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please help! Invalid file format [wav]
by ran zhang 09 Aug '11

09 Aug '11

i'm trying to play a sound when entering the conference, FS gives an error that 'switch_core_file.c122 Invalid file format [wav] for [/sounds/welcomef.wav] I put '/sounds' as the sound prefix and welcomef.wav as the wave file for entering conference. please help!!!

3 2

FS/MRCP/SIP issue....
by Mike Ghicas 08 Aug '11

08 Aug '11

I have a very odd one - what to see if anyone else has had a similar issue. Randomly after starting freeswitch, we get timeouts talking to nuance (different host via MRCP v2) - at first we thought it was an issue with nuance - however after digging a little bit we saw that FS did not even attempt to open a connection to the Nuance host (tcpdump shows noting at all...) and the connection times out. Have tried running with sqllite and mysql via odbc Have tied different combinations of FS/OS/vmware/etc No common pattern. When this fails I stop and restart FS and sometimes it works, sometimes it does not. Sometimes I have to restart 2-10 times, other times it will work right away. Has been working find on my dev host for a while, and I can not replicate the issue. Have installed and updated vmware tools, kernel, etc Had is sync to VMwarehost and not for time NTP is in use. Tried FS with -norc and -nocal options Nothing seems to make this behavior stop - or at least be consistent. Have reserved CPU cycles on Vmware Have run under strace - nothing interesting - just sits there sleeping... Spinning up a physical box for testing - but has anyone seen anything like this? Only seems to affect it when we are opening an MRCP connection Valgrind is next on the list..... Feels like a vmware/os/fs clocking/timing issue Thoughts? Some more details below including log - would love to hear if anyone has experiences anything like this.. As i was writing this I tried the Physical host - same issue - so it does not looks like a timing issue.... RHEL 5.2 -------- We have FS running in a VM on Vmware Vsphere (4.1.0 320137, 4.1.0 208167, 4.1.0 260247) and on a physical machine. OS that has been tried RHEL 5.2, 5.5, 5.6 Working: Vmware 4.1.0 260247 Guest 5.5 Freeswitch - pulled from Git on 2011-07-27 Sofia and uni_mrcp enabled We have tried multiple combinations of everything - only similarity is VMware Our extension that we test with looks like this: <extension name="unimrcp"> <condition field="destination_number" expression="5189"> <action application="answer"/> <action application="set" data="tts_engine=unimrcp:nuance5-mrcp2-prod"/> <action application="set" data="tts_voice=tom"/> <action application="speak" data="The roof, the roof, the roof is on fire"/> <action application="sleep" data="100"/> </condition> </extension> This talks to Nuance Speech Server (via MRC v2) on another host. -- Here is the log from the console with debug enabled: freeswitch(a)devipc02.nj3.ip-soft.net> tport_wakeup_pri(0x6f996f0): events IN tport_recv_event(0x6f996f0) tport_recv_iovec(0x6f996f0) msg 0x2aaac000c420 from (udp/10.140.20.182:5060) has 1171 bytes, veclen = 1 tport_deliver(0x6f996f0): msg 0x2aaac000c420 (1171 bytes) from udp/192.168.8.11:5060/sip next=(nil) nta: received INVITE sip:5189@10.140.20.182:5060 SIP/2.0 (CSeq 101) nta: canonizing sip:5189@10.140.20.182:5060 with contact nta: INVITE (101) going to a default leg nta: timer set to 200 ms nua: nua_stack_process_request: entering nua: nh_create: entering nua: nh_create_handle: entering nua: nua_stack_set_params: entering soa_clone(static::0x6f80bd0, 0x6f67ec0, 0x2aaac000c110) called soa_set_params(static::0x2aaac000d300, ...) called nta_leg_tcreate(0x2aaac000da30) soa_init_offer_answer(static::0x2aaac000d300) called soa_set_remote_sdp(static::0x2aaac000d300, (nil), 0x2aaac000a0af, 212) called nua(0x2aaac000c110): adding session usage tport_tsend(0x6f996f0) tpn = UDP/192.168.8.11:5060 tport_resolve addrinfo = 192.168.8.11:5060 tport_by_addrinfo(0x6f996f0): not found by name UDP/192.168.8.11:5060 tport_vsend(0x6f996f0): 350 bytes of 350 to udp/192.168.8.11:5060 tport_vsend returned 350 nta: sent 100 Trying for INVITE (101) nua(0x2aaac000c110): event i_invite 100 Trying nua(0x2aaac000c110): call state changed: init -> received, received offer soa_get_remote_sdp(static::0x2aaac000d300, [0x400ea8c8], [0x400ea8c0], [(nil)]) called nua(0x2aaac000c110): event i_state 100 Trying nua: nua_application_event: entering 2011-08-08 18:13:36.135540 [NOTICE] switch_channel.c:812 New Channel sofia/internal/5502(a)192.168.8.11 [148a57a7-c2af-45f2-8539-462b21f77a87] nua: nua_handle_bind: entering 2011-08-08 18:13:36.135540 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/5502(a)192.168.8.11) Running State Change CS_NEW 2011-08-08 18:13:36.135540 [DEBUG] switch_core_state_machine.c:343 (sofia/internal/5502(a)192.168.8.11) State NEW nua: nua_handle_magic: entering nua: nua_application_event: entering 2011-08-08 18:13:36.141412 [DEBUG] sofia.c:4761 Channel sofia/internal/5502(a)192.168.8.11 entering state [received][100] 2011-08-08 18:13:36.141412 [DEBUG] sofia.c:4772 Remote SDP: v=0 o=CiscoSystemsCCM-SIP 2000 1 IN IP4 192.168.8.11 s=SIP Call c=IN IP4 10.150.25.10 t=0 0 m=audio 16388 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 2011-08-08 18:13:36.141412 [DEBUG] sofia_glue.c:4651 Audio Codec Compare [PCMU:0:8000:20:64000]/[G7221:115:32000:20:48000] 2011-08-08 18:13:36.141412 [DEBUG] sofia_glue.c:4651 Audio Codec Compare [PCMU:0:8000:20:64000]/[G7221:107:16000:20:32000] 2011-08-08 18:13:36.141412 [DEBUG] sofia_glue.c:4651 Audio Codec Compare [PCMU:0:8000:20:64000]/[G722:9:8000:20:64000] 2011-08-08 18:13:36.141412 [DEBUG] sofia_glue.c:4651 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] 2011-08-08 18:13:36.141412 [DEBUG] sofia_glue.c:2774 Set Codec sofia/internal/5502(a)192.168.8.11 PCMU/8000 20 ms 160 samples 64000 bits 2011-08-08 18:13:36.141412 [DEBUG] sofia_glue.c:4765 Set 2833 dtmf send/recv payload to 101 2011-08-08 18:13:36.141412 [DEBUG] sofia.c:4943 (sofia/internal/5502(a)192.168.8.11) State Change CS_NEW -> CS_INIT 2011-08-08 18:13:36.141412 [DEBUG] switch_core_session.c:1116 Send signal sofia/internal/5502(a)192.168.8.11 [BREAK] nua: nua_handle_magic: entering 2011-08-08 18:13:36.141412 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/5502(a)192.168.8.11) Running State Change CS_INIT 2011-08-08 18:13:36.141412 [DEBUG] switch_core_state_machine.c:361 (sofia/internal/5502(a)192.168.8.11) State INIT 2011-08-08 18:13:36.141412 [DEBUG] mod_sofia.c:84 sofia/internal/5502(a)192.168.8.11 SOFIA INIT 2011-08-08 18:13:36.141412 [DEBUG] mod_sofia.c:124 (sofia/internal/5502(a)192.168.8.11) State Change CS_INIT -> CS_ROUTING 2011-08-08 18:13:36.141412 [DEBUG] switch_core_session.c:1116 Send signal sofia/internal/5502(a)192.168.8.11 [BREAK] 2011-08-08 18:13:36.141412 [DEBUG] switch_core_state_machine.c:361 (sofia/internal/5502(a)192.168.8.11) State INIT going to sleep 2011-08-08 18:13:36.141412 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/5502(a)192.168.8.11) Running State Change CS_ROUTING 2011-08-08 18:13:36.141412 [DEBUG] switch_channel.c:1668 (sofia/internal/5502(a)192.168.8.11) Callstate Change DOWN -> RINGING 2011-08-08 18:13:36.141412 [DEBUG] switch_core_state_machine.c:364 (sofia/internal/5502(a)192.168.8.11) State ROUTING 2011-08-08 18:13:36.141412 [DEBUG] mod_sofia.c:147 sofia/internal/5502(a)192.168.8.11 SOFIA ROUTING 2011-08-08 18:13:36.141412 [DEBUG] switch_core_state_machine.c:77 sofia/internal/5502(a)192.168.8.11 Standard ROUTING 2011-08-08 18:13:36.141412 [INFO] mod_dialplan_xml.c:331 Processing Mike Ghicas <5502>->5189 in context public Dialplan: sofia/internal/5502(a)192.168.8.11 parsing [public->unloop] continue=false Dialplan: sofia/internal/5502(a)192.168.8.11 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/internal/5502(a)192.168.8.11 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: sofia/internal/5502(a)192.168.8.11 parsing [public->outside_call] continue=true Dialplan: sofia/internal/5502(a)192.168.8.11 Absolute Condition [outside_call] Dialplan: sofia/internal/5502(a)192.168.8.11 Action set(outside_call=true) Dialplan: sofia/internal/5502(a)192.168.8.11 Action set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) Dialplan: sofia/internal/5502(a)192.168.8.11 parsing [public->call_debug] continue=true Dialplan: sofia/internal/5502(a)192.168.8.11 Regex (FAIL) [call_debug] ${call_debug}(false) =~ /^true$/ break=never Dialplan: sofia/internal/5502(a)192.168.8.11 parsing [public->public_extensions] continue=false Dialplan: sofia/internal/5502(a)192.168.8.11 Regex (FAIL) [public_extensions] destination_number(5189) =~ /^(10[01][0-9])$/ break=on-false Dialplan: sofia/internal/5502(a)192.168.8.11 parsing [public->public_did] continue=false Dialplan: sofia/internal/5502(a)192.168.8.11 Regex (FAIL) [public_did] destination_number(5189) =~ /^(5551212)$/ break=on-false Dialplan: sofia/internal/5502(a)192.168.8.11 parsing [public->unimrcp] continue=false Dialplan: sofia/internal/5502(a)192.168.8.11 Regex (PASS) [unimrcp] destination_number(5189) =~ /5189/ break=on-false Dialplan: sofia/internal/5502(a)192.168.8.11 Action answer() Dialplan: sofia/internal/5502(a)192.168.8.11 Action set(tts_engine=unimrcp:nuance5-mrcp2-prod) Dialplan: sofia/internal/5502(a)192.168.8.11 Action set(tts_voice=tom) Dialplan: sofia/internal/5502(a)192.168.8.11 Action speak(The roof, the roof, the roof is on fire. we don't need no water let that mother fucker burn.) Dialplan: sofia/internal/5502(a)192.168.8.11 Action sleep(100) 2011-08-08 18:13:36.141412 [DEBUG] switch_core_state_machine.c:119 (sofia/internal/5502(a)192.168.8.11) State Change CS_ROUTING -> CS_EXECUTE 2011-08-08 18:13:36.141412 [DEBUG] switch_core_session.c:1116 Send signal sofia/internal/5502(a)192.168.8.11 [BREAK] 2011-08-08 18:13:36.141412 [DEBUG] switch_core_state_machine.c:364 (sofia/internal/5502(a)192.168.8.11) State ROUTING going to sleep 2011-08-08 18:13:36.141412 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/5502(a)192.168.8.11) Running State Change CS_EXECUTE 2011-08-08 18:13:36.141412 [DEBUG] switch_core_state_machine.c:371 (sofia/internal/5502(a)192.168.8.11) State EXECUTE 2011-08-08 18:13:36.141412 [DEBUG] mod_sofia.c:240 sofia/internal/5502(a)192.168.8.11 SOFIA EXECUTE 2011-08-08 18:13:36.141412 [DEBUG] switch_core_state_machine.c:157 sofia/internal/5502(a)192.168.8.11 Standard EXECUTE EXECUTE sofia/internal/5502(a)192.168.8.11 set(outside_call=true) 2011-08-08 18:13:36.141412 [DEBUG] mod_dptools.c:1060 sofia/internal/5502(a)192.168.8.11 SET [outside_call]=[true] EXECUTE sofia/internal/5502(a)192.168.8.11 set(RFC2822_DATE=Mon, 08 Aug 2011 18:13:36 +0000) 2011-08-08 18:13:36.143981 [DEBUG] mod_dptools.c:1060 sofia/internal/5502(a)192.168.8.11 SET [RFC2822_DATE]=[Mon, 08 Aug 2011 18:13:36 +0000] EXECUTE sofia/internal/5502(a)192.168.8.11 answer() 2011-08-08 18:13:36.152965 [DEBUG] sofia_glue.c:3015 AUDIO RTP [sofia/internal/5502(a)192.168.8.11] 10.140.20.182 port 16858 -> 10.150.25.10 port 16388 codec: 0 ms: 20 2011-08-08 18:13:36.152965 [DEBUG] switch_rtp.c:1623 Starting timer [soft] 160 bytes per 20ms 2011-08-08 18:13:36.154960 [DEBUG] sofia_glue.c:3277 Set 2833 dtmf send payload to 101 2011-08-08 18:13:36.154960 [DEBUG] sofia_glue.c:3282 Set 2833 dtmf receive payload to 101 2011-08-08 18:13:36.154960 [DEBUG] mod_sofia.c:681 Local SDP sofia/internal/5502(a)192.168.8.11: v=0 o=FreeSWITCH 1312810358 1312810359 IN IP4 10.140.20.182 s=FreeSWITCH c=IN IP4 10.140.20.182 t=0 0 m=audio 16858 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 nua: nua_respond: entering nua(0x2aaac000c110): sent signal r_respond 2011-08-08 18:13:36.154960 [DEBUG] switch_core_session.c:709 Send signal sofia/internal/5502(a)192.168.8.11 [BREAK] 2011-08-08 18:13:36.154960 [DEBUG] switch_channel.c:2830 (sofia/internal/5502(a)192.168.8.11) Callstate Change RINGING -> ACTIVE 2011-08-08 18:13:36.154960 [NOTICE] mod_dptools.c:930 Channel [sofia/internal/5502(a)192.168.8.11] has been answered EXECUTE sofia/internal/5502(a)192.168.8.11 set(tts_engine=unimrcp:nuance5-mrcp2-prod) 2011-08-08 18:13:36.156012 [DEBUG] mod_dptools.c:1060 sofia/internal/5502(a)192.168.8.11 SET [tts_engine]=[unimrcp:nuance5-mrcp2-prod] EXECUTE sofia/internal/5502(a)192.168.8.11 set(tts_voice=tom) 2011-08-08 18:13:36.156012 [DEBUG] mod_dptools.c:1060 sofia/internal/5502(a)192.168.8.11 SET [tts_voice]=[tom] EXECUTE sofia/internal/5502(a)192.168.8.11 speak(The roof, the roof, the roof is on fire. we don't need no water let that mother fucker burn.) 2011-08-08 18:13:36.156947 [INFO] mod_unimrcp.c:1567 speech_handle: name = unimrcp, rate = 8000, speed = 0, samples = 160, voice = , engine = unimrcp, param = nuance5-mrcp2-prod 2011-08-08 18:13:36.156947 [INFO] mod_unimrcp.c:1570 voice = tom, rate = 8000 2011-08-08 18:13:36.156947 [DEBUG] mod_unimrcp.c:652 (TTS-1) audio queue created 2011-08-08 18:13:36.156947 [NOTICE] mrcp_client.c:549 Create MRCP Handle 0x703f1f0 [nuance5-mrcp2-prod] 2011-08-08 18:13:36.156947 [INFO] mrcp_client_session.c:142 Create Channel 0x703f1f0 <new> 2011-08-08 18:13:36.156947 [DEBUG] mrcp_client.c:1038 Signal Application Task Message nua(0x2aaac000c110): recv signal r_respond 200 OK nua: nua_stack_set_params: entering soa_set_params(static::0x2aaac000d300, ...) called soa_set_user_sdp(static::0x2aaac000d300, (nil), 0x2aaac0010a32, -1) called 2011-08-08 18:13:36.156947 [DEBUG] mrcp_client.c:1006 Receive Application Task Message [0] 2011-08-08 18:13:36.156947 [INFO] mrcp_client_session.c:398 Receive App Request 0x703f1f0 <new> [2] 2011-08-08 18:13:36.156947 [INFO] mrcp_client.c:901 Add MRCP Handle 0x703f1f0 2011-08-08 18:13:36.156947 [DEBUG] mrcp_client_session.c:1203 Dispatch Application Request 0x703f1f0 <new> [2] 2011-08-08 18:13:36.156947 [NOTICE] mrcp_client_session.c:718 Add Control Channel 0x703f1f0 <new@speechsynth> soa_set_capability_sdp(static::0x2aaac000d300, (nil), 0x2aaac0010a32, -1) called 2011-08-08 18:13:36.156947 [DEBUG] mrcp_client_session.c:762 Add RTP Termination 0x703f1f0 <new> 2011-08-08 18:13:36.156947 [DEBUG] apt_consumer_task.c:90 Wait for Task Messages [MRCP Client] nua: nua_invite_server_respond: entering soa_generate_answer(static::0x2aaac000d300) called soa_static_offer_answer_action(0x2aaac000d300, soa_generate_answer): called soa_static(0x2aaac000d300, soa_generate_answer): generating local description soa_static(0x2aaac000d300, soa_generate_answer): upgrade with remote description soa_sdp_mode_set(0x400eacd0, 0x2aaac000e990, ""): called soa_static(0x2aaac000d300, soa_generate_answer): storing local description soa_activate(static::0x2aaac000d300, (nil)) called soa_get_local_sdp(static::0x2aaac000d300, [(nil)], [0x400eae38], [0x400eae44]) called tport_tsend(0x6f996f0) tpn = UDP/192.168.8.11:5060 tport_resolve addrinfo = 192.168.8.11:5060 tport_by_addrinfo(0x6f996f0): not found by name UDP/192.168.8.11:5060 tport_vsend(0x6f996f0): 1175 bytes of 1175 to udp/192.168.8.11:5060 tport_vsend returned 1175 nta: sent 200 OK for INVITE (101) nua(0x2aaac000c110): call state changed: received -> completed, sent answer soa_get_local_sdp(static::0x2aaac000d300, [0x400eaef8], [0x400eaef0], [(nil)]) called soa_get_params(static::0x2aaac000d300, ...) called nua(0x2aaac000c110): event i_state 200 OK 2011-08-08 18:13:36.156947 [DEBUG] mrcp_client_connection.c:603 Process Control Message 2011-08-08 18:13:36.156947 [DEBUG] mrcp_client.c:1104 Signal Connection Task Message 2011-08-08 18:13:36.156947 [DEBUG] mrcp_client.c:974 Receive Connection Task Message [0] nua: nua_application_event: entering 2011-08-08 18:13:36.156947 [DEBUG] mrcp_client_session.c:311 On Control Channel Add 0x703f1f0 <new> 2011-08-08 18:13:36.156947 [DEBUG] apt_consumer_task.c:90 Wait for Task Messages [MRCP Client] 2011-08-08 18:13:36.156947 [DEBUG] sofia.c:4761 Channel sofia/internal/5502(a)192.168.8.11 entering state [completed][200] nua: nua_handle_magic: entering 2011-08-08 18:13:36.159000 [DEBUG] mpf_engine.c:302 Process MPF Message 2011-08-08 18:13:36.159000 [DEBUG] mpf_context.c:172 Add Context 2011-08-08 18:13:36.159000 [DEBUG] mpf_context.c:176 Add Termination 2011-08-08 18:13:36.159000 [DEBUG] mpf_context.c:176 Add Termination 2011-08-08 18:13:36.159000 [DEBUG] mrcp_client.c:999 Receive Media Task Message 2011-08-08 18:13:36.159000 [DEBUG] mrcp_client_session.c:1039 On Termination Add 0x703f1f0 <new> 2011-08-08 18:13:36.159000 [DEBUG] mrcp_client_session.c:1039 On Termination Add 0x703f1f0 <new> 2011-08-08 18:13:36.159000 [INFO] mrcp_client_session.c:420 Send Offer 0x703f1f0 <new> [c:1 a:1 v:0] 2011-08-08 18:13:36.159000 [INFO] mrcp_sofiasip_client_agent.c:316 Local SDP 0x703f1f0 <new> v=0 o=FreeSWITCH 0 0 IN IP4 10.140.20.182 s=- c=IN IP4 10.140.20.182 t=0 0 m=application 9 TCP/MRCPv2 1 a=setup:active a=connection:new a=resource:speechsynth a=cmid:1 m=audio 4002 RTP/AVP 0 8 96 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:96 L16/8000 a=recvonly a=mid:1 2011-08-08 18:13:36.159000 [DEBUG] apt_consumer_task.c:90 Wait for Task Messages [MRCP Client] tport_wakeup_pri(0x6f996f0): events IN tport_recv_event(0x6f996f0) tport_recv_iovec(0x6f996f0) msg 0x2aaac000fb00 from (udp/10.140.20.182:5060) has 431 bytes, veclen = 1 tport_deliver(0x6f996f0): msg 0x2aaac000fb00 (431 bytes) from udp/192.168.8.11:5060/sip next=(nil) nta: received ACK sip:5189@10.140.20.182:5060;transport=udp SIP/2.0 (CSeq 101) nta: ACK (101) is going to INVITE (101) nua: process_ack_or_cancel: entering soa_clear_remote_sdp(static::0x2aaac000d300) called nua(0x2aaac000c110): event i_ack 200 OK nua(0x2aaac000c110): call state changed: completed -> ready nua(0x2aaac000c110): event i_state 200 OK nua(0x2aaac000c110): event i_active 200 Call active nua(): refresh session after 1768 seconds (in [1768..1768]) nua: nua_application_event: entering nua: nua_handle_magic: entering nua: nua_application_event: entering 2011-08-08 18:13:36.161958 [DEBUG] sofia.c:4761 Channel sofia/internal/5502(a)192.168.8.11 entering state [ready][200] nua: nua_handle_magic: entering nua: nua_application_event: entering nua: nua_handle_magic: entering nta: timer set next to 4827 ms 2011-08-08 18:13:41.158959 [WARNING] mod_unimrcp.c:1010 (TTS-1) MRCP session has not opened after 5000 ms nta: timer I fired, terminate 200 response incoming_reclaim_all((nil), (nil), 0x400ead40) nta_incoming_timer: 0/0 resent, 0/0 tout, 1/1 term, 1/1 free nta: timer not set

