advance for any suggestion. > > > > Roberto > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference 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/20090708/6d62c035/attachment-0002.html From larclap at yahoo.com Wed Jul 8 14:37:51 2009 From: larclap at yahoo.com (Lars Zeb) Date: Wed, 8 Jul 2009 14:37:51 -0700 Subject: [Freeswitch-users] Error in my dialplan Message-ID: <00ff01ca0014$58a20700$09e61500$@com> I receive an error on an inbound call from my dialplan. I don't have a clue what it means. Can someone help? from log: 2009-07-08 09:54:50.172590 [DEBUG] switch_core_state_machine.c:78 sofia/external/+13105551212 at 66.53.188.187 Standard ROUTING 2009-07-08 09:54:50.172590 [INFO] mod_dialplan_xml.c:310 Processing +13105551212->1000 in context default .... Dialplan: sofia/external/+13105551212 at 66.53.188.187 parsing [default->Local_Extension_Lars] continue=false Dialplan: sofia/external/+13105551212 at 66.53.188.187 Regex (PASS) [Local_Extension_Lars] destination_number(1000) =~ /^(100[0-9])$/ break=on-false Dialplan: sofia/external/+13105551212 at 66.53.188.187 Action set(dialed_ext=1000) Dialplan: sofia/external/+13105551212 at 66.53.188.187 Action export(dialed_ext=1000) 2009-07-08 09:54:50.173586 [ERR] switch_regex.c:101 COMPILE ERROR: 1 [nothing to repeat][^+13105551212$] Dialplan: sofia/external/+13105551212 at 66.53.188.187 Regex (FAIL) [Local_Extension_Lars] destination_number(1000) =~ /^+13105551212$/ break=on-false Dialplan: sofia/external/+13104647614 at 66.53.188.187 ANTI-Action bind_meta_app(1 b s execute_extension::dx XML features) Dialplan: sofia/external/+13104647614 at 66.53.188.187 ANTI-Action bind_meta_app(2 b s record_session::/usr/local/freeswitch/recordings/${caller_id_number}.${strft ime(%Y-%m-%d-%H-%M-%S)}.wav) .... Caller-Dialplan: [XML] Caller-Caller-ID-Name: [+13105551212] Caller-Caller-ID-Number: [+13105551212] My dialplan: ... .... Thanks, Lars -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090708/037e29a1/attachment-0002.html From brian at freeswitch.org Wed Jul 8 14:46:44 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 8 Jul 2009 16:46:44 -0500 Subject: [Freeswitch-users] Error in my dialplan In-Reply-To: <00ff01ca0014$58a20700$09e61500$@com> References: <00ff01ca0014$58a20700$09e61500$@com> Message-ID: <92324E8F-B612-4328-8F30-197562963EB1@freeswitch.org> You have to escape the + with \+ /b On Jul 8, 2009, at 4:37 PM, Lars Zeb wrote: > 2009-07-08 09:54:50.173586 [ERR] switch_regex.c:101 COMPILE ERROR: 1 > [nothing to repeat][^+13105551212$] -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090708/b9154a63/attachment-0002.html From rupa at rupa.com Wed Jul 8 14:48:42 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Wed, 8 Jul 2009 16:48:42 -0500 Subject: [Freeswitch-users] Error in my dialplan In-Reply-To: <92324E8F-B612-4328-8F30-197562963EB1@freeswitch.org> References: <00ff01ca0014$58a20700$09e61500$@com> <92324E8F-B612-4328-8F30-197562963EB1@freeswitch.org> Message-ID: the problem is the + is coming from the network... On Wed, Jul 8, 2009 at 4:46 PM, Brian West wrote: > You have to escape the + with \+ > /b > > On Jul 8, 2009, at 4:37 PM, Lars Zeb wrote: > > 2009-07-08 09:54:50.173586 [ERR] switch_regex.c:101 COMPILE ERROR: 1 > [nothing to repeat][^+13105551212$] > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090708/6b1f8a80/attachment-0002.html From brian at freeswitch.org Wed Jul 8 14:50:53 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 8 Jul 2009 16:50:53 -0500 Subject: [Freeswitch-users] Error in my dialplan In-Reply-To: References: <00ff01ca0014$58a20700$09e61500$@com> <92324E8F-B612-4328-8F30-197562963EB1@freeswitch.org> Message-ID: Ah yes this line We have since removed that ability to login without password from the default configs. /b On Jul 8, 2009, at 4:48 PM, Rupa Schomaker wrote: > the problem is the + is coming from the network... > > On Wed, Jul 8, 2009 at 4:46 PM, Brian West > wrote: > You have to escape the + with \+ > > /b > > On Jul 8, 2009, at 4:37 PM, Lars Zeb wrote: > >> 2009-07-08 09:54:50.173586 [ERR] switch_regex.c:101 COMPILE ERROR: >> 1 [nothing to repeat][^+13105551212$] > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > -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 From larclap at yahoo.com Wed Jul 8 15:14:37 2009 From: larclap at yahoo.com (Lars Zeb) Date: Wed, 8 Jul 2009 15:14:37 -0700 Subject: [Freeswitch-users] Error in my dialplan In-Reply-To: References: <00ff01ca0014$58a20700$09e61500$@com> <92324E8F-B612-4328-8F30-197562963EB1@freeswitch.org> Message-ID: <011901ca0019$7b2f72d0$718e5870$@com> Still lost. What is the solution? 1) Remove the ability to login without password (and the comparison between destination_number and ${caller_id_number}, 2) Create a condition which strips the + sign and creates a new variable like caller_id_number, or 3) ??? > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch- > users-bounces at lists.freeswitch.org] On Behalf Of Brian West > Sent: Wednesday, July 08, 2009 2:51 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Error in my dialplan > > Ah yes this line > > We have since removed that ability to login without password from the > default configs. > > /b > > On Jul 8, 2009, at 4:48 PM, Rupa Schomaker wrote: > > > the problem is the + is coming from the network... > > > > On Wed, Jul 8, 2009 at 4:46 PM, Brian West > > wrote: > > You have to escape the + with \+ > > > > /b > > > > On Jul 8, 2009, at 4:37 PM, Lars Zeb wrote: > > > >> 2009-07-08 09:54:50.173586 [ERR] switch_regex.c:101 COMPILE ERROR: > >> 1 [nothing to repeat][^+13105551212$] > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > > http://www.freeswitch.org > > > > > > > > > > -- > > -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 > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From rupa at rupa.com Wed Jul 8 15:37:19 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Wed, 8 Jul 2009 17:37:19 -0500 Subject: [Freeswitch-users] Error in my dialplan In-Reply-To: <011901ca0019$7b2f72d0$718e5870$@com> References: <00ff01ca0014$58a20700$09e61500$@com> <92324E8F-B612-4328-8F30-197562963EB1@freeswitch.org> <011901ca0019$7b2f72d0$718e5870$@com> Message-ID: Your choice, but I'd suggest 1 -- you don't want to trust the network callerid for authenticating voicemail access. On Wed, Jul 8, 2009 at 5:14 PM, Lars Zeb wrote: > Still lost. What is the solution? 1) Remove the ability to login without > password (and the comparison between destination_number and > ${caller_id_number}, 2) Create a condition which strips the + sign and > creates a new variable like caller_id_number, or 3) ??? > > > -----Original Message----- > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch- > > users-bounces at lists.freeswitch.org] On Behalf Of Brian West > > Sent: Wednesday, July 08, 2009 2:51 PM > > To: freeswitch-users at lists.freeswitch.org > > Subject: Re: [Freeswitch-users] Error in my dialplan > > > > Ah yes this line > > > > We have since removed that ability to login without password from the > > default configs. > > > > /b > > > > On Jul 8, 2009, at 4:48 PM, Rupa Schomaker wrote: > > > > > the problem is the + is coming from the network... > > > > > > On Wed, Jul 8, 2009 at 4:46 PM, Brian West > > > wrote: > > > You have to escape the + with \+ > > > > > > /b > > > > > > On Jul 8, 2009, at 4:37 PM, Lars Zeb wrote: > > > > > >> 2009-07-08 09:54:50.173586 [ERR] switch_regex.c:101 COMPILE ERROR: > > >> 1 [nothing to repeat][^+13105551212$] > > > > > > > > > _______________________________________________ > > > Freeswitch-users mailing list > > > Freeswitch-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > > users > > > http://www.freeswitch.org > > > > > > > > > > > > > > > -- > > > -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 > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > > users > > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090708/74dedc87/attachment-0002.html From brian at freeswitch.org Wed Jul 8 15:47:19 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 8 Jul 2009 17:47:19 -0500 Subject: [Freeswitch-users] Error in my dialplan In-Reply-To: References: <00ff01ca0014$58a20700$09e61500$@com> <92324E8F-B612-4328-8F30-197562963EB1@freeswitch.org> <011901ca0019$7b2f72d0$718e5870$@com> Message-ID: <45A1B0D7-5018-4989-9ED5-64E738EB1755@freeswitch.org> Hence the reason I removed it from the defaults. :p /b On Jul 8, 2009, at 5:37 PM, Rupa Schomaker wrote: > Your choice, but I'd suggest 1 -- you don't want to trust the > network callerid for authenticating voicemail access. From msc at freeswitch.org Wed Jul 8 16:31:31 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 8 Jul 2009 16:31:31 -0700 Subject: [Freeswitch-users] FreeSWITCH article for OSnews In-Reply-To: <86a32abc0907081332m3e25c1c7x73d8753e8b1cc616@mail.gmail.com> References: <86a32abc0907081332m3e25c1c7x73d8753e8b1cc616@mail.gmail.com> Message-ID: <87f2f3b90907081631x563dc073n10235c5a0f24c81c@mail.gmail.com> Thanks for taking the initiative. I'll catch up with you offline and we'll work up a gameplan. -MC On Wed, Jul 8, 2009 at 1:32 PM, Diego Viola wrote: > Hey guys. > > I wrote the OSnews staff about the possibility to post some FreeSWITCH > articles in the OSnews site, as they have published some Asterisk articles > before, I thought that it would be nice to post something about FreeSWITCH > as well, since it deserves more attention. > > This is what they said: > > "I don?t think any of the staff have any knowledge on this topic to write > an article, so the best we could do is a page 2 item. As I see it, there?s > two things you can do here: > > 1. Send the item to us using the "Submit News" link at the top of the page, > it will appear in the back end for all of us to see and one of the staff may > pick it up. > > 2. Write an article about FreeSWITCH, what it is, and how it differs from > Asterisk using your knowledge an submit it in the usual manner, you?re much > more likely to get on the front page then. > > Kind regards, > Kroc." > > So if you guys are interested and want to help me to write a FreeSWITCH > article, maybe we could send them so they publish it there. > > Diego > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090708/10ef69eb/attachment-0002.html From kees at mroffice.org Wed Jul 8 18:21:03 2009 From: kees at mroffice.org (Kees Varekamp) Date: Thu, 9 Jul 2009 13:21:03 +1200 Subject: [Freeswitch-users] play file and stop play file on 2 channels Message-ID: <98d38dcf0907081821x797dc2b9q7ff9f1a9e64a34af@mail.gmail.com> Hi there, I am trying to play a file to both channels of a bridged conversation through a socket. This can be done with: api uuid_broadcast both However, I would like to be able to stop playing the file as well. This can be done with: api uuid_displace start 3600 mux and api uuid_displace stop But that one seems to work on only one channel. The other channel can mix its output but can not hear the file. Are there any other options? Thanks, Kees -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090709/d9b834cc/attachment-0002.html From mashudiflexi at telkom.co.id Wed Jul 8 19:35:04 2009 From: mashudiflexi at telkom.co.id (mashudi) Date: Thu, 09 Jul 2009 09:35:04 +0700 Subject: [Freeswitch-users] FreeSwitch & Sangoma Q.SIG Message-ID: <4A555758.9000006@telkom.co.id> Dear All, do FreeSwitch and Sangoma card can support Q.SIG signalling protocol? thank you in advance. best regards, mashudi ==================================== Mau GRATIS TELPON LOKAL, DISCOUNT 50% SMS, DISCOUNT 20% SLJJ, dan DISCOUNT FLEXI MILIS? Ikuti Dahsyatnya FLEXI KOMUNITAS. Ketik CREATE[NAMA GRUP], sms ke 345. Contoh: CREATE SMU2, sms ke 345. Informasi selanjutnya: - hubungi 147 - http://www.telkomflexi.com - ketik INFO, sms ke 345. From krivushinme at rn-inform.tomsk.ru Thu Jul 9 00:41:26 2009 From: krivushinme at rn-inform.tomsk.ru (=?utf-8?b?0JrRgNC40LLRg9GI0LjQvSDQnNC40YXQsNC40Ls=?=) Date: Thu, 9 Jul 2009 14:41:26 +0700 Subject: [Freeswitch-users] Conference, ask to unmute Message-ID: <200907091441.26456.krivushinme@rn-inform.tomsk.ru> Hello! Is any ability to ask to unmute in conference? -- ? ?????????, ???????? ?????? ??????? ?????????? ?????? ????????????????, ??? "??-??????" ?????? ? ?.??????, ?. ????? ???. +7 913 865 78 66 icq: 218 744 127 xmpp: KrivushinME at jabber.ru mail: KrivushinME at rn-inform.tomsk.ru ?????????? ????: 86 099 192726 From krivushinme at rn-inform.tomsk.ru Thu Jul 9 02:25:26 2009 From: krivushinme at rn-inform.tomsk.ru (=?utf-8?b?0JrRgNC40LLRg9GI0LjQvSDQnNC40YXQsNC40Ls=?=) Date: Thu, 9 Jul 2009 16:25:26 +0700 Subject: [Freeswitch-users] Conference, ask to unmute Message-ID: <200907091625.26630.krivushinme@rn-inform.tomsk.ru> Hello! Is any ability to ask to unmute in conference? -- ? ?????????, ???????? ?????? ??????? ?????????? ?????? ????????????????, ??? "??-??????" ?????? ? ?.??????, ?. ????? ???. +7 913 865 78 66 icq: 218 744 127 xmpp: KrivushinME at jabber.ru mail: KrivushinME at rn-inform.tomsk.ru ?????????? ????: 86 099 192726 From anthony.minessale at gmail.com Thu Jul 9 06:55:03 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 9 Jul 2009 08:55:03 -0500 Subject: [Freeswitch-users] play file and stop play file on 2 channels In-Reply-To: <98d38dcf0907081821x797dc2b9q7ff9f1a9e64a34af@mail.gmail.com> References: <98d38dcf0907081821x797dc2b9q7ff9f1a9e64a34af@mail.gmail.com> Message-ID: <191c3a030907090655x574907d7td2f0403e7408a66f@mail.gmail.com> you can use the first way then send the "break" api command to each leg individually On Wed, Jul 8, 2009 at 8:21 PM, Kees Varekamp wrote: > Hi there, > > I am trying to play a file to both channels of a bridged conversation > through a socket. This can be done with: > > api uuid_broadcast both > > However, I would like to be able to stop playing the file as well. This can > be done with: > > api uuid_displace start 3600 mux > > and > > api uuid_displace stop > > But that one seems to work on only one channel. The other channel can mix > its output but can not hear the file. Are there any other options? > > Thanks, > > Kees > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference 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/20090709/43d0f2e2/attachment-0002.html From larclap at yahoo.com Thu Jul 9 07:08:31 2009 From: larclap at yahoo.com (Lars Zeb) Date: Thu, 9 Jul 2009 07:08:31 -0700 Subject: [Freeswitch-users] Documentation error? Message-ID: <003801ca009e$bda23f00$38e6bd00$@com> In http://wiki.freeswitch.org/wiki/Mod_cidlookup, at the third line after CLI, it states: which would the options from your config file: Should this be?: which would use the options from your config file: Thanks, Lars -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090709/7f38892b/attachment-0002.html From brian at freeswitch.org Thu Jul 9 07:11:42 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 9 Jul 2009 09:11:42 -0500 Subject: [Freeswitch-users] Documentation error? In-Reply-To: <003801ca009e$bda23f00$38e6bd00$@com> References: <003801ca009e$bda23f00$38e6bd00$@com> Message-ID: <9CB32C44-9C6D-4D42-8ED4-FD395A761968@freeswitch.org> If it makes better sense to you then please login and fix it! ;) /b On Jul 9, 2009, at 9:08 AM, Lars Zeb wrote: > In http://wiki.freeswitch.org/wiki/Mod_cidlookup, at the third line > after CLI, it states: > > which would the options from your config file: > > Should this be?: > > which would use the options from your config file: > > Thanks, Lars -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090709/3b8f0c5c/attachment-0002.html From larclap at yahoo.com Thu Jul 9 07:43:23 2009 From: larclap at yahoo.com (Lars Zeb) Date: Thu, 9 Jul 2009 07:43:23 -0700 Subject: [Freeswitch-users] Documentation error? In-Reply-To: <9CB32C44-9C6D-4D42-8ED4-FD395A761968@freeswitch.org> References: <003801ca009e$bda23f00$38e6bd00$@com> <9CB32C44-9C6D-4D42-8ED4-FD395A761968@freeswitch.org> Message-ID: <004901ca00a3$9c653c70$d52fb550$@com> Done. Not knowing much, I'm reluctant to make changes without asking. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Thursday, July 09, 2009 7:12 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Documentation error? If it makes better sense to you then please login and fix it! ;) /b On Jul 9, 2009, at 9:08 AM, Lars Zeb wrote: In http://wiki.freeswitch.org/wiki/Mod_cidlookup, at the third line after CLI, it states: which would the options from your config file: Should this be?: which would use the options from your config file: Thanks, Lars -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090709/2df62404/attachment-0002.html From msc at freeswitch.org Thu Jul 9 08:20:38 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 9 Jul 2009 08:20:38 -0700 Subject: [Freeswitch-users] Conference, ask to unmute In-Reply-To: <200907091441.26456.krivushinme@rn-inform.tomsk.ru> References: <200907091441.26456.krivushinme@rn-inform.tomsk.ru> Message-ID: <87f2f3b90907090820g2edcc38oa2a1a25c2c982b64@mail.gmail.com> 2009/7/9 ???????? ?????? > Hello! > > Is any ability to ask to unmute in conference? > Not sure if I understand the question. Are you talking about the caller pressing zero to mute/unmute? -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090709/934bb104/attachment-0002.html From mike at jerris.com Thu Jul 9 08:28:45 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 9 Jul 2009 11:28:45 -0400 Subject: [Freeswitch-users] Conference, ask to unmute In-Reply-To: <87f2f3b90907090820g2edcc38oa2a1a25c2c982b64@mail.gmail.com> References: <200907091441.26456.krivushinme@rn-inform.tomsk.ru> <87f2f3b90907090820g2edcc38oa2a1a25c2c982b64@mail.gmail.com> Message-ID: mute/unmute is a toggle. Mike On Jul 9, 2009, at 11:20 AM, Michael Collins wrote: > > > 2009/7/9 ???????? ?????? inform.tomsk.ru> > Hello! > > Is any ability to ask to unmute in conference? > > Not sure if I understand the question. Are you talking about the > caller pressing zero to mute/unmute? > > -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/20090709/8404bee1/attachment-0002.html From krivushinme at rn-inform.tomsk.ru Thu Jul 9 08:47:10 2009 From: krivushinme at rn-inform.tomsk.ru (=?utf-8?b?0JrRgNC40LLRg9GI0LjQvSDQnNC40YXQsNC40Ls=?=) Date: Thu, 9 Jul 2009 22:47:10 +0700 Subject: [Freeswitch-users] Conference, ask to unmute In-Reply-To: <87f2f3b90907090820g2edcc38oa2a1a25c2c982b64@mail.gmail.com> References: <200907091441.26456.krivushinme@rn-inform.tomsk.ru> <87f2f3b90907090820g2edcc38oa2a1a25c2c982b64@mail.gmail.com> Message-ID: <200907092247.10336.krivushinme@rn-inform.tomsk.ru> My guys want to work with operator - I wrote an WEB-face for conferencing. And he wants to mute all participants, and give voice by order. May be say caller_id on press any button to all conference. I think to make execute_application + say caller_id. I will try to introspect channel vars in "exec_app" in conference context tomorrow. Finally - we always has esl to make anything we want! : ) -- ? ?????????, ???????? ?????? ??????? ?????????? ?????? ????????????????, ??? "??-??????" ?????? ? ?.??????, ?. ????? ???. +7 913 865 78 66 icq: 218 744 127 xmpp: KrivushinME at jabber.ru mail: KrivushinME at rn-inform.tomsk.ru ?????????? ????: 86 099 192726 From rupa at rupa.com Thu Jul 9 08:56:23 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Thu, 9 Jul 2009 10:56:23 -0500 Subject: [Freeswitch-users] Documentation error? In-Reply-To: <004901ca00a3$9c653c70$d52fb550$@com> References: <003801ca009e$bda23f00$38e6bd00$@com> <9CB32C44-9C6D-4D42-8ED4-FD395A761968@freeswitch.org> <004901ca00a3$9c653c70$d52fb550$@com> Message-ID: Thanks. At least a few of us monitor every change to the wiki. I especially monitor stuff related to my modules. So... feel free to make changes, we can always roll back to an earlier version of the page or clarify further. The wiki is a community resource, everyone should feel like they can make (reasonable) changes to it. On Thu, Jul 9, 2009 at 9:43 AM, Lars Zeb wrote: > Done. > > > > Not knowing much, I?m reluctant to make changes without asking. > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Brian West > *Sent:* Thursday, July 09, 2009 7:12 AM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Documentation error? > > > > If it makes better sense to you then please login and fix it! ;) > > > > /b > > > > On Jul 9, 2009, at 9:08 AM, Lars Zeb wrote: > > > > In http://wiki.freeswitch.org/wiki/Mod_cidlookup, at the third line after > CLI, it states: > > > > which would the options from your config file: > > > > Should this be?: > > > > which would use the options from your config file: > > > > Thanks, Lars > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090709/549f01c5/attachment-0002.html From anthony.minessale at gmail.com Thu Jul 9 09:01:29 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 9 Jul 2009 11:01:29 -0500 Subject: [Freeswitch-users] Conference, ask to unmute In-Reply-To: <200907092247.10336.krivushinme@rn-inform.tomsk.ru> References: <200907091441.26456.krivushinme@rn-inform.tomsk.ru> <87f2f3b90907090820g2edcc38oa2a1a25c2c982b64@mail.gmail.com> <200907092247.10336.krivushinme@rn-inform.tomsk.ru> Message-ID: <191c3a030907090901q2c8fdf0at9c1238b54b59bda1@mail.gmail.com> conference xyz mute all On Thu, Jul 9, 2009 at 10:47 AM, ???????? ?????? < krivushinme at rn-inform.tomsk.ru> wrote: > My guys want to work with operator - I wrote an WEB-face for conferencing. > And he wants to mute all participants, and give voice by order. > May be say caller_id on press any button to all conference. > I think to make execute_application + say caller_id. I will try to > introspect > channel vars in "exec_app" in conference context tomorrow. > > Finally - we always has esl to make anything we want! : ) > > -- > ? ?????????, ???????? ?????? > ??????? ?????????? ?????? ????????????????, > ??? "??-??????" ?????? ? ?.??????, > ?. ????? ???. +7 913 865 78 66 > icq: 218 744 127 > xmpp: KrivushinME at jabber.ru > mail: KrivushinME at rn-inform.tomsk.ru > ?????????? ????: 86 099 192726 > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference 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/20090709/c2ac7e2c/attachment-0002.html From pjintheusa at gmail.com Thu Jul 9 09:06:33 2009 From: pjintheusa at gmail.com (Phillip Jones) Date: Thu, 9 Jul 2009 12:06:33 -0400 Subject: [Freeswitch-users] mod_say_en directory location Message-ID: <367751820907090906if8493dfoc927b07f8bc86cfd@mail.gmail.com> Hi there, I have a very simply script that speaks back some digits, as so: session:execute("say", "en number iterated 1234"); However, to get this to work successfully I have had to move the 'digits' directory to: C:\Program Files (x86)\Freeswitch\sounds\en from the default: C:\Program Files (x86)\Freeswitch\sounds\en\us\callie\digits\8000 This is a clean install of FreeSWITCH - so I am wondering why I needed to do this, what have not configured correctly? As you can see I am using windows with a resent build (3 days) from svn. Any help appreciated. Thanks Phillip Jones -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090709/0d39366d/attachment-0002.html From msc at freeswitch.org Thu Jul 9 09:07:47 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 9 Jul 2009 09:07:47 -0700 Subject: [Freeswitch-users] Documentation error? In-Reply-To: References: <003801ca009e$bda23f00$38e6bd00$@com> <9CB32C44-9C6D-4D42-8ED4-FD395A761968@freeswitch.org> <004901ca00a3$9c653c70$d52fb550$@com> Message-ID: <87f2f3b90907090907p17aea315ob3d5ade9423c0005@mail.gmail.com> On Thu, Jul 9, 2009 at 8:56 AM, Rupa Schomaker wrote: > Thanks. > > At least a few of us monitor every change to the wiki. I especially > monitor stuff related to my modules. > > So... feel free to make changes, we can always roll back to an earlier > version of the page or clarify further. > > The wiki is a community resource, everyone should feel like they can make > (reasonable) changes to it. Also, several of us check the changes each day so if anything really goofy happens we can always roll it back. Feel free to make changes, especially obvious ones. If you have questions don't hesitate to ask. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090709/a88df4ed/attachment-0002.html From msc at freeswitch.org Thu Jul 9 09:38:31 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 9 Jul 2009 09:38:31 -0700 Subject: [Freeswitch-users] mod_say_en directory location In-Reply-To: <367751820907090906if8493dfoc927b07f8bc86cfd@mail.gmail.com> References: <367751820907090906if8493dfoc927b07f8bc86cfd@mail.gmail.com> Message-ID: <87f2f3b90907090938u53ee6145l825beb30275e1a20@mail.gmail.com> Is the call perhaps at 16kHz and it's looking for non-installed 16kHz sound files? -MC On Thu, Jul 9, 2009 at 9:06 AM, Phillip Jones wrote: > Hi there, > > I have a very simply script that speaks back some digits, as so: > > session:execute("say", "en number iterated 1234"); > > However, to get this to work successfully I have had to move the 'digits' > directory to: > > C:\Program Files (x86)\Freeswitch\sounds\en > > from the default: > > C:\Program Files (x86)\Freeswitch\sounds\en\us\callie\digits\8000 > > > This is a clean install of FreeSWITCH - so I am wondering why I needed to > do this, what have not configured correctly? > > As you can see I am using windows with a resent build (3 days) from svn. > > Any help appreciated. > > Thanks > > > Phillip Jones > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090709/decd2304/attachment-0002.html From brian at freeswitch.org Thu Jul 9 10:19:11 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 9 Jul 2009 12:19:11 -0500 Subject: [Freeswitch-users] Less Than a Month Until ClueCon In-Reply-To: <191c3a030907081329q53ec3609i37b9ee87a31c9247@mail.gmail.com> References: <191c3a030907081329q53ec3609i37b9ee87a31c9247@mail.gmail.com> Message-ID: <09DA50D4-C82E-425D-9CB2-467159B673DB@freeswitch.org> We have info of where the meet and greet will take place here http://www.cluecon.com/node/35 Also info on travel from both O'Hare and Midway will be posted today via taxi or CTA. Thanks, /b On Jul 8, 2009, at 3:29 PM, Anthony Minessale wrote: > Hi everyone, > > We are less than a month away from ClueCon 2009 and I would like to > urge anyone who is considering attending to > sign up ASAP to make sure you are properly counted in the food > totals and the early bird pricing. > Bring your laptops and all your gizmos and get ready to dive into > telephony for 3 fun-filled days. > > We will be having FULL BREAKFAST on the first 2 mornings > (continental on day 3) FULL LUNCH, and OPEN BAR for 2 hours the > first 2 nights for ALL atendees included in your attendance fee. > We also will be giving away several prizes provided by our various > sponsors. Every paid registration gives you a chance to win one of > many goodies such as phones/tdm cards etc. > > REGISTER NOW http://www.cluecon.com > > or CALL (877) 742 -CLUE > or INSTALL FreeSWITCH and dial 5000 and choose the "register for > cluecon" option on the ivr. > or E-MAIL marketing at cluecon.com to discuss sponsoring ot > participating in helping out with logistics etc for a reduced > attendance fee. > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > 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/20090709/1619ce39/attachment-0002.html From msc at freeswitch.org Thu Jul 9 10:47:24 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 9 Jul 2009 10:47:24 -0700 Subject: [Freeswitch-users] Less Than a Month Until ClueCon In-Reply-To: <09DA50D4-C82E-425D-9CB2-467159B673DB@freeswitch.org> References: <191c3a030907081329q53ec3609i37b9ee87a31c9247@mail.gmail.com> <09DA50D4-C82E-425D-9CB2-467159B673DB@freeswitch.org> Message-ID: <87f2f3b90907091047p21c536d5re65acd6c5141ba99@mail.gmail.com> FYI, visit the hotel details page . Scroll to the bottom and there are links both to online maps and directions as well as a handy document that you can download. We have directions for getting to/from the hotel from both airports (O'Hare and Midway) via public transport as well as for those driving into town. -Michael On Thu, Jul 9, 2009 at 10:19 AM, Brian West wrote: > We have info of where the meet and greet will take place here > http://www.cluecon.com/node/35 Also info on travel from both O'Hare and > Midway will be posted today via taxi or CTA. > Thanks, > /b > > On Jul 8, 2009, at 3:29 PM, Anthony Minessale wrote: > > Hi everyone, > > We are less than a month away from ClueCon 2009 and I would like to urge > anyone who is considering attending to > sign up ASAP to make sure you are properly counted in the food totals and > the early bird pricing. > Bring your laptops and all your gizmos and get ready to dive into telephony > for 3 fun-filled days. > > We will be having FULL BREAKFAST on the first 2 mornings (continental on > day 3) FULL LUNCH, and OPEN BAR for 2 hours the first 2 nights for ALL > atendees included in your attendance fee. > We also will be giving away several prizes provided by our various > sponsors. Every paid registration gives you a chance to win one of many > goodies such as phones/tdm cards etc. > > REGISTER NOW http://www.cluecon.com > > or CALL (877) 742 -CLUE > or INSTALL FreeSWITCH and dial 5000 and choose the "register for cluecon" > option on the ivr. > or E-MAIL marketing at cluecon.com to discuss sponsoring ot participating in > helping out with logistics etc for a reduced attendance fee. > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > 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/20090709/0d9cffe0/attachment-0002.html From brian at freeswitch.org Thu Jul 9 10:54:11 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 9 Jul 2009 12:54:11 -0500 Subject: [Freeswitch-users] Less Than a Month Until ClueCon In-Reply-To: <87f2f3b90907091047p21c536d5re65acd6c5141ba99@mail.gmail.com> References: <191c3a030907081329q53ec3609i37b9ee87a31c9247@mail.gmail.com> <09DA50D4-C82E-425D-9CB2-467159B673DB@freeswitch.org> <87f2f3b90907091047p21c536d5re65acd6c5141ba99@mail.gmail.com> Message-ID: Also if you need room share info email Michael or Myself offlist we can help you find a roomie! /b On Jul 9, 2009, at 12:47 PM, Michael Collins wrote: > FYI, visit the hotel details page. Scroll to the bottom and there > are links both to online maps and directions as well as a handy > document that you can download. We have directions for getting to/ > from the hotel from both airports (O'Hare and Midway) via public > transport as well as for those driving into town. > > -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090709/908db0cf/attachment-0002.html From pjintheusa at gmail.com Thu Jul 9 11:46:10 2009 From: pjintheusa at gmail.com (Phillip Jones) Date: Thu, 9 Jul 2009 14:46:10 -0400 Subject: [Freeswitch-users] mod_say_en directory location In-Reply-To: <87f2f3b90907090938u53ee6145l825beb30275e1a20@mail.gmail.com> References: <367751820907090906if8493dfoc927b07f8bc86cfd@mail.gmail.com> <87f2f3b90907090938u53ee6145l825beb30275e1a20@mail.gmail.com> Message-ID: <367751820907091146y61b78b4v2b7525afe38c43a5@mail.gmail.com> Thanks for the response. I don't think so - the trace states: [ERR] mod_sndfile.c: 192 Error Opening File [C:\Program Files (x86)\Freeswitch\sounds\en\digits/1.wav] [System error : The system cannot find the path specified.] I created a 16000 directory to see whether that would help, and it did not. My C:\Program Files (x86)\Freeswitch\conf\lang\en\en.xml contains: Am I correct in thinking this is where the sound file dir for digits would be specified? On Thu, Jul 9, 2009 at 12:38 PM, Michael Collins wrote: > Is the call perhaps at 16kHz and it's looking for non-installed 16kHz sound > files? > -MC > > On Thu, Jul 9, 2009 at 9:06 AM, Phillip Jones wrote: > >> Hi there, >> >> I have a very simply script that speaks back some digits, as so: >> >> session:execute("say", "en number iterated 1234"); >> >> However, to get this to work successfully I have had to move the 'digits' >> directory to: >> >> C:\Program Files (x86)\Freeswitch\sounds\en >> >> from the default: >> >> C:\Program Files (x86)\Freeswitch\sounds\en\us\callie\digits\8000 >> >> >> This is a clean install of FreeSWITCH - so I am wondering why I needed to >> do this, what have not configured correctly? >> >> As you can see I am using windows with a resent build (3 days) from svn. >> >> Any help appreciated. >> >> Thanks >> >> >> Phillip Jones >> >> >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090709/1c8bce4c/attachment-0002.html From pjintheusa at gmail.com Thu Jul 9 13:38:22 2009 From: pjintheusa at gmail.com (Phillip Jones) Date: Thu, 9 Jul 2009 16:38:22 -0400 Subject: [Freeswitch-users] mod_say_en directory location In-Reply-To: <367751820907091146y61b78b4v2b7525afe38c43a5@mail.gmail.com> References: <367751820907090906if8493dfoc927b07f8bc86cfd@mail.gmail.com> <87f2f3b90907090938u53ee6145l825beb30275e1a20@mail.gmail.com> <367751820907091146y61b78b4v2b7525afe38c43a5@mail.gmail.com> Message-ID: <367751820907091338n2e5f8312r50a743ec6d18ec26@mail.gmail.com> Ok - forget this one - I did a fresh install from the pre-compiled windows binary/msi - referenced on the wiki - and every thing is working as it should be. Thanks On Thu, Jul 9, 2009 at 2:46 PM, Phillip Jones wrote: > Thanks for the response. > > I don't think so - the trace states: > > [ERR] mod_sndfile.c: 192 Error Opening File [C:\Program Files > (x86)\Freeswitch\sounds\en\digits/1.wav] [System error : The system cannot > find the path specified.] > > I created a 16000 directory to see whether that would help, and it did not. > > My C:\Program Files (x86)\Freeswitch\conf\lang\en\en.xml contains: > > > tts-engine="cepstral" tts-voice="callie"> > > > > > > > Am I correct in thinking this is where the sound file dir for digits would > be specified? > > > > On Thu, Jul 9, 2009 at 12:38 PM, Michael Collins wrote: > >> Is the call perhaps at 16kHz and it's looking for non-installed 16kHz >> sound files? >> -MC >> >> On Thu, Jul 9, 2009 at 9:06 AM, Phillip Jones wrote: >> >>> Hi there, >>> >>> I have a very simply script that speaks back some digits, as so: >>> >>> session:execute("say", "en number iterated 1234"); >>> >>> However, to get this to work successfully I have had to move the 'digits' >>> directory to: >>> >>> C:\Program Files (x86)\Freeswitch\sounds\en >>> >>> from the default: >>> >>> C:\Program Files (x86)\Freeswitch\sounds\en\us\callie\digits\8000 >>> >>> >>> This is a clean install of FreeSWITCH - so I am wondering why I needed to >>> do this, what have not configured correctly? >>> >>> As you can see I am using windows with a resent build (3 days) from svn. >>> >>> Any help appreciated. >>> >>> Thanks >>> >>> >>> Phillip Jones >>> >>> >>> >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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/20090709/e2eddb63/attachment-0002.html From tayeb.meftah at gmail.com Thu Jul 9 16:58:22 2009 From: tayeb.meftah at gmail.com (Meftah Tayeb) Date: Thu, 09 Jul 2009 23:58:22 +0000 Subject: [Freeswitch-users] Skypiax Parameters Informations Request Message-ID: <4A56841E.4080206@gmail.com> hello, i have the folowing parameter in Skypiax.conf.xml: each call that will to by routed to this destination?? Each Call will to by routed to this destination? each codecs that is pocible to use it with Skypiax? all? speex? this codecs is used beetwan skypiax and the remote peer? thanks __________ Information from ESET NOD32 Antivirus, version of virus signature database 4229 (20090709) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com From andrew at hijacked.us Thu Jul 9 17:21:05 2009 From: andrew at hijacked.us (Andrew Thompson) Date: Thu, 9 Jul 2009 20:21:05 -0400 Subject: [Freeswitch-users] mod_snom dialplan demo not working Message-ID: <20090710002104.GE28401@hijacked.us> I'm trying to replace our aging nortel system with a FreeSWITCH based system using snom 3[267]0 phones. I've been doing okay so far but I'm running into a brick wall when I try to run the mod_snom demo in the dialplan. I've setup the 'line 2' function key to type button with the number being 'message' like it says on the snom wiki and when I dial extension 9000 the button lights up, but pressing the button does absolutely nothing. I can't figure out what I'm missing here. Any advice? Andrew From brian at freeswitch.org Thu Jul 9 18:10:41 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 9 Jul 2009 20:10:41 -0500 Subject: [Freeswitch-users] mod_snom dialplan demo not working In-Reply-To: <20090710002104.GE28401@hijacked.us> References: <20090710002104.GE28401@hijacked.us> Message-ID: <96EB3A77-7230-4EFE-9541-8A0AC747DBB8@freeswitch.org> It needs tons of work its not so demo tastic ... /b On Jul 9, 2009, at 7:21 PM, Andrew Thompson wrote: > I'm trying to replace our aging nortel system with a FreeSWITCH based > system using snom 3[267]0 phones. I've been doing okay so far but I'm > running into a brick wall when I try to run the mod_snom demo in the > dialplan. I've setup the 'line 2' function key to type button with the > number being 'message' like it says on the snom wiki and when I dial > extension 9000 the button lights up, but pressing the button does > absolutely nothing. I can't figure out what I'm missing here. Any > advice? > > 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 andrew at hijacked.us Thu Jul 9 19:00:28 2009 From: andrew at hijacked.us (Andrew Thompson) Date: Thu, 9 Jul 2009 22:00:28 -0400 Subject: [Freeswitch-users] mod_snom dialplan demo not working In-Reply-To: <96EB3A77-7230-4EFE-9541-8A0AC747DBB8@freeswitch.org> References: <20090710002104.GE28401@hijacked.us> <96EB3A77-7230-4EFE-9541-8A0AC747DBB8@freeswitch.org> Message-ID: <20090710020027.GF28401@hijacked.us> On Thu, Jul 09, 2009 at 08:10:41PM -0500, Brian West wrote: > It needs tons of work its not so demo tastic ... > Is it incomplete or just there's no documentation on how to do it? From the sip trace it looks like FreeSWITCH is sending the phone the right stuff just when I hit the button that was programmed nothing happens. I want to be able to *use* this feature of the snom phones, not just play with the demo anyway, so if there's some development needed to make it work I'd be happy to take a stab at it. Andrew From shaheryarkh at googlemail.com Thu Jul 9 22:40:00 2009 From: shaheryarkh at googlemail.com (Muhammad Shahzad) Date: Fri, 10 Jul 2009 11:40:00 +0600 Subject: [Freeswitch-users] Skypiax Parameters Informations Request In-Reply-To: <4A56841E.4080206@gmail.com> References: <4A56841E.4080206@gmail.com> Message-ID: Destination parameter actually specifies the extension on which this Skype user is reachable within FreeSWITCH dialplan for incoming calls. If this parameter is specified in per_interface_settings xml tag then it will override the value of this parameter in global_settings xml tag, otherwise value of this parameter from global_settings xml tag will be used. Here is an example (see below), the user test.01 is reachable on dialplan extension 2000 (since it has its own destination defined in per_interface_settings xml tag), whereas test.02 is reachable on dialplan extension 5000 (since it does not have destination parameter defined and thus it will use value for this parameter in global_settings xml tag). Now the codec, Skype has its own proprietory code for Skype to Skype calls. The codec we specify in Skypiax configuration file is actually used for Skype to/from non-Skype calls. Consider following dial plan example (with skypiax configuration given above), If a remote Skype user dials test.01 from his/her Skype client, then FreeSWITCH will route this call to SIP user 1000 and codecs specified in Skypiax configuration will be offered to destination SIP endpoint (SIP user 1000 in this case). Thank you. On Fri, Jul 10, 2009 at 5:58 AM, Meftah Tayeb wrote: > hello, > i have the folowing parameter in Skypiax.conf.xml: > > > > each call that will to by routed to this destination?? > > > > > > Each Call will to by routed to this destination? > > each codecs that is pocible to use it with Skypiax? all? speex? > this codecs is used beetwan skypiax and the remote peer? > thanks > > > __________ Information from ESET NOD32 Antivirus, version of virus > signature database 4229 (20090709) __________ > > The message was checked by ESET NOD32 Antivirus. > > http://www.eset.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 > -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090710/db9c354a/attachment-0002.html From shaheryarkh at googlemail.com Thu Jul 9 22:55:17 2009 From: shaheryarkh at googlemail.com (Muhammad Shahzad) Date: Fri, 10 Jul 2009 11:55:17 +0600 Subject: [Freeswitch-users] Interactive Connectivity Establishment (ICE) support in FS Message-ID: Hi, Do we have ICE support in FreeSWITCH. If so, any module as example that is using it? If not then i would like to write one for my mod_msn module, do we have any FS API that i would need to implement in this case? Thank you. -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090710/10ceea11/attachment-0002.html From tayeb.meftah at gmail.com Fri Jul 10 02:48:27 2009 From: tayeb.meftah at gmail.com (Meftah Tayeb) Date: Fri, 10 Jul 2009 09:48:27 +0000 Subject: [Freeswitch-users] Skypiax Parameters Informations Request In-Reply-To: References: <4A56841E.4080206@gmail.com> Message-ID: <4A570E6B.1000106@gmail.com> hello Muhammad , thank you what about hig cality audio codec to use? speex is good? thanks Muhammad Shahzad wrote: > Destination parameter actually specifies the extension on which this > Skype user is reachable within FreeSWITCH dialplan for incoming calls. > > If this parameter is specified in per_interface_settings xml tag then > it will override the value of this parameter in global_settings xml > tag, otherwise value of this parameter from global_settings xml tag > will be used. > > Here is an example (see below), the user test.01 is reachable on > dialplan extension 2000 (since it has its own destination defined in > per_interface_settings xml tag), whereas test.02 is reachable on > dialplan extension 5000 (since it does not have destination parameter > defined and thus it will use value for this parameter in > global_settings xml tag). > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > Now the codec, Skype has its own proprietory code for Skype to Skype > calls. The codec we specify in Skypiax configuration file is actually > used for Skype to/from non-Skype calls. Consider following dial plan > example (with skypiax configuration given above), > > > > > > > > If a remote Skype user dials test.01 from his/her Skype client, then > FreeSWITCH will route this call to SIP user 1000 and codecs specified > in Skypiax configuration will be offered to destination SIP endpoint > (SIP user 1000 in this case). > > Thank you. > > > On Fri, Jul 10, 2009 at 5:58 AM, Meftah Tayeb > wrote: > > hello, > i have the folowing parameter in Skypiax.conf.xml: > > > > each call that will to by routed to this destination?? > > > > > > Each Call will to by routed to this destination? > > each codecs that is pocible to use it with Skypiax? all? speex? > this codecs is used beetwan skypiax and the remote peer? > thanks > > > __________ Information from ESET NOD32 Antivirus, version of virus > signature database 4229 (20090709) __________ > > The message was checked by ESET NOD32 Antivirus. > > http://www.eset.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 > > > > > -- > Muhammad Shahzad > ----------------------------------- > CISCO Rich Media Communication Specialist (CRMCS) > CISCO Certified Network Associate (CCNA) > Cell: +92 334 422 40 88 > MSN: shari_786pk at hotmail.com > Email: shaheryarkh at googlemail.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 > > > > __________ Information from ESET NOD32 Antivirus, version of virus signature database 4231 (20090710) __________ > > The message was checked by ESET NOD32 Antivirus. > > http://www.eset.com > > __________ Information from ESET NOD32 Antivirus, version of virus signature database 4231 (20090710) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090710/f68d2fed/attachment-0002.html From mcampbellsmith at gmail.com Fri Jul 10 02:57:55 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Fri, 10 Jul 2009 19:57:55 +1000 Subject: [Freeswitch-users] 2 voicemail questions Message-ID: <33c87fa30907100257w29c628cbnb641d09a75bc22b8@mail.gmail.com> Hi! >> 1. Can I email the voicemail message to multiple email addresses? A comma separated list does not work in the extensions.xml file (1000.xml), but it does work if I hard code the email addresses into the notify-voicemail.tpl file. Could this be added to the switch so that it can handle comma separated lists? >> 2. How can I make Freeswitch dial a number AFTER a voicemail is left? >api Hangup hook? i want the 'voicemail' application to appear to call the extension to notify the user that there is a waiting message. This is an extract from my dialplan.xml: This only works if the B leg (ie voicemail application) hangs up first. This would be an unusual situation and does not achieve what I want... is there any other way to achieve this? Thanks > Hi! > > I have 2 questions regarding voicemail ... > > 1. Can I email the voicemail message to multiple email addresses? If > so, what format is this in? > Try a comma sep. list. Not sure if it will work. > > 2. How can I make Freeswitch dial a number AFTER a voicemail is left? api Hangup hook? I g > From: Brian West > On Jul 7, 2009, at 9:11 AM, Mark Campbell-Smith wrote: > Hi! > > I have 2 questions regarding voicemail ... > > 1. Can I email the voicemail message to multiple email addresses? If > so, what format is this in? > > Try a comma sep. list. Not sure if it will work. From shaheryarkh at googlemail.com Fri Jul 10 04:02:34 2009 From: shaheryarkh at googlemail.com (Muhammad Shahzad) Date: Fri, 10 Jul 2009 17:02:34 +0600 Subject: [Freeswitch-users] Skypiax Parameters Informations Request In-Reply-To: <4A570E6B.1000106@gmail.com> References: <4A56841E.4080206@gmail.com> <4A570E6B.1000106@gmail.com> Message-ID: I think you can use it has long as remote end-point supports it. Thank you. On Fri, Jul 10, 2009 at 3:48 PM, Meftah Tayeb wrote: > hello Muhammad , > thank you > what about hig cality audio codec to use? > speex is good? > thanks > > Muhammad Shahzad wrote: > > Destination parameter actually specifies the extension on which this Skype > user is reachable within FreeSWITCH dialplan for incoming calls. > > If this parameter is specified in per_interface_settings xml tag then it > will override the value of this parameter in global_settings xml tag, > otherwise value of this parameter from global_settings xml tag will be used. > > Here is an example (see below), the user test.01 is reachable on dialplan > extension 2000 (since it has its own destination defined in > per_interface_settings xml tag), whereas test.02 is reachable on dialplan > extension 5000 (since it does not have destination parameter defined and > thus it will use value for this parameter in global_settings xml tag). > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > Now the codec, Skype has its own proprietory code for Skype to Skype calls. > The codec we specify in Skypiax configuration file is actually used for > Skype to/from non-Skype calls. Consider following dial plan example (with > skypiax configuration given above), > > > > > > > > If a remote Skype user dials test.01 from his/her Skype client, then > FreeSWITCH will route this call to SIP user 1000 and codecs specified in > Skypiax configuration will be offered to destination SIP endpoint (SIP user > 1000 in this case). > > Thank you. > > > On Fri, Jul 10, 2009 at 5:58 AM, Meftah Tayeb wrote: > >> hello, >> i have the folowing parameter in Skypiax.conf.xml: >> >> >> >> each call that will to by routed to this destination?? >> >> >> >> >> >> Each Call will to by routed to this destination? >> >> each codecs that is pocible to use it with Skypiax? all? speex? >> this codecs is used beetwan skypiax and the remote peer? >> thanks >> >> >> __________ Information from ESET NOD32 Antivirus, version of virus >> signature database 4229 (20090709) __________ >> >> The message was checked by ESET NOD32 Antivirus. >> >> http://www.eset.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 >> > > > > -- > Muhammad Shahzad > ----------------------------------- > CISCO Rich Media Communication Specialist (CRMCS) > CISCO Certified Network Associate (CCNA) > Cell: +92 334 422 40 88 > MSN: shari_786pk at hotmail.com > Email: shaheryarkh at googlemail.com > > ------------------------------ > > _______________________________________________ > 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 > > > > > __________ Information from ESET NOD32 Antivirus, version of virus signature database 4231 (20090710) __________ > > The message was checked by ESET NOD32 Antivirus. > http://www.eset.com > > > > > __________ Information from ESET NOD32 Antivirus, version of virus > signature database 4231 (20090710) __________ > > The message was checked by ESET NOD32 Antivirus. > > http://www.eset.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 > > -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090710/8dab18a1/attachment-0002.html From helmut.kuper at ewetel.de Fri Jul 10 05:53:50 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Fri, 10 Jul 2009 14:53:50 +0200 Subject: [Freeswitch-users] pocketsphinx Message-ID: <4A5739DE.1080800@ewetel.de> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I try to change pocketsphinx's grammar from default (english) to german. I found this archive (http://www.repository.voxforge1.org/downloads/de/Trunk/AcousticModels/), which contains similar files like those which can be found in grammar/model/communicator directory. Unfortunately FS crashed without writing a core file nor logfile enries as soon as as pizza demo trys to detect speech. Any Ideas? Maybe someone has already working grammar/model files for german language? regards helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) iD8DBQFKVzne4tZeNddg3dwRAthzAJ4hvonJLgaTWc3kCQXhESb2wsTu8QCeP/DD skUOkNgUHLRaKqVGWWk1uM8= =HkcN -----END PGP SIGNATURE----- From anthony.minessale at gmail.com Fri Jul 10 06:25:04 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 10 Jul 2009 08:25:04 -0500 Subject: [Freeswitch-users] Less Than a Month Until ClueCon In-Reply-To: References: <191c3a030907081329q53ec3609i37b9ee87a31c9247@mail.gmail.com> <09DA50D4-C82E-425D-9CB2-467159B673DB@freeswitch.org> <87f2f3b90907091047p21c536d5re65acd6c5141ba99@mail.gmail.com> Message-ID: <191c3a030907100625g77a6fe32ub7a948931144e90f@mail.gmail.com> Yes, This is very important. The rates on the room will soon rise out of control as the conference date nears so it's important to book now or find someone to share with before it's too late. On Thu, Jul 9, 2009 at 12:54 PM, Brian West wrote: > Also if you need room share info email Michael or Myself offlist we can > help you find a roomie! > /b > > On Jul 9, 2009, at 12:47 PM, Michael Collins wrote: > > FYI, visit the hotel details page . Scroll to > the bottom and there are links both to online maps and directions as well as > a handy document that you can download. We have directions for getting > to/from the hotel from both airports (O'Hare and Midway) via public > transport as well as for those driving into town. > > -Michael > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference 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/20090710/876bed77/attachment-0002.html From ney at frota.net Fri Jul 10 00:00:28 2009 From: ney at frota.net (Ney Frota) Date: Fri, 10 Jul 2009 04:00:28 -0300 Subject: [Freeswitch-users] Help Message-ID: <5E41B7C2-B163-4364-A2A9-8B90AD58F47E@frota.net> Help From vkozak at abisoft.spb.ru Fri Jul 10 02:39:28 2009 From: vkozak at abisoft.spb.ru (Kozak Vladimir) Date: Fri, 10 Jul 2009 13:39:28 +0400 Subject: [Freeswitch-users] FS not wait respond from called and send 200 Ok Message-ID: <5F5ABAC4334545CE8323DACB10805DE1@abisoft.biz> Hello, I have the following problem: I send Invite without SDP to Freeswitch on destination_number "xxx_123" And I want Freeswitch to make "bridge", but it doesn't wait respond from "123" and sends 200 Ok with SDP to me. Does nybody know a clue about this? Best regards vkozak -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090710/8ee67c81/attachment-0002.html From velu.technical at gmail.com Fri Jul 10 05:36:14 2009 From: velu.technical at gmail.com (velusamy velu) Date: Fri, 10 Jul 2009 18:06:14 +0530 Subject: [Freeswitch-users] How to configure SIP(Sofia) profiles Message-ID: <1452e2980907100536s15818474k901093b742510064@mail.gmail.com> Dear Friends, I am a newbie for FreeSWITCH. I was installed FreeSWITCH locally. I just wanted to test whether my FreeSWITCH is working fine. I need help from you that how to configure my Softphone(Twinkle) to use FreeSWITCH. I need steps to check my FreeSWITCH working with Twinkle. Please help me in this... Thanks in Advance. Regards, K.Velusamy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090710/66f36fc9/attachment-0002.html From Prometheus001 at gmx.net Fri Jul 10 07:12:46 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Fri, 10 Jul 2009 16:12:46 +0200 Subject: [Freeswitch-users] pocketsphinx In-Reply-To: <4A5739DE.1080800@ewetel.de> References: <4A5739DE.1080800@ewetel.de> Message-ID: <4A574C5E.2000401@gmx.net> Hello Helmut, I looked at these dic files. Their content (look at all the qq's) is quite different from the dic files supplied with freeswitch pocketsphinx. As I remember the CMU dict file format has changed in April 2008. Best regards Peter Helmut Kuper schrieb: > Hi, > > I try to change pocketsphinx's grammar from default (english) to german. > I found this archive > (http://www.repository.voxforge1.org/downloads/de/Trunk/AcousticModels/), > which > contains similar files like those which can be found in > grammar/model/communicator directory. > > Unfortunately FS crashed without writing a core file nor logfile enries > as soon as as pizza demo trys to detect speech. > > Any Ideas? Maybe someone has already working grammar/model files for > german language? > > > 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 brian at freeswitch.org Fri Jul 10 07:24:01 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 10 Jul 2009 09:24:01 -0500 Subject: [Freeswitch-users] pocketsphinx In-Reply-To: <4A574C5E.2000401@gmx.net> References: <4A5739DE.1080800@ewetel.de> <4A574C5E.2000401@gmx.net> Message-ID: <51760CDC-8560-4C2B-B618-2E97A0B3F8C2@freeswitch.org> Yes you have to make sure you use the one that comes with Pocketsphinx and not the 7.x one you download from the website. They aren't compatible last I checked. /b On Jul 10, 2009, at 9:12 AM, Peter P GMX wrote: > Hello Helmut, > > I looked at these dic files. Their content (look at all the qq's) is > quite different from the dic files supplied with freeswitch > pocketsphinx. > As I remember the CMU dict file format has changed in April 2008. > > Best regards > Peter From Prometheus001 at gmx.net Fri Jul 10 07:24:13 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Fri, 10 Jul 2009 16:24:13 +0200 Subject: [Freeswitch-users] pocketsphinx In-Reply-To: <4A5739DE.1080800@ewetel.de> References: <4A5739DE.1080800@ewetel.de> Message-ID: <4A574F0D.7040604@gmx.net> Hello Helmut, I looked at these dic files. Their content (look at all the qq's) is quite different from the dic files supplied with freeswitch pocketsphinx. As I remember the CMU dict file format has changed in April 2008. Maybe there is a converter somewhere? I was thinking of just enhancing the current dict file for some german words I need, but did not test it so far. This should be possible without modifying the underlying grammar. http://en.wikipedia.org/wiki/CMU_Pronouncing_Dictionary I would love to hear when you have had any progress on this. Best regards Peter Helmut Kuper schrieb: > Hi, > > I try to change pocketsphinx's grammar from default (english) to german. > I found this archive > (http://www.repository.voxforge1.org/downloads/de/Trunk/AcousticModels/), > which > contains similar files like those which can be found in > grammar/model/communicator directory. > > Unfortunately FS crashed without writing a core file nor logfile enries > as soon as as pizza demo trys to detect speech. > > Any Ideas? Maybe someone has already working grammar/model files for > german language? > > > 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 dftoro at yahoo.com Fri Jul 10 08:54:34 2009 From: dftoro at yahoo.com (Diego Toro) Date: Fri, 10 Jul 2009 08:54:34 -0700 (PDT) Subject: [Freeswitch-users] mod_say_en directory location Message-ID: <45224.76458.qm@web33505.mail.mud.yahoo.com> Hi all, ? This happens becouse sound_prefix variable is not used to make path to sound files, case "Language Handling: call for assistance" ? Diego --- On Thu, 7/9/09, Phillip Jones wrote: From: Phillip Jones Subject: Re: [Freeswitch-users] mod_say_en directory location To: freeswitch-users at lists.freeswitch.org Date: Thursday, July 9, 2009, 3:38 PM Ok - forget this one - I did a fresh install from the pre-compiled windows binary/msi - referenced on the wiki - and every thing is working as it should be. Thanks On Thu, Jul 9, 2009 at 2:46 PM, Phillip Jones wrote: Thanks for the response. I don't think so - the trace states: [ERR] mod_sndfile.c: 192 Error Opening File [C:\Program Files (x86)\Freeswitch\sounds\en\digits/1.wav] [System error : The system cannot find the path specified.] I created a 16000 directory to see whether that would help, and it did not. My C:\Program Files (x86)\Freeswitch\conf\lang\en\en.xml contains: ? ??? ??? ??? ? ? Am I correct in thinking this is where the sound file dir for digits would be specified? On Thu, Jul 9, 2009 at 12:38 PM, Michael Collins wrote: Is the call perhaps at 16kHz and it's looking for non-installed 16kHz sound files? -MC On Thu, Jul 9, 2009 at 9:06 AM, Phillip Jones wrote: Hi there, I have a very simply script that speaks back some digits, as so: session:execute("say", "en number iterated 1234"); However, to get this to work successfully I have had to move the 'digits' directory to: C:\Program Files (x86)\Freeswitch\sounds\en from the default: C:\Program Files (x86)\Freeswitch\sounds\en\us\callie\digits\8000 This is a clean install of FreeSWITCH - so I am wondering why I needed to do this, what have not configured correctly? As you can see I am using windows with a resent build (3 days) from svn. Any help appreciated. Thanks Phillip Jones _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -----Inline Attachment Follows----- _______________________________________________ Freeswitch-users mailing list Freeswitch-users at 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/20090710/c5f5dfc9/attachment-0002.html From woof at iwoof.org Fri Jul 10 08:56:16 2009 From: woof at iwoof.org (Andy Spitzer) Date: Fri, 10 Jul 2009 11:56:16 -0400 Subject: [Freeswitch-users] Question on auth-calls Message-ID: Woof! It is my understanding, that if I set in a SIP profile, it shouldn't challenge for authentication under any circumstances. However, if an INVITE contains a a Proxy-Authorization header from another proxy, Sofia DOES challenge with a 407. I'm aware one can set accept-blind-auth to work around this, but I'm wondering if my understanding of auth-calls is wrong, or the behavior I'm seeing is wrong. --Woof! From mike at jerris.com Fri Jul 10 09:04:01 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 10 Jul 2009 12:04:01 -0400 Subject: [Freeswitch-users] 2 voicemail questions In-Reply-To: <33c87fa30907100257w29c628cbnb641d09a75bc22b8@mail.gmail.com> References: <33c87fa30907100257w29c628cbnb641d09a75bc22b8@mail.gmail.com> Message-ID: <344FC083-5916-4232-AEB5-3504626566DF@jerris.com> could you post how you tired to do it in dialplan that didn't work? Mike On Jul 10, 2009, at 5:57 AM, Mark Campbell-Smith wrote: > Hi! >>> 1. Can I email the voicemail message to multiple email addresses? > A comma separated list does not work in the extensions.xml file > (1000.xml), but it does work if I hard code the email addresses into > the notify-voicemail.tpl file. > > Could this be added to the switch so that it can handle comma > separated lists? > >>> 2. How can I make Freeswitch dial a number AFTER a voicemail is >>> left? > >> api Hangup hook? > > i want the 'voicemail' application to appear to call the extension to > notify the user that there is a waiting message. This is an extract > from my dialplan.xml: > > > > > > > > > data="user/${dialed_extension}@${domain_name}"/> > > > > > field="${vm_boxcount(${destination_number}@${domain_name})}" > expression="^(1)$"> > > > > This only works if the B leg (ie voicemail application) hangs up > first. This would be an unusual situation and does not achieve what I > want... is there any other way to achieve this? > > Thanks > >> Hi! >> >> I have 2 questions regarding voicemail ... >> >> 1. Can I email the voicemail message to multiple email addresses? If >> so, what format is this in? >> > > Try a comma sep. list. Not sure if it will work. > >> >> 2. How can I make Freeswitch dial a number AFTER a voicemail is left? > > api Hangup hook? > > I g >> From: Brian West > >> On Jul 7, 2009, at 9:11 AM, Mark Campbell-Smith wrote: > >> Hi! >> >> I have 2 questions regarding voicemail ... >> >> 1. Can I email the voicemail message to multiple email addresses? If >> so, what format is this in? >> > >> Try a comma sep. list. Not sure if it will work. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.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 Jul 10 09:06:07 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 10 Jul 2009 11:06:07 -0500 Subject: [Freeswitch-users] Question on auth-calls In-Reply-To: References: Message-ID: <19E3739C-73C3-4B8D-BC54-18079CB65CEA@freeswitch.org> accept-blind-auth is for this scenario.... /b On Jul 10, 2009, at 10:56 AM, Andy Spitzer wrote: > Woof! > > It is my understanding, that if I set > > in a SIP profile, it shouldn't challenge for authentication under > any circumstances. > > However, if an INVITE contains a a Proxy-Authorization header from > another proxy, Sofia DOES challenge with a 407. > > I'm aware one can set accept-blind-auth to work around this, but I'm > wondering if my understanding of auth-calls is wrong, or the > behavior I'm seeing is wrong. > > --Woof! From anthony.minessale at gmail.com Fri Jul 10 09:09:05 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 10 Jul 2009 11:09:05 -0500 Subject: [Freeswitch-users] Question on auth-calls In-Reply-To: References: Message-ID: <191c3a030907100909h6a0031cfv3bbc0f80e0cdbcec@mail.gmail.com> The way it works by default is that if you send a www-authenticate, we *always* try to process it. HOWEVER, we have a accept-blind-auth sofia profile param (in fact it was invented just for sipX) On Fri, Jul 10, 2009 at 10:56 AM, Andy Spitzer wrote: > Woof! > > It is my understanding, that if I set > > in a SIP profile, it shouldn't challenge for authentication under any > circumstances. > > However, if an INVITE contains a a Proxy-Authorization header from another > proxy, Sofia DOES challenge with a 407. > > I'm aware one can set accept-blind-auth to work around this, but I'm > wondering if my understanding of auth-calls is wrong, or the behavior I'm > seeing is wrong. > > --Woof! > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference 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/20090710/8d650d75/attachment-0002.html From dome at tel.co.th Fri Jul 10 09:10:23 2009 From: dome at tel.co.th (Dome Charoenyost) Date: Fri, 10 Jul 2009 23:10:23 +0700 Subject: [Freeswitch-users] How to check hangup cause before try next route in mod_lcr Message-ID: <8ccbff060907100910s2f12827fu18f9ff267b8b8e51@mail.gmail.com> Dear all, I'm testing mod_lcr. i found example in wiki If i want to use ${lcr_route_1} and ${lcr_route_2} What's best way to check hangup cause before try lcr_route_2 ? BG Dome C. From mrene_lists at avgs.ca Fri Jul 10 09:11:22 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Fri, 10 Jul 2009 12:11:22 -0400 Subject: [Freeswitch-users] How to check hangup cause before try next route in mod_lcr In-Reply-To: <8ccbff060907100910s2f12827fu18f9ff267b8b8e51@mail.gmail.com> References: <8ccbff060907100910s2f12827fu18f9ff267b8b8e51@mail.gmail.com> Message-ID: <7E454794-5270-4BA0-A11E-9CF220FA4BE4@avgs.ca> That'll make your dialplan stop if the call is hung up from the B-leg. Math On 10-Jul-09, at 12:10 PM, Dome Charoenyost wrote: > Dear all, > > I'm testing mod_lcr. > i found example in wiki > > > If i want to use ${lcr_route_1} and ${lcr_route_2} What's best way to > check hangup cause before try lcr_route_2 ? > > > BG > > Dome C. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.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 Fri Jul 10 09:12:03 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 10 Jul 2009 12:12:03 -0400 Subject: [Freeswitch-users] FS not wait respond from called and send 200 Ok In-Reply-To: <5F5ABAC4334545CE8323DACB10805DE1@abisoft.biz> References: <5F5ABAC4334545CE8323DACB10805DE1@abisoft.biz> Message-ID: <74C6337E-6F03-4485-B69B-372658E0D79B@jerris.com> Look closer at the logs, we don't send a 200ok in a bridge until we get one from the b leg. Mike On Jul 10, 2009, at 5:39 AM, Kozak Vladimir wrote: > Hello, > > I have the following problem: I send Invite without SDP to > Freeswitch on destination_number "xxx_123" > > > > > > > And I want Freeswitch to make "bridge", but it doesn't wait respond > from "123" and sends 200 Ok with SDP to me. > Does nybody know a clue about this? > > Best regards > vkozak > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090710/a5643c1f/attachment-0002.html From woof at iwoof.org Fri Jul 10 09:17:47 2009 From: woof at iwoof.org (Andy Spitzer) Date: Fri, 10 Jul 2009 12:17:47 -0400 Subject: [Freeswitch-users] Question on auth-calls In-Reply-To: <191c3a030907100909h6a0031cfv3bbc0f80e0cdbcec@mail.gmail.com> References: <191c3a030907100909h6a0031cfv3bbc0f80e0cdbcec@mail.gmail.com> Message-ID: Woof! On Fri, 10 Jul 2009 12:09:05 -0400, Anthony Minessale wrote: > The way it works by default is that if you send a www-authenticate, we > *always* try to process it. > HOWEVER, we have a accept-blind-auth sofia profile param (in fact it was > invented just for sipX) Then my understanding was incorrect. Thanks for the clarification. --Woof! From Christian.Jensen at Teligence.Net Fri Jul 10 09:32:08 2009 From: Christian.Jensen at Teligence.Net (Christian Jensen) Date: Fri, 10 Jul 2009 09:32:08 -0700 Subject: [Freeswitch-users] 302 Redirect Message-ID: Hi everyone! I have a question - I need to change the "From" SIP header during a "Redirect" to make it look like the number that called is a different number. I need to be able to change it but it is not taking - I have tried just about every combination of variable settings that I know of but the SIP message is just not getting the data. Here is my config: In fact - it would appear that no channel variables are making it out the door during a redirect. Any help would be greatly appreciated. Thanks! Christian Jensen Software Development Manager - Back Office Teligence T: 604-629-6055 Ext. 3304 M: 778-996-4283 F: 604-257-5578 christian.jensen at teligence.net www.teligence.net -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090710/f6b41a3d/attachment-0002.html From dome at tel.co.th Fri Jul 10 09:37:29 2009 From: dome at tel.co.th (Dome Charoenyost) Date: Fri, 10 Jul 2009 23:37:29 +0700 Subject: [Freeswitch-users] How to check hangup cause before try next route in mod_lcr In-Reply-To: <7E454794-5270-4BA0-A11E-9CF220FA4BE4@avgs.ca> References: <8ccbff060907100910s2f12827fu18f9ff267b8b8e51@mail.gmail.com> <7E454794-5270-4BA0-A11E-9CF220FA4BE4@avgs.ca> Message-ID: <8ccbff060907100937l319b85e9t4200a1a8a630a1c9@mail.gmail.com> 2009/7/10 Mathieu Rene : > thanks. let's me ask some question. i want to provide callback solution. 1. customer call to FS 2. FS hangup call and check balance (by callerid) 3. Make call to customer number 4. customer answer call and input destination this step when B-leg hangup (Or A-leg put *) customer can input other number until A-leg Hangup Now my solution use javascripts. but someone tell me JS not good for handle call. So now i testing mod_lcr and nibblebill Can someone show me dialplan for do like that. Best Regards. Dome C. > > That'll make your dialplan stop if the call is hung up from the B-leg. > > Math > > On 10-Jul-09, at 12:10 PM, Dome Charoenyost wrote: > >> Dear all, >> >> ? ? ? ? I'm testing mod_lcr. >> i found example in wiki >> ? ? ? >> ? ? ? >> If i want to use ${lcr_route_1} and ${lcr_route_2} What's best way to >> check hangup cause before try lcr_route_2 ? >> >> >> BG >> >> Dome C. >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.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 Jul 10 09:41:03 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 10 Jul 2009 11:41:03 -0500 Subject: [Freeswitch-users] 302 Redirect In-Reply-To: References: Message-ID: <9F8A6292-3BE0-4E29-B331-9EA7A413BE37@freeswitch.org> Ok you need the nice new feature I added to FreeSWITCH that lets you handle all the 302 redirects in your own dialplan/context. Set the param manual-redirect on your sofia profile then you can define sip_redirect_profile, sip_redirect_context, sip_redirect_dialplan or you can create a context called redirected and do anything you wish with the 302'ed call 100% manually. You'll get sip_redirect_contact_X sip_redirected_to sip_redirect_contact_user_X sip_redirect_contact_host_X sip_redirect_contact_params_X sip_redirected_by All of these will help you process this via the dialplan. /b On Jul 10, 2009, at 11:32 AM, Christian Jensen wrote: > Hi everyone! > > I have a question ? I need to change the ?From? SIP header during a > ?Redirect? to make it look like the number that called is a > different number. > > I need to be able to change it but it is not taking ? I have tried > just about every combination of variable settings that I know of but > the SIP message is just not getting the data. > > Here is my config: > > > > > data="sip_from_user_stripped=false"/> > data="sip_from_user=0445555555"/> > data="sip_invite_domain=sip:0446666666 at 192.168.8.2"/> > > > data="myani=0449999999"/> > > > > > > In fact ? it would appear that no channel variables are making it > out the door during a redirect. > > Any help would be greatly appreciated. > > Thanks! > > Christian Jensen > Software Development Manager ? Back Office > Teligence > T: 604-629-6055 Ext. 3304 > M: 778-996-4283 > F: 604-257-5578 > christian.jensen at teligence.net > www.teligence.net > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090710/213ac79a/attachment-0002.html From mrene_lists at avgs.ca Fri Jul 10 09:43:46 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Fri, 10 Jul 2009 12:43:46 -0400 Subject: [Freeswitch-users] 302 Redirect In-Reply-To: <9F8A6292-3BE0-4E29-B331-9EA7A413BE37@freeswitch.org> References: <9F8A6292-3BE0-4E29-B331-9EA7A413BE37@freeswitch.org> Message-ID: You can also set the following variables to control where the call will go once you get a 302: sip_redirect_profile (used to build the dialstrings) sip_redirect_context (the dialplan context to send the call) sip_redirect_dialplan (the dialplan to send the call) Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 10-Jul-09, at 12:41 PM, Brian West wrote: > Ok you need the nice new feature I added to FreeSWITCH that lets you > handle all the 302 redirects in your own dialplan/context. > > Set the param manual-redirect on your sofia profile then you can > define sip_redirect_profile, sip_redirect_context, > sip_redirect_dialplan or you can create a context called redirected > and do anything you wish with the 302'ed call 100% manually. > > You'll get > > sip_redirect_contact_X > sip_redirected_to > sip_redirect_contact_user_X > sip_redirect_contact_host_X > sip_redirect_contact_params_X > sip_redirected_by > > All of these will help you process this via the dialplan. > > /b > > On Jul 10, 2009, at 11:32 AM, Christian Jensen wrote: > >> Hi everyone! >> >> I have a question ? I need to change the ?From? SIP header during a >> ?Redirect? to make it look like the number that called is a >> different number. >> >> I need to be able to change it but it is not taking ? I have tried >> just about every combination of variable settings that I know of >> but the SIP message is just not getting the data. >> >> Here is my config: >> >> >> >> >> > data="sip_from_user_stripped=false"/> >> > data="sip_from_user=0445555555"/> >> > data="sip_invite_domain=sip:0446666666 at 192.168.8.2"/> >> >> >> > data="myani=0449999999"/> >> >> >> >> >> >> In fact ? it would appear that no channel variables are making it >> out the door during a redirect. >> >> Any help would be greatly appreciated. >> >> Thanks! >> >> Christian Jensen >> Software Development Manager ? Back Office >> Teligence >> T: 604-629-6055 Ext. 3304 >> M: 778-996-4283 >> F: 604-257-5578 >> christian.jensen at teligence.net >> www.teligence.net >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090710/1ce4e939/attachment-0002.html From jens at vegeby.nu Fri Jul 10 09:44:34 2009 From: jens at vegeby.nu (Jens Vegeby) Date: Fri, 10 Jul 2009 18:44:34 +0200 Subject: [Freeswitch-users] Help In-Reply-To: <5E41B7C2-B163-4364-A2A9-8B90AD58F47E@frota.net> References: <5E41B7C2-B163-4364-A2A9-8B90AD58F47E@frota.net> Message-ID: <30ee97110907100944y15d1e90fu29b543cd87f12c5f@mail.gmail.com> You might wanna write what you need help with :) On 7/10/09, Ney Frota wrote: > Help > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 Mvh/Regards Jens From brian at freeswitch.org Fri Jul 10 09:46:08 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 10 Jul 2009 11:46:08 -0500 Subject: [Freeswitch-users] 302 Redirect In-Reply-To: References: <9F8A6292-3BE0-4E29-B331-9EA7A413BE37@freeswitch.org> Message-ID: We really need a wiki page for this. This is handy for those people that do CNAM dips via 302's. /b On Jul 10, 2009, at 11:43 AM, Mathieu Rene wrote: > You can also set the following variables to control where the call > will go once you get a 302: > > sip_redirect_profile (used to build the dialstrings) > sip_redirect_context (the dialplan context to send the call) > sip_redirect_dialplan (the dialplan to send the call) > > Mathieu Rene > Avant-Garde Solutions Inc > Office: + 1 (514) 664-1044 x100 > Cell: +1 (514) 664-1044 x200 > mrene at avgs.ca > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090710/d61a885d/attachment-0002.html From Christian.Jensen at Teligence.Net Fri Jul 10 09:52:00 2009 From: Christian.Jensen at Teligence.Net (Christian Jensen) Date: Fri, 10 Jul 2009 09:52:00 -0700 Subject: [Freeswitch-users] 302 Redirect In-Reply-To: <9F8A6292-3BE0-4E29-B331-9EA7A413BE37@freeswitch.org> References: <9F8A6292-3BE0-4E29-B331-9EA7A413BE37@freeswitch.org> Message-ID: I will give it a try. Gracias! Christian Jensen Software Development Manager Back Office ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Friday, July 10, 2009 9:41 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] 302 Redirect Ok you need the nice new feature I added to FreeSWITCH that lets you handle all the 302 redirects in your own dialplan/context. Set the param manual-redirect on your sofia profile then you can define sip_redirect_profile, sip_redirect_context, sip_redirect_dialplan or you can create a context called redirected and do anything you wish with the 302'ed call 100% manually. You'll get sip_redirect_contact_X sip_redirected_to sip_redirect_contact_user_X sip_redirect_contact_host_X sip_redirect_contact_params_X sip_redirected_by All of these will help you process this via the dialplan. /b On Jul 10, 2009, at 11:32 AM, Christian Jensen wrote: Hi everyone! I have a question - I need to change the "From" SIP header during a "Redirect" to make it look like the number that called is a different number. I need to be able to change it but it is not taking - I have tried just about every combination of variable settings that I know of but the SIP message is just not getting the data. Here is my config: In fact - it would appear that no channel variables are making it out the door during a redirect. Any help would be greatly appreciated. Thanks! Christian Jensen Software Development Manager - Back Office Teligence T: 604-629-6055 Ext. 3304 M: 778-996-4283 F: 604-257-5578 christian.jensen at teligence.net www.teligence.net _______________________________________________ Freeswitch-users mailing list Freeswitch-users at 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/20090710/8f974121/attachment-0002.html From msc at freeswitch.org Fri Jul 10 09:52:03 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 10 Jul 2009 09:52:03 -0700 Subject: [Freeswitch-users] How to configure SIP(Sofia) profiles In-Reply-To: <1452e2980907100536s15818474k901093b742510064@mail.gmail.com> References: <1452e2980907100536s15818474k901093b742510064@mail.gmail.com> Message-ID: <87f2f3b90907100952q104339ebqe26999fb38705136@mail.gmail.com> The wiki has some basic information. If you already have FS installed then you will have 20 "users" pre-configured, 1000-1019, all with password of "1234" so you can use that to set up your softphone. Just be sure to have your soft-phone on a different computer than your FreeSWITCH server! (Technically it can work sometimes but we don't recommend it at all.) See here for more info: http://wiki.freeswitch.org/wiki/Getting_Started_Guide#Some_stuff_to_try_out.21 -MC On Fri, Jul 10, 2009 at 5:36 AM, velusamy velu wrote: > Dear Friends, > I am a newbie for FreeSWITCH. I was installed FreeSWITCH locally. I > just wanted to test whether my FreeSWITCH is working fine. I need help from > you that how to configure my Softphone(Twinkle) to use FreeSWITCH. I need > steps to check my FreeSWITCH working with Twinkle. > Please help me in this... > Thanks in Advance. > > Regards, > K.Velusamy > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090710/ba858d92/attachment-0002.html From larclap at yahoo.com Fri Jul 10 10:35:03 2009 From: larclap at yahoo.com (Lars Zeb) Date: Fri, 10 Jul 2009 10:35:03 -0700 Subject: [Freeswitch-users] Intercom call failing with "No Matching gateway found" error Message-ID: <00f001ca0184$c226e440$4674acc0$@com> Trying to make an intercom call (8+extension#) gives me an error. I don't know what I've done wrong, but I think it used to work. I am on Centos 5 with 14196M. Can someone point me in the right direction? The sofia status, dialplan and log are in http://pastebin.freeswitch.org/9681. Thanks, Lars -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090710/9588972a/attachment-0002.html From pjintheusa at gmail.com Fri Jul 10 10:46:02 2009 From: pjintheusa at gmail.com (Phillip Jones) Date: Fri, 10 Jul 2009 13:46:02 -0400 Subject: [Freeswitch-users] managed_mod directories Message-ID: <367751820907101046l7b41a2fdl7cac3d05b37a80f7@mail.gmail.com> Hi there, Using windows with the pre-compiled binary / msi found via the WIKI Using mod_managed with no problems however: mod_managed appears to require I create a directory 'managed' under C:\Program Files (x86)\FreeSWITCH\mod BUT also requires that I place my .dll in C:\Program Files (x86)\FreeSWITCH\mod and NOT C:\Program Files (x86)\FreeSWITCH\mod\managed thus making C:\Program Files (x86)\FreeSWITCH\mod\managed redundant Anyone else seen this behavior? Thanks! Phillip Jones -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090710/69c2baf0/attachment-0002.html From Christian.Jensen at Teligence.Net Fri Jul 10 10:54:55 2009 From: Christian.Jensen at Teligence.Net (Christian Jensen) Date: Fri, 10 Jul 2009 10:54:55 -0700 Subject: [Freeswitch-users] 302 Redirect In-Reply-To: <9F8A6292-3BE0-4E29-B331-9EA7A413BE37@freeswitch.org> References: <9F8A6292-3BE0-4E29-B331-9EA7A413BE37@freeswitch.org> Message-ID: I think I may not have been clear enough on my original post - not unusual for me :-) The information below is for handling incoming redirects - I am certain that it works perfectly from what I see in the source code - however... What I am trying to do is adjust the "From" header in an Outgoing redirect that I am sending to another device (a nextone). Effectively what is happening is I am receiving a call and then telling the originator that I would like them to go somewhere else but at the same time I am changing the "From" field to look like a different number is calling - changing the Caller Id won't cut it. What I need to be able to do is have a parameter on the "redirect" application or have a "From" field override channel variable. Is this doable? I can build and test from source if need be - I am just not familiar enough with the code to do it the "right" way. Thanks! Christian Jensen Software Development Manager Back Office ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Friday, July 10, 2009 9:41 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] 302 Redirect Ok you need the nice new feature I added to FreeSWITCH that lets you handle all the 302 redirects in your own dialplan/context. Set the param manual-redirect on your sofia profile then you can define sip_redirect_profile, sip_redirect_context, sip_redirect_dialplan or you can create a context called redirected and do anything you wish with the 302'ed call 100% manually. You'll get sip_redirect_contact_X sip_redirected_to sip_redirect_contact_user_X sip_redirect_contact_host_X sip_redirect_contact_params_X sip_redirected_by All of these will help you process this via the dialplan. /b On Jul 10, 2009, at 11:32 AM, Christian Jensen wrote: Hi everyone! I have a question - I need to change the "From" SIP header during a "Redirect" to make it look like the number that called is a different number. I need to be able to change it but it is not taking - I have tried just about every combination of variable settings that I know of but the SIP message is just not getting the data. Here is my config: In fact - it would appear that no channel variables are making it out the door during a redirect. Any help would be greatly appreciated. Thanks! Christian Jensen Software Development Manager - Back Office Teligence T: 604-629-6055 Ext. 3304 M: 778-996-4283 F: 604-257-5578 christian.jensen at teligence.net www.teligence.net _______________________________________________ Freeswitch-users mailing list Freeswitch-users at 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/20090710/6375cf26/attachment-0002.html From mrene_lists at avgs.ca Fri Jul 10 10:59:25 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Fri, 10 Jul 2009 13:59:25 -0400 Subject: [Freeswitch-users] 302 Redirect In-Reply-To: References: <9F8A6292-3BE0-4E29-B331-9EA7A413BE37@freeswitch.org> Message-ID: Add that to your gateway: Then set the callerid. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 10-Jul-09, at 1:54 PM, Christian Jensen wrote: > I think I may not have been clear enough on my original post ? not > unusual for me J > > The information below is for handling incoming redirects ? I am > certain that it works perfectly from what I see in the source code ? > however? > > What I am trying to do is adjust the ?From? header in an Outgoing > redirect that I am sending to another device (a nextone). > > Effectively what is happening is I am receiving a call and then > telling the originator that I would like them to go somewhere else > but at the same time I am changing the ?From? field to look like a > different number is calling ? changing the Caller Id won?t cut it. > > What I need to be able to do is have a parameter on the ?redirect? > application or have a ?From? field override channel variable. > > Is this doable? I can build and test from source if need be ? I am > just not familiar enough with the code to do it the ?right? way. > > Thanks! > > Christian Jensen > Software Development Manager > Back Office > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Brian West > Sent: Friday, July 10, 2009 9:41 AM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] 302 Redirect > > Ok you need the nice new feature I added to FreeSWITCH that lets you > handle all the 302 redirects in your own dialplan/context. > > Set the param manual-redirect on your sofia profile then you can > define sip_redirect_profile, sip_redirect_context, > sip_redirect_dialplan or you can create a context called redirected > and do anything you wish with the 302'ed call 100% manually. > > You'll get > > sip_redirect_contact_X > sip_redirected_to > sip_redirect_contact_user_X > sip_redirect_contact_host_X > sip_redirect_contact_params_X > sip_redirected_by > > All of these will help you process this via the dialplan. > > /b > > On Jul 10, 2009, at 11:32 AM, Christian Jensen wrote: > > > Hi everyone! > > I have a question ? I need to change the ?From? SIP header during a > ?Redirect? to make it look like the number that called is a > different number. > > I need to be able to change it but it is not taking ? I have tried > just about every combination of variable settings that I know of but > the SIP message is just not getting the data. > > Here is my config: > > > > > data="sip_from_user_stripped=false"/> > data="sip_from_user=0445555555"/> > data="sip_invite_domain=sip:0446666666 at 192.168.8.2"/> > > > data="myani=0449999999"/> > > > > > > In fact ? it would appear that no channel variables are making it > out the door during a redirect. > > Any help would be greatly appreciated. > > Thanks! > > Christian Jensen > Software Development Manager ? Back Office > Teligence > T: 604-629-6055 Ext. 3304 > M: 778-996-4283 > F: 604-257-5578 > christian.jensen at teligence.net > www.teligence.net > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090710/a40dfeb5/attachment-0002.html From brian at freeswitch.org Fri Jul 10 11:05:35 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 10 Jul 2009 13:05:35 -0500 Subject: [Freeswitch-users] 302 Redirect In-Reply-To: References: <9F8A6292-3BE0-4E29-B331-9EA7A413BE37@freeswitch.org> Message-ID: <76E5EAB2-66A0-40F8-93A5-6534744376B9@freeswitch.org> Can you show me what we send now vs what you want to send in a sip packet? /b On Jul 10, 2009, at 12:54 PM, Christian Jensen wrote: > I think I may not have been clear enough on my original post ? not > unusual for me J > > The information below is for handling incoming redirects ? I am > certain that it works perfectly from what I see in the source code ? > however? > > What I am trying to do is adjust the ?From? header in an Outgoing > redirect that I am sending to another device (a nextone). > > Effectively what is happening is I am receiving a call and then > telling the originator that I would like them to go somewhere else > but at the same time I am changing the ?From? field to look like a > different number is calling ? changing the Caller Id won?t cut it. > > What I need to be able to do is have a parameter on the ?redirect? > application or have a ?From? field override channel variable. > > Is this doable? I can build and test from source if need be ? I am > just not familiar enough with the code to do it the ?right? way. > > Thanks! > > Christian Jensen > Software Development Manager > Back Office -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090710/afe9e380/attachment-0002.html From msc at freeswitch.org Fri Jul 10 11:26:56 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 10 Jul 2009 11:26:56 -0700 Subject: [Freeswitch-users] Intercom call failing with "No Matching gateway found" error In-Reply-To: <00f001ca0184$c226e440$4674acc0$@com> References: <00f001ca0184$c226e440$4674acc0$@com> Message-ID: <87f2f3b90907101126o1c80fe4fyc86aa4c7944a398f@mail.gmail.com> Lars, If I read your dialplan correctly I believe this line is a problem: Try this: Let us know if that works... -MC On Fri, Jul 10, 2009 at 10:35 AM, Lars Zeb wrote: > Trying to make an intercom call (8+extension#) gives me an error. I don?t > know what I?ve done wrong, but I think it used to work. I am on Centos 5 > with 14196M. > > > > Can someone point me in the right direction? The sofia status, dialplan and > log are in http://pastebin.freeswitch.org/9681. > > > > Thanks, Lars > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090710/8cbd5704/attachment-0002.html From dftoro at yahoo.com Fri Jul 10 12:04:10 2009 From: dftoro at yahoo.com (dftoro at yahoo.com) Date: Fri, 10 Jul 2009 12:04:10 -0700 (PDT) Subject: [Freeswitch-users] managed_mod directories Message-ID: <547062.98354.qm@web33504.mail.mud.yahoo.com> Hi, check whether the dll references to other?dll's, in that case you should put the references in managed directory. ? Check this link http://lists.freeswitch.org/pipermail/freeswitch-dev/2009-April/002232.html?, may be your case ? Diego ? ? --- On Fri, 7/10/09, Phillip Jones wrote: From: Phillip Jones Subject: [Freeswitch-users] managed_mod directories To: freeswitch-users at lists.freeswitch.org Date: Friday, July 10, 2009, 12:46 PM Hi there, Using windows with the pre-compiled binary / msi found via the WIKI Using mod_managed with no problems however: mod_managed appears to require I create a directory 'managed' under C:\Program Files (x86)\FreeSWITCH\mod BUT also requires that I place my .dll in C:\Program Files (x86)\FreeSWITCH\mod and NOT C:\Program Files (x86)\FreeSWITCH\mod\managed thus making C:\Program Files (x86)\FreeSWITCH\mod\managed redundant Anyone else seen this behavior? Thanks! Phillip Jones -----Inline Attachment Follows----- _______________________________________________ Freeswitch-users mailing list Freeswitch-users at 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/20090710/41c965ab/attachment-0002.html From larclap at yahoo.com Fri Jul 10 12:04:55 2009 From: larclap at yahoo.com (Lars Zeb) Date: Fri, 10 Jul 2009 12:04:55 -0700 Subject: [Freeswitch-users] Intercom call failing with "No Matching gateway found" error In-Reply-To: <87f2f3b90907101126o1c80fe4fyc86aa4c7944a398f@mail.gmail.com> References: <00f001ca0184$c226e440$4674acc0$@com> <87f2f3b90907101126o1c80fe4fyc86aa4c7944a398f@mail.gmail.com> Message-ID: <013801ca0191$50388ed0$f0a9ac70$@com> Michael, The extension-intercom is from the conf/dialplan/default.xml. I checked the file in the source tree, and it's the same as I originally used. But I did try your suggestion: . The result was the same after reloadxml. To check if it had been reloaded, I opened conf/freeswitch.xml. I was surprised to see only 64 lines in the file. Something is hosed in my configuration. Should I try to rebuild from scratch and move over my changes to the xml files? Lars From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Friday, July 10, 2009 11:27 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Intercom call failing with "No Matching gateway found" error Lars, If I read your dialplan correctly I believe this line is a problem: Try this: Let us know if that works... -MC On Fri, Jul 10, 2009 at 10:35 AM, Lars Zeb wrote: Trying to make an intercom call (8+extension#) gives me an error. I don't know what I've done wrong, but I think it used to work. I am on Centos 5 with 14196M. Can someone point me in the right direction? The sofia status, dialplan and log are in http://pastebin.freeswitch.org/9681. Thanks, Lars _______________________________________________ Freeswitch-users mailing list Freeswitch-users at 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/20090710/33643197/attachment-0002.html From msc at freeswitch.org Fri Jul 10 12:18:53 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 10 Jul 2009 12:18:53 -0700 Subject: [Freeswitch-users] Intercom call failing with "No Matching gateway found" error In-Reply-To: <013801ca0191$50388ed0$f0a9ac70$@com> References: <00f001ca0184$c226e440$4674acc0$@com> <87f2f3b90907101126o1c80fe4fyc86aa4c7944a398f@mail.gmail.com> <013801ca0191$50388ed0$f0a9ac70$@com> Message-ID: <87f2f3b90907101218l1fba314at62fabb596e0f8603@mail.gmail.com> Yeah, you definitely have an issue somewhere. Move your existing configs to a safe place, reinstall the defaults configs, then slowly merge your customizations back in. I recommend creating a separate dialplan file in conf/dialplan/default/ directory so that you can keep track of your custom stuff. -MC On Fri, Jul 10, 2009 at 12:04 PM, Lars Zeb wrote: > Michael, > > > > The extension-intercom is from the conf/dialplan/default.xml. I checked the > file in the source tree, and it?s the same as I originally used. > > > > But I did try your suggestion: data="user/${dialed_extension}"/>. The result was the same after reloadxml. > > > > To check if it had been reloaded, I opened conf/freeswitch.xml. I was > surprised to see only 64 lines in the file. Something is hosed in my > configuration. Should I try to rebuild from scratch and move over my changes > to the xml files? > > > > Lars > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael > Collins > *Sent:* Friday, July 10, 2009 11:27 AM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Intercom call failing with "No Matching > gateway found" error > > > > Lars, > > If I read your dialplan correctly I believe this line is a problem: > > > Try this: > > > Let us know if that works... > > -MC > > On Fri, Jul 10, 2009 at 10:35 AM, Lars Zeb wrote: > > Trying to make an intercom call (8+extension#) gives me an error. I don?t > know what I?ve done wrong, but I think it used to work. I am on Centos 5 > with 14196M. > > > > Can someone point me in the right direction? The sofia status, dialplan and > log are in http://pastebin.freeswitch.org/9681. > > > > Thanks, Lars > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090710/20d124e0/attachment-0002.html From pjintheusa at gmail.com Fri Jul 10 13:06:27 2009 From: pjintheusa at gmail.com (Phillip Jones) Date: Fri, 10 Jul 2009 16:06:27 -0400 Subject: [Freeswitch-users] managed_mod directories In-Reply-To: <547062.98354.qm@web33504.mail.mud.yahoo.com> References: <547062.98354.qm@web33504.mail.mud.yahoo.com> Message-ID: <367751820907101306n39a71912r2e2361b0b2b17d23@mail.gmail.com> Hi, Thanks for the reply. My DDL is working fine. Just not in the mod\managed directory. What is the mod\managed directory for? It is required but not used? Phil On Fri, Jul 10, 2009 at 3:04 PM, wrote: > Hi, check whether the dll references to other dll's, in that case you > should put the references in managed directory. > > Check this link > http://lists.freeswitch.org/pipermail/freeswitch-dev/2009-April/002232.html , > may be your case > > Diego > > > > --- On *Fri, 7/10/09, Phillip Jones * wrote: > > > From: Phillip Jones > Subject: [Freeswitch-users] managed_mod directories > To: freeswitch-users at lists.freeswitch.org > Date: Friday, July 10, 2009, 12:46 PM > > > Hi there, > > Using windows with the pre-compiled binary / msi found via the WIKI > > Using mod_managed with no problems however: > > mod_managed appears to require I create a directory 'managed' under > C:\Program Files (x86)\FreeSWITCH\mod > > BUT also requires that I place my .dll in C:\Program Files > (x86)\FreeSWITCH\mod and NOT C:\Program Files (x86)\FreeSWITCH\mod\managed > > thus making C:\Program Files (x86)\FreeSWITCH\mod\managed redundant > > Anyone else seen this behavior? > > Thanks! > > > Phillip Jones > > -----Inline Attachment Follows----- > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090710/41571b2d/attachment-0002.html From mgg at giagnocavo.net Fri Jul 10 13:48:02 2009 From: mgg at giagnocavo.net (Michael Giagnocavo) Date: Fri, 10 Jul 2009 16:48:02 -0400 Subject: [Freeswitch-users] managed_mod directories In-Reply-To: <367751820907101046l7b41a2fdl7cac3d05b37a80f7@mail.gmail.com> References: <367751820907101046l7b41a2fdl7cac3d05b37a80f7@mail.gmail.com> Message-ID: <6E8D2069C08AA84A83D336E996AE4C67027C6C2A1C@mse17be1.mse17.exchange.ms> You're saying that it requires the managed DLL to be in both the mod and mod\managed directory? What error do you get if it's only in mod? It's been months, but I just checked loader.cs and it looks explicitly in the managed directory to resolve assemblies as well as to scan to load them. -Michael From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Phillip Jones Sent: Friday, July 10, 2009 11:46 AM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] managed_mod directories Hi there, Using windows with the pre-compiled binary / msi found via the WIKI Using mod_managed with no problems however: mod_managed appears to require I create a directory 'managed' under C:\Program Files (x86)\FreeSWITCH\mod BUT also requires that I place my .dll in C:\Program Files (x86)\FreeSWITCH\mod and NOT C:\Program Files (x86)\FreeSWITCH\mod\managed thus making C:\Program Files (x86)\FreeSWITCH\mod\managed redundant Anyone else seen this behavior? Thanks! Phillip Jones -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090710/e77db563/attachment-0002.html From pjintheusa at gmail.com Fri Jul 10 16:22:06 2009 From: pjintheusa at gmail.com (Phillip Jones) Date: Fri, 10 Jul 2009 19:22:06 -0400 Subject: [Freeswitch-users] managed_mod directories In-Reply-To: <6E8D2069C08AA84A83D336E996AE4C67027C6C2A1C@mse17be1.mse17.exchange.ms> References: <367751820907101046l7b41a2fdl7cac3d05b37a80f7@mail.gmail.com> <6E8D2069C08AA84A83D336E996AE4C67027C6C2A1C@mse17be1.mse17.exchange.ms> Message-ID: <367751820907101622u5b561dd1qcd058c6c649f49c7@mail.gmail.com> It is looking in mod. It required the mod\managed directory, but if I place my dll in mod\managed it fails. DLL must be in mod - mod\managed is empty. My app works fine though Phil On Fri, Jul 10, 2009 at 4:48 PM, Michael Giagnocavo wrote: > You?re saying that it requires the managed DLL to be in both the mod and > mod\managed directory? What error do you get if it?s only in mod? It?s been > months, but I just checked loader.cs and it looks explicitly in the managed > directory to resolve assemblies as well as to scan to load them. > > > > -Michael > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Phillip > Jones > *Sent:* Friday, July 10, 2009 11:46 AM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* [Freeswitch-users] managed_mod directories > > > > Hi there, > > > Using windows with the pre-compiled binary / msi found via the WIKI > > Using mod_managed with no problems however: > > mod_managed appears to require I create a directory 'managed' under > C:\Program Files (x86)\FreeSWITCH\mod > > BUT also requires that I place my .dll in C:\Program Files > (x86)\FreeSWITCH\mod and NOT C:\Program Files (x86)\FreeSWITCH\mod\managed > > thus making C:\Program Files (x86)\FreeSWITCH\mod\managed redundant > > Anyone else seen this behavior? > > Thanks! > > > Phillip Jones > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090710/406fa883/attachment-0002.html From velu.technical at gmail.com Fri Jul 10 22:23:51 2009 From: velu.technical at gmail.com (velusamy velu) Date: Sat, 11 Jul 2009 10:53:51 +0530 Subject: [Freeswitch-users] Error in default Sofia profile checking Message-ID: <1452e2980907102223r681f59e9kad0cf6883cda4df2@mail.gmail.com> Dear Friends, When I register my Softphone(Twinkle) with predefined sofia registration("1000" with password "1234"). I have got the following error in FreeSWITCH console. "2009-07-11 09:37:16 [ERR] sofia_reg.c:1135 sofia_reg_handle_sip_i_register() NO CONTACT!" Please help me to solve this problem... Regards, K.Velusamy. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090711/c97d496b/attachment-0002.html From jason at jasonjgw.net Fri Jul 10 23:59:40 2009 From: jason at jasonjgw.net (Jason White) Date: Sat, 11 Jul 2009 16:59:40 +1000 Subject: [Freeswitch-users] Error in default Sofia profile checking In-Reply-To: <1452e2980907102223r681f59e9kad0cf6883cda4df2@mail.gmail.com> References: <1452e2980907102223r681f59e9kad0cf6883cda4df2@mail.gmail.com> Message-ID: <20090711065940.GA28162@jdc.jasonjgw.net> velusamy velu wrote: > When I register my Softphone(Twinkle) with predefined sofia > registration("1000" with password "1234"). I have got the following error > in FreeSWITCH console. > > "2009-07-11 09:37:16 [ERR] sofia_reg.c:1135 > sofia_reg_handle_sip_i_register() NO CONTACT!" Activate sip tracing on the profile (e.g., sofia profile internal siptrace on), try to register again and save the trace. This should help you to solve the problem. From mrene_lists at avgs.ca Sat Jul 11 00:28:56 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Sat, 11 Jul 2009 03:28:56 -0400 Subject: [Freeswitch-users] Error in default Sofia profile checking In-Reply-To: <1452e2980907102223r681f59e9kad0cf6883cda4df2@mail.gmail.com> References: <1452e2980907102223r681f59e9kad0cf6883cda4df2@mail.gmail.com> Message-ID: <3EFB9073-9339-4A06-BADF-130E69A104CF@avgs.ca> Chances are the registering UA didnt provide a Contact header (required by rfc3261) Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 11-Jul-09, at 1:23 AM, velusamy velu wrote: > Dear Friends, > When I register my Softphone(Twinkle) with predefined > sofia registration("1000" with password "1234"). I have got the > following error in FreeSWITCH console. > > "2009-07-11 09:37:16 [ERR] sofia_reg.c:1135 > sofia_reg_handle_sip_i_ > register() NO CONTACT!" > > Please help me to solve this problem... > > Regards, > K.Velusamy. > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.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 Sat Jul 11 00:48:18 2009 From: jason at jasonjgw.net (Jason White) Date: Sat, 11 Jul 2009 17:48:18 +1000 Subject: [Freeswitch-users] Error in default Sofia profile checking In-Reply-To: <3EFB9073-9339-4A06-BADF-130E69A104CF@avgs.ca> References: <1452e2980907102223r681f59e9kad0cf6883cda4df2@mail.gmail.com> <3EFB9073-9339-4A06-BADF-130E69A104CF@avgs.ca> Message-ID: <20090711074818.GA31938@jdc.jasonjgw.net> Mathieu Rene wrote: > Chances are the registering UA didnt provide a Contact header > (required by rfc3261) Just what I thought, hence the suggestion to obtain a sip trace. From velu.technical at gmail.com Sat Jul 11 01:29:13 2009 From: velu.technical at gmail.com (velusamy velu) Date: Sat, 11 Jul 2009 13:59:13 +0530 Subject: [Freeswitch-users] ERROR in Sofia internal profile Message-ID: <1452e2980907110129i145cbc3fpe53028c92d9b127a@mail.gmail.com> Dear Friends, When I reload the mod_sofia I have got the following error. "2009-07-11 13:19:32 [ERR] sofia.c:739 sofia_profile_thread_run() Error Creating SIP UA for profile: internal" Please any one explain about this error and please give any suggestions to solve this problem.. Thanks in Advance.. Regards, K.Velusamy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090711/12547731/attachment-0002.html From dujinfang at gmail.com Sat Jul 11 02:10:55 2009 From: dujinfang at gmail.com (Seven Du) Date: Sat, 11 Jul 2009 17:10:55 +0800 Subject: [Freeswitch-users] ERROR in Sofia internal profile In-Reply-To: <1452e2980907110129i145cbc3fpe53028c92d9b127a@mail.gmail.com> References: <1452e2980907110129i145cbc3fpe53028c92d9b127a@mail.gmail.com> Message-ID: <096BCE35-ABA3-4CA0-AF59-CCA3806F40BE@gmail.com> chances are the tcp/udp port (5060?) already used by other software, are you running softphone on the same computer? On Jul 11, 2009, at 4:29 PM, velusamy velu wrote: > Dear Friends, > When I reload the mod_sofia I have got the following error. > > "2009-07-11 13:19:32 [ERR] sofia.c:739 sofia_profile_thread_run() > Error Creating SIP UA for profile: internal" > > Please any one explain about this error and please give any > suggestions to solve this problem.. > > Thanks in Advance.. > > Regards, > K.Velusamy > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mcampbellsmith at gmail.com Sat Jul 11 06:15:42 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Sat, 11 Jul 2009 23:15:42 +1000 Subject: [Freeswitch-users] 2 voicemail questions In-Reply-To: <344FC083-5916-4232-AEB5-3504626566DF@jerris.com> References: <33c87fa30907100257w29c628cbnb641d09a75bc22b8@mail.gmail.com> <344FC083-5916-4232-AEB5-3504626566DF@jerris.com> Message-ID: <33c87fa30907110615n3b7b7baclc6276d659fda1e85@mail.gmail.com> Hi Mike, This was my dialplan (extracted from my last email): >> 2. How can I make Freeswitch dial a number AFTER a voicemail is left? >api Hangup hook? i want the 'voicemail' application to appear to call the extension to notify the user that there is a waiting message. This is an extract from my dialplan.xml: This only works if the B leg (ie voicemail application) hangs up first. This would be an unusual situation and does not achieve what I want... is there any other way to achieve this? On Sat, Jul 11, 2009 at 2:04 AM, Michael Jerris wrote: > could you post how you tired to do it in dialplan that didn't work? > > Mike > > On Jul 10, 2009, at 5:57 AM, Mark Campbell-Smith wrote: > >> Hi! >>>> 1. Can I email the voicemail message to multiple email addresses? >> A comma separated list does not work in the extensions.xml file >> (1000.xml), but it does work if I hard code the email addresses into >> the notify-voicemail.tpl file. >> >> Could this be added to the switch so that it can handle comma >> separated lists? >> >>>> 2. How can I make Freeswitch dial a number AFTER a voicemail is >>>> left? >> >>> api Hangup hook? >> >> i want the 'voicemail' application to appear to call the extension to >> notify the user that there is a waiting message. ?This is an extract >> from my dialplan.xml: >> >> ? ? ? >> ? ? ? ? >> ? ? ? ? >> ? ? ? ? >> ? ? ? ? >> ? ? ? ? >> ? ? ? ? >> ? ? ? ?> data="user/${dialed_extension}@${domain_name}"/> >> ? ? ? ? >> ? ? ? ? >> ? ? ? ? >> ? ? ? >> ? ? ?> field="${vm_boxcount(${destination_number}@${domain_name})}" >> expression="^(1)$"> >> ? ? ? ? >> ? ? ? ? >> >> This only works if the B leg (ie voicemail application) hangs up >> first. ?This would be an unusual situation and does not achieve what I >> want... is there any other way to achieve this? >> >> Thanks >> >>> Hi! >>> >>> I have 2 questions regarding voicemail ... >>> >>> 1. Can I email the voicemail message to multiple email addresses? ?If >>> so, what format is this in? >>> ? ? >> >> Try a comma sep. list. ?Not sure if it will work. >> >>> >>> 2. How can I make Freeswitch dial a number AFTER a voicemail is left? >> >> api Hangup hook? >> >> I g >>> From: Brian West >> >>> On Jul 7, 2009, at 9:11 AM, Mark Campbell-Smith wrote: >> >>> Hi! >>> >>> I have 2 questions regarding voicemail ... >>> >>> 1. Can I email the voicemail message to multiple email addresses? ?If >>> so, what format is this in? >>> ? ? >> >>> Try a comma sep. list. ?Not sure if it will work. >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From dftoro at yahoo.com Sat Jul 11 06:15:57 2009 From: dftoro at yahoo.com (Diego Toro) Date: Sat, 11 Jul 2009 06:15:57 -0700 (PDT) Subject: [Freeswitch-users] managed_mod directories Message-ID: <509787.62310.qm@web33504.mail.mud.yahoo.com> Hello, ? What error do you get when dll is put on mod/managed ?, I work with dll's on mod/managed although I changed loadfile by loadfrom on loader.cs. ? Diego --- On Fri, 7/10/09, Phillip Jones wrote: From: Phillip Jones Subject: Re: [Freeswitch-users] managed_mod directories To: freeswitch-users at lists.freeswitch.org Date: Friday, July 10, 2009, 6:22 PM It is looking in mod. It required the mod\managed directory, but if I place my dll in mod\managed it fails. DLL must be in mod - mod\managed is empty. My app works fine though Phil On Fri, Jul 10, 2009 at 4:48 PM, Michael Giagnocavo wrote: You?re saying that it requires the managed DLL to be in both the mod and mod\managed directory? ?What error do you get if it?s only in mod? It?s been months, but I just checked loader.cs and it looks explicitly in the managed directory to resolve assemblies as well as to scan to load them. ? -Michael ? From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Phillip Jones Sent: Friday, July 10, 2009 11:46 AM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] managed_mod directories ? Hi there, Using windows with the pre-compiled binary / msi found via the WIKI Using mod_managed with no problems however: mod_managed appears to require I create a directory 'managed' under C:\Program Files (x86)\FreeSWITCH\mod BUT also requires that I place my .dll in C:\Program Files (x86)\FreeSWITCH\mod and NOT C:\Program Files (x86)\FreeSWITCH\mod\managed thus making C:\Program Files (x86)\FreeSWITCH\mod\managed redundant Anyone else seen this behavior? Thanks! Phillip Jones _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -----Inline Attachment Follows----- _______________________________________________ Freeswitch-users mailing list Freeswitch-users at 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/20090711/c95949a3/attachment-0002.html From Prometheus001 at gmx.net Sat Jul 11 06:46:56 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Sat, 11 Jul 2009 15:46:56 +0200 Subject: [Freeswitch-users] Error in default Sofia profile checking In-Reply-To: <3EFB9073-9339-4A06-BADF-130E69A104CF@avgs.ca> References: <1452e2980907102223r681f59e9kad0cf6883cda4df2@mail.gmail.com> <3EFB9073-9339-4A06-BADF-130E69A104CF@avgs.ca> Message-ID: <4A5897D0.7060005@gmx.net> I have several Twinkles running against freeswitch on a locally installed machine (FS acts as a SIP/TLS proxy). So in general Twinkle works (on various Ubuntu machines from 7 upto 9 with various Twinkle versions). It must be some kind of setting in Twinkle. E.g. * set the local Twinkle SIP UDP port to 5062 in general settings * Set the right network interface (e.g. eth0) * In the profile do not set the realm * Allow missing contact header on 200 OK Best regards Peter Mathieu Rene schrieb: > Chances are the registering UA didnt provide a Contact header > (required by rfc3261) > > Mathieu Rene > Avant-Garde Solutions Inc > Office: + 1 (514) 664-1044 x100 > Cell: +1 (514) 664-1044 x200 > mrene at avgs.ca > > > > > On 11-Jul-09, at 1:23 AM, velusamy velu wrote: > > >> Dear Friends, >> When I register my Softphone(Twinkle) with predefined >> sofia registration("1000" with password "1234"). I have got the >> following error in FreeSWITCH console. >> >> "2009-07-11 09:37:16 [ERR] sofia_reg.c:1135 >> sofia_reg_handle_sip_i_ >> register() NO CONTACT!" >> >> Please help me to solve this problem... >> >> Regards, >> K.Velusamy. >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.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 Jul 11 07:07:29 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 11 Jul 2009 09:07:29 -0500 Subject: [Freeswitch-users] Error in default Sofia profile checking In-Reply-To: <4A5897D0.7060005@gmx.net> References: <1452e2980907102223r681f59e9kad0cf6883cda4df2@mail.gmail.com> <3EFB9073-9339-4A06-BADF-130E69A104CF@avgs.ca> <4A5897D0.7060005@gmx.net> Message-ID: <0FE28F43-2AAA-4928-AF72-73845A129801@freeswitch.org> http://jira.freeswitch.org/browse/MODENDP-86 /b On Jul 11, 2009, at 8:46 AM, Peter P GMX wrote: > I have several Twinkles running against freeswitch on a locally > installed machine (FS acts as a SIP/TLS proxy). > So in general Twinkle works (on various Ubuntu machines from 7 upto 9 > with various Twinkle versions). It must be some kind of setting in > Twinkle. E.g. > > * set the local Twinkle SIP UDP port to 5062 in general settings > * Set the right network interface (e.g. eth0) > * In the profile do not set the realm > * Allow missing contact header on 200 OK > > Best regards > Peter > > > > Mathieu Rene schrieb: >> Chances are the registering UA didnt provide a Contact header >> (required by rfc3261) >> >> Mathieu Rene >> Avant-Garde Solutions Inc >> Office: + 1 (514) 664-1044 x100 >> Cell: +1 (514) 664-1044 x200 >> mrene at avgs.ca >> >> >> >> >> On 11-Jul-09, at 1:23 AM, velusamy velu wrote: >> >> >>> Dear Friends, >>> When I register my Softphone(Twinkle) with predefined >>> sofia registration("1000" with password "1234"). I have got the >>> following error in FreeSWITCH console. >>> >>> "2009-07-11 09:37:16 [ERR] sofia_reg.c:1135 >>> sofia_reg_handle_sip_i_ >>> register() NO CONTACT!" >>> >>> Please help me to solve this problem... >>> >>> Regards, >>> K.Velusamy. >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From pjintheusa at gmail.com Sat Jul 11 09:25:28 2009 From: pjintheusa at gmail.com (Phillip Jones) Date: Sat, 11 Jul 2009 12:25:28 -0400 Subject: [Freeswitch-users] managed_mod directories In-Reply-To: <509787.62310.qm@web33504.mail.mud.yahoo.com> References: <509787.62310.qm@web33504.mail.mud.yahoo.com> Message-ID: <367751820907110925o61168b64t1db5b6944e0dac7e@mail.gmail.com> Hi, If I place the DLL in mod\managed I get the following error: [err] mod_managed.cpp:287 Assembly::LoadFrom failed: system.IO.FileNotFoundException: Could not load file or assembly 'file:///c:program files (x86)\Freeswitch\mod\freeSWITCH.Managed.dll' or one of its dependencies. The system could not find the file specified. As I said. When I place freeSWITCH.Managed.dll straight into \mod then everything works fine. Thanks Phil On Sat, Jul 11, 2009 at 9:15 AM, Diego Toro wrote: > Hello, > > What error do you get when dll is put on mod/managed ?, I work with dll's > on mod/managed although I changed loadfile by loadfrom on loader.cs. > > Diego > > > --- On *Fri, 7/10/09, Phillip Jones * wrote: > > > From: Phillip Jones > Subject: Re: [Freeswitch-users] managed_mod directories > To: freeswitch-users at lists.freeswitch.org > Date: Friday, July 10, 2009, 6:22 PM > > > It is looking in mod. > > It required the mod\managed directory, but if I place my dll in mod\managed > it fails. DLL must be in mod - mod\managed is empty. > > My app works fine though > > > Phil > > > On Fri, Jul 10, 2009 at 4:48 PM, Michael Giagnocavo > > wrote: > >> You?re saying that it requires the managed DLL to be in both the mod and >> mod\managed directory? What error do you get if it?s only in mod? It?s been >> months, but I just checked loader.cs and it looks explicitly in the managed >> directory to resolve assemblies as well as to scan to load them. >> >> -Michael >> >> *From:* freeswitch-users-bounces at lists.freeswitch.org[mailto: >> freeswitch-users-bounces at lists.freeswitch.org] >> *On Behalf Of *Phillip Jones >> *Sent:* Friday, July 10, 2009 11:46 AM >> *To:* freeswitch-users at lists.freeswitch.org >> *Subject:* [Freeswitch-users] managed_mod directories >> >> Hi there, >> >> >> Using windows with the pre-compiled binary / msi found via the WIKI >> >> Using mod_managed with no problems however: >> >> mod_managed appears to require I create a directory 'managed' under >> C:\Program Files (x86)\FreeSWITCH\mod >> >> BUT also requires that I place my .dll in C:\Program Files >> (x86)\FreeSWITCH\mod and NOT C:\Program Files (x86)\FreeSWITCH\mod\managed >> >> thus making C:\Program Files (x86)\FreeSWITCH\mod\managed redundant >> >> Anyone else seen this behavior? >> >> Thanks! >> >> >> Phillip Jones >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > -----Inline Attachment Follows----- > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090711/c9820dc7/attachment-0002.html From jlenk at frontiernet.net Sat Jul 11 10:43:43 2009 From: jlenk at frontiernet.net (Jeff Lenk) Date: Sat, 11 Jul 2009 10:43:43 -0700 (PDT) Subject: [Freeswitch-users] managed_mod directories In-Reply-To: <367751820907110925o61168b64t1db5b6944e0dac7e@mail.gmail.com> References: <367751820907101046l7b41a2fdl7cac3d05b37a80f7@mail.gmail.com> <509787.62310.qm@web33504.mail.mud.yahoo.com> <367751820907110925o61168b64t1db5b6944e0dac7e@mail.gmail.com> Message-ID: <1247334223093-3243183.post@n2.nabble.com> Hi, The base dll (FreeSWITCH.Managed.dll) is loaded from /mod - additional managed dlls are loaded from /mod/managed. This is designed to allow your dll's to be built and maintained independant of the FS build files. You can simply just drop your dlls into mod/managed and they will be loaded and available for use(this happens at FS startup). The base managed dll (FreeSWITCH.Managed.dll) is only really supposed to be used for loader support and the demo classes - you should place your code in your own dll. - Jeff Phillip Jones-2 wrote: > > Hi, > > If I place the DLL in mod\managed I get the following error: > > [err] mod_managed.cpp:287 Assembly::LoadFrom failed: > system.IO.FileNotFoundException: Could not load file or assembly > 'file:///c:program files (x86)\Freeswitch\mod\freeSWITCH.Managed.dll' or > one > of its dependencies. The system could not find the file specified. > > As I said. When I place freeSWITCH.Managed.dll straight into \mod then > everything works fine. > > Thanks > > > Phil > > > On Sat, Jul 11, 2009 at 9:15 AM, Diego Toro wrote: > >> Hello, >> >> What error do you get when dll is put on mod/managed ?, I work with dll's >> on mod/managed although I changed loadfile by loadfrom on loader.cs. >> >> Diego >> >> >> --- On *Fri, 7/10/09, Phillip Jones * wrote: >> >> >> From: Phillip Jones >> Subject: Re: [Freeswitch-users] managed_mod directories >> To: freeswitch-users at lists.freeswitch.org >> Date: Friday, July 10, 2009, 6:22 PM >> >> >> It is looking in mod. >> >> It required the mod\managed directory, but if I place my dll in >> mod\managed >> it fails. DLL must be in mod - mod\managed is empty. >> >> My app works fine though >> >> >> Phil >> >> >> On Fri, Jul 10, 2009 at 4:48 PM, Michael Giagnocavo >> >> > wrote: >> >>> You?re saying that it requires the managed DLL to be in both the mod >>> and >>> mod\managed directory? What error do you get if it?s only in mod? It?s >>> been >>> months, but I just checked loader.cs and it looks explicitly in the >>> managed >>> directory to resolve assemblies as well as to scan to load them. >>> >>> -Michael >>> >>> *From:* >>> freeswitch-users-bounces at lists.freeswitch.org[mailto: >>> freeswitch-users-bounces at lists.freeswitch.org] >>> *On Behalf Of *Phillip Jones >>> *Sent:* Friday, July 10, 2009 11:46 AM >>> *To:* >>> freeswitch-users at lists.freeswitch.org >>> *Subject:* [Freeswitch-users] managed_mod directories >>> >>> Hi there, >>> >>> >>> Using windows with the pre-compiled binary / msi found via the WIKI >>> >>> Using mod_managed with no problems however: >>> >>> mod_managed appears to require I create a directory 'managed' under >>> C:\Program Files (x86)\FreeSWITCH\mod >>> >>> BUT also requires that I place my .dll in C:\Program Files >>> (x86)\FreeSWITCH\mod and NOT C:\Program Files >>> (x86)\FreeSWITCH\mod\managed >>> >>> thus making C:\Program Files (x86)\FreeSWITCH\mod\managed redundant >>> >>> Anyone else seen this behavior? >>> >>> Thanks! >>> >>> >>> Phillip Jones >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> -----Inline Attachment Follows----- >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/managed_mod-directories-tp3240303p3243183.html Sent from the freeswitch-users mailing list archive at Nabble.com. From nik.middleton at noblesolutions.co.uk Sat Jul 11 12:19:02 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Sat, 11 Jul 2009 20:19:02 +0100 Subject: [Freeswitch-users] Setting channel variables using event socket Message-ID: Hi Guys, Is it possible to set a channel variable while a call is in progress using an outbound event socket? I have a listening process that examines the hang-up events and it would be neat if it could also get some variables that I have set mid call as well. Note: I know it's possible to set them in the originate but that's not what I'm after Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090711/41c4f3df/attachment-0002.html From brian at freeswitch.org Sat Jul 11 12:50:57 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 11 Jul 2009 14:50:57 -0500 Subject: [Freeswitch-users] Setting channel variables using event socket In-Reply-To: References: Message-ID: <7525E12C-1213-4502-8993-F01300D5D644@freeswitch.org> uuid_setvar /b On Jul 11, 2009, at 2:19 PM, Nik Middleton wrote: > Hi Guys, > > Is it possible to set a channel variable while a call is in progress > using an outbound event socket? I have a listening process that > examines the hang-up events and it would be neat if it could also > get some variables that I have set mid call as well. Note: I know > it?s possible to set them in the originate but that?s not what I?m > after > > Regards, > ________ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090711/7e248e5c/attachment-0002.html From nik.middleton at noblesolutions.co.uk Sat Jul 11 14:08:50 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Sat, 11 Jul 2009 22:08:50 +0100 Subject: [Freeswitch-users] Setting channel variables using event socket In-Reply-To: <7525E12C-1213-4502-8993-F01300D5D644@freeswitch.org> References: <7525E12C-1213-4502-8993-F01300D5D644@freeswitch.org> Message-ID: Excellent. Do I need to supply uuid on an outbound socket? Regards ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: 11 July 2009 20:51 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Setting channel variables using event socket uuid_setvar /b On Jul 11, 2009, at 2:19 PM, Nik Middleton wrote: Hi Guys, Is it possible to set a channel variable while a call is in progress using an outbound event socket? I have a listening process that examines the hang-up events and it would be neat if it could also get some variables that I have set mid call as well. Note: I know it's possible to set them in the originate but that's not what I'm after Regards, ________ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090711/0212f069/attachment-0002.html From brian at freeswitch.org Sat Jul 11 14:21:42 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 11 Jul 2009 16:21:42 -0500 Subject: [Freeswitch-users] Setting channel variables using event socket In-Reply-To: References: <7525E12C-1213-4502-8993-F01300D5D644@freeswitch.org> Message-ID: I think you do ... /b On Jul 11, 2009, at 4:08 PM, Nik Middleton wrote: > Excellent. Do I need to supply uuid on an outbound socket? > > Regards > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090711/17242f4c/attachment-0002.html From msc at freeswitch.org Sat Jul 11 14:50:55 2009 From: msc at freeswitch.org (Michael S Collins) Date: Sat, 11 Jul 2009 14:50:55 -0700 Subject: [Freeswitch-users] Setting channel variables using event socket In-Reply-To: References: <7525E12C-1213-4502-8993-F01300D5D644@freeswitch.org> Message-ID: <0E283956-3903-4822-9BD9-684803A15EC4@freeswitch.org> IIRC you need to supply the uuid because the socket doesn't make any assumptions about the APIs you send. -MC Sent from my iPhone On Jul 11, 2009, at 2:21 PM, Brian West wrote: > I think you do ... > > /b > > On Jul 11, 2009, at 4:08 PM, Nik Middleton wrote: > >> Excellent. Do I need to supply uuid on an outbound socket? >> >> 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/20090711/4cde569e/attachment-0002.html From anthony.minessale at gmail.com Sat Jul 11 14:51:03 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sat, 11 Jul 2009 16:51:03 -0500 Subject: [Freeswitch-users] Setting channel variables using event socket In-Reply-To: References: <7525E12C-1213-4502-8993-F01300D5D644@freeswitch.org> Message-ID: <191c3a030907111451u7ddb17c3r66cc4371c58cc63d@mail.gmail.com> I think also sendmsg with command execute works in this case if you are using async socket but uuid_setvar always works in all cases On Jul 11, 2009 4:27 PM, "Brian West" wrote: I think you do ... /b On Jul 11, 2009, at 4:08 PM, Nik Middleton wrote: > Excellent. Do I need to supply uuid on an out... _______________________________________________ Freeswitch-users mailing list Freeswitch-users at 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/20090711/acdb945f/attachment-0002.html From elihayun at gmail.com Sun Jul 12 01:49:20 2009 From: elihayun at gmail.com (Eli Hayun) Date: Sun, 12 Jul 2009 11:49:20 +0300 Subject: [Freeswitch-users] Getting xml_request in LUA Message-ID: <4A59A390.7030000@savion.huji.ac.il> In the Perl example I found: How to access request parameters and how to return data You have two hashes that are populated for you by freeswitch. Those hashes are: * %XML_REQUEST * %XML_DATA I want to use LUA to set the directory and dialplan xml. How do I get the XML_REQUEST/XML_DATA from LUA? Thanks Eli Hayun From info at nalawo.com Sun Jul 12 06:54:57 2009 From: info at nalawo.com (Maarten De Maeyer) Date: Sun, 12 Jul 2009 13:54:57 +0000 (GMT) Subject: [Freeswitch-users] QSIG Message-ID: <1964152093.1076.1247406897795.JavaMail.mail@webmail25> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090712/97dafa9b/attachment-0002.html From dome at tel.co.th Sun Jul 12 09:59:05 2009 From: dome at tel.co.th (Dome Charoenyost) Date: Sun, 12 Jul 2009 23:59:05 +0700 Subject: [Freeswitch-users] Originate in Dial plan Message-ID: <8ccbff060907120959ma7c0ey22adeae196f872c3@mail.gmail.com> Dear sir, I want to user dialplan callback to customer. is posible to to this is dialplan XML ? Now i use javascript. my call flow. 1. customer call 2. FS rining and wait until customer hangup 3. callback to customer number Best Regards. Dome C. From mike at jerris.com Sun Jul 12 09:58:15 2009 From: mike at jerris.com (Michael Jerris) Date: Sun, 12 Jul 2009 12:58:15 -0400 Subject: [Freeswitch-users] QSIG In-Reply-To: <1964152093.1076.1247406897795.JavaMail.mail@webmail25> References: <1964152093.1076.1247406897795.JavaMail.mail@webmail25> Message-ID: We have no qsig support. On Jul 12, 2009, at 9:54 AM, Maarten De Maeyer wrote: > > Hi, > > Can someone tell me how complete QSIG support is in FS ? Is there a > config example available ? I need to connect FS to another pbx with > QSIG. > Any tips/advice more than welcome. > > Thanks. > > MdM > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090712/e7c73975/attachment-0002.html From q.edward at gmail.com Sun Jul 12 13:45:16 2009 From: q.edward at gmail.com (Edward Q.) Date: Sun, 12 Jul 2009 16:45:16 -0400 Subject: [Freeswitch-users] outbound_caller_id dynamic from mysql ? Message-ID: <89313a90907121345v76a1fe41o35fd70753dded41c@mail.gmail.com> Hi guys. I was just wondering if it is possible to have the outbound_caller_id dynamically pulled from MySQL db ? If it is can anyone please point me in the right direction ? Thanks in advanced to all for all your help. Thanks Ed -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090712/1594cec7/attachment-0002.html From mike at jerris.com Sun Jul 12 13:48:46 2009 From: mike at jerris.com (Michael Jerris) Date: Sun, 12 Jul 2009 16:48:46 -0400 Subject: [Freeswitch-users] outbound_caller_id dynamic from mysql ? In-Reply-To: <89313a90907121345v76a1fe41o35fd70753dded41c@mail.gmail.com> References: <89313a90907121345v76a1fe41o35fd70753dded41c@mail.gmail.com> Message-ID: See mod_xml_curl On Jul 12, 2009, at 4:45 PM, "Edward Q." wrote: > Hi guys. > > I was just wondering if it is possible to have the > outbound_caller_id dynamically pulled from MySQL db ? > If it is can anyone please point me in the right direction ? > Thanks in advanced to all for all your help. > Thanks > Ed > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From q.edward at gmail.com Sun Jul 12 13:59:13 2009 From: q.edward at gmail.com (Edward Q.) Date: Sun, 12 Jul 2009 16:59:13 -0400 Subject: [Freeswitch-users] outbound_caller_id dynamic from mysql ? In-Reply-To: References: <89313a90907121345v76a1fe41o35fd70753dded41c@mail.gmail.com> Message-ID: <89313a90907121359n7aa219afoc14e45da55ff60af@mail.gmail.com> Thank you for the prompt reply Michael .. Looking through the wiki now Thanks to all and have all of you a great day. Ed On Sun, Jul 12, 2009 at 4:48 PM, Michael Jerris wrote: > See mod_xml_curl > > On Jul 12, 2009, at 4:45 PM, "Edward Q." wrote: > > > Hi guys. > > > > I was just wondering if it is possible to have the > > outbound_caller_id dynamically pulled from MySQL db ? > > If it is can anyone please point me in the right direction ? > > Thanks in advanced to all for all your help. > > Thanks > > Ed > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090712/13c52525/attachment-0002.html From nik.middleton at noblesolutions.co.uk Sun Jul 12 15:10:37 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Sun, 12 Jul 2009 23:10:37 +0100 Subject: [Freeswitch-users] Setting channel variables using event socket In-Reply-To: <191c3a030907111451u7ddb17c3r66cc4371c58cc63d@mail.gmail.com> References: <7525E12C-1213-4502-8993-F01300D5D644@freeswitch.org> <191c3a030907111451u7ddb17c3r66cc4371c58cc63d@mail.gmail.com> Message-ID: HI Guys, Can't seem to get this to work call-command: execute execute-app-name: uuid_setvar execute-app-arg: cad8eb99-cdcd-4d0d-9453-20b8d9d71859 fred=out_to_lunch Tried various permutations, but still nothing stored when the channel is hung up Can anyone tell me what I'm doing wrong? Regards, ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: 11 July 2009 22:51 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Setting channel variables using event socket I think also sendmsg with command execute works in this case if you are using async socket but uuid_setvar always works in all cases On Jul 11, 2009 4:27 PM, "Brian West" wrote: I think you do ... /b On Jul 11, 2009, at 4:08 PM, Nik Middleton wrote: > Excellent. Do I need to supply uuid on an out... _______________________________________________ Freeswitch-users mailing list Freeswitch-users at 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/20090712/99431a7c/attachment-0002.html From brian at freeswitch.org Sun Jul 12 15:23:45 2009 From: brian at freeswitch.org (Brian West) Date: Sun, 12 Jul 2009 17:23:45 -0500 Subject: [Freeswitch-users] Setting channel variables using event socket In-Reply-To: References: <7525E12C-1213-4502-8993-F01300D5D644@freeswitch.org> <191c3a030907111451u7ddb17c3r66cc4371c58cc63d@mail.gmail.com> Message-ID: If you're going to do it that way you can just use set. uuid_setvar is an api call... /b On Jul 12, 2009, at 5:10 PM, Nik Middleton wrote: > HI Guys, > > Can?t seem to get this to work > > call-command: execute > execute-app-name: uuid_setvar > execute-app-arg: cad8eb99-cdcd-4d0d-9453-20b8d9d71859 > fred=out_to_lunch > > Tried various permutations, but still nothing stored when the > channel is hung up > > Can anyone tell me what I?m doing wrong? > > Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090712/f6a62091/attachment-0002.html From nik.middleton at noblesolutions.co.uk Sun Jul 12 15:33:45 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Sun, 12 Jul 2009 23:33:45 +0100 Subject: [Freeswitch-users] Setting channel variables using event socket In-Reply-To: References: <7525E12C-1213-4502-8993-F01300D5D644@freeswitch.org><191c3a030907111451u7ddb17c3r66cc4371c58cc63d@mail.gmail.com> Message-ID: As in call-command: set joe=out_to_lunch ? ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: 12 July 2009 23:24 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Setting channel variables using event socket If you're going to do it that way you can just use set. uuid_setvar is an api call... /b On Jul 12, 2009, at 5:10 PM, Nik Middleton wrote: HI Guys, Can't seem to get this to work call-command: execute execute-app-name: uuid_setvar execute-app-arg: cad8eb99-cdcd-4d0d-9453-20b8d9d71859 fred=out_to_lunch Tried various permutations, but still nothing stored when the channel is hung up Can anyone tell me what I'm doing wrong? Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090712/fea97d65/attachment-0002.html From brian at freeswitch.org Sun Jul 12 16:00:24 2009 From: brian at freeswitch.org (Brian West) Date: Sun, 12 Jul 2009 18:00:24 -0500 Subject: [Freeswitch-users] Setting channel variables using event socket In-Reply-To: References: <7525E12C-1213-4502-8993-F01300D5D644@freeswitch.org><191c3a030907111451u7ddb17c3r66cc4371c58cc63d@mail.gmail.com> Message-ID: call-command: execute execute-app-name: set execute-app-arg: fred=out_to_lunch On Jul 12, 2009, at 5:33 PM, Nik Middleton wrote: > As in > > call-command: set joe=out_to_lunch ? > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090712/14d15cc6/attachment-0002.html From timuckun at gmail.com Sun Jul 12 21:07:06 2009 From: timuckun at gmail.com (Tim Uckun) Date: Mon, 13 Jul 2009 16:07:06 +1200 Subject: [Freeswitch-users] Dialogic cards Message-ID: <855e4dcf0907122107q392a9ca1h851dfbd57e2518ca@mail.gmail.com> We have some older dialogic cards (D300 series E1 cards) and I am wondering if freeswitch can support these cards. Thanks. From velu.technical at gmail.com Sun Jul 12 22:38:48 2009 From: velu.technical at gmail.com (velusamy velu) Date: Mon, 13 Jul 2009 11:08:48 +0530 Subject: [Freeswitch-users] Error in default Sofia profile checking Message-ID: <1452e2980907122238h1392b903o6b1179e423fb7893@mail.gmail.com> Dear Peter, I have followed your steps, For me my FS and Twinkle running in separate machine. But, I am still receiving the same error "[ERR] sofia_reg.c:1135 sofia_reg_handle_sip_i_register() NO CONTACT!" Please give any suggestions to rectify this error.. Thanks in Advance, Regards, K.Velusamy. > > > On Jul 11, 2009, at 8:46 AM, Peter P GMX wrote: > > > I have several Twinkles running against freeswitch on a locally > > installed machine (FS acts as a SIP/TLS proxy). > > So in general Twinkle works (on various Ubuntu machines from 7 upto 9 > > with various Twinkle versions). It must be some kind of setting in > > Twinkle. E.g. > > > > * set the local Twinkle SIP UDP port to 5062 in general settings > > * Set the right network interface (e.g. eth0) > > * In the profile do not set the realm > > * Allow missing contact header on 200 OK > > > > Best regards > > Peter > > > > > > > > Mathieu Rene schrieb: > >> Chances are the registering UA didnt provide a Contact header > >> (required by rfc3261) > >> > >> Mathieu Rene > >> Avant-Garde Solutions Inc > >> Office: + 1 (514) 664-1044 x100 > >> Cell: +1 (514) 664-1044 x200 > >> mrene at avgs.ca > >> > >> > >> > >> > >> On 11-Jul-09, at 1:23 AM, velusamy velu wrote: > >> > >> > >>> Dear Friends, > >>> When I register my Softphone(Twinkle) with predefined > >>> sofia registration("1000" with password "1234"). I have got the > >>> following error in FreeSWITCH console. > >>> > >>> "2009-07-11 09:37:16 [ERR] sofia_reg.c:1135 > >>> sofia_reg_handle_sip_i_ > >>> register() NO CONTACT!" > >>> > >>> Please help me to solve this problem... > >>> > >>> Regards, > >>> K.Velusamy. > >>> _______________________________________________ > >>> Freeswitch-users mailing list > >>> Freeswitch-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> http://www.freeswitch.org > >>> > >> > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > >> > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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/20090713/54b29a3f/attachment-0002.html From sridhart at alcatel-lucent.com Mon Jul 13 00:07:26 2009 From: sridhart at alcatel-lucent.com (Rajagopal, Sridhar (Sridhar)) Date: Mon, 13 Jul 2009 12:37:26 +0530 Subject: [Freeswitch-users] Help Regarding memory leak with freeswitch Message-ID: <9389DD3DDD6B9144B147CE564C6599B9082FC07020@INBANSXCHMBSA3.in.alcatel-lucent.com> Hi all, I am running freeswitch on powerpc processor. I see memory being allocated for each subsequent REGISTER requests coming to freeswitch. But not all the memory allocated is not freed. If I run the code for two days the system is running out of memory (RAM available to me is very less). The same memory issue is happening even for calls. Please let me know if any body has seen this issue. Please let me know how I can go ahead and debug this issue. Thanks in advance for the help. Regards, Sridhar -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090713/9fc85a3d/attachment-0002.html From helmut.kuper at ewetel.de Mon Jul 13 00:53:51 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Mon, 13 Jul 2009 09:53:51 +0200 Subject: [Freeswitch-users] pocketsphinx In-Reply-To: <4A574F0D.7040604@gmx.net> References: <4A5739DE.1080800@ewetel.de> <4A574F0D.7040604@gmx.net> Message-ID: <4A5AE80F.1000904@ewetel.de> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi Peter, hmmm well, I had the same idea and I tested it! Buuuuut ... you have to make sure that the english grammar/acousticModel is able to cover all german noises. E.g. I tried to detect "Burke", "Jan" and "Gerd". I was able to map Burke successfully in default.dic. Otherwise I had to say "B?hrki" ... I did the same with "Jan" - but when I tried to detect Jan I always got Gerd (with and witout mapping ind default.dic) ... Quite strange and not really usable for (german) customers. But some typical software magic happened on my way: During my tests I had somehow a configuration using the voxforge files which was working within FS. But I can't reproduce it. I configured serveral files at the same time and used for reloading "reloadxml" and "reload mod_pocketsphinx" instead of rebooting FS. When it worked, FS was able to detect "Burke", "Jan" and "Gerd" correctly without modifying the dictionary... Is there any manual about pocketsphinx and its config files, which can explain how PS is working in more detail? Currently I walk with a flashlight in the dark ... regards Helmut On 10.07.2009 16:24, Peter P GMX wrote: > Hello Helmut, > > I looked at these dic files. Their content (look at all the qq's) is > quite different from the dic files supplied with freeswitch pocketsphinx. > As I remember the CMU dict file format has changed in April 2008. > Maybe there is a converter somewhere? > > I was thinking of just enhancing the current dict file for some german > words I need, but did not test it so far. This should be possible > without modifying the underlying grammar. > http://en.wikipedia.org/wiki/CMU_Pronouncing_Dictionary > I would love to hear when you have had any progress on this. > > Best regards > Peter -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) iD8DBQFKWugP4tZeNddg3dwRAhjkAKConTWen4bq5BxSg23F6keZeY2CIACffAks yyOVZOkROr8tfUNGMv4t9o8= =lC+W -----END PGP SIGNATURE----- From nik.middleton at noblesolutions.co.uk Mon Jul 13 01:51:32 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Mon, 13 Jul 2009 09:51:32 +0100 Subject: [Freeswitch-users] Help Regarding memory leak with freeswitch In-Reply-To: <9389DD3DDD6B9144B147CE564C6599B9082FC07020@INBANSXCHMBSA3.in.alcatel-lucent.com> References: <9389DD3DDD6B9144B147CE564C6599B9082FC07020@INBANSXCHMBSA3.in.alcatel-lucent.com> Message-ID: If this is Linux, there's nothing wrong with it using most of the memory, if it starts to use the swap, then there might be an issue. Utilizing the memory does not mean there is a memory leak Regards, ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rajagopal, Sridhar (Sridhar) Sent: 13 July 2009 08:07 To: 'freeswitch-users at lists.freeswitch.org' Subject: [Freeswitch-users] Help Regarding memory leak with freeswitch Hi all, I am running freeswitch on powerpc processor. I see memory being allocated for each subsequent REGISTER requests coming to freeswitch. But not all the memory allocated is not freed. If I run the code for two days the system is running out of memory (RAM available to me is very less). The same memory issue is happening even for calls. Please let me know if any body has seen this issue. Please let me know how I can go ahead and debug this issue. Thanks in advance for the help. Regards, Sridhar -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090713/c870589a/attachment-0002.html From jingwei.yang at gmail.com Mon Jul 13 02:27:21 2009 From: jingwei.yang at gmail.com (Jingwei Yang) Date: Mon, 13 Jul 2009 17:27:21 +0800 Subject: [Freeswitch-users] mod_dingaling no audio In-Reply-To: <507898380906290346w20cfea6bq82114efc63439f52@mail.gmail.com> References: <13529f9d0906240240x55f0c19ao7246de1d34b1f661@mail.gmail.com> <13529f9d0906242042j598a2a21v3f32c2d99a28b58@mail.gmail.com> <507898380906242053p155e16edve8bc797c22f11c12@mail.gmail.com> <13529f9d0906242231s7384908ayb30340f5442b2ab@mail.gmail.com> <507898380906250602r4c666b06xfdeea5943ad0cbff@mail.gmail.com> <13529f9d0906251915m440a1c46s534a5c7c735b8ce8@mail.gmail.com> <507898380906251930v6f82d786l9d1bf0da61212b05@mail.gmail.com> <13529f9d0906252034o5cb0b7b7y8a9ffe750edf2b20@mail.gmail.com> <13529f9d0906290225v4abd1552l4a3b72e0ff1afc66@mail.gmail.com> <507898380906290346w20cfea6bq82114efc63439f52@mail.gmail.com> Message-ID: <13529f9d0907130227u377b7459l8143a37b9ccfc761@mail.gmail.com> Hi Chris, sorry for the late reply. Have been quite busy last few days. I had shifted 888 from default.xml to public.xml and the dialplan is simply having an echo action now. I've turned on dl_debug but unfortunately didn't find anything useful. Logs are attached for your reference. I don't think there's something wrong with the dialplan as two external parties can talk to each other perfectly (with ext-rtp-ip uncommented, at this time my ip was interpreted to be an external one). With ext-rtp-ip commented, I can hear the echo and I saw my ip was translated into an internal one (at this time, external party's audio failed). I tried the method on this wiki page as well: http://wiki.freeswitch.org/wiki/NAT_Traversal (the last FreeSwitch behind NAT portion) but still no luck. Please kindly let me know what other configs I should change. Thanks, -Jingwei On Mon, Jun 29, 2009 at 6:46 PM, Chris Chen wrote: > Jingwei, I don't know if you have the 888 defined in default.xml? also you > have to define $${domain}. > please do " dl_debug on" from fs_cli, and watch the console logs and see > what's going on when you try calling from external. Most likely your > dialplan is not correctly defined. > > Chris > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090713/d5061ea6/attachment-0002.html -------------- next part -------------- freeswitch at localhost.localdomain> originate dingaling/gmail.com/xxxxxx at gmail.com &echo 2009-07-13 15:19:34.950696 [NOTICE] libdingaling.c:1309 SecSEND: ------------------------------------------------------------------------------- Call Me! 2009-07-13 15:19:34.950696 [NOTICE] libdingaling.c:1309 SecSEND: ------------------------------------------------------------------------------- 2009-07-13 15:19:35.643651 [INFO] libdingaling.c:1307 SecRECV: ------------------------------------------------------------------------------- 24 2009-07-13 15:19:35.647665 [NOTICE] switch_channel.c:602 New Channel dingaling/gmail.com/xxxxxx at gmail.com [dde6f921-ed98-46c0-9628-8f2c0d1b2835] 2009-07-13 15:19:35.648943 [NOTICE] mod_dingaling.c:1084 Ring-Ready dingaling/gmail.com/xxxxxx at gmail.com! 2009-07-13 15:19:35.651651 [NOTICE] libdingaling.c:1309 SecSEND: ------------------------------------------------------------------------------- Incoming Call From FreeSWITCH 0000000000 2009-07-13 15:19:35.651651 [NOTICE] libdingaling.c:1309 SecSEND: ------------------------------------------------------------------------------- 2009-07-13 15:19:35.662651 [INFO] libdingaling.c:1307 SecRECV: ------------------------------------------------------------------------------- 24 2009-07-13 15:19:36.458618 [INFO] libdingaling.c:1307 SecRECV: ------------------------------------------------------------------------------- 2009-07-13 15:19:36.631611 [INFO] libdingaling.c:1307 SecRECV: ------------------------------------------------------------------------------- 2009-07-13 15:19:36.657730 [NOTICE] libdingaling.c:1309 SecSEND: ------------------------------------------------------------------------------- 2009-07-13 15:19:36.872615 [INFO] libdingaling.c:1307 SecRECV: ------------------------------------------------------------------------------- 2009-07-13 15:19:36.959598 [NOTICE] libdingaling.c:1309 SecSEND: ------------------------------------------------------------------------------- 2009-07-13 15:19:37.14595 [INFO] libdingaling.c:1307 SecRECV: ------------------------------------------------------------------------------- 2009-07-13 15:19:37.60593 [NOTICE] libdingaling.c:1309 SecSEND: ------------------------------------------------------------------------------- 2009-07-13 15:19:40.791443 [INFO] libdingaling.c:1307 SecRECV: ------------------------------------------------------------------------------- 2009-07-13 15:19:41.680379 [INFO] mod_dingaling.c:974 Stun Success 59.189.194.106:26746 2009-07-13 15:19:41.691378 [NOTICE] mod_dingaling.c:1142 Channel [dingaling/gmail.com/xxxxxx at gmail.com] has been answered API CALL [originate(dingaling/gmail.com/xxxxxx at gmail.com &echo)] output: +OK dde6f921-ed98-46c0-9628-8f2c0d1b2835 freeswitch at localhost.localdomain> 2009-07-13 15:19:41.762376 [NOTICE] libdingaling.c:1309 SecSEND: ------------------------------------------------------------------------------- 2009-07-13 15:19:41.762376 [NOTICE] libdingaling.c:1309 SecSEND: ------------------------------------------------------------------------------- 2009-07-13 15:19:42.502589 [INFO] libdingaling.c:1307 SecRECV: ------------------------------------------------------------------------------- 2009-07-13 15:19:43.672335 [INFO] libdingaling.c:1307 SecRECV: ------------------------------------------------------------------------------- 2009-07-13 15:19:43.762360 [NOTICE] libdingaling.c:1309 SecSEND: ------------------------------------------------------------------------------- 2009-07-13 15:19:44.499288 [INFO] libdingaling.c:1307 SecRECV: ------------------------------------------------------------------------------- 2009-07-13 15:19:44.562353 [NOTICE] libdingaling.c:1309 SecSEND: ------------------------------------------------------------------------------- 2009-07-13 15:19:44.691280 [INFO] libdingaling.c:1307 SecRECV: ------------------------------------------------------------------------------- 2009-07-13 15:19:44.762321 [NOTICE] libdingaling.c:1309 SecSEND: ------------------------------------------------------------------------------- 2009-07-13 15:19:45.498291 [INFO] libdingaling.c:1307 SecRECV: ------------------------------------------------------------------------------- 2009-07-13 15:19:45.562530 [NOTICE] libdingaling.c:1309 SecSEND: ------------------------------------------------------------------------------- 2009-07-13 15:19:52.922946 [INFO] libdingaling.c:1307 SecRECV: ------------------------------------------------------------------------------- 2009-07-13 15:19:52.922946 [NOTICE] mod_dingaling.c:718 Hangup dingaling/gmail.com/xxxxxx at gmail.com [CS_EXECUTE] [NORMAL_CLEARING] 2009-07-13 15:19:52.936942 [NOTICE] switch_core_session.c:1085 Session 2 (dingaling/gmail.com/xxxxxx at gmail.com) Ended 2009-07-13 15:19:52.936942 [NOTICE] switch_core_session.c:1087 Close Channel dingaling/gmail.com/xxxxxx at gmail.com [CS_DESTROY] 2009-07-13 15:19:52.961986 [NOTICE] libdingaling.c:1309 SecSEND: ------------------------------------------------------------------------------- 2009-07-13 15:19:52.961986 [NOTICE] libdingaling.c:1309 SecSEND: ------------------------------------------------------------------------------- 2009-07-13 15:19:53.702945 [INFO] libdingaling.c:1307 SecRECV: ------------------------------------------------------------------------------- unknown session From Prometheus001 at gmx.net Mon Jul 13 02:30:14 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Mon, 13 Jul 2009 11:30:14 +0200 Subject: [Freeswitch-users] Error in default Sofia profile checking In-Reply-To: <1452e2980907122238h1392b903o6b1179e423fb7893@mail.gmail.com> References: <1452e2980907122238h1392b903o6b1179e423fb7893@mail.gmail.com> Message-ID: <4A5AFEA6.9000806@gmx.net> Which twinkle version are you using? I use Twinkle 1.0.1 and my SIP register looks as follows. As you can see, the contact header is there. U 127.0.0.1:5062 -> 127.0.0.1:5060 REGISTER sip:127.0.0.1 SIP/2.0. Via: SIP/2.0/UDP 192.168.178.146:5062;rport;branch=z9hG4bKtvnvzdwy. Max-Forwards: 70. To: "8353310" . From: "8353310" ;tag=avpju. Call-ID: ibubkykiithqlne at 192.168.178.146. CSeq: 5792 REGISTER. Contact: ;expires=60. Authorization: Digest username="8353310",realm="127.0.0.1",nonce="4bcfe1b0-6f8f-11de-bc32-2dff86a04420",uri="sip:127.0.0.1",response="922690317852a402052da6f74f7196df",algorithm=MD5,cnonce="k9662kmk64",qop=auth,nc=00000001. Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO. User-Agent: Twinkle/1.0.1. Content-Length: 0. . # U 127.0.0.1:5060 -> 127.0.0.1:5062 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.178.146:5062;rport=5062;branch=z9hG4bKtvnvzdwy;received=127.0.0.1. From: "8353310" ;tag=avpju. To: "8353310" ;tag=4p5K211F33N2c. Call-ID: ibubkykiithqlne at 192.168.178.146. CSeq: 5792 REGISTER. Contact: ;expires=60. Date: Mon, 13 Jul 2009 09:26:51 GMT. User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-12955M. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. Supported: timer, precondition, path, replaces. Content-Length: 0. Can you ngrep your traffic and port your register request? ngrep -d any port 5060 -W byline Best regards Peter velusamy velu schrieb: > > Dear Peter, > I have followed your steps, For me my FS and Twinkle running > in separate machine. But, I am still receiving the same error > "[ERR] sofia_reg.c:1135 > sofia_reg_handle_sip_i_register() NO CONTACT!" > > Please give any suggestions to rectify this error.. > > Thanks in Advance, > > Regards, > K.Velusamy. > > > > On Jul 11, 2009, at 8:46 AM, Peter P GMX wrote: > > > I have several Twinkles running against freeswitch on a locally > > installed machine (FS acts as a SIP/TLS proxy). > > So in general Twinkle works (on various Ubuntu machines from 7 > upto 9 > > with various Twinkle versions). It must be some kind of setting in > > Twinkle. E.g. > > > > * set the local Twinkle SIP UDP port to 5062 in general settings > > * Set the right network interface (e.g. eth0) > > * In the profile do not set the realm > > * Allow missing contact header on 200 OK > > > > Best regards > > Peter > > > > > > > > Mathieu Rene schrieb: > >> Chances are the registering UA didnt provide a Contact header > >> (required by rfc3261) > >> > >> Mathieu Rene > >> Avant-Garde Solutions Inc > >> Office: + 1 (514) 664-1044 x100 > >> Cell: +1 (514) 664-1044 x200 > >> mrene at avgs.ca > >> > >> > >> > >> > >> On 11-Jul-09, at 1:23 AM, velusamy velu wrote: > >> > >> > >>> Dear Friends, > >>> When I register my Softphone(Twinkle) with predefined > >>> sofia registration("1000" with password "1234"). I have got the > >>> following error in FreeSWITCH console. > >>> > >>> "2009-07-11 09:37:16 [ERR] sofia_reg.c:1135 > >>> sofia_reg_handle_sip_i_ > >>> register() NO CONTACT!" > >>> > >>> Please help me to solve this problem... > >>> > >>> Regards, > >>> K.Velusamy. > >>> _______________________________________________ > >>> Freeswitch-users mailing list > >>> Freeswitch-users at lists.freeswitch.org > > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> http://www.freeswitch.org > >>> > >> > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > >> > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From chris.chen2004 at gmail.com Mon Jul 13 04:29:23 2009 From: chris.chen2004 at gmail.com (Chris Chen) Date: Mon, 13 Jul 2009 07:29:23 -0400 Subject: [Freeswitch-users] mod_dingaling no audio In-Reply-To: <13529f9d0907130227u377b7459l8143a37b9ccfc761@mail.gmail.com> References: <13529f9d0906240240x55f0c19ao7246de1d34b1f661@mail.gmail.com> <507898380906242053p155e16edve8bc797c22f11c12@mail.gmail.com> <13529f9d0906242231s7384908ayb30340f5442b2ab@mail.gmail.com> <507898380906250602r4c666b06xfdeea5943ad0cbff@mail.gmail.com> <13529f9d0906251915m440a1c46s534a5c7c735b8ce8@mail.gmail.com> <507898380906251930v6f82d786l9d1bf0da61212b05@mail.gmail.com> <13529f9d0906252034o5cb0b7b7y8a9ffe750edf2b20@mail.gmail.com> <13529f9d0906290225v4abd1552l4a3b72e0ff1afc66@mail.gmail.com> <507898380906290346w20cfea6bq82114efc63439f52@mail.gmail.com> <13529f9d0907130227u377b7459l8143a37b9ccfc761@mail.gmail.com> Message-ID: <507898380907130429haeb7ef9haeaf31b7e409da5c@mail.gmail.com> Jingwei, can you show your console log when somebody is calling you from gtalk client? Will it really hit 888 in your dialplan? Thanks, Chris On Mon, Jul 13, 2009 at 5:27 AM, Jingwei Yang wrote: > Hi Chris, sorry for the late reply. Have been quite busy last few days. > > I had shifted 888 from default.xml to public.xml and the dialplan is simply > having an echo action now. I've turned on dl_debug but unfortunately didn't > find anything useful. Logs are attached for your reference. > > I don't think there's something wrong with the dialplan as two external > parties can talk to each other perfectly (with ext-rtp-ip uncommented, at > this time my ip was interpreted to be an external one). With ext-rtp-ip > commented, I can hear the echo and I saw my ip was translated into an > internal one (at this time, external party's audio failed). > > I tried the method on this wiki page as well: > http://wiki.freeswitch.org/wiki/NAT_Traversal (the last FreeSwitch behind > NAT portion) but still no luck. Please kindly let me know what other configs > I should change. > > Thanks, > -Jingwei > > On Mon, Jun 29, 2009 at 6:46 PM, Chris Chen wrote: > >> Jingwei, I don't know if you have the 888 defined in default.xml? also >> you have to define $${domain}. >> please do " dl_debug on" from fs_cli, and watch the console logs and see >> what's going on when you try calling from external. Most likely your >> dialplan is not correctly defined. >> >> Chris >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090713/69780b63/attachment-0002.html From Prometheus001 at gmx.net Mon Jul 13 05:55:00 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Mon, 13 Jul 2009 14:55:00 +0200 Subject: [Freeswitch-users] pocketsphinx In-Reply-To: <4A5AE80F.1000904@ewetel.de> References: <4A5739DE.1080800@ewetel.de> <4A574F0D.7040604@gmx.net> <4A5AE80F.1000904@ewetel.de> Message-ID: <4A5B2EA4.8050608@gmx.net> Hello Helmut, the 3 mentioned words are already part of the englisch standard dictionary, so maybe this causes the problem? You may test with words which are outside of the standard grammar files or delete the original ones? So far I have no other documentation available. This part of PocketSphinx is rather poorly documented. And for the FS part I've only got some information from this mailing list. Best regards Peter Helmut Kuper schrieb: > Hi Peter, > > hmmm well, I had the same idea and I tested it! Buuuuut ... you have to > make sure that the english grammar/acousticModel is able to cover all > german noises. E.g. I tried to detect "Burke", "Jan" and "Gerd". I was > able to map Burke successfully in default.dic. Otherwise I had to say > "B?hrki" ... I did the same with "Jan" - but when I tried to detect Jan > I always got Gerd (with and witout mapping ind default.dic) ... Quite > strange and not really usable for (german) customers. > > But some typical software magic happened on my way: > During my tests I had somehow a configuration using the voxforge files > which was working within FS. But I can't reproduce it. I configured > serveral files at the same time and used for reloading "reloadxml" and > "reload mod_pocketsphinx" instead of rebooting FS. When it worked, FS > was able to detect "Burke", "Jan" and "Gerd" correctly without modifying > the dictionary... > > > Is there any manual about pocketsphinx and its config files, which can > explain how PS is working in more detail? Currently I walk with a > flashlight in the dark ... > > regards > Helmut > > > On 10.07.2009 16:24, Peter P GMX wrote: > > Hello Helmut, > > > I looked at these dic files. Their content (look at all the qq's) is > > quite different from the dic files supplied with freeswitch > pocketsphinx. > > As I remember the CMU dict file format has changed in April 2008. > > Maybe there is a converter somewhere? > > > I was thinking of just enhancing the current dict file for some german > > words I need, but did not test it so far. This should be possible > > without modifying the underlying grammar. > > http://en.wikipedia.org/wiki/CMU_Pronouncing_Dictionary > > I would love to hear when you have had any progress on this. > > > Best regards > > Peter _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From andy at fabulous4.co.uk Mon Jul 13 05:59:55 2009 From: andy at fabulous4.co.uk (Andy) Date: Mon, 13 Jul 2009 13:59:55 +0100 Subject: [Freeswitch-users] Problems with Ping and re-registering broken gateways Message-ID: <0324DD608A074940AAC173B85A3978F2@D810> Hi, I'm fairly sure my problem lies with my voip provider VoipTalk but wonder if you could help me understand a couple of things. My config is very simple, I'm using freeswitch to accept incoming calls via a voip gateway and record messages. Here's the problem: - When freeswitch starts the gateways are all created and register correctly - I have the ping parameter set to make sure the gateway stays alive. - The first time freeswitch pings the gateway it fails even though the registration appears intact as calls are still coming through to freeswitch - Freeswitch then tries to re-register the gateway but this fails. The SIP trace shows an Unauthorized message and the actual log entry is 'Registration Failed with status Operation has no matching challenge [904]' - eventually the registration times out with my provider and all is lost. - if I call 'sofia profile external restart' or restart the software this fixes the problem My questions are: 1) Why would the ping fail when the registration appears to be intact? 2) Whay would the auto re-register not work but a restart would? This ones driving me nuts so any help greatly appreciated. regards Andy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090713/f4e8db59/attachment-0002.html From brian at freeswitch.org Mon Jul 13 06:27:25 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 13 Jul 2009 08:27:25 -0500 Subject: [Freeswitch-users] Error in default Sofia profile checking In-Reply-To: <4A5AFEA6.9000806@gmx.net> References: <1452e2980907122238h1392b903o6b1179e423fb7893@mail.gmail.com> <4A5AFEA6.9000806@gmx.net> Message-ID: Its not a bug... its just something we do not support in FreeSWITCH yet... Register with no contact is a fetch operation. /b On Jul 13, 2009, at 4:30 AM, Peter P GMX wrote: > Which twinkle version are you using? I use Twinkle 1.0.1 and my SIP > register looks as follows. As you can see, the contact header is > there. > > U 127.0.0.1:5062 -> 127.0.0.1:5060 > REGISTER sip:127.0.0.1 SIP/2.0. > Via: SIP/2.0/UDP 192.168.178.146:5062;rport;branch=z9hG4bKtvnvzdwy. > Max-Forwards: 70. > To: "8353310" . > From: "8353310" ;tag=avpju. > Call-ID: ibubkykiithqlne at 192.168.178.146. > CSeq: 5792 REGISTER. > Contact: ;expires=60. > Authorization: Digest > username="8353310",realm="127.0.0.1",nonce="4bcfe1b0-6f8f-11de- > bc32-2dff86a04420",uri="sip: > 127.0.0.1 > ",response > = > "922690317852a402052da6f74f7196df > ",algorithm=MD5,cnonce="k9662kmk64",qop=auth,nc=00000001. > Allow: > INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO. > User-Agent: Twinkle/1.0.1. > Content-Length: 0. > . > > # > U 127.0.0.1:5060 -> 127.0.0.1:5062 > SIP/2.0 200 OK. > Via: SIP/2.0/UDP > 192.168.178.146 > :5062;rport=5062;branch=z9hG4bKtvnvzdwy;received=127.0.0.1. > From: "8353310" ;tag=avpju. > To: "8353310" ;tag=4p5K211F33N2c. > Call-ID: ibubkykiithqlne at 192.168.178.146. > CSeq: 5792 REGISTER. > Contact: ;expires=60. > Date: Mon, 13 Jul 2009 09:26:51 GMT. > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-12955M. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. > Supported: timer, precondition, path, replaces. > Content-Length: 0. > > Can you ngrep your traffic and port your register request? > ngrep -d any port 5060 -W byline > > > Best regards > Peter > > velusamy velu schrieb: >> >> Dear Peter, >> I have followed your steps, For me my FS and Twinkle running >> in separate machine. But, I am still receiving the same error >> "[ERR] sofia_reg.c:1135 >> sofia_reg_handle_sip_i_register() NO CONTACT!" >> >> Please give any suggestions to rectify this error.. >> >> Thanks in Advance, >> >> Regards, >> K.Velusamy. >> >> >> >> On Jul 11, 2009, at 8:46 AM, Peter P GMX wrote: >> >>> I have several Twinkles running against freeswitch on a locally >>> installed machine (FS acts as a SIP/TLS proxy). >>> So in general Twinkle works (on various Ubuntu machines from 7 >> upto 9 >>> with various Twinkle versions). It must be some kind of setting in >>> Twinkle. E.g. >>> >>> * set the local Twinkle SIP UDP port to 5062 in general settings >>> * Set the right network interface (e.g. eth0) >>> * In the profile do not set the realm >>> * Allow missing contact header on 200 OK >>> >>> Best regards >>> Peter >>> >>> >>> >>> Mathieu Rene schrieb: >>>> Chances are the registering UA didnt provide a Contact header >>>> (required by rfc3261) >>>> >>>> Mathieu Rene >>>> Avant-Garde Solutions Inc >>>> Office: + 1 (514) 664-1044 x100 >>>> Cell: +1 (514) 664-1044 x200 >>>> mrene at avgs.ca >>>> >>>> >>>> >>>> >>>> On 11-Jul-09, at 1:23 AM, velusamy velu wrote: >>>> >>>> >>>>> Dear Friends, >>>>> When I register my Softphone(Twinkle) with predefined >>>>> sofia registration("1000" with password "1234"). I have got the >>>>> following error in FreeSWITCH console. >>>>> >>>>> "2009-07-11 09:37:16 [ERR] sofia_reg.c:1135 >>>>> sofia_reg_handle_sip_i_ >>>>> register() NO CONTACT!" >>>>> >>>>> Please help me to solve this problem... >>>>> >>>>> Regards, >>>>> K.Velusamy. >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.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 Jul 13 08:08:26 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 13 Jul 2009 10:08:26 -0500 Subject: [Freeswitch-users] Dialogic cards In-Reply-To: <855e4dcf0907122107q392a9ca1h851dfbd57e2518ca@mail.gmail.com> References: <855e4dcf0907122107q392a9ca1h851dfbd57e2518ca@mail.gmail.com> Message-ID: <191c3a030907130808s37e52ddg3ba567d6a5a1034b@mail.gmail.com> Dialogic is coming to ClueCon this year (this aug 4th) and they are sponsoring the conference. I can discuss the possibility of supporting their cards at that time. On Sun, Jul 12, 2009 at 11:07 PM, Tim Uckun wrote: > We have some older dialogic cards (D300 series E1 cards) and I am > wondering if freeswitch can support these cards. > > 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/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference 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/20090713/25e82735/attachment-0002.html From anthony.minessale at gmail.com Mon Jul 13 08:20:53 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 13 Jul 2009 10:20:53 -0500 Subject: [Freeswitch-users] Help Regarding memory leak with freeswitch In-Reply-To: <9389DD3DDD6B9144B147CE564C6599B9082FC07020@INBANSXCHMBSA3.in.alcatel-lucent.com> References: <9389DD3DDD6B9144B147CE564C6599B9082FC07020@INBANSXCHMBSA3.in.alcatel-lucent.com> Message-ID: <191c3a030907130820y46985521y1a01270a98cc2fcf@mail.gmail.com> Which revision of FreeSWITCH are you using? Several memory leaks have been fixed since the last formal release. One specifically in REGISTER. You should probably try SVN trunk or the latest pre-release of 1.0.4 On Mon, Jul 13, 2009 at 2:07 AM, Rajagopal, Sridhar (Sridhar) < sridhart at alcatel-lucent.com> wrote: > Hi all, > > I am running freeswitch on powerpc processor. I see memory being allocated > for each subsequent REGISTER requests coming to freeswitch. But not all the > memory allocated is not freed. If I run the code for two days the system is > running out of memory (RAM available to me is very less). > The same memory issue is happening even for calls. > > Please let me know if any body has seen this issue. Please let me know how > I can go ahead and debug this issue. > > Thanks in advance for the help. > > Regards, > Sridhar > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference 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/20090713/b97f4578/attachment-0002.html From anthony.minessale at gmail.com Mon Jul 13 08:25:10 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 13 Jul 2009 10:25:10 -0500 Subject: [Freeswitch-users] Setting channel variables using event socket In-Reply-To: References: <7525E12C-1213-4502-8993-F01300D5D644@freeswitch.org> <191c3a030907111451u7ddb17c3r66cc4371c58cc63d@mail.gmail.com> Message-ID: <191c3a030907130825q178fdfa9q1f3098258288ed3b@mail.gmail.com> and if you go the uuid_setvar route you do this: api uuid_setvar joe=out_to_lunch On Sun, Jul 12, 2009 at 6:00 PM, Brian West wrote: > call-command: execute > execute-app-name: set > execute-app-arg: fred=out_to_lunch > > > On Jul 12, 2009, at 5:33 PM, Nik Middleton wrote: > > As in > > call-command: set joe=out_to_lunch ? > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference 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/20090713/113023df/attachment-0002.html From steveu at coppice.org Mon Jul 13 08:39:39 2009 From: steveu at coppice.org (Steve Underwood) Date: Mon, 13 Jul 2009 23:39:39 +0800 Subject: [Freeswitch-users] Dialogic cards In-Reply-To: <855e4dcf0907122107q392a9ca1h851dfbd57e2518ca@mail.gmail.com> References: <855e4dcf0907122107q392a9ca1h851dfbd57e2518ca@mail.gmail.com> Message-ID: <4A5B553B.9020000@coppice.org> Tim Uckun wrote: > We have some older dialogic cards (D300 series E1 cards) and I am > wondering if freeswitch can support these cards. > Oh, I like the easy questions. No. It lacks the hardware features to do anything useful with Freeswitch, or Asterisk, or Callweaver, or Yate, or anything else that expects to do two way telephony through the host CPU. Steve From steveu at coppice.org Mon Jul 13 08:39:47 2009 From: steveu at coppice.org (Steve Underwood) Date: Mon, 13 Jul 2009 23:39:47 +0800 Subject: [Freeswitch-users] Dialogic cards In-Reply-To: <191c3a030907130808s37e52ddg3ba567d6a5a1034b@mail.gmail.com> References: <855e4dcf0907122107q392a9ca1h851dfbd57e2518ca@mail.gmail.com> <191c3a030907130808s37e52ddg3ba567d6a5a1034b@mail.gmail.com> Message-ID: <4A5B5543.6080405@coppice.org> Being now a mashup of several CTI companies, there are now a number of disparate things called Dialogic cards. Some, like the cards previously known as Prince..... er, Eicon are perfectly supportable. The old Dialogic cards, like the D300 series, are not duplex to and from the host. They are only duplex across the mezzanine bus. There's nothing you can do with them. Steve Anthony Minessale wrote: > Dialogic is coming to ClueCon this year (this aug 4th) and they are > sponsoring the conference. > I can discuss the possibility of supporting their cards at that time. > > > On Sun, Jul 12, 2009 at 11:07 PM, Tim Uckun > wrote: > > We have some older dialogic cards (D300 series E1 cards) and I am > wondering if freeswitch can support these cards. > > Thanks. > From anthony.minessale at gmail.com Mon Jul 13 08:46:58 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 13 Jul 2009 10:46:58 -0500 Subject: [Freeswitch-users] Dialogic cards In-Reply-To: <4A5B553B.9020000@coppice.org> References: <855e4dcf0907122107q392a9ca1h851dfbd57e2518ca@mail.gmail.com> <4A5B553B.9020000@coppice.org> Message-ID: <191c3a030907130846u4c134fc4n578589c196617557@mail.gmail.com> ok, or we could ask Steve I guess. =D On Mon, Jul 13, 2009 at 10:39 AM, Steve Underwood wrote: > Tim Uckun wrote: > > We have some older dialogic cards (D300 series E1 cards) and I am > > wondering if freeswitch can support these cards. > > > Oh, I like the easy questions. No. It lacks the hardware features to do > anything useful with Freeswitch, or Asterisk, or Callweaver, or Yate, or > anything else that expects to do two way telephony through the host CPU. > > 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/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference 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/20090713/2912d77b/attachment-0002.html From valentin.doroga at pronexus.com Mon Jul 13 08:57:07 2009 From: valentin.doroga at pronexus.com (Valentin Doroga) Date: Mon, 13 Jul 2009 11:57:07 -0400 Subject: [Freeswitch-users] Dialogic cards In-Reply-To: <4A5B5543.6080405@coppice.org> Message-ID: <20090713155712.OYAB273.tomts27-srv.bellnexxia.net@toip37-bus.srvr.bell.ca> Maybe Dialogic would add support for "thin blades", currently used for HMP (DNI series). Val. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steve Underwood Sent: Monday, July 13, 2009 11:40 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Dialogic cards Being now a mashup of several CTI companies, there are now a number of disparate things called Dialogic cards. Some, like the cards previously known as Prince..... er, Eicon are perfectly supportable. The old Dialogic cards, like the D300 series, are not duplex to and from the host. They are only duplex across the mezzanine bus. There's nothing you can do with them. Steve Anthony Minessale wrote: > Dialogic is coming to ClueCon this year (this aug 4th) and they are > sponsoring the conference. > I can discuss the possibility of supporting their cards at that time. > > > On Sun, Jul 12, 2009 at 11:07 PM, Tim Uckun > wrote: > > We have some older dialogic cards (D300 series E1 cards) and I am > wondering if freeswitch can support these cards. > > 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 From anthony.minessale at gmail.com Mon Jul 13 08:59:28 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 13 Jul 2009 10:59:28 -0500 Subject: [Freeswitch-users] Problems with Ping and re-registering broken gateways In-Reply-To: <0324DD608A074940AAC173B85A3978F2@D810> References: <0324DD608A074940AAC173B85A3978F2@D810> Message-ID: <191c3a030907130859y2665da03xf704f9db6070fb3c@mail.gmail.com> Are they ignoring the options packet we send them or are they maybe getting lost behind NAT? we send an OPTIONS and even if we get a error back we consider that a successful reply. We did have a patch into SVN very recently to correct a problem with OPTIONS ping in a NAT situation. Maybe try latest trunk first then capture the console log with sip traffic in place if it still does not work so we can have a look. to capture the log use these 2 commands from the cli. sofia profile internal siptrace on console loglevel debug On Mon, Jul 13, 2009 at 7:59 AM, Andy wrote: > Hi, > > I'm fairly sure my problem lies with my voip provider VoipTalk but wonder > if you could help me understand a couple of things. My config is very > simple, I'm using freeswitch to accept incoming calls via a voip gateway and > record messages. Here's the problem: > > - When freeswitch starts the gateways are all created and register > correctly > > - I have the ping parameter set to make sure the gateway stays alive. > > - The first time freeswitch pings the gateway it fails even though the > registration appears intact as calls are still coming through to freeswitch > > - Freeswitch then tries to re-register the gateway but this fails. The SIP > trace shows an Unauthorized message and the actual log entry is > 'Registration Failed with status Operation has no matching challenge [904]' > > - eventually the registration times out with my provider and all is lost. > > - if I call 'sofia profile external restart' or restart the software this > fixes the problem > > My questions are: > > 1) Why would the ping fail when the registration appears to be intact? > 2) Whay would the auto re-register not work but a restart would? > > This ones driving me nuts so any help greatly appreciated. > > regards > Andy > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference 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/20090713/d54f755e/attachment-0002.html From pjintheusa at gmail.com Mon Jul 13 09:03:17 2009 From: pjintheusa at gmail.com (Phillip Jones) Date: Mon, 13 Jul 2009 12:03:17 -0400 Subject: [Freeswitch-users] managed_mod directories In-Reply-To: <1247334223093-3243183.post@n2.nabble.com> References: <367751820907101046l7b41a2fdl7cac3d05b37a80f7@mail.gmail.com> <509787.62310.qm@web33504.mail.mud.yahoo.com> <367751820907110925o61168b64t1db5b6944e0dac7e@mail.gmail.com> <1247334223093-3243183.post@n2.nabble.com> Message-ID: <367751820907130903p36734e0eyb010da5409dfb765@mail.gmail.com> Got it! Thanks very much for that clarification. Phil On Sat, Jul 11, 2009 at 1:43 PM, Jeff Lenk wrote: > > Hi, > > The base dll (FreeSWITCH.Managed.dll) is loaded from /mod - additional > managed dlls are loaded from /mod/managed. This is designed to allow your > dll's to be built and maintained independant of the FS build files. You can > simply just drop your dlls into mod/managed and they will be loaded and > available for use(this happens at FS startup). > > The base managed dll (FreeSWITCH.Managed.dll) is only really supposed to be > used for loader support and the demo classes - you should place your code > in > your own dll. > > - Jeff > > > Phillip Jones-2 wrote: > > > > Hi, > > > > If I place the DLL in mod\managed I get the following error: > > > > [err] mod_managed.cpp:287 Assembly::LoadFrom failed: > > system.IO.FileNotFoundException: Could not load file or assembly > > 'file:///c:program files (x86)\Freeswitch\mod\freeSWITCH.Managed.dll' or > > one > > of its dependencies. The system could not find the file specified. > > > > As I said. When I place freeSWITCH.Managed.dll straight into \mod then > > everything works fine. > > > > Thanks > > > > > > Phil > > > > > > On Sat, Jul 11, 2009 at 9:15 AM, Diego Toro wrote: > > > >> Hello, > >> > >> What error do you get when dll is put on mod/managed ?, I work with > dll's > >> on mod/managed although I changed loadfile by loadfrom on loader.cs. > >> > >> Diego > >> > >> > >> --- On *Fri, 7/10/09, Phillip Jones * wrote: > >> > >> > >> From: Phillip Jones > >> Subject: Re: [Freeswitch-users] managed_mod directories > >> To: freeswitch-users at lists.freeswitch.org > >> Date: Friday, July 10, 2009, 6:22 PM > >> > >> > >> It is looking in mod. > >> > >> It required the mod\managed directory, but if I place my dll in > >> mod\managed > >> it fails. DLL must be in mod - mod\managed is empty. > >> > >> My app works fine though > >> > >> > >> Phil > >> > >> > >> On Fri, Jul 10, 2009 at 4:48 PM, Michael Giagnocavo > >> http://us.mc335.mail.yahoo.com/mc/compose?to=mgg at giagnocavo.net> > >> > wrote: > >> > >>> You?re saying that it requires the managed DLL to be in both the mod > >>> and > >>> mod\managed directory? What error do you get if it?s only in mod? It?s > >>> been > >>> months, but I just checked loader.cs and it looks explicitly in the > >>> managed > >>> directory to resolve assemblies as well as to scan to load them. > >>> > >>> -Michael > >>> > >>> *From:* > >>> freeswitch-users-bounces at lists.freeswitch.org< > http://us.mc335.mail.yahoo.com/mc/compose?to=freeswitch-users-bounces at lists.freeswitch.org > >[mailto: > >>> freeswitch-users-bounces at lists.freeswitch.org< > http://us.mc335.mail.yahoo.com/mc/compose?to=freeswitch-users-bounces at lists.freeswitch.org > >] > >>> *On Behalf Of *Phillip Jones > >>> *Sent:* Friday, July 10, 2009 11:46 AM > >>> *To:* > >>> freeswitch-users at lists.freeswitch.org< > http://us.mc335.mail.yahoo.com/mc/compose?to=freeswitch-users at lists.freeswitch.org > > > >>> *Subject:* [Freeswitch-users] managed_mod directories > >>> > >>> Hi there, > >>> > >>> > >>> Using windows with the pre-compiled binary / msi found via the WIKI > >>> > >>> Using mod_managed with no problems however: > >>> > >>> mod_managed appears to require I create a directory 'managed' under > >>> C:\Program Files (x86)\FreeSWITCH\mod > >>> > >>> BUT also requires that I place my .dll in C:\Program Files > >>> (x86)\FreeSWITCH\mod and NOT C:\Program Files > >>> (x86)\FreeSWITCH\mod\managed > >>> > >>> thus making C:\Program Files (x86)\FreeSWITCH\mod\managed redundant > >>> > >>> Anyone else seen this behavior? > >>> > >>> Thanks! > >>> > >>> > >>> Phillip Jones > >>> > >>> _______________________________________________ > >>> Freeswitch-users mailing list > >>> Freeswitch-users at lists.freeswitch.org< > http://us.mc335.mail.yahoo.com/mc/compose?to=Freeswitch-users at lists.freeswitch.org > > > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> http://www.freeswitch.org > >>> > >>> > >> > >> -----Inline Attachment Follows----- > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org< > http://us.mc335.mail.yahoo.com/mc/compose?to=Freeswitch-users at lists.freeswitch.org > > > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > >> > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > >> > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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/managed_mod-directories-tp3240303p3243183.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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090713/fbfd9270/attachment-0002.html From excelsio at gmx.net Mon Jul 13 09:26:59 2009 From: excelsio at gmx.net (excelsio at gmx.net) Date: Mon, 13 Jul 2009 18:26:59 +0200 Subject: [Freeswitch-users] DID with 10 numbers Message-ID: <20090713162659.129030@gmx.net> Hi, I purchased a block of 10 did numbers. Base number is 01234/56789. The numbers themselves range from 01234/567890 to 01234/567899 What works? Well, I can dial in to a "hardcoded" 01234/56789 which belongs to user 1000. I can?t dial out. The main problem is, that I do not know how I can assing those numbers to the users. Of course I want to dial out with each user and the corresponding number should be displayed. Also I want to dial in to the appropriate number. I can?t find an example within the wiki, where there a several numbers to be configured. It would be great if you could give me some hints or links. thanks in advance Michael -- Neu: GMX Doppel-FLAT mit Internet-Flatrate + Telefon-Flatrate f?r nur 19,99 Euro/mtl.!* http://portal.gmx.net/de/go/dsl02 From mrene_lists at avgs.ca Mon Jul 13 09:31:26 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Mon, 13 Jul 2009 12:31:26 -0400 Subject: [Freeswitch-users] DID with 10 numbers In-Reply-To: <20090713162659.129030@gmx.net> References: <20090713162659.129030@gmx.net> Message-ID: You need to define variables within the user's entry, you can then re- use those in the dialplan to route the call using the proper line. If you are using a single trunk you can set the effective_caller_id_number to the number you want to call from and it'll set the callerid accordignly. PS: Meld dich bei unserem irc-channel und wir koennen dir einfacher weiter helfen. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 13-Jul-09, at 12:26 PM, excelsio at gmx.net wrote: > Hi, > > I purchased a block of 10 did numbers. Base number is 01234/56789. > The numbers themselves range from 01234/567890 to 01234/567899 > > What works? Well, I can dial in to a "hardcoded" 01234/56789 which > belongs to user 1000. > I can?t dial out. > > The main problem is, that I do not know how I can assing those > numbers to the users. > Of course I want to dial out with each user and the corresponding > number should be displayed. Also I want to dial in to the > appropriate number. > > I can?t find an example within the wiki, where there a several > numbers to be configured. > > It would be great if you could give me some hints or links. > > thanks in advance > > Michael > -- > Neu: GMX Doppel-FLAT mit Internet-Flatrate + Telefon-Flatrate > f?r nur 19,99 Euro/mtl.!* http://portal.gmx.net/de/go/dsl02 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From vkozak at abisoft.spb.ru Mon Jul 13 10:08:20 2009 From: vkozak at abisoft.spb.ru (Kozak Vladimir) Date: Mon, 13 Jul 2009 21:08:20 +0400 Subject: [Freeswitch-users] FS not wait respond from called and send 200Ok References: <5F5ABAC4334545CE8323DACB10805DE1@abisoft.biz> <74C6337E-6F03-4485-B69B-372658E0D79B@jerris.com> Message-ID: Please, look at these logs. Fist invite witchout SDP (from 1007 at uat.agent.starpoundtech.net to vk_1008 at uat.agent.starpoundtech.net) not wait respond from 1008 and send 200 Ok to 1007. Called phone didn't accept call. ip:ports 172.26.200.252:5071 - userAgent (starpound), originator call 172.26.200.252:5080 - FS external profile 172.26.200.252:5090 - FS doubleNat profile 172.26.10.65:38464 - phone1 (1007, registered on FS) 172.26.10.65:17748 - phone2 (1008, registered on FS) ----- Original Message ----- From: Michael Jerris To: freeswitch-users at lists.freeswitch.org Sent: Friday, July 10, 2009 8:12 PM Subject: Re: [Freeswitch-users] FS not wait respond from called and send 200Ok Look closer at the logs, we don't send a 200ok in a bridge until we get one from the b leg. Mike On Jul 10, 2009, at 5:39 AM, Kozak Vladimir wrote: Hello, I have the following problem: I send Invite without SDP to Freeswitch on destination_number "xxx_123" And I want Freeswitch to make "bridge", but it doesn't wait respond from "123" and sends 200 Ok with SDP to me. Does nybody know a clue about this? Best regards vkozak _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ------------------------------------------------------------------------------ _______________________________________________ Freeswitch-users mailing list Freeswitch-users at 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/20090713/abeb13b5/attachment-0002.html -------------- next part -------------- A non-text attachment was scrubbed... Name: fs.log Type: application/octet-stream Size: 56155 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090713/abeb13b5/attachment-0006.obj -------------- next part -------------- A non-text attachment was scrubbed... Name: sip.log Type: application/octet-stream Size: 13359 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090713/abeb13b5/attachment-0007.obj -------------- next part -------------- A non-text attachment was scrubbed... Name: sip4wireshark.log Type: application/octet-stream Size: 14186 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090713/abeb13b5/attachment-0008.obj From brian at freeswitch.org Mon Jul 13 10:19:42 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 13 Jul 2009 12:19:42 -0500 Subject: [Freeswitch-users] FS not wait respond from called and send 200Ok In-Reply-To: References: <5F5ABAC4334545CE8323DACB10805DE1@abisoft.biz> <74C6337E-6F03-4485-B69B-372658E0D79B@jerris.com> Message-ID: <03192D98-85A9-4A7A-8D06-837C6FDB0BBE@freeswitch.org> If you're on SVN trunk you no longer have to use a double nat profile. You can set the local-network-acl and ext-[rtp|sip]-ip settings correctly. /b On Jul 13, 2009, at 12:08 PM, Kozak Vladimir wrote: > Please, look at these logs. > Fist invite witchout SDP (from 1007 at uat.agent.starpoundtech.net to vk_1008 at uat.agent.starpoundtech.net > ) not wait respond from 1008 and send 200 Ok to 1007. Called phone > didn't accept call. > > > > > > > > > ip:ports > 172.26.200.252:5071 - userAgent (starpound), originator call > 172.26.200.252:5080 - FS external profile > 172.26.200.252:5090 - FS doubleNat profile > 172.26.10.65:38464 - phone1 (1007, registered on FS) > 172.26.10.65:17748 - phone2 (1008, registered on FS) > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090713/123fcb00/attachment-0002.html From sridhart at alcatel-lucent.com Mon Jul 13 10:32:12 2009 From: sridhart at alcatel-lucent.com (Rajagopal, Sridhar (Sridhar)) Date: Mon, 13 Jul 2009 23:02:12 +0530 Subject: [Freeswitch-users] Help Regarding memory leak with freeswitch Message-ID: <9389DD3DDD6B9144B147CE564C6599B9082F92A2ED@INBANSXCHMBSA3.in.alcatel-lucent.com> Hi , Thanks very much for the help. Please let me know which module of the code has fix. Do I need to update freeswitch core library or sofia-sip library or is it necessary to update entire freeswitch code. Regards, Sridhar ________________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of freeswitch-users-request at lists.freeswitch.org [freeswitch-users-request at lists.freeswitch.org] Sent: Monday, July 13, 2009 9:17 PM To: freeswitch-users at lists.freeswitch.org Subject: Freeswitch-users Digest, Vol 37, Issue 67 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: Dialogic cards (Anthony Minessale) 2. Re: Help Regarding memory leak with freeswitch (Anthony Minessale) 3. Re: Setting channel variables using event socket (Anthony Minessale) 4. Re: Dialogic cards (Steve Underwood) 5. Re: Dialogic cards (Steve Underwood) 6. Re: Dialogic cards (Anthony Minessale) ---------------------------------------------------------------------- Message: 1 Date: Mon, 13 Jul 2009 10:08:26 -0500 From: Anthony Minessale Subject: Re: [Freeswitch-users] Dialogic cards To: freeswitch-users at lists.freeswitch.org Message-ID: <191c3a030907130808s37e52ddg3ba567d6a5a1034b at mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" Dialogic is coming to ClueCon this year (this aug 4th) and they are sponsoring the conference. I can discuss the possibility of supporting their cards at that time. On Sun, Jul 12, 2009 at 11:07 PM, Tim Uckun wrote: > We have some older dialogic cards (D300 series E1 cards) and I am > wondering if freeswitch can support these cards. > > 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/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference 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/20090713/25e82735/attachment-0001.html ------------------------------ Message: 2 Date: Mon, 13 Jul 2009 10:20:53 -0500 From: Anthony Minessale Subject: Re: [Freeswitch-users] Help Regarding memory leak with freeswitch To: freeswitch-users at lists.freeswitch.org Message-ID: <191c3a030907130820y46985521y1a01270a98cc2fcf at mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" Which revision of FreeSWITCH are you using? Several memory leaks have been fixed since the last formal release. One specifically in REGISTER. You should probably try SVN trunk or the latest pre-release of 1.0.4 On Mon, Jul 13, 2009 at 2:07 AM, Rajagopal, Sridhar (Sridhar) < sridhart at alcatel-lucent.com> wrote: > Hi all, > > I am running freeswitch on powerpc processor. I see memory being allocated > for each subsequent REGISTER requests coming to freeswitch. But not all the > memory allocated is not freed. If I run the code for two days the system is > running out of memory (RAM available to me is very less). > The same memory issue is happening even for calls. > > Please let me know if any body has seen this issue. Please let me know how > I can go ahead and debug this issue. > > Thanks in advance for the help. > > Regards, > Sridhar > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference 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/20090713/b97f4578/attachment-0001.html ------------------------------ Message: 3 Date: Mon, 13 Jul 2009 10:25:10 -0500 From: Anthony Minessale Subject: Re: [Freeswitch-users] Setting channel variables using event socket To: freeswitch-users at lists.freeswitch.org Message-ID: <191c3a030907130825q178fdfa9q1f3098258288ed3b at mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" and if you go the uuid_setvar route you do this: api uuid_setvar joe=out_to_lunch On Sun, Jul 12, 2009 at 6:00 PM, Brian West wrote: > call-command: execute > execute-app-name: set > execute-app-arg: fred=out_to_lunch > > > On Jul 12, 2009, at 5:33 PM, Nik Middleton wrote: > > As in > > call-command: set joe=out_to_lunch ? > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference 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/20090713/113023df/attachment-0001.html ------------------------------ Message: 4 Date: Mon, 13 Jul 2009 23:39:39 +0800 From: Steve Underwood Subject: Re: [Freeswitch-users] Dialogic cards To: freeswitch-users at lists.freeswitch.org Message-ID: <4A5B553B.9020000 at coppice.org> Content-Type: text/plain; charset=ISO-8859-1; format=flowed Tim Uckun wrote: > We have some older dialogic cards (D300 series E1 cards) and I am > wondering if freeswitch can support these cards. > Oh, I like the easy questions. No. It lacks the hardware features to do anything useful with Freeswitch, or Asterisk, or Callweaver, or Yate, or anything else that expects to do two way telephony through the host CPU. Steve ------------------------------ Message: 5 Date: Mon, 13 Jul 2009 23:39:47 +0800 From: Steve Underwood Subject: Re: [Freeswitch-users] Dialogic cards To: freeswitch-users at lists.freeswitch.org Message-ID: <4A5B5543.6080405 at coppice.org> Content-Type: text/plain; charset=ISO-8859-1; format=flowed Being now a mashup of several CTI companies, there are now a number of disparate things called Dialogic cards. Some, like the cards previously known as Prince..... er, Eicon are perfectly supportable. The old Dialogic cards, like the D300 series, are not duplex to and from the host. They are only duplex across the mezzanine bus. There's nothing you can do with them. Steve Anthony Minessale wrote: > Dialogic is coming to ClueCon this year (this aug 4th) and they are > sponsoring the conference. > I can discuss the possibility of supporting their cards at that time. > > > On Sun, Jul 12, 2009 at 11:07 PM, Tim Uckun > wrote: > > We have some older dialogic cards (D300 series E1 cards) and I am > wondering if freeswitch can support these cards. > > Thanks. > ------------------------------ Message: 6 Date: Mon, 13 Jul 2009 10:46:58 -0500 From: Anthony Minessale Subject: Re: [Freeswitch-users] Dialogic cards To: freeswitch-users at lists.freeswitch.org Message-ID: <191c3a030907130846u4c134fc4n578589c196617557 at mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" ok, or we could ask Steve I guess. =D On Mon, Jul 13, 2009 at 10:39 AM, Steve Underwood wrote: > Tim Uckun wrote: > > We have some older dialogic cards (D300 series E1 cards) and I am > > wondering if freeswitch can support these cards. > > > Oh, I like the easy questions. No. It lacks the hardware features to do > anything useful with Freeswitch, or Asterisk, or Callweaver, or Yate, or > anything else that expects to do two way telephony through the host CPU. > > 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/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference 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/20090713/2912d77b/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 37, Issue 67 ************************************************ From anthony.minessale at gmail.com Mon Jul 13 10:50:02 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 13 Jul 2009 12:50:02 -0500 Subject: [Freeswitch-users] Help Regarding memory leak with freeswitch In-Reply-To: <9389DD3DDD6B9144B147CE564C6599B9082F92A2ED@INBANSXCHMBSA3.in.alcatel-lucent.com> References: <9389DD3DDD6B9144B147CE564C6599B9082F92A2ED@INBANSXCHMBSA3.in.alcatel-lucent.com> Message-ID: <191c3a030907131050i7b39ef91s631d3ab9ea189600@mail.gmail.com> I am saying you should update the entire freeswitch code to either latest trunk or at least 1.0.4pre9 On Mon, Jul 13, 2009 at 12:32 PM, Rajagopal, Sridhar (Sridhar) < sridhart at alcatel-lucent.com> wrote: > Hi , > > Thanks very much for the help. > > Please let me know which module of the code has fix. Do I need to update > freeswitch core library or sofia-sip library or is it necessary to update > entire freeswitch code. > > Regards, > Sridhar > > ________________________________________ > From: freeswitch-users-bounces at lists.freeswitch.org [ > freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > freeswitch-users-request at lists.freeswitch.org [ > freeswitch-users-request at lists.freeswitch.org] > Sent: Monday, July 13, 2009 9:17 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Freeswitch-users Digest, Vol 37, Issue 67 > > 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: Dialogic cards (Anthony Minessale) > 2. Re: Help Regarding memory leak with freeswitch (Anthony Minessale) > 3. Re: Setting channel variables using event socket > (Anthony Minessale) > 4. Re: Dialogic cards (Steve Underwood) > 5. Re: Dialogic cards (Steve Underwood) > 6. Re: Dialogic cards (Anthony Minessale) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Mon, 13 Jul 2009 10:08:26 -0500 > From: Anthony Minessale > Subject: Re: [Freeswitch-users] Dialogic cards > To: freeswitch-users at lists.freeswitch.org > Message-ID: > <191c3a030907130808s37e52ddg3ba567d6a5a1034b at mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > Dialogic is coming to ClueCon this year (this aug 4th) and they are > sponsoring the conference. > I can discuss the possibility of supporting their cards at that time. > > > On Sun, Jul 12, 2009 at 11:07 PM, Tim Uckun wrote: > > > We have some older dialogic cards (D300 series E1 cards) and I am > > wondering if freeswitch can support these cards. > > > > 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/ > Twitter: http://twitter.com/FreeSWITCH_wire > > 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 > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090713/25e82735/attachment-0001.html > > ------------------------------ > > Message: 2 > Date: Mon, 13 Jul 2009 10:20:53 -0500 > From: Anthony Minessale > Subject: Re: [Freeswitch-users] Help Regarding memory leak with > freeswitch > To: freeswitch-users at lists.freeswitch.org > Message-ID: > <191c3a030907130820y46985521y1a01270a98cc2fcf at mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > Which revision of FreeSWITCH are you using? Several memory leaks have been > fixed since the last formal release. One specifically in REGISTER. > You should probably try SVN trunk or the latest pre-release of 1.0.4 > > > > > On Mon, Jul 13, 2009 at 2:07 AM, Rajagopal, Sridhar (Sridhar) < > sridhart at alcatel-lucent.com> wrote: > > > Hi all, > > > > I am running freeswitch on powerpc processor. I see memory being > allocated > > for each subsequent REGISTER requests coming to freeswitch. But not all > the > > memory allocated is not freed. If I run the code for two days the system > is > > running out of memory (RAM available to me is very less). > > The same memory issue is happening even for calls. > > > > Please let me know if any body has seen this issue. Please let me know > how > > I can go ahead and debug this issue. > > > > Thanks in advance for the help. > > > > Regards, > > Sridhar > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > 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 > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090713/b97f4578/attachment-0001.html > > ------------------------------ > > Message: 3 > Date: Mon, 13 Jul 2009 10:25:10 -0500 > From: Anthony Minessale > Subject: Re: [Freeswitch-users] Setting channel variables using event > socket > To: freeswitch-users at lists.freeswitch.org > Message-ID: > <191c3a030907130825q178fdfa9q1f3098258288ed3b at mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > and if you go the uuid_setvar route you do this: > > api uuid_setvar joe=out_to_lunch > > > > On Sun, Jul 12, 2009 at 6:00 PM, Brian West wrote: > > > call-command: execute > > execute-app-name: set > > execute-app-arg: fred=out_to_lunch > > > > > > On Jul 12, 2009, at 5:33 PM, Nik Middleton wrote: > > > > As in > > > > call-command: set joe=out_to_lunch ? > > > > > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > 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 > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090713/113023df/attachment-0001.html > > ------------------------------ > > Message: 4 > Date: Mon, 13 Jul 2009 23:39:39 +0800 > From: Steve Underwood > Subject: Re: [Freeswitch-users] Dialogic cards > To: freeswitch-users at lists.freeswitch.org > Message-ID: <4A5B553B.9020000 at coppice.org> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Tim Uckun wrote: > > We have some older dialogic cards (D300 series E1 cards) and I am > > wondering if freeswitch can support these cards. > > > Oh, I like the easy questions. No. It lacks the hardware features to do > anything useful with Freeswitch, or Asterisk, or Callweaver, or Yate, or > anything else that expects to do two way telephony through the host CPU. > > Steve > > > > > > ------------------------------ > > Message: 5 > Date: Mon, 13 Jul 2009 23:39:47 +0800 > From: Steve Underwood > Subject: Re: [Freeswitch-users] Dialogic cards > To: freeswitch-users at lists.freeswitch.org > Message-ID: <4A5B5543.6080405 at coppice.org> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Being now a mashup of several CTI companies, there are now a number of > disparate things called Dialogic cards. Some, like the cards previously > known as Prince..... er, Eicon are perfectly supportable. The old > Dialogic cards, like the D300 series, are not duplex to and from the > host. They are only duplex across the mezzanine bus. There's nothing you > can do with them. > > Steve > > > Anthony Minessale wrote: > > Dialogic is coming to ClueCon this year (this aug 4th) and they are > > sponsoring the conference. > > I can discuss the possibility of supporting their cards at that time. > > > > > > On Sun, Jul 12, 2009 at 11:07 PM, Tim Uckun > > wrote: > > > > We have some older dialogic cards (D300 series E1 cards) and I am > > wondering if freeswitch can support these cards. > > > > Thanks. > > > > > > > > ------------------------------ > > Message: 6 > Date: Mon, 13 Jul 2009 10:46:58 -0500 > From: Anthony Minessale > Subject: Re: [Freeswitch-users] Dialogic cards > To: freeswitch-users at lists.freeswitch.org > Message-ID: > <191c3a030907130846u4c134fc4n578589c196617557 at mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > ok, > > or we could ask Steve I guess. =D > > > On Mon, Jul 13, 2009 at 10:39 AM, Steve Underwood >wrote: > > > Tim Uckun wrote: > > > We have some older dialogic cards (D300 series E1 cards) and I am > > > wondering if freeswitch can support these cards. > > > > > Oh, I like the easy questions. No. It lacks the hardware features to do > > anything useful with Freeswitch, or Asterisk, or Callweaver, or Yate, or > > anything else that expects to do two way telephony through the host CPU. > > > > 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/ > Twitter: http://twitter.com/FreeSWITCH_wire > > 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 > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090713/2912d77b/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 37, Issue 67 > ************************************************ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference 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/20090713/4a541591/attachment-0002.html From fdhege at gmail.com Mon Jul 13 12:04:05 2009 From: fdhege at gmail.com (Dale) Date: Mon, 13 Jul 2009 15:04:05 -0400 Subject: [Freeswitch-users] Gateway Settings from-domain and caller-id-in-from Message-ID: <3DB91FF8-24DA-4031-9A7E-8B5390907D3B@gmail.com> Hello, I have been playing around with gateway settings today and noticed something that I wasn't sure if it was a bug or if its just the way it works. When I have from-domain set in my gateway config it correctly uses the configured from domain. If I then set caller-id-in-from to true the configured from domain is no longer used and it reverts back to the ip address configured for that sip_profile. Is that expected or is that a bug? Thanks, -Dale From brian at freeswitch.org Mon Jul 13 12:09:30 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 13 Jul 2009 14:09:30 -0500 Subject: [Freeswitch-users] Gateway Settings from-domain and caller-id-in-from In-Reply-To: <3DB91FF8-24DA-4031-9A7E-8B5390907D3B@gmail.com> References: <3DB91FF8-24DA-4031-9A7E-8B5390907D3B@gmail.com> Message-ID: <41E67BD3-25A5-41B6-B605-D7A75DA2342B@freeswitch.org> Can you collect up sip traces and open a jira please. /b On Jul 13, 2009, at 2:04 PM, Dale wrote: > > Hello, > > I have been playing around with gateway settings today and noticed > something that I wasn't sure if it was a bug or if its just the way it > works. > > When I have from-domain set in my gateway config it correctly uses the > configured from domain. If I then set caller-id-in-from to true the > configured from domain is no longer used and it reverts back to the ip > address configured for that sip_profile. > > Is that expected or is that a bug? > > Thanks, > > -Dale From saeedahmad1981 at gmail.com Mon Jul 13 12:38:42 2009 From: saeedahmad1981 at gmail.com (Saeed Ahmad) Date: Mon, 13 Jul 2009 21:38:42 +0200 Subject: [Freeswitch-users] Help In-Reply-To: <30ee97110907100944y15d1e90fu29b543cd87f12c5f@mail.gmail.com> References: <5E41B7C2-B163-4364-A2A9-8B90AD58F47E@frota.net> <30ee97110907100944y15d1e90fu29b543cd87f12c5f@mail.gmail.com> Message-ID: helpless On Fri, Jul 10, 2009 at 6:44 PM, Jens Vegeby wrote: > You might wanna write what you need help with :) > > On 7/10/09, Ney Frota wrote: > > Help > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at 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 > > Mvh/Regards Jens > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090713/c2daeb59/attachment-0002.html From siniypin at gmail.com Mon Jul 13 13:53:46 2009 From: siniypin at gmail.com (=?KOI8-R?B?8s/CxdLUIPTXxdLJ1M7F0g==?=) Date: Tue, 14 Jul 2009 00:53:46 +0400 Subject: [Freeswitch-users] port restricted NAT, SRTP problem Message-ID: <2160023e0907131353v39fe88a5i3b1c05ee9ee4dd23@mail.gmail.com> Hi guys! I'm a novice in VoIP world, and may be missing some important concepts, but recently I've faced a problem with client softphone residing behind a port-restricted NAT and a public FS server and can't find an explanation on why it is happening and how to escape it. Okey, the problem is as follows. I have a client residing behind port restricted NAT. It can register at our public server and can issue a looped call and hear itself perfectly well. But when I call to speek to this natted guy from computer exposed to web without any routers he able to hear me for merely a second and then I become muted and he hear nothing. On the other side I can hear this guy quite good, though with slight jittery sound. If I set bypass_media param in our server's external profile to true - everything works as it supposed to - we hear one another. But still there is a problem with call originating from that guy - it is being interrupted after some time (after about 30 sec). Both clients are capable to do STUN and ICE and have these options enabled. Calls are secured with TLS and SRTP enabled on our server. FreeSWITCH is installed on Windows Server 2008 box with open UDP traffic and TCP, UDP ports 5080,5081 opened in order to expose an external profile. As far as I understand, with bypass_media param disabled FreeSWITCH is acting as media proxy and it is unable to do ICE and that should be a reason why that guy can't hear me. In overmentioned peer2peer scenario switching to no media mode is acceptable, but still there is a question whether this call with media flow bypassing FreeSWITCH is secured? I guess not. Cause I don't have any certificates installed on clients. Also, we've plans to use our FreeSWITCH as a media conference server. And of course this guy failes to connect to the testing one. Below are some of configurations: vars.xml ... ... external.xml ... ... public dialplan My SDP: Remote SDP: v=0 o=- 3456517465 3456517465 IN IP4 91.79.44.168 s=pjmedia c=IN IP4 91.79.44.168 t=0 0 a=X-nat:2 m=audio 1142 RTP/SAVP 103 102 104 117 3 0 8 9 101 a=rtpmap:103 speex/16000 ... a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtcp:1143 IN IP4 91.79.44.168 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:ee+Xv9etM5t5w3DH5B1hR+i9lrt7BHQhzJIwFv7d a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:ELeudfX0mgsL+0u7qzGOqdEfg891fMn281BJszkS a=ice-ufrag:70db7dd4 a=ice-pwd:708804fe a=candidate:H 1 UDP 39 91.79.44.168 1142 typ host ... And this is the other guy SDP: Remote SDP: v=0 o=- 3456517469 3456517470 IN IP4 85.140.191.254 s=pjmedia c=IN IP4 85.140.191.254 t=0 0 a=X-nat:8 m=audio 23374 RTP/SAVP 103 101 a=rtpmap:103 speex/16000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtcp:23375 IN IP4 85.140.191.254 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:2m4YBCRid0h+5AIaIGmXaqelQsSuK3HP1jtAMoiG a=ice-ufrag:182f5a61 a=ice-pwd:1d690863 a=candidate:S 1 UDP 31 85.140.191.254 23374 typ srflx raddr 192.168.2.2 rport 2796 ... PS sorry for a long post -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090714/0b021252/attachment-0002.html From timuckun at gmail.com Mon Jul 13 14:33:54 2009 From: timuckun at gmail.com (Tim Uckun) Date: Tue, 14 Jul 2009 09:33:54 +1200 Subject: [Freeswitch-users] Dialogic cards In-Reply-To: <4A5B553B.9020000@coppice.org> References: <855e4dcf0907122107q392a9ca1h851dfbd57e2518ca@mail.gmail.com> <4A5B553B.9020000@coppice.org> Message-ID: <855e4dcf0907131433q33a21d2v935797013046aa81@mail.gmail.com> On Tue, Jul 14, 2009 at 3:39 AM, Steve Underwood wrote: > Tim Uckun wrote: >> We have some older dialogic cards (D300 series E1 cards) and I am >> wondering if freeswitch can support these cards. >> > Oh, I like the easy questions. No. It lacks the hardware features to do > anything useful with Freeswitch, or Asterisk, or Callweaver, or Yate, or > anything else that expects to do two way telephony through the host CPU. What are the recommended cards to be used with freeswitch? From msc at freeswitch.org Mon Jul 13 14:45:40 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 13 Jul 2009 14:45:40 -0700 Subject: [Freeswitch-users] Dialogic cards In-Reply-To: <855e4dcf0907131433q33a21d2v935797013046aa81@mail.gmail.com> References: <855e4dcf0907122107q392a9ca1h851dfbd57e2518ca@mail.gmail.com> <4A5B553B.9020000@coppice.org> <855e4dcf0907131433q33a21d2v935797013046aa81@mail.gmail.com> Message-ID: <87f2f3b90907131445r3bda46e9wa8650ce487c0b62c@mail.gmail.com> > What are the recommended cards to be used with freeswitch? > Sangoma cards and Zaptel/DAHDI compatible cards work well. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090713/07f49276/attachment-0002.html From eweaver at meetingone.com Mon Jul 13 16:35:17 2009 From: eweaver at meetingone.com (Weaver, Eric) Date: Mon, 13 Jul 2009 16:35:17 -0700 Subject: [Freeswitch-users] Preventing disconnect on event_sockte close Message-ID: <7DB369DA44D5CF42B6C23BD7D611A50FDA7E1D5FD6@VA3DIAXVS061.RED001.local> Using mod_event_socket in outbound mode, is there any to prevent a call from being disconnected when the outbound socket is closed ? I would like to handle the initial inbound call using outbound but after the disposition of the call is determined, close the socket and have that call managed using an inbound socket instead. Eric Weaver -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090713/6de37657/attachment-0002.html From msc at freeswitch.org Mon Jul 13 18:08:22 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 13 Jul 2009 18:08:22 -0700 Subject: [Freeswitch-users] Preventing disconnect on event_sockte close In-Reply-To: <7DB369DA44D5CF42B6C23BD7D611A50FDA7E1D5FD6@VA3DIAXVS061.RED001.local> References: <7DB369DA44D5CF42B6C23BD7D611A50FDA7E1D5FD6@VA3DIAXVS061.RED001.local> Message-ID: <87f2f3b90907131808l100afe7dwc95d0f624ff38d50@mail.gmail.com> I don't know if this will work for you but I just tested this scenario with uuid_park. After parking the call I disconnected the socket and the call continued. I did the same thing with uuid_transfer. After the transfer I disconnected the socket and the call continued. How are you handling the call and how is the socket getting disconnected? -MC On Mon, Jul 13, 2009 at 4:35 PM, Weaver, Eric wrote: > Using mod_event_socket in outbound mode, is there any to prevent a call > from being disconnected when the outbound socket is closed ? I would like to > handle the initial inbound call using outbound but after the disposition of > the call is determined, close the socket and have that call managed using an > inbound socket instead. > > > > Eric Weaver > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090713/a143bc38/attachment-0002.html From msc at freeswitch.org Mon Jul 13 18:10:13 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 13 Jul 2009 18:10:13 -0700 Subject: [Freeswitch-users] Originate in Dial plan In-Reply-To: <8ccbff060907120959ma7c0ey22adeae196f872c3@mail.gmail.com> References: <8ccbff060907120959ma7c0ey22adeae196f872c3@mail.gmail.com> Message-ID: <87f2f3b90907131810q219c04f3ra385e0e1d252d31c@mail.gmail.com> What phone number do you call back? I mean, how do you know what the customer's number is? Do you go by the caller id number? -MC On Sun, Jul 12, 2009 at 9:59 AM, Dome Charoenyost wrote: > Dear sir, > I want to user dialplan callback to customer. is posible to > to this is dialplan XML ? > Now i use javascript. > my call flow. > 1. customer call > 2. FS rining and wait until customer hangup > 3. callback to customer number > > > Best Regards. > > Dome C. > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090713/4f700998/attachment-0002.html From msc at freeswitch.org Mon Jul 13 18:11:56 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 13 Jul 2009 18:11:56 -0700 Subject: [Freeswitch-users] Getting xml_request in LUA In-Reply-To: <4A59A390.7030000@savion.huji.ac.il> References: <4A59A390.7030000@savion.huji.ac.il> Message-ID: <87f2f3b90907131811w312c802fofe04f2be158c1521@mail.gmail.com> On Sun, Jul 12, 2009 at 1:49 AM, Eli Hayun wrote: > In the Perl example I found: > > > How to access request parameters and how to return data > > You have two hashes that are populated for you by freeswitch. Those > hashes are: > > * %XML_REQUEST > * %XML_DATA > > I want to use LUA to set the directory and dialplan xml. How do I get > the XML_REQUEST/XML_DATA from LUA? Is this the information you are looking for? http://wiki.freeswitch.org/wiki/Mod_lua#For_serving_configuration > > > Thanks > Eli Hayun > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090713/304d2fe8/attachment-0002.html From eweaver at meetingone.com Mon Jul 13 18:47:12 2009 From: eweaver at meetingone.com (Weaver, Eric) Date: Mon, 13 Jul 2009 18:47:12 -0700 Subject: [Freeswitch-users] Preventing disconnect on event_sockte close In-Reply-To: <87f2f3b90907131808l100afe7dwc95d0f624ff38d50@mail.gmail.com> References: <7DB369DA44D5CF42B6C23BD7D611A50FDA7E1D5FD6@VA3DIAXVS061.RED001.local>, <87f2f3b90907131808l100afe7dwc95d0f624ff38d50@mail.gmail.com> Message-ID: <7DB369DA44D5CF42B6C23BD7D611A50FDA7E210D73@VA3DIAXVS061.RED001.local> I noticed it in testing last night using net cat. I killed netcat and the inbound call was disconnected, I'll try your suggestions tonight. Thanks for the reply, eric ________________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins [msc at freeswitch.org] Sent: Monday, July 13, 2009 7:08 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Preventing disconnect on event_sockte close I don't know if this will work for you but I just tested this scenario with uuid_park. After parking the call I disconnected the socket and the call continued. I did the same thing with uuid_transfer. After the transfer I disconnected the socket and the call continued. How are you handling the call and how is the socket getting disconnected? -MC On Mon, Jul 13, 2009 at 4:35 PM, Weaver, Eric > wrote: Using mod_event_socket in outbound mode, is there any to prevent a call from being disconnected when the outbound socket is closed ? I would like to handle the initial inbound call using outbound but after the disposition of the call is determined, close the socket and have that call managed using an inbound socket instead. Eric Weaver _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From klaus.teller at gmx.net Mon Jul 13 19:34:11 2009 From: klaus.teller at gmx.net (Klaus Teller) Date: Tue, 14 Jul 2009 04:34:11 +0200 Subject: [Freeswitch-users] Gafachi no passing caller number Message-ID: <20090714023411.151100@gmx.net> Hi, I tend to believe that we already had this working. Here is my origination string: {effective_caller_id_name=Paul Gascogne,effective_caller_id_number=16478343812}sofia/gateway/sip.gafachi.com/164783486421 The caller number is not being passed to the destination. Is there something i'm missing? Thanks, Klaus. -- GRATIS f?r alle GMX-Mitglieder: Die maxdome Movie-FLAT! Jetzt freischalten unter http://portal.gmx.net/de/go/maxdome01 From mrene_lists at avgs.ca Mon Jul 13 19:36:37 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Mon, 13 Jul 2009 22:36:37 -0400 Subject: [Freeswitch-users] Gafachi no passing caller number In-Reply-To: <20090714023411.151100@gmx.net> References: <20090714023411.151100@gmx.net> Message-ID: You need to escape the spaces with \s in the caller id name. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 13-Jul-09, at 10:34 PM, Klaus Teller wrote: > Hi, > > I tend to believe that we already had this working. Here is my > origination string: > > {effective_caller_id_name=Paul > Gascogne,effective_caller_id_number=16478343812}sofia/gateway/ > sip.gafachi.com/164783486421 > > The caller number is not being passed to the destination. Is there > something i'm missing? > > Thanks, > > Klaus. > > -- > GRATIS f?r alle GMX-Mitglieder: Die maxdome Movie-FLAT! > Jetzt freischalten unter http://portal.gmx.net/de/go/maxdome01 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From klaus.teller at gmx.net Mon Jul 13 19:43:20 2009 From: klaus.teller at gmx.net (Klaus Teller) Date: Tue, 14 Jul 2009 04:43:20 +0200 Subject: [Freeswitch-users] Gafachi no passing caller number In-Reply-To: References: <20090714023411.151100@gmx.net> Message-ID: <20090714024320.198030@gmx.net> It doesn't seem to work though. I tried removing the space completely as well as removing the caller name parameter. -------- Original-Nachricht -------- > Datum: Mon, 13 Jul 2009 22:36:37 -0400 > Von: Mathieu Rene > An: freeswitch-users at lists.freeswitch.org > Betreff: Re: [Freeswitch-users] Gafachi no passing caller number > You need to escape the spaces with \s in the caller id name. > > Mathieu Rene > Avant-Garde Solutions Inc > Office: + 1 (514) 664-1044 x100 > Cell: +1 (514) 664-1044 x200 > mrene at avgs.ca > > > > > On 13-Jul-09, at 10:34 PM, Klaus Teller wrote: > > > Hi, > > > > I tend to believe that we already had this working. Here is my > > origination string: > > > > {effective_caller_id_name=Paul > > Gascogne,effective_caller_id_number=16478343812}sofia/gateway/ > > sip.gafachi.com/164783486421 > > > > The caller number is not being passed to the destination. Is there > > something i'm missing? > > > > Thanks, > > > > Klaus. > > > > -- > > GRATIS f?r alle GMX-Mitglieder: Die maxdome Movie-FLAT! > > Jetzt freischalten unter http://portal.gmx.net/de/go/maxdome01 > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- GRATIS f?r alle GMX-Mitglieder: Die maxdome Movie-FLAT! Jetzt freischalten unter http://portal.gmx.net/de/go/maxdome01 From mrene_lists at avgs.ca Mon Jul 13 19:47:18 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Mon, 13 Jul 2009 22:47:18 -0400 Subject: [Freeswitch-users] Gafachi no passing caller number In-Reply-To: <20090714024320.198030@gmx.net> References: <20090714023411.151100@gmx.net> <20090714024320.198030@gmx.net> Message-ID: <3DD2DE9A-7726-4167-B968-7434A7BEFBEC@avgs.ca> Oh you're using effective_caller_id_number, those vars are only checked when an a-leg exists. Use origination_caller_id_number and origination_caller_id_name. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 13-Jul-09, at 10:43 PM, Klaus Teller wrote: > It doesn't seem to work though. I tried removing the space > completely as well as removing the caller name parameter. > > -------- Original-Nachricht -------- >> Datum: Mon, 13 Jul 2009 22:36:37 -0400 >> Von: Mathieu Rene >> An: freeswitch-users at lists.freeswitch.org >> Betreff: Re: [Freeswitch-users] Gafachi no passing caller number > >> You need to escape the spaces with \s in the caller id name. >> >> Mathieu Rene >> Avant-Garde Solutions Inc >> Office: + 1 (514) 664-1044 x100 >> Cell: +1 (514) 664-1044 x200 >> mrene at avgs.ca >> >> >> >> >> On 13-Jul-09, at 10:34 PM, Klaus Teller wrote: >> >>> Hi, >>> >>> I tend to believe that we already had this working. Here is my >>> origination string: >>> >>> {effective_caller_id_name=Paul >>> Gascogne,effective_caller_id_number=16478343812}sofia/gateway/ >>> sip.gafachi.com/164783486421 >>> >>> The caller number is not being passed to the destination. Is there >>> something i'm missing? >>> >>> Thanks, >>> >>> Klaus. >>> >>> -- >>> GRATIS f?r alle GMX-Mitglieder: Die maxdome Movie-FLAT! >>> Jetzt freischalten unter http://portal.gmx.net/de/go/maxdome01 >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > -- > GRATIS f?r alle GMX-Mitglieder: Die maxdome Movie-FLAT! > Jetzt freischalten unter http://portal.gmx.net/de/go/maxdome01 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.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 Mon Jul 13 19:48:15 2009 From: sicfslist at gmail.com (Shelby Ramsey) Date: Mon, 13 Jul 2009 21:48:15 -0500 Subject: [Freeswitch-users] Gafachi no passing caller number In-Reply-To: <20090714024320.198030@gmx.net> References: <20090714023411.151100@gmx.net> <20090714024320.198030@gmx.net> Message-ID: <35b355e90907131948r651f1fdcle139031d8f6a9521@mail.gmail.com> Klaus, Use ngrep and see if the From / RPID headers are correct in the SIP message. This will let you know if FS is doing the correct thing. SDR -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090713/a8caa856/attachment-0002.html From klaus.teller at gmx.net Mon Jul 13 19:53:15 2009 From: klaus.teller at gmx.net (Klaus Teller) Date: Tue, 14 Jul 2009 04:53:15 +0200 Subject: [Freeswitch-users] Gafachi no passing caller number In-Reply-To: <3DD2DE9A-7726-4167-B968-7434A7BEFBEC@avgs.ca> References: <20090714023411.151100@gmx.net> <20090714024320.198030@gmx.net> <3DD2DE9A-7726-4167-B968-7434A7BEFBEC@avgs.ca> Message-ID: <20090714025315.198040@gmx.net> Thanks folks. Indeed i had to use origination_caller_id_number. Cheers, Klaus. -------- Original-Nachricht -------- > Datum: Mon, 13 Jul 2009 22:47:18 -0400 > Von: Mathieu Rene > An: freeswitch-users at lists.freeswitch.org > Betreff: Re: [Freeswitch-users] Gafachi no passing caller number > Oh you're using effective_caller_id_number, those vars are only > checked when an a-leg exists. > > Use origination_caller_id_number and origination_caller_id_name. > > Mathieu Rene > Avant-Garde Solutions Inc > Office: + 1 (514) 664-1044 x100 > Cell: +1 (514) 664-1044 x200 > mrene at avgs.ca > > > > > On 13-Jul-09, at 10:43 PM, Klaus Teller wrote: > > > It doesn't seem to work though. I tried removing the space > > completely as well as removing the caller name parameter. > > > > -------- Original-Nachricht -------- > >> Datum: Mon, 13 Jul 2009 22:36:37 -0400 > >> Von: Mathieu Rene > >> An: freeswitch-users at lists.freeswitch.org > >> Betreff: Re: [Freeswitch-users] Gafachi no passing caller number > > > >> You need to escape the spaces with \s in the caller id name. > >> > >> Mathieu Rene > >> Avant-Garde Solutions Inc > >> Office: + 1 (514) 664-1044 x100 > >> Cell: +1 (514) 664-1044 x200 > >> mrene at avgs.ca > >> > >> > >> > >> > >> On 13-Jul-09, at 10:34 PM, Klaus Teller wrote: > >> > >>> Hi, > >>> > >>> I tend to believe that we already had this working. Here is my > >>> origination string: > >>> > >>> {effective_caller_id_name=Paul > >>> Gascogne,effective_caller_id_number=16478343812}sofia/gateway/ > >>> sip.gafachi.com/164783486421 > >>> > >>> The caller number is not being passed to the destination. Is there > >>> something i'm missing? > >>> > >>> Thanks, > >>> > >>> Klaus. > >>> > >>> -- > >>> GRATIS f?r alle GMX-Mitglieder: Die maxdome Movie-FLAT! > >>> Jetzt freischalten unter http://portal.gmx.net/de/go/maxdome01 > >>> > >>> _______________________________________________ > >>> Freeswitch-users mailing list > >>> Freeswitch-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> http://www.freeswitch.org > >> > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > -- > > GRATIS f?r alle GMX-Mitglieder: Die maxdome Movie-FLAT! > > Jetzt freischalten unter http://portal.gmx.net/de/go/maxdome01 > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Neu: GMX Doppel-FLAT mit Internet-Flatrate + Telefon-Flatrate f?r nur 19,99 Euro/mtl.!* http://portal.gmx.net/de/go/dsl02 From jingwei.yang at gmail.com Mon Jul 13 20:34:32 2009 From: jingwei.yang at gmail.com (Jingwei Yang) Date: Tue, 14 Jul 2009 11:34:32 +0800 Subject: [Freeswitch-users] mod_dingaling no audio In-Reply-To: <507898380907130429haeb7ef9haeaf31b7e409da5c@mail.gmail.com> References: <13529f9d0906240240x55f0c19ao7246de1d34b1f661@mail.gmail.com> <13529f9d0906242231s7384908ayb30340f5442b2ab@mail.gmail.com> <507898380906250602r4c666b06xfdeea5943ad0cbff@mail.gmail.com> <13529f9d0906251915m440a1c46s534a5c7c735b8ce8@mail.gmail.com> <507898380906251930v6f82d786l9d1bf0da61212b05@mail.gmail.com> <13529f9d0906252034o5cb0b7b7y8a9ffe750edf2b20@mail.gmail.com> <13529f9d0906290225v4abd1552l4a3b72e0ff1afc66@mail.gmail.com> <507898380906290346w20cfea6bq82114efc63439f52@mail.gmail.com> <13529f9d0907130227u377b7459l8143a37b9ccfc761@mail.gmail.com> <507898380907130429haeb7ef9haeaf31b7e409da5c@mail.gmail.com> Message-ID: <13529f9d0907132034j546feeabj2aa7185f9711186a@mail.gmail.com> Hi Chris, I've attached the console logs for your reference. It really hits 888 in the dialplan and the external call can hear the echo without any problem. One thing attracts me is how the ip addresses are translated. Here's the working external example: *(external party's local addr)* * (external party's global addr)* *(server's global addr)* Here's the non-working internal exmaple: *(internal party's local addr)* *(internal party's global addr) * * (server's global addr)* *(internal party's local addr, again!) * *(internal party's global addr) * *(google's global addr)* *(google's global addr) * *(internal party's local addr, 3rd time!) * *(google's global addr)* And finally, when the call was hung up, the internal one showed an error like this: unknown session Regards, -Jingwei On Mon, Jul 13, 2009 at 7:29 PM, Chris Chen wrote: > Jingwei, can you show your console log when somebody is calling you from > gtalk client? Will it really hit 888 in your dialplan? > Thanks, > Chris > > > On Mon, Jul 13, 2009 at 5:27 AM, Jingwei Yang wrote: > >> Hi Chris, sorry for the late reply. Have been quite busy last few days. >> >> I had shifted 888 from default.xml to public.xml and the dialplan is >> simply having an echo action now. I've turned on dl_debug but unfortunately >> didn't find anything useful. Logs are attached for your reference. >> >> I don't think there's something wrong with the dialplan as two external >> parties can talk to each other perfectly (with ext-rtp-ip uncommented, at >> this time my ip was interpreted to be an external one). With ext-rtp-ip >> commented, I can hear the echo and I saw my ip was translated into an >> internal one (at this time, external party's audio failed). >> >> I tried the method on this wiki page as well: >> http://wiki.freeswitch.org/wiki/NAT_Traversal (the last FreeSwitch behind >> NAT portion) but still no luck. Please kindly let me know what other configs >> I should change. >> >> Thanks, >> -Jingwei >> >> On Mon, Jun 29, 2009 at 6:46 PM, Chris Chen wrote: >> >>> Jingwei, I don't know if you have the 888 defined in default.xml? also >>> you have to define $${domain}. >>> please do " dl_debug on" from fs_cli, and watch the console logs and see >>> what's going on when you try calling from external. Most likely your >>> dialplan is not correctly defined. >>> >>> Chris >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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... 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Name: external_call_in.txt Url: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090714/e4a3eb1a/attachment-0002.txt -------------- next part -------------- freeswitch at localhost.localdomain> 2009-07-14 10:20:12.779916 [INFO] libdingaling.c:1307 SecRECV: ------------------------------------------------------------------------------- 2009-07-14 10:20:12.779916 [INFO] libdingaling.c:1307 SecRECV: ------------------------------------------------------------------------------- 2009-07-14 10:20:12.779916 [NOTICE] switch_channel.c:602 New Channel dingaling/888 [2ed89b9b-bb8b-4606-a71d-2fe6dbd66d1d] 2009-07-14 10:20:12.779916 [NOTICE] switch_channel.c:600 Rename Channel dingaling/888->DingaLing/new [2ed89b9b-bb8b-4606-a71d-2fe6dbd66d1d] 2009-07-14 10:20:12.782051 [NOTICE] mod_dingaling.c:1084 Ring-Ready DingaLing/new! 2009-07-14 10:20:12.803916 [NOTICE] libdingaling.c:1309 SecSEND: ------------------------------------------------------------------------------- 2009-07-14 10:20:12.803916 [NOTICE] libdingaling.c:1309 SecSEND: ------------------------------------------------------------------------------- 2009-07-14 10:20:13.679871 [INFO] libdingaling.c:1307 SecRECV: ------------------------------------------------------------------------------- 2009-07-14 10:20:13.679871 [INFO] libdingaling.c:1307 SecRECV: ------------------------------------------------------------------------------- 2009-07-14 10:20:13.711870 [NOTICE] libdingaling.c:1309 SecSEND: ------------------------------------------------------------------------------- 2009-07-14 10:20:14.729818 [INFO] mod_dingaling.c:974 Stun Success 59.189.194.244:30578 2009-07-14 10:20:14.729818 [INFO] libdingaling.c:1307 SecRECV: ------------------------------------------------------------------------------- 2009-07-14 10:20:14.748836 [INFO] mod_dialplan_xml.c:252 Processing xxxxxx at gmail.com/Talk.v1046D90E88C->888 in context public 2009-07-14 10:20:14.748836 [NOTICE] switch_core_session.c:1391 Pre-Answer DingaLing/new! 2009-07-14 10:20:14.814814 [NOTICE] libdingaling.c:1309 SecSEND: ------------------------------------------------------------------------------- 2009-07-14 10:20:14.814814 [NOTICE] libdingaling.c:1309 SecSEND: ------------------------------------------------------------------------------- 2009-07-14 10:20:15.544777 [INFO] libdingaling.c:1307 SecRECV: ------------------------------------------------------------------------------- 2009-07-14 10:20:15.615820 [NOTICE] libdingaling.c:1309 SecSEND: ------------------------------------------------------------------------------- 2009-07-14 10:20:15.826763 [INFO] libdingaling.c:1307 SecRECV: ------------------------------------------------------------------------------- 2009-07-14 10:20:16.740726 [INFO] libdingaling.c:1307 SecRECV: ------------------------------------------------------------------------------- 2009-07-14 10:20:16.816760 [NOTICE] libdingaling.c:1309 SecSEND: ------------------------------------------------------------------------------- 2009-07-14 10:20:17.533727 [INFO] libdingaling.c:1307 SecRECV: ------------------------------------------------------------------------------- 2009-07-14 10:20:17.617673 [NOTICE] libdingaling.c:1309 SecSEND: ------------------------------------------------------------------------------- 2009-07-14 10:20:17.774664 [INFO] libdingaling.c:1307 SecRECV: ------------------------------------------------------------------------------- 2009-07-14 10:20:17.818663 [NOTICE] libdingaling.c:1309 SecSEND: ------------------------------------------------------------------------------- 2009-07-14 10:20:18.532660 [INFO] libdingaling.c:1307 SecRECV: ------------------------------------------------------------------------------- 2009-07-14 10:20:18.619622 [NOTICE] libdingaling.c:1309 SecSEND: ------------------------------------------------------------------------------- 2009-07-14 10:20:24.211338 [INFO] libdingaling.c:1307 SecRECV: ------------------------------------------------------------------------------- 2009-07-14 10:20:24.211338 [NOTICE] mod_dingaling.c:718 Hangup DingaLing/new [CS_EXECUTE] [NORMAL_CLEARING] 2009-07-14 10:20:24.216352 [NOTICE] switch_core_session.c:1085 Session 3 (DingaLing/new) Ended 2009-07-14 10:20:24.216352 [NOTICE] switch_core_session.c:1087 Close Channel DingaLing/new [CS_DESTROY] 2009-07-14 10:20:24.222338 [NOTICE] libdingaling.c:1309 SecSEND: ------------------------------------------------------------------------------- 2009-07-14 10:20:24.222338 [NOTICE] libdingaling.c:1309 SecSEND: ------------------------------------------------------------------------------- 2009-07-14 10:20:24.944314 [INFO] libdingaling.c:1307 SecRECV: ------------------------------------------------------------------------------- unknown session From dome at tel.co.th Mon Jul 13 21:30:56 2009 From: dome at tel.co.th (Dome Charoenyost) Date: Tue, 14 Jul 2009 11:30:56 +0700 Subject: [Freeswitch-users] Originate in Dial plan In-Reply-To: <87f2f3b90907131810q219c04f3ra385e0e1d252d31c@mail.gmail.com> References: <8ccbff060907120959ma7c0ey22adeae196f872c3@mail.gmail.com> <87f2f3b90907131810q219c04f3ra385e0e1d252d31c@mail.gmail.com> Message-ID: <8ccbff060907132130l329f7e1at87a16da0c2108e2@mail.gmail.com> 2009/7/14 Michael Collins : > What phone number do you call back? I mean, how do you know what the > customer's number is? Do you go by the caller id number? yes callback to caller id > > -MC > > On Sun, Jul 12, 2009 at 9:59 AM, Dome Charoenyost wrote: >> >> Dear sir, >> ? ? ? ? I want to user dialplan callback to customer. is posible to >> to this is dialplan XML ? >> Now i use javascript. >> my call flow. >> 1. customer call >> 2. FS rining and wait until customer hangup >> 3. callback to customer number >> >> >> Best Regards. >> >> Dome C. >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From velu.technical at gmail.com Mon Jul 13 23:37:50 2009 From: velu.technical at gmail.com (velusamy velu) Date: Tue, 14 Jul 2009 12:07:50 +0530 Subject: [Freeswitch-users] Problem in Adding another user in default directory Message-ID: <1452e2980907132337w45cc550ci587f811d9f3851f@mail.gmail.com> Dear All, How to create another user agent like 1000 to 1919 in internal profile. Please provide some steps to do it.. Thanks in Advance, Velusamy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090714/5f6c7ccc/attachment-0002.html From jason at jasonjgw.net Tue Jul 14 00:10:02 2009 From: jason at jasonjgw.net (Jason White) Date: Tue, 14 Jul 2009 17:10:02 +1000 Subject: [Freeswitch-users] Problem in Adding another user in default directory In-Reply-To: <1452e2980907132337w45cc550ci587f811d9f3851f@mail.gmail.com> References: <1452e2980907132337w45cc550ci587f811d9f3851f@mail.gmail.com> Message-ID: <20090714071002.GA6485@jdc.jasonjgw.net> velusamy velu wrote: > How to create another user agent like 1000 to 1919 in internal > profile. Copy one of the existing files, edit it, and make all of the obvious changes. Then edit your dial plan so that the extension can be called. From msc at freeswitch.org Tue Jul 14 00:48:28 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 14 Jul 2009 00:48:28 -0700 Subject: [Freeswitch-users] Originate in Dial plan In-Reply-To: <8ccbff060907132130l329f7e1at87a16da0c2108e2@mail.gmail.com> References: <8ccbff060907120959ma7c0ey22adeae196f872c3@mail.gmail.com> <87f2f3b90907131810q219c04f3ra385e0e1d252d31c@mail.gmail.com> <8ccbff060907132130l329f7e1at87a16da0c2108e2@mail.gmail.com> Message-ID: <87f2f3b90907140048m7e1eea18x46e9418e43fb4007@mail.gmail.com> On Mon, Jul 13, 2009 at 9:30 PM, Dome Charoenyost wrote: > 2009/7/14 Michael Collins : > > What phone number do you call back? I mean, how do you know what the > > customer's number is? Do you go by the caller id number? > yes callback to caller id > Okay, here's a dialplan snippet that I used to successfully do the autocallback. In my case I used ext 1001 as the customer and portaudio as the "agent" if you get my meaning. Extension 1001 dials 9902, hangs up, and immediately the api_hangup_hook's originate command is executed. In this case it calls portaudio/auto_answer for the A-leg and user/1001 as the B-leg. I don't claim that it's the prettiest thing in the world but it definitely works. You'll need to adjust according to your specific situation. Let us know how it goes. BTW, what is the reason for this type of scenario? Just curious. -MC > > > > > > -MC > > > > On Sun, Jul 12, 2009 at 9:59 AM, Dome Charoenyost > wrote: > >> > >> Dear sir, > >> I want to user dialplan callback to customer. is posible to > >> to this is dialplan XML ? > >> Now i use javascript. > >> my call flow. > >> 1. customer call > >> 2. FS rining and wait until customer hangup > >> 3. callback to customer number > >> > >> > >> Best Regards. > >> > >> Dome C. > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090714/54ad43f0/attachment-0002.html From andy at fabulous4.co.uk Tue Jul 14 02:07:09 2009 From: andy at fabulous4.co.uk (Andy) Date: Tue, 14 Jul 2009 10:07:09 +0100 Subject: [Freeswitch-users] Problems with Ping and re-registering brokengateways In-Reply-To: <191c3a030907130859y2665da03xf704f9db6070fb3c@mail.gmail.com> References: <0324DD608A074940AAC173B85A3978F2@D810> <191c3a030907130859y2665da03xf704f9db6070fb3c@mail.gmail.com> Message-ID: <78A03DDB5A5043C597807D0DFCF00DCA@D810> Hi Anthony, Thanks for your reply. The trace of the ping request looks like this. Any clues? send 674 bytes to udp/[77.240.48.94]:5060 at 09:07:25.479313: ------------------------------------------------------------------------ OPTIONS sip:voiptalk.org;transport=udp SIP/2.0 Via: SIP/2.0/UDP 77.86.49.249;rport;branch=z9hG4bK8rBQ4a33Ny02K Max-Forwards: 70 From: ;tag=a8U21NNZ23tBB To: Call-ID: 96da25e1-eaf8-122c-02ae-0019d15ec7e1 CSeq: 117662779 OPTIONS Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13850 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 ------------------------------------------------------------------------ recv 352 bytes from udp/[77.240.48.94]:5060 at 09:07:25.486124: ------------------------------------------------------------------------ SIP/2.0 484 Address Incomplete Via: SIP/2.0/UDP 77.86.49.249;rport=5060;branch=z9hG4bK8rBQ4a33Ny02K From: ;tag=a8U21NNZ23tBB To: ;tag=fd79486175647ed1617969929fdaad02.f21c Call-ID: 96da25e1-eaf8-122c-02ae-0019d15ec7e1 CSeq: 117662779 OPTIONS Server: OpenSIPS (1.5.1-notls (x86_64/linux)) Content-Length: 0 ------------------------------------------------------------------------ 2009-07-14 10:07:25.483507 [WARNING] sofia.c:2810 Ping failed voiptalk.org 2009-07-14 10:07:25.483507 [WARNING] sofia.c:2813 Unregister voiptalk.org Andy _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: 13 July 2009 16:59 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Problems with Ping and re-registering brokengateways Are they ignoring the options packet we send them or are they maybe getting lost behind NAT? we send an OPTIONS and even if we get a error back we consider that a successful reply. We did have a patch into SVN very recently to correct a problem with OPTIONS ping in a NAT situation. Maybe try latest trunk first then capture the console log with sip traffic in place if it still does not work so we can have a look. to capture the log use these 2 commands from the cli. sofia profile internal siptrace on console loglevel debug On Mon, Jul 13, 2009 at 7:59 AM, Andy wrote: Hi, I'm fairly sure my problem lies with my voip provider VoipTalk but wonder if you could help me understand a couple of things. My config is very simple, I'm using freeswitch to accept incoming calls via a voip gateway and record messages. Here's the problem: - When freeswitch starts the gateways are all created and register correctly - I have the ping parameter set to make sure the gateway stays alive. - The first time freeswitch pings the gateway it fails even though the registration appears intact as calls are still coming through to freeswitch - Freeswitch then tries to re-register the gateway but this fails. The SIP trace shows an Unauthorized message and the actual log entry is 'Registration Failed with status Operation has no matching challenge [904]' - eventually the registration times out with my provider and all is lost. - if I call 'sofia profile external restart' or restart the software this fixes the problem My questions are: 1) Why would the ping fail when the registration appears to be intact? 2) Whay would the auto re-register not work but a restart would? This ones driving me nuts so any help greatly appreciated. regards Andy _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference 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/20090714/2bf45931/attachment-0002.html From dome at tel.co.th Tue Jul 14 04:52:47 2009 From: dome at tel.co.th (Dome Charoenyost) Date: Tue, 14 Jul 2009 18:52:47 +0700 Subject: [Freeswitch-users] Originate in Dial plan In-Reply-To: <87f2f3b90907140048m7e1eea18x46e9418e43fb4007@mail.gmail.com> References: <8ccbff060907120959ma7c0ey22adeae196f872c3@mail.gmail.com> <87f2f3b90907131810q219c04f3ra385e0e1d252d31c@mail.gmail.com> <8ccbff060907132130l329f7e1at87a16da0c2108e2@mail.gmail.com> <87f2f3b90907140048m7e1eea18x46e9418e43fb4007@mail.gmail.com> Message-ID: <8ccbff060907140452s7b8bc06esd03b45424a0a545c@mail.gmail.com> Thanks it's work api_hangup_hook i'm looking for :) Dome C. 2009/7/14 Michael Collins : > > > On Mon, Jul 13, 2009 at 9:30 PM, Dome Charoenyost wrote: >> >> 2009/7/14 Michael Collins : >> > What phone number do you call back? I mean, how do you know what the >> > customer's number is? Do you go by the caller id number? >> yes callback to caller id > > Okay, here's a dialplan snippet that I used to successfully do the > autocallback. In my case I used ext 1001 as the customer and portaudio as > the "agent" if you get my meaning. Extension 1001 dials 9902, hangs up, and > immediately the api_hangup_hook's originate command is executed. In this > case it calls portaudio/auto_answer for the A-leg and user/1001 as the > B-leg. I don't claim that it's the prettiest thing in the world but it > definitely works. You'll need to adjust according to your specific > situation. > > ? > ??? > ??? > ????? > ????? > ????? > ????? > ????? > ????? > ??? > ? > > ? > ??? > ????? > ??? > ? > > > Let us know how it goes. BTW, what is the reason for this type of scenario? > Just curious. > -MC >> >> >> > >> > -MC >> > >> > On Sun, Jul 12, 2009 at 9:59 AM, Dome Charoenyost >> > wrote: >> >> >> >> Dear sir, >> >> ? ? ? ? I want to user dialplan callback to customer. is posible to >> >> to this is dialplan XML ? >> >> Now i use javascript. >> >> my call flow. >> >> 1. customer call >> >> 2. FS rining and wait until customer hangup >> >> 3. callback to customer number >> >> >> >> >> >> Best Regards. >> >> >> >> Dome C. >> >> >> >> _______________________________________________ >> >> Freeswitch-users mailing list >> >> Freeswitch-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> > >> > >> > _______________________________________________ >> > Freeswitch-users mailing list >> > Freeswitch-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From elihayun at gmail.com Tue Jul 14 05:02:18 2009 From: elihayun at gmail.com (Eli Hayun) Date: Tue, 14 Jul 2009 15:02:18 +0300 Subject: [Freeswitch-users] Get voicemail messages Message-ID: <4A5C73CA.40306@savion.huji.ac.il> Hi I am not using fixed xml files for the extension registration. I have LUA script to return an XML string to FS. Everything goes fine until I am trying to get the voice messages. When am entering my id, FS (or voicemail module) try to get the xml for that id, but it cant find it. My lua script did NOT recieved any xml request at that point. What should I do to solve the problem. Thanks Eli Hayun From yehavi.bourvine at gmail.com Tue Jul 14 05:12:27 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Tue, 14 Jul 2009 15:12:27 +0300 Subject: [Freeswitch-users] Originate in Dial plan In-Reply-To: <87f2f3b90907140048m7e1eea18x46e9418e43fb4007@mail.gmail.com> References: <8ccbff060907120959ma7c0ey22adeae196f872c3@mail.gmail.com> <87f2f3b90907131810q219c04f3ra385e0e1d252d31c@mail.gmail.com> <8ccbff060907132130l329f7e1at87a16da0c2108e2@mail.gmail.com> <87f2f3b90907140048m7e1eea18x46e9418e43fb4007@mail.gmail.com> Message-ID: 2009/7/14 Michael Collins > > > On Mon, Jul 13, 2009 at 9:30 PM, Dome Charoenyost wrote: > >> 2009/7/14 Michael Collins : >> > What phone number do you call back? I mean, how do you know what the >> > customer's number is? Do you go by the caller id number? >> yes callback to caller id >> > > Okay, here's a dialplan snippet that I used to successfully do the > autocallback. In my case I used ext 1001 as the customer and portaudio as > the "agent" if you get my meaning. Extension 1001 dials 9902, hangs up, and > immediately the api_hangup_hook's originate command is executed. In this > case it calls portaudio/auto_answer for the A-leg and user/1001 as the > B-leg. I don't claim that it's the prettiest thing in the world but it > definitely works. You'll need to adjust according to your specific > situation. > > > > > > > > > > > > > > > > > > > > > Let us know how it goes. BTW, what is the reason for this type of scenario? > Just curious. > -MC > >> >> >> > >> > -MC >> > >> > On Sun, Jul 12, 2009 at 9:59 AM, Dome Charoenyost >> wrote: >> >> >> >> Dear sir, >> >> I want to user dialplan callback to customer. is posible to >> >> to this is dialplan XML ? >> >> Now i use javascript. >> >> my call flow. >> >> 1. customer call >> >> 2. FS rining and wait until customer hangup >> >> 3. callback to customer number >> >> >> >> >> >> Best Regards. >> >> >> >> Dome C. >> >> >> >> _______________________________________________ >> >> Freeswitch-users mailing list >> >> Freeswitch-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> > >> > >> > _______________________________________________ >> > Freeswitch-users mailing list >> > Freeswitch-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090714/133488d3/attachment-0002.html From elihayun at gmail.com Tue Jul 14 05:19:33 2009 From: elihayun at gmail.com (Eli Hayun) Date: Tue, 14 Jul 2009 15:19:33 +0300 Subject: [Freeswitch-users] Getting xml_request in LUA Message-ID: <4A5C77D5.6040105@savion.huji.ac.il> In the Perl example I found: > > > How to access request parameters and how to return data > > You have two hashes that are populated for you by freeswitch. Those > hashes are: > > * %XML_REQUEST > * %XML_DATA > > I want to use LUA to set the directory and dialplan xml. How do I get > the XML_REQUEST/XML_DATA from LUA? Is this the information you are looking for? http://wiki.freeswitch.org/wiki/Mod_lua#For_serving_configuration Hi Thanks fro your answer No, this is not the information i am looking for. What I need is the "section" value that FS pass when using HTTP to get the XML from. This information is not available (at least, I don't know how to get it) Thanks Eli From msc at freeswitch.org Tue Jul 14 06:00:10 2009 From: msc at freeswitch.org (Michael S Collins) Date: Tue, 14 Jul 2009 06:00:10 -0700 Subject: [Freeswitch-users] Get voicemail messages In-Reply-To: <4A5C73CA.40306@savion.huji.ac.il> References: <4A5C73CA.40306@savion.huji.ac.il> Message-ID: <4429558B-4DA5-43BF-9824-D5B407546CCB@freeswitch.org> On Jul 14, 2009, at 5:02 AM, Eli Hayun wrote: > Hi > I am not using fixed xml files for the extension registration. I have > LUA script to return an XML string to FS. > Everything goes fine until I am trying to get the voice messages. > When am entering my id, FS (or voicemail module) try to get the xml > for > that id, but it cant find it. My lua script did NOT recieved any xml > request at that point. > What should I do to solve the problem. > > Thanks > Eli Hayun Can you pastebin the log? Be sure to press F8 to turn up the debug level. -MC From brian at freeswitch.org Tue Jul 14 06:02:43 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 14 Jul 2009 08:02:43 -0500 Subject: [Freeswitch-users] Gafachi no passing caller number In-Reply-To: <20090714023411.151100@gmx.net> References: <20090714023411.151100@gmx.net> Message-ID: <9376D332-3935-4019-A440-CADD8B4DBD6E@freeswitch.org> You single quote them. {effective_caller_id_name='Paul Gascogne',effective_caller_id_number=16478343812}sofia/gateway/ sip.gafachi.com/164783486421 /b On Jul 13, 2009, at 9:34 PM, Klaus Teller wrote: > Hi, > > I tend to believe that we already had this working. Here is my > origination string: > > {effective_caller_id_name=Paul > Gascogne,effective_caller_id_number=16478343812}sofia/gateway/ > sip.gafachi.com/164783486421 > > The caller number is not being passed to the destination. Is there > something i'm missing? > > Thanks, > > Klaus. From brian at freeswitch.org Tue Jul 14 06:04:07 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 14 Jul 2009 08:04:07 -0500 Subject: [Freeswitch-users] Problems with Ping and re-registering brokengateways In-Reply-To: <78A03DDB5A5043C597807D0DFCF00DCA@D810> References: <0324DD608A074940AAC173B85A3978F2@D810> <191c3a030907130859y2665da03xf704f9db6070fb3c@mail.gmail.com> <78A03DDB5A5043C597807D0DFCF00DCA@D810> Message-ID: <6C973FA2-4148-46C5-B98C-13569CB7533E@freeswitch.org> They probably shouldn't be responding 484 to them... hrm /b On Jul 14, 2009, at 4:07 AM, Andy wrote: > Hi Anthony, > > Thanks for your reply. The trace of the ping request looks like > this. Any clues? > > send 674 bytes to udp/[77.240.48.94]:5060 at 09:07:25.479313: > > ------------------------------------------------------------------------ > OPTIONS sip:voiptalk.org;transport=udp SIP/2.0 > Via: SIP/2.0/UDP 77.86.49.249;rport;branch=z9hG4bK8rBQ4a33Ny02K > Max-Forwards: 70 > From: ;tag=a8U21NNZ23tBB > To: > Call-ID: 96da25e1-eaf8-122c-02ae-0019d15ec7e1 > CSeq: 117662779 OPTIONS > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13850 > 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 > > > ------------------------------------------------------------------------ > recv 352 bytes from udp/[77.240.48.94]:5060 at 09:07:25.486124: > > ------------------------------------------------------------------------ > SIP/2.0 484 Address Incomplete > Via: SIP/2.0/UDP > 77.86.49.249;rport=5060;branch=z9hG4bK8rBQ4a33Ny02K > From: ;tag=a8U21NNZ23tBB > To: ;tag=fd79486175647ed1617969929fdaad02.f21c > Call-ID: 96da25e1-eaf8-122c-02ae-0019d15ec7e1 > CSeq: 117662779 OPTIONS > Server: OpenSIPS (1.5.1-notls (x86_64/linux)) > Content-Length: 0 > > ------------------------------------------------------------------------ > 2009-07-14 10:07:25.483507 [WARNING] sofia.c:2810 Ping failed > voiptalk.org > 2009-07-14 10:07:25.483507 [WARNING] sofia.c:2813 Unregister > voiptalk.org > > Andy > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090714/6410520b/attachment-0002.html From yivzhenko at mksat.net Tue Jul 14 06:18:03 2009 From: yivzhenko at mksat.net (Yuriy Ivzhenko) Date: Tue, 14 Jul 2009 16:18:03 +0300 Subject: [Freeswitch-users] Freeswitch architecture In-Reply-To: <347F12AB-EA05-4090-A203-BDC011F02411@freeswitch.org> References: <89c9bbf80907080542y39f2b234tff9163ea34c0b968@mail.gmail.com> <347F12AB-EA05-4090-A203-BDC011F02411@freeswitch.org> Message-ID: <200907141618.03295.yivzhenko@mksat.net> On Wednesday 08 July 2009 16:29:50 Brian West wrote: > http://wiki.freeswitch.org i not found any essential information about architecture :-((((( .... may be bad looking? > > /b > > On Jul 8, 2009, at 7:42 AM, Maxim Tsvetov wrote: > > Hi All > > > > Where can I get information about internal Freeswitch architecture: > > 1) how modules interoperates with each other (maybe using corba or > > com > > objects or something else) > > 2) how core interoperates with other modules > > 3) how javascript function is translated to internal commands. > > > > In addition if you cand send me some schemas of Freeswitch > > architecture > > that will be great. > > > > Regards, > > Maxim Tsvetov > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090714/8df8bc10/attachment-0002.html From brian at freeswitch.org Tue Jul 14 06:18:58 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 14 Jul 2009 08:18:58 -0500 Subject: [Freeswitch-users] Freeswitch architecture In-Reply-To: <200907141618.03295.yivzhenko@mksat.net> References: <89c9bbf80907080542y39f2b234tff9163ea34c0b968@mail.gmail.com> <347F12AB-EA05-4090-A203-BDC011F02411@freeswitch.org> <200907141618.03295.yivzhenko@mksat.net> Message-ID: <722CF8A4-9768-4F05-AAC9-3FE4305E6370@freeswitch.org> http://wiki.freeswitch.org/wiki/Main_Page#FreeSWITCH.E2.84.A2_Architecture /b On Jul 14, 2009, at 8:18 AM, Yuriy Ivzhenko wrote: > On Wednesday 08 July 2009 16:29:50 Brian West wrote: > > http://wiki.freeswitch.org > i not found any essential information about architecture :-((((( > .... may be bad looking? > > > > /b > > > > On Jul 8, 2009, at 7:42 AM, Maxim Tsvetov wrote: > > > Hi All > > > > > > Where can I get information about internal Freeswitch > architecture: > > > 1) how modules interoperates with each other (maybe using > corba or > > > com > > > objects or something else) > > > 2) how core interoperates with other modules > > > 3) how javascript function is translated to internal commands. > > > > > > In addition if you cand send me some schemas of Freeswitch > > > architecture > > > that will be great. > > > > > > Regards, > > > Maxim Tsvetov > > > > > > _______________________________________________ > > > Freeswitch-users mailing list > > > Freeswitch-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090714/031d19a8/attachment-0002.html From elihayun at gmail.com Tue Jul 14 06:26:42 2009 From: elihayun at gmail.com (Eli Hayun) Date: Tue, 14 Jul 2009 16:26:42 +0300 Subject: [Freeswitch-users] Freeswitch-users Digest, Vol 37, Issue 75 In-Reply-To: References: Message-ID: <4A5C8792.1030807@savion.huji.ac.il> Message: 5 Date: Tue, 14 Jul 2009 06:00:10 -0700 From: Michael S Collins Subject: Re: [Freeswitch-users] Get voicemail messages To: "freeswitch-users at lists.freeswitch.org" Message-ID: <4429558B-4DA5-43BF-9824-D5B407546CCB at freeswitch.org> Content-Type: text/plain; charset=us-ascii; format=flowed; delsp=yes On Jul 14, 2009, at 5:02 AM, Eli Hayun wrote: > > Hi > > I am not using fixed xml files for the extension registration. I have > > LUA script to return an XML string to FS. > > Everything goes fine until I am trying to get the voice messages. > > When am entering my id, FS (or voicemail module) try to get the xml > > for > > that id, but it cant find it. My lua script did NOT recieved any xml > > request at that point. > > What should I do to solve the problem. > > > > Thanks > > Eli Hayun > Can you pastebin the log? Be sure to press F8 to turn up the debug level. Here is the log ( with the XML string for the requested id (80670) freeswitch at tst-sip-srv> API CALL [console(loglevel 7)] output: +OK console log level set to DEBUG freeswitch at tst-sip-srv> 2009-07-14 16:05:15.860230 [DEBUG] sofia.c:4554 IP 132.64.4.238 Rejected by acl "domains". Falling back to Digest auth. 2009-07-14 16:05:16.88367 [DEBUG] sofia.c:4554 IP 132.64.4.238 Rejected by acl "domains". Falling back to Digest auth. ->
