- FreeSWITCH-users - FreeSWITCH Mailing Lists

Sangoma WANPipe and libpri
by Michael Collins 22 Mar '07

22 Mar '07

Gang, I saw this Sangoma tutorial on the wiki: http://wiki.freeswitch.org/wiki/FreeSwitch_Enpoint_Wanpipe I noticed that it said that libpri was a prerequisite. Question: is there a GPL issue with using libpri in this fashion? Just curious. I'd hate to buy a Sangoma card and not be able to use it with my PRI circuits. Thanks, MC

2 2

Re: [Freeswitch-users] Bridging dingaling to sofia
by Kieran O'Loughlin 22 Mar '07

22 Mar '07

Hey Anthony, I found one other thing. I registered an account with freeworlddialup and tried bridging the call to that account rather than the blueface account. That worked! So my guess now is that blueface is silently rejecting the call for some reason. I don't think the issue is really on the freeswitch side. Of course if you have any ideas on what the problem might be or how I could troubleshoot it that would be a great help. Thanks, Kieran.

2 1

Re: [Freeswitch-users] Bridging dingaling to sofia
by Kieran O'Loughlin 21 Mar '07

21 Mar '07

Thanks Anthony, Here is the complete console trace you mentioned. Let me first explain the architecture. I wanted to get NAT out of the picture so I installed Freeswitch on a linux box with a public internet IP address. I've masked that public IP address with <FreeswitchServerIP>. My googletalk client is running on 192.168.2.53 behind a NAT firewall from my home. The public IP address for my home is masked with the text <MyExternalIPHome>. So during this call Googletalk should talk with freeswitch. Freeswitch should choose to communicate back on <MyExternalIPHome> rather than 192.168.2.53 which it does. Blueface's sip server at sip.blueface.ie resolves with no problem on the box. As I mentioned in my earlier email if I register a sip phone with freeswitch directly i can call to blueface using the same extension 1004j as I use for google without any problem. If I modify the extension to bridge to this local sip registration rather than to blueface it also works fine. So bridging local sip to a remote sip registration works and bridging googletalk to a local sip registration works, it only fails when bridging googletalk to a remote sip registration. Strange isn't it? Anyway any help you can offer will be greatly appreciated. Thanks, Kieran. 2007-03-21 16:59:45 [CONSOLE] switch_core.c:4697 switch_core_init_and_modload() Bringing up environment. 2007-03-21 16:59:45 [CONSOLE] switch_core.c:4698 switch_core_init_and_modload() Loading Modules. 2007-03-21 16:59:45 [CONSOLE] switch_loadable_module.c:454 switch_loadable_module_load_file() Successfully Loaded [mod_console] 2007-03-21 16:59:45 [CONSOLE] switch_loadable_module.c:454 switch_loadable_module_load_file() Successfully Loaded [mod_enum] 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Dialplan 'enum' 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:215 switch_loadable_module_process() Adding Application 'enum' 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:235 switch_loadable_module_process() Adding API Function 'enum' 2007-03-21 16:59:45 [DEBUG] mod_dingaling.c:1772 init_profile() Started Thread for <myUser>@gmail.com/talk@XML 2007-03-21 16:59:45 [CONSOLE] switch_loadable_module.c:454 switch_loadable_module_load_file() Successfully Loaded [mod_dingaling] 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:128 switch_loadable_module_process() Adding Endpoint 'dingaling' 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:235 switch_loadable_module_process() Adding API Function 'dl_login' 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:235 switch_loadable_module_process() Adding API Function 'dl_logout' 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:330 switch_loadable_module_process() Adding Chat interface 'jingle' PortAudio version number = 1899 PortAudio version text = 'PortAudio V19-devel' Number of devices = 0 -------------------------------------------------------------------------------- 2007-03-21 16:59:45 [ERR] mod_portaudio.c:1018 load_config() Cannot find an input device! 2007-03-21 16:59:45 [ERR] mod_portaudio.c:1028 load_config() Cannot find an input device! 2007-03-21 16:59:45 [ERR] switch_loadable_module.c:438 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_portaudio.so **Module load routine returned an error** 2007-03-21 16:59:45 [CRIT] mod_sofia.c:5558 config_sofia() The <registrations> syntax has been discontinued, please see the new syntax in the default configuration examples 2007-03-21 16:59:45 [NOTICE] mod_sofia.c:5682 config_sofia() Started Profile <FreeswitchServerIP> [sofia_reg_<FreeswitchServerIP>] 2007-03-21 16:59:45 [CONSOLE] switch_loadable_module.c:454 switch_loadable_module_load_file() Successfully Loaded [mod_sofia] 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:128 switch_loadable_module_process() Adding Endpoint 'sofia' 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:330 switch_loadable_module_process() Adding Chat interface 'sip' 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:370 switch_loadable_module_process() Adding Management interface 'mod_sofia' OID[.1.3.6.1.4.1.27880.1] 2007-03-21 16:59:45 [CONSOLE] switch_loadable_module.c:454 