1 1

openssl deprecated on OS X Lion and MacPorts with latest Git head?
by Gavin Henry 08 Aug '11

08 Aug '11

Hi all, Anyone seeing this? My port tree is up to date and on latest MacPorts: In file included from src/switch_core.c:39: ./src/include/switch_ssl.h: In function ‘switch_ssl_init_ssl_locks’: ./src/include/switch_ssl.h:63: warning: ‘CRYPTO_num_locks’ is deprecated (declared at /usr/include/openssl/crypto.h:415) ./src/include/switch_ssl.h:65: warning: ‘CRYPTO_malloc’ is deprecated (declared at /usr/include/openssl/crypto.h:478) ./src/include/switch_ssl.h:65: warning: ‘CRYPTO_num_locks’ is deprecated (declared at /usr/include/openssl/crypto.h:415) ./src/include/switch_ssl.h:75: warning: ‘CRYPTO_set_id_callback’ is deprecated (declared at /usr/include/openssl/crypto.h:425) ./src/include/switch_ssl.h:76: warning: ‘CRYPTO_set_locking_callback’ is deprecated (declared at /usr/include/openssl/crypto.h:418) ./src/include/switch_ssl.h: In function ‘switch_ssl_destroy_ssl_locks’: ./src/include/switch_ssl.h:91: warning: ‘CRYPTO_set_locking_callback’ is deprecated (declared at /usr/include/openssl/crypto.h:418) ./src/include/switch_ssl.h:92: warning: ‘CRYPTO_num_locks’ is deprecated (declared at /usr/include/openssl/crypto.h:415) ./src/include/switch_ssl.h:98: warning: ‘CRYPTO_free’ is deprecated (declared at /usr/include/openssl/crypto.h:480) make[2]: *** [libfreeswitch_la-switch_core.lo] Error 1 make[1]: *** [all] Error 2 make: *** [current] Error 2 -- http://www.suretecsystems.com/services/openldap/ http://www.surevoip.co.uk