<-- 2009-07-14 16:05:16.108310 [NOTICE] switch_channel.c:602 New Channel sofia/internal/80670 at 132.64.3.86 [f9716554-7076-11de-9237-7d312efadfc4] 2009-07-14 16:05:16.112125 [DEBUG] sofia.c:3215 Channel sofia/internal/80670 at 132.64.3.86 entering state [received][100] 2009-07-14 16:05:16.112125 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/80670 at 132.64.3.86) Running State Change CS_NEW 2009-07-14 16:05:16.112125 [DEBUG] sofia.c:3222 Remote SDP: v=0 o=root 1385664886 1385664886 IN IP4 132.64.4.238 s=call c=IN IP4 132.64.4.238 t=0 0 m=audio 57682 RTP/AVP 0 8 9 99 3 18 4 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:99 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:18 g729/8000 a=rtpmap:4 g723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:both a=ptime:20 2009-07-14 16:05:16.112125 [DEBUG] sofia_glue.c:3070 Audio Codec Compare [pcmu:0:8000:20]/[G7221:115:32000:20] 2009-07-14 16:05:16.112125 [DEBUG] sofia_glue.c:3070 Audio Codec Compare [pcmu:0:8000:20]/[G7221:107:16000:20] 2009-07-14 16:05:16.112125 [DEBUG] sofia_glue.c:3070 Audio Codec Compare [pcmu:0:8000:20]/[G722:9:8000:20] 2009-07-14 16:05:16.112125 [DEBUG] sofia_glue.c:3070 Audio Codec Compare [pcmu:0:8000:20]/[PCMU:0:8000:20] 2009-07-14 16:05:16.112125 [DEBUG] sofia_glue.c:2028 Set Codec sofia/internal/80670 at 132.64.3.86 PCMU/8000 20 ms 160 samples 2009-07-14 16:05:16.112125 [DEBUG] sofia_glue.c:3030 Set 2833 dtmf payload to 101 2009-07-14 16:05:16.112125 [DEBUG] switch_core_state_machine.c:403 (sofia/internal/80670 at 132.64.3.86) State NEW 2009-07-14 16:05:16.112125 [DEBUG] sofia.c:3381 (sofia/internal/80670 at 132.64.3.86) State Change CS_NEW -> CS_INIT 2009-07-14 16:05:16.112125 [DEBUG] switch_core_session.c:933 Send signal sofia/internal/80670 at 132.64.3.86 [BREAK] 2009-07-14 16:05:16.112125 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/80670 at 132.64.3.86) Running State Change CS_INIT 2009-07-14 16:05:16.116244 [DEBUG] switch_core_state_machine.c:480 (sofia/internal/80670 at 132.64.3.86) State INIT 2009-07-14 16:05:16.116244 [DEBUG] mod_sofia.c:83 sofia/internal/80670 at 132.64.3.86 SOFIA INIT 2009-07-14 16:05:16.116244 [DEBUG] mod_sofia.c:111 (sofia/internal/80670 at 132.64.3.86) State Change CS_INIT -> CS_ROUTING 2009-07-14 16:05:16.116244 [DEBUG] switch_core_session.c:933 Send signal sofia/internal/80670 at 132.64.3.86 [BREAK] 2009-07-14 16:05:16.116244 [DEBUG] switch_core_state_machine.c:480 (sofia/internal/80670 at 132.64.3.86) State INIT going to sleep 2009-07-14 16:05:16.116244 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/80670 at 132.64.3.86) Running State Change CS_ROUTING 2009-07-14 16:05:16.116244 [DEBUG] switch_core_state_machine.c:483 (sofia/internal/80670 at 132.64.3.86) State ROUTING 2009-07-14 16:05:16.116244 [DEBUG] mod_sofia.c:130 sofia/internal/80670 at 132.64.3.86 SOFIA ROUTING 2009-07-14 16:05:16.116244 [DEBUG] switch_core_state_machine.c:78 sofia/internal/80670 at 132.64.3.86 Standard ROUTING 2009-07-14 16:05:16.116244 [INFO] mod_dialplan_xml.c:252 Processing phone-1->*98 in context default Dialplan: sofia/internal/80670 at 132.64.3.86 parsing [default->unloop] continue=false Dialplan: sofia/internal/80670 at 132.64.3.86 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/internal/80670 at 132.64.3.86 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: sofia/internal/80670 at 132.64.3.86 parsing [default->tod_example] continue=true Dialplan: sofia/internal/80670 at 132.64.3.86 Regex (PASS) [tod_example] ${strftime(%w)}(2) =~ /^([1-5])$/ break=on-false Dialplan: sofia/internal/80670 at 132.64.3.86 Regex (PASS) [tod_example] ${strftime(%H%M)}(1605) =~ /^((09|1[0-7])[0-5][0-9]|1800)$/ break=on-false Dialplan: sofia/internal/80670 at 132.64.3.86 Action set(open=true) Dialplan: sofia/internal/80670 at 132.64.3.86 parsing [default->global-intercept] continue=false Dialplan: sofia/internal/80670 at 132.64.3.86 Regex (FAIL) [global-intercept] destination_number(*98) =~ /^886$/ break=on-false Dialplan: sofia/internal/80670 at 132.64.3.86 parsing [default->group-intercept] continue=false Dialplan: sofia/internal/80670 at 132.64.3.86 Regex (FAIL) [group-intercept] destination_number(*98) =~ /^\*8$/ break=on-false Dialplan: sofia/internal/80670 at 132.64.3.86 parsing [default->intercept-ext] continue=false Dialplan: sofia/internal/80670 at 132.64.3.86 Regex (FAIL) [intercept-ext] destination_number(*98) =~ /^\*\*(\d+)$/ break=on-false Dialplan: sofia/internal/80670 at 132.64.3.86 parsing [default->redial] continue=false Dialplan: sofia/internal/80670 at 132.64.3.86 Regex (FAIL) [redial] destination_number(*98) =~ /^870$/ break=on-false Dialplan: sofia/internal/80670 at 132.64.3.86 parsing [default->global] continue=true Dialplan: sofia/internal/80670 at 132.64.3.86 Regex (FAIL) [global] ${call_debug}(false) =~ /^true$/ break=never Dialplan: sofia/internal/80670 at 132.64.3.86 Regex (FAIL) [global] ${sip_has_crypto}() =~ /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=never Dialplan: sofia/internal/80670 at 132.64.3.86 Absolute Condition [global] Dialplan: sofia/internal/80670 at 132.64.3.86 Action hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) Dialplan: sofia/internal/80670 at 132.64.3.86 Action hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}) Dialplan: sofia/internal/80670 at 132.64.3.86 Action hash(insert/${domain_name}-last_dial/global/${uuid}) Dialplan: sofia/internal/80670 at 132.64.3.86 parsing [default->snom-demo-2] continue=false Dialplan: sofia/internal/80670 at 132.64.3.86 Regex (FAIL) [snom-demo-2] destination_number(*98) =~ /^9001$/ break=on-false Dialplan: sofia/internal/80670 at 132.64.3.86 parsing [default->snom-demo-1] continue=false Dialplan: sofia/internal/80670 at 132.64.3.86 Regex (FAIL) [snom-demo-1] destination_number(*98) =~ /^9000$/ break=on-false Dialplan: sofia/internal/80670 at 132.64.3.86 parsing [default->eavesdrop] continue=false Dialplan: sofia/internal/80670 at 132.64.3.86 Regex (FAIL) [eavesdrop] destination_number(*98) =~ /^88(.*)$|^\*0(.*)$/ break=on-false Dialplan: sofia/internal/80670 at 132.64.3.86 parsing [default->eavesdrop] continue=false Dialplan: sofia/internal/80670 at 132.64.3.86 Regex (FAIL) [eavesdrop] destination_number(*98) =~ /^779$/ break=on-false Dialplan: sofia/internal/80670 at 132.64.3.86 parsing [default->call_return] continue=false Dialplan: sofia/internal/80670 at 132.64.3.86 Regex (FAIL) [call_return] destination_number(*98) =~ /^\*69$|^869$|^lcr$/ break=on-false Dialplan: sofia/internal/80670 at 132.64.3.86 parsing [default->del-group] continue=false Dialplan: sofia/internal/80670 at 132.64.3.86 Regex (FAIL) [del-group] destination_number(*98) =~ /^80(\d{2})$/ break=on-false Dialplan: sofia/internal/80670 at 132.64.3.86 parsing [default->add-group] continue=false Dialplan: sofia/internal/80670 at 132.64.3.86 Regex (FAIL) [add-group] destination_number(*98) =~ /^81(\d{2})$/ break=on-false Dialplan: sofia/internal/80670 at 132.64.3.86 parsing [default->call-group-simo] continue=false Dialplan: sofia/internal/80670 at 132.64.3.86 Regex (FAIL) [call-group-simo] destination_number(*98) =~ /^82(\d{2})$/ break=on-false Dialplan: sofia/internal/80670 at 132.64.3.86 parsing [default->call-group-order] continue=false Dialplan: sofia/internal/80670 at 132.64.3.86 Regex (FAIL) [call-group-order] destination_number(*98) =~ /^83(\d{2})$/ break=on-false Dialplan: sofia/internal/80670 at 132.64.3.86 parsing [default->extension-intercom] continue=false Dialplan: sofia/internal/80670 at 132.64.3.86 Regex (FAIL) [extension-intercom] destination_number(*98) =~ /^(8888)$/ break=on-false Dialplan: sofia/internal/80670 at 132.64.3.86 parsing [default->Local_Extension] continue=false Dialplan: sofia/internal/80670 at 132.64.3.86 Regex (FAIL) [Local_Extension] destination_number(*98) =~ /^(80\d{3})$/ break=on-false Dialplan: sofia/internal/80670 at 132.64.3.86 parsing [default->group_dial_sales] continue=false Dialplan: sofia/internal/80670 at 132.64.3.86 Regex (FAIL) [group_dial_sales] destination_number(*98) =~ /^2000$/ break=on-false Dialplan: sofia/internal/80670 at 132.64.3.86 parsing [default->group_dial_support] continue=false Dialplan: sofia/internal/80670 at 132.64.3.86 Regex (FAIL) [group_dial_support] destination_number(*98) =~ /^2001$/ break=on-false Dialplan: sofia/internal/80670 at 132.64.3.86 parsing [default->group_dial_billing] continue=false Dialplan: sofia/internal/80670 at 132.64.3.86 Regex (FAIL) [group_dial_billing] destination_number(*98) =~ /^2002$/ break=on-false Dialplan: sofia/internal/80670 at 132.64.3.86 parsing [default->operator] continue=false Dialplan: sofia/internal/80670 at 132.64.3.86 Regex (FAIL) [operator] destination_number(*98) =~ /^(operator|0)$/ break=on-false Dialplan: sofia/internal/80670 at 132.64.3.86 parsing [default->vmain] continue=false Dialplan: sofia/internal/80670 at 132.64.3.86 Regex (PASS) [vmain] destination_number(*98) =~ /^vmain$|^86111$|^\*98$/ break=on-false Dialplan: sofia/internal/80670 at 132.64.3.86 Action answer() Dialplan: sofia/internal/80670 at 132.64.3.86 Action sleep(1000) Dialplan: sofia/internal/80670 at 132.64.3.86 Action voicemail(check default ${domain_name} ${caller_id_number}) 2009-07-14 16:05:16.120891 [DEBUG] switch_core_state_machine.c:114 (sofia/internal/80670 at 132.64.3.86) State Change CS_ROUTING -> CS_EXECUTE 2009-07-14 16:05:16.120891 [DEBUG] switch_core_session.c:933 Send signal sofia/internal/80670 at 132.64.3.86 [BREAK] 2009-07-14 16:05:16.120891 [DEBUG] switch_core_state_machine.c:483 (sofia/internal/80670 at 132.64.3.86) State ROUTING going to sleep 2009-07-14 16:05:16.120891 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/80670 at 132.64.3.86) Running State Change CS_EXECUTE 2009-07-14 16:05:16.120891 [DEBUG] switch_core_state_machine.c:490 (sofia/internal/80670 at 132.64.3.86) State EXECUTE 2009-07-14 16:05:16.120891 [DEBUG] mod_sofia.c:173 sofia/internal/80670 at 132.64.3.86 SOFIA EXECUTE 2009-07-14 16:05:16.120891 [DEBUG] switch_core_state_machine.c:151 sofia/internal/80670 at 132.64.3.86 Standard EXECUTE EXECUTE sofia/internal/80670 at 132.64.3.86 set(open=true) 2009-07-14 16:05:16.120891 [DEBUG] mod_dptools.c:748 sofia/internal/80670 at 132.64.3.86 SET [open]=[true] EXECUTE sofia/internal/80670 at 132.64.3.86 hash(insert/132.64.3.86-spymap/80670/f9716554-7076-11de-9237-7d312efadfc4) EXECUTE sofia/internal/80670 at 132.64.3.86 hash(insert/132.64.3.86-last_dial/80670/*98) EXECUTE sofia/internal/80670 at 132.64.3.86 hash(insert/132.64.3.86-last_dial/global/f9716554-7076-11de-9237-7d312efadfc4) EXECUTE sofia/internal/80670 at 132.64.3.86 answer() 2009-07-14 16:05:16.124221 [DEBUG] mod_dptools.c:649 sofia/internal/80670 at 132.64.3.86 receive message [ANSWER] 2009-07-14 16:05:16.124221 [DEBUG] sofia_glue.c:2262 AUDIO RTP [sofia/internal/80670 at 132.64.3.86] 132.64.3.86 port 31372 -> 132.64.4.238 port 57682 codec: 0 ms: 20 2009-07-14 16:05:16.124221 [DEBUG] switch_rtp.c:1138 Starting timer [soft] 160 bytes per 20ms 2009-07-14 16:05:16.138142 [DEBUG] mod_sofia.c:549 Local SDP sofia/internal/80670 at 132.64.3.86: v=0 o=FreeSWITCH 1247545344 1247545345 IN IP4 132.64.3.86 s=FreeSWITCH c=IN IP4 132.64.3.86 t=0 0 m=audio 31372 RTP/AVP 0 101 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 2009-07-14 16:05:16.138142 [DEBUG] switch_core_session.c:630 Send signal sofia/internal/80670 at 132.64.3.86 [BREAK] 2009-07-14 16:05:16.138142 [NOTICE] mod_dptools.c:649 Channel [sofia/internal/80670 at 132.64.3.86] has been answered 2009-07-14 16:05:16.138142 [DEBUG] switch_channel.c:182 sofia/internal/80670 at 132.64.3.86 receive message [AUDIO_SYNC] EXECUTE sofia/internal/80670 at 132.64.3.86 sleep(1000) 2009-07-14 16:05:16.138142 [DEBUG] switch_channel.c:182 sofia/internal/80670 at 132.64.3.86 receive message [AUDIO_SYNC] 2009-07-14 16:05:16.138142 [DEBUG] sofia.c:3215 Channel sofia/internal/80670 at 132.64.3.86 entering state [completed][200] 2009-07-14 16:05:16.216623 [DEBUG] sofia.c:3215 Channel sofia/internal/80670 at 132.64.3.86 entering state [ready][200] EXECUTE sofia/internal/80670 at 132.64.3.86 voicemail(check default 132.64.3.86 80670) 2009-07-14 16:05:17.148904 [DEBUG] mod_voicemail.c:776 [default] rwlock 2009-07-14 16:05:17.148904 [DEBUG] switch_ivr_play_say.c:118 No language specified - Using [en] 2009-07-14 16:05:17.152174 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[voicemail/vm-hello.wav] (en:en) 2009-07-14 16:05:17.152174 [DEBUG] switch_ivr_play_say.c:1097 Codec Activated L16 at 8000hz 1 channels 20ms 2009-07-14 16:05:17.152174 [DEBUG] switch_core_io.c:649 sofia/internal/80670 at 132.64.3.86 receive message [TRANSCODING_NECESSARY] 2009-07-14 16:05:18.368781 [DEBUG] switch_ivr_play_say.c:1391 done playing file 2009-07-14 16:05:18.488598 [DEBUG] switch_channel.c:182 sofia/internal/80670 at 132.64.3.86 receive message [AUDIO_SYNC] 2009-07-14 16:05:18.608382 [DEBUG] switch_channel.c:182 sofia/internal/80670 at 132.64.3.86 receive message [AUDIO_SYNC] 2009-07-14 16:05:18.708720 [WARNING] mod_voicemail.c:2072 Can't find user [80670 at 132.64.3.86] 2009-07-14 16:05:18.708720 [DEBUG] switch_ivr_play_say.c:118 No language specified - Using [en] 2009-07-14 16:05:18.712630 [DEBUG] switch_ivr_play_say.c:273 Handle play-file:[voicemail/vm-goodbye.wav] (en:en) 2009-07-14 16:05:18.712630 [DEBUG] switch_ivr_play_say.c:1097 Codec Activated L16 at 8000hz 1 channels 20ms 2009-07-14 16:05:18.712630 [DEBUG] switch_core_io.c:649 sofia/internal/80670 at 132.64.3.86 receive message [TRANSCODING_NECESSARY] 2009-07-14 16:05:19.188996 [DEBUG] switch_ivr_play_say.c:1391 done playing file 2009-07-14 16:05:19.308974 [NOTICE] switch_core_state_machine.c:179 Hangup sofia/internal/80670 at 132.64.3.86 [CS_EXECUTE] [NORMAL_CLEARING] 2009-07-14 16:05:19.308974 [DEBUG] switch_channel.c:1683 Send signal sofia/internal/80670 at 132.64.3.86 [KILL] 2009-07-14 16:05:19.308974 [DEBUG] switch_core_session.c:933 Send signal sofia/internal/80670 at 132.64.3.86 [BREAK] 2009-07-14 16:05:19.308974 [DEBUG] switch_core_state_machine.c:490 (sofia/internal/80670 at 132.64.3.86) State EXECUTE going to sleep 2009-07-14 16:05:19.308974 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/80670 at 132.64.3.86) Running State Change CS_HANGUP 2009-07-14 16:05:19.308974 [DEBUG] switch_core_state_machine.c:433 (sofia/internal/80670 at 132.64.3.86) State HANGUP 2009-07-14 16:05:19.308974 [DEBUG] mod_sofia.c:338 Channel sofia/internal/80670 at 132.64.3.86 hanging up, cause: NORMAL_CLEARING 2009-07-14 16:05:19.308974 [DEBUG] mod_sofia.c:393 Sending BYE to sofia/internal/80670 at 132.64.3.86 2009-07-14 16:05:19.312163 [DEBUG] switch_core_state_machine.c:46 sofia/internal/80670 at 132.64.3.86 Standard HANGUP, cause: NORMAL_CLEARING 2009-07-14 16:05:19.312163 [DEBUG] switch_core_state_machine.c:433 (sofia/internal/80670 at 132.64.3.86) State HANGUP going to sleep 2009-07-14 16:05:19.312163 [DEBUG] switch_core_state_machine.c:475 (sofia/internal/80670 at 132.64.3.86) State Change CS_HANGUP -> CS_REPORTING 2009-07-14 16:05:19.312163 [DEBUG] switch_core_session.c:933 Send signal sofia/internal/80670 at 132.64.3.86 [BREAK] 2009-07-14 16:05:19.312163 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/80670 at 132.64.3.86) Running State Change CS_REPORTING 2009-07-14 16:05:19.312163 [DEBUG] switch_core_state_machine.c:607 (sofia/internal/80670 at 132.64.3.86) State REPORTING 2009-07-14 16:05:19.312163 [DEBUG] switch_core_state_machine.c:53 sofia/internal/80670 at 132.64.3.86 Standard REPORTING, cause: NORMAL_CLEARING 2009-07-14 16:05:19.312163 [DEBUG] switch_core_state_machine.c:607 (sofia/internal/80670 at 132.64.3.86) State REPORTING going to sleep 2009-07-14 16:05:19.312163 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/80670 at 132.64.3.86) State Change CS_REPORTING -> CS_DESTROY 2009-07-14 16:05:19.312163 [DEBUG] switch_core_session.c:1067 Session 7 (sofia/internal/80670 at 132.64.3.86) Locked, Waiting on external entities 2009-07-14 16:05:19.312163 [NOTICE] switch_core_session.c:1085 Session 7 (sofia/internal/80670 at 132.64.3.86) Ended 2009-07-14 16:05:19.312163 [NOTICE] switch_core_session.c:1087 Close Channel sofia/internal/80670 at 132.64.3.86 [CS_DESTROY] 2009-07-14 16:05:19.312163 [DEBUG] switch_core_state_machine.c:559 (sofia/internal/80670 at 132.64.3.86) State DESTROY 2009-07-14 16:05:19.312163 [DEBUG] mod_sofia.c:255 sofia/internal/80670 at 132.64.3.86 SOFIA DESTROY 2009-07-14 16:05:19.312163 [DEBUG] switch_core_state_machine.c:60 sofia/internal/80670 at 132.64.3.86 Standard DESTROY 2009-07-14 16:05:19.312163 [DEBUG] switch_core_state_machine.c:559 (sofia/internal/80670 at 132.64.3.86) State DESTROY going to sleep - Thanks Eli Hayun From dftoro at yahoo.com Tue Jul 14 06:32:53 2009 From: dftoro at yahoo.com (Diego Toro) Date: Tue, 14 Jul 2009 06:32:53 -0700 (PDT) Subject: [Freeswitch-users] Dialogic cards Message-ID: <999132.13204.qm@web33505.mail.mud.yahoo.com> hi,?I think that may be... ? Analog cards:?D/41JCT-LS, D/120JCT-LS? (jct serie) Digital cards: D/600JCT-1E1 and DMV?serie. ? Diego --- On Mon, 7/13/09, Michael Collins wrote: From: Michael Collins Subject: Re: [Freeswitch-users] Dialogic cards To: freeswitch-users at lists.freeswitch.org Date: Monday, July 13, 2009, 4:45 PM What are the recommended cards to be used with freeswitch? Sangoma cards and Zaptel/DAHDI compatible cards work well. -MC -----Inline Attachment Follows----- _______________________________________________ Freeswitch-users mailing list Freeswitch-users at 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/20090714/f875361f/attachment-0002.html From anthony.minessale at gmail.com Tue Jul 14 07:37:30 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 14 Jul 2009 09:37:30 -0500 Subject: [Freeswitch-users] Problems with Ping and re-registering brokengateways In-Reply-To: <78A03DDB5A5043C597807D0DFCF00DCA@D810> References: <0324DD608A074940AAC173B85A3978F2@D810> <191c3a030907130859y2665da03xf704f9db6070fb3c@mail.gmail.com> <78A03DDB5A5043C597807D0DFCF00DCA@D810> Message-ID: <191c3a030907140737s6c7cb091r91628802a38d6c3c@mail.gmail.com> yes this was plenty of information try r14242 On Tue, Jul 14, 2009 at 4:07 AM, Andy wrote: > Hi Anthony, > > Thanks for your reply. The trace of the ping request looks like this. Any > clues? > > send 674 bytes to udp/[77.240.48.94]:5060 at 09:07:25.479313: > ------------------------------------------------------------------------ > OPTIONS sip:voiptalk.org;transport=udp SIP/2.0 > Via: SIP/2.0/UDP 77.86.49.249;rport;branch=z9hG4bK8rBQ4a33Ny02K > Max-Forwards: 70 > From: ;tag=a8U21NNZ23tBB > To: > Call-ID: 96da25e1-eaf8-122c-02ae-0019d15ec7e1 > CSeq: 117662779 OPTIONS > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-13850 > 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 > > ------------------------------------------------------------------------ > recv 352 bytes from udp/[77.240.48.94]:5060 at 09:07:25.486124: > ------------------------------------------------------------------------ > SIP/2.0 484 Address Incomplete > Via: SIP/2.0/UDP 77.86.49.249;rport=5060;branch=z9hG4bK8rBQ4a33Ny02K > From: ;tag=a8U21NNZ23tBB > To: ;tag=fd79486175647ed1617969929fdaad02.f21c > Call-ID: 96da25e1-eaf8-122c-02ae-0019d15ec7e1 > CSeq: 117662779 OPTIONS > Server: OpenSIPS (1.5.1-notls (x86_64/linux)) > Content-Length: 0 > ------------------------------------------------------------------------ > 2009-07-14 10:07:25.483507 [WARNING] sofia.c:2810 Ping failed voiptalk.org > 2009-07-14 10:07:25.483507 [WARNING] sofia.c:2813 Unregister voiptalk.org > > Andy > > ------------------------------ > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony > Minessale > *Sent:* 13 July 2009 16:59 > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Problems with Ping and re-registering > brokengateways > > Are they ignoring the options packet we send them or are they maybe getting > lost behind NAT? > we send an OPTIONS and even if we get a error back we consider that a > successful reply. > > We did have a patch into SVN very recently to correct a problem with > OPTIONS ping in a NAT situation. > > Maybe try latest trunk first then capture the console log with sip traffic > in place if it still does not work so we can have a look. > > to capture the log use these 2 commands from the cli. > > sofia profile internal siptrace on > console loglevel debug > > > > > On Mon, Jul 13, 2009 at 7:59 AM, Andy wrote: > >> Hi, >> >> I'm fairly sure my problem lies with my voip provider VoipTalk but wonder >> if you could help me understand a couple of things. My config is very >> simple, I'm using freeswitch to accept incoming calls via a voip gateway and >> record messages. Here's the problem: >> >> - When freeswitch starts the gateways are all created and register >> correctly >> >> - I have the ping parameter set to make sure the gateway stays alive. >> >> - The first time freeswitch pings the gateway it fails even though the >> registration appears intact as calls are still coming through to freeswitch >> >> - Freeswitch then tries to re-register the gateway but this fails. The SIP >> trace shows an Unauthorized message and the actual log entry is >> 'Registration Failed with status Operation has no matching challenge [904]' >> >> - eventually the registration times out with my provider and all is lost. >> >> - if I call 'sofia profile external restart' or restart the software this >> fixes the problem >> >> My questions are: >> >> 1) Why would the ping fail when the registration appears to be intact? >> 2) Whay would the auto re-register not work but a restart would? >> >> This ones driving me nuts so any help greatly appreciated. >> >> regards >> Andy >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > 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/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference 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/20090714/fe645332/attachment-0002.html From anthony.minessale at gmail.com Tue Jul 14 08:04:32 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 14 Jul 2009 10:04:32 -0500 Subject: [Freeswitch-users] Get voicemail messages In-Reply-To: <4A5C73CA.40306@savion.huji.ac.il> References: <4A5C73CA.40306@savion.huji.ac.il> Message-ID: <191c3a030907140804y775562a3wca745dae4f67c1d5@mail.gmail.com> did you bind your lua script to directory lookups in addition to the dialplan? On Tue, Jul 14, 2009 at 7:02 AM, Eli Hayun wrote: > Hi > I am not using fixed xml files for the extension registration. I have > LUA script to return an XML string to FS. > Everything goes fine until I am trying to get the voice messages. > When am entering my id, FS (or voicemail module) try to get the xml for > that id, but it cant find it. My lua script did NOT recieved any xml > request at that point. > What should I do to solve the problem. > > Thanks > Eli Hayun > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference 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/20090714/6cc4a8d0/attachment-0002.html From klaus.teller at gmx.net Tue Jul 14 08:14:50 2009 From: klaus.teller at gmx.net (Klaus Teller) Date: Tue, 14 Jul 2009 17:14:50 +0200 Subject: [Freeswitch-users] Where is the country code? Message-ID: <20090714151450.79890@gmx.net> Hi, I'm playing with Freeswitch and Les.NET right now. It strikes me that the caller id as passed to javascript doesn't contain the country code. Anyone knows where teh issue lies? Thanks, Klaus. -- Jetzt kostenlos herunterladen: Internet Explorer 8 und Mozilla Firefox 3 - sicherer, schneller und einfacher! http://portal.gmx.net/de/go/chbrowser From pjintheusa at gmail.com Tue Jul 14 08:35:59 2009 From: pjintheusa at gmail.com (Phillip Jones) Date: Tue, 14 Jul 2009 11:35:59 -0400 Subject: [Freeswitch-users] leg_timeout Message-ID: <367751820907140835y322c576en1eb2ba931388d732@mail.gmail.com> Hi there, Here is my call flow: 1) leg A is bridged to leg B 2) when leg B is answered I play a confirm script - "please 1 to accept this call" I only want leg B to ring 20 seconds. BUT when the caller party answers, he should have as long as he needs to press 1. "leg_timeout" seems to be in play until the bridge is completed. I need it to reset when leg b is answered. I tried resetting the leg_timeout in the confirm script after leg b is answered. I also tried using leg_progress_timeout. Neither seemed to work. Any help or suggestions would be welcome. Phillip Jones -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090714/7be21ec1/attachment-0002.html From svetikvoip at gmail.com Tue Jul 14 08:39:15 2009 From: svetikvoip at gmail.com (Svetik VOIP) Date: Tue, 14 Jul 2009 11:39:15 -0400 Subject: [Freeswitch-users] How to pass Message Waiting Ind. from VOIP provider directly to the phone Message-ID: <94790b850907140839m431f80cdu74d37dae17c5f7b@mail.gmail.com> Hi Guys, I have a question about configuring Message Waiting Indicator (MWI). I am running Freeswitch on Ubuntu 8.04 Desktop. I have Freeswitch connected to the external VOIP provider (voip.ms) and to the internal SIP box (Linksys RTP300) with one phone hooked to it. I let my VOIP provider to handle voicemail, and I would like Freeswitch to passthrough Message Waiting Indicator to the SIP box (and eventially to the phone), so it turns on when I have a new message in the VOIP provider voicemail . How to configure Freeswitch to achieve this? Before using Freeswitch I had SIP box hooked to my external provider directly, and I was getting Message Waiting Indicator on the phone no problem, but since I put FreeSwitch, it does not work anymore. I suspect it is because Freeswitch has its own voicemail system and it triggers MWI based on its state. I want to keep things simple and keep my voicemail at my VOIP provider. Thank you, Svetik -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090714/9f6af00d/attachment-0002.html From larclap at yahoo.com Tue Jul 14 08:53:07 2009 From: larclap at yahoo.com (Lars Zeb) Date: Tue, 14 Jul 2009 08:53:07 -0700 Subject: [Freeswitch-users] Intercom error with SNOM Message-ID: <00a001ca049b$2e9e03b0$8bda0b10$@com> I am getting an error when I try to make an intercom call from a softphone to a SNOM 320. I get a "401 Unauthorized" in the siptrace and a "No Matching gateway found" in the log. I can successfully make an intercom call between my softphone and a Polycom 501, so it must be something with the SNOM. bkw suggested that the problem was in the challenge/response between FreeSWITCH and SNOM. In the SNOM's Setup/Advanced/Behavior page I have set "Challenge/Response" off, "Enable intercom" on and "Type of Intercom Answering" to "Handsfree". The error is still "401 Unauthorized". What more do I need to do? Thanks, Lars http://pastebin.freeswitch.org/9709 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090714/a77c8a3d/attachment-0002.html From msc at freeswitch.org Tue Jul 14 09:16:38 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 14 Jul 2009 09:16:38 -0700 Subject: [Freeswitch-users] leg_timeout In-Reply-To: <367751820907140835y322c576en1eb2ba931388d732@mail.gmail.com> References: <367751820907140835y322c576en1eb2ba931388d732@mail.gmail.com> Message-ID: <87f2f3b90907140916o3f008a3br26bfca075a9e3d2e@mail.gmail.com> On Tue, Jul 14, 2009 at 8:35 AM, Phillip Jones wrote: > Hi there, > > Here is my call flow: > > 1) leg A is bridged to leg B > 2) when leg B is answered I play a confirm script - "please 1 to accept > this call" > > I only want leg B to ring 20 seconds. BUT when the caller party answers, he > should have as long as he needs to press 1. > > "leg_timeout" seems to be in play until the bridge is completed. I need it > to reset when leg b is answered. > Correct. Both "leg_timeout" and "leg_progress_timeout" are for controlling how long to wait prior to the B-leg answering. (leg_progress_timeout specifies how long to wait for any kind of progress, be it early media of some sort, ringing, or an answer.) > > I tried resetting the leg_timeout in the confirm script after leg b is > answered. I also tried using leg_progress_timeout. Neither seemed to work. > What exactly are you trying to do? The two variables you've mentioned shouldn't have any effect on the call after it has been established. > > Any help or suggestions would be welcome. > Could you pastebin your dialplan and a debug log of a call that does not work? See this page for some handy tips on using pastebin and collecting information for debugging purposes: http://wiki.freeswitch.org/wiki/Reporting_Bugs#Debugging_Steps -MC > > Phillip Jones > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090714/67646679/attachment-0002.html From eweaver at meetingone.com Tue Jul 14 09:59:17 2009 From: eweaver at meetingone.com (Weaver, Eric) Date: Tue, 14 Jul 2009 09:59:17 -0700 Subject: [Freeswitch-users] Preventing disconnect on event_sockte close In-Reply-To: <87f2f3b90907131808l100afe7dwc95d0f624ff38d50@mail.gmail.com> References: <7DB369DA44D5CF42B6C23BD7D611A50FDA7E1D5FD6@VA3DIAXVS061.RED001.local> <87f2f3b90907131808l100afe7dwc95d0f624ff38d50@mail.gmail.com> Message-ID: <7DB369DA44D5CF42B6C23BD7D611A50FDA7E1D604D@VA3DIAXVS061.RED001.local> Tried it using FreeSWITCH Version 1.0.4pre9 (14036M) and the following steps Start netcat netcat -v -l -p 14000 place call, socket is connected via dial plan, enter the following. connect\n\n sendmsg call-command: execute execute-app-name: answer\n\n sendmsg call-command: execute execute-app-name: playback execute-app-arg: /home/eweaver/holdmusic.wav sendmsg call-command: execute execute-app-name: park Console window displays this message: 2009-07-14 10:42:29.245833 [ERR] switch_ivr.c:654 Cannot park channels that are under control already. at this point, ^C in the netcat window. Call is disconnected. Need to be able to park these calls so they can then be handled from an inbound event socket connection. eric From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Monday, July 13, 2009 7:08 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Preventing disconnect on event_sockte close I don't know if this will work for you but I just tested this scenario with uuid_park. After parking the call I disconnected the socket and the call continued. I did the same thing with uuid_transfer. After the transfer I disconnected the socket and the call continued. How are you handling the call and how is the socket getting disconnected? -MC On Mon, Jul 13, 2009 at 4:35 PM, Weaver, Eric > wrote: Using mod_event_socket in outbound mode, is there any to prevent a call from being disconnected when the outbound socket is closed ? I would like to handle the initial inbound call using outbound but after the disposition of the call is determined, close the socket and have that call managed using an inbound socket instead. Eric Weaver _______________________________________________ Freeswitch-users mailing list Freeswitch-users at 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/20090714/6dfdc4a1/attachment-0002.html From pjintheusa at gmail.com Tue Jul 14 10:02:37 2009 From: pjintheusa at gmail.com (Phillip Jones) Date: Tue, 14 Jul 2009 13:02:37 -0400 Subject: [Freeswitch-users] leg_timeout In-Reply-To: <87f2f3b90907140916o3f008a3br26bfca075a9e3d2e@mail.gmail.com> References: <367751820907140835y322c576en1eb2ba931388d732@mail.gmail.com> <87f2f3b90907140916o3f008a3br26bfca075a9e3d2e@mail.gmail.com> Message-ID: <367751820907141002g3a837a09meabdbc9743c3bee7@mail.gmail.com> Hi, Thanks for the reply. >> Both "leg_timeout" and "leg_progress_timeout" are for controlling how long to wait prior to the B-leg answering. I think this is my point. leg_timeout seems to control how long to wait prior to the bridge completeing, not the B-leg answering. In my situation I am using: Session.Execute("set", "group_confirm_key=exec"); Session.Execute("set", "group_confirm_file=javascript confirm.js"); my confirm.js prompts leg_b for 1 (to take the call). If leg_timeout is set to 10 you have 10 seconds to answer the call AND press 1. I just want call_timeout to be satisfied when the call is answered. Not when the called party presses 1 and the bridge is complete. I am new all this so I will work out how to use the pastebin etc. Thanks for your help. Phillip Jones On Tue, Jul 14, 2009 at 12:16 PM, Michael Collins wrote: > > > On Tue, Jul 14, 2009 at 8:35 AM, Phillip Jones wrote: > >> Hi there, >> >> Here is my call flow: >> >> 1) leg A is bridged to leg B >> 2) when leg B is answered I play a confirm script - "please 1 to accept >> this call" >> >> I only want leg B to ring 20 seconds. BUT when the caller party answers, >> he should have as long as he needs to press 1. >> >> "leg_timeout" seems to be in play until the bridge is completed. I need it >> to reset when leg b is answered. >> > > Correct. Both "leg_timeout" and "leg_progress_timeout" are for controlling > how long to wait prior to the B-leg answering. (leg_progress_timeout > specifies how long to wait for any kind of progress, be it early media of > some sort, ringing, or an answer.) > >> >> I tried resetting the leg_timeout in the confirm script after leg b is >> answered. I also tried using leg_progress_timeout. Neither seemed to work. >> > > What exactly are you trying to do? The two variables you've mentioned > shouldn't have any effect on the call after it has been established. > >> >> Any help or suggestions would be welcome. >> > > Could you pastebin your dialplan and a debug log of a call that does not > work? See this page for some handy tips on using pastebin and collecting > information for debugging purposes: > http://wiki.freeswitch.org/wiki/Reporting_Bugs#Debugging_Steps > > -MC > >> >> Phillip Jones >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090714/2a2904c6/attachment-0002.html From msc at freeswitch.org Tue Jul 14 10:09:39 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 14 Jul 2009 10:09:39 -0700 Subject: [Freeswitch-users] Intercom error with SNOM In-Reply-To: <00a001ca049b$2e9e03b0$8bda0b10$@com> References: <00a001ca049b$2e9e03b0$8bda0b10$@com> Message-ID: <87f2f3b90907141009j28b6da38g4e1403a131fdcf6d@mail.gmail.com> On Tue, Jul 14, 2009 at 8:53 AM, Lars Zeb wrote: > I am getting an error when I try to make an intercom call from a > softphone to a SNOM 320. I get a ?401 Unauthorized? in the siptrace and a > ?No Matching gateway found? in the log. I can successfully make an intercom > call between my softphone and a Polycom 501, so it must be something with > the SNOM. > > > > bkw suggested that the problem was in the challenge/response between > FreeSWITCH and SNOM. In the SNOM?s Setup/Advanced/Behavior page I have set > ?Challenge/Response? off, ?Enable intercom? on and ?Type of Intercom > Answering? to ?Handsfree?. The error is still ?401 Unauthorized?. > > > > What more do I need to do? > Lars, It looks like FreeSWITCH is sending the challenge to the Snom. Note this line from the debug log: 2009-07-14 08:12:32.333079 [DEBUG] sofia.c:4587 IP 192.168.10.11 Rejected by acl "domains". Falling back to Digest auth. For testing purposes you can edit acl.conf.xml and add a new line right after: Add: Restart FS or issue this command at the CLI: reloadacl reloadxml Then try your call again. To learn more about how you can have your local users bypass the "domains" acl without editing acl.conf.xml then look in conf/directory/default/brian.xml. At the top of that file you will see a note about how adding a cidr= attribute to your user tag will let you bypass the domains ACL check. Enjoy! -MC > > > Thanks, Lars > > > > http://pastebin.freeswitch.org/9709 > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090714/cb8b333a/attachment-0002.html From msc at freeswitch.org Tue Jul 14 10:11:42 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 14 Jul 2009 10:11:42 -0700 Subject: [Freeswitch-users] Preventing disconnect on event_sockte close In-Reply-To: <7DB369DA44D5CF42B6C23BD7D611A50FDA7E1D604D@VA3DIAXVS061.RED001.local> References: <7DB369DA44D5CF42B6C23BD7D611A50FDA7E1D5FD6@VA3DIAXVS061.RED001.local> <87f2f3b90907131808l100afe7dwc95d0f624ff38d50@mail.gmail.com> <7DB369DA44D5CF42B6C23BD7D611A50FDA7E1D604D@VA3DIAXVS061.RED001.local> Message-ID: <87f2f3b90907141011g46686396se27fa3cc0320963d@mail.gmail.com> On Tue, Jul 14, 2009 at 9:59 AM, Weaver, Eric wrote: > Tried it using FreeSWITCH Version 1.0.4pre9 (14036M) and the > following steps > > > > Start netcat > > > > netcat -v -l -p 14000 > > > > place call, socket is connected via dial plan, enter the following. > > > > connect\n\n > > > > sendmsg > > call-command: execute > > execute-app-name: answer\n\n > > > > > > sendmsg > > call-command: execute > > execute-app-name: playback > > execute-app-arg: /home/eweaver/holdmusic.wav > > > > > > > > > > sendmsg > > call-command: execute > > execute-app-name: park > > Try this: api uuid_park You'll need to capture the uuid at some point and store it. For testing I just manually copied and pasted it to/from the console screen. -MC > > > Console window displays this message: > > > > 2009-07-14 10:42:29.245833 [ERR] switch_ivr.c:654 Cannot park channels that > are under control already. > > > > at this point, ^C in the netcat window. Call is disconnected. > > > > > > Need to be able to park these calls so they can then be handled from an > inbound event socket connection. > > > > eric > > > > > > > > > > > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael > Collins > *Sent:* Monday, July 13, 2009 7:08 PM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Preventing disconnect on event_sockte > close > > > > I don't know if this will work for you but I just tested this scenario with > uuid_park. After parking the call I disconnected the socket and the call > continued. I did the same thing with uuid_transfer. After the transfer I > disconnected the socket and the call continued. > > How are you handling the call and how is the socket getting disconnected? > > -MC > > On Mon, Jul 13, 2009 at 4:35 PM, Weaver, Eric > wrote: > > Using mod_event_socket in outbound mode, is there any to prevent a call > from being disconnected when the outbound socket is closed ? I would like to > handle the initial inbound call using outbound but after the disposition of > the call is determined, close the socket and have that call managed using an > inbound socket instead. > > > > Eric Weaver > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090714/a11cca5c/attachment-0002.html From anthony.minessale at gmail.com Tue Jul 14 10:20:18 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 14 Jul 2009 12:20:18 -0500 Subject: [Freeswitch-users] leg_timeout In-Reply-To: <367751820907141002g3a837a09meabdbc9743c3bee7@mail.gmail.com> References: <367751820907140835y322c576en1eb2ba931388d732@mail.gmail.com> <87f2f3b90907140916o3f008a3br26bfca075a9e3d2e@mail.gmail.com> <367751820907141002g3a837a09meabdbc9743c3bee7@mail.gmail.com> Message-ID: <191c3a030907141020l48e8cb2cm6106ff4442ee1bed@mail.gmail.com> update to trunk and try setting group_confirm_cancel_timeout=true let me know if it works On Tue, Jul 14, 2009 at 12:02 PM, Phillip Jones wrote: > Hi, > > Thanks for the reply. > > >> Both "leg_timeout" and "leg_progress_timeout" are for controlling how > long to wait prior to the B-leg answering. > > I think this is my point. leg_timeout seems to control how long to wait > prior to the bridge completeing, not the B-leg answering. > > In my situation I am using: > > Session.Execute("set", "group_confirm_key=exec"); > Session.Execute("set", "group_confirm_file=javascript confirm.js"); > > my confirm.js prompts leg_b for 1 (to take the call). If leg_timeout is set > to 10 you have 10 seconds to answer the call AND press 1. > > I just want call_timeout to be satisfied when the call is answered. Not > when the called party presses 1 and the bridge is complete. > > I am new all this so I will work out how to use the pastebin etc. > > Thanks for your help. > > > Phillip Jones > > > > On Tue, Jul 14, 2009 at 12:16 PM, Michael Collins wrote: > >> >> >> On Tue, Jul 14, 2009 at 8:35 AM, Phillip Jones wrote: >> >>> Hi there, >>> >>> Here is my call flow: >>> >>> 1) leg A is bridged to leg B >>> 2) when leg B is answered I play a confirm script - "please 1 to accept >>> this call" >>> >>> I only want leg B to ring 20 seconds. BUT when the caller party answers, >>> he should have as long as he needs to press 1. >>> >>> "leg_timeout" seems to be in play until the bridge is completed. I need >>> it to reset when leg b is answered. >>> >> >> Correct. Both "leg_timeout" and "leg_progress_timeout" are for controlling >> how long to wait prior to the B-leg answering. (leg_progress_timeout >> specifies how long to wait for any kind of progress, be it early media of >> some sort, ringing, or an answer.) >> >>> >>> I tried resetting the leg_timeout in the confirm script after leg b is >>> answered. I also tried using leg_progress_timeout. Neither seemed to work. >>> >> >> What exactly are you trying to do? The two variables you've mentioned >> shouldn't have any effect on the call after it has been established. >> >>> >>> Any help or suggestions would be welcome. >>> >> >> Could you pastebin your dialplan and a debug log of a call that does not >> work? See this page for some handy tips on using pastebin and collecting >> information for debugging purposes: >> http://wiki.freeswitch.org/wiki/Reporting_Bugs#Debugging_Steps >> >> -MC >> >>> >>> Phillip Jones >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference 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/20090714/7c8a0205/attachment-0002.html From anthony.minessale at gmail.com Tue Jul 14 10:24:37 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 14 Jul 2009 12:24:37 -0500 Subject: [Freeswitch-users] Preventing disconnect on event_sockte close In-Reply-To: <87f2f3b90907141011g46686396se27fa3cc0320963d@mail.gmail.com> References: <7DB369DA44D5CF42B6C23BD7D611A50FDA7E1D5FD6@VA3DIAXVS061.RED001.local> <87f2f3b90907131808l100afe7dwc95d0f624ff38d50@mail.gmail.com> <7DB369DA44D5CF42B6C23BD7D611A50FDA7E1D604D@VA3DIAXVS061.RED001.local> <87f2f3b90907141011g46686396se27fa3cc0320963d@mail.gmail.com> Message-ID: <191c3a030907141024u35c0509aqdc7112d82dba18cb@mail.gmail.com> or you can also do api uuid_transfer park inline or sendmsg call-command: execute execute-app-name: transfer execute-app-arg: park inline all of these will pull the call out of the control of your socket and into the care of the core. On Tue, Jul 14, 2009 at 12:11 PM, Michael Collins wrote: > > > On Tue, Jul 14, 2009 at 9:59 AM, Weaver, Eric wrote: > >> Tried it using FreeSWITCH Version 1.0.4pre9 (14036M) and the >> following steps >> >> >> >> Start netcat >> >> >> >> netcat -v -l -p 14000 >> >> >> >> place call, socket is connected via dial plan, enter the following. >> >> >> >> connect\n\n >> >> >> >> sendmsg >> >> call-command: execute >> >> execute-app-name: answer\n\n >> >> >> >> >> >> sendmsg >> >> call-command: execute >> >> execute-app-name: playback >> >> execute-app-arg: /home/eweaver/holdmusic.wav >> >> >> >> >> >> >> >> >> >> sendmsg >> >> call-command: execute >> >> execute-app-name: park >> >> > Try this: > api uuid_park > > You'll need to capture the uuid at some point and store it. For testing I > just manually copied and pasted it to/from the console screen. > -MC > >> >> >> Console window displays this message: >> >> >> >> 2009-07-14 10:42:29.245833 [ERR] switch_ivr.c:654 Cannot park channels >> that are under control already. >> >> >> >> at this point, ^C in the netcat window. Call is disconnected. >> >> >> >> >> >> Need to be able to park these calls so they can then be handled from an >> inbound event socket connection. >> >> >> >> eric >> >> >> >> >> >> >> >> >> >> >> >> >> >> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael >> Collins >> *Sent:* Monday, July 13, 2009 7:08 PM >> *To:* freeswitch-users at lists.freeswitch.org >> *Subject:* Re: [Freeswitch-users] Preventing disconnect on event_sockte >> close >> >> >> >> I don't know if this will work for you but I just tested this scenario >> with uuid_park. After parking the call I disconnected the socket and the >> call continued. I did the same thing with uuid_transfer. After the transfer >> I disconnected the socket and the call continued. >> >> How are you handling the call and how is the socket getting disconnected? >> >> -MC >> >> On Mon, Jul 13, 2009 at 4:35 PM, Weaver, Eric >> wrote: >> >> Using mod_event_socket in outbound mode, is there any to prevent a call >> from being disconnected when the outbound socket is closed ? I would like to >> handle the initial inbound call using outbound but after the disposition of >> the call is determined, close the socket and have that call managed using an >> inbound socket instead. >> >> >> >> Eric Weaver >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference 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/20090714/81383a23/attachment-0002.html From msc at freeswitch.org Tue Jul 14 10:37:46 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 14 Jul 2009 10:37:46 -0700 Subject: [Freeswitch-users] leg_timeout In-Reply-To: <191c3a030907141020l48e8cb2cm6106ff4442ee1bed@mail.gmail.com> References: <367751820907140835y322c576en1eb2ba931388d732@mail.gmail.com> <87f2f3b90907140916o3f008a3br26bfca075a9e3d2e@mail.gmail.com> <367751820907141002g3a837a09meabdbc9743c3bee7@mail.gmail.com> <191c3a030907141020l48e8cb2cm6106ff4442ee1bed@mail.gmail.com> Message-ID: <87f2f3b90907141037r260001d4i2c41dcc0e4c25c5@mail.gmail.com> FYI, This has been added to the wiki: http://wiki.freeswitch.org/wiki/Channel_Variables#group_confirm_cancel_timeout -MC On Tue, Jul 14, 2009 at 10:20 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > update to trunk and try setting > group_confirm_cancel_timeout=true > > let me know if it works > > > On Tue, Jul 14, 2009 at 12:02 PM, Phillip Jones wrote: > >> Hi, >> >> Thanks for the reply. >> >> >> Both "leg_timeout" and "leg_progress_timeout" are for controlling how >> long to wait prior to the B-leg answering. >> >> I think this is my point. leg_timeout seems to control how long to wait >> prior to the bridge completeing, not the B-leg answering. >> >> In my situation I am using: >> >> Session.Execute("set", "group_confirm_key=exec"); >> Session.Execute("set", "group_confirm_file=javascript confirm.js"); >> >> my confirm.js prompts leg_b for 1 (to take the call). If leg_timeout is >> set to 10 you have 10 seconds to answer the call AND press 1. >> >> I just want call_timeout to be satisfied when the call is answered. Not >> when the called party presses 1 and the bridge is complete. >> >> I am new all this so I will work out how to use the pastebin etc. >> >> Thanks for your help. >> >> >> Phillip Jones >> >> >> >> On Tue, Jul 14, 2009 at 12:16 PM, Michael Collins wrote: >> >>> >>> >>> On Tue, Jul 14, 2009 at 8:35 AM, Phillip Jones wrote: >>> >>>> Hi there, >>>> >>>> Here is my call flow: >>>> >>>> 1) leg A is bridged to leg B >>>> 2) when leg B is answered I play a confirm script - "please 1 to accept >>>> this call" >>>> >>>> I only want leg B to ring 20 seconds. BUT when the caller party answers, >>>> he should have as long as he needs to press 1. >>>> >>>> "leg_timeout" seems to be in play until the bridge is completed. I need >>>> it to reset when leg b is answered. >>>> >>> >>> Correct. Both "leg_timeout" and "leg_progress_timeout" are for >>> controlling how long to wait prior to the B-leg answering. >>> (leg_progress_timeout specifies how long to wait for any kind of progress, >>> be it early media of some sort, ringing, or an answer.) >>> >>>> >>>> I tried resetting the leg_timeout in the confirm script after leg b is >>>> answered. I also tried using leg_progress_timeout. Neither seemed to work. >>>> >>> >>> What exactly are you trying to do? The two variables you've mentioned >>> shouldn't have any effect on the call after it has been established. >>> >>>> >>>> Any help or suggestions would be welcome. >>>> >>> >>> Could you pastebin your dialplan and a debug log of a call that does not >>> work? See this page for some handy tips on using pastebin and collecting >>> information for debugging purposes: >>> http://wiki.freeswitch.org/wiki/Reporting_Bugs#Debugging_Steps >>> >>> -MC >>> >>>> >>>> Phillip Jones >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > 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/20090714/e06c16b6/attachment-0002.html From larclap at yahoo.com Tue Jul 14 10:46:58 2009 From: larclap at yahoo.com (Lars Zeb) Date: Tue, 14 Jul 2009 10:46:58 -0700 Subject: [Freeswitch-users] Intercom error with SNOM In-Reply-To: <87f2f3b90907141009j28b6da38g4e1403a131fdcf6d@mail.gmail.com> References: <00a001ca049b$2e9e03b0$8bda0b10$@com> <87f2f3b90907141009j28b6da38g4e1403a131fdcf6d@mail.gmail.com> Message-ID: <00e301ca04ab$16337430$429a5c90$@com> Michael, I made the changes you suggested, but the result is the same. If it matters, I am at 14229. I stopped and restarted FreeSWITCH and then reran the intercom call. Lars 2009-07-14 10:41:40.553145 [DEBUG] sofia.c:4587 IP 192.168.10.11 Rejected by acl "domains". Falling back to Digest auth. 2009-07-14 10:41:40.562147 [DEBUG] sofia.c:4587 IP 192.168.10.11 Rejected by acl "domains". Falling back to Digest auth. 2009-07-14 10:41:40.565166 [NOTICE] switch_channel.c:602 New Channel sofia/internal/1009 at 192.168.10.29 [fac3f284-3710-4492-a9e7-667c7e816ea0] From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Tuesday, July 14, 2009 10:10 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Intercom error with SNOM On Tue, Jul 14, 2009 at 8:53 AM, Lars Zeb wrote: I am getting an error when I try to make an intercom call from a softphone to a SNOM 320. I get a "401 Unauthorized" in the siptrace and a "No Matching gateway found" in the log. I can successfully make an intercom call between my softphone and a Polycom 501, so it must be something with the SNOM. bkw suggested that the problem was in the challenge/response between FreeSWITCH and SNOM. In the SNOM's Setup/Advanced/Behavior page I have set "Challenge/Response" off, "Enable intercom" on and "Type of Intercom Answering" to "Handsfree". The error is still "401 Unauthorized". What more do I need to do? Lars, It looks like FreeSWITCH is sending the challenge to the Snom. Note this line from the debug log: 2009-07-14 08:12:32.333079 [DEBUG] sofia.c:4587 IP 192.168.10.11 Rejected by acl "domains". Falling back to Digest auth. For testing purposes you can edit acl.conf.xml and add a new line right after: Add: Restart FS or issue this command at the CLI: reloadacl reloadxml Then try your call again. To learn more about how you can have your local users bypass the "domains" acl without editing acl.conf.xml then look in conf/directory/default/brian.xml. At the top of that file you will see a note about how adding a cidr= attribute to your user tag will let you bypass the domains ACL check. Enjoy! -MC Thanks, Lars http://pastebin.freeswitch.org/9709 _______________________________________________ Freeswitch-users mailing list Freeswitch-users at 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/20090714/0864b1ae/attachment-0002.html From msc at freeswitch.org Tue Jul 14 10:58:09 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 14 Jul 2009 10:58:09 -0700 Subject: [Freeswitch-users] Intercom error with SNOM In-Reply-To: <00e301ca04ab$16337430$429a5c90$@com> References: <00a001ca049b$2e9e03b0$8bda0b10$@com> <87f2f3b90907141009j28b6da38g4e1403a131fdcf6d@mail.gmail.com> <00e301ca04ab$16337430$429a5c90$@com> Message-ID: <87f2f3b90907141058v4cf6d8bds62873fb0bb61553c@mail.gmail.com> Lars, Brian pointed out that the challenge is coming from the phone. Is 192.168.10.104 the Snom? -MC On Tue, Jul 14, 2009 at 10:46 AM, Lars Zeb wrote: > Michael, > > > > I made the changes you suggested, but the result is the same. If it > matters, I am at 14229. I stopped and restarted FreeSWITCH and then reran > the intercom call. > > > > Lars > > > > > > > > > > > > > > > > > > > > > > 2009-07-14 10:41:40.553145 [DEBUG] sofia.c:4587 IP 192.168.10.11 Rejected > by acl "domains". Falling back to Digest auth. > > 2009-07-14 10:41:40.562147 [DEBUG] sofia.c:4587 IP 192.168.10.11 Rejected > by acl "domains". Falling back to Digest auth. > > 2009-07-14 10:41:40.565166 [NOTICE] switch_channel.c:602 New Channel > sofia/internal/1009 at 192.168.10.29 [fac3f284-3710-4492-a9e7-667c7e816ea0] > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael > Collins > *Sent:* Tuesday, July 14, 2009 10:10 AM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Intercom error with SNOM > > > > > > On Tue, Jul 14, 2009 at 8:53 AM, Lars Zeb wrote: > > I am getting an error when I try to make an intercom call from a softphone > to a SNOM 320. I get a ?401 Unauthorized? in the siptrace and a ?No Matching > gateway found? in the log. I can successfully make an intercom call between > my softphone and a Polycom 501, so it must be something with the SNOM. > > > > bkw suggested that the problem was in the challenge/response between > FreeSWITCH and SNOM. In the SNOM?s Setup/Advanced/Behavior page I have set > ?Challenge/Response? off, ?Enable intercom? on and ?Type of Intercom > Answering? to ?Handsfree?. The error is still ?401 Unauthorized?. > > > > What more do I need to do? > > > Lars, > It looks like FreeSWITCH is sending the challenge to the Snom. Note this > line from the debug log: > 2009-07-14 08:12:32.333079 [DEBUG] sofia.c:4587 IP 192.168.10.11 Rejected > by acl "domains". Falling back to Digest auth. > > For testing purposes you can edit acl.conf.xml and add a new line right > after: > > > Add: > > > Restart FS or issue this command at the CLI: > reloadacl reloadxml > > Then try your call again. > > To learn more about how you can have your local users bypass the "domains" > acl without editing acl.conf.xml then look in > conf/directory/default/brian.xml. At the top of that file you will see a > note about how adding a cidr= attribute to your user tag will let you bypass > the domains ACL check. > > Enjoy! > -MC > > > > Thanks, Lars > > > > http://pastebin.freeswitch.org/9709 > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090714/d1049faf/attachment-0002.html From peder at networkoblivion.com Tue Jul 14 11:05:37 2009 From: peder at networkoblivion.com (Peder) Date: Tue, 14 Jul 2009 13:05:37 -0500 Subject: [Freeswitch-users] Intercom error with SNOM In-Reply-To: <00e301ca04ab$16337430$429a5c90$@com> References: <00a001ca049b$2e9e03b0$8bda0b10$@com> <87f2f3b90907141009j28b6da38g4e1403a131fdcf6d@mail.gmail.com> <00e301ca04ab$16337430$429a5c90$@com> Message-ID: <06b101ca04ad$b0fc2b90$12f482b0$@com> I haven't followed the whole thread, but the acl listed is named "lan" and the rejected acl is named "domains". From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Lars Zeb Sent: Tuesday, July 14, 2009 12:47 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Intercom error with SNOM Michael, I made the changes you suggested, but the result is the same. If it matters, I am at 14229. I stopped and restarted FreeSWITCH and then reran the intercom call. Lars 2009-07-14 10:41:40.553145 [DEBUG] sofia.c:4587 IP 192.168.10.11 Rejected by acl "domains". Falling back to Digest auth. 2009-07-14 10:41:40.562147 [DEBUG] sofia.c:4587 IP 192.168.10.11 Rejected by acl "domains". Falling back to Digest auth. 2009-07-14 10:41:40.565166 [NOTICE] switch_channel.c:602 New Channel sofia/internal/1009 at 192.168.10.29 [fac3f284-3710-4492-a9e7-667c7e816ea0] From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Tuesday, July 14, 2009 10:10 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Intercom error with SNOM On Tue, Jul 14, 2009 at 8:53 AM, Lars Zeb wrote: I am getting an error when I try to make an intercom call from a softphone to a SNOM 320. I get a "401 Unauthorized" in the siptrace and a "No Matching gateway found" in the log. I can successfully make an intercom call between my softphone and a Polycom 501, so it must be something with the SNOM. bkw suggested that the problem was in the challenge/response between FreeSWITCH and SNOM. In the SNOM's Setup/Advanced/Behavior page I have set "Challenge/Response" off, "Enable intercom" on and "Type of Intercom Answering" to "Handsfree". The error is still "401 Unauthorized". What more do I need to do? Lars, It looks like FreeSWITCH is sending the challenge to the Snom. Note this line from the debug log: 2009-07-14 08:12:32.333079 [DEBUG] sofia.c:4587 IP 192.168.10.11 Rejected by acl "domains". Falling back to Digest auth. For testing purposes you can edit acl.conf.xml and add a new line right after: Add: Restart FS or issue this command at the CLI: reloadacl reloadxml Then try your call again. To learn more about how you can have your local users bypass the "domains" acl without editing acl.conf.xml then look in conf/directory/default/brian.xml. At the top of that file you will see a note about how adding a cidr= attribute to your user tag will let you bypass the domains ACL check. Enjoy! -MC Thanks, Lars http://pastebin.freeswitch.org/9709 _______________________________________________ Freeswitch-users mailing list Freeswitch-users at 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/20090714/3fd8f8e2/attachment-0002.html From lon at kickasspixels.com Tue Jul 14 11:10:54 2009 From: lon at kickasspixels.com (Lon Baker) Date: Tue, 14 Jul 2009 11:10:54 -0700 Subject: [Freeswitch-users] Even socket packets/chunks Message-ID: <5d3e0dc60907141110u4f4084ecyb8fdee94c5983a2d@mail.gmail.com> Hi, You confirm if FS ever sends partial or incomplete commands over the event socket? I have heard from other developers that the commands I am listening for may come across incomplete. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090714/cdc45973/attachment-0002.html From pjintheusa at gmail.com Tue Jul 14 11:14:19 2009 From: pjintheusa at gmail.com (Phillip Jones) Date: Tue, 14 Jul 2009 14:14:19 -0400 Subject: [Freeswitch-users] leg_timeout In-Reply-To: <87f2f3b90907141037r260001d4i2c41dcc0e4c25c5@mail.gmail.com> References: <367751820907140835y322c576en1eb2ba931388d732@mail.gmail.com> <87f2f3b90907140916o3f008a3br26bfca075a9e3d2e@mail.gmail.com> <367751820907141002g3a837a09meabdbc9743c3bee7@mail.gmail.com> <191c3a030907141020l48e8cb2cm6106ff4442ee1bed@mail.gmail.com> <87f2f3b90907141037r260001d4i2c41dcc0e4c25c5@mail.gmail.com> Message-ID: <367751820907141114w56ae9adey4836238740a1ed43@mail.gmail.com> Thanks for your response. That does not seem to work. Here is my code: if(Session.Ready()) { Session.Execute("set", "ignore_early_media=true"); Session.Execute("set", "hangup_after_bridge=true"); Session.Execute("set", "ringback=${us-ring}"); Session.Answer(); string Caller_ID_Number = this.Session.GetVariable("caller_id_number"); Session.Execute("set", "group_confirm_key=exec"); *Session.Execute("set", "group_confirm_cancel_timeout=true"); * Session.Execute("set", "group_confirm_file=javascript confirm.js"); Session.Execute("bridge", "[leg_timeout=20,leg_confirm=y]sofia/gateway/broadvox/6095553828"); } Session.Hangup("USER_BUSY"); I also tried *group_confirm_cancel_leg_timeout* just in case. Am I missing something? On Tue, Jul 14, 2009 at 1:37 PM, Michael Collins wrote: > FYI, > This has been added to the wiki: > > http://wiki.freeswitch.org/wiki/Channel_Variables#group_confirm_cancel_timeout > > -MC > > > On Tue, Jul 14, 2009 at 10:20 AM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> update to trunk and try setting >> group_confirm_cancel_timeout=true >> >> let me know if it works >> >> >> On Tue, Jul 14, 2009 at 12:02 PM, Phillip Jones wrote: >> >>> Hi, >>> >>> Thanks for the reply. >>> >>> >> Both "leg_timeout" and "leg_progress_timeout" are for controlling how >>> long to wait prior to the B-leg answering. >>> >>> I think this is my point. leg_timeout seems to control how long to wait >>> prior to the bridge completeing, not the B-leg answering. >>> >>> In my situation I am using: >>> >>> Session.Execute("set", "group_confirm_key=exec"); >>> Session.Execute("set", "group_confirm_file=javascript confirm.js"); >>> >>> my confirm.js prompts leg_b for 1 (to take the call). If leg_timeout is >>> set to 10 you have 10 seconds to answer the call AND press 1. >>> >>> I just want call_timeout to be satisfied when the call is answered. Not >>> when the called party presses 1 and the bridge is complete. >>> >>> I am new all this so I will work out how to use the pastebin etc. >>> >>> Thanks for your help. >>> >>> >>> Phillip Jones >>> >>> >>> >>> On Tue, Jul 14, 2009 at 12:16 PM, Michael Collins wrote: >>> >>>> >>>> >>>> On Tue, Jul 14, 2009 at 8:35 AM, Phillip Jones wrote: >>>> >>>>> Hi there, >>>>> >>>>> Here is my call flow: >>>>> >>>>> 1) leg A is bridged to leg B >>>>> 2) when leg B is answered I play a confirm script - "please 1 to accept >>>>> this call" >>>>> >>>>> I only want leg B to ring 20 seconds. BUT when the caller party >>>>> answers, he should have as long as he needs to press 1. >>>>> >>>>> "leg_timeout" seems to be in play until the bridge is completed. I need >>>>> it to reset when leg b is answered. >>>>> >>>> >>>> Correct. Both "leg_timeout" and "leg_progress_timeout" are for >>>> controlling how long to wait prior to the B-leg answering. >>>> (leg_progress_timeout specifies how long to wait for any kind of progress, >>>> be it early media of some sort, ringing, or an answer.) >>>> >>>>> >>>>> I tried resetting the leg_timeout in the confirm script after leg b is >>>>> answered. I also tried using leg_progress_timeout. Neither seemed to work. >>>>> >>>> >>>> What exactly are you trying to do? The two variables you've mentioned >>>> shouldn't have any effect on the call after it has been established. >>>> >>>>> >>>>> Any help or suggestions would be welcome. >>>>> >>>> >>>> Could you pastebin your dialplan and a debug log of a call that does not >>>> work? See this page for some handy tips on using pastebin and collecting >>>> information for debugging purposes: >>>> http://wiki.freeswitch.org/wiki/Reporting_Bugs#Debugging_Steps >>>> >>>> -MC >>>> >>>>> >>>>> Phillip Jones >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> 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/20090714/8dab1cb2/attachment-0002.html From msc at freeswitch.org Tue Jul 14 11:18:14 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 14 Jul 2009 11:18:14 -0700 Subject: [Freeswitch-users] leg_timeout In-Reply-To: <367751820907141114w56ae9adey4836238740a1ed43@mail.gmail.com> References: <367751820907140835y322c576en1eb2ba931388d732@mail.gmail.com> <87f2f3b90907140916o3f008a3br26bfca075a9e3d2e@mail.gmail.com> <367751820907141002g3a837a09meabdbc9743c3bee7@mail.gmail.com> <191c3a030907141020l48e8cb2cm6106ff4442ee1bed@mail.gmail.com> <87f2f3b90907141037r260001d4i2c41dcc0e4c25c5@mail.gmail.com> <367751820907141114w56ae9adey4836238740a1ed43@mail.gmail.com> Message-ID: <87f2f3b90907141118j55ec9960xaa0c329e499e1cdf@mail.gmail.com> Can you capture a debug output and put it in pastebin? That will help us track it down. -MC On Tue, Jul 14, 2009 at 11:14 AM, Phillip Jones wrote: > Thanks for your response. > > That does not seem to work. Here is my code: > > if(Session.Ready()) > { > Session.Execute("set", "ignore_early_media=true"); > Session.Execute("set", "hangup_after_bridge=true"); > Session.Execute("set", "ringback=${us-ring}"); > > Session.Answer(); > string Caller_ID_Number = this.Session.GetVariable("caller_id_number"); > Session.Execute("set", "group_confirm_key=exec"); > *Session.Execute("set", "group_confirm_cancel_timeout=true"); > * Session.Execute("set", "group_confirm_file=javascript confirm.js"); > Session.Execute("bridge", > "[leg_timeout=20,leg_confirm=y]sofia/gateway/broadvox/6095553828"); > } > Session.Hangup("USER_BUSY"); > > I also tried *group_confirm_cancel_leg_timeout* just in case. > > Am I missing something? > > > > > On Tue, Jul 14, 2009 at 1:37 PM, Michael Collins wrote: > >> FYI, >> This has been added to the wiki: >> >> http://wiki.freeswitch.org/wiki/Channel_Variables#group_confirm_cancel_timeout >> >> -MC >> >> >> On Tue, Jul 14, 2009 at 10:20 AM, Anthony Minessale < >> anthony.minessale at gmail.com> wrote: >> >>> update to trunk and try setting >>> group_confirm_cancel_timeout=true >>> >>> let me know if it works >>> >>> >>> On Tue, Jul 14, 2009 at 12:02 PM, Phillip Jones wrote: >>> >>>> Hi, >>>> >>>> Thanks for the reply. >>>> >>>> >> Both "leg_timeout" and "leg_progress_timeout" are for controlling how >>>> long to wait prior to the B-leg answering. >>>> >>>> I think this is my point. leg_timeout seems to control how long to wait >>>> prior to the bridge completeing, not the B-leg answering. >>>> >>>> In my situation I am using: >>>> >>>> Session.Execute("set", "group_confirm_key=exec"); >>>> Session.Execute("set", "group_confirm_file=javascript confirm.js"); >>>> >>>> my confirm.js prompts leg_b for 1 (to take the call). If leg_timeout is >>>> set to 10 you have 10 seconds to answer the call AND press 1. >>>> >>>> I just want call_timeout to be satisfied when the call is answered. Not >>>> when the called party presses 1 and the bridge is complete. >>>> >>>> I am new all this so I will work out how to use the pastebin etc. >>>> >>>> Thanks for your help. >>>> >>>> >>>> Phillip Jones >>>> >>>> >>>> >>>> On Tue, Jul 14, 2009 at 12:16 PM, Michael Collins wrote: >>>> >>>>> >>>>> >>>>> On Tue, Jul 14, 2009 at 8:35 AM, Phillip Jones wrote: >>>>> >>>>>> Hi there, >>>>>> >>>>>> Here is my call flow: >>>>>> >>>>>> 1) leg A is bridged to leg B >>>>>> 2) when leg B is answered I play a confirm script - "please 1 to >>>>>> accept this call" >>>>>> >>>>>> I only want leg B to ring 20 seconds. BUT when the caller party >>>>>> answers, he should have as long as he needs to press 1. >>>>>> >>>>>> "leg_timeout" seems to be in play until the bridge is completed. I >>>>>> need it to reset when leg b is answered. >>>>>> >>>>> >>>>> Correct. Both "leg_timeout" and "leg_progress_timeout" are for >>>>> controlling how long to wait prior to the B-leg answering. >>>>> (leg_progress_timeout specifies how long to wait for any kind of progress, >>>>> be it early media of some sort, ringing, or an answer.) >>>>> >>>>>> >>>>>> I tried resetting the leg_timeout in the confirm script after leg b is >>>>>> answered. I also tried using leg_progress_timeout. Neither seemed to work. >>>>>> >>>>> >>>>> What exactly are you trying to do? The two variables you've mentioned >>>>> shouldn't have any effect on the call after it has been established. >>>>> >>>>>> >>>>>> Any help or suggestions would be welcome. >>>>>> >>>>> >>>>> Could you pastebin your dialplan and a debug log of a call that does >>>>> not work? See this page for some handy tips on using pastebin and collecting >>>>> information for debugging purposes: >>>>> http://wiki.freeswitch.org/wiki/Reporting_Bugs#Debugging_Steps >>>>> >>>>> -MC >>>>> >>>>>> >>>>>> Phillip Jones >>>>>> >>>>>> _______________________________________________ >>>>>> Freeswitch-users mailing list >>>>>> Freeswitch-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> Anthony Minessale II >>> >>> FreeSWITCH http://www.freeswitch.org/ >>> ClueCon http://www.cluecon.com/ >>> Twitter: http://twitter.com/FreeSWITCH_wire >>> >>> AIM: anthm >>> MSN:anthony_minessale at hotmail.com >>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>> IRC: irc.freenode.net #freeswitch >>> >>> FreeSWITCH Developer Conference >>> 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090714/01847f87/attachment-0002.html From anthony.minessale at gmail.com Tue Jul 14 11:27:20 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 14 Jul 2009 13:27:20 -0500 Subject: [Freeswitch-users] leg_timeout In-Reply-To: <367751820907141114w56ae9adey4836238740a1ed43@mail.gmail.com> References: <367751820907140835y322c576en1eb2ba931388d732@mail.gmail.com> <87f2f3b90907140916o3f008a3br26bfca075a9e3d2e@mail.gmail.com> <367751820907141002g3a837a09meabdbc9743c3bee7@mail.gmail.com> <191c3a030907141020l48e8cb2cm6106ff4442ee1bed@mail.gmail.com> <87f2f3b90907141037r260001d4i2c41dcc0e4c25c5@mail.gmail.com> <367751820907141114w56ae9adey4836238740a1ed43@mail.gmail.com> Message-ID: <191c3a030907141127x255f21c5q7d80cf26ea8451a9@mail.gmail.com> You actually updated your code and recompiled it all too? This param was added about 30 seconds before I sent you the email. On Tue, Jul 14, 2009 at 1:14 PM, Phillip Jones wrote: > Thanks for your response. > > That does not seem to work. Here is my code: > > if(Session.Ready()) > { > Session.Execute("set", "ignore_early_media=true"); > Session.Execute("set", "hangup_after_bridge=true"); > Session.Execute("set", "ringback=${us-ring}"); > > Session.Answer(); > string Caller_ID_Number = this.Session.GetVariable("caller_id_number"); > Session.Execute("set", "group_confirm_key=exec"); > *Session.Execute("set", "group_confirm_cancel_timeout=true"); > * Session.Execute("set", "group_confirm_file=javascript confirm.js"); > Session.Execute("bridge", > "[leg_timeout=20,leg_confirm=y]sofia/gateway/broadvox/6095553828"); > } > Session.Hangup("USER_BUSY"); > > I also tried *group_confirm_cancel_leg_timeout* just in case. > > Am I missing something? > > > > > On Tue, Jul 14, 2009 at 1:37 PM, Michael Collins wrote: > >> FYI, >> This has been added to the wiki: >> >> http://wiki.freeswitch.org/wiki/Channel_Variables#group_confirm_cancel_timeout >> >> -MC >> >> >> On Tue, Jul 14, 2009 at 10:20 AM, Anthony Minessale < >> anthony.minessale at gmail.com> wrote: >> >>> update to trunk and try setting >>> group_confirm_cancel_timeout=true >>> >>> let me know if it works >>> >>> >>> On Tue, Jul 14, 2009 at 12:02 PM, Phillip Jones wrote: >>> >>>> Hi, >>>> >>>> Thanks for the reply. >>>> >>>> >> Both "leg_timeout" and "leg_progress_timeout" are for controlling how >>>> long to wait prior to the B-leg answering. >>>> >>>> I think this is my point. leg_timeout seems to control how long to wait >>>> prior to the bridge completeing, not the B-leg answering. >>>> >>>> In my situation I am using: >>>> >>>> Session.Execute("set", "group_confirm_key=exec"); >>>> Session.Execute("set", "group_confirm_file=javascript confirm.js"); >>>> >>>> my confirm.js prompts leg_b for 1 (to take the call). If leg_timeout is >>>> set to 10 you have 10 seconds to answer the call AND press 1. >>>> >>>> I just want call_timeout to be satisfied when the call is answered. Not >>>> when the called party presses 1 and the bridge is complete. >>>> >>>> I am new all this so I will work out how to use the pastebin etc. >>>> >>>> Thanks for your help. >>>> >>>> >>>> Phillip Jones >>>> >>>> >>>> >>>> On Tue, Jul 14, 2009 at 12:16 PM, Michael Collins wrote: >>>> >>>>> >>>>> >>>>> On Tue, Jul 14, 2009 at 8:35 AM, Phillip Jones wrote: >>>>> >>>>>> Hi there, >>>>>> >>>>>> Here is my call flow: >>>>>> >>>>>> 1) leg A is bridged to leg B >>>>>> 2) when leg B is answered I play a confirm script - "please 1 to >>>>>> accept this call" >>>>>> >>>>>> I only want leg B to ring 20 seconds. BUT when the caller party >>>>>> answers, he should have as long as he needs to press 1. >>>>>> >>>>>> "leg_timeout" seems to be in play until the bridge is completed. I >>>>>> need it to reset when leg b is answered. >>>>>> >>>>> >>>>> Correct. Both "leg_timeout" and "leg_progress_timeout" are for >>>>> controlling how long to wait prior to the B-leg answering. >>>>> (leg_progress_timeout specifies how long to wait for any kind of progress, >>>>> be it early media of some sort, ringing, or an answer.) >>>>> >>>>>> >>>>>> I tried resetting the leg_timeout in the confirm script after leg b is >>>>>> answered. I also tried using leg_progress_timeout. Neither seemed to work. >>>>>> >>>>> >>>>> What exactly are you trying to do? The two variables you've mentioned >>>>> shouldn't have any effect on the call after it has been established. >>>>> >>>>>> >>>>>> Any help or suggestions would be welcome. >>>>>> >>>>> >>>>> Could you pastebin your dialplan and a debug log of a call that does >>>>> not work? See this page for some handy tips on using pastebin and collecting >>>>> information for debugging purposes: >>>>> http://wiki.freeswitch.org/wiki/Reporting_Bugs#Debugging_Steps >>>>> >>>>> -MC >>>>> >>>>>> >>>>>> Phillip Jones >>>>>> >>>>>> _______________________________________________ >>>>>> Freeswitch-users mailing list >>>>>> Freeswitch-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> _______________________________________________ >>>> Freeswitch-users mailing list >>>> Freeswitch-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> Anthony Minessale II >>> >>> FreeSWITCH http://www.freeswitch.org/ >>> ClueCon http://www.cluecon.com/ >>> Twitter: http://twitter.com/FreeSWITCH_wire >>> >>> AIM: anthm >>> MSN:anthony_minessale at hotmail.com >>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>> IRC: irc.freenode.net #freeswitch >>> >>> FreeSWITCH Developer Conference >>> 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 > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference 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/20090714/88ef3ad1/attachment-0002.html From dave at 3c.co.uk Tue Jul 14 11:33:56 2009 From: dave at 3c.co.uk (David Knell) Date: Tue, 14 Jul 2009 15:33:56 -0300 Subject: [Freeswitch-users] Even socket packets/chunks In-Reply-To: <5d3e0dc60907141110u4f4084ecyb8fdee94c5983a2d@mail.gmail.com> References: <5d3e0dc60907141110u4f4084ecyb8fdee94c5983a2d@mail.gmail.com> Message-ID: <1247596436.4254.6.camel@dk-d820> Hi - You can't assume that 1 packet=1 command/event - it's true often enough to lull you in to a false sense of security, but false often enough that you'll end up with odd problems unless you do things properly. In any case, it's not hard to get it right - there's plenty of other instances where applications have to read from a socket until they hit \n\n, then possibly read content-length bytes: HTTP for a starter. --Dave > Hi, > > > You confirm if FS ever sends partial or incomplete commands over the > event socket? I have heard from other developers that the commands I > am listening for may come across incomplete. > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- David Knell, Director, 3C Limited T: +44 20 3298 2000 E: dave at 3c.co.uk W: http://www.3c.co.uk From pjintheusa at gmail.com Tue Jul 14 11:35:05 2009 From: pjintheusa at gmail.com (Phillip Jones) Date: Tue, 14 Jul 2009 14:35:05 -0400 Subject: [Freeswitch-users] leg_timeout In-Reply-To: <191c3a030907141127x255f21c5q7d80cf26ea8451a9@mail.gmail.com> References: <367751820907140835y322c576en1eb2ba931388d732@mail.gmail.com> <87f2f3b90907140916o3f008a3br26bfca075a9e3d2e@mail.gmail.com> <367751820907141002g3a837a09meabdbc9743c3bee7@mail.gmail.com> <191c3a030907141020l48e8cb2cm6106ff4442ee1bed@mail.gmail.com> <87f2f3b90907141037r260001d4i2c41dcc0e4c25c5@mail.gmail.com> <367751820907141114w56ae9adey4836238740a1ed43@mail.gmail.com> <191c3a030907141127x255f21c5q7d80cf26ea8451a9@mail.gmail.com> Message-ID: <367751820907141135t2c374338y54226c9721cfa739@mail.gmail.com> Ah - SVN Trunk - thought you meant DID trunk!!! My bad. Sorry - understand now! Will recompile and let you know. On Tue, Jul 14, 2009 at 2:27 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > You actually updated your code and recompiled it all too? > This param was added about 30 seconds before I sent you the email. > > > > On Tue, Jul 14, 2009 at 1:14 PM, Phillip Jones wrote: > >> Thanks for your response. >> >> That does not seem to work. Here is my code: >> >> if(Session.Ready()) >> { >> Session.Execute("set", "ignore_early_media=true"); >> Session.Execute("set", "hangup_after_bridge=true"); >> Session.Execute("set", "ringback=${us-ring}"); >> >> Session.Answer(); >> string Caller_ID_Number = >> this.Session.GetVariable("caller_id_number"); >> Session.Execute("set", "group_confirm_key=exec"); >> *Session.Execute("set", "group_confirm_cancel_timeout=true"); >> * Session.Execute("set", "group_confirm_file=javascript confirm.js"); >> Session.Execute("bridge", >> "[leg_timeout=20,leg_confirm=y]sofia/gateway/broadvox/6095553828"); >> } >> Session.Hangup("USER_BUSY"); >> >> I also tried *group_confirm_cancel_leg_timeout* just in case. >> >> Am I missing something? >> >> >> >> >> On Tue, Jul 14, 2009 at 1:37 PM, Michael Collins wrote: >> >>> FYI, >>> This has been added to the wiki: >>> >>> http://wiki.freeswitch.org/wiki/Channel_Variables#group_confirm_cancel_timeout >>> >>> -MC >>> >>> >>> On Tue, Jul 14, 2009 at 10:20 AM, Anthony Minessale < >>> anthony.minessale at gmail.com> wrote: >>> >>>> update to trunk and try setting >>>> group_confirm_cancel_timeout=true >>>> >>>> let me know if it works >>>> >>>> >>>> On Tue, Jul 14, 2009 at 12:02 PM, Phillip Jones wrote: >>>> >>>>> Hi, >>>>> >>>>> Thanks for the reply. >>>>> >>>>> >> Both "leg_timeout" and "leg_progress_timeout" are for controlling >>>>> how long to wait prior to the B-leg answering. >>>>> >>>>> I think this is my point. leg_timeout seems to control how long to wait >>>>> prior to the bridge completeing, not the B-leg answering. >>>>> >>>>> In my situation I am using: >>>>> >>>>> Session.Execute("set", "group_confirm_key=exec"); >>>>> Session.Execute("set", "group_confirm_file=javascript confirm.js"); >>>>> >>>>> my confirm.js prompts leg_b for 1 (to take the call). If leg_timeout is >>>>> set to 10 you have 10 seconds to answer the call AND press 1. >>>>> >>>>> I just want call_timeout to be satisfied when the call is answered. Not >>>>> when the called party presses 1 and the bridge is complete. >>>>> >>>>> I am new all this so I will work out how to use the pastebin etc. >>>>> >>>>> Thanks for your help. >>>>> >>>>> >>>>> Phillip Jones >>>>> >>>>> >>>>> >>>>> On Tue, Jul 14, 2009 at 12:16 PM, Michael Collins wrote: >>>>> >>>>>> >>>>>> >>>>>> On Tue, Jul 14, 2009 at 8:35 AM, Phillip Jones >>>>> > wrote: >>>>>> >>>>>>> Hi there, >>>>>>> >>>>>>> Here is my call flow: >>>>>>> >>>>>>> 1) leg A is bridged to leg B >>>>>>> 2) when leg B is answered I play a confirm script - "please 1 to >>>>>>> accept this call" >>>>>>> >>>>>>> I only want leg B to ring 20 seconds. BUT when the caller party >>>>>>> answers, he should have as long as he needs to press 1. >>>>>>> >>>>>>> "leg_timeout" seems to be in play until the bridge is completed. I >>>>>>> need it to reset when leg b is answered. >>>>>>> >>>>>> >>>>>> Correct. Both "leg_timeout" and "leg_progress_timeout" are for >>>>>> controlling how long to wait prior to the B-leg answering. >>>>>> (leg_progress_timeout specifies how long to wait for any kind of progress, >>>>>> be it early media of some sort, ringing, or an answer.) >>>>>> >>>>>>> >>>>>>> I tried resetting the leg_timeout in the confirm script after leg b >>>>>>> is answered. I also tried using leg_progress_timeout. Neither seemed to >>>>>>> work. >>>>>>> >>>>>> >>>>>> What exactly are you trying to do? The two variables you've mentioned >>>>>> shouldn't have any effect on the call after it has been established. >>>>>> >>>>>>> >>>>>>> Any help or suggestions would be welcome. >>>>>>> >>>>>> >>>>>> Could you pastebin your dialplan and a debug log of a call that does >>>>>> not work? See this page for some handy tips on using pastebin and collecting >>>>>> information for debugging purposes: >>>>>> http://wiki.freeswitch.org/wiki/Reporting_Bugs#Debugging_Steps >>>>>> >>>>>> -MC >>>>>> >>>>>>> >>>>>>> Phillip Jones >>>>>>> >>>>>>> _______________________________________________ >>>>>>> Freeswitch-users mailing list >>>>>>> Freeswitch-users at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE: >>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>>> >>>>>> >>>>>> _______________________________________________ >>>>>> Freeswitch-users mailing list >>>>>> Freeswitch-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>> >>>>> _______________________________________________ >>>>> Freeswitch-users mailing list >>>>> Freeswitch-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> -- >>>> Anthony Minessale II >>>> >>>> FreeSWITCH http://www.freeswitch.org/ >>>> ClueCon http://www.cluecon.com/ >>>> Twitter: http://twitter.com/FreeSWITCH_wire >>>> >>>> AIM: anthm >>>> MSN:anthony_minessale at hotmail.com >>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>>> IRC: irc.freenode.net #freeswitch >>>> >>>> FreeSWITCH Developer Conference >>>> 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 >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > 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/20090714/38b4aa69/attachment-0002.html From anthony.minessale at gmail.com Tue Jul 14 11:56:29 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 14 Jul 2009 13:56:29 -0500 Subject: [Freeswitch-users] Even socket packets/chunks In-Reply-To: <1247596436.4254.6.camel@dk-d820> References: <5d3e0dc60907141110u4f4084ecyb8fdee94c5983a2d@mail.gmail.com> <1247596436.4254.6.camel@dk-d820> Message-ID: <191c3a030907141156y4614b28fy7feac44fe92b8532@mail.gmail.com> I took the time to write the ESL lib and release it BSD licensed. It does everything you need to talk to event socket from a series of languages from C all the way up... On Tue, Jul 14, 2009 at 1:33 PM, David Knell wrote: > Hi - > > You can't assume that 1 packet=1 command/event - it's true often enough > to lull you in to a false sense of security, but false often enough that > you'll end up with odd problems unless you do things properly. > > In any case, it's not hard to get it right - there's plenty of other > instances where applications have to read from a socket until they hit > \n\n, then possibly read content-length bytes: HTTP for a starter. > > --Dave > > > Hi, > > > > > > You confirm if FS ever sends partial or incomplete commands over the > > event socket? I have heard from other developers that the commands I > > am listening for may come across incomplete. > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > -- > David Knell, Director, 3C Limited > T: +44 20 3298 2000 > E: dave at 3c.co.uk > W: 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/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference 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/20090714/cca46e1f/attachment-0002.html From larclap at yahoo.com Tue Jul 14 12:34:10 2009 From: larclap at yahoo.com (Lars Zeb) Date: Tue, 14 Jul 2009 12:34:10 -0700 Subject: [Freeswitch-users] Intercom error with SNOM In-Reply-To: <87f2f3b90907141058v4cf6d8bds62873fb0bb61553c@mail.gmail.com> References: <00a001ca049b$2e9e03b0$8bda0b10$@com> <87f2f3b90907141009j28b6da38g4e1403a131fdcf6d@mail.gmail.com> <00e301ca04ab$16337430$429a5c90$@com> <87f2f3b90907141058v4cf6d8bds62873fb0bb61553c@mail.gmail.com> Message-ID: <014501ca04ba$0faa07f0$2efe17d0$@com> Yes. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Tuesday, July 14, 2009 10:58 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Intercom error with SNOM Lars, Brian pointed out that the challenge is coming from the phone. Is 192.168.10.104 the Snom? -MC On Tue, Jul 14, 2009 at 10:46 AM, Lars Zeb wrote: Michael, I made the changes you suggested, but the result is the same. If it matters, I am at 14229. I stopped and restarted FreeSWITCH and then reran the intercom call. Lars 2009-07-14 10:41:40.553145 [DEBUG] sofia.c:4587 IP 192.168.10.11 Rejected by acl "domains". Falling back to Digest auth. 2009-07-14 10:41:40.562147 [DEBUG] sofia.c:4587 IP 192.168.10.11 Rejected by acl "domains". Falling back to Digest auth. 2009-07-14 10:41:40.565166 [NOTICE] switch_channel.c:602 New Channel sofia/internal/1009 at 192.168.10.29 [fac3f284-3710-4492-a9e7-667c7e816ea0] From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Tuesday, July 14, 2009 10:10 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Intercom error with SNOM On Tue, Jul 14, 2009 at 8:53 AM, Lars Zeb wrote: I am getting an error when I try to make an intercom call from a softphone to a SNOM 320. I get a "401 Unauthorized" in the siptrace and a "No Matching gateway found" in the log. I can successfully make an intercom call between my softphone and a Polycom 501, so it must be something with the SNOM. bkw suggested that the problem was in the challenge/response between FreeSWITCH and SNOM. In the SNOM's Setup/Advanced/Behavior page I have set "Challenge/Response" off, "Enable intercom" on and "Type of Intercom Answering" to "Handsfree". The error is still "401 Unauthorized". What more do I need to do? Lars, It looks like FreeSWITCH is sending the challenge to the Snom. Note this line from the debug log: 2009-07-14 08:12:32.333079 [DEBUG] sofia.c:4587 IP 192.168.10.11 Rejected by acl "domains". Falling back to Digest auth. For testing purposes you can edit acl.conf.xml and add a new line right after: Add: Restart FS or issue this command at the CLI: reloadacl reloadxml Then try your call again. To learn more about how you can have your local users bypass the "domains" acl without editing acl.conf.xml then look in conf/directory/default/brian.xml. At the top of that file you will see a note about how adding a cidr= attribute to your user tag will let you bypass the domains ACL check. Enjoy! -MC Thanks, Lars http://pastebin.freeswitch.org/9709 _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at 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/20090714/26816e66/attachment-0002.html From andrew at hijacked.us Tue Jul 14 12:40:53 2009 From: andrew at hijacked.us (Andrew Thompson) Date: Tue, 14 Jul 2009 15:40:53 -0400 Subject: [Freeswitch-users] Intercom error with SNOM In-Reply-To: <00a001ca049b$2e9e03b0$8bda0b10$@com> References: <00a001ca049b$2e9e03b0$8bda0b10$@com> Message-ID: <20090714194053.GH28401@hijacked.us> On Tue, Jul 14, 2009 at 08:53:07AM -0700, Lars Zeb wrote: > I am getting an error when I try to make an intercom call from a softphone > to a SNOM 320. I get a "401 Unauthorized" in the siptrace and a "No Matching > gateway found" in the log. I can successfully make an intercom call between > my softphone and a Polycom 501, so it must be something with the SNOM. > > > > bkw suggested that the problem was in the challenge/response between > FreeSWITCH and SNOM. In the SNOM's Setup/Advanced/Behavior page I have set > "Challenge/Response" off, "Enable intercom" on and "Type of Intercom > Answering" to "Handsfree". The error is still "401 Unauthorized". > > > > What more do I need to do? > > Try disabling the intercom option and setting the 'answer after policy' to 'only in idle'. I think there's a bug in the snom firmware, see also: http://forum.snom.com/index.php?showtopic=1790&st=0&gopid=3688&#entry3688 Andrew From brian at freeswitch.org Tue Jul 14 12:41:21 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 14 Jul 2009 14:41:21 -0500 Subject: [Freeswitch-users] Intercom error with SNOM In-Reply-To: <014501ca04ba$0faa07f0$2efe17d0$@com> References: <00a001ca049b$2e9e03b0$8bda0b10$@com> <87f2f3b90907141009j28b6da38g4e1403a131fdcf6d@mail.gmail.com> <00e301ca04ab$16337430$429a5c90$@com> <87f2f3b90907141058v4cf6d8bds62873fb0bb61553c@mail.gmail.com> <014501ca04ba$0faa07f0$2efe17d0$@com> Message-ID: <81196D62-B915-4B03-A8E2-6E189FD7BC32@freeswitch.org> Make sure you 1. reboot. 2. make sure the setting is correct to not auth. 3. what firmware are you on? /b On Jul 14, 2009, at 2:34 PM, Lars Zeb wrote: > Yes. > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Michael Collins > Sent: Tuesday, July 14, 2009 10:58 AM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Intercom error with SNOM > > Lars, > > Brian pointed out that the challenge is coming from the phone. Is > 192.168.10.104 the Snom? > > -MC > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090714/a9a57b27/attachment-0002.html From larclap at yahoo.com Tue Jul 14 12:51:58 2009 From: larclap at yahoo.com (Lars Zeb) Date: Tue, 14 Jul 2009 12:51:58 -0700 Subject: [Freeswitch-users] Intercom error with SNOM In-Reply-To: <81196D62-B915-4B03-A8E2-6E189FD7BC32@freeswitch.org> References: <00a001ca049b$2e9e03b0$8bda0b10$@com> <87f2f3b90907141009j28b6da38g4e1403a131fdcf6d@mail.gmail.com> <00e301ca04ab$16337430$429a5c90$@com> <87f2f3b90907141058v4cf6d8bds62873fb0bb61553c@mail.gmail.com> <014501ca04ba$0faa07f0$2efe17d0$@com> <81196D62-B915-4B03-A8E2-6E189FD7BC32@freeswitch.org> Message-ID: <015e01ca04bc$8cab9be0$a602d3a0$@com> 2. Do you mean setting "Challenge Response on phone" on the SNOM? It is already set to no. 3. 7.3.14 From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Tuesday, July 14, 2009 12:41 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Intercom error with SNOM Make sure you 1. reboot. 2. make sure the setting is correct to not auth. 3. what firmware are you on? /b On Jul 14, 2009, at 2:34 PM, Lars Zeb wrote: Yes. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Tuesday, July 14, 2009 10:58 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Intercom error with SNOM Lars, Brian pointed out that the challenge is coming from the phone. Is 192.168.10.104 the Snom? -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090714/77fe412f/attachment-0002.html From brian at freeswitch.org Tue Jul 14 12:59:15 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 14 Jul 2009 14:59:15 -0500 Subject: [Freeswitch-users] Intercom error with SNOM In-Reply-To: <015e01ca04bc$8cab9be0$a602d3a0$@com> References: <00a001ca049b$2e9e03b0$8bda0b10$@com> <87f2f3b90907141009j28b6da38g4e1403a131fdcf6d@mail.gmail.com> <00e301ca04ab$16337430$429a5c90$@com> <87f2f3b90907141058v4cf6d8bds62873fb0bb61553c@mail.gmail.com> <014501ca04ba$0faa07f0$2efe17d0$@com> <81196D62-B915-4B03-A8E2-6E189FD7BC32@freeswitch.org> <015e01ca04bc$8cab9be0$a602d3a0$@com> Message-ID: Do what Andrew said.. it has to be a bug :P (in the snom) /b On Jul 14, 2009, at 2:51 PM, Lars Zeb wrote: > 2. Do you mean setting ?Challenge Response on phone? on the SNOM? It > is already set to no. > 3. 7.3.14 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090714/db8fc1c7/attachment-0002.html From larclap at yahoo.com Tue Jul 14 13:42:59 2009 From: larclap at yahoo.com (Lars Zeb) Date: Tue, 14 Jul 2009 13:42:59 -0700 Subject: [Freeswitch-users] Intercom error with SNOM In-Reply-To: References: <00a001ca049b$2e9e03b0$8bda0b10$@com> <87f2f3b90907141009j28b6da38g4e1403a131fdcf6d@mail.gmail.com> <00e301ca04ab$16337430$429a5c90$@com> <87f2f3b90907141058v4cf6d8bds62873fb0bb61553c@mail.gmail.com> <014501ca04ba$0faa07f0$2efe17d0$@com> <81196D62-B915-4B03-A8E2-6E189FD7BC32@freeswitch.org> <015e01ca04bc$8cab9be0$a602d3a0$@com> Message-ID: <017501ca04c3$ad21ae80$07650b80$@com> Thanks Michael and Brian and Andrew and Peder. I changed "Enable Intercom" to off and "'Answer After' Policy" to "only in idle" and it works. The next firmware will probably change the settings to mean what they should mean. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Tuesday, July 14, 2009 12:59 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Intercom error with SNOM Do what Andrew said.. it has to be a bug :P (in the snom) /b On Jul 14, 2009, at 2:51 PM, Lars Zeb wrote: 2. Do you mean setting "Challenge Response on phone" on the SNOM? It is already set to no. 3. 7.3.14 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090714/9dcfa426/attachment-0002.html From eweaver at meetingone.com Tue Jul 14 14:14:56 2009 From: eweaver at meetingone.com (Weaver, Eric) Date: Tue, 14 Jul 2009 14:14:56 -0700 Subject: [Freeswitch-users] Preventing disconnect on event_sockte close In-Reply-To: <87f2f3b90907141011g46686396se27fa3cc0320963d@mail.gmail.com> References: <7DB369DA44D5CF42B6C23BD7D611A50FDA7E1D5FD6@VA3DIAXVS061.RED001.local> <87f2f3b90907131808l100afe7dwc95d0f624ff38d50@mail.gmail.com> <7DB369DA44D5CF42B6C23BD7D611A50FDA7E1D604D@VA3DIAXVS061.RED001.local> <87f2f3b90907141011g46686396se27fa3cc0320963d@mail.gmail.com> Message-ID: <7DB369DA44D5CF42B6C23BD7D611A50FDABBFA760F@VA3DIAXVS061.RED001.local> Using api uuid_park 81b65478-70b9-11de-bf26-b962186102f7 before terminating the NC session works, the call is not disconnected. Once that is done, I do not receive DTMF and cannot play prompts to the caller, they seem to be in limbo. I can uuid_kill the call but I need to get dtmf and play prompts to them. Perhaps Park is not where I need to put these calls ? From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Tuesday, July 14, 2009 11:12 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Preventing disconnect on event_sockte close On Tue, Jul 14, 2009 at 9:59 AM, Weaver, Eric > wrote: Tried it using FreeSWITCH Version 1.0.4pre9 (14036M) and the following steps Start netcat netcat -v -l -p 14000 place call, socket is connected via dial plan, enter the following. connect\n\n sendmsg call-command: execute execute-app-name: answer\n\n sendmsg call-command: execute execute-app-name: playback execute-app-arg: /home/eweaver/holdmusic.wav sendmsg call-command: execute execute-app-name: park Try this: api uuid_park You'll need to capture the uuid at some point and store it. For testing I just manually copied and pasted it to/from the console screen. -MC Console window displays this message: 2009-07-14 10:42:29.245833 [ERR] switch_ivr.c:654 Cannot park channels that are under control already. at this point, ^C in the netcat window. Call is disconnected. Need to be able to park these calls so they can then be handled from an inbound event socket connection. eric From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Monday, July 13, 2009 7:08 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Preventing disconnect on event_sockte close I don't know if this will work for you but I just tested this scenario with uuid_park. After parking the call I disconnected the socket and the call continued. I did the same thing with uuid_transfer. After the transfer I disconnected the socket and the call continued. How are you handling the call and how is the socket getting disconnected? -MC On Mon, Jul 13, 2009 at 4:35 PM, Weaver, Eric > wrote: Using mod_event_socket in outbound mode, is there any to prevent a call from being disconnected when the outbound socket is closed ? I would like to handle the initial inbound call using outbound but after the disposition of the call is determined, close the socket and have that call managed using an inbound socket instead. Eric Weaver _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at 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/20090714/43cafd72/attachment-0002.html From eweaver at meetingone.com Tue Jul 14 14:17:18 2009 From: eweaver at meetingone.com (Weaver, Eric) Date: Tue, 14 Jul 2009 14:17:18 -0700 Subject: [Freeswitch-users] Preventing disconnect on event_sockte close In-Reply-To: <87f2f3b90907141011g46686396se27fa3cc0320963d@mail.gmail.com> References: <7DB369DA44D5CF42B6C23BD7D611A50FDA7E1D5FD6@VA3DIAXVS061.RED001.local> <87f2f3b90907131808l100afe7dwc95d0f624ff38d50@mail.gmail.com> <7DB369DA44D5CF42B6C23BD7D611A50FDA7E1D604D@VA3DIAXVS061.RED001.local> <87f2f3b90907141011g46686396se27fa3cc0320963d@mail.gmail.com> Message-ID: <7DB369DA44D5CF42B6C23BD7D611A50FDABBFA7611@VA3DIAXVS061.RED001.local> Tried uuid_displace to play prompt, got this message on console 2009-07-14 15:16:17.189764 [ERR] switch_ivr_async.c:367 Can not displace session. Media not enabled on channel So how can the call be "parked" and still have media ? From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Tuesday, July 14, 2009 11:12 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Preventing disconnect on event_sockte close On Tue, Jul 14, 2009 at 9:59 AM, Weaver, Eric > wrote: Tried it using FreeSWITCH Version 1.0.4pre9 (14036M) and the following steps Start netcat netcat -v -l -p 14000 place call, socket is connected via dial plan, enter the following. connect\n\n sendmsg call-command: execute execute-app-name: answer\n\n sendmsg call-command: execute execute-app-name: playback execute-app-arg: /home/eweaver/holdmusic.wav sendmsg call-command: execute execute-app-name: park Try this: api uuid_park You'll need to capture the uuid at some point and store it. For testing I just manually copied and pasted it to/from the console screen. -MC Console window displays this message: 2009-07-14 10:42:29.245833 [ERR] switch_ivr.c:654 Cannot park channels that are under control already. at this point, ^C in the netcat window. Call is disconnected. Need to be able to park these calls so they can then be handled from an inbound event socket connection. eric From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Monday, July 13, 2009 7:08 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Preventing disconnect on event_sockte close I don't know if this will work for you but I just tested this scenario with uuid_park. After parking the call I disconnected the socket and the call continued. I did the same thing with uuid_transfer. After the transfer I disconnected the socket and the call continued. How are you handling the call and how is the socket getting disconnected? -MC On Mon, Jul 13, 2009 at 4:35 PM, Weaver, Eric > wrote: Using mod_event_socket in outbound mode, is there any to prevent a call from being disconnected when the outbound socket is closed ? I would like to handle the initial inbound call using outbound but after the disposition of the call is determined, close the socket and have that call managed using an inbound socket instead. Eric Weaver _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at 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/20090714/eeeb578c/attachment-0002.html From pjintheusa at gmail.com Tue Jul 14 14:27:51 2009 From: pjintheusa at gmail.com (Phillip Jones) Date: Tue, 14 Jul 2009 17:27:51 -0400 Subject: [Freeswitch-users] Pastebin.freeswitch.org Message-ID: <367751820907141427o636fc28bsf36f8320f3b8783b@mail.gmail.com> Hi there, I am sure I am missing something. Can someone point out where to signup for a username / password to pastebin.freeswitch.org. I am pulling my hair out and feel kinda stupid asking this. Thanks Phillip Jones -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090714/8404b3aa/attachment-0002.html From brian at freeswitch.org Tue Jul 14 14:33:41 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 14 Jul 2009 16:33:41 -0500 Subject: [Freeswitch-users] Pastebin.freeswitch.org In-Reply-To: <367751820907141427o636fc28bsf36f8320f3b8783b@mail.gmail.com> References: <367751820907141427o636fc28bsf36f8320f3b8783b@mail.gmail.com> Message-ID: Phillip, Well did you read the dialog box? :P It tells you the user/pass to use ... it keeps the spammers out! /b On Jul 14, 2009, at 4:27 PM, Phillip Jones wrote: > Hi there, > > I am sure I am missing something. Can someone point out where to > signup for a username / password to pastebin.freeswitch.org. > > I am pulling my hair out and feel kinda stupid asking this. > > Thanks > > > Phillip Jones > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090714/afb1b7b8/attachment-0002.html From msc at freeswitch.org Tue Jul 14 14:40:02 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 14 Jul 2009 14:40:02 -0700 Subject: [Freeswitch-users] Preventing disconnect on event_sockte close In-Reply-To: <7DB369DA44D5CF42B6C23BD7D611A50FDABBFA760F@VA3DIAXVS061.RED001.local> References: <7DB369DA44D5CF42B6C23BD7D611A50FDA7E1D5FD6@VA3DIAXVS061.RED001.local> <87f2f3b90907131808l100afe7dwc95d0f624ff38d50@mail.gmail.com> <7DB369DA44D5CF42B6C23BD7D611A50FDA7E1D604D@VA3DIAXVS061.RED001.local> <87f2f3b90907141011g46686396se27fa3cc0320963d@mail.gmail.com> <7DB369DA44D5CF42B6C23BD7D611A50FDABBFA760F@VA3DIAXVS061.RED001.local> Message-ID: <87f2f3b90907141440l5235ab70h7d48ee961367f5e@mail.gmail.com> On Tue, Jul 14, 2009 at 2:14 PM, Weaver, Eric wrote: > Using api uuid_park 81b65478-70b9-11de-bf26-b962186102f7 before > terminating the NC session works, the call is not disconnected. > > > > Once that is done, I do not receive DTMF and cannot play prompts to the > caller, they seem to be in limbo. I can uuid_kill the call but I need to get > dtmf and play prompts to them. Perhaps Park is not where I need to put > these calls ? > > > To get a call out of park you need to bridge to it or transfer it to another extension. If you have an extension you can just uuid_transfer the parked call's uuid. If you have an existing call's uuid you can use uuid_bridge to bridge the two together. Could you remind me of the application you're building? Just curious what the big picture is. -MC > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael > Collins > *Sent:* Tuesday, July 14, 2009 11:12 AM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Preventing disconnect on event_sockte > close > > > > > > On Tue, Jul 14, 2009 at 9:59 AM, Weaver, Eric > wrote: > > Tried it using FreeSWITCH Version 1.0.4pre9 (14036M) and the following > steps > > > > Start netcat > > > > netcat -v -l -p 14000 > > > > place call, socket is connected via dial plan, enter the following. > > > > connect\n\n > > > > sendmsg > > call-command: execute > > execute-app-name: answer\n\n > > > > > > sendmsg > > call-command: execute > > execute-app-name: playback > > execute-app-arg: /home/eweaver/holdmusic.wav > > > > > > > > > > sendmsg > > call-command: execute > > execute-app-name: park > > > Try this: > api uuid_park > > You'll need to capture the uuid at some point and store it. For testing I > just manually copied and pasted it to/from the console screen. > -MC > > > > Console window displays this message: > > > > 2009-07-14 10:42:29.245833 [ERR] switch_ivr.c:654 Cannot park channels that > are under control already. > > > > at this point, ^C in the netcat window. Call is disconnected. > > > > > > Need to be able to park these calls so they can then be handled from an > inbound event socket connection. > > > > eric > > > > > > > > > > > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael > Collins > *Sent:* Monday, July 13, 2009 7:08 PM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Preventing disconnect on event_sockte > close > > > > I don't know if this will work for you but I just tested this scenario with > uuid_park. After parking the call I disconnected the socket and the call > continued. I did the same thing with uuid_transfer. After the transfer I > disconnected the socket and the call continued. > > How are you handling the call and how is the socket getting disconnected? > > -MC > > On Mon, Jul 13, 2009 at 4:35 PM, Weaver, Eric > wrote: > > Using mod_event_socket in outbound mode, is there any to prevent a call > from being disconnected when the outbound socket is closed ? I would like to > handle the initial inbound call using outbound but after the disposition of > the call is determined, close the socket and have that call managed using an > inbound socket instead. > > > > Eric Weaver > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090714/49dd88d8/attachment-0002.html From msc at freeswitch.org Tue Jul 14 14:43:55 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 14 Jul 2009 14:43:55 -0700 Subject: [Freeswitch-users] Pastebin.freeswitch.org In-Reply-To: <367751820907141427o636fc28bsf36f8320f3b8783b@mail.gmail.com> References: <367751820907141427o636fc28bsf36f8320f3b8783b@mail.gmail.com> Message-ID: <87f2f3b90907141443y3033c364n78f5dd33fd99c318@mail.gmail.com> On Tue, Jul 14, 2009 at 2:27 PM, Phillip Jones wrote: > Hi there, > > I am sure I am missing something. Can someone point out where to signup for > a username / password to pastebin.freeswitch.org. > > I am pulling my hair out and feel kinda stupid asking this. That's okay, you are neither the first nor the last person to get caught by this one. :) -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090714/795662e4/attachment-0002.html From pjintheusa at gmail.com Tue Jul 14 15:01:43 2009 From: pjintheusa at gmail.com (Phillip Jones) Date: Tue, 14 Jul 2009 18:01:43 -0400 Subject: [Freeswitch-users] Pastebin.freeswitch.org In-Reply-To: <87f2f3b90907141443y3033c364n78f5dd33fd99c318@mail.gmail.com> References: <367751820907141427o636fc28bsf36f8320f3b8783b@mail.gmail.com> <87f2f3b90907141443y3033c364n78f5dd33fd99c318@mail.gmail.com> Message-ID: <367751820907141501v3a71fbe2v683393803147da5d@mail.gmail.com> Thank you! Not that obvious in IE actually - I should stick to FF. On Tue, Jul 14, 2009 at 5:43 PM, Michael Collins wrote: > > > On Tue, Jul 14, 2009 at 2:27 PM, Phillip Jones wrote: > >> Hi there, >> >> I am sure I am missing something. Can someone point out where to signup >> for a username / password to pastebin.freeswitch.org. >> >> I am pulling my hair out and feel kinda stupid asking this. > > > That's okay, you are neither the first nor the last person to get caught by > this one. :) > -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/20090714/1decbc49/attachment-0002.html From eweaver at meetingone.com Tue Jul 14 15:03:11 2009 From: eweaver at meetingone.com (Weaver, Eric) Date: Tue, 14 Jul 2009 15:03:11 -0700 Subject: [Freeswitch-users] Preventing disconnect on event_sockte close In-Reply-To: <87f2f3b90907141440l5235ab70h7d48ee961367f5e@mail.gmail.com> References: <7DB369DA44D5CF42B6C23BD7D611A50FDA7E1D5FD6@VA3DIAXVS061.RED001.local> <87f2f3b90907131808l100afe7dwc95d0f624ff38d50@mail.gmail.com> <7DB369DA44D5CF42B6C23BD7D611A50FDA7E1D604D@VA3DIAXVS061.RED001.local> <87f2f3b90907141011g46686396se27fa3cc0320963d@mail.gmail.com> <7DB369DA44D5CF42B6C23BD7D611A50FDABBFA760F@VA3DIAXVS061.RED001.local> <87f2f3b90907141440l5235ab70h7d48ee961367f5e@mail.gmail.com> Message-ID: <7DB369DA44D5CF42B6C23BD7D611A50FDACD801F9A@VA3DIAXVS061.RED001.local> Looking a FS to use as Media mixer for conferencing platform. Not really doing call to call bridging. We really don't have extensions.... Conferences are created o the fly as needed. Already have the conf and call control app done and in production using a different audio mixer, I would like to put FS in place of it. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Tuesday, July 14, 2009 3:40 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Preventing disconnect on event_sockte close On Tue, Jul 14, 2009 at 2:14 PM, Weaver, Eric > wrote: Using api uuid_park 81b65478-70b9-11de-bf26-b962186102f7 before terminating the NC session works, the call is not disconnected. Once that is done, I do not receive DTMF and cannot play prompts to the caller, they seem to be in limbo. I can uuid_kill the call but I need to get dtmf and play prompts to them. Perhaps Park is not where I need to put these calls ? To get a call out of park you need to bridge to it or transfer it to another extension. If you have an extension you can just uuid_transfer the parked call's uuid. If you have an existing call's uuid you can use uuid_bridge to bridge the two together. Could you remind me of the application you're building? Just curious what the big picture is. -MC From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Tuesday, July 14, 2009 11:12 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Preventing disconnect on event_sockte close On Tue, Jul 14, 2009 at 9:59 AM, Weaver, Eric > wrote: Tried it using FreeSWITCH Version 1.0.4pre9 (14036M) and the following steps Start netcat netcat -v -l -p 14000 place call, socket is connected via dial plan, enter the following. connect\n\n sendmsg call-command: execute execute-app-name: answer\n\n sendmsg call-command: execute execute-app-name: playback execute-app-arg: /home/eweaver/holdmusic.wav sendmsg call-command: execute execute-app-name: park Try this: api uuid_park You'll need to capture the uuid at some point and store it. For testing I just manually copied and pasted it to/from the console screen. -MC Console window displays this message: 2009-07-14 10:42:29.245833 [ERR] switch_ivr.c:654 Cannot park channels that are under control already. at this point, ^C in the netcat window. Call is disconnected. Need to be able to park these calls so they can then be handled from an inbound event socket connection. eric From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Monday, July 13, 2009 7:08 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Preventing disconnect on event_sockte close I don't know if this will work for you but I just tested this scenario with uuid_park. After parking the call I disconnected the socket and the call continued. I did the same thing with uuid_transfer. After the transfer I disconnected the socket and the call continued. How are you handling the call and how is the socket getting disconnected? -MC On Mon, Jul 13, 2009 at 4:35 PM, Weaver, Eric > wrote: Using mod_event_socket in outbound mode, is there any to prevent a call from being disconnected when the outbound socket is closed ? I would like to handle the initial inbound call using outbound but after the disposition of the call is determined, close the socket and have that call managed using an inbound socket instead. Eric Weaver _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ Freeswitch-users mailing list Freeswitch-users at 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/20090714/f5f4b6e5/attachment-0002.html From pjintheusa at gmail.com Tue Jul 14 15:22:42 2009 From: pjintheusa at gmail.com (Phillip Jones) Date: Tue, 14 Jul 2009 18:22:42 -0400 Subject: [Freeswitch-users] leg_timeout In-Reply-To: <367751820907141135t2c374338y54226c9721cfa739@mail.gmail.com> References: <367751820907140835y322c576en1eb2ba931388d732@mail.gmail.com> <87f2f3b90907140916o3f008a3br26bfca075a9e3d2e@mail.gmail.com> <367751820907141002g3a837a09meabdbc9743c3bee7@mail.gmail.com> <191c3a030907141020l48e8cb2cm6106ff4442ee1bed@mail.gmail.com> <87f2f3b90907141037r260001d4i2c41dcc0e4c25c5@mail.gmail.com> <367751820907141114w56ae9adey4836238740a1ed43@mail.gmail.com> <191c3a030907141127x255f21c5q7d80cf26ea8451a9@mail.gmail.com> <367751820907141135t2c374338y54226c9721cfa739@mail.gmail.com> Message-ID: <367751820907141522o491b40f4jbf79c25f3c053139@mail.gmail.com> Hi there, I downloaded the latest trunk, compiled and updated. Still no joy I am afraid. This is the log file in pastebin - http://pastebin.freeswitch.org/9712 Code in my managed DLL is at: http://pastebin.freeswitch.org/9715 Dialplan binds to above: confirm.js is at: http://pastebin.freeswitch.org/9713 Thanks again for your help on this. On Tue, Jul 14, 2009 at 2:35 PM, Phillip Jones wrote: > Ah - SVN Trunk - thought you meant DID trunk!!! My bad. > > Sorry - understand now! Will recompile and let you know. > > On Tue, Jul 14, 2009 at 2:27 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> You actually updated your code and recompiled it all too? >> This param was added about 30 seconds before I sent you the email. >> >> >> >> On Tue, Jul 14, 2009 at 1:14 PM, Phillip Jones wrote: >> >>> Thanks for your response. >>> >>> That does not seem to work. Here is my code: >>> >>> if(Session.Ready()) >>> { >>> Session.Execute("set", "ignore_early_media=true"); >>> Session.Execute("set", "hangup_after_bridge=true"); >>> Session.Execute("set", "ringback=${us-ring}"); >>> >>> Session.Answer(); >>> string Caller_ID_Number = >>> this.Session.GetVariable("caller_id_number"); >>> Session.Execute("set", "group_confirm_key=exec"); >>> *Session.Execute("set", "group_confirm_cancel_timeout=true"); >>> * Session.Execute("set", "group_confirm_file=javascript confirm.js"); >>> Session.Execute("bridge", >>> "[leg_timeout=20,leg_confirm=y]sofia/gateway/broadvox/6095553828"); >>> } >>> Session.Hangup("USER_BUSY"); >>> >>> I also tried *group_confirm_cancel_leg_timeout* just in case. >>> >>> Am I missing something? >>> >>> >>> >>> >>> On Tue, Jul 14, 2009 at 1:37 PM, Michael Collins wrote: >>> >>>> FYI, >>>> This has been added to the wiki: >>>> >>>> http://wiki.freeswitch.org/wiki/Channel_Variables#group_confirm_cancel_timeout >>>> >>>> -MC >>>> >>>> >>>> On Tue, Jul 14, 2009 at 10:20 AM, Anthony Minessale < >>>> anthony.minessale at gmail.com> wrote: >>>> >>>>> update to trunk and try setting >>>>> group_confirm_cancel_timeout=true >>>>> >>>>> let me know if it works >>>>> >>>>> >>>>> On Tue, Jul 14, 2009 at 12:02 PM, Phillip Jones wrote: >>>>> >>>>>> Hi, >>>>>> >>>>>> Thanks for the reply. >>>>>> >>>>>> >> Both "leg_timeout" and "leg_progress_timeout" are for controlling >>>>>> how long to wait prior to the B-leg answering. >>>>>> >>>>>> I think this is my point. leg_timeout seems to control how long to >>>>>> wait prior to the bridge completeing, not the B-leg answering. >>>>>> >>>>>> In my situation I am using: >>>>>> >>>>>> Session.Execute("set", "group_confirm_key=exec"); >>>>>> Session.Execute("set", "group_confirm_file=javascript confirm.js"); >>>>>> >>>>>> my confirm.js prompts leg_b for 1 (to take the call). If leg_timeout >>>>>> is set to 10 you have 10 seconds to answer the call AND press 1. >>>>>> >>>>>> I just want call_timeout to be satisfied when the call is answered. >>>>>> Not when the called party presses 1 and the bridge is complete. >>>>>> >>>>>> I am new all this so I will work out how to use the pastebin etc. >>>>>> >>>>>> Thanks for your help. >>>>>> >>>>>> >>>>>> Phillip Jones >>>>>> >>>>>> >>>>>> >>>>>> On Tue, Jul 14, 2009 at 12:16 PM, Michael Collins >>>>> > wrote: >>>>>> >>>>>>> >>>>>>> >>>>>>> On Tue, Jul 14, 2009 at 8:35 AM, Phillip Jones < >>>>>>> pjintheusa at gmail.com> wrote: >>>>>>> >>>>>>>> Hi there, >>>>>>>> >>>>>>>> Here is my call flow: >>>>>>>> >>>>>>>> 1) leg A is bridged to leg B >>>>>>>> 2) when leg B is answered I play a confirm script - "please 1 to >>>>>>>> accept this call" >>>>>>>> >>>>>>>> I only want leg B to ring 20 seconds. BUT when the caller party >>>>>>>> answers, he should have as long as he needs to press 1. >>>>>>>> >>>>>>>> "leg_timeout" seems to be in play until the bridge is completed. I >>>>>>>> need it to reset when leg b is answered. >>>>>>>> >>>>>>> >>>>>>> Correct. Both "leg_timeout" and "leg_progress_timeout" are for >>>>>>> controlling how long to wait prior to the B-leg answering. >>>>>>> (leg_progress_timeout specifies how long to wait for any kind of progress, >>>>>>> be it early media of some sort, ringing, or an answer.) >>>>>>> >>>>>>>> >>>>>>>> I tried resetting the leg_timeout in the confirm script after leg b >>>>>>>> is answered. I also tried using leg_progress_timeout. Neither seemed to >>>>>>>> work. >>>>>>>> >>>>>>> >>>>>>> What exactly are you trying to do? The two variables you've mentioned >>>>>>> shouldn't have any effect on the call after it has been established. >>>>>>> >>>>>>>> >>>>>>>> Any help or suggestions would be welcome. >>>>>>>> >>>>>>> >>>>>>> Could you pastebin your dialplan and a debug log of a call that does >>>>>>> not work? See this page for some handy tips on using pastebin and collecting >>>>>>> information for debugging purposes: >>>>>>> http://wiki.freeswitch.org/wiki/Reporting_Bugs#Debugging_Steps >>>>>>> >>>>>>> -MC >>>>>>> >>>>>>>> >>>>>>>> Phillip Jones >>>>>>>> >>>>>>>> _______________________________________________ >>>>>>>> Freeswitch-users mailing list >>>>>>>> Freeswitch-users at lists.freeswitch.org >>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>> UNSUBSCRIBE: >>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>> http://www.freeswitch.org >>>>>>>> >>>>>>>> >>>>>>> >>>>>>> _______________________________________________ >>>>>>> Freeswitch-users mailing list >>>>>>> Freeswitch-users at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE: >>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>>> >>>>>> >>>>>> _______________________________________________ >>>>>> Freeswitch-users mailing list >>>>>> Freeswitch-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>> >>>>> >>>>> -- >>>>> Anthony Minessale II >>>>> >>>>> FreeSWITCH http://www.freeswitch.org/ >>>>> ClueCon http://www.cluecon.com/ >>>>> Twitter: http://twitter.com/FreeSWITCH_wire >>>>> >>>>> AIM: anthm >>>>> MSN:anthony_minessale at hotmail.com >>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>>>> IRC: irc.freenode.net #freeswitch >>>>> >>>>> FreeSWITCH Developer Conference >>>>> 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 >>> >>> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> 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/20090714/3c69cb83/attachment-0002.html From Kareem.Hamdy at trustvesta.com Tue Jul 14 17:04:05 2009 From: Kareem.Hamdy at trustvesta.com (Kareem Hamdy) Date: Tue, 14 Jul 2009 17:04:05 -0700 Subject: [Freeswitch-users] OpenZAP and FreeSWITCH w/ Sangoma In-Reply-To: References: Message-ID: <1134625859513549B3B943E0133490E202AC614768@TDCP-EXSTORE-01.ad.trustvesta.com> Hello: I would like to connect a Sangoma T1 card to FreeSWITCH. Most of the docs I see pertain to a PRI. When I leave out the d-chan notation, I get errors regarding not able to get the d-chan up and running in the CLI. Here's my info: [span wanpipe T1] trunk_type => t1 b-channel => 1:1-24 [span wanpipe T2] trunk_type => t1 b-channel => 2:1-24 ---- #================================================ # WANPIPE1 Configuration File #================================================ # # Date: Wed Dec 6 20:29:03 UTC 2006 # # Note: This file was generated automatically # by /usr/local/sbin/setup-sangoma program. # # If you want to edit this file, it is # recommended that you use wancfg program # to do so. #================================================ # Sangoma Technologies Inc. #================================================ [devices] wanpipe1 = WAN_AFT_TE1, Comment [interfaces] w1g1 = wanpipe1, , TDM_VOICE_API, Comment [wanpipe1] CARD_TYPE = AFT S514CPU = A CommPort = PRI AUTO_PCISLOT = NO PCISLOT = 1 PCIBUS = 6 FE_MEDIA = T1 FE_LCODE = B8ZS FE_FRAME = ESF FE_LINE = 1 TE_CLOCK = NORMAL TE_REF_CLOCK = 0 TE_HIGHIMPEDANCE = NO LBO = 0DB FE_TXTRISTATE = NO MTU = 1500 UDPPORT = 9000 TTL = 255 IGNORE_FRONT_END = NO TDMV_SPAN = 1 TDMV_DCHAN = 0 TDMV_HW_DTMF = YES TDMV_HW_FAX_DETECT = NO [w1g1] ACTIVE_CH = ALL TDMV_HWEC = YES MTU = 80 --- In freeSWITCH's openzap.conf.xml, I've tried pri_span as well as analog. I cannot find a straight up T1 wiki anywhere. Would someone please provide an example? Thanks, Kareem From msc at freeswitch.org Tue Jul 14 17:24:18 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 14 Jul 2009 17:24:18 -0700 Subject: [Freeswitch-users] OpenZAP and FreeSWITCH w/ Sangoma In-Reply-To: <1134625859513549B3B943E0133490E202AC614768@TDCP-EXSTORE-01.ad.trustvesta.com> References: <1134625859513549B3B943E0133490E202AC614768@TDCP-EXSTORE-01.ad.trustvesta.com> Message-ID: <87f2f3b90907141724q2735fac1jdacea3994db62782@mail.gmail.com> See inline comments On Tue, Jul 14, 2009 at 5:04 PM, Kareem Hamdy wrote: > Hello: > > I would like to connect a Sangoma T1 card to FreeSWITCH. Most of the docs > I see pertain to a PRI. When I leave out the d-chan notation, I get errors > regarding not able to get the d-chan up and running in the CLI. > > Here's my info: > > [span wanpipe T1] > trunk_type => t1 > b-channel => 1:1-24 b-channel => 1:1-23 d-channel => 1:24 > > > [span wanpipe T2] > trunk_type => t1 > b-channel => 2:1-24 > set up like span 1 example > > ---- > > #================================================ > # WANPIPE1 Configuration File > #================================================ > # > # Date: Wed Dec 6 20:29:03 UTC 2006 > # > # Note: This file was generated automatically > # by /usr/local/sbin/setup-sangoma program. > # > # If you want to edit this file, it is > # recommended that you use wancfg program > # to do so. > #================================================ > # Sangoma Technologies Inc. > #================================================ > > [devices] > wanpipe1 = WAN_AFT_TE1, Comment > > [interfaces] > w1g1 = wanpipe1, , TDM_VOICE_API, Comment > > [wanpipe1] > CARD_TYPE = AFT > S514CPU = A > CommPort = PRI > AUTO_PCISLOT = NO > PCISLOT = 1 > PCIBUS = 6 > FE_MEDIA = T1 > FE_LCODE = B8ZS > FE_FRAME = ESF > FE_LINE = 1 > TE_CLOCK = NORMAL > TE_REF_CLOCK = 0 > > TE_HIGHIMPEDANCE = NO > LBO = 0DB > FE_TXTRISTATE = NO > MTU = 1500 > UDPPORT = 9000 > TTL = 255 > IGNORE_FRONT_END = NO > TDMV_SPAN = 1 > TDMV_DCHAN = 0 TDMV_DCAHN = 24 > > TDMV_HW_DTMF = YES > TDMV_HW_FAX_DETECT = NO > > [w1g1] > ACTIVE_CH = ALL > TDMV_HWEC = YES > MTU = 80 > > > --- > > > > In freeSWITCH's openzap.conf.xml, I've tried pri_span as well as analog. > > I cannot find a straight up T1 wiki anywhere. Would someone please provide > an example? > > > Thanks, > Kareem > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090714/da16d89e/attachment-0002.html From digilord at me.com Tue Jul 14 19:20:19 2009 From: digilord at me.com (DigiLord) Date: Tue, 14 Jul 2009 19:20:19 -0700 Subject: [Freeswitch-users] GXW4104 & FreeSwitch Message-ID: <8DC39E34-A395-42D5-B299-070605A2DCEE@me.com> Hello all, I am getting my feet wet with FreeSwitch by migrating my Asterisk box over. I have run into a few things that I am not sure how to accomplish. I have a Grandstream GXW4104 with one analog line connected. I have it connected and I am able to receive calls on my Polycom 501 (ext 2101) that is registered to the FreeSwitch server. The one problem is that CallerID is not the CallerID from the caller, it's the CallerID from the Grandstream device (ext 2100). On the same device there is HORRIBLE echo. I have set echo cancellation on the device to enabled and disabled to no avail. Under Asterisk there was no echo. I setup the device as a provider. Was that the right way to accomplish connecting this device to FS? Is there a way to enable sending an e-mail containing my voicemail messages like Asterisk does? Thanks in advance for any help you can give! Dan From brian at freeswitch.org Tue Jul 14 19:31:23 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 14 Jul 2009 21:31:23 -0500 Subject: [Freeswitch-users] GXW4104 & FreeSwitch In-Reply-To: <8DC39E34-A395-42D5-B299-070605A2DCEE@me.com> References: <8DC39E34-A395-42D5-B299-070605A2DCEE@me.com> Message-ID: <085EE9F4-A513-45FD-89E9-C66A0BE3715F@freeswitch.org> On Jul 14, 2009, at 9:20 PM, DigiLord wrote: > Hello all, > I am getting my feet wet with FreeSwitch by migrating my Asterisk box > over. I have run into a few things that I am not sure how to > accomplish. > > I have a Grandstream GXW4104 with one analog line connected. I have > it connected and I am able to receive calls on my Polycom 501 (ext > 2101) that is registered to the FreeSwitch server. The one problem is > that CallerID is not the CallerID from the caller, it's the CallerID > from the Grandstream device (ext 2100). How is the callerid passed on this device? > On the same device there is HORRIBLE echo. I have set echo > cancellation on the device to enebled and disabled to no avail. Under > Asterisk there was no echo. If it didn't have echo on asterisk it shouldn't have echo on FreeSWITCH, Can you describe the echo better? Are you using speaker phone? What codecs? > > > I setup the device as a provider. Was that the right way to > accomplish connecting this device to FS? > > Is there a way to enable sending an e-mail containing my voicemail > messages like Asterisk does? Yes check the mod_voicemail page on the wiki. /b > > > Thanks in advance for any help you can give! > > Dan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090714/eed9a9d7/attachment-0002.html From shaheryarkh at googlemail.com Tue Jul 14 21:19:48 2009 From: shaheryarkh at googlemail.com (Muhammad Shahzad) Date: Wed, 15 Jul 2009 10:19:48 +0600 Subject: [Freeswitch-users] SIP Trace Option at Runtime Message-ID: Hi, Is there any CLI command to enable / disable SIP packet trace at runtime. I do know an option in SIP profile which enables / disable SIP trace but it to apply it i have reload mod_sofia, which at many times fail due to a running call. Thank you. -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090715/ca7ce7b2/attachment-0002.html From jason at jasonjgw.net Tue Jul 14 21:32:25 2009 From: jason at jasonjgw.net (Jason White) Date: Wed, 15 Jul 2009 14:32:25 +1000 Subject: [Freeswitch-users] SIP Trace Option at Runtime In-Reply-To: References: Message-ID: <20090715043225.GA21117@jdc.jasonjgw.net> Muhammad Shahzad wrote: > Is there any CLI command to enable / disable SIP packet trace at runtime. sofia profile siptrace on sofia profile siptrace off sofia help would have answered your question. From elihayun at gmail.com Tue Jul 14 21:49:07 2009 From: elihayun at gmail.com (Eli Hayun) Date: Wed, 15 Jul 2009 07:49:07 +0300 Subject: [Freeswitch-users] Get voicemail messages In-Reply-To: References: Message-ID: <4A5D5FC3.4050701@savion.huji.ac.il> did you bind your lua script to directory lookups in addition to the dialplan? On Tue, Jul 14, 2009 at 7:02 AM, Eli Hayun wrote: > > Hi > > I am not using fixed xml files for the extension registration. I have > > LUA script to return an XML string to FS. > > Everything goes fine until I am trying to get the voice messages. > > When am entering my id, FS (or voicemail module) try to get the xml for > > that id, but it cant find it. My lua script did NOT recieved any xml > > request at that point. > > What should I do to solve the problem. > > > > Thanks > > Eli Hayun > > > Yes I did bind it: my lua.conf.xml is like this When an extension tried to register, I have no problem. But when I want to use VoiceMail to retrieve my messeges, I got a problem. Here is the partial log: 2009-07-15 07:44:49.373089 [INFO] mod_dialplan_xml.c:252 Processing Phone2->*98 in context default 2009-07-15 07:44:49.386466 [NOTICE] mod_dptools.c:649 Channel [sofia/internal/80671 at 132.64.3.86] has been answered 2009-07-15 07:44:51.933664 [WARNING] mod_voicemail.c:2072 Can't find user [80671 at 132.64.3.86] 2009-07-15 07:44:52.533435 [NOTICE] switch_core_state_machine.c:179 Hangup sofia/internal/80671 at 132.64.3.86 [CS_EXECUTE] [NORMAL_CLEARING] 2009-07-15 07:44:52.545698 [NOTICE] switch_core_session.c:1085 Session 3 (sofia/internal/80671 at 132.64.3.86) Ended 2009-07-15 07:44:52.545698 [NOTICE] switch_core_session.c:1087 Close Channel sofia/internal/80671 at 132.64.3.86 [CS_DESTROY] From brad.tuan at gmail.com Wed Jul 15 01:05:24 2009 From: brad.tuan at gmail.com (Brad Tuan) Date: Wed, 15 Jul 2009 16:05:24 +0800 Subject: [Freeswitch-users] How to set the IVR of VM menu?? Message-ID: How to set the date format , and the IVR flow ........?? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090715/c35eaf18/attachment-0002.html From tzury.by at reguluslabs.com Wed Jul 15 02:57:29 2009 From: tzury.by at reguluslabs.com (Tzury Bar Yochay) Date: Wed, 15 Jul 2009 12:57:29 +0300 Subject: [Freeswitch-users] SIP TLS (and SRTP) Message-ID: <10128ef10907150257u3ae1c20jb7772b66e8c1a738@mail.gmail.com> Hi all, I was following the instruction found at http://wiki.freeswitch.org/wiki/SIP_TLS When I got to Step 4 I saw that instruction of editing the dial-string. However, in my conf/dialplan/default.xml I did not found any matched entry . Version I am using: typing version at my FS CLI yields: FreeSWITCH 1.0.trunk (14144M) thanks, Tzury Bar Yochay From rupa at rupa.com Wed Jul 15 06:26:23 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Wed, 15 Jul 2009 08:26:23 -0500 Subject: [Freeswitch-users] SIP TLS (and SRTP) In-Reply-To: <10128ef10907150257u3ae1c20jb7772b66e8c1a738@mail.gmail.com> References: <10128ef10907150257u3ae1c20jb7772b66e8c1a738@mail.gmail.com> Message-ID: hmm... When I put that section in, I put the wrong filename. It should be directory/default.xml. I'll update the wiki. Also, it is a valid configuration to support tls but not srtp. I'll put a bit of a discussion in there talking about that. Setting sip_secure_media to true requires the endpoint do srtp. There is no way (that I know of) to say "do srtp if possible but if not fallback to clear". zrtp does fallback to clear if it can't negotiate keys. But zrtp is supported by far fewer endpoints and no hardphones (as of yet). On Wed, Jul 15, 2009 at 4:57 AM, Tzury Bar Yochay wrote: > Hi all, > > I was following the instruction found at > http://wiki.freeswitch.org/wiki/SIP_TLS > When I got to Step 4 I saw that instruction of editing the dial-string. > However, in my conf/dialplan/default.xml I did not found any matched entry > . > > Version I am using: > typing version at my FS CLI yields: FreeSWITCH 1.0.trunk (14144M) > > thanks, > Tzury Bar Yochay > > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090715/e465368c/attachment-0002.html From brian at freeswitch.org Wed Jul 15 06:41:08 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 15 Jul 2009 08:41:08 -0500 Subject: [Freeswitch-users] SIP TLS (and SRTP) In-Reply-To: <10128ef10907150257u3ae1c20jb7772b66e8c1a738@mail.gmail.com> References: <10128ef10907150257u3ae1c20jb7772b66e8c1a738@mail.gmail.com> Message-ID: <8594764D-2F2A-431C-BA91-1C2D5A97C90D@freeswitch.org> It tells you to edit conf/directory/default.xml not dialplan/ default.xml and put as the dial-string. /b On Jul 15, 2009, at 4:57 AM, Tzury Bar Yochay wrote: > Hi all, > > I was following the instruction found at http://wiki.freeswitch.org/wiki/SIP_TLS > When I got to Step 4 I saw that instruction of editing the dial- > string. > However, in my conf/dialplan/default.xml I did not found any matched > entry . > > Version I am using: > typing version at my FS CLI yields: FreeSWITCH 1.0.trunk (14144M) > > thanks, > Tzury Bar Yochay From larclap at yahoo.com Wed Jul 15 07:29:28 2009 From: larclap at yahoo.com (Lars Zeb) Date: Wed, 15 Jul 2009 07:29:28 -0700 Subject: [Freeswitch-users] fs_cli - display variable values? Message-ID: <003401ca0558$a944a6b0$fbcdf410$@com> Is it possible to display the value of a variable in fs_cli? I tried "echo ${domain_name}", but it just echoed what I typed (${domain_name}), rather than its value. I do not know how to get help on an individual command from the help facility in fs_cli. I tried fs_cli itself and also the docs but could find nothing. Thanks, Lars -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090715/9bf80956/attachment-0002.html From mrene_lists at avgs.ca Wed Jul 15 07:36:46 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Wed, 15 Jul 2009 10:36:46 -0400 Subject: [Freeswitch-users] fs_cli - display variable values? In-Reply-To: <003401ca0558$a944a6b0$fbcdf410$@com> References: <003401ca0558$a944a6b0$fbcdf410$@com> Message-ID: <3C946296-7580-44EE-B8E1-35417E26B182@avgs.ca> eval [expression] or eval uuid: [expression] or global_getvar varname or uuid_getvar uuid varname Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca Am 15-Jul-09 um 10:29 AM schrieb Lars Zeb: > Is it possible to display the value of a variable in fs_cli? I tried > ?echo ${domain_name}?, but it just echoed what I typed ($ > {domain_name}), rather than its value. > > I do not know how to get help on an individual command from the help > facility in fs_cli. I tried fs_cli itself and also the docs but > could find nothing. > > Thanks, Lars > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090715/8a9efc91/attachment-0002.html From pjintheusa at gmail.com Wed Jul 15 07:38:24 2009 From: pjintheusa at gmail.com (Phillip Jones) Date: Wed, 15 Jul 2009 10:38:24 -0400 Subject: [Freeswitch-users] leg_timeout In-Reply-To: <367751820907141522o491b40f4jbf79c25f3c053139@mail.gmail.com> References: <367751820907140835y322c576en1eb2ba931388d732@mail.gmail.com> <87f2f3b90907140916o3f008a3br26bfca075a9e3d2e@mail.gmail.com> <367751820907141002g3a837a09meabdbc9743c3bee7@mail.gmail.com> <191c3a030907141020l48e8cb2cm6106ff4442ee1bed@mail.gmail.com> <87f2f3b90907141037r260001d4i2c41dcc0e4c25c5@mail.gmail.com> <367751820907141114w56ae9adey4836238740a1ed43@mail.gmail.com> <191c3a030907141127x255f21c5q7d80cf26ea8451a9@mail.gmail.com> <367751820907141135t2c374338y54226c9721cfa739@mail.gmail.com> <367751820907141522o491b40f4jbf79c25f3c053139@mail.gmail.com> Message-ID: <367751820907150738k4b0fae48u4ede9d87ff916324@mail.gmail.com> Hey Guys, I took a look at the source that Anthony updated. I see this: } else if (!strcasecmp((char *) hi->name, "group_confirm_file")) { ok = 1; } else if (!strcasecmp((char *) hi->name, "group_confirm_*clear*_timeout")) { ok = 1; } else if (!strcasecmp((char *) hi->name, "forked_dial")) { and: if (switch_true(switch_event_get_header(var_event, "group_confirm_*cancel*_timeout"))) { oglobals.cancel_timeout = -1; } I updated the *group_confirm_clear_timeout *to *group_confirm_cancel_timeout * and recompiled and this is now working just great. Thanks very much for incorporating this. It is much appreciated. Phillip Jones On Tue, Jul 14, 2009 at 6:22 PM, Phillip Jones wrote: > Hi there, > > I downloaded the latest trunk, compiled and updated. Still no joy I am > afraid. > > This is the log file in pastebin - http://pastebin.freeswitch.org/9712 > > Code in my managed DLL is at: http://pastebin.freeswitch.org/9715 > > Dialplan binds to above: > > > > > confirm.js is at: http://pastebin.freeswitch.org/9713 > > > Thanks again for your help on this. > > > > > > On Tue, Jul 14, 2009 at 2:35 PM, Phillip Jones wrote: > >> Ah - SVN Trunk - thought you meant DID trunk!!! My bad. >> >> Sorry - understand now! Will recompile and let you know. >> >> On Tue, Jul 14, 2009 at 2:27 PM, Anthony Minessale < >> anthony.minessale at gmail.com> wrote: >> >>> You actually updated your code and recompiled it all too? >>> This param was added about 30 seconds before I sent you the email. >>> >>> >>> >>> On Tue, Jul 14, 2009 at 1:14 PM, Phillip Jones wrote: >>> >>>> Thanks for your response. >>>> >>>> That does not seem to work. Here is my code: >>>> >>>> if(Session.Ready()) >>>> { >>>> Session.Execute("set", "ignore_early_media=true"); >>>> Session.Execute("set", "hangup_after_bridge=true"); >>>> Session.Execute("set", "ringback=${us-ring}"); >>>> >>>> Session.Answer(); >>>> string Caller_ID_Number = >>>> this.Session.GetVariable("caller_id_number"); >>>> Session.Execute("set", "group_confirm_key=exec"); >>>> *Session.Execute("set", "group_confirm_cancel_timeout=true"); >>>> * Session.Execute("set", "group_confirm_file=javascript >>>> confirm.js"); >>>> Session.Execute("bridge", >>>> "[leg_timeout=20,leg_confirm=y]sofia/gateway/broadvox/6095553828"); >>>> } >>>> Session.Hangup("USER_BUSY"); >>>> >>>> I also tried *group_confirm_cancel_leg_timeout* just in case. >>>> >>>> Am I missing something? >>>> >>>> >>>> >>>> >>>> On Tue, Jul 14, 2009 at 1:37 PM, Michael Collins wrote: >>>> >>>>> FYI, >>>>> This has been added to the wiki: >>>>> >>>>> http://wiki.freeswitch.org/wiki/Channel_Variables#group_confirm_cancel_timeout >>>>> >>>>> -MC >>>>> >>>>> >>>>> On Tue, Jul 14, 2009 at 10:20 AM, Anthony Minessale < >>>>> anthony.minessale at gmail.com> wrote: >>>>> >>>>>> update to trunk and try setting >>>>>> group_confirm_cancel_timeout=true >>>>>> >>>>>> let me know if it works >>>>>> >>>>>> >>>>>> On Tue, Jul 14, 2009 at 12:02 PM, Phillip Jones >>>>> > wrote: >>>>>> >>>>>>> Hi, >>>>>>> >>>>>>> Thanks for the reply. >>>>>>> >>>>>>> >> Both "leg_timeout" and "leg_progress_timeout" are for controlling >>>>>>> how long to wait prior to the B-leg answering. >>>>>>> >>>>>>> I think this is my point. leg_timeout seems to control how long to >>>>>>> wait prior to the bridge completeing, not the B-leg answering. >>>>>>> >>>>>>> In my situation I am using: >>>>>>> >>>>>>> Session.Execute("set", "group_confirm_key=exec"); >>>>>>> Session.Execute("set", "group_confirm_file=javascript confirm.js"); >>>>>>> >>>>>>> my confirm.js prompts leg_b for 1 (to take the call). If leg_timeout >>>>>>> is set to 10 you have 10 seconds to answer the call AND press 1. >>>>>>> >>>>>>> I just want call_timeout to be satisfied when the call is answered. >>>>>>> Not when the called party presses 1 and the bridge is complete. >>>>>>> >>>>>>> I am new all this so I will work out how to use the pastebin etc. >>>>>>> >>>>>>> Thanks for your help. >>>>>>> >>>>>>> >>>>>>> Phillip Jones >>>>>>> >>>>>>> >>>>>>> >>>>>>> On Tue, Jul 14, 2009 at 12:16 PM, Michael Collins < >>>>>>> msc at freeswitch.org> wrote: >>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> On Tue, Jul 14, 2009 at 8:35 AM, Phillip Jones < >>>>>>>> pjintheusa at gmail.com> wrote: >>>>>>>> >>>>>>>>> Hi there, >>>>>>>>> >>>>>>>>> Here is my call flow: >>>>>>>>> >>>>>>>>> 1) leg A is bridged to leg B >>>>>>>>> 2) when leg B is answered I play a confirm script - "please 1 to >>>>>>>>> accept this call" >>>>>>>>> >>>>>>>>> I only want leg B to ring 20 seconds. BUT when the caller party >>>>>>>>> answers, he should have as long as he needs to press 1. >>>>>>>>> >>>>>>>>> "leg_timeout" seems to be in play until the bridge is completed. I >>>>>>>>> need it to reset when leg b is answered. >>>>>>>>> >>>>>>>> >>>>>>>> Correct. Both "leg_timeout" and "leg_progress_timeout" are for >>>>>>>> controlling how long to wait prior to the B-leg answering. >>>>>>>> (leg_progress_timeout specifies how long to wait for any kind of progress, >>>>>>>> be it early media of some sort, ringing, or an answer.) >>>>>>>> >>>>>>>>> >>>>>>>>> I tried resetting the leg_timeout in the confirm script after leg b >>>>>>>>> is answered. I also tried using leg_progress_timeout. Neither seemed to >>>>>>>>> work. >>>>>>>>> >>>>>>>> >>>>>>>> What exactly are you trying to do? The two variables you've >>>>>>>> mentioned shouldn't have any effect on the call after it has been >>>>>>>> established. >>>>>>>> >>>>>>>>> >>>>>>>>> Any help or suggestions would be welcome. >>>>>>>>> >>>>>>>> >>>>>>>> Could you pastebin your dialplan and a debug log of a call that does >>>>>>>> not work? See this page for some handy tips on using pastebin and collecting >>>>>>>> information for debugging purposes: >>>>>>>> http://wiki.freeswitch.org/wiki/Reporting_Bugs#Debugging_Steps >>>>>>>> >>>>>>>> -MC >>>>>>>> >>>>>>>>> >>>>>>>>> Phillip Jones >>>>>>>>> >>>>>>>>> _______________________________________________ >>>>>>>>> Freeswitch-users mailing list >>>>>>>>> Freeswitch-users at lists.freeswitch.org >>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>> UNSUBSCRIBE: >>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>> http://www.freeswitch.org >>>>>>>>> >>>>>>>>> >>>>>>>> >>>>>>>> _______________________________________________ >>>>>>>> Freeswitch-users mailing list >>>>>>>> Freeswitch-users at lists.freeswitch.org >>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>> UNSUBSCRIBE: >>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>> http://www.freeswitch.org >>>>>>>> >>>>>>>> >>>>>>> >>>>>>> _______________________________________________ >>>>>>> Freeswitch-users mailing list >>>>>>> Freeswitch-users at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE: >>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> -- >>>>>> Anthony Minessale II >>>>>> >>>>>> FreeSWITCH http://www.freeswitch.org/ >>>>>> ClueCon http://www.cluecon.com/ >>>>>> Twitter: http://twitter.com/FreeSWITCH_wire >>>>>> >>>>>> AIM: anthm >>>>>> MSN:anthony_minessale at hotmail.com >>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>>>>> IRC: irc.freenode.net #freeswitch >>>>>> >>>>>> FreeSWITCH Developer Conference >>>>>> 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 >>>> >>>> >>> >>> >>> -- >>> Anthony Minessale II >>> >>> FreeSWITCH http://www.freeswitch.org/ >>> ClueCon http://www.cluecon.com/ >>> Twitter: http://twitter.com/FreeSWITCH_wire >>> >>> AIM: anthm >>> MSN:anthony_minessale at hotmail.com >>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>> IRC: irc.freenode.net #freeswitch >>> >>> FreeSWITCH Developer Conference >>> 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/20090715/440c585b/attachment-0002.html From brian at freeswitch.org Wed Jul 15 07:38:21 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 15 Jul 2009 09:38:21 -0500 Subject: [Freeswitch-users] fs_cli - display variable values? In-Reply-To: <003401ca0558$a944a6b0$fbcdf410$@com> References: <003401ca0558$a944a6b0$fbcdf410$@com> Message-ID: global_getvar will list all globals... "uuid_getvar uuid var" get the var off the uuid. /b On Jul 15, 2009, at 9:29 AM, Lars Zeb wrote: > Is it possible to display the value of a variable in fs_cli? I tried > ?echo ${domain_name}?, but it just echoed what I typed ($ > {domain_name}), rather than its value. > > I do not know how to get help on an individual command from the help > facility in fs_cli. I tried fs_cli itself and also the docs but > could find nothing. > > Thanks, Lars -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090715/c7e02bb7/attachment-0002.html From anthony.minessale at gmail.com Wed Jul 15 07:51:51 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 15 Jul 2009 09:51:51 -0500 Subject: [Freeswitch-users] Preventing disconnect on event_sockte close In-Reply-To: <7DB369DA44D5CF42B6C23BD7D611A50FDACD801F9A@VA3DIAXVS061.RED001.local> References: <7DB369DA44D5CF42B6C23BD7D611A50FDA7E1D5FD6@VA3DIAXVS061.RED001.local> <87f2f3b90907131808l100afe7dwc95d0f624ff38d50@mail.gmail.com> <7DB369DA44D5CF42B6C23BD7D611A50FDA7E1D604D@VA3DIAXVS061.RED001.local> <87f2f3b90907141011g46686396se27fa3cc0320963d@mail.gmail.com> <7DB369DA44D5CF42B6C23BD7D611A50FDABBFA760F@VA3DIAXVS061.RED001.local> <87f2f3b90907141440l5235ab70h7d48ee961367f5e@mail.gmail.com> <7DB369DA44D5CF42B6C23BD7D611A50FDACD801F9A@VA3DIAXVS061.RED001.local> Message-ID: <191c3a030907150751u39c6b483w23f22b7667639d47@mail.gmail.com> You could dynamically transfer it to an empty conference. api uuid_transfer conference:foo inline You may want to consider joining irc and getting some realtime help to avoid a really long thread and report your solutions back here in a follow up email. On Tue, Jul 14, 2009 at 5:03 PM, Weaver, Eric wrote: > Looking a FS to use as Media mixer for conferencing platform. Not really > doing call to call bridging. We really don?t have extensions?. Conferences > are created o the fly as needed. Already have the conf and call control app > done and in production using a different audio mixer, I would like to put FS > in place of it. > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael > Collins > *Sent:* Tuesday, July 14, 2009 3:40 PM > > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Preventing disconnect on event_sockte > close > > > > > > On Tue, Jul 14, 2009 at 2:14 PM, Weaver, Eric > wrote: > > Using api uuid_park 81b65478-70b9-11de-bf26-b962186102f7 before terminating > the NC session works, the call is not disconnected. > > > > Once that is done, I do not receive DTMF and cannot play prompts to the > caller, they seem to be in limbo. I can uuid_kill the call but I need to get > dtmf and play prompts to them. Perhaps Park is not where I need to put > these calls ? > > > > To get a call out of park you need to bridge to it or transfer it to > another extension. If you have an extension you can just uuid_transfer the > parked call's uuid. If you have an existing call's uuid you can use > uuid_bridge to bridge the two together. > > Could you remind me of the application you're building? Just curious what > the big picture is. > -MC > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael > Collins > *Sent:* Tuesday, July 14, 2009 11:12 AM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Preventing disconnect on event_sockte > close > > > > > > On Tue, Jul 14, 2009 at 9:59 AM, Weaver, Eric > wrote: > > Tried it using FreeSWITCH Version 1.0.4pre9 (14036M) and the following > steps > > > > Start netcat > > > > netcat -v -l -p 14000 > > > > place call, socket is connected via dial plan, enter the following. > > > > connect\n\n > > > > sendmsg > > call-command: execute > > execute-app-name: answer\n\n > > > > > > sendmsg > > call-command: execute > > execute-app-name: playback > > execute-app-arg: /home/eweaver/holdmusic.wav > > > > > > > > > > sendmsg > > call-command: execute > > execute-app-name: park > > > Try this: > api uuid_park > > You'll need to capture the uuid at some point and store it. For testing I > just manually copied and pasted it to/from the console screen. > -MC > > > > Console window displays this message: > > > > 2009-07-14 10:42:29.245833 [ERR] switch_ivr.c:654 Cannot park channels that > are under control already. > > > > at this point, ^C in the netcat window. Call is disconnected. > > > > > > Need to be able to park these calls so they can then be handled from an > inbound event socket connection. > > > > eric > > > > > > > > > > > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael > Collins > *Sent:* Monday, July 13, 2009 7:08 PM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Preventing disconnect on event_sockte > close > > > > I don't know if this will work for you but I just tested this scenario with > uuid_park. After parking the call I disconnected the socket and the call > continued. I did the same thing with uuid_transfer. After the transfer I > disconnected the socket and the call continued. > > How are you handling the call and how is the socket getting disconnected? > > -MC > > On Mon, Jul 13, 2009 at 4:35 PM, Weaver, Eric > wrote: > > Using mod_event_socket in outbound mode, is there any to prevent a call > from being disconnected when the outbound socket is closed ? I would like to > handle the initial inbound call using outbound but after the disposition of > the call is determined, close the socket and have that call managed using an > inbound socket instead. > > > > Eric Weaver > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference 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/20090715/2e40fb09/attachment-0002.html From anthony.minessale at gmail.com Wed Jul 15 08:25:03 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 15 Jul 2009 10:25:03 -0500 Subject: [Freeswitch-users] leg_timeout In-Reply-To: <367751820907150738k4b0fae48u4ede9d87ff916324@mail.gmail.com> References: <367751820907140835y322c576en1eb2ba931388d732@mail.gmail.com> <87f2f3b90907140916o3f008a3br26bfca075a9e3d2e@mail.gmail.com> <367751820907141002g3a837a09meabdbc9743c3bee7@mail.gmail.com> <191c3a030907141020l48e8cb2cm6106ff4442ee1bed@mail.gmail.com> <87f2f3b90907141037r260001d4i2c41dcc0e4c25c5@mail.gmail.com> <367751820907141114w56ae9adey4836238740a1ed43@mail.gmail.com> <191c3a030907141127x255f21c5q7d80cf26ea8451a9@mail.gmail.com> <367751820907141135t2c374338y54226c9721cfa739@mail.gmail.com> <367751820907141522o491b40f4jbf79c25f3c053139@mail.gmail.com> <367751820907150738k4b0fae48u4ede9d87ff916324@mail.gmail.com> Message-ID: <191c3a030907150825w283d45c6l3642342f6b11f228@mail.gmail.com> doh, i'm slipping. fixed in tree On Wed, Jul 15, 2009 at 9:38 AM, Phillip Jones wrote: > Hey Guys, > > I took a look at the source that Anthony updated. I see this: > > } else if (!strcasecmp((char *) hi->name, "group_confirm_file")) { > ok = 1; > } else if (!strcasecmp((char *) hi->name, "group_confirm_*clear*_timeout")) > { > ok = 1; > } else if (!strcasecmp((char *) hi->name, "forked_dial")) { > > and: > > if (switch_true(switch_event_get_header(var_event, "group_confirm_*cancel*_timeout"))) > { > oglobals.cancel_timeout = -1; > } > > I updated the *group_confirm_clear_timeout *to * > group_confirm_cancel_timeout* and recompiled and this is now working just > great. > > Thanks very much for incorporating this. It is much appreciated. > > > Phillip Jones > > > > > On Tue, Jul 14, 2009 at 6:22 PM, Phillip Jones wrote: > >> Hi there, >> >> I downloaded the latest trunk, compiled and updated. Still no joy I am >> afraid. >> >> This is the log file in pastebin - http://pastebin.freeswitch.org/9712 >> >> Code in my managed DLL is at: http://pastebin.freeswitch.org/9715 >> >> Dialplan binds to above: >> >> >> >> >> confirm.js is at: http://pastebin.freeswitch.org/9713 >> >> >> Thanks again for your help on this. >> >> >> >> >> >> On Tue, Jul 14, 2009 at 2:35 PM, Phillip Jones wrote: >> >>> Ah - SVN Trunk - thought you meant DID trunk!!! My bad. >>> >>> Sorry - understand now! Will recompile and let you know. >>> >>> On Tue, Jul 14, 2009 at 2:27 PM, Anthony Minessale < >>> anthony.minessale at gmail.com> wrote: >>> >>>> You actually updated your code and recompiled it all too? >>>> This param was added about 30 seconds before I sent you the email. >>>> >>>> >>>> >>>> On Tue, Jul 14, 2009 at 1:14 PM, Phillip Jones wrote: >>>> >>>>> Thanks for your response. >>>>> >>>>> That does not seem to work. Here is my code: >>>>> >>>>> if(Session.Ready()) >>>>> { >>>>> Session.Execute("set", "ignore_early_media=true"); >>>>> Session.Execute("set", "hangup_after_bridge=true"); >>>>> Session.Execute("set", "ringback=${us-ring}"); >>>>> >>>>> Session.Answer(); >>>>> string Caller_ID_Number = >>>>> this.Session.GetVariable("caller_id_number"); >>>>> Session.Execute("set", "group_confirm_key=exec"); >>>>> *Session.Execute("set", "group_confirm_cancel_timeout=true"); >>>>> * Session.Execute("set", "group_confirm_file=javascript >>>>> confirm.js"); >>>>> Session.Execute("bridge", >>>>> "[leg_timeout=20,leg_confirm=y]sofia/gateway/broadvox/6095553828"); >>>>> } >>>>> Session.Hangup("USER_BUSY"); >>>>> >>>>> I also tried *group_confirm_cancel_leg_timeout* just in case. >>>>> >>>>> Am I missing something? >>>>> >>>>> >>>>> >>>>> >>>>> On Tue, Jul 14, 2009 at 1:37 PM, Michael Collins wrote: >>>>> >>>>>> FYI, >>>>>> This has been added to the wiki: >>>>>> >>>>>> http://wiki.freeswitch.org/wiki/Channel_Variables#group_confirm_cancel_timeout >>>>>> >>>>>> -MC >>>>>> >>>>>> >>>>>> On Tue, Jul 14, 2009 at 10:20 AM, Anthony Minessale < >>>>>> anthony.minessale at gmail.com> wrote: >>>>>> >>>>>>> update to trunk and try setting >>>>>>> group_confirm_cancel_timeout=true >>>>>>> >>>>>>> let me know if it works >>>>>>> >>>>>>> >>>>>>> On Tue, Jul 14, 2009 at 12:02 PM, Phillip Jones < >>>>>>> pjintheusa at gmail.com> wrote: >>>>>>> >>>>>>>> Hi, >>>>>>>> >>>>>>>> Thanks for the reply. >>>>>>>> >>>>>>>> >> Both "leg_timeout" and "leg_progress_timeout" are for controlling >>>>>>>> how long to wait prior to the B-leg answering. >>>>>>>> >>>>>>>> I think this is my point. leg_timeout seems to control how long to >>>>>>>> wait prior to the bridge completeing, not the B-leg answering. >>>>>>>> >>>>>>>> In my situation I am using: >>>>>>>> >>>>>>>> Session.Execute("set", "group_confirm_key=exec"); >>>>>>>> Session.Execute("set", "group_confirm_file=javascript confirm.js"); >>>>>>>> >>>>>>>> my confirm.js prompts leg_b for 1 (to take the call). If leg_timeout >>>>>>>> is set to 10 you have 10 seconds to answer the call AND press 1. >>>>>>>> >>>>>>>> I just want call_timeout to be satisfied when the call is answered. >>>>>>>> Not when the called party presses 1 and the bridge is complete. >>>>>>>> >>>>>>>> I am new all this so I will work out how to use the pastebin etc. >>>>>>>> >>>>>>>> Thanks for your help. >>>>>>>> >>>>>>>> >>>>>>>> Phillip Jones >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> On Tue, Jul 14, 2009 at 12:16 PM, Michael Collins < >>>>>>>> msc at freeswitch.org> wrote: >>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> On Tue, Jul 14, 2009 at 8:35 AM, Phillip Jones < >>>>>>>>> pjintheusa at gmail.com> wrote: >>>>>>>>> >>>>>>>>>> Hi there, >>>>>>>>>> >>>>>>>>>> Here is my call flow: >>>>>>>>>> >>>>>>>>>> 1) leg A is bridged to leg B >>>>>>>>>> 2) when leg B is answered I play a confirm script - "please 1 to >>>>>>>>>> accept this call" >>>>>>>>>> >>>>>>>>>> I only want leg B to ring 20 seconds. BUT when the caller party >>>>>>>>>> answers, he should have as long as he needs to press 1. >>>>>>>>>> >>>>>>>>>> "leg_timeout" seems to be in play until the bridge is completed. I >>>>>>>>>> need it to reset when leg b is answered. >>>>>>>>>> >>>>>>>>> >>>>>>>>> Correct. Both "leg_timeout" and "leg_progress_timeout" are for >>>>>>>>> controlling how long to wait prior to the B-leg answering. >>>>>>>>> (leg_progress_timeout specifies how long to wait for any kind of progress, >>>>>>>>> be it early media of some sort, ringing, or an answer.) >>>>>>>>> >>>>>>>>>> >>>>>>>>>> I tried resetting the leg_timeout in the confirm script after leg >>>>>>>>>> b is answered. I also tried using leg_progress_timeout. Neither seemed to >>>>>>>>>> work. >>>>>>>>>> >>>>>>>>> >>>>>>>>> What exactly are you trying to do? The two variables you've >>>>>>>>> mentioned shouldn't have any effect on the call after it has been >>>>>>>>> established. >>>>>>>>> >>>>>>>>>> >>>>>>>>>> Any help or suggestions would be welcome. >>>>>>>>>> >>>>>>>>> >>>>>>>>> Could you pastebin your dialplan and a debug log of a call that >>>>>>>>> does not work? See this page for some handy tips on using pastebin and >>>>>>>>> collecting information for debugging purposes: >>>>>>>>> http://wiki.freeswitch.org/wiki/Reporting_Bugs#Debugging_Steps >>>>>>>>> >>>>>>>>> -MC >>>>>>>>> >>>>>>>>>> >>>>>>>>>> Phillip Jones >>>>>>>>>> >>>>>>>>>> _______________________________________________ >>>>>>>>>> Freeswitch-users mailing list >>>>>>>>>> Freeswitch-users at lists.freeswitch.org >>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>>> UNSUBSCRIBE: >>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>>> http://www.freeswitch.org >>>>>>>>>> >>>>>>>>>> >>>>>>>>> >>>>>>>>> _______________________________________________ >>>>>>>>> Freeswitch-users mailing list >>>>>>>>> Freeswitch-users at lists.freeswitch.org >>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>> UNSUBSCRIBE: >>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>> http://www.freeswitch.org >>>>>>>>> >>>>>>>>> >>>>>>>> >>>>>>>> _______________________________________________ >>>>>>>> Freeswitch-users mailing list >>>>>>>> Freeswitch-users at lists.freeswitch.org >>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>> UNSUBSCRIBE: >>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>> http://www.freeswitch.org >>>>>>>> >>>>>>>> >>>>>>> >>>>>>> >>>>>>> -- >>>>>>> Anthony Minessale II >>>>>>> >>>>>>> FreeSWITCH http://www.freeswitch.org/ >>>>>>> ClueCon http://www.cluecon.com/ >>>>>>> Twitter: http://twitter.com/FreeSWITCH_wire >>>>>>> >>>>>>> AIM: anthm >>>>>>> MSN:anthony_minessale at hotmail.com >>>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>>>>>> IRC: irc.freenode.net #freeswitch >>>>>>> >>>>>>> FreeSWITCH Developer Conference >>>>>>> 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 >>>>> >>>>> >>>> >>>> >>>> -- >>>> Anthony Minessale II >>>> >>>> FreeSWITCH http://www.freeswitch.org/ >>>> ClueCon http://www.cluecon.com/ >>>> Twitter: http://twitter.com/FreeSWITCH_wire >>>> >>>> AIM: anthm >>>> MSN:anthony_minessale at hotmail.com >>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>>> IRC: irc.freenode.net #freeswitch >>>> >>>> FreeSWITCH Developer Conference >>>> 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/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference 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/20090715/bacf2162/attachment-0002.html From Kareem.Hamdy at trustvesta.com Wed Jul 15 08:29:49 2009 From: Kareem.Hamdy at trustvesta.com (Kareem Hamdy) Date: Wed, 15 Jul 2009 08:29:49 -0700 Subject: [Freeswitch-users] Freeswitch-users Digest, Vol 37, Issue 90 In-Reply-To: References: Message-ID: <1134625859513549B3B943E0133490E202AC61485E@TDCP-EXSTORE-01.ad.trustvesta.com> Thanks Michael, but I'm setting up a T1, not a PRI. I should be able to use all 24 channels. ? -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of freeswitch-users-request at lists.freeswitch.org Sent: Wednesday, July 15, 2009 1:06 AM To: freeswitch-users at lists.freeswitch.org Subject: Freeswitch-users Digest, Vol 37, Issue 90 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: OpenZAP and FreeSWITCH w/ Sangoma (Michael Collins) 2. GXW4104 & FreeSwitch (DigiLord) 3. Re: GXW4104 & FreeSwitch (Brian West) 4. SIP Trace Option at Runtime (Muhammad Shahzad) 5. Re: SIP Trace Option at Runtime (Jason White) 6. Re: Get voicemail messages (Eli Hayun) 7. How to set the IVR of VM menu?? (Brad Tuan) ---------------------------------------------------------------------- Message: 1 Date: Tue, 14 Jul 2009 17:24:18 -0700 From: Michael Collins Subject: Re: [Freeswitch-users] OpenZAP and FreeSWITCH w/ Sangoma To: freeswitch-users at lists.freeswitch.org Message-ID: <87f2f3b90907141724q2735fac1jdacea3994db62782 at mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" See inline comments On Tue, Jul 14, 2009 at 5:04 PM, Kareem Hamdy wrote: > Hello: > > I would like to connect a Sangoma T1 card to FreeSWITCH. Most of the docs > I see pertain to a PRI. When I leave out the d-chan notation, I get errors > regarding not able to get the d-chan up and running in the CLI. > > Here's my info: > > [span wanpipe T1] > trunk_type => t1 > b-channel => 1:1-24 b-channel => 1:1-23 d-channel => 1:24 > > > [span wanpipe T2] > trunk_type => t1 > b-channel => 2:1-24 > set up like span 1 example > > ---- > > #================================================ > # WANPIPE1 Configuration File > #================================================ > # > # Date: Wed Dec 6 20:29:03 UTC 2006 > # > # Note: This file was generated automatically > # by /usr/local/sbin/setup-sangoma program. > # > # If you want to edit this file, it is > # recommended that you use wancfg program > # to do so. > #================================================ > # Sangoma Technologies Inc. > #================================================ > > [devices] > wanpipe1 = WAN_AFT_TE1, Comment > > [interfaces] > w1g1 = wanpipe1, , TDM_VOICE_API, Comment > > [wanpipe1] > CARD_TYPE = AFT > S514CPU = A > CommPort = PRI > AUTO_PCISLOT = NO > PCISLOT = 1 > PCIBUS = 6 > FE_MEDIA = T1 > FE_LCODE = B8ZS > FE_FRAME = ESF > FE_LINE = 1 > TE_CLOCK = NORMAL > TE_REF_CLOCK = 0 > > TE_HIGHIMPEDANCE = NO > LBO = 0DB > FE_TXTRISTATE = NO > MTU = 1500 > UDPPORT = 9000 > TTL = 255 > IGNORE_FRONT_END = NO > TDMV_SPAN = 1 > TDMV_DCHAN = 0 TDMV_DCAHN = 24 > > TDMV_HW_DTMF = YES > TDMV_HW_FAX_DETECT = NO > > [w1g1] > ACTIVE_CH = ALL > TDMV_HWEC = YES > MTU = 80 > > > --- > > > > In freeSWITCH's openzap.conf.xml, I've tried pri_span as well as analog. > > I cannot find a straight up T1 wiki anywhere. Would someone please provide > an example? > > > Thanks, > Kareem > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090714/da16d89e/attachment-0001.html ------------------------------ Message: 2 Date: Tue, 14 Jul 2009 19:20:19 -0700 From: DigiLord Subject: [Freeswitch-users] GXW4104 & FreeSwitch To: freeswitch-users at lists.freeswitch.org Message-ID: <8DC39E34-A395-42D5-B299-070605A2DCEE at me.com> Content-Type: text/plain; charset=us-ascii; format=flowed; delsp=yes Hello all, I am getting my feet wet with FreeSwitch by migrating my Asterisk box over. I have run into a few things that I am not sure how to accomplish. I have a Grandstream GXW4104 with one analog line connected. I have it connected and I am able to receive calls on my Polycom 501 (ext 2101) that is registered to the FreeSwitch server. The one problem is that CallerID is not the CallerID from the caller, it's the CallerID from the Grandstream device (ext 2100). On the same device there is HORRIBLE echo. I have set echo cancellation on the device to enabled and disabled to no avail. Under Asterisk there was no echo. I setup the device as a provider. Was that the right way to accomplish connecting this device to FS? Is there a way to enable sending an e-mail containing my voicemail messages like Asterisk does? Thanks in advance for any help you can give! Dan ------------------------------ Message: 3 Date: Tue, 14 Jul 2009 21:31:23 -0500 From: Brian West Subject: Re: [Freeswitch-users] GXW4104 & FreeSwitch To: freeswitch-users at lists.freeswitch.org Message-ID: <085EE9F4-A513-45FD-89E9-C66A0BE3715F at freeswitch.org> Content-Type: text/plain; charset="us-ascii" On Jul 14, 2009, at 9:20 PM, DigiLord wrote: > Hello all, > I am getting my feet wet with FreeSwitch by migrating my Asterisk box > over. I have run into a few things that I am not sure how to > accomplish. > > I have a Grandstream GXW4104 with one analog line connected. I have > it connected and I am able to receive calls on my Polycom 501 (ext > 2101) that is registered to the FreeSwitch server. The one problem is > that CallerID is not the CallerID from the caller, it's the CallerID > from the Grandstream device (ext 2100). How is the callerid passed on this device? > On the same device there is HORRIBLE echo. I have set echo > cancellation on the device to enebled and disabled to no avail. Under > Asterisk there was no echo. If it didn't have echo on asterisk it shouldn't have echo on FreeSWITCH, Can you describe the echo better? Are you using speaker phone? What codecs? > > > I setup the device as a provider. Was that the right way to > accomplish connecting this device to FS? > > Is there a way to enable sending an e-mail containing my voicemail > messages like Asterisk does? Yes check the mod_voicemail page on the wiki. /b > > > Thanks in advance for any help you can give! > > Dan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090714/eed9a9d7/attachment-0001.html ------------------------------ Message: 4 Date: Wed, 15 Jul 2009 10:19:48 +0600 From: Muhammad Shahzad Subject: [Freeswitch-users] SIP Trace Option at Runtime To: freeswitch-users at lists.freeswitch.org Message-ID: Content-Type: text/plain; charset="utf-8" Hi, Is there any CLI command to enable / disable SIP packet trace at runtime. I do know an option in SIP profile which enables / disable SIP trace but it to apply it i have reload mod_sofia, which at many times fail due to a running call. Thank you. -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090715/ca7ce7b2/attachment-0001.html ------------------------------ Message: 5 Date: Wed, 15 Jul 2009 14:32:25 +1000 From: Jason White Subject: Re: [Freeswitch-users] SIP Trace Option at Runtime To: freeswitch-users at lists.freeswitch.org Message-ID: <20090715043225.GA21117 at jdc.jasonjgw.net> Content-Type: text/plain; charset=us-ascii Muhammad Shahzad wrote: > Is there any CLI command to enable / disable SIP packet trace at runtime. sofia profile siptrace on sofia profile siptrace off sofia help would have answered your question. ------------------------------ Message: 6 Date: Wed, 15 Jul 2009 07:49:07 +0300 From: Eli Hayun Subject: Re: [Freeswitch-users] Get voicemail messages To: "freeswitch-users at lists.freeswitch.org" Message-ID: <4A5D5FC3.4050701 at savion.huji.ac.il> Content-Type: text/plain; charset=ISO-8859-1 did you bind your lua script to directory lookups in addition to the dialplan? On Tue, Jul 14, 2009 at 7:02 AM, Eli Hayun wrote: > > Hi > > I am not using fixed xml files for the extension registration. I have > > LUA script to return an XML string to FS. > > Everything goes fine until I am trying to get the voice messages. > > When am entering my id, FS (or voicemail module) try to get the xml for > > that id, but it cant find it. My lua script did NOT recieved any xml > > request at that point. > > What should I do to solve the problem. > > > > Thanks > > Eli Hayun > > > Yes I did bind it: my lua.conf.xml is like this When an extension tried to register, I have no problem. But when I want to use VoiceMail to retrieve my messeges, I got a problem. Here is the partial log: 2009-07-15 07:44:49.373089 [INFO] mod_dialplan_xml.c:252 Processing Phone2->*98 in context default 2009-07-15 07:44:49.386466 [NOTICE] mod_dptools.c:649 Channel [sofia/internal/80671 at 132.64.3.86] has been answered 2009-07-15 07:44:51.933664 [WARNING] mod_voicemail.c:2072 Can't find user [80671 at 132.64.3.86] 2009-07-15 07:44:52.533435 [NOTICE] switch_core_state_machine.c:179 Hangup sofia/internal/80671 at 132.64.3.86 [CS_EXECUTE] [NORMAL_CLEARING] 2009-07-15 07:44:52.545698 [NOTICE] switch_core_session.c:1085 Session 3 (sofia/internal/80671 at 132.64.3.86) Ended 2009-07-15 07:44:52.545698 [NOTICE] switch_core_session.c:1087 Close Channel sofia/internal/80671 at 132.64.3.86 [CS_DESTROY] ------------------------------ Message: 7 Date: Wed, 15 Jul 2009 16:05:24 +0800 From: Brad Tuan Subject: [Freeswitch-users] How to set the IVR of VM menu?? To: freeswitch-users Message-ID: Content-Type: text/plain; charset="iso-8859-1" How to set the date format , and the IVR flow ........?? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090715/c35eaf18/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 37, Issue 90 ************************************************ From brian at freeswitch.org Wed Jul 15 08:34:12 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 15 Jul 2009 10:34:12 -0500 Subject: [Freeswitch-users] Freeswitch-users Digest, Vol 37, Issue 90 In-Reply-To: <1134625859513549B3B943E0133490E202AC61485E@TDCP-EXSTORE-01.ad.trustvesta.com> References: <1134625859513549B3B943E0133490E202AC61485E@TDCP-EXSTORE-01.ad.trustvesta.com> Message-ID: <2E49E0A3-397F-4000-AADF-85796D18E301@freeswitch.org> Are you trying to do E&M? /b On Jul 15, 2009, at 10:29 AM, Kareem Hamdy wrote: > Thanks Michael, but I'm setting up a T1, not a PRI. I should be > able to use all 24 channels. From saeedahmad1981 at gmail.com Wed Jul 15 08:49:52 2009 From: saeedahmad1981 at gmail.com (Saeed Ahmad) Date: Wed, 15 Jul 2009 17:49:52 +0200 Subject: [Freeswitch-users] Originate in Dial plan In-Reply-To: <87f2f3b90907140048m7e1eea18x46e9418e43fb4007@mail.gmail.com> References: <8ccbff060907120959ma7c0ey22adeae196f872c3@mail.gmail.com> <87f2f3b90907131810q219c04f3ra385e0e1d252d31c@mail.gmail.com> <8ccbff060907132130l329f7e1at87a16da0c2108e2@mail.gmail.com> <87f2f3b90907140048m7e1eea18x46e9418e43fb4007@mail.gmail.com> Message-ID: Call back is quite cool where users are in areas where no callshops, internet and other calling facilities are available except mobile phones, users will pay both calls. There might be some other usages as well. - Saeed On Tue, Jul 14, 2009 at 9:48 AM, Michael Collins wrote: > > > On Mon, Jul 13, 2009 at 9:30 PM, Dome Charoenyost wrote: > >> 2009/7/14 Michael Collins : >> > What phone number do you call back? I mean, how do you know what the >> > customer's number is? Do you go by the caller id number? >> yes callback to caller id >> > > Okay, here's a dialplan snippet that I used to successfully do the > autocallback. In my case I used ext 1001 as the customer and portaudio as > the "agent" if you get my meaning. Extension 1001 dials 9902, hangs up, and > immediately the api_hangup_hook's originate command is executed. In this > case it calls portaudio/auto_answer for the A-leg and user/1001 as the > B-leg. I don't claim that it's the prettiest thing in the world but it > definitely works. You'll need to adjust according to your specific > situation. > > > > > > > > > > > > > > > > > > > > > Let us know how it goes. BTW, what is the reason for this type of scenario? > Just curious. > -MC > >> >> >> > >> > -MC >> > >> > On Sun, Jul 12, 2009 at 9:59 AM, Dome Charoenyost >> wrote: >> >> >> >> Dear sir, >> >> I want to user dialplan callback to customer. is posible to >> >> to this is dialplan XML ? >> >> Now i use javascript. >> >> my call flow. >> >> 1. customer call >> >> 2. FS rining and wait until customer hangup >> >> 3. callback to customer number >> >> >> >> >> >> Best Regards. >> >> >> >> Dome C. >> >> >> >> _______________________________________________ >> >> Freeswitch-users mailing list >> >> Freeswitch-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> > >> > >> > _______________________________________________ >> > Freeswitch-users mailing list >> > Freeswitch-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090715/aacfbd88/attachment-0001.html From mike at jerris.com Wed Jul 15 08:56:39 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 15 Jul 2009 11:56:39 -0400 Subject: [Freeswitch-users] How to set the IVR of VM menu?? In-Reply-To: References: Message-ID: <930DA7E7-DD57-44D2-914B-9D301041EEC3@jerris.com> Please try looking on the wiki, this and many other questions should be answered for you there. Mike On Jul 15, 2009, at 4:05 AM, Brad Tuan wrote: > How to set the date format , and the IVR flow ........?? > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.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 Wed Jul 15 09:03:26 2009 From: saeedahmad1981 at gmail.com (Saeed Ahmad) Date: Wed, 15 Jul 2009 18:03:26 +0200 Subject: [Freeswitch-users] SIP Trace Option at Runtime In-Reply-To: <20090715043225.GA21117@jdc.jasonjgw.net> References: <20090715043225.GA21117@jdc.jasonjgw.net> Message-ID: and how can we do the same for console message? I noticed that if then even we press F8 on cli then it won't turn on the log message, is there anyway to enable them even loglevel is set to 'err' - Saeed On Wed, Jul 15, 2009 at 6:32 AM, Jason White wrote: > Muhammad Shahzad wrote: > > Is there any CLI command to enable / disable SIP packet trace at > runtime. > > sofia profile siptrace on > sofia profile siptrace off > > sofia help would have answered your question. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090715/93a0c247/attachment-0001.html From brian at freeswitch.org Wed Jul 15 09:06:30 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 15 Jul 2009 11:06:30 -0500 Subject: [Freeswitch-users] SIP Trace Option at Runtime In-Reply-To: References: <20090715043225.GA21117@jdc.jasonjgw.net> Message-ID: where are you setting the loglevel to err at? /b On Jul 15, 2009, at 11:03 AM, Saeed Ahmad wrote: > and how can we do the same for console message? > > I noticed that if > > > > then even we press F8 on cli then it won't turn on the log message, > is there anyway to enable them even loglevel is set to 'err' > > - Saeed > From saeedahmad1981 at gmail.com Wed Jul 15 09:08:14 2009 From: saeedahmad1981 at gmail.com (Saeed Ahmad) Date: Wed, 15 Jul 2009 18:08:14 +0200 Subject: [Freeswitch-users] SIP Trace Option at Runtime In-Reply-To: References: <20090715043225.GA21117@jdc.jasonjgw.net> Message-ID: switch.conf.xml On Wed, Jul 15, 2009 at 6:06 PM, Brian West wrote: > where are you setting the loglevel to err at? > > /b > > On Jul 15, 2009, at 11:03 AM, Saeed Ahmad wrote: > > > and how can we do the same for console message? > > > > I noticed that if > > > > > > > > then even we press F8 on cli then it won't turn on the log message, > > is there anyway to enable them even loglevel is set to 'err' > > > > - Saeed > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090715/d1ed0522/attachment-0001.html From brian at freeswitch.org Wed Jul 15 09:11:15 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 15 Jul 2009 11:11:15 -0500 Subject: [Freeswitch-users] SIP Trace Option at Runtime In-Reply-To: References: <20090715043225.GA21117@jdc.jasonjgw.net> Message-ID: <60FEB0EB-321A-431F-A8A2-1D92253D7040@freeswitch.org> you're lowering the core loglevel so you'll fsctl logelvel debug then console loglevel debug../. If you wish to start in err then edit console.conf.xml /b On Jul 15, 2009, at 11:08 AM, Saeed Ahmad wrote: > switch.conf.xml > From mrene_lists at avgs.ca Wed Jul 15 09:15:42 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Wed, 15 Jul 2009 12:15:42 -0400 Subject: [Freeswitch-users] SIP Trace Option at Runtime In-Reply-To: References: <20090715043225.GA21117@jdc.jasonjgw.net> Message-ID: Thats the global loglevel.... here's a summary of how logging works: When a log message is issued, FS checks the global level (fsctl loglevel xxx, or the one in switch.conf.xml) and discards anything less important than this loglevel. Once this is passed, the module which you use control FS will filter the logs it displays for you. This can be mod_console, mod_event_socket (for fs_cli), and even mod_logfile. It is clear that most messages wont make it to your screen if your global loglevel is at error. Change your switch.conf.xml level to debug and then you'll see them. Pressing F8 only control whatever you are using to connect to FS, not the global level. Let me know if that makes any sense to you, and if it does, a little documentation on the wiki could be of great use. Regards,, Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca Am 15-Jul-09 um 12:08 PM schrieb Saeed Ahmad: > switch.conf.xml > > > On Wed, Jul 15, 2009 at 6:06 PM, Brian West > wrote: > where are you setting the loglevel to err at? > > /b > > On Jul 15, 2009, at 11:03 AM, Saeed Ahmad wrote: > > > and how can we do the same for console message? > > > > I noticed that if > > > > > > > > then even we press F8 on cli then it won't turn on the log message, > > is there anyway to enable them even loglevel is set to 'err' > > > > - Saeed > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090715/09908ba1/attachment-0001.html From mrene_lists at avgs.ca Wed Jul 15 09:16:57 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Wed, 15 Jul 2009 12:16:57 -0400 Subject: [Freeswitch-users] SIP Trace Option at Runtime In-Reply-To: References: <20090715043225.GA21117@jdc.jasonjgw.net> Message-ID: <45002ACD-36EC-4BBE-8D8F-9F74823938E7@avgs.ca> Oh and the default level for "siptrace" is CONSOLE, you can change it runtime (if you want to log it into log files) by doing sofia tracelevel [xxx] (where [xxx] is the loglevel) Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca Am 15-Jul-09 um 12:08 PM schrieb Saeed Ahmad: > switch.conf.xml > > > On Wed, Jul 15, 2009 at 6:06 PM, Brian West > wrote: > where are you setting the loglevel to err at? > > /b > > On Jul 15, 2009, at 11:03 AM, Saeed Ahmad wrote: > > > and how can we do the same for console message? > > > > I noticed that if > > > > > > > > then even we press F8 on cli then it won't turn on the log message, > > is there anyway to enable them even loglevel is set to 'err' > > > > - Saeed > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090715/b0f321ea/attachment-0001.html From msc at freeswitch.org Wed Jul 15 09:25:56 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 15 Jul 2009 09:25:56 -0700 Subject: [Freeswitch-users] leg_timeout In-Reply-To: <367751820907150738k4b0fae48u4ede9d87ff916324@mail.gmail.com> References: <367751820907140835y322c576en1eb2ba931388d732@mail.gmail.com> <87f2f3b90907140916o3f008a3br26bfca075a9e3d2e@mail.gmail.com> <367751820907141002g3a837a09meabdbc9743c3bee7@mail.gmail.com> <191c3a030907141020l48e8cb2cm6106ff4442ee1bed@mail.gmail.com> <87f2f3b90907141037r260001d4i2c41dcc0e4c25c5@mail.gmail.com> <367751820907141114w56ae9adey4836238740a1ed43@mail.gmail.com> <191c3a030907141127x255f21c5q7d80cf26ea8451a9@mail.gmail.com> <367751820907141135t2c374338y54226c9721cfa739@mail.gmail.com> <367751820907141522o491b40f4jbf79c25f3c053139@mail.gmail.com> <367751820907150738k4b0fae48u4ede9d87ff916324@mail.gmail.com> Message-ID: <87f2f3b90907150925k448169c0iebf34d9def2aecd2@mail.gmail.com> On Wed, Jul 15, 2009 at 7:38 AM, Phillip Jones wrote: > Hey Guys, > > I took a look at the source that Anthony updated. I see this: > > } else if (!strcasecmp((char *) hi->name, "group_confirm_file")) { > ok = 1; > } else if (!strcasecmp((char *) hi->name, "group_confirm_*clear*_timeout")) > { > ok = 1; > } else if (!strcasecmp((char *) hi->name, "forked_dial")) { > > and: > > if (switch_true(switch_event_get_header(var_event, "group_confirm_*cancel*_timeout"))) > { > oglobals.cancel_timeout = -1; > } > > I updated the *group_confirm_clear_timeout *to * > group_confirm_cancel_timeout* and recompiled and this is now working just > great. > > Thanks very much for incorporating this. It is much appreciated. > > > Phillip Jones > > Phillip, Thank you very much for taking the time and initiative to dig a little. The devs definitely appreciate it when community members roll up their sleeves and do some investigative work. Karma++ for you! :) -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090715/f1618b2d/attachment-0001.html From woof at iwoof.org Wed Jul 15 09:42:41 2009 From: woof at iwoof.org (Andy Spitzer) Date: Wed, 15 Jul 2009 12:42:41 -0400 Subject: [Freeswitch-users] copy and past "oops" in mod_event_socket.c Message-ID: Woof! Too simple to open a JIRA with a patch (and it actually works as written): 1804: } else if (!strncasecmp(cmd, "nolinger", 6)) { That should be an 8 as nolinger is 8 characters long. --Woof! From saeedahmad1981 at gmail.com Wed Jul 15 09:45:12 2009 From: saeedahmad1981 at gmail.com (Saeed Ahmad) Date: Wed, 15 Jul 2009 18:45:12 +0200 Subject: [Freeswitch-users] SIP Trace Option at Runtime In-Reply-To: <45002ACD-36EC-4BBE-8D8F-9F74823938E7@avgs.ca> References: <20090715043225.GA21117@jdc.jasonjgw.net> <45002ACD-36EC-4BBE-8D8F-9F74823938E7@avgs.ca> Message-ID: still a little confusion, what should be set in switch.conf.xml and console.conf.xml when system is in production and we only want to enable/disable logs on runtime. please also consider log files freeswitch.log & freeswitch.xml.fsxml, we don't want to log debug here. So main goal is without restarting FS we should be able to enable/disable logs on runtime (when logs are enabled then its ok to write logs in log files, but on disable it should stop writing) and everything on runtime. Thanks - Saeed On Wed, Jul 15, 2009 at 6:16 PM, Mathieu Rene wrote: > Oh and the default level for "siptrace" is CONSOLE, you can change it > runtime (if you want to log it into log files) by doing sofia tracelevel > [xxx] (where [xxx] is the loglevel) > Mathieu Rene > Avant-Garde Solutions Inc > Office: + 1 (514) 664-1044 x100 > Cell: +1 (514) 664-1044 x200 > mrene at avgs.ca > > > > Am 15-Jul-09 um 12:08 PM schrieb Saeed Ahmad: > > switch.conf.xml > > > On Wed, Jul 15, 2009 at 6:06 PM, Brian West wrote: > >> where are you setting the loglevel to err at? >> >> /b >> >> On Jul 15, 2009, at 11:03 AM, Saeed Ahmad wrote: >> >> > and how can we do the same for console message? >> > >> > I noticed that if >> > >> > >> > >> > then even we press F8 on cli then it won't turn on the log message, >> > is there anyway to enable them even loglevel is set to 'err' >> > >> > - Saeed >> > >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090715/0ff94499/attachment-0001.html From mrene_lists at avgs.ca Wed Jul 15 09:45:48 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Wed, 15 Jul 2009 12:45:48 -0400 Subject: [Freeswitch-users] copy and past "oops" in mod_event_socket.c In-Reply-To: References: Message-ID: Thx Committed revision 14258. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca Am 15-Jul-09 um 12:42 PM schrieb Andy Spitzer: > nolinger From brian at freeswitch.org Wed Jul 15 09:51:42 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 15 Jul 2009 11:51:42 -0500 Subject: [Freeswitch-users] SIP Trace Option at Runtime In-Reply-To: References: <20090715043225.GA21117@jdc.jasonjgw.net> <45002ACD-36EC-4BBE-8D8F-9F74823938E7@avgs.ca> Message-ID: fsctl loglevel xxx - controls core loglevel console loglevel xxx - controls console loglevel If you fsctl loglevel 0 you basically turn it of and console loglevel 8 won't work anymore because you turned it off in the core. /b On Jul 15, 2009, at 11:45 AM, Saeed Ahmad wrote: > still a little confusion, > > what should be set in switch.conf.xml and console.conf.xml when > system is in production and we only want to enable/disable logs on > runtime. please also consider log files freeswitch.log & > freeswitch.xml.fsxml, we don't want to log debug here. > > So main goal is without restarting FS we should be able to enable/ > disable logs on runtime (when logs are enabled then its ok to write > logs in log files, but on disable it should stop writing) and > everything on runtime. > > Thanks > - Saeed > From larclap at yahoo.com Wed Jul 15 09:59:28 2009 From: larclap at yahoo.com (Lars Zeb) Date: Wed, 15 Jul 2009 09:59:28 -0700 Subject: [Freeswitch-users] contrib directory location Message-ID: <007301ca056d$9d6cd550$d8467ff0$@com> What is the address of the contrib directory? I would like to download it and its contents for study. Thanks, Lars -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090715/a53b9d5e/attachment-0001.html From brian at freeswitch.org Wed Jul 15 10:05:46 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 15 Jul 2009 12:05:46 -0500 Subject: [Freeswitch-users] contrib directory location In-Reply-To: <007301ca056d$9d6cd550$d8467ff0$@com> References: <007301ca056d$9d6cd550$d8467ff0$@com> Message-ID: <3D133974-DCA3-40FB-9D83-03E3B4BAD4CB@freeswitch.org> If you have the freeswitch source tarball or svn check out just cd contrib /b On Jul 15, 2009, at 11:59 AM, Lars Zeb wrote: > What is the address of the contrib directory? I would like to > download it and its contents for study. > > Thanks, Lars > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090715/c2921277/attachment-0001.html From saeedahmad1981 at gmail.com Wed Jul 15 10:28:42 2009 From: saeedahmad1981 at gmail.com (Saeed Ahmad) Date: Wed, 15 Jul 2009 19:28:42 +0200 Subject: [Freeswitch-users] SIP Trace Option at Runtime In-Reply-To: References: <20090715043225.GA21117@jdc.jasonjgw.net> <45002ACD-36EC-4BBE-8D8F-9F74823938E7@avgs.ca> Message-ID: fsctl loglevel 8 console loglevel now freeswitch.log is showing what i wanted but nothing on console, i am connected to ./fs_cli in vars.conf.xml i think its nothing to do with that! On Wed, Jul 15, 2009 at 6:51 PM, Brian West wrote: > fsctl loglevel xxx - controls core loglevel > console loglevel xxx - controls console loglevel > > If you fsctl loglevel 0 you basically turn it of and console loglevel > 8 won't work anymore because you turned it off in the core. > > /b > > > On Jul 15, 2009, at 11:45 AM, Saeed Ahmad wrote: > > > still a little confusion, > > > > what should be set in switch.conf.xml and console.conf.xml when > > system is in production and we only want to enable/disable logs on > > runtime. please also consider log files freeswitch.log & > > freeswitch.xml.fsxml, we don't want to log debug here. > > > > So main goal is without restarting FS we should be able to enable/ > > disable logs on runtime (when logs are enabled then its ok to write > > logs in log files, but on disable it should stop writing) and > > everything on runtime. > > > > Thanks > > - Saeed > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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/20090715/88f5efe1/attachment-0001.html From saeedahmad1981 at gmail.com Wed Jul 15 10:29:01 2009 From: saeedahmad1981 at gmail.com (Saeed Ahmad) Date: Wed, 15 Jul 2009 19:29:01 +0200 Subject: [Freeswitch-users] SIP Trace Option at Runtime In-Reply-To: References: <20090715043225.GA21117@jdc.jasonjgw.net> <45002ACD-36EC-4BBE-8D8F-9F74823938E7@avgs.ca> Message-ID: console loglevel 8 too On Wed, Jul 15, 2009 at 7:28 PM, Saeed Ahmad wrote: > fsctl loglevel 8 > console loglevel > now freeswitch.log is showing what i wanted but nothing on console, i am > connected to ./fs_cli > > in vars.conf.xml > > > > i think its nothing to do with that! > > On Wed, Jul 15, 2009 at 6:51 PM, Brian West wrote: > >> fsctl loglevel xxx - controls core loglevel >> console loglevel xxx - controls console loglevel >> >> If you fsctl loglevel 0 you basically turn it of and console loglevel >> 8 won't work anymore because you turned it off in the core. >> >> /b >> >> >> On Jul 15, 2009, at 11:45 AM, Saeed Ahmad wrote: >> >> > still a little confusion, >> > >> > what should be set in switch.conf.xml and console.conf.xml when >> > system is in production and we only want to enable/disable logs on >> > runtime. please also consider log files freeswitch.log & >> > freeswitch.xml.fsxml, we don't want to log debug here. >> > >> > So main goal is without restarting FS we should be able to enable/ >> > disable logs on runtime (when logs are enabled then its ok to write >> > logs in log files, but on disable it should stop writing) and >> > everything on runtime. >> > >> > Thanks >> > - Saeed >> > >> >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users at 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/20090715/c69d84dc/attachment-0001.html From saeedahmad1981 at gmail.com Wed Jul 15 10:41:39 2009 From: saeedahmad1981 at gmail.com (Saeed Ahmad) Date: Wed, 15 Jul 2009 19:41:39 +0200 Subject: [Freeswitch-users] SIP Trace Option at Runtime In-Reply-To: References: <20090715043225.GA21117@jdc.jasonjgw.net> <45002ACD-36EC-4BBE-8D8F-9F74823938E7@avgs.ca> Message-ID: it works! thanks On Wed, Jul 15, 2009 at 7:29 PM, Saeed Ahmad wrote: > console loglevel 8 > too > > On Wed, Jul 15, 2009 at 7:28 PM, Saeed Ahmad wrote: > >> fsctl loglevel 8 >> console loglevel >> now freeswitch.log is showing what i wanted but nothing on console, i am >> connected to ./fs_cli >> >> in vars.conf.xml >> >> >> >> i think its nothing to do with that! >> >> On Wed, Jul 15, 2009 at 6:51 PM, Brian West wrote: >> >>> fsctl loglevel xxx - controls core loglevel >>> console loglevel xxx - controls console loglevel >>> >>> If you fsctl loglevel 0 you basically turn it of and console loglevel >>> 8 won't work anymore because you turned it off in the core. >>> >>> /b >>> >>> >>> On Jul 15, 2009, at 11:45 AM, Saeed Ahmad wrote: >>> >>> > still a little confusion, >>> > >>> > what should be set in switch.conf.xml and console.conf.xml when >>> > system is in production and we only want to enable/disable logs on >>> > runtime. please also consider log files freeswitch.log & >>> > freeswitch.xml.fsxml, we don't want to log debug here. >>> > >>> > So main goal is without restarting FS we should be able to enable/ >>> > disable logs on runtime (when logs are enabled then its ok to write >>> > logs in log files, but on disable it should stop writing) and >>> > everything on runtime. >>> > >>> > Thanks >>> > - Saeed >>> > >>> >>> >>> _______________________________________________ >>> Freeswitch-users mailing list >>> Freeswitch-users at 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/20090715/59c5a959/attachment-0001.html From larclap at yahoo.com Wed Jul 15 11:33:33 2009 From: larclap at yahoo.com (Lars Zeb) Date: Wed, 15 Jul 2009 11:33:33 -0700 Subject: [Freeswitch-users] Dialplan with lua - error missing closing angle bracket? Message-ID: <00be01ca057a$c2572340$470569c0$@com> I copied the action element from http://wiki.freeswitch.org/wiki/Lua, "Sample Dialplan". When I try to reloadxml, the cli tells me that there is a missing right angle bracket. +OK [[error near line 3130]: missing >] Do the docs need updating or have I totally blown it? Thanks, Lars -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090715/d9a4ee37/attachment-0001.html From sprice at gmail.com Wed Jul 15 11:47:37 2009 From: sprice at gmail.com (SP) Date: Wed, 15 Jul 2009 13:47:37 -0500 Subject: [Freeswitch-users] Dialplan with lua - error missing closing angle bracket? In-Reply-To: <00be01ca057a$c2572340$470569c0$@com> References: <00be01ca057a$c2572340$470569c0$@com> Message-ID: <7e2ac3270907151147r576eb502j4d9431673a7e8e3@mail.gmail.com> check your whitespace On Wed, Jul 15, 2009 at 13:33, Lars Zeb wrote: > I copied the action element from http://wiki.freeswitch.org/wiki/Lua, > ?Sample Dialplan?. > > > > When I try to reloadxml, the cli tells me that there is a missing right > angle bracket. > > +OK [[error near line 3130]: missing >] > > > > Do the docs need updating or have I totally blown it? > > > > Thanks, Lars > > > > > > > > > > > > > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users at 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090715/fabcc9d6/attachment-0001.html From freeswitch-users at lists.freeswitch.org Wed Jul 15 17:40:18 2009 From: freeswitch-users at lists.freeswitch.org (freeswitch-users at lists.freeswitch.org) Date: Thu, 16 Jul 2009 08:40:18 +0800 Subject: [Freeswitch-users] How to set the IVR of VM menu?? Message-ID: Could you please just tell me where to set it?? >Please try looking on the wiki, this and many other questions should >be answered for you there. > >Mike > >On Jul 15, 2009, at 4:05 AM, Brad Tuan wrote: > >>* How to set the date format , and the IVR flow ........?? *>>* _______________________________________________ *>>* Freeswitch-users mailing list *>>* Freeswitch-users at 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/20090716/05832b1b/attachment-0001.html From freeswitch-users at lists.freeswitch.org Wed Jul 15 22:07:21 2009 From: freeswitch-users at lists.freeswitch.org (freeswitch-users at lists.freeswitch.org) Date: Thu, 16 Jul 2009 01:07:21 -0400 Subject: [Freeswitch-users] Freeswitch architecture In-Reply-To: <200907141618.03295.yivzhenko@mksat.net> References: <89c9bbf80907080542y39f2b234tff9163ea34c0b968@mail.gmail.com> <347F12AB-EA05-4090-A203-BDC011F02411@freeswitch.org> <200907141618.03295.yivzhenko@mksat.net> Message-ID: <20090716050720.GJ28401@hijacked.us> On Tue, Jul 14, 2009 at 04:18:03PM +0300, Yuriy Ivzhenko wrote: > On Wednesday 08 July 2009 16:29:50 Brian West wrote: > > http://wiki.freeswitch.org > > i not found any essential information about architecture :-((((( > .... may be bad looking? I'd recommend just reading the code and looking at some of the simpler modules. Most of FreeSWITCH is remarkably readable (just steer clear of the XML parsing stuff :) ). Andrew From freeswitch-users at lists.freeswitch.org Thu Jul 16 00:47:17 2009 From: freeswitch-users at lists.freeswitch.org (freeswitch-users at lists.freeswitch.org) Date: Thu, 16 Jul 2009 10:47:17 +0300 Subject: [Freeswitch-users] SIP TLS (and SRTP) In-Reply-To: <8594764D-2F2A-431C-BA91-1C2D5A97C90D@freeswitch.org> References: <10128ef10907150257u3ae1c20jb7772b66e8c1a738@mail.gmail.com> <8594764D-2F2A-431C-BA91-1C2D5A97C90D@freeswitch.org> Message-ID: <10128ef10907160047o3ca4dc25o93b58ae97b94bb4c@mail.gmail.com> thanks allot, this was my mistake. /Tzury > It tells you to edit conf/directory/default.xml not dialplan/ > default.xml and put > > > as the dial-string. > > /b From freeswitch-users at lists.freeswitch.org Thu Jul 16 06:30:26 2009 From: freeswitch-users at lists.freeswitch.org (freeswitch-users at lists.freeswitch.org) Date: Thu, 16 Jul 2009 20:30:26 +0700 Subject: [Freeswitch-users] sip extermal profile for all IP Message-ID: <8ccbff060907160630r2188c8d8od0c5c677f81eb71b@mail.gmail.com> Dear All, How to set ${local_ip_v4} to 0.0.0.0 ? i want to set up external profile for All IP Best regards. Dome C. From freeswitch-users at lists.freeswitch.org Thu Jul 16 06:48:02 2009 From: freeswitch-users at lists.freeswitch.org (freeswitch-users at lists.freeswitch.org) Date: Thu, 16 Jul 2009 19:48:02 +0600 Subject: [Freeswitch-users] sip extermal profile for all IP In-Reply-To: <8ccbff060907160630r2188c8d8od0c5c677f81eb71b@mail.gmail.com> References: <8ccbff060907160630r2188c8d8od0c5c677f81eb71b@mail.gmail.com> Message-ID: I think by default it maps to all IPs unless you mention one here (by replacing ${local_ip_v4} to some ip address). Thank you. On Thu, Jul 16, 2009 at 7:30 PM, wrote: > Dear All, > > How to set ${local_ip_v4} to 0.0.0.0 ? i want to set up > external profile for All IP > > > Best regards. > > Dome C. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090716/32926e09/attachment-0001.html From freeswitch-users at lists.freeswitch.org Thu Jul 16 07:02:46 2009 From: freeswitch-users at lists.freeswitch.org (freeswitch-users at lists.freeswitch.org) Date: Thu, 16 Jul 2009 17:02:46 +0300 Subject: [Freeswitch-users] "show channels" command with duration - patch included Message-ID: <4A5F3306.1000300@kinetix.gr> Hi, I usually find it very useful when I can retrieve a list of the currents calls along with durations. I noticed that the 'show channels' format does not include the duration (or the answered timestamp - so that one can extract it from there). So, I made a patch that includes the answered timestamp, the answered timestamp in epoch, and the duration in seconds. Of course these fields remain empty when the call hasn't been answered yet. I don't know if anyone else finds this functionality useful, so I am posting this patch here first (instead of JIRA) in order to get feedback from the users. If many of you (or the maintainers) find it interesting I can then proceed in posting it to JIRA. -- ------------------------------------------- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: regs at kinetix.gr ------------------------------------------- -------------- next part -------------- Index: src/mod/applications/mod_commands/mod_commands.c =================================================================== --- src/mod/applications/mod_commands/mod_commands.c (revision 14256) +++ src/mod/applications/mod_commands/mod_commands.c (working copy) @@ -2827,10 +2827,10 @@ } } if (strchr(argv[2], '%')) { - sprintf(sql, "select * from channels where uuid like '%s' or name like '%s' or cid_name like '%s' or cid_num like '%s' order by created_epoch", + sprintf(sql, "select *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration from channels where uuid like '%s' or name like '%s' or cid_name like '%s' or cid_num like '%s' order by created_epoch", argv[2], argv[2], argv[2], argv[2]); } else { - sprintf(sql, "select * from channels where uuid like '%%%s%%' or name like '%%%s%%' or cid_name like '%%%s%%' or cid_num like '%%%s%%' order by created_epoch", + sprintf(sql, "select *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration from channels where uuid like '%%%s%%' or name like '%%%s%%' or cid_name like '%%%s%%' or cid_num like '%%%s%%' order by created_epoch", argv[2], argv[2], argv[2], argv[2]); } @@ -2839,10 +2839,10 @@ as = argv[4]; } } else { - sprintf(sql, "select * from channels order by created_epoch"); + sprintf(sql, "select *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration from channels order by created_epoch"); } } else if (!strcasecmp(command, "channels")) { - sprintf(sql, "select * from channels order by created_epoch"); + sprintf(sql, "select *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration from channels order by created_epoch"); if (argv[1] && !strcasecmp(argv[1],"count")) { holder.justcount = 1; if (argv[3] && !strcasecmp(argv[2], "as")) { @@ -2850,7 +2850,7 @@ } } } else if (!strcasecmp(command, "distinct_channels")) { - sprintf(sql, "select * from channels left join calls on " + sprintf(sql, "select *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration from channels left join calls on " "channels.uuid=calls.caller_uuid where channels.uuid not in (select callee_uuid from calls) order by created_epoch"); if (argv[2] && !strcasecmp(argv[1], "as")) { as = argv[2]; Index: src/switch_core_sqldb.c =================================================================== --- src/switch_core_sqldb.c (revision 14256) +++ src/switch_core_sqldb.c (working copy) @@ -309,9 +309,21 @@ ); break; + case SWITCH_EVENT_CHANNEL_ANSWER: + { + + sql = switch_mprintf("update channels set answered='%s',answered_epoch='%ld' where uuid='%s'", + switch_event_get_header_nil(event, "event-date-local"), + (long)switch_epoch_time_now(NULL), + switch_event_get_header_nil(event, "unique-id") + ); + + } + break; case SWITCH_EVENT_CHANNEL_STATE: { char *state = switch_event_get_header_nil(event, "channel-state-number"); + switch_channel_state_t state_i = CS_DESTROY; if (!switch_strlen_zero(state)) { @@ -492,7 +504,9 @@ " read_rate VARCHAR(255),\n" " write_codec VARCHAR(255),\n" " write_rate VARCHAR(255),\n" - " secure VARCHAR(255)\n" + " secure VARCHAR(255),\n" + " answered VARCHAR(255),\n" + " answered_epoch INTEGER\n" ");\ncreate index uuindex on channels (uuid);\n"; char create_calls_sql[] = "CREATE TABLE calls (\n" From anthony.minessale at gmail.com Thu Jul 16 07:15:34 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 16 Jul 2009 09:15:34 -0500 Subject: [Freeswitch-users] "show channels" command with duration - patch included In-Reply-To: <4A5F3306.1000300@kinetix.gr> References: <4A5F3306.1000300@kinetix.gr> Message-ID: <191c3a030907160715k74e785c8j99b68a42dc019bb7@mail.gmail.com> I'm ok with the idea as long as it's thoroughly tested. If there is any more info you want to save from those events you should consider it now while we are modifying it. On Thu, Jul 16, 2009 at 9:02 AM, wrote: > Hi, > > I usually find it very useful when I can retrieve a list of the currents > calls along with durations. I noticed that the 'show channels' format does > not include the duration (or the answered timestamp - so that one can > extract it from there). So, I made a patch that includes the answered > timestamp, the answered timestamp in epoch, and the duration in seconds. Of > course these fields remain empty when the call hasn't been > answered yet. > > I don't know if anyone else finds this functionality useful, so I am > posting this patch here first (instead of JIRA) in order to get feedback > from the users. If many of you (or the maintainers) find it interesting I > can then proceed in posting it to JIRA. > > -- > ------------------------------------------- > Apostolos Pantsiopoulos > Kinetix Tele.com R & D > email: regs at kinetix.gr > ------------------------------------------- > > Index: src/mod/applications/mod_commands/mod_commands.c > =================================================================== > --- src/mod/applications/mod_commands/mod_commands.c (revision 14256) > +++ src/mod/applications/mod_commands/mod_commands.c (working copy) > @@ -2827,10 +2827,10 @@ > } > } > if (strchr(argv[2], '%')) { > - sprintf(sql, "select * from channels where > uuid like '%s' or name like '%s' or cid_name like '%s' or cid_num like '%s' > order by created_epoch", > + sprintf(sql, "select > *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration from channels > where uuid like '%s' or name like '%s' or cid_name like '%s' or cid_num like > '%s' order by created_epoch", > argv[2], argv[2], argv[2], > argv[2]); > } else { > - sprintf(sql, "select * from channels where > uuid like '%%%s%%' or name like '%%%s%%' or cid_name like '%%%s%%' or > cid_num like '%%%s%%' order by created_epoch", > + sprintf(sql, "select > *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration from channels > where uuid like '%%%s%%' or name like '%%%s%%' or cid_name like '%%%s%%' or > cid_num like '%%%s%%' order by created_epoch", > argv[2], argv[2], argv[2], > argv[2]); > > } > @@ -2839,10 +2839,10 @@ > as = argv[4]; > } > } else { > - sprintf(sql, "select * from channels order by > created_epoch"); > + sprintf(sql, "select > *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration from channels > order by created_epoch"); > } > } else if (!strcasecmp(command, "channels")) { > - sprintf(sql, "select * from channels order by > created_epoch"); > + sprintf(sql, "select > *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration from channels > order by created_epoch"); > if (argv[1] && !strcasecmp(argv[1],"count")) { > holder.justcount = 1; > if (argv[3] && !strcasecmp(argv[2], "as")) { > @@ -2850,7 +2850,7 @@ > } > } > } else if (!strcasecmp(command, "distinct_channels")) { > - sprintf(sql, "select * from channels left join calls on " > + sprintf(sql, "select > *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration from channels > left join calls on " > "channels.uuid=calls.caller_uuid where > channels.uuid not in (select callee_uuid from calls) order by > created_epoch"); > if (argv[2] && !strcasecmp(argv[1], "as")) { > as = argv[2]; > Index: src/switch_core_sqldb.c > =================================================================== > --- src/switch_core_sqldb.c (revision 14256) > +++ src/switch_core_sqldb.c (working copy) > @@ -309,9 +309,21 @@ > ); > > break; > + case SWITCH_EVENT_CHANNEL_ANSWER: > + { > + > + sql = switch_mprintf("update channels set > answered='%s',answered_epoch='%ld' where uuid='%s'", > + > switch_event_get_header_nil(event, "event-date-local"), > + > (long)switch_epoch_time_now(NULL), > + > switch_event_get_header_nil(event, "unique-id") > + ); > + > + } > + break; > case SWITCH_EVENT_CHANNEL_STATE: > { > char *state = switch_event_get_header_nil(event, > "channel-state-number"); > + > switch_channel_state_t state_i = CS_DESTROY; > > if (!switch_strlen_zero(state)) { > @@ -492,7 +504,9 @@ > " read_rate VARCHAR(255),\n" > " write_codec VARCHAR(255),\n" > " write_rate VARCHAR(255),\n" > - " secure VARCHAR(255)\n" > + " secure VARCHAR(255),\n" > + " answered VARCHAR(255),\n" > + " answered_epoch INTEGER\n" > ");\ncreate index uuindex on channels (uuid);\n"; > char create_calls_sql[] = > "CREATE TABLE calls (\n" > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference 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/20090716/ef6b146b/attachment-0001.html From Prometheus001 at gmx.net Thu Jul 16 07:25:23 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Thu, 16 Jul 2009 16:25:23 +0200 Subject: [Freeswitch-users] "show channels" command with duration - patch included In-Reply-To: <4A5F3306.1000300@kinetix.gr> References: <4A5F3306.1000300@kinetix.gr> Message-ID: <4A5F3853.1030005@gmx.net> Well, that's very useful for us in order to have this info in our FS Operator panel. Best regards Peter freeswitch-users at lists.freeswitch.org schrieb: > Hi, > > I usually find it very useful when I can retrieve a list of the > currents calls along with durations. I noticed that the 'show > channels' format does not include the duration (or the answered > timestamp - so that one can extract it from there). So, I made a patch > that includes the answered timestamp, the answered timestamp in epoch, > and the duration in seconds. Of course these fields remain empty when > the call hasn't been > answered yet. > > I don't know if anyone else finds this functionality useful, so I am > posting this patch here first (instead of JIRA) in order to get > feedback from the users. If many of you (or the maintainers) find it > interesting I can then proceed in posting it to JIRA. > > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From fdhege at gmail.com Thu Jul 16 07:30:34 2009 From: fdhege at gmail.com (Dale) Date: Thu, 16 Jul 2009 10:30:34 -0400 Subject: [Freeswitch-users] Setting P-Asserted-ID to something other than the callerid Message-ID: <723C30E7-4AD0-418D-A4B4-10148367A9AA@gmail.com> Hello again, I wanted to first say thanks to Brain for helping me fix my from domain issue the other day. It helped quite a bit. Now with more testing and talking with the vendor (please don't shoot the messenger :) ) They want the caller id info in the from and the charge number/ screening number in the P-Asserted-ID. I have tested this and verified that this does work like they say it does by setting the callerid number to my charge number and setting the from user in the gateway config to the callerid I want displayed. But this solution doesn't scale very well. I know I can set the gateway option caller-id-in-from to get that part done. But is there a way to set the P-Asserted-ID to something other than the callerid? Any hints would be welcomed. Thanks, -Dale From brian at freeswitch.org Thu Jul 16 07:37:02 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 16 Jul 2009 09:37:02 -0500 Subject: [Freeswitch-users] Setting P-Asserted-ID to something other than the callerid In-Reply-To: <723C30E7-4AD0-418D-A4B4-10148367A9AA@gmail.com> References: <723C30E7-4AD0-418D-A4B4-10148367A9AA@gmail.com> Message-ID: <6BAA8BE5-BFA9-47C9-A1FC-651AAA303E6E@freeswitch.org> On Jul 16, 2009, at 9:30 AM, Dale wrote: > They want the caller id info in the from and the charge number/ > screening number in the P-Asserted-ID. > set the sip_h_P-Asserted-ID=contents of header its just a variable you need to set now. > > > -Dale From regs at kinetix.gr Thu Jul 16 07:37:13 2009 From: regs at kinetix.gr (Apostolos Pantsiopoulos) Date: Thu, 16 Jul 2009 17:37:13 +0300 Subject: [Freeswitch-users] "show channels" command with duration - patch included In-Reply-To: <191c3a030907160715k74e785c8j99b68a42dc019bb7@mail.gmail.com> References: <4A5F3306.1000300@kinetix.gr> <191c3a030907160715k74e785c8j99b68a42dc019bb7@mail.gmail.com> Message-ID: <4A5F3B19.8020507@kinetix.gr> Now that I come to think of it... It would be useful if we had the timestamp (and epoch) of the events PROGRESS and PROGRESS WITH MEDIA so that we can extract the PDD (Post Dial Delay) which is a very useful statistic. Adding the user's (for the incoming) and the gateway's (for the outbound) id would also be useful. In case these fields are empty the show channels command could ommit the (empty string). Are you planning to implement them yourselves or should I begin looking at the code? Anthony Minessale wrote: > I'm ok with the idea as long as it's thoroughly tested. > If there is any more info you want to save from those events you should > consider it now while we are modifying it. > > > On Thu, Jul 16, 2009 at 9:02 AM, > wrote: > > Hi, > > I usually find it very useful when I can retrieve a list of the > currents calls along with durations. I noticed that the 'show > channels' format does not include the duration (or the answered > timestamp - so that one can extract it from there). So, I made a > patch that includes the answered timestamp, the answered timestamp > in epoch, and the duration in seconds. Of course these fields remain > empty when the call hasn't been > answered yet. > > I don't know if anyone else finds this functionality useful, so I am > posting this patch here first (instead of JIRA) in order to get > feedback from the users. If many of you (or the maintainers) find it > interesting I can then proceed in posting it to JIRA. > > -- > ------------------------------------------- > Apostolos Pantsiopoulos > Kinetix Tele.com R & D > email: regs at kinetix.gr > ------------------------------------------- > > Index: src/mod/applications/mod_commands/mod_commands.c > =================================================================== > --- src/mod/applications/mod_commands/mod_commands.c (revision 14256) > +++ src/mod/applications/mod_commands/mod_commands.c (working copy) > @@ -2827,10 +2827,10 @@ > } > } > if (strchr(argv[2], '%')) { > - sprintf(sql, "select * from channels > where uuid like '%s' or name like '%s' or cid_name like '%s' or > cid_num like '%s' order by created_epoch", > + sprintf(sql, "select > *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration from > channels where uuid like '%s' or name like '%s' or cid_name like > '%s' or cid_num like '%s' order by created_epoch", > argv[2], argv[2], > argv[2], argv[2]); > } else { > - sprintf(sql, "select * from channels > where uuid like '%%%s%%' or name like '%%%s%%' or cid_name like > '%%%s%%' or cid_num like '%%%s%%' order by created_epoch", > + sprintf(sql, "select > *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration from > channels where uuid like '%%%s%%' or name like '%%%s%%' or cid_name > like '%%%s%%' or cid_num like '%%%s%%' order by created_epoch", > argv[2], argv[2], > argv[2], argv[2]); > > } > @@ -2839,10 +2839,10 @@ > as = argv[4]; > } > } else { > - sprintf(sql, "select * from channels order > by created_epoch"); > + sprintf(sql, "select > *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration from > channels order by created_epoch"); > } > } else if (!strcasecmp(command, "channels")) { > - sprintf(sql, "select * from channels order by > created_epoch"); > + sprintf(sql, "select > *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration from > channels order by created_epoch"); > if (argv[1] && !strcasecmp(argv[1],"count")) { > holder.justcount = 1; > if (argv[3] && !strcasecmp(argv[2], "as")) { > @@ -2850,7 +2850,7 @@ > } > } > } else if (!strcasecmp(command, "distinct_channels")) { > - sprintf(sql, "select * from channels left join calls > on " > + sprintf(sql, "select > *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration from > channels left join calls on " > "channels.uuid=calls.caller_uuid > where channels.uuid not in (select callee_uuid from calls) order by > created_epoch"); > if (argv[2] && !strcasecmp(argv[1], "as")) { > as = argv[2]; > Index: src/switch_core_sqldb.c > =================================================================== > --- src/switch_core_sqldb.c (revision 14256) > +++ src/switch_core_sqldb.c (working copy) > @@ -309,9 +309,21 @@ > ); > > break; > + case SWITCH_EVENT_CHANNEL_ANSWER: > + { > + > + sql = switch_mprintf("update channels set > answered='%s',answered_epoch='%ld' where uuid='%s'", > + > switch_event_get_header_nil(event, "event-date-local"), > + > (long)switch_epoch_time_now(NULL), > + > switch_event_get_header_nil(event, "unique-id") > + ); > + > + } > + break; > case SWITCH_EVENT_CHANNEL_STATE: > { > char *state = > switch_event_get_header_nil(event, "channel-state-number"); > + > switch_channel_state_t state_i = CS_DESTROY; > > if (!switch_strlen_zero(state)) { > @@ -492,7 +504,9 @@ > " read_rate VARCHAR(255),\n" > " write_codec VARCHAR(255),\n" > " write_rate VARCHAR(255),\n" > - " secure VARCHAR(255)\n" > + " secure VARCHAR(255),\n" > + " answered VARCHAR(255),\n" > + " answered_epoch INTEGER\n" > ");\ncreate index uuindex on channels (uuid);\n"; > char create_calls_sql[] = > "CREATE TABLE calls (\n" > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > 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 -- ------------------------------------------- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: regs at kinetix.gr ------------------------------------------- From krice at suspicious.org Thu Jul 16 07:41:25 2009 From: krice at suspicious.org (Ken Rice) Date: Thu, 16 Jul 2009 09:41:25 -0500 Subject: [Freeswitch-users] Setting P-Asserted-ID to something other than the callerid In-Reply-To: <723C30E7-4AD0-418D-A4B4-10148367A9AA@gmail.com> Message-ID: That's just right down screwing with the standards... PAID is the caller id... This particular definition is from the RFCs and 3GPP docs for IMS which is why we have standardized P- headers... Can your vendor not look at the P-Charging-Vector field? Also, From when used with PAID is more like an ANI not a CLID > From: Dale > Reply-To: > Date: Thu, 16 Jul 2009 10:30:34 -0400 > To: > Subject: [Freeswitch-users] Setting P-Asserted-ID to something other than the > callerid > > > Hello again, > > I wanted to first say thanks to Brain for helping me fix my from > domain issue the other day. It helped quite a bit. > > Now with more testing and talking with the vendor (please don't shoot > the messenger :) ) > > They want the caller id info in the from and the charge number/ > screening number in the P-Asserted-ID. > > I have tested this and verified that this does work like they say it > does by setting the callerid number to my charge number and setting > the from user in the gateway config to the callerid I want displayed. > But this solution doesn't scale very well. > > I know I can set the gateway option caller-id-in-from to get that part > done. But is there a way to set the P-Asserted-ID to something other > than the callerid? > > Any hints would be welcomed. > > Thanks, > > -Dale > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.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 Jul 16 07:43:45 2009 From: msc at freeswitch.org (Michael S Collins) Date: Thu, 16 Jul 2009 07:43:45 -0700 Subject: [Freeswitch-users] How to set the IVR of VM menu?? Message-ID: On Jul 15, 2009, at 5:40 PM, freeswitch-users at lists.freeswitch.org wrote: > Could you please just tell me where to set it?? The menu actions are defined in conf/autoload_configs/ivr.conf.xml The audio played for the menus is defined in conf/lang/en/demo/demo- ivr.xml -MC > > >Please try looking on the wiki, this and many other questions should > >be answered for you there. > > > >Mike > > > >On Jul 15, 2009, at 4:05 AM, Brad Tuan wrote: > > > >> How to set the date format , and the IVR flow ........?? > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20090716/50bf7815/attachment-0001.html From anthony.minessale at gmail.com Thu Jul 16 07:46:29 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 16 Jul 2009 09:46:29 -0500 Subject: [Freeswitch-users] "show channels" command with duration - patch included In-Reply-To: <4A5F3B19.8020507@kinetix.gr> References: <4A5F3306.1000300@kinetix.gr> <191c3a030907160715k74e785c8j99b68a42dc019bb7@mail.gmail.com> <4A5F3B19.8020507@kinetix.gr> Message-ID: <191c3a030907160746p3ed81c31s58caf5fb39c2a47a@mail.gmail.com> I wasn't planning on implementing it but I was just mentioning that if you were going to do your patch, consider if there is any other info to store while the patient is on the operating table. The line we should not cross is to store all the info in the table since, really, you could be collecting those events in your application as well to store that info. But for the casual user, some more fields may be interesting. On Thu, Jul 16, 2009 at 9:37 AM, Apostolos Pantsiopoulos wrote: > Now that I come to think of it... > > It would be useful if we had the timestamp (and epoch) > of the events PROGRESS and PROGRESS WITH MEDIA so that we can extract > the PDD (Post Dial Delay) which is a very useful statistic. > > Adding the user's (for the incoming) and the gateway's (for the > outbound) id would also be useful. In case these fields are empty the > show channels command could ommit the (empty string). > > Are you planning to implement them yourselves or should I begin looking > at the code? > > > Anthony Minessale wrote: > > I'm ok with the idea as long as it's thoroughly tested. > > If there is any more info you want to save from those events you should > > consider it now while we are modifying it. > > > > > > On Thu, Jul 16, 2009 at 9:02 AM, > > wrote: > > > > Hi, > > > > I usually find it very useful when I can retrieve a list of the > > currents calls along with durations. I noticed that the 'show > > channels' format does not include the duration (or the answered > > timestamp - so that one can extract it from there). So, I made a > > patch that includes the answered timestamp, the answered timestamp > > in epoch, and the duration in seconds. Of course these fields remain > > empty when the call hasn't been > > answered yet. > > > > I don't know if anyone else finds this functionality useful, so I am > > posting this patch here first (instead of JIRA) in order to get > > feedback from the users. If many of you (or the maintainers) find it > > interesting I can then proceed in posting it to JIRA. > > > > -- > > ------------------------------------------- > > Apostolos Pantsiopoulos > > Kinetix Tele.com R & D > > email: regs at kinetix.gr > > ------------------------------------------- > > > > Index: src/mod/applications/mod_commands/mod_commands.c > > =================================================================== > > --- src/mod/applications/mod_commands/mod_commands.c (revision > 14256) > > +++ src/mod/applications/mod_commands/mod_commands.c (working > copy) > > @@ -2827,10 +2827,10 @@ > > } > > } > > if (strchr(argv[2], '%')) { > > - sprintf(sql, "select * from channels > > where uuid like '%s' or name like '%s' or cid_name like '%s' or > > cid_num like '%s' order by created_epoch", > > + sprintf(sql, "select > > *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration from > > channels where uuid like '%s' or name like '%s' or cid_name like > > '%s' or cid_num like '%s' order by created_epoch", > > argv[2], argv[2], > > argv[2], argv[2]); > > } else { > > - sprintf(sql, "select * from channels > > where uuid like '%%%s%%' or name like '%%%s%%' or cid_name like > > '%%%s%%' or cid_num like '%%%s%%' order by created_epoch", > > + sprintf(sql, "select > > *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration from > > channels where uuid like '%%%s%%' or name like '%%%s%%' or cid_name > > like '%%%s%%' or cid_num like '%%%s%%' order by created_epoch", > > argv[2], argv[2], > > argv[2], argv[2]); > > > > } > > @@ -2839,10 +2839,10 @@ > > as = argv[4]; > > } > > } else { > > - sprintf(sql, "select * from channels order > > by created_epoch"); > > + sprintf(sql, "select > > *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration from > > channels order by created_epoch"); > > } > > } else if (!strcasecmp(command, "channels")) { > > - sprintf(sql, "select * from channels order by > > created_epoch"); > > + sprintf(sql, "select > > *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration from > > channels order by created_epoch"); > > if (argv[1] && !strcasecmp(argv[1],"count")) { > > holder.justcount = 1; > > if (argv[3] && !strcasecmp(argv[2], "as")) { > > @@ -2850,7 +2850,7 @@ > > } > > } > > } else if (!strcasecmp(command, "distinct_channels")) { > > - sprintf(sql, "select * from channels left join calls > > on " > > + sprintf(sql, "select > > *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration from > > channels left join calls on " > > "channels.uuid=calls.caller_uuid > > where channels.uuid not in (select callee_uuid from calls) order by > > created_epoch"); > > if (argv[2] && !strcasecmp(argv[1], "as")) { > > as = argv[2]; > > Index: src/switch_core_sqldb.c > > =================================================================== > > --- src/switch_core_sqldb.c (revision 14256) > > +++ src/switch_core_sqldb.c (working copy) > > @@ -309,9 +309,21 @@ > > ); > > > > break; > > + case SWITCH_EVENT_CHANNEL_ANSWER: > > + { > > + > > + sql = switch_mprintf("update channels set > > answered='%s',answered_epoch='%ld' where uuid='%s'", > > + > > switch_event_get_header_nil(event, "event-date-local"), > > + > > (long)switch_epoch_time_now(NULL), > > + > > switch_event_get_header_nil(event, "unique-id") > > + ); > > + > > + } > > + break; > > case SWITCH_EVENT_CHANNEL_STATE: > > { > > char *state = > > switch_event_get_header_nil(event, "channel-state-number"); > > + > > switch_channel_state_t state_i = CS_DESTROY; > > > > if (!switch_strlen_zero(state)) { > > @@ -492,7 +504,9 @@ > > " read_rate VARCHAR(255),\n" > > " write_codec VARCHAR(255),\n" > > " write_rate VARCHAR(255),\n" > > - " secure VARCHAR(255)\n" > > + " secure VARCHAR(255),\n" > > + " answered VARCHAR(255),\n" > > + " answered_epoch INTEGER\n" > > ");\ncreate index uuindex on channels > (uuid);\n"; > > char create_calls_sql[] = > > "CREATE TABLE calls (\n" > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com > > > > > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > > > > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > 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 > > > -- > ------------------------------------------- > 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 > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference 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/20090716/9fdb77aa/attachment-0001.html From fdhege at gmail.com Thu Jul 16 07:48:24 2009 From: fdhege at gmail.com (Dale) Date: Thu, 16 Jul 2009 10:48:24 -0400 Subject: [Freeswitch-users] Setting P-Asserted-ID to something other than the callerid In-Reply-To: <6BAA8BE5-BFA9-47C9-A1FC-651AAA303E6E@freeswitch.org> References: <723C30E7-4AD0-418D-A4B4-10148367A9AA@gmail.com> <6BAA8BE5-BFA9-47C9-A1FC-651AAA303E6E@freeswitch.org> Message-ID: <98E42E2F-642A-41AD-A6FC-7D92B412F1A1@gmail.com> That works perfectly. Thanks, -Dale On Jul 16, 2009, at 10:37 AM, Brian West wrote: > > On Jul 16, 2009, at 9:30 AM, Dale wrote: > >> They want the caller id info in the from and the charge number/ >> screening number in the P-Asserted-ID. >> > > set the sip_h_P-Asserted-ID=contents of header > > its just a variable you need to set now. >> >> >> -Dale > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.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 Jul 16 07:50:28 2009 From: msc at freeswitch.org (Michael S Collins) Date: Thu, 16 Jul 2009 07:50:28 -0700 Subject: [Freeswitch-users] "show channels" command with duration - patch included In-Reply-To: <4A5F3B19.8020507@kinetix.gr> References: <4A5F3306.1000300@kinetix.gr> <191c3a030907160715k74e785c8j99b68a42dc019bb7@mail.gmail.com> <4A5F3B19.8020507@kinetix.gr> Message-ID: <6079D960-F5EB-4774-93DA-87402CFC1157@freeswitch.org> I wonder if it would make sense to create a separate sub-command like "show channels stats" or something. That way we could put all sorts of nifty info there without breaking the existing command. Thoughts? -MC Sent from my iPhone On Jul 16, 2009, at 7:37 AM, Apostolos Pantsiopoulos wrote: > Now that I come to think of it... > > It would be useful if we had the timestamp (and epoch) > of the events PROGRESS and PROGRESS WITH MEDIA so that we can extract > the PDD (Post Dial Delay) which is a very useful statistic. > > Adding the user's (for the incoming) and the gateway's (for the > outbound) id would also be useful. In case these fields are empty the > show channels command could ommit the (empty string). > > Are you planning to implement them yourselves or should I begin > looking > at the code? > > > Anthony Minessale wrote: >> I'm ok with the idea as long as it's thoroughly tested. >> If there is any more info you want to save from those events you >> should >> consider it now while we are modifying it. >> >> >> On Thu, Jul 16, 2009 at 9:02 AM, > > wrote: >> >> Hi, >> >> I usually find it very useful when I can retrieve a list of the >> currents calls along with durations. I noticed that the 'show >> channels' format does not include the duration (or the answered >> timestamp - so that one can extract it from there). So, I made a >> patch that includes the answered timestamp, the answered timestamp >> in epoch, and the duration in seconds. Of course these fields >> remain >> empty when the call hasn't been >> answered yet. >> >> I don't know if anyone else finds this functionality useful, so >> I am >> posting this patch here first (instead of JIRA) in order to get >> feedback from the users. If many of you (or the maintainers) >> find it >> interesting I can then proceed in posting it to JIRA. >> >> -- >> ------------------------------------------- >> Apostolos Pantsiopoulos >> Kinetix Tele.com R & D >> email: regs at kinetix.gr >> ------------------------------------------- >> >> Index: src/mod/applications/mod_commands/mod_commands.c >> >> =================================================================== >> --- src/mod/applications/mod_commands/mod_commands.c >> (revision 14256) >> +++ src/mod/applications/mod_commands/mod_commands.c (working >> copy) >> @@ -2827,10 +2827,10 @@ >> } >> } >> if (strchr(argv[2], '%')) { >> - sprintf(sql, "select * from >> channels >> where uuid like '%s' or name like '%s' or cid_name like '%s' or >> cid_num like '%s' order by created_epoch", >> + sprintf(sql, "select >> *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration from >> channels where uuid like '%s' or name like '%s' or cid_name like >> '%s' or cid_num like '%s' order by created_epoch", >> argv[2], argv[2], >> argv[2], argv[2]); >> } else { >> - sprintf(sql, "select * from >> channels >> where uuid like '%%%s%%' or name like '%%%s%%' or cid_name like >> '%%%s%%' or cid_num like '%%%s%%' order by created_epoch", >> + sprintf(sql, "select >> *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration from >> channels where uuid like '%%%s%%' or name like '%%%s%%' or >> cid_name >> like '%%%s%%' or cid_num like '%%%s%%' order by created_epoch", >> argv[2], argv[2], >> argv[2], argv[2]); >> >> } >> @@ -2839,10 +2839,10 @@ >> as = argv[4]; >> } >> } else { >> - sprintf(sql, "select * from channels order >> by created_epoch"); >> + sprintf(sql, "select >> *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration from >> channels order by created_epoch"); >> } >> } else if (!strcasecmp(command, "channels")) { >> - sprintf(sql, "select * from channels order by >> created_epoch"); >> + sprintf(sql, "select >> *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration from >> channels order by created_epoch"); >> if (argv[1] && !strcasecmp(argv[1],"count")) { >> holder.justcount = 1; >> if (argv[3] && !strcasecmp(argv[2], "as")) { >> @@ -2850,7 +2850,7 @@ >> } >> } >> } else if (!strcasecmp(command, "distinct_channels")) { >> - sprintf(sql, "select * from channels left join >> calls >> on " >> + sprintf(sql, "select >> *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration from >> channels left join calls on " >> "channels.uuid=calls.caller_uuid >> where channels.uuid not in (select callee_uuid from calls) order >> by >> created_epoch"); >> if (argv[2] && !strcasecmp(argv[1], "as")) { >> as = argv[2]; >> Index: src/switch_core_sqldb.c >> >> =================================================================== >> --- src/switch_core_sqldb.c (revision 14256) >> +++ src/switch_core_sqldb.c (working copy) >> @@ -309,9 +309,21 @@ >> ); >> >> break; >> + case SWITCH_EVENT_CHANNEL_ANSWER: >> + { >> + >> + sql = switch_mprintf("update channels set >> answered='%s',answered_epoch='%ld' where uuid='%s'", >> + >> switch_event_get_header_nil(event, "event-date-local"), >> + >> (long)switch_epoch_time_now(NULL), >> + >> switch_event_get_header_nil(event, "unique-id") >> + ); >> + >> + } >> + break; >> case SWITCH_EVENT_CHANNEL_STATE: >> { >> char *state = >> switch_event_get_header_nil(event, "channel-state-number"); >> + >> switch_channel_state_t state_i = >> CS_DESTROY; >> >> if (!switch_strlen_zero(state)) { >> @@ -492,7 +504,9 @@ >> " read_rate VARCHAR(255),\n" >> " write_codec VARCHAR(255),\n" >> " write_rate VARCHAR(255),\n" >> - " secure VARCHAR(255)\n" >> + " secure VARCHAR(255),\n" >> + " answered VARCHAR(255),\n" >> + " answered_epoch INTEGER\n" >> ");\ncreate index uuindex on channels >> (uuid);\n"; >> char create_calls_sql[] = >> "CREATE TABLE calls (\n" >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> 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 > > > -- > ------------------------------------------- > 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 From dave at 3c.co.uk Thu Jul 16 07:51:18 2009 From: dave at 3c.co.uk (David Knell) Date: Thu, 16 Jul 2009 11:51:18 -0300 Subject: [Freeswitch-users] Setting P-Asserted-ID to something other than the callerid In-Reply-To: <723C30E7-4AD0-418D-A4B4-10148367A9AA@gmail.com> References: <723C30E7-4AD0-418D-A4B4-10148367A9AA@gmail.com> Message-ID: <1247755878.4289.64.camel@dk-d820> Hi Dale, You can set the header to anything you like by including something along the lines of in your dialplan. Cheers -- Dave > Hello again, > > I wanted to first say thanks to Brain for helping me fix my from > domain issue the other day. It helped quite a bit. > > Now with more testing and talking with the vendor (please don't shoot > the messenger :) ) > > They want the caller id info in the from and the charge number/ > screening number in the P-Asserted-ID. > > I have tested this and verified that this does work like they say it > does by setting the callerid number to my charge number and setting > the from user in the gateway config to the callerid I want displayed. > But this solution doesn't scale very well. > > I know I can set the gateway option caller-id-in-from to get that part > done. But is there a way to set the P-Asserted-ID to something other > than the callerid? > > Any hints would be welcomed. > > Thanks, > > -Dale > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- David Knell, Director, 3C Limited T: +44 20 3298 2000 E: dave at 3c.co.uk W: http://www.3c.co.uk From anthony.minessale at gmail.com Thu Jul 16 07:54:55 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 16 Jul 2009 09:54:55 -0500 Subject: [Freeswitch-users] "show channels" command with duration - patch included In-Reply-To: <6079D960-F5EB-4774-93DA-87402CFC1157@freeswitch.org> References: <4A5F3306.1000300@kinetix.gr> <191c3a030907160715k74e785c8j99b68a42dc019bb7@mail.gmail.com> <4A5F3B19.8020507@kinetix.gr> <6079D960-F5EB-4774-93DA-87402CFC1157@freeswitch.org> Message-ID: <191c3a030907160754g62aac895ge10a90476d5e890@mail.gmail.com> it doesn't really break anything to add more fields besides the ability to read it but it's already fairly wide as it is. Thats why we have "show channels as xml" On Thu, Jul 16, 2009 at 9:50 AM, Michael S Collins wrote: > I wonder if it would make sense to create a separate sub-command like > "show channels stats" or something. That way we could put all sorts of > nifty info there without breaking the existing command. > > Thoughts? > -MC > > Sent from my iPhone > > On Jul 16, 2009, at 7:37 AM, Apostolos Pantsiopoulos > wrote: > > > Now that I come to think of it... > > > > It would be useful if we had the timestamp (and epoch) > > of the events PROGRESS and PROGRESS WITH MEDIA so that we can extract > > the PDD (Post Dial Delay) which is a very useful statistic. > > > > Adding the user's (for the incoming) and the gateway's (for the > > outbound) id would also be useful. In case these fields are empty the > > show channels command could ommit the (empty string). > > > > Are you planning to implement them yourselves or should I begin > > looking > > at the code? > > > > > > Anthony Minessale wrote: > >> I'm ok with the idea as long as it's thoroughly tested. > >> If there is any more info you want to save from those events you > >> should > >> consider it now while we are modifying it. > >> > >> > >> On Thu, Jul 16, 2009 at 9:02 AM, >> > wrote: > >> > >> Hi, > >> > >> I usually find it very useful when I can retrieve a list of the > >> currents calls along with durations. I noticed that the 'show > >> channels' format does not include the duration (or the answered > >> timestamp - so that one can extract it from there). So, I made a > >> patch that includes the answered timestamp, the answered timestamp > >> in epoch, and the duration in seconds. Of course these fields > >> remain > >> empty when the call hasn't been > >> answered yet. > >> > >> I don't know if anyone else finds this functionality useful, so > >> I am > >> posting this patch here first (instead of JIRA) in order to get > >> feedback from the users. If many of you (or the maintainers) > >> find it > >> interesting I can then proceed in posting it to JIRA. > >> > >> -- > >> ------------------------------------------- > >> Apostolos Pantsiopoulos > >> Kinetix Tele.com R & D > >> email: regs at kinetix.gr > >> ------------------------------------------- > >> > >> Index: src/mod/applications/mod_commands/mod_commands.c > >> > >> =================================================================== > >> --- src/mod/applications/mod_commands/mod_commands.c > >> (revision 14256) > >> +++ src/mod/applications/mod_commands/mod_commands.c (working > >> copy) > >> @@ -2827,10 +2827,10 @@ > >> } > >> } > >> if (strchr(argv[2], '%')) { > >> - sprintf(sql, "select * from > >> channels > >> where uuid like '%s' or name like '%s' or cid_name like '%s' or > >> cid_num like '%s' order by created_epoch", > >> + sprintf(sql, "select > >> *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration from > >> channels where uuid like '%s' or name like '%s' or cid_name like > >> '%s' or cid_num like '%s' order by created_epoch", > >> argv[2], argv[2], > >> argv[2], argv[2]); > >> } else { > >> - sprintf(sql, "select * from > >> channels > >> where uuid like '%%%s%%' or name like '%%%s%%' or cid_name like > >> '%%%s%%' or cid_num like '%%%s%%' order by created_epoch", > >> + sprintf(sql, "select > >> *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration from > >> channels where uuid like '%%%s%%' or name like '%%%s%%' or > >> cid_name > >> like '%%%s%%' or cid_num like '%%%s%%' order by created_epoch", > >> argv[2], argv[2], > >> argv[2], argv[2]); > >> > >> } > >> @@ -2839,10 +2839,10 @@ > >> as = argv[4]; > >> } > >> } else { > >> - sprintf(sql, "select * from channels order > >> by created_epoch"); > >> + sprintf(sql, "select > >> *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration from > >> channels order by created_epoch"); > >> } > >> } else if (!strcasecmp(command, "channels")) { > >> - sprintf(sql, "select * from channels order by > >> created_epoch"); > >> + sprintf(sql, "select > >> *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration from > >> channels order by created_epoch"); > >> if (argv[1] && !strcasecmp(argv[1],"count")) { > >> holder.justcount = 1; > >> if (argv[3] && !strcasecmp(argv[2], "as")) { > >> @@ -2850,7 +2850,7 @@ > >> } > >> } > >> } else if (!strcasecmp(command, "distinct_channels")) { > >> - sprintf(sql, "select * from channels left join > >> calls > >> on " > >> + sprintf(sql, "select > >> *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration from > >> channels left join calls on " > >> "channels.uuid=calls.caller_uuid > >> where channels.uuid not in (select callee_uuid from calls) order > >> by > >> created_epoch"); > >> if (argv[2] && !strcasecmp(argv[1], "as")) { > >> as = argv[2]; > >> Index: src/switch_core_sqldb.c > >> > >> =================================================================== > >> --- src/switch_core_sqldb.c (revision 14256) > >> +++ src/switch_core_sqldb.c (working copy) > >> @@ -309,9 +309,21 @@ > >> ); > >> > >> break; > >> + case SWITCH_EVENT_CHANNEL_ANSWER: > >> + { > >> + > >> + sql = switch_mprintf("update channels set > >> answered='%s',answered_epoch='%ld' where uuid='%s'", > >> + > >> switch_event_get_header_nil(event, "event-date-local"), > >> + > >> (long)switch_epoch_time_now(NULL), > >> + > >> switch_event_get_header_nil(event, "unique-id") > >> + ); > >> + > >> + } > >> + break; > >> case SWITCH_EVENT_CHANNEL_STATE: > >> { > >> char *state = > >> switch_event_get_header_nil(event, "channel-state-number"); > >> + > >> switch_channel_state_t state_i = > >> CS_DESTROY; > >> > >> if (!switch_strlen_zero(state)) { > >> @@ -492,7 +504,9 @@ > >> " read_rate VARCHAR(255),\n" > >> " write_codec VARCHAR(255),\n" > >> " write_rate VARCHAR(255),\n" > >> - " secure VARCHAR(255)\n" > >> + " secure VARCHAR(255),\n" > >> + " answered VARCHAR(255),\n" > >> + " answered_epoch INTEGER\n" > >> ");\ncreate index uuindex on channels > >> (uuid);\n"; > >> char create_calls_sql[] = > >> "CREATE TABLE calls (\n" > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > >> > >> > >> > >> -- > >> Anthony Minessale II > >> > >> FreeSWITCH http://www.freeswitch.org/ > >> ClueCon http://www.cluecon.com/ > >> Twitter: http://twitter.com/FreeSWITCH_wire > >> > >> AIM: anthm > >> MSN:anthony_minessale at hotmail.com > >> > > > >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > >> > > > >> IRC: irc.freenode.net #freeswitch > >> > >> FreeSWITCH Developer Conference > >> 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 > > > > > > -- > > ------------------------------------------- > > 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/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference 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/20090716/781ef4d2/attachment-0001.html From brian at freeswitch.org Thu Jul 16 07:58:16 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 16 Jul 2009 09:58:16 -0500 Subject: [Freeswitch-users] Setting P-Asserted-ID to something other than the callerid In-Reply-To: <1247755878.4289.64.camel@dk-d820> References: <723C30E7-4AD0-418D-A4B4-10148367A9AA@gmail.com> <1247755878.4289.64.camel@dk-d820> Message-ID: Kinda wrong there! Gotta use CDATA because it has < and > in the data you're setting. And you'll wanna export it I suspect. ]]> /b On Jul 16, 2009, at 9:51 AM, David Knell wrote: > Hi Dale, > > You can set the header to anything you like by including something > along > the lines of > > in your dialplan. > > Cheers -- -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090716/15b3fd87/attachment-0001.html From regs at kinetix.gr Thu Jul 16 08:10:53 2009 From: regs at kinetix.gr (Apostolos Pantsiopoulos) Date: Thu, 16 Jul 2009 18:10:53 +0300 Subject: [Freeswitch-users] "show channels" command with duration - patch included In-Reply-To: <191c3a030907160754g62aac895ge10a90476d5e890@mail.gmail.com> References: <4A5F3306.1000300@kinetix.gr> <191c3a030907160715k74e785c8j99b68a42dc019bb7@mail.gmail.com> <4A5F3B19.8020507@kinetix.gr> <6079D960-F5EB-4774-93DA-87402CFC1157@freeswitch.org> <191c3a030907160754g62aac895ge10a90476d5e890@mail.gmail.com> Message-ID: <4A5F42FD.60504@kinetix.gr> OK I 'll start implementing the progress timestamp field. The only reason I mentioned the user/gateway id field is that FS admins gain a lot by looking at a row of "show channels" result and be able to see who is the caller and who is the callee. Anthony Minessale wrote: > it doesn't really break anything to add more fields besides the ability > to read it but it's already fairly wide as it is. > Thats why we have "show channels as xml" > > > On Thu, Jul 16, 2009 at 9:50 AM, Michael S Collins > wrote: > > I wonder if it would make sense to create a separate sub-command like > "show channels stats" or something. That way we could put all sorts of > nifty info there without breaking the existing command. > > Thoughts? > -MC > > Sent from my iPhone > > On Jul 16, 2009, at 7:37 AM, Apostolos Pantsiopoulos > > > wrote: > > > Now that I come to think of it... > > > > It would be useful if we had the timestamp (and epoch) > > of the events PROGRESS and PROGRESS WITH MEDIA so that we can extract > > the PDD (Post Dial Delay) which is a very useful statistic. > > > > Adding the user's (for the incoming) and the gateway's (for the > > outbound) id would also be useful. In case these fields are empty the > > show channels command could ommit the (empty string). > > > > Are you planning to implement them yourselves or should I begin > > looking > > at the code? > > > > > > Anthony Minessale wrote: > >> I'm ok with the idea as long as it's thoroughly tested. > >> If there is any more info you want to save from those events you > >> should > >> consider it now while we are modifying it. > >> > >> > >> On Thu, Jul 16, 2009 at 9:02 AM, > > >> >> wrote: > >> > >> Hi, > >> > >> I usually find it very useful when I can retrieve a list of the > >> currents calls along with durations. I noticed that the 'show > >> channels' format does not include the duration (or the answered > >> timestamp - so that one can extract it from there). So, I made a > >> patch that includes the answered timestamp, the answered > timestamp > >> in epoch, and the duration in seconds. Of course these fields > >> remain > >> empty when the call hasn't been > >> answered yet. > >> > >> I don't know if anyone else finds this functionality useful, so > >> I am > >> posting this patch here first (instead of JIRA) in order to get > >> feedback from the users. If many of you (or the maintainers) > >> find it > >> interesting I can then proceed in posting it to JIRA. > >> > >> -- > >> ------------------------------------------- > >> Apostolos Pantsiopoulos > >> Kinetix Tele.com R & D > >> email: regs at kinetix.gr > > > >> ------------------------------------------- > >> > >> Index: src/mod/applications/mod_commands/mod_commands.c > >> > >> =================================================================== > >> --- src/mod/applications/mod_commands/mod_commands.c > >> (revision 14256) > >> +++ src/mod/applications/mod_commands/mod_commands.c (working > >> copy) > >> @@ -2827,10 +2827,10 @@ > >> } > >> } > >> if (strchr(argv[2], '%')) { > >> - sprintf(sql, "select * from > >> channels > >> where uuid like '%s' or name like '%s' or cid_name like '%s' or > >> cid_num like '%s' order by created_epoch", > >> + sprintf(sql, "select > >> *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration from > >> channels where uuid like '%s' or name like '%s' or cid_name like > >> '%s' or cid_num like '%s' order by created_epoch", > >> argv[2], argv[2], > >> argv[2], argv[2]); > >> } else { > >> - sprintf(sql, "select * from > >> channels > >> where uuid like '%%%s%%' or name like '%%%s%%' or cid_name like > >> '%%%s%%' or cid_num like '%%%s%%' order by created_epoch", > >> + sprintf(sql, "select > >> *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration from > >> channels where uuid like '%%%s%%' or name like '%%%s%%' or > >> cid_name > >> like '%%%s%%' or cid_num like '%%%s%%' order by created_epoch", > >> argv[2], argv[2], > >> argv[2], argv[2]); > >> > >> } > >> @@ -2839,10 +2839,10 @@ > >> as = argv[4]; > >> } > >> } else { > >> - sprintf(sql, "select * from channels > order > >> by created_epoch"); > >> + sprintf(sql, "select > >> *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration from > >> channels order by created_epoch"); > >> } > >> } else if (!strcasecmp(command, "channels")) { > >> - sprintf(sql, "select * from channels order by > >> created_epoch"); > >> + sprintf(sql, "select > >> *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration from > >> channels order by created_epoch"); > >> if (argv[1] && !strcasecmp(argv[1],"count")) { > >> holder.justcount = 1; > >> if (argv[3] && !strcasecmp(argv[2], "as")) { > >> @@ -2850,7 +2850,7 @@ > >> } > >> } > >> } else if (!strcasecmp(command, "distinct_channels")) { > >> - sprintf(sql, "select * from channels left join > >> calls > >> on " > >> + sprintf(sql, "select > >> *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration from > >> channels left join calls on " > >> "channels.uuid=calls.caller_uuid > >> where channels.uuid not in (select callee_uuid from calls) order > >> by > >> created_epoch"); > >> if (argv[2] && !strcasecmp(argv[1], "as")) { > >> as = argv[2]; > >> Index: src/switch_core_sqldb.c > >> > >> =================================================================== > >> --- src/switch_core_sqldb.c (revision 14256) > >> +++ src/switch_core_sqldb.c (working copy) > >> @@ -309,9 +309,21 @@ > >> ); > >> > >> break; > >> + case SWITCH_EVENT_CHANNEL_ANSWER: > >> + { > >> + > >> + sql = switch_mprintf("update channels set > >> answered='%s',answered_epoch='%ld' where uuid='%s'", > >> + > >> switch_event_get_header_nil(event, "event-date-local"), > >> + > >> (long)switch_epoch_time_now(NULL), > >> + > >> switch_event_get_header_nil(event, "unique-id") > >> + ); > >> + > >> + } > >> + break; > >> case SWITCH_EVENT_CHANNEL_STATE: > >> { > >> char *state = > >> switch_event_get_header_nil(event, "channel-state-number"); > >> + > >> switch_channel_state_t state_i = > >> CS_DESTROY; > >> > >> if (!switch_strlen_zero(state)) { > >> @@ -492,7 +504,9 @@ > >> " read_rate VARCHAR(255),\n" > >> " write_codec VARCHAR(255),\n" > >> " write_rate VARCHAR(255),\n" > >> - " secure VARCHAR(255)\n" > >> + " secure VARCHAR(255),\n" > >> + " answered VARCHAR(255),\n" > >> + " answered_epoch INTEGER\n" > >> ");\ncreate index uuindex on channels > >> (uuid);\n"; > >> char create_calls_sql[] = > >> "CREATE TABLE calls (\n" > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > > >> > > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > >> > >> > >> > >> -- > >> Anthony Minessale II > >> > >> FreeSWITCH http://www.freeswitch.org/ > >> ClueCon http://www.cluecon.com/ > >> Twitter: http://twitter.com/FreeSWITCH_wire > >> > >> AIM: anthm > >> MSN:anthony_minessale at hotmail.com > > >> > > >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > >> > > >> IRC: irc.freenode.net > #freeswitch > >> > >> FreeSWITCH Developer Conference > >> 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 > > > > > > -- > > ------------------------------------------- > > 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/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > 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 -- ------------------------------------------- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: regs at kinetix.gr ------------------------------------------- From saeedahmad1981 at gmail.com Thu Jul 16 08:38:02 2009 From: saeedahmad1981 at gmail.com (Saeed Ahmed) Date: Thu, 16 Jul 2009 17:38:02 +0200 Subject: [Freeswitch-users] "show channels" command with duration - patch included In-Reply-To: <4A5F42FD.60504@kinetix.gr> References: <4A5F3306.1000300@kinetix.gr> <191c3a030907160715k74e785c8j99b68a42dc019bb7@mail.gmail.com> <4A5F3B19.8020507@kinetix.gr> <6079D960-F5EB-4774-93DA-87402CFC1157@freeswitch.org><191c3a030907160754g62aac895ge10a90476d5e890@mail.gmail.com> <4A5F42FD.60504@kinetix.gr> Message-ID: <9AEB5B242EBF4DD0A9B1544B7EAFC82C@saeedlaptop> Hi, its very useful feature for monitoring, I am doing it with XML RPC and getting the result on webpage. There is one issue which is nothing to do with that patch but in general: if we are using absolute_codec_string variable and codes are like G729,G723 then this *comma* between codec ruin the array. I think there should be other separator. - Saeed -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Apostolos Pantsiopoulos Sent: Thursday, July 16, 2009 5:11 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] "show channels" command with duration - patch included OK I 'll start implementing the progress timestamp field. The only reason I mentioned the user/gateway id field is that FS admins gain a lot by looking at a row of "show channels" result and be able to see who is the caller and who is the callee. Anthony Minessale wrote: > it doesn't really break anything to add more fields besides the ability > to read it but it's already fairly wide as it is. > Thats why we have "show channels as xml" > > > On Thu, Jul 16, 2009 at 9:50 AM, Michael S Collins > wrote: > > I wonder if it would make sense to create a separate sub-command like > "show channels stats" or something. That way we could put all sorts of > nifty info there without breaking the existing command. > > Thoughts? > -MC > > Sent from my iPhone > > On Jul 16, 2009, at 7:37 AM, Apostolos Pantsiopoulos > > > wrote: > > > Now that I come to think of it... > > > > It would be useful if we had the timestamp (and epoch) > > of the events PROGRESS and PROGRESS WITH MEDIA so that we can extract > > the PDD (Post Dial Delay) which is a very useful statistic. > > > > Adding the user's (for the incoming) and the gateway's (for the > > outbound) id would also be useful. In case these fields are empty the > > show channels command could ommit the (empty string). > > > > Are you planning to implement them yourselves or should I begin > > looking > > at the code? > > > > > > Anthony Minessale wrote: > >> I'm ok with the idea as long as it's thoroughly tested. > >> If there is any more info you want to save from those events you > >> should > >> consider it now while we are modifying it. > >> > >> > >> On Thu, Jul 16, 2009 at 9:02 AM, > > >> >> wrote: > >> > >> Hi, > >> > >> I usually find it very useful when I can retrieve a list of the > >> currents calls along with durations. I noticed that the 'show > >> channels' format does not include the duration (or the answered > >> timestamp - so that one can extract it from there). So, I made a > >> patch that includes the answered timestamp, the answered > timestamp > >> in epoch, and the duration in seconds. Of course these fields > >> remain > >> empty when the call hasn't been > >> answered yet. > >> > >> I don't know if anyone else finds this functionality useful, so > >> I am > >> posting this patch here first (instead of JIRA) in order to get > >> feedback from the users. If many of you (or the maintainers) > >> find it > >> interesting I can then proceed in posting it to JIRA. > >> > >> -- > >> ------------------------------------------- > >> Apostolos Pantsiopoulos > >> Kinetix Tele.com R & D > >> email: regs at kinetix.gr > > > >> ------------------------------------------- > >> > >> Index: src/mod/applications/mod_commands/mod_commands.c > >> > >> =================================================================== > >> --- src/mod/applications/mod_commands/mod_commands.c > >> (revision 14256) > >> +++ src/mod/applications/mod_commands/mod_commands.c (working > >> copy) > >> @@ -2827,10 +2827,10 @@ > >> } > >> } > >> if (strchr(argv[2], '%')) { > >> - sprintf(sql, "select * from > >> channels > >> where uuid like '%s' or name like '%s' or cid_name like '%s' or > >> cid_num like '%s' order by created_epoch", > >> + sprintf(sql, "select > >> *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration from > >> channels where uuid like '%s' or name like '%s' or cid_name like > >> '%s' or cid_num like '%s' order by created_epoch", > >> argv[2], argv[2], > >> argv[2], argv[2]); > >> } else { > >> - sprintf(sql, "select * from > >> channels > >> where uuid like '%%%s%%' or name like '%%%s%%' or cid_name like > >> '%%%s%%' or cid_num like '%%%s%%' order by created_epoch", > >> + sprintf(sql, "select > >> *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration from > >> channels where uuid like '%%%s%%' or name like '%%%s%%' or > >> cid_name > >> like '%%%s%%' or cid_num like '%%%s%%' order by created_epoch", > >> argv[2], argv[2], > >> argv[2], argv[2]); > >> > >> } > >> @@ -2839,10 +2839,10 @@ > >> as = argv[4]; > >> } > >> } else { > >> - sprintf(sql, "select * from channels > order > >> by created_epoch"); > >> + sprintf(sql, "select > >> *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration from > >> channels order by created_epoch"); > >> } > >> } else if (!strcasecmp(command, "channels")) { > >> - sprintf(sql, "select * from channels order by > >> created_epoch"); > >> + sprintf(sql, "select > >> *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration from > >> channels order by created_epoch"); > >> if (argv[1] && !strcasecmp(argv[1],"count")) { > >> holder.justcount = 1; > >> if (argv[3] && !strcasecmp(argv[2], "as")) { > >> @@ -2850,7 +2850,7 @@ > >> } > >> } > >> } else if (!strcasecmp(command, "distinct_channels")) { > >> - sprintf(sql, "select * from channels left join > >> calls > >> on " > >> + sprintf(sql, "select > >> *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration from > >> channels left join calls on " > >> "channels.uuid=calls.caller_uuid > >> where channels.uuid not in (select callee_uuid from calls) order > >> by > >> created_epoch"); > >> if (argv[2] && !strcasecmp(argv[1], "as")) { > >> as = argv[2]; > >> Index: src/switch_core_sqldb.c > >> > >> =================================================================== > >> --- src/switch_core_sqldb.c (revision 14256) > >> +++ src/switch_core_sqldb.c (working copy) > >> @@ -309,9 +309,21 @@ > >> ); > >> > >> break; > >> + case SWITCH_EVENT_CHANNEL_ANSWER: > >> + { > >> + > >> + sql = switch_mprintf("update channels set > >> answered='%s',answered_epoch='%ld' where uuid='%s'", > >> + > >> switch_event_get_header_nil(event, "event-date-local"), > >> + > >> (long)switch_epoch_time_now(NULL), > >> + > >> switch_event_get_header_nil(event, "unique-id") > >> + ); > >> + > >> + } > >> + break; > >> case SWITCH_EVENT_CHANNEL_STATE: > >> { > >> char *state = > >> switch_event_get_header_nil(event, "channel-state-number"); > >> + > >> switch_channel_state_t state_i = > >> CS_DESTROY; > >> > >> if (!switch_strlen_zero(state)) { > >> @@ -492,7 +504,9 @@ > >> " read_rate VARCHAR(255),\n" > >> " write_codec VARCHAR(255),\n" > >> " write_rate VARCHAR(255),\n" > >> - " secure VARCHAR(255)\n" > >> + " secure VARCHAR(255),\n" > >> + " answered VARCHAR(255),\n" > >> + " answered_epoch INTEGER\n" > >> ");\ncreate index uuindex on channels > >> (uuid);\n"; > >> char create_calls_sql[] = > >> "CREATE TABLE calls (\n" > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > > >> > > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > >> > >> > >> > >> -- > >> Anthony Minessale II > >> > >> FreeSWITCH http://www.freeswitch.org/ > >> ClueCon http://www.cluecon.com/ > >> Twitter: http://twitter.com/FreeSWITCH_wire > >> > >> AIM: anthm > >> MSN:anthony_minessale at hotmail.com > > >> > > >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > >> > > >> IRC: irc.freenode.net > #freeswitch > >> > >> FreeSWITCH Developer Conference > >> 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 > > > > > > -- > > ------------------------------------------- > > 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/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > 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 -- ------------------------------------------- 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 From dome at tel.co.th Thu Jul 16 08:38:37 2009 From: dome at tel.co.th (Dome Charoenyost) Date: Thu, 16 Jul 2009 22:38:37 +0700 Subject: [Freeswitch-users] sip extermal profile for all IP In-Reply-To: References: <8ccbff060907160630r2188c8d8od0c5c677f81eb71b@mail.gmail.com> Message-ID: <8ccbff060907160838y2d61e7a3me668250c18fb420@mail.gmail.com> I have eth0 and tap0 (vpn) FS bind only eth0 ip Dome C. 2009/7/16 : > I think by default it maps to all IPs unless you mention one here (by > replacing ${local_ip_v4} to some ip address). > > Thank you. > > > On Thu, Jul 16, 2009 at 7:30 PM, > wrote: >> >> Dear All, >> >> ? ? ? ? How to set ${local_ip_v4} to 0.0.0.0 ? i want to set up >> external profile for All IP >> >> >> Best regards. >> >> Dome C. >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > -- > Muhammad Shahzad > ----------------------------------- > CISCO Rich Media Communication Specialist (CRMCS) > CISCO Certified Network Associate (CCNA) > Cell: +92 334 422 40 88 > MSN: shari_786pk at hotmail.com > Email: shaheryarkh at googlemail.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 mrene_lists at avgs.ca Thu Jul 16 08:39:44 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Thu, 16 Jul 2009 11:39:44 -0400 Subject: [Freeswitch-users] "show channels" command with duration - patch included In-Reply-To: <9AEB5B242EBF4DD0A9B1544B7EAFC82C@saeedlaptop> References: <4A5F3306.1000300@kinetix.gr> <191c3a030907160715k74e785c8j99b68a42dc019bb7@mail.gmail.com> <4A5F3B19.8020507@kinetix.gr> <6079D960-F5EB-4774-93DA-87402CFC1157@freeswitch.org><191c3a030907160754g62aac895ge10a90476d5e890@mail.gmail.com> <4A5F42FD.60504@kinetix.gr> <9AEB5B242EBF4DD0A9B1544B7EAFC82C@saeedlaptop> Message-ID: If you want to use , within a { } block you can escape it with \, Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca Am 16-Jul-09 um 11:38 AM schrieb Saeed Ahmed: > Hi, > > its very useful feature for monitoring, I am doing it with XML RPC and > getting the result on webpage. > > There is one issue which is nothing to do with that patch but in > general: if > we are using absolute_codec_string variable and codes are like > G729,G723 > then this *comma* between codec ruin the array. I think there should > be > other separator. > > - Saeed > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Apostolos Pantsiopoulos > Sent: Thursday, July 16, 2009 5:11 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] "show channels" command with > duration - > patch included > > OK I 'll start implementing the progress timestamp field. > > The only reason I mentioned the user/gateway id field is that FS > admins > gain a lot by looking at a row of "show channels" result and be able > to > see who is the caller and who is the callee. > > Anthony Minessale wrote: >> it doesn't really break anything to add more fields besides the >> ability >> to read it but it's already fairly wide as it is. >> Thats why we have "show channels as xml" >> >> >> On Thu, Jul 16, 2009 at 9:50 AM, Michael S Collins >> > > wrote: >> >> I wonder if it would make sense to create a separate sub-command >> like >> "show channels stats" or something. That way we could put all >> sorts of >> nifty info there without breaking the existing command. >> >> Thoughts? >> -MC >> >> Sent from my iPhone >> >> On Jul 16, 2009, at 7:37 AM, Apostolos Pantsiopoulos >> > >> wrote: >> >>> Now that I come to think of it... >>> >>> It would be useful if we had the timestamp (and epoch) >>> of the events PROGRESS and PROGRESS WITH MEDIA so that we can > extract >>> the PDD (Post Dial Delay) which is a very useful statistic. >>> >>> Adding the user's (for the incoming) and the gateway's (for the >>> outbound) id would also be useful. In case these fields are empty > the >>> show channels command could ommit the (empty string). >>> >>> Are you planning to implement them yourselves or should I begin >>> looking >>> at the code? >>> >>> >>> Anthony Minessale wrote: >>>> I'm ok with the idea as long as it's thoroughly tested. >>>> If there is any more info you want to save from those events you >>>> should >>>> consider it now while we are modifying it. >>>> >>>> >>>> On Thu, Jul 16, 2009 at 9:02 AM, >> > >>>> > >> wrote: >>>> >>>> Hi, >>>> >>>> I usually find it very useful when I can retrieve a list of the >>>> currents calls along with durations. I noticed that the 'show >>>> channels' format does not include the duration (or the answered >>>> timestamp - so that one can extract it from there). So, I made > a >>>> patch that includes the answered timestamp, the answered >> timestamp >>>> in epoch, and the duration in seconds. Of course these fields >>>> remain >>>> empty when the call hasn't been >>>> answered yet. >>>> >>>> I don't know if anyone else finds this functionality useful, so >>>> I am >>>> posting this patch here first (instead of JIRA) in order to get >>>> feedback from the users. If many of you (or the maintainers) >>>> find it >>>> interesting I can then proceed in posting it to JIRA. >>>> >>>> -- >>>> ------------------------------------------- >>>> Apostolos Pantsiopoulos >>>> Kinetix Tele.com R & D >>>> email: regs at kinetix.gr >> > >>>> ------------------------------------------- >>>> >>>> Index: src/mod/applications/mod_commands/mod_commands.c >>>> >>>> > =================================================================== >>>> --- src/mod/applications/mod_commands/mod_commands.c >>>> (revision 14256) >>>> +++ src/mod/applications/mod_commands/mod_commands.c > (working >>>> copy) >>>> @@ -2827,10 +2827,10 @@ >>>> } >>>> } >>>> if (strchr(argv[2], '%')) { >>>> - sprintf(sql, "select * from >>>> channels >>>> where uuid like '%s' or name like '%s' or cid_name like '%s' or >>>> cid_num like '%s' order by created_epoch", >>>> + sprintf(sql, "select >>>> *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration > from >>>> channels where uuid like '%s' or name like '%s' or cid_name > like >>>> '%s' or cid_num like '%s' order by created_epoch", >>>> argv[2], > argv[2], >>>> argv[2], argv[2]); >>>> } else { >>>> - sprintf(sql, "select * from >>>> channels >>>> where uuid like '%%%s%%' or name like '%%%s%%' or cid_name like >>>> '%%%s%%' or cid_num like '%%%s%%' order by created_epoch", >>>> + sprintf(sql, "select >>>> *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration > from >>>> channels where uuid like '%%%s%%' or name like '%%%s%%' or >>>> cid_name >>>> like '%%%s%%' or cid_num like '%%%s%%' order by created_epoch", >>>> argv[2], > argv[2], >>>> argv[2], argv[2]); >>>> >>>> } >>>> @@ -2839,10 +2839,10 @@ >>>> as = argv[4]; >>>> } >>>> } else { >>>> - sprintf(sql, "select * from channels >> order >>>> by created_epoch"); >>>> + sprintf(sql, "select >>>> *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration > from >>>> channels order by created_epoch"); >>>> } >>>> } else if (!strcasecmp(command, "channels")) { >>>> - sprintf(sql, "select * from channels order by >>>> created_epoch"); >>>> + sprintf(sql, "select >>>> *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration > from >>>> channels order by created_epoch"); >>>> if (argv[1] && !strcasecmp(argv[1],"count")) { >>>> holder.justcount = 1; >>>> if (argv[3] && !strcasecmp(argv[2], "as")) { >>>> @@ -2850,7 +2850,7 @@ >>>> } >>>> } >>>> } else if (!strcasecmp(command, "distinct_channels")) { >>>> - sprintf(sql, "select * from channels left join >>>> calls >>>> on " >>>> + sprintf(sql, "select >>>> *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration > from >>>> channels left join calls on " >>>> "channels.uuid=calls.caller_uuid >>>> where channels.uuid not in (select callee_uuid from calls) > order >>>> by >>>> created_epoch"); >>>> if (argv[2] && !strcasecmp(argv[1], "as")) { >>>> as = argv[2]; >>>> Index: src/switch_core_sqldb.c >>>> >>>> > =================================================================== >>>> --- src/switch_core_sqldb.c (revision 14256) >>>> +++ src/switch_core_sqldb.c (working copy) >>>> @@ -309,9 +309,21 @@ >>>> ); >>>> >>>> break; >>>> + case SWITCH_EVENT_CHANNEL_ANSWER: >>>> + { >>>> + >>>> + sql = switch_mprintf("update channels > set >>>> answered='%s',answered_epoch='%ld' where uuid='%s'", >>>> + >>>> switch_event_get_header_nil(event, "event-date-local"), >>>> + >>>> (long)switch_epoch_time_now(NULL), >>>> + >>>> switch_event_get_header_nil(event, "unique-id") >>>> + ); >>>> + >>>> + } >>>> + break; >>>> case SWITCH_EVENT_CHANNEL_STATE: >>>> { >>>> char *state = >>>> switch_event_get_header_nil(event, "channel-state-number"); >>>> + >>>> switch_channel_state_t state_i = >>>> CS_DESTROY; >>>> >>>> if (!switch_strlen_zero(state)) { >>>> @@ -492,7 +504,9 @@ >>>> " read_rate VARCHAR(255),\n" >>>> " write_codec VARCHAR(255),\n" >>>> " write_rate VARCHAR(255),\n" >>>> - " secure VARCHAR(255)\n" >>>> + " secure VARCHAR(255),\n" >>>> + " answered VARCHAR(255),\n" >>>> + " answered_epoch INTEGER\n" >>>> ");\ncreate index uuindex on channels >>>> (uuid);\n"; >>>> char create_calls_sql[] = >>>> "CREATE TABLE calls (\n" >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >> >>>> > > >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> >>>> >>>> -- >>>> Anthony Minessale II >>>> >>>> FreeSWITCH http://www.freeswitch.org/ >>>> ClueCon http://www.cluecon.com/ >>>> Twitter: http://twitter.com/FreeSWITCH_wire >>>> >>>> AIM: anthm >>>> MSN:anthony_minessale at hotmail.com >> >>>> > > >>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> >>>> > > >>>> IRC: irc.freenode.net >> #freeswitch >>>> >>>> FreeSWITCH Developer Conference >>>> 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 >>> >>> >>> -- >>> ------------------------------------------- >>> 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/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> 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 > > > -- > ------------------------------------------- > 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 From regs at kinetix.gr Thu Jul 16 08:47:50 2009 From: regs at kinetix.gr (Apostolos Pantsiopoulos) Date: Thu, 16 Jul 2009 18:47:50 +0300 Subject: [Freeswitch-users] "show channels" command with duration - patch included In-Reply-To: <4A5F42FD.60504@kinetix.gr> References: <4A5F3306.1000300@kinetix.gr> <191c3a030907160715k74e785c8j99b68a42dc019bb7@mail.gmail.com> <4A5F3B19.8020507@kinetix.gr> <6079D960-F5EB-4774-93DA-87402CFC1157@freeswitch.org> <191c3a030907160754g62aac895ge10a90476d5e890@mail.gmail.com> <4A5F42FD.60504@kinetix.gr> Message-ID: <4A5F4BA6.50204@kinetix.gr> On a second thought... I am starting to agree with Michael. Nobody really collects stats about PDD from the 'show channels' command. This is usually done by using the cdrs. So, forget about the second request. The patch I sent earlier covers my needs (and I hope everybody else's too.) Apostolos Pantsiopoulos wrote: > OK I 'll start implementing the progress timestamp field. > > The only reason I mentioned the user/gateway id field is that FS admins > gain a lot by looking at a row of "show channels" result and be able to > see who is the caller and who is the callee. > > Anthony Minessale wrote: >> it doesn't really break anything to add more fields besides the ability >> to read it but it's already fairly wide as it is. >> Thats why we have "show channels as xml" >> >> >> On Thu, Jul 16, 2009 at 9:50 AM, Michael S Collins > > wrote: >> >> I wonder if it would make sense to create a separate sub-command like >> "show channels stats" or something. That way we could put all sorts of >> nifty info there without breaking the existing command. >> >> Thoughts? >> -MC >> >> Sent from my iPhone >> >> On Jul 16, 2009, at 7:37 AM, Apostolos Pantsiopoulos >> > >> wrote: >> >> > Now that I come to think of it... >> > >> > It would be useful if we had the timestamp (and epoch) >> > of the events PROGRESS and PROGRESS WITH MEDIA so that we can extract >> > the PDD (Post Dial Delay) which is a very useful statistic. >> > >> > Adding the user's (for the incoming) and the gateway's (for the >> > outbound) id would also be useful. In case these fields are empty the >> > show channels command could ommit the (empty string). >> > >> > Are you planning to implement them yourselves or should I begin >> > looking >> > at the code? >> > >> > >> > Anthony Minessale wrote: >> >> I'm ok with the idea as long as it's thoroughly tested. >> >> If there is any more info you want to save from those events you >> >> should >> >> consider it now while we are modifying it. >> >> >> >> >> >> On Thu, Jul 16, 2009 at 9:02 AM, >> > >> >> > >> wrote: >> >> >> >> Hi, >> >> >> >> I usually find it very useful when I can retrieve a list of the >> >> currents calls along with durations. I noticed that the 'show >> >> channels' format does not include the duration (or the answered >> >> timestamp - so that one can extract it from there). So, I made a >> >> patch that includes the answered timestamp, the answered >> timestamp >> >> in epoch, and the duration in seconds. Of course these fields >> >> remain >> >> empty when the call hasn't been >> >> answered yet. >> >> >> >> I don't know if anyone else finds this functionality useful, so >> >> I am >> >> posting this patch here first (instead of JIRA) in order to get >> >> feedback from the users. If many of you (or the maintainers) >> >> find it >> >> interesting I can then proceed in posting it to JIRA. >> >> >> >> -- >> >> ------------------------------------------- >> >> Apostolos Pantsiopoulos >> >> Kinetix Tele.com R & D >> >> email: regs at kinetix.gr >> > >> >> ------------------------------------------- >> >> >> >> Index: src/mod/applications/mod_commands/mod_commands.c >> >> >> >> =================================================================== >> >> --- src/mod/applications/mod_commands/mod_commands.c >> >> (revision 14256) >> >> +++ src/mod/applications/mod_commands/mod_commands.c (working >> >> copy) >> >> @@ -2827,10 +2827,10 @@ >> >> } >> >> } >> >> if (strchr(argv[2], '%')) { >> >> - sprintf(sql, "select * from >> >> channels >> >> where uuid like '%s' or name like '%s' or cid_name like '%s' or >> >> cid_num like '%s' order by created_epoch", >> >> + sprintf(sql, "select >> >> *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration from >> >> channels where uuid like '%s' or name like '%s' or cid_name like >> >> '%s' or cid_num like '%s' order by created_epoch", >> >> argv[2], argv[2], >> >> argv[2], argv[2]); >> >> } else { >> >> - sprintf(sql, "select * from >> >> channels >> >> where uuid like '%%%s%%' or name like '%%%s%%' or cid_name like >> >> '%%%s%%' or cid_num like '%%%s%%' order by created_epoch", >> >> + sprintf(sql, "select >> >> *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration from >> >> channels where uuid like '%%%s%%' or name like '%%%s%%' or >> >> cid_name >> >> like '%%%s%%' or cid_num like '%%%s%%' order by created_epoch", >> >> argv[2], argv[2], >> >> argv[2], argv[2]); >> >> >> >> } >> >> @@ -2839,10 +2839,10 @@ >> >> as = argv[4]; >> >> } >> >> } else { >> >> - sprintf(sql, "select * from channels >> order >> >> by created_epoch"); >> >> + sprintf(sql, "select >> >> *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration from >> >> channels order by created_epoch"); >> >> } >> >> } else if (!strcasecmp(command, "channels")) { >> >> - sprintf(sql, "select * from channels order by >> >> created_epoch"); >> >> + sprintf(sql, "select >> >> *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration from >> >> channels order by created_epoch"); >> >> if (argv[1] && !strcasecmp(argv[1],"count")) { >> >> holder.justcount = 1; >> >> if (argv[3] && !strcasecmp(argv[2], "as")) { >> >> @@ -2850,7 +2850,7 @@ >> >> } >> >> } >> >> } else if (!strcasecmp(command, "distinct_channels")) { >> >> - sprintf(sql, "select * from channels left join >> >> calls >> >> on " >> >> + sprintf(sql, "select >> >> *,strftime('%%s',DATETIME('NOW'))-answered_epoch as duration from >> >> channels left join calls on " >> >> "channels.uuid=calls.caller_uuid >> >> where channels.uuid not in (select callee_uuid from calls) order >> >> by >> >> created_epoch"); >> >> if (argv[2] && !strcasecmp(argv[1], "as")) { >> >> as = argv[2]; >> >> Index: src/switch_core_sqldb.c >> >> >> >> =================================================================== >> >> --- src/switch_core_sqldb.c (revision 14256) >> >> +++ src/switch_core_sqldb.c (working copy) >> >> @@ -309,9 +309,21 @@ >> >> ); >> >> >> >> break; >> >> + case SWITCH_EVENT_CHANNEL_ANSWER: >> >> + { >> >> + >> >> + sql = switch_mprintf("update channels set >> >> answered='%s',answered_epoch='%ld' where uuid='%s'", >> >> + >> >> switch_event_get_header_nil(event, "event-date-local"), >> >> + >> >> (long)switch_epoch_time_now(NULL), >> >> + >> >> switch_event_get_header_nil(event, "unique-id") >> >> + ); >> >> + >> >> + } >> >> + break; >> >> case SWITCH_EVENT_CHANNEL_STATE: >> >> { >> >> char *state = >> >> switch_event_get_header_nil(event, "channel-state-number"); >> >> + >> >> switch_channel_state_t state_i = >> >> CS_DESTROY; >> >> >> >> if (!switch_strlen_zero(state)) { >> >> @@ -492,7 +504,9 @@ >> >> " read_rate VARCHAR(255),\n" >> >> " write_codec VARCHAR(255),\n" >> >> " write_rate VARCHAR(255),\n" >> >> - " secure VARCHAR(255)\n" >> >> + " secure VARCHAR(255),\n" >> >> + " answered VARCHAR(255),\n" >> >> + " answered_epoch INTEGER\n" >> >> ");\ncreate index uuindex on channels >> >> (uuid);\n"; >> >> char create_calls_sql[] = >> >> "CREATE TABLE calls (\n" >> >> >> >> _______________________________________________ >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> >> > > >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> >> >> >> >> >> >> >> >> >> -- >> >> Anthony Minessale II >> >> >> >> FreeSWITCH http://www.freeswitch.org/ >> >> ClueCon http://www.cluecon.com/ >> >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> >> >> AIM: anthm >> >> MSN:anthony_minessale at hotmail.com >> >> >> > > >> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> >> >> > > >> >> IRC: irc.freenode.net >> #freeswitch >> >> >> >> FreeSWITCH Developer Conference >> >> 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 >> > >> > >> > -- >> > ------------------------------------------- >> > 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/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> 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 > > -- ------------------------------------------- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: regs at kinetix.gr ------------------------------------------- From fdhege at gmail.com Thu Jul 16 09:37:40 2009 From: fdhege at gmail.com (Dale) Date: Thu, 16 Jul 2009 12:37:40 -0400 Subject: [Freeswitch-users] Setting P-Asserted-ID to something other than the callerid In-Reply-To: References: Message-ID: <3953E116-DC6C-4324-B063-374CE37822F2@gmail.com> Who knows. I have asked why they are backwards when it comes to the RFC's. -Dale On Jul 16, 2009, at 10:41 AM, Ken Rice wrote: > That's just right down screwing with the standards... > > PAID is the caller id... This particular definition is from the RFCs > and > 3GPP docs for IMS which is why we have standardized P- headers... > > Can your vendor not look at the P-Charging-Vector field? > > Also, From when used with PAID is more like an ANI not a CLID > > >> From: Dale >> Reply-To: >> Date: Thu, 16 Jul 2009 10:30:34 -0400 >> To: >> Subject: [Freeswitch-users] Setting P-Asserted-ID to something >> other than the >> callerid >> >> >> Hello again, >> >> I wanted to first say thanks to Brain for helping me fix my from >> domain issue the other day. It helped quite a bit. >> >> Now with more testing and talking with the vendor (please don't shoot >> the messenger :) ) >> >> They want the caller id info in the from and the charge number/ >> screening number in the P-Asserted-ID. >> >> I have tested this and verified that this does work like they say it >> does by setting the callerid number to my charge number and setting >> the from user in the gateway config to the callerid I want displayed. >> But this solution doesn't scale very well. >> >> I know I can set the gateway option caller-id-in-from to get that >> part >> done. But is there a way to set the P-Asserted-ID to something other >> than the callerid? >> >> Any hints would be welcomed. >> >> Thanks, >> >> -Dale >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From fdhege at gmail.com Thu Jul 16 10:04:35 2009 From: fdhege at gmail.com (Dale) Date: Thu, 16 Jul 2009 13:04:35 -0400 Subject: [Freeswitch-users] Setting P-Asserted-ID to something other than the callerid In-Reply-To: References: Message-ID: Hey, you happen to have the RFC number and section they are violating? I'll dig it up. But figured I'd ask since you all seem to have them memorized by now. :) -Dale On Jul 16, 2009, at 10:41 AM, Ken Rice wrote: > That's just right down screwing with the standards... > > PAID is the caller id... This particular definition is from the RFCs > and > 3GPP docs for IMS which is why we have standardized P- headers... > > Can your vendor not look at the P-Charging-Vector field? > > Also, From when used with PAID is more like an ANI not a CLID > > >> From: Dale >> Reply-To: >> Date: Thu, 16 Jul 2009 10:30:34 -0400 >> To: >> Subject: [Freeswitch-users] Setting P-Asserted-ID to something >> other than the >> callerid >> >> >> Hello again, >> >> I wanted to first say thanks to Brain for helping me fix my from >> domain issue the other day. It helped quite a bit. >> >> Now with more testing and talking with the vendor (please don't shoot >> the messenger :) ) >> >> They want the caller id info in the from and the charge number/ >> screening number in the P-Asserted-ID. >> >> I have tested this and verified that this does work like they say it >> does by setting the callerid number to my charge number and setting >> the from user in the gateway config to the callerid I want displayed. >> But this solution doesn't scale very well. >> >> I know I can set the gateway option caller-id-in-from to get that >> part >> done. But is there a way to set the P-Asserted-ID to something other >> than the callerid? >> >> Any hints would be welcomed. >> >> Thanks, >> >> -Dale >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.