switch_loadable_module_load_file() Successfully Loaded [mod_bridgecall] 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:215 switch_loadable_module_process() Adding Application 'bridge' 2007-03-21 16:59:45 [CONSOLE] switch_loadable_module.c:454 switch_loadable_module_load_file() Successfully Loaded [mod_commands] 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:235 switch_loadable_module_process() Adding API Function 'originate' 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:235 switch_loadable_module_process() Adding API Function 'killchan' 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:235 switch_loadable_module_process() Adding API Function 'reloadxml' 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:235 switch_loadable_module_process() Adding API Function 'load' 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:235 switch_loadable_module_process() Adding API Function 'transfer' 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:235 switch_loadable_module_process() Adding API Function 'pause' 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:235 switch_loadable_module_process() Adding API Function 'show' 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:235 switch_loadable_module_process() Adding API Function 'status' 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:235 switch_loadable_module_process() Adding API Function 'uuid_bridge' 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:235 switch_loadable_module_process() Adding API Function 'session_record' 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:235 switch_loadable_module_process() Adding API Function 'broadcast' 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:235 switch_loadable_module_process() Adding API Function 'hold' 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:235 switch_loadable_module_process() Adding API Function 'media' 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:235 switch_loadable_module_process() Adding API Function 'fsctl' 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:235 switch_loadable_module_process() Adding API Function 'help' 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:235 switch_loadable_module_process() Adding API Function 'version' 2007-03-21 16:59:45 [CONSOLE] switch_loadable_module.c:454 switch_loadable_module_load_file() Successfully Loaded [mod_conference] 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:215 switch_loadable_module_process() Adding Application 'conference' 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:235 switch_loadable_module_process() Adding API Function 'conference' 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:330 switch_loadable_module_process() Adding Chat interface 'conf' 2007-03-21 16:59:45 [CONSOLE] switch_loadable_module.c:454 switch_loadable_module_load_file() Successfully Loaded [mod_dptools] 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:215 switch_loadable_module_process() Adding Application 'privacy' 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:215 switch_loadable_module_process() Adding Application 'transfer' 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:215 switch_loadable_module_process() Adding Application 'sleep' 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:215 switch_loadable_module_process() Adding Application 'strftime' 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:215 switch_loadable_module_process() Adding Application 'phrase' 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:215 switch_loadable_module_process() Adding Application 'eval' 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:215 switch_loadable_module_process() Adding Application 'pre_answer' 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:215 switch_loadable_module_process() Adding Application 'answer' 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:215 switch_loadable_module_process() Adding Application 'hangup' 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:215 switch_loadable_module_process() Adding Application 'log' 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:215 switch_loadable_module_process() Adding Application 'info' 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:215 switch_loadable_module_process() Adding Application 'export' 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:215 switch_loadable_module_process() Adding Application 'set' 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:215 switch_loadable_module_process() Adding Application 'unset' 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:215 switch_loadable_module_process() Adding Application 'ring_ready' 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:215 switch_loadable_module_process() Adding Application 'detect_speech' 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:215 switch_loadable_module_process() Adding Application 'ivr' 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:215 switch_loadable_module_process() Adding Application 'redirect' 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:215 switch_loadable_module_process() Adding Application 'queue_dtmf' 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:235 switch_loadable_module_process() Adding API Function 'presence' 