January 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

488 Not Acceptable Here (INCOMPATIBLE_DESTINATION) with JsSIP and OverSIP + FreeSWITCH
by Paan Singh 17 Jan '13

17 Jan '13

Hi, I'm using JsSIP from a webpage to make a SIP call to FS, using OverSIP as a Websocket->SIP proxy. I'm able to Register successfully, but when I make a call from JsSIP UA to FreeSWITCH, I get a 180 Ringing, but after that I get a 488 Not Acceptable here.. Please help me figure out what I'm doing wrong here. My inbound_codec_prefs is : <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729,G7221@32000h ,G7221@16000h,G722,PCMU,PCMA,GSM,isac@16000h,isac,opus"/> Here is FreeSWITCH's log for the incoming call : ============================================= port.c:2730 tport_wakeup_pri() tport_wakeup_pri(0x26036f0): events IN tport.c:2845 tport_recv_event() tport_recv_event(0x26036f0) tport.c:3186 tport_recv_iovec() tport_recv_iovec(0x26036f0) msg 0x2913be0 from (udp/116.214.15.45:5080) has 3113 bytes, veclen = 1 tport.c:3004 tport_deliver() tport_deliver(0x26036f0): msg 0x2913be0 (3113 bytes) from udp/121.245.35.150:5080/sip next=(nil) nta.c:2803 agent_recv_request() nta: received INVITE sip:30201009@116.214.15.45 SIP/2.0 (CSeq 2141) nta.c:3161 agent_aliases() nta: canonizing sip:30201009@116.214.15.45 with contact nta.c:3002 agent_recv_request() nta: INVITE (2141) going to a default leg nta.c:1348 set_timeout() nta: timer set to 200 ms nua_server.c:102 nua_stack_process_request() nua: nua_stack_process_request: entering nua_stack.c:899 nh_create() nua: nh_create: entering nua_common.c:108 nh_create_handle() nua: nh_create_handle: entering nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering soa.c:282 soa_clone() soa_clone(static::0x26122e0, 0x2606e20, 0x7f91840c4c50) called soa.c:405 soa_set_params() soa_set_params(static::0x7f91840b25e0, ...) called nta.c:4313 nta_leg_tcreate() nta_leg_tcreate(0x7f91840356d0) soa.c:1302 soa_init_offer_answer() soa_init_offer_answer(static::0x7f91840b25e0) called soa.c:1171 soa_set_remote_sdp() soa_set_remote_sdp(static::0x7f91840b25e0, (nil), 0x285676c, 2333) called nua_dialog.c:338 nua_dialog_usage_add() nua(0x7f91840c4c50): adding session usage tport.c:3238 tport_tsend() tport_tsend(0x26036f0) tpn = UDP/ 121.245.35.150:5065 tport.c:4026 tport_resolve() tport_resolve addrinfo = 121.245.35.150:5065 tport.c:4660 tport_by_addrinfo() tport_by_addrinfo(0x26036f0): not found by name UDP/121.245.35.150:5065 tport.c:3574 tport_vsend() tport_vsend(0x26036f0): 564 bytes of 564 to udp/ 121.245.35.150:5065 tport.c:3472 tport_send_msg() tport_vsend returned 564 nta.c:6678 incoming_reply() nta: sent 100 Trying for INVITE (2141) nua_stack.c:271 nua_stack_event() nua(0x7f91840c4c50): event i_invite 100 Trying nua_session.c:4135 signal_call_state_change() nua(0x7f91840c4c50): call state changed: init -> received, received offer soa.c:1098 soa_get_remote_sdp() soa_get_remote_sdp(static::0x7f91840b25e0, [0x7f91aa98a5e8], [0x7f91aa98a5e0], [(nil)]) called nua_stack.c:271 nua_stack_event() nua(0x7f91840c4c50): event i_state 100 Trying nua_stack.c:359 nua_application_event() nua: nua_application_event: entering nua.c:342 nua_handle_bind() nua: nua_handle_bind: entering 2013-01-17 00:25:27.733506 [NOTICE] switch_channel.c:968 New Channel sofia/external/1000@116.214.15.45:5080[4aec7fd8-600e-11e2-8706-1365ba01ad52] 2013-01-17 00:25:27.733506 [DEBUG] switch_core_session.c:975 Send signal sofia/external/1000@116.214.15.45:5080 [BREAK] nua_stack.c:359 nua_application_event() nua: nua_application_event: entering 2013-01-17 00:25:27.733506 [DEBUG] switch_core_session.c:975 Send signal sofia/external/1000@116.214.15.45:5080 [BREAK] 2013-01-17 00:25:27.733506 [DEBUG] switch_core_state_machine.c:415 (sofia/external/1000@116.214.15.45:5080) Running State Change CS_NEW 2013-01-17 00:25:27.733506 [DEBUG] switch_core_state_machine.c:433 (sofia/external/1000@116.214.15.45:5080) State NEW nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering 2013-01-17 00:25:27.753511 [DEBUG] sofia.c:5576 Channel sofia/external/ 1000@116.214.15.45:5080 entering state [received][100] 2013-01-17 00:25:27.753511 [DEBUG] sofia.c:5587 Remote SDP: v=0 o=- 2957375511 2 IN IP4 127.0.0.1 s=- t=0 0 a=group:BUNDLE audio video m=audio 40111 RTP/SAVPF 103 104 0 8 106 105 13 126 c=IN IP4 121.245.35.150 a=rtcp:40111 IN IP4 121.245.35.150 a=candidate:877273017 1 udp 2113937151 121.245.35.150 39842 typ host generation 0 a=candidate:877273017 2 udp 2113937151 121.245.35.150 39842 typ host generation 0 a=candidate:4173799466 1 udp 1677729535 121.245.35.150 40111 typ srflx generation 0 a=candidate:4173799466 2 udp 1677729535 121.245.35.150 40111 typ srflx generation 0 a=candidate:2060051273 1 tcp 1509957375 121.245.35.150 54291 typ host generation 0 a=candidate:2060051273 2 tcp 1509957375 121.245.35.150 54291 typ host generation 0 a=ice-ufrag:qiKF0ErbxaXldKI0 a=ice-pwd:XlMXg/i8jwnCAgwnuo9KyZBw a=ice-options:google-ice a=mid:audio a=rtcp-mux a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:bX1U9Kr56lIcZBLQvS2pmRW3nAcTG9H0uahawqj2 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:126 telephone-event/8000 a=ssrc:4278252985 cname:fMHr4uVHPcpHFfGu a=ssrc:4278252985 mslabel:OEZOjX953DJU0k0J3IKmuasdqH13b96rGxhx a=ssrc:4278252985 label:OEZOjX953DJU0k0J3IKmuasdqH13b96rGxhx00 m=video 40111 RTP/SAVPF 100 101 102 c=IN IP4 121.245.35.150 a=rtcp:40111 IN IP4 121.245.35.150 a=candidate:877273017 1 udp 2113937151 121.245.35.150 39842 typ host generation 0 a=candidate:877273017 2 udp 2113937151 121.245.35.150 39842 typ host generation 0 a=candidate:4173799466 1 udp 1677729535 121.245.35.150 40111 typ srflx generation 0 a=candidate:4173799466 2 udp 1677729535 121.245.35.150 40111 typ srflx generation 0 a=candidate:2060051273 1 tcp 1509957375 121.245.35.150 54291 typ host generation 0 a=candidate:2060051273 2 tcp 1509957375 121.245.35.150 54291 typ host generation 0 a=ice-ufrag:qiKF0ErbxaXldKI0 a=ice-pwd:XlMXg/i8jwnCAgwnuo9KyZBw a=ice-options:google-ice a=mid:video a=rtcp-mux a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:bX1U9Kr56lIcZBLQvS2pmRW3nAcTG9H0uahawqj2 a=rtpmap:100 VP8/90000 a=rtpmap:101 red/90000 a=rtpmap:102 ulpfec/90000 a=ssrc:169275022 cname:fMHr4uVHPcpHFfGu a=ssrc:169275022 mslabel:OEZOjX953DJU0k0J3IKmuasdqH13b96rGxhx a=ssrc:169275022 label:OEZOjX953DJU0k0J3IKmuasdqH13b96rGxhx10 2013-01-17 00:25:27.753511 [DEBUG] sofia_glue.c:4978 Set Remote Key [1 AES_CM_128_HMAC_SHA1_80 inline:bX1U9Kr56lIcZBLQvS2pmRW3nAcTG9H0uahawqj2] 2013-01-17 00:25:27.753511 [DEBUG] sofia_glue.c:3155 Set Local Key [1 AES_CM_128_HMAC_SHA1_80 inline:i8vqmBeU4+/sjJBndCbEQ4G87gmpYeVnqxh8Xuzu] 2013-01-17 00:25:27.753511 [DEBUG] sofia_glue.c:5277 No 2833 in SDP. Disable 2833 dtmf and switch to INFO 2013-01-17 00:25:27.753511 [DEBUG] switch_channel.c:2994 (sofia/external/1000@116.214.15:45:5080) Callstate Change DOWN -> HANGUP 2013-01-17 00:25:27.753511 [NOTICE] sofia.c:5878 Hangup sofia/external/ 1000@116.214.15.45:5080 [CS_NEW] [INCOMPATIBLE_DESTINATION] 2013-01-17 00:25:27.753511 [DEBUG] switch_channel.c:3017 Send signal sofia/external/1000@116.214.15.45:5080 [KILL] 2013-01-17 00:25:27.753511 [DEBUG] switch_core_session.c:1291 Send signal sofia/external/1000@116.214.15.45:5080 [BREAK] nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:415 (sofia/external/1000@116.214.15.45:5080) Running State Change CS_HANGUP 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:667 (sofia/external/1000@116.214.15.45:5080) State HANGUP 2013-01-17 00:25:27.753511 [DEBUG] mod_sofia.c:503 Channel sofia/external/ 1000@116.214.15.45:5080 hanging up, cause: INCOMPATIBLE_DESTINATION 2013-01-17 00:25:27.753511 [DEBUG] mod_sofia.c:633 Responding to INVITE