3 2

Best way to route Large block of DIDs
by Luis F Urrea 08 Aug '11

08 Aug '11

Hi all, I have a relatively large block of DIDs coming from a T1 gateway to a FS box. These DIDs are mapped to user extensions in most cases and in others they go to an IVR. My question is what would be the best way to write an extension in public.xml dialplan if the block of DIDs that go directly to users is not exactly divided so that I can write a regexp to cover the block. Does writing a single rule for each user can bring performance implications? What if I cannot arrange the block of DIDs that belong to users in a consecutive group? What are the best practices for such case? TIA!!! Luis

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conference count
by Sam 08 Aug '11

08 Aug '11

*I have a dial plan ,* <extension name="dynamic_conferences_guest"> <condition field="destination_number" expression="^1234"> <action application="answer"/> <action application="info"/> <action application="set" data="conf_count=${conference ${conference_name} list count}"/> <action application="log" data="INFO ${conf_count}"/> <action application="say" data="en number pronounced ${conf_count}"/> <action application="conference" data="1000146@meeting"/> </condition> </extension> *and the output of it as below:-* 2011-08-06 14:32:53.205895 [DEBUG] sofia.c:5084 Channel sofia/internal/ 61323123999(a)192.153.53.158 entering state [ready][200] EXECUTE sofia/internal/61323123999(a)192.153.53.158 set(conf_count=Conference 1000146 (1 member rate: 16000) 14;sofia/internal/883510001282001(a)sip.ntfone.in ;97f5f386-c00a-11e0-9517-738827116f37;883510001282001;883510001282001;hear|speak|floor;0;0;0;300 ) 2011-08-06 14:32:53.205895 [DEBUG] mod_dptools.c:1063 sofia/internal/ 61323123999(a)192.153.53.158 SET [conf_count]=[Conference 1000146 (1 member rate: 16000) 14;sofia/internal/883510001282001(a)sip.nfone.in ;97f5f386-c00a-11e0-9517-738827116f37;883510001282001;883510001282001;hear|speak|floor;0;0;0;300 ] EXECUTE sofia/internal/61323123999(a)192.153.53.158 log(INFO Conference 1000146 (1 member rate: 16000) 14;sofia/internal/883510001282001(a)sip.ntfone.in ;97f5f386-c00a-11e0-9517-738827116f37;883510001282001;883510001282001;hear|speak|floor;0;0;0;300 ) 2011-08-06 14:32:53.205895 [INFO] mod_dptools.c:1202 Conference 1000146 (1 member rate: 16000) 14;sofia/internal/883510001282001(a)sip.nfone.in ;97f5f386-c00a-11e0-9517-738827116f37;883510001282001;883510001282001;hear|speak|floor;0;0;0;300 EXECUTE sofia/internal/61323123999(a)192.153.53.158 say(en number pronounced Conference 1000146 (1 member rate: 16000) 14;sofia/internal/883510001282001(a)sip.nfone.in ;97f5f386-c00a-11e0-9517-738827116f37;883510001282001;883510001282001;hear|speak|floor;0;0;0;300 ) 2011-08-06 14:32:53.215905 [ERR] mod_say_en.c:130 Parse Error! EXECUTE sofia/internal/61323123999(a)192.153.53.158 conference(1000146@meeting ) *What should be my dialplan to get the exact conf_count ?* Thanks in advance Sam

2 2

media proxy
by Simon Leck 07 Aug '11

07 Aug '11

Hi guys, Has anybody been successful in trying to get mediaproxy to work with Freeswitch? If so, it would be great if any of you could give us leads on how this is been done. Mediaproxy URL http://mediaproxy.ag-projects.com/ Thanks a lot. Simon

2 1

Hangup time is blank
by Sam 07 Aug '11

07 Aug '11

How come the "hangup_time" channel variable is 0 even after a call has been bridged successfully and hung up? I am trying to find out what time the bridged call hung up by looking at the channel variable "hangup_time".