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:235 switch_loadable_module_process() Adding API Function 'strftime' 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:235 switch_loadable_module_process() Adding API Function 'chat' su_port(0x80a3228): epoll_create(): Function not implemented 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:235 switch_loadable_module_process() Adding API Function 'strepoch' su_port(0x80a4058): epoll_create(): Function not implemented 2007-03-21 16:59:45 [CONSOLE] switch_loadable_module.c:454 switch_loadable_module_load_file() Successfully Loaded [mod_echo] 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:215 switch_loadable_module_process() Adding Application 'echo' 2007-03-21 16:59:45 [CONSOLE] switch_loadable_module.c:454 switch_loadable_module_load_file() Successfully Loaded [mod_playback] 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:215 switch_loadable_module_process() Adding Application 'playback' 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:215 switch_loadable_module_process() Adding Application 'stop_record_session' 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:215 switch_loadable_module_process() Adding Application 'record_session' 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:215 switch_loadable_module_process() Adding Application 'record' 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:215 switch_loadable_module_process() Adding Application 'speak' 2007-03-21 16:59:45 [CONSOLE] switch_loadable_module.c:454 switch_loadable_module_load_file() Successfully Loaded [mod_dialplan_xml] 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:179 switch_loadable_module_process() Adding Dialplan 'XML' 2007-03-21 16:59:45 [CONSOLE] switch_loadable_module.c:454 switch_loadable_module_load_file() Successfully Loaded [mod_g711] 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:153 switch_loadable_module_process() Adding Codec 'PCMU' (g711 ulaw) 8000hz 10ms 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:153 switch_loadable_module_process() Adding Codec 'PCMU' (g711 ulaw) 8000hz 20ms 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:153 switch_loadable_module_process() Adding Codec 'PCMU' (g711 ulaw) 8000hz 30ms 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:153 switch_loadable_module_process() Adding Codec 'PCMU' (g711 ulaw) 8000hz 60ms 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:153 switch_loadable_module_process() Adding Codec 'PCMU' (g711 ulaw) 8000hz 120ms 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:153 switch_loadable_module_process() Adding Codec 'PCMA' (g711 alaw) 8000hz 10ms 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:153 switch_loadable_module_process() Adding Codec 'PCMA' (g711 alaw) 8000hz 20ms 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:153 switch_loadable_module_process() Adding Codec 'PCMA' (g711 alaw) 8000hz 30ms 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:153 switch_loadable_module_process() Adding Codec 'PCMA' (g711 alaw) 8000hz 60ms 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:153 switch_loadable_module_process() Adding Codec 'PCMA' (g711 alaw) 8000hz 120ms 2007-03-21 16:59:45 [CONSOLE] switch_loadable_module.c:454 switch_loadable_module_load_file() Successfully Loaded [mod_gsm] 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:153 switch_loadable_module_process() Adding Codec 'GSM' (gsm) 8000hz 20ms 2007-03-21 16:59:45 [CONSOLE] switch_loadable_module.c:454 switch_loadable_module_load_file() Successfully Loaded [mod_l16] 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:153 switch_loadable_module_process() Adding Codec 'L16' (raw signed linear (16 bit)) 8000hz 10ms 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:153 switch_loadable_module_process() Adding Codec 'L16' (raw signed linear (16 bit)) 8000hz 20ms 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:153 switch_loadable_module_process() Adding Codec 'L16' (raw signed linear (16 bit)) 8000hz 30ms 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:153 switch_loadable_module_process() Adding Codec 'L16' (raw signed linear (16 bit)) 8000hz 60ms 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:153 switch_loadable_module_process() Adding Codec 'L16' (raw signed linear (16 bit)) 8000hz 120ms 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:153 switch_loadable_module_process() Adding Codec 'L16' (raw signed linear (16 bit)) 16000hz 10ms 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:153 switch_loadable_module_process() Adding Codec 'L16' (raw signed linear (16 bit)) 16000hz 20ms 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:153 switch_loadable_module_process() Adding Codec 'L16' (raw signed linear (16 bit)) 16000hz 30ms 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:153 switch_loadable_module_process() Adding Codec 'L16' (raw signed linear (16 bit)) 16000hz 60ms 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:153 switch_loadable_module_process() Adding Codec 'L16' (raw signed linear (16 bit)) 8000hz 120ms 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:153 switch_loadable_module_process() Adding Codec 'L16' (raw signed linear (16 bit)) 22050hz 20ms 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:153 