with: 488 nua.c:879 nua_respond() nua: nua_respond: entering nua_stack.c:529 nua_signal() nua(0x7f91840c4c50): sent signal r_respond 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:48 sofia/external/1000@116.214.15.45:5080 Standard HANGUP, cause: INCOMPATIBLE_DESTINATION 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:667 (sofia/external/1000@116.214.15.45:5080) State HANGUP going to sleep nua_stack.c:573 nua_stack_signal() nua(0x7f91840c4c50): recv signal r_respond 488 Not Acceptable Here 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:446 (sofia/external/1000@116.214.15.45:5080) State Change CS_HANGUP -> CS_REPORTING nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering 2013-01-17 00:25:27.753511 [DEBUG] switch_core_session.c:1291 Send signal sofia/external/1000@116.214.15.45:5080 [BREAK] soa.c:405 soa_set_params() soa_set_params(static::0x7f91840b25e0, ...) called 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:415 (sofia/external/1000@116.214.15.45:5080) Running State Change CS_REPORTING nua_session.c:2316 nua_invite_server_respond() nua: nua_invite_server_respond: entering soa.c:1214 soa_clear_remote_sdp() soa_clear_remote_sdp(static::0x7f91840b25e0) called tport.c:3238 tport_tsend() tport_tsend(0x26036f0) tpn = UDP/ 121.245.35.150:5065 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:749 (sofia/external/1000@116.214.15.45:5080) State REPORTING tport.c:4026 tport_resolve() tport_resolve addrinfo = 121.245.35.150:5065 tport.c:4660 tport_by_addrinfo() tport_by_addrinfo(0x26036f0): not found by name UDP/121.245.35.150:5065 tport.c:3574 tport_vsend() tport_vsend(0x26036f0): 784 bytes of 784 to udp/ 121.245.35.150:5065 tport.c:3472 tport_send_msg() tport_vsend returned 784 nta.c:6678 incoming_reply() nta: sent 488 Not Acceptable Here for INVITE (2141) nua_dialog.c:397 nua_dialog_usage_remove_at() nua(0x7f91840c4c50): removing session usage nua_session.c:4135 signal_call_state_change() nua(0x7f91840c4c50): call state changed: received -> terminated nua_stack.c:271 nua_stack_event() nua(0x7f91840c4c50): event i_state 488 Not Acceptable Here nua_stack.c:271 nua_stack_event() nua(0x7f91840c4c50): event i_terminated 488 Not Acceptable Here soa.c:358 soa_destroy() soa_destroy(static::0x7f91840b25e0) called nta.c:4366 nta_leg_destroy() nta_leg_destroy(0x7f91840356d0) nua_stack.c:359 nua_application_event() nua: nua_application_event: entering nua_stack.c:359 nua_application_event() nua: nua_application_event: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nua.c:342 nua_handle_bind() nua: nua_handle_bind: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nua.c:921 nua_handle_destroy() nua: nua_handle_destroy: entering nua_stack.c:569 nua_stack_signal() nua(0x7f91840c4c50): recv signal r_destroy nta.c:4366 nta_leg_destroy() nta_leg_destroy((nil)) nua_stack.c:529 nua_signal() nua(0x7f91840c4c50): sent signal r_destroy 2013-01-17 00:25:27.753511 [DEBUG] mod_cdr_sqlite.c:102 Writing SQL to DB: INSERT INTO cdr VALUES ("1000","1000","30201009","public","2013-01-17 00:25:27","","2013-01-17 00:25:27",0,0,"INCOMPATIBLE_DESTINATION","4aec7fd8-600e-11e2-8706-1365ba01ad52","","") 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:92 sofia/external/1000@116.214.15.45:5080 Standard REPORTING, cause: INCOMPATIBLE_DESTINATION 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:749 (sofia/external/1000@116.214.15.45:5080) State REPORTING going to sleep 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:440 (sofia/external/1000@116.214.15.45:5080) State Change CS_REPORTING -> CS_DESTROY 2013-01-17 00:25:27.753511 [DEBUG] switch_core_session.c:1291 Send signal sofia/external/1000@116.214.15.45:5080 [BREAK] 2013-01-17 00:25:27.753511 [DEBUG] switch_core_session.c:1499 Session 1794 (sofia/external/1000@116.214.15.45:5080) Locked, Waiting on external entities 2013-01-17 00:25:27.753511 [NOTICE] switch_core_session.c:1517 Session 1794 (sofia/external/1000@116.214.15.45:5080) Ended 2013-01-17 00:25:27.753511 [NOTICE] switch_core_session.c:1521 Close Channel sofia/external/1000@116.214.15.45:5080 [CS_DESTROY] 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:556 (sofia/external/1000@116.214.15.45:5080) Callstate Change HANGUP -> DOWN 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:559 (sofia/external/1000@116.214.15.45:5080) Running State Change CS_DESTROY 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:569 (sofia/external/1000@116.214.15.45:5080) State DESTROY 2013-01-17 00:25:27.753511 [DEBUG] mod_sofia.c:396 sofia/external/ 1000@116.214.15.45:5080 SOFIA DESTROY 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:99 sofia/external/1000@116.214.15.45:5080 Standard DESTROY 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:569 (sofia/external/1000@116.214.15.45:5080) State DESTROY going to sleep nta.c:1294 agent_timer() nta: timer set next to 322 ms tport.c:2730 tport_wakeup_pri() tport_wakeup_pri(0x26036f0): events IN tport.c:2845 tport_recv_event() tport_recv_event(0x26036f0) tport.c:3186 tport_recv_iovec() tport_recv_iovec(0x26036f0) msg 0x7f9198220070 from (udp/116.214.15.45:5080) has 339 bytes, veclen = 1 tport.c:3004 tport_deliver() tport_deliver(0x26036f0): msg 0x7f9198220070 (339 bytes) from udp/121.245.35.150:5080/sip next=(nil) nta.c:2803 agent_recv_request() nta: received ACK sip:30201009@116.214.15.45SIP/2.0 (CSeq 2141) nta.c:2938 agent_recv_request() nta: ACK (2141) is going to INVITE (2141) nta.c:1294 agent_timer() nta: timer set next to 4823 ms nta.c:7010 _nta_incoming_timer() nta: timer I fired, terminate 488 response nta.c:5718 incoming_reclaim_queued() incoming_reclaim_all((nil), (nil), 0x7f91aa98ac20) nta.c:7061 _nta_incoming_timer() nta_incoming_timer: 0/0 resent, 0/0 tout, 1/1 term, 1/1 free nta.c:1279 agent_timer() nta: timer not set ================================================================ Here is the Log from JsSIP in the browser that receives messages from FreeSWITCH : =========================================================== JsSIP | TRANSPORT | Received WebSocket text message: SIP/2.0 180 Ringing Via: SIP/2.0/TCP 29.62.109.242;branch=z9hG4bK1565615 Record-Route: <sip:192.168.1.8:5060;transport=udp;lr;ovid=69d4d6a0> Record-Route: <sip:774f80554e@192.168.1.8:10080;transport=ws;lr;ovid=69d4d6a0> From: < sip:1000@116.214.15.45:5080>;tag=rj2nc38qem To: <sip:30201009@116.214.15.45>;tag=j4X0e04aZjU2K Call-ID: joshempnx2kuh7smrt3x CSeq: 1974 INVITE Contact: <sip:30201009@116.214.15.45:5080;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.3.13b+git~20130108T190107Z~30e4c71bbe Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, refer Content-Length: 0 P-Asserted-Identity: "30201009" <sip:30201009@116.214.15.45> jssip-0.2.1.js:684 <http://example.com:8080/static/did/tryit/jssip-0.2.1.js> JsSIP | DIALOG | New UAC dialog created: 1 jssip-0.2.1.js:2171<http://example.com:8080/static/did/tryit/jssip-0.2.1.js> JsSIP | EVENT EMITTER | Emitting event: progress jssip-0.2.1.js:170<http://example.com:8080/static/did/tryit/jssip-0.2.1.js> JsSIP | TRANSPORT | Received WebSocket text message: SIP/2.0 488 Not Acceptable Here Via: SIP/2.0/TCP 29.62.109.242;branch=z9hG4bK1565615 Max-Forwards: 10 From: <sip:1000@116.214.15.45:5080>;tag=rj2nc38qem To: < sip:30201009@116.214.15.45>;tag=j4X0e04aZjU2K Call-ID: joshempnx2kuh7smrt3x CSeq: 1974 INVITE User-Agent: FreeSWITCH-mod_sofia/1.3.13b+git~20130108T190107Z~30e4c71bbe Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, refer Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION" Content-Length: 0 P-Asserted-Identity: "30201009" < sip:30201009@116.214.15.45> ============================================================================= What am I doing wrong? Thanks, Paan