1 0

FS for SIP registrar server
by Dome Charoenyost 07 Aug '11

07 Aug '11

Dear All, I use FS for my wholesale solution. it's work fine over 2000 concurrent in peak hr. so in experience when i use FS for SIP registrar server with out ODBC (but use tmp for db path) i can handle 12,000 active in 1 server (25% CPU). All user send register packet every 3 minutes. Now i need system for handle 500,000 subscriber I have 2 choice 1. use Kamailio for SIP regitrar and use FS for routing and billing 2. still use multiple FS with ODBC So anyone can help? BG Dome C.

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Re: [Freeswitch-users] Radius/Authentication/Authorization
by Abid Saleem 06 Aug '11

06 Aug '11

Dear Tihomir, Any update and help on the below please. Abid From: abid_freeswitch(a)live.com To: tculjaga(a)gmail.com CC: neal(a)wanlink.com Subject: Re: [Freeswitch-users] Radius/Authentication/Authorization Date: Wed, 31 Mar 2010 19:35:32 +0500 Dear Tihomir, Thank you very much for the configuration example but in which files to place these configurations. Please bear with me because I am new to FreeSwitch and if you could provide complete steps. Also when I compile FS and load the module mod_rad_auth in conf/autoload_configs/modules.conf.xml, I get an error while starting FS as follows. 2010-03-31 19:32:29.399466 [CRIT] switch_loadable_module.c:882 Error Loading module /usr/local/freeswitch/mod/mod_rad_auth.so**/usr/local/freeswitch/mod/mod_rad_auth.so: undefined symbol: rc_conf_str** Please advise. Thanks. Regards-----------Abid Saleem --Forwarded Message Attachment-- From: tculjaga(a)gmail.com CC: neal(a)wanlink.com To: freeswitch-users(a)lists.freeswitch.org Date: Tue, 30 Mar 2010 21:42:03 +0200 Subject: Re: [Freeswitch-users] Radius/Authentication/Authorization hello, here is an example <configuration name="rad_auth.conf" description="radius authentification module"> <settings>   </settings> <client> <param name="authserver" value="10.1.1.10:1812:gateway"/> <param name="dictionary" value="/usr/local/etc/radiusclient/dictionary.all"/> <param name="seqfile" value="/var/run/radius.seq"/> <param name="mapfile" value="/usr/local/etc/radiusclient/port-id-map"/> <param name="default_realm" value=""/> <param name="radius_timeout" value="3"/> <param name="radius_retries" value="2"/> <param name="radius_deadtime" value="0"/> <param name="bindaddr" value="*"/> </client> <vsas>   <param name="h323-conf-id" id="24" value="CALLID" pec="9" expr="1" direction="in"/> <param name="h323-prompt-id" id="104" value="SERVICENUM" pec="9" expr="1" direction="in"/> <param name="Cisco-AVPair" id="1" value="TRANSACTIONID" pec="9" expr="1" direction="in"/> <param name="Calling-Station-Id" id="31" value="CALLINGNUMBER" pec="0" expr="1" direction="in"/> <param name="NAS-Port-Type" id="61" value="0" pec="0" expr="0" direction="in"/> <param name="NAS-Port-Id" id="87" value="ISDN 3/0:D:14" pec="0" expr="0" direction="in"/> <param name="Login-User" id="1" value="1" pec="0" expr="0" direction="in"/>  <param name="BILING_MODEL" id="109" value="billing_model" pec="9" expr="0" direction="out"/> <param name="CREDIT_AMOUNT" id="101" value="credit_amount" pec="9" expr="0" direction="out"/> <param name="CURRENCY" id="110" value="currency" pec="9" expr="0" direction="out"/> <param name="PREFFERED_LANG" id="107" value="preffered_lang" pec="9" expr="0" direction="out"/> <param name="CREDIT_TIME" id="102" value="credit_time" pec="9" expr="0" direction="out"/> <param name="H323-IVR-IN:DIRATION" id="1" value="h323_ivr_duration" pec="9" expr="0" direction="out"/> <param name="RADIUS_RETURN_CODE" id="103" value="return_code" pec="9" expr="0" direction="out"/>  </vsas> </configuration> in the dialplan you need to trigger auth as: <action application="auth_function" data="in ${DIALED_NUMBER}, in ${USERNAME}, in ${PASSWD}, out AUTH_RESULT"/> there are two behaviours: 1. authorize the call according to username&pass and dialed number - if authorized, the radius server returns credit time towards the dialed number 2. authorize the call according to username&pass - if authorized, the radius server returns the current account balance <extension name="unitest_rad-ANI-auth"> <condition field="destination_number" expression="^601$"> <action application="log" data="INFO PRIJE RAD_AUTH "/> <action inline="true" application="set" data="CALLID=h323-conf-id=${uuid}"/> <action inline="true" application="set" data="SERVICENUM=h323-prompt-id=${destination_number}"/> <action inline="true" application="set" data="TRANSACTIONID=h323-ivr-out=transactionID:1234"/>  <action inline="true" application="set" data="CALLINGNUMBER=38516060333"/> <action inline="true" application="set" data="USERNAME=38516060333"/>  <action inline="true" application="set" data="PASSWD=003282"/>  <action application="sleep" data="2000"/> <action application="auth_function" data="in ${DIALED_NUMBER}, in ${USERNAME}, in ${PASSWD}, out AUTH_RESULT"/> <action application="log" data="INFO AUTH_RESULT=${AUTH_RESULT}"/> <action application="log" data="INFO billing_model=${billing_model}"/> <action application="log" data="INFO credit_amount=${credit_amount}"/> <action application="log" data="INFO currency=${currency}"/> <action application="log" data="INFO preffered_lang=${preffered_lang}"/> <action application="log" data="INFO credit_time=${credit_time}"/> <action application="log" data="INFO h323_ivr_duration=${h323_ivr_duration}"/> <action application="log" data="INFO return_code=${return_code}"/>  </condition> </extension> <extension name="unitest_rad-ANI-balance"> <condition field="destination_number" expression="^602$"> <action application="log" data="INFO PRIJE RAD_AUTH "/> <action inline="true" application="set" data="CALLID=h323-conf-id=${uuid}"/> <action inline="true" application="set" data="SERVICENUM=h323-prompt-id=${destination_number}"/> <action inline="true" application="set" data="TRANSACTIONID=h323-ivr-out=transactionID:1234"/>  <action inline="true" application="set" data="CALLINGNUMBER=38516060333"/> <action inline="true" application="set" data="USERNAME=38516060333"/>  <action inline="true" application="set" data="PASSWD=003282"/> <action inline="true" application="set" data="DIALED_NUMBER=16094191500"/> <action application="sleep" data="2000"/> <action application="auth_function" data="in ${DIALED_NUMBER}, in ${USERNAME}, in ${PASSWD}, out AUTH_RESULT"/> <action application="log" data="INFO AUTH_RESULT=${AUTH_RESULT}"/> <action application="log" data="INFO billing_model=${billing_model}"/> <action application="log" data="INFO credit_amount=${credit_amount}"/> <action application="log" data="INFO currency=${currency}"/> <action application="log" data="INFO preffered_lang=${preffered_lang}"/> <action application="log" data="INFO credit_time=${credit_time}"/> <action application="log" data="INFO h323_ivr_duration=${h323_ivr_duration}"/> <action application="log" data="INFO return_code=${return_code}"/>  </condition> </extension> will update the wiki by the end of the week. you have enough information for now. Tihomir. On Tue, Mar 30, 2010 at 1:52 PM, Abid Saleem <abid_freeswitch(a)live.com> wrote: Hi Neal and other Contributors to FS, I recieved an answer on the list that mod_rad_auth is ready. I upgraded FS to download and install it by "make current" and it is successfully built and installed. Could you please outline the detailed steps to do all this configuration. If it is still not ready, is there any other method that somebody has already implemented like perl scripts etc. Any help in this regard is badly required. THanks for your cooperation. Regards----------Abid SaleemSr. Product Manager Hotmail: Powerful Free email with security by Microsoft. Get it now. _________________________________________________________________ Hotmail: Powerful Free email with security by Microsoft. https://signup.live.com/signup.aspx?id=60969