switch_loadable_module_process() Adding Codec 'L16' (raw signed linear (16 bit)) 32000hz 10ms 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:153 switch_loadable_module_process() Adding Codec 'L16' (raw signed linear (16 bit)) 32000hz 20ms 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:153 switch_loadable_module_process() Adding Codec 'L16' (raw signed linear (16 bit)) 32000hz 30ms 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:153 switch_loadable_module_process() Adding Codec 'L16' (raw signed linear (16 bit)) 32000hz 60ms 2007-03-21 16:59:45 [INFO] mod_sndfile.c:298 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") ================================================================================ 2007-03-21 16:59:45 [CONSOLE] switch_loadable_module.c:454 switch_loadable_module_load_file() Successfully Loaded [mod_sndfile] 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:257 switch_loadable_module_process() Adding File Format 'aiff' 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:257 switch_loadable_module_process() Adding File Format 'au' 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:257 switch_loadable_module_process() Adding File Format 'avr' 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:257 switch_loadable_module_process() Adding File Format 'caf' 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:257 switch_loadable_module_process() Adding File Format 'htk' 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:257 switch_loadable_module_process() Adding File Format 'iff' 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:257 switch_loadable_module_process() Adding File Format 'mat' 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:257 switch_loadable_module_process() Adding File Format 'paf' 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:257 switch_loadable_module_process() Adding File Format 'pvf' 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:257 switch_loadable_module_process() Adding File Format 'raw' 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:257 switch_loadable_module_process() Adding File Format 'sd2' 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:257 switch_loadable_module_process() Adding File Format 'sds' 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:257 switch_loadable_module_process() Adding File Format 'sf' 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:257 switch_loadable_module_process() Adding File Format 'voc' 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:257 switch_loadable_module_process() Adding File Format 'w64' 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:257 switch_loadable_module_process() Adding File Format 'wav' 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:257 switch_loadable_module_process() Adding File Format 'xi' 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:257 switch_loadable_module_process() Adding File Format 'r8' 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:257 switch_loadable_module_process() Adding File Format 'r16' 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:257 switch_loadable_module_process() Adding File Format 'r24' 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:257 switch_loadable_module_process() Adding File Format 'r32' 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:257 switch_loadable_module_process() Adding File Format 'gsm' 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:257 switch_loadable_module_process() Adding File Format 'ul' 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:257 switch_loadable_module_process() Adding File Format 'al' 2007-03-21 16:59:45 [CONSOLE] switch_loadable_module.c:454 switch_loadable_module_load_file() Successfully Loaded [mod_native_file] 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:257 switch_loadable_module_process() Adding File Format 'PCMA' 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:257 switch_loadable_module_process() Adding File Format 'L16' 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:257 switch_loadable_module_process() Adding File Format 'PCMU' 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:257 switch_loadable_module_process() Adding File Format 'GSM' 2007-03-21 16:59:45 [CONSOLE] switch_loadable_module.c:454 switch_loadable_module_load_file() Successfully Loaded [mod_softtimer] 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:197 switch_loadable_module_process() Adding Timer 'soft' 2007-03-21 16:59:45 [CONSOLE] switch_loadable_module.c:454 switch_loadable_module_load_file() Successfully Loaded [mod_say_en] 2007-03-21 16:59:45 [NOTICE] switch_loadable_module.c:348 switch_loadable_module_process() Adding Say interface 'en' 2007-03-21 16:59:45 [CONSOLE] switch_loadable_module.c:677 switch_loadable_module_init() open of post_load_modules.conf failed 2007-03-21 16:59:45 [CONSOLE] switch_loadable_module.c:110 switch_loadable_module_runtime() Starting runtime thread for mod_softtimer 2007-03-21 16:59:45 [CONSOLE] switch_core.c:4710 switch_core_init_and_modload() FreeSWITCH Version 1.0.4712 Started. Crash Protection [DISABLED] Max Sessions[1000] freeswitch@spock> 2007-03-21 16:59:46 [DEBUG] libdingaling.c:1151 on_stream() XMPP server