3 3

Process REFER and not Hangup the channel
by ashou 17 Jan '13

17 Jan '13

Hi There, I use FS as proxy to handle a call with REFER to. My question is when get a REFER request, how to do the Transfer regarding the REFER TO before hangup the current channel. My mean before send back the NOTIFY, do the INVITE (based on REFER TO), when the INVITE is 200 OK, then send back NOTIFY. my logs: 11:24:14.141700 [DEBUG] sofia.c:6616 Process REFER to [11004(a)111.22.33.1] 11:24:14.141700 [DEBUG] switch_ivr.c:1773 (sofia/external/6175552222(a)12.13.14.15) State Change CS_EXECUTE -> CS_ROUTING 11:24:14.141700 [DEBUG] switch_core_session.c:1287 Send signal sofia/external/6175552222(a)12.13.14.15 [BREAK] 11:24:14.141700 [DEBUG] switch_core_session.c:830 Send signal sofia/external/6175552222(a)12.13.14.15 [BREAK] 11:24:14.141700 [NOTICE] switch_ivr.c:1779 Transfer sofia/external/6175552222(a)12.13.14.15 to XML[11004@public] 11:24:14.161701 [DEBUG] switch_ivr_play_say.c:1682 done playing file local_stream://moh 11:24:14.161701 [DEBUG] switch_ivr_bridge.c:597 BRIDGE THREAD DONE [sofia/external/+18550000000@1.2.3.4:5060] 11:24:14.161701 [DEBUG] switch_ivr_bridge.c:622 Send signal sofia/external/6175552222(a)12.13.14.15 [BREAK] 11:24:14.161701 [DEBUG] switch_channel.c:2979 (sofia/external/+18550000000@1.2.3.4:5060) Callstate Change HELD -> HANGUP the Transfer to XML[11004@public] not DONE yet, the (sofia/external/+18550000000@1.2.3.4:5060) already Hangup, How to delay the Hangup after Transfer is done? Thanks, Ashou -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Process-REFER-and-not-Hangup-… Sent from the freeswitch-users mailing list archive at Nabble.com.

1 0

pre_answer and dtmf
by Frank Carmickle 17 Jan '13

17 Jan '13

Hello I can't seem to read digits with pre_answer. Answer works just fine. I know this used to work. I'm not sure if it's upstreams problem or mine. Where should I look? Thanks --FC

3 3

Freeswitch as SBC and refer method
by Alex Kostenko 17 Jan '13

17 Jan '13

Hi all We have next situation: We use Freeswitch as SBC. And now we stumbled to the situation when we need to translate "refer method" through freeswitch. Now freeswitch processes this option. This question appeared because of wrong amount of session from customer (more then limit for this customer). In some way customer can create more sessions using "refer method" more then he can.