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 Thu Jul 16 10:13:28 2009 From: sicfslist at gmail.com (Shelby Ramsey) Date: Thu, 16 Jul 2009 12:13:28 -0500 Subject: [Freeswitch-users] Setting P-Asserted-ID to something other than the callerid In-Reply-To: References: Message-ID: <35b355e90907161013j31fb4603r6d632fa0f4cc5abe@mail.gmail.com> PAI is defined @ http://www.ietf.org/rfc/rfc3325.txt For what they are trying to do Sonus suggested P-Charge-Info (which I think it is still in the draft stage) --> http://www.ietf.org/internet-drafts/draft-york-sipping-p-charge-info-06.txt -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090716/3f9eb95f/attachment-0001.html From marketing at cluecon.com Thu Jul 16 11:29:38 2009 From: marketing at cluecon.com (Michael Collins) Date: Thu, 16 Jul 2009 11:29:38 -0700 Subject: [Freeswitch-users] ClueCon 2009 - News and Notes Message-ID: <87f2f3b90907161129m29921db0q611bb87b791ab522@mail.gmail.com> Only 19 more days until ClueCon! If you haven't registered then please do so right away by calling 877.742.CLUE. This year's event is going to be awesome - you don't want to miss it! Some news and notes: We have two more sponsors for our event this year: Nokia and Twilio! We are happy to welcome them both to the conference this year. Many of our sponsors are supplying items for each attendee as well as some nice prizes that will be raffled off at the end of events on Thursday afternoon. We also have a very special guest who will be speaking at ClueCon: Philip Zimmermann! Philip is the author of PGP and co-author of ZRTP. We look forward to his presentation on Wednesday. Another security expert, Dan York of Voxeo, will also be speaking on Wednesday. All of those who are interested in security will want to pay special attention to Wednesday's program. Thanks for your support and we look forward to seeing you in a few weeks! -Michael Collins http://www.cluecon.com 877.742.CLUE -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090716/dc1c2cbd/attachment-0001.html From dave at 3c.co.uk Thu Jul 16 12:20:25 2009 From: dave at 3c.co.uk (David Knell) Date: Thu, 16 Jul 2009 16:20:25 -0300 Subject: [Freeswitch-users] Setting P-Asserted-ID to something other than the callerid In-Reply-To: References: <723C30E7-4AD0-418D-A4B4-10148367A9AA@gmail.com> <1247755878.4289.64.camel@dk-d820> Message-ID: <1247772025.4289.88.camel@dk-d820> Really? That line (+/- the IP address) came directly out of a working dialplan. To be fair, the box is running a faintly prehistoric FreeSWITCH - you crazy cats haven't been chewing on the tail of my cherished mouse of backwards compatibility again, have you?! What has been incorrect in this discussion is the name of the header: it's P-Asserted-Identity, not P-Asserted-ID. The fact that it's usually shortened to PAID doesn't help; nor does the fact that Remote-Party-ID (which is deprecated, but still widely used for the same job as P-Asserted-Identity) is about as well. --Dave > Kinda wrong there! > > > Gotta use CDATA because it has < and > in the data you're setting. > And you'll wanna export it I suspect. > > > ${caller_id_number}@1.2.3.4>]]> > > > /b > > > > > > On Jul 16, 2009, at 9:51 AM, David Knell wrote: > > > Hi Dale, > > > > You can set the header to anything you like by including something > > along > > the lines of > > > > in your dialplan. > > > > 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 -- David Knell, Director, 3C Limited T: +44 20 3298 2000 E: dave at 3c.co.uk W: http://www.3c.co.uk From brian at freeswitch.org Thu Jul 16 12:27:30 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 16 Jul 2009 14:27:30 -0500 Subject: [Freeswitch-users] Setting P-Asserted-ID to something other than the callerid In-Reply-To: <1247772025.4289.88.camel@dk-d820> References: <723C30E7-4AD0-418D-A4B4-10148367A9AA@gmail.com> <1247755878.4289.64.camel@dk-d820> <1247772025.4289.88.camel@dk-d820> Message-ID: Not sure how you were able to set the variable with < and > in it at all thats not been possible cuz the XML parser will barf on it usually. /b On Jul 16, 2009, at 2:20 PM, David Knell wrote: > Really? That line (+/- the IP address) came directly out of a working > dialplan. To be fair, the box is running a faintly prehistoric > FreeSWITCH - you crazy cats haven't been chewing on the tail of my > cherished mouse of backwards compatibility again, have you?! > > What has been incorrect in this discussion is the name of the header: > it's P-Asserted-Identity, not P-Asserted-ID. The fact that it's > usually > shortened to PAID doesn't help; nor does the fact that Remote-Party-ID > (which is deprecated, but still widely used for the same job as > P-Asserted-Identity) is about as well. > > --Dave From jens at vegeby.nu Thu Jul 16 12:36:35 2009 From: jens at vegeby.nu (Jens Vegeby) Date: Thu, 16 Jul 2009 21:36:35 +0200 Subject: [Freeswitch-users] ClueCon 2009 - News and Notes In-Reply-To: <87f2f3b90907161129m29921db0q611bb87b791ab522@mail.gmail.com> References: <87f2f3b90907161129m29921db0q611bb87b791ab522@mail.gmail.com> Message-ID: <30ee97110907161236n33bbfe39qcf1d8a009fded6b9@mail.gmail.com> Will there be video recordings availble online? On 7/16/09, Michael Collins wrote: > Only 19 more days until ClueCon! If you haven't registered then please do so > right away by calling 877.742.CLUE. This year's event is going to be awesome > - you don't want to miss it! > > Some news and notes: > We have two more sponsors for our event this year: Nokia and Twilio! We are > happy to welcome them both to the conference this year. Many of our sponsors > are supplying items for each attendee as well as some nice prizes that will > be raffled off at the end of events on Thursday afternoon. > > We also have a very special guest who will be speaking at ClueCon: Philip > Zimmermann! Philip is the author of PGP and co-author of ZRTP. We look > forward to his presentation on Wednesday. Another security expert, Dan York > of Voxeo, will also be speaking on Wednesday. All of those who are > interested in security will want to pay special attention to Wednesday's > program. > > Thanks for your support and we look forward to seeing you in a few weeks! > -Michael Collins > http://www.cluecon.com > 877.742.CLUE > -- Sent from my mobile device Mvh/Regards Jens From fdhege at gmail.com Thu Jul 16 12:42:59 2009 From: fdhege at gmail.com (Dale) Date: Thu, 16 Jul 2009 15:42:59 -0400 Subject: [Freeswitch-users] Setting P-Asserted-ID to something other than the callerid In-Reply-To: References: <723C30E7-4AD0-418D-A4B4-10148367A9AA@gmail.com> <1247755878.4289.64.camel@dk-d820> <1247772025.4289.88.camel@dk-d820> Message-ID: <53727858-AFDF-47EE-A0F8-5C69BB3DB204@gmail.com> Hmm, it seems to work on a copy of trunk from a few days ago. :) Both in the dialplan and the gateway config. But I could see how that might cause a problem with the xml. -Dale On Jul 16, 2009, at 3:27 PM, Brian West wrote: > Not sure how you were able to set the variable with < and > in it at > all thats not been possible cuz the XML parser will barf on it > usually. > > /b > > On Jul 16, 2009, at 2:20 PM, David Knell wrote: > >> Really? That line (+/- the IP address) came directly out of a >> working >> dialplan. To be fair, the box is running a faintly prehistoric >> FreeSWITCH - you crazy cats haven't been chewing on the tail of my >> cherished mouse of backwards compatibility again, have you?! >> >> What has been incorrect in this discussion is the name of the header: >> it's P-Asserted-Identity, not P-Asserted-ID. The fact that it's >> usually >> shortened to PAID doesn't help; nor does the fact that Remote-Party- >> ID >> (which is deprecated, but still widely used for the same job as >> P-Asserted-Identity) is about as well. >> >> --Dave > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.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 Jul 16 12:43:04 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 16 Jul 2009 14:43:04 -0500 Subject: [Freeswitch-users] ClueCon 2009 - News and Notes In-Reply-To: <30ee97110907161236n33bbfe39qcf1d8a009fded6b9@mail.gmail.com> References: <87f2f3b90907161129m29921db0q611bb87b791ab522@mail.gmail.com> <30ee97110907161236n33bbfe39qcf1d8a009fded6b9@mail.gmail.com> Message-ID: Yes their will be a couple of weeks after cluecon! /b On Jul 16, 2009, at 2:36 PM, Jens Vegeby wrote: > Will there be video recordings availble online? From msc at freeswitch.org Thu Jul 16 12:57:47 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 16 Jul 2009 12:57:47 -0700 Subject: [Freeswitch-users] Setting P-Asserted-ID to something other than the callerid In-Reply-To: <53727858-AFDF-47EE-A0F8-5C69BB3DB204@gmail.com> References: <723C30E7-4AD0-418D-A4B4-10148367A9AA@gmail.com> <1247755878.4289.64.camel@dk-d820> <1247772025.4289.88.camel@dk-d820> <53727858-AFDF-47EE-A0F8-5C69BB3DB204@gmail.com> Message-ID: <87f2f3b90907161257y65e39c07g595db93bd96b1fe3@mail.gmail.com> I think the parser is probably supposed to barf on consecutive < chars but it doesn't. I learned that the hard way when I left the closing tag off of an action in the dp. Consider this dp snippet: The above works but the 2nd log does not display. Just an FYI. -MC On Thu, Jul 16, 2009 at 12:42 PM, Dale wrote: > > Hmm, it seems to work on a copy of trunk from a few days ago. :) Both > in the dialplan and the gateway config. > > But I could see how that might cause a problem with the xml. > > -Dale > > On Jul 16, 2009, at 3:27 PM, Brian West wrote: > > > Not sure how you were able to set the variable with < and > in it at > > all thats not been possible cuz the XML parser will barf on it > > usually. > > > > /b > > > > On Jul 16, 2009, at 2:20 PM, David Knell wrote: > > > >> Really? That line (+/- the IP address) came directly out of a > >> working > >> dialplan. To be fair, the box is running a faintly prehistoric > >> FreeSWITCH - you crazy cats haven't been chewing on the tail of my > >> cherished mouse of backwards compatibility again, have you?! > >> > >> What has been incorrect in this discussion is the name of the header: > >> it's P-Asserted-Identity, not P-Asserted-ID. The fact that it's > >> usually > >> shortened to PAID doesn't help; nor does the fact that Remote-Party- > >> ID > >> (which is deprecated, but still widely used for the same job as > >> P-Asserted-Identity) is about as well. > >> > >> --Dave > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20090716/ac160dfc/attachment-0001.html From tayeb.meftah at gmail.com Thu Jul 16 13:34:27 2009 From: tayeb.meftah at gmail.com (Meftah Tayeb) Date: Thu, 16 Jul 2009 20:34:27 +0000 Subject: [Freeswitch-users] Freeswitch ASR application example Message-ID: <4A5F8ED3.1060900@gmail.com> hello please cool anyone give me a Speech ASR script in LUA ? only recognise the speech and put it into a variable thanks! __________ Information from ESET NOD32 Antivirus, version of virus signature database 4251 (20090716) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com From jens at vegeby.nu Thu Jul 16 13:51:47 2009 From: jens at vegeby.nu (Jens Vegeby) Date: Thu, 16 Jul 2009 22:51:47 +0200 Subject: [Freeswitch-users] ClueCon 2009 - News and Notes In-Reply-To: References: <87f2f3b90907161129m29921db0q611bb87b791ab522@mail.gmail.com> <30ee97110907161236n33bbfe39qcf1d8a009fded6b9@mail.gmail.com> Message-ID: <30ee97110907161351p47f98e38xdb623f6ef897859e@mail.gmail.com> Great! On 7/16/09, Brian West wrote: > Yes their will be a couple of weeks after cluecon! > > /b > > On Jul 16, 2009, at 2:36 PM, Jens Vegeby wrote: > >> Will there be video recordings availble online? > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 Mvh/Regards Jens From krice at suspicious.org Thu Jul 16 13:55:00 2009 From: krice at suspicious.org (Ken Rice) Date: Thu, 16 Jul 2009 15:55:00 -0500 Subject: [Freeswitch-users] Setting P-Asserted-ID to something other than the callerid In-Reply-To: Message-ID: Hey Dale RFC4694 defines how to pass JIP LRN and CIC (and a few other things) as part of the tel URI... This has been adapted for use in SIP as the user part of the invite and from URIs... Theres also RFC3325 for PAID and if you check out the wikipedia page for IMS (IP Multimedia Subsystem) there are several other RFCs that are related for various "P headers" there. Most of these headers come out from the 3GPP/IMS working groups > From: Dale > Reply-To: > Date: Thu, 16 Jul 2009 13:04:35 -0400 > To: > Subject: Re: [Freeswitch-users] Setting P-Asserted-ID to something other than > the callerid > > > Hey, you happen to have the RFC number and section they are violating? > > I'll dig it up. But figured I'd ask since you all seem to have them > memorized by now. :) > > -Dale > > On Jul 16, 2009, at 10:41 AM, Ken Rice wrote: > >> That's just right down screwing with the standards... >> >> PAID is the caller id... This particular definition is from the RFCs >> and >> 3GPP docs for IMS which is why we have standardized P- headers... >> >> Can your vendor not look at the P-Charging-Vector field? >> >> Also, From when used with PAID is more like an ANI not a CLID >> >> >>> From: Dale >>> Reply-To: >>> Date: Thu, 16 Jul 2009 10:30:34 -0400 >>> To: >>> Subject: [Freeswitch-users] Setting P-Asserted-ID to something >>> other than the >>> callerid >>> >>> >>> Hello again, >>> >>> I wanted to first say thanks to Brain for helping me fix my from >>> domain issue the other day. It helped quite a bit. >>> >>> Now with more testing and talking with the vendor (please don't shoot >>> the messenger :) ) >>> >>> They want the caller id info in the from and the charge number/ >>> screening number in the P-Asserted-ID. >>> >>> I have tested this and verified that this does work like they say it >>> does by setting the callerid number to my charge number and setting >>> the from user in the gateway config to the callerid I want displayed. >>> But this solution doesn't scale very well. >>> >>> I know I can set the gateway option caller-id-in-from to get that >>> part >>> done. But is there a way to set the P-Asserted-ID to something other >>> than the callerid? >>> >>> Any hints would be welcomed. >>> >>> Thanks, >>> >>> -Dale >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From pjintheusa at gmail.com Thu Jul 16 14:07:32 2009 From: pjintheusa at gmail.com (Phillip Jones) Date: Thu, 16 Jul 2009 17:07:32 -0400 Subject: [Freeswitch-users] .NET demo / hangupHook() Message-ID: <367751820907161407x57c9f81cwe4465ad1c1e20bb6@mail.gmail.com> Hi there, I am looking at the FreeSWITCH.Demo.AppDemo class. Have it running nicely from a soft phone - all is well. However I am not seeing the hangupHook() method fired, when I hangup. Debug log for an example call is at: http://pastebin.freeswitch.org/9744 Reminder of demo code is here: http://pastebin.freeswitch.org/9745 dialplan.xml is simply: Are there changes to the demo required to get this method firing? Any help much appreciated. Phillip Jones -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090716/f42ef880/attachment-0001.html From raul at etellicom.com Thu Jul 16 14:15:00 2009 From: raul at etellicom.com (Raul Fragoso) Date: Thu, 16 Jul 2009 18:15:00 -0300 Subject: [Freeswitch-users] sip extermal profile for all IP In-Reply-To: <8ccbff060907160630r2188c8d8od0c5c677f81eb71b@mail.gmail.com> References: <8ccbff060907160630r2188c8d8od0c5c677f81eb71b@mail.gmail.com> Message-ID: <1247778900.21971.8.camel@raul-laptop> You can not do that with a single profile. Each profile is bound to only one local IP, so if you need to bind to more than one you will have to create a new profile and set the specific sip-ip/rtp-ip params for them. Regards, RAul On Thu, 2009-07-16 at 20:30 +0700, freeswitch-users at lists.freeswitch.org wrote: > Dear All, > > How to set ${local_ip_v4} to 0.0.0.0 ? i want to set up > external profile for All IP > > > Best regards. > > Dome C. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From tayeb.meftah at gmail.com Thu Jul 16 14:17:05 2009 From: tayeb.meftah at gmail.com (Meftah Tayeb) Date: Thu, 16 Jul 2009 21:17:05 +0000 Subject: [Freeswitch-users] Freeswitch ASR application example Message-ID: <4A5F98D1.2060909@gmail.com> hello please cool anyone give me a Speech ASR script in LUA ? only recognise the speech and put it into a variable thanks! __________ Information from ESET NOD32 Antivirus, version of virus signature database 4251 (20090716) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com From mgg at giagnocavo.net Thu Jul 16 14:32:07 2009 From: mgg at giagnocavo.net (Michael Giagnocavo) Date: Thu, 16 Jul 2009 17:32:07 -0400 Subject: [Freeswitch-users] .NET demo / hangupHook() In-Reply-To: <367751820907161407x57c9f81cwe4465ad1c1e20bb6@mail.gmail.com> References: <367751820907161407x57c9f81cwe4465ad1c1e20bb6@mail.gmail.com> Message-ID: <6E8D2069C08AA84A83D336E996AE4C67027C6C33E7@mse17be1.mse17.exchange.ms> The debug log has this: 65.2009-07-16 16:48:41.432200 [DEBUG] switch_cpp.cpp:1124 AppFunction is in hangupCallback. 66.2009-07-16 16:48:41.432200 [WARNING] switch_cpp.cpp:1124 Thread will not be aborted because Hangup was called from the Run thread. The problem is the Demo doesn't have code to actually set hangupHook as the handler. Adding something like this to the app demo code: Session.HangupFunction = hangupHook; Should fix it. -Michael From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Phillip Jones Sent: Thursday, July 16, 2009 3:08 PM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] .NET demo / hangupHook() Hi there, I am looking at the FreeSWITCH.Demo.AppDemo class. Have it running nicely from a soft phone - all is well. However I am not seeing the hangupHook() method fired, when I hangup. Debug log for an example call is at: http://pastebin.freeswitch.org/9744 Reminder of demo code is here: http://pastebin.freeswitch.org/9745 dialplan.xml is simply: Are there changes to the demo required to get this method firing? Any help much appreciated. Phillip Jones -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090716/f64e98b5/attachment-0001.html From larclap at yahoo.com Thu Jul 16 14:36:14 2009 From: larclap at yahoo.com (Lars Zeb) Date: Thu, 16 Jul 2009 14:36:14 -0700 Subject: [Freeswitch-users] Error in lua script with session:getVariable Message-ID: <001801ca065d$71f669e0$55e33da0$@com> I am getting an error in a lua script which I don't understand. Why is it returning nil in the script yet something in the cli? lua snippet: user_data = session:getVariable('user_data 1000 at 192.168.10.29 var callgroup'); freeswitch.console_log("INFO", " UserData group " .. user_data .. "\n") log: 2009-07-16 14:30:49.237343 [ERR] mod_lua.cpp:182 /usr/local/freeswitch/scripts/helloworld.lua:14: attempt to concatenate global 'user_data' (a nil value) stack traceback: /usr/local/freeswitch/scripts/helloworld.lua:14: in main chunk cli: freeswitch at internal> user_data 1000 at 192.168.10.29 var callgroup techsupport Thanks, Lars -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090716/fb3edfb5/attachment-0001.html From mrene_lists at avgs.ca Thu Jul 16 14:42:19 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Thu, 16 Jul 2009 17:42:19 -0400 Subject: [Freeswitch-users] Error in lua script with session:getVariable In-Reply-To: <001801ca065d$71f669e0$55e33da0$@com> References: <001801ca065d$71f669e0$55e33da0$@com> Message-ID: <55B40657-61AE-46DE-952D-E5F496048B88@avgs.ca> You need to make an API call, not get a variable. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca Am 16-Jul-09 um 5:36 PM schrieb Lars Zeb: > I am getting an error in a lua script which I don?t understand. Why > is it returning nil in the script yet something in the cli? > > lua snippet: > user_data = session:getVariable('user_data 1000 at 192.168.10.29 var > callgroup'); > freeswitch.console_log("INFO", " UserData group " .. user_data .. > "\n") > > log: > 2009-07-16 14:30:49.237343 [ERR] mod_lua.cpp:182 /usr/local/ > freeswitch/scripts/helloworld.lua:14: attempt to concatenate global > 'user_data' (a nil value) > stack traceback: > /usr/local/freeswitch/scripts/helloworld.lua:14: in main chunk > > > cli: > freeswitch at internal> user_data 1000 at 192.168.10.29 var callgroup > techsupport > > Thanks, Lars > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20090716/68533c3d/attachment-0001.html From mgg at giagnocavo.net Thu Jul 16 14:43:08 2009 From: mgg at giagnocavo.net (Michael Giagnocavo) Date: Thu, 16 Jul 2009 17:43:08 -0400 Subject: [Freeswitch-users] mod_managed users? Message-ID: <6E8D2069C08AA84A83D336E996AE4C67027C6C33F0@mse17be1.mse17.exchange.ms> Hey, if there are any mod_managed users on this list, I'd love it if you were able to let me know. I'd like to get feedback, positive or negative, on what worked, what didn't, and how mod_managed can improve for you. Feel free to write on list or directly to me: mgg at giagnocavo.net Thanks! -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090716/de66d029/attachment-0001.html From dftoro at yahoo.com Thu Jul 16 14:54:30 2009 From: dftoro at yahoo.com (Diego Toro) Date: Thu, 16 Jul 2009 14:54:30 -0700 (PDT) Subject: [Freeswitch-users] mod_managed users? In-Reply-To: <6E8D2069C08AA84A83D336E996AE4C67027C6C33F0@mse17be1.mse17.exchange.ms> Message-ID: <36860.21084.qm@web33505.mail.mud.yahoo.com> Hey, I am here? :) ? I am working with mod_managed on Windows 2003 and Windows Vista with sucessfull.? I noted on user list the issue with LoadFile on Loader.cs when a assembly had reference to others assemblies, I change LoadFile by LoadFrom and the load is made fine. ? I use c# application and sqlserver 2005, using FS and mod_managed. ? Diego --- On Thu, 7/16/09, Michael Giagnocavo wrote: From: Michael Giagnocavo Subject: [Freeswitch-users] mod_managed users? To: "freeswitch-users at lists.freeswitch.org" Date: Thursday, July 16, 2009, 4:43 PM Hey, if there are any mod_managed users on this list, I?d love it if you were able to let me know. I?d like to get feedback, positive or negative, on what worked, what didn?t, and how mod_managed can improve for you. Feel free to write on list or directly to me: mgg at giagnocavo.net ? Thanks! -Michael -----Inline Attachment Follows----- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at 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/20090716/c759066b/attachment-0001.html From tayeb.meftah at gmail.com Thu Jul 16 14:56:00 2009 From: tayeb.meftah at gmail.com (Meftah Tayeb) Date: Thu, 16 Jul 2009 21:56:00 +0000 Subject: [Freeswitch-users] mod_managed users? In-Reply-To: <6E8D2069C08AA84A83D336E996AE4C67027C6C33F0@mse17be1.mse17.exchange.ms> References: <6E8D2069C08AA84A83D336E996AE4C67027C6C33F0@mse17be1.mse17.exchange.ms> Message-ID: <4A5FA1F0.90202@gmail.com> hello, yes, i'm trying to develope modules using Mod_Managed is working very perfectly, except that i don't know how i can run it using MONO (no .Net framework) also i'm trying to develope a EndPoint using it thanks Michael Giagnocavo wrote: > > Hey, if there are any mod_managed users on this list, I'd love it if > you were able to let me know. I'd like to get feedback, positive or > negative, on what worked, what didn't, and how mod_managed can improve > for you. Feel free to write on list or directly to me: mgg at > giagnocavo.net > > > > Thanks! > > -Michael > > > > __________ Information from ESET NOD32 Antivirus, version of virus > signature database 4251 (20090716) __________ > > The message was checked by ESET NOD32 Antivirus. > > http://www.eset.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 > > > > __________ Information from ESET NOD32 Antivirus, version of virus signature database 4251 (20090716) __________ > > The message was checked by ESET NOD32 Antivirus. > > http://www.eset.com > > __________ Information from ESET NOD32 Antivirus, version of virus signature database 4251 (20090716) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090716/c19f14f4/attachment-0001.html From pjintheusa at gmail.com Thu Jul 16 16:25:09 2009 From: pjintheusa at gmail.com (Phillip Jones) Date: Thu, 16 Jul 2009 19:25:09 -0400 Subject: [Freeswitch-users] .NET demo / hangupHook() In-Reply-To: <6E8D2069C08AA84A83D336E996AE4C67027C6C33E7@mse17be1.mse17.exchange.ms> References: <367751820907161407x57c9f81cwe4465ad1c1e20bb6@mail.gmail.com> <6E8D2069C08AA84A83D336E996AE4C67027C6C33E7@mse17be1.mse17.exchange.ms> Message-ID: <367751820907161625m1b040984ufe39e83d64bd16dc@mail.gmail.com> Perfect - thank you very much! On Thu, Jul 16, 2009 at 5:32 PM, Michael Giagnocavo wrote: > The debug log has this: > > 65.2009-07-16 16:48:41.432200 [DEBUG] switch_cpp.cpp:1124 AppFunction is in > hangupCallback. > > 66.2009-07-16 16:48:41.432200 [WARNING] switch_cpp.cpp:1124 Thread will not > be aborted because Hangup was called from the Run thread. > > > > The problem is the Demo doesn?t have code to actually set hangupHook as the > handler. Adding something like this to the app demo code: > > > > Session.HangupFunction = hangupHook; > > > > Should fix it. > > > > -Michael > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Phillip > Jones > *Sent:* Thursday, July 16, 2009 3:08 PM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* [Freeswitch-users] .NET demo / hangupHook() > > > > Hi there, > > I am looking at the FreeSWITCH.Demo.AppDemo class. Have it running nicely > from a soft phone - all is well. > > However I am not seeing the hangupHook() method fired, when I hangup. > > Debug log for an example call is at: http://pastebin.freeswitch.org/9744 > > Reminder of demo code is here: http://pastebin.freeswitch.org/9745 > > dialplan.xml is simply: > > > > > > > > Are there changes to the demo required to get this method firing? > > > Any help much appreciated. > > > Phillip Jones > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20090716/41e66cb9/attachment-0001.html From larclap at yahoo.com Thu Jul 16 17:07:09 2009 From: larclap at yahoo.com (Lars Zeb) Date: Thu, 16 Jul 2009 17:07:09 -0700 Subject: [Freeswitch-users] Error in lua script with session:getVariable In-Reply-To: <55B40657-61AE-46DE-952D-E5F496048B88@avgs.ca> References: <001801ca065d$71f669e0$55e33da0$@com> <55B40657-61AE-46DE-952D-E5F496048B88@avgs.ca> Message-ID: <006501ca0672$87039dc0$950ad940$@com> Mathieu, Thanks for the reply. I'm very new with FreeSWITCH and not familiar with api calls. I tried: user_data = apiExecute("user_data", "1000 at 192.168.10.29 var callgroup"); But got a similar error: 2009-07-16 17:01:47.212904 [ERR] mod_lua.cpp:182 /usr/local/freeswitch/scripts/helloworld.lua:13: attempt to call global 'apiExecute' (a nil value) stack traceback: /usr/local/freeswitch/scripts/helloworld.lua:13: in main chunk I would appreciate an example or a link to the pertinent documentation. Thanks, Lars From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mathieu Rene Sent: Thursday, July 16, 2009 2:42 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Error in lua script with session:getVariable You need to make an API call, not get a variable. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca Am 16-Jul-09 um 5:36 PM schrieb Lars Zeb: I am getting an error in a lua script which I don't understand. Why is it returning nil in the script yet something in the cli? lua snippet: user_data = session:getVariable('user_data 1000 at 192.168.10.29 var callgroup'); freeswitch.console_log("INFO", " UserData group " .. user_data .. "\n") log: 2009-07-16 14:30:49.237343 [ERR] mod_lua.cpp:182 /usr/local/freeswitch/scripts/helloworld.lua:14: attempt to concatenate global 'user_data' (a nil value) stack traceback: /usr/local/freeswitch/scripts/helloworld.lua:14: in main chunk cli: freeswitch at internal> user_data 1000 at 192.168.10.29 var callgroup techsupport Thanks, Lars _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at 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/20090716/13ab0e0a/attachment-0001.html From brian at freeswitch.org Thu Jul 16 17:11:28 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 16 Jul 2009 19:11:28 -0500 Subject: [Freeswitch-users] Error in lua script with session:getVariable In-Reply-To: <006501ca0672$87039dc0$950ad940$@com> References: <001801ca065d$71f669e0$55e33da0$@com> <55B40657-61AE-46DE-952D-E5F496048B88@avgs.ca> <006501ca0672$87039dc0$950ad940$@com> Message-ID: <0F52BC07-1CFE-4C0D-8A29-91D3C6F18D9B@freeswitch.org> http://wiki.freeswitch.org/wiki/Mod_lua#For_making_API_calls Please READ that page. These details are there! /b On Jul 16, 2009, at 7:07 PM, Lars Zeb wrote: > Mathieu, > > Thanks for the reply. I?m very new with FreeSWITCH and not familiar > with api calls. I tried: > > user_data = apiExecute("user_data", "1000 at 192.168.10.29 var > callgroup"); > > But got a similar error: > > 2009-07-16 17:01:47.212904 [ERR] mod_lua.cpp:182 /usr/local/ > freeswitch/scripts/helloworld.lua:13: attempt to call global > 'apiExecute' (a nil value) > stack traceback: > /usr/local/freeswitch/scripts/helloworld.lua:13: in main chunk > > I would appreciate an example or a link to the pertinent > documentation. > > Thanks, Lars > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Mathieu Rene > Sent: Thursday, July 16, 2009 2:42 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Error in lua script with > session:getVariable > > You need to make an API call, not get a variable. > > Mathieu Rene > Avant-Garde Solutions Inc > Office: + 1 (514) 664-1044 x100 > Cell: +1 (514) 664-1044 x200 > mrene at avgs.ca > > > > > Am 16-Jul-09 um 5:36 PM schrieb Lars Zeb: > > > I am getting an error in a lua script which I don?t understand. Why > is it returning nil in the script yet something in the cli? > > lua snippet: > user_data = session:getVariable('user_data 1000 at 192.168.10.29 var > callgroup'); > freeswitch.console_log("INFO", " UserData group " .. user_data .. > "\n") > > log: > 2009-07-16 14:30:49.237343 [ERR] mod_lua.cpp:182 /usr/local/ > freeswitch/scripts/helloworld.lua:14: attempt to concatenate global > 'user_data' (a nil value) > stack traceback: > /usr/local/freeswitch/scripts/helloworld.lua:14: in main chunk > > > cli: > freeswitch at internal> user_data 1000 at 192.168.10.29 var callgroup > techsupport > > Thanks, Lars > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20090716/f4c15437/attachment-0001.html From dome at tel.co.th Thu Jul 16 18:19:35 2009 From: dome at tel.co.th (Dome Charoenyost) Date: Fri, 17 Jul 2009 08:19:35 +0700 Subject: [Freeswitch-users] sip extermal profile for all IP In-Reply-To: <1247778900.21971.8.camel@raul-laptop> References: <8ccbff060907160630r2188c8d8od0c5c677f81eb71b@mail.gmail.com> <1247778900.21971.8.camel@raul-laptop> Message-ID: <8ccbff060907161819o4b2fc44cp5087ec5553cdd2f8@mail.gmail.com> 2009/7/17 Raul Fragoso : > You can not do that with a single profile. Each profile is bound to only > one local IP, so if you need to bind to more than one you will have to > create a new profile and set the specific sip-ip/rtp-ip params for them. > Thanks. > Regards, > > RAul > > On Thu, 2009-07-16 at 20:30 +0700, freeswitch-users at lists.freeswitch.org > wrote: >> Dear All, >> >> ? ? ? ? ?How to set ${local_ip_v4} to 0.0.0.0 ? i want to set up >> external profile for All IP >> >> >> Best regards. >> >> Dome C. >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From shaheryarkh at googlemail.com Thu Jul 16 21:59:53 2009 From: shaheryarkh at googlemail.com (Muhammad Shahzad) Date: Fri, 17 Jul 2009 10:59:53 +0600 Subject: [Freeswitch-users] mod_managed users? In-Reply-To: <36860.21084.qm@web33505.mail.mud.yahoo.com> References: <6E8D2069C08AA84A83D336E996AE4C67027C6C33F0@mse17be1.mse17.exchange.ms> <36860.21084.qm@web33505.mail.mud.yahoo.com> Message-ID: I tried to install mod_managed on ubuntu-9.04, mono framework version 2.0. It gave me a lots of errors in Loader.cs, which seems to be SWIG related. Since i am not a expert in SWIG so i disabled this module. This happend long ago, i think FS svn revision 136xx. Let me try to compile it from latest FS revision and see if it works. I will let you know the results. Thank you. On Fri, Jul 17, 2009 at 3:54 AM, Diego Toro wrote: > Hey, I am here :) > > I am working with mod_managed on Windows 2003 and Windows Vista with > sucessfull. I noted on user list the issue with LoadFile on Loader.cs when > a assembly had reference to others assemblies, I change LoadFile by LoadFrom > and the load is made fine. > > I use c# application and sqlserver 2005, using FS and mod_managed. > > Diego > > --- On *Thu, 7/16/09, Michael Giagnocavo * wrote: > > > From: Michael Giagnocavo > Subject: [Freeswitch-users] mod_managed users? > To: "freeswitch-users at lists.freeswitch.org" < > freeswitch-users at lists.freeswitch.org> > Date: Thursday, July 16, 2009, 4:43 PM > > Hey, if there are any mod_managed users on this list, I?d love it if you > were able to let me know. I?d like to get feedback, positive or negative, on > what worked, what didn?t, and how mod_managed can improve for you. Feel free > to write on list or directly to me: mgg at giagnocavo.net > > > > Thanks! > > -Michael > -----Inline Attachment Follows----- > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090717/bab4453a/attachment-0001.html From nicolas at medularis.com Fri Jul 17 13:35:34 2009 From: nicolas at medularis.com (Nicolas Brenner) Date: Fri, 17 Jul 2009 16:35:34 -0400 Subject: [Freeswitch-users] No credit = NETWORK_OUT_OF_ORDER ? Message-ID: <1b46b4e80907171335j729505c3oae8492b6582af8b8@mail.gmail.com> Hi, Today I ran out of credit in one of my voip providers. When this happened, all my outgoing calls started failing with hangup cause NETWORK_OUT_OF_ORDER. Once I got some more credit, the calls kept failing. I restarted freeswitch and then everything worked fine again. Unfortunately this is not something I'd like to reproduce, and the only thing I have is the logs (no SIP trace). But I was wodering if someone here has had a similar experience or could tell if this is plausible or even likely to happen. Another part of the platform I'm running, runs on Asterisk, using the same voip providers, nevertheless the calls originating there only failed during the no credit period, and began working again automatically as soon as credit was added to the account. Thanks, Nicolas -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090717/1d33b510/attachment-0001.html From brian at freeswitch.org Fri Jul 17 13:58:47 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 17 Jul 2009 15:58:47 -0500 Subject: [Freeswitch-users] No credit = NETWORK_OUT_OF_ORDER ? In-Reply-To: <1b46b4e80907171335j729505c3oae8492b6582af8b8@mail.gmail.com> References: <1b46b4e80907171335j729505c3oae8492b6582af8b8@mail.gmail.com> Message-ID: <305E86FA-95BC-4D04-837D-AAAC291B34B1@freeswitch.org> Open a jira with everything you can provide.. I'll try my best to reproduce the issue. /b On Jul 17, 2009, at 3:35 PM, Nicolas Brenner wrote: > Hi, > > Today I ran out of credit in one of my voip providers. When this > happened, all my outgoing calls started failing with hangup cause > NETWORK_OUT_OF_ORDER. Once I got some more credit, the calls kept > failing. I restarted freeswitch and then everything worked fine again. > > Unfortunately this is not something I'd like to reproduce, and the > only thing I have is the logs (no SIP trace). But I was wodering if > someone here has had a similar experience or could tell if this is > plausible or even likely to happen. > > Another part of the platform I'm running, runs on Asterisk, using > the same voip providers, nevertheless the calls originating there > only failed during the no credit period, and began working again > automatically as soon as credit was added to the account. > > Thanks, > > Nicolas > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.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 Fri Jul 17 14:13:26 2009 From: nicolas at medularis.com (Nicolas Brenner) Date: Fri, 17 Jul 2009 17:13:26 -0400 Subject: [Freeswitch-users] No credit = NETWORK_OUT_OF_ORDER ? In-Reply-To: <1b46b4e80907171335j729505c3oae8492b6582af8b8@mail.gmail.com> References: <1b46b4e80907171335j729505c3oae8492b6582af8b8@mail.gmail.com> Message-ID: <1b46b4e80907171413i260e70d4u43d4b1d88fe9a492@mail.gmail.com> A little bit more info: When the calls failed, the following was recorded in the log: 2009-07-17 15:19:07.880175 [ERR] switch_ivr_originate.c:1495 Cannot create outgoing channel of type [sofia] cause: [NETWORK_OUT_OF_ORDER] 2009-07-17 15:19:07.880175 [DEBUG] switch_ivr_originate.c:2123 Originate Resulted in Error Cause: 38 [NETWORK_OUT_OF_ORDER] 2009-07-17 15:19:07.880175 [WARNING] mod_spidermonkey.c:3013 Cannot Create Outgoing Channel! [{ignore_early_media=true,originate_timeout=30,execute_on_answer='sched_hangup +30 ALLOTED_TIMEOUT'}sofia/gateway/mygateway/005698793046] 2009-07-17 15:19:07.880175 [NOTICE] new_energizer_async.js:15 *********** CAUSE: NETWORK_OUT_OF_ORDER *********** 2009-07-17 15:19:34.158980 [NOTICE] sofia_reg.c:319 Registering mygateway 2009-07-17 15:19:34.294518 [ERR] sofia_reg.c:1445 mygateway Registration Failed with status Operation has no matching challenge [904]. failure #37 2009-07-17 15:19:34.365065 [WARNING] sofia_reg.c:348 mygateway Failed Registration, setting retry to 190 seconds. I searched for the "Registration Failed with status Operation has no matching challenge" error on the list, and someone else had a similar issue, but apparently it had something to do with NAT, and in this case there's no NAT involved. Anyway, I'm running a rev 13973 so I'll update to the latest svn rev and hope it doesn't happen again. On Fri, Jul 17, 2009 at 4:35 PM, Nicolas Brenner wrote: > Hi, > > Today I ran out of credit in one of my voip providers. When this happened, > all my outgoing calls started failing with hangup cause > NETWORK_OUT_OF_ORDER. Once I got some more credit, the calls kept failing. I > restarted freeswitch and then everything worked fine again. > > Unfortunately this is not something I'd like to reproduce, and the only > thing I have is the logs (no SIP trace). But I was wodering if someone here > has had a similar experience or could tell if this is plausible or even > likely to happen. > > Another part of the platform I'm running, runs on Asterisk, using the same > voip providers, nevertheless the calls originating there only failed during > the no credit period, and began working again automatically as soon as > credit was added to the account. > > Thanks, > > Nicolas > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090717/6e043f87/attachment-0001.html From lfurrea at gmail.com Fri Jul 17 16:58:12 2009 From: lfurrea at gmail.com (Luis F Urrea) Date: Fri, 17 Jul 2009 17:58:12 -0600 Subject: [Freeswitch-users] mod_conference: behavior when the conference is torn down before 31seconds Message-ID: Hi all, I am experiencing a behavior that I cannot clearly understand. Basically I "autocall" a few phones into a conference with the sip_auto_answer set to true, as follows: The conference establishes just fine and everyone can hear just fine. The "strange" behavior comes when the person calling to ext 773 hangs up before 31 seconds have passed, the rest of the phones stay up until they reach second 31 into the "conference". I am using snom phones and I see the BYE message arriving at the phones exactly at second 31 after the call establishes. The conference itself however does not exist after the person calling 773 hangs up (doing conference list on CLI shows NO active conferences). If the conference stays up more than 31 seconds, then when the person calling 773 hangs up, the rest of the phones hang up immediately as desired. Here's the log for a "page" that lasts less than 31 seconds: http://pastebin.freeswitch.org/9773 Here's the log of the phone for a "page" that lasts less than 31 seconds: http://pastebin.freeswitch.org/9774 Your inout is appreciated. Regards, Luis -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090717/f1ca8d62/attachment-0001.html From jaybinks at gmail.com Fri Jul 17 20:49:27 2009 From: jaybinks at gmail.com (Jay Binks) Date: Sat, 18 Jul 2009 13:49:27 +1000 Subject: [Freeswitch-users] 302 redirects and continue_on_fail=true Message-ID: <0A05EBC4-8DC0-4D4E-9029-E1BA2D5E443F@gmail.com> I have an upstream provider that utilizes a load balancer that spits back 302 redirects with contact headers SIP/2.0 302 Moved Temporarily Via: SIP/2.0/UDP xxx.xxx.xxx.xxx: 5080;rport=5080;branch=z9hG4bKB6peFDvXZ5S2F;received=xxx.xxx.xxx.xxx From: "test" ;tag=ByFF2244HHvmj To: ;tag=1288540-274799759-385876096-3219999652 Call-ID: f6dc8d30-edd7-122c-c98e-000e7f301839 CSeq: 117820807 INVITE Contact: Server: MERA MVTS3G v.3.10.2-49-Release Content-Length: 0 in my dialplan I have multiple upstream suppliers in a failover setup so I setup some vars and sip headers then attempt the bridge. if it fails I then go on to do the same thing for a few other suppliers ( setup headers, attempt bridge ) so because of this I use continue_on_fail=true it appears Freeswitch sees the 302 as a temp failure and does not follow the redirect, and instead moves on to the next upstream and bridges there. ive read that I can selectively exclude temporary failures from continue_on_fail but im not sure thats exact enough for this situation. I do wish for continue_on_fail to ignore 302 moved temporarily but not ALL temporary failures ( for which there are probably many more causes ) any help would be greatly appreciated as Im not sure the best way to resolve this. Jay -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090718/007c8854/attachment-0001.html From markmorreny at gmail.com Fri Jul 17 21:04:26 2009 From: markmorreny at gmail.com (mark morreny) Date: Sat, 18 Jul 2009 12:04:26 +0800 Subject: [Freeswitch-users] freeswitch on blackfin Message-ID: <20ad6b920907172104x3354cacal9567d91576b8ef75@mail.gmail.com> Hi, Have anyone tried getting freeswitch to work on uclinux/blackfin platform? Is there any info out there on how that can be done? Thanks for any info. Best Regards, Mark -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090718/328408bd/attachment-0001.html From jmesquita at gmail.com Fri Jul 17 21:44:13 2009 From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Sat, 18 Jul 2009 01:44:13 -0300 Subject: [Freeswitch-users] FsGUI Message-ID: <5a8712120907172144u991d4c7o2b743116877ef0c0@mail.gmail.com> Dear folks, Even tho it might be premature, I would like to already spread the word. Check out FsGUI and feel free give feedback if this is a wanted tool and what direction it should take. Beware that the code is still contrib code and might now be yet mature for production use. http://wiki.freeswitch.org/wiki/Fsgui Thanks, Jo?o Mesquita -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090718/a6215ed7/attachment-0001.html From steveu at coppice.org Fri Jul 17 22:19:12 2009 From: steveu at coppice.org (Steve Underwood) Date: Sat, 18 Jul 2009 13:19:12 +0800 Subject: [Freeswitch-users] freeswitch on blackfin In-Reply-To: <20ad6b920907172104x3354cacal9567d91576b8ef75@mail.gmail.com> References: <20ad6b920907172104x3354cacal9567d91576b8ef75@mail.gmail.com> Message-ID: <4A615B50.30703@coppice.org> mark morreny wrote: > Hi, > > Have anyone tried getting freeswitch to work on uclinux/blackfin > platform? > > Is there any info out there on how that can be done? > > Thanks for any info. > Look in the mailing list archive. This question comes up regularly. Steve From dome at tel.co.th Fri Jul 17 23:19:15 2009 From: dome at tel.co.th (Dome Charoenyost) Date: Sat, 18 Jul 2009 13:19:15 +0700 Subject: [Freeswitch-users] Nibble bill in B Leg Message-ID: <8ccbff060907172319s7cad6446o59887029a0c9dc42@mail.gmail.com> Dear sir, I found some problem when try to enable nibblebill in B-Leg My Dialplan problem is niblle do nothing until hanup call. i try to debug nibblebill and found some issue. nibblebill can't get billrate , billaccount from channel billrate = switch_channel_get_variable(channel, "nibble_rate"); billaccount = switch_channel_get_variable(channel, "nibble_account"); if (!billrate || !billaccount) { return SWITCH_STATUS_SUCCESS; } Dome C. From tayeb.meftah at gmail.com Sat Jul 18 02:25:24 2009 From: tayeb.meftah at gmail.com (Meftah Tayeb) Date: Sat, 18 Jul 2009 09:25:24 +0000 Subject: [Freeswitch-users] freeswitch on blackfin In-Reply-To: <20ad6b920907172104x3354cacal9567d91576b8ef75@mail.gmail.com> References: <20ad6b920907172104x3354cacal9567d91576b8ef75@mail.gmail.com> Message-ID: <4A619504.60503@gmail.com> hi mark, please go to the freeswitch web site (http://www.freeswitch.org and open Download / Install Guide. this contin a step by step guide to cross compil Freeswitch for Embedded system thanks; mark morreny wrote: > Hi, > > Have anyone tried getting freeswitch to work on uclinux/blackfin > platform? > > Is there any info out there on how that can be done? > > Thanks for any info. > > > Best Regards, > > 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 > > > > __________ Information from ESET NOD32 Antivirus, version of virus signature database 4256 (20090718) __________ > > The message was checked by ESET NOD32 Antivirus. > > http://www.eset.com > > __________ Information from ESET NOD32 Antivirus, version of virus signature database 4256 (20090718) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090718/c96c97d5/attachment-0001.html From dome at tel.co.th Sat Jul 18 02:48:22 2009 From: dome at tel.co.th (Dome Charoenyost) Date: Sat, 18 Jul 2009 16:48:22 +0700 Subject: [Freeswitch-users] Nibble bill in B Leg In-Reply-To: <8ccbff060907172319s7cad6446o59887029a0c9dc42@mail.gmail.com> References: <8ccbff060907172319s7cad6446o59887029a0c9dc42@mail.gmail.com> Message-ID: <8ccbff060907180248l6d5a2dbfoa127d928b62e34@mail.gmail.com> I found enable_heartbeat_events variable can help this case. it's work fine. I update http://wiki.freeswitch.org/wiki/Mod_nibblebill already Dome C. 2009/7/18 Dome Charoenyost : > Dear sir, > > ? ? ?I found some problem when try to enable nibblebill in B-Leg > ? ? ?My Dialplan > > ? ? ? data="{nibble_rate=1,nibble_account=0838833133}sofia/external/191$1 at 203.xxx.xxx.xxx" > /> > > ? ? ?problem is niblle do nothing until hanup call. i try to debug > nibblebill and found some issue. nibblebill can't get billrate , > billaccount from channel > > ? ? ? ?billrate = switch_channel_get_variable(channel, "nibble_rate"); > ? ? ? ?billaccount = switch_channel_get_variable(channel, "nibble_account"); > ? ? ? ?if (!billrate || !billaccount) { > ? ? ? ? ? ? ? ?return SWITCH_STATUS_SUCCESS; > ? ? ? ?} > > > Dome C. > From hads at nice.net.nz Sat Jul 18 02:59:44 2009 From: hads at nice.net.nz (Hadley Rich) Date: Sat, 18 Jul 2009 21:59:44 +1200 Subject: [Freeswitch-users] freeswitch on blackfin In-Reply-To: <4A619504.60503@gmail.com> References: <20ad6b920907172104x3354cacal9567d91576b8ef75@mail.gmail.com> <4A619504.60503@gmail.com> Message-ID: <1247911184.1988.8.camel@sodium> On Sat, 2009-07-18 at 09:25 +0000, Meftah Tayeb wrote: > please go to the freeswitch web site (http://www.freeswitch.org and > open Download / Install Guide. > this contin a step by step guide to cross compil Freeswitch for > Embedded system That's not for blackfin. hads -- http://nicegear.co.nz New Zealand's Open Source Hardware Supplier From Prometheus001 at gmx.net Sat Jul 18 05:01:23 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Sat, 18 Jul 2009 14:01:23 +0200 Subject: [Freeswitch-users] mod_conference: behavior when the conference is torn down before 31seconds In-Reply-To: References: Message-ID: <4A61B993.6090402@gmx.net> Hello Luis, are you using encrypted TLS instead on SIP on this phone? I experienced a similar behaviour with 31 seocnds on TLS. Best regards Peter Luis F Urrea schrieb: > Hi all, > > I am experiencing a behavior that I cannot clearly understand. > Basically I "autocall" a few phones into a conference with the > sip_auto_answer set to true, as follows: > > > > data="conference_auto_outcall_prefix={sip_auto_answer=true}"/> > data="user/305"/> > data="user/303"/> > data="user/201"/> > > > > > > The conference establishes just fine and everyone can hear just fine. > > The "strange" behavior comes when the person calling to ext 773 hangs > up before 31 seconds have passed, the rest of the phones stay up until > they reach second 31 into the "conference". > > I am using snom phones and I see the BYE message arriving at the > phones exactly at second 31 after the call establishes. > > The conference itself however does not exist after the person calling > 773 hangs up (doing conference list on CLI shows NO active conferences). > > If the conference stays up more than 31 seconds, then when the person > calling 773 hangs up, the rest of the phones hang up immediately as > desired. > > Here's the log for a "page" that lasts less than 31 seconds: > > http://pastebin.freeswitch.org/9773 > > Here's the log of the phone for a "page" that lasts less than 31 seconds: > > http://pastebin.freeswitch.org/9774 > > Your inout is appreciated. > > Regards, > > 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 Prometheus001 at gmx.net Sat Jul 18 05:06:00 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Sat, 18 Jul 2009 14:06:00 +0200 Subject: [Freeswitch-users] FsGUI In-Reply-To: <5a8712120907172144u991d4c7o2b743116877ef0c0@mail.gmail.com> References: <5a8712120907172144u991d4c7o2b743116877ef0c0@mail.gmail.com> Message-ID: <4A61BAA8.7080403@gmx.net> Thanks, I have found the sources in contrib/jmesquita/fsgui Any recommendatioins how to compile it under Linux? Best regards Peter Jo?o Mesquita schrieb: > Dear folks, > > Even tho it might be premature, I would like to already spread the word. > > Check out FsGUI and feel free give feedback if this is a wanted tool > and what direction it should take. Beware that the code is still > contrib code and might now be yet mature for production use. > > http://wiki.freeswitch.org/wiki/Fsgui > > Thanks, > > Jo?o Mesquita > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From shaheryarkh at googlemail.com Sat Jul 18 05:57:37 2009 From: shaheryarkh at googlemail.com (Muhammad Shahzad) Date: Sat, 18 Jul 2009 18:57:37 +0600 Subject: [Freeswitch-users] mod_managed users? In-Reply-To: References: <6E8D2069C08AA84A83D336E996AE4C67027C6C33F0@mse17be1.mse17.exchange.ms> <36860.21084.qm@web33505.mail.mud.yahoo.com> Message-ID: Sorry for replying late. I have tried mod_managed again on same machine (Lenovo 3000 N200), same OS (Ubuntu-9.04) and Mono version 2.0.1 but with latest FS revision 14249. It compiles correctly this time but gives following error upon "make install", ===================================================================== making install mod_managed make[5]: *** No rule to make target `/usr/local/freeswitch/mod/mod_managed.so', needed by `local_install'. Stop. make[4]: *** [install] Error 1 make[3]: *** [mod_managed-install] Error 1 make[2]: *** [install-recursive] Error 1 ===================================================================== Here is compilation log when executing "make", if it could of any help. ===================================================================== making all mod_managed Compiling freeswitch_managed.cpp... g++ -I/usr/src/svn-src/freeswitch/src/include -I/usr/src/svn-src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE -D_REENTRANT -pthread -I/usr/include/mono-1.0 -I/usr/include/glib-2.0 -I/usr/lib/glib-2.0/include -DHAVE_CONFIG_H -c -o freeswitch_managed.o freeswitch_managed.cpp Compiling freeswitch_wrap.cpp... g++ -I/usr/src/svn-src/freeswitch/src/include -I/usr/src/svn-src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE -D_REENTRANT -pthread -I/usr/include/mono-1.0 -I/usr/include/glib-2.0 -I/usr/lib/glib-2.0/include -DHAVE_CONFIG_H -c -o freeswitch_wrap.o freeswitch_wrap.cpp Demo.cs(58,14): warning CS0169: The private method `FreeSWITCH.Demo.AppDemo.hangupHook()' is never used Compilation succeeded - 1 warning(s) Compiling mod_managed.cpp... /usr/src/svn-src/freeswitch/libtool --mode=compile --tag=CXX g++ -I/usr/src/svn-src/freeswitch/src/include -I/usr/src/svn-src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE -D_REENTRANT -pthread -I/usr/include/mono-1.0 -I/usr/include/glib-2.0 -I/usr/lib/glib-2.0/include -DHAVE_CONFIG_H -c -o mod_managed.lo mod_managed.cpp libtool: compile: g++ -I/usr/src/svn-src/freeswitch/src/include -I/usr/src/svn-src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE -D_REENTRANT -pthread -I/usr/include/mono-1.0 -I/usr/include/glib-2.0 -I/usr/lib/glib-2.0/include -DHAVE_CONFIG_H -c mod_managed.cpp -fPIC -DPIC -o .libs/mod_managed.o mod_managed.cpp: In function ?void InitManagedSession(ManagedSession*, char* (*)(void*, switch_input_type_t), void (*)())?: mod_managed.cpp:97: warning: deprecated conversion from string constant to ?char*? libtool: compile: g++ -I/usr/src/svn-src/freeswitch/src/include -I/usr/src/svn-src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE -D_REENTRANT -pthread -I/usr/include/mono-1.0 -I/usr/include/glib-2.0 -I/usr/lib/glib-2.0/include -DHAVE_CONFIG_H -c mod_managed.cpp -o mod_managed.o >/dev/null 2>&1 Creating mod_managed.la... cat: .libs/mod_managed.log: No such file or directory ===================================================================== Thank you. On Fri, Jul 17, 2009 at 10:59 AM, Muhammad Shahzad < shaheryarkh at googlemail.com> wrote: > I tried to install mod_managed on ubuntu-9.04, mono framework version 2.0. > It gave me a lots of errors in Loader.cs, which seems to be SWIG related. > Since i am not a expert in SWIG so i disabled this module. This happend long > ago, i think FS svn revision 136xx. > > Let me try to compile it from latest FS revision and see if it works. I > will let you know the results. > > Thank you. > > > > On Fri, Jul 17, 2009 at 3:54 AM, Diego Toro wrote: > >> Hey, I am here :) >> >> I am working with mod_managed on Windows 2003 and Windows Vista with >> sucessfull. I noted on user list the issue with LoadFile on Loader.cs when >> a assembly had reference to others assemblies, I change LoadFile by LoadFrom >> and the load is made fine. >> >> I use c# application and sqlserver 2005, using FS and mod_managed. >> >> Diego >> >> --- On *Thu, 7/16/09, Michael Giagnocavo * wrote: >> >> >> From: Michael Giagnocavo >> Subject: [Freeswitch-users] mod_managed users? >> To: "freeswitch-users at lists.freeswitch.org" < >> freeswitch-users at lists.freeswitch.org> >> Date: Thursday, July 16, 2009, 4:43 PM >> >> Hey, if there are any mod_managed users on this list, I?d love it if you >> were able to let me know. I?d like to get feedback, positive or negative, on >> what worked, what didn?t, and how mod_managed can improve for you. Feel free >> to write on list or directly to me: mgg at giagnocavo.net >> >> >> >> Thanks! >> >> -Michael >> -----Inline Attachment Follows----- >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Muhammad Shahzad > ----------------------------------- > CISCO Rich Media Communication Specialist (CRMCS) > CISCO Certified Network Associate (CCNA) > Cell: +92 334 422 40 88 > MSN: shari_786pk at hotmail.com > Email: shaheryarkh at googlemail.com > -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090718/abd4e22b/attachment-0001.html From jmesquita at gmail.com Sat Jul 18 06:03:01 2009 From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Sat, 18 Jul 2009 10:03:01 -0300 Subject: [Freeswitch-users] FsGUI In-Reply-To: <4A61BAA8.7080403@gmx.net> References: <5a8712120907172144u991d4c7o2b743116877ef0c0@mail.gmail.com> <4A61BAA8.7080403@gmx.net> Message-ID: <5a8712120907180603g5a5053e6p9910e0d02f7cf18d@mail.gmail.com> Added to the wiki: http://wiki.freeswitch.org/wiki/Fsgui#Ubuntu jmesquita On Sat, Jul 18, 2009 at 9:06 AM, Peter P GMX wrote: > Thanks, > > I have found the sources in > contrib/jmesquita/fsgui > Any recommendatioins how to compile it under Linux? > > Best regards > Peter > > Jo?o Mesquita schrieb: > > Dear folks, > > > > Even tho it might be premature, I would like to already spread the word. > > > > Check out FsGUI and feel free give feedback if this is a wanted tool > > and what direction it should take. Beware that the code is still > > contrib code and might now be yet mature for production use. > > > > http://wiki.freeswitch.org/wiki/Fsgui > > > > Thanks, > > > > Jo?o Mesquita > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20090718/a43f23da/attachment-0001.html From d at unwire.it Sat Jul 18 10:04:31 2009 From: d at unwire.it (Darin Weeks) Date: Sat, 18 Jul 2009 10:04:31 -0700 Subject: [Freeswitch-users] FsGUI In-Reply-To: <5a8712120907180603g5a5053e6p9910e0d02f7cf18d@mail.gmail.com> References: <5a8712120907172144u991d4c7o2b743116877ef0c0@mail.gmail.com> <4A61BAA8.7080403@gmx.net> <5a8712120907180603g5a5053e6p9910e0d02f7cf18d@mail.gmail.com> Message-ID: <989132e70907181004u2c15449n2e445ae7489960fc@mail.gmail.com> Thanks guys! Can't wait to check it out. Would you mind adding some screenshots to the wiki? -- Darin 2009/7/18 Jo?o Mesquita > Added to the wiki: > > http://wiki.freeswitch.org/wiki/Fsgui#Ubuntu > > jmesquita > > > On Sat, Jul 18, 2009 at 9:06 AM, Peter P GMX wrote: > >> Thanks, >> >> I have found the sources in >> contrib/jmesquita/fsgui >> Any recommendatioins how to compile it under Linux? >> >> Best regards >> Peter >> >> Jo?o Mesquita schrieb: >> > Dear folks, >> > >> > Even tho it might be premature, I would like to already spread the word. >> > >> > Check out FsGUI and feel free give feedback if this is a wanted tool >> > and what direction it should take. Beware that the code is still >> > contrib code and might now be yet mature for production use. >> > >> > http://wiki.freeswitch.org/wiki/Fsgui >> > >> > Thanks, >> > >> > Jo?o Mesquita >> > ------------------------------------------------------------------------ >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- ) ) ) ) ) ) ) ) ) UNWIRE.IT ( ( ( ( ( ( ( ( ( broadband internet for west hollywood d at unwire.it http://unwire.it -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090718/9e506d72/attachment-0001.html From d at unwire.it Sat Jul 18 10:06:34 2009 From: d at unwire.it (Darin Weeks) Date: Sat, 18 Jul 2009 10:06:34 -0700 Subject: [Freeswitch-users] FsGUI In-Reply-To: <989132e70907181004u2c15449n2e445ae7489960fc@mail.gmail.com> References: <5a8712120907172144u991d4c7o2b743116877ef0c0@mail.gmail.com> <4A61BAA8.7080403@gmx.net> <5a8712120907180603g5a5053e6p9910e0d02f7cf18d@mail.gmail.com> <989132e70907181004u2c15449n2e445ae7489960fc@mail.gmail.com> Message-ID: <989132e70907181006h35a24c00h8f540096cbfc3f84@mail.gmail.com> DOH! Sorry... I just noticed the link at the bottom of the page to the graphic. Thanks! On Sat, Jul 18, 2009 at 10:04 AM, Darin Weeks wrote: > Thanks guys! Can't wait to check it out. Would you mind adding some > screenshots to the wiki? > > -- Darin > > 2009/7/18 Jo?o Mesquita > > Added to the wiki: >> >> http://wiki.freeswitch.org/wiki/Fsgui#Ubuntu >> >> jmesquita >> >> >> On Sat, Jul 18, 2009 at 9:06 AM, Peter P GMX wrote: >> >>> Thanks, >>> >>> I have found the sources in >>> contrib/jmesquita/fsgui >>> Any recommendatioins how to compile it under Linux? >>> >>> Best regards >>> Peter >>> >>> Jo?o Mesquita schrieb: >>> > Dear folks, >>> > >>> > Even tho it might be premature, I would like to already spread the >>> word. >>> > >>> > Check out FsGUI and feel free give feedback if this is a wanted tool >>> > and what direction it should take. Beware that the code is still >>> > contrib code and might now be yet mature for production use. >>> > >>> > http://wiki.freeswitch.org/wiki/Fsgui >>> > >>> > Thanks, >>> > >>> > Jo?o Mesquita >>> > >>> ------------------------------------------------------------------------ >>> > >>> > _______________________________________________ >>> > FreeSWITCH-users mailing list >>> > FreeSWITCH-users at lists.freeswitch.org >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> > UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> > http://www.freeswitch.org >>> > >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > > ) ) ) ) ) ) ) ) ) UNWIRE.IT ( ( ( ( ( ( ( ( ( > broadband internet for west hollywood > d at unwire.it http://unwire.it > -- ) ) ) ) ) ) ) ) ) UNWIRE.IT ( ( ( ( ( ( ( ( ( broadband internet for west hollywood d at unwire.it http://unwire.it -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090718/55ae3ccc/attachment-0001.html From mgg at giagnocavo.net Sat Jul 18 11:47:36 2009 From: mgg at giagnocavo.net (Michael Giagnocavo) Date: Sat, 18 Jul 2009 14:47:36 -0400 Subject: [Freeswitch-users] mod_managed users? In-Reply-To: References: <6E8D2069C08AA84A83D336E996AE4C67027C6C33F0@mse17be1.mse17.exchange.ms> <36860.21084.qm@web33505.mail.mud.yahoo.com> Message-ID: <6E8D2069C08AA84A83D336E996AE4C67027C7F85D6@mse17be1.mse17.exchange.ms> Thanks for this report. I?ll look into the linux build shortly and perhaps be able to help get it working. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Muhammad Shahzad Sent: Saturday, July 18, 2009 6:58 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_managed users? Sorry for replying late. I have tried mod_managed again on same machine (Lenovo 3000 N200), same OS (Ubuntu-9.04) and Mono version 2.0.1 but with latest FS revision 14249. It compiles correctly this time but gives following error upon "make install", ===================================================================== making install mod_managed make[5]: *** No rule to make target `/usr/local/freeswitch/mod/mod_managed.so', needed by `local_install'. Stop. make[4]: *** [install] Error 1 make[3]: *** [mod_managed-install] Error 1 make[2]: *** [install-recursive] Error 1 ===================================================================== Here is compilation log when executing "make", if it could of any help. ===================================================================== making all mod_managed Compiling freeswitch_managed.cpp... g++ -I/usr/src/svn-src/freeswitch/src/include -I/usr/src/svn-src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE -D_REENTRANT -pthread -I/usr/include/mono-1.0 -I/usr/include/glib-2.0 -I/usr/lib/glib-2.0/include -DHAVE_CONFIG_H -c -o freeswitch_managed.o freeswitch_managed.cpp Compiling freeswitch_wrap.cpp... g++ -I/usr/src/svn-src/freeswitch/src/include -I/usr/src/svn-src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE -D_REENTRANT -pthread -I/usr/include/mono-1.0 -I/usr/include/glib-2.0 -I/usr/lib/glib-2.0/include -DHAVE_CONFIG_H -c -o freeswitch_wrap.o freeswitch_wrap.cpp Demo.cs(58,14): warning CS0169: The private method `FreeSWITCH.Demo.AppDemo.hangupHook()' is never used Compilation succeeded - 1 warning(s) Compiling mod_managed.cpp... /usr/src/svn-src/freeswitch/libtool --mode=compile --tag=CXX g++ -I/usr/src/svn-src/freeswitch/src/include -I/usr/src/svn-src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE -D_REENTRANT -pthread -I/usr/include/mono-1.0 -I/usr/include/glib-2.0 -I/usr/lib/glib-2.0/include -DHAVE_CONFIG_H -c -o mod_managed.lo mod_managed.cpp libtool: compile: g++ -I/usr/src/svn-src/freeswitch/src/include -I/usr/src/svn-src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE -D_REENTRANT -pthread -I/usr/include/mono-1.0 -I/usr/include/glib-2.0 -I/usr/lib/glib-2.0/include -DHAVE_CONFIG_H -c mod_managed.cpp -fPIC -DPIC -o .libs/mod_managed.o mod_managed.cpp: In function ?void InitManagedSession(ManagedSession*, char* (*)(void*, switch_input_type_t), void (*)())?: mod_managed.cpp:97: warning: deprecated conversion from string constant to ?char*? libtool: compile: g++ -I/usr/src/svn-src/freeswitch/src/include -I/usr/src/svn-src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE -D_REENTRANT -pthread -I/usr/include/mono-1.0 -I/usr/include/glib-2.0 -I/usr/lib/glib-2.0/include -DHAVE_CONFIG_H -c mod_managed.cpp -o mod_managed.o >/dev/null 2>&1 Creating mod_managed.la... cat: .libs/mod_managed.log: No such file or directory ===================================================================== Thank you. On Fri, Jul 17, 2009 at 10:59 AM, Muhammad Shahzad > wrote: I tried to install mod_managed on ubuntu-9.04, mono framework version 2.0. It gave me a lots of errors in Loader.cs, which seems to be SWIG related. Since i am not a expert in SWIG so i disabled this module. This happend long ago, i think FS svn revision 136xx. Let me try to compile it from latest FS revision and see if it works. I will let you know the results. Thank you. On Fri, Jul 17, 2009 at 3:54 AM, Diego Toro > wrote: Hey, I am here :) I am working with mod_managed on Windows 2003 and Windows Vista with sucessfull. I noted on user list the issue with LoadFile on Loader.cs when a assembly had reference to others assemblies, I change LoadFile by LoadFrom and the load is made fine. I use c# application and sqlserver 2005, using FS and mod_managed. Diego --- On Thu, 7/16/09, Michael Giagnocavo > wrote: From: Michael Giagnocavo > Subject: [Freeswitch-users] mod_managed users? To: "freeswitch-users at lists.freeswitch.org" > Date: Thursday, July 16, 2009, 4:43 PM Hey, if there are any mod_managed users on this list, I?d love it if you were able to let me know. I?d like to get feedback, positive or negative, on what worked, what didn?t, and how mod_managed can improve for you. Feel free to write on list or directly to me: mgg at giagnocavo.net Thanks! -Michael -----Inline Attachment Follows----- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090718/c98b8292/attachment-0001.html From raul at etellicom.com Sat Jul 18 15:29:44 2009 From: raul at etellicom.com (Raul Fragoso) Date: Sat, 18 Jul 2009 19:29:44 -0300 Subject: [Freeswitch-users] 302 redirects and continue_on_fail=true In-Reply-To: <0A05EBC4-8DC0-4D4E-9029-E1BA2D5E443F@gmail.com> References: <0A05EBC4-8DC0-4D4E-9029-E1BA2D5E443F@gmail.com> Message-ID: <1247956185.30330.4.camel@raul-laptop> You can set continue_on_fail to the major (or all) causes that you want to handle and don't include 302 in those so it will continue to execute the extension, for example: continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION,CALL_REJECTED,USER_NOT_REGISTERED You can know more about the hangup causes here: http://wiki.freeswitch.org/wiki/Hangup_causes Regards, Raul On Sat, 2009-07-18 at 13:49 +1000, Jay Binks wrote: > I have an upstream provider that utilizes a load balancer that spits > back 302 redirects with contact headers > > > SIP/2.0 302 Moved Temporarily > Via: SIP/2.0/UDP > xxx.xxx.xxx.xxx:5080;rport=5080;branch=z9hG4bKB6peFDvXZ5S2F;received=xxx.xxx.xxx.xxx > From: "test" > ;tag=ByFF2244HHvmj > To: > ;tag=1288540-274799759-385876096-3219999652 > Call-ID: f6dc8d30-edd7-122c-c98e-000e7f301839 > CSeq: 117820807 INVITE > Contact: > Server: MERA MVTS3G v.3.10.2-49-Release > Content-Length: 0 > > > in my dialplan I have multiple upstream suppliers in a failover setup > so I setup some vars and sip headers then attempt the bridge. > if it fails I then go on to do the same thing for a few other > suppliers ( setup headers, attempt bridge ) so because of this I > use continue_on_fail=true > > > it appears Freeswitch sees the 302 as a temp failure and does not > follow the redirect, and instead moves on to the next upstream and > bridges there. > > > ive read that I can selectively exclude temporary failures from > continue_on_fail but im not sure thats exact enough for this > situation. > I do wish for continue_on_fail to ignore 302 moved temporarily but not > ALL temporary failures ( for which there are probably many more > causes ) > > > any help would be greatly appreciated as Im not sure the best way to > resolve this. > > > Jay > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.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 Sat Jul 18 16:55:09 2009 From: jaybinks at gmail.com (Jay Binks) Date: Sun, 19 Jul 2009 09:55:09 +1000 Subject: [Freeswitch-users] 302 redirects and continue_on_fail=true In-Reply-To: <1247956185.30330.4.camel@raul-laptop> References: <0A05EBC4-8DC0-4D4E-9029-E1BA2D5E443F@gmail.com> <1247956185.30330.4.camel@raul-laptop> Message-ID: Sure however my concern is that NORMAL_TEMPORARY_FAILURE is a generic failure and not only from 302 redirects. I'm not sure I'm game to do this, unless Im being completly paranoid which is possible. Can I get advice on this from a few Other users ? Jay On 19/07/2009, at 8:29 AM, Raul Fragoso wrote: > You can set continue_on_fail to the major (or all) causes that you > want > to handle and don't include 302 in those so it will continue to > execute > the extension, for example: > continue_on_fail= > NORMAL_TEMPORARY_FAILURE, > USER_BUSY, > NO_ANSWER, > TIMEOUT,NO_ROUTE_DESTINATION,CALL_REJECTED,USER_NOT_REGISTERED > > You can know more about the hangup causes here: > http://wiki.freeswitch.org/wiki/Hangup_causes > > Regards, > > Raul > > On Sat, 2009-07-18 at 13:49 +1000, Jay Binks wrote: >> I have an upstream provider that utilizes a load balancer that spits >> back 302 redirects with contact headers >> >> >> SIP/2.0 302 Moved Temporarily >> Via: SIP/2.0/UDP >> xxx.xxx.xxx.xxx: >> 5080;rport=5080;branch=z9hG4bKB6peFDvXZ5S2F;received=xxx.xxx.xxx.xxx >> From: "test" >> ;tag=ByFF2244HHvmj >> To: >> ;tag=1288540-274799759-385876096-3219999652 >> Call-ID: f6dc8d30-edd7-122c-c98e-000e7f301839 >> CSeq: 117820807 INVITE >> Contact: >> Server: MERA MVTS3G v.3.10.2-49-Release >> Content-Length: 0 >> >> >> in my dialplan I have multiple upstream suppliers in a failover setup >> so I setup some vars and sip headers then attempt the bridge. >> if it fails I then go on to do the same thing for a few other >> suppliers ( setup headers, attempt bridge ) so because of this I >> use continue_on_fail=true >> >> >> it appears Freeswitch sees the 302 as a temp failure and does not >> follow the redirect, and instead moves on to the next upstream and >> bridges there. >> >> >> ive read that I can selectively exclude temporary failures from >> continue_on_fail but im not sure thats exact enough for this >> situation. >> I do wish for continue_on_fail to ignore 302 moved temporarily but >> not >> ALL temporary failures ( for which there are probably many more >> causes ) >> >> >> any help would be greatly appreciated as Im not sure the best way to >> resolve this. >> >> >> Jay >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> users >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.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 Sun Jul 19 00:26:24 2009 From: woodydickson at gmail.com (Woody Dickson) Date: Sun, 19 Jul 2009 15:26:24 +0800 Subject: [Freeswitch-users] Can FreeSWITCH send and receive SIP MESSAGE Message-ID: Hi, I would like to use freeswitch as a gateway for sending and receiving short message. Does Freeswitch have the capability to send and recevie SIP MESSAGE? How can I set it up? I can't find any document on how to use Freeswitch for text message. Thanks, Woody -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090719/a938b903/attachment-0001.html From mcampbellsmith at gmail.com Sun Jul 19 05:11:28 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Sun, 19 Jul 2009 22:11:28 +1000 Subject: [Freeswitch-users] NAT'd FS / publice softphone problems Message-ID: <33c87fa30907190511s1c4a9a9fi71f03d6491efeb95@mail.gmail.com> Hi All, I know this question has come up before but I couldn't find the answer that I could understand! Sorry in advance. My setup is: Freeswtch NAT'd (192.168.x.x) -> Router -> Internet -> Softphone with public IP I can easily get the softphones to register, but when I try to call from the softphone to voicemail (for example), I don't get any audio. I checked out this page: http://wiki.freeswitch.org/wiki/External_profile (section Switch with External SoftPhone) but I am not clear how I can get this to work. I have played around with the rtp-ip and external-rtp-ip but without success. Is it possible for someone to help me configure this so softphones that are outside the nat'd lan get audio correctly? Help appreciated! Thanks From mike at jerris.com Sun Jul 19 08:35:43 2009 From: mike at jerris.com (Michael Jerris) Date: Sun, 19 Jul 2009 11:35:43 -0400 Subject: [Freeswitch-users] Can FreeSWITCH send and receive SIP MESSAGE In-Reply-To: References: Message-ID: See the chat_send api command and it also should just work with presence peers. On Jul 19, 2009, at 3:26 AM, Woody Dickson wrote: > Hi, > > I would like to use freeswitch as a gateway for sending and > receiving short message. > > Does Freeswitch have the capability to send and recevie SIP MESSAGE? > > How can I set it up? I can't find any document on how to use > Freeswitch for text message. > > 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 From lzwierko at gmail.com Sun Jul 19 04:27:13 2009 From: lzwierko at gmail.com (=?UTF-8?B?xYF1a2FzeiBad2llcmtv?=) Date: Sun, 19 Jul 2009 13:27:13 +0200 Subject: [Freeswitch-users] Dial up from confernece issue In-Reply-To: References: <0A05EBC4-8DC0-4D4E-9029-E1BA2D5E443F@gmail.com> <1247956185.30330.4.camel@raul-laptop> Message-ID: <4A630311.9010003@gmail.com> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I want to use bgdial command to add a person to a already started conference (that is, call that person and when answered - add the channel to conference). The scenario is I have two sip clients registered in default context - 1000 and 1001. 1000 dials conference number (3001 in this case) and new conference is started. I want to dial out to second using bgdial, unfortunately mod_sofia drops the call with 'Cannot Blind Transfer 1 Legged calls' message. Should I use the bgdial command differently? Or perhaps I should do this totally differently? Logs attached below. Thanks for any help, Lukasz freeswitch at Zwierko-laptop> conference list API CALL [conference(list)] output: Conference 3001-192.168.0.1 (1 member) 3;sofia/internal/1000 at 192.168.0.1;25f125a5-3556-a449-85ec-e4001336f313;1000;1000;hear|speak|floor;0;0;300 freeswitch at Zwierko-laptop> conference 3001-192.168.0.1 bgdial sofia/default/1001 at 192.168.0.1 API CALL [conference(3001-192.168.0.1 bgdial sofia/default/1001 at 192.168.0.1)] output: OK freeswitch at Zwierko-laptop> 2009-07-19 13:19:44.451100 [NOTICE] switch_channel.c:602 New Channel sofia/internal/1001 at 192.168.0.1 [b9fada7f-9c1d-4949-af8a-a8220ce f9c5b] 2009-07-19 13:19:44.459100 [NOTICE] switch_channel.c:602 New Channel sofia/internal/48228882211 at 192.168.0.1 [4f6b26dd-a0cb-2846-ad17-5f517e60e2e7] 2009-07-19 13:19:44.488100 [INFO] mod_dialplan_xml.c:252 Processing TelkaSwitch->1001 in context public 2009-07-19 13:19:44.495100 [ERR] sofia.c:4174 Cannot Blind Transfer 1 Legged calls 2009-07-19 13:19:44.498100 [NOTICE] sofia.c:3775 Hangup sofia/internal/1001 at 192.168.0.1 [CS_CONSUME_MEDIA] [NO_USER_RESPONSE] 2009-07-19 13:19:44.501100 [ERR] mod_conference.c:4351 Cannot create outgoing channel, cause: NO_USER_RESPONSE 2009-07-19 13:19:44.512100 [NOTICE] switch_core_session.c:1085 Session 9 (sofia/internal/1001 at 192.168.0.1) Ended 2009-07-19 13:19:44.512100 [NOTICE] switch_core_session.c:1087 Close Channel sofia/internal/1001 at 192.168.0.1 [CS_DESTROY] 2009-07-19 13:19:44.628100 [NOTICE] switch_core_state_machine.c:179 Hangup sofia/internal/48228882211 at 192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING] 2009-07-19 13:19:44.708100 [NOTICE] switch_core_session.c:1085 Session 10 (sofia/internal/48228882211 at 192.168.0.1) Ended 2009-07-19 13:19:44.708100 [NOTICE] switch_core_session.c:1087 Close Channel sofia/internal/48228882211 at 192.168.0.1 [CS_DESTROY] -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.9 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iQEcBAEBAgAGBQJKYwMRAAoJED7LBosr0F2ullAH/j6AebaezM2/RQ4PVKeNbMEm yWYqC1bDmp5F56owRH6Vq7BRnXKB4roqV2NLFqLNRYwzq/S4bzc9p417/NckrACg DhmZ6tFd4ujLb6B1HvJMTKsDvnYCpn5EVCbENfKVIY4INDAcEYbncwUA21XxILI+ ztz+6qNPwOMOjY9aZaf1qpTcTcG2yn62mpvesmVeYS1vNpZFVpnQq4PrukDg+1xs N8EpJetP0FxhYzT/IiD9fS2wAzQSgJPgo0m7R4ezk/1NIF9f+o0irgc8zx+VgKw1 UhJ1FLhs8ObzhYclvwJxwTlG+ppI28uIVO8EItiPB4/ZhEjyPfNVHcqTvMG6wb4= =3Bm/ -----END PGP SIGNATURE----- From carlos.talbot at gmail.com Sun Jul 19 12:46:47 2009 From: carlos.talbot at gmail.com (Carlos Talbot) Date: Sun, 19 Jul 2009 14:46:47 -0500 Subject: [Freeswitch-users] FsGUI In-Reply-To: <5a8712120907180603g5a5053e6p9910e0d02f7cf18d@mail.gmail.com> References: <5a8712120907172144u991d4c7o2b743116877ef0c0@mail.gmail.com> <4A61BAA8.7080403@gmx.net> <5a8712120907180603g5a5053e6p9910e0d02f7cf18d@mail.gmail.com> Message-ID: <5800526b0907191246i5377b6acx37c5f9b2bffe9a4f@mail.gmail.com> FYI, for those interested I've built an FsGui MSI file compiled for Windows via VS 2008 & the QT SDK library. It includes 2 necessary QT dlls. Future builds of the MSI for Freeswitch will include this. Here's the temporary link http://pbxinaflash.com/tusc/fsgui.msi (until I can sync it up to files.freeswitch.org) Carlos 2009/7/18 Jo?o Mesquita > Added to the wiki: > > http://wiki.freeswitch.org/wiki/Fsgui#Ubuntu > > jmesquita > > > On Sat, Jul 18, 2009 at 9:06 AM, Peter P GMX wrote: > >> Thanks, >> >> I have found the sources in >> contrib/jmesquita/fsgui >> Any recommendatioins how to compile it under Linux? >> >> Best regards >> Peter >> >> Jo?o Mesquita schrieb: >> > Dear folks, >> > >> > Even tho it might be premature, I would like to already spread the word. >> > >> > Check out FsGUI and feel free give feedback if this is a wanted tool >> > and what direction it should take. Beware that the code is still >> > contrib code and might now be yet mature for production use. >> > >> > http://wiki.freeswitch.org/wiki/Fsgui >> > >> > Thanks, >> > >> > Jo?o Mesquita >> > ------------------------------------------------------------------------ >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20090719/b5985077/attachment-0001.html From lzwierko at gmail.com Sun Jul 19 13:19:17 2009 From: lzwierko at gmail.com (=?UTF-8?B?xYF1a2FzeiBad2llcmtv?=) Date: Sun, 19 Jul 2009 22:19:17 +0200 Subject: [Freeswitch-users] Dial up from confernece issue Message-ID: <4A637FC5.6000301@gmail.com> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, sorry if you're getting this again, I'm not sure if this mail got deliverd to the mail-list (I didn't get a copy...) Anyway, I want to use bgdial command to add a person to a already started conference (that is, call that person and when answered - add the channel to conference). The scenario is I have two sip clients registered in default context - 1000 and 1001. 1000 dials conference number (3001 in this case) and new conference is started. I want to dial out to second using bgdial, unfortunately mod_sofia drops the call with 'Cannot Blind Transfer 1 Legged calls' message. Should I use the bgdial command differently? Or perhaps I should do this totally differently? Logs attached below. Thanks for any help, Lukasz freeswitch at Zwierko-laptop> conference list API CALL [conference(list)] output: Conference 3001-192.168.0.1 (1 member) 3;sofia/internal/1000 at 192.168.0.1;25f125a5-3556-a449-85ec-e4001336f313;1000;1000;hear|speak|floor;0;0;300 freeswitch at Zwierko-laptop> conference 3001-192.168.0.1 bgdial sofia/default/1001 at 192.168.0.1 API CALL [conference(3001-192.168.0.1 bgdial sofia/default/1001 at 192.168.0.1)] output: OK freeswitch at Zwierko-laptop> 2009-07-19 13:19:44.451100 [NOTICE] switch_channel.c:602 New Channel sofia/internal/1001 at 192.168.0.1 [b9fada7f-9c1d-4949-af8a-a8220ce f9c5b] 2009-07-19 13:19:44.459100 [NOTICE] switch_channel.c:602 New Channel sofia/internal/48228882211 at 192.168.0.1 [4f6b26dd-a0cb-2846-ad17-5f517e60e2e7] 2009-07-19 13:19:44.488100 [INFO] mod_dialplan_xml.c:252 Processing TelkaSwitch->1001 in context public 2009-07-19 13:19:44.495100 [ERR] sofia.c:4174 Cannot Blind Transfer 1 Legged calls 2009-07-19 13:19:44.498100 [NOTICE] sofia.c:3775 Hangup sofia/internal/1001 at 192.168.0.1 [CS_CONSUME_MEDIA] [NO_USER_RESPONSE] 2009-07-19 13:19:44.501100 [ERR] mod_conference.c:4351 Cannot create outgoing channel, cause: NO_USER_RESPONSE 2009-07-19 13:19:44.512100 [NOTICE] switch_core_session.c:1085 Session 9 (sofia/internal/1001 at 192.168.0.1) Ended 2009-07-19 13:19:44.512100 [NOTICE] switch_core_session.c:1087 Close Channel sofia/internal/1001 at 192.168.0.1 [CS_DESTROY] 2009-07-19 13:19:44.628100 [NOTICE] switch_core_state_machine.c:179 Hangup sofia/internal/48228882211 at 192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING] 2009-07-19 13:19:44.708100 [NOTICE] switch_core_session.c:1085 Session 10 (sofia/internal/48228882211 at 192.168.0.1) Ended 2009-07-19 13:19:44.708100 [NOTICE] switch_core_session.c:1087 Close Channel sofia/internal/48228882211 at 192.168.0.1 [CS_DESTROY] -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.9 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iQEcBAEBAgAGBQJKY3/FAAoJED7LBosr0F2uWo8H/iJEHblsAENCjRh/dsZvj9br Mq6txy7iafLE970XxvToaa0+FGBFxN+S6yQ6ampNPd8t+jl6WwC79Btwr+NLgXEc NcWpVQp65QxKxA+MgQOyqWIskcMcxdf4Uht3wuLPZtre0BpjcAFhykweYjOy1jFp AYAM61ShogHlpXtl9Z6upDvWPoOzdY4m13EM7f0NmpbC32Sg+OOULEtsxvSkZ8ah DBKDyDdXFo9iIcReDqjsu/kzAgrBsAZvOiEbSPoQTjZgzX+UrbgqIc+rhmP60vyt 8u8ufDgzh7MC/VQObHKHLe8e/Zbpaf+3JiGxZBtFyoUFyP3DjHoR5TYu5IsENPU= =QlRU -----END PGP SIGNATURE----- From brian at freeswitch.org Sun Jul 19 13:23:03 2009 From: brian at freeswitch.org (Brian West) Date: Sun, 19 Jul 2009 15:23:03 -0500 Subject: [Freeswitch-users] FsGUI In-Reply-To: <5800526b0907191246i5377b6acx37c5f9b2bffe9a4f@mail.gmail.com> References: <5a8712120907172144u991d4c7o2b743116877ef0c0@mail.gmail.com> <4A61BAA8.7080403@gmx.net> <5a8712120907180603g5a5053e6p9910e0d02f7cf18d@mail.gmail.com> <5800526b0907191246i5377b6acx37c5f9b2bffe9a4f@mail.gmail.com> Message-ID: <57FE57A0-24AE-462D-87B7-114D073C4F6D@freeswitch.org> Its sycned to files. now. /b On Jul 19, 2009, at 2:46 PM, Carlos Talbot wrote: > FYI, > > for those interested I've built an FsGui MSI file compiled for > Windows via VS 2008 & the QT SDK library. It includes 2 necessary QT > dlls. > > Future builds of the MSI for Freeswitch will include this. > > Here's the temporary link http://pbxinaflash.com/tusc/fsgui.msi > (until I can sync it up to files.freeswitch.org) > > Carlos > > 2009/7/18 Jo?o Mesquita > Added to the wiki: > > http://wiki.freeswitch.org/wiki/Fsgui#Ubuntu > > jmesquita > > > On Sat, Jul 18, 2009 at 9:06 AM, Peter P GMX > wrote: > Thanks, > > I have found the sources in > contrib/jmesquita/fsgui > Any recommendatioins how to compile it under Linux? > > Best regards > Peter > > Jo?o Mesquita schrieb: > > Dear folks, > > > > Even tho it might be premature, I would like to already spread the > word. > > > > Check out FsGUI and feel free give feedback if this is a wanted tool > > and what direction it should take. Beware that the code is still > > contrib code and might now be yet mature for production use. > > > > http://wiki.freeswitch.org/wiki/Fsgui > > > > Thanks, > > > > Jo?o Mesquita > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20090719/ba585ff9/attachment-0001.html From brian at freeswitch.org Sun Jul 19 13:24:09 2009 From: brian at freeswitch.org (Brian West) Date: Sun, 19 Jul 2009 15:24:09 -0500 Subject: [Freeswitch-users] Dial up from confernece issue In-Reply-To: <4A637FC5.6000301@gmail.com> References: <4A637FC5.6000301@gmail.com> Message-ID: <41A88E3D-C420-4A1F-8B80-098EA18F9544@freeswitch.org> If a single leg call gets a 302 you can't really "transfer" it anywhere... What SVN rev are you on? /b On Jul 19, 2009, at 3:19 PM, ?ukasz Zwierko wrote: > Hi, > > sorry if you're getting this again, I'm not sure if this mail got > deliverd to the mail-list (I didn't get a copy...) > > Anyway, > > I want to use bgdial command to add a person to a already started > conference (that is, call that person and when answered - add the > channel to conference). > > The scenario is I have two sip clients registered in default context - > 1000 and 1001. 1000 dials conference number (3001 in this case) and > new > conference is started. I want to dial out to second using bgdial, > unfortunately mod_sofia drops the call with 'Cannot Blind Transfer 1 > Legged calls' message. > > Should I use the bgdial command differently? Or perhaps I should do > this > totally differently? Logs attached below. > > Thanks for any help, > > Lukasz From jmesquita at gmail.com Sun Jul 19 14:33:45 2009 From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Sun, 19 Jul 2009 18:33:45 -0300 Subject: [Freeswitch-users] FsGUI In-Reply-To: <57FE57A0-24AE-462D-87B7-114D073C4F6D@freeswitch.org> References: <5a8712120907172144u991d4c7o2b743116877ef0c0@mail.gmail.com> <4A61BAA8.7080403@gmx.net> <5a8712120907180603g5a5053e6p9910e0d02f7cf18d@mail.gmail.com> <5800526b0907191246i5377b6acx37c5f9b2bffe9a4f@mail.gmail.com> <57FE57A0-24AE-462D-87B7-114D073C4F6D@freeswitch.org> Message-ID: <5a8712120907191433t2239f8fkf97fce67639415ea@mail.gmail.com> Thank you very much for your support. Brian, how can I put MacOSX dmg and linux binaries on files.freeswitch.org? jmesquita On Sun, Jul 19, 2009 at 5:23 PM, Brian West wrote: > Its sycned to files. now. > /b > > On Jul 19, 2009, at 2:46 PM, Carlos Talbot wrote: > > FYI, > for those interested I've built an FsGui MSI file compiled for Windows via > VS 2008 & the QT SDK library. It includes 2 necessary QT dlls. > > Future builds of the MSI for Freeswitch will include this. > > Here's the temporary link http://pbxinaflash.com/tusc/fsgui.msi (until I > can sync it up to files.freeswitch.org) > > Carlos > > 2009/7/18 Jo?o Mesquita > >> Added to the wiki: >> >> http://wiki.freeswitch.org/wiki/Fsgui#Ubuntu >> >> jmesquita >> >> >> On Sat, Jul 18, 2009 at 9:06 AM, Peter P GMX wrote: >> >>> Thanks, >>> >>> I have found the sources in >>> contrib/jmesquita/fsgui >>> Any recommendatioins how to compile it under Linux? >>> >>> Best regards >>> Peter >>> >>> Jo?o Mesquita schrieb: >>> > Dear folks, >>> > >>> > Even tho it might be premature, I would like to already spread the >>> word. >>> > >>> > Check out FsGUI and feel free give feedback if this is a wanted tool >>> > and what direction it should take. Beware that the code is still >>> > contrib code and might now be yet mature for production use. >>> > >>> > http://wiki.freeswitch.org/wiki/Fsgui >>> > >>> > Thanks, >>> > >>> > Jo?o Mesquita >>> > >>> ------------------------------------------------------------------------ >>> > >>> > _______________________________________________ >>> > FreeSWITCH-users mailing list >>> > FreeSWITCH-users at lists.freeswitch.org >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> > UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> > http://www.freeswitch.org >>> > >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20090719/1bb160f8/attachment-0001.html From msc at freeswitch.org Sun Jul 19 16:00:03 2009 From: msc at freeswitch.org (Michael S Collins) Date: Sun, 19 Jul 2009 16:00:03 -0700 Subject: [Freeswitch-users] NAT'd FS / publice softphone problems In-Reply-To: <33c87fa30907190511s1c4a9a9fi71f03d6491efeb95@mail.gmail.com> References: <33c87fa30907190511s1c4a9a9fi71f03d6491efeb95@mail.gmail.com> Message-ID: <4E741DC1-E045-4429-9CD4-E76941609FA3@freeswitch.org> What svn rev are you on? There have been some important changes recently. -MC Sent from my iPhone On Jul 19, 2009, at 5:11 AM, Mark Campbell-Smith wrote: > Hi All, > > I know this question has come up before but I couldn't find the answer > that I could understand! Sorry in advance. > > My setup is: > Freeswtch NAT'd (192.168.x.x) -> Router -> Internet -> Softphone > with public IP > > I can easily get the softphones to register, but when I try to call > from the softphone to voicemail (for example), I don't get any audio. > > I checked out this page: > http://wiki.freeswitch.org/wiki/External_profile (section Switch with > External SoftPhone) but I am not clear how I can get this to work. I > have played around with the rtp-ip and external-rtp-ip but without > success. > > Is it possible for someone to help me configure this so softphones > that are outside the nat'd lan get audio correctly? > > Help appreciated! > > 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 From yudha2008 at gmail.com Sun Jul 19 23:22:23 2009 From: yudha2008 at gmail.com (Baskar) Date: Mon, 20 Jul 2009 11:52:23 +0530 Subject: [Freeswitch-users] Freeswitch Windows Issues Message-ID: *Hi, I have installed freeswitch in windows, but when i star the freeswitch i get this error. Due to this i cant able to register my extension in softphone. how can i resolve this problem. 2009-07-20 10:55:05.390625 [CONSOLE] switch_core.c:1465 FreeSWITCH Version 1.0.trunk (13754M) Started. Crash Protection [Disabled] Max Sessions[1000] Session Rate[30] SQL [Enabled] freeswitch at Baskar> 2009-07-20 10:55:05.812500 [ERR] sofia.c:801 Error Creating SIP UA for profile: internal-ipv6 One more question in windows whether it is possible to connect the ODBC connection through JavaScript in freeswitch. I have configured inbound in Linux it is working fine but same script i tried in windows but i get this error. I have installed and configured MYSQL connector ODBC in window. But when is call the script i get this error. 2009-07-20 10:55:42.781250 [ERR] mod_spidermonkey.c:3393 Error loading ODBC 2009-07-20 10:55:42.781250 [ERR] Inbound1.js:5 ReferenceError: ODBC is not defined Can some one assist me to resolve this above error Thanks in advance. -- Thanks with Regards, N.Baskar * -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090720/43211174/attachment-0001.html From velu.technical at gmail.com Sun Jul 19 23:30:50 2009 From: velu.technical at gmail.com (velusamy velu) Date: Mon, 20 Jul 2009 12:00:50 +0530 Subject: [Freeswitch-users] Creating a new User Agent Message-ID: <1452e2980907192330o14904fb8ke0aec88fd6afde0a@mail.gmail.com> Dear All, I want to create a new User Agent like sip configurations in Asterisk. I checked default user agents 1000 to 1001. But I have bit confused the relationship between default user agents and sip_profiles. I need some help from you all for the following questions, How to create new user agent ? How to relate the new user agent with sip internal profile ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090720/4c66dd72/attachment-0001.html From helmut.kuper at ewetel.de Mon Jul 20 02:44:29 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Mon, 20 Jul 2009 11:44:29 +0200 Subject: [Freeswitch-users] Q931 TE State Timer In-Reply-To: <4A35F0F0.50406@ewetel.de> References: <4A3266E5.2000702@ewetel.de> <4A328E8E.6030607@freeswitch.org> <4A35F0F0.50406@ewetel.de> Message-ID: <4A643C7D.7010209@ewetel.de> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello community, to keep you informed about this. I got an early copy of the new openzap ISDN stack. I did some changes to get it running with my uplink provider. Most were dedicated to my uplink provider (AVAYA PBX, e.g. needed IEs in CALL_PROC, missed CON ACK, debug messages, ...) and a few were more general (like auto recover after a long term E1 link disconnect, not handled PROGRESS messages in cetrain call states, addind Timerhandlinf or call tear down) Since last friday I use it my prod environment (E1, Q931-TE, enBlock-Dialing). I have currently no problems. Timers work as expected and seems to keep the stack clean, although not all timers are implemented, yet ... So far Stefan and Tony did a very good job so far. Here is a snippet from my log showing the handling of my old originate problem of a missed RELEASE message from uplink after sending a DISCONNECT by the NEW stack: 2009-07-20 11:05:40.208698 [DEBUG] ozmod_isdn.c:1999 WRITE 9 - -------------------------------------------------------------------------------- [08 02 80 1e 45 08 02 81 90] 2009-07-20 11:05:40.208698 [NOTICE] Span:1 Starting timer 6 (timeout: 30000 ms) for call 30 [0x1e] 2009-07-20 11:05:40.208698 [DEBUG] Span:1 Call going from state 8 -> 11 2009-07-20 11:05:40.208698 [DEBUG] Span:1 Q931Rx43 return code: 0 [Waiting for RELEASE] 2009-07-20 11:06:10.279950 [NOTICE] Span:1 Stopping timer 6 for call 30 [0x1e] 2009-07-20 11:06:10.279950 [NOTICE] Span:1 Timer T305 timed out for call 30 2009-07-20 11:06:10.279950 [DEBUG] Span:1 Receiving message from Layer4 (size: 109, type: 77 [0x4d], CRV: 30, CRVFlag: 0) 2009-07-20 11:06:10.279950 [NOTICE] Span:1 Handling message from Layer4 in call state: 11 2009-07-20 11:06:10.279950 [DEBUG] Span:1 Processing RELEASE message from Local for CRV: 30 (0x1e) 2009-07-20 11:06:10.279950 [DEBUG] Span:1 Sending message to Q.921 (size: 109) 2009-07-20 11:06:10.279950 [DEBUG] Span:1 Creating Q.931 Message Header: ProtDisc 8 (0x8), CRV 30 (0x1e), CRVflag: 1 (0x1), MesType: 77 (0x4d) 2009-07-20 11:06:10.279950 [DEBUG] Span:1 XXX Adding IE 0x8 2009-07-20 11:06:10.279950 [DEBUG] Span:1 XXX Adding IE 0x28 2009-07-20 11:06:10.279950 [DEBUG] Span:1 XXX Adding IE 0x34 2009-07-20 11:06:10.279950 [DEBUG] Span:1 XXX Adding IE 0x1c 2009-07-20 11:06:10.279950 [NOTICE] Span:1 Unable to get reference for IE 28 (0x1c) 2009-07-20 11:06:10.279950 [DEBUG] Span:1 XXX Adding IE 0xd 2009-07-20 11:06:10.279950 [NOTICE] Span:1 Unable to get reference for IE 13 (0xd) 2009-07-20 11:06:10.279950 [DEBUG] ozmod_isdn.c:1999 WRITE 9 - -------------------------------------------------------------------------------- [08 02 80 1e 4d 08 02 81 e6] 2009-07-20 11:06:10.279950 [NOTICE] Span:1 Starting timer 9 (timeout: 4000 ms) for call 30 [0x1e] 2009-07-20 11:06:10.279950 [DEBUG] Span:1 Call going from state 11 -> 19 2009-07-20 11:06:10.279950 [DEBUG] Span:1 Q931Rx43 return code: 0 2009-07-20 11:06:10.419951 [DEBUG] ozmod_isdn.c:1181 READ 9 - -------------------------------------------------------------------------------- [08 02 00 1e 5a 08 02 81 d1] 2009-07-20 11:06:10.419951 [DEBUG] Span:1 Call is in state 19 [Release request] 2009-07-20 11:06:10.419951 [DEBUG] Span:1 Received message from Q.921 (ind 4, tei 0, size 13) MesType: 90, CRVFlag 0 (Originator), CRV 30 (Dialect: Q.931 TE) 2009-07-20 11:06:10.419951 [DEBUG] Span:1 Processing RELEASE COMPLETE message from Remote for CRV: 30 (0x1e) 2009-07-20 11:06:10.419951 [NOTICE] Span:1 Stopping timer 9 for call 30 [0x1e] 2009-07-20 11:06:10.419951 [DEBUG] Span:1 Sending message to Layer4 (size: 110) 2009-07-20 11:06:10.419951 [DEBUG] ozmod_isdn.c:696 Yay I got an event! Type:[5a] Size:[110] CRV: 30 (0x1e, CTX: Originator) 2009-07-20 11:06:10.419951 [DEBUG] ozmod_isdn.c:728 zchan 8124e08 (1:25) source isdn_data->channels_remote_crv[0x1e] 2009-07-20 11:06:10.419951 [DEBUG] ozmod_isdn.c:799 Received Release Complete message for channel 0 2009-07-20 11:06:10.419951 [DEBUG] ozmod_isdn.c:804 Changing state on 1:25 from HANGUP to DOWN :D A known bug is the handling of incomming calls aiming for non registered, but known targets. Currently caller isn't informed about that and so its call stays open. Will try to fix that. regards Helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) iD8DBQFKZDx94tZeNddg3dwRApjxAJ4tynu+xrzM3uMWlAhgnp2fdjmi0wCfZvgU 1wxVCgyskanESU0UxHQ4Ars= =4xqu -----END PGP SIGNATURE----- From tayeb.meftah at gmail.com Mon Jul 20 03:15:20 2009 From: tayeb.meftah at gmail.com (Meftah Tayeb) Date: Mon, 20 Jul 2009 10:15:20 +0000 Subject: [Freeswitch-users] Freeswitch Windows Issues In-Reply-To: References: Message-ID: <4A6443B8.5000106@gmail.com> hello, this is no an error this is only a problem if your Ipv6 is not installed go to your network connection (LAN) and install, select protocol and select Microsoft TCP/IP version 6 thanks Baskar wrote: > *Hi, > > I have installed freeswitch in windows, but when i star the > freeswitch i get this error. Due to this i cant able to register my > extension in softphone. how can i resolve this problem. > > 2009-07-20 10:55:05.390625 [CONSOLE] switch_core.c:1465 > FreeSWITCH Version 1.0.trunk (13754M) Started. > Crash Protection [Disabled] > Max Sessions[1000] > Session Rate[30] > SQL [Enabled] > > freeswitch at Baskar> 2009-07-20 10:55:05.812500 [ERR] sofia.c:801 Error > Creating SIP UA for profile: internal-ipv6 > > One more question in windows whether it is possible to connect the > ODBC connection through JavaScript in freeswitch. > > I have configured inbound in Linux it is working fine but same script > i tried in windows but i get this error. I have installed and > configured MYSQL connector ODBC in window. But when is call the script > i get this error. > > 2009-07-20 10:55:42.781250 [ERR] mod_spidermonkey.c:3393 Error loading > ODBC > 2009-07-20 10:55:42.781250 [ERR] Inbound1.js:5 ReferenceError: ODBC > is not defined > > Can some one assist me to resolve this above error > > Thanks in advance. > > -- > Thanks with 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 > > > > __________ Information from ESET NOD32 Antivirus, version of virus signature database 4260 (20090720) __________ > > The message was checked by ESET NOD32 Antivirus. > > http://www.eset.com > > __________ Information from ESET NOD32 Antivirus, version of virus signature database 4260 (20090720) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090720/6ff9bb19/attachment-0001.html From yudha2008 at gmail.com Mon Jul 20 03:21:11 2009 From: yudha2008 at gmail.com (Baskar) Date: Mon, 20 Jul 2009 15:51:11 +0530 Subject: [Freeswitch-users] Freeswitch Windows Issues In-Reply-To: <4A6443B8.5000106@gmail.com> References: <4A6443B8.5000106@gmail.com> Message-ID: *Hi Meftah Tayeb**,* *One more question in windows whether it is possible to connect the ODBC connection through JavaScript in freeswitch. I have configured inbound in Linux it is working fine but same script i tried in windows but i get this error. I have installed and configured MYSQL connector ODBC in window. But when is call the script i get this error. 2009-07-20 10:55:42.781250 [ERR] mod_spidermonkey.c:3393 Error loading ODBC 2009-07-20 10:55:42.781250 [ERR] Inbound1.js:5 ReferenceError: ODBC is not defined Can some one assist me to resolve this above error Thanks in advance. -- Thanks with Regards, N.Baskar * -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090720/a5af8dbc/attachment-0001.html From regs at kinetix.gr Mon Jul 20 04:01:58 2009 From: regs at kinetix.gr (Apostolos Pantsiopoulos) Date: Mon, 20 Jul 2009 14:01:58 +0300 Subject: [Freeswitch-users] Possible memory leak - need a second opinion Message-ID: <4A644EA6.70209@kinetix.gr> Hi I noticed that after a day of relatively moderate traffic (about 400 simultaneous channels average) the memory used by FS reached 1.3 GB of RAM. I tried to trace the leak (if any) with valgrind and got that : ==18894== 572 bytes in 1 blocks are definitely lost in loss record 148 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x7AFDE93: sofia_glue_do_invite (sofia_glue.c:1599) ==18894== by 0x7ACB191: sofia_on_init (mod_sofia.c:102) ==18894== by 0x405B124: switch_core_session_run (switch_core_state_machine.c:480) ==18894== by 0x4058204: switch_core_session_thread (switch_core_session.c:1064) ==18894== by 0x40CD6C5: dummy_worker (thread.c:138) ==18894== by 0xB5346A: start_thread (in /lib/libpthread-2.5.so) ==18894== by 0xAAADBD: clone (in /lib/libc-2.5.so) So, I assummed that this happens for every call. I tried testing it again by placing two calls before shutting down FS, but it only came up once. I wanted to get a second opinion before posting this to JIRA as an issue. I used revision 14269 of the SVN. I am attaching the valgrind output as well. I also noticed that only one of my CPU cores gets really busy when dealing with moderate traffic. From what I read in the mailing list users are encouraged to use 64bit multi core servers for FS because it scales up better. But this is not what I am seeing in practice. Could the single threaded architecture of libsofia be the cause of that behavior? -- ------------------------------------------- Apostolos Pantsiopoulos Kinetix Tele.com R & D email: regs at kinetix.gr ------------------------------------------- -------------- next part -------------- ==18894== Memcheck, a memory error detector. ==18894== Copyright (C) 2002-2006, and GNU GPL'd, by Julian Seward et al. ==18894== Using LibVEX rev 1658, a library for dynamic binary translation. ==18894== Copyright (C) 2004-2006, and GNU GPL'd, by OpenWorks LLP. ==18894== Using valgrind-3.2.1, a dynamic binary instrumentation framework. ==18894== Copyright (C) 2000-2006, and GNU GPL'd, by Julian Seward et al. ==18894== For more details, rerun with: -v ==18894== ==18894== My PID = 18894, parent PID = 18741. Prog and args are: ==18894== /usr/local/bin/freeswitch ==18894== -vg ==18894== ==18894== Thread 13: ==18894== Syscall param epoll_ctl(event) points to uninitialised byte(s) ==18894== at 0xAAB3AE: epoll_ctl (in /lib/libc-2.5.so) ==18894== by 0x403E62C: switch_pollset_add (switch_apr.c:802) ==18894== by 0x403E72D: switch_socket_create_pollfd (switch_apr.c:842) ==18894== by 0x407A785: switch_rtp_set_local_address (switch_rtp.c:824) ==18894== by 0x407AD82: switch_rtp_new (switch_rtp.c:1242) ==18894== by 0x7AF8C74: sofia_glue_activate_rtp (sofia_glue.c:2322) ==18894== by 0x7ADECA5: sofia_event_callback (sofia.c:3772) ==18894== by 0x7B72DD8: nua_application_event (nua_stack.c:393) ==18894== by 0x7BB784C: su_base_port_execute_msgs (su_base_port.c:280) ==18894== by 0x7BB75F3: su_base_port_getmsgs (su_base_port.c:202) ==18894== by 0x7BB7B63: su_base_port_step (su_base_port.c:473) ==18894== by 0x7BC863A: su_port_step (su_port.h:340) ==18894== Address 0xCE52B58 is on thread 13's stack ==18894== ==18894== ERROR SUMMARY: 2 errors from 1 contexts (suppressed: 281 from 1) ==18894== malloc/free: in use at exit: 927,131 bytes in 606 blocks. ==18894== malloc/free: 360,322 allocs, 359,716 frees, 661,660,377 bytes allocated. ==18894== For counts of detected errors, rerun with: -v ==18894== searching for pointers to 606 not-freed blocks. ==18894== checked 22,451,028 bytes. ==18894== ==18894== Thread 1: ==18894== ==18894== 3 bytes in 1 blocks are still reachable in loss record 1 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0xA48DEF: strdup (in /lib/libc-2.5.so) ==18894== by 0x127B7848: PR_NewLogModule (prlog.c:363) ==18894== by 0x127BE34B: _PR_InitStuff (prinit.c:193) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== ==18894== ==18894== 3 bytes in 1 blocks are still reachable in loss record 2 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0xA48DEF: strdup (in /lib/libc-2.5.so) ==18894== by 0x127B7848: PR_NewLogModule (prlog.c:363) ==18894== by 0x127BE2BB: _PR_InitStuff (prinit.c:187) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== ==18894== ==18894== 3 bytes in 1 blocks are still reachable in loss record 3 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x127BB923: PR_Malloc (prmem.c:448) ==18894== by 0x127BDCFD: PR_CreateStack (pratom.c:386) ==18894== by 0x127B2AAB: _PR_InitFdCache (prfdcach.c:285) ==18894== by 0x127C6520: _PR_InitIO (ptio.c:1109) ==18894== by 0x127BE40E: _PR_InitStuff (prinit.c:236) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== ==18894== ==18894== 4 bytes in 1 blocks are still reachable in loss record 4 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0xA48DEF: strdup (in /lib/libc-2.5.so) ==18894== by 0x127B7848: PR_NewLogModule (prlog.c:363) ==18894== by 0x127BE363: _PR_InitStuff (prinit.c:194) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== ==18894== ==18894== 4 bytes in 1 blocks are still reachable in loss record 5 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0xA48DEF: strdup (in /lib/libc-2.5.so) ==18894== by 0x127B7848: PR_NewLogModule (prlog.c:363) ==18894== by 0x127BE2D3: _PR_InitStuff (prinit.c:188) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== ==18894== ==18894== 4 bytes in 1 blocks are still reachable in loss record 6 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x22AC7D: (within /lib/libcrypto.so.0.9.8b) ==18894== by 0x22B2FE: CRYPTO_malloc (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x26E4D6: (within /lib/libcrypto.so.0.9.8b) ==18894== by 0x26E51D: engine_cleanup_add_last (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x26EFC6: ENGINE_add (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x272711: ENGINE_load_dynamic (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x270BE6: ENGINE_load_builtin_engines (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x127FE136: Curl_ossl_init (in /usr/lib/libcurl.so.3.0.0) ==18894== by 0x1280F3AB: Curl_ssl_init (in /usr/lib/libcurl.so.3.0.0) ==18894== by 0x12807162: curl_global_init (in /usr/lib/libcurl.so.3.0.0) ==18894== by 0x1279FAB3: mod_spidermonkey_load (mod_spidermonkey.c:3751) ==18894== ==18894== ==18894== 5 bytes in 1 blocks are still reachable in loss record 7 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0xA48DEF: strdup (in /lib/libc-2.5.so) ==18894== by 0x127B7848: PR_NewLogModule (prlog.c:363) ==18894== by 0x127BE37B: _PR_InitStuff (prinit.c:195) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== ==18894== ==18894== 5 bytes in 1 blocks are still reachable in loss record 8 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0xA48DEF: strdup (in /lib/libc-2.5.so) ==18894== by 0x127B7848: PR_NewLogModule (prlog.c:363) ==18894== by 0x127BE303: _PR_InitStuff (prinit.c:190) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== ==18894== ==18894== 5 bytes in 1 blocks are still reachable in loss record 9 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0xA48DEF: strdup (in /lib/libc-2.5.so) ==18894== by 0x127B7848: PR_NewLogModule (prlog.c:363) ==18894== by 0x127BE2A3: _PR_InitStuff (prinit.c:186) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== ==18894== ==18894== 6 bytes in 1 blocks are still reachable in loss record 10 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0xA48DEF: strdup (in /lib/libc-2.5.so) ==18894== by 0x127B7848: PR_NewLogModule (prlog.c:363) ==18894== by 0x127BE31B: _PR_InitStuff (prinit.c:191) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== ==18894== ==18894== 6 bytes in 1 blocks are still reachable in loss record 11 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0xA48DEF: strdup (in /lib/libc-2.5.so) ==18894== by 0x127B7848: PR_NewLogModule (prlog.c:363) ==18894== by 0x127BE28B: _PR_InitStuff (prinit.c:185) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== ==18894== ==18894== 6 bytes in 1 blocks are still reachable in loss record 12 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0xA48DEF: strdup (in /lib/libc-2.5.so) ==18894== by 0xC23E31: _nc_setupterm (in /usr/lib/libncurses.so.5.5) ==18894== by 0xC24747: tgetent (in /usr/lib/libncurses.so.5.5) ==18894== by 0x4141441: term_set (term.c:925) ==18894== by 0x4141843: term_init (term.c:350) ==18894== by 0x4139E84: el_init (el.c:84) ==18894== by 0x4049449: switch_console_loop (switch_console.c:747) ==18894== by 0x4061CE7: switch_core_runtime_loop (switch_core.c:687) ==18894== by 0x804A453: main (switch.c:715) ==18894== ==18894== ==18894== 7 bytes in 1 blocks are still reachable in loss record 13 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0xA48DEF: strdup (in /lib/libc-2.5.so) ==18894== by 0x127B7848: PR_NewLogModule (prlog.c:363) ==18894== by 0x127BE333: _PR_InitStuff (prinit.c:192) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== ==18894== ==18894== 7 bytes in 1 blocks are still reachable in loss record 14 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0xA48DEF: strdup (in /lib/libc-2.5.so) ==18894== by 0x127B7848: PR_NewLogModule (prlog.c:363) ==18894== by 0x127BE2EB: _PR_InitStuff (prinit.c:189) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== ==18894== ==18894== 8 bytes in 1 blocks are still reachable in loss record 15 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0xA48DEF: strdup (in /lib/libc-2.5.so) ==18894== by 0xC2A532: (within /usr/lib/libncurses.so.5.5) ==18894== by 0xC2A837: _nc_trim_sgr0 (in /usr/lib/libncurses.so.5.5) ==18894== by 0xC248BE: tgetent (in /usr/lib/libncurses.so.5.5) ==18894== by 0x4141441: term_set (term.c:925) ==18894== by 0x4141843: term_init (term.c:350) ==18894== by 0x4139E84: el_init (el.c:84) ==18894== by 0x4049449: switch_console_loop (switch_console.c:747) ==18894== by 0x4061CE7: switch_core_runtime_loop (switch_core.c:687) ==18894== by 0x804A453: main (switch.c:715) ==18894== ==18894== ==18894== 8 bytes in 1 blocks are still reachable in loss record 16 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40DED94: sqlite3MallocX (sqliteInt.h:278) ==18894== by 0x40D3D03: sqlite3HashInsert (hash.c:366) ==18894== by 0x40531EE: switch_core_hash_insert (switch_core_hash.c:67) ==18894== by 0x7AA4EF1: mod_cdr_csv_load (mod_cdr_csv.c:314) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 8 bytes in 1 blocks are still reachable in loss record 17 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40DED94: sqlite3MallocX (sqliteInt.h:278) ==18894== by 0x40D3D03: sqlite3HashInsert (hash.c:366) ==18894== by 0x40531EE: switch_core_hash_insert (switch_core_hash.c:67) ==18894== by 0x400EC26: mod_logfile_load (mod_logfile.c:290) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 8 bytes in 1 blocks are still reachable in loss record 18 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x407176E: my_dup (switch_event.c:95) ==18894== by 0x40737A4: switch_event_bind_removable (switch_event.c:1176) ==18894== by 0x4073960: switch_event_bind (switch_event.c:1205) ==18894== by 0x4053B7A: switch_core_sqldb_start (switch_core_sqldb.c:584) ==18894== by 0x4063A91: switch_core_init (switch_core.c:1248) ==18894== by 0x406585C: switch_core_init_and_modload (switch_core.c:1422) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 9 bytes in 1 blocks are still reachable in loss record 19 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40DED94: sqlite3MallocX (sqliteInt.h:278) ==18894== by 0x40D3D03: sqlite3HashInsert (hash.c:366) ==18894== by 0x40531EE: switch_core_hash_insert (switch_core_hash.c:67) ==18894== by 0x4064603: switch_load_network_lists (switch_core.c:909) ==18894== by 0x40659E4: switch_core_init_and_modload (switch_core.c:1457) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 9 bytes in 1 blocks are still reachable in loss record 20 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40DED94: sqlite3MallocX (sqliteInt.h:278) ==18894== by 0x40D3D03: sqlite3HashInsert (hash.c:366) ==18894== by 0x40531EE: switch_core_hash_insert (switch_core_hash.c:67) ==18894== by 0x4064724: switch_load_network_lists (switch_core.c:921) ==18894== by 0x40659E4: switch_core_init_and_modload (switch_core.c:1457) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 12 bytes in 1 blocks are still reachable in loss record 21 of 193 ==18894== at 0x40046FF: calloc (vg_replace_malloc.c:279) ==18894== by 0x127BB985: PR_Calloc (prmem.c:455) ==18894== by 0x127B7834: PR_NewLogModule (prlog.c:361) ==18894== by 0x127BE28B: _PR_InitStuff (prinit.c:185) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== ==18894== ==18894== 12 bytes in 1 blocks are still reachable in loss record 22 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x22AC7D: (within /lib/libcrypto.so.0.9.8b) ==18894== by 0x22B2FE: CRYPTO_malloc (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x27E53F: lh_insert (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x280EDF: (within /lib/libcrypto.so.0.9.8b) ==18894== by 0x28141B: ERR_get_state (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x281ADA: ERR_clear_error (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x27271E: ENGINE_load_dynamic (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x270BE6: ENGINE_load_builtin_engines (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x127FE136: Curl_ossl_init (in /usr/lib/libcurl.so.3.0.0) ==18894== by 0x1280F3AB: Curl_ssl_init (in /usr/lib/libcurl.so.3.0.0) ==18894== by 0x12807162: curl_global_init (in /usr/lib/libcurl.so.3.0.0) ==18894== ==18894== ==18894== 12 bytes in 1 blocks are still reachable in loss record 23 of 193 ==18894== at 0x40046FF: calloc (vg_replace_malloc.c:279) ==18894== by 0x127BB985: PR_Calloc (prmem.c:455) ==18894== by 0x127B7834: PR_NewLogModule (prlog.c:361) ==18894== by 0x127BE2BB: _PR_InitStuff (prinit.c:187) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== ==18894== ==18894== 12 bytes in 1 blocks are still reachable in loss record 24 of 193 ==18894== at 0x40046FF: calloc (vg_replace_malloc.c:279) ==18894== by 0x127BB985: PR_Calloc (prmem.c:455) ==18894== by 0x127B2C12: _PR_InitMW (prmwait.c:243) ==18894== by 0x127BE41D: _PR_InitStuff (prinit.c:239) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== ==18894== ==18894== 12 bytes in 1 blocks are still reachable in loss record 25 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x127BB923: PR_Malloc (prmem.c:448) ==18894== by 0x127BDCBA: PR_CreateStack (pratom.c:382) ==18894== by 0x127B2AAB: _PR_InitFdCache (prfdcach.c:285) ==18894== by 0x127C6520: _PR_InitIO (ptio.c:1109) ==18894== by 0x127BE40E: _PR_InitStuff (prinit.c:236) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== ==18894== ==18894== 12 bytes in 1 blocks are still reachable in loss record 26 of 193 ==18894== at 0x40046FF: calloc (vg_replace_malloc.c:279) ==18894== by 0x127BB985: PR_Calloc (prmem.c:455) ==18894== by 0x127B7834: PR_NewLogModule (prlog.c:361) ==18894== by 0x127BE37B: _PR_InitStuff (prinit.c:195) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== ==18894== ==18894== 12 bytes in 1 blocks are still reachable in loss record 27 of 193 ==18894== at 0x40046FF: calloc (vg_replace_malloc.c:279) ==18894== by 0x127BB985: PR_Calloc (prmem.c:455) ==18894== by 0x127B7834: PR_NewLogModule (prlog.c:361) ==18894== by 0x127BE363: _PR_InitStuff (prinit.c:194) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== ==18894== ==18894== 12 bytes in 1 blocks are still reachable in loss record 28 of 193 ==18894== at 0x40046FF: calloc (vg_replace_malloc.c:279) ==18894== by 0x127BB985: PR_Calloc (prmem.c:455) ==18894== by 0x127B7834: PR_NewLogModule (prlog.c:361) ==18894== by 0x127BE34B: _PR_InitStuff (prinit.c:193) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== ==18894== ==18894== 12 bytes in 1 blocks are still reachable in loss record 29 of 193 ==18894== at 0x40046FF: calloc (vg_replace_malloc.c:279) ==18894== by 0x127BB985: PR_Calloc (prmem.c:455) ==18894== by 0x127B7834: PR_NewLogModule (prlog.c:361) ==18894== by 0x127BE333: _PR_InitStuff (prinit.c:192) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== ==18894== ==18894== 12 bytes in 1 blocks are still reachable in loss record 30 of 193 ==18894== at 0x40046FF: calloc (vg_replace_malloc.c:279) ==18894== by 0x127BB985: PR_Calloc (prmem.c:455) ==18894== by 0x127B7834: PR_NewLogModule (prlog.c:361) ==18894== by 0x127BE31B: _PR_InitStuff (prinit.c:191) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== ==18894== ==18894== 12 bytes in 1 blocks are still reachable in loss record 31 of 193 ==18894== at 0x40046FF: calloc (vg_replace_malloc.c:279) ==18894== by 0x127BB985: PR_Calloc (prmem.c:455) ==18894== by 0x127B7834: PR_NewLogModule (prlog.c:361) ==18894== by 0x127BE303: _PR_InitStuff (prinit.c:190) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== ==18894== ==18894== 12 bytes in 1 blocks are still reachable in loss record 32 of 193 ==18894== at 0x40046FF: calloc (vg_replace_malloc.c:279) ==18894== by 0x127BB985: PR_Calloc (prmem.c:455) ==18894== by 0x127B7834: PR_NewLogModule (prlog.c:361) ==18894== by 0x127BE2EB: _PR_InitStuff (prinit.c:189) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== ==18894== ==18894== 12 bytes in 1 blocks are still reachable in loss record 33 of 193 ==18894== at 0x40046FF: calloc (vg_replace_malloc.c:279) ==18894== by 0x127BB985: PR_Calloc (prmem.c:455) ==18894== by 0x127B7834: PR_NewLogModule (prlog.c:361) ==18894== by 0x127BE2D3: _PR_InitStuff (prinit.c:188) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== ==18894== ==18894== 12 bytes in 1 blocks are still reachable in loss record 34 of 193 ==18894== at 0x40046FF: calloc (vg_replace_malloc.c:279) ==18894== by 0x127BB985: PR_Calloc (prmem.c:455) ==18894== by 0x127B7834: PR_NewLogModule (prlog.c:361) ==18894== by 0x127BE2A3: _PR_InitStuff (prinit.c:186) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== ==18894== ==18894== 12 bytes in 2 blocks are still reachable in loss record 35 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40DED94: sqlite3MallocX (sqliteInt.h:278) ==18894== by 0x40D3D03: sqlite3HashInsert (hash.c:366) ==18894== by 0x40531EE: switch_core_hash_insert (switch_core_hash.c:67) ==18894== by 0x4064E08: switch_load_network_lists (switch_core.c:1077) ==18894== by 0x40659E4: switch_core_init_and_modload (switch_core.c:1457) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 12 bytes in 1 blocks are still reachable in loss record 36 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x22AC7D: (within /lib/libcrypto.so.0.9.8b) ==18894== by 0x22B2FE: CRYPTO_malloc (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x4C048E: (within /lib/libssl.so.0.9.8b) ==18894== by 0x4C06B6: SSL_COMP_get_compression_methods (in /lib/libssl.so.0.9.8b) ==18894== by 0x4C61F4: SSL_library_init (in /lib/libssl.so.0.9.8b) ==18894== by 0x127FE140: Curl_ossl_init (in /usr/lib/libcurl.so.3.0.0) ==18894== by 0x1280F3AB: Curl_ssl_init (in /usr/lib/libcurl.so.3.0.0) ==18894== by 0x12807162: curl_global_init (in /usr/lib/libcurl.so.3.0.0) ==18894== by 0x1279FAB3: mod_spidermonkey_load (mod_spidermonkey.c:3751) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== ==18894== ==18894== 12 bytes in 1 blocks are still reachable in loss record 37 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x9BE859: decompose_rpath (in /lib/ld-2.5.so) ==18894== by 0x9BF831: _dl_map_object (in /lib/ld-2.5.so) ==18894== by 0x9C32D5: openaux (in /lib/ld-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0x9C3894: _dl_map_object_deps (in /lib/ld-2.5.so) ==18894== by 0x9C8A8C: dl_open_worker (in /lib/ld-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0x9C8411: _dl_open (in /lib/ld-2.5.so) ==18894== by 0xB1FC4C: dlopen_doit (in /lib/libdl-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0xB202CB: _dlerror_run (in /lib/libdl-2.5.so) ==18894== ==18894== ==18894== 13 bytes in 1 blocks are definitely lost in loss record 38 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x412F754: crypto_alloc (alloc.c:97) ==18894== by 0x412FF9D: null_cipher_alloc (null_cipher.c:68) ==18894== by 0x412B53C: cipher_type_self_test (cipher.c:264) ==18894== by 0x412F0B5: crypto_kernel_load_cipher_type (crypto_kernel.c:310) ==18894== by 0x412F62C: crypto_kernel_init (crypto_kernel.c:151) ==18894== by 0x4129026: srtp_init (srtp.c:1081) ==18894== by 0x407AF62: switch_rtp_init (switch_rtp.c:611) ==18894== by 0x40639D6: switch_core_init (switch_core.c:1252) ==18894== by 0x406585C: switch_core_init_and_modload (switch_core.c:1422) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 13 bytes in 1 blocks are still reachable in loss record 39 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40DED94: sqlite3MallocX (sqliteInt.h:278) ==18894== by 0x40D3D03: sqlite3HashInsert (hash.c:366) ==18894== by 0x40531EE: switch_core_hash_insert (switch_core_hash.c:67) ==18894== by 0x4064511: switch_load_network_lists (switch_core.c:901) ==18894== by 0x40659E4: switch_core_init_and_modload (switch_core.c:1457) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 14 bytes in 1 blocks are still reachable in loss record 40 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40DED94: sqlite3MallocX (sqliteInt.h:278) ==18894== by 0x40D3D03: sqlite3HashInsert (hash.c:366) ==18894== by 0x40531EE: switch_core_hash_insert (switch_core_hash.c:67) ==18894== by 0x4064881: switch_load_network_lists (switch_core.c:937) ==18894== by 0x40659E4: switch_core_init_and_modload (switch_core.c:1457) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 14 bytes in 1 blocks are still reachable in loss record 41 of 193 ==18894== at 0x40054BB: realloc (vg_replace_malloc.c:306) ==18894== by 0xC21BDF: _nc_doalloc (in /usr/lib/libncurses.so.5.5) ==18894== by 0xC25667: tparm (in /usr/lib/libncurses.so.5.5) ==18894== by 0xC2A526: (within /usr/lib/libncurses.so.5.5) ==18894== by 0xC2A82B: _nc_trim_sgr0 (in /usr/lib/libncurses.so.5.5) ==18894== by 0xC248BE: tgetent (in /usr/lib/libncurses.so.5.5) ==18894== by 0x4141441: term_set (term.c:925) ==18894== by 0x4141843: term_init (term.c:350) ==18894== by 0x4139E84: el_init (el.c:84) ==18894== by 0x4049449: switch_console_loop (switch_console.c:747) ==18894== by 0x4061CE7: switch_core_runtime_loop (switch_core.c:687) ==18894== by 0x804A453: main (switch.c:715) ==18894== ==18894== ==18894== 14 bytes in 1 blocks are still reachable in loss record 42 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40DED94: sqlite3MallocX (sqliteInt.h:278) ==18894== by 0x40D3D03: sqlite3HashInsert (hash.c:366) ==18894== by 0x40531EE: switch_core_hash_insert (switch_core_hash.c:67) ==18894== by 0x40647CC: switch_load_network_lists (switch_core.c:927) ==18894== by 0x40659E4: switch_core_init_and_modload (switch_core.c:1457) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 16 bytes in 1 blocks are still reachable in loss record 43 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x22AC7D: (within /lib/libcrypto.so.0.9.8b) ==18894== by 0x22B2FE: CRYPTO_malloc (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x27DED9: sk_new (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x27DF4D: sk_new_null (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x26E3FA: (within /lib/libcrypto.so.0.9.8b) ==18894== by 0x26E50B: engine_cleanup_add_last (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x26EFC6: ENGINE_add (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x272711: ENGINE_load_dynamic (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x270BE6: ENGINE_load_builtin_engines (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x127FE136: Curl_ossl_init (in /usr/lib/libcurl.so.3.0.0) ==18894== by 0x1280F3AB: Curl_ssl_init (in /usr/lib/libcurl.so.3.0.0) ==18894== ==18894== ==18894== 