connected 2007-03-21 16:59:46 [DEBUG] libdingaling.c:1163 on_stream() XMPP authenticated 2007-03-21 16:59:59 [DEBUG] mod_dingaling.c:2418 handle_signalling() Creating a session for 3776263765 2007-03-21 16:59:59 [NOTICE] switch_channel.c:347 switch_channel_set_name() New Chan DingaLing/new [9a5ea39a-d7cd-11db-96d7-896d5659d9ab] 2007-03-21 16:59:59 [DEBUG] mod_dingaling.c:2422 handle_signalling() DingaLing/new State Change CS_NEW -> CS_INIT 2007-03-21 16:59:59 [DEBUG] switch_core.c:3393 switch_core_session_run() (DingaLing/new) State INIT 2007-03-21 16:59:59 [NOTICE] mod_dingaling.c:959 negotiate_media() Ring-Ready DingaLing/new! 2007-03-21 16:59:59 [DEBUG] mod_dingaling.c:2507 handle_signalling() 12 payloads 2007-03-21 16:59:59 [DEBUG] mod_dingaling.c:2509 handle_signalling() Available Payload ISAC 103 2007-03-21 16:59:59 [DEBUG] mod_dingaling.c:2516 handle_signalling() compare ISAC 103/16000 to PCMU 0/8000 2007-03-21 16:59:59 [DEBUG] mod_dingaling.c:2509 handle_signalling() Available Payload IPCMWB 97 2007-03-21 16:59:59 [DEBUG] mod_dingaling.c:2516 handle_signalling() compare IPCMWB 97/16000 to PCMU 0/8000 2007-03-21 16:59:59 [DEBUG] mod_dingaling.c:2509 handle_signalling() Available Payload speex 99 2007-03-21 16:59:59 [DEBUG] mod_dingaling.c:2516 handle_signalling() compare speex 99/16000 to PCMU 0/8000 2007-03-21 16:59:59 [DEBUG] mod_dingaling.c:2509 handle_signalling() Available Payload G723 4 2007-03-21 16:59:59 [DEBUG] mod_dingaling.c:2516 handle_signalling() compare G723 4/8000 to PCMU 0/8000 2007-03-21 16:59:59 [DEBUG] mod_dingaling.c:2509 handle_signalling() Available Payload speex 98 2007-03-21 16:59:59 [DEBUG] mod_dingaling.c:2516 handle_signalling() compare speex 98/8000 to PCMU 0/8000 2007-03-21 16:59:59 [DEBUG] mod_dingaling.c:2509 handle_signalling() Available Payload EG711U 100 2007-03-21 16:59:59 [DEBUG] mod_dingaling.c:2516 handle_signalling() compare EG711U 100/8000 to PCMU 0/8000 2007-03-21 16:59:59 [DEBUG] mod_dingaling.c:2509 handle_signalling() Available Payload EG711A 101 2007-03-21 16:59:59 [DEBUG] mod_dingaling.c:2516 handle_signalling() compare EG711A 101/8000 to PCMU 0/8000 2007-03-21 16:59:59 [DEBUG] mod_dingaling.c:2509 handle_signalling() Available Payload PCMU 0 2007-03-21 16:59:59 [DEBUG] mod_dingaling.c:2516 handle_signalling() compare PCMU 0/8000 to PCMU 0/8000 2007-03-21 16:59:59 [DEBUG] mod_dingaling.c:2527 handle_signalling() Choosing Payload index 0 PCMU 0 2007-03-21 16:59:59 [DEBUG] mod_dingaling.c:929 do_describe() Send Describe [PCMU@8000] nta_agent: received garbage from udp/<MyExternalIPHome>:5060/sip 2007-03-21 17:00:01 [DEBUG] mod_dingaling.c:2381 handle_signalling() using Existing session for 3776263765 2007-03-21 17:00:01 [DEBUG] mod_dingaling.c:2575 handle_signalling() 1 candidates 2007-03-21 17:00:01 [DEBUG] mod_dingaling.c:2583 handle_signalling() candidates 192.168.2.53:2686 2007-03-21 17:00:01 [DEBUG] mod_dingaling.c:2381 handle_signalling() using Existing session for 3776263765 2007-03-21 17:00:01 [DEBUG] mod_dingaling.c:2575 handle_signalling() 2 candidates 2007-03-21 17:00:01 [DEBUG] mod_dingaling.c:2583 handle_signalling() candidates 192.168.2.53:2686 2007-03-21 17:00:01 [DEBUG] mod_dingaling.c:2583 handle_signalling() candidates <MyExternalIPHome>:2687 2007-03-21 17:00:01 [DEBUG] mod_dingaling.c:2613 handle_signalling() Acceptable Candidate <MyExternalIPHome>:2687 2007-03-21 17:00:01 [DEBUG] mod_dingaling.c:2678 handle_signalling() Creating an identity for 3776263765 <myCallingUser>@ gmail.com/Talk.v104B94A6705 <<myCallingUser>@gmail.com/Talk.v104B94A6705> 1004j 2007-03-21 17:00:01 [NOTICE] switch_channel.c:347 switch_channel_set_name() New Chan DingaLing/1004j [9a5ea39a-d7cd-11db-96d7-896d5659d9ab] 2007-03-21 17:00:01 [DEBUG] mod_dingaling.c:861 do_candidates() Send Candidate <FreeswitchServerIP>:16384 [oXcJOdeXEZN9avL6] 2007-03-21 17:00:01 [DEBUG] mod_dingaling.c:725 activate_rtp() Set Read Codec to PCMU@8000 2007-03-21 17:00:01 [DEBUG] mod_dingaling.c:740 activate_rtp() Set Write Codec to PCMU@8000 2007-03-21 17:00:01 [DEBUG] mod_dingaling.c:747 activate_rtp() SETUP RTP <FreeswitchServerIP>:16384 -> <MyExternalIPHome>:2687 2007-03-21 17:00:01 [DEBUG] switch_rtp.c:487 switch_rtp_create() Starting timer [soft] 160 bytes per 20000ms 2007-03-21 17:00:01 [DEBUG] mod_dingaling.c:1052 channel_on_init() DingaLing/1004j State Change CS_INIT -> CS_RING 2007-03-21 17:00:01 [DEBUG] switch_core.c:3429 switch_core_session_run() (DingaLing/1004j) State RING 2007-03-21 17:00:01 [DEBUG] mod_dingaling.c:1069 channel_on_ring() DingaLing/1004j CHANNEL RING 2007-03-21 17:00:01 [DEBUG] switch_core.c:3058 switch_core_standard_on_ring() Standard RING DingaLing/1004j 2007-03-21 17:00:01 [INFO] mod_dialplan_xml.c:219 dialplan_hunt() Processing <myCallingUser>@gmail.com/Talk.v104B94A6705->1004j! 2007-03-21 17:00:01 [DEBUG] mod_dialplan_xml.c:120 parse_exten() test conditions destination_number(1004j) =~ /^1004j/ 2007-03-21 17:00:01 [DEBUG] mod_dialplan_xml.c:355 dialplan_hunt() DingaLing/1004j State Change CS_RING -> CS_EXECUTE 2007-03-21 17:00:01 [DEBUG] switch_core.c:3465 switch_core_session_run() (DingaLing/1004j) State EXECUTE 2007-03-21 17:00:01 [DEBUG] mod_dingaling.c:1086 channel_on_execute() DingaLing/1004j CHANNEL EXECUTE 2007-03-21 