1 0

Re: [Freeswitch-users] need help on call recording - Solved
by haloha 17 Jan '13

17 Jan '13

hi Benoit Raymond it works properly now :) Thank you for your guide Ha On 1/16/13, Benoit Raymond <benoit.raymond(a)amiconcept.com> wrote: > Hi, > > It is most likely recorded but since you have RECORD_STEREO=true, the audio > is ... stereo! > > If you load your wav in an editor (like Goldwave), that should be evident. > So drop the RECORD_STEREO parameter or set it to false (this is the default > anyway) and check if that works. > > Benoit Raymond > AMI Concept Inc. > Business VoIP Solutions / Web Design > Phone: (450) 553-1231 > http://www.amiconcept.com > > -----Message d'origine----- > De : freeswitch-users-bounces(a)lists.freeswitch.org > [mailto:freeswitch-users-bounces@lists.freeswitch.org] De la part de haloha > Envoyé : 16 janvier 2013 03:50 > À : FreeSWITCH Users Help > Objet : [Freeswitch-users] need help on call recording > > hi all > > i try to record all calls to/from local extension so my dialplan > > > <extension name="Local_Extension"> > <condition field="destination_number" expression="^(10[01][0-9])$"> > <action application="export" data="dialed_extension=$1"/> > <action application="set" data="RECORD_TITLE=Recording > ${destination_number} ${caller_id_number} ${strftime(%Y-%m-%d %H:%M)}"/> > <action application="set" data="RECORD_COPYRIGHT=(c) 2011"/> > <action application="set" data="RECORD_SOFTWARE=FreeSWITCH"/> > <action application="set" data="RECORD_ARTIST=FreeSWITCH"/> > <action application="set" data="RECORD_COMMENT=FreeSWITCH"/> > <action application="set" > data="RECORD_DATE=${strftime(%Y-%m-%d %H:%M)}"/> > <action application="set" data="RECORD_STEREO=true"/> > <action application="record_session" > data="/etc/freeswitch/recordings/${strftime(%Y-%m-%d-%H-%M-%S)}_${caller_id_ > number}_${destination_number}.wav"/> >  > <action application="bind_meta_app" data="1 b s > execute_extension::dx XML features"/> > <action application="bind_meta_app" data="2 b s > record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-% > H-%M-%S)}.wav"/> > <action application="bind_meta_app" data="3 b s > execute_extension::cf XML features"/> > <action application="bind_meta_app" data="4 b s > execute_extension::att_xfer XML features"/> > <action application="set" data="ringback=${us-ring}"/> >  > <action application="set" data="call_timeout=30"/> >  > <action application="set" data="hangup_after_bridge=true"/> >  > <action application="set" data="continue_on_fail=true"/> > <action application="hash" > data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_numb > er}"/> > <action application="hash" > data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/> > <action application="set" > data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} > var callgroup)}"/> > <action application="hash" > data="insert/${domain_name}-last_dial_ext/${called_party_callgroup}/${uuid}" > /> > <action application="hash" > data="insert/${domain_name}-last_dial_ext/global/${uuid}"/> >  > <action application="hash" > data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/> > <action application="bridge" > data="user/${dialed_extension}@${domain_name}"/> > <action application="answer"/> > <action application="sleep" data="1000"/> > <action application="bridge" > data="loopback/app=voicemail:default ${domain_name} ${dialed_extension}"/> > </condition> > </extension> > > > FS only records the caller voicem FS doesnt record the called voice > > How do i record the entire calls(caller and called) > > Thank you > Ha` > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org >

1 0

Polycom with TLS?
by Yehavi Bourvine 17 Jan '13

17 Jan '13

Hello, I have a long standing issue with Polycom and Freeswitch: The TLS session starts ok, I can make 1-2 calls, but then the TLS session dies. It re-connect automatically, but once ebwery fwe minuts it disconnects. Has anyone managed to make this working successfully? Thanks, __Yehavi:

1 0

Re: [Freeswitch-users] Mod_Conference Improvement (Denoise)
by Usama Zaidi 17 Jan '13