4 8

outgoing SIP profiles setup
by lloyd thomas 06 Aug '11

06 Aug '11

Hi All, I am trying to create a new sip profile as my provider does not use sip registrations, but uses IP address and port number to identify wether a connection should be allowed. I am currently using the 'internal' profile to make calls '<action application="bridge" data="sofia/internal/$1(a)80.40.150.150"/>', but this is causing a couple of strange anomalies. I have created a new xml file in sip_profiles and use 'sofia profile phisys-2circles start' to add it. But I get the following errors and the profile does not show in 'sofia status' errors----------------------------------------------- 2011-08-06 17:38:28.620292 [NOTICE] sofia_reg.c:2775 Added gateway ' example.com' to profile 'phisys-2circles' 2011-08-06 17:38:28.620292 [WARNING] sofia.c:1898 Ignoring duplicate gateway 'example.com' 2011-08-06 17:38:28.624397 [WARNING] sofia.c:1898 Ignoring duplicate gateway 'example.com' 2011-08-06 17:38:28.625645 [NOTICE] sofia.c:3999 Started Profile phisys-2circles [sofia_reg_phisys-2circles] 2011-08-06 17:38:28.643631 [DEBUG] sofia.c:1471 Creating agent for phisys-2circles 2011-08-06 17:38:28.800994 [ERR] sofia.c:1533 Error Creating SIP UA for profile: phisys-2circles 2011-08-06 17:38:28.800994 [NOTICE] sofia_glue.c:5198 deleted gateway example.com I have added phisys_2circles.xml to the sip_profiles directory and the contents are here http://pastebin.freeswitch.org/16998 Any advice welcome as I am at a lost what to do next. Thanks Lloyd T

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