17:00:01 [DEBUG] switch_core.c:3117 switch_core_standard_on_execute() Standard EXECUTE 2007-03-21 17:00:01 [NOTICE] switch_core.c:3127 switch_core_standard_on_execute() Execute answer() 2007-03-21 17:00:01 [NOTICE] mod_dptools.c:162 answer_function() Channel [DingaLing/1004j] has been answered 2007-03-21 17:00:01 [NOTICE] switch_core.c:3127 switch_core_standard_on_execute() Execute bridge(sofia/<FreeswitchServerIP>/<exten>@sip.blueface.ie) 2007-03-21 17:00:01 [NOTICE] switch_channel.c:347 switch_channel_set_name() New Chan sofia/<FreeswitchServerIP>/<exten>@sip.blueface.ie[9b63301c-d7cd-11db-96d7-896d5659d9ab] 2007-03-21 17:00:01 [DEBUG] mod_sofia.c:2484 sofia_outgoing_channel() sofia/<FreeswitchServerIP>/<exten>@sip.blueface.ie State Change CS_NEW -> CS_INIT 2007-03-21 17:00:01 [DEBUG] switch_core.c:3393 switch_core_session_run() (sofia/<FreeswitchServerIP>/<exten>@sip.blueface.ie) State INIT 2007-03-21 17:00:01 [DEBUG] mod_sofia.c:1323 sofia_on_init() SOFIA INIT 2007-03-21 17:00:01 [DEBUG] mod_sofia.c:4982 event_callback() event [nua_i_state] status [0][INVITE sent] session: sofia/<FreeswitchServerIP>/<exten>@sip.blueface.ie 2007-03-21 17:00:01 [DEBUG] mod_sofia.c:2991 sip_i_state() Channel sofia/<FreeswitchServerIP>/<exten>@sip.blueface.ie entering state [calling] 2007-03-21 17:00:01 [DEBUG] mod_sofia.c:1336 sofia_on_init() sofia/<FreeswitchServerIP>/<exten>@sip.blueface.ie State Change CS_INIT -> CS_RING 2007-03-21 17:00:01 [DEBUG] switch_core.c:3429 switch_core_session_run() (sofia/<FreeswitchServerIP>/<exten>@sip.blueface.ie) State RING 2007-03-21 17:00:01 [DEBUG] mod_sofia.c:1351 sofia_on_ring() SOFIA RING 2007-03-21 17:00:01 [DEBUG] switch_ivr.c:2182 originate_on_ring() sofia/<FreeswitchServerIP>/<exten>@sip.blueface.ie State Change CS_RING -> CS_HOLD 2007-03-21 17:00:01 [DEBUG] switch_core.c:3574 switch_core_session_run() (sofia/<FreeswitchServerIP>/<exten>@sip.blueface.ie) State HOLD 2007-03-21 17:00:01 [DEBUG] switch_core.c:3208 switch_core_standard_on_hold() Standard HOLD 2007-03-21 17:00:01 [DEBUG] switch_ivr.c:2836 switch_ivr_originate() Raw Codec Activation Success L16@8000hz 1 channel 20ms 2007-03-21 17:00:14 [DEBUG] mod_dingaling.c:2381 handle_signalling() using Existing session for 3776263765 2007-03-21 17:00:14 [DEBUG] mod_dingaling.c:2744 handle_signalling() hungup DingaLing/1004j 2007-03-21 17:00:14 [DEBUG] mod_dingaling.c:594 terminate_session() Terminate called from line 2745 state=CS_EXECUTE 2007-03-21 17:00:14 [NOTICE] mod_dingaling.c:609 terminate_session() Hangup DingaLing/1004j [CS_EXECUTE] [NORMAL_CLEARING] 2007-03-21 17:00:14 [INFO] switch_channel.c:993 switch_channel_perform_hangup() Kill DingaLing/1004j [KILL] 2007-03-21 17:00:14 [DEBUG] mod_dingaling.c:1182 channel_kill_channel() DingaLing/1004j CHANNEL KILL 2007-03-21 17:00:14 [DEBUG] mod_dingaling.c:2746 handle_signalling() End Call 2007-03-21 17:00:14 [NOTICE] switch_ivr.c:3029 switch_ivr_originate() Hangup sofia/<FreeswitchServerIP>/<exten>@sip.blueface.ie [CS_HOLD] [ORIGINATOR_CANCEL] 2007-03-21 17:00:14 [INFO] switch_channel.c:993 switch_channel_perform_hangup() Kill sofia/<FreeswitchServerIP>/<exten>@ sip.blueface.ie [KILL] 2007-03-21 17:00:14 [DEBUG] switch_ivr.c:3101 switch_ivr_originate() Originate Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL] 2007-03-21 17:00:14 [INFO] mod_bridgecall.c:74 audio_bridge_function() Originate Failed. Cause: ORIGINATOR_CANCEL 2007-03-21 17:00:14 [DEBUG] switch_core.c:3356 switch_core_session_run() (DingaLing/1004j) State HANGUP 2007-03-21 17:00:14 [DEBUG] mod_dingaling.c:1125 channel_on_hangup() NUKE RTP 2007-03-21 17:00:14 [DEBUG] mod_dingaling.c:1137 channel_on_hangup() DingaLing/1004j CHANNEL HANGUP 2007-03-21 17:00:14 [DEBUG] switch_core.c:3048 switch_core_standard_on_hangup() Standard HANGUP DingaLing/1004j, cause: NORMAL_CLEARING 2007-03-21 17:00:14 [DEBUG] switch_core.c:3799 switch_core_session_thread() Session 1 (DingaLing/1004j) Locked, Waiting on external entities 2007-03-21 17:00:14 [INFO] switch_core.c:3804 switch_core_session_thread() Session 1 (DingaLing/1004j) Ended 2007-03-21 17:00:14 [NOTICE] switch_core.c:3680 switch_core_session_destroy() Close Channel DingaLing/1004j 2007-03-21 17:00:14 [DEBUG] switch_core.c:3356 switch_core_session_run() (sofia/<FreeswitchServerIP>/<exten>@sip.blueface.ie) State HANGUP 2007-03-21 17:00:14 [DEBUG] mod_sofia.c:1448 sofia_on_hangup() Channel sofia/<FreeswitchServerIP>/<exten>@sip.blueface.ie hanging up, cause: ORIGINATOR_CANCEL 2007-03-21 17:00:14 [DEBUG] mod_sofia.c:1471 sofia_on_hangup() Sending CANCEL to sofia/<FreeswitchServerIP>/<exten>@sip.blueface.ie 2007-03-21 17:00:14 [DEBUG] switch_core.c:3048 switch_core_standard_on_hangup() Standard HANGUP sofia/<FreeswitchServerIP>/<exten>@sip.blueface.ie, cause: ORIGINATOR_CANCEL 2007-03-21 17:00:14 [DEBUG] switch_core.c:3799 switch_core_session_thread() Session 2 (sofia/<FreeswitchServerIP>/<exten>@sip.blueface.ie) Locked, Waiting on external entities 2007-03-21 17:00:14 [INFO] switch_core.c:3804 switch_core_session_thread() Session 2 (sofia/<FreeswitchServerIP>/<exten>@sip.blueface.ie) Ended 2007-03-21 17:00:14 [NOTICE] switch_core.c:3680 switch_core_session_destroy() Close Channel sofia/<FreeswitchServerIP>/<exten>@sip.blueface.ie