17 Jan '13

Hi, So I'm trying to borrow some code from sox now and trying to make an adaptive noise filter which would basically be equivalent of using the noiseprof and noisered filters on a stored file, I'll be calling the noiseprof function whenever there is silence on the line to update my noise profile after regular intervals lets see how that goes. On Thu, Jan 17, 2013 at 12:25 PM, < freeswitch-users-request(a)lists.freeswitch.org> wrote: > Send FreeSWITCH-users mailing list submissions to > freeswitch-users(a)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(a)lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner(a)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: Mod_Conference Improvement (Denoise) (Valer Nur) > 2. Re: Mod_Conference Improvement (Denoise) (William King) > 3. Re: 488 Not Acceptable Here (INCOMPATIBLE_DESTINATION) with > JsSIP and OverSIP + FreeSWITCH (royj) > > > ---------- Forwarded message ---------- > From: Valer Nur <valernur(a)yahoo.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Wed, 16 Jan 2013 22:29:59 -0800 (PST) > Subject: Re: [Freeswitch-users] Mod_Conference Improvement (Denoise) > I think the best way to solve this is by doing AGC first. This should make > sure audio level is similar for all participants. > > > ------------------------------ > *From:* Anthony Minessale <anthony.minessale(a)gmail.com> > *To:* FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > *Sent:* Wednesday, January 16, 2013 6:23 PM > *Subject:* Re: [Freeswitch-users] Mod_Conference Improvement (Denoise) > > There is also the preprocess app which uses a media bug in the core for a > variety of things but is incomplete it's using a bit of code from speex so > we can certainly improve that as well. > > > > > On Tue, Jan 15, 2013 at 8:58 PM, Michael Jerris <mike(a)jerris.com> wrote: > > If you wanted to implement something like this you could do so as a media > bug, it does not need to touch mod_conference in any way. Take a look at > mod_ladspa for an example of a similar module that modifies audio using a > media bug. Give it a try and see if you can get something working. > > Mike > > On Jan 15, 2013, at 8:03 PM, Michael Collins <msc(a)freeswitch.org> wrote: > > While I don't have a problem with the concept of noise filtering I have to > point out that each member of the conference can have his or her own noise > level setting. From fs_cli: > > conference <conf name> energy <member_id|all|last|non_moderator> [<newval>] > > You only need to apply it to the person who is in the noisy location. In > fact, in the example configuration the user can dial 9 to increase the > energy threshold (for when he's in a noisy environment) or press 7 to > decrease the threshold. Pressing 8 will reset the energy to normal, which I > believe is 200. > > -MC > > On Tue, Jan 15, 2013 at 4:24 PM, Usama Zaidi <itsusama(a)gmail.com> wrote: > > Hi, > > I talked about this a while back on the channel, the way how energy-level > is implemented in mod_conference isn't the right way to handle noise in my > opinion because the participants that get bridged would have a different > noise floor depending on where they are located, so if I set the value of > energy-level too high to accommodate one participant, any other participant > in a quite sitting would never get bridged, I proposed we should tackle > this the way Asterisk does using a denoise dialplan function ( > https://wiki.asterisk.org/wiki/display/AST/Asterisk+10+Function_DENOISE) > we can use libav for noise filtering on the muxed audio. I'm willing to > contribute if one of the devs would help me out regarding implementation. > Any comments? > > -Regards > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://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@hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888@conference.freeswitch.org > googletalk:conf+888@conference.freeswitch.org > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > ---------- Forwarded message ---------- > From: William King <william.king(a)quentustech.com> > To: freeswitch-users(a)lists.freeswitch.org > Cc: > Date: Wed, 16 Jan 2013 22:37:19 -0800 > Subject: Re: [Freeswitch-users] Mod_Conference Improvement (Denoise) > That feature already exists, but appears to still be undocumented. Added > just over a year ago. > http://wiki.freeswitch.org/wiki/FS_weekly_2011_03_23 > > William King > Senior Engineer > Quentus Technologies, INC > 1037 NE 65th St Suite 273 > Seattle, WA 98115 > Main: (877) 211-9337 > Office: (206) 388-4772 > Cell: (253) 686-5518 > william.king(a)quentustech.com > > On 01/16/2013 10:29 PM, Valer Nur wrote: > > I think the best way to solve this is by doing AGC first. This should > > make sure audio level is similar for all participants. > > > > > > ------------------------------------------------------------------------ > > *From:* Anthony Minessale <anthony.minessale(a)gmail.com> > > *To:* FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > > *Sent:* Wednesday, January 16, 2013 6:23 PM > > *Subject:* Re: [Freeswitch-users] Mod_Conference Improvement (Denoise) > > > > There is also the preprocess app which uses a media bug in the core for > > a variety of things but is incomplete it's using a bit of code from > > speex so we can certainly improve that as well. > > > > > > > > > > On Tue, Jan 15, 2013 at 8:58 PM, Michael Jerris <mike(a)jerris.com > > <mailto:mike@jerris.com>> wrote: > > > > If you wanted to implement something like this you could do so as a > > media bug, it does not need to touch mod_conference in any way. > > Take a look at mod_ladspa for an example of a similar module that > > modifies audio using a media bug. Give it a try and see if you can > > get something working. > > > > Mike > > > > On Jan 15, 2013, at 8:03 PM, Michael Collins <msc(a)freeswitch.org > > <mailto:msc@freeswitch.org>> wrote: > > > >> While I don't have a problem with the concept of noise filtering I > >> have to point out that each member of the conference can have his > >> or her own noise level setting. From fs_cli: > >> > >> conference <conf name> energy <member_id|all|last|non_moderator> > >> [<newval>] > >> > >> You only need to apply it to the person who is in the noisy > >> location. In fact, in the example configuration the user can dial > >> 9 to increase the energy threshold (for when he's in a noisy > >> environment) or press 7 to decrease the threshold. Pressing 8 will > >> reset the energy to normal, which I believe is 200. > >> > >> -MC > >> > >> On Tue, Jan 15, 2013 at 4:24 PM, Usama Zaidi <itsusama(a)gmail.com > >> <mailto:itsusama@gmail.com>> wrote: > >> > >> Hi, > >> > >> I talked about this a while back on the channel, the way how > >> energy-level is implemented in mod_conference isn't the right > >> way to handle noise in my opinion because the participants > >> that get bridged would have a different noise floor depending > >> on where they are located, so if I set the value of > >> energy-level too high to accommodate one participant, any > >> other participant in a quite sitting would never get bridged, > >> I proposed we should tackle this the way Asterisk does using a > >> denoise dialplan function > >> ( > https://wiki.asterisk.org/wiki/display/AST/Asterisk+10+Function_DENOISE) > >> we can use libav for noise filtering on the muxed audio. I'm > >> willing to contribute if one of the devs would help me out > >> regarding implementation. Any comments? > >> > >> -Regards > >> > >> > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting(a)freeswitch.org <mailto:consulting@freeswitch.org> > > http://www.freeswitchsolutions.com > > > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > > http://www.cudatel.com > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users(a)lists.freeswitch.org > > <mailto:FreeSWITCH-users@lists.freeswitch.org> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org <http://www.freeswitch.org/> > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > AIM: anthm > > MSN:anthony_minessale@hotmail.com > > <mailto:MSN%3Aanthony_minessale@hotmail.com> > > GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com > > <mailto:PAYPAL%3Aanthony.minessale@gmail.com> > > IRC: irc.freenode.net <http://irc.freenode.net/> #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888@conference.freeswitch.org > > <mailto:sip%3A888@conference.freeswitch.org> > > googletalk:conf+888@conference.freeswitch.org > > <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> > > pstn:+19193869900 > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting(a)freeswitch.org <mailto:consulting@freeswitch.org> > > http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/> > > > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > > http://www.cudatel.com <http://www.cudatel.com/> > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org <http://www.freeswitch.org/> > > http://wiki.freeswitch.org <http://wiki.freeswitch.org/> > > http://www.cluecon.com <http://www.cluecon.com/> > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users(a)lists.freeswitch.org > > <mailto:FreeSWITCH-users@lists.freeswitch.org> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org <http://www.freeswitch.org/> > > > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting(a)freeswitch.org > > http://www.freeswitchsolutions.com > > > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > > http://www.cudatel.com > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users(a)lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > ---------- Forwarded message ---------- > From: royj <royj(a)yandex.ru> > To: <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Thu, 17 Jan 2013 11:24:25 +0400 > Subject: Re: [Freeswitch-users] 488 Not Acceptable Here > (INCOMPATIBLE_DESTINATION) with JsSIP and OverSIP + FreeSWITCH > > http://jira.freeswitch.org/browse/FS-4606 > > On Thu, 17 Jan 2013 02:05:34 +0530 > Paan Singh <paan.singh.coder(a)gmail.com> wrote: > > > New SIP capture og with FreeSWITCH log syntax highlighting : > > http://pastebin.freeswitch.com/20457 > > > > > > On Thu, Jan 17, 2013 at 12:50 AM, Paan Singh <paan.singh.coder(a)gmail.com > >wrote: > > > > > Hi, > > > > > > I'm using JsSIP from a webpage to make a SIP call to FS, using OverSIP > as > > > a Websocket->SIP proxy. > > > > > > I'm able to Register successfully, but when I make a call from JsSIP > UA to > > > FreeSWITCH, I get a 180 Ringing, but after that I get a 488 Not > Acceptable > > > here.. Please help me figure out what I'm doing wrong here. > > > > > > > > > My inbound_codec_prefs is : > > > > > > <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729,G7221@32000h > > > ,G7221@16000h,G722,PCMU,PCMA,GSM,isac@16000h,isac,opus"/> > > > > > > Here is FreeSWITCH's log for the incoming call : > > > > > > ============================================= > > > port.c:2730 tport_wakeup_pri() tport_wakeup_pri(0x26036f0): events IN > > > tport.c:2845 tport_recv_event() tport_recv_event(0x26036f0) > > > tport.c:3186 tport_recv_iovec() tport_recv_iovec(0x26036f0) msg > 0x2913be0 > > > from (udp/116.214.15.45:5080) has 3113 bytes, veclen = 1 > > > tport.c:3004 tport_deliver() tport_deliver(0x26036f0): msg 0x2913be0 > (3113 > > > bytes) from udp/121.245.35.150:5080/sip next=(nil) > > > nta.c:2803 agent_recv_request() nta: received INVITE > > > sip:30201009@116.214.15.45 SIP/2.0 (CSeq 2141) > > > nta.c:3161 agent_aliases() nta: canonizing > sip:30201009@116.214.15.45with contact > > > nta.c:3002 agent_recv_request() nta: INVITE (2141) going to a default > leg > > > nta.c:1348 set_timeout() nta: timer set to 200 ms > > > nua_server.c:102 nua_stack_process_request() nua: > > > nua_stack_process_request: entering > > > nua_stack.c:899 nh_create() nua: nh_create: entering > > > nua_common.c:108 nh_create_handle() nua: nh_create_handle: entering > > > nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: > entering > > > soa.c:282 soa_clone() soa_clone(static::0x26122e0, 0x2606e20, > > > 0x7f91840c4c50) called > > > soa.c:405 soa_set_params() soa_set_params(static::0x7f91840b25e0, ...) > > > called > > > nta.c:4313 nta_leg_tcreate() nta_leg_tcreate(0x7f91840356d0) > > > soa.c:1302 soa_init_offer_answer() > > > soa_init_offer_answer(static::0x7f91840b25e0) called > > > soa.c:1171 soa_set_remote_sdp() > soa_set_remote_sdp(static::0x7f91840b25e0, > > > (nil), 0x285676c, 2333) called > > > nua_dialog.c:338 nua_dialog_usage_add() nua(0x7f91840c4c50): adding > > > session usage > > > tport.c:3238 tport_tsend() tport_tsend(0x26036f0) tpn = UDP/ > > > 121.245.35.150:5065 > > > tport.c:4026 tport_resolve() tport_resolve addrinfo = > 121.245.35.150:5065 > > > tport.c:4660 tport_by_addrinfo() tport_by_addrinfo(0x26036f0): not > found > > > by name UDP/121.245.35.150:5065 > > > tport.c:3574 tport_vsend() tport_vsend(0x26036f0): 564 bytes of 564 to > udp/ > > > 121.245.35.150:5065 > > > tport.c:3472 tport_send_msg() tport_vsend returned 564 > > > nta.c:6678 incoming_reply() nta: sent 100 Trying for INVITE (2141) > > > nua_stack.c:271 nua_stack_event() nua(0x7f91840c4c50): event i_invite > 100 > > > Trying > > > nua_session.c:4135 signal_call_state_change() nua(0x7f91840c4c50): call > > > state changed: init -> received, received offer > > > soa.c:1098 soa_get_remote_sdp() > soa_get_remote_sdp(static::0x7f91840b25e0, > > > [0x7f91aa98a5e8], [0x7f91aa98a5e0], [(nil)]) called > > > nua_stack.c:271 nua_stack_event() nua(0x7f91840c4c50): event i_state > 100 > > > Trying > > > nua_stack.c:359 nua_application_event() nua: nua_application_event: > > > entering > > > nua.c:342 nua_handle_bind() nua: nua_handle_bind: entering > > > 2013-01-17 00:25:27.733506 [NOTICE] switch_channel.c:968 New Channel > > > sofia/external/1000@116.214.15.45:5080 > [4aec7fd8-600e-11e2-8706-1365ba01ad52] > > > 2013-01-17 00:25:27.733506 [DEBUG] switch_core_session.c:975 Send > signal > > > sofia/external/1000@116.214.15.45:5080 [BREAK] > > > nua_stack.c:359 nua_application_event() nua: nua_application_event: > > > entering > > > 2013-01-17 00:25:27.733506 [DEBUG] switch_core_session.c:975 Send > signal > > > sofia/external/1000@116.214.15.45:5080 [BREAK] > > > 2013-01-17 00:25:27.733506 [DEBUG] switch_core_state_machine.c:415 > > > (sofia/external/1000@116.214.15.45:5080) Running State Change CS_NEW > > > 2013-01-17 00:25:27.733506 [DEBUG] switch_core_state_machine.c:433 > > > (sofia/external/1000@116.214.15.45:5080) State NEW > > > nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering > > > nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering > > > nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering > > > 2013-01-17 00:25:27.753511 [DEBUG] sofia.c:5576 Channel sofia/external/ > > > 1000@116.214.15.45:5080 entering state [received][100] > > > 2013-01-17 00:25:27.753511 [DEBUG] sofia.c:5587 Remote SDP: > > > v=0 > > > o=- 2957375511 2 IN IP4 127.0.0.1 > > > s=- > > > t=0 0 > > > a=group:BUNDLE audio video > > > m=audio 40111 RTP/SAVPF 103 104 0 8 106 105 13 126 > > > c=IN IP4 121.245.35.150 > > > a=rtcp:40111 IN IP4 121.245.35.150 > > > a=candidate:877273017 1 udp 2113937151 121.245.35.150 39842 typ host > > > generation 0 > > > a=candidate:877273017 2 udp 2113937151 121.245.35.150 39842 typ host > > > generation 0 > > > a=candidate:4173799466 1 udp 1677729535 121.245.35.150 40111 typ srflx > > > generation 0 > > > a=candidate:4173799466 2 udp 1677729535 121.245.35.150 40111 typ srflx > > > generation 0 > > > a=candidate:2060051273 1 tcp 1509957375 121.245.35.150 54291 typ host > > > generation 0 > > > a=candidate:2060051273 2 tcp 1509957375 121.245.35.150 54291 typ host > > > generation 0 > > > a=ice-ufrag:qiKF0ErbxaXldKI0 > > > a=ice-pwd:XlMXg/i8jwnCAgwnuo9KyZBw > > > a=ice-options:google-ice > > > a=mid:audio > > > a=rtcp-mux > > > a=crypto:1 AES_CM_128_HMAC_SHA1_80 > > > inline:bX1U9Kr56lIcZBLQvS2pmRW3nAcTG9H0uahawqj2 > > > a=rtpmap:103 ISAC/16000 > > > a=rtpmap:104 ISAC/32000 > > > a=rtpmap:0 PCMU/8000 > > > a=rtpmap:8 PCMA/8000 > > > a=rtpmap:106 CN/32000 > > > a=rtpmap:105 CN/16000 > > > a=rtpmap:13 CN/8000 > > > a=rtpmap:126 telephone-event/8000 > > > a=ssrc:4278252985 cname:fMHr4uVHPcpHFfGu > > > a=ssrc:4278252985 mslabel:OEZOjX953DJU0k0J3IKmuasdqH13b96rGxhx > > > a=ssrc:4278252985 label:OEZOjX953DJU0k0J3IKmuasdqH13b96rGxhx00 > > > m=video 40111 RTP/SAVPF 100 101 102 > > > c=IN IP4 121.245.35.150 > > > a=rtcp:40111 IN IP4 121.245.35.150 > > > a=candidate:877273017 1 udp 2113937151 121.245.35.150 39842 typ host > > > generation 0 > > > a=candidate:877273017 2 udp 2113937151 121.245.35.150 39842 typ host > > > generation 0 > > > a=candidate:4173799466 1 udp 1677729535 121.245.35.150 40111 typ srflx > > > generation 0 > > > a=candidate:4173799466 2 udp 1677729535 121.245.35.150 40111 typ srflx > > > generation 0 > > > a=candidate:2060051273 1 tcp 1509957375 121.245.35.150 54291 typ host > > > generation 0 > > > a=candidate:2060051273 2 tcp 1509957375 121.245.35.150 54291 typ host > > > generation 0 > > > a=ice-ufrag:qiKF0ErbxaXldKI0 > > > a=ice-pwd:XlMXg/i8jwnCAgwnuo9KyZBw > > > a=ice-options:google-ice > > > a=mid:video > > > a=rtcp-mux > > > a=crypto:1 AES_CM_128_HMAC_SHA1_80 > > > inline:bX1U9Kr56lIcZBLQvS2pmRW3nAcTG9H0uahawqj2 > > > a=rtpmap:100 VP8/90000 > > > a=rtpmap:101 red/90000 > > > a=rtpmap:102 ulpfec/90000 > > > a=ssrc:169275022 cname:fMHr4uVHPcpHFfGu > > > a=ssrc:169275022 mslabel:OEZOjX953DJU0k0J3IKmuasdqH13b96rGxhx > > > a=ssrc:169275022 label:OEZOjX953DJU0k0J3IKmuasdqH13b96rGxhx10 > > > > > > 2013-01-17 00:25:27.753511 [DEBUG] sofia_glue.c:4978 Set Remote Key [1 > > > AES_CM_128_HMAC_SHA1_80 > inline:bX1U9Kr56lIcZBLQvS2pmRW3nAcTG9H0uahawqj2] > > > 2013-01-17 00:25:27.753511 [DEBUG] sofia_glue.c:3155 Set Local Key [1 > > > AES_CM_128_HMAC_SHA1_80 > inline:i8vqmBeU4+/sjJBndCbEQ4G87gmpYeVnqxh8Xuzu] > > > 2013-01-17 00:25:27.753511 [DEBUG] sofia_glue.c:5277 No 2833 in SDP. > > > Disable 2833 dtmf and switch to INFO > > > 2013-01-17 00:25:27.753511 [DEBUG] switch_channel.c:2994 > > > (sofia/external/1000@116.214.15:45:5080) Callstate Change DOWN -> > HANGUP > > > 2013-01-17 00:25:27.753511 [NOTICE] sofia.c:5878 Hangup sofia/external/ > > > 1000@116.214.15.45:5080 [CS_NEW] [INCOMPATIBLE_DESTINATION] > > > 2013-01-17 00:25:27.753511 [DEBUG] switch_channel.c:3017 Send signal > > > sofia/external/1000@116.214.15.45:5080 [KILL] > > > 2013-01-17 00:25:27.753511 [DEBUG] switch_core_session.c:1291 Send > signal > > > sofia/external/1000@116.214.15.45:5080 [BREAK] > > > nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering > > > 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:415 > > > (sofia/external/1000@116.214.15.45:5080) Running State Change > CS_HANGUP > > > 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:667 > > > (sofia/external/1000@116.214.15.45:5080) State HANGUP > > > 2013-01-17 00:25:27.753511 [DEBUG] mod_sofia.c:503 Channel > sofia/external/ > > > 1000@116.214.15.45:5080 hanging up, cause: INCOMPATIBLE_DESTINATION > > > 2013-01-17 00:25:27.753511 [DEBUG] mod_sofia.c:633 Responding to INVITE > > > with: 488 > > > nua.c:879 nua_respond() nua: nua_respond: entering > > > nua_stack.c:529 nua_signal() nua(0x7f91840c4c50): sent signal r_respond > > > 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:48 > > > sofia/external/1000@116.214.15.45:5080 Standard HANGUP, cause: > > > INCOMPATIBLE_DESTINATION > > > 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:667 > > > (sofia/external/1000@116.214.15.45:5080) State HANGUP going to sleep > > > nua_stack.c:573 nua_stack_signal() nua(0x7f91840c4c50): recv signal > > > r_respond 488 Not Acceptable Here > > > 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:446 > > > (sofia/external/1000@116.214.15.45:5080) State Change CS_HANGUP -> > > > CS_REPORTING > > > nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: > entering > > > 2013-01-17 00:25:27.753511 [DEBUG] switch_core_session.c:1291 Send > signal > > > sofia/external/1000@116.214.15.45:5080 [BREAK] > > > soa.c:405 soa_set_params() soa_set_params(static::0x7f91840b25e0, ...) > > > called > > > 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:415 > > > (sofia/external/1000@116.214.15.45:5080) Running State Change > CS_REPORTING > > > nua_session.c:2316 nua_invite_server_respond() nua: > > > nua_invite_server_respond: entering > > > soa.c:1214 soa_clear_remote_sdp() > > > soa_clear_remote_sdp(static::0x7f91840b25e0) called > > > tport.c:3238 tport_tsend() tport_tsend(0x26036f0) tpn = UDP/ > > > 121.245.35.150:5065 > > > 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:749 > > > (sofia/external/1000@116.214.15.45:5080) State REPORTING > > > tport.c:4026 tport_resolve() tport_resolve addrinfo = > 121.245.35.150:5065 > > > tport.c:4660 tport_by_addrinfo() tport_by_addrinfo(0x26036f0): not > found > > > by name UDP/121.245.35.150:5065 > > > tport.c:3574 tport_vsend() tport_vsend(0x26036f0): 784 bytes of 784 to > udp/ > > > 121.245.35.150:5065 > > > tport.c:3472 tport_send_msg() tport_vsend returned 784 > > > nta.c:6678 incoming_reply() nta: sent 488 Not Acceptable Here for > INVITE > > > (2141) > > > nua_dialog.c:397 nua_dialog_usage_remove_at() nua(0x7f91840c4c50): > > > removing session usage > > > nua_session.c:4135 signal_call_state_change() nua(0x7f91840c4c50): call > > > state changed: received -> terminated > > > nua_stack.c:271 nua_stack_event() nua(0x7f91840c4c50): event i_state > 488 > > > Not Acceptable Here > > > nua_stack.c:271 nua_stack_event() nua(0x7f91840c4c50): event > i_terminated > > > 488 Not Acceptable Here > > > soa.c:358 soa_destroy() soa_destroy(static::0x7f91840b25e0) called > > > nta.c:4366 nta_leg_destroy() nta_leg_destroy(0x7f91840356d0) > > > nua_stack.c:359 nua_application_event() nua: nua_application_event: > > > entering > > > nua_stack.c:359 nua_application_event() nua: nua_application_event: > > > entering > > > nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering > > > nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering > > > nua.c:342 nua_handle_bind() nua: nua_handle_bind: entering > > > nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering > > > nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering > > > nua.c:921 nua_handle_destroy() nua: nua_handle_destroy: entering > > > nua_stack.c:569 nua_stack_signal() nua(0x7f91840c4c50): recv signal > > > r_destroy > > > nta.c:4366 nta_leg_destroy() nta_leg_destroy((nil)) > > > nua_stack.c:529 nua_signal() nua(0x7f91840c4c50): sent signal r_destroy > > > 2013-01-17 00:25:27.753511 [DEBUG] mod_cdr_sqlite.c:102 Writing SQL to > DB: > > > INSERT INTO cdr VALUES ("1000","1000","30201009","public","2013-01-17 > > > 00:25:27","","2013-01-17 > > > > 00:25:27",0,0,"INCOMPATIBLE_DESTINATION","4aec7fd8-600e-11e2-8706-1365ba01ad52","","") > > > 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:92 > > > sofia/external/1000@116.214.15.45:5080 Standard REPORTING, cause: > > > INCOMPATIBLE_DESTINATION > > > 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:749 > > > (sofia/external/1000@116.214.15.45:5080) State REPORTING going to > sleep > > > 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:440 > > > (sofia/external/1000@116.214.15.45:5080) State Change CS_REPORTING -> > > > CS_DESTROY > > > 2013-01-17 00:25:27.753511 [DEBUG] switch_core_session.c:1291 Send > signal > > > sofia/external/1000@116.214.15.45:5080 [BREAK] > > > 2013-01-17 00:25:27.753511 [DEBUG] switch_core_session.c:1499 Session > 1794 > > > (sofia/external/1000@116.214.15.45:5080) Locked, Waiting on external > > > entities > > > 2013-01-17 00:25:27.753511 [NOTICE] switch_core_session.c:1517 Session > > > 1794 (sofia/external/1000@116.214.15.45:5080) Ended > > > 2013-01-17 00:25:27.753511 [NOTICE] switch_core_session.c:1521 Close > > > Channel sofia/external/1000@116.214.15.45:5080 [CS_DESTROY] > > > 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:556 > > > (sofia/external/1000@116.214.15.45:5080) Callstate Change HANGUP -> > DOWN > > > 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:559 > > > (sofia/external/1000@116.214.15.45:5080) Running State Change > CS_DESTROY > > > 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:569 > > > (sofia/external/1000@116.214.15.45:5080) State DESTROY > > > 2013-01-17 00:25:27.753511 [DEBUG] mod_sofia.c:396 sofia/external/ > > > 1000@116.214.15.45:5080 SOFIA DESTROY > > > 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:99 > > > sofia/external/1000@116.214.15.45:5080 Standard DESTROY > > > 2013-01-17 00:25:27.753511 [DEBUG] switch_core_state_machine.c:569 > > > (sofia/external/1000@116.214.15.45:5080) State DESTROY going to sleep > > > nta.c:1294 agent_timer() nta: timer set next to 322 ms > > > tport.c:2730 tport_wakeup_pri() tport_wakeup_pri(0x26036f0): events IN > > > tport.c:2845 tport_recv_event() tport_recv_event(0x26036f0) > > > tport.c:3186 tport_recv_iovec() tport_recv_iovec(0x26036f0) msg > > > 0x7f9198220070 from (udp/116.214.15.45:5080) has 339 bytes, veclen = 1 > > > tport.c:3004 tport_deliver() tport_deliver(0x26036f0): msg > 0x7f9198220070 > > > (339 bytes) from udp/121.245.35.150:5080/sip next=(nil) > > > nta.c:2803 agent_recv_request() nta: received ACK > > > sip:30201009@116.214.15.45 SIP/2.0 (CSeq 2141) > > > nta.c:2938 agent_recv_request() nta: ACK (2141) is going to INVITE > (2141) > > > nta.c:1294 agent_timer() nta: timer set next to 4823 ms > > > nta.c:7010 _nta_incoming_timer() nta: timer I fired, terminate 488 > response > > > nta.c:5718 incoming_reclaim_queued() incoming_reclaim_all((nil), (nil), > > > 0x7f91aa98ac20) > > > nta.c:7061 _nta_incoming_timer() nta_incoming_timer: 0/0 resent, 0/0 > tout, > > > 1/1 term, 1/1 free > > > nta.c:1279 agent_timer() nta: timer not set > > > ================================================================ > > > > > > Here is the Log from JsSIP in the browser that receives messages from > > > FreeSWITCH : > > > =========================================================== > > > JsSIP | TRANSPORT | Received WebSocket text message: SIP/2.0 180 > Ringing > > > Via: SIP/2.0/TCP 29.62.109.242;branch=z9hG4bK1565615 Record-Route: > > > <sip:192.168.1.8:5060;transport=udp;lr;ovid=69d4d6a0> Record-Route: > > > <sip:774f80554e@192.168.1.8:10080;transport=ws;lr;ovid=69d4d6a0> > From: < > > > sip:1000@116.214.15.45:5080>;tag=rj2nc38qem To: < > > > sip:30201009@116.214.15.45>;tag=j4X0e04aZjU2K Call-ID: > > > joshempnx2kuh7smrt3x CSeq: 1974 INVITE Contact: > > > <sip:30201009@116.214.15.45:5080;transport=udp> User-Agent: > > > FreeSWITCH-mod_sofia/1.3.13b+git~20130108T190107Z~30e4c71bbe Accept: > > > application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, > INFO, > > > UPDATE, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, > > > replaces Allow-Events: talk, hold, conference, refer Content-Length: 0 > > > P-Asserted-Identity: "30201009" <sip:30201009@116.214.15.45> > > > jssip-0.2.1.js:684< > http://example.com:8080/static/did/tryit/jssip-0.2.1.js> > > > JsSIP | DIALOG | New UAC dialog created: 1 jssip-0.2.1.js:2171< > http://example.com:8080/static/did/tryit/jssip-0.2.1.js> > > > JsSIP | EVENT EMITTER | Emitting event: progress jssip-0.2.1.js:170< > http://example.com:8080/static/did/tryit/jssip-0.2.1.js> > > > JsSIP | TRANSPORT | Received WebSocket text message: SIP/2.0 488 Not > > > Acceptable Here Via: SIP/2.0/TCP 29.62.109.242;branch=z9hG4bK1565615 > > > Max-Forwards: 10 From: <sip:1000@116.214.15.45:5080>;tag=rj2nc38qem > To: < > > > sip:30201009@116.214.15.45>;tag=j4X0e04aZjU2K Call-ID: > > > joshempnx2kuh7smrt3x CSeq: 1974 INVITE User-Agent: > > > FreeSWITCH-mod_sofia/1.3.13b+git~20130108T190107Z~30e4c71bbe Allow: > INVITE, > > > ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, > NOTIFY > > > Supported: timer, precondition, path, replaces Allow-Events: talk, > hold, > > > conference, refer Reason: > Q.850;cause=88;text="INCOMPATIBLE_DESTINATION" > > > Content-Length: 0 P-Asserted-Identity: "30201009" < > > > sip:30201009@116.214.15.45> > > > > ============================================================================= > > > > > > > > > > > > > > > > > > > > > What am I doing wrong? Thanks, > > > Paan > > > > > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- I'd love to change the world, but they wont gimme the source code to it