16 bytes in 1 blocks are still reachable in loss record 44 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x22AC7D: (within /lib/libcrypto.so.0.9.8b) ==18894== by 0x22B2FE: CRYPTO_malloc (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x27DED9: sk_new (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x4C046C: (within /lib/libssl.so.0.9.8b) ==18894== by 0x4C06B6: SSL_COMP_get_compression_methods (in /lib/libssl.so.0.9.8b) ==18894== by 0x4C61F4: SSL_library_init (in /lib/libssl.so.0.9.8b) ==18894== by 0x127FE140: Curl_ossl_init (in /usr/lib/libcurl.so.3.0.0) ==18894== by 0x1280F3AB: Curl_ssl_init (in /usr/lib/libcurl.so.3.0.0) ==18894== by 0x12807162: curl_global_init (in /usr/lib/libcurl.so.3.0.0) ==18894== by 0x1279FAB3: mod_spidermonkey_load (mod_spidermonkey.c:3751) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== ==18894== ==18894== 18 bytes in 1 blocks are still reachable in loss record 45 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0xC21DCE: _nc_home_terminfo (in /usr/lib/libncurses.so.5.5) ==18894== by 0xC29449: _nc_next_db (in /usr/lib/libncurses.so.5.5) ==18894== by 0xC29F31: _nc_read_entry (in /usr/lib/libncurses.so.5.5) ==18894== by 0xC23FDC: _nc_setupterm (in /usr/lib/libncurses.so.5.5) ==18894== by 0xC24747: tgetent (in /usr/lib/libncurses.so.5.5) ==18894== by 0x4141441: term_set (term.c:925) ==18894== by 0x4141843: term_init (term.c:350) ==18894== by 0x4139E84: el_init (el.c:84) ==18894== by 0x4049449: switch_console_loop (switch_console.c:747) ==18894== by 0x4061CE7: switch_core_runtime_loop (switch_core.c:687) ==18894== by 0x804A453: main (switch.c:715) ==18894== ==18894== ==18894== 20 bytes in 1 blocks are still reachable in loss record 46 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x22AC7D: (within /lib/libcrypto.so.0.9.8b) ==18894== by 0x22B2FE: CRYPTO_malloc (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x27DEB6: sk_new (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x27DF4D: sk_new_null (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x26E3FA: (within /lib/libcrypto.so.0.9.8b) ==18894== by 0x26E50B: engine_cleanup_add_last (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x26EFC6: ENGINE_add (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x272711: ENGINE_load_dynamic (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x270BE6: ENGINE_load_builtin_engines (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x127FE136: Curl_ossl_init (in /usr/lib/libcurl.so.3.0.0) ==18894== by 0x1280F3AB: Curl_ssl_init (in /usr/lib/libcurl.so.3.0.0) ==18894== ==18894== ==18894== 20 bytes in 1 blocks are still reachable in loss record 47 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40DED94: sqlite3MallocX (sqliteInt.h:278) ==18894== by 0x40D3CD2: sqlite3HashInsert (hash.c:363) ==18894== by 0x40531EE: switch_core_hash_insert (switch_core_hash.c:67) ==18894== by 0x4064881: switch_load_network_lists (switch_core.c:937) ==18894== by 0x40659E4: switch_core_init_and_modload (switch_core.c:1457) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 20 bytes in 1 blocks are still reachable in loss record 48 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40DED94: sqlite3MallocX (sqliteInt.h:278) ==18894== by 0x40D3CD2: sqlite3HashInsert (hash.c:363) ==18894== by 0x40531EE: switch_core_hash_insert (switch_core_hash.c:67) ==18894== by 0x40647CC: switch_load_network_lists (switch_core.c:927) ==18894== by 0x40659E4: switch_core_init_and_modload (switch_core.c:1457) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 20 bytes in 1 blocks are still reachable in loss record 49 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40DED94: sqlite3MallocX (sqliteInt.h:278) ==18894== by 0x40D3CD2: sqlite3HashInsert (hash.c:363) ==18894== by 0x40531EE: switch_core_hash_insert (switch_core_hash.c:67) ==18894== by 0x4064724: switch_load_network_lists (switch_core.c:921) ==18894== by 0x40659E4: switch_core_init_and_modload (switch_core.c:1457) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 20 bytes in 1 blocks are still reachable in loss record 50 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40DED94: sqlite3MallocX (sqliteInt.h:278) ==18894== by 0x40D3CD2: sqlite3HashInsert (hash.c:363) ==18894== by 0x40531EE: switch_core_hash_insert (switch_core_hash.c:67) ==18894== by 0x7AA4EF1: mod_cdr_csv_load (mod_cdr_csv.c:314) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 20 bytes in 1 blocks are still reachable in loss record 51 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40DED94: sqlite3MallocX (sqliteInt.h:278) ==18894== by 0x40D3CD2: sqlite3HashInsert (hash.c:363) ==18894== by 0x40531EE: switch_core_hash_insert (switch_core_hash.c:67) ==18894== by 0x400EC26: mod_logfile_load (mod_logfile.c:290) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 20 bytes in 1 blocks are still reachable in loss record 52 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40DED94: sqlite3MallocX (sqliteInt.h:278) ==18894== by 0x40D3CD2: sqlite3HashInsert (hash.c:363) ==18894== by 0x40531EE: switch_core_hash_insert (switch_core_hash.c:67) ==18894== by 0x4064603: switch_load_network_lists (switch_core.c:909) ==18894== by 0x40659E4: switch_core_init_and_modload (switch_core.c:1457) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 20 bytes in 1 blocks are still reachable in loss record 53 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x400544A: realloc (vg_replace_malloc.c:306) ==18894== by 0x9C163F: _dl_lookup_symbol_x (in /lib/ld-2.5.so) ==18894== by 0x9C2334: _dl_relocate_object (in /lib/ld-2.5.so) ==18894== by 0x9C8B67: dl_open_worker (in /lib/ld-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0x9C8411: _dl_open (in /lib/ld-2.5.so) ==18894== by 0xB1FC4C: dlopen_doit (in /lib/libdl-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0xB202CB: _dlerror_run (in /lib/libdl-2.5.so) ==18894== by 0xB1FB83: dlopen@@GLIBC_2.1 (in /lib/libdl-2.5.so) ==18894== by 0x4067486: switch_dso_open (switch_dso.c:94) ==18894== ==18894== ==18894== 20 bytes in 1 blocks are still reachable in loss record 54 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x22AC7D: (within /lib/libcrypto.so.0.9.8b) ==18894== by 0x22B2FE: CRYPTO_malloc (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x27DEB6: sk_new (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x4C046C: (within /lib/libssl.so.0.9.8b) ==18894== by 0x4C06B6: SSL_COMP_get_compression_methods (in /lib/libssl.so.0.9.8b) ==18894== by 0x4C61F4: SSL_library_init (in /lib/libssl.so.0.9.8b) ==18894== by 0x127FE140: Curl_ossl_init (in /usr/lib/libcurl.so.3.0.0) ==18894== by 0x1280F3AB: Curl_ssl_init (in /usr/lib/libcurl.so.3.0.0) ==18894== by 0x12807162: curl_global_init (in /usr/lib/libcurl.so.3.0.0) ==18894== by 0x1279FAB3: mod_spidermonkey_load (mod_spidermonkey.c:3751) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== ==18894== ==18894== 20 bytes in 1 blocks are still reachable in loss record 55 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40DED94: sqlite3MallocX (sqliteInt.h:278) ==18894== by 0x40D3CD2: sqlite3HashInsert (hash.c:363) ==18894== by 0x40531EE: switch_core_hash_insert (switch_core_hash.c:67) ==18894== by 0x4064511: switch_load_network_lists (switch_core.c:901) ==18894== by 0x40659E4: switch_core_init_and_modload (switch_core.c:1457) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 20 bytes in 1 blocks are still reachable in loss record 56 of 193 ==18894== at 0x40046FF: calloc (vg_replace_malloc.c:279) ==18894== by 0xB2033B: _dlerror_run (in /lib/libdl-2.5.so) ==18894== by 0xB1FB83: dlopen@@GLIBC_2.1 (in /lib/libdl-2.5.so) ==18894== by 0x40674AF: switch_dso_open (switch_dso.c:96) ==18894== by 0x406C800: switch_loadable_module_load_module_ex (switch_loadable_module.c:795) ==18894== by 0x406D073: switch_loadable_module_init (switch_loadable_module.c:1159) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 20 bytes in 1 blocks are still reachable in loss record 57 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40DED94: sqlite3MallocX (sqliteInt.h:278) ==18894== by 0x40D3CD2: sqlite3HashInsert (hash.c:363) ==18894== by 0x40531EE: switch_core_hash_insert (switch_core_hash.c:67) ==18894== by 0x7AA4812: ??? (mod_cdr_csv.c:140) ==18894== by 0x7AA4B55: ??? (mod_cdr_csv.c:233) ==18894== by 0x405A631: switch_core_session_reporting_state (switch_core_state_machine.c:607) ==18894== by 0x405AC1A: switch_core_session_run (switch_core_state_machine.c:409) ==18894== by 0x4058204: switch_core_session_thread (switch_core_session.c:1064) ==18894== by 0x40CD6C5: dummy_worker (thread.c:138) ==18894== by 0xB5346A: start_thread (in /lib/libpthread-2.5.so) ==18894== by 0xAAADBD: clone (in /lib/libc-2.5.so) ==18894== ==18894== ==18894== 20 bytes in 1 blocks are still reachable in loss record 58 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40DED94: sqlite3MallocX (sqliteInt.h:278) ==18894== by 0x40D3CD2: sqlite3HashInsert (hash.c:363) ==18894== by 0x40531EE: switch_core_hash_insert (switch_core_hash.c:67) ==18894== by 0x7AA4812: ??? (mod_cdr_csv.c:140) ==18894== by 0x7AA4AA0: ??? (mod_cdr_csv.c:251) ==18894== by 0x405A631: switch_core_session_reporting_state (switch_core_state_machine.c:607) ==18894== by 0x405AC1A: switch_core_session_run (switch_core_state_machine.c:409) ==18894== by 0x4058204: switch_core_session_thread (switch_core_session.c:1064) ==18894== by 0x40CD6C5: dummy_worker (thread.c:138) ==18894== by 0xB5346A: start_thread (in /lib/libpthread-2.5.so) ==18894== by 0xAAADBD: clone (in /lib/libc-2.5.so) ==18894== ==18894== ==18894== 21 bytes in 1 blocks are definitely lost in loss record 59 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0xA48DEF: strdup (in /lib/libc-2.5.so) ==18894== by 0x40A9A52: switch_xml_config_parse_event (switch_xml_config.c:267) ==18894== by 0x11CAC65C: ??? (mod_voicemail.c:643) ==18894== by 0x11CAD21F: mod_voicemail_load (mod_voicemail.c:821) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 22 bytes in 1 blocks are still reachable in loss record 60 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x407176E: my_dup (switch_event.c:95) ==18894== by 0x40737A4: switch_event_bind_removable (switch_event.c:1176) ==18894== by 0x40ABD58: softtimer_load (switch_time.c:753) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D073: switch_loadable_module_init (switch_loadable_module.c:1159) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 24 bytes in 1 blocks are still reachable in loss record 61 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x127BB923: PR_Malloc (prmem.c:448) ==18894== by 0x127B26D2: _PR_Getfd (prfdcach.c:141) ==18894== by 0x127C8EBA: pt_SetMethods (ptio.c:3248) ==18894== by 0x127C65A0: _PR_InitIO (ptio.c:1113) ==18894== by 0x127BE40E: _PR_InitStuff (prinit.c:236) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== ==18894== ==18894== 24 bytes in 1 blocks are still reachable in loss record 62 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x127BB923: PR_Malloc (prmem.c:448) ==18894== by 0x127B26D2: _PR_Getfd (prfdcach.c:141) ==18894== by 0x127C8EBA: pt_SetMethods (ptio.c:3248) ==18894== by 0x127C6572: _PR_InitIO (ptio.c:1112) ==18894== by 0x127BE40E: _PR_InitStuff (prinit.c:236) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== ==18894== ==18894== 24 bytes in 1 blocks are still reachable in loss record 63 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x127BB923: PR_Malloc (prmem.c:448) ==18894== by 0x127B26D2: _PR_Getfd (prfdcach.c:141) ==18894== by 0x127C8EBA: pt_SetMethods (ptio.c:3248) ==18894== by 0x127C6544: _PR_InitIO (ptio.c:1111) ==18894== by 0x127BE40E: _PR_InitStuff (prinit.c:236) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== ==18894== ==18894== 24 bytes in 1 blocks are still reachable in loss record 64 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x407374D: switch_event_bind_removable (switch_event.c:1172) ==18894== by 0x40ABD58: softtimer_load (switch_time.c:753) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D073: switch_loadable_module_init (switch_loadable_module.c:1159) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 24 bytes in 1 blocks are still reachable in loss record 65 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x407374D: switch_event_bind_removable (switch_event.c:1172) ==18894== by 0x4073960: switch_event_bind (switch_event.c:1205) ==18894== by 0x4053B7A: switch_core_sqldb_start (switch_core_sqldb.c:584) ==18894== by 0x4063A91: switch_core_init (switch_core.c:1248) ==18894== by 0x406585C: switch_core_init_and_modload (switch_core.c:1422) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 28 bytes in 1 blocks are still reachable in loss record 66 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x9C3B6B: _dl_map_object_deps (in /lib/ld-2.5.so) ==18894== by 0x9C8A8C: dl_open_worker (in /lib/ld-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0x9C8411: _dl_open (in /lib/ld-2.5.so) ==18894== by 0xAE07F1: do_dlopen (in /lib/libc-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0xAE09A4: __libc_dlopen_mode (in /lib/libc-2.5.so) ==18894== by 0xB5BB06: pthread_cancel_init (in /lib/libpthread-2.5.so) ==18894== by 0xB5BC30: _Unwind_ForcedUnwind (in /lib/libpthread-2.5.so) ==18894== by 0xB59700: __pthread_unwind (in /lib/libpthread-2.5.so) ==18894== by 0xB543CF: pthread_exit (in /lib/libpthread-2.5.so) ==18894== ==18894== ==18894== 28 bytes in 1 blocks are still reachable in loss record 67 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x127BB923: PR_Malloc (prmem.c:448) ==18894== by 0x127B26E7: _PR_Getfd (prfdcach.c:144) ==18894== by 0x127C8EBA: pt_SetMethods (ptio.c:3248) ==18894== by 0x127C65A0: _PR_InitIO (ptio.c:1113) ==18894== by 0x127BE40E: _PR_InitStuff (prinit.c:236) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== ==18894== ==18894== 28 bytes in 1 blocks are still reachable in loss record 68 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x127BB923: PR_Malloc (prmem.c:448) ==18894== by 0x127B26E7: _PR_Getfd (prfdcach.c:144) ==18894== by 0x127C8EBA: pt_SetMethods (ptio.c:3248) ==18894== by 0x127C6572: _PR_InitIO (ptio.c:1112) ==18894== by 0x127BE40E: _PR_InitStuff (prinit.c:236) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== ==18894== ==18894== 28 bytes in 1 blocks are still reachable in loss record 69 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x127BB923: PR_Malloc (prmem.c:448) ==18894== by 0x127B26E7: _PR_Getfd (prfdcach.c:144) ==18894== by 0x127C8EBA: pt_SetMethods (ptio.c:3248) ==18894== by 0x127C6544: _PR_InitIO (ptio.c:1111) ==18894== by 0x127BE40E: _PR_InitStuff (prinit.c:236) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== ==18894== ==18894== 31 bytes in 1 blocks are still reachable in loss record 70 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0xA48DEF: strdup (in /lib/libc-2.5.so) ==18894== by 0x400EBEB: mod_logfile_load (mod_logfile.c:283) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 33 bytes in 1 blocks are still reachable in loss record 71 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40DED94: sqlite3MallocX (sqliteInt.h:278) ==18894== by 0x40D3D03: sqlite3HashInsert (hash.c:366) ==18894== by 0x40531EE: switch_core_hash_insert (switch_core_hash.c:67) ==18894== by 0x7AA4812: ??? (mod_cdr_csv.c:140) ==18894== by 0x7AA4B55: ??? (mod_cdr_csv.c:233) ==18894== by 0x405A631: switch_core_session_reporting_state (switch_core_state_machine.c:607) ==18894== by 0x405AC1A: switch_core_session_run (switch_core_state_machine.c:409) ==18894== by 0x4058204: switch_core_session_thread (switch_core_session.c:1064) ==18894== by 0x40CD6C5: dummy_worker (thread.c:138) ==18894== by 0xB5346A: start_thread (in /lib/libpthread-2.5.so) ==18894== by 0xAAADBD: clone (in /lib/libc-2.5.so) ==18894== ==18894== ==18894== 34 bytes in 5 blocks are still reachable in loss record 72 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40DED94: sqlite3MallocX (sqliteInt.h:278) ==18894== by 0x40D3D03: sqlite3HashInsert (hash.c:366) ==18894== by 0x40531EE: switch_core_hash_insert (switch_core_hash.c:67) ==18894== by 0x7AA518D: mod_cdr_csv_load (mod_cdr_csv.c:361) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 35 bytes in 1 blocks are still reachable in loss record 73 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40DED94: sqlite3MallocX (sqliteInt.h:278) ==18894== by 0x40D3D03: sqlite3HashInsert (hash.c:366) ==18894== by 0x40531EE: switch_core_hash_insert (switch_core_hash.c:67) ==18894== by 0x7AA4812: ??? (mod_cdr_csv.c:140) ==18894== by 0x7AA4AA0: ??? (mod_cdr_csv.c:251) ==18894== by 0x405A631: switch_core_session_reporting_state (switch_core_state_machine.c:607) ==18894== by 0x405AC1A: switch_core_session_run (switch_core_state_machine.c:409) ==18894== by 0x4058204: switch_core_session_thread (switch_core_session.c:1064) ==18894== by 0x40CD6C5: dummy_worker (thread.c:138) ==18894== by 0xB5346A: start_thread (in /lib/libpthread-2.5.so) ==18894== by 0xAAADBD: clone (in /lib/libc-2.5.so) ==18894== ==18894== ==18894== 40 bytes in 1 blocks are still reachable in loss record 74 of 193 ==18894== at 0x40046FF: calloc (vg_replace_malloc.c:279) ==18894== by 0x127BB985: PR_Calloc (prmem.c:455) ==18894== by 0x127CC5A2: _PR_InitThreads (ptthread.c:900) ==18894== by 0x127BE404: _PR_InitStuff (prinit.c:215) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== ==18894== ==18894== 40 bytes in 2 blocks are still reachable in loss record 75 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40DED94: sqlite3MallocX (sqliteInt.h:278) ==18894== by 0x40D3CD2: sqlite3HashInsert (hash.c:363) ==18894== by 0x40531EE: switch_core_hash_insert (switch_core_hash.c:67) ==18894== by 0x4064E08: switch_load_network_lists (switch_core.c:1077) ==18894== by 0x40659E4: switch_core_init_and_modload (switch_core.c:1457) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 44 bytes in 2 blocks are still reachable in loss record 76 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x9BF9B7: _dl_map_object (in /lib/ld-2.5.so) ==18894== by 0x9C32D5: openaux (in /lib/ld-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0x9C3894: _dl_map_object_deps (in /lib/ld-2.5.so) ==18894== by 0x9C8A8C: dl_open_worker (in /lib/ld-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0x9C8411: _dl_open (in /lib/ld-2.5.so) ==18894== by 0xB1FC4C: dlopen_doit (in /lib/libdl-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0xB202CB: _dlerror_run (in /lib/libdl-2.5.so) ==18894== by 0xB1FB83: dlopen@@GLIBC_2.1 (in /lib/libdl-2.5.so) ==18894== ==18894== ==18894== 45 bytes in 1 blocks are still reachable in loss record 77 of 193 ==18894== at 0x40054BB: realloc (vg_replace_malloc.c:306) ==18894== by 0xC21BDF: _nc_doalloc (in /usr/lib/libncurses.so.5.5) ==18894== by 0xC29ADC: _nc_read_file_entry (in /usr/lib/libncurses.so.5.5) ==18894== by 0xC29FA4: _nc_read_entry (in /usr/lib/libncurses.so.5.5) ==18894== by 0xC23FDC: _nc_setupterm (in /usr/lib/libncurses.so.5.5) ==18894== by 0xC24747: tgetent (in /usr/lib/libncurses.so.5.5) ==18894== by 0x4141441: term_set (term.c:925) ==18894== by 0x4141843: term_init (term.c:350) ==18894== by 0x4139E84: el_init (el.c:84) ==18894== by 0x4049449: switch_console_loop (switch_console.c:747) ==18894== by 0x4061CE7: switch_core_runtime_loop (switch_core.c:687) ==18894== by 0x804A453: main (switch.c:715) ==18894== ==18894== ==18894== 48 bytes in 1 blocks are still reachable in loss record 78 of 193 ==18894== at 0x40046FF: calloc (vg_replace_malloc.c:279) ==18894== by 0xC29C23: _nc_read_file_entry (in /usr/lib/libncurses.so.5.5) ==18894== by 0xC29FA4: _nc_read_entry (in /usr/lib/libncurses.so.5.5) ==18894== by 0xC23FDC: _nc_setupterm (in /usr/lib/libncurses.so.5.5) ==18894== by 0xC24747: tgetent (in /usr/lib/libncurses.so.5.5) ==18894== by 0x4141441: term_set (term.c:925) ==18894== by 0x4141843: term_init (term.c:350) ==18894== by 0x4139E84: el_init (el.c:84) ==18894== by 0x4049449: switch_console_loop (switch_console.c:747) ==18894== by 0x4061CE7: switch_core_runtime_loop (switch_core.c:687) ==18894== by 0x804A453: main (switch.c:715) ==18894== ==18894== ==18894== 55 bytes in 2 blocks are still reachable in loss record 79 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x9BCC9F: open_path (in /lib/ld-2.5.so) ==18894== by 0x9BF7C4: _dl_map_object (in /lib/ld-2.5.so) ==18894== by 0x9C32D5: openaux (in /lib/ld-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0x9C3894: _dl_map_object_deps (in /lib/ld-2.5.so) ==18894== by 0x9C8A8C: dl_open_worker (in /lib/ld-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0x9C8411: _dl_open (in /lib/ld-2.5.so) ==18894== by 0xB1FC4C: dlopen_doit (in /lib/libdl-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0xB202CB: _dlerror_run (in /lib/libdl-2.5.so) ==18894== ==18894== ==18894== 56 bytes in 1 blocks are still reachable in loss record 80 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x127BB923: PR_Malloc (prmem.c:448) ==18894== by 0x127C3F77: PR_NewCondVar (ptsynch.c:341) ==18894== by 0x12867B4E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 56 bytes in 1 blocks are still reachable in loss record 81 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x127BB923: PR_Malloc (prmem.c:448) ==18894== by 0x127C3F77: PR_NewCondVar (ptsynch.c:341) ==18894== by 0x12867B36: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 56 bytes in 1 blocks are still reachable in loss record 82 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x127BB923: PR_Malloc (prmem.c:448) ==18894== by 0x127C3F77: PR_NewCondVar (ptsynch.c:341) ==18894== by 0x12867B15: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 56 bytes in 1 blocks are still reachable in loss record 83 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x127BB923: PR_Malloc (prmem.c:448) ==18894== by 0x127C3F77: PR_NewCondVar (ptsynch.c:341) ==18894== by 0x12867AC7: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 56 bytes in 1 blocks are still reachable in loss record 84 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x127BB923: PR_Malloc (prmem.c:448) ==18894== by 0x127C3F77: PR_NewCondVar (ptsynch.c:341) ==18894== by 0x12867AAF: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 56 bytes in 1 blocks are still reachable in loss record 85 of 193 ==18894== at 0x40046FF: calloc (vg_replace_malloc.c:279) ==18894== by 0x127BB985: PR_Calloc (prmem.c:455) ==18894== by 0x127C43FF: PR_NewMonitor (ptsynch.c:457) ==18894== by 0x127CDA2D: _PR_UnixInit (unix.c:2877) ==18894== by 0x127BE422: _PR_InitStuff (prinit.c:240) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== ==18894== ==18894== 56 bytes in 1 blocks are still reachable in loss record 86 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x127BB923: PR_Malloc (prmem.c:448) ==18894== by 0x127C3F77: PR_NewCondVar (ptsynch.c:341) ==18894== by 0x127CC47A: _PR_InitThreads (ptthread.c:876) ==18894== by 0x127BE404: _PR_InitStuff (prinit.c:215) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== ==18894== ==18894== 56 bytes in 1 blocks are still reachable in loss record 87 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x127BB923: PR_Malloc (prmem.c:448) ==18894== by 0x127C3F77: PR_NewCondVar (ptsynch.c:341) ==18894== by 0x127C64B6: _PR_InitIO (ptio.c:1104) ==18894== by 0x127BE40E: _PR_InitStuff (prinit.c:236) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== ==18894== ==18894== 56 bytes in 1 blocks are still reachable in loss record 88 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x127BB923: PR_Malloc (prmem.c:448) ==18894== by 0x127C3F77: PR_NewCondVar (ptsynch.c:341) ==18894== by 0x127BEAEE: _PR_InitCallOnce (prinit.c:688) ==18894== by 0x127BE418: _PR_InitStuff (prinit.c:238) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== ==18894== ==18894== 63 bytes in 2 blocks are still reachable in loss record 89 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x9BE714: expand_dynamic_string_token (in /lib/ld-2.5.so) ==18894== by 0x9BF3A9: _dl_map_object (in /lib/ld-2.5.so) ==18894== by 0x9C8A30: dl_open_worker (in /lib/ld-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0x9C8411: _dl_open (in /lib/ld-2.5.so) ==18894== by 0xB1FC4C: dlopen_doit (in /lib/libdl-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0xB202CB: _dlerror_run (in /lib/libdl-2.5.so) ==18894== by 0xB1FB83: dlopen@@GLIBC_2.1 (in /lib/libdl-2.5.so) ==18894== by 0x4067486: switch_dso_open (switch_dso.c:94) ==18894== by 0x406C96B: switch_loadable_module_load_module_ex (switch_loadable_module.c:831) ==18894== ==18894== ==18894== 64 bytes in 1 blocks are still reachable in loss record 90 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x22AC7D: (within /lib/libcrypto.so.0.9.8b) ==18894== by 0x22B2FE: CRYPTO_malloc (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x27E5F6: lh_new (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x281272: (within /lib/libcrypto.so.0.9.8b) ==18894== by 0x280EA0: (within /lib/libcrypto.so.0.9.8b) ==18894== by 0x28141B: ERR_get_state (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x281ADA: ERR_clear_error (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x27271E: ENGINE_load_dynamic (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x270BE6: ENGINE_load_builtin_engines (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x127FE136: Curl_ossl_init (in /usr/lib/libcurl.so.3.0.0) ==18894== by 0x1280F3AB: Curl_ssl_init (in /usr/lib/libcurl.so.3.0.0) ==18894== ==18894== ==18894== 64 bytes in 1 blocks are still reachable in loss record 91 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40DED94: sqlite3MallocX (sqliteInt.h:278) ==18894== by 0x40D3A1C: rehash (hash.c:227) ==18894== by 0x40D3D42: sqlite3HashInsert (hash.c:378) ==18894== by 0x40531EE: switch_core_hash_insert (switch_core_hash.c:67) ==18894== by 0x7AA4EF1: mod_cdr_csv_load (mod_cdr_csv.c:314) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 64 bytes in 2 blocks are still reachable in loss record 92 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x9C8E78: dl_open_worker (in /lib/ld-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0x9C8411: _dl_open (in /lib/ld-2.5.so) ==18894== by 0xB1FC4C: dlopen_doit (in /lib/libdl-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0xB202CB: _dlerror_run (in /lib/libdl-2.5.so) ==18894== by 0xB1FB83: dlopen@@GLIBC_2.1 (in /lib/libdl-2.5.so) ==18894== by 0x4067486: switch_dso_open (switch_dso.c:94) ==18894== by 0x1279F92E: mod_spidermonkey_load (mod_spidermonkey.c:894) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== ==18894== ==18894== 64 bytes in 1 blocks are still reachable in loss record 93 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40DED94: sqlite3MallocX (sqliteInt.h:278) ==18894== by 0x40D3A1C: rehash (hash.c:227) ==18894== by 0x40D3D42: sqlite3HashInsert (hash.c:378) ==18894== by 0x40531EE: switch_core_hash_insert (switch_core_hash.c:67) ==18894== by 0x400EC26: mod_logfile_load (mod_logfile.c:290) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 64 bytes in 1 blocks are still reachable in loss record 94 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40DED94: sqlite3MallocX (sqliteInt.h:278) ==18894== by 0x40D3A1C: rehash (hash.c:227) ==18894== by 0x40D3D42: sqlite3HashInsert (hash.c:378) ==18894== by 0x40531EE: switch_core_hash_insert (switch_core_hash.c:67) ==18894== by 0x4064511: switch_load_network_lists (switch_core.c:901) ==18894== by 0x40659E4: switch_core_init_and_modload (switch_core.c:1457) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 64 bytes in 1 blocks are still reachable in loss record 95 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x128A3CBB: js_SetupLocks (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867AE5: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 64 bytes in 1 blocks are still reachable in loss record 96 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40DED94: sqlite3MallocX (sqliteInt.h:278) ==18894== by 0x40D3A1C: rehash (hash.c:227) ==18894== by 0x40D3D42: sqlite3HashInsert (hash.c:378) ==18894== by 0x40531EE: switch_core_hash_insert (switch_core_hash.c:67) ==18894== by 0x7AA4812: ??? (mod_cdr_csv.c:140) ==18894== by 0x7AA4B55: ??? (mod_cdr_csv.c:233) ==18894== by 0x405A631: switch_core_session_reporting_state (switch_core_state_machine.c:607) ==18894== by 0x405AC1A: switch_core_session_run (switch_core_state_machine.c:409) ==18894== by 0x4058204: switch_core_session_thread (switch_core_session.c:1064) ==18894== by 0x40CD6C5: dummy_worker (thread.c:138) ==18894== by 0xB5346A: start_thread (in /lib/libpthread-2.5.so) ==18894== ==18894== ==18894== 78 bytes in 1 blocks are still reachable in loss record 97 of 193 ==18894== at 0x40054BB: realloc (vg_replace_malloc.c:306) ==18894== by 0xC21BDF: _nc_doalloc (in /usr/lib/libncurses.so.5.5) ==18894== by 0xC29AFA: _nc_read_file_entry (in /usr/lib/libncurses.so.5.5) ==18894== by 0xC29FA4: _nc_read_entry (in /usr/lib/libncurses.so.5.5) ==18894== by 0xC23FDC: _nc_setupterm (in /usr/lib/libncurses.so.5.5) ==18894== by 0xC24747: tgetent (in /usr/lib/libncurses.so.5.5) ==18894== by 0x4141441: term_set (term.c:925) ==18894== by 0x4141843: term_init (term.c:350) ==18894== by 0x4139E84: el_init (el.c:84) ==18894== by 0x4049449: switch_console_loop (switch_console.c:747) ==18894== by 0x4061CE7: switch_core_runtime_loop (switch_core.c:687) ==18894== by 0x804A453: main (switch.c:715) ==18894== ==18894== ==18894== 81 bytes in 3 blocks are definitely lost in loss record 98 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0xA48DEF: strdup (in /lib/libc-2.5.so) ==18894== by 0x11CBE451: ??? (mod_limit.c:538) ==18894== by 0x4058785: switch_core_session_exec (switch_core_session.c:1474) ==18894== by 0x4058CD8: switch_core_session_execute_application (switch_core_session.c:1396) ==18894== by 0x405C794: switch_core_session_run (switch_core_state_machine.c:166) ==18894== by 0x4058204: switch_core_session_thread (switch_core_session.c:1064) ==18894== by 0x40CD6C5: dummy_worker (thread.c:138) ==18894== by 0xB5346A: start_thread (in /lib/libpthread-2.5.so) ==18894== by 0xAAADBD: clone (in /lib/libc-2.5.so) ==18894== ==18894== ==18894== 83 bytes in 1 blocks are still reachable in loss record 99 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0xC21BFC: _nc_doalloc (in /usr/lib/libncurses.so.5.5) ==18894== by 0xC2522E: _nc_tparm_analyze (in /usr/lib/libncurses.so.5.5) ==18894== by 0xC25391: tparm (in /usr/lib/libncurses.so.5.5) ==18894== by 0xC2A526: (within /usr/lib/libncurses.so.5.5) ==18894== by 0xC2A82B: _nc_trim_sgr0 (in /usr/lib/libncurses.so.5.5) ==18894== by 0xC248BE: tgetent (in /usr/lib/libncurses.so.5.5) ==18894== by 0x4141441: term_set (term.c:925) ==18894== by 0x4141843: term_init (term.c:350) ==18894== by 0x4139E84: el_init (el.c:84) ==18894== by 0x4049449: switch_console_loop (switch_console.c:747) ==18894== by 0x4061CE7: switch_core_runtime_loop (switch_core.c:687) ==18894== ==18894== ==18894== 88 bytes in 1 blocks are still reachable in loss record 100 of 193 ==18894== at 0x40046FF: calloc (vg_replace_malloc.c:279) ==18894== by 0x127BB985: PR_Calloc (prmem.c:455) ==18894== by 0x127C37E3: PR_NewLock (ptsynch.c:174) ==18894== by 0x12867B24: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 88 bytes in 1 blocks are still reachable in loss record 101 of 193 ==18894== at 0x40046FF: calloc (vg_replace_malloc.c:279) ==18894== by 0x127BB985: PR_Calloc (prmem.c:455) ==18894== by 0x127C37E3: PR_NewLock (ptsynch.c:174) ==18894== by 0x12867A9D: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 88 bytes in 1 blocks are still reachable in loss record 102 of 193 ==18894== at 0x40046FF: calloc (vg_replace_malloc.c:279) ==18894== by 0x127BB985: PR_Calloc (prmem.c:455) ==18894== by 0x127C37E3: PR_NewLock (ptsynch.c:174) ==18894== by 0x12868DFC: JS_InitArenaPool (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12893089: js_InitGC (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A94: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 88 bytes in 1 blocks are still reachable in loss record 103 of 193 ==18894== at 0x40046FF: calloc (vg_replace_malloc.c:279) ==18894== by 0x127BB985: PR_Calloc (prmem.c:455) ==18894== by 0x127C37E3: PR_NewLock (ptsynch.c:174) ==18894== by 0x127BEAB0: _PR_InitCallOnce (prinit.c:686) ==18894== by 0x127BE418: _PR_InitStuff (prinit.c:238) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== ==18894== ==18894== 88 bytes in 1 blocks are still reachable in loss record 104 of 193 ==18894== at 0x40046FF: calloc (vg_replace_malloc.c:279) ==18894== by 0x127BB985: PR_Calloc (prmem.c:455) ==18894== by 0x127C37E3: PR_NewLock (ptsynch.c:174) ==18894== by 0x12867AFD: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 88 bytes in 1 blocks are still reachable in loss record 105 of 193 ==18894== at 0x40046FF: calloc (vg_replace_malloc.c:279) ==18894== by 0x127BB985: PR_Calloc (prmem.c:455) ==18894== by 0x127C37E3: PR_NewLock (ptsynch.c:174) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 88 bytes in 1 blocks are still reachable in loss record 106 of 193 ==18894== at 0x40046FF: calloc (vg_replace_malloc.c:279) ==18894== by 0x127BB985: PR_Calloc (prmem.c:455) ==18894== by 0x127C37E3: PR_NewLock (ptsynch.c:174) ==18894== by 0x127CD9F8: _PR_UnixInit (unix.c:2875) ==18894== by 0x127BE422: _PR_InitStuff (prinit.c:240) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== ==18894== ==18894== 88 bytes in 1 blocks are still reachable in loss record 107 of 193 ==18894== at 0x40046FF: calloc (vg_replace_malloc.c:279) ==18894== by 0x127BB985: PR_Calloc (prmem.c:455) ==18894== by 0x127C37E3: PR_NewLock (ptsynch.c:174) ==18894== by 0x127B2BCE: _PR_InitMW (prmwait.c:241) ==18894== by 0x127BE41D: _PR_InitStuff (prinit.c:239) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== ==18894== ==18894== 88 bytes in 1 blocks are still reachable in loss record 108 of 193 ==18894== at 0x40046FF: calloc (vg_replace_malloc.c:279) ==18894== by 0x127BB985: PR_Calloc (prmem.c:455) ==18894== by 0x127C37E3: PR_NewLock (ptsynch.c:174) ==18894== by 0x127B73D5: _PR_InitLog (prlog.c:209) ==18894== by 0x127BE413: _PR_InitStuff (prinit.c:237) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== ==18894== ==18894== 88 bytes in 1 blocks are still reachable in loss record 109 of 193 ==18894== at 0x40046FF: calloc (vg_replace_malloc.c:279) ==18894== by 0x127BB985: PR_Calloc (prmem.c:455) ==18894== by 0x127C37E3: PR_NewLock (ptsynch.c:174) ==18894== by 0x127BDD50: PR_CreateStack (pratom.c:396) ==18894== by 0x127B2AAB: _PR_InitFdCache (prfdcach.c:285) ==18894== by 0x127C6520: _PR_InitIO (ptio.c:1109) ==18894== by 0x127BE40E: _PR_InitStuff (prinit.c:236) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== ==18894== ==18894== 88 bytes in 1 blocks are still reachable in loss record 110 of 193 ==18894== at 0x40046FF: calloc (vg_replace_malloc.c:279) ==18894== by 0x127BB985: PR_Calloc (prmem.c:455) ==18894== by 0x127C37E3: PR_NewLock (ptsynch.c:174) ==18894== by 0x127CC43C: _PR_InitThreads (ptthread.c:874) ==18894== by 0x127BE404: _PR_InitStuff (prinit.c:215) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== ==18894== ==18894== 88 bytes in 1 blocks are still reachable in loss record 111 of 193 ==18894== at 0x40046FF: calloc (vg_replace_malloc.c:279) ==18894== by 0x127BB985: PR_Calloc (prmem.c:455) ==18894== by 0x127C37E3: PR_NewLock (ptsynch.c:174) ==18894== by 0x127BE3B2: _PR_InitStuff (prinit.c:208) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== ==18894== ==18894== 88 bytes in 1 blocks are still reachable in loss record 112 of 193 ==18894== at 0x40046FF: calloc (vg_replace_malloc.c:279) ==18894== by 0x127BB985: PR_Calloc (prmem.c:455) ==18894== by 0x127C37E3: PR_NewLock (ptsynch.c:174) ==18894== by 0x127B72DB: _PR_InitLayerCache (prlayer.c:745) ==18894== by 0x127BE3A8: _PR_InitStuff (prinit.c:205) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== ==18894== ==18894== 88 bytes in 1 blocks are still reachable in loss record 113 of 193 ==18894== at 0x40046FF: calloc (vg_replace_malloc.c:279) ==18894== by 0x127BB985: PR_Calloc (prmem.c:455) ==18894== by 0x127C37E3: PR_NewLock (ptsynch.c:174) ==18894== by 0x127C64EB: _PR_InitIO (ptio.c:1106) ==18894== by 0x127BE40E: _PR_InitStuff (prinit.c:236) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== ==18894== ==18894== 88 bytes in 1 blocks are still reachable in loss record 114 of 193 ==18894== at 0x40046FF: calloc (vg_replace_malloc.c:279) ==18894== by 0x127BB985: PR_Calloc (prmem.c:455) ==18894== by 0x127C37E3: PR_NewLock (ptsynch.c:174) ==18894== by 0x127C6478: _PR_InitIO (ptio.c:1102) ==18894== by 0x127BE40E: _PR_InitStuff (prinit.c:236) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== ==18894== ==18894== 88 bytes in 1 blocks are still reachable in loss record 115 of 193 ==18894== at 0x40046FF: calloc (vg_replace_malloc.c:279) ==18894== by 0x127BB985: PR_Calloc (prmem.c:455) ==18894== by 0x127C37E3: PR_NewLock (ptsynch.c:174) ==18894== by 0x127B2A6D: _PR_InitFdCache (prfdcach.c:283) ==18894== by 0x127C6520: _PR_InitIO (ptio.c:1109) ==18894== by 0x127BE40E: _PR_InitStuff (prinit.c:236) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== ==18894== ==18894== 96 bytes in 1 blocks are still reachable in loss record 116 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x22AC7D: (within /lib/libcrypto.so.0.9.8b) ==18894== by 0x22B2FE: CRYPTO_malloc (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x27E5C7: lh_new (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x281272: (within /lib/libcrypto.so.0.9.8b) ==18894== by 0x280EA0: (within /lib/libcrypto.so.0.9.8b) ==18894== by 0x28141B: ERR_get_state (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x281ADA: ERR_clear_error (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x27271E: ENGINE_load_dynamic (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x270BE6: ENGINE_load_builtin_engines (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x127FE136: Curl_ossl_init (in /usr/lib/libcurl.so.3.0.0) ==18894== by 0x1280F3AB: Curl_ssl_init (in /usr/lib/libcurl.so.3.0.0) ==18894== ==18894== ==18894== 99 bytes in 4 blocks are still reachable in loss record 117 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x9C1937: _dl_new_object (in /lib/ld-2.5.so) ==18894== by 0x9BD020: _dl_map_object_from_fd (in /lib/ld-2.5.so) ==18894== by 0x9BF46B: _dl_map_object (in /lib/ld-2.5.so) ==18894== by 0x9C32D5: openaux (in /lib/ld-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0x9C3894: _dl_map_object_deps (in /lib/ld-2.5.so) ==18894== by 0x9C8A8C: dl_open_worker (in /lib/ld-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0x9C8411: _dl_open (in /lib/ld-2.5.so) ==18894== by 0xB1FC4C: dlopen_doit (in /lib/libdl-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== ==18894== ==18894== 100 bytes in 1 blocks are still reachable in loss record 118 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x22AC7D: (within /lib/libcrypto.so.0.9.8b) ==18894== by 0x22B2FE: CRYPTO_malloc (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x26E6F0: ENGINE_new (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x2737DB: ENGINE_load_padlock (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x270BEB: ENGINE_load_builtin_engines (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x127FE136: Curl_ossl_init (in /usr/lib/libcurl.so.3.0.0) ==18894== by 0x1280F3AB: Curl_ssl_init (in /usr/lib/libcurl.so.3.0.0) ==18894== by 0x12807162: curl_global_init (in /usr/lib/libcurl.so.3.0.0) ==18894== by 0x1279FAB3: mod_spidermonkey_load (mod_spidermonkey.c:3751) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== ==18894== ==18894== 100 bytes in 1 blocks are still reachable in loss record 119 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x22AC7D: (within /lib/libcrypto.so.0.9.8b) ==18894== by 0x22B2FE: CRYPTO_malloc (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x26E6F0: ENGINE_new (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x27264B: ENGINE_load_dynamic (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x270BE6: ENGINE_load_builtin_engines (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x127FE136: Curl_ossl_init (in /usr/lib/libcurl.so.3.0.0) ==18894== by 0x1280F3AB: Curl_ssl_init (in /usr/lib/libcurl.so.3.0.0) ==18894== by 0x12807162: curl_global_init (in /usr/lib/libcurl.so.3.0.0) ==18894== by 0x1279FAB3: mod_spidermonkey_load (mod_spidermonkey.c:3751) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== ==18894== ==18894== 100 bytes in 5 blocks are still reachable in loss record 120 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40DED94: sqlite3MallocX (sqliteInt.h:278) ==18894== by 0x40D3CD2: sqlite3HashInsert (hash.c:363) ==18894== by 0x40531EE: switch_core_hash_insert (switch_core_hash.c:67) ==18894== by 0x7AA518D: mod_cdr_csv_load (mod_cdr_csv.c:361) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 104 bytes in 1 blocks are still reachable in loss record 121 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40C7FFE: apr_allocator_create (apr_pools.c:91) ==18894== by 0x40502BF: switch_core_perform_new_memory_pool (switch_core_memory.c:353) ==18894== by 0x8036763: mod_loopback_load (mod_loopback.c:871) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 104 bytes in 1 blocks are still reachable in loss record 122 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40C7FFE: apr_allocator_create (apr_pools.c:91) ==18894== by 0x40502BF: switch_core_perform_new_memory_pool (switch_core_memory.c:353) ==18894== by 0x1279F75A: mod_spidermonkey_load (mod_spidermonkey.c:1007) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 104 bytes in 1 blocks are still reachable in loss record 123 of 193 ==18894== at 0x40046FF: calloc (vg_replace_malloc.c:279) ==18894== by 0x127BB985: PR_Calloc (prmem.c:455) ==18894== by 0x127C443C: PR_NewMonitor (ptsynch.c:463) ==18894== by 0x127CDA2D: _PR_UnixInit (unix.c:2877) ==18894== by 0x127BE422: _PR_InitStuff (prinit.c:240) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== ==18894== ==18894== 104 bytes in 1 blocks are still reachable in loss record 124 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40C7FFE: apr_allocator_create (apr_pools.c:91) ==18894== by 0x40502BF: switch_core_perform_new_memory_pool (switch_core_memory.c:353) ==18894== by 0x40643E4: switch_load_network_lists (switch_core.c:891) ==18894== by 0x40659E4: switch_core_init_and_modload (switch_core.c:1457) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 104 bytes in 1 blocks are still reachable in loss record 125 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40C7FFE: apr_allocator_create (apr_pools.c:91) ==18894== by 0x40502BF: switch_core_perform_new_memory_pool (switch_core_memory.c:353) ==18894== by 0x406C7CD: switch_loadable_module_load_module_ex (switch_loadable_module.c:785) ==18894== by 0x406D097: switch_loadable_module_init (switch_loadable_module.c:1160) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 104 bytes in 1 blocks are still reachable in loss record 126 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40C7FFE: apr_allocator_create (apr_pools.c:91) ==18894== by 0x40502BF: switch_core_perform_new_memory_pool (switch_core_memory.c:353) ==18894== by 0x11D2FDC6: mod_sndfile_load (mod_sndfile.c:400) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 104 bytes in 1 blocks are still reachable in loss record 127 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40C7FFE: apr_allocator_create (apr_pools.c:91) ==18894== by 0x40502BF: switch_core_perform_new_memory_pool (switch_core_memory.c:353) ==18894== by 0x406C7CD: switch_loadable_module_load_module_ex (switch_loadable_module.c:785) ==18894== by 0x406D073: switch_loadable_module_init (switch_loadable_module.c:1159) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 104 bytes in 1 blocks are still reachable in loss record 128 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40C7FFE: apr_allocator_create (apr_pools.c:91) ==18894== by 0x40502BF: switch_core_perform_new_memory_pool (switch_core_memory.c:353) ==18894== by 0x406CE99: switch_loadable_module_init (switch_loadable_module.c:1136) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 104 bytes in 1 blocks are still reachable in loss record 129 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40C7FFE: apr_allocator_create (apr_pools.c:91) ==18894== by 0x40502BF: switch_core_perform_new_memory_pool (switch_core_memory.c:353) ==18894== by 0x4065C3C: switch_scheduler_task_thread_start (switch_scheduler.c:295) ==18894== by 0x40639C5: switch_core_init (switch_core.c:1250) ==18894== by 0x406585C: switch_core_init_and_modload (switch_core.c:1422) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 104 bytes in 1 blocks are still reachable in loss record 130 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40C7FFE: apr_allocator_create (apr_pools.c:91) ==18894== by 0x40C88ED: apr_pool_initialize (apr_pools.c:522) ==18894== by 0x40C971E: apr_initialize (start.c:55) ==18894== by 0x8049C7B: main (switch.c:596) ==18894== ==18894== ==18894== 128 bytes in 1 blocks are still reachable in loss record 131 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0xC29E6D: _nc_read_file_entry (in /usr/lib/libncurses.so.5.5) ==18894== by 0xC29FA4: _nc_read_entry (in /usr/lib/libncurses.so.5.5) ==18894== by 0xC23FDC: _nc_setupterm (in /usr/lib/libncurses.so.5.5) ==18894== by 0xC24747: tgetent (in /usr/lib/libncurses.so.5.5) ==18894== by 0x4141441: term_set (term.c:925) ==18894== by 0x4141843: term_init (term.c:350) ==18894== by 0x4139E84: el_init (el.c:84) ==18894== by 0x4049449: switch_console_loop (switch_console.c:747) ==18894== by 0x4061CE7: switch_core_runtime_loop (switch_core.c:687) ==18894== by 0x804A453: main (switch.c:715) ==18894== ==18894== ==18894== 128 bytes in 1 blocks are still reachable in loss record 132 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x1287870C: JS_DHashAllocTable (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12878532: JS_DHashTableInit (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x128CA989: js_InitPropertyTree (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867B74: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 128 bytes in 1 blocks are still reachable in loss record 133 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x128A3D24: js_SetupLocks (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867AE5: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 132 bytes in 3 blocks are still reachable in loss record 134 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x9BE714: expand_dynamic_string_token (in /lib/ld-2.5.so) ==18894== by 0x9BF3A9: _dl_map_object (in /lib/ld-2.5.so) ==18894== by 0x9C8A30: dl_open_worker (in /lib/ld-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0x9C8411: _dl_open (in /lib/ld-2.5.so) ==18894== by 0xB1FC4C: dlopen_doit (in /lib/libdl-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0xB202CB: _dlerror_run (in /lib/libdl-2.5.so) ==18894== by 0xB1FB83: dlopen@@GLIBC_2.1 (in /lib/libdl-2.5.so) ==18894== by 0x4067486: switch_dso_open (switch_dso.c:94) ==18894== by 0x1279F92E: mod_spidermonkey_load (mod_spidermonkey.c:894) ==18894== ==18894== ==18894== 160 bytes in 1 blocks are possibly lost in loss record 135 of 193 ==18894== at 0x40046FF: calloc (vg_replace_malloc.c:279) ==18894== by 0x9C7CC9: _dl_allocate_tls (in /lib/ld-2.5.so) ==18894== by 0xB53B59: pthread_create@@GLIBC_2.1 (in /lib/libpthread-2.5.so) ==18894== by 0xB541D7: pthread_create at GLIBC_2.0 (in /lib/libpthread-2.5.so) ==18894== by 0x40CDA64: apr_thread_create (thread.c:176) ==18894== by 0x403ECF8: switch_thread_create (switch_apr.c:631) ==18894== by 0x4062738: switch_core_launch_thread (switch_core.c:400) ==18894== by 0x406D4D6: switch_loadable_module_init (switch_loadable_module.c:121) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 160 bytes in 1 blocks are still reachable in loss record 136 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x9C8881: add_to_global (in /lib/ld-2.5.so) ==18894== by 0x9C8D75: dl_open_worker (in /lib/ld-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0x9C8411: _dl_open (in /lib/ld-2.5.so) ==18894== by 0xB1FC4C: dlopen_doit (in /lib/libdl-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0xB202CB: _dlerror_run (in /lib/libdl-2.5.so) ==18894== by 0xB1FB83: dlopen@@GLIBC_2.1 (in /lib/libdl-2.5.so) ==18894== by 0x4067486: switch_dso_open (switch_dso.c:94) ==18894== by 0x406C96B: switch_loadable_module_load_module_ex (switch_loadable_module.c:831) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== ==18894== ==18894== 160 bytes in 1 blocks are possibly lost in loss record 137 of 193 ==18894== at 0x40046FF: calloc (vg_replace_malloc.c:279) ==18894== by 0x9C7CC9: _dl_allocate_tls (in /lib/ld-2.5.so) ==18894== by 0xB53B59: pthread_create@@GLIBC_2.1 (in /lib/libpthread-2.5.so) ==18894== by 0xB541D7: pthread_create at GLIBC_2.0 (in /lib/libpthread-2.5.so) ==18894== by 0x40CDA64: apr_thread_create (thread.c:176) ==18894== by 0x403ECF8: switch_thread_create (switch_apr.c:631) ==18894== by 0x11D90A0E: mod_local_stream_load (mod_local_stream.c:507) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 172 bytes in 1 blocks are still reachable in loss record 138 of 193 ==18894== at 0x40046FF: calloc (vg_replace_malloc.c:279) ==18894== by 0xC23C86: _nc_setupterm (in /usr/lib/libncurses.so.5.5) ==18894== by 0xC24747: tgetent (in /usr/lib/libncurses.so.5.5) ==18894== by 0x4141441: term_set (term.c:925) ==18894== by 0x4141843: term_init (term.c:350) ==18894== by 0x4139E84: el_init (el.c:84) ==18894== by 0x4049449: switch_console_loop (switch_console.c:747) ==18894== by 0x4061CE7: switch_core_runtime_loop (switch_core.c:687) ==18894== by 0x804A453: main (switch.c:715) ==18894== ==18894== ==18894== 172 bytes in 1 blocks are still reachable in loss record 139 of 193 ==18894== at 0x40046FF: calloc (vg_replace_malloc.c:279) ==18894== by 0x127BB985: PR_Calloc (prmem.c:455) ==18894== by 0x127CC4BE: _PR_InitThreads (ptthread.c:878) ==18894== by 0x127BE404: _PR_InitStuff (prinit.c:215) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== ==18894== ==18894== 176 bytes in 4 blocks are still reachable in loss record 140 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x9C3905: _dl_map_object_deps (in /lib/ld-2.5.so) ==18894== by 0x9C8A8C: dl_open_worker (in /lib/ld-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0x9C8411: _dl_open (in /lib/ld-2.5.so) ==18894== by 0xB1FC4C: dlopen_doit (in /lib/libdl-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0xB202CB: _dlerror_run (in /lib/libdl-2.5.so) ==18894== by 0xB1FB83: dlopen@@GLIBC_2.1 (in /lib/libdl-2.5.so) ==18894== by 0x4067486: switch_dso_open (switch_dso.c:94) ==18894== by 0x406C96B: switch_loadable_module_load_module_ex (switch_loadable_module.c:831) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== ==18894== ==18894== 192 bytes in 3 blocks are still reachable in loss record 141 of 193 ==18894== at 0x40046FF: calloc (vg_replace_malloc.c:279) ==18894== by 0x9C6573: _dl_check_map_versions (in /lib/ld-2.5.so) ==18894== by 0x9C8D68: dl_open_worker (in /lib/ld-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0x9C8411: _dl_open (in /lib/ld-2.5.so) ==18894== by 0xB1FC4C: dlopen_doit (in /lib/libdl-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0xB202CB: _dlerror_run (in /lib/libdl-2.5.so) ==18894== by 0xB1FB83: dlopen@@GLIBC_2.1 (in /lib/libdl-2.5.so) ==18894== by 0x4067486: switch_dso_open (switch_dso.c:94) ==18894== by 0x1279F92E: mod_spidermonkey_load (mod_spidermonkey.c:894) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== ==18894== ==18894== 195 bytes in 5 blocks are still reachable in loss record 142 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x9C1937: _dl_new_object (in /lib/ld-2.5.so) ==18894== by 0x9BD020: _dl_map_object_from_fd (in /lib/ld-2.5.so) ==18894== by 0x9BF46B: _dl_map_object (in /lib/ld-2.5.so) ==18894== by 0x9C8A30: dl_open_worker (in /lib/ld-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0x9C8411: _dl_open (in /lib/ld-2.5.so) ==18894== by 0xB1FC4C: dlopen_doit (in /lib/libdl-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0xB202CB: _dlerror_run (in /lib/libdl-2.5.so) ==18894== by 0xB1FB83: dlopen@@GLIBC_2.1 (in /lib/libdl-2.5.so) ==18894== by 0x4067486: switch_dso_open (switch_dso.c:94) ==18894== ==18894== ==18894== 220 bytes in 1 blocks are definitely lost in loss record 143 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x412F754: crypto_alloc (alloc.c:97) ==18894== by 0x41320C7: aes_cbc_alloc (aes_cbc.c:71) ==18894== by 0x412B53C: cipher_type_self_test (cipher.c:264) ==18894== by 0x412F0B5: crypto_kernel_load_cipher_type (crypto_kernel.c:310) ==18894== by 0x412F668: crypto_kernel_init (crypto_kernel.c:157) ==18894== by 0x4129026: srtp_init (srtp.c:1081) ==18894== by 0x407AF62: switch_rtp_init (switch_rtp.c:611) ==18894== by 0x40639D6: switch_core_init (switch_core.c:1252) ==18894== by 0x406585C: switch_core_init_and_modload (switch_core.c:1422) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 240 bytes in 1 blocks are definitely lost in loss record 144 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x412F754: crypto_alloc (alloc.c:97) ==18894== by 0x412CF0B: aes_icm_alloc_ismacryp (aes_icm.c:115) ==18894== by 0x412CF9B: aes_icm_alloc (aes_icm.c:134) ==18894== by 0x412B53C: cipher_type_self_test (cipher.c:264) ==18894== by 0x412F0B5: crypto_kernel_load_cipher_type (crypto_kernel.c:310) ==18894== by 0x412F64A: crypto_kernel_init (crypto_kernel.c:154) ==18894== by 0x4129026: srtp_init (srtp.c:1081) ==18894== by 0x407AF62: switch_rtp_init (switch_rtp.c:611) ==18894== by 0x40639D6: switch_core_init (switch_core.c:1252) ==18894== by 0x406585C: switch_core_init_and_modload (switch_core.c:1422) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 396 bytes in 1 blocks are still reachable in loss record 145 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x22AC7D: (within /lib/libcrypto.so.0.9.8b) ==18894== by 0x22B2FE: CRYPTO_malloc (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x2813CD: ERR_get_state (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x281ADA: ERR_clear_error (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x27271E: ENGINE_load_dynamic (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x270BE6: ENGINE_load_builtin_engines (in /lib/libcrypto.so.0.9.8b) ==18894== by 0x127FE136: Curl_ossl_init (in /usr/lib/libcurl.so.3.0.0) ==18894== by 0x1280F3AB: Curl_ssl_init (in /usr/lib/libcurl.so.3.0.0) ==18894== by 0x12807162: curl_global_init (in /usr/lib/libcurl.so.3.0.0) ==18894== by 0x1279FAB3: mod_spidermonkey_load (mod_spidermonkey.c:3751) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== ==18894== ==18894== 440 bytes in 2 blocks are still reachable in loss record 146 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x9C3B6B: _dl_map_object_deps (in /lib/ld-2.5.so) ==18894== by 0x9C8A8C: dl_open_worker (in /lib/ld-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0x9C8411: _dl_open (in /lib/ld-2.5.so) ==18894== by 0xB1FC4C: dlopen_doit (in /lib/libdl-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0xB202CB: _dlerror_run (in /lib/libdl-2.5.so) ==18894== by 0xB1FB83: dlopen@@GLIBC_2.1 (in /lib/libdl-2.5.so) ==18894== by 0x4067486: switch_dso_open (switch_dso.c:94) ==18894== by 0x406C96B: switch_loadable_module_load_module_ex (switch_loadable_module.c:831) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== ==18894== ==18894== 512 bytes in 1 blocks are still reachable in loss record 147 of 193 ==18894== at 0x40046FF: calloc (vg_replace_malloc.c:279) ==18894== by 0x127BB985: PR_Calloc (prmem.c:455) ==18894== by 0x127BA8F9: _PR_InitTPD (prtpd.c:96) ==18894== by 0x127BE3A3: _PR_InitStuff (prinit.c:204) ==18894== by 0x127BE433: _PR_ImplicitInitialization (prinit.c:245) ==18894== by 0x127C37CF: PR_NewLock (ptsynch.c:172) ==18894== by 0x128D452B: js_InitStringGlobals (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A0E: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== ==18894== ==18894== 572 bytes in 1 blocks are definitely lost in loss record 148 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x7AFDE93: sofia_glue_do_invite (sofia_glue.c:1599) ==18894== by 0x7ACB191: sofia_on_init (mod_sofia.c:102) ==18894== by 0x405B124: switch_core_session_run (switch_core_state_machine.c:480) ==18894== by 0x4058204: switch_core_session_thread (switch_core_session.c:1064) ==18894== by 0x40CD6C5: dummy_worker (thread.c:138) ==18894== by 0xB5346A: start_thread (in /lib/libpthread-2.5.so) ==18894== by 0xAAADBD: clone (in /lib/libc-2.5.so) ==18894== ==18894== ==18894== 660 bytes in 3 blocks are still reachable in loss record 149 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x9C3B6B: _dl_map_object_deps (in /lib/ld-2.5.so) ==18894== by 0x9C8A8C: dl_open_worker (in /lib/ld-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0x9C8411: _dl_open (in /lib/ld-2.5.so) ==18894== by 0xB1FC4C: dlopen_doit (in /lib/libdl-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0xB202CB: _dlerror_run (in /lib/libdl-2.5.so) ==18894== by 0xB1FB83: dlopen@@GLIBC_2.1 (in /lib/libdl-2.5.so) ==18894== by 0x4067486: switch_dso_open (switch_dso.c:94) ==18894== by 0x1279F92E: mod_spidermonkey_load (mod_spidermonkey.c:894) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== ==18894== ==18894== 944 bytes in 6 blocks are still reachable in loss record 150 of 193 ==18894== at 0x40046FF: calloc (vg_replace_malloc.c:279) ==18894== by 0x9C6573: _dl_check_map_versions (in /lib/ld-2.5.so) ==18894== by 0x9C8D68: dl_open_worker (in /lib/ld-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0x9C8411: _dl_open (in /lib/ld-2.5.so) ==18894== by 0xB1FC4C: dlopen_doit (in /lib/libdl-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0xB202CB: _dlerror_run (in /lib/libdl-2.5.so) ==18894== by 0xB1FB83: dlopen@@GLIBC_2.1 (in /lib/libdl-2.5.so) ==18894== by 0x4067486: switch_dso_open (switch_dso.c:94) ==18894== by 0x406C96B: switch_loadable_module_load_module_ex (switch_loadable_module.c:831) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== ==18894== ==18894== 1,024 bytes in 1 blocks are still reachable in loss record 151 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x4062333: switch_core_set_globals (switch_core.c:449) ==18894== by 0x804A06D: main (switch.c:667) ==18894== ==18894== ==18894== 1,061 bytes in 34 blocks are still reachable in loss record 152 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x9BE714: expand_dynamic_string_token (in /lib/ld-2.5.so) ==18894== by 0x9BF3A9: _dl_map_object (in /lib/ld-2.5.so) ==18894== by 0x9C8A30: dl_open_worker (in /lib/ld-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0x9C8411: _dl_open (in /lib/ld-2.5.so) ==18894== by 0xB1FC4C: dlopen_doit (in /lib/libdl-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0xB202CB: _dlerror_run (in /lib/libdl-2.5.so) ==18894== by 0xB1FB83: dlopen@@GLIBC_2.1 (in /lib/libdl-2.5.so) ==18894== by 0x40674AF: switch_dso_open (switch_dso.c:96) ==18894== by 0x406CC16: switch_loadable_module_load_module_ex (switch_loadable_module.c:804) ==18894== ==18894== ==18894== 1,061 bytes in 34 blocks are still reachable in loss record 153 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x9C1937: _dl_new_object (in /lib/ld-2.5.so) ==18894== by 0x9BD020: _dl_map_object_from_fd (in /lib/ld-2.5.so) ==18894== by 0x9BF46B: _dl_map_object (in /lib/ld-2.5.so) ==18894== by 0x9C8A30: dl_open_worker (in /lib/ld-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0x9C8411: _dl_open (in /lib/ld-2.5.so) ==18894== by 0xB1FC4C: dlopen_doit (in /lib/libdl-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0xB202CB: _dlerror_run (in /lib/libdl-2.5.so) ==18894== by 0xB1FB83: dlopen@@GLIBC_2.1 (in /lib/libdl-2.5.so) ==18894== by 0x40674AF: switch_dso_open (switch_dso.c:96) ==18894== ==18894== ==18894== 1,344 bytes in 1 blocks are still reachable in loss record 154 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x7B17A2E: msg_mclass_clone (msg_mclass.c:112) ==18894== by 0x7B94268: sip_extend_mclass (sip_parser.c:132) ==18894== by 0x7AE4887: config_sofia (sofia.c:2007) ==18894== by 0x7AC85C9: mod_sofia_load (mod_sofia.c:3264) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 1,381 bytes in 1 blocks are still reachable in loss record 155 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0xC29666: _nc_read_file_entry (in /usr/lib/libncurses.so.5.5) ==18894== by 0xC29FA4: _nc_read_entry (in /usr/lib/libncurses.so.5.5) ==18894== by 0xC23FDC: _nc_setupterm (in /usr/lib/libncurses.so.5.5) ==18894== by 0xC24747: tgetent (in /usr/lib/libncurses.so.5.5) ==18894== by 0x4141441: term_set (term.c:925) ==18894== by 0x4141843: term_init (term.c:350) ==18894== by 0x4139E84: el_init (el.c:84) ==18894== by 0x4049449: switch_console_loop (switch_console.c:747) ==18894== by 0x4061CE7: switch_core_runtime_loop (switch_core.c:687) ==18894== by 0x804A453: main (switch.c:715) ==18894== ==18894== ==18894== 1,408 bytes in 16 blocks are still reachable in loss record 156 of 193 ==18894== at 0x40046FF: calloc (vg_replace_malloc.c:279) ==18894== by 0x127BB985: PR_Calloc (prmem.c:455) ==18894== by 0x127C37E3: PR_NewLock (ptsynch.c:174) ==18894== by 0x128A3CF4: js_SetupLocks (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867AE5: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 1,600 (44 direct, 1,556 indirect) bytes in 1 blocks are definitely lost in loss record 157 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40724DA: switch_event_create_subclass_detailed (switch_event.c:628) ==18894== by 0x40A902E: switch_event_import_xml (switch_xml_config.c:55) ==18894== by 0x11CAC4EE: ??? (mod_voicemail.c:619) ==18894== by 0x11CAD21F: mod_voicemail_load (mod_voicemail.c:821) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 64 bytes in 4 blocks are indirectly lost in loss record 158 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40720BF: switch_event_base_add_header (switch_event.c:737) ==18894== by 0x40721A5: switch_event_add_header_string (switch_event.c:788) ==18894== by 0x11CAC625: ??? (mod_voicemail.c:637) ==18894== by 0x11CAD21F: mod_voicemail_load (mod_voicemail.c:821) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 79 bytes in 4 blocks are indirectly lost in loss record 159 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x407176E: my_dup (switch_event.c:95) ==18894== by 0x40720E8: switch_event_base_add_header (switch_event.c:745) ==18894== by 0x40721A5: switch_event_add_header_string (switch_event.c:788) ==18894== by 0x11CAC625: ??? (mod_voicemail.c:637) ==18894== by 0x11CAD21F: mod_voicemail_load (mod_voicemail.c:821) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 99 bytes in 4 blocks are indirectly lost in loss record 160 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x407176E: my_dup (switch_event.c:95) ==18894== by 0x4072188: switch_event_add_header_string (switch_event.c:788) ==18894== by 0x11CAC625: ??? (mod_voicemail.c:637) ==18894== by 0x11CAD21F: mod_voicemail_load (mod_voicemail.c:821) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 150 bytes in 37 blocks are indirectly lost in loss record 161 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x407176E: my_dup (switch_event.c:95) ==18894== by 0x4072188: switch_event_add_header_string (switch_event.c:788) ==18894== by 0x40A8FE4: switch_event_import_xml (switch_xml_config.c:63) ==18894== by 0x11CAC4EE: ??? (mod_voicemail.c:619) ==18894== by 0x11CAD21F: mod_voicemail_load (mod_voicemail.c:821) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 572 bytes in 37 blocks are indirectly lost in loss record 162 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x407176E: my_dup (switch_event.c:95) ==18894== by 0x40720E8: switch_event_base_add_header (switch_event.c:745) ==18894== by 0x40721A5: switch_event_add_header_string (switch_event.c:788) ==18894== by 0x40A8FE4: switch_event_import_xml (switch_xml_config.c:63) ==18894== by 0x11CAC4EE: ??? (mod_voicemail.c:619) ==18894== by 0x11CAD21F: mod_voicemail_load (mod_voicemail.c:821) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 592 bytes in 37 blocks are indirectly lost in loss record 163 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40720BF: switch_event_base_add_header (switch_event.c:737) ==18894== by 0x40721A5: switch_event_add_header_string (switch_event.c:788) ==18894== by 0x40A8FE4: switch_event_import_xml (switch_xml_config.c:63) ==18894== by 0x11CAC4EE: ??? (mod_voicemail.c:619) ==18894== by 0x11CAD21F: mod_voicemail_load (mod_voicemail.c:821) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 1,700 bytes in 1 blocks are still reachable in loss record 164 of 193 ==18894== at 0x40054BB: realloc (vg_replace_malloc.c:306) ==18894== by 0xC21BDF: _nc_doalloc (in /usr/lib/libncurses.so.5.5) ==18894== by 0xC29B19: _nc_read_file_entry (in /usr/lib/libncurses.so.5.5) ==18894== by 0xC29FA4: _nc_read_entry (in /usr/lib/libncurses.so.5.5) ==18894== by 0xC23FDC: _nc_setupterm (in /usr/lib/libncurses.so.5.5) ==18894== by 0xC24747: tgetent (in /usr/lib/libncurses.so.5.5) ==18894== by 0x4141441: term_set (term.c:925) ==18894== by 0x4141843: term_init (term.c:350) ==18894== by 0x4139E84: el_init (el.c:84) ==18894== by 0x4049449: switch_console_loop (switch_console.c:747) ==18894== by 0x4061CE7: switch_core_runtime_loop (switch_core.c:687) ==18894== by 0x804A453: main (switch.c:715) ==18894== ==18894== ==18894== 2,481 bytes in 4 blocks are still reachable in loss record 165 of 193 ==18894== at 0x40046FF: calloc (vg_replace_malloc.c:279) ==18894== by 0x9C16CA: _dl_new_object (in /lib/ld-2.5.so) ==18894== by 0x9BD020: _dl_map_object_from_fd (in /lib/ld-2.5.so) ==18894== by 0x9BF46B: _dl_map_object (in /lib/ld-2.5.so) ==18894== by 0x9C32D5: openaux (in /lib/ld-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0x9C3894: _dl_map_object_deps (in /lib/ld-2.5.so) ==18894== by 0x9C8A8C: dl_open_worker (in /lib/ld-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0x9C8411: _dl_open (in /lib/ld-2.5.so) ==18894== by 0xB1FC4C: dlopen_doit (in /lib/libdl-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== ==18894== ==18894== 2,544 bytes in 34 blocks are still reachable in loss record 166 of 193 ==18894== at 0x40046FF: calloc (vg_replace_malloc.c:279) ==18894== by 0x9C6573: _dl_check_map_versions (in /lib/ld-2.5.so) ==18894== by 0x9C8D68: dl_open_worker (in /lib/ld-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0x9C8411: _dl_open (in /lib/ld-2.5.so) ==18894== by 0xB1FC4C: dlopen_doit (in /lib/libdl-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0xB202CB: _dlerror_run (in /lib/libdl-2.5.so) ==18894== by 0xB1FB83: dlopen@@GLIBC_2.1 (in /lib/libdl-2.5.so) ==18894== by 0x40674AF: switch_dso_open (switch_dso.c:96) ==18894== by 0x406CC16: switch_loadable_module_load_module_ex (switch_loadable_module.c:804) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== ==18894== ==18894== 3,072 bytes in 1 blocks are still reachable in loss record 167 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x1287870C: JS_DHashAllocTable (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12878532: JS_DHashTableInit (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x128930BF: js_InitGC (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x12867A94: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 3,235 bytes in 5 blocks are still reachable in loss record 168 of 193 ==18894== at 0x40046FF: calloc (vg_replace_malloc.c:279) ==18894== by 0x9C16CA: _dl_new_object (in /lib/ld-2.5.so) ==18894== by 0x9BD020: _dl_map_object_from_fd (in /lib/ld-2.5.so) ==18894== by 0x9BF46B: _dl_map_object (in /lib/ld-2.5.so) ==18894== by 0x9C8A30: dl_open_worker (in /lib/ld-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0x9C8411: _dl_open (in /lib/ld-2.5.so) ==18894== by 0xB1FC4C: dlopen_doit (in /lib/libdl-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0xB202CB: _dlerror_run (in /lib/libdl-2.5.so) ==18894== by 0xB1FB83: dlopen@@GLIBC_2.1 (in /lib/libdl-2.5.so) ==18894== by 0x4067486: switch_dso_open (switch_dso.c:94) ==18894== ==18894== ==18894== 3,744 bytes in 36 blocks are still reachable in loss record 169 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40C7FFE: apr_allocator_create (apr_pools.c:91) ==18894== by 0x40502BF: switch_core_perform_new_memory_pool (switch_core_memory.c:353) ==18894== by 0x406C7CD: switch_loadable_module_load_module_ex (switch_loadable_module.c:785) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 6,920 bytes in 34 blocks are still reachable in loss record 170 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x9C3B6B: _dl_map_object_deps (in /lib/ld-2.5.so) ==18894== by 0x9C8A8C: dl_open_worker (in /lib/ld-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0x9C8411: _dl_open (in /lib/ld-2.5.so) ==18894== by 0xB1FC4C: dlopen_doit (in /lib/libdl-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0xB202CB: _dlerror_run (in /lib/libdl-2.5.so) ==18894== by 0xB1FB83: dlopen@@GLIBC_2.1 (in /lib/libdl-2.5.so) ==18894== by 0x40674AF: switch_dso_open (switch_dso.c:96) ==18894== by 0x406CC16: switch_loadable_module_load_module_ex (switch_loadable_module.c:804) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== ==18894== ==18894== 8,192 bytes in 1 blocks are still reachable in loss record 171 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40C8710: apr_pool_create_ex (apr_pools.c:300) ==18894== by 0x40502E6: switch_core_perform_new_memory_pool (switch_core_memory.c:357) ==18894== by 0x40643E4: switch_load_network_lists (switch_core.c:891) ==18894== by 0x40659E4: switch_core_init_and_modload (switch_core.c:1457) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 8,192 bytes in 1 blocks are still reachable in loss record 172 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40C8710: apr_pool_create_ex (apr_pools.c:300) ==18894== by 0x40502E6: switch_core_perform_new_memory_pool (switch_core_memory.c:357) ==18894== by 0x1279F75A: mod_spidermonkey_load (mod_spidermonkey.c:1007) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 8,192 bytes in 1 blocks are still reachable in loss record 173 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x4144CD8: libedit_fgetln (fgetln.c:60) ==18894== by 0x4145E8F: history (history.c:685) ==18894== by 0x4049C9E: switch_console_loop (switch_console.c:804) ==18894== by 0x4061CE7: switch_core_runtime_loop (switch_core.c:687) ==18894== by 0x804A453: main (switch.c:715) ==18894== ==18894== ==18894== 8,192 bytes in 1 blocks are still reachable in loss record 174 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40C8B7C: apr_palloc (apr_pools.c:300) ==18894== by 0x404FC7F: switch_core_perform_alloc (switch_core_memory.c:442) ==18894== by 0x40677B3: switch_loadable_module_create_interface (switch_loadable_module.c:1632) ==18894== by 0x10D88DC9: mod_dptools_load (mod_dptools.c:2754) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 8,192 bytes in 1 blocks are still reachable in loss record 175 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40C8710: apr_pool_create_ex (apr_pools.c:300) ==18894== by 0x40CDA36: apr_thread_create (thread.c:171) ==18894== by 0x403ECF8: switch_thread_create (switch_apr.c:631) ==18894== by 0x7AFF689: sofia_presence_event_thread_start (sofia_presence.c:755) ==18894== by 0x7AFF791: sofia_presence_event_handler (sofia_presence.c:768) ==18894== by 0x4073C1B: switch_event_deliver (switch_event.c:342) ==18894== by 0x4073DAA: switch_event_dispatch_thread (switch_event.c:255) ==18894== by 0x40CD6C5: dummy_worker (thread.c:138) ==18894== by 0xB5346A: start_thread (in /lib/libpthread-2.5.so) ==18894== by 0xAAADBD: clone (in /lib/libc-2.5.so) ==18894== ==18894== ==18894== 8,192 bytes in 1 blocks are still reachable in loss record 176 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40C8710: apr_pool_create_ex (apr_pools.c:300) ==18894== by 0x40502E6: switch_core_perform_new_memory_pool (switch_core_memory.c:357) ==18894== by 0x11D2FDC6: mod_sndfile_load (mod_sndfile.c:400) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 8,192 bytes in 1 blocks are still reachable in loss record 177 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40C8B7C: apr_palloc (apr_pools.c:300) ==18894== by 0x404FC7F: switch_core_perform_alloc (switch_core_memory.c:442) ==18894== by 0x11CD6624: ??? (switch_loadable_module.h:396) ==18894== by 0x11CD6E91: mod_voipcodecs_load (mod_voipcodecs.c:712) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 8,192 bytes in 1 blocks are still reachable in loss record 178 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40C8710: apr_pool_create_ex (apr_pools.c:300) ==18894== by 0x40502E6: switch_core_perform_new_memory_pool (switch_core_memory.c:357) ==18894== by 0x8036763: mod_loopback_load (mod_loopback.c:871) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 8,192 bytes in 1 blocks are still reachable in loss record 179 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40C8710: apr_pool_create_ex (apr_pools.c:300) ==18894== by 0x40502E6: switch_core_perform_new_memory_pool (switch_core_memory.c:357) ==18894== by 0x406C7CD: switch_loadable_module_load_module_ex (switch_loadable_module.c:785) ==18894== by 0x406D073: switch_loadable_module_init (switch_loadable_module.c:1159) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 8,192 bytes in 1 blocks are still reachable in loss record 180 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40C8710: apr_pool_create_ex (apr_pools.c:300) ==18894== by 0x40502E6: switch_core_perform_new_memory_pool (switch_core_memory.c:357) ==18894== by 0x406CE99: switch_loadable_module_init (switch_loadable_module.c:1136) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 8,192 bytes in 1 blocks are still reachable in loss record 181 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40C8710: apr_pool_create_ex (apr_pools.c:300) ==18894== by 0x40502E6: switch_core_perform_new_memory_pool (switch_core_memory.c:357) ==18894== by 0x4065C3C: switch_scheduler_task_thread_start (switch_scheduler.c:295) ==18894== by 0x40639C5: switch_core_init (switch_core.c:1250) ==18894== by 0x406585C: switch_core_init_and_modload (switch_core.c:1422) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 8,192 bytes in 1 blocks are still reachable in loss record 182 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40C8710: apr_pool_create_ex (apr_pools.c:300) ==18894== by 0x40502E6: switch_core_perform_new_memory_pool (switch_core_memory.c:357) ==18894== by 0x406C7CD: switch_loadable_module_load_module_ex (switch_loadable_module.c:785) ==18894== by 0x406D097: switch_loadable_module_init (switch_loadable_module.c:1160) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 8,192 bytes in 1 blocks are still reachable in loss record 183 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40C8710: apr_pool_create_ex (apr_pools.c:300) ==18894== by 0x40CDA36: apr_thread_create (thread.c:171) ==18894== by 0x403ECF8: switch_thread_create (switch_apr.c:631) ==18894== by 0x4065CA6: switch_scheduler_task_thread_start (switch_scheduler.c:300) ==18894== by 0x40639C5: switch_core_init (switch_core.c:1250) ==18894== by 0x406585C: switch_core_init_and_modload (switch_core.c:1422) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 8,192 bytes in 1 blocks are still reachable in loss record 184 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40C8B7C: apr_palloc (apr_pools.c:300) ==18894== by 0x404FC7F: switch_core_perform_alloc (switch_core_memory.c:442) ==18894== by 0x4067843: switch_loadable_module_create_interface (switch_loadable_module.c:1635) ==18894== by 0x1035CCB3: mod_commands_load (mod_commands.c:3565) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 8,192 bytes in 1 blocks are still reachable in loss record 185 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40C8710: apr_pool_create_ex (apr_pools.c:300) ==18894== by 0x804A15D: main (switch.c:676) ==18894== ==18894== ==18894== 8,192 bytes in 1 blocks are still reachable in loss record 186 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40C8710: apr_pool_create_ex (apr_pools.c:300) ==18894== by 0x40C9747: apr_initialize (start.c:58) ==18894== by 0x8049C7B: main (switch.c:596) ==18894== ==18894== ==18894== 8,192 bytes in 1 blocks are still reachable in loss record 187 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40C8710: apr_pool_create_ex (apr_pools.c:300) ==18894== by 0x40C892E: apr_pool_initialize (apr_pools.c:527) ==18894== by 0x40C971E: apr_initialize (start.c:55) ==18894== by 0x8049C7B: main (switch.c:596) ==18894== ==18894== ==18894== 9,388 bytes in 1 blocks are still reachable in loss record 188 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x12867A2F: JS_Init (in /usr/local/lib/libjs.so.1.0.6) ==18894== by 0x1279F6FB: mod_spidermonkey_load (mod_spidermonkey.c:1046) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 16,384 bytes in 2 blocks are still reachable in loss record 189 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40C8710: apr_pool_create_ex (apr_pools.c:300) ==18894== by 0x40CDA36: apr_thread_create (thread.c:171) ==18894== by 0x403ECF8: switch_thread_create (switch_apr.c:631) ==18894== by 0x4062738: switch_core_launch_thread (switch_core.c:400) ==18894== by 0x406D4D6: switch_loadable_module_init (switch_loadable_module.c:121) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 21,733 bytes in 34 blocks are still reachable in loss record 190 of 193 ==18894== at 0x40046FF: calloc (vg_replace_malloc.c:279) ==18894== by 0x9C16CA: _dl_new_object (in /lib/ld-2.5.so) ==18894== by 0x9BD020: _dl_map_object_from_fd (in /lib/ld-2.5.so) ==18894== by 0x9BF46B: _dl_map_object (in /lib/ld-2.5.so) ==18894== by 0x9C8A30: dl_open_worker (in /lib/ld-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0x9C8411: _dl_open (in /lib/ld-2.5.so) ==18894== by 0xB1FC4C: dlopen_doit (in /lib/libdl-2.5.so) ==18894== by 0x9C4E25: _dl_catch_error (in /lib/ld-2.5.so) ==18894== by 0xB202CB: _dlerror_run (in /lib/libdl-2.5.so) ==18894== by 0xB1FB83: dlopen@@GLIBC_2.1 (in /lib/libdl-2.5.so) ==18894== by 0x40674AF: switch_dso_open (switch_dso.c:96) ==18894== ==18894== ==18894== 200,704 bytes in 1 blocks are still reachable in loss record 191 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40C8B7C: apr_palloc (apr_pools.c:300) ==18894== by 0x40BEA4A: apr_queue_create (apr_queue.c:129) ==18894== by 0x403E4AA: switch_queue_create (switch_apr.c:892) ==18894== by 0x7AC85B5: mod_sofia_load (mod_sofia.c:3262) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 200,704 bytes in 1 blocks are still reachable in loss record 192 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40C8B7C: apr_palloc (apr_pools.c:300) ==18894== by 0x40BEA4A: apr_queue_create (apr_queue.c:129) ==18894== by 0x403E4AA: switch_queue_create (switch_apr.c:892) ==18894== by 0x7AC8595: mod_sofia_load (mod_sofia.c:3261) ==18894== by 0x406C884: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== ==18894== 294,912 bytes in 36 blocks are still reachable in loss record 193 of 193 ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) ==18894== by 0x40C8710: apr_pool_create_ex (apr_pools.c:300) ==18894== by 0x40502E6: switch_core_perform_new_memory_pool (switch_core_memory.c:357) ==18894== by 0x406C7CD: switch_loadable_module_load_module_ex (switch_loadable_module.c:785) ==18894== by 0x406D22F: switch_loadable_module_init (switch_loadable_module.c:1174) ==18894== by 0x4065980: switch_core_init_and_modload (switch_core.c:1451) ==18894== by 0x804A3FF: main (switch.c:710) ==18894== ==18894== LEAK SUMMARY: ==18894== definitely lost: 1,191 bytes in 9 blocks. ==18894== indirectly lost: 1,556 bytes in 123 blocks. ==18894== possibly lost: 320 bytes in 2 blocks. ==18894== still reachable: 924,064 bytes in 472 blocks. ==18894== suppressed: 0 bytes in 0 blocks. From talk2ram at gmail.com Mon Jul 20 04:19:55 2009 From: talk2ram at gmail.com (ram) Date: Mon, 20 Jul 2009 16:49:55 +0530 Subject: [Freeswitch-users] Creating a new User Agent In-Reply-To: <1452e2980907192330o14904fb8ke0aec88fd6afde0a@mail.gmail.com> References: <1452e2980907192330o14904fb8ke0aec88fd6afde0a@mail.gmail.com> Message-ID: On Mon, Jul 20, 2009 at 12:00 PM, velusamy velu wrote: > Dear All, > I want to create a new User Agent like sip configurations in > Asterisk. I checked default user agents 1000 to 1001. But I have bit > confused the relationship between default user agents and sip_profiles. > > I need some help from you all for the following questions, > How to create new user agent ? > How to relate the new user agent with sip internal > profile ? > Hi have you looked at this link http://wiki.freeswitch.org/wiki/Getting_Started_Guide#What_SIP_Profiles_Do ram -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090720/3492ea21/attachment-0001.html From technical at ttnc.co.uk Mon Jul 20 04:22:53 2009 From: technical at ttnc.co.uk (TTNC - Adnan Barakat) Date: Mon, 20 Jul 2009 12:22:53 +0100 Subject: [Freeswitch-users] call fails when using leg_timeout Message-ID: <4A64538D.6070301@ttnc.co.uk> Hi, I seem to have come across a strange problem; Basically I'm trying to dial 3 destinations one after another, until the destination dialled is answered, and I only want the destination to ring for 20 seconds. If I do this from the console what I'm trying to achieve works fine; originate {leg_timeout=20,ignore_early_media=true}sofia/internal/123 at 1.2.3.4|sofia/internal/456 at 1.2.3.4|sofia/internal/789 at 1.2.3.4 &park() But if I do it from dialplan it doesn't; Doing this below dials the first destination for 20 seconds, then the second destination for 5 seconds or so, then the call terminates with ORIGINATOR_CANCEL Any ideas what might be causing this and any solutions would be appreciated. Many thanks Adnan From technical at ttnc.co.uk Mon Jul 20 04:47:46 2009 From: technical at ttnc.co.uk (TTNC - Adnan Barakat) Date: Mon, 20 Jul 2009 12:47:46 +0100 Subject: [Freeswitch-users] caller_id 0000000000 Message-ID: <4A645962.6080507@ttnc.co.uk> Hi We have a FreeSWITCH server receiving calls from a provider, we process the call (play greeting messages etc...) then pass the call out again to an end destination via the same provider. But if the caller is witholding their cli our provider send the call to us with the sip_from_user variable to 'nobody'. Is there anyway to remove the caller_id when we don't receive it rather than override it with 0000000000? or send the call in the same way we receive it ie. from 'nobody'? Thanks Adnan From tayeb.meftah at gmail.com Mon Jul 20 06:36:17 2009 From: tayeb.meftah at gmail.com (Meftah Tayeb) Date: Mon, 20 Jul 2009 13:36:17 +0000 Subject: [Freeswitch-users] Freeswitch Windows Issues In-Reply-To: References: <4A6443B8.5000106@gmail.com> Message-ID: <4A6472D1.6030603@gmail.com> hello baskar, i think Freeswitch ODBC Support is not enabled for Windows you must compile it with ODBC Support enabled thanks Baskar wrote: > *Hi Meftah Tayeb**,* > > *One more question in windows whether it is possible to connect the > ODBC connection through JavaScript in freeswitch. > > I have configured inbound in Linux it is working fine but same script > i tried in windows but i get this error. I have installed and > configured MYSQL connector ODBC in window. But when is call the script > i get this error. > > 2009-07-20 10:55:42.781250 [ERR] mod_spidermonkey.c:3393 Error loading > ODBC > 2009-07-20 10:55:42.781250 [ERR] Inbound1.js:5 ReferenceError: ODBC > is not defined > > Can some one assist me to resolve this above error > > Thanks in advance. > > -- > Thanks with 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 > > > > __________ Information from ESET NOD32 Antivirus, version of virus signature database 4260 (20090720) __________ > > The message was checked by ESET NOD32 Antivirus. > > http://www.eset.com > > __________ Information from ESET NOD32 Antivirus, version of virus signature database 4260 (20090720) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com From mike at jerris.com Mon Jul 20 06:58:49 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 20 Jul 2009 09:58:49 -0400 Subject: [Freeswitch-users] Freeswitch Windows Issues In-Reply-To: <4A6472D1.6030603@gmail.com> References: <4A6443B8.5000106@gmail.com> <4A6472D1.6030603@gmail.com> Message-ID: <5E024C30-C98D-4EC0-9E9F-F126FAED0BE3@jerris.com> This is not correct. Do you have the odbc mod for spidermonkey loaded? On Jul 20, 2009, at 9:36 AM, Meftah Tayeb wrote: > hello baskar, > i think Freeswitch ODBC Support is not enabled for Windows > you must compile it with ODBC Support enabled > thanks > Baskar wrote: >> *Hi Meftah Tayeb**,* >> >> *One more question in windows whether it is possible to connect the >> ODBC connection through JavaScript in freeswitch. >> >> I have configured inbound in Linux it is working fine but same script >> i tried in windows but i get this error. I have installed and >> configured MYSQL connector ODBC in window. But when is call the >> script >> i get this error. >> >> 2009-07-20 10:55:42.781250 [ERR] mod_spidermonkey.c:3393 Error >> loading >> ODBC >> 2009-07-20 10:55:42.781250 [ERR] Inbound1.js:5 ReferenceError: ODBC >> is not defined >> >> Can some one assist me to resolve this above error >> >> Thanks in advance. >> >> -- >> Thanks with 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 >> >> >> >> __________ Information from ESET NOD32 Antivirus, version of virus >> signature database 4260 (20090720) __________ >> >> The message was checked by ESET NOD32 Antivirus. >> >> http://www.eset.com >> >> > > > > __________ Information from ESET NOD32 Antivirus, version of virus > signature database 4260 (20090720) __________ > > The message was checked by ESET NOD32 Antivirus. > > http://www.eset.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 zolotov at altron.ua Mon Jul 20 03:12:07 2009 From: zolotov at altron.ua (Evgeniy Zolotov) Date: Mon, 20 Jul 2009 13:12:07 +0300 Subject: [Freeswitch-users] mod_fax problems Message-ID: <3210356507.20090720131207@altron.ua> Hello! I have some problems with receiving fax messages. My FreeSWITCH cann't decode some faxes. As I understood this problem has occurred with Panasonic and Panasonic-like fax-machines. I have recorded wav-file, which corresponds to the receiving fax (see fax.wav.tgz) and log-file of this session (fax.log.tgz). I tried decode this fax with the help of transferring them from one FreeSWITCH to another, but with no luck (I used for this SIP and openzap - result the same). Transferring from one FreeSWITCH to another with the help of 'txfax' works perfect. OS: Centos 5.2 FreeSWITCH: 1.0.2 (11053) Configuration file for mod_fax - original. So, please, help me to understand this situation. Thanks, Evgeniy. mailto:zolotov at altron.ua -------------- next part -------------- A non-text attachment was scrubbed... Name: fax.log.tgz Type: application/x-compressed Size: 2672 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090720/e164ae4e/attachment-0002.bin -------------- next part -------------- A non-text attachment was scrubbed... Name: fax.wav.tgz Type: application/x-compressed Size: 595910 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090720/e164ae4e/attachment-0003.bin From brian at freeswitch.org Mon Jul 20 07:23:22 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 20 Jul 2009 09:23:22 -0500 Subject: [Freeswitch-users] mod_fax problems In-Reply-To: <3210356507.20090720131207@altron.ua> References: <3210356507.20090720131207@altron.ua> Message-ID: I would highly recommend you update to SVN trunk. /b On Jul 20, 2009, at 5:12 AM, Evgeniy Zolotov wrote: > Hello! > > I have some problems with receiving fax messages. My FreeSWITCH > cann't decode some faxes. As I understood this problem has occurred > with Panasonic and Panasonic-like fax-machines. > I have recorded wav-file, which corresponds to the receiving fax > (see fax.wav.tgz) and log-file of this session (fax.log.tgz). > I tried decode this fax with the help of transferring them from one > FreeSWITCH to another, but with no luck (I used for this SIP and > openzap - result the same). > Transferring from one FreeSWITCH to another with the help of 'txfax' > works perfect. > > OS: Centos 5.2 > FreeSWITCH: 1.0.2 (11053) > > Configuration file for mod_fax - original. > > So, please, help me to understand this situation. > > Thanks, Evgeniy. From anthony.minessale at gmail.com Mon Jul 20 07:29:28 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 20 Jul 2009 09:29:28 -0500 Subject: [Freeswitch-users] mod_conference: behavior when the conference is torn down before 31seconds In-Reply-To: <4A61B993.6090402@gmx.net> References: <4A61B993.6090402@gmx.net> Message-ID: <191c3a030907200729l5783d877m3ef5159dba25624e@mail.gmail.com> the problem stems from the fact that you did an outcall to those other addresses. The 30 sec is the timeout waiting for those calls to establish. The outcome of those outbound calls must be determined before the conf will end. On Sat, Jul 18, 2009 at 7:01 AM, Peter P GMX wrote: > Hello Luis, > > are you using encrypted TLS instead on SIP on this phone? I experienced > a similar behaviour with 31 seocnds on TLS. > > Best regards > Peter > > Luis F Urrea schrieb: > > Hi all, > > > > I am experiencing a behavior that I cannot clearly understand. > > Basically I "autocall" a few phones into a conference with the > > sip_auto_answer set to true, as follows: > > > > > > > > > data="conference_auto_outcall_prefix={sip_auto_answer=true}"/> > > > data="user/305"/> > > > data="user/303"/> > > > data="user/201"/> > > > > > > > > > > > > The conference establishes just fine and everyone can hear just fine. > > > > The "strange" behavior comes when the person calling to ext 773 hangs > > up before 31 seconds have passed, the rest of the phones stay up until > > they reach second 31 into the "conference". > > > > I am using snom phones and I see the BYE message arriving at the > > phones exactly at second 31 after the call establishes. > > > > The conference itself however does not exist after the person calling > > 773 hangs up (doing conference list on CLI shows NO active conferences). > > > > If the conference stays up more than 31 seconds, then when the person > > calling 773 hangs up, the rest of the phones hang up immediately as > > desired. > > > > Here's the log for a "page" that lasts less than 31 seconds: > > > > http://pastebin.freeswitch.org/9773 > > > > Here's the log of the phone for a "page" that lasts less than 31 seconds: > > > > http://pastebin.freeswitch.org/9774 > > > > Your inout is appreciated. > > > > Regards, > > > > 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 > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference 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/20090720/77f5af7c/attachment-0001.html From anthony.minessale at gmail.com Mon Jul 20 07:58:57 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 20 Jul 2009 09:58:57 -0500 Subject: [Freeswitch-users] call fails when using leg_timeout In-Reply-To: <4A64538D.6070301@ttnc.co.uk> References: <4A64538D.6070301@ttnc.co.uk> Message-ID: <191c3a030907200758p24978436kcdbe5259a2e2d865@mail.gmail.com> in a bridge situation you also have to set originate_timeout to the total time you are willing to wait for the combined leg timeouts i.e. 60 for 20x20x20 On Mon, Jul 20, 2009 at 6:22 AM, TTNC - Adnan Barakat wrote: > Hi, > > I seem to have come across a strange problem; Basically I'm trying to > dial 3 destinations one after another, until the destination dialled is > answered, and I only want the destination to ring for 20 seconds. > > If I do this from the console what I'm trying to achieve works fine; > > originate > {leg_timeout=20,ignore_early_media=true}sofia/internal/123 at 1.2.3.4 > |sofia/internal/456 at 1.2.3.4|sofia/internal/789 at 1.2.3.4 > &park() > > But if I do it from dialplan it doesn't; Doing this below dials the > first destination for 20 seconds, then the second destination for 5 > seconds or so, then the call terminates with ORIGINATOR_CANCEL > > data="{leg_timeout=20,ignore_early_media=true}sofia/internal/123 at 1.2.3.4 > |sofia/internal/456 at 1.2.3.4|sofia/internal/789 at 1.2.3.4" > /> > > Any ideas what might be causing this and any solutions would be > appreciated. > > > Many thanks > > Adnan > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference 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/20090720/632fdf0c/attachment-0001.html From zolotov at altron.ua Mon Jul 20 08:06:25 2009 From: zolotov at altron.ua (Evgeniy Zolotov) Date: Mon, 20 Jul 2009 18:06:25 +0300 Subject: [Freeswitch-users] mod_fax problems In-Reply-To: References: <3210356507.20090720131207@altron.ua> Message-ID: <1445760366.20090720180625@altron.ua> Hello, Brian. I understand that this is the first thing that I should make. But we have many our own changes in modules and after update of all project we'll need to transfer them too. But this is the great peace of job. So probably update of mod_fax and spandsp would be enough? ?? ?????? 20 ???? 2009 ?., 17:23:22: > I would highly recommend you update to SVN trunk. > /b > On Jul 20, 2009, at 5:12 AM, Evgeniy Zolotov wrote: >> Hello! >> >> I have some problems with receiving fax messages. My FreeSWITCH >> cann't decode some faxes. As I understood this problem has occurred >> with Panasonic and Panasonic-like fax-machines. >> I have recorded wav-file, which corresponds to the receiving fax >> (see fax.wav.tgz) and log-file of this session (fax.log.tgz). >> I tried decode this fax with the help of transferring them from one >> FreeSWITCH to another, but with no luck (I used for this SIP and >> openzap - result the same). >> Transferring from one FreeSWITCH to another with the help of 'txfax' >> works perfect. >> >> OS: Centos 5.2 >> FreeSWITCH: 1.0.2 (11053) >> >> Configuration file for mod_fax - original. >> >> So, please, help me to understand this situation. >> >> 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 -- Thanks, Evgeniy. mailto:zolotov at altron.ua From brian at freeswitch.org Mon Jul 20 08:09:37 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 20 Jul 2009 10:09:37 -0500 Subject: [Freeswitch-users] mod_fax problems In-Reply-To: <1445760366.20090720180625@altron.ua> References: <3210356507.20090720131207@altron.ua> <1445760366.20090720180625@altron.ua> Message-ID: <0C683BCA-1F4D-42CA-8ADA-4F1A5B744EAB@freeswitch.org> If you have made changes to the included modules you'll need to report your changes to jira http://jira.freeswitch.org Thanks, /b On Jul 20, 2009, at 10:06 AM, Evgeniy Zolotov wrote: > Hello, Brian. > > I understand that this is the first thing that I should make. > But we have many our own changes in modules and after update of all > project we'll need to transfer them too. But this is the great peace > of job. > So probably update of mod_fax and spandsp would be enough? From zolotov at altron.ua Mon Jul 20 08:30:31 2009 From: zolotov at altron.ua (Evgeniy Zolotov) Date: Mon, 20 Jul 2009 18:30:31 +0300 Subject: [Freeswitch-users] mod_fax problems In-Reply-To: <0C683BCA-1F4D-42CA-8ADA-4F1A5B744EAB@freeswitch.org> References: <3210356507.20090720131207@altron.ua> <1445760366.20090720180625@altron.ua> <0C683BCA-1F4D-42CA-8ADA-4F1A5B744EAB@freeswitch.org> Message-ID: <151636129.20090720183031@altron.ua> Hello, Brian. Ok, I understand. So I'll try to update to SVN trunk and then post log, if any problems still exists. ?? ?????? 20 ???? 2009 ?., 18:09:37: > If you have made changes to the included modules you'll need to report > your changes to jira http://jira.freeswitch.org > Thanks, > /b > On Jul 20, 2009, at 10:06 AM, Evgeniy Zolotov wrote: >> Hello, Brian. >> >> I understand that this is the first thing that I should make. >> But we have many our own changes in modules and after update of all >> project we'll need to transfer them too. But this is the great peace >> of job. >> So probably update of mod_fax and spandsp would be enough? > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Thanks, Evgeniy. mailto:zolotov at altron.ua From zolotov at altron.ua Mon Jul 20 08:40:15 2009 From: zolotov at altron.ua (Evgeniy Zolotov) Date: Mon, 20 Jul 2009 18:40:15 +0300 Subject: [Freeswitch-users] mod_fax problems Message-ID: <1810546946.20090720184015@altron.ua> Hello, Brian. Ok, I understand. So I'll try to update to SVN trunk and then post log, if any problems still exists. ?? ?????? 20 ???? 2009 ?., 18:09:37: > If you have made changes to the included modules you'll need to report > your changes to jira http://jira.freeswitch.org > Thanks, > /b > On Jul 20, 2009, at 10:06 AM, Evgeniy Zolotov wrote: >> Hello, Brian. >> >> I understand that this is the first thing that I should make. >> But we have many our own changes in modules and after update of all >> project we'll need to transfer them too. But this is the great peace >> of job. >> So probably update of mod_fax and spandsp would be enough? > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Thanks, Evgeniy. mailto:zolotov at altron.ua From msc at freeswitch.org Mon Jul 20 09:40:51 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 20 Jul 2009 09:40:51 -0700 Subject: [Freeswitch-users] Creating a new User Agent In-Reply-To: <1452e2980907192330o14904fb8ke0aec88fd6afde0a@mail.gmail.com> References: <1452e2980907192330o14904fb8ke0aec88fd6afde0a@mail.gmail.com> Message-ID: <87f2f3b90907200940tdfa995bk6bf465c2f0880788@mail.gmail.com> On Sun, Jul 19, 2009 at 11:30 PM, velusamy velu wrote: > Dear All, > I want to create a new User Agent like sip configurations in > Asterisk. I checked default user agents 1000 to 1001. But I have bit > confused the relationship between default user agents and sip_profiles. > > I need some help from you all for the following questions, > How to create new user agent ? > How to relate the new user agent with sip internal > profile ? I believe you might be mixing terminology. There is a difference between a "user" and a "user agent." In FreeSWITCH, a SIP profile *is* a user agent. In the default configuration, in conf/directory/default/ there are 20 pre-configured users. 1000, 1001, etc. are simply SIP users that are ready for use. Point a SIP phone at your FreeSWITCH IP address and set the auth user name to "1000" and the password to "1234" and it should register just fine. To learn more about SIP profiles I do recommend the link that ram posted. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090720/99cfb45b/attachment-0001.html From technical at ttnc.co.uk Mon Jul 20 09:48:41 2009 From: technical at ttnc.co.uk (TTNC - Adnan Barakat) Date: Mon, 20 Jul 2009 17:48:41 +0100 Subject: [Freeswitch-users] call fails when using leg_timeout In-Reply-To: <191c3a030907200758p24978436kcdbe5259a2e2d865@mail.gmail.com> References: <4A64538D.6070301@ttnc.co.uk> <191c3a030907200758p24978436kcdbe5259a2e2d865@mail.gmail.com> Message-ID: <4A649FE9.3060104@ttnc.co.uk> Anthony Minessale wrote: > in a bridge situation you also have to set originate_timeout to the > total time you are willing to wait for the combined leg timeouts > > i.e. 60 for 20x20x20 I added originate_timeout=60 but now only the first destination rings for 30 seconds, then the call is still terminated with ORIGINATOR_CANCEL Any other ideas? Thanks Adnan From dujinfang at gmail.com Mon Jul 20 10:39:46 2009 From: dujinfang at gmail.com (Seven Du) Date: Tue, 21 Jul 2009 01:39:46 +0800 Subject: [Freeswitch-users] FreeSWITCH-Air, Another GUI? Message-ID: ALL, I know you guys more prefer a CLI version of softphone to a GUI version. But I still would like to share this: http://wiki.freeswitch.org/wiki/FsAir And feel free to give me feedbacks. I'v only played a few days of ActionScript, it's highly appreciated if someone can give me help on the following problems. 1) Bounce the icon on Mac on incoming call. 2) Show a small window on incoming call. 3) Is it possible to block read/write a socket? I want to implement sendRecv() in ActionScript like in the C version of ESL. Thanks. From zolotov at altron.ua Mon Jul 20 11:02:25 2009 From: zolotov at altron.ua (Evgeniy Zolotov) Date: Mon, 20 Jul 2009 21:02:25 +0300 Subject: [Freeswitch-users] mod_fax problems In-Reply-To: <1810546946.20090720184015@altron.ua> References: <1810546946.20090720184015@altron.ua> Message-ID: <1717692390.20090720210225@altron.ua> I've updated to current trunk: freeswitch at test> version FreeSWITCH Version 1.0.trunk (14299) But problem still exists. I've attached a new log. So if anybody has an idea it would be greatly appreciated. ?? ?????? 20 ???? 2009 ?., 18:40:15: > Hello, Brian. > Ok, I understand. So I'll try to update to SVN trunk and then post > log, if any problems still exists. > ?? ?????? 20 ???? 2009 ?., 18:09:37: >> If you have made changes to the included modules you'll need to report >> your changes to jira http://jira.freeswitch.org >> Thanks, >> /b >> On Jul 20, 2009, at 10:06 AM, Evgeniy Zolotov wrote: >>> Hello, Brian. >>> >>> I understand that this is the first thing that I should make. >>> But we have many our own changes in modules and after update of all >>> project we'll need to transfer them too. But this is the great peace >>> of job. >>> So probably update of mod_fax and spandsp would be enough? >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > Thanks, Evgeniy. > mailto:zolotov at altron.ua > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Thanks, Evgeniy. mailto:zolotov at altron.ua -------------- next part -------------- A non-text attachment was scrubbed... Name: fax.log.tgz Type: application/x-compressed Size: 2746 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090720/9f80c6a0/attachment-0001.bin From marketing at cluecon.com Mon Jul 20 10:58:59 2009 From: marketing at cluecon.com (Michael Collins) Date: Mon, 20 Jul 2009 10:58:59 -0700 Subject: [Freeswitch-users] ClueCon 2009 - Last Chance To Book At The Wyndham Message-ID: <87f2f3b90907201058i33465e82k7a3608436064fe06@mail.gmail.com> Greetings! We would like to let everyone know that the hotel where ClueCon 2009 is being held - The Wyndham Chicago - is completely booked up outside of the block of rooms that we have reserved. There are still rooms available. However, after noon CST tomorrow, July 21, those rooms will no longer be reserved for ClueCon attendees so you have until then to get your reservations in. After noon tomorrow we will no longer be able to guarantee that you have an chance to register with the hotel, so please act fast! See you in Chicago. Thanks for supporting ClueCon! -Michael http://www.cluecon.com 877.742.CLUE -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090720/536fa5be/attachment-0001.html From brian at freeswitch.org Mon Jul 20 11:29:05 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 20 Jul 2009 13:29:05 -0500 Subject: [Freeswitch-users] mod_fax problems In-Reply-To: <1717692390.20090720210225@altron.ua> References: <1810546946.20090720184015@altron.ua> <1717692390.20090720210225@altron.ua> Message-ID: <3B4D48C6-4556-4B76-98F7-3A0BC34C133F@freeswitch.org> Please follow the bug reporting guidelines here http://wiki.freeswitch.org/wiki/Reporting_Bugs If you can open a jira and attach all the info for the issue you're having. Thanks, Brian On Jul 20, 2009, at 1:02 PM, Evgeniy Zolotov wrote: > I've updated to current trunk: > > freeswitch at test> version > FreeSWITCH Version 1.0.trunk (14299) > > But problem still exists. I've attached a new log. > > So if anybody has an idea it would be greatly appreciated. From tayeb.meftah at gmail.com Mon Jul 20 11:45:43 2009 From: tayeb.meftah at gmail.com (Meftah Tayeb) Date: Mon, 20 Jul 2009 18:45:43 +0000 Subject: [Freeswitch-users] FreeSWITCH-Air, Another GUI? In-Reply-To: References: Message-ID: <4A64BB57.50206@gmail.com> hellok, i can help about Action Script by providing some EBooks (CHM/PDF) to you DelphiWorld in #Freeswitch thanks Seven Du wrote: > ALL, > > I know you guys more prefer a CLI version of softphone to a GUI > version. But I still would like to share this: > > http://wiki.freeswitch.org/wiki/FsAir > > And feel free to give me feedbacks. > > I'v only played a few days of ActionScript, it's highly appreciated if > someone can give me help on the following problems. > > > 1) Bounce the icon on Mac on incoming call. > 2) Show a small window on incoming call. > 3) Is it possible to block read/write a socket? I want to implement > sendRecv() in ActionScript like in the C version of ESL. > > 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 > > > __________ Information from ESET NOD32 Antivirus, version of virus signature database 4261 (20090720) __________ > > The message was checked by ESET NOD32 Antivirus. > > http://www.eset.com > > > __________ Information from ESET NOD32 Antivirus, version of virus signature database 4261 (20090720) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com From gabe at gundy.org Mon Jul 20 13:09:27 2009 From: gabe at gundy.org (Gabriel Gunderson) Date: Mon, 20 Jul 2009 14:09:27 -0600 Subject: [Freeswitch-users] Sonus - what's the latest? Message-ID: <903da5680907201309m45d84647ucc2041f65b655f7@mail.gmail.com> All, Anyone setup FS with Sonus lately? I've just ordered service for a customer and it should be hooked-up pretty soon. Since ordering, I was made aware the they're using Sonus on the back-end. Well, I've been reading up on it and I'm getting a bit worried :( Anyone know if this is still the state of things? http://wiki.freeswitch.org/wiki/RTP_Issues#Sonus Thanks, Gabe From kristian.kielhofner at gmail.com Mon Jul 20 14:06:14 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Mon, 20 Jul 2009 17:06:14 -0400 Subject: [Freeswitch-users] Sonus - what's the latest? In-Reply-To: <903da5680907201309m45d84647ucc2041f65b655f7@mail.gmail.com> References: <903da5680907201309m45d84647ucc2041f65b655f7@mail.gmail.com> Message-ID: <2d9149cd0907201406h3a550817ya2b29ec40661662@mail.gmail.com> On Mon, Jul 20, 2009 at 4:09 PM, Gabriel Gunderson wrote: > All, > > Anyone setup FS with Sonus lately? ?I've just ordered service for a > customer and it should be hooked-up pretty soon. ?Since ordering, I > was made aware the they're using Sonus on the back-end. ?Well, I've > been reading up on it and I'm getting a bit worried :( > > Anyone know if this is still the state of things? > http://wiki.freeswitch.org/wiki/RTP_Issues#Sonus > > Thanks, > Gabe > Gabe, Every day... I don't know of any new issues with Sonus as used by the carriers I deal with but YMMV. -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From msc at freeswitch.org Mon Jul 20 15:20:42 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 20 Jul 2009 15:20:42 -0700 Subject: [Freeswitch-users] call fails when using leg_timeout In-Reply-To: <4A649FE9.3060104@ttnc.co.uk> References: <4A64538D.6070301@ttnc.co.uk> <191c3a030907200758p24978436kcdbe5259a2e2d865@mail.gmail.com> <4A649FE9.3060104@ttnc.co.uk> Message-ID: <87f2f3b90907201520t210b6f6chd743fa6550377013@mail.gmail.com> On Mon, Jul 20, 2009 at 9:48 AM, TTNC - Adnan Barakat wrote: > Anthony Minessale wrote: > > in a bridge situation you also have to set originate_timeout to the > > total time you are willing to wait for the combined leg timeouts > > > > i.e. 60 for 20x20x20 > I added originate_timeout=60 but now only the first destination rings > for 30 seconds, then the call is still terminated with ORIGINATOR_CANCEL > > Any other ideas? > I just did a test with this syntax and it worked for me. Please try it and report back. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090720/a1523e2a/attachment-0001.html From steveu at coppice.org Mon Jul 20 17:22:48 2009 From: steveu at coppice.org (Steve Underwood) Date: Tue, 21 Jul 2009 08:22:48 +0800 Subject: [Freeswitch-users] mod_fax problems In-Reply-To: <3210356507.20090720131207@altron.ua> References: <3210356507.20090720131207@altron.ua> Message-ID: <4A650A58.3000607@coppice.org> Hi Evgeniy, Evgeniy Zolotov wrote: > Hello! > > I have some problems with receiving fax messages. My FreeSWITCH > cann't decode some faxes. As I understood this problem has occurred > with Panasonic and Panasonic-like fax-machines. > I have recorded wav-file, which corresponds to the receiving fax > (see fax.wav.tgz) and log-file of this session (fax.log.tgz). > I tried decode this fax with the help of transferring them from one > FreeSWITCH to another, but with no luck (I used for this SIP and > openzap - result the same). > Transferring from one FreeSWITCH to another with the help of 'txfax' > works perfect. > > OS: Centos 5.2 > FreeSWITCH: 1.0.2 (11053) > > Configuration file for mod_fax - original. > > So, please, help me to understand this situation. > > Thanks, Evgeniy The audio you posted doesn't seem to come from the call in your log file. The audio is very odd, with sections of a FAX called joined together in a strange order. The log file is very simple, and tells you exactly what went wrong. It says: Fax processing not successful - result (10) Far end is not able to transmit. and the reason it says this is because the DIS message from the far end says: .... ...0= Ready to transmit a fax document (polling): Not set Regards, Steve From yudha2008 at gmail.com Mon Jul 20 21:25:49 2009 From: yudha2008 at gmail.com (Baskar) Date: Tue, 21 Jul 2009 09:55:49 +0530 Subject: [Freeswitch-users] Freeswitch Windows Issues In-Reply-To: <5E024C30-C98D-4EC0-9E9F-F126FAED0BE3@jerris.com> References: <4A6443B8.5000106@gmail.com> <4A6472D1.6030603@gmail.com> <5E024C30-C98D-4EC0-9E9F-F126FAED0BE3@jerris.com> Message-ID: *Hi Michael Jerris, In linux if ODBC modules to be load in freeswitch i can load by this command make mod_spidermonkey_odbc-install make install But in windows how can i enable the modules for mod_spidermonkey? I checked whether mod spidermonkey is loaded by this command load mod_spidermonkey . freeswitch at Baskar>load mod_spidermonkey 2009-07-21 09:42:14.859375 [WARNING] switch_loadable_module.c:939 Module mod_spidermonkey Already **Loaded! ** API CALL [load(mod_spidermonkey)] output:-ERR [Module already loaded] **But output say is already loaded but still i get this error**: **2009-07-20 10:55:42.781250 [ERR] mod_spidermonkey.c:3393 Error loading ODBC 2009-07-20 10:55:42.781250 [ERR] Inbound1.js:5 ReferenceError: ODBC is not defined* * Can some one assist me to resolve this above error. Thanks for reply from Meftah Tayeb,** Michael Jerris*. * **-- Thanks with Regards, N.Baskar * -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090721/fabee107/attachment-0001.html From yehavi.bourvine at gmail.com Mon Jul 20 21:41:45 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Tue, 21 Jul 2009 07:41:45 +0300 Subject: [Freeswitch-users] BLF & Directed call pickup on Polycom phones Message-ID: Hello, I am trying to integrate Polycom phones with a FrewSwitch server, and have some problems with BLF and directed pickup. I've defined a buddy list with BW (buddy watch) on. One of the phone's line buttons (one fo the 3 ones on a Polycom-501 model) is assigned to this buddy and indeed shows its status. I would like to pickup a call to this buddy by pressing its button when his phone rings; however, this generates a second call to him... Using a SNOM phones this works ok. Has anyone managed to make it working with Polycom? Thanks! __Yehavi: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090721/1cb95215/attachment-0001.html From technical at ttnc.co.uk Mon Jul 20 21:55:13 2009 From: technical at ttnc.co.uk (TTNC - Adnan Barakat) Date: Tue, 21 Jul 2009 05:55:13 +0100 Subject: [Freeswitch-users] call fails when using leg_timeout In-Reply-To: <87f2f3b90907201520t210b6f6chd743fa6550377013@mail.gmail.com> References: <4A64538D.6070301@ttnc.co.uk> <191c3a030907200758p24978436kcdbe5259a2e2d865@mail.gmail.com> <4A649FE9.3060104@ttnc.co.uk> <87f2f3b90907201520t210b6f6chd743fa6550377013@mail.gmail.com> Message-ID: <4A654A31.4040508@ttnc.co.uk> Michael Collins wrote: > I just did a test with this syntax and it worked for me. Please try it > and report back. > > data="{ignore_early_media=true}[leg_timeout=20]sofia/internal/123 at 1.2.3.4 > |[leg_timeout=20]sofia/internal/456 at 1.2.3.4 > |[leg_timeout=20]sofia/internal/789 at 1.2.3.4 > "/> Thanks Michael, just tried that, but unfortunately still doesn't work. It seems that there is a hard limit somewhere of 30 sec, I've just tried different timeout values, and it's terminating everytime at 30 sec. Any other ideas? Thanks Adnan From mrene_lists at avgs.ca Mon Jul 20 21:59:35 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Tue, 21 Jul 2009 00:59:35 -0400 Subject: [Freeswitch-users] call fails when using leg_timeout In-Reply-To: <4A654A31.4040508@ttnc.co.uk> References: <4A64538D.6070301@ttnc.co.uk> <191c3a030907200758p24978436kcdbe5259a2e2d865@mail.gmail.com> <4A649FE9.3060104@ttnc.co.uk> <87f2f3b90907201520t210b6f6chd743fa6550377013@mail.gmail.com> <4A654A31.4040508@ttnc.co.uk> Message-ID: It is possible that your inbound carrier applies some timeout rules. Try the following before your bridge: Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca Am 21-Jul-09 um 12:55 AM schrieb TTNC - Adnan Barakat: > Michael Collins wrote: >> I just did a test with this syntax and it worked for me. Please try >> it >> and report back. >> >> > data="{ignore_early_media=true}[leg_timeout=20]sofia/internal/123 at 1.2.3.4 >> |[leg_timeout=20]sofia/internal/456 at 1.2.3.4 >> |[leg_timeout=20]sofia/internal/789 at 1.2.3.4 >> "/> > Thanks Michael, just tried that, but unfortunately still doesn't work. > It seems that there is a hard limit somewhere of 30 sec, I've just > tried > different timeout values, and it's terminating everytime at 30 sec. > > Any other ideas? > > > Thanks > > Adnan > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From rupa at rupa.com Mon Jul 20 22:04:03 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Tue, 21 Jul 2009 00:04:03 -0500 Subject: [Freeswitch-users] Freeswitch Windows Issues In-Reply-To: References: <4A6443B8.5000106@gmail.com> <4A6472D1.6030603@gmail.com> <5E024C30-C98D-4EC0-9E9F-F126FAED0BE3@jerris.com> Message-ID: mod_spidermonkey_odbc is a mod for spidermonkey, not freeswitch. So, you need to enable this mod in the spidermonkey config file, not freeswitch's. This is in conf/spidermonkey.conf.xml. The default has mod_spidermonkey_odbc commented out. On Mon, Jul 20, 2009 at 11:25 PM, Baskar wrote: > *Hi Michael Jerris, > > In linux if ODBC modules to be load in freeswitch i can load by this > command > > make mod_spidermonkey_odbc-install > make install > > But in windows how can i enable the modules for mod_spidermonkey? > > I checked whether mod spidermonkey is loaded by this command load > mod_spidermonkey . > > freeswitch at Baskar>load > mod_spidermonkey > 2009-07-21 09:42:14.859375 [WARNING] switch_loadable_module.c:939 Module > mod_spidermonkey Already **Loaded! ** > > API CALL [load(mod_spidermonkey)] > output:-ERR [Module already loaded] > > **But output say is already loaded but still i get this error**: > > **2009-07-20 10:55:42.781250 [ERR] mod_spidermonkey.c:3393 Error loading > ODBC > 2009-07-20 10:55:42.781250 [ERR] Inbound1.js:5 ReferenceError: ODBC is not > defined* > * > Can some one assist me to resolve this above error. > > Thanks for reply from Meftah Tayeb,** Michael Jerris*. > > * > **-- > Thanks with 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 > > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090721/5edaf5f5/attachment-0001.html From technical at ttnc.co.uk Mon Jul 20 23:06:17 2009 From: technical at ttnc.co.uk (TTNC - Adnan Barakat) Date: Tue, 21 Jul 2009 07:06:17 +0100 Subject: [Freeswitch-users] call fails when using leg_timeout In-Reply-To: References: <4A64538D.6070301@ttnc.co.uk> <191c3a030907200758p24978436kcdbe5259a2e2d865@mail.gmail.com> <4A649FE9.3060104@ttnc.co.uk> <87f2f3b90907201520t210b6f6chd743fa6550377013@mail.gmail.com> <4A654A31.4040508@ttnc.co.uk> Message-ID: <4A655AD9.4050006@ttnc.co.uk> Mathieu Rene wrote: > It is possible that your inbound carrier applies some timeout rules. > Try the following before your bridge: > > Not that I know of, I just tried with ring_ready, and it still doesn't work. Thanks, Adnan From technical at ttnc.co.uk Mon Jul 20 23:25:29 2009 From: technical at ttnc.co.uk (TTNC - Adnan Barakat) Date: Tue, 21 Jul 2009 07:25:29 +0100 Subject: [Freeswitch-users] call fails when using leg_timeout In-Reply-To: <4A655AD9.4050006@ttnc.co.uk> References: <4A64538D.6070301@ttnc.co.uk> <191c3a030907200758p24978436kcdbe5259a2e2d865@mail.gmail.com> <4A649FE9.3060104@ttnc.co.uk> <87f2f3b90907201520t210b6f6chd743fa6550377013@mail.gmail.com> <4A654A31.4040508@ttnc.co.uk> <4A655AD9.4050006@ttnc.co.uk> Message-ID: <4A655F59.6090604@ttnc.co.uk> TTNC - Adnan Barakat wrote: > Mathieu Rene wrote: >> It is possible that your inbound carrier applies some timeout rules. >> Try the following before your bridge: >> >> > Not that I know of, I just tried with ring_ready, and it still doesn't work. Sorry guys, turns out it's a timeout on the VoIP phone I'm using. Thanks Adnan From elihayun at gmail.com Mon Jul 20 23:51:54 2009 From: elihayun at gmail.com (Eli Hayun) Date: Tue, 21 Jul 2009 09:51:54 +0300 Subject: [Freeswitch-users] How to detect that a party is in a conversation? Message-ID: <4A65658A.10009@savion.huji.ac.il> I want to know if the party that I am calling to is in a middle of a conversation. I did not get a busy line because he had more then one line defined. I tried to set max_calls=1 with no luck. any suggestions? Thanx Eli From elihayun at gmail.com Tue Jul 21 01:12:25 2009 From: elihayun at gmail.com (Eli Hayun) Date: Tue, 21 Jul 2009 11:12:25 +0300 Subject: [Freeswitch-users] Getting error without continuation Message-ID: <4A657869.5010202@savion.huji.ac.il> Hi I set continue_on_fail=true but I keep getting error : 2009-07-21 11:00:53.284148 [INFO] mod_dialplan_xml.c:252 Processing phone-1->limit_exceeded in context default 2 instead of continue to the line after the "bridge" What am I doing wrong? Eli From tayeb.meftah at gmail.com Tue Jul 21 01:50:17 2009 From: tayeb.meftah at gmail.com (Meftah Tayeb) Date: Tue, 21 Jul 2009 08:50:17 +0000 Subject: [Freeswitch-users] Freeswitch Windows Issues In-Reply-To: References: <4A6443B8.5000106@gmail.com> <4A6472D1.6030603@gmail.com> <5E024C30-C98D-4EC0-9E9F-F126FAED0BE3@jerris.com> Message-ID: <4A658149.9070407@gmail.com> hello baskar please open: c:\program files\freeswitch\conf\autoload_configs\modules.conf.xml and try to see if the module mod_spidermonkey_odbc is loaded are you using the binary installer (MSI) or you are compiling it? i think if you use the .MSI file the odbc module is not installed and loaded, i don't see it in my modules.conf.xml thanks, Meftah Tayeb DelphiWorld@#Freeswitch Global Voice Communication Baskar wrote: > *Hi Michael Jerris, > > In linux if ODBC modules to be load in freeswitch i can load by this > command > > make mod_spidermonkey_odbc-install > make install > > But in windows how can i enable the modules for mod_spidermonkey? > > I checked whether mod spidermonkey is loaded by this command load > mod_spidermonkey . > > freeswitch at Baskar>load > mod_spidermonkey > > 2009-07-21 09:42:14.859375 [WARNING] switch_loadable_module.c:939 > Module mod_spidermonkey Already > **Loaded! ** > API CALL [load(mod_spidermonkey)] > output:-ERR [Module already loaded] > > **But output say is already loaded but still i get this error**: > > **2009-07-20 10:55:42.781250 [ERR] mod_spidermonkey.c:3393 Error > loading ODBC > 2009-07-20 10:55:42.781250 [ERR] Inbound1.js:5 ReferenceError: ODBC > is not defined* > * > Can some one assist me to resolve this above error. > > Thanks for reply from Meftah Tayeb,** Michael Jerris*. > * > **-- > Thanks with 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 > > > > __________ Information from ESET NOD32 Antivirus, version of virus signature database 4262 (20090720) __________ > > The message was checked by ESET NOD32 Antivirus. > > http://www.eset.com > > __________ Information from ESET NOD32 Antivirus, version of virus signature database 4262 (20090720) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090721/776af1a2/attachment-0001.html From tayeb.meftah at gmail.com Tue Jul 21 02:00:31 2009 From: tayeb.meftah at gmail.com (Meftah Tayeb) Date: Tue, 21 Jul 2009 09:00:31 +0000 Subject: [Freeswitch-users] Freeswitch Windows Issues In-Reply-To: References: <4A6443B8.5000106@gmail.com> <4A6472D1.6030603@gmail.com> <5E024C30-C98D-4EC0-9E9F-F126FAED0BE3@jerris.com> Message-ID: <4A6583AF.5020504@gmail.com> hello baskar, sory, i see it now in spidermonkey.conf.xml line: encommant it and restart your freeswitch thanks Baskar wrote: > *Hi Michael Jerris, > > In linux if ODBC modules to be load in freeswitch i can load by this > command > > make mod_spidermonkey_odbc-install > make install > > But in windows how can i enable the modules for mod_spidermonkey? > > I checked whether mod spidermonkey is loaded by this command load > mod_spidermonkey . > > freeswitch at Baskar>load > mod_spidermonkey > > 2009-07-21 09:42:14.859375 [WARNING] switch_loadable_module.c:939 > Module mod_spidermonkey Already > **Loaded! ** > API CALL [load(mod_spidermonkey)] > output:-ERR [Module already loaded] > > **But output say is already loaded but still i get this error**: > > **2009-07-20 10:55:42.781250 [ERR] mod_spidermonkey.c:3393 Error > loading ODBC > 2009-07-20 10:55:42.781250 [ERR] Inbound1.js:5 ReferenceError: ODBC > is not defined* > * > Can some one assist me to resolve this above error. > > Thanks for reply from Meftah Tayeb,** Michael Jerris*. > * > **-- > Thanks with 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 > > > > __________ Information from ESET NOD32 Antivirus, version of virus signature database 4262 (20090720) __________ > > The message was checked by ESET NOD32 Antivirus. > > http://www.eset.com > > __________ Information from ESET NOD32 Antivirus, version of virus signature database 4262 (20090720) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090721/835f86ce/attachment-0001.html From helmut.kuper at ewetel.de Tue Jul 21 02:52:26 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Tue, 21 Jul 2009 11:52:26 +0200 Subject: [Freeswitch-users] Q931 TE State Timer In-Reply-To: <4A643C7D.7010209@ewetel.de> References: <4A3266E5.2000702@ewetel.de> <4A328E8E.6030607@freeswitch.org> <4A35F0F0.50406@ewetel.de> <4A643C7D.7010209@ewetel.de> Message-ID: <4A658FDA.8080908@ewetel.de> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, I fixed the bug that openzap doesn't send a RELEASE on incomming calls for not registered but existing extensions. Day 3 running the new Q931-TE stack is still successful :) It seems I have still no open calls left after 1000+ of total calls ... regards helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) iD8DBQFKZY/a4tZeNddg3dwRAjYRAJ9fV1BJiJyyyrG2A5BWEUbhJZQ1bgCgkazo G8lw40GrRvfynDmYDZrLUU0= =pZ2r -----END PGP SIGNATURE----- From elihayun at gmail.com Tue Jul 21 03:49:20 2009 From: elihayun at gmail.com (Eli Hayun) Date: Tue, 21 Jul 2009 13:49:20 +0300 Subject: [Freeswitch-users] How to initiate a call without dialing Message-ID: <4A659D30.5020600@savion.huji.ac.il> Is there is a way to initiate a call without making any dial manually? Suppose that I want to initiate a call let say every day at 17:00, is there is a way to do it? Eli From yehavi.bourvine at gmail.com Tue Jul 21 05:22:09 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Tue, 21 Jul 2009 15:22:09 +0300 Subject: [Freeswitch-users] BLF & Directed call pickup on Polycom phones In-Reply-To: References: Message-ID: After playing a little with SNOM phones I see that for doing BLF the SNOM subscribes for number XXXX (the real number), but when I want to pickup a ringing extension it dials **XXXX which is catched by FreeSwitch and handled by the pickup code (probably the intercept function). I would like to mimic this on Polycom phones. Thus, I want the phone to subscribe for *ZXXXX and catch the *Z prefix: - If it is a subscribe command, then strip *Z and subscribe to it. - If this is INVITE and the destination is ringing - strip *Z and and call intercept. - If this is INVITE and the destination is free - ring it. I know roughly how to do the last two items, but how can I catch the SUBSCRIBE, modify the destination number and then call the actual function? Thanks! __Yehavi: 2009/7/21 Yehavi Bourvine > Hello, > > I am trying to integrate Polycom phones with a FrewSwitch server, and > have some problems with BLF and directed pickup. > > I've defined a buddy list with BW (buddy watch) on. One of > the phone's line buttons (one fo the 3 ones on a Polycom-501 model) is > assigned to this buddy and indeed shows its status. I would like to pickup a > call to this buddy by pressing its button when his phone rings; however, > this generates a second call to him... > > Using a SNOM phones this works ok. Has anyone managed to make it working > with Polycom? > > Thanks! __Yehavi: > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090721/3e5ac6bb/attachment-0001.html From javieraristizabal at gmail.com Tue Jul 21 06:19:09 2009 From: javieraristizabal at gmail.com (=?ISO-8859-1?Q?Javier_Aristiz=E1bal?