1 0

Re: [Freeswitch-users] Bridging dingaling to sofia
by Anthony Minessale 21 Mar '07

21 Mar '07

A *complete* console trace of the call that doesnt work would make it easier to debug. including the ip (the XXX actually hides an important piece of info as to if the ip is internal or public so at least provide XXLOCAL XXPUBLIC). The easiest way to do this is to execute /usr/local/freeswitch/bin/freeswitch | tee /tmp/console.log then reproduce the problem, stop freeswitch and post console.log. 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: Kieran O'Loughlin <kieran(a)alumni.virginia.edu> To: freeswitch-users(a)lists.freeswitch.org Sent: Wednesday, March 21, 2007 7:24:49 AM Subject: [Freeswitch-users] Bridging dingaling to sofia Hey all, I've been playing around with trying to bridge dingaling to sofia for some time now. I've had some success. 1) I can call to googletalk using a local sip registration 2) I can call to googletalk using a remote sip registration 3) I can call from googletalk to a local sip registration 4) I can call from a local sip registration to a remote sip registration In case my terminology is bad when I say local registration I mean that the soft-phone is registered with freeswitch. When I say remote sip registration I mean that the soft-phone is registered with my standard SIP provider. The problem is if I call from googletalk and attempt to bridge the call to my remote sip provider the call rings, but there is no audio. I've tracked through the console and here is the piece that's different. This piece never shows up if I attempt to bridge the call to a remote sip registration. I copied this from the console when bridging to a local sip registration. 2007-03-21 12:56:13 [DEBUG] mod_sofia.c:4982 event_callback() event [nua_r_invite] status [180][Ringing] session: sofia/XX.XX.XX.XX/kieran 2007-03-21 12:56:13 [DEBUG] mod_sofia.c:4982 event_callback() event [nua_i_state] status [180][Ringing] session: sofia/XX.XX.XX.XX/kieran 2007-03-21 12:56:13 [DEBUG] mod_sofia.c:2991 sip_i_state() Channel sofia/XX.XX.XX.XX/kieran entering state [proceeding] 2007-03-21 12:56:13 [NOTICE] mod_sofia.c:3018 sip_i_state() Ring-Ready sofia/XX.XX.XX.XX/kieran! 2007-03-21 12:56:13 [INFO] switch_core.c:1872 2007-03-21 12:56:13 [INFO] switch_core.c:1872 switch_core_session_receive_message() Kill DingaLing/1004j [BREAK] 2007-03-21 12:56:13 [DEBUG] mod_dingaling.c:1182 channel_kill_channel() DingaLing/1004j CHANNEL KILL 2007-03-21 12:56:15 [DEBUG] mod_sofia.c:2991 sip_i_state() Channel sofia/XX.XX.XX.XX/kieran entering state [ready] 2007-03-21 12:56:15 [DEBUG] mod_sofia.c:2996 sip_i_state() Remote SDP: v=0 o=- 7 2 IN IP4 192.168.2.53 s=CounterPath X-Lite 3.0 c=IN IP4 192.168.2.53 t=0 0 m=audio 37468 RTP/AVP 0 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 2007-03-21 12:56:15 [DEBUG] mod_sofia.c:2595 negotiate_sdp() Codec Compare [PCMU:0]/[PCMU:0] 2007-03-21 12:56:15 [INFO] mod_sofia.c:1578 tech_set_codec() Set Codec sofia/XX.XX.XX.XX/kieran PCMU/8000 20 ms 2007-03-21 12:56:15 [DEBUG] mod_sofia.c:2571 negotiate_sdp() Set 2833 dtmf payload to 101 2007-03-21 12:56:15 [INFO] mod_sofia.c:1635 activate_rtp() RTP [sofia/XX.XX.XX.XX/kieran] XX.XX.XX.XX:16386->192.168.2.53:37468 codec: 0 ms: 20 2007-03-21 12:56:15 [DEBUG] switch_rtp.c:487 switch_rtp_create() Starting timer [soft] 160 bytes per 20000ms 2007-03-21 12:56:15 [NOTICE] mod_sofia.c:3247 sip_i_state() Channel [sofia/XX.XX.XX.XX/kieran] has been answered 2007-03-21 12:56:15 [DEBUG] switch_ivr.c:3074 switch_ivr_originate() Originate Resulted in Success: [sofia/XX.XX.XX.XX/kieran] The weird thing is that if I call the same extension in default_context.xml using a sip phone registered locally it bridges without any problem to the remote sip registration. Can anyone please help with this? I've been battling it for a long time now. I've learned a lot which is good though :-). By the way I just downloaded the last svn version today, so I couldn't be on a more recent version :-) Also if this isn't the best place to address this type of question if anyone could point me in the right direction that would be greatly appreciated. Thanks for any help. Kieran. _______________________________________________ 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 ____________________________________________________________________________________ Sucker-punch spam with award-winning protection. Try the free Yahoo! Mail Beta. http://advision.webevents.yahoo.com/mailbeta/features_spam.html

1 0

Re: [Freeswitch-users] G722 8000 vs 16000
by Anthony Minessale 21 Mar '07

21 Mar '07

yes it's already checked in as of the send date on my last email. 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: "Sluschny, Thomas" <Thomas.Sluschny(a)siemens.com> To: freeswitch-users(a)lists.freeswitch.org Sent: Wednesday, March 21, 2007 7:31:55 AM Subject: Re: [Freeswitch-users] G722 8000 vs 16000 DIV { MARGIN:0px;} Thank you very much, it is a really silly problem ;) So i can take the latest trunk version with your patch ? (is it already checked in?) Von: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] Im Auftrag von Anthony Minessale Gesendet: Mittwoch, 21. März 2007 00:18 An: freeswitch-users(a)lists.freeswitch.org Betreff: Re: [Freeswitch-users] G722 8000 vs 16000 I fixed it the best I could (At least for SIP) The stupid thing here is that there really is an 8khz g722 and the moronic lemmings who would implement the *wideband* version to say 8000 despite the fact that they *know* it is 16000 and are too stupid to realize should be the ones to suffer not us. sigh, I moved actual 8k g722 to payload 109 name G722_8 and put 16k g722 on payload 9 by itself and added a hack to the sdp generator in mod_sofia to display 8000 This makes my stomach turn but I guess we must do whatever it takes to preserve stupidity. This makes me wonder if more of the SIP and SDP RFC's are just typo's which would explain a lot.... 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: "Sluschny, Thomas" <Thomas.Sluschny(a)siemens.com> To: freeswitch-users(a)lists.freeswitch.org Sent: Tuesday, March 20, 2007 11:42:17 AM Subject: [Freeswitch-users] G722 8000 vs 16000 Hi, i want to use G.722 with its wide audio bandwith of 7kHz. But i have found that FreeSwitch resamples always from 16000 to 8000 Samples/s if i connect a "standard device", i. e. a SIP device with a "G722:9/8000" in its SDP. On the other side it is possible to use full bandwith if there is a "G722:9/16000" in SDP. I think this is not correct! FreeSwitch should always use 16000S/s and also use "G722:9/8000" in SDP! Read RFC3551, 4.5.2: ...Even though the actual sampling rate for G.722 audio is 16,000 Hz, the RTP clock rate for the G722 payload format is 8,000 Hz because that value was erroneously assigned in RFC 1890 and must remain unchanged for backward compatibility. ... I tried the 16kHz-implementation with a SIP phone (siemens optipoint 410 with "G722:9/8000") and it seems to work. In the g722.h file is mentioned that the 8k option is used for fast and flexible interworking with narrow band telephony. I think this is a special case (we also support G711 for this). Our default should be 16000S/s and "G722:9/8000". What do u think? Thomas _______________________________________________ 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 Now that's room service! Choose from over 150,000 hotels in 45,000 destinations on Yahoo! Travel to find your fit._______________________________________________ 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 your own web address. Have a HUGE year through Yahoo! Small Business. http://smallbusiness.yahoo.com/domains/?p=BESTDEAL