1 0

Issue - Recording and Playing MP4 video
by Muqtheear S 17 Jan '13

17 Jan '13

Hi all, My SIP client *Ekiga 4.0.0* which has *MP4V-ES* video codec support. I have enabled *MP4V-ES* video codec in Freeswitch. I am trying to record MP4 video as shown below. *<action application="record_mp4" data="/tmp/record_and_play.mp4"/>* After recording, if I look at *mp4info* of *record_and_play.mp4, *it is showing as below #*mp4info /tmp/record_and_play.mp4* mp4info version 1.5.0.1 /tmp/record_and_play.mp4: mp4info: can't open /tmp/record_and_play.mp4 I am neither able to open in *VLC* player. Please tell me where I have gone wrong?. Thanks & Regards, Muqtheear.S

2 1

Issue while playing MP4 video - Suggest me.
by Muqtheear S 17 Jan '13

17 Jan '13

Hi all, I am trying to play *minnale.mp4 *whose *mp4info* is as shown below. *# mp4info minnale.mp4* Track Type Info 1 video MPEG-4 Simple @ L1, 97.833 secs, 136 kbps, 176x144 @ 12.000041 fps 2 audio MPEG-4 AAC LC, 97.568 secs, 64 kbps, 32000 Hz 3 hint Payload MP4V-ES for track 1 4 hint Payload mpeg4-generic for track 2 My SIP client *Ekiga 4.0.0* which has *MP4V-ES* video codec support. I have enabled *MP4V-ES* video codec in Freeswitch, but there is no * mpeg4-generic* audio codec in both Ekiga client and Freeswitch server. I am streaming *minnale.mp4* from freeswitch server to *Ekiga* client. Now on my client side I could see only the video, but there is distortion in audio. How can I overcome above issue? Is there any alternate way to play *.mp4* files?. Please let me know. Thanks & Regards, Muqtheear.S

1 0

Freeswitch and lcr
by stephen＠picardogroup.com 17 Jan '13

17 Jan '13

2 1