=) Date: Tue, 21 Jul 2009 08:19:09 -0500 Subject: [Freeswitch-users] BLF & Directed call pickup on Polycom phones In-Reply-To: References: Message-ID: I dunno about BLF on Polycom phones. But for the call pickup, check the phone dialplan if permit **XXXX. Javier -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090721/6ed7f48a/attachment-0001.html From regs at kinetix.gr Tue Jul 21 06:29:15 2009 From: regs at kinetix.gr (Apostolos Pantsiopoulos) Date: Tue, 21 Jul 2009 16:29:15 +0300 Subject: [Freeswitch-users] Possible memory leak - need a second opinion In-Reply-To: <4A644EA6.70209@kinetix.gr> References: <4A644EA6.70209@kinetix.gr> Message-ID: <4A65C2AB.5020001@kinetix.gr> Since nobody replied I am posting it to JIRA. Apostolos Pantsiopoulos wrote: > Hi I noticed that after a day of relatively moderate traffic (about 400 > simultaneous channels average) the memory used by FS reached 1.3 GB of > RAM. I tried to trace the leak (if any) with valgrind and got that : > > ==18894== 572 bytes in 1 blocks are definitely lost in loss record 148 > of 193 > ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) > ==18894== by 0x7AFDE93: sofia_glue_do_invite (sofia_glue.c:1599) > ==18894== by 0x7ACB191: sofia_on_init (mod_sofia.c:102) > ==18894== by 0x405B124: switch_core_session_run > (switch_core_state_machine.c:480) > ==18894== by 0x4058204: switch_core_session_thread > (switch_core_session.c:1064) > ==18894== by 0x40CD6C5: dummy_worker (thread.c:138) > ==18894== by 0xB5346A: start_thread (in /lib/libpthread-2.5.so) > ==18894== by 0xAAADBD: clone (in /lib/libc-2.5.so) > > > So, I assummed that this happens for every call. I tried testing it > again by placing two calls before shutting down FS, but it only came up > once. I wanted to get a second opinion before posting this to JIRA as an > issue. > > I used revision 14269 of the SVN. I am attaching the valgrind output as > well. > > I also noticed that only one of my CPU cores gets really busy when > dealing with moderate traffic. From what I read in the mailing list > users are encouraged to use 64bit multi core servers for FS because it > scales up better. But this is not what I am seeing in practice. Could > the single threaded architecture of libsofia be the cause of that behavior? > > > > > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 ------------------------------------------- From yudha2008 at gmail.com Tue Jul 21 06:47:59 2009 From: yudha2008 at gmail.com (Baskar) Date: Tue, 21 Jul 2009 19:17:59 +0530 Subject: [Freeswitch-users] Freeswitch Windows Issues In-Reply-To: <4A6583AF.5020504@gmail.com> References: <4A6443B8.5000106@gmail.com> <4A6472D1.6030603@gmail.com> <5E024C30-C98D-4EC0-9E9F-F126FAED0BE3@jerris.com> <4A6583AF.5020504@gmail.com> Message-ID: *Hi, Problem has been resolved. **Thanks for reply from Meftah Tayeb,** Michael Jerris, Rupa Schomaker and freeswitch-users. -- Thanks with Regards, N.Baskar * -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090721/fcae9e60/attachment-0001.html From brian at freeswitch.org Tue Jul 21 06:55:15 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 21 Jul 2009 08:55:15 -0500 Subject: [Freeswitch-users] Possible memory leak - need a second opinion In-Reply-To: <4A65C2AB.5020001@kinetix.gr> References: <4A644EA6.70209@kinetix.gr> <4A65C2AB.5020001@kinetix.gr> Message-ID: <0238ECAC-FC98-42AA-8824-B90FB6232520@freeswitch.org> If you're gonna open a jira we'll need a sipp scenario file that will reproduce the issue along with sip traces that can show what is going on. /b On Jul 21, 2009, at 8:29 AM, Apostolos Pantsiopoulos wrote: > Since nobody replied I am posting it to JIRA. > > Apostolos Pantsiopoulos wrote: >> Hi I noticed that after a day of relatively moderate traffic (about >> 400 >> simultaneous channels average) the memory used by FS reached 1.3 GB >> of >> RAM. I tried to trace the leak (if any) with valgrind and got that : >> >> ==18894== 572 bytes in 1 blocks are definitely lost in loss record >> 148 >> of 193 >> ==18894== at 0x40053C0: malloc (vg_replace_malloc.c:149) >> ==18894== by 0x7AFDE93: sofia_glue_do_invite (sofia_glue.c:1599) >> ==18894== by 0x7ACB191: sofia_on_init (mod_sofia.c:102) >> ==18894== by 0x405B124: switch_core_session_run >> (switch_core_state_machine.c:480) >> ==18894== by 0x4058204: switch_core_session_thread >> (switch_core_session.c:1064) >> ==18894== by 0x40CD6C5: dummy_worker (thread.c:138) >> ==18894== by 0xB5346A: start_thread (in /lib/libpthread-2.5.so) >> ==18894== by 0xAAADBD: clone (in /lib/libc-2.5.so) >> >> >> So, I assummed that this happens for every call. I tried testing it >> again by placing two calls before shutting down FS, but it only >> came up >> once. I wanted to get a second opinion before posting this to JIRA >> as an >> issue. >> >> I used revision 14269 of the SVN. I am attaching the valgrind >> output as >> well. >> >> I also noticed that only one of my CPU cores gets really busy when >> dealing with moderate traffic. From what I read in the mailing list >> users are encouraged to use 64bit multi core servers for FS because >> it >> scales up better. But this is not what I am seeing in practice. Could >> the single threaded architecture of libsofia be the cause of that >> behavior? >> >> >> >> >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 From intralanman at freeswitch.org Tue Jul 21 06:56:33 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Tue, 21 Jul 2009 09:56:33 -0400 Subject: [Freeswitch-users] How to initiate a call without dialing In-Reply-To: <4A659D30.5020600@savion.huji.ac.il> References: <4A659D30.5020600@savion.huji.ac.il> Message-ID: <4A65C911.50707@freeswitch.org> Eli Hayun wrote: > Is there is a way to initiate a call without making any dial manually? > i think the api command "originate" is what you're looking for -Ray From helmut.kuper at ewetel.de Tue Jul 21 07:01:54 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Tue, 21 Jul 2009 16:01:54 +0200 Subject: [Freeswitch-users] Question: Capturing VoiceData received from Sagnoma E1 card Message-ID: <4A65CA52.90002@ewetel.de> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, For outgoing calls I'm hunting the cause for missing some 100ms of voice data send from remote right after pickup the remote phone (e.g. initial "Hello?" sound like "o?" or even nothing) On FreeSwitch server I captured the VoIP data to the called VoIP-Phone on the sofia interface. Using wireshark it also shows that the voice data from remote is missed. Using Mobil phones or ISDN phones calling the same remote party there is never a bit missed. This problem occurs rare - once or twice per day and per local voip phone, but it's quite anoying. So is there a way to capture the correspondig ISDN voice data FS receives before it is transmitted via RTP or just droped? I want to c whether FS drops the early RTP packets or whether FS never got the data from ISDN. Sofia Profile is using The dialplan portion is: Any ideas to refine my debugging? regard helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) iD8DBQFKZcpS4tZeNddg3dwRAnVDAKCxXXkdbf0RKeeSMFYucCIno3tA9gCfUzbD 148BfuKavTtBoJNScRQDmSk= =JbtY -----END PGP SIGNATURE----- From rupa at rupa.com Tue Jul 21 07:13:34 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Tue, 21 Jul 2009 09:13:34 -0500 Subject: [Freeswitch-users] Getting error without continuation In-Reply-To: <4A657869.5010202@savion.huji.ac.il> References: <4A657869.5010202@savion.huji.ac.il> Message-ID: Didn't you just set max_calls=1? Maybe takes that out so you can handle more than 1 call.... On Tue, Jul 21, 2009 at 3:12 AM, Eli Hayun wrote: > Hi > I set continue_on_fail=true but I keep getting error : > 2009-07-21 11:00:53.284148 [INFO] mod_dialplan_xml.c:252 Processing > phone-1->limit_exceeded in context default > 2 > instead of continue to the line after the "bridge" > What am I doing wrong? > > Eli > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090721/01cfdaf1/attachment-0001.html From vkozak at abisoft.spb.ru Tue Jul 21 01:46:03 2009 From: vkozak at abisoft.spb.ru (Kozak Vladimir) Date: Tue, 21 Jul 2009 12:46:03 +0400 Subject: [Freeswitch-users] Bridge command with domain, using FS outbound connection API not work [NORMAL_TEMPORARY_FAILURE]. Message-ID: Hello, Problem with bridge command using FS outbound connection API. Configuration: 1. X-lite: auth/disp/user name: 1001; domain: rantipin.starpoundtech.net; Register with domain and receive incomming calls: true; proxy: 172.26.200.250:5060 2 eyeBeam: auth/disp/user name: 1000; domain: master.agent.rantipin.starpoundtech.net; Register with domain and receive incomming calls: true; proxy: 172.26.200.250:5060 I start nc -v -l 127.0.0.1 -p 8084 Make call from eyeBeam, recive connected event. I execute commands at nc: 1. sendmsg call-command: execute execute-app-name: bridge execute-app-arg: sofia/internal/1001 at 172.26.200.250 - OK, work. 2. sendmsg call-command: execute execute-app-name: bridge execute-app-arg: sofia/internal/1001 at rantipin.starpoundtech.net - Not work. nc response: Content-Type: command/reply Reply-Text: +OK Content-Type: text/disconnect-notice Controlled-Session-UUID: a7e94e62-2c4b-48ba-81c2-019300b420d6 Content-Disposition: disconnect FS log: 2009-07-21 19:12:24 [DEBUG] switch_core_session.c:706 switch_core_session_queue_private_event() Send signal sofia/internal/1000 at master.agent.rantipin.starpoundtech.net [BREAK] 2009-07-21 19:12:24 [DEBUG] switch_ivr.c:540 switch_ivr_parse_event() sofia/internal/1000 at master.agent.rantipin.starpoundtech.net Command Execute bridge(sofia/internal/1001 at rantipin.starpoundtech.net) 2009-07-21 19:12:24 [NOTICE] switch_channel.c:567 switch_channel_set_name() New Channel sofia/internal/1001 at rantipin.starpoundtech.net [84861d3f-88a5-49c7-82cc-52f7580c44a8] 2009-07-21 19:12:24 [DEBUG] mod_sofia.c:2535 sofia_outgoing_channel() (sofia/internal/1001 at rantipin.starpoundtech.net) State Change CS_NEW -> CS_INIT 2009-07-21 19:12:24 [DEBUG] switch_core_session.c:820 switch_core_session_signal_state_change() Send signal sofia/internal/1001 at rantipin.starpoundtech.net [BREAK] 2009-07-21 19:12:24 [DEBUG] switch_core_state_machine.c:383 switch_core_session_run() (sofia/internal/1001 at rantipin.starpoundtech.net) Running State Change CS_INIT 2009-07-21 19:12:24 [DEBUG] switch_core_state_machine.c:454 switch_core_session_run() (sofia/internal/1001 at rantipin.starpoundtech.net) State INIT 2009-07-21 19:12:24 [DEBUG] mod_sofia.c:83 sofia_on_init() sofia/internal/1001 at rantipin.starpoundtech.net SOFIA INIT 2009-07-21 19:12:24 [DEBUG] mod_sofia.c:111 sofia_on_init() (sofia/internal/1001 at rantipin.starpoundtech.net) State Change CS_INIT -> CS_ROUTING 2009-07-21 19:12:24 [DEBUG] switch_core_session.c:820 switch_core_session_signal_state_change() Send signal sofia/internal/1001 at rantipin.starpoundtech.net [BREAK] 2009-07-21 19:12:24 [DEBUG] switch_core_state_machine.c:454 switch_core_session_run() (sofia/internal/1001 at rantipin.starpoundtech.net) State INIT going to sleep 2009-07-21 19:12:24 [DEBUG] switch_core_state_machine.c:383 switch_core_session_run() (sofia/internal/1001 at rantipin.starpoundtech.net) Running State Change CS_ROUTING 2009-07-21 19:12:24 [DEBUG] switch_core_state_machine.c:457 switch_core_session_run() (sofia/internal/1001 at rantipin.starpoundtech.net) State ROUTING 2009-07-21 19:12:24 [DEBUG] mod_sofia.c:130 sofia_on_routing() sofia/internal/1001 at rantipin.starpoundtech.net SOFIA ROUTING 2009-07-21 19:12:24 [DEBUG] switch_ivr_originate.c:63 originate_on_routing() (sofia/internal/1001 at rantipin.starpoundtech.net) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2009-07-21 19:12:24 [DEBUG] switch_core_session.c:820 switch_core_session_signal_state_change() Send signal sofia/internal/1001 at rantipin.starpoundtech.net [BREAK] 2009-07-21 19:12:24 [DEBUG] switch_core_state_machine.c:457 switch_core_session_run() (sofia/internal/1001 at rantipin.starpoundtech.net) State ROUTING going to sleep 2009-07-21 19:12:24 [DEBUG] switch_core_state_machine.c:383 switch_core_session_run() (sofia/internal/1001 at rantipin.starpoundtech.net) Running State Change CS_CONSUME_MEDIA 2009-07-21 19:12:24 [DEBUG] switch_core_state_machine.c:476 switch_core_session_run() (sofia/internal/1001 at rantipin.starpoundtech.net) State CONSUME_MEDIA 2009-07-21 19:12:24 [DEBUG] sofia.c:2728 sofia_handle_sip_i_state() Channel sofia/internal/1001 at rantipin.starpoundtech.net entering state [calling] 2009-07-21 19:12:24 [DEBUG] sofia.c:2728 sofia_handle_sip_i_state() Channel sofia/internal/1001 at rantipin.starpoundtech.net entering state [terminated] 2009-07-21 19:12:24 [NOTICE] sofia.c:3279 sofia_handle_sip_i_state() Hangup sofia/internal/1001 at rantipin.starpoundtech.net [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE] 2009-07-21 19:12:24 [DEBUG] switch_channel.c:1566 switch_channel_perform_hangup() Send signal sofia/internal/1001 at rantipin.starpoundtech.net [KILL] 2009-07-21 19:12:24 [DEBUG] switch_core_session.c:820 switch_core_session_signal_state_change() Send signal sofia/internal/1001 at rantipin.starpoundtech.net [BREAK] 2009-07-21 19:12:24 [DEBUG] switch_core_state_machine.c:476 switch_core_session_run() (sofia/internal/1001 at rantipin.starpoundtech.net) State CONSUME_MEDIA going to sleep 2009-07-21 19:12:24 [DEBUG] switch_core_state_machine.c:383 switch_core_session_run() (sofia/internal/1001 at rantipin.starpoundtech.net) Running State Change CS_HANGUP 2009-07-21 19:12:24 [DEBUG] switch_core_state_machine.c:414 switch_core_session_run() (sofia/internal/1001 at rantipin.starpoundtech.net) State HANGUP 2009-07-21 19:12:24 [DEBUG] mod_sofia.c:253 sofia_on_hangup() sofia/internal/1001 at rantipin.starpoundtech.net Overriding SIP cause 503 with 503 from the other leg 2009-07-21 19:12:24 [DEBUG] mod_sofia.c:287 sofia_on_hangup() Channel sofia/internal/1001 at rantipin.starpoundtech.net hanging up, cause: NORMAL_TEMPORARY_FAILURE 2009-07-21 19:12:24 [DEBUG] switch_core_state_machine.c:46 switch_core_standard_on_hangup() sofia/internal/1001 at rantipin.starpoundtech.net Standard HANGUP, cause: NORMAL_TEMPORARY_FAILURE 2009-07-21 19:12:24 [DEBUG] switch_core_state_machine.c:414 switch_core_session_run() (sofia/internal/1001 at rantipin.starpoundtech.net) State HANGUP going to sleep 2009-07-21 19:12:24 [DEBUG] switch_core_session.c:952 switch_core_session_thread() Session 2 (sofia/internal/1001 at rantipin.starpoundtech.net) Locked, Waiting on external entities 2009-07-21 19:12:24 [DEBUG] switch_ivr_originate.c:2014 switch_ivr_originate() Originate Resulted in Error Cause: 41 [NORMAL_TEMPORARY_FAILURE] 2009-07-21 19:12:24 [INFO] mod_dptools.c:1998 audio_bridge_function() Originate Failed. Cause: NORMAL_TEMPORARY_FAILURE 2009-07-21 19:12:24 [NOTICE] mod_dptools.c:2030 audio_bridge_function() Hangup sofia/internal/1000 at master.agent.rantipin.starpoundtech.net [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE] 2009-07-21 19:12:24 [DEBUG] switch_channel.c:1566 switch_channel_perform_hangup() Send signal sofia/internal/1000 at master.agent.rantipin.starpoundtech.net [KILL] 2009-07-21 19:12:24 [DEBUG] switch_core_session.c:820 switch_core_session_signal_state_change() Send signal sofia/internal/1000 at master.agent.rantipin.starpoundtech.net [BREAK] 2009-07-21 19:12:24 [DEBUG] switch_core_state_machine.c:464 switch_core_session_run() (sofia/internal/1000 at master.agent.rantipin.starpoundtech.net) State EXECUTE going to sleep 2009-07-21 19:12:24 [DEBUG] switch_core_state_machine.c:383 switch_core_session_run() (sofia/internal/1000 at master.agent.rantipin.starpoundtech.net) Running State Change CS_HANGUP 2009-07-21 19:12:24 [DEBUG] switch_core_state_machine.c:414 switch_core_session_run() (sofia/internal/1000 at master.agent.rantipin.starpoundtech.net) State HANGUP 2009-07-21 19:12:24 [DEBUG] mod_sofia.c:253 sofia_on_hangup() sofia/internal/1000 at master.agent.rantipin.starpoundtech.net Overriding SIP cause 503 with 503 from the other leg 2009-07-21 19:12:24 [DEBUG] mod_sofia.c:287 sofia_on_hangup() Channel sofia/internal/1000 at master.agent.rantipin.starpoundtech.net hanging up, cause: NORMAL_TEMPORARY_FAILURE 2009-07-21 19:12:24 [DEBUG] mod_sofia.c:361 sofia_on_hangup() Responding to INVITE with: 503 2009-07-21 19:12:24 [DEBUG] switch_core_state_machine.c:46 switch_core_standard_on_hangup() sofia/internal/1000 at master.agent.rantipin.starpoundtech.net Standard HANGUP, cause: NORMAL_TEMPORARY_FAILURE 2009-07-21 19:12:24 [DEBUG] switch_core_state_machine.c:414 switch_core_session_run() (sofia/internal/1000 at master.agent.rantipin.starpoundtech.net) State HANGUP going to sleep 2009-07-21 19:12:24 [DEBUG] switch_core_session.c:952 switch_core_session_thread() Session 1 (sofia/internal/1000 at master.agent.rantipin.starpoundtech.net) Locked, Waiting on external entities 2009-07-21 19:12:24 [NOTICE] switch_core_session.c:970 switch_core_session_thread() Session 2 (sofia/internal/1001 at rantipin.starpoundtech.net) Ended 2009-07-21 19:12:24 [NOTICE] switch_core_session.c:972 switch_core_session_thread() Close Channel sofia/internal/1001 at rantipin.starpoundtech.net [CS_HANGUP] 2009-07-21 19:12:24 [DEBUG] mod_event_socket.c:1979 listener_run() Session complete, waiting for children 2009-07-21 19:12:24 [DEBUG] mod_event_socket.c:2012 listener_run() Connection Closed 2009-07-21 19:12:24 [NOTICE] switch_core_session.c:970 switch_core_session_thread() Session 1 (sofia/internal/1000 at master.agent.rantipin.starpoundtech.net) Ended 2009-07-21 19:12:24 [NOTICE] switch_core_session.c:972 switch_core_session_thread() Close Channel sofia/internal/1000 at master.agent.rantipin.starpoundtech.net [CS_HANGUP] What can be not correct? Thank you. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090721/60728d9c/attachment-0001.html From brian at freeswitch.org Tue Jul 21 08:03:54 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 21 Jul 2009 10:03:54 -0500 Subject: [Freeswitch-users] Bridge command with domain, using FS outbound connection API not work [NORMAL_TEMPORARY_FAILURE]. In-Reply-To: References: Message-ID: <74240E91-6797-4D03-80D4-A5F9EBF77853@freeswitch.org> http://wiki.freeswitch.org/wiki/Dialplan_XML#SIP-Specific_Dialstrings "Dialing A Registered User" is what you should refer to /b On Jul 21, 2009, at 3:46 AM, Kozak Vladimir wrote: > Hello, > > Problem with bridge command using FS outbound connection API. > Configuration: > 1. X-lite: auth/disp/user name: 1001; domain: > rantipin.starpoundtech.net; Register with domain and receive > incomming calls: true; proxy: 172.26.200.250:5060 > 2 eyeBeam: auth/disp/user name: 1000; domain: > master.agent.rantipin.starpoundtech.net; Register with domain and > receive incomming calls: true; proxy: 172.26.200.250:5060 > > I start nc -v -l 127.0.0.1 -p 8084 > Make call from eyeBeam, recive connected event. > > I execute commands at nc: > 1. > sendmsg > call-command: execute > execute-app-name: bridge > execute-app-arg: sofia/internal/1001 at 172.26.200.250 > > - OK, work. > > 2. > sendmsg > call-command: execute > execute-app-name: bridge > execute-app-arg: sofia/internal/1001 at rantipin.starpoundtech.net > > - Not work. > > nc response: > > Content-Type: command/reply > Reply-Text: +OK > > Content-Type: text/disconnect-notice > Controlled-Session-UUID: a7e94e62-2c4b-48ba-81c2-019300b420d6 > Content-Disposition: disconnect > > FS log: > > 2009-07-21 19:12:24 [DEBUG] switch_core_session.c:706 > switch_core_session_queue_private_event() Send signal sofia/internal/1000 at master.agent.rantipin.starpoundtech.net > [BREAK] > 2009-07-21 19:12:24 [DEBUG] switch_ivr.c:540 > switch_ivr_parse_event() sofia/internal/1000 at master.agent.rantipin.starpoundtech.net > Command Execute bridge(sofia/internal/ > 1001 at rantipin.starpoundtech.net) > 2009-07-21 19:12:24 [NOTICE] switch_channel.c:567 > switch_channel_set_name() New Channel sofia/internal/1001 at rantipin.starpoundtech.net > [84861d3f-88a5-49c7-82cc-52f7580c44a8] > 2009-07-21 19:12:24 [DEBUG] mod_sofia.c:2535 > sofia_outgoing_channel() (sofia/internal/1001 at rantipin.starpoundtech.net > ) State Change CS_NEW -> CS_INIT > 2009-07-21 19:12:24 [DEBUG] switch_core_session.c:820 > switch_core_session_signal_state_change() Send signal sofia/internal/1001 at rantipin.starpoundtech.net > [BREAK] > 2009-07-21 19:12:24 [DEBUG] switch_core_state_machine.c:383 > switch_core_session_run() (sofia/internal/1001 at rantipin.starpoundtech.net > ) Running State Change CS_INIT > 2009-07-21 19:12:24 [DEBUG] switch_core_state_machine.c:454 > switch_core_session_run() (sofia/internal/1001 at rantipin.starpoundtech.net > ) State INIT > 2009-07-21 19:12:24 [DEBUG] mod_sofia.c:83 sofia_on_init() sofia/internal/1001 at rantipin.starpoundtech.net > SOFIA INIT > 2009-07-21 19:12:24 [DEBUG] mod_sofia.c:111 sofia_on_init() (sofia/internal/1001 at rantipin.starpoundtech.net > ) State Change CS_INIT -> CS_ROUTING > 2009-07-21 19:12:24 [DEBUG] switch_core_session.c:820 > switch_core_session_signal_state_change() Send signal sofia/internal/1001 at rantipin.starpoundtech.net > [BREAK] > 2009-07-21 19:12:24 [DEBUG] switch_core_state_machine.c:454 > switch_core_session_run() (sofia/internal/1001 at rantipin.starpoundtech.net > ) State INIT going to sleep > 2009-07-21 19:12:24 [DEBUG] switch_core_state_machine.c:383 > switch_core_session_run() (sofia/internal/1001 at rantipin.starpoundtech.net > ) Running State Change CS_ROUTING > 2009-07-21 19:12:24 [DEBUG] switch_core_state_machine.c:457 > switch_core_session_run() (sofia/internal/1001 at rantipin.starpoundtech.net > ) State ROUTING > 2009-07-21 19:12:24 [DEBUG] mod_sofia.c:130 sofia_on_routing() sofia/internal/1001 at rantipin.starpoundtech.net > SOFIA ROUTING > 2009-07-21 19:12:24 [DEBUG] switch_ivr_originate.c:63 > originate_on_routing() (sofia/internal/ > 1001 at rantipin.starpoundtech.net) State Change CS_ROUTING -> > CS_CONSUME_MEDIA > 2009-07-21 19:12:24 [DEBUG] switch_core_session.c:820 > switch_core_session_signal_state_change() Send signal sofia/internal/1001 at rantipin.starpoundtech.net > [BREAK] > 2009-07-21 19:12:24 [DEBUG] switch_core_state_machine.c:457 > switch_core_session_run() (sofia/internal/1001 at rantipin.starpoundtech.net > ) State ROUTING going to sleep > 2009-07-21 19:12:24 [DEBUG] switch_core_state_machine.c:383 > switch_core_session_run() (sofia/internal/1001 at rantipin.starpoundtech.net > ) Running State Change CS_CONSUME_MEDIA > 2009-07-21 19:12:24 [DEBUG] switch_core_state_machine.c:476 > switch_core_session_run() (sofia/internal/1001 at rantipin.starpoundtech.net > ) State CONSUME_MEDIA > 2009-07-21 19:12:24 [DEBUG] sofia.c:2728 sofia_handle_sip_i_state() > Channel sofia/internal/1001 at rantipin.starpoundtech.net entering > state [calling] > 2009-07-21 19:12:24 [DEBUG] sofia.c:2728 sofia_handle_sip_i_state() > Channel sofia/internal/1001 at rantipin.starpoundtech.net entering > state [terminated] > 2009-07-21 19:12:24 [NOTICE] sofia.c:3279 sofia_handle_sip_i_state() > Hangup sofia/internal/1001 at rantipin.starpoundtech.net > [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE] > 2009-07-21 19:12:24 [DEBUG] switch_channel.c:1566 > switch_channel_perform_hangup() Send signal sofia/internal/1001 at rantipin.starpoundtech.net > [KILL] > 2009-07-21 19:12:24 [DEBUG] switch_core_session.c:820 > switch_core_session_signal_state_change() Send signal sofia/internal/1001 at rantipin.starpoundtech.net > [BREAK] > 2009-07-21 19:12:24 [DEBUG] switch_core_state_machine.c:476 > switch_core_session_run() (sofia/internal/1001 at rantipin.starpoundtech.net > ) State CONSUME_MEDIA going to sleep > 2009-07-21 19:12:24 [DEBUG] switch_core_state_machine.c:383 > switch_core_session_run() (sofia/internal/1001 at rantipin.starpoundtech.net > ) Running State Change CS_HANGUP > 2009-07-21 19:12:24 [DEBUG] switch_core_state_machine.c:414 > switch_core_session_run() (sofia/internal/1001 at rantipin.starpoundtech.net > ) State HANGUP > 2009-07-21 19:12:24 [DEBUG] mod_sofia.c:253 sofia_on_hangup() sofia/internal/1001 at rantipin.starpoundtech.net > Overriding SIP cause 503 with 503 from the other leg > 2009-07-21 19:12:24 [DEBUG] mod_sofia.c:287 sofia_on_hangup() > Channel sofia/internal/1001 at rantipin.starpoundtech.net hanging up, > cause: NORMAL_TEMPORARY_FAILURE > 2009-07-21 19:12:24 [DEBUG] switch_core_state_machine.c:46 > switch_core_standard_on_hangup() sofia/internal/1001 at rantipin.starpoundtech.net > Standard HANGUP, cause: NORMAL_TEMPORARY_FAILURE > 2009-07-21 19:12:24 [DEBUG] switch_core_state_machine.c:414 > switch_core_session_run() (sofia/internal/1001 at rantipin.starpoundtech.net > ) State HANGUP going to sleep > 2009-07-21 19:12:24 [DEBUG] switch_core_session.c:952 > switch_core_session_thread() Session 2 (sofia/internal/1001 at rantipin.starpoundtech.net > ) Locked, Waiting on external entities > 2009-07-21 19:12:24 [DEBUG] switch_ivr_originate.c:2014 > switch_ivr_originate() Originate Resulted in Error Cause: 41 > [NORMAL_TEMPORARY_FAILURE] > 2009-07-21 19:12:24 [INFO] mod_dptools.c:1998 > audio_bridge_function() Originate Failed. Cause: > NORMAL_TEMPORARY_FAILURE > 2009-07-21 19:12:24 [NOTICE] mod_dptools.c:2030 > audio_bridge_function() Hangup sofia/internal/1000 at master.agent.rantipin.starpoundtech.net > [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE] > 2009-07-21 19:12:24 [DEBUG] switch_channel.c:1566 > switch_channel_perform_hangup() Send signal sofia/internal/1000 at master.agent.rantipin.starpoundtech.net > [KILL] > 2009-07-21 19:12:24 [DEBUG] switch_core_session.c:820 > switch_core_session_signal_state_change() Send signal sofia/internal/1000 at master.agent.rantipin.starpoundtech.net > [BREAK] > 2009-07-21 19:12:24 [DEBUG] switch_core_state_machine.c:464 > switch_core_session_run() (sofia/internal/1000 at master.agent.rantipin.starpoundtech.net > ) State EXECUTE going to sleep > 2009-07-21 19:12:24 [DEBUG] switch_core_state_machine.c:383 > switch_core_session_run() (sofia/internal/1000 at master.agent.rantipin.starpoundtech.net > ) Running State Change CS_HANGUP > 2009-07-21 19:12:24 [DEBUG] switch_core_state_machine.c:414 > switch_core_session_run() (sofia/internal/1000 at master.agent.rantipin.starpoundtech.net > ) State HANGUP > 2009-07-21 19:12:24 [DEBUG] mod_sofia.c:253 sofia_on_hangup() sofia/internal/1000 at master.agent.rantipin.starpoundtech.net > Overriding SIP cause 503 with 503 from the other leg > 2009-07-21 19:12:24 [DEBUG] mod_sofia.c:287 sofia_on_hangup() > Channel sofia/internal/1000 at master.agent.rantipin.starpoundtech.net > hanging up, cause: NORMAL_TEMPORARY_FAILURE > 2009-07-21 19:12:24 [DEBUG] mod_sofia.c:361 sofia_on_hangup() > Responding to INVITE with: 503 > 2009-07-21 19:12:24 [DEBUG] switch_core_state_machine.c:46 > switch_core_standard_on_hangup() sofia/internal/1000 at master.agent.rantipin.starpoundtech.net > Standard HANGUP, cause: NORMAL_TEMPORARY_FAILURE > 2009-07-21 19:12:24 [DEBUG] switch_core_state_machine.c:414 > switch_core_session_run() (sofia/internal/1000 at master.agent.rantipin.starpoundtech.net > ) State HANGUP going to sleep > 2009-07-21 19:12:24 [DEBUG] switch_core_session.c:952 > switch_core_session_thread() Session 1 (sofia/internal/1000 at master.agent.rantipin.starpoundtech.net > ) Locked, Waiting on external entities > 2009-07-21 19:12:24 [NOTICE] switch_core_session.c:970 > switch_core_session_thread() Session 2 (sofia/internal/1001 at rantipin.starpoundtech.net > ) Ended > 2009-07-21 19:12:24 [NOTICE] switch_core_session.c:972 > switch_core_session_thread() Close Channel sofia/internal/1001 at rantipin.starpoundtech.net > [CS_HANGUP] > 2009-07-21 19:12:24 [DEBUG] mod_event_socket.c:1979 listener_run() > Session complete, waiting for children > 2009-07-21 19:12:24 [DEBUG] mod_event_socket.c:2012 listener_run() > Connection Closed > 2009-07-21 19:12:24 [NOTICE] switch_core_session.c:970 > switch_core_session_thread() Session 1 (sofia/internal/1000 at master.agent.rantipin.starpoundtech.net > ) Ended > 2009-07-21 19:12:24 [NOTICE] switch_core_session.c:972 > switch_core_session_thread() Close Channel sofia/internal/1000 at master.agent.rantipin.starpoundtech.net > [CS_HANGUP] > > What can be not correct? > 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 msc at freeswitch.org Tue Jul 21 08:29:42 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 21 Jul 2009 08:29:42 -0700 Subject: [Freeswitch-users] Q931 TE State Timer In-Reply-To: <4A658FDA.8080908@ewetel.de> References: <4A3266E5.2000702@ewetel.de> <4A328E8E.6030607@freeswitch.org> <4A35F0F0.50406@ewetel.de> <4A643C7D.7010209@ewetel.de> <4A658FDA.8080908@ewetel.de> Message-ID: <87f2f3b90907210829o38f18c23xf893f88fd4690ba5@mail.gmail.com> Helmut, This is wonderful news. Please keep up the good work. -MC On Tue, Jul 21, 2009 at 2:52 AM, Helmut Kuper wrote: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > Hello, > > I fixed the bug that openzap doesn't send a RELEASE on incomming calls > for not registered but existing extensions. > > Day 3 running the new Q931-TE stack is still successful :) It seems I > have still no open calls left after 1000+ of total calls ... > > regards > helmut > > -----BEGIN PGP SIGNATURE----- > Version: GnuPG v1.4.7 (MingW32) > > iD8DBQFKZY/a4tZeNddg3dwRAjYRAJ9fV1BJiJyyyrG2A5BWEUbhJZQ1bgCgkazo > G8lw40GrRvfynDmYDZrLUU0= > =pZ2r > -----END PGP SIGNATURE----- > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20090721/3959200c/attachment-0001.html From nicolas at medularis.com Tue Jul 21 08:35:00 2009 From: nicolas at medularis.com (Nicolas Brenner) Date: Tue, 21 Jul 2009 11:35:00 -0400 Subject: [Freeswitch-users] Best way to bridge 2 calls with LCR? Message-ID: <1b46b4e80907210835s6e516e2ft84f3d7329eee1535@mail.gmail.com> I would like to originate 2 calls from FS and then bridge them. There's no incoming call so I think there's no dialplan involved. What I'd like to do now is apply lcr rules to these calls. I've come up with 2 options so far: 1) call lcr through the socket twice (once for each phonenumber) and then originate the calls through the socket too 2) have a javascript file which runs the actions above, run the script through the socket with 'jsrun' How would you do it? For what I've read on the list, usually the recommended way is to stay away from javascript as much as possible because it is not as efficient as doing everything from the dialplan. Does this mean the first option is the best? or is there a "dialplan way" of doing it? Thank you very much for your help! Nicolas -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090721/793f2c18/attachment-0001.html From rupa at rupa.com Tue Jul 21 08:43:26 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Tue, 21 Jul 2009 10:43:26 -0500 Subject: [Freeswitch-users] Best way to bridge 2 calls with LCR? In-Reply-To: <1b46b4e80907210835s6e516e2ft84f3d7329eee1535@mail.gmail.com> References: <1b46b4e80907210835s6e516e2ft84f3d7329eee1535@mail.gmail.com> Message-ID: lcr api command doesn't really return a usable dialstring (it was originally done for debug purposes). I could add an "as xml" option if needed... Anyway, to do this from the dialplan: remember that originate's usage is: -USAGE |&() [] [] [] [] [] so, the first argument is the call url and the second would be an extension. so: 1) execute lcr for the first leg of the call 2) execute originate with: originate ${lcr_auto_route} extension extension just needs to match something in your dialplan. In extension, you'd do another lcr lookup and then bridge to that leg's ${lcr_auto_route} value. On Tue, Jul 21, 2009 at 10:35 AM, Nicolas Brenner wrote: > I would like to originate 2 calls from FS and then bridge them. There's no > incoming call so I think there's no dialplan involved. > What I'd like to do now is apply lcr rules to these calls. I've come up > with 2 options so far: > > 1) call lcr through the socket twice (once for each phonenumber) and then > originate the calls through the socket too > 2) have a javascript file which runs the actions above, run the script > through the socket with 'jsrun' > > How would you do it? > > For what I've read on the list, usually the recommended way is to stay away > from javascript as much as possible because it is not as efficient as doing > everything from the dialplan. Does this mean the first option is the best? > or is there a "dialplan way" of doing it? > > Thank you very much for your help! > > Nicolas > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090721/2ade0ec8/attachment-0001.html From marketing at cluecon.com Tue Jul 21 09:15:56 2009 From: marketing at cluecon.com (Michael Collins) Date: Tue, 21 Jul 2009 09:15:56 -0700 Subject: [Freeswitch-users] ClueCon 2009 - GREAT NEWS - Hotel and early registration extended through July 27! Message-ID: <87f2f3b90907210915i2c98d072nb784c9f1676d04b3@mail.gmail.com> Hello everyone! We are happy to announce that the ClueCon team has been able to extend the early bird registration price AND the hotel room reservations through Monday July 27th. This is important because, outside of our ClueCon block of rooms, the Wyndham is completely filled up. Please make your hotel reservations right away. If you haven't secured your ClueCon registration yet then please call 877.742.CLUE and we will get you set up with the early bird special of $499. Don't delay! ClueCon 2009 starts two weeks from today. It will be here before you know it. See you all in Chicago! -Michael http://www.cluecon.com 877.742.CLUE -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090721/5ba1b846/attachment-0001.html From nicolas at medularis.com Tue Jul 21 09:27:07 2009 From: nicolas at medularis.com (Nicolas Brenner) Date: Tue, 21 Jul 2009 12:27:07 -0400 Subject: [Freeswitch-users] Best way to bridge 2 calls with LCR? In-Reply-To: References: <1b46b4e80907210835s6e516e2ft84f3d7329eee1535@mail.gmail.com> Message-ID: <1b46b4e80907210927n4885b4c5g180a06e32bedc7c4@mail.gmail.com> That looks like a good way to go about it. How can I access channel variables through the socket using the api? I mean, how do I recover the value of ${lcr_auto_route}? I would need to add some other variables, like ignore_early_media=true and a uuid that 'links' the two calls so I can track it listening for events. Thanks! Nicolas On Tue, Jul 21, 2009 at 11:43 AM, Rupa Schomaker wrote: > lcr api command doesn't really return a usable dialstring (it was > originally done for debug purposes). I could add an "as xml" option if > needed... > > Anyway, to do this from the dialplan: > > remember that originate's usage is: > > -USAGE |&() [] > [] [] [] [] > > so, the first argument is the call url and the second would be an > extension. so: > > 1) execute lcr for the first leg of the call > 2) execute originate with: > > originate ${lcr_auto_route} extension > > extension just needs to match something in your dialplan. > > In extension, you'd do another lcr lookup and then bridge to that leg's > ${lcr_auto_route} value. > > > > On Tue, Jul 21, 2009 at 10:35 AM, Nicolas Brenner wrote: > >> I would like to originate 2 calls from FS and then bridge them. There's no >> incoming call so I think there's no dialplan involved. >> What I'd like to do now is apply lcr rules to these calls. I've come up >> with 2 options so far: >> >> 1) call lcr through the socket twice (once for each phonenumber) and then >> originate the calls through the socket too >> 2) have a javascript file which runs the actions above, run the script >> through the socket with 'jsrun' >> >> How would you do it? >> >> For what I've read on the list, usually the recommended way is to stay >> away from javascript as much as possible because it is not as efficient as >> doing everything from the dialplan. Does this mean the first option is the >> best? or is there a "dialplan way" of doing it? >> >> Thank you very much for your help! >> >> Nicolas >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > -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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090721/191199b9/attachment-0001.html From larclap at yahoo.com Tue Jul 21 09:40:06 2009 From: larclap at yahoo.com (Lars Zeb) Date: Tue, 21 Jul 2009 09:40:06 -0700 Subject: [Freeswitch-users] Inbound call routing help In-Reply-To: References: <004a01c9da6f$c1e40120$45ac0360$@com> Message-ID: <00a501ca0a21$e77e67e0$b67b37a0$@com> Brian, Pressing * no longer works. I don't know if my dialplan is causing the error or something in FreeSWITCH has changed. When pressing * during the greeting, the call immediately hangs up. Any ideas? http://pastebin.freeswitch.org/9803 Thanks, Lars From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Thursday, May 21, 2009 5:01 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Inbound call routing help Try pressing * during the greeting and make sure you have the vmain extension so you can login. /b On May 21, 2009, at 6:56 PM, Lars Zeb wrote: I want to setup a dialplan for a single DID. I would like it to go to a specific extension, and if not picked up in 15 seconds, go to voicemail. I have set this scenario up and it works. But I would also like this person to be able to call this DID from outside FS via a phone and be able to retrieve their voicemail. I've seen the example of how to pick up an extension's voicemail while inside FS by checking to see if the destination_number is the same as the caller_id_number, and if so, listen to voicemail, otherwise leave the message with voicemail. But I don't have a clue how to accomplish this from outside, other than dedicating another DID to solely retrieving voicemail from outside. Any ideas? Thanks, Lars ______________ Brian West brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090721/3bdc38dc/attachment-0001.html From rupa at rupa.com Tue Jul 21 09:54:22 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Tue, 21 Jul 2009 11:54:22 -0500 Subject: [Freeswitch-users] Best way to bridge 2 calls with LCR? In-Reply-To: <1b46b4e80907210927n4885b4c5g180a06e32bedc7c4@mail.gmail.com> References: <1b46b4e80907210835s6e516e2ft84f3d7329eee1535@mail.gmail.com> <1b46b4e80907210927n4885b4c5g180a06e32bedc7c4@mail.gmail.com> Message-ID: Ok, if you want to do it from the socket api, then I need to make a 'as xml' option to mod_lcr and give you lcr_auto_route as one of the nodes in the returned xml. Then you can do your own substitution in the originate line... In that case, you'd call lcr twice and do: originate lcr_auto_route1 &bridge(lcr_auto_route2) How soon do you need this? On Tue, Jul 21, 2009 at 11:27 AM, Nicolas Brenner wrote: > That looks like a good way to go about it. > > How can I access channel variables through the socket using the api? I > mean, how do I recover the value of ${lcr_auto_route}? I would need to add > some other variables, like ignore_early_media=true and a uuid that 'links' > the two calls so I can track it listening for events. > > Thanks! > > Nicolas > > > On Tue, Jul 21, 2009 at 11:43 AM, Rupa Schomaker wrote: > >> lcr api command doesn't really return a usable dialstring (it was >> originally done for debug purposes). I could add an "as xml" option if >> needed... >> >> Anyway, to do this from the dialplan: >> >> remember that originate's usage is: >> >> -USAGE |&() [] >> [] [] [] [] >> >> so, the first argument is the call url and the second would be an >> extension. so: >> >> 1) execute lcr for the first leg of the call >> 2) execute originate with: >> >> originate ${lcr_auto_route} extension >> >> extension just needs to match something in your dialplan. >> >> In extension, you'd do another lcr lookup and then bridge to that leg's >> ${lcr_auto_route} value. >> >> >> >> On Tue, Jul 21, 2009 at 10:35 AM, Nicolas Brenner wrote: >> >>> I would like to originate 2 calls from FS and then bridge them. There's >>> no incoming call so I think there's no dialplan involved. >>> What I'd like to do now is apply lcr rules to these calls. I've come up >>> with 2 options so far: >>> >>> 1) call lcr through the socket twice (once for each phonenumber) and then >>> originate the calls through the socket too >>> 2) have a javascript file which runs the actions above, run the script >>> through the socket with 'jsrun' >>> >>> How would you do it? >>> >>> For what I've read on the list, usually the recommended way is to stay >>> away from javascript as much as possible because it is not as efficient as >>> doing everything from the dialplan. Does this mean the first option is the >>> best? or is there a "dialplan way" of doing it? >>> >>> Thank you very much for your help! >>> >>> Nicolas >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> -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 >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090721/d731caa8/attachment-0001.html From nicolas at medularis.com Tue Jul 21 11:00:41 2009 From: nicolas at medularis.com (Nicolas Brenner) Date: Tue, 21 Jul 2009 14:00:41 -0400 Subject: [Freeswitch-users] Best way to bridge 2 calls with LCR? In-Reply-To: References: <1b46b4e80907210835s6e516e2ft84f3d7329eee1535@mail.gmail.com> <1b46b4e80907210927n4885b4c5g180a06e32bedc7c4@mail.gmail.com> Message-ID: <1b46b4e80907211100r4f9967fesb1b2c5fba4785a66@mail.gmail.com> Thank you very much for the offer, but I don't want to bother you with this. I can just parse the string returned by lcr and get the gateway, that's all I really need to create my complete originate command. I am using the socket api because it is easier for me to understand how to do it, nevertheless I'd really like to know how to do it with the dialplan. What I don't understand very well about using the dialplan for this, is how to do the first originate command (which I need to do using the socket api). What puzzles me is that according to the originate syntax, I need to use an extension or call an application, yet for the first call I would have to use a dummy extension as I only need to hit the dialplan section that calls lcr once to originate the first call with an extension that hits the section of the dialplan where lcr gets called again and the calls get bridged. I'm thinking something like this: 1) call originate from socket api to hit dialplan section that does all the work (this originate command is what I don't understand, is there another way of "hitting the dialplan" besides calling originate?) 2) hit dialplan section which calls lcr for first number and bridges to an extension 3) the extension calls lcr fir the second number and originates the second call On steps 2 and 3 I could just use set data to set the additional variables I need. The first step is what troubles me. Thank you! Nicolas On Tue, Jul 21, 2009 at 12:54 PM, Rupa Schomaker wrote: > Ok, if you want to do it from the socket api, then I need to make a 'as > xml' option to mod_lcr and give you lcr_auto_route as one of the nodes in > the returned xml. Then you can do your own substitution in the originate > line... In that case, you'd call lcr twice and do: > > originate lcr_auto_route1 &bridge(lcr_auto_route2) > > How soon do you need this? > > > On Tue, Jul 21, 2009 at 11:27 AM, Nicolas Brenner wrote: > >> That looks like a good way to go about it. >> >> How can I access channel variables through the socket using the api? I >> mean, how do I recover the value of ${lcr_auto_route}? I would need to add >> some other variables, like ignore_early_media=true and a uuid that 'links' >> the two calls so I can track it listening for events. >> >> Thanks! >> >> Nicolas >> >> >> On Tue, Jul 21, 2009 at 11:43 AM, Rupa Schomaker wrote: >> >>> lcr api command doesn't really return a usable dialstring (it was >>> originally done for debug purposes). I could add an "as xml" option if >>> needed... >>> >>> Anyway, to do this from the dialplan: >>> >>> remember that originate's usage is: >>> >>> -USAGE |&() [] >>> [] [] [] [] >>> >>> so, the first argument is the call url and the second would be an >>> extension. so: >>> >>> 1) execute lcr for the first leg of the call >>> 2) execute originate with: >>> >>> originate ${lcr_auto_route} extension >>> >>> extension just needs to match something in your dialplan. >>> >>> In extension, you'd do another lcr lookup and then bridge to that leg's >>> ${lcr_auto_route} value. >>> >>> >>> >>> On Tue, Jul 21, 2009 at 10:35 AM, Nicolas Brenner >> > wrote: >>> >>>> I would like to originate 2 calls from FS and then bridge them. There's >>>> no incoming call so I think there's no dialplan involved. >>>> What I'd like to do now is apply lcr rules to these calls. I've come up >>>> with 2 options so far: >>>> >>>> 1) call lcr through the socket twice (once for each phonenumber) and >>>> then originate the calls through the socket too >>>> 2) have a javascript file which runs the actions above, run the script >>>> through the socket with 'jsrun' >>>> >>>> How would you do it? >>>> >>>> For what I've read on the list, usually the recommended way is to stay >>>> away from javascript as much as possible because it is not as efficient as >>>> doing everything from the dialplan. Does this mean the first option is the >>>> best? or is there a "dialplan way" of doing it? >>>> >>>> Thank you very much for your help! >>>> >>>> Nicolas >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> -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 >>> >>> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > -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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090721/93726fc6/attachment-0001.html From rupa at rupa.com Tue Jul 21 11:21:00 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Tue, 21 Jul 2009 13:21:00 -0500 Subject: [Freeswitch-users] Best way to bridge 2 calls with LCR? In-Reply-To: <1b46b4e80907211100r4f9967fesb1b2c5fba4785a66@mail.gmail.com> References: <1b46b4e80907210835s6e516e2ft84f3d7329eee1535@mail.gmail.com> <1b46b4e80907210927n4885b4c5g180a06e32bedc7c4@mail.gmail.com> <1b46b4e80907211100r4f9967fesb1b2c5fba4785a66@mail.gmail.com> Message-ID: Well, the "as xml" is something I've been meaning to do, so I'm gonna get that checked in today sometime anyway. If you want to do any programmatic processing of the lcr data, the as xml is the way to go rather than parsing the strings. As for originate + lcr.... You can use the loopback endpoint and do it all in the dialplan: originate loopback/firstnumber secondnumber This will hit your dialplan with firstnumber first which you can lcr route. Then when that call establishes, it'll hit the dialplan with the second number which will also be routed through lcr. Is that more what you are looking for? This way all the 'routing' logic can be done via the dialplan. On Tue, Jul 21, 2009 at 1:00 PM, Nicolas Brenner wrote: > Thank you very much for the offer, but I don't want to bother you with > this. > > I can just parse the string returned by lcr and get the gateway, that's all > I really need to create my complete originate command. > > I am using the socket api because it is easier for me to understand how to > do it, nevertheless I'd really like to know how to do it with the dialplan. > > What I don't understand very well about using the dialplan for this, is how > to do the first originate command (which I need to do using the socket api). > What puzzles me is that according to the originate syntax, I need to use an > extension or call an application, yet for the first call I would have to use > a dummy extension as I only need to hit the dialplan section that calls lcr > once to originate the first call with an extension that hits the section of > the dialplan where lcr gets called again and the calls get bridged. > > I'm thinking something like this: > > 1) call originate from socket api to hit dialplan section that does all the > work (this originate command is what I don't understand, is there another > way of "hitting the dialplan" besides calling originate?) > > 2) hit dialplan section which calls lcr for first number and bridges to an > extension > > 3) the extension calls lcr fir the second number and originates the second > call > > On steps 2 and 3 I could just use set data to set the additional variables > I need. The first step is what troubles me. > > > Thank you! > > > Nicolas > > > On Tue, Jul 21, 2009 at 12:54 PM, Rupa Schomaker wrote: > >> Ok, if you want to do it from the socket api, then I need to make a 'as >> xml' option to mod_lcr and give you lcr_auto_route as one of the nodes in >> the returned xml. Then you can do your own substitution in the originate >> line... In that case, you'd call lcr twice and do: >> >> originate lcr_auto_route1 &bridge(lcr_auto_route2) >> >> How soon do you need this? >> >> >> On Tue, Jul 21, 2009 at 11:27 AM, Nicolas Brenner wrote: >> >>> That looks like a good way to go about it. >>> >>> How can I access channel variables through the socket using the api? I >>> mean, how do I recover the value of ${lcr_auto_route}? I would need to add >>> some other variables, like ignore_early_media=true and a uuid that 'links' >>> the two calls so I can track it listening for events. >>> >>> Thanks! >>> >>> Nicolas >>> >>> >>> On Tue, Jul 21, 2009 at 11:43 AM, Rupa Schomaker wrote: >>> >>>> lcr api command doesn't really return a usable dialstring (it was >>>> originally done for debug purposes). I could add an "as xml" option if >>>> needed... >>>> >>>> Anyway, to do this from the dialplan: >>>> >>>> remember that originate's usage is: >>>> >>>> -USAGE |&() [] >>>> [] [] [] [] >>>> >>>> so, the first argument is the call url and the second would be an >>>> extension. so: >>>> >>>> 1) execute lcr for the first leg of the call >>>> 2) execute originate with: >>>> >>>> originate ${lcr_auto_route} extension >>>> >>>> extension just needs to match something in your dialplan. >>>> >>>> In extension, you'd do another lcr lookup and then bridge to that leg's >>>> ${lcr_auto_route} value. >>>> >>>> >>>> >>>> On Tue, Jul 21, 2009 at 10:35 AM, Nicolas Brenner < >>>> nicolas at medularis.com> wrote: >>>> >>>>> I would like to originate 2 calls from FS and then bridge them. There's >>>>> no incoming call so I think there's no dialplan involved. >>>>> What I'd like to do now is apply lcr rules to these calls. I've come up >>>>> with 2 options so far: >>>>> >>>>> 1) call lcr through the socket twice (once for each phonenumber) and >>>>> then originate the calls through the socket too >>>>> 2) have a javascript file which runs the actions above, run the script >>>>> through the socket with 'jsrun' >>>>> >>>>> How would you do it? >>>>> >>>>> For what I've read on the list, usually the recommended way is to stay >>>>> away from javascript as much as possible because it is not as efficient as >>>>> doing everything from the dialplan. Does this mean the first option is the >>>>> best? or is there a "dialplan way" of doing it? >>>>> >>>>> Thank you very much for your help! >>>>> >>>>> Nicolas >>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> -- >>>> -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 >>>> >>>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> -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 >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090721/96f270b0/attachment-0001.html From rupa at rupa.com Tue Jul 21 11:51:54 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Tue, 21 Jul 2009 13:51:54 -0500 Subject: [Freeswitch-users] Best way to bridge 2 calls with LCR? In-Reply-To: References: <1b46b4e80907210835s6e516e2ft84f3d7329eee1535@mail.gmail.com> <1b46b4e80907210927n4885b4c5g180a06e32bedc7c4@mail.gmail.com> <1b46b4e80907211100r4f9967fesb1b2c5fba4785a66@mail.gmail.com> Message-ID: Just a note that the "as xml" syntax has been added to current trunk. On Tue, Jul 21, 2009 at 1:21 PM, Rupa Schomaker wrote: > Well, the "as xml" is something I've been meaning to do, so I'm gonna get > that checked in today sometime anyway. If you want to do any programmatic > processing of the lcr data, the as xml is the way to go rather than parsing > the strings. > > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090721/3c03681d/attachment-0001.html From lon at kickasspixels.com Tue Jul 21 12:01:56 2009 From: lon at kickasspixels.com (Lon Baker) Date: Tue, 21 Jul 2009 12:01:56 -0700 Subject: [Freeswitch-users] Call confirm ivr Message-ID: <5d3e0dc60907211201t2be7ea8dq2b2931f854bdc455@mail.gmail.com> Hi there, I am putting together an ivr to allow the recipient of a call to accept, route to voice mail or eavesdrop on voicemail. The current path is to answer the inbound call, park it, using the bgapi call to the recipient and play the IVR. Basically: 1. Answer 2. Playback greeting 3. UUID_PARK 4. Set filter for BACKGROUND_JOB 5. BGAPI Originate to the recipient with customer variables for processing 6. Do new IVR for recipient and process their input to route call. I am not sure is this the right path. Is there a better way? Lon -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090721/f1981cba/attachment-0001.html From lzwierko at gmail.com Tue Jul 21 12:38:01 2009 From: lzwierko at gmail.com (=?UTF-8?B?xYF1a2FzeiBad2llcmtv?=) Date: Tue, 21 Jul 2009 21:38:01 +0200 Subject: [Freeswitch-users] Dial up from confernece issue In-Reply-To: <41A88E3D-C420-4A1F-8B80-098EA18F9544@freeswitch.org> References: <4A637FC5.6000301@gmail.com> <41A88E3D-C420-4A1F-8B80-098EA18F9544@freeswitch.org> Message-ID: <4A661919.8030500@gmail.com> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Brian, I've just updated to 14310 and it's the same. The thing seems that sofia module rejects the call in quite early stage, so there is no 302 answer from remote SIP peer (as no INVITE was sent). Again, I'm exercising a very simple scenario with default FS configuration (just downloaded from svn), so I don't really know what's wrong here... Perhaps there is a different way to attach a call to a existing conference? Perhaps I should just originate new call (with the 'originate' command), and when received, pass it it conference application with the conference-id of the conference that I want to attach it to? Does that make any sense? Thanks ? freeswitch at Zwierko-laptop> conference list API CALL [conference(list)] output: Conference 3001-192.168.0.1 (1 member) 3;sofia/internal/1000 at 192.168.0.1;25f125a5-3556-a449-85ec-e4001336f313;1000;1000;hear|speak|floor;0;0;300 freeswitch at Zwierko-laptop> conference 3001-192.168.0.1 bgdial sofia/default/1001 at 192.168.0.1 API CALL [conference(3001-192.168.0.1 bgdial sofia/default/1001 at 192.168.0.1)] output: OK freeswitch at Zwierko-laptop> 2009-07-19 13:19:44.451100 [NOTICE] switch_channel.c:602 New Channel sofia/internal/1001 at 192.168.0.1 [b9fada7f-9c1d-4949-af8a-a8220ce f9c5b] 2009-07-19 13:19:44.459100 [NOTICE] switch_channel.c:602 New Channel sofia/internal/48228882211 at 192.168.0.1 [4f6b26dd-a0cb-2846-ad17-5f517e60e2e7] 2009-07-19 13:19:44.488100 [INFO] mod_dialplan_xml.c:252 Processing TelkaSwitch->1001 in context public 2009-07-19 13:19:44.495100 [ERR] sofia.c:4174 Cannot Blind Transfer 1 Legged calls 2009-07-19 13:19:44.498100 [NOTICE] sofia.c:3775 Hangup sofia/internal/1001 at 192.168.0.1 [CS_CONSUME_MEDIA] [NO_USER_RESPONSE] 2009-07-19 13:19:44.501100 [ERR] mod_conference.c:4351 Cannot create outgoing channel, cause: NO_USER_RESPONSE 2009-07-19 13:19:44.512100 [NOTICE] switch_core_session.c:1085 Session 9 (sofia/internal/1001 at 192.168.0.1) Ended 2009-07-19 13:19:44.512100 [NOTICE] switch_core_session.c:1087 Close Channel sofia/internal/1001 at 192.168.0.1 [CS_DESTROY] 2009-07-19 13:19:44.628100 [NOTICE] switch_core_state_machine.c:179 Hangup sofia/internal/48228882211 at 192.168.0.1 [CS_EXECUTE] [NORMAL_CLEARING] 2009-07-19 13:19:44.708100 [NOTICE] switch_core_session.c:1085 Session 10 (sofia/internal/48228882211 at 192.168.0.1) Ended 2009-07-19 13:19:44.708100 [NOTICE] switch_core_session.c:1087 Close Channel sofia/internal/48228882211 at 192.168.0.1 [CS_DESTROY] Brian West wrote: > If a single leg call gets a 302 you can't really "transfer" it > anywhere... What SVN rev are you on? > > /b > > On Jul 19, 2009, at 3:19 PM, ?ukasz Zwierko wrote: > >> Hi, >> >> sorry if you're getting this again, I'm not sure if this mail got >> deliverd to the mail-list (I didn't get a copy...) >> >> Anyway, >> >> I want to use bgdial command to add a person to a already started >> conference (that is, call that person and when answered - add the >> channel to conference). >> >> The scenario is I have two sip clients registered in default context - >> 1000 and 1001. 1000 dials conference number (3001 in this case) and >> new >> conference is started. I want to dial out to second using bgdial, >> unfortunately mod_sofia drops the call with 'Cannot Blind Transfer 1 >> Legged calls' message. >> >> Should I use the bgdial command differently? Or perhaps I should do >> this >> totally differently? Logs attached below. >> >> Thanks for any help, >> >> Lukasz > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.9 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iQEcBAEBAgAGBQJKZhkZAAoJED7LBosr0F2uE0wIAILk4StkVFGr4QVNsn7dob3d C1UBQnHOPezxlRmyT/lZjeN0Ddw+LZdvC5/Z14V8qjItsar2BDxT65AtVdryaKZq 9wlaEpGCoE377YGKM/k+hi8FYzvTkL1/Oz7aFGW/wpe2gbxKk1YWFSeU13iGpsZV 2byaY0qLdsGrs3CL3XMs69tKHmnnPcdM5p6xSYlOpKeE8/jUNJ+W7cOo0CcmVFf8 Mybwlhq7S7g6cKOD3WqgmBzMJi0pZRBgdz6x6uinAGmiSmTJIWO6+8BNjSIN373U OS7ivn8Gu4Tub50NBhkjhIEM3Kf+2JLQBRkwT0Mr4heIle9ZFe5UWbMFy0g8GbE= =xG/h -----END PGP SIGNATURE----- From brian at freeswitch.org Tue Jul 21 12:42:07 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 21 Jul 2009 14:42:07 -0500 Subject: [Freeswitch-users] Dial up from confernece issue In-Reply-To: <4A661919.8030500@gmail.com> References: <4A637FC5.6000301@gmail.com> <41A88E3D-C420-4A1F-8B80-098EA18F9544@freeswitch.org> <4A661919.8030500@gmail.com> Message-ID: Its a 302 on a single leg call right? /b On Jul 21, 2009, at 2:38 PM, ?ukasz Zwierko wrote: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > Brian, > > I've just updated to 14310 and it's the same. The thing seems that > sofia > module rejects the call in quite early stage, so there is no 302 > answer > from remote SIP peer (as no INVITE was sent). > Again, I'm exercising a very simple scenario with default FS > configuration (just downloaded from svn), so I don't really know > what's > wrong here... > Perhaps there is a different way to attach a call to a existing > conference? > Perhaps I should just originate new call (with the 'originate' > command), > and when received, pass it it conference application with the > conference-id of the conference that I want to attach it to? Does that > make any sense? > > Thanks > > ? From rupa at rupa.com Tue Jul 21 13:01:03 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Tue, 21 Jul 2009 15:01:03 -0500 Subject: [Freeswitch-users] Call confirm ivr In-Reply-To: <5d3e0dc60907211201t2be7ea8dq2b2931f854bdc455@mail.gmail.com> References: <5d3e0dc60907211201t2be7ea8dq2b2931f854bdc455@mail.gmail.com> Message-ID: Maybe look at the group_confirm_* stuff. http://wiki.freeswitch.org/wiki/Freeswitch_IVR_Originate#Answer_confirmation There is a way to get it to execute a script as well which is probably waht you want. This would be simpler than doing the work you are saying below. On Tue, Jul 21, 2009 at 2:01 PM, Lon Baker wrote: > Hi there, > I am putting together an ivr to allow the recipient of a call to accept, > route to voice mail or eavesdrop on voicemail. > > The current path is to answer the inbound call, park it, using the bgapi > call to the recipient and play the IVR. > > Basically: > > 1. Answer > 2. Playback greeting > 3. UUID_PARK > 4. Set filter for BACKGROUND_JOB > 5. BGAPI Originate to the recipient with customer variables for > processing > 6. Do new IVR for recipient and process their input to route call. > > I am not sure is this the right path. > > Is there a better way? > > Lon > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090721/07027e0f/attachment-0001.html From nicolas at medularis.com Tue Jul 21 13:05:22 2009 From: nicolas at medularis.com (Nicolas Brenner) Date: Tue, 21 Jul 2009 16:05:22 -0400 Subject: [Freeswitch-users] Best way to bridge 2 calls with LCR? In-Reply-To: References: <1b46b4e80907210835s6e516e2ft84f3d7329eee1535@mail.gmail.com> <1b46b4e80907210927n4885b4c5g180a06e32bedc7c4@mail.gmail.com> <1b46b4e80907211100r4f9967fesb1b2c5fba4785a66@mail.gmail.com> Message-ID: <1b46b4e80907211305g17b838b1s9e5acd72da7f1792@mail.gmail.com> Now I understand! thank you very much for your explanation, very clear! On Tue, Jul 21, 2009 at 2:21 PM, Rupa Schomaker wrote: > Well, the "as xml" is something I've been meaning to do, so I'm gonna get > that checked in today sometime anyway. If you want to do any programmatic > processing of the lcr data, the as xml is the way to go rather than parsing > the strings. > > As for originate + lcr.... You can use the loopback endpoint and do it all > in the dialplan: > > originate loopback/firstnumber secondnumber > > This will hit your dialplan with firstnumber first which you can lcr > route. Then when that call establishes, it'll hit the dialplan with the > second number which will also be routed through lcr. > > Is that more what you are looking for? > > This way all the 'routing' logic can be done via the dialplan. > > > On Tue, Jul 21, 2009 at 1:00 PM, Nicolas Brenner wrote: > >> Thank you very much for the offer, but I don't want to bother you with >> this. >> >> I can just parse the string returned by lcr and get the gateway, that's >> all I really need to create my complete originate command. >> >> I am using the socket api because it is easier for me to understand how to >> do it, nevertheless I'd really like to know how to do it with the dialplan. >> >> What I don't understand very well about using the dialplan for this, is >> how to do the first originate command (which I need to do using the socket >> api). What puzzles me is that according to the originate syntax, I need to >> use an extension or call an application, yet for the first call I would have >> to use a dummy extension as I only need to hit the dialplan section that >> calls lcr once to originate the first call with an extension that hits the >> section of the dialplan where lcr gets called again and the calls get >> bridged. >> >> I'm thinking something like this: >> >> 1) call originate from socket api to hit dialplan section that does all >> the work (this originate command is what I don't understand, is there >> another way of "hitting the dialplan" besides calling originate?) >> >> 2) hit dialplan section which calls lcr for first number and bridges to an >> extension >> >> 3) the extension calls lcr fir the second number and originates the second >> call >> >> On steps 2 and 3 I could just use set data to set the additional variables >> I need. The first step is what troubles me. >> >> >> Thank you! >> >> >> Nicolas >> >> >> On Tue, Jul 21, 2009 at 12:54 PM, Rupa Schomaker wrote: >> >>> Ok, if you want to do it from the socket api, then I need to make a 'as >>> xml' option to mod_lcr and give you lcr_auto_route as one of the nodes in >>> the returned xml. Then you can do your own substitution in the originate >>> line... In that case, you'd call lcr twice and do: >>> >>> originate lcr_auto_route1 &bridge(lcr_auto_route2) >>> >>> How soon do you need this? >>> >>> >>> On Tue, Jul 21, 2009 at 11:27 AM, Nicolas Brenner >> > wrote: >>> >>>> That looks like a good way to go about it. >>>> >>>> How can I access channel variables through the socket using the api? I >>>> mean, how do I recover the value of ${lcr_auto_route}? I would need to add >>>> some other variables, like ignore_early_media=true and a uuid that 'links' >>>> the two calls so I can track it listening for events. >>>> >>>> Thanks! >>>> >>>> Nicolas >>>> >>>> >>>> On Tue, Jul 21, 2009 at 11:43 AM, Rupa Schomaker wrote: >>>> >>>>> lcr api command doesn't really return a usable dialstring (it was >>>>> originally done for debug purposes). I could add an "as xml" option if >>>>> needed... >>>>> >>>>> Anyway, to do this from the dialplan: >>>>> >>>>> remember that originate's usage is: >>>>> >>>>> -USAGE |&() [] >>>>> [] [] [] [] >>>>> >>>>> so, the first argument is the call url and the second would be an >>>>> extension. so: >>>>> >>>>> 1) execute lcr for the first leg of the call >>>>> 2) execute originate with: >>>>> >>>>> originate ${lcr_auto_route} extension >>>>> >>>>> extension just needs to match something in your dialplan. >>>>> >>>>> In extension, you'd do another lcr lookup and then bridge to that leg's >>>>> ${lcr_auto_route} value. >>>>> >>>>> >>>>> >>>>> On Tue, Jul 21, 2009 at 10:35 AM, Nicolas Brenner < >>>>> nicolas at medularis.com> wrote: >>>>> >>>>>> I would like to originate 2 calls from FS and then bridge them. >>>>>> There's no incoming call so I think there's no dialplan involved. >>>>>> What I'd like to do now is apply lcr rules to these calls. I've come >>>>>> up with 2 options so far: >>>>>> >>>>>> 1) call lcr through the socket twice (once for each phonenumber) and >>>>>> then originate the calls through the socket too >>>>>> 2) have a javascript file which runs the actions above, run the script >>>>>> through the socket with 'jsrun' >>>>>> >>>>>> How would you do it? >>>>>> >>>>>> For what I've read on the list, usually the recommended way is to stay >>>>>> away from javascript as much as possible because it is not as efficient as >>>>>> doing everything from the dialplan. Does this mean the first option is the >>>>>> best? or is there a "dialplan way" of doing it? >>>>>> >>>>>> Thank you very much for your help! >>>>>> >>>>>> Nicolas >>>>>> >>>>>> _______________________________________________ >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>> >>>>> >>>>> -- >>>>> -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 >>>>> >>>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> -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 >>> >>> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > -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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090721/476cfe9b/attachment-0001.html From nicolas at medularis.com Tue Jul 21 13:05:37 2009 From: nicolas at medularis.com (Nicolas Brenner) Date: Tue, 21 Jul 2009 16:05:37 -0400 Subject: [Freeswitch-users] Best way to bridge 2 calls with LCR? In-Reply-To: References: <1b46b4e80907210835s6e516e2ft84f3d7329eee1535@mail.gmail.com> <1b46b4e80907210927n4885b4c5g180a06e32bedc7c4@mail.gmail.com> <1b46b4e80907211100r4f9967fesb1b2c5fba4785a66@mail.gmail.com> Message-ID: <1b46b4e80907211305m2b21f576n757245b78200939c@mail.gmail.com> Great! Thanks! On Tue, Jul 21, 2009 at 2:51 PM, Rupa Schomaker wrote: > Just a note that the "as xml" syntax has been added to current trunk. > > On Tue, Jul 21, 2009 at 1:21 PM, Rupa Schomaker wrote: > >> Well, the "as xml" is something I've been meaning to do, so I'm gonna get >> that checked in today sometime anyway. If you want to do any programmatic >> processing of the lcr data, the as xml is the way to go rather than parsing >> the strings. >> >> > -- > -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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090721/19812cfc/attachment-0001.html From lon at kickasspixels.com Tue Jul 21 13:29:40 2009 From: lon at kickasspixels.com (Lon Baker) Date: Tue, 21 Jul 2009 13:29:40 -0700 Subject: [Freeswitch-users] Call confirm ivr In-Reply-To: References: <5d3e0dc60907211201t2be7ea8dq2b2931f854bdc455@mail.gmail.com> Message-ID: <576617A9-AA8E-47BB-96FA-4D882592E50A@kickasspixels.com> Thanks. I have looked at that, but everything has to run over the event_socket to the application logic we are building. I didn't see a way to exec a remote script/URL. Lon On Jul 21, 2009, at 1:01 PM, Rupa Schomaker wrote: > Maybe look at the group_confirm_* stuff. > > http://wiki.freeswitch.org/wiki/Freeswitch_IVR_Originate#Answer_confirmation > > There is a way to get it to execute a script as well which is > probably waht you want. This would be simpler than doing the work > you are saying below. > > On Tue, Jul 21, 2009 at 2:01 PM, Lon Baker > wrote: > Hi there, > > I am putting together an ivr to allow the recipient of a call to > accept, route to voice mail or eavesdrop on voicemail. > > The current path is to answer the inbound call, park it, using the > bgapi call to the recipient and play the IVR. > > Basically: > Answer > Playback greeting > UUID_PARK > Set filter for BACKGROUND_JOB > BGAPI Originate to the recipient with customer variables for > processing > Do new IVR for recipient and process their input to route call. > I am not sure is this the right path. > > Is there a better way? > > Lon > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org > > > > > -- > -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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090721/3228aeaa/attachment-0001.html From msc at freeswitch.org Tue Jul 21 14:58:30 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 21 Jul 2009 14:58:30 -0700 Subject: [Freeswitch-users] ClueCon 2009 - Check Out The 15" MacBook Pro That We're Giving Away Message-ID: <87f2f3b90907211458x109d7611m2d64a5dfa0e5fcab@mail.gmail.com> This just in: We have a picture of the incredible MacBook Pro that we will be giving away this year: http://cluecon.com/node/38 Remember, all paid attendees are eligible to win this beautiful unit, so register today! Call 877.742.CLUE and get signed up. -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090721/fc3cf52b/attachment-0001.html From pete at privateconnect.com Tue Jul 21 21:35:12 2009 From: pete at privateconnect.com (Pete Mueller) Date: Tue, 21 Jul 2009 21:35:12 -0700 Subject: [Freeswitch-users] Confusing handling of incoming calls Message-ID: <20090721213512.2ad02225396a31c9de30536f2e338977.fae79d6455.wbe@email04.secureserver.net> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090721/7160996a/attachment-0001.html From lzwierko at gmail.com Tue Jul 21 22:45:54 2009 From: lzwierko at gmail.com (=?ISO-8859-2?Q?=A3ukasz_Zwierko?=) Date: Wed, 22 Jul 2009 07:45:54 +0200 Subject: [Freeswitch-users] Dial up from confernece issue In-Reply-To: References: <4A637FC5.6000301@gmail.com> <41A88E3D-C420-4A1F-8B80-098EA18F9544@freeswitch.org> <4A661919.8030500@gmail.com> Message-ID: I'm not sure how this exactly works, but I suppose that it is a single leg call, which upon answer would be attached to the conference (?) somehow. But again, this call does not originate outside FS so what would be the cause for 302? 2009/7/21 Brian West : > Its a 302 on a single leg call right? > > /b > > On Jul 21, 2009, at 2:38 PM, ?ukasz Zwierko wrote: > >> -----BEGIN PGP SIGNED MESSAGE----- >> Hash: SHA1 >> >> Brian, >> >> I've just updated to 14310 and it's the same. The thing seems that >> sofia >> module rejects the call in quite early stage, so there is no 302 >> answer >> from remote SIP peer (as no INVITE was sent). >> Again, I'm exercising a very simple scenario with default FS >> configuration (just downloaded from svn), so I don't really know >> what's >> wrong here... >> Perhaps there is a different way to attach a call to a existing >> conference? >> Perhaps I should just originate new call (with the 'originate' >> command), >> and when received, pass it it conference application with the >> conference-id of the conference that I want to attach it to? Does that >> make any sense? >> >> 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 > From elihayun at gmail.com Wed Jul 22 00:19:43 2009 From: elihayun at gmail.com (Eli Hayun) Date: Wed, 22 Jul 2009 10:19:43 +0300 Subject: [Freeswitch-users] How to initiate a call without dialing In-Reply-To: <4A65C911.50707@freeswitch.org> References: <4A659D30.5020600@savion.huji.ac.il> <4A65C911.50707@freeswitch.org> Message-ID: <4A66BD8F.8050108@savion.huji.ac.il> Raymond Chandler wrote: > Eli Hayun wrote: > >> Is there is a way to initiate a call without making any dial manually? >> >> > i think the api command "originate" is what you're looking for > > -Ray > > _______________________________________________ > Thanks, I figure that out, but now I have another problem. When I do that, the name display as "FreeSwitch" and the number is display as "00000000000" I tried to set "outbound_caller_name" with no success. How should I solve that? Thanks Eli From solko at gcdf.pl Wed Jul 22 01:06:01 2009 From: solko at gcdf.pl (Szymon Olko) Date: Wed, 22 Jul 2009 10:06:01 +0200 Subject: [Freeswitch-users] How to initiate a call without dialing In-Reply-To: <4A66BD8F.8050108@savion.huji.ac.il> References: <4A659D30.5020600@savion.huji.ac.il> <4A65C911.50707@freeswitch.org> <4A66BD8F.8050108@savion.huji.ac.il> Message-ID: <4A66C869.3070806@gcdf.pl> Eli Hayun pisze: > Raymond Chandler wrote: >> Eli Hayun wrote: >> >>> Is there is a way to initiate a call without making any dial manually? >>> >>> >> i think the api command "originate" is what you're looking for >> >> -Ray >> >> _______________________________________________ >> > Thanks, I figure that out, but now I have another problem. When I do > that, the name display as "FreeSwitch" and the number is display as > "00000000000" > I tried to set "outbound_caller_name" with no success. > How should I solve that? > > Thanks > Eli > Read wiki, it explains a lot. http://wiki.freeswitch.org/wiki/Mod_commands#originate use it like that: originate sofia/internal/1001%192.168.1.1 3001 XML default Name 1213232 OR originate sofia/internal/1001%192.168.1.1 &conference(test) '' '' Name 1213232 From elihayun at gmail.com Wed Jul 22 01:44:52 2009 From: elihayun at gmail.com (Eli Hayun) Date: Wed, 22 Jul 2009 11:44:52 +0300 Subject: [Freeswitch-users] How to initiate a call without dialing In-Reply-To: <4A66C869.3070806@gcdf.pl> References: <4A659D30.5020600@savion.huji.ac.il> <4A65C911.50707@freeswitch.org> <4A66BD8F.8050108@savion.huji.ac.il> <4A66C869.3070806@gcdf.pl> Message-ID: <4A66D184.3090201@savion.huji.ac.il> Szymon Olko wrote: > Eli Hayun pisze: > >> Raymond Chandler wrote: >> >>> Eli Hayun wrote: >>> >>> >>>> Is there is a way to initiate a call without making any dial manually? >>>> >>>> >>>> >>> i think the api command "originate" is what you're looking for >>> >>> -Ray >>> >>> _______________________________________________ >>> >>> >> Thanks, I figure that out, but now I have another problem. When I do >> that, the name display as "FreeSwitch" and the number is display as >> "00000000000" >> I tried to set "outbound_caller_name" with no success. >> How should I solve that? >> >> Thanks >> Eli >> >> > Read wiki, it explains a lot. > > http://wiki.freeswitch.org/wiki/Mod_commands#originate > > use it like that: > originate sofia/internal/1001%192.168.1.1 3001 XML default Name 1213232 > > OR > > originate sofia/internal/1001%192.168.1.1 &conference(test) '' '' Name 1213232 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > Thanks alot. Its working now. Eli From rupa at rupa.com Wed Jul 22 02:12:30 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Wed, 22 Jul 2009 04:12:30 -0500 Subject: [Freeswitch-users] Confusing handling of incoming calls In-Reply-To: <20090721213512.2ad02225396a31c9de30536f2e338977.fae79d6455.wbe@email04.secureserver.net> References: <20090721213512.2ad02225396a31c9de30536f2e338977.fae79d6455.wbe@email04.secureserver.net> Message-ID: On Tue, Jul 21, 2009 at 11:35 PM, Pete Mueller wrote: > My goal is: > 0) figure out why the bandwidth gateway is being processed as "internal" > (this is more of a security thing) > they are probably terminating traffic on port 5060 rather than 5080. 5060 is internal, 5080 is external. > > 1) have both gateways enter at the same point in the dialplan (this seems > to be the purpose of the "Extension" param) > I'd drop the extension param and instead match on the destination_number (the DID used to reach you). > 2) be able to identify which gateway the call came in on. I was hoping to > set a param in the gateway configuration that would be passed through onto > the channel, but have not found one. Worst case, I could have each gateway > enter at a different extension in the dialplan, however, that doesn't seem > to be working if the channel comes in the "internal" profile. > Not sure here... gateways are an outbound thing. Inbound calls just hit your dialplan and you process from there. A sledgehammer approach would be to have a different sip_profile for each gateway. But that is just silly. Flowroute at least puts their name in the sdp.... > Thanks for your help. I've provided INFO dumps from both gateways if they > help... > -pete > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090722/61a4b30b/attachment-0001.html From pete at privateconnect.com Wed Jul 22 03:11:26 2009 From: pete at privateconnect.com (Pete Mueller) Date: Wed, 22 Jul 2009 03:11:26 -0700 Subject: [Freeswitch-users] Confusing handling of incoming calls Message-ID: <20090722031126.2ad02225396a31c9de30536f2e338977.ca479dcfc2.wbe@email04.secureserver.net> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090722/11590c21/attachment-0001.html From rdenert at tng.de Wed Jul 22 03:23:02 2009 From: rdenert at tng.de (Rudolf Denert) Date: Wed, 22 Jul 2009 12:23:02 +0200 (CEST) Subject: [Freeswitch-users] Playing sound files in a conference In-Reply-To: <25598690.126581248258109020.JavaMail.root@zimbra.tng.de> Message-ID: <4286351.126631248258182227.JavaMail.root@zimbra.tng.de> Hallo everybody! I would like to play soundfiles in a existing conference. The procedure is this: Someone calls the number of the conference. Then this person types the pin in to his phone. The next step is that he has to say the name for example "John". This file is saved in a special folder. When he enters the conference room everybody in this existing conference should here: "John" "has enterd the room". If he leaves then everybody should hear: "John" "has left the room". Of course there are two soundfiles. "John" is what the caller has spoken into his phone and the second one a generated file from me. They should be played in succession. Is it possible to implement this with lua? If yes, how can I do that. Thanks for your help. Greetz From brian at freeswitch.org Wed Jul 22 03:43:37 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 22 Jul 2009 05:43:37 -0500 Subject: [Freeswitch-users] Confusing handling of incoming calls In-Reply-To: <20090722031126.2ad02225396a31c9de30536f2e338977.ca479dcfc2.wbe@email04.secureserver.net> References: <20090722031126.2ad02225396a31c9de30536f2e338977.ca479dcfc2.wbe@email04.secureserver.net> Message-ID: On Jul 22, 2009, at 5:11 AM, Pete Mueller wrote: > 0) Rupa, you are absolutely right, I forgot that. ports was never > an issue because previous gateways all REGISTERed. I will have to > swap my ports around as bandwidth is not flexible. What do you mean here? > > 1) I thought of this, but I have hundreds of DID, (around 600 at the > moment) and maintaining that mapping in the dialplan would be a > mess. AFTER I know what gateway the call arrived on, I have a > database for each gateway that helps me process from there. XML_CURL? > > 2) Yes, separate profiles would work, but does sound gross. I'm > going to swap my ports around and see if that clears things up... > > -pete -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090722/ba51e974/attachment-0001.html From brian at freeswitch.org Wed Jul 22 03:50:02 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 22 Jul 2009 05:50:02 -0500 Subject: [Freeswitch-users] Dial up from confernece issue In-Reply-To: References: <4A637FC5.6000301@gmail.com> <41A88E3D-C420-4A1F-8B80-098EA18F9544@freeswitch.org> <4A661919.8030500@gmail.com> Message-ID: <122E4883-9AF7-4F7A-AAC0-2562ECBDBFDF@freeswitch.org> The far end you're calling is sending a 302 can you check the sip traffic please. /b On Jul 22, 2009, at 12:45 AM, ?ukasz Zwierko wrote: > I'm not sure how this exactly works, but I suppose that it is a single > leg call, which upon answer would be attached to the conference (?) > somehow. But again, this call does not originate outside FS so what > would be the cause for 302? From elihayun at gmail.com Wed Jul 22 05:45:08 2009 From: elihayun at gmail.com (Eli Hayun) Date: Wed, 22 Jul 2009 15:45:08 +0300 Subject: [Freeswitch-users] Limit is not working when originate a call Message-ID: <4A6709D4.6000704@savion.huji.ac.il> Hi I set the limit to 1 on the extension like that When I am trying to make a call the that destination i transfered to limit_exceeded dialplan, just like I want The problem is, that when I am trying to make a call using "originate" I am not getting the limitation. Why is that? From rupa at rupa.com Wed Jul 22 06:04:35 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Wed, 22 Jul 2009 08:04:35 -0500 Subject: [Freeswitch-users] Confusing handling of incoming calls In-Reply-To: <20090722031126.2ad02225396a31c9de30536f2e338977.ca479dcfc2.wbe@email04.secureserver.net> References: <20090722031126.2ad02225396a31c9de30536f2e338977.ca479dcfc2.wbe@email04.secureserver.net> Message-ID: On Wed, Jul 22, 2009 at 5:11 AM, Pete Mueller wrote: > 0) Rupa, you are absolutely right, I forgot that. ports was never an issue > because previous gateways all REGISTERed. I will have to swap my ports > around as bandwidth is not flexible. > You can't tell bandwidth.com to use port 5080? > > 1) I thought of this, but I have hundreds of DID, (around 600 at the > moment) and maintaining that mapping in the dialplan would be a mess. AFTER > I know what gateway the call arrived on, I have a database for each gateway > that helps me process from there. > You have cases where the same DID maps differently for one gateway or another? If not, why is the gateway part of the database query? > > 2) Yes, separate profiles would work, but does sound gross. I'm going to > swap my ports around and see if that clears things up... > > -pete > > -------- Original Message -------- > Subject: Re: [Freeswitch-users] Confusing handling of incoming calls > From: Rupa Schomaker > Date: Wed, July 22, 2009 2:12 am > To: freeswitch-users at lists.freeswitch.org > > > > On Tue, Jul 21, 2009 at 11:35 PM, Pete Mueller wrote: > >> My goal is: >> 0) figure out why the bandwidth gateway is being processed as "internal" >> (this is more of a security thing) >> > > they are probably terminating traffic on port 5060 rather than 5080. 5060 > is internal, 5080 is external. > > >> >> 1) have both gateways enter at the same point in the dialplan (this seems >> to be the purpose of the "Extension" param) >> > > I'd drop the extension param and instead match on the destination_number > (the DID used to reach you). > > >> 2) be able to identify which gateway the call came in on. I was hoping to >> set a param in the gateway configuration that would be passed through onto >> the channel, but have not found one. Worst case, I could have each gateway >> enter at a different extension in the dialplan, however, that doesn't seem >> to be working if the channel comes in the "internal" profile. >> > > Not sure here... gateways are an outbound thing. Inbound calls just hit > your dialplan and you process from there. A sledgehammer approach would be > to have a different sip_profile for each gateway. But that is just silly. > Flowroute at least puts their name in the sdp.... > > >> Thanks for your help. I've provided INFO dumps from both gateways if they >> help... >> -pete >> > > > -- > -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 > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090722/0e3267c6/attachment-0001.html From brian at freeswitch.org Wed Jul 22 06:10:03 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 22 Jul 2009 08:10:03 -0500 Subject: [Freeswitch-users] Confusing handling of incoming calls In-Reply-To: References: <20090722031126.2ad02225396a31c9de30536f2e338977.ca479dcfc2.wbe@email04.secureserver.net> Message-ID: <140A2687-B8F6-407D-9B67-A91DB5496AC4@freeswitch.org> On Jul 22, 2009, at 8:04 AM, Rupa Schomaker wrote: > > > On Wed, Jul 22, 2009 at 5:11 AM, Pete Mueller > wrote: > 0) Rupa, you are absolutely right, I forgot that. ports was never > an issue because previous gateways all REGISTERed. I will have to > swap my ports around as bandwidth is not flexible. > > You can't tell bandwidth.com to use port 5080? Yes you can... I do it all the time. > > 1) I thought of this, but I have hundreds of DID, (around 600 at the > moment) and maintaining that mapping in the dialplan would be a > mess. AFTER I know what gateway the call arrived on, I have a > database for each gateway that helps me process from there. > > You have cases where the same DID maps differently for one gateway > or another? If not, why is the gateway part of the database query? > > 2) Yes, separate profiles would work, but does sound gross. I'm > going to swap my ports around and see if that clears things up... > > -pete -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090722/393f04e9/attachment-0001.html From intralanman at freeswitch.org Wed Jul 22 06:47:58 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Wed, 22 Jul 2009 09:47:58 -0400 Subject: [Freeswitch-users] Playing sound files in a conference In-Reply-To: <4286351.126631248258182227.JavaMail.root@zimbra.tng.de> References: <4286351.126631248258182227.JavaMail.root@zimbra.tng.de> Message-ID: <4A67188E.5060802@freeswitch.org> you could hang on the event socket and catch the conference events, then play the sounds via the "conference" api commands -Ray Rudolf Denert wrote: > Hallo everybody! > > I would like to play soundfiles in a existing conference. > > The procedure is this: > > Someone calls the number of the conference. Then this person types the pin in to his phone. The next step is that he has to say the name for example "John". This file is saved in a special folder. When he enters the conference room everybody in this existing conference should here: "John" "has enterd the room". If he leaves then everybody should hear: "John" "has left the room". Of course there are two soundfiles. "John" is what the caller has spoken into his phone and the second one a generated file from me. They should be played in succession. > > Is it possible to implement this with lua? If yes, how can I do that. > > Thanks for your help. > > Greetz > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.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 Wed Jul 22 07:46:06 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 22 Jul 2009 10:46:06 -0400 Subject: [Freeswitch-users] Limit is not working when originate a call In-Reply-To: <4A6709D4.6000704@savion.huji.ac.il> References: <4A6709D4.6000704@savion.huji.ac.il> Message-ID: <2C85EB6C-1D80-4DC7-8D14-06B17CD1046D@jerris.com> because your not running limit at all when you are doing an originate directly. You can use loopback to originate through a dialplan extension. Mike On Jul 22, 2009, at 8:45 AM, Eli Hayun wrote: > Hi > I set the limit to 1 on the extension like that > > > > When I am trying to make a call the that destination i transfered to > limit_exceeded dialplan, just like I want > > The problem is, that when I am trying to make a call using > "originate" I > am not getting the limitation. > Why is 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 msc at freeswitch.org Wed Jul 22 10:30:16 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 22 Jul 2009 10:30:16 -0700 Subject: [Freeswitch-users] Good Information On How To Submit Bug Reports Message-ID: <87f2f3b90907221030y3f2cc840y504cdbb3ac6194ec@mail.gmail.com> FYI, Brian West called to my attention that one of our community members, John Wehle, has been very good at submitting useful bug reports, in many cases with patches. His style of reporting is worthy of imitation, so I've added a few links to the JIRA section of the Reporting Bugs wiki page: http://wiki.freeswitch.org/wiki/Reporting_Bugs#Examples_Of_Well-Written_JIRA_Submissions Please feel free to check it out. If you have any questions on what a bug report should look like then definitely read some of John's submissions and emulate his style. By submitting useful bug reports you will save the FreeSWITCH developers countless hours and headaches, not to mention the warm fuzzies you'll feel inside. :) -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090722/01374541/attachment-0002.html From pete at privateconnect.com Wed Jul 22 11:30:11 2009 From: pete at privateconnect.com (Pete Mueller) Date: Wed, 22 Jul 2009 11:30:11 -0700 Subject: [Freeswitch-users] Confusing handling of incoming calls Message-ID: <20090722113011.2ad02225396a31c9de30536f2e338977.3d5b2c362e.wbe@email04.secureserver.net> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090722/ea917137/attachment-0002.html From larclap at yahoo.com Wed Jul 22 13:13:56 2009 From: larclap at yahoo.com (Lars Zeb) Date: Wed, 22 Jul 2009 13:13:56 -0700 Subject: [Freeswitch-users] Asterisk key during message hangs up call In-Reply-To: References: <004a01c9da6f$c1e40120$45ac0360$@com> Message-ID: <00b501ca0b08$f178a520$d469ef60$@com> Brian, When calling into FreeSWITCH and pressing * during the greeting, the call immediately hangs up. It used to ask for the mailbox number to retrieve messages. It no longer works. I don't know if my dialplan is causing the error or something in FreeSWITCH has changed. Any ideas? http://pastebin.freeswitch.org/9803 Thanks, Lars -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090722/37516c17/attachment-0002.html From lfurrea at gmail.com Wed Jul 22 16:21:07 2009 From: lfurrea at gmail.com (Luis F Urrea) Date: Wed, 22 Jul 2009 17:21:07 -0600 Subject: [Freeswitch-users] Asterisk key during message hangs up call In-Reply-To: <00b501ca0b08$f178a520$d469ef60$@com> References: <004a01c9da6f$c1e40120$45ac0360$@com> <00b501ca0b08$f178a520$d469ef60$@com> Message-ID: I don't know if this may be related but in voicemail.conf.xml by default the two params that follow are defined: And pressing 9 during the greeting does not send me to the operator. I am on trunk rev 14123M On Wed, Jul 22, 2009 at 2:13 PM, Lars Zeb wrote: > Brian, > > > > When calling into FreeSWITCH and pressing * during the greeting, the call > immediately hangs up. > > > > It used to ask for the mailbox number to retrieve messages. It no longer > works. I don?t know if my dialplan is causing the error or something in > FreeSWITCH has changed. > > > > Any ideas? > > > > http://pastebin.freeswitch.org/9803 > > > > Thanks, Lars > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20090722/cdd6efc0/attachment-0002.html From anthony.minessale at gmail.com Wed Jul 22 16:33:08 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 22 Jul 2009 18:33:08 -0500 Subject: [Freeswitch-users] Asterisk key during message hangs up call In-Reply-To: <00b501ca0b08$f178a520$d469ef60$@com> References: <004a01c9da6f$c1e40120$45ac0360$@com> <00b501ca0b08$f178a520$d469ef60$@com> Message-ID: <191c3a030907221633k6594901dj22b9c7ded8af19e4@mail.gmail.com> you are using a channel created with a script and you did not set js session.autoHangup(0) lua session:autoHangup(0) so when the * makes the call transfer the script kills the channel. On Wed, Jul 22, 2009 at 3:13 PM, Lars Zeb wrote: > Brian, > > > > When calling into FreeSWITCH and pressing * during the greeting, the call > immediately hangs up. > > > > It used to ask for the mailbox number to retrieve messages. It no longer > works. I don?t know if my dialplan is causing the error or something in > FreeSWITCH has changed. > > > > Any ideas? > > > > http://pastebin.freeswitch.org/9803 > > > > Thanks, Lars > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference 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/20090722/9bbc547a/attachment-0002.html From mike at jerris.com Wed Jul 22 16:46:30 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 22 Jul 2009 19:46:30 -0400 Subject: [Freeswitch-users] Asterisk key during message hangs up call In-Reply-To: References: <004a01c9da6f$c1e40120$45ac0360$@com> <00b501ca0b08$f178a520$d469ef60$@com> Message-ID: <01355490-8C46-4E90-A4C1-51F4921AEFBF@jerris.com> Do you have anything on that extension? On Jul 22, 2009, at 7:21 PM, Luis F Urrea wrote: > I don't know if this may be related but in voicemail.conf.xml by > default the two params that follow are defined: > > > > > And pressing 9 during the greeting does not send me to the operator. > > I am on trunk rev 14123M > > On Wed, Jul 22, 2009 at 2:13 PM, Lars Zeb wrote: > Brian, > > > > When calling into FreeSWITCH and pressing * during the greeting, the > call immediately hangs up. > > > > It used to ask for the mailbox number to retrieve messages. It no > longer works. I don?t know if my dialplan is causing the error or so > mething in FreeSWITCH has changed. > > > > Any ideas? > > > > http://pastebin.freeswitch.org/9803 > > > > Thanks, Lars > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20090722/c99dda3d/attachment-0002.html From elihayun at gmail.com Wed Jul 22 22:04:07 2009 From: elihayun at gmail.com (Eli Hayun) Date: Thu, 23 Jul 2009 08:04:07 +0300 Subject: [Freeswitch-users] Limit is not working when originate a call In-Reply-To: <2C85EB6C-1D80-4DC7-8D14-06B17CD1046D@jerris.com> References: <4A6709D4.6000704@savion.huji.ac.il> <2C85EB6C-1D80-4DC7-8D14-06B17CD1046D@jerris.com> Message-ID: <4A67EF47.6000402@savion.huji.ac.il> Michael Jerris wrote: > because your not running limit at all when you are doing an originate > directly. You can use loopback to originate through a dialplan > extension. > > Mike > > On Jul 22, 2009, at 8:45 AM, Eli Hayun wrote: > > >> Hi >> I set the limit to 1 on the extension like that >> >> >> >> When I am trying to make a call the that destination i transfered to >> limit_exceeded dialplan, just like I want >> >> The problem is, that when I am trying to make a call using >> "originate" I >> am not getting the limitation. >> Why is 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 >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > Thanks for answer. I am calling "Originate" from JS. I tried to call "limit_hash" from JS but with no success. I did it like that: lmt = apiExecute("limit_hash", dialed_ext + " " + dialed_ext + " 1"); I could't find any documentation on that. can u help ? Thanks Eli From anthony.minessale at gmail.com Thu Jul 23 05:39:47 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 23 Jul 2009 07:39:47 -0500 Subject: [Freeswitch-users] Limit is not working when originate a call In-Reply-To: <4A67EF47.6000402@savion.huji.ac.il> References: <4A6709D4.6000704@savion.huji.ac.il> <2C85EB6C-1D80-4DC7-8D14-06B17CD1046D@jerris.com> <4A67EF47.6000402@savion.huji.ac.il> Message-ID: <191c3a030907230539i5a640e2do4377d8797ff8f279@mail.gmail.com> limit is for inbound calls you cannot call it after you already made the call. The correct approach would be to not make the call at all. you could maybe use the limit FSAPI interface with apiExecute to check if the limit was exceeded and then not bother to place the call to begin with. otherwise it's sort of like putting a prisoner in the electric chair then giving him his trial. On Thu, Jul 23, 2009 at 12:04 AM, Eli Hayun wrote: > Michael Jerris wrote: > > because your not running limit at all when you are doing an originate > > directly. You can use loopback to originate through a dialplan > > extension. > > > > Mike > > > > On Jul 22, 2009, at 8:45 AM, Eli Hayun wrote: > > > > > >> Hi > >> I set the limit to 1 on the extension like that > >> > >> > >> > >> When I am trying to make a call the that destination i transfered to > >> limit_exceeded dialplan, just like I want > >> > >> The problem is, that when I am trying to make a call using > >> "originate" I > >> am not getting the limitation. > >> Why is 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 > >> > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > Thanks for answer. > I am calling "Originate" from JS. I tried to call "limit_hash" from JS > but with no success. I did it like that: > > lmt = apiExecute("limit_hash", dialed_ext + " " + dialed_ext + " 1"); > > I could't find any documentation on that. > can u help ? > > Thanks > Eli > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference 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/20090723/e19fe0fa/attachment-0002.html From larclap at yahoo.com Thu Jul 23 07:23:11 2009 From: larclap at yahoo.com (Lars Zeb) Date: Thu, 23 Jul 2009 07:23:11 -0700 Subject: [Freeswitch-users] Asterisk key during message hangs up call In-Reply-To: <191c3a030907221633k6594901dj22b9c7ded8af19e4@mail.gmail.com> References: <004a01c9da6f$c1e40120$45ac0360$@com> <00b501ca0b08$f178a520$d469ef60$@com> <191c3a030907221633k6594901dj22b9c7ded8af19e4@mail.gmail.com> Message-ID: <004c01ca0ba1$1bb6de90$53249bb0$@com> Thanks for the reply. This is my first attempt at using a script. I tried: session:autoHangup(0) or session:autoHangup(false) but got an error: 2009-07-23 07:19:09.652238 [ERR] mod_lua.cpp:182 /usr/local/freeswitch/scripts/inbound_calls.lua:29: attempt to call method 'autoHangup' (a nil value) stack traceback: /usr/local/freeswitch/scripts/inbound_calls.lua:29: in main chunk I looked at the documentation and tried: session:setAutoHangup(false) and the script proceeded without error. However, looking at the log, I do not see the setAutoHangup being called. Also, when pressing *, I get a fast, busy signal. I have pasted the script and log at http://pastebin.freeswitch.org/9836 Thanks again, Lars From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Wednesday, July 22, 2009 4:33 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Asterisk key during message hangs up call you are using a channel created with a script and you did not set js session.autoHangup(0) lua session:autoHangup(0) so when the * makes the call transfer the script kills the channel. On Wed, Jul 22, 2009 at 3:13 PM, Lars Zeb wrote: Brian, When calling into FreeSWITCH and pressing * during the greeting, the call immediately hangs up. It used to ask for the mailbox number to retrieve messages. It no longer works. I don't know if my dialplan is causing the error or something in FreeSWITCH has changed. Any ideas? http://pastebin.freeswitch.org/9803 Thanks, Lars _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference 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/20090723/bea25341/attachment-0002.html From anthony.minessale at gmail.com Thu Jul 23 07:41:28 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 23 Jul 2009 09:41:28 -0500 Subject: [Freeswitch-users] Asterisk key during message hangs up call In-Reply-To: <004c01ca0ba1$1bb6de90$53249bb0$@com> References: <004a01c9da6f$c1e40120$45ac0360$@com> <00b501ca0b08$f178a520$d469ef60$@com> <191c3a030907221633k6594901dj22b9c7ded8af19e4@mail.gmail.com> <004c01ca0ba1$1bb6de90$53249bb0$@com> Message-ID: <191c3a030907230741uc04b0cdoafcee3f79be55850@mail.gmail.com> I meant to pick one based on whichever lang you were using not to literally write what i said. anyway, yes so now you solved your autoHangup make a new debug trace like the one you looked at before now which should be different. On Thu, Jul 23, 2009 at 9:23 AM, Lars Zeb wrote: > Thanks for the reply. This is my first attempt at using a script. > > > > I tried: > > > > session:autoHangup(0) or session:autoHangup(false) > > > > but got an error: > > > > 2009-07-23 07:19:09.652238 [ERR] mod_lua.cpp:182 > /usr/local/freeswitch/scripts/inbound_calls.lua:29: attempt to call method > 'autoHangup' (a nil value) > > stack traceback: > > /usr/local/freeswitch/scripts/inbound_calls.lua:29: in main chunk > > > > I looked at the documentation and tried: > > > > session:setAutoHangup(false) > > > > and the script proceeded without error. However, looking at the log, I do > not see the setAutoHangup being called. Also, when pressing *, I get a fast, > busy signal. > > > > I have pasted the script and log at http://pastebin.freeswitch.org/9836 > > > > Thanks again, Lars > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony > Minessale > *Sent:* Wednesday, July 22, 2009 4:33 PM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Asterisk key during message hangs up > call > > > > you are using a channel created with a script and you did not set > > js > session.autoHangup(0) > > lua > session:autoHangup(0) > > so when the * makes the call transfer the script kills the channel. > > On Wed, Jul 22, 2009 at 3:13 PM, Lars Zeb wrote: > > Brian, > > > > When calling into FreeSWITCH and pressing * during the greeting, the call > immediately hangs up. > > > > It used to ask for the mailbox number to retrieve messages. It no longer > works. I don?t know if my dialplan is causing the error or something in > FreeSWITCH has changed. > > > > Any ideas? > > > > http://pastebin.freeswitch.org/9803 > > > > Thanks, Lars > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > 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/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference 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/20090723/74a679fb/attachment-0002.html From rdenert at tng.de Thu Jul 23 08:59:42 2009 From: rdenert at tng.de (Rudolf Denert) Date: Thu, 23 Jul 2009 17:59:42 +0200 (CEST) Subject: [Freeswitch-users] Problem with Caller controls Message-ID: <13405311.137361248364782823.JavaMail.root@zimbra.tng.de> Hello, I?m getting always the message: [ERR] mod_conference.c:5463 conference_new() Unable to install caller controls group 'test' I made a new caller-controls in the conference.conf.xml. It has the name "test". I also implemented the line: param name="caller-controls" value="test" in my conference-profil. Did I forget something? Thanks for your help! Greetz From larclap at yahoo.com Thu Jul 23 11:12:07 2009 From: larclap at yahoo.com (Lars Zeb) Date: Thu, 23 Jul 2009 11:12:07 -0700 Subject: [Freeswitch-users] Asterisk key during message hangs up call In-Reply-To: <191c3a030907230741uc04b0cdoafcee3f79be55850@mail.gmail.com> References: <004a01c9da6f$c1e40120$45ac0360$@com> <00b501ca0b08$f178a520$d469ef60$@com> <191c3a030907221633k6594901dj22b9c7ded8af19e4@mail.gmail.com> <004c01ca0ba1$1bb6de90$53249bb0$@com> <191c3a030907230741uc04b0cdoafcee3f79be55850@mail.gmail.com> Message-ID: <00a501ca0bc1$16f180c0$44d48240$@com> Your message made me look at the documentation, which was helpful. http://pastebin.freeswitch.org/9838 When I press *, I get a busy signal. Please disregard the USER_NOT_REGISTERED error in the log; one of the endpoints I bridged to is off-line. Thanks, Lars From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Thursday, July 23, 2009 7:41 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Asterisk key during message hangs up call I meant to pick one based on whichever lang you were using not to literally write what i said. anyway, yes so now you solved your autoHangup make a new debug trace like the one you looked at before now which should be different. On Thu, Jul 23, 2009 at 9:23 AM, Lars Zeb wrote: Thanks for the reply. This is my first attempt at using a script. I tried: session:autoHangup(0) or session:autoHangup(false) but got an error: 2009-07-23 07:19:09.652238 [ERR] mod_lua.cpp:182 /usr/local/freeswitch/scripts/inbound_calls.lua:29: attempt to call method 'autoHangup' (a nil value) stack traceback: /usr/local/freeswitch/scripts/inbound_calls.lua:29: in main chunk I looked at the documentation and tried: session:setAutoHangup(false) and the script proceeded without error. However, looking at the log, I do not see the setAutoHangup being called. Also, when pressing *, I get a fast, busy signal. I have pasted the script and log at http://pastebin.freeswitch.org/9836 Thanks again, Lars From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Wednesday, July 22, 2009 4:33 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Asterisk key during message hangs up call you are using a channel created with a script and you did not set js session.autoHangup(0) lua session:autoHangup(0) so when the * makes the call transfer the script kills the channel. On Wed, Jul 22, 2009 at 3:13 PM, Lars Zeb wrote: Brian, When calling into FreeSWITCH and pressing * during the greeting, the call immediately hangs up. It used to ask for the mailbox number to retrieve messages. It no longer works. I don't know if my dialplan is causing the error or something in FreeSWITCH has changed. Any ideas? http://pastebin.freeswitch.org/9803 Thanks, Lars _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference 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/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference 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/20090723/48cd1586/attachment-0002.html From pjintheusa at gmail.com Thu Jul 23 11:13:52 2009 From: pjintheusa at gmail.com (Phillip Jones) Date: Thu, 23 Jul 2009 14:13:52 -0400 Subject: [Freeswitch-users] Barge on on prompts Message-ID: <367751820907231113p1a5d772eta359d9e25146dec1@mail.gmail.com> Hi there, Very simple scenario: Session.DtmfReceivedFunction = (d, t) => { Log.WriteLine(LogLevel.Info, "Received {0} for {1}.", d, t); CollectedDigits = d.ToString().Trim(); return ""; }; Session.flushDigits(); Session.StreamFile(VoicemailPromptsDirectory + "abigfile.wav", 0); Question is, it there a way to kill the streaming when the a digit is pressed? I would use the Session.PlayAndGetDigits() but that does not help when want to string things together like: Session.StreamFile(VoicemailPromptsDirectory + "vm-to_delete_the_message.wav", 0); Session.StreamFile(VoicemailPromptsDirectory + "vm-press.wav", 0); Session.Say("7", "en", "number", "pronounced"); Any help would be appreciated. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090723/6435af85/attachment-0002.html From msc at freeswitch.org Thu Jul 23 11:40:07 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 23 Jul 2009 11:40:07 -0700 Subject: [Freeswitch-users] Barge on on prompts In-Reply-To: <367751820907231113p1a5d772eta359d9e25146dec1@mail.gmail.com> References: <367751820907231113p1a5d772eta359d9e25146dec1@mail.gmail.com> Message-ID: <87f2f3b90907231140j24c6d2efhdf3316bee4d87d73@mail.gmail.com> I think you might want to check out phrase macros... http://wiki.freeswitch.org/wiki/Speech_Phrase_Management -MC On Thu, Jul 23, 2009 at 11:13 AM, Phillip Jones wrote: > Hi there, > > Very simple scenario: > > Session.DtmfReceivedFunction = (d, t) => > { > Log.WriteLine(LogLevel.Info, "Received {0} for {1}.", d, t); > CollectedDigits = d.ToString().Trim(); > return ""; > }; > > > Session.flushDigits(); > Session.StreamFile(VoicemailPromptsDirectory + "abigfile.wav", 0); > > Question is, it there a way to kill the streaming when the a digit is > pressed? > > I would use the Session.PlayAndGetDigits() > > but that does not help when want to string things together like: > > Session.StreamFile(VoicemailPromptsDirectory + > "vm-to_delete_the_message.wav", 0); > Session.StreamFile(VoicemailPromptsDirectory + "vm-press.wav", 0); > Session.Say("7", "en", "number", "pronounced"); > > Any help would be 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090723/0cec4539/attachment-0002.html From pjintheusa at gmail.com Thu Jul 23 12:39:02 2009 From: pjintheusa at gmail.com (Phillip Jones) Date: Thu, 23 Jul 2009 15:39:02 -0400 Subject: [Freeswitch-users] Barge on on prompts In-Reply-To: <87f2f3b90907231140j24c6d2efhdf3316bee4d87d73@mail.gmail.com> References: <367751820907231113p1a5d772eta359d9e25146dec1@mail.gmail.com> <87f2f3b90907231140j24c6d2efhdf3316bee4d87d73@mail.gmail.com> Message-ID: <367751820907231239m308a3ee2s4be1cefdadf8e192@mail.gmail.com> Hi there, Thanks for the reply. That information is extremely useful. Given the code below though - when if I press '1' when the phrase is playing - playing does not stop. It continues. I am looking for a method to barge in and collect & react to digits immediately. Session.DtmfReceivedFunction = (d, t) => { Log.WriteLine(LogLevel.Info, "Received {0} for {1}.", d, t); Session.StreamFile("", 0); CollectedDigits = d.ToString().Trim(); return ""; }; Session.SayPhrase("msgcount", "187346", "en"); Any ideas? I am sure I must be missing something simple. Thanks a lot. Phillip Jones On Thu, Jul 23, 2009 at 2:40 PM, Michael Collins wrote: > I think you might want to check out phrase macros... > http://wiki.freeswitch.org/wiki/Speech_Phrase_Management > -MC > > On Thu, Jul 23, 2009 at 11:13 AM, Phillip Jones wrote: > >> Hi there, >> >> Very simple scenario: >> >> Session.DtmfReceivedFunction = (d, t) => >> { >> Log.WriteLine(LogLevel.Info, "Received {0} for {1}.", d, t); >> CollectedDigits = d.ToString().Trim(); >> return ""; >> }; >> >> >> Session.flushDigits(); >> Session.StreamFile(VoicemailPromptsDirectory + "abigfile.wav", 0); >> >> Question is, it there a way to kill the streaming when the a digit is >> pressed? >> >> I would use the Session.PlayAndGetDigits() >> >> but that does not help when want to string things together like: >> >> Session.StreamFile(VoicemailPromptsDirectory + >> "vm-to_delete_the_message.wav", 0); >> Session.StreamFile(VoicemailPromptsDirectory + "vm-press.wav", 0); >> Session.Say("7", "en", "number", "pronounced"); >> >> Any help would be 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090723/fb2f57f6/attachment-0002.html From dave at 3c.co.uk Thu Jul 23 13:17:14 2009 From: dave at 3c.co.uk (David Knell) Date: Thu, 23 Jul 2009 17:17:14 -0300 Subject: [Freeswitch-users] Barge on on prompts In-Reply-To: <367751820907231239m308a3ee2s4be1cefdadf8e192@mail.gmail.com> References: <367751820907231113p1a5d772eta359d9e25146dec1@mail.gmail.com> <87f2f3b90907231140j24c6d2efhdf3316bee4d87d73@mail.gmail.com> <367751820907231239m308a3ee2s4be1cefdadf8e192@mail.gmail.com> Message-ID: <1248380234.16040.17.camel@dk-d820> Hi Phillip, You need to call FreeSWITCH's break function - I'd guess Session.Break(); might do it for you, but no guarantees. --Dave > Hi there, > > Thanks for the reply. That information is extremely useful. > > Given the code below though - when if I press '1' when the phrase is > playing - playing does not stop. It continues. I am looking for a > method to barge in and collect & react to digits immediately. > > > Session.DtmfReceivedFunction = (d, t) => > { > Log.WriteLine(LogLevel.Info, "Received {0} for > {1}.", d, t); > Session.StreamFile("", 0); > CollectedDigits = d.ToString().Trim(); > return ""; > > }; > > Session.SayPhrase("msgcount", "187346", "en"); > > > Any ideas? I am sure I must be missing something simple. > > Thanks a lot. > > > Phillip Jones > > > > > > On Thu, Jul 23, 2009 at 2:40 PM, Michael Collins > wrote: > I think you might want to check out phrase macros... > http://wiki.freeswitch.org/wiki/Speech_Phrase_Management > -MC > > > On Thu, Jul 23, 2009 at 11:13 AM, Phillip Jones > wrote: > > > Hi there, > > Very simple scenario: > > Session.DtmfReceivedFunction = (d, t) => > { > Log.WriteLine(LogLevel.Info, "Received {0} for > {1}.", d, t); > CollectedDigits = d.ToString().Trim(); > return ""; > }; > > > Session.flushDigits(); > Session.StreamFile(VoicemailPromptsDirectory + > "abigfile.wav", 0); > > Question is, it there a way to kill the streaming when > the a digit is pressed? > > I would use the Session.PlayAndGetDigits() > > but that does not help when want to string things > together like: > > Session.StreamFile(VoicemailPromptsDirectory + > "vm-to_delete_the_message.wav", 0); > Session.StreamFile(VoicemailPromptsDirectory + > "vm-press.wav", 0); > Session.Say("7", "en", "number", "pronounced"); > > Any help would be 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 > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- David Knell, Director, 3C Limited T: +44 20 3298 2000 E: dave at 3c.co.uk W: http://www.3c.co.uk From pjintheusa at gmail.com Thu Jul 23 14:12:24 2009 From: pjintheusa at gmail.com (Phillip Jones) Date: Thu, 23 Jul 2009 17:12:24 -0400 Subject: [Freeswitch-users] Barge on on prompts In-Reply-To: <1248380234.16040.17.camel@dk-d820> References: <367751820907231113p1a5d772eta359d9e25146dec1@mail.gmail.com> <87f2f3b90907231140j24c6d2efhdf3316bee4d87d73@mail.gmail.com> <367751820907231239m308a3ee2s4be1cefdadf8e192@mail.gmail.com> <1248380234.16040.17.camel@dk-d820> Message-ID: <367751820907231412u70560cbs92c1ad316bb5b552@mail.gmail.com> Ah! That you very much. Not Session.Break() but: Session.DtmfReceivedFunction = (d, t) => { Log.WriteLine(LogLevel.Info, "Received {0} for {1}.", d, t); CollectedDigits = d.ToString().Trim(); return "break"; } Thanks to you both for your help on this. On Thu, Jul 23, 2009 at 4:17 PM, David Knell wrote: > Hi Phillip, > > You need to call FreeSWITCH's break function - I'd guess > Session.Break(); might do it for you, but no guarantees. > > --Dave > > > Hi there, > > > > Thanks for the reply. That information is extremely useful. > > > > Given the code below though - when if I press '1' when the phrase is > > playing - playing does not stop. It continues. I am looking for a > > method to barge in and collect & react to digits immediately. > > > > > > Session.DtmfReceivedFunction = (d, t) => > > { > > Log.WriteLine(LogLevel.Info, "Received {0} for > > {1}.", d, t); > > Session.StreamFile("", 0); > > CollectedDigits = d.ToString().Trim(); > > return ""; > > > > }; > > > > Session.SayPhrase("msgcount", "187346", "en"); > > > > > > Any ideas? I am sure I must be missing something simple. > > > > Thanks a lot. > > > > > > Phillip Jones > > > > > > > > > > > > On Thu, Jul 23, 2009 at 2:40 PM, Michael Collins > > wrote: > > I think you might want to check out phrase macros... > > http://wiki.freeswitch.org/wiki/Speech_Phrase_Management > > -MC > > > > > > On Thu, Jul 23, 2009 at 11:13 AM, Phillip Jones > > wrote: > > > > > > Hi there, > > > > Very simple scenario: > > > > Session.DtmfReceivedFunction = (d, t) => > > { > > Log.WriteLine(LogLevel.Info, "Received {0} for > > {1}.", d, t); > > CollectedDigits = d.ToString().Trim(); > > return ""; > > }; > > > > > > Session.flushDigits(); > > Session.StreamFile(VoicemailPromptsDirectory + > > "abigfile.wav", 0); > > > > Question is, it there a way to kill the streaming when > > the a digit is pressed? > > > > I would use the Session.PlayAndGetDigits() > > > > but that does not help when want to string things > > together like: > > > > Session.StreamFile(VoicemailPromptsDirectory + > > "vm-to_delete_the_message.wav", 0); > > Session.StreamFile(VoicemailPromptsDirectory + > > "vm-press.wav", 0); > > Session.Say("7", "en", "number", "pronounced"); > > > > Any help would be 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 > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > -- > David Knell, Director, 3C Limited > T: +44 20 3298 2000 > E: dave at 3c.co.uk > W: 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090723/a7851342/attachment-0002.html From velu.technical at gmail.com Thu Jul 23 23:22:48 2009 From: velu.technical at gmail.com (velusamy velu) Date: Fri, 24 Jul 2009 11:52:48 +0530 Subject: [Freeswitch-users] A stun server lookup Message-ID: <1452e2980907232322h2229bd8bp25d50b73ed59fb7b@mail.gmail.com> Dear All, When I start the freeSWITCH, I am receiving the following errors, 2009-07-24 16:56:23 [ERR] sofia_glue.c:566 sofia_glue_ext_address_lookup() STUN Failed! stun.freeswitch.org:3478[Remote Address Error!] 2009-07-24 16:56:23 [ERR] sofia.c:1972 config_sofia() Failed to get external ip. I commented the stun configurations in vars.xml.conf file eventhough I am receiving the same error. Pleas any one give solution to solve this error.... Regards, Velusamy. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090724/96b49d8c/attachment-0002.html From jason at jasonjgw.net Fri Jul 24 00:12:41 2009 From: jason at jasonjgw.net (Jason White) Date: Fri, 24 Jul 2009 17:12:41 +1000 Subject: [Freeswitch-users] A stun server lookup In-Reply-To: <1452e2980907232322h2229bd8bp25d50b73ed59fb7b@mail.gmail.com> References: <1452e2980907232322h2229bd8bp25d50b73ed59fb7b@mail.gmail.com> Message-ID: <20090724071241.GA31649@jdc.jasonjgw.net> velusamy velu wrote: > I commented the stun configurations in vars.xml.conf file eventhough I > am receiving the same error. > > Pleas any one give solution to solve this error.... Edit vars.xml, change the variables that use Stun to be wahtever you want your ext-sip-ip and ext-rtp-ip addresses to be, then restart the external profile sofia profile external restart reloadxml or restart FreeSWITCH. From lzwierko at gmail.com Fri Jul 24 00:17:14 2009 From: lzwierko at gmail.com (=?ISO-8859-2?Q?=A3ukasz_Zwierko?=) Date: Fri, 24 Jul 2009 09:17:14 +0200 Subject: [Freeswitch-users] Dial up from confernece issue In-Reply-To: <122E4883-9AF7-4F7A-AAC0-2562ECBDBFDF@freeswitch.org> References: <4A637FC5.6000301@gmail.com> <41A88E3D-C420-4A1F-8B80-098EA18F9544@freeswitch.org> <4A661919.8030500@gmail.com> <122E4883-9AF7-4F7A-AAC0-2562ECBDBFDF@freeswitch.org> Message-ID: Ok Brian, you were right after all - I've had my X-lite incorrectly configured, sorry for wasting your time. thanks, LZ W dniu 22 lipca 2009 12:50 u?ytkownik Brian West napisa?: > The far end you're calling is sending a 302 can you check the sip > traffic please. > > /b > > On Jul 22, 2009, at 12:45 AM, ?ukasz Zwierko wrote: > >> I'm not sure how this exactly works, but I suppose that it is a single >> leg call, which upon answer would be attached to the conference (?) >> somehow. But again, this call does not originate outside FS so what >> would be the cause for 302? > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From hoaianh at gmx.de Fri Jul 24 06:45:29 2009 From: hoaianh at gmx.de (Ngo-Vi Hoai-Anh) Date: Fri, 24 Jul 2009 15:45:29 +0200 Subject: [Freeswitch-users] newbie question Message-ID: <4A69BAF9.2000408@gmx.de> Hi folk, I'm very new to FreeSwitch. I've read all the FAQs and traced the mailing list back to 12.2008 but still not found the answers for my questions. Please help! 1. Is it possible to make unauthenticated call to FS in the manner 1000@? 2. Is there already a java implementation for FS like http://asterisk-java.org/ for Asterisk? Thank you . Hoai-Anh From thangappan143 at gmail.com Thu Jul 23 22:42:17 2009 From: thangappan143 at gmail.com (Thangappan.M) Date: Fri, 24 Jul 2009 11:12:17 +0530 Subject: [Freeswitch-users] Problem in mod_perl Message-ID: <7aa29e790907232242v5f6db729p7b1158114fcfeb9@mail.gmail.com> I am new to Freeswitch, I started to write a dial plan using perl instead of xml in the case of IVR. I used the following statement in the dialplan/default.xml file I am using Twinkle Soft phone.When I am calling to 5000 in the freeswitch console it tells the following error. Invalid application perl. >From that I understood there is no Perl module has been installed.Then I uncommented the line from modules.conf.xml. Again I checked with my Perl version it also supports usemultiplicity. Where I made a mistake? Can anyone please solve my problem? I want to execute the Perl script in the dial plan. -- Regards, Thangappan.M -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090724/dedac132/attachment-0002.html From niall.crosby at gmail.com Fri Jul 24 07:29:51 2009 From: niall.crosby at gmail.com (Niall Crosby) Date: Fri, 24 Jul 2009 15:29:51 +0100 Subject: [Freeswitch-users] newbie question In-Reply-To: <4A69BAF9.2000408@gmx.de> References: <4A69BAF9.2000408@gmx.de> Message-ID: <4aec92830907240729l1892125fy127f17a42cd4bb9d@mail.gmail.com> Hi Hoai-Anh, a) Disable forced registration: In sip_profiles\internal.xml set auth-calls = false b) Enable calls from any IP: In sip_profiles\zinternal take out ) This is what I had to do to get SIPP working without registering first. I also program Java, the best I could find is the socket event interface. Hope this helps, Niall. 2009/7/24 Ngo-Vi Hoai-Anh : > Hi folk, > > I'm very new to FreeSwitch. I've read all the FAQs and traced the > mailing list back to 12.2008 but still not found the answers for my > questions. Please help! > > 1. Is it possible to make unauthenticated call to FS in the manner > 1000@? > 2. Is there already a java implementation for FS like > http://asterisk-java.org/ for Asterisk? > > Thank you . > Hoai-Anh > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- -- Sremium Ltd. Reg Number: 451937 Mobile: +353 (0)87 2393174 Web: www.sremium.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. Statements and opinions expressed in this e-mail may not represent those of Sremium. 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 immediately and delete the material from any computer. From hoaianh at gmx.de Fri Jul 24 08:02:41 2009 From: hoaianh at gmx.de (Ngo-Vi Hoai-Anh) Date: Fri, 24 Jul 2009 17:02:41 +0200 Subject: [Freeswitch-users] newbie question In-Reply-To: <4aec92830907240729l1892125fy127f17a42cd4bb9d@mail.gmail.com> References: <4A69BAF9.2000408@gmx.de> <4aec92830907240729l1892125fy127f17a42cd4bb9d@mail.gmail.com> Message-ID: <4A69CD11.5040905@gmx.de> Thank you Niall :-) Niall Crosby schrieb: > Hi Hoai-Anh, > > a) Disable forced registration: > In sip_profiles\internal.xml set auth-calls = false > > b) Enable calls from any IP: > In sip_profiles\zinternal take out value="domains"/>) > > This is what I had to do to get SIPP working without registering first. > > I also program Java, the best I could find is the socket event interface. > > Hope this helps, > Niall. > > > 2009/7/24 Ngo-Vi Hoai-Anh : > >> Hi folk, >> >> I'm very new to FreeSwitch. I've read all the FAQs and traced the >> mailing list back to 12.2008 but still not found the answers for my >> questions. Please help! >> >> 1. Is it possible to make unauthenticated call to FS in the manner >> 1000@? >> 2. Is there already a java implementation for FS like >> http://asterisk-java.org/ for Asterisk? >> >> Thank you . >> Hoai-Anh >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > > From shawn at sboyle.com Fri Jul 24 08:36:31 2009 From: shawn at sboyle.com (Shawn Boyle) Date: Fri, 24 Jul 2009 11:36:31 -0400 Subject: [Freeswitch-users] Problem in mod_perl In-Reply-To: <7aa29e790907232242v5f6db729p7b1158114fcfeb9@mail.gmail.com> Message-ID: Did you also uncomment the line: languages/mod_perl in modules.conf when you compiled FS? I believe it's commented out by default. [Something I personally disagree with...but I would bear Larry Wall's children if I could manage it physiologically.] -Shawn ________________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Thangappan.M Sent: Friday, July 24, 2009 1:42 AM To: FreeSWITCH-users at lists.freeswitch.org Subject: [Freeswitch-users] Problem in mod_perl Where I made a mistake? Can anyone please solve my problem? I want to execute the Perl script? in the dial plan. -- Regards, Thangappan.M From gshfreesw at gmail.com Fri Jul 24 08:45:15 2009 From: gshfreesw at gmail.com (Gu Sh) Date: Fri, 24 Jul 2009 11:45:15 -0400 Subject: [Freeswitch-users] IAX Transfer support Message-ID: <5070fcbd0907240845q246b80d6i5837c0d4f4451634@mail.gmail.com> I have been using freeswitch for over a year and I love all of the features, extensibility etc. Recently one of the clients wanted to use a IAX client and call from the IAX client works fine but there was one feature requested by my client that did not work. The feature is the "IAX Transfer" and I see the Transfer message come through by turning up debugging in the iax.conf file but freeswitch does not do anything with it. What is the current status of IAX support on freeswitch? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090724/57e2294b/attachment-0002.html From anthony.minessale at gmail.com Fri Jul 24 08:51:56 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 24 Jul 2009 10:51:56 -0500 Subject: [Freeswitch-users] Problem in mod_perl In-Reply-To: <7aa29e790907232242v5f6db729p7b1158114fcfeb9@mail.gmail.com> References: <7aa29e790907232242v5f6db729p7b1158114fcfeb9@mail.gmail.com> Message-ID: <191c3a030907240851t3bd46f9n65d86063cdc460a1@mail.gmail.com> edit modules.conf in th build root for FS uncomment the line that builds mod_perl issue "make mod_perl-install" from the shell edit /usr/local/freeswitch/conf/autoload_configs/modules.conf.xml and uncomment the mod_perl line. On Fri, Jul 24, 2009 at 12:42 AM, Thangappan.M wrote: > I am new to Freeswitch, I started to write a dial plan using perl instead > of xml in the case of IVR. > I used the following statement in the dialplan/default.xml file > > > > > > > I am using Twinkle Soft phone.When I am calling to 5000 in the freeswitch > console it tells the following error. > Invalid application perl. > > From that I understood there is no Perl module has been installed.Then I > uncommented the line from modules.conf.xml. Again > I checked with my Perl version it also supports > usemultiplicity. > > Where I made a mistake? > Can anyone please solve my problem? > I want to execute the Perl script in the dial plan. > > > -- > Regards, > Thangappan.M > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference 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/20090724/86356eaf/attachment-0002.html From anthony.minessale at gmail.com Fri Jul 24 08:58:58 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 24 Jul 2009 10:58:58 -0500 Subject: [Freeswitch-users] IAX Transfer support In-Reply-To: <5070fcbd0907240845q246b80d6i5837c0d4f4451634@mail.gmail.com> References: <5070fcbd0907240845q246b80d6i5837c0d4f4451634@mail.gmail.com> Message-ID: <191c3a030907240858g308f5e20x3d9173c1e9dd359f@mail.gmail.com> mod_iax was one of the first endpoint modules made in FS and it has seen little attention since it was first added. The primary purpose of the module was to have alternate endpoints to help work on the abstraction concepts in the core. Since it actually works, we left it in. But we really don't have any plans to do much else with it. The IAX lib we use in mod_iax is a heavily modified version of the iax client lib used in most iax softphones, its was really only designed for one or 2 calls max. An open source and liberally licensed (BSD/MIT) IAX library that is scalable and meets all of the programmatic challenges presented by the iax spec is really needed to advance the protocol any further. In addition a few enhancements should be made to the protocol to improve it's scalability in general. On Fri, Jul 24, 2009 at 10:45 AM, Gu Sh wrote: > I have been using freeswitch for over a year and I love all of the > features, extensibility etc. Recently one of the clients wanted to use a IAX > client and call from the IAX client works fine but there was one feature > requested by my client that did not work. The feature is the "IAX Transfer" > and I see the Transfer message come through by turning up debugging in the > iax.conf file but freeswitch does not do anything with it. > > What is the current status of IAX support on freeswitch? > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference 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/20090724/afd8c47e/attachment-0002.html From gregt at cgicommunications.com Fri Jul 24 08:59:41 2009 From: gregt at cgicommunications.com (Greg Thoen) Date: Fri, 24 Jul 2009 11:59:41 -0400 Subject: [Freeswitch-users] setInputCallback not working with Javascript? In-Reply-To: <554DE828-B96A-4E17-A974-151CAC50A0E5@freeswitch.org> References: <8CB7A629F5159D9-17A0-AA5@webmail-de13.sysops.aol.com> <554DE828-B96A-4E17-A974-151CAC50A0E5@freeswitch.org> Message-ID: Regarding the second part of his question, I am having a hard time stripping SpeechTools.jm into a very simple speech recognition example. I also cannot get collectInput to receive the type of "event", only "dtmf" -- Greg Thoen, Vice President CGI Communications, Inc. 1-585-427-0020 x260 On Mar 24, 2009, at 9:47 AM, Brian West wrote: > Javascript doesn't use the Core Session constructor. Its not the > same as the other languages. > > /b > > On Mar 24, 2009, at 1:40 AM, mszlazak at aol.com wrote: > >> I'm getting in build 12653M: >> >> [ERR] notify.js:130 mod_spidermonkey() TypeError: >> session.setInputCallback is not a function >> >> The wiki says this function should work in Javascript. >> >> http://wiki.freeswitch.org/wiki/ >> CoreSession_Constructor#session:setInputCallback >> >> Also, has there been changes to session.collectInput with >> type="event"? I get dtmf type events with my callback function but >> can't seem to get type="event" with speech events. >> >> 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090724/4c25d099/attachment-0002.html From intralanman at freeswitch.org Fri Jul 24 09:02:06 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Fri, 24 Jul 2009 12:02:06 -0400 Subject: [Freeswitch-users] IAX Transfer support In-Reply-To: <5070fcbd0907240845q246b80d6i5837c0d4f4451634@mail.gmail.com> References: <5070fcbd0907240845q246b80d6i5837c0d4f4451634@mail.gmail.com> Message-ID: <4A69DAFE.2050709@freeswitch.org> Gu Sh wrote: > > What is the current status of IAX support on freeswitch? Basically unsupported, the module still builds and sort of works, but there isn't much interest in maintaining it. It would definitely be a lot better for IAX lovers if someone wanted to take on maintaining it, or fund its maintenance. -Ray From mike at jerris.com Fri Jul 24 09:03:55 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 24 Jul 2009 12:03:55 -0400 Subject: [Freeswitch-users] Problem in mod_perl In-Reply-To: References: Message-ID: <2079FE7F-23F1-44A9-B2EA-0B69238E89EC@jerris.com> It is not built by default because it requires manual intervention to make sure you have a proper threadsafe perl and all its dev libs installed first. We work hard to make sure all default modules build out of the box with minimal external dependencies. Also, this module still does not work 100% on some platforms (solaris?) Mike On Jul 24, 2009, at 11:36 AM, Shawn Boyle wrote: > Did you also uncomment the line: > > languages/mod_perl > > in modules.conf when you compiled FS? I believe it's commented out > by default. [Something I personally disagree with...but I would bear > Larry Wall's children if I could manage it physiologically.] > > -Shawn From mike at jerris.com Fri Jul 24 09:05:16 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 24 Jul 2009 12:05:16 -0400 Subject: [Freeswitch-users] IAX Transfer support In-Reply-To: <5070fcbd0907240845q246b80d6i5837c0d4f4451634@mail.gmail.com> References: <5070fcbd0907240845q246b80d6i5837c0d4f4451634@mail.gmail.com> Message-ID: On Jul 24, 2009, at 11:45 AM, Gu Sh wrote: > I have been using freeswitch for over a year and I love all of the > features, extensibility etc. Recently one of the clients wanted to > use a IAX client and call from the IAX client works fine but there > was one feature requested by my client that did not work. The > feature is the "IAX Transfer" and I see the Transfer message come > through by turning up debugging in the iax.conf file but freeswitch > does not do anything with it. > > What is the current status of IAX support on freeswitch? > _______________________________________________ IAX transfer is not supported and no one has really touched that module in a couple years. If anyone is interested in enhancing that module we would be glad to accept patches to improve it but we have no roadmap plans to add anything to it. Mike From mattdfong at gmail.com Fri Jul 24 10:58:48 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Fri, 24 Jul 2009 10:58:48 -0700 Subject: [Freeswitch-users] Application to Record Calls - Out of Band Message-ID: <4256bf830907241058h67d1c798r9ddb56c8d2845611@mail.gmail.com> Hi, I'm trying to build an application that provides statistics of calls and call recording. Someone told me this could be done out of band with a SPAN (?) port that would replicate SIP and media packets to a separate NIC without having to actually pass the real-calls thru FreeSWITCH. It was explained that this SPAN port would in the SBC would replicate data received. If this is done, is there a way I can utilize FreeSWITCH to interpret these packets without actually having any control of the calls? If so how? Sorry, I'm new to telco, so hopefully this post makes sense to someone. --matt -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090724/ffbb04d2/attachment-0002.html From dome at tel.co.th Fri Jul 24 12:25:02 2009 From: dome at tel.co.th (Dome Charoenyost) Date: Sat, 25 Jul 2009 02:25:02 +0700 Subject: [Freeswitch-users] freeswitch-1.0.4pre10 ? Message-ID: <8ccbff060907241225m476078xb87dcbfa7c51e5f7@mail.gmail.com> I found freeswitch-1.0.4pre10 in http://files.freeswitch.org/freeswitch-1.0.4pre10.tar.bz2 But no news about this version Dome C. From dave at 3c.co.uk Fri Jul 24 14:56:09 2009 From: dave at 3c.co.uk (David Knell) Date: Fri, 24 Jul 2009 18:56:09 -0300 Subject: [Freeswitch-users] Application to Record Calls - Out of Band In-Reply-To: <4256bf830907241058h67d1c798r9ddb56c8d2845611@mail.gmail.com> References: <4256bf830907241058h67d1c798r9ddb56c8d2845611@mail.gmail.com> Message-ID: <1248472569.4360.6.camel@dk-d820> Hi Matt, FreeSWITCH probably isn't what you want. A quick Google for 'sip call sniffer' found this: http://www.enderunix.org/voipong/ which might well be a more appropriate starting point. A SPAN port is just a port on a network switch which has the traffic going to/from another port (or ports) replicated to it. Cheers -- Dave > Hi, > > > I'm trying to build an application that provides statistics of calls > and call recording. Someone told me this could be done out of band > with a SPAN (?) port that would replicate SIP and media packets to a > separate NIC without having to actually pass the real-calls thru > FreeSWITCH. It was explained that this SPAN port would in the SBC > would replicate data received. > > > If this is done, is there a way I can utilize FreeSWITCH to interpret > these packets without actually having any control of the calls? If so > how? Sorry, I'm new to telco, so hopefully this post makes sense to > someone. > > > --matt > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- David Knell, Director, 3C Limited T: +44 20 3298 2000 E: dave at 3c.co.uk W: http://www.3c.co.uk From thangappan143 at gmail.com Fri Jul 24 22:50:44 2009 From: thangappan143 at gmail.com (Thangappan.M) Date: Sat, 25 Jul 2009 11:20:44 +0530 Subject: [Freeswitch-users] Regarding IVR Message-ID: <7aa29e790907242250o152c539x16a4b25c1e2e53c0@mail.gmail.com> I am learning freeswitch for implementing IVR in this software. In our organization we are using Perl language.So I decided to implement a IVR in Perl on Freeswitch. What are the steps I need to do here. I am also new to IVR.Can you little bit explain about IVR (not basic) how to define a menu like that.. -- Regards, Thangappan.M -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090725/6e3e8b6b/attachment-0002.html From testeador01 at gmail.com Sat Jul 25 07:20:26 2009 From: testeador01 at gmail.com (Milena) Date: Sat, 25 Jul 2009 09:20:26 -0500 Subject: [Freeswitch-users] Command to get FreeSWITCH's internal time Message-ID: Hello everyone, I'm using the inbound event socket to receive some information about the status of my FreeSWITCH system and i wanted to know if there is an api command that can be used to get the FreeSWITCH time, I tried searching around in the docs and in google but i couldn't find an answer. Thanks for your attention and thanks in advance if anyone can assist me with this. Have a nice time and lots of cookies :) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090725/c1150850/attachment-0002.html From asannucci at gmail.com Sat Jul 25 07:40:23 2009 From: asannucci at gmail.com (bakko) Date: Sat, 25 Jul 2009 16:40:23 +0200 Subject: [Freeswitch-users] FS and Nokia E71 Message-ID: <3E9424AC46E1484E9AEEBA20FEA8C46A@voztovoice> Hi all, i have some problems tu make calls from a nokia E71 connect to FS. This is the scenario: Nokia E71 -> NAT - Internet - FS (public IP) I'm using the 5059 UDP port in FS. A can receive calls from others phones to nokia E71 but i can't make calls from nokia E71. If a change the UDP port to 5060 in vars.xml all work fine. I think the problem is the VoIP cllient in the nokia E71 but i dont?t know how resolve the problem. Thank's in advance. BR From talk2ram at gmail.com Sat Jul 25 08:12:35 2009 From: talk2ram at gmail.com (ram) Date: Sat, 25 Jul 2009 20:42:35 +0530 Subject: [Freeswitch-users] Regarding IVR In-Reply-To: <7aa29e790907242250o152c539x16a4b25c1e2e53c0@mail.gmail.com> References: <7aa29e790907242250o152c539x16a4b25c1e2e53c0@mail.gmail.com> Message-ID: On Sat, Jul 25, 2009 at 11:20 AM, Thangappan.M wrote: > I am learning freeswitch for implementing IVR in this software. > In our organization we are using Perl language.So I decided to implement a > IVR in Perl on Freeswitch. > What are the steps I need to do here. > > I am also new to IVR.Can you little bit explain about IVR (not basic) how > to define a menu like that.. > > Hi have you looked this examples http://wiki.freeswitch.org/wiki/SOHO_PBX_Example Ram -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090725/3aaade76/attachment-0002.html From msc at freeswitch.org Sat Jul 25 18:45:36 2009 From: msc at freeswitch.org (Michael S Collins) Date: Sat, 25 Jul 2009 18:45:36 -0700 Subject: [Freeswitch-users] Command to get FreeSWITCH's internal time In-Reply-To: References: Message-ID: How about this: strftime Try it at the CLI -MC Sent from my iPhone On Jul 25, 2009, at 7:20 AM, Milena wrote: > Hello everyone, > > I'm using the inbound event socket to receive some information about > the status of my FreeSWITCH system and i wanted to know if there is > an api command that can be used to get the FreeSWITCH time, I tried > searching around in the docs and in google but i couldn't find an > answer. Thanks for your attention and thanks in advance if anyone > can assist me with this. > > Have a nice time and lots of cookies :) > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mgg at giagnocavo.net Sun Jul 26 00:18:37 2009 From: mgg at giagnocavo.net (Michael Giagnocavo) Date: Sun, 26 Jul 2009 03:18:37 -0400 Subject: [Freeswitch-users] mod_managed users? In-Reply-To: References: <6E8D2069C08AA84A83D336E996AE4C67027C6C33F0@mse17be1.mse17.exchange.ms> <36860.21084.qm@web33505.mail.mud.yahoo.com> Message-ID: <6E8D2069C08AA84A83D336E996AE4C6702C479459E@mse17be1.mse17.exchange.ms> Hello, I just checked in a new mod_managed. It breaks backwards compatibility, but adds scripting and reloading support. I tested it on CentOS 5.3 x64 with Mono 2.4.2.2. Just make & make install seemed to take care of everything. Let me know if you have better luck with this version. Thanks, Michael From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Muhammad Shahzad Sent: Saturday, July 18, 2009 6:58 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_managed users? Sorry for replying late. I have tried mod_managed again on same machine (Lenovo 3000 N200), same OS (Ubuntu-9.04) and Mono version 2.0.1 but with latest FS revision 14249. It compiles correctly this time but gives following error upon "make install", ===================================================================== making install mod_managed make[5]: *** No rule to make target `/usr/local/freeswitch/mod/mod_managed.so', needed by `local_install'. Stop. make[4]: *** [install] Error 1 make[3]: *** [mod_managed-install] Error 1 make[2]: *** [install-recursive] Error 1 ===================================================================== Here is compilation log when executing "make", if it could of any help. ===================================================================== making all mod_managed Compiling freeswitch_managed.cpp... g++ -I/usr/src/svn-src/freeswitch/src/include -I/usr/src/svn-src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE -D_REENTRANT -pthread -I/usr/include/mono-1.0 -I/usr/include/glib-2.0 -I/usr/lib/glib-2.0/include -DHAVE_CONFIG_H -c -o freeswitch_managed.o freeswitch_managed.cpp Compiling freeswitch_wrap.cpp... g++ -I/usr/src/svn-src/freeswitch/src/include -I/usr/src/svn-src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE -D_REENTRANT -pthread -I/usr/include/mono-1.0 -I/usr/include/glib-2.0 -I/usr/lib/glib-2.0/include -DHAVE_CONFIG_H -c -o freeswitch_wrap.o freeswitch_wrap.cpp Demo.cs(58,14): warning CS0169: The private method `FreeSWITCH.Demo.AppDemo.hangupHook()' is never used Compilation succeeded - 1 warning(s) Compiling mod_managed.cpp... /usr/src/svn-src/freeswitch/libtool --mode=compile --tag=CXX g++ -I/usr/src/svn-src/freeswitch/src/include -I/usr/src/svn-src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE -D_REENTRANT -pthread -I/usr/include/mono-1.0 -I/usr/include/glib-2.0 -I/usr/lib/glib-2.0/include -DHAVE_CONFIG_H -c -o mod_managed.lo mod_managed.cpp libtool: compile: g++ -I/usr/src/svn-src/freeswitch/src/include -I/usr/src/svn-src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE -D_REENTRANT -pthread -I/usr/include/mono-1.0 -I/usr/include/glib-2.0 -I/usr/lib/glib-2.0/include -DHAVE_CONFIG_H -c mod_managed.cpp -fPIC -DPIC -o .libs/mod_managed.o mod_managed.cpp: In function ?void InitManagedSession(ManagedSession*, char* (*)(void*, switch_input_type_t), void (*)())?: mod_managed.cpp:97: warning: deprecated conversion from string constant to ?char*? libtool: compile: g++ -I/usr/src/svn-src/freeswitch/src/include -I/usr/src/svn-src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE -D_REENTRANT -pthread -I/usr/include/mono-1.0 -I/usr/include/glib-2.0 -I/usr/lib/glib-2.0/include -DHAVE_CONFIG_H -c mod_managed.cpp -o mod_managed.o >/dev/null 2>&1 Creating mod_managed.la... cat: .libs/mod_managed.log: No such file or directory ===================================================================== Thank you. On Fri, Jul 17, 2009 at 10:59 AM, Muhammad Shahzad > wrote: I tried to install mod_managed on ubuntu-9.04, mono framework version 2.0. It gave me a lots of errors in Loader.cs, which seems to be SWIG related. Since i am not a expert in SWIG so i disabled this module. This happend long ago, i think FS svn revision 136xx. Let me try to compile it from latest FS revision and see if it works. I will let you know the results. Thank you. On Fri, Jul 17, 2009 at 3:54 AM, Diego Toro > wrote: Hey, I am here :) I am working with mod_managed on Windows 2003 and Windows Vista with sucessfull. I noted on user list the issue with LoadFile on Loader.cs when a assembly had reference to others assemblies, I change LoadFile by LoadFrom and the load is made fine. I use c# application and sqlserver 2005, using FS and mod_managed. Diego --- On Thu, 7/16/09, Michael Giagnocavo > wrote: From: Michael Giagnocavo > Subject: [Freeswitch-users] mod_managed users? To: "freeswitch-users at lists.freeswitch.org" > Date: Thursday, July 16, 2009, 4:43 PM Hey, if there are any mod_managed users on this list, I?d love it if you were able to let me know. I?d like to get feedback, positive or negative, on what worked, what didn?t, and how mod_managed can improve for you. Feel free to write on list or directly to me: mgg at giagnocavo.net Thanks! -Michael -----Inline Attachment Follows----- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090726/593d5a26/attachment-0002.html From elihayun at gmail.com Sun Jul 26 00:29:58 2009 From: elihayun at gmail.com (Eli Hayun) Date: Sun, 26 Jul 2009 10:29:58 +0300 Subject: [Freeswitch-users] Limit is not working when originate a call In-Reply-To: <191c3a030907230539i5a640e2do4377d8797ff8f279@mail.gmail.com> References: <4A6709D4.6000704@savion.huji.ac.il> <2C85EB6C-1D80-4DC7-8D14-06B17CD1046D@jerris.com> <4A67EF47.6000402@savion.huji.ac.il> <191c3a030907230539i5a640e2do4377d8797ff8f279@mail.gmail.com> Message-ID: <4A6C05F6.4000501@savion.huji.ac.il> Anthony Minessale wrote: > limit is for inbound calls > you cannot call it after you already made the call. > The correct approach would be to not make the call at all. > > you could maybe use the limit FSAPI interface with apiExecute to check > if the limit was exceeded and > then not bother to place the call to begin with. > > otherwise it's sort of like putting a prisoner in the electric chair > then giving him his trial. > > > On Thu, Jul 23, 2009 at 12:04 AM, Eli Hayun > wrote: > > Michael Jerris wrote: > > because your not running limit at all when you are doing an > originate > > directly. You can use loopback to originate through a dialplan > > extension. > > > > Mike > > > > On Jul 22, 2009, at 8:45 AM, Eli Hayun wrote: > > > > > >> Hi > >> I set the limit to 1 on the extension like that > >> > >> > >> > >> When I am trying to make a call the that destination i > transfered to > >> limit_exceeded dialplan, just like I want > >> > >> The problem is, that when I am trying to make a call using > >> "originate" I > >> am not getting the limitation. > >> Why is 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 > >> > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > Thanks for answer. > I am calling "Originate" from JS. I tried to call "limit_hash" from JS > but with no success. I did it like that: > > lmt = apiExecute("limit_hash", dialed_ext + " " + dialed_ext + " 1"); > > I could't find any documentation on that. > can u help ? > > Thanks > Eli > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 Thanks for replay, but how do I do that? I tried to use : lmt = apiExecute("limit_hash", extno + " " + extno + " 1"); console_log("info","*** Limit ***" + lmt + "\n"); But it gave me "Invalid command". What is the exact way to do that. The documentation on that, is missing. From saeedahmad1981 at gmail.com Sun Jul 26 02:27:54 2009 From: saeedahmad1981 at gmail.com (Saeed Ahmad) Date: Sun, 26 Jul 2009 11:27:54 +0200 Subject: [Freeswitch-users] Mod_Limit - Limiting different destinations on same IP Message-ID: Dear All, Is it possible to apply different limits on a same IP by different destinations. Example: IP: 1.2.3.4 Destination: Germany Mobile (491) => max-channels=10 Destination: Germany (49) => max-channel=20 Sometimes the supplier provides limited capacity on different destinations; so in this case its necessary to apply the limit so that after limit exceeds the call can go to next endpoint. Thanks Saeed. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090726/f3ddd1b6/attachment-0002.html From krice at freeswitch.org Sun Jul 26 02:47:37 2009 From: krice at freeswitch.org (Ken Rice) Date: Sun, 26 Jul 2009 04:47:37 -0500 Subject: [Freeswitch-users] Mod_Limit - Limiting different destinations on same IP In-Reply-To: Message-ID: They don?t just 503 the call once you hit the limit? And look at the options for using limit... Be a little creative and you can do just want you want to do with some regex From: Saeed Ahmad Reply-To: Date: Sun, 26 Jul 2009 11:27:54 +0200 To: Subject: [Freeswitch-users] Mod_Limit - Limiting different destinations on same IP Dear All, Is it possible to apply different limits on a same IP by different destinations. Example: IP: 1.2.3.4 Destination: Germany??Mobile (491) =>?max-channels=10 Destination: Germany (49) => max-channel=20 ? Sometimes the supplier provides limited capacity on different destinations; so in this case its necessary to apply the limit so that after limit exceeds the call can go to next endpoint. ? Thanks Saeed. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at 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/20090726/a0ecb6d5/attachment-0002.html From saeedahmad1981 at gmail.com Sun Jul 26 03:33:06 2009 From: saeedahmad1981 at gmail.com (Saeed Ahmad) Date: Sun, 26 Jul 2009 12:33:06 +0200 Subject: [Freeswitch-users] Mod_Limit - Limiting different destinations on same IP In-Reply-To: References: Message-ID: Can mod limit also be used to apply limit on outbound supplier ip? i was unable to find a way to do that... - Saeed On Sun, Jul 26, 2009 at 11:47 AM, Ken Rice wrote: > They don?t just 503 the call once you hit the limit? > > And look at the options for using limit... Be a little creative and you can > do just want you want to do with some regex > > ------------------------------ > *From: *Saeed Ahmad > *Reply-To: * > *Date: *Sun, 26 Jul 2009 11:27:54 +0200 > *To: * > *Subject: *[Freeswitch-users] Mod_Limit - Limiting different destinations > on same IP > > > Dear All, > > Is it possible to apply different limits on a same IP by different > destinations. > > Example: > > IP: 1.2.3.4 > Destination: Germany Mobile (491) => max-channels=10 > Destination: Germany (49) => max-channel=20 > > Sometimes the supplier provides limited capacity on different destinations; > so in this case its necessary to apply the limit so that after limit exceeds > the call can go to next endpoint. > > Thanks > Saeed. > > ------------------------------ > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20090726/b2ebc6fe/attachment-0002.html From shaheryarkh at googlemail.com Sun Jul 26 05:19:43 2009 From: shaheryarkh at googlemail.com (Muhammad Shahzad) Date: Sun, 26 Jul 2009 17:19:43 +0500 Subject: [Freeswitch-users] A possible bug in FS causing Linux Kernel crash Message-ID: Hi, I am having random Linux Kernel crash problems while running FreeSWITCH as Skype to/from SIP gateway on one of our production servers. This machine is running CentOS 5.2, Kernel version 2.6.18-92.1.22.el5.centos.plusPAE with FS svn revision number 13754. At time of Kernel crash i could find following crash messages which point to some source code file in FS source tree. --------------------- Kernel Begin ------------------------ 3 Time(s): ======================= 3 Time(s): [] syscall_call+0x7/0xb 3 Time(s): [] sys_delete_module+0x192/0x1b8 3 Time(s): [] audit_syscall_entry+0x14b/0x17d 3 Time(s): [] remove_proc_entry+0x139/0x18c 3 Time(s): [] alsa_sound_exit+0xa/0x30 [snd] 3 Time(s): [] snd_info_done+0x46/0x49 [snd] 3 Time(s): BUG: warning at fs/proc/generic.c:732/remove_proc_entry() (Not tainted) 1 Time(s): snd-malloc: Memory leak? pages not freed = 1 ---------------------- Kernel End ------------------------- While the problem seems to arise from ALSA kernel module but it blames FS file fs/proc/generic.c:732 for this. The only FS module that is using ALSA is mod_skypiax but as far as i remember that module is using FS internal routines to allocate and de-allocate sound driver services for Skype client. Please suggest a solution. Thank you. -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090726/b0e5d401/attachment-0002.html From gmaruzz at celliax.org Sun Jul 26 05:37:39 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Sun, 26 Jul 2009 14:37:39 +0200 Subject: [Freeswitch-users] A possible bug in FS causing Linux Kernel crash In-Reply-To: References: Message-ID: <7b197bef0907260537n5b88a583g3a37139a64aa452f@mail.gmail.com> Ciao Muhammad, I've got many problems with ALSA drivers, including various kind of crashes. To make a looong story short, use the alsa_drivers version 1.0.20, they have not yet crashed on me. Also, if you want to test it, you can compile the customized snd-dummy driver you find in the svn code, it is a try to have much more efficiency bot in softirqs and context switches, allows for 64 Skype instances (128 subdevices), etc. it is to be compiled with alsa_drivers 1.0.20 too. Is my feeling (I mean, almost sure) they got spin_locking wrong in previous versions, and it crashes the kernel when you "really" use it (Skype clients have a demented usage of alsa). BTW, I'm in the process of revamp the code, fix the bugs and apply patches. Please, have a look at the new wiki page with lots of new content, I'll send a mail to the ML tomorrow :-) 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 Sun, Jul 26, 2009 at 2:19 PM, Muhammad Shahzad wrote: > Hi, > > I am having random Linux Kernel crash problems while running FreeSWITCH as > Skype to/from SIP gateway on one of our production servers. This machine is > running CentOS 5.2, Kernel version 2.6.18-92.1.22.el5.centos.plusPAE with FS > svn revision number 13754. > > At time of Kernel crash i could find following crash messages which point to > some source code file in FS source tree. > > ?--------------------- Kernel Begin ------------------------ > > > ?3 Time(s):? ======================= > ?3 Time(s):? [] syscall_call+0x7/0xb > ?3 Time(s):? [] sys_delete_module+0x192/0x1b8 > ?3 Time(s):? [] audit_syscall_entry+0x14b/0x17d > ?3 Time(s):? [] remove_proc_entry+0x139/0x18c > ?3 Time(s):? [] alsa_sound_exit+0xa/0x30 [snd] > ?3 Time(s):? [] snd_info_done+0x46/0x49 [snd] > ?3 Time(s): BUG: warning at fs/proc/generic.c:732/remove_proc_entry() (Not > tainted) > ?1 Time(s): snd-malloc: Memory leak?? pages not freed = 1 > > ?---------------------- Kernel End ------------------------- > > While the problem seems to arise from ALSA kernel module but it blames FS > file fs/proc/generic.c:732 for this. The only FS module that is using ALSA > is mod_skypiax but as far as i remember that module is using FS internal > routines to allocate and de-allocate sound driver services for Skype client. > > Please suggest a solution. > > Thank you. > > > -- > Muhammad Shahzad > ----------------------------------- > CISCO Rich Media Communication Specialist (CRMCS) > CISCO Certified Network Associate (CCNA) > Cell: +92 334 422 40 88 > MSN: shari_786pk at hotmail.com > Email: shaheryarkh at googlemail.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 gmaruzz at celliax.org Sun Jul 26 05:40:11 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Sun, 26 Jul 2009 14:40:11 +0200 Subject: [Freeswitch-users] A possible bug in FS causing Linux Kernel crash In-Reply-To: <7b197bef0907260537n5b88a583g3a37139a64aa452f@mail.gmail.com> References: <7b197bef0907260537n5b88a583g3a37139a64aa452f@mail.gmail.com> Message-ID: <7b197bef0907260540v434463ck5fc2943074a0e5bc@mail.gmail.com> Performance problems and other issues (eg crashes on ALSA drivers) has been reported for Skypiax on CentOS, albeit various users got good success on same CentOS. The section down below, "Extreme" Performances on Linux solves all problems for the user that got issues on CentOS. http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk#.22Extreme.22_Performances_on_Linux On Sun, Jul 26, 2009 at 2:37 PM, Giovanni Maruzzelli wrote: > Ciao Muhammad, > > I've got many problems with ALSA drivers, including various kind of crashes. > > To make a looong story short, use the alsa_drivers version 1.0.20, > they have not yet crashed on me. > > Also, if you want to test it, you can compile the customized snd-dummy > driver you find in the svn code, it is a try to have much more > efficiency bot in softirqs and context switches, allows for 64 Skype > instances (128 subdevices), etc. it is to be compiled with > alsa_drivers 1.0.20 too. > > Is my feeling (I mean, almost sure) they got spin_locking wrong in > previous versions, and it crashes the kernel when you "really" use it > (Skype clients have a demented usage of alsa). > > BTW, I'm in the process of revamp the code, fix the bugs and apply > patches. Please, have a look at the new wiki page with lots of new > content, I'll send a mail to the ML tomorrow :-) > > > 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 Sun, Jul 26, 2009 at 2:19 PM, Muhammad > Shahzad wrote: >> Hi, >> >> I am having random Linux Kernel crash problems while running FreeSWITCH as >> Skype to/from SIP gateway on one of our production servers. This machine is >> running CentOS 5.2, Kernel version 2.6.18-92.1.22.el5.centos.plusPAE with FS >> svn revision number 13754. >> >> At time of Kernel crash i could find following crash messages which point to >> some source code file in FS source tree. >> >> ?--------------------- Kernel Begin ------------------------ >> >> >> ?3 Time(s):? ======================= >> ?3 Time(s):? [] syscall_call+0x7/0xb >> ?3 Time(s):? [] sys_delete_module+0x192/0x1b8 >> ?3 Time(s):? [] audit_syscall_entry+0x14b/0x17d >> ?3 Time(s):? [] remove_proc_entry+0x139/0x18c >> ?3 Time(s):? [] alsa_sound_exit+0xa/0x30 [snd] >> ?3 Time(s):? [] snd_info_done+0x46/0x49 [snd] >> ?3 Time(s): BUG: warning at fs/proc/generic.c:732/remove_proc_entry() (Not >> tainted) >> ?1 Time(s): snd-malloc: Memory leak?? pages not freed = 1 >> >> ?---------------------- Kernel End ------------------------- >> >> While the problem seems to arise from ALSA kernel module but it blames FS >> file fs/proc/generic.c:732 for this. The only FS module that is using ALSA >> is mod_skypiax but as far as i remember that module is using FS internal >> routines to allocate and de-allocate sound driver services for Skype client. >> >> Please suggest a solution. >> >> Thank you. >> >> >> -- >> Muhammad Shahzad >> ----------------------------------- >> CISCO Rich Media Communication Specialist (CRMCS) >> CISCO Certified Network Associate (CCNA) >> Cell: +92 334 422 40 88 >> MSN: shari_786pk at hotmail.com >> Email: shaheryarkh at googlemail.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 shaheryarkh at googlemail.com Sun Jul 26 05:44:02 2009 From: shaheryarkh at googlemail.com (Muhammad Shahzad) Date: Sun, 26 Jul 2009 17:44:02 +0500 Subject: [Freeswitch-users] A possible bug in FS causing Linux Kernel crash In-Reply-To: <7b197bef0907260540v434463ck5fc2943074a0e5bc@mail.gmail.com> References: <7b197bef0907260537n5b88a583g3a37139a64aa452f@mail.gmail.com> <7b197bef0907260540v434463ck5fc2943074a0e5bc@mail.gmail.com> Message-ID: Thanks. Let me try it and let you know the results. Thank you. On Sun, Jul 26, 2009 at 5:40 PM, Giovanni Maruzzelli wrote: > Performance problems and other issues (eg crashes on ALSA drivers) has > been reported for Skypiax on CentOS, albeit various users got good > success on same CentOS. The section down below, "Extreme" Performances > on Linux solves all problems for the user that got issues on CentOS. > > > http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk#.22Extreme.22_Performances_on_Linux > > > > On Sun, Jul 26, 2009 at 2:37 PM, Giovanni Maruzzelli > wrote: > > Ciao Muhammad, > > > > I've got many problems with ALSA drivers, including various kind of > crashes. > > > > To make a looong story short, use the alsa_drivers version 1.0.20, > > they have not yet crashed on me. > > > > Also, if you want to test it, you can compile the customized snd-dummy > > driver you find in the svn code, it is a try to have much more > > efficiency bot in softirqs and context switches, allows for 64 Skype > > instances (128 subdevices), etc. it is to be compiled with > > alsa_drivers 1.0.20 too. > > > > Is my feeling (I mean, almost sure) they got spin_locking wrong in > > previous versions, and it crashes the kernel when you "really" use it > > (Skype clients have a demented usage of alsa). > > > > BTW, I'm in the process of revamp the code, fix the bugs and apply > > patches. Please, have a look at the new wiki page with lots of new > > content, I'll send a mail to the ML tomorrow :-) > > > > > > 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 Sun, Jul 26, 2009 at 2:19 PM, Muhammad > > Shahzad wrote: > >> Hi, > >> > >> I am having random Linux Kernel crash problems while running FreeSWITCH > as > >> Skype to/from SIP gateway on one of our production servers. This machine > is > >> running CentOS 5.2, Kernel version 2.6.18-92.1.22.el5.centos.plusPAE > with FS > >> svn revision number 13754. > >> > >> At time of Kernel crash i could find following crash messages which > point to > >> some source code file in FS source tree. > >> > >> --------------------- Kernel Begin ------------------------ > >> > >> > >> 3 Time(s): ======================= > >> 3 Time(s): [] syscall_call+0x7/0xb > >> 3 Time(s): [] sys_delete_module+0x192/0x1b8 > >> 3 Time(s): [] audit_syscall_entry+0x14b/0x17d > >> 3 Time(s): [] remove_proc_entry+0x139/0x18c > >> 3 Time(s): [] alsa_sound_exit+0xa/0x30 [snd] > >> 3 Time(s): [] snd_info_done+0x46/0x49 [snd] > >> 3 Time(s): BUG: warning at fs/proc/generic.c:732/remove_proc_entry() > (Not > >> tainted) > >> 1 Time(s): snd-malloc: Memory leak? pages not freed = 1 > >> > >> ---------------------- Kernel End ------------------------- > >> > >> While the problem seems to arise from ALSA kernel module but it blames > FS > >> file fs/proc/generic.c:732 for this. The only FS module that is using > ALSA > >> is mod_skypiax but as far as i remember that module is using FS internal > >> routines to allocate and de-allocate sound driver services for Skype > client. > >> > >> Please suggest a solution. > >> > >> Thank you. > >> > >> > >> -- > >> Muhammad Shahzad > >> ----------------------------------- > >> CISCO Rich Media Communication Specialist (CRMCS) > >> CISCO Certified Network Associate (CCNA) > >> Cell: +92 334 422 40 88 > >> MSN: shari_786pk at hotmail.com > >> Email: shaheryarkh at googlemail.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 > -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090726/22a45ab5/attachment-0002.html From gmaruzz at celliax.org Sun Jul 26 06:26:51 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Sun, 26 Jul 2009 15:26:51 +0200 Subject: [Freeswitch-users] A possible bug in FS causing Linux Kernel crash In-Reply-To: References: Message-ID: <7b197bef0907260626u5e273b6bj235a7edb3fdfd949@mail.gmail.com> On Sun, Jul 26, 2009 at 2:19 PM, Muhammad Shahzad wrote: > Hi, > > I am having random Linux Kernel crash problems while running FreeSWITCH as > Skype to/from SIP gateway on one of our production servers. This machine is > running CentOS 5.2, Kernel version 2.6.18-92.1.22.el5.centos.plusPAE with FS > svn revision number 13754. > > While the problem seems to arise from ALSA kernel module but it blames FS > file fs/proc/generic.c:732 for this. The only FS module that is using ALSA > is mod_skypiax but as far as i remember that module is using FS internal > routines to allocate and de-allocate sound driver services for Skype client. Also, please note that neither mod_skypiax nor FreeSWITCH have nothing to do with ALSA (eg: no ALSA code at all in mod_skypiax or FreeSWITCH). Is the Skype client instance that uses the sound driver, just like on a desktop Skype client usage The Skype client instances are started by a shell script, but you could as well start them from the command line, and are completely autonomous from FreeSWITCH (FS do not allocate or deallocate sound driver services for them). Summary: it's just the ALSA drivers that are to blame :-) -giovanni From dftoro at yahoo.com Sun Jul 26 07:47:28 2009 From: dftoro at yahoo.com (Diego Toro) Date: Sun, 26 Jul 2009 07:47:28 -0700 (PDT) Subject: [Freeswitch-users] mod_managed users? In-Reply-To: <6E8D2069C08AA84A83D336E996AE4C6702C479459E@mse17be1.mse17.exchange.ms> Message-ID: <250325.39922.qm@web33504.mail.mud.yahoo.com> Hi Michael, ? Thank you for your job with mod_managed, I get lastest version with mod_managed but the files PluginInterfaces.cs, PluginManager.cs and ScriptPluginManager.cs were not downloaded. ? Diego --- On Sun, 7/26/09, Michael Giagnocavo wrote: From: Michael Giagnocavo Subject: Re: [Freeswitch-users] mod_managed users? To: "freeswitch-users at lists.freeswitch.org" Date: Sunday, July 26, 2009, 2:18 AM Hello, ? ??????????????? I just checked in a new mod_managed. It breaks backwards compatibility, but adds scripting and reloading support. ? ??????????????? I tested it on CentOS 5.3 x64 with Mono 2.4.2.2. Just make & make install seemed to take care of everything. ? ??????????????? Let me know if you have better luck with this version. ? Thanks, Michael ? From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Muhammad Shahzad Sent: Saturday, July 18, 2009 6:58 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_managed users? ? Sorry for replying late. I have tried mod_managed again on same machine (Lenovo 3000 N200), same OS (Ubuntu-9.04) and Mono version 2.0.1 but with latest FS revision 14249. It compiles correctly this time but gives following error upon "make install", ===================================================================== making install mod_managed make[5]: *** No rule to make target `/usr/local/freeswitch/mod/mod_managed.so', needed by `local_install'.? Stop. make[4]: *** [install] Error 1 make[3]: *** [mod_managed-install] Error 1 make[2]: *** [install-recursive] Error 1 ===================================================================== Here is compilation log when executing "make", if it could of any help. ===================================================================== making all mod_managed Compiling freeswitch_managed.cpp... g++ -I/usr/src/svn-src/freeswitch/src/include -I/usr/src/svn-src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE -D_REENTRANT -pthread -I/usr/include/mono-1.0 -I/usr/include/glib-2.0 -I/usr/lib/glib-2.0/include -DHAVE_CONFIG_H -c -o freeswitch_managed.o freeswitch_managed.cpp Compiling freeswitch_wrap.cpp... g++ -I/usr/src/svn-src/freeswitch/src/include -I/usr/src/svn-src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE -D_REENTRANT -pthread -I/usr/include/mono-1.0 -I/usr/include/glib-2.0 -I/usr/lib/glib-2.0/include -DHAVE_CONFIG_H -c -o freeswitch_wrap.o freeswitch_wrap.cpp Demo.cs(58,14): warning CS0169: The private method `FreeSWITCH.Demo.AppDemo.hangupHook()' is never used Compilation succeeded - 1 warning(s) Compiling mod_managed.cpp... /usr/src/svn-src/freeswitch/libtool --mode=compile --tag=CXX g++ -I/usr/src/svn-src/freeswitch/src/include -I/usr/src/svn-src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE -D_REENTRANT -pthread -I/usr/include/mono-1.0 -I/usr/include/glib-2.0 -I/usr/lib/glib-2.0/include -DHAVE_CONFIG_H -c -o mod_managed.lo mod_managed.cpp libtool: compile:? g++ -I/usr/src/svn-src/freeswitch/src/include -I/usr/src/svn-src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE -D_REENTRANT -pthread -I/usr/include/mono-1.0 -I/usr/include/glib-2.0 -I/usr/lib/glib-2.0/include -DHAVE_CONFIG_H -c mod_managed.cpp? -fPIC -DPIC -o .libs/mod_managed.o mod_managed.cpp: In function ?void InitManagedSession(ManagedSession*, char* (*)(void*, switch_input_type_t), void (*)())?: mod_managed.cpp:97: warning: deprecated conversion from string constant to ?char*? libtool: compile:? g++ -I/usr/src/svn-src/freeswitch/src/include -I/usr/src/svn-src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE -D_REENTRANT -pthread -I/usr/include/mono-1.0 -I/usr/include/glib-2.0 -I/usr/lib/glib-2.0/include -DHAVE_CONFIG_H -c mod_managed.cpp -o mod_managed.o >/dev/null 2>&1 Creating mod_managed.la... cat: .libs/mod_managed.log: No such file or directory ===================================================================== Thank you. On Fri, Jul 17, 2009 at 10:59 AM, Muhammad Shahzad wrote: I tried to install mod_managed on ubuntu-9.04, mono framework version 2.0. It gave me a lots of errors in Loader.cs, which seems to be SWIG related. Since i am not a expert in SWIG so i disabled this module. This happend long ago, i think FS svn revision 136xx. Let me try to compile it from latest FS revision and see if it works. I will let you know the results. Thank you. On Fri, Jul 17, 2009 at 3:54 AM, Diego Toro wrote: Hey, I am here? :) ? I am working with mod_managed on Windows 2003 and Windows Vista with sucessfull.? I noted on user list the issue with LoadFile on Loader.cs when a assembly had reference to others assemblies, I change LoadFile by LoadFrom and the load is made fine. ? I use c# application and sqlserver 2005, using FS and mod_managed. ? Diego --- On Thu, 7/16/09, Michael Giagnocavo wrote: From: Michael Giagnocavo Subject: [Freeswitch-users] mod_managed users? To: "freeswitch-users at lists.freeswitch.org" Date: Thursday, July 16, 2009, 4:43 PM ? Hey, if there are any mod_managed users on this list, I?d love it if you were able to let me know. I?d like to get feedback, positive or negative, on what worked, what didn?t, and how mod_managed can improve for you. Feel free to write on list or directly to me: mgg at giagnocavo.net ? Thanks! -Michael -----Inline Attachment Follows----- ? _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -----Inline Attachment Follows----- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at 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/20090726/92aee1cf/attachment-0002.html From raul at etellicom.com Sun Jul 26 09:25:01 2009 From: raul at etellicom.com (Raul Fragoso) Date: Sun, 26 Jul 2009 13:25:01 -0300 Subject: [Freeswitch-users] Command to get FreeSWITCH's internal time In-Reply-To: References: Message-ID: <1248625501.5655.79.camel@raul-laptop> Every event has a header 'Event-Date-Local', which has the local date and time. If want to actively retrieve the date and time, you can send this API request to the server: api strftime Regards, Raul On Sat, 2009-07-25 at 09:20 -0500, Milena wrote: > Hello everyone, > > I'm using the inbound event socket to receive some information about > the status of my FreeSWITCH system and i wanted to know if there is an > api command that can be used to get the FreeSWITCH time, I tried > searching around in the docs and in google but i couldn't find an > answer. Thanks for your attention and thanks in advance if anyone can > assist me with this. > > Have a nice time and lots of cookies :) > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mgg at giagnocavo.net Sun Jul 26 09:37:05 2009 From: mgg at giagnocavo.net (Michael Giagnocavo) Date: Sun, 26 Jul 2009 12:37:05 -0400 Subject: [Freeswitch-users] mod_managed users? In-Reply-To: <250325.39922.qm@web33504.mail.mud.yahoo.com> References: <6E8D2069C08AA84A83D336E996AE4C6702C479459E@mse17be1.mse17.exchange.ms> <250325.39922.qm@web33504.mail.mud.yahoo.com> Message-ID: <6E8D2069C08AA84A83D336E996AE4C6702C47945B3@mse17be1.mse17.exchange.ms> Ah, that?s embarrassing. I added them and tried building FreeSWITCH.Managed from svn and it worked fine now. (I?ll kick off a new complete build in a minute.) -Michael From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Diego Toro Sent: Sunday, July 26, 2009 8:47 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_managed users? Hi Michael, Thank you for your job with mod_managed, I get lastest version with mod_managed but the files PluginInterfaces.cs, PluginManager.cs and ScriptPluginManager.cs were not downloaded. Diego --- On Sun, 7/26/09, Michael Giagnocavo wrote: From: Michael Giagnocavo Subject: Re: [Freeswitch-users] mod_managed users? To: "freeswitch-users at lists.freeswitch.org" Date: Sunday, July 26, 2009, 2:18 AM Hello, I just checked in a new mod_managed. It breaks backwards compatibility, but adds scripting and reloading support. I tested it on CentOS 5.3 x64 with Mono 2.4.2.2. Just make & make install seemed to take care of everything. Let me know if you have better luck with this version. Thanks, Michael From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Muhammad Shahzad Sent: Saturday, July 18, 2009 6:58 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_managed users? Sorry for replying late. I have tried mod_managed again on same machine (Lenovo 3000 N200), same OS (Ubuntu-9.04) and Mono version 2.0.1 but with latest FS revision 14249. It compiles correctly this time but gives following error upon "make install", ===================================================================== making install mod_managed make[5]: *** No rule to make target `/usr/local/freeswitch/mod/mod_managed.so', needed by `local_install'. Stop. make[4]: *** [install] Error 1 make[3]: *** [mod_managed-install] Error 1 make[2]: *** [install-recursive] Error 1 ===================================================================== Here is compilation log when executing "make", if it could of any help. ===================================================================== making all mod_managed Compiling freeswitch_managed.cpp... g++ -I/usr/src/svn-src/freeswitch/src/include -I/usr/src/svn-src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE -D_REENTRANT -pthread -I/usr/include/mono-1.0 -I/usr/include/glib-2.0 -I/usr/lib/glib-2.0/include -DHAVE_CONFIG_H -c -o freeswitch_managed.o freeswitch_managed.cpp Compiling freeswitch_wrap.cpp... g++ -I/usr/src/svn-src/freeswitch/src/include -I/usr/src/svn-src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE -D_REENTRANT -pthread -I/usr/include/mono-1.0 -I/usr/include/glib-2.0 -I/usr/lib/glib-2.0/include -DHAVE_CONFIG_H -c -o freeswitch_wrap.o freeswitch_wrap.cpp Demo.cs(58,14): warning CS0169: The private method `FreeSWITCH.Demo.AppDemo.hangupHook()' is never used Compilation succeeded - 1 warning(s) Compiling mod_managed.cpp... /usr/src/svn-src/freeswitch/libtool --mode=compile --tag=CXX g++ -I/usr/src/svn-src/freeswitch/src/include -I/usr/src/svn-src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE -D_REENTRANT -pthread -I/usr/include/mono-1.0 -I/usr/include/glib-2.0 -I/usr/lib/glib-2.0/include -DHAVE_CONFIG_H -c -o mod_managed.lo mod_managed.cpp libtool: compile: g++ -I/usr/src/svn-src/freeswitch/src/include -I/usr/src/svn-src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE -D_REENTRANT -pthread -I/usr/include/mono-1.0 -I/usr/include/glib-2.0 -I/usr/lib/glib-2.0/include -DHAVE_CONFIG_H -c mod_managed.cpp -fPIC -DPIC -o .libs/mod_managed.o mod_managed.cpp: In function ?void InitManagedSession(ManagedSession*, char* (*)(void*, switch_input_type_t), void (*)())?: mod_managed.cpp:97: warning: deprecated conversion from string constant to ?char*? libtool: compile: g++ -I/usr/src/svn-src/freeswitch/src/include -I/usr/src/svn-src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE -D_REENTRANT -pthread -I/usr/include/mono-1.0 -I/usr/include/glib-2.0 -I/usr/lib/glib-2.0/include -DHAVE_CONFIG_H -c mod_managed.cpp -o mod_managed.o >/dev/null 2>&1 Creating mod_managed.la... cat: .libs/mod_managed.log: No such file or directory ===================================================================== Thank you. On Fri, Jul 17, 2009 at 10:59 AM, Muhammad Shahzad > wrote: I tried to install mod_managed on ubuntu-9.04, mono framework version 2.0. It gave me a lots of errors in Loader.cs, which seems to be SWIG related. Since i am not a expert in SWIG so i disabled this module. This happend long ago, i think FS svn revision 136xx. Let me try to compile it from latest FS revision and see if it works. I will let you know the results. Thank you. On Fri, Jul 17, 2009 at 3:54 AM, Diego Toro > wrote: Hey, I am here :) I am working with mod_managed on Windows 2003 and Windows Vista with sucessfull. I noted on user list the issue with LoadFile on Loader.cs when a assembly had reference to others assemblies, I change LoadFile by LoadFrom and the load is made fine. I use c# application and sqlserver 2005, using FS and mod_managed. Diego --- On Thu, 7/16/09, Michael Giagnocavo > wrote: From: Michael Giagnocavo > Subject: [Freeswitch-users] mod_managed users? To: "freeswitch-users at lists.freeswitch.org" > Date: Thursday, July 16, 2009, 4:43 PM Hey, if there are any mod_managed users on this list, I?d love it if you were able to let me know. I?d like to get feedback, positive or negative, on what worked, what didn?t, and how mod_managed can improve for you. Feel free to write on list or directly to me: mgg at giagnocavo.net Thanks! -Michael -----Inline Attachment Follows----- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -----Inline Attachment Follows----- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at 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/20090726/1f1e9e80/attachment-0002.html From a.afzali2003 at gmail.com Sun Jul 26 09:00:21 2009 From: a.afzali2003 at gmail.com (afshin afzali) Date: Sun, 26 Jul 2009 20:30:21 +0430 Subject: [Freeswitch-users] Sofia SIP Subscription For External Events Message-ID: Hi Guys, I'm going to use OpenSER as SIP Platform (Registrar, Proxy, Presence) and FreeSWITCH in my application. So I need to subscribe the FreeSWITCH for presence information of users who involve in the application. After some looking of Sofia.conf.xml , it seems there is not support to doing so, Is it right ? appreciate all comments, -- afshin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090726/fc91d5f1/attachment-0002.html From thangappan143 at gmail.com Sun Jul 26 22:07:10 2009 From: thangappan143 at gmail.com (Thangappan.M) Date: Mon, 27 Jul 2009 10:37:10 +0530 Subject: [Freeswitch-users] Which method Can I use in IVR Message-ID: <7aa29e790907262207n5d50c96cl2f19e2d152399db6@mail.gmail.com> Dear all, I am learning how to implement a IVR in Freeswitch.In our organization we are using Perl scripting language for doing this.So In freeswitch also I need to use Perl. So far I heard two methods for executing IVR. One is in dial plan using perl application.( In perl I create IVR menu and play the voice files) Another one is using event socket.In dial plan I specified socket application and write a Perl script which is listening that particular port and get the session Id. Have I understood correctly?.If it is correct means tell which method can I use?. Other make me understand well. I have seen downloaded perl IVR menu from freeswitch site.In that they called some internal functions like playandGetDigits,StreamFile,ready ...etc. These functions is been called by using $session variable.Where these functions are defined.? -- Regards, Thangappan.M -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090727/bf394aa0/attachment-0002.html From velu.technical at gmail.com Sun Jul 26 22:23:19 2009 From: velu.technical at gmail.com (velusamy velu) Date: Mon, 27 Jul 2009 10:53:19 +0530 Subject: [Freeswitch-users] IAX configurations Message-ID: <1452e2980907262223u677eef21o282f39e2a56209cc@mail.gmail.com> Dear All, I have tried to call a Asterisk extension from FreeSWITCH. I have done the following configurations, * I have enabled mod_iax module in modules.conf.xml file. * Next I have configure following extension in dialplan. * Next I have configured a 222 user in sip.conf file at Asterisk machine. * I wrote dialplan for that extension in extension.conf file. When I tried to call 222 from FreeSWITCH, I have received following error in Console. "[ERR] switch_core_session.c:255 switch_core_session_outgoing_channel() Could not locate channel type iax" What would be the problem? Is there any configuration I missed? Please help me ..... Regards, K.Velusamy. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090727/9d12c4ca/attachment-0002.html From mike at jerris.com Sun Jul 26 22:41:58 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 27 Jul 2009 01:41:58 -0400 Subject: [Freeswitch-users] IAX configurations In-Reply-To: <1452e2980907262223u677eef21o282f39e2a56209cc@mail.gmail.com> References: <1452e2980907262223u677eef21o282f39e2a56209cc@mail.gmail.com> Message-ID: <98E444F7-ECF8-40DF-82D2-89A48DC11EAC@jerris.com> mod_iax isn't loaded. I suggest using sip anyways. Mike On Jul 27, 2009, at 1:23 AM, velusamy velu wrote: > Dear All, > I have tried to call a Asterisk extension from FreeSWITCH. I > have done the following configurations, > * I have enabled mod_iax module in > modules.conf.xml file. > * Next I have configure following > extension in dialplan. > > field="destination_number" expression="^(222)$"> > application="bridge" data="iax/222:222 at 192.168.6.94/$1"/> > > > * Next I have configured a 222 user in > sip.conf file at Asterisk machine. > * I wrote dialplan for that extension in > extension.conf file. > > When I tried to call 222 from FreeSWITCH, I have received > following error in Console. > "[ERR] switch_core_session.c:255 > switch_core_session_outgoing_channel() Could not locate channel type > iax" > > What would be the problem? Is there any configuration I > missed? Please help me ..... > > Regards, > K.Velusamy. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20090727/e83ccb54/attachment-0002.html From velu.technical at gmail.com Sun Jul 26 22:59:47 2009 From: velu.technical at gmail.com (velusamy velu) Date: Mon, 27 Jul 2009 11:29:47 +0530 Subject: [Freeswitch-users] IAX configurations In-Reply-To: <98E444F7-ECF8-40DF-82D2-89A48DC11EAC@jerris.com> References: <1452e2980907262223u677eef21o282f39e2a56209cc@mail.gmail.com> <98E444F7-ECF8-40DF-82D2-89A48DC11EAC@jerris.com> Message-ID: <1452e2980907262259k30713f4ft363592ac77ee3e33@mail.gmail.com> I have loaded mod_iax now that error didn't come. But, When I call I have received following message in the console. "[INFO] mod_dptools.c:1998 audio_bridge_function() Originate Failed. Cause: FACILITY_REJECTED" What configuration I missed? How to use sip to connect the Asterisk? Please give solutions above questions... __ Velusamy On Mon, Jul 27, 2009 at 11:11 AM, Michael Jerris wrote: > mod_iax isn't loaded. I suggest using sip anyways. > Mike > > On Jul 27, 2009, at 1:23 AM, velusamy velu wrote: > > Dear All, > I have tried to call a Asterisk extension from FreeSWITCH. I have done > the following configurations, > * I have enabled mod_iax module in > modules.conf.xml file. > * Next I have configure following extension in > dialplan. > > field="destination_number" expression="^(222)$"> > data="iax/222:222 at 192.168.6.94/$1"/> > > > * Next I have configured a 222 user in sip.conf > file at Asterisk machine. > * I wrote dialplan for that extension in > extension.conf file. > > When I tried to call 222 from FreeSWITCH, I have received following > error in Console. > "[ERR] switch_core_session.c:255 > switch_core_session_outgoing_channel() Could not locate channel type iax" > > What would be the problem? Is there any configuration I missed? > Please help me ..... > > Regards, > K.Velusamy. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20090727/a7d0e128/attachment-0002.html From yudha2008 at gmail.com Sun Jul 26 23:40:42 2009 From: yudha2008 at gmail.com (Baskar) Date: Mon, 27 Jul 2009 12:10:42 +0530 Subject: [Freeswitch-users] core dump Message-ID: Hi, I get core dump segmentation fault in freeswitch machine frequently. can any one assist me what is error in the freeswitch. i have pasted the logs in freeswitch pastebin. This is the link http://pastebin.freeswitch.org/9851 -- Thanks with Regards, N.Baskar -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090727/5e4bfde5/attachment-0002.html From darklion11 at yahoo.com Sun Jul 26 23:41:29 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Sun, 26 Jul 2009 23:41:29 -0700 (PDT) Subject: [Freeswitch-users] Multiple gateways: Priority the first bridge with prefix Message-ID: <24674260.post@talk.nabble.com> Hi FS Users, I just want to try multiple gateways. It works actually like this... But I test call like 5133333 at 222.333.444.555, it also calls the second bridge 111.222.333.333. It there any way to determine which prefix will call to a bridge specified. E.g. for bridge 1: with prefix of 51 the call with run to 222.333.444.555 not at the second bridge and vice versa. Please help.. -- View this message in context: http://www.nabble.com/Multiple-gateways%3A-Priority-the-first-bridge-with-prefix-tp24674260p24674260.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From darklion11 at yahoo.com Sun Jul 26 23:42:41 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Sun, 26 Jul 2009 23:42:41 -0700 (PDT) Subject: [Freeswitch-users] Multiple gateways: Priority the first bridge with prefix Message-ID: <24674279.post@talk.nabble.com> Hi FS Users, I just want to try multiple gateways. It works actually like this... But I test call like 5133333 at 222.333.444.555, it also calls the second bridge 111.222.333.333. It there any way to determine which prefix will call to a bridge specified. E.g. for bridge 1: with prefix of 51 the call with run to 222.333.444.555 not at the second bridge and vice versa. Please help.. -- View this message in context: http://www.nabble.com/Multiple-gateways%3A-Priority-the-first-bridge-with-prefix-tp24674279p24674279.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From darklion11 at yahoo.com Sun Jul 26 23:42:50 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Sun, 26 Jul 2009 23:42:50 -0700 (PDT) Subject: [Freeswitch-users] Multiple gateways: Priority the first bridge with prefix Message-ID: <24674280.post@talk.nabble.com> Hi FS Users, I just want to try multiple gateways. It works actually like this... But I test call like 5133333 at 222.333.444.555, it also calls the second bridge 111.222.333.333. It there any way to determine which prefix will call to a bridge specified. E.g. for bridge 1: with prefix of 51 the call with run to 222.333.444.555 not at the second bridge and vice versa. Please help.. -- View this message in context: http://www.nabble.com/Multiple-gateways%3A-Priority-the-first-bridge-with-prefix-tp24674280p24674280.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From darklion11 at yahoo.com Sun Jul 26 23:52:32 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Sun, 26 Jul 2009 23:52:32 -0700 (PDT) Subject: [Freeswitch-users] Multiple gateways: Priority the first bridge with prefix Message-ID: <24674381.post@talk.nabble.com> Hi FS Users, I just want to try multiple gateways. It works actually like this... But I test call like 5133333 at 222.333.444.555, it also calls the second bridge 111.222.333.333. It there any way to determine which prefix will call to a bridge specified. E.g. for bridge 1: with prefix of 51 the call with run to 222.333.444.555 not at the second bridge and vice versa. Please help.. -- View this message in context: http://www.nabble.com/Multiple-gateways%3A-Priority-the-first-bridge-with-prefix-tp24674381p24674381.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From gcd at i.ph Mon Jul 27 00:04:34 2009 From: gcd at i.ph (Nandy Dagondon) Date: Mon, 27 Jul 2009 15:04:34 +0800 Subject: [Freeswitch-users] Multiple gateways: Priority the first bridge with prefix In-Reply-To: <24674280.post@talk.nabble.com> References: <24674280.post@talk.nabble.com> Message-ID: <7d0bfd8c0907270004s26db6c05j4d760a5a6ab5fcf3@mail.gmail.com> in my implementation, i would use 2 separate conditions that looks like this: On Mon, Jul 27, 2009 at 2:42 PM, Edmar Cruz wrote: > > Hi FS Users, > > I just want to try multiple gateways. It works actually like this... > > > > But I test call like 5133333 at 222.333.444.555, it also calls the > second bridge 111.222.333.333. > > It there any way to determine which prefix will call to a bridge > specified. > > E.g. > > for bridge 1: with prefix of 51 the call with run to 222.333.444.555 not > at the second bridge and vice versa. Please help.. > -- > View this message in context: > http://www.nabble.com/Multiple-gateways%3A-Priority-the-first-bridge-with-prefix-tp24674280p24674280.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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090727/c8a7d177/attachment-0002.html From jason at jasonjgw.net Mon Jul 27 00:07:56 2009 From: jason at jasonjgw.net (Jason White) Date: Mon, 27 Jul 2009 17:07:56 +1000 Subject: [Freeswitch-users] Dial plan contexts Message-ID: <20090727070756.GA22463@jdc.jasonjgw.net> Has anything changed in the handling of dial plan contexts recently? As of rev. 14363, the context setting in the Sofia profile seems to be overriding the context setting in the user's definition in the directory. As per the default configuration, I have user-context set to public in my internal profile, my user has its context set to "default", but calls made from the phone registered to that user ID end up in public context when they reach the dial plan. Either something has changed or there's something wierd in my configuration that I haven't tracked down. I haven't made any changes to any of the profiles or users recently, though, and it was working under an older revision. From darklion11 at yahoo.com Mon Jul 27 00:47:40 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Mon, 27 Jul 2009 00:47:40 -0700 (PDT) Subject: [Freeswitch-users] Multiple gateways: Priority the first bridge with prefix In-Reply-To: <7d0bfd8c0907270004s26db6c05j4d760a5a6ab5fcf3@mail.gmail.com> References: <24674260.post@talk.nabble.com> <7d0bfd8c0907270004s26db6c05j4d760a5a6ab5fcf3@mail.gmail.com> Message-ID: <24675073.post@talk.nabble.com> Not working just the same both of them are running Nandy Dagondon wrote: > > in my implementation, i would use 2 separate conditions that looks like > this: > > > > > > > > > On Mon, Jul 27, 2009 at 2:42 PM, Edmar Cruz wrote: > >> >> Hi FS Users, >> >> I just want to try multiple gateways. It works actually like this... >> >> >> >> But I test call like 5133333 at 222.333.444.555, it also calls the >> second bridge 111.222.333.333. >> >> It there any way to determine which prefix will call to a bridge >> specified. >> >> E.g. >> >> for bridge 1: with prefix of 51 the call with run to 222.333.444.555 >> not >> at the second bridge and vice versa. Please help.. >> -- >> View this message in context: >> http://www.nabble.com/Multiple-gateways%3A-Priority-the-first-bridge-with-prefix-tp24674280p24674280.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/Multiple-gateways%3A-Priority-the-first-bridge-with-prefix-tp24674260p24675073.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From elihayun at gmail.com Mon Jul 27 00:59:52 2009 From: elihayun at gmail.com (Eli Hayun) Date: Mon, 27 Jul 2009 10:59:52 +0300 Subject: [Freeswitch-users] Limit is not working when originate a call In-Reply-To: <4A6C05F6.4000501@savion.huji.ac.il> References: <4A6709D4.6000704@savion.huji.ac.il> <2C85EB6C-1D80-4DC7-8D14-06B17CD1046D@jerris.com> <4A67EF47.6000402@savion.huji.ac.il> <191c3a030907230539i5a640e2do4377d8797ff8f279@mail.gmail.com> <4A6C05F6.4000501@savion.huji.ac.il> Message-ID: <4A6D5E78.1060307@savion.huji.ac.il> Anthony Minessale wrote: >> limit is for inbound calls >> you cannot call it after you already made the call. >> The correct approach would be to not make the call at all. >> >> you could maybe use the limit FSAPI interface with apiExecute to check >> if the limit was exceeded and >> then not bother to place the call to begin with. >> >> otherwise it's sort of like putting a prisoner in the electric chair >> then giving him his trial. >> >> >> Can you tell me how to do that? I set the limit as: Now, how do I know what is the current limit of ${destination_number} Can you give me a JS (or lua) example? Thanks Eli From jason at jasonjgw.net Mon Jul 27 01:16:45 2009 From: jason at jasonjgw.net (Jason White) Date: Mon, 27 Jul 2009 18:16:45 +1000 Subject: [Freeswitch-users] Multiple gateways: Priority the first bridge with prefix In-Reply-To: <24675073.post@talk.nabble.com> References: <24674260.post@talk.nabble.com> <7d0bfd8c0907270004s26db6c05j4d760a5a6ab5fcf3@mail.gmail.com> <24675073.post@talk.nabble.com> Message-ID: <20090727081645.GA31511@jdc.jasonjgw.net> Edmar Cruz wrote: > > Not working just the same both of them are running Do you have them as separate extensions in the dial plan? From gcd at i.ph Mon Jul 27 01:44:15 2009 From: gcd at i.ph (Nandy Dagondon) Date: Mon, 27 Jul 2009 16:44:15 +0800 Subject: [Freeswitch-users] Multiple gateways: Priority the first bridge with prefix In-Reply-To: <20090727081645.GA31511@jdc.jasonjgw.net> References: <24674260.post@talk.nabble.com> <7d0bfd8c0907270004s26db6c05j4d760a5a6ab5fcf3@mail.gmail.com> <24675073.post@talk.nabble.com> <20090727081645.GA31511@jdc.jasonjgw.net> Message-ID: <7d0bfd8c0907270144q74e4d0td2b810a49302ccae@mail.gmail.com> ed, i mean you use separate extension names: btw, you should also use separate gateway names "sip1" and "sip2". so differentiate them in the bridge application. On Mon, Jul 27, 2009 at 4:16 PM, Jason White wrote: > Edmar Cruz wrote: > > > > Not working just the same both of them are running > > Do you have them as separate extensions in the dial plan? > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20090727/64739dc6/attachment-0002.html From helmut.kuper at ewetel.de Mon Jul 27 02:34:49 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Mon, 27 Jul 2009 11:34:49 +0200 Subject: [Freeswitch-users] Question: Capturing VoiceData received from Sagnoma E1 card In-Reply-To: <4A65CA52.90002@ewetel.de> References: <4A65CA52.90002@ewetel.de> Message-ID: <4A6D74B9.3040306@ewetel.de> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, any ideas? regards Helmut On 21.07.2009 16:01, Helmut Kuper wrote: > Hello, > > > For outgoing calls I'm hunting the cause for missing some 100ms of voice > data send from remote right after pickup the remote phone (e.g. initial > "Hello?" sound like "o?" or even nothing) > > On FreeSwitch server I captured the VoIP data to the called VoIP-Phone > on the sofia interface. Using wireshark it also shows that the voice > data from remote is missed. Using Mobil phones or ISDN phones calling > the same remote party there is never a bit missed. > > This problem occurs rare - once or twice per day and per local voip > phone, but it's quite anoying. > > So is there a way to capture the correspondig ISDN voice data FS > receives before it is transmitted via RTP or just droped? I want to c > whether FS drops the early RTP packets or whether FS never got the data > from ISDN. > > > > Sofia Profile is using > > > The dialplan portion is: > > expression="^94([0-9]+)$" break="never"> > > data="effective_caller_id_name=anonymous"/> > data="effective_caller_id_number=anonymous"/> > > expression="^([0-9]+)$"> > data="ignore_early_media=true"/> > data="absolute_codec_string=PCMA"/> > data="continue_on_fail=true"/> > > data="${destination_number} XML et_internal_error"/> > > > Any ideas to refine my debugging? > > regard > 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 -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) iD8DBQFKbXS54tZeNddg3dwRAp07AJ9e9gNY/MR4byUvpeR6so9Ap3cx8ACaA9SP EodxfZrtLAZiYtzYtQsBldY= =ZfJl -----END PGP SIGNATURE----- From rupa at rupa.com Mon Jul 27 05:38:14 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Mon, 27 Jul 2009 07:38:14 -0500 Subject: [Freeswitch-users] core dump In-Reply-To: References: Message-ID: That backtrace is not useful because gdb was unable to locate the freeswitch binary. Since you are running it from /opt/freeswitch/bin directory, don't use 'bin/freeswitch' to point gdb to the binary: gdb freeswitch corefile On Mon, Jul 27, 2009 at 1:40 AM, Baskar wrote: > Hi, > > I get core dump segmentation fault in freeswitch machine frequently. can > any one assist me what is error in the freeswitch. i have pasted the logs in > freeswitch pastebin. > > This is the link http://pastebin.freeswitch.org/9851 > > > > -- > Thanks with 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 > > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090727/ca477a25/attachment-0002.html From yudha2008 at gmail.com Mon Jul 27 06:00:32 2009 From: yudha2008 at gmail.com (Baskar) Date: Mon, 27 Jul 2009 06:00:32 -0700 Subject: [Freeswitch-users] core dump In-Reply-To: References: Message-ID: *Hi Rupa, I get core dump segmentation fault in freeswitch machine frequently. can any one assist me what is error in the freeswitch. i have pasted the logs in freeswitch pastebin. This is the link http://pastebin.freeswitch.org/9854 Can some one assist me what is error in freeswitch to hit core dump. -- Thanks with Regards, N.Baskar * -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090727/a7e16fb6/attachment-0002.html From rupa at rupa.com Mon Jul 27 06:10:33 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Mon, 27 Jul 2009 08:10:33 -0500 Subject: [Freeswitch-users] core dump In-Reply-To: References: Message-ID: That backtrace is useless since you are not giving the right path to freeswitch. Look at line 9 of the pastebin. Until you fix that we can't look at the backtrace. On Mon, Jul 27, 2009 at 8:00 AM, Baskar wrote: > *Hi Rupa, > > I get core dump segmentation fault in freeswitch machine frequently. can > any one assist me what is error in the freeswitch. i have pasted the logs in > freeswitch pastebin. > > This is the link http://pastebin.freeswitch.org/9854 > > Can some one assist me what is error in freeswitch to hit core dump. > > > > -- > Thanks with 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 > > -- -Rupa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090727/00c52c72/attachment-0002.html From mike at jerris.com Mon Jul 27 06:17:28 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 27 Jul 2009 09:17:28 -0400 Subject: [Freeswitch-users] core dump In-Reply-To: References: Message-ID: Did you read his response to you? Please generate a usable backtrace as rupa explained and post a bug to jira freeswitch.org On Jul 27, 2009, at 9:00 AM, Baskar wrote: > Hi Rupa, > > I get core dump segmentation fault in freeswitch machine > frequently. can any one assist me what is error in the freeswitch. > i have pasted the logs in freeswitch pastebin. > > This is the link http://pastebin.freeswitch.org/9854 > > Can some one assist me what is error in freeswitch to hit core dump. > > > > -- > Thanks with 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090727/0731f784/attachment-0002.html From gmaruzz at celliax.org Mon Jul 27 07:08:55 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Mon, 27 Jul 2009 16:08:55 +0200 Subject: [Freeswitch-users] Skypiax, Skype endpoint and trunk, revamped Message-ID: <7b197bef0907270708h1dd55de2m92b5de869da6404d@mail.gmail.com> Ciao FreeSWITCHers, please have a look at the much changed wiki page: http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk and checkout and test the code in svn. Much has happened, various bug fixes and features added. Most notable: - multiple instances of the same Skype username on Linux (eg: running 20 concurrent channels as "Bob" Skype user) - adding and removing interfaces on the fly (patch sent by Muhammad Shahzad) - easier creation of Skype clients configuration directory - reduced latency - better robustness - running as Windows Service - customized ALSA driver for more devices with less IRQs and context switches - custom kernel tickless and 100HZ (eg. solves high load problems in CentOS and in virtualization) - interactive command line client for prototyping Also, please note that ALSA drivers version 1.0.20 seems to be much more stable in our kind of usage (snd-dummy). Various other enhancements will come, but in the mean time please give feedback on the current svn code (we want to be robust for the 1.0.4 Release :-) ) See you all at www.cluecon.com, talk on Skypiax August 4th at 4.30 pm ! -giovanni 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 Mon Jul 27 07:09:43 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 27 Jul 2009 09:09:43 -0500 Subject: [Freeswitch-users] core dump In-Reply-To: References: Message-ID: <191c3a030907270709pa530cey4dc70af085853e6a@mail.gmail.com> Again, I like I have to say weekly, Please do not report bugs on the mailing list. http://jira.freeswitch.org Also, please completely re-checkout and rebuild latest trunk and erase your prior freeswitch install before filing the jira. We only accept bug reports of this nature confirmed on a fresh build of latest SVN. On Mon, Jul 27, 2009 at 8:17 AM, Michael Jerris wrote: > Did you read his response to you? Please generate a usable backtrace as > rupa explained and post a bug to jira freeswitch.org > > > On Jul 27, 2009, at 9:00 AM, Baskar wrote: > > *Hi Rupa, > > I get core dump segmentation fault in freeswitch machine frequently. can > any one assist me what is error in the freeswitch. i have pasted the logs in > freeswitch pastebin. > > This is the link > http://pastebin.freeswitch.org/9854 > > Can some one assist me what is error in freeswitch to hit core dump. > > > > -- > Thanks with 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 > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference 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/20090727/07d74b71/attachment-0002.html From brian at freeswitch.org Mon Jul 27 07:20:38 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 27 Jul 2009 09:20:38 -0500 Subject: [Freeswitch-users] Dial plan contexts In-Reply-To: <20090727070756.GA22463@jdc.jasonjgw.net> References: <20090727070756.GA22463@jdc.jasonjgw.net> Message-ID: Jason, You need to set the context to on the profile and the user_context variable on the user to default. There is no such thing as a user- context param on the profile. There is a user_context variable on the user. /b On Jul 27, 2009, at 2:07 AM, Jason White wrote: > As per the default configuration, I have user-context set to public > in my > internal profile, my user has its context set to "default", but > calls made > from the phone registered to that user ID end up in public context > when they > reach the dial plan. From anthony.minessale at gmail.com Mon Jul 27 07:36:26 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 27 Jul 2009 09:36:26 -0500 Subject: [Freeswitch-users] Limit is not working when originate a call In-Reply-To: <4A6D5E78.1060307@savion.huji.ac.il> References: <4A6709D4.6000704@savion.huji.ac.il> <2C85EB6C-1D80-4DC7-8D14-06B17CD1046D@jerris.com> <4A67EF47.6000402@savion.huji.ac.il> <191c3a030907230539i5a640e2do4377d8797ff8f279@mail.gmail.com> <4A6C05F6.4000501@savion.huji.ac.il> <4A6D5E78.1060307@savion.huji.ac.il> Message-ID: <191c3a030907270736r303abfadraadc0527b1d45e34@mail.gmail.com> I suggest you study FS more because if you can't tell what to do with the info provided you have some fundamentals to review before proceeding. On Mon, Jul 27, 2009 at 2:59 AM, Eli Hayun wrote: > Anthony Minessale wrote: > >> limit is for inbound calls > >> you cannot call it after you already made the call. > >> The correct approach would be to not make the call at all. > >> > >> you could maybe use the limit FSAPI interface with apiExecute to check > >> if the limit was exceeded and > >> then not bother to place the call to begin with. > >> > >> otherwise it's sort of like putting a prisoner in the electric chair > >> then giving him his trial. > >> > >> > >> > Can you tell me how to do that? > I set the limit as: > > Now, how do I know what is the current limit of ${destination_number} > Can you give me a JS (or lua) example? > Thanks > Eli > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference 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/20090727/22895d3e/attachment-0002.html From xengelpublicx at gmail.com Mon Jul 27 07:42:11 2009 From: xengelpublicx at gmail.com (Vladimir Elizarov) Date: Mon, 27 Jul 2009 18:42:11 +0400 Subject: [Freeswitch-users] linksys spa962+spa932 blf, hold and intercept of ringing extensions Message-ID: <4A6DBCC3.8010107@gmail.com> Hello. I am trying to configure the linksys spa-932 (at http://wiki.freeswitch.org/wiki/Interop_List#Linksys_SPA932 stated that he works with freeswitch). I said "Server-Type option set to" RFC3265_4235 "" added to the unit 1 key 1 string: "fnc = blf + sd; sub = 1002 at pbx0.test.lan; nme = test". The button blinks orange. if call on 1000 (spa962). This subscription runs spa932 and starts to show the status of the phone 1002. Thanks. -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 261 bytes Desc: OpenPGP digital signature Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090727/2fdbdbe7/attachment-0002.bin From jgonzalez at sqli.com Mon Jul 27 08:58:29 2009 From: jgonzalez at sqli.com (julien) Date: Mon, 27 Jul 2009 17:58:29 +0200 Subject: [Freeswitch-users] SIP trunk (PBX - FreeSwitch) routing problem. Message-ID: <4A6DCEA5.2090502@sqli.com> I'm currently trying to connect FreeSwitch to a PBX (Alcatel-Lucent), thanks to a SIP trunk. SIP trunks are available and working on the PBX thanks to a recent update. My problem is that I can't call phones linked to the PBX. When I try to call 300, I've got this message in freeswitch console : 2009-07-27 17:38:48.514105 [NOTICE] switch_channel.c:602 New Channel sofia/internal/[EMAIL PROTECTED] [93d5a10e-7ac3-11de-b456-e5e56113066d] 2009-07-27 17:38:48.516907 [INFO] mod_dialplan_xml.c:252 Processing jgonzalez jgonzalez->300 in context default 2009-07-27 17:38:48.521084 [NOTICE] switch_channel.c:602 New Channel sofia/external/[EMAIL PROTECTED] [93d69816-7ac3-11de-b456-e5e56113066d] 2009-07-27 17:38:48.636073 [NOTICE] sofia.c:3775 Hangup sofia/external/[EMAIL PROTECTED] [CS_CONSUME_MEDIA] [NO_ROUTE_DESTINATION] 2009-07-27 17:38:48.636073 [INFO] mod_dptools.c:2091 Originate Failed. Cause: NO_ROUTE_DESTINATION 2009-07-27 17:38:48.637788 [NOTICE] mod_dptools.c:633 Hangup sofia/internal/[EMAIL PROTECTED] [CS_EXECUTE] [NORMAL_CLEARING] 2009-07-27 17:38:48.639047 [NOTICE] switch_core_session.c:1085 Session 15 (sofia/internal/[EMAIL PROTECTED]) Ended 2009-07-27 17:38:48.639047 [NOTICE] switch_core_session.c:1087 Close Channel sofia/internal/[EMAIL PROTECTED] [CS_DESTROY] 2009-07-27 17:38:48.643042 [NOTICE] switch_core_session.c:1085 Session 16 (sofia/external/[EMAIL PROTECTED]) Ended 2009-07-27 17:38:48.643042 [NOTICE] switch_core_session.c:1087 Close Channel sofia/external/[EMAIL PROTECTED] [CS_DESTROY] I've defined, in sip_profiles/external, a gateway to the PBX this way : And in the dialplan default.xml : (for the moment, I'm trying only with the number 300 which is a correct number of the phone system). As you can see, I'm far from being an expert of FreeSwitch, SIP or even VoIP in general. I'm learning. I hope you can help me. Regards, Julien Gonzalez. From brian at freeswitch.org Mon Jul 27 09:09:54 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 27 Jul 2009 11:09:54 -0500 Subject: [Freeswitch-users] SIP trunk (PBX - FreeSwitch) routing problem. In-Reply-To: <4A6DCEA5.2090502@sqli.com> References: <4A6DCEA5.2090502@sqli.com> Message-ID: <60EC1D48-40FD-4BF9-B5D2-D8FAE1EBEEEC@freeswitch.org> I have to guess that you put this at the bottom of the default.xml? /b On Jul 27, 2009, at 10:58 AM, julien wrote: > And in the dialplan default.xml : > > > > > > > > > > From jgonzalez at sqli.com Mon Jul 27 09:16:20 2009 From: jgonzalez at sqli.com (julien) Date: Mon, 27 Jul 2009 18:16:20 +0200 Subject: [Freeswitch-users] SIP trunk (PBX - FreeSwitch) routing problem. In-Reply-To: <60EC1D48-40FD-4BF9-B5D2-D8FAE1EBEEEC@freeswitch.org> References: <4A6DCEA5.2090502@sqli.com> <60EC1D48-40FD-4BF9-B5D2-D8FAE1EBEEEC@freeswitch.org> Message-ID: <4A6DD2D4.8040800@sqli.com> No totally at the bottom. Before : Brian West a ?crit : > I have to guess that you put this at the bottom of the default.xml? > > /b > > On Jul 27, 2009, at 10:58 AM, julien wrote: > > >> And in the dialplan default.xml : >> >> >> >> >> >> >> >> >> >> >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From jesse.peterson at exbiblio.com Mon Jul 27 09:25:12 2009 From: jesse.peterson at exbiblio.com (Jesse Peterson) Date: Mon, 27 Jul 2009 09:25:12 -0700 Subject: [Freeswitch-users] Operation has no matching challenge Message-ID: <1B7E58CE-5BF3-4C45-9A27-02E9B84F4BA4@exbiblio.com> Hello, I am getting some SIP registration problems with FreeSWITCH 1.0.4pre10 and my ITSP (Vitelity). The error in the logs is such: 2009-07-27 08:54:48.217815 [ERR] sofia_reg.c:1460 vitelity-out Registration Failed with status Operation has no matching challenge [904]. failure #56 While these errors are happening the gateway state (via "sofia status") is FAIL_WAIT. With the ever-increasing back-off wait (60, 90, 120, 150, ..., seconds) the registration never resumes. Now one might suspect that there is something wrong with the configuration/ authorization but this problem is intermittent: a simple "sofia profile external restart" restores the registration and all is well (state turns to REGED) and of course the initial registration succeeds just fine, too. Quite an annoying problem as you never quite know when your gateway is registered when you pick up the receiver of your phone giving the impression of unreliable service! I suspect this to be the same problem, but with a different error message, that has been reported before[1][2]. Thoughts? Anything I should try? Thanks, - Jesse [1] http://lists.freeswitch.org/pipermail/freeswitch-users/2009-February/011269.html [2] http://lists.freeswitch.org/pipermail/freeswitch-users/2008-September/006272.html From brian at freeswitch.org Mon Jul 27 09:30:00 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 27 Jul 2009 11:30:00 -0500 Subject: [Freeswitch-users] Operation has no matching challenge In-Reply-To: <1B7E58CE-5BF3-4C45-9A27-02E9B84F4BA4@exbiblio.com> References: <1B7E58CE-5BF3-4C45-9A27-02E9B84F4BA4@exbiblio.com> Message-ID: Can you get a sofia loglevel all 9 and a sip trace? /b On Jul 27, 2009, at 11:25 AM, Jesse Peterson wrote: > Hello, > > I am getting some SIP registration problems with FreeSWITCH 1.0.4pre10 > and my ITSP (Vitelity). The error in the logs is such: > > 2009-07-27 08:54:48.217815 [ERR] sofia_reg.c:1460 vitelity-out > Registration Failed with status Operation has no matching challenge > [904]. failure #56 > > While these errors are happening the gateway state (via "sofia > status") is FAIL_WAIT. With the ever-increasing back-off wait (60, 90, > 120, 150, ..., seconds) the registration never resumes. Now one might > suspect that there is something wrong with the configuration/ > authorization but this problem is intermittent: a simple "sofia > profile external restart" restores the registration and all is well > (state turns to REGED) and of course the initial registration succeeds > just fine, too. Quite an annoying problem as you never quite know when > your gateway is registered when you pick up the receiver of your phone > giving the impression of unreliable service! > > I suspect this to be the same problem, but with a different error > message, that has been reported before[1][2]. > > Thoughts? Anything I should try? Thanks, > - Jesse From miconda at gmail.com Mon Jul 27 09:45:35 2009 From: miconda at gmail.com (Daniel-Constantin Mierla) Date: Mon, 27 Jul 2009 18:45:35 +0200 Subject: [Freeswitch-users] freeswitch and siremis v0.9.3 Message-ID: <4A6DD9AF.2080406@gmail.com> Hello, recently released siremis v0.9.3 adds support for communication with freeswitch via event socket. siremis is an open source web management interface targeting the SIP routing engines kamailio (openser) and sip-router.org. freeswitch fits perfectly in the picture since it completes the routing engines with rich media services. The new release includes php code to communicate with freeswitch via tcp/event socket and a panel to send commands/display response. Code is grouped like a library, new features being straightforward to develop. For some commands, the output is pretty formatted - screenshot: http://www.asipto.com/gallery/v/siremis/siremis_20.jpg.html?g2_imageViewsIndex=1 More is planned for the future (e.g., display active calls of a certain user, click to end an active call). Feedback and contributions are welcome, visit: http://siremis.asipto.com Cheers, Daniel -- Daniel-Constantin Mierla * SIP Router Bootcamp * Kamailio (OpenSER) and Asterisk Training * Berlin, Germany, Sep 1-4, 2009 * http://www.asipto.com/index.php/sip-router-bootcamp/ From msc at freeswitch.org Mon Jul 27 10:00:42 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 27 Jul 2009 10:00:42 -0700 Subject: [Freeswitch-users] Which method Can I use in IVR In-Reply-To: <7aa29e790907262207n5d50c96cl2f19e2d152399db6@mail.gmail.com> References: <7aa29e790907262207n5d50c96cl2f19e2d152399db6@mail.gmail.com> Message-ID: <87f2f3b90907271000h557e3c05jbd3839b9de595292@mail.gmail.com> See comments inline... On Sun, Jul 26, 2009 at 10:07 PM, Thangappan.M wrote: > Dear all, > > I am learning how to implement a IVR in Freeswitch.In our organization we > are using Perl scripting language for doing this.So In freeswitch also I > need to use Perl. Tony, Brian, and I all like Perl. :) > > > So far I heard two methods for executing IVR. > One is in dial plan using perl application.( In perl I create IVR > menu and play the voice files) > Another one is using event socket.In dial plan I specified socket > application and write a Perl script which is listening that particular port > and get the session Id. > Yes, you can call a script from the dialplan using syntax like this: OR You can call an outbound socket connection like this: > > Have I understood correctly?.If it is correct means tell which method can I > use?. Other make me understand well. You're on the right track. As to which method to use, that depends on your circumstances. How much does it need to scale? Do you want the IVR "brain" to reside physically on a different server than the FS server? Think about those things. > > I have seen downloaded perl IVR menu from freeswitch site.In that they > called some internal functions like playandGetDigits,StreamFile,ready > ...etc. > > These functions is been called by using $session variable.Where these > functions are defined.? > When you call a Perl script from the dialplan the script automatically has access to a variable called $session. Check this for more information: http://wiki.freeswitch.org/wiki/Mod_perl#Programming_with_mod_perl Of course, when using the outbound event socket you will not have this magic $session variable. Your best bet to learn more about the socket interface is to look at the sample scripts in src/libs/esl/perl/. (server.pl, server2.pl, and server3.pl) If you are building an IVR with Perl and the event socket be sure to check out src/libs/esl/perl/ESL/IVR.pm which is a small Perl module with some simple abstractions to make IVR programming a bit more convenient. I recommend that you try and create a simple IVR using each method and get a feel for how each one works. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090727/6902070a/attachment-0002.html From miles.chet at gmail.com Mon Jul 27 10:35:48 2009 From: miles.chet at gmail.com (roberto) Date: Mon, 27 Jul 2009 14:35:48 -0300 Subject: [Freeswitch-users] Which method Can I use in IVR In-Reply-To: <87f2f3b90907271000h557e3c05jbd3839b9de595292@mail.gmail.com> References: <7aa29e790907262207n5d50c96cl2f19e2d152399db6@mail.gmail.com> <87f2f3b90907271000h557e3c05jbd3839b9de595292@mail.gmail.com> Message-ID: Michael, For scale reasons is the best choice event socket? thanks, On Mon, Jul 27, 2009 at 2:00 PM, Michael Collins wrote: > See comments inline... > > On Sun, Jul 26, 2009 at 10:07 PM, Thangappan.M > wrote: >> >> Dear all, >> >> ?I am learning how to implement a IVR in Freeswitch.In our organization we >> are using Perl scripting language for doing this.So In freeswitch also I >> need to use Perl. > > Tony, Brian, and I all like Perl. :) > >> >> >> ?So far I heard two methods for executing IVR. >> ??????? One is in dial plan using perl application.( In perl I create IVR >> menu and play the voice files) >> ??????? Another one is using event socket.In dial plan I specified socket >> application and write a Perl script which is listening that particular port >> and get the session Id. > > Yes, you can call a script from the dialplan using syntax like this: > > > OR > > You can call an outbound socket connection like this: > > >> >> Have I understood correctly?.If it is correct means tell which method can >> I use?. Other make me understand well. > > You're on the right track. As to which method to use, that depends on your > circumstances. How much does it need to scale? Do you want the IVR "brain" > to reside physically on a different server than the FS server? Think about > those things. > >> >> >> I have seen downloaded perl IVR menu from freeswitch site.In that they >> called some internal functions like playandGetDigits,StreamFile,ready >> ...etc. >> >> These functions is been called by using $session variable.Where these >> functions are defined.? > > When you call a Perl script from the dialplan the script automatically has > access to a variable called $session.? Check this for more information: > http://wiki.freeswitch.org/wiki/Mod_perl#Programming_with_mod_perl > > Of course, when using the outbound event socket you will not have this magic > $session variable. Your best bet to learn more about the socket interface is > to look at the sample scripts in src/libs/esl/perl/. (server.pl, server2.pl, > and server3.pl) If you are building an IVR with Perl and the event socket be > sure to check out src/libs/esl/perl/ESL/IVR.pm which is a small Perl module > with some simple abstractions to make IVR programming a bit more convenient. > > I recommend that you try and create a simple IVR using each method and get a > feel for how each one works. > > -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 marketing at cluecon.com Mon Jul 27 11:50:01 2009 From: marketing at cluecon.com (Michael Collins) Date: Mon, 27 Jul 2009 11:50:01 -0700 Subject: [Freeswitch-users] ClueCon 2009 - Last Chance For Early Bird Special! Message-ID: <87f2f3b90907271150y2b5c5db1pb7466590a87c0a48@mail.gmail.com> ClueCon is next week! We're all gearing up for a great event. Here is some important information for those who've not already paid: Today is the last day to receive the $499 early bird special. After today, the price will go up to $699. If you have registered at the ClueCon website but you have not yet paid then please call 877.742.CLUE immediately! We want to make sure that you get the early bird rate. All registrations after today (Monday July 27) will be $699. Thank you for your support of ClueCon 2009! We are looking forward to seeing everyone in person in Chicago. -Michael Collins http://www.cluecon.com 877.742.CLUE -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090727/ad6aaa33/attachment-0002.html From msc at freeswitch.org Mon Jul 27 11:57:46 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 27 Jul 2009 11:57:46 -0700 Subject: [Freeswitch-users] Which method Can I use in IVR In-Reply-To: References: <7aa29e790907262207n5d50c96cl2f19e2d152399db6@mail.gmail.com> <87f2f3b90907271000h557e3c05jbd3839b9de595292@mail.gmail.com> Message-ID: <87f2f3b90907271157h2aefb8ceu581c0fd2b6b4a89d@mail.gmail.com> On Mon, Jul 27, 2009 at 10:35 AM, roberto wrote: > Michael, > > For scale reasons is the best choice event socket? > Yes. You can have the IVR stuff running on a separate server altogether. It also gives you great flexibility in designing a setup where you can have a db backend and/or a backup IVR server. The socket method requires a little more effort up front but it pays off in power and flexibility. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090727/2ed51b57/attachment-0002.html From gmaruzz at celliax.org Mon Jul 27 12:31:01 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Mon, 27 Jul 2009 21:31:01 +0200 Subject: [Freeswitch-users] Slashdot: How-to-Help-With-a-University-ICT-Strategy Message-ID: <7b197bef0907271231y262c7b06v3a83ebcef86f7f54@mail.gmail.com> http://ask.slashdot.org/story/09/07/27/1652247/How-to-Help-With-a-University-ICT-Strategy " An anonymous reader writes "I have been asked to contribute to my university's revised ICT (Information and Communication Technology) strategy and I am curious what fellow Slashdot members consider to be the main advice in this context. What are the major mistakes that organizations like universities make? Given the complexity of the different participants in a university, how does one have a coherent strategy that fulfills the needs of such a wide audience? How does one promote open source in a managerial culture? How does one deal with the curse of the virtual learning environment?"" http://ask.slashdot.org/comments.pl?sid=1316571&cid=28842157 From testeador01 at gmail.com Mon Jul 27 12:34:54 2009 From: testeador01 at gmail.com (Milena) Date: Mon, 27 Jul 2009 14:34:54 -0500 Subject: [Freeswitch-users] Command to get FreeSWITCH's internal time In-Reply-To: <1248625501.5655.79.camel@raul-laptop> References: <1248625501.5655.79.camel@raul-laptop> Message-ID: *Hello everyone, strftime *does what i want (now I can't figure out why didn't I see it before in the wiki page), thank you very much for your replies, have a nice time :) 2009/7/26 Raul Fragoso > Every event has a header 'Event-Date-Local', which has the local date > and time. If want to actively retrieve the date and time, you can send > this API request to the server: api strftime > > Regards, > > Raul > > On Sat, 2009-07-25 at 09:20 -0500, Milena wrote: > > Hello everyone, > > > > I'm using the inbound event socket to receive some information about > > the status of my FreeSWITCH system and i wanted to know if there is an > > api command that can be used to get the FreeSWITCH time, I tried > > searching around in the docs and in google but i couldn't find an > > answer. Thanks for your attention and thanks in advance if anyone can > > assist me with this. > > > > Have a nice time and lots of cookies :) > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20090727/d887db3b/attachment-0002.html From william.suffill at gmail.com Mon Jul 27 13:17:41 2009 From: william.suffill at gmail.com (William Suffill) Date: Mon, 27 Jul 2009 16:17:41 -0400 Subject: [Freeswitch-users] IAX configurations In-Reply-To: <1452e2980907262259k30713f4ft363592ac77ee3e33@mail.gmail.com> References: <1452e2980907262223u677eef21o282f39e2a56209cc@mail.gmail.com> <98E444F7-ECF8-40DF-82D2-89A48DC11EAC@jerris.com> <1452e2980907262259k30713f4ft363592ac77ee3e33@mail.gmail.com> Message-ID: <6b65470d0907271317q7476ed6x7b85ef5eb0c478d4@mail.gmail.com> http://wiki.freeswitch.org/wiki/Connecting_Freeswitch_And_Asterisk Goes into some detail with connecting to Asterisk via SIP From oseslija at gmail.com Mon Jul 27 13:45:06 2009 From: oseslija at gmail.com (Ognjen Seslija) Date: Mon, 27 Jul 2009 22:45:06 +0200 Subject: [Freeswitch-users] linksys spa962+spa932 blf, hold and intercept of ringing extensions In-Reply-To: <4A6DBCC3.8010107@gmail.com> References: <4A6DBCC3.8010107@gmail.com> Message-ID: <4468a6770907271345j5062e625pffc77fb26ed72ebb@mail.gmail.com> Hello, I authored that wiki article. The following key will work: fnc=blf+sd+cp;sub=1002@$PROXY You need to make sure that presence is not off in the profile. Also "cp" in the key will enable you to do the intercept of "ringing" call to watched extension. For further help please join #freeswitch IRC channel. Regards, Ognjen fnc=blf+sd+cp;sub=4601@$PROXY On Mon, Jul 27, 2009 at 4:42 PM, Vladimir Elizarov wrote: > Hello. > > I am trying to configure the linksys spa-932 (at > http://wiki.freeswitch.org/wiki/Interop_List#Linksys_SPA932 stated that > he works with freeswitch). > I said "Server-Type option set to" RFC3265_4235 "" added to the unit 1 > key 1 string: "fnc = blf + sd; sub = 1002 at pbx0.test.lan; nme = test". > > The button blinks orange. if call on 1000 (spa962). This subscription > runs spa932 and starts to show the status of the phone 1002. > > 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/20090727/44498024/attachment-0002.html From keithl at voxtelecom.co.za Mon Jul 27 14:23:59 2009 From: keithl at voxtelecom.co.za (Keith Laaks) Date: Mon, 27 Jul 2009 23:23:59 +0200 Subject: [Freeswitch-users] Distortion on approx first 200ms of G722 prompts on DECT based CPE Message-ID: <1B99233662E2104983E3550185D3ED73628FA0@xena.internal.datapro.co.za> Hi All, I am testing a range of G722 capable DECT based CPE. With one range, I have noticed that the first 200ms or so of each separate prompt file being played back is played out distorted from the DECT handset. When having a normal conversation, the quality is excellent, but when accessing your vmail, all the individual audio files making up the menu choices exhibit the distortion, which is pretty annoying. The same unit using G729, alaw or ulaw works 100%. I wonder if anybody else has uncounted this issue? My guess at this point - There may be a short break in the RTP between the separate files being played out by FS that makes up any menu. During this time the DECT handset's AGC probably goes to MAX amplification (as its not receiving any input during the short break in RTP). Then, when the RTP returns at the start of the next file, the AGC boosts the audio into clipping zone and takes 200ms to dampen down back to normal good levels. Looks like in these devices the G722 encode/decode is actually done in the DECT handset and not the voip-base unit. Is there any parameter that can be set in FS to ensure that the RTP keeps flowing, sending 'silence' between prompts ? Would be interesting to validate the above 'guess'. Best Regards Keith -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090727/2ece9638/attachment-0002.html From jesse.peterson at exbiblio.com Mon Jul 27 16:34:21 2009 From: jesse.peterson at exbiblio.com (Jesse Peterson) Date: Mon, 27 Jul 2009 16:34:21 -0700 Subject: [Freeswitch-users] Operation has no matching challenge In-Reply-To: References: <1B7E58CE-5BF3-4C45-9A27-02E9B84F4BA4@exbiblio.com> Message-ID: <5F25DBDD-B65C-444C-9189-09416D66042E@exbiblio.com> Just to keep those interested informed this thread is being tracked as: http://jira.freeswitch.org/browse/SFSIP-169 On Jul 27, 2009, at 9:30 AM, Brian West wrote: > Can you get a sofia loglevel all 9 > and a sip trace? > > /b > > On Jul 27, 2009, at 11:25 AM, Jesse Peterson wrote: > >> Hello, >> >> I am getting some SIP registration problems with FreeSWITCH >> 1.0.4pre10 >> and my ITSP (Vitelity). The error in the logs is such: >> >> 2009-07-27 08:54:48.217815 [ERR] sofia_reg.c:1460 vitelity-out >> Registration Failed with status Operation has no matching challenge >> [904]. failure #56 Thanks, - Jesse From jason at jasonjgw.net Mon Jul 27 17:01:26 2009 From: jason at jasonjgw.net (Jason White) Date: Tue, 28 Jul 2009 10:01:26 +1000 Subject: [Freeswitch-users] Dial plan contexts In-Reply-To: References: <20090727070756.GA22463@jdc.jasonjgw.net> Message-ID: <20090728000126.GA9387@jdc.jasonjgw.net> >From the profile: >From the user's entry in the directory: but under rev. 14363 when the phone registered to that user makes a call, the dial plan is searched in public context. I hope this helps to clarify. I tried resetting my configuration using Git to a known good state, but with no change to the above behaviour. I'm going to rebuild with the latest from svn soon. From anthony.minessale at gmail.com Mon Jul 27 17:06:00 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 27 Jul 2009 19:06:00 -0500 Subject: [Freeswitch-users] Distortion on approx first 200ms of G722 prompts on DECT based CPE In-Reply-To: <1B99233662E2104983E3550185D3ED73628FA0@xena.internal.datapro.co.za> References: <1B99233662E2104983E3550185D3ED73628FA0@xena.internal.datapro.co.za> Message-ID: <191c3a030907271706yb3d55e4m1da94a18b6f9bb0b@mail.gmail.com> you can set the global var send_silence_when_idle=true in vars.xml On Mon, Jul 27, 2009 at 4:23 PM, Keith Laaks wrote: > Hi All, > > > > I am testing a range of G722 capable DECT based CPE. > > With one range, I have noticed that the first 200ms or so of each separate > prompt file being played back is played out distorted from the DECT handset. > > When having a normal conversation, the quality is excellent, but when > accessing your vmail, all the individual audio files making up the menu > choices exhibit the distortion, which is pretty annoying. > > The same unit using G729, alaw or ulaw works 100%. > > > > I wonder if anybody else has uncounted this issue? > > > > My guess at this point ? > > There may be a short break in the RTP between the separate files being > played out by FS that makes up any menu. > > During this time the DECT handset?s AGC probably goes to MAX amplification > (as its not receiving any input during the short break in RTP). > > Then, when the RTP returns at the start of the next file, the AGC boosts > the audio into clipping zone and takes 200ms to dampen down back to normal > good levels. > > > > Looks like in these devices the G722 encode/decode is actually done in the > DECT handset and not the voip-base unit. > > > > Is there any parameter that can be set in FS to ensure that the RTP keeps > flowing, sending ?silence? between prompts ? Would be interesting to > validate the above ?guess?. > > > > > > Best Regards > > > > Keith > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference 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/20090727/3517e201/attachment-0002.html From jason at jasonjgw.net Mon Jul 27 18:13:42 2009 From: jason at jasonjgw.net (Jason White) Date: Tue, 28 Jul 2009 11:13:42 +1000 Subject: [Freeswitch-users] Dial plan contexts In-Reply-To: <20090728000126.GA9387@jdc.jasonjgw.net> References: <20090727070756.GA22463@jdc.jasonjgw.net> <20090728000126.GA9387@jdc.jasonjgw.net> Message-ID: <20090728011342.GA4273@jdc.jasonjgw.net> With apologies to all, it was something that sneaked into my configuration that I'm still tracking down. From darklion11 at yahoo.com Mon Jul 27 18:23:10 2009 From: darklion11 at yahoo.com (Edmar Cruz) Date: Mon, 27 Jul 2009 18:23:10 -0700 (PDT) Subject: [Freeswitch-users] Multiple gateways: Priority the first bridge with prefix In-Reply-To: <7d0bfd8c0907270144q74e4d0td2b810a49302ccae@mail.gmail.com> References: <24674260.post@talk.nabble.com> <7d0bfd8c0907270004s26db6c05j4d760a5a6ab5fcf3@mail.gmail.com> <24675073.post@talk.nabble.com> <20090727081645.GA31511@jdc.jasonjgw.net> <7d0bfd8c0907270144q74e4d0td2b810a49302ccae@mail.gmail.com> Message-ID: <24691020.post@talk.nabble.com> Yes, I actually just want to not be able to communicate with the other bridges. I have already this extension name = "sample-1". Freeswitch gets the first extension the 2nd also trigger it. When the calls finds the match it suits perfectly but I just want that I do not want to view the bridges with CS_DESTROY or hangup_after_false if not found. Nandy Dagondon wrote: > > ed, > > i mean you use separate extension names: > > > > > > > > > > > > btw, you should also use separate gateway names "sip1" and "sip2". so > differentiate them in the bridge application. > > On Mon, Jul 27, 2009 at 4:16 PM, Jason White wrote: > >> Edmar Cruz wrote: >> > >> > Not working just the same both of them are running >> >> Do you have them as separate extensions in the dial plan? >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/Multiple-gateways%3A-Priority-the-first-bridge-with-prefix-tp24674260p24691020.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From dujinfang at gmail.com Mon Jul 27 21:37:11 2009 From: dujinfang at gmail.com (Seven Du) Date: Tue, 28 Jul 2009 12:37:11 +0800 Subject: [Freeswitch-users] Skypiax, Skype endpoint and trunk, revamped In-Reply-To: <7b197bef0907270708h1dd55de2m92b5de869da6404d@mail.gmail.com> References: <7b197bef0907270708h1dd55de2m92b5de869da6404d@mail.gmail.com> Message-ID: <23f91030907272137r1c4879bp22dcb23cda7cc5fd@mail.gmail.com> Thanks for the great work. Just want you know that 20 channels with the same username works well on my server. And echo() works without any problem. An updated version of Round Robin hunt and a minor bug posted on jira. Thanks again. 2009/7/27 Giovanni Maruzzelli > Ciao FreeSWITCHers, > > please have a look at the much changed wiki page: > http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk and > checkout and test the code in svn. > > Much has happened, various bug fixes and features added. > > Most notable: > - multiple instances of the same Skype username on Linux (eg: running > 20 concurrent channels as "Bob" Skype user) > - adding and removing interfaces on the fly (patch sent by Muhammad > Shahzad) > - easier creation of Skype clients configuration directory > - reduced latency > - better robustness > - running as Windows Service > - customized ALSA driver for more devices with less IRQs and context > switches > - custom kernel tickless and 100HZ (eg. solves high load problems in > CentOS and in virtualization) > - interactive command line client for prototyping > > Also, please note that ALSA drivers version 1.0.20 seems to be much > more stable in our kind of usage (snd-dummy). > > Various other enhancements will come, but in the mean time please give > feedback on the current svn code (we want to be robust for the 1.0.4 > Release :-) ) > > See you all at www.cluecon.com, talk on Skypiax August 4th at 4.30 pm ! > > -giovanni > > > > > > 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 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20090728/5c561c02/attachment-0002.html From jason at jasonjgw.net Mon Jul 27 22:25:18 2009 From: jason at jasonjgw.net (Jason White) Date: Tue, 28 Jul 2009 15:25:18 +1000 Subject: [Freeswitch-users] Multiple gateways: Priority the first bridge with prefix In-Reply-To: <24691020.post@talk.nabble.com> References: <24674260.post@talk.nabble.com> <7d0bfd8c0907270004s26db6c05j4d760a5a6ab5fcf3@mail.gmail.com> <24675073.post@talk.nabble.com> <20090727081645.GA31511@jdc.jasonjgw.net> <7d0bfd8c0907270144q74e4d0td2b810a49302ccae@mail.gmail.com> <24691020.post@talk.nabble.com> Message-ID: <20090728052518.GA10571@jdc.jasonjgw.net> Edmar Cruz wrote: > > Yes, I actually just want to not be able to communicate with the other > bridges. I have already this extension name = "sample-1". Freeswitch gets > the first extension the 2nd also trigger it. When the calls finds the match > it suits perfectly but I just want that I do not want to view the bridges > with CS_DESTROY or hangup_after_false if not found. The above text is absolutely incoherent and incomprehensible, so I don't understand what you are trying to say. Try setting on the first extension and see whether that does what you want. I hope this help. From velu.technical at gmail.com Mon Jul 27 23:10:00 2009 From: velu.technical at gmail.com (velusamy velu) Date: Tue, 28 Jul 2009 11:40:00 +0530 Subject: [Freeswitch-users] Connecting FreeSWITCH with Asterisk Message-ID: <1452e2980907272310y49bdab1fg39b490ba83c845@mail.gmail.com> Dear All, I have tried to connect the FreeSWITCH with Asterisk I have followed steps which is provided in the following link, http://wiki.freeswitch.org/wiki/Connecting_Freeswitch_And_Asterisk I have tried to call "2000" from FreeSWITCH, but I have received the following message in Asterisk console "NOTICE[6346]: chan_sip.c:10543 handle_request_invite: Failed to authenticate user "Velusamy" >;tag=69Q648H9NjrSK" I have read "Using Authentication" topic in the link, But I did understand that topic.. They have mentioned "HOSTNAME.DOMAIN.COM" in that topic. Which hostname I have to specify here? Please help me.... Regards, Velusamy.K -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090728/75c1324c/attachment-0002.html From thangappan143 at gmail.com Mon Jul 27 23:16:43 2009 From: thangappan143 at gmail.com (Thangappan.M) Date: Tue, 28 Jul 2009 11:46:43 +0530 Subject: [Freeswitch-users] ESL problem Message-ID: <7aa29e790907272316m43671f92jf9f5442f6c7caf22@mail.gmail.com> Dear all, In the previous post, I got the information that using event outbound socket we can implement the IVR and also see the example in libs/esl/perl/server2.pl. I have seen it and understood the flow of the script.But when I was running that script it tells the following error. Can't locate loadable object for module ESL in @INC (@INC contains: /etc/perl /usr/local/lib/perl/5.8.8 /usr/local/share/perl/5.8.8 /usr/lib/perl5 /usr/share/perl5 /usr/lib/perl/5.8 /usr/share/perl/5.8 /usr/local/lib/site_perl .) at ESL.pm line 11 Compilation failed in require at server2.pl line 1. Then I checked the 11th line in the ESL.pm that line is (bootstrap ESL;) But in perl directory there is no directory called ESL. What would be the issue?. Is ESL necessary is necessary for implementing IVR using event outbound socket? -- Regards, Thangappan.M -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090728/052ac2b0/attachment-0002.html From msc at freeswitch.org Mon Jul 27 23:53:47 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 27 Jul 2009 23:53:47 -0700 Subject: [Freeswitch-users] ESL problem In-Reply-To: <7aa29e790907272316m43671f92jf9f5442f6c7caf22@mail.gmail.com> References: <7aa29e790907272316m43671f92jf9f5442f6c7caf22@mail.gmail.com> Message-ID: <87f2f3b90907272353n142f268atb804509038147f0b@mail.gmail.com> Make certain that you've built both libesl and the Perl mod. Change directory into /usr/src/freeswitch.trunk/libs/esl (or whatever your path is to libs/esl) and do these commands: make make perlmod Then give it another shot. -MC On Mon, Jul 27, 2009 at 11:16 PM, Thangappan.M wrote: > Dear all, > > In the previous post, I got the information that using event outbound > socket we can implement the IVR and also see the example in > libs/esl/perl/server2.pl. > > I have seen it and understood the flow of the script.But when I was > running that script it tells the following error. > > Can't locate loadable object for module ESL in @INC (@INC contains: > /etc/perl /usr/local/lib/perl/5.8.8 /usr/local/share/perl/5.8.8 > /usr/lib/perl5 /usr/share/perl5 /usr/lib/perl/5.8 /usr/share/perl/5.8 > /usr/local/lib/site_perl .) at ESL.pm line 11 > Compilation failed in require at server2.pl line 1. > > Then I checked the 11th line in the ESL.pm that line is (bootstrap ESL;) > But in perl directory there is no directory called ESL. > > What would be the issue?. > Is ESL necessary is necessary for implementing IVR using event outbound > socket? > > > -- > Regards, > Thangappan.M > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20090727/4184f352/attachment-0002.html From msc at freeswitch.org Tue Jul 28 00:02:06 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 28 Jul 2009 00:02:06 -0700 Subject: [Freeswitch-users] Connecting FreeSWITCH with Asterisk In-Reply-To: <1452e2980907272310y49bdab1fg39b490ba83c845@mail.gmail.com> References: <1452e2980907272310y49bdab1fg39b490ba83c845@mail.gmail.com> Message-ID: <87f2f3b90907280002v442c1071vb8bd0525dce290c8@mail.gmail.com> Before you go any further, could you let us know what you are trying to accomplish? What kind of calls need to go to/from Asterisk/FreeSWITCH? Do you require some sort of authentication? Are the FS and Ast machines on the same LAN? It might help for you to pastebin the output from the FS CLI when you make a test call to the Asterisk box as that might give you some clue as to what isn't working. -MC On Mon, Jul 27, 2009 at 11:10 PM, velusamy velu wrote: > Dear All, > > I have tried to connect the FreeSWITCH with Asterisk > > I have followed steps which is provided in the following link, > http://wiki.freeswitch.org/wiki/Connecting_Freeswitch_And_Asterisk > > I have tried to call "2000" from FreeSWITCH, but I have received the > following message in Asterisk console > > "NOTICE[6346]: chan_sip.c:10543 handle_request_invite: Failed to > authenticate user "Velusamy" > >;tag=69Q648H9NjrSK" > > I have read "Using Authentication" topic in the link, But I did understand > that topic.. > They have mentioned "HOSTNAME.DOMAIN.COM" in that topic. Which hostname I > have to specify here? > > Please help me.... > > Regards, > Velusamy.K > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20090728/498b4a08/attachment-0002.html From velu.technical at gmail.com Tue Jul 28 00:32:22 2009 From: velu.technical at gmail.com (velusamy velu) Date: Tue, 28 Jul 2009 13:02:22 +0530 Subject: [Freeswitch-users] Connecting FreeSWITCH with Asterisk In-Reply-To: <87f2f3b90907280002v442c1071vb8bd0525dce290c8@mail.gmail.com> References: <1452e2980907272310y49bdab1fg39b490ba83c845@mail.gmail.com> <87f2f3b90907280002v442c1071vb8bd0525dce290c8@mail.gmail.com> Message-ID: <1452e2980907280032r46a0760ekdef3cb6c29242b84@mail.gmail.com> Dear, I am just testing that how to connect FreeSWITCH with Asterisk. I don't want any sort of authentication. Yes, the FS and Asterisk are on the same LAN.. My intention is that When I call an extension from FS, the dial plan should bridge a user in Asterisk.. Please give some suggestions... Thanks in Advance. Regards, Velusamy. On Tue, Jul 28, 2009 at 12:32 PM, Michael Collins wrote: > Before you go any further, could you let us know what you are trying to > accomplish? What kind of calls need to go to/from Asterisk/FreeSWITCH? Do > you require some sort of authentication? Are the FS and Ast machines on the > same LAN? > > It might help for you to pastebin the output from the FS CLI when you make > a test call to the Asterisk box as that might give you some clue as to what > isn't working. > > -MC > > On Mon, Jul 27, 2009 at 11:10 PM, velusamy velu wrote: > >> Dear All, >> >> I have tried to connect the FreeSWITCH with Asterisk >> >> I have followed steps which is provided in the following link, >> http://wiki.freeswitch.org/wiki/Connecting_Freeswitch_And_Asterisk >> >> I have tried to call "2000" from FreeSWITCH, but I have received the >> following message in Asterisk console >> >> "NOTICE[6346]: chan_sip.c:10543 handle_request_invite: Failed to >> authenticate user "Velusamy" >> >;tag=69Q648H9NjrSK" >> >> I have read "Using Authentication" topic in the link, But I did understand >> that topic.. >> They have mentioned "HOSTNAME.DOMAIN.COM" in that topic. Which hostname >> I have to specify here? >> >> Please help me.... >> >> Regards, >> Velusamy.K >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20090728/de45ac70/attachment-0002.html From brian at freeswitch.org Tue Jul 28 01:26:05 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 28 Jul 2009 03:26:05 -0500 Subject: [Freeswitch-users] ESL problem In-Reply-To: <87f2f3b90907272353n142f268atb804509038147f0b@mail.gmail.com> References: <7aa29e790907272316m43671f92jf9f5442f6c7caf22@mail.gmail.com> <87f2f3b90907272353n142f268atb804509038147f0b@mail.gmail.com> Message-ID: Don't forget you need to install libs/esl/perl/ESL.so and libs/esl/ perl/ESL.pm into your system perl library path. /b On Jul 28, 2009, at 1:53 AM, Michael Collins wrote: > Make certain that you've built both libesl and the Perl mod. Change > directory into /usr/src/freeswitch.trunk/libs/esl (or whatever your > path is to libs/esl) and do these commands: > make > make perlmod > > Then give it another shot. > -MC From jgonzalez at sqli.com Tue Jul 28 02:32:23 2009 From: jgonzalez at sqli.com (julien) Date: Tue, 28 Jul 2009 11:32:23 +0200 Subject: [Freeswitch-users] SIP trunk (PBX - FreeSwitch) routing problem. In-Reply-To: <60EC1D48-40FD-4BF9-B5D2-D8FAE1EBEEEC@freeswitch.org> References: <4A6DCEA5.2090502@sqli.com> <60EC1D48-40FD-4BF9-B5D2-D8FAE1EBEEEC@freeswitch.org> Message-ID: <4A6EC5A7.60802@sqli.com> Hello brian, It was not exactly at the bottom but before I tried to put it higher in the dialplan but it still doesn't work (with the same error). Thanks for your help. Brian West a ?crit : > I have to guess that you put this at the bottom of the default.xml? > > /b > > On Jul 27, 2009, at 10:58 AM, julien wrote: > > >> And in the dialplan default.xml : >> >> >> >> >> >> >> >> >> >> >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From gregory.charles at sogeti.com Tue Jul 28 02:43:42 2009 From: gregory.charles at sogeti.com (Gregory Charles) Date: Tue, 28 Jul 2009 11:43:42 +0200 Subject: [Freeswitch-users] SIP instant messaging presence signaling doesn't work. Message-ID: <20090728114342.7ej52d57dick0kss@mail.sogeti.com> Hi everybody, ? ?I intend to use Freeswitch with two Ekiga Softphones. SIP Instant? messaging works between the two softphones but SIP presence signaling? is not managed by the softphones. I try to use other softphones? (QuteCom and SIPCommunicator) and it is the same. I have the following? error in my FreeSwitch console: ? ?[WARNING] sofia.c:75 sofia_handle_sip_r_notify() delete? subscriptions for failed notify ? ?Is there any special configuration for SIP instant messaging presence? ? ?Thanks. ? ?G.C. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090728/07ced992/attachment-0004.html -------------- next part -------------- Hi everybody, I intend to use Freeswitch with two Ekiga Softphones. SIP Instant messaging works between the two softphones but SIP presence signaling is not managed by the softphones. I try to use other softphones (QuteCom and SIPCommunicator) and it is the same. I have the following error in my FreeSwitch console: [WARNING] sofia.c:75 sofia_handle_sip_r_notify() delete subscriptions for failed notify Is there any special configuration for SIP instant messaging presence? Thanks. G.C. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090728/07ced992/attachment-0005.html From brian at freeswitch.org Tue Jul 28 02:46:29 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 28 Jul 2009 04:46:29 -0500 Subject: [Freeswitch-users] SIP trunk (PBX - FreeSwitch) routing problem. In-Reply-To: <4A6EC5A7.60802@sqli.com> References: <4A6DCEA5.2090502@sqli.com> <60EC1D48-40FD-4BF9-B5D2-D8FAE1EBEEEC@freeswitch.org> <4A6EC5A7.60802@sqli.com> Message-ID: Well press F8 and increase the debug level.. then try again you'll prob. see that its not finding it NOR matching it anywhere in your dialplan. /b On Jul 28, 2009, at 4:32 AM, julien wrote: > Hello brian, > It was not exactly at the bottom but before > > > > I tried to put it higher in the dialplan but it still doesn't work > (with > the same error). > > Thanks for your help. From jason at jasonjgw.net Tue Jul 28 02:58:18 2009 From: jason at jasonjgw.net (Jason White) Date: Tue, 28 Jul 2009 19:58:18 +1000 Subject: [Freeswitch-users] SIP trunk (PBX - FreeSwitch) routing problem. In-Reply-To: <4A6EC5A7.60802@sqli.com> References: <4A6DCEA5.2090502@sqli.com> <60EC1D48-40FD-4BF9-B5D2-D8FAE1EBEEEC@freeswitch.org> <4A6EC5A7.60802@sqli.com> Message-ID: <20090728095818.GA23876@jdc.jasonjgw.net> julien wrote: > It was not exactly at the bottom but before > > Why not put it in the default directory, from which it will be included by the above line? If necessary, you could comment out any entries in default.xml that might be matched first. I've debugged this kind of problem before, and the best solution has always been to read the logs carefully to see which extensions matched (or didn't match). Also, if necessary, check out freeswitch/log/freeswitch.xml.fsxml to see where your extension ends up in the final dial plan. From mike at jerris.com Tue Jul 28 05:37:25 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 28 Jul 2009 08:37:25 -0400 Subject: [Freeswitch-users] SIP instant messaging presence signaling doesn't work. In-Reply-To: <20090728114342.7ej52d57dick0kss@mail.sogeti.com> References: <20090728114342.7ej52d57dick0kss@mail.sogeti.com> Message-ID: You must turn on the option to manage presence in the sip profile. Mike On Jul 28, 2009, at 5:43 AM, Gregory Charles wrote: > Hi everybody, > > I intend to use Freeswitch with two Ekiga Softphones. SIP Instant > messaging works between the two softphones but SIP presence signaling > is not managed by the softphones. I try to use other softphones > (QuteCom and SIPCommunicator) and it is the same. I have the following > error in my FreeSwitch console: > > [WARNING] sofia.c:75 sofia_handle_sip_r_notify() delete > subscriptions for failed notify > > Is there any special configuration for SIP instant messaging > presence? > > Thanks. > > G.C. > Hi everybody, > > I intend to use Freeswitch with two Ekiga Softphones. SIP Instant > messaging works between the two softphones but SIP presence > signaling is not managed by the softphones. I try to use other > softphones (QuteCom and SIPCommunicator) and it is the same. I have > the following error in my FreeSwitch console: > > [WARNING] sofia.c:75 sofia_handle_sip_r_notify() delete > subscriptions for failed notify > > Is there any special configuration for SIP instant messaging presence? > > Thanks. > > G.C. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From jgonzalez at sqli.com Tue Jul 28 08:00:58 2009 From: jgonzalez at sqli.com (julien) Date: Tue, 28 Jul 2009 17:00:58 +0200 Subject: [Freeswitch-users] SIP trunk (PBX - FreeSwitch) routing problem. In-Reply-To: References: <4A6DCEA5.2090502@sqli.com> <60EC1D48-40FD-4BF9-B5D2-D8FAE1EBEEEC@freeswitch.org> <4A6EC5A7.60802@sqli.com> Message-ID: <4A6F12AA.1030605@sqli.com> Thanks for the tip Brian. It seems that the extension matches successfully in the dialplan (PASS, instead of FAIL for all other entries of the dialplan) : Dialplan: sofia/internal/[EMAIL PROTECTED] parsing [default->pbxlyon] continue=false Dialplan: sofia/internal/[EMAIL PROTECTED] Regex (PASS) [pbxlyon] destination_number(300) =~ /300/ break=on-false But it leads nowhere. After the match the connection to the PBX fails : 2009-07-28 16:16:43.963836 [NOTICE] switch_channel.c:602 New Channel sofia/external/300 [46fca878-7b81-11de-a9c2-0f49fee5280a] 2009-07-28 16:16:43.963836 [DEBUG] mod_sofia.c:2751 (sofia/external/300) State Change CS_NEW -> CS_INIT 2009-07-28 16:16:43.963836 [DEBUG] switch_core_session.c:933 Send signal sofia/external/300 [BREAK] 2009-07-28 16:16:43.973759 [DEBUG] switch_core_state_machine.c:397 (sofia/external/300) Running State Change CS_INIT 2009-07-28 16:16:43.973759 [DEBUG] switch_core_state_machine.c:480 (sofia/external/300) State INIT 2009-07-28 16:16:43.973759 [DEBUG] mod_sofia.c:83 sofia/external/300 SOFIA INIT 2009-07-28 16:16:43.975221 [DEBUG] mod_sofia.c:111 (sofia/external/300) State Change CS_INIT -> CS_ROUTING 2009-07-28 16:16:43.975221 [DEBUG] switch_core_session.c:933 Send signal sofia/external/300 [BREAK] 2009-07-28 16:16:43.975221 [DEBUG] sofia.c:3215 Channel sofia/external/300 entering state [calling][0] 2009-07-28 16:16:43.975221 [DEBUG] switch_core_state_machine.c:480 (sofia/external/300) State INIT going to sleep 2009-07-28 16:16:43.975221 [DEBUG] switch_core_state_machine.c:397 (sofia/external/300) Running State Change CS_ROUTING 2009-07-28 16:16:43.975221 [DEBUG] switch_core_state_machine.c:483 (sofia/external/300) State ROUTING 2009-07-28 16:16:43.975221 [DEBUG] mod_sofia.c:130 sofia/external/300 SOFIA ROUTING 2009-07-28 16:16:43.975221 [DEBUG] switch_ivr_originate.c:63 (sofia/external/300) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2009-07-28 16:16:43.975221 [DEBUG] switch_core_session.c:933 Send signal sofia/external/300 [BREAK] 2009-07-28 16:16:43.976763 [DEBUG] switch_core_state_machine.c:483 (sofia/external/300) State ROUTING going to sleep 2009-07-28 16:16:43.976763 [DEBUG] switch_core_state_machine.c:397 (sofia/external/300) Running State Change CS_CONSUME_MEDIA 2009-07-28 16:16:43.976763 [DEBUG] switch_core_state_machine.c:502 (sofia/external/300) State CONSUME_MEDIA 2009-07-28 16:16:44.82786 [DEBUG] sofia.c:3215 Channel sofia/external/300 entering state [terminated][404] 2009-07-28 16:16:44.82786 [NOTICE] sofia.c:3775 Hangup sofia/external/300 [CS_CONSUME_MEDIA] [NO_ROUTE_DESTINATION] It looks to me that it's more a problem from the gateway than from the dialplan? Don't you think so? Do you think the way I defined my gateway is good for a connexion to a PBX ? Thanks for your replies Brian and Jason. Brian West a ?crit : > Well press F8 and increase the debug level.. then try again you'll > prob. see that its not finding it NOR matching it anywhere in your > dialplan. > > /b > > On Jul 28, 2009, at 4:32 AM, julien wrote: > > >> Hello brian, >> It was not exactly at the bottom but before >> >> >> >> I tried to put it higher in the dialplan but it still doesn't work >> (with >> the same error). >> >> Thanks for your help. >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.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 Jul 28 08:11:17 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 28 Jul 2009 10:11:17 -0500 Subject: [Freeswitch-users] SIP trunk (PBX - FreeSwitch) routing problem. In-Reply-To: <4A6F12AA.1030605@sqli.com> References: <4A6DCEA5.2090502@sqli.com> <60EC1D48-40FD-4BF9-B5D2-D8FAE1EBEEEC@freeswitch.org> <4A6EC5A7.60802@sqli.com> <4A6F12AA.1030605@sqli.com> Message-ID: The remote end said 404 /b On Jul 28, 2009, at 10:00 AM, julien wrote: > 2009-07-28 16:16:44.82786 [DEBUG] sofia.c:3215 Channel > sofia/external/300 entering state [terminated][404] From vkozak at abisoft.spb.ru Tue Jul 28 08:16:45 2009 From: vkozak at abisoft.spb.ru (Kozak Vladimir) Date: Tue, 28 Jul 2009 19:16:45 +0400 Subject: [Freeswitch-users] originate in dialplan Message-ID: <793928086CB844E0852A49DE57CADEC8@abisoft.biz> Hello, Please tell me, how can I execute originate new call and uuid_bridge in dial plan. I tried to make like thise: result: [ERR] switch_core_session.c:1239 switch_core_session_execute_application() Invalid Application originate [ERR] switch_core_session.c:1239 switch_core_session_execute_application() Invalid Application uuid_bridge Best regards. vkozak -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090728/49be300c/attachment-0002.html From msc at freeswitch.org Tue Jul 28 09:07:20 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 28 Jul 2009 09:07:20 -0700 Subject: [Freeswitch-users] originate in dialplan In-Reply-To: <793928086CB844E0852A49DE57CADEC8@abisoft.biz> References: <793928086CB844E0852A49DE57CADEC8@abisoft.biz> Message-ID: <87f2f3b90907280907r1bcf4705i695dcc9f3ed4091a@mail.gmail.com> What exactly are you trying to accomplish with this dialplan entry? That will help us answer your question. -MC 2009/7/28 Kozak Vladimir > Hello, > > Please tell me, how can I execute originate new call and uuid_bridge in > dial plan. > I tried to make like thise: > data="user/$${destination_end_point} &playback(${hold_music})"/> > data="user/$${destination_end_point}, &playback($${hold_music})"/> > > > result: > [ERR] switch_core_session.c:1239 > switch_core_session_execute_application() Invalid Application originate > [ERR] switch_core_session.c:1239 > switch_core_session_execute_application() Invalid Application uuid_bridge > Best regards. > vkozak > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20090728/6012eb5e/attachment-0002.html From helmut.kuper at ewetel.de Tue Jul 28 09:14:03 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Tue, 28 Jul 2009 18:14:03 +0200 Subject: [Freeswitch-users] CELT codec code number Message-ID: <4A6F23CB.6090405@ewetel.de> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, today I finylly got a working Ekiga Softphone version which is able to use high quality celt codec with FS :) On my way to get it work with FS I found that Ekiga currently uses codec code 95 in SDP while FS uses 114. Changing FS to 95 made it works so I'm now able to listen to my PBX-based MP3-Player on Windows Desktop instead of using Ubuntu. veeeeeerrry cool work of FS team !!!!! Concerning the codec code 95, 114 or whatever I found the link below, which states that every codec code between 96 and 127 is OK but it seems they prefer 97 ... http://tools.ietf.org/html/draft-valin-celt-rtp-profile-02#section-5 You can get windows Ekiga 3.2.5 (with celt 0.5.1) here: http://wwwuser.gwdg.de/~mrickma/ekiga/ekiga-setup-3.2.5-release.exe regards Helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) iD8DBQFKbyPL4tZeNddg3dwRAiDMAKChqgWeirYklgra5nN7NGwZSpK6wQCgjYox Q/okubHauhgjtoiogzFM9mI= =Ml4q -----END PGP SIGNATURE----- From zolotov at altron.ua Tue Jul 28 09:24:38 2009 From: zolotov at altron.ua (Evgeniy Zolotov) Date: Tue, 28 Jul 2009 19:24:38 +0300 Subject: [Freeswitch-users] originate in dialplan In-Reply-To: <793928086CB844E0852A49DE57CADEC8@abisoft.biz> References: <793928086CB844E0852A49DE57CADEC8@abisoft.biz> Message-ID: <1783498449.20090728192438@altron.ua> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090728/579a61a4/attachment-0002.html From brian at freeswitch.org Tue Jul 28 09:23:36 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 28 Jul 2009 11:23:36 -0500 Subject: [Freeswitch-users] CELT codec code number In-Reply-To: <4A6F23CB.6090405@ewetel.de> References: <4A6F23CB.6090405@ewetel.de> Message-ID: <8AD48BD2-8752-4F99-9CF7-96955C485E98@freeswitch.org> On Jul 28, 2009, at 11:14 AM, Helmut Kuper wrote: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > Hello, > > today I finylly got a working Ekiga Softphone version which is able to > use high quality celt codec with FS :) > > > On my way to get it work with FS I found that Ekiga currently uses > codec > code 95 in SDP while FS uses 114. Changing FS to 95 made it works so > I'm > now able to listen to my PBX-based MP3-Player on Windows Desktop > instead > of using Ubuntu. It should work if they use different codec numbers.... I suspect we are sending on 114 and receiving on 95 which is what should take place. This is one of those areas most people fail to implement properly. We send the remote our RTP map they send us theirs... Can you get a packet capture of this taking place so I can verify who is at fault? /b > veeeeeerrry cool work of FS team !!!!! > > Concerning the codec code 95, 114 or whatever I found the link below, > which states that every codec code between 96 and 127 is OK but it > seems > they prefer 97 ... > > http://tools.ietf.org/html/draft-valin-celt-rtp-profile-02#section-5 > > > You can get windows Ekiga 3.2.5 (with celt 0.5.1) here: > http://wwwuser.gwdg.de/~mrickma/ekiga/ekiga-setup-3.2.5-release.exe > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090728/e946c19f/attachment-0002.html From mike at jerris.com Tue Jul 28 10:06:21 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 28 Jul 2009 13:06:21 -0400 Subject: [Freeswitch-users] CELT codec code number In-Reply-To: <8AD48BD2-8752-4F99-9CF7-96955C485E98@freeswitch.org> References: <4A6F23CB.6090405@ewetel.de> <8AD48BD2-8752-4F99-9CF7-96955C485E98@freeswitch.org> Message-ID: <206116BA-A08A-45A4-B72F-990E5CC680FE@jerris.com> using 95 is wrong. That is not part of the dynamic range for unassigned codecs. This needs to be fixed on their side. MIke On Jul 28, 2009, at 12:23 PM, Brian West wrote: > > On Jul 28, 2009, at 11:14 AM, Helmut Kuper wrote: > >> -----BEGIN PGP SIGNED MESSAGE----- >> Hash: SHA1 >> >> Hello, >> >> today I finylly got a working Ekiga Softphone version which is able >> to >> use high quality celt codec with FS :) >> >> >> On my way to get it work with FS I found that Ekiga currently uses >> codec >> code 95 in SDP while FS uses 114. Changing FS to 95 made it works >> so I'm >> now able to listen to my PBX-based MP3-Player on Windows Desktop >> instead >> of using Ubuntu. > > It should work if they use different codec numbers.... I suspect we > are sending on 114 and receiving on 95 which is what should take > place. This is one of those areas most people fail to implement > properly. We send the remote our RTP map they send us theirs... Can > you get a packet capture of this taking place so I can verify who is > at fault? > > /b > > > >> veeeeeerrry cool work of FS team !!!!! >> >> Concerning the codec code 95, 114 or whatever I found the link below, >> which states that every codec code between 96 and 127 is OK but it >> seems >> they prefer 97 ... >> >> http://tools.ietf.org/html/draft-valin-celt-rtp-profile-02#section-5 >> >> >> You can get windows Ekiga 3.2.5 (with celt 0.5.1) here: >> http://wwwuser.gwdg.de/~mrickma/ekiga/ekiga-setup-3.2.5-release.exe >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20090728/358c4680/attachment-0002.html From brian at freeswitch.org Tue Jul 28 10:13:31 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 28 Jul 2009 12:13:31 -0500 Subject: [Freeswitch-users] CELT codec code number In-Reply-To: <206116BA-A08A-45A4-B72F-990E5CC680FE@jerris.com> References: <4A6F23CB.6090405@ewetel.de> <8AD48BD2-8752-4F99-9CF7-96955C485E98@freeswitch.org> <206116BA-A08A-45A4-B72F-990E5CC680FE@jerris.com> Message-ID: I totally missed this at first... but 95 wouldn't dynamically work because its not 96-127 /b On Jul 28, 2009, at 12:06 PM, Michael Jerris wrote: > using 95 is wrong. That is not part of the dynamic range for > unassigned codecs. This needs to be fixed on their side. > > MIke From kristian.kielhofner at gmail.com Tue Jul 28 10:28:03 2009 From: kristian.kielhofner at gmail.com (Kristian Kielhofner) Date: Tue, 28 Jul 2009 13:28:03 -0400 Subject: [Freeswitch-users] AbsoluteTimeout: The FreeSWITCH way Message-ID: <2d9149cd0907281028u7bb8ed52ufe72370aaae9311c@mail.gmail.com> Hello everyone, I need to set a maximum call duration. What is the current recommended way to implement this in FreeSWITCH? I'm looking for something similar to AbsoluteTimeout() in Asterisk. Thanks! -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com From saeedahmad1981 at gmail.com Tue Jul 28 10:48:11 2009 From: saeedahmad1981 at gmail.com (Saeed Ahmad) Date: Tue, 28 Jul 2009 19:48:11 +0200 Subject: [Freeswitch-users] AbsoluteTimeout: The FreeSWITCH way In-Reply-To: <2d9149cd0907281028u7bb8ed52ufe72370aaae9311c@mail.gmail.com> References: <2d9149cd0907281028u7bb8ed52ufe72370aaae9311c@mail.gmail.com> Message-ID: On Tue, Jul 28, 2009 at 7:28 PM, Kristian Kielhofner < kristian.kielhofner at gmail.com> wrote: > Hello everyone, > > I need to set a maximum call duration. What is the current > recommended way to implement this in FreeSWITCH? I'm looking for > something similar to AbsoluteTimeout() in Asterisk. > > Thanks! > > -- > Kristian Kielhofner > http://www.astlinux.org > http://blog.krisk.org > http://www.star2star.com > http://www.submityoursip.com > http://www.voalte.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/20090728/acb9d27e/attachment-0002.html From msc at freeswitch.org Tue Jul 28 10:50:49 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 28 Jul 2009 10:50:49 -0700 Subject: [Freeswitch-users] AbsoluteTimeout: The FreeSWITCH way In-Reply-To: <2d9149cd0907281028u7bb8ed52ufe72370aaae9311c@mail.gmail.com> References: <2d9149cd0907281028u7bb8ed52ufe72370aaae9311c@mail.gmail.com> Message-ID: <87f2f3b90907281050g14cd8c74g3bfa3e5481fa2ecf@mail.gmail.com> What needs to happen at the end of the timeout? In any case you can use the sched_XXX APIs: sched_api sched_transfer sched_hangup You can get fancy or just hangup up on the call after X number of seconds... :) -MC On Tue, Jul 28, 2009 at 10:28 AM, Kristian Kielhofner < kristian.kielhofner at gmail.com> wrote: > Hello everyone, > > I need to set a maximum call duration. What is the current > recommended way to implement this in FreeSWITCH? I'm looking for > something similar to AbsoluteTimeout() in Asterisk. > > Thanks! > > -- > Kristian Kielhofner > http://www.astlinux.org > http://blog.krisk.org > http://www.star2star.com > http://www.submityoursip.com > http://www.voalte.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/20090728/39c90253/attachment-0002.html From mrene_lists at avgs.ca Tue Jul 28 10:54:58 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Tue, 28 Jul 2009 13:54:58 -0400 Subject: [Freeswitch-users] AbsoluteTimeout: The FreeSWITCH way In-Reply-To: <87f2f3b90907281050g14cd8c74g3bfa3e5481fa2ecf@mail.gmail.com> References: <2d9149cd0907281028u7bb8ed52ufe72370aaae9311c@mail.gmail.com> <87f2f3b90907281050g14cd8c74g3bfa3e5481fa2ecf@mail.gmail.com> Message-ID: You can also schedule a playback then a hangup, what comes after the ! is the hangup cause. sched_broadcast,Schedule a broadcast in the future,[+]