1 0

Re: [Freeswitch-users] two problems with mod_conference
by Anthony Minessale 21 Mar '07

21 Mar '07

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: Mark D. Anderson <mda(a)discerning.com> To: freeswitch-users(a)lists.freeswitch.org Sent: Tuesday, March 20, 2007 11:55:42 PM Subject: [Freeswitch-users] two problems with mod_conference I'm currently running head (4687) but these problems have been unchanged since I've started playing with mod_conference about 2 months ago. I have tried a cross-product of two ITSPs (vitelity and teliax) either landline caller or cell caller mod_conference "native", or a javascript based on confroom.js I have found some differences in behavior in these 8 cases. 1. Missing 1 second of initial audio (callers don't hear the beginning of the "please enter your pin" message). It doesn't matter whether I move the session.answer() earlier. This happens except for {teliax, landline, *} You can avoid this by adding a sleep action to your dialplan first answer the call from the dialplan and then use sleep to run off 1.5 lead-in seconds to avoid missing some of the audio <action application="answer"/> <action application="sleep" data="1500"/> <action application="conference" data="???"/> 2. Doubling (and tripling) of incoming dtmf digits. It is quite hard to get it to accept a pin, because it keeps thinking I've entered a digit twice, when I haven't. As far as I can see, there is no user configuration to tune dtmf configuration (like min/max length of a tone, or echo cancellation). I also don't know if there is a way to force freeswitch/sofia to use only a particular dtmf system. Note that the problem seems worse when running javascript confroom.js, but it happens both ways. This happens with both ITSPs. I see it except for {*, landline, native mod_conference.c}. Like Mike said, we are only getting the DTMF as rfc 2833 and it has been tested to parse the packets with a duration of as little as 1ms up to several seconds with accuracy. As a receiver of rfc2833 there is really nothing to tune but I know all too well from to much time using rtp that a disagreement in the protocol could cause some issues so I recommend you take Mike up on his offer to investigate it. Any suggestion for causes, fixes, ways to diagnose further, etc., are welcome. -mda _______________________________________________ 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 ____________________________________________________________________________________ Bored stiff? Loosen up... Download and play hundreds of games for free on Yahoo! Games. http://games.yahoo.com/games/front

1 0

GalaxyVoice
by Dave 21 Mar '07

21 Mar '07

Hi guys, I DLed, compiled, and installed FreeSWITCH a couple of days ago. First off, thanks a ton for making it. I got a kick out of dialing 1234 on my SIP phone. Anyway, I won't waste your time on hearing how much I like FreeSWITCH; instead, I'd like to waste it on helping me figure out where I went wrong ;-) I'm trying to connect to my GalaxyVoice (RNK reseller) account. Under Asterisk, it works fine, but I must have screwed something up in my FreeSWITCH config: <registration name='galaxyvoice'>  <param name='register-scheme' value='Basic'/> <param name='register-realm' value='216.143.130.36'/> <param name='register-username' value='R352928'/> <param name='register-password' value='whatever'/>  <param name='register-from' value='sip:R352928:whatever@216.143.130.36'/>  <param name='register-to' value='sip:R352928@freeswitch1.asnetinc.net'/> <param name='register-proxy' value='sip:216.143.130.36:5060'/> <param name='register-frequency' value='105'/> </registration> 216.143.130.36 is RNK's Asterisk server. freeswitch1.asnetinc.net (64.118.94.19) is my new FreeSWITCH installation. I tried Basic authentication since it was complaining: 2007-03-18 13:20:03 [NOTICE] mod_sofia.c:4989 check_oreg() registering galaxyvoice 2007-03-18 13:20:03 [DEBUG] mod_sofia.c:4841 event_callback() event [nua_r_register] status [407][Proxy Authentication Required] session: n/a 2007-03-18 13:20:03 [ERR] mod_sofia.c:4775 sip_r_challenge() No Match for Scheme [Digest] Realm [rnktel.com] FreeSWITCH doesn't seem to be respecting my request for Basic authentication, though. I've tried every realm imagineable, including rnktel.com. They all produce the exact same error. Obviously my password isn't whatever. I've tried a bunch of different combinations of from and to, to no avail. The frequency is 105 on Asterisk, since RNK's server times you out after 120. I just copied the 105 into my FreeSWITCH config. If you need more info, just tell me how to get it, and I'll post it :-) Thanks in advance for any help you might be able to offer, - Dave

3 6

Re: [Freeswitch-users] G722 8000 vs 16000
by Anthony Minessale 21 Mar '07

21 Mar '07

I fixed it the best I could (At least for SIP) The stupid thing here is that there really is an 8khz g722 and the moronic lemmings who would implement the *wideband* version to say 8000 despite the fact that they *know* it is 16000 and are too stupid to realize should be the ones to suffer not us. sigh, I moved actual 8k g722 to payload 109 name G722_8 and put 16k g722 on payload 9 by itself and added a hack to the sdp generator in mod_sofia to display 8000 This makes my stomach turn but I guess we must do whatever it takes to preserve stupidity. This makes me wonder if more of the SIP and SDP RFC's are just typo's which would explain a lot.... 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: "Sluschny, Thomas" <Thomas.Sluschny(a)siemens.com> To: freeswitch-users(a)lists.freeswitch.org Sent: Tuesday, March 20, 2007 11:42:17 AM Subject: [Freeswitch-users] G722 8000 vs 16000 Hi, i want to use G.722 with its wide audio bandwith of 7kHz. But i have found that FreeSwitch resamples always from 16000 to 8000 Samples/s if i connect a "standard device", i. e. a SIP device with a "G722:9/8000" in its SDP. On the other side it is possible to use full bandwith if there is a "G722:9/16000" in SDP. I think this is not correct! FreeSