May 2007 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] mfcr2
by Mike Murdock 11 Apr '08

11 Apr '08

What is mfcr2 used to interface with? Michael Murdock VP Operations & Development Switchmaxx Product Line 125 N. Market. Suite 1520 Wichita KS. 67202 (316) 932-2693 -----Original Message----- From: Brian West [mailto:brian.west@mac.com] Sent: Tuesday, January 30, 2007 11:46 AM Eastern Standard Time To: Yeruult[ICN]; freeswitch-users(a)lists.freeswitch.org Subject: Re: [Freeswitch-users] mfcr2 Do you happen to know of an open source libmfcr2 that is license compatible with FreeSWITCH? /b On Jan 29, 2007, at 7:10 AM, Yeruult[ICN] wrote: > Hello all, > > > > Does Freeswitch supports mfcr2? I know mfcr2 is old protocol but > that doesn't mean ppl don't use it. As I know there is still many > guys from everywhere. As I understood asterisk v1.4 becoming > complicated for Steve Underwood's unicall library. Is there any > other open sources which supports mfcr2? > > > > -- > > Best regards, > > Yeruult[ICN] mailto:yeruult@icn-mn.net > > _______________________________________________ > 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

10 16

Re: [Freeswitch-users] freeswitch: src/switch_core.c:1360: switch_core_session_strdup: Assertion `session != ((void *)0)' failed.
by Anthony Minessale 23 Jul '07

23 Jul '07

Try executing "make sure" from the build directory this will erase all the object code and all the modules and rebuild them all and reinstall it (takes about a minute or 2) If you still have a problem please let us know. Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com JABBER:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: Pekka K. Kurki <pekka.kurki(a)intellectics.com> To: freeswitch-users(a)lists.freeswitch.org Sent: Monday, December 25, 2006 6:17:03 PM Subject: Re: [Freeswitch-users] freeswitch: src/switch_core.c:1360: switch_core_session_strdup: Assertion `session != ((void *)0)' failed. here the trace: [root@salo freeswitch]# bin/freeswitch 2006-12-26 00:57:52 [CONSOLE] switch_core.c:4072 switch_core_init_and_modload() Bringing up environment. 2006-12-26 00:57:52 [CONSOLE] switch_core.c:4073 switch_core_init_and_modload() Loading Modules. 2006-12-26 00:57:52 [CONSOLE] switch_loadable_module.c:347 switch_loadable_module_load_file() Successfully Loaded [mod_console] 2006-12-26 00:57:52 [ERR] mod_portaudio.c:615 load_config() open of portaudio.conf failed Number of devices = 1 ---------------------------------------------- #0 DefaultInput DefaultOutput Name = /dev/dsp Max Inputs = 16, Max Outputs = 16 Sample Rates = 96000.00, 48000.00, 44100.00, 32000.00, 24000.00, 22050.00, 16000.00, 11025.00, 8000.00, Native Sample Formats = paUInt8, paInt16, ---------------------------------------------- 2006-12-26 00:57:52 [CONSOLE] switch_loadable_module.c:347 switch_loadable_module_load_file() Successfully Loaded [mod_portaudio] 2006-12-26 00:57:52 [NOTICE] switch_loadable_module.c:102 switch_loadable_module_process() Adding Endpoint 'portaudio' 2006-12-26 00:57:52 [NOTICE] switch_loadable_module.c:180 switch_loadable_module_process() Adding API Function 'pacall' 2006-12-26 00:57:52 [NOTICE] switch_loadable_module.c:180 switch_loadable_module_process() Adding API Function 'pahup' 2006-12-26 00:57:52 [NOTICE] switch_loadable_module.c:180 switch_loadable_module_process() Adding API Function 'painfo' 2006-12-26 00:57:52 [NOTICE] switch_loadable_module.c:180 switch_loadable_module_process() Adding API Function 'paoffhook' 2006-12-26 00:57:52 [NOTICE] switch_loadable_module.c:180 switch_loadable_module_process() Adding API Function 'padtmf' 2006-12-26 00:57:52 [NOTICE] mod_sofia.c:5000 config_sofia() Started Profile salo.intellectics.com [sip:mod_sofia@85.126.75.149:5061] 2006-12-26 00:57:52 [CONSOLE] switch_loadable_module.c:347 switch_loadable_module_load_file() Successfully Loaded [mod_sofia] 2006-12-26 00:57:52 [NOTICE] switch_loadable_module.c:102 switch_loadable_module_process() Adding Endpoint 'sofia' 2006-12-26 00:57:52 [NOTICE] switch_loadable_module.c:255 switch_loadable_module_process() Adding Chat interface 'sip' 2006-12-26 00:57:52 [CONSOLE] switch_loadable_module.c:347 switch_loadable_module_load_file() Successfully Loaded [mod_bridgecall] 2006-12-26 00:57:52 [NOTICE] switch_loadable_module.c:163 switch_loadable_module_process() Adding Application 'bridge' su_port_create() called su_port(0x96b6760): epoll_create() => 10: OK su_port_create() returns 0x96b6760 2006-12-26 00:57:52 [CONSOLE] switch_loadable_module.c:347 switch_loadable_module_load_file() Successfully Loaded [mod_commands] 2006-12-26 00:57:52 [NOTICE] switch_loadable_module.c:180 switch_loadable_module_process() Adding API Function 'originate' 2006-12-26 00:57:52 [NOTICE] switch_loadable_module.c:180 switch_loadable_module_process() Adding API Function 'killchan' 2006-12-26 00:57:52 [NOTICE] switch_loadable_module.c:180 switch_loadable_module_process() Adding API Function 'reloadxml' 2006-12-26 00:57:52 [NOTICE] switch_loadable_module.c:180 switch_loadable_module_process() Adding API Function 'load' 2006-12-26 00:57:52 [NOTICE] switch_loadable_module.c:180 switch_loadable_module_process() Adding API Function 'transfer' 2006-12-26 00:57:52 [NOTICE] switch_loadable_module.c:180 switch_loadable_module_process() Adding API Function 'pause' 2006-12-26 00:57:52 [NOTICE] switch_loadable_module.c:180 switch_loadable_module_process() Adding API Function 'show' 2006-12-26 00:57:52 [NOTICE] switch_loadable_module.c:180 switch_loadable_module_process() Adding API Function 'status' 2006-12-26 00:57:52 [NOTICE] switch_loadable_module.c:180 switch_loadable_module_process() Adding API Function 'uuid_bridge' 2006-12-26 00:57:52 [NOTICE] switch_loadable_module.c:180 switch_loadable_module_process() Adding API Function 'broadcast' 2006-12-26 00:57:52 [NOTICE] switch_loadable_module.c:180 switch_loadable_module_process() Adding API Function 'hold' 2006-12-26 00:57:52 [NOTICE] switch_loadable_module.c:180 switch_loadable_module_process() Adding API Function 'media' 2006-12-26 00:57:52 [NOTICE] switch_loadable_module.c:180 switch_loadable_module_process() Adding API Function 'fsctl' 2006-12-26 00:57:52 [NOTICE] switch_loadable_module.c:180 switch_loadable_module_process() Adding API Function 'help' 2006-12-26 00:57:52 [NOTICE] switch_loadable_module.c:180 switch_loadable_module_process() Adding API Function 'version' 2006-12-26 00:57:52 [CONSOLE] switch_loadable_module.c:347 switch_loadable_module_load_file() Successfully Loaded [mod_dptools] 2006-12-26 00:57:52 [NOTICE] switch_loadable_module.c:163 switch_loadable_module_process() Adding Application 'privacy' 2006-12-26 00:57:52 [NOTICE] switch_loadable_module.c:163 switch_loadable_module_process() Adding Application 'transfer' 2006-12-26 00:57:52 [NOTICE] switch_loadable_module.c:163 switch_loadable_module_process() Adding Application 'sleep' 2006-12-26 00:57:52 [NOTICE] switch_loadable_module.c:163 switch_loadable_module_process() Adding Application 'strftime' 2006-12-26 00:57:52 [NOTICE] switch_loadable_module.c:163 switch_loadable_module_process() Adding Application 'phrase' 2006-12-26 00:57:52 [NOTICE] switch_loadable_module.c:163 switch_loadable_module_process() Adding Application 'eval' 2006-12-26 00:57:52 [NOTICE] switch_loadable_module.c:163 switch_loadable_module_process() Adding Application 'answer' 2006-12-26 00:57:52 [NOTICE] switch_loadable_module.c:163 switch_loadable_module_process() Adding Application 'log' 2006-12-26 00:57:52 [NOTICE] switch_loadable_module.c:163 switch_loadable_module_process() Adding Application 'info' 2006-12-26 00:57:52 [NOTICE] switch_loadable_module.c:163 switch_loadable_module_process() Adding Application 'set' 2006-12-26 00:57:53 [NOTICE] switch_loadable_module.c:163 switch_loadable_module_process() Adding Application 'ringback' 2006-12-26 00:57:53 [NOTICE] switch_loadable_module.c:163 switch_loadable_module_process() Adding Application 'detect_speech' 2006-12-26 00:57:53 [NOTICE] switch_loadable_module.c:163 switch_loadable_module_process() Adding Application 'ivr' 2006-12-26 00:57:53 [NOTICE] switch_loadable_module.c:180 switch_loadable_module_process() Adding API Function 'presence' 2006-12-26 00:57:53 [NOTICE] switch_loadable_module.c:180 switch_loadable_module_process() Adding API Function 'strftime' 2006-12-26 00:57:53 [NOTICE] switch_loadable_module.c:180 switch_loadable_module_process() Adding API Function 'chat' 2006-12-26 00:57:53 [CONSOLE] switch_loadable_module.c:347 switch_loadable_module_load_file() Successfully Loaded [mod_echo] 2006-12-26 00:57:53 [NOTICE] switch_loadable_module.c:163 switch_loadable_module_process() Adding Application 'echo' 2006-12-26 00:57:53 [CONSOLE] switch_loadable_module.c:347 switch_loadable_module_load_file() Successfully Loaded [mod_playback] 2006-12-26 00:57:53 [NOTICE] switch_loadable_module.c:163 switch_loadable_module_process() Adding Application 'playback' 2006-12-26 00:57:53 [NOTICE] switch_loadable_module.c:163 switch_loadable_module_process() Adding Application 'stop_record_session' 2006-12-26 00:57:53 [NOTICE] switch_loadable_module.c:163 switch_loadable_module_process() Adding Application 'record_session' 2006-12-26 00:57:53 [NOTICE] switch_loadable_module.c:163 switch_loadable_module_process() Adding Application 'record' 2006-12-26 00:57:53 [NOTICE] switch_loadable_module.c:163 switch_loadable_module_process() Adding Application 'speak' 2006-12-26 00:57:53 [CONSOLE] switch_loadable_module.c:347 switch_loadable_module_load_file() Successfully Loaded [mod_dialplan_xml] 2006-12-26 00:57:53 [NOTICE] switch_loadable_module.c:135 switch_loadable_module_process() Adding Dialplan 'XML' 2006-12-26 00:57:53 [CONSOLE] switch_loadable_module.c:347 switch_loadable_module_load_file() Successfully Loaded [mod_g711] 2006-12-26 00:57:53 [NOTICE] switch_loadable_module.c:113 switch_loadable_module_process() Adding Codec 'PCMU' (g711 ulaw) 8000hz 20ms 2006-12-26 00:57:53 [NOTICE] switch_loadable_module.c:113 switch_loadable_module_process() Adding Codec 'PCMU' (g711 ulaw) 8000hz 60ms 2006-12-26 00:57:53 [NOTICE] switch_loadable_module.c:113 switch_loadable_module_process() Adding Codec 'PCMA' (g711 alaw) 8000hz 20ms 2006-12-26 00:57:53 [CONSOLE] switch_loadable_module.c:347 switch_loadable_module_load_file() Successfully Loaded [mod_gsm] 2006-12-26 00:57:53 [NOTICE] switch_loadable_module.c:113 switch_loadable_module_process() Adding Codec 'GSM' (gsm) 8000hz 20ms nua: nua_create: entering su_port_create() called 2006-12-26 00:57:53 [CONSOLE] switch_loadable_module.c:347 switch_loadable_module_load_file() Successfully Loaded [mod_l16] 2006-12-26 00:57:53 [NOTICE] switch_loadable_module.c:113 switch_loadable_module_process() Adding Codec 'L16' (raw signed linear (16 bit)) 8000hz 120ms 2006-12-26 00:57:53 [NOTICE] switch_loadable_module.c:113 switch_loadable_module_process() Adding Codec 'L16' (raw signed linear (16 bit)) 8000hz 60ms 2006-12-26 00:57:53 [NOTICE] switch_loadable_module.c:113 switch_loadable_module_process() Adding Codec 'L16' (raw signed linear (16 bit)) 8000hz 30ms 2006-12-26 00:57:53 [NOTICE] switch_loadable_module.c:113 switch_loadable_module_process() Adding Codec 'L16' (raw signed linear (16 bit)) 8000hz 20ms 2006-12-26 00:57:53 [NOTICE] switch_loadable_module.c:113 switch_loadable_module_process() Adding Codec 'L16' (raw signed linear (16 bit)) 16000hz 20ms 2006-12-26 00:57:53 [NOTICE] switch_loadable_module.c:113 switch_loadable_module_process() Adding Codec 'L16' (raw signed linear (16 bit)) 22050hz 20ms 2006-12-26 00:57:53 [NOTICE] switch_loadable_module.c:113 switch_loadable_module_process() Adding Codec 'L16' (raw signed linear (16 bit)) 32000hz 20ms 2006-12-26 00:57:53 [INFO] mod_sndfile.c:286 setup_formats() LibSndFile Version : libsndfile-1.0.17 Supported Formats ================================================================================ AIFF (Apple/SGI) (extension "aiff") AU (Sun/NeXT) (extension "au") AVR (Audio Visual Research) (extension "avr") CAF (Apple Core Audio File) (extension "caf") HTK (HMM Tool Kit) (extension "htk") IFF (Amiga IFF/SVX8/SV16) (extension "iff") MAT4 (GNU Octave 2.0 / Matlab 4.2) (extension "mat") MAT5 (GNU Octave 2.1 / Matlab 5.0) (extension "mat") PAF (Ensoniq PARIS) (extension "paf") PVF (Portable Voice Format) (extension "pvf") RAW (header-less) (extension "raw") SD2 (Sound Designer II) (extension "sd2") SDS (Midi Sample Dump Standard) (extension "sds") SF (Berkeley/IRCAM/CARL) (extension "sf") VOC (Creative Labs) (extension "voc") W64 (SoundFoundry WAVE 64) (extension "w64") WAV (Microsoft) (extension "wav") WAV (NIST Sphere) (extension "wav") WAVEX (Microsoft) (extension "wav") XI (FastTracker 2) (extension "xi") ================================================================================ 2006-12-26 00:57:53 [CONSOLE] switch_loadable_module.c:347 switch_loadable_module_load_file() Successfully Loaded [mod_sndfile] 2006-12-26 00:57:53 [NOTICE] switch_loadable_module.c:198 switch_loadable_module_process() Adding File Format 'aiff' 2006-12-26 00:57:53 [NOTICE] switch_loadable_module.c:198 switch_loadable_module_process() Adding File Format 'au' 2006-12-26 00:57:53 [NOTICE] switch_loadable_module.c:198 switch_loadable_module_process() Adding File Format 'avr' 2006-12-26 00:57:53 [NOTICE] switch_loadable_module.c:198 switch_loadable_module_process() Adding File Format 'caf' 2006-12-26 00:57:53 [NOTICE] switch_loadable_module.c:198 switch_loadable_module_process() Adding File Format 'htk' 2006-12-26 00:57:53 [NOTICE] switch_loadable_module.c:198 switch_loadable_module_process() Adding File Format 'iff' 2006-12-26 00:57:53 [NOTICE] switch_loadable_module.c:198 switch_loadable_module_process() Adding File Format 'mat' 2006-12-26 00:57:53 [NOTICE] switch_loadable_module.c:198 switch_loadable_module_process() Adding File Format 'paf' 2006-12-26 00:57:53 [NOTICE] switch_loadable_module.c:198 switch_loadable_module_process() Adding File Format 'pvf' 2006-12-26 00:57:53 [NOTICE] switch_loadable_module.c:198 switch_loadable_module_process() Adding File Format 'raw' 2006-12-26 00:57:53 [NOTICE] switch_loadable_module.c:198 switch_loadable_module_process() Adding File Format 'sd2' 2006-12-26 00:57:53 [NOTICE] switch_loadable_module.c:198 switch_loadable_module_process() Adding File Format 'sds' 2006-12-26 00:57:53 [NOTICE] switch_loadable_module.c:198 switch_loadable_module_process() Adding File Format 'sf' 2006-12-26 00:57:53 [NOTICE] switch_loadable_module.c:198 switch_loadable_module_process() Adding File Format 'voc' 2006-12-26 00:57:53 [NOTICE] switch_loadable_module.c:198 switch_loadable_module_process() Adding File Format 'w64' 2006-12-26 00:57:53 [NOTICE] switch_loadable_module.c:198 switch_loadable_module_process() Adding File Format 'wav' 2006-12-26 00:57:53 [NOTICE] switch_loadable_module.c:198 switch_loadable_module_process() Adding File Format 'xi' 2006-12-26 00:57:53 [NOTICE] switch_loadable_module.c:198 switch_loadable_module_process() Adding File Format 'r8' 2006-12-26 00:57:53 [NOTICE] switch_loadable_module.c:198 switch_loadable_module_process() Adding File Format 'r16' 2006-12-26 00:57:53 [NOTICE] switch_loadable_module.c:198 switch_loadable_module_process() Adding File Format 'r24' 2006-12-26 00:57:53 [NOTICE] switch_loadable_module.c:198 switch_loadable_module_process() Adding File Format 'r32' 2006-12-26 00:57:53 [NOTICE] switch_loadable_module.c:198 switch_loadable_module_process() Adding File Format 'gsm' 2006-12-26 00:57:53 [CONSOLE] switch_loadable_module.c:347 switch_loadable_module_load_file() Successfully Loaded [mod_native_file] 2006-12-26 00:57:53 [NOTICE] switch_loadable_module.c:198 switch_loadable_module_process() Adding File Format 'PCMA' 2006-12-26 00:57:53 [NOTICE] switch_loadable_module.c:198 switch_loadable_module_process() Adding File Format 'L16' 2006-12-26 00:57:53 [NOTICE] switch_loadable_module.c:198 switch_loadable_module_process() Adding File Format 'PCMU' 2006-12-26 00:57:53 [NOTICE] switch_loadable_module.c:198 switch_loadable_module_process() Adding File Format 'GSM' 2006-12-26 00:57:53 [CONSOLE] switch_loadable_module.c:347 switch_loadable_module_load_file() Successfully Loaded [mod_softtimer] 2006-12-26 00:57:53 [NOTICE] switch_loadable_module.c:149 switch_loadable_module_process() Adding Timer 'soft' 2006-12-26 00:57:53 [CONSOLE] switch_loadable_module.c:564 switch_loadable_module_init() open of post_load_modules.conf failed 2006-12-26 00:57:53 [CONSOLE] switch_core.c:4085 switch_core_init_and_modload() FreeSWITCH Version 1.0.3827 Started. Crash Protection [DISABLED] Max Sessions[1000] su_port(0x96b8738): epoll_create() => 13: OK su_port_create() returns 0x96b8738 freeswitch(a)salo.intellectics.com> nua: nua_stack_init: entering nua: nua_stack_set_params: entering soa_create("default", 0x96a44b0, 0x96b8cf8) called soa_set_params(static::0x96b9a00, ...) called nta_agent_create: initialized hash tables nta_agent_create: initialized transports nta_agent_create: initialized random identifiers nta_agent_create: initialized timer nta_agent_create: initialized resolver tport_create(): 0x96ba978 nta: master transport created tport_bind_server(0x96ba978) to */85.126.75.149:5061/sip tport_bind_server(0x96ba978): calling tport_listen for udp tport_alloc_primary(0x96ba978): new primary tport 0x96bae60 tport_listen(0x96bae60): listening at udp/85.126.75.149:5061/sip tport_bind_server(0x96ba978): calling tport_listen for tcp tport_alloc_primary(0x96ba978): new primary tport 0x96bfdb8 tport_listen(0x96bfdb8): listening at tcp/85.126.75.149:5061/sip nta: bound to (85.126.75.149:5061;transport=*) nta: agent_init_via: SIP/2.0/udp 85.126.75.149:5061 (sip) nta: agent_init_via: SIP/2.0/tcp 85.126.75.149:5061 (sip) nta: Via fields initialized nta: Contact header created nua_register: Adding contact URL '85.126.75.149' to list. nua: nua_set_params: entering nua: nh_create_handle: entering 2006-12-26 00:57:53 [NOTICE] mod_sofia.c:4555 check_oreg() registering asterlink nua: nua_handle_bind: entering nua: nua_register: entering nua((nil)): signal r_set_params nua: nua_stack_set_params: entering soa_set_params(static::0x96b9a00, ...) called nua: nua_stack_event: entering nua(0x96b8cf8): r_set_params 200 OK nua: nua_event: entering nua(0x96c0538): signal r_register nua: nua_stack_set_params: entering soa_clone(static::0x96b9a00, 0x96a44b0, 0x96c0538) called soa_set_params(static::0x96c2348, ...) called soa_set_params(static::0x96c2348, ...) called soa_set_params(static::0x96c2348, ...) called nua(0x96c0538): adding register usage nta_leg_create(0x96c1b50) nta: selecting scheme sip tport(0x96bae60): EXPENSIVE unresolved */85.126.75.147:5060 tport_tsend(0x96bae60) tpn = */85.126.75.147:5060 tport_resolve addrinfo = 85.126.75.147:5060 tport_by_addrinfo(0x96bae60): not found by name */85.126.75.147:5060 tport_vsend(0x96bae60): 534 bytes of 534 to */85.126.75.147:5060 tport_vsend returned 534 send 534 bytes to udp/[85.126.75.147]:5060 at 23:57:53.979911: ------------------------------------------------------------------------ REGISTER sip:85.126.75.147:5060 SIP/2.0 Via: SIP/2.0/UDP 85.126.75.149:5061;rport;branch=z9hG4bKB9a0jSr3eXj2F Max-Forwards: 70 From: <sip:1002@85.126.75.147>;tag=B9gBa1jtQ5HZQ To: <sip:1002@85.126.75.147> Call-ID: 93b81520-0f16-122a-ea9d-00079556e80f CSeq: 77427200 REGISTER Contact: <sip:1002@85.126.75.147> Expires: 600 User-Agent: FreeSWITCH(mod_sofia) Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: 100rel, precondition, path Content-Length: 0 ------------------------------------------------------------------------ nta: sent REGISTER (77427200) to */85.126.75.147:5060 tport_pend(0x96bae60): pending 0x96c0b88 for udp/85.126.75.149:5061 (already 0) tport_wakeup_pri(0x96bae60): events IN tport_recv_event(0x96bae60) tport_recv_iovec(0x96bae60) msg 0x96c0338 from (udp/85.126.75.149:5061) has 65536 bytes, veclen = 1 recv 434 bytes from udp/[85.126.75.147]:5060 at 23:57:54.090576: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 85.126.75.149:5061;rport;branch=z9hG4bKB9a0jSr3eXj2F;received=85.126.75.149 From: <sip:1002@85.126.75.147>;tag=B9gBa1jtQ5HZQ To: <sip:1002@85.126.75.147> Call-ID: 93b81520-0f16-122a-ea9d-00079556e80f CSeq: 77427200 REGISTER User-Agent: intellectics Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Contact: <sip:1002@85.126.75.147> Content-Length: 0 ------------------------------------------------------------------------ tport_deliver(0x96bae60): msg 0x96c0338 (434 bytes) from udp/85.126.75.147:5061/sip next=(nil) nta: received 100 Trying for REGISTER (77427200) nta: 100 Trying going to a transaction nta_outgoing: RTT is 137.032 ms tport_release(0x96bae60): 0x96c0b88 by 0x96c3508 with 0x96c0338 (preliminary) tport_wakeup_pri(0x96bae60): events IN tport_recv_event(0x96bae60) tport_recv_iovec(0x96bae60) msg 0x96c0338 from (udp/85.126.75.149:5061) has 65536 bytes, veclen = 1 recv 508 bytes from udp/[85.126.75.147]:5060 at 23:57:54.208309: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 85.126.75.149:5061;rport;branch=z9hG4bKB9a0jSr3eXj2F;received=85.126.75.149 From: <sip:1002@85.126.75.147>;tag=B9gBa1jtQ5HZQ To: <sip:1002@85.126.75.147>;tag=as0cd8e88e Call-ID: 93b81520-0f16-122a-ea9d-00079556e80f CSeq: 77427200 REGISTER User-Agent: intellectics Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Expires: 600 Contact: <sip:1002@85.126.75.147>;expires=600 Date: Mon, 25 Dec 2006 23:58:24 GMT Content-Length: 0 ------------------------------------------------------------------------ tport_deliver(0x96bae60): msg 0x96c0338 (508 bytes) from udp/85.126.75.147:5061/sip next=(nil) nta: received 200 OK for REGISTER (77427200) nta: 200 OK going to a transaction tport_release(0x96bae60): 0x96c0b88 by 0x96c3508 with 0x96c0338 nua(): refresh register after 247 seconds (in [150..450]) nua: nua_stack_event: entering nua(0x96c0538): r_register 200 OK nua: nua_event: entering 2006-12-26 00:57:54 [NOTICE] mod_sofia.c:4545 check_oreg() registered asterlink nta: timer K fired, terminate REGISTER (77427200) outgoing_reclaim_all((nil), (nil), 0xb736c1fc) nta_outgoing_timer: 0/0 resent, 0/0 tout, 1/1 term, 1/1 free tport_wakeup_pri(0x96bae60): events IN tport_recv_event(0x96bae60) tport_recv_iovec(0x96bae60) msg 0x96c4818 from (udp/85.126.75.149:5061) has 65536 bytes, veclen = 1 recv 1013 bytes from udp/[85.126.75.147]:5060 at 23:58:38.848482: ------------------------------------------------------------------------ INVITE sip:1002@85.126.75.147 SIP/2.0 Via: SIP/2.0/UDP 85.126.75.147:5060;branch=z9hG4bK4fad7691 From: "Pekka Kurki" <sip:+432256815811@85.126.75.147>;tag=as0c642fe4 To: <sip:1002@85.126.75.147> Contact: <sip:+432256815811@85.126.75.147> Call-ID: 49c54c511b82059c2a91829103ccabbe(a)85.126.75.147 CSeq: 102 INVITE User-Agent: intellectics Max-Forwards: 70 Date: Mon, 25 Dec 2006 23:59:09 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 492 v=0 o=root 11091 11091 IN IP4 85.126.75.147 s=session c=IN IP4 85.126.75.147 t=0 0 m=audio 16414 RTP/AVP 0 4 3 8 111 5 10 7 18 110 97 101 a=rtpmap:0 PCMU/8000 a=rtpmap:4 G723/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:7 LPC/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=noa=rtpmap:110 speex/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - ------------------------------------------------------------------------ tport_deliver(0x96bae60): msg 0x96c4818 (1013 bytes) from udp/85.126.75.147:5061/sip next=(nil) nta: received INVITE sip:1002@85.126.75.147 SIP/2.0 (CSeq 102) nta: INVITE (102) going to a default leg nua: nua_stack_process_request: entering nua: nh_create: entering nua: nh_create_handle: entering nua: nua_stack_set_params: entering soa_clone(static::0x96b9a00, 0x96a44b0, 0x96c3508) called soa_set_params(static::0x96c05e8, ...) called soa_set_params(static::0x96c05e8, ...) called nta_leg_create(0x96c0d50) soa_init_offer_answer(static::0x96c05e8) called soa_set_remote_sdp(static::0x96c05e8, (nil), 0x96c4c11, 492) called nua(0x96c3508): adding session usage tport_tsend(0x96bae60) tpn = UDP/85.126.75.147:5060 tport_resolve addrinfo = 85.126.75.147:5060 tport_by_addrinfo(0x96bae60): not found by name UDP/85.126.75.147:5060 tport_vsend(0x96bae60): 311 bytes of 311 to UDP/85.126.75.147:5060 tport_vsend returned 311 send 311 bytes to udp/[85.126.75.147]:5060 at 23:58:38.851717: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 85.126.75.147:5060;branch=z9hG4bK4fad7691 From: "Pekka Kurki" <sip:+432256815811@85.126.75.147>;tag=as0c642fe4 To: <sip:1002@85.126.75.147> Call-ID: 49c54c511b82059c2a91829103ccabbe(a)85.126.75.147 CSeq: 102 INVITE User-Agent: FreeSWITCH(mod_sofia) Content-Length: 0 ------------------------------------------------------------------------ nta: sent 100 Trying for INVITE (102) nua: nua_stack_event: entering nua(0x96c3508): i_invite 100 Trying nua(0x96c3508): call state changed: init -> received, received offer soa_get_remote_sdp(static::0x96c05e8, [0xb736bb58], [0xb736bb54], [(nil)]) called nua: nua_stack_event: entering nua(0x96c3508): i_state 100 Trying nua: nua_event: entering nua: nua_handle_bind: entering nua: nua_event: entering 2006-12-26 00:58:38 [NOTICE] switch_channel.c:347 switch_channel_set_name() New Chan sofia/salo.intellectics.com/+432256815811@85.126.75.147 [68b21b7f-b866-4f40-9b04-94d6cbc592e3] 2006-12-26 00:58:38 [DEBUG] mod_sofia.c:2521 sip_i_state() Channel sofia/salo.intellectics.com/+432256815811@85.126.75.147 entering state [received] 2006-12-26 00:58:38 [DEBUG] mod_sofia.c:2526 sip_i_state() Remote SDP: v=0 o=root 11091 11091 IN IP4 85.126.75.147 s=session c=IN IP4 85.126.75.147 t=0 0 m=audio 16414 RTP/AVP 0 4 3 8 111 5 10 7 18 110 97 101 a=rtpmap:0 PCMU/8000 a=rtpmap:4 G723/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:7 LPC/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=noa=rtpmap:110 speex/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - 2006-12-26 00:58:38 [DEBUG] mod_sofia.c:2172 negotiate_sdp() Codec Compare [PCMU:0]/[PCMU:0] 2006-12-26 00:58:38 [INFO] mod_sofia.c:1368 tech_set_codec() Set Codec sofia/salo.intellectics.com/+432256815811@85.126.75.147 PCMU/8000 20 ms 2006-12-26 00:58:38 [DEBUG] mod_sofia.c:2644 sip_i_state() sofia/salo.intellectics.com/+432256815811@85.126.75.147 State Change CS_NEW -> CS_INIT 2006-12-26 00:58:38 [DEBUG] switch_core.c:2945 switch_core_session_run() (sofia/salo.intellectics.com/+432256815811@85.126.75.147) State INIT 2006-12-26 00:58:38 [DEBUG] mod_sofia.c:1119 sofia_on_init() SOFIA INIT 2006-12-26 00:58:38 [DEBUG] mod_sofia.c:1130 sofia_on_init() sofia/salo.intellectics.com/+432256815811@85.126.75.147 State Change CS_INIT -> CS_RING 2006-12-26 00:58:38 [DEBUG] switch_core.c:2981 switch_core_session_run() (sofia/salo.intellectics.com/+432256815811@85.126.75.147) State RING 2006-12-26 00:58:38 [DEBUG] mod_sofia.c:1146 sofia_on_ring() SOFIA RING 2006-12-26 00:58:38 [DEBUG] switch_core.c:2622 switch_core_standard_on_ring() Standard RING sofia/salo.intellectics.com/+432256815811@85.126.75.147 2006-12-26 00:58:38 [INFO] mod_dialplan_xml.c:218 dialplan_hunt() Processing Pekka Kurki->1002! 2006-12-26 00:58:38 [DEBUG] mod_dialplan_xml.c:120 parse_exten() test conditions destination_number(1002) =~ /^1002$/ 2006-12-26 00:58:39 [DEBUG] mod_dialplan_xml.c:337 dialplan_hunt() sofia/salo.intellectics.com/+432256815811@85.126.75.147 State Change CS_RING -> CS_EXECUTE 2006-12-26 00:58:39 [DEBUG] switch_core.c:3017 switch_core_session_run() (sofia/salo.intellectics.com/+432256815811@85.126.75.147) State EXECUTE 2006-12-26 00:58:39 [DEBUG] mod_sofia.c:1164 sofia_on_execute() SOFIA EXECUTE 2006-12-26 00:58:39 [DEBUG] switch_core.c:2673 switch_core_standard_on_execute() Standard EXECUTE 2006-12-26 00:58:39 [NOTICE] switch_core.c:2683 switch_core_standard_on_execute() Execute bridge(sofia/salo.intellectics.com/111@sip.intellectics.com) 2006-12-26 00:58:39 [NOTICE] switch_channel.c:347 switch_channel_set_name() New Chan sofia/salo.intellectics.com/111@sip.intellectics.com [e2e18ecc-ac1e-421f-b2dd-c2b3ce1a021a] 2006-12-26 00:58:39 [DEBUG] mod_sofia.c:2116 sofia_outgoing_channel() sofia/salo.intellectics.com/111@sip.intellectics.com State Change CS_NEW -> CS_INIT 2006-12-26 00:58:39 [DEBUG] switch_core.c:2945 switch_core_session_run() (sofia/salo.intellectics.com/111@sip.intellectics.com) State INIT 2006-12-26 00:58:39 [DEBUG] mod_sofia.c:1119 sofia_on_init() SOFIA INIT freeswitch: src/switch_core.c:1360: switch_core_session_strdup: Assertion `session != ((void *)0)' failed. Aborted [root@salo freeswitch]# and my conf file: [root@salo freeswitch]# cat /usr/local/freeswitch/conf/freeswitch.xml xml version="1.0"?> <document type="freeswitch/xml"> <section name="configuration" description="Various Configuration"> <configuration name="switch.conf" description="Modules"> <settings>  <param name="max-sessions" value="1000"/> </settings>  <variables> <variable name="uk-ring" value="%(400,200,400,450);%(400,2200,400,450)"/> <variable name="us-ring" value="%(2000, 4000, 440.0, 480.0)"/> <variable name="bong-ring" value="v=4000;>=0;+=2;#(60,0);v=2000;%(940,0,350,440)"/> </variables> </configuration> <configuration name="modules.conf" description="Modules"> <modules>  <load module="mod_console"/>               <load module="mod_portaudio"/> <load module="mod_sofia"/>    <load module="mod_bridgecall"/> <load module="mod_commands"/>  <load module="mod_dptools"/> <load module="mod_echo"/>  <load module="mod_playback"/>   <load module="mod_dialplan_xml"/>  <load module="mod_g711"/> <load module="mod_gsm"/>  <load module="mod_l16"/>   <load module="mod_sndfile"/> <load module="mod_native_file"/>  <load module="mod_softtimer"/>       </modules> </configuration> <configuration name="console.conf" description="Console Logger">   <mappings>  <param name="all" value="DEBUG"/> </mappings> </configuration> <configuration name="sofia.conf" description="Sofia Endpoint"> <profiles> <profile name="salo.intellectics.com"> <registrations> <registration name="asterlink"> <param name="register-scheme" value="Digest"/> <param name="register-realm" value=""/> <param name="register-username" value="1002"/> <param name="register-password" value="xxxxx"/> <param name="register-from" value="sip:1002@85.126.75.147"/> <param name="register-to" value="sip:1002@85.126.75.147"/> <param name="register-proxy" value="sip:85.126.75.147:5060"/> <param name="register-frequency" value="600"/> </registration> </registrations> <settings> <param name="debug" value="1"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5061"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU@20i"/> <param name="codec-ms" value="20"/> <param name="use-rtp-timer" value="false"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-ip" value="85.126.75.149"/> <param name="sip-ip" value="85.126.75.149"/> <param name="ext-sip-ip" value="85.126.75.149"/>   <param name="accept-blind-reg" value="true"/>     <param name="ext-rtp-ip" value="stun:stun.voipdiscount.com"/>       </settings> </profile> </profiles> </configuration> <configuration name="syslog.conf" description="Syslog Logger">          <settings> <param name="ident" value="freeswitch"/> <param name="facility" value="user"/> <param name="format" value="${time} - ${message}"/> <param name="level" value="debug,info,warning-alert"/> </settings> </configuration> <configuration name="xml_rpc.conf" description="XML RPC"> <settings>  <param name="http-port" value="8080"/>  <param name="auth-realm" value="freeswitch"/> <param name="auth-user" value="pekka"/> <param name="auth-pass" value="kalevi"/>    <param name="gateway-url" value="http://salo.intellectics.com:8080/gateway.cgi" bindings="configuration|directory|dialplan"/> </settings> </configuration> </section> <section name="dialplan" description="Regex/XML Dialplan">     <context name="default"> <extension name="43720503918"> <condition field="source" expression="mod_sofia"/> <condition field="destination_number" expression="43720503918"> <action application="bridge" data="sofia/salo.intellectics.com/1001@${server-domain-name}"/> </condition> </extension> <extension name="111"> <condition field="source" expression="mod_sofia"/> <condition field="destination_number" expression="111"> <action application="bridge" data="sofia/salo.intellectics.com/111@sip.intellectics.com"/> </condition> </extension> <extension name="1001"> <condition field="destination_number" expression="^1001$"> <action application="bridge" data="sofia/salo.intellectics.com/1001%${server-domain-name}"/> <action application=“echo” /> </condition> </extension> <extension name="1002"> <condition field="destination_number" expression="^1002$"> <action application="bridge" data="sofia/salo.intellectics.com/111@sip.intellectics.com"/> <action application=“echo” /> </condition> </extension> </context> </section> </document> [root@salo freeswitch]# Michael Jerris wrote: Can you get a back-trace on this and report it either here, or in irc so we can address this issue. Thanks Mike From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Pekka K. Kurki Sent: Monday, December 25, 2006 1:45 PM To: freeswitch-users(a)lists.freeswitch.org Subject: Re: [Freeswitch-users] freeswitch: src/switch_core.c:1360: switch_core_session_strdup: Assertion `session != ((void *)0)' failed. same error... Brian West wrote: Update and try again. /b On Dec 25, 2006, at 7:27 AM, Pekka K. Kurki wrote: any time i try to bridge sofia channels i get: freeswitch: src/switch_core.c:1360: switch_core_session_strdup: Assertion `session != ((void *)0)' failed. Aborted (FC5, svn version of last week) br pekka -- (Mr.) Pekka K. Kurki Intellectics GesmbH http://www.intellectics.com Helenenstrasse 44, A-2500 Baden , Austria Tel. +43 2256 815811 Mobile +43 699 12532539 Fax +43 2256 815819 (Tel. US : 360-2267729 UK : 0870 4787064) sip:pekka.kurki@sip.intellectics.com h323:432256815811@sip.intellectics.com email: Pekka.Kurki(a)intellectics.com www: http://pekka.kurki.at <pekka.kurki.vcf> _______________________________________________ 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 _______________________________________________ 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 -- (Mr.) Pekka K. Kurki Intellectics GesmbH http://www.intellectics.com Helenenstrasse 44, A-2500 Baden , Austria Tel. +43 2256 815811 Mobile +43 699 12532539 Fax +43 2256 815819 (Tel. US : 360-2267729 UK : 0870 4787064) sip:pekka.kurki@sip.intellectics.com h323:432256815811@sip.intellectics.com email: Pekka.Kurki(a)intellectics.com www: http://pekka.kurki.at _______________________________________________ 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 -- (Mr.) Pekka K. Kurki Intellectics GesmbH http://www.intellectics.com Helenenstrasse 44, A-2500 Baden, Austria Tel. +43 2256 815811 Mobile +43 699 12532539 Fax +43 2256 815819 (Tel. US: 360-2267729 UK: 0870 4787064) sip:pekka.kurki@sip.intellectics.com h323:432256815811@sip.intellectics.com email: Pekka.Kurki(a)intellectics.com www: http://pekka.kurki.at begin:vcard fn:Pekka Kurki n:Kurki;Pekka org:Intellectics GesmbH adr:;;Helenenstr. 44;Baden;;A-2500;Austria email;internet:Pekka.Kurki@intellectics.com tel;work:+43 2256 815811 tel;fax:+43 2256 815819 tel;pager:skype:pkkurki tel;home:+43 2256 815813 tel;cell:+43 699 12532539 url:http://www.intellectics.com version:2.1 end:vcard _______________________________________________ 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 __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

4 3

Broadvoice
by Jonathan Rogers 17 Jun '07

17 Jun '07

Has anyone tried to use FreeSwitch with Broadvoice? I'm having no luck getting it to authenticate. -- Jonathan Rogers <JonathanRRogers(a)gmail.com>

6 20

Re: [Freeswitch-users] Dial from inside application
by Anthony Minessale 02 Jun '07

02 Jun '07

The exact equiv of what you showed would be this: static int pbx_dial(switch_core_session_t *session, void *data) { const switch_application_interface_t *application_interface; int ret; char *app_name = "bridge"; char *app_arg = (char *) data; if ((application_interface = switch_loadable_module_get_application_interface(app_name))) { application_interface->application_function(session, app_arg); return 0; } else { return -1; } } Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com JABBER:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: Mindaugas Kezys <mkezys(a)gmail.com> To: freeswitch-users(a)lists.freeswitch.org Sent: Tuesday, April 17, 2007 11:59:19 AM Subject: [Freeswitch-users] Dial from inside application  Hello, How to dial from inside application in FreeSwitch? I have such Asterisk code: static int pbx_dial(struct ast_channel *chan, void *data) { struct ast_app *app; int ret; app = pbx_findapp("Dial"); if (app) { ret = pbx_exec(chan, app, data); } else { ast_log(LOG_WARNING, "Could not find application (Dial)\n"); ret = -2; } return ret; } What is the equivalent in FreeSwitch world? Thank you for your help. Regards/Pagarbiai, Mindaugas Kezys http://www.kolmisoft.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 __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

2 2

Cluecon 2007 is almost here don't forget to register. June 26th - 28th
by Brian West 31 May '07

31 May '07

FreeSWITCHers, If you haven't please take a moment and look at http:// www.cluecon.com, This year is on track to be even better than our previous two years. If you can't make it please tell all your friends and co-workers that might be interested. Meet the people that make Open Source VoIP possible. Surj Patel and I will be working the conference together this year so please show up.. interact.. have fun. Cluecon Telephony Developer Conference 2007 (June 26th - 28th) will be held in: Best Western Inn of Chicago 162 E Ohio Street Chicago, Illinois 60611 Phone: (312) 787-3100 Toll Free: (800) 557-2378 US/Canada Call 877 742-CLUE for more information about the conference or to just chat with me. Thanks, Brian West PS: <subliminal-message>YOU MUST GO!!!</subliminal-message>

1 0

Re: [Freeswitch-users] Strange Reg
by Anthony Minessale 28 May '07

28 May '07

you need to monitor both the sip and the rtp traffic. Run freeswitch with TPORT_LOG=1 and also run wireshark on the machine capturing udp traffic on the interface where the sip calls will take place. Also make sure the sofia.conf.xml profile is setup the same as the default example in the source tree apart from the options necessary to enable the directory auth. Make sure the box freeswitch is on has no firewall enabled while you are testing (we do not support getting things to work with a firewall that's your job) Also make sure the nat router the ata is behind does not also have a firewall that is intentionally blocking RTP traffic (getting a trace from there with wireshark also would be helpful) Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com JABBER:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: Dave <lists+freeswitch(a)bigfatdave.com> To: freeswitch-users(a)lists.freeswitch.org Sent: Friday, May 25, 2007 1:24:36 AM Subject: Re: [Freeswitch-users] Strange Reg Now that I've documented what works, let's move on to what doesn't :-) I can't get outgoing calls to work at all (from the ATA out), and incoming calls ring, but have no audio in either direction for about half a minute, before just dying completely. Before I go around posting tons of logs, can somebody just give me some pointers about what to look for? Thank you, - Dave On Mon, May 21, 2007 at 05:17:30PM -0700, Anthony Minessale wrote: > You need to make sure you setup the user directory correctly. > > http://wiki.freeswitch.org/wiki/FreeSwitch_XML_User_Directory_Guide > > > My earlier email provided a precise example and warnings about making sure the tags etc match up. > > When you setup the <domain> tag in the directory > you must make sure there is a domain tag that has the same name as the host portion of the to: header in the register. > > Using forensics from your traces here is the exact domain tag to use > in directory.xml with 1 caveat: > > ----------------------------------------------------------------------- > **replace the string 1234 with the password you specified in the ata** > ----------------------------------------------------------------------- > > *NOTE* > normally $${domain} would be ok here as the name > assuming that the domain variable is set in freeswitch.xml > but I put the exact domain name to avoid confusion. > > *ALSO NOTE* > You can comment the password param if you want it to > accept any password for this user. > > <domain name="freeswitch1.asnetinc.net"> > <user id="homebase"> > <params> > <param name="password" value="1234"/> > </params> > <variables> > <variable name="force-contact" value="nat-connectile-dysfunction"/> > </variables> > </user> > </domain> > > I ask that if you get this working you please document this on our wiki in exchange for this exhaustive description of what you need to do. > > > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale@hotmail.com > JABBER:anthony.minessale@gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888@conference.freeswitch.org > iax:guest@conference.freeswitch.org/888 > googletalk:conf+888@conference.freeswitch.org > pstn:213-799-1400 > > > ----- Original Message ---- > From: Dave <lists+freeswitch(a)bigfatdave.com> > To: freeswitch-users(a)lists.freeswitch.org > Sent: Monday, May 21, 2007 6:49:00 PM > Subject: Re: [Freeswitch-users] Strange Reg > > Okay, I tried: > > recv 434 bytes from udp/[67.80.171.57]:48406 at 23:41:21.754015: > ------------------------------------------------------------------------ > REGISTER sip:freeswitch1.asnetinc.net SIP/2.0 > From: <sip:homebase@freeswitch1.asnetinc.net>;tag=a010033-13c4-108-40e02-46ec > To: <sip:homebase@freeswitch1.asnetinc.net> > Call-ID: a010033-13c4-108-40dfd-133e > CSeq: 1 REGISTER > Via: SIP/2.0/UDP 10.0.0.221:48406;branch=z9hG4bK-108-40e02-4ab5 > Max-Forwards: 70 > Supported: replaces > User-Agent: FXSO_GW > Contact: <sip:homebase@10.0.0.221:48406>;q=0.5 > Expires: 60 > Content-Length: 0 > > ------------------------------------------------------------------------ > 2007-05-21 23:41:21 [DEBUG] sofia_reg.c:387 sofia_reg_handle_register() Requesti > ng Registration from: [homebase(a)freeswitch1.asnetinc.net] > > ...and it responds with: > > send 640 bytes to udp/[67.80.171.57]:48406 at 23:41:21.840657: > ------------------------------------------------------------------------ > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP 10.0.0.221:48406;branch=z9hG4bK-108-40e02-4ab5;received=67.8 > 0.171.57 > From: <sip:homebase@freeswitch1.asnetinc.net>;tag=a010033-13c4-108-40e02-46ec > To: <sip:homebase@freeswitch1.asnetinc.net>;tag=e1arjB5HrFcHF > Call-ID: a010033-13c4-108-40dfd-133e > CSeq: 1 REGISTER > User-Agent: FreeSWITCH(mod_sofia) > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, > REFER, UPDATE, REGISTER, INFO > Supported: 100rel, precondition > WWW-Authenticate: Digest realm="freeswitch1.asnetinc.net", nonce="119b66e2-04 > 43-49a9-980f-d5b150274ff6", algorithm=MD5, qop="auth" > Content-Length: 0 > > ------------------------------------------------------------------------ > > Any ideas as to what's wrong here? > > Thanks, > - Dave > > > On Mon, May 21, 2007 at 12:17:12PM -0700, Anthony Minessale wrote: > > ok how about this: > > > > try rev 5207 and do everything I described before only in the > > force-contact set it to nat-connectile-dysfunction > > > > <variable name="force-contact" value="nat-connectile-dysfunction"/> > > > > This will rewrite the host and ip of the contact to use the > > ip and port the req was received from. To benefit from this feature, you will probably need to set the ata to constantly register to maintain this dynamic reverse mapping created by the nat routers. Most of them will destroy any mappings with inactivity after a certain period of time so you must make your reg freq more often that that time period whatever it may be. > > > > > > > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > > > AIM: anthm > > MSN:anthony_minessale@hotmail.com > > JABBER:anthony.minessale@gmail.com > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888@conference.freeswitch.org > > iax:guest@conference.freeswitch.org/888 > > googletalk:conf+888@conference.freeswitch.org > > pstn:213-799-1400 > > > > > > ----- Original Message ---- > > From: Dave <lists+freeswitch(a)bigfatdave.com> > > To: freeswitch-users(a)lists.freeswitch.org > > Sent: Monday, May 21, 2007 11:24:55 AM > > Subject: Re: [Freeswitch-users] Strange Reg > > > > Hi Anthony, > > > > I assume that means we can change the topolgy to something like this: > > ATA<->NAT<->Internet<->FS1 > > > > Basically, what we're doing here is telling FS1 how to find ATA. > > There are only two problems: > > 1) NAT is on DHCP. (I assume this can be worked around by using DynDNS > > instead of an IP.) > > 2) As I discovered while looking for the mystery 10.0.0.221 IP, NAT is > > actually two NATs, one behind the other, and I only have access to one > > of them. > > > > While you're at it, why don't you just make an option (<variable > > name='my-ata-sucks' value='true'/>) to totally break RFCs and mimic > > Asterisk's treatment? :-) > > > > Thanks a bunch for all your help, > > - Dave > > > > > > On Mon, May 21, 2007 at 07:25:01AM -0700, Anthony Minessale wrote: > > > Hi, > > > > > > You sound like a good candidate for my new experimental force contact feature. > > > > > > For this to work you need to use the directory for auth instead of the blind reg mode. > > > > > > *) Edit sofia.conf.xml and comment the blind reg option. > > > *) Edit directory.xml and add a <user> tag to the <domain name="$${domain}"> > > > section making sure $${domain} is also the name of the domain portion of > > > the sip's client registration. > > > > > > edit the file to look something like this and put in a absolute contact to use when this user registers regardless of what he advertises. This means that if you are using nat you should setup a permenant port forwarding for whatver uri you mention. > > > > > > e.g. > > > > > > Assuming your public ip is 66.2.2.2 and your phone is 10.0.0.2 > > > > > > add a rule to always port forward udp port 5555 of 66.2.2.2 to 10.0.0.2 5060 > > > then edit the following so the contact uri is user@66.2.2.2:5555 > > > > > > Whenever the phone registers with this user the specified contact will be added to the database instead of the real one this should make it possible for your phone to receive sip traffic even if it does not support STUN. > > > > > > <domain name="$${domain}"> > > > <user id="myuser"> > > > <params> > > > <param name="password" value="mypass"/> > > > </params> > > > <variables> > > > <variable name="force-contact" value="[TESTING] sip:myuser@66.2.2.2:5555"/> > > > </variables> > > > </user> > > > </domain> > > > > > > Anthony Minessale II > > > > > > FreeSWITCH http://www.freeswitch.org/ > > > ClueCon http://www.cluecon.com/ > > > > > > AIM: anthm > > > MSN:anthony_minessale@hotmail.com > > > JABBER:anthony.minessale@gmail.com > > > IRC: irc.freenode.net #freeswitch > > > > > > FreeSWITCH Developer Conference > > > sip:888@conference.freeswitch.org > > > iax:guest@conference.freeswitch.org/888 > > > googletalk:conf+888@conference.freeswitch.org > > > pstn:213-799-1400 > > > > > > > > > ----- Original Message ---- > > > From: Dave <lists+freeswitch(a)bigfatdave.com> > > > To: FSwtch Usrs <freeswitch-users(a)lists.freeswitch.org> > > > Sent: Monday, May 21, 2007 2:20:53 AM > > > Subject: [Freeswitch-users] Strange Reg > > > > > > Hi guys, > > > > > > I've got a rather interesting situation, here. I've got an ATA > > > registering to a FreeSWITCH(TM) (FS2), both behind NAT. So far, so good. > > > Now, the FreeSWITCH(TM) registers to another FreeSWITCH(TM) outside the > > > NAT (FS2), and that's where the problems start. Here's the configuration: > > > > > > ATA<->FS2<->NAT<->Internet<->FS1 > > > > > > ATA: 10.1.0.51 (behind NAT) > > > FS2: 10.1.0.30 (behind NAT) > > > FS1: freeswitch1.asnetinc.net (64.118.94.19) > > > > > > Now, here's where it gets very interesting. This is what FS1 gets when > > > FS2 registers: > > > > From: [R352928(a)freeswitch1.asnetinc.net] > > > > Contact: ["user" <sip:R352928@10.0.0.221:39260>] > > > > Expires: [3600] > > > > nta_agent: tport: Bad message > > > > > > Now, where on Earth did 10.0.0.221 come from??? (I've actually hardcoded > > > both external_rtp_ip and server_name to the public IP of the NAT in the > > > FS2 config, but for some strange reason, FS2 apparently claims to be at > > > 10.0.0.221, which isn't even its local IP.) > > > > > > As if that's not interesting enough, I get this from FS1, too, when it > > > tries to connect to another SIP server on the public Internet: > > > > outbound(0xb4e1e1a0): detected NAT: freeswitch1.asnetinc.net != 64.118.94.19 > > > > > > Now, since a forward DNS resolution claims that the inequality is false, I > > > assume it attempted a reverse DNS lookup, which would fail (since my ISP > > > takes forever to update PTR records). > > > > > > As usual, any hints, tips, tricks, ideas, suggestions, flames, questions, > > > or (ideally) answers are greatly appreciated :-) > > > > > > Thank you, > > > - Dave > > > > > > _______________________________________________ > > > 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 > > > > > > > > > > > > > > > > > > > > > > > > > > > ____________________________________________________________________________________Be a better Heartthrob. Get better relationship answers from someone who knows. Yahoo! Answers - Check it out. > > > http://answers.yahoo.com/dir/?link=list&sid=396545433 > > > _______________________________________________ > > > 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 > > > > > > _______________________________________________ > > 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 > > > > > > > > > > > > > > > > > > > > ____________________________________________________________________________________ > > Need Mail bonding? > > Go to the Yahoo! Mail Q&A for great tips from Yahoo! Answers users. > > http://answers.yahoo.com/dir/?link=list&sid=396546091 > > _______________________________________________ > > 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 > > > _______________________________________________ > 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 > > > > > > > > > ____________________________________________________________________________________ > No need to miss a message. Get email on-the-go > with Yahoo! Mail for Mobile. Get started. > http://mobile.yahoo.com/mail > _______________________________________________ > 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 _______________________________________________ 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 ____________________________________________________________________________________Need a vacation? Get great deals to amazing places on Yahoo! Travel. http://travel.yahoo.com/

3 3

Re: [Freeswitch-users] Strange Reg
by Anthony Minessale 25 May '07

25 May '07

You need to make sure you setup the user directory correctly. http://wiki.freeswitch.org/wiki/FreeSwitch_XML_User_Directory_Guide My earlier email provided a precise example and warnings about making sure the tags etc match up. When you setup the <domain> tag in the directory you must make sure there is a domain tag that has the same name as the host portion of the to: header in the register. Using forensics from your traces here is the exact domain tag to use in directory.xml with 1 caveat: ----------------------------------------------------------------------- **replace the string 1234 with the password you specified in the ata** ----------------------------------------------------------------------- *NOTE* normally $${domain} would be ok here as the name assuming that the domain variable is set in freeswitch.xml but I put the exact domain name to avoid confusion. *ALSO NOTE* You can comment the password param if you want it to accept any password for this user. <domain name="freeswitch1.asnetinc.net"> <user id="homebase"> <params> <param name="password" value="1234"/> </params> <variables> <variable name="force-contact" value="nat-connectile-dysfunction"/> </variables> </user> </domain> I ask that if you get this working you please document this on our wiki in exchange for this exhaustive description of what you need to do. Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com JABBER:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: Dave <lists+freeswitch(a)bigfatdave.com> To: freeswitch-users(a)lists.freeswitch.org Sent: Monday, May 21, 2007 6:49:00 PM Subject: Re: [Freeswitch-users] Strange Reg Okay, I tried: recv 434 bytes from udp/[67.80.171.57]:48406 at 23:41:21.754015: ------------------------------------------------------------------------ REGISTER sip:freeswitch1.asnetinc.net SIP/2.0 From: <sip:homebase@freeswitch1.asnetinc.net>;tag=a010033-13c4-108-40e02-46ec To: <sip:homebase@freeswitch1.asnetinc.net> Call-ID: a010033-13c4-108-40dfd-133e CSeq: 1 REGISTER Via: SIP/2.0/UDP 10.0.0.221:48406;branch=z9hG4bK-108-40e02-4ab5 Max-Forwards: 70 Supported: replaces User-Agent: FXSO_GW Contact: <sip:homebase@10.0.0.221:48406>;q=0.5 Expires: 60 Content-Length: 0 ------------------------------------------------------------------------ 2007-05-21 23:41:21 [DEBUG] sofia_reg.c:387 sofia_reg_handle_register() Requesti ng Registration from: [homebase(a)freeswitch1.asnetinc.net] ...and it responds with: send 640 bytes to udp/[67.80.171.57]:48406 at 23:41:21.840657: ------------------------------------------------------------------------ SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.0.0.221:48406;branch=z9hG4bK-108-40e02-4ab5;received=67.8 0.171.57 From: <sip:homebase@freeswitch1.asnetinc.net>;tag=a010033-13c4-108-40e02-46ec To: <sip:homebase@freeswitch1.asnetinc.net>;tag=e1arjB5HrFcHF Call-ID: a010033-13c4-108-40dfd-133e CSeq: 1 REGISTER User-Agent: FreeSWITCH(mod_sofia) Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: 100rel, precondition WWW-Authenticate: Digest realm="freeswitch1.asnetinc.net", nonce="119b66e2-04 43-49a9-980f-d5b150274ff6", algorithm=MD5, qop="auth" Content-Length: 0 ------------------------------------------------------------------------ Any ideas as to what's wrong here? Thanks, - Dave On Mon, May 21, 2007 at 12:17:12PM -0700, Anthony Minessale wrote: > ok how about this: > > try rev 5207 and do everything I described before only in the > force-contact set it to nat-connectile-dysfunction > > <variable name="force-contact" value="nat-connectile-dysfunction"/> > > This will rewrite the host and ip of the contact to use the > ip and port the req was received from. To benefit from this feature, you will probably need to set the ata to constantly register to maintain this dynamic reverse mapping created by the nat routers. Most of them will destroy any mappings with inactivity after a certain period of time so you must make your reg freq more often that that time period whatever it may be. > > > > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale@hotmail.com > JABBER:anthony.minessale@gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888@conference.freeswitch.org > iax:guest@conference.freeswitch.org/888 > googletalk:conf+888@conference.freeswitch.org > pstn:213-799-1400 > > > ----- Original Message ---- > From: Dave <lists+freeswitch(a)bigfatdave.com> > To: freeswitch-users(a)lists.freeswitch.org > Sent: Monday, May 21, 2007 11:24:55 AM > Subject: Re: [Freeswitch-users] Strange Reg > > Hi Anthony, > > I assume that means we can change the topolgy to something like this: > ATA<->NAT<->Internet<->FS1 > > Basically, what we're doing here is telling FS1 how to find ATA. > There are only two problems: > 1) NAT is on DHCP. (I assume this can be worked around by using DynDNS > instead of an IP.) > 2) As I discovered while looking for the mystery 10.0.0.221 IP, NAT is > actually two NATs, one behind the other, and I only have access to one > of them. > > While you're at it, why don't you just make an option (<variable > name='my-ata-sucks' value='true'/>) to totally break RFCs and mimic > Asterisk's treatment? :-) > > Thanks a bunch for all your help, > - Dave > > > On Mon, May 21, 2007 at 07:25:01AM -0700, Anthony Minessale wrote: > > Hi, > > > > You sound like a good candidate for my new experimental force contact feature. > > > > For this to work you need to use the directory for auth instead of the blind reg mode. > > > > *) Edit sofia.conf.xml and comment the blind reg option. > > *) Edit directory.xml and add a <user> tag to the <domain name="$${domain}"> > > section making sure $${domain} is also the name of the domain portion of > > the sip's client registration. > > > > edit the file to look something like this and put in a absolute contact to use when this user registers regardless of what he advertises. This means that if you are using nat you should setup a permenant port forwarding for whatver uri you mention. > > > > e.g. > > > > Assuming your public ip is 66.2.2.2 and your phone is 10.0.0.2 > > > > add a rule to always port forward udp port 5555 of 66.2.2.2 to 10.0.0.2 5060 > > then edit the following so the contact uri is user@66.2.2.2:5555 > > > > Whenever the phone registers with this user the specified contact will be added to the database instead of the real one this should make it possible for your phone to receive sip traffic even if it does not support STUN. > > > > <domain name="$${domain}"> > > <user id="myuser"> > > <params> > > <param name="password" value="mypass"/> > > </params> > > <variables> > > <variable name="force-contact" value="[TESTING] sip:myuser@66.2.2.2:5555"/> > > </variables> > > </user> > > </domain> > > > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > > > AIM: anthm > > MSN:anthony_minessale@hotmail.com > > JABBER:anthony.minessale@gmail.com > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888@conference.freeswitch.org > > iax:guest@conference.freeswitch.org/888 > > googletalk:conf+888@conference.freeswitch.org > > pstn:213-799-1400 > > > > > > ----- Original Message ---- > > From: Dave <lists+freeswitch(a)bigfatdave.com> > > To: FSwtch Usrs <freeswitch-users(a)lists.freeswitch.org> > > Sent: Monday, May 21, 2007 2:20:53 AM > > Subject: [Freeswitch-users] Strange Reg > > > > Hi guys, > > > > I've got a rather interesting situation, here. I've got an ATA > > registering to a FreeSWITCH(TM) (FS2), both behind NAT. So far, so good. > > Now, the FreeSWITCH(TM) registers to another FreeSWITCH(TM) outside the > > NAT (FS2), and that's where the problems start. Here's the configuration: > > > > ATA<->FS2<->NAT<->Internet<->FS1 > > > > ATA: 10.1.0.51 (behind NAT) > > FS2: 10.1.0.30 (behind NAT) > > FS1: freeswitch1.asnetinc.net (64.118.94.19) > > > > Now, here's where it gets very interesting. This is what FS1 gets when > > FS2 registers: > > > From: [R352928(a)freeswitch1.asnetinc.net] > > > Contact: ["user" <sip:R352928@10.0.0.221:39260>] > > > Expires: [3600] > > > nta_agent: tport: Bad message > > > > Now, where on Earth did 10.0.0.221 come from??? (I've actually hardcoded > > both external_rtp_ip and server_name to the public IP of the NAT in the > > FS2 config, but for some strange reason, FS2 apparently claims to be at > > 10.0.0.221, which isn't even its local IP.) > > > > As if that's not interesting enough, I get this from FS1, too, when it > > tries to connect to another SIP server on the public Internet: > > > outbound(0xb4e1e1a0): detected NAT: freeswitch1.asnetinc.net != 64.118.94.19 > > > > Now, since a forward DNS resolution claims that the inequality is false, I > > assume it attempted a reverse DNS lookup, which would fail (since my ISP > > takes forever to update PTR records). > > > > As usual, any hints, tips, tricks, ideas, suggestions, flames, questions, > > or (ideally) answers are greatly appreciated :-) > > > > Thank you, > > - Dave > > > > _______________________________________________ > > 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 > > > > > > > > > > > > > > > > > > ____________________________________________________________________________________Be a better Heartthrob. Get better relationship answers from someone who knows. Yahoo! Answers - Check it out. > > http://answers.yahoo.com/dir/?link=list&sid=396545433 > > _______________________________________________ > > 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 > > > _______________________________________________ > 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 > > > > > > > > > > ____________________________________________________________________________________ > Need Mail bonding? > Go to the Yahoo! Mail Q&A for great tips from Yahoo! Answers users. > http://answers.yahoo.com/dir/?link=list&sid=396546091 > _______________________________________________ > 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 _______________________________________________ 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 ____________________________________________________________________________________ No need to miss a message. Get email on-the-go with Yahoo! Mail for Mobile. Get started. http://mobile.yahoo.com/mail

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pastebin
by Brian West 25 May '07

25 May '07

Can someone that is good with php provide me with a patched version of pastebin that has captcha support .. I have had to do some drastic things to stop spammers. I'm not that great with php but someone out there can whip up a patched version ? /b

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Re: [Freeswitch-users] Does FS handle remote NATed phones?
by Anthony Minessale 24 May '07

24 May '07

thanks =D Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com JABBER:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: Jason Garland <jgarland(a)gmail.com> To: freeswitch-users(a)lists.freeswitch.org Sent: Thursday, May 24, 2007 12:05:26 PM Subject: Re: [Freeswitch-users] Does FS handle remote NATed phones? http://wiki.freeswitch.org/wiki/Talk:Event_Socket http://wiki.freeswitch.org/wiki/Event_Socket On 5/24/07, Anthony Minessale <anthmct(a)yahoo.com> wrote: > > We do have mod_event_socket's "socket" app which is similar to the network > based fast-agi (FAGI). This application makes an outbound tcp connection to > the specified ip:port and the other end can control the call in similar ways > to an agi only the dialect is much different and it supports async > operations e.g. telling the call to play foo.wav and getting an immediate > return to listen for dtmf as the file plays rather than block till the file > is over. > > there is an example perl script in tree: > > "scripts/socket/sock.pl" > > The data syntax from the dialplan is > <ip>:<port> [<keywords>] > > keywords: > > async: > > The "async" keyword indicates that all commands will return instantly making > it possible to monitor the socket for events etc while the > stack of commands are executing. > > If the async keyword is absent then all calls will block until the > command has finished. > > full: > > The "full" keyword indicates that the other end will have the full command > set for event_socket this is the same command set an inbound connection to > event_socket has so you can execute api commands, get global events etc. > > If the "full" keyword is absent the command set and events are limited > to that paticular call > > Here are examples of how to use it in the dialplan. > > <action application="socket" data="127.0.0.1:8084"/> > <action application="socket" data="127.0.0.1:8084 async"/> > <action application="socket" data="127.0.0.1:8084 full"/> > <action application="socket" data="127.0.0.1:8084 async full"/> > > > I think if wasim's angel want's to implement it a good idea > is to write it as a separate application and use this interface > to control the call then let the forked processes in the remote > application provide the AGI emulation. > > We have someone working on a C library to use as an event socket library > also. > but in the meantime there is also a crude C example with no client library > but just inline commands in "scripts/socket/socket2me" this small C program > demonstrates how you can use the interface to not only control the call but > to also request a 2 way media stream and process the data. This application > uses spandsp to implement a fax send/recv similar to asterisk's rxfax txfax > only over the loopback interface. > > > > /me hopes somone makes a wiki page out of this info =D > > > > > > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale@hotmail.com > JABBER:anthony.minessale@gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888@conference.freeswitch.org > iax:guest@conference.freeswitch.org/888 > googletalk:conf+888@conference.freeswitch.org > pstn:213-799-1400 > > > ----- Original Message ---- > From: Wasim Baig <wasim(a)convergence.pk> > To: freeswitch-users(a)lists.freeswitch.org > Sent: Thursday, May 24, 2007 7:13:24 AM > Subject: Re: [Freeswitch-users] Does FS handle remote NATed phones? > > On 5/24/07, Fred <codecomplete(a)free.fr> wrote: > > > Hopefully, the AGI interface works the same, so that existing scripts will > > still work in FS. > > nope, not as yet, but i'm convincing an angel to write up a mod_agi that > would allow using most if not all of the AGI functions in freeswitch > (atleast SetVar and GetVar, if not Dial) > > imho thats one of the main bottlenecks for people who have existing > asterisk deployments on AGI from switching to FS, lets see if we can get > some good news on this shortly > > -- > wasim h. baig | principal consultant | convergence pk | +92 300 8508070 > _______________________________________________ > 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 > > > ________________________________ > Ready for the edge of your seat? Check out tonight's top picks on Yahoo! TV. > _______________________________________________ > 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 > > _______________________________________________ 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 ____________________________________________________________________________________ Don't pick lemons. See all the new 2007 cars at Yahoo! Autos. http://autos.yahoo.com/new_cars.html

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Re: [Freeswitch-users] Does FS handle remote NATed phones?
by Anthony Minessale 24 May '07

24 May '07

We do have mod_event_socket's "socket" app which is similar to the network based fast-agi (FAGI). This application makes an outbound tcp connection to the specified ip:port and the other end can control the call in similar ways to an agi only the dialect is much different and it supports async operations e.g. telling the call to play foo.wav and getting an immediate return to listen for dtmf as the file plays rather than block till the file is over. there is an example perl script in tree: "scripts/socket/sock.pl" The data syntax from the dialplan is <ip>:<port> [<keywords>] keywords: async: The "async" keyword indicates that all commands will return instantly making it possible to monitor the socket for events etc while the stack of commands are executing. If the async keyword is absent then all calls will block until the command has finished. full: The "full" keyword indicates that the other end will have the full command set for event_socket this is the same command set an inbound connection to event_socket has so you can execute api commands, get global events etc. If the "full" keyword is absent the command set and events are limited to that paticular call Here are examples of how to use it in the dialplan. <action application="socket" data="127.0.0.1:8084"/> <action application="socket" data="127.0.0.1:8084 async"/> <action application="socket" data="127.0.0.1:8084 full"/> <action application="socket" data="127.0.0.1:8084 async full"/> I think if wasim's angel want's to implement it a good idea is to write it as a separate application and use this interface to control the call then let the forked processes in the remote application provide the AGI emulation. We have someone working on a C library to use as an event socket library also. but in the meantime there is also a crude C example with no client library but just inline commands in "scripts/socket/socket2me" this small C program demonstrates how you can use the interface to not only control the call but to also request a 2 way media stream and process the data. This application uses spandsp to implement a fax send/recv similar to asterisk's rxfax txfax only over the loopback interface. /me hopes somone makes a wiki page out of this info =D Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com JABBER:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: Wasim Baig <wasim(a)convergence.pk> To: freeswitch-users(a)lists.freeswitch.org Sent: Thursday, May 24, 2007 7:13:24 AM Subject: Re: [Freeswitch-users] Does FS handle remote NATed phones? On 5/24/07, Fred <codecomplete(a)free.fr> wrote: Hopefully, the AGI interface works the same, so that existing scripts will still work in FS. nope, not as yet, but i'm convincing an angel to write up a mod_agi that would allow using most if not all of the AGI functions in freeswitch (atleast SetVar and GetVar, if not Dial) imho thats one of the main bottlenecks for people who have existing asterisk deployments on AGI from switching to FS, lets see if we can get some good news on this shortly -- wasim h. baig | principal consultant | convergence pk | +92 300 8508070 _______________________________________________ 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 ____________________________________________________________________________________Get the Yahoo! toolbar and be alerted to new email wherever you're surfing. http://new.toolbar.yahoo.com/toolbar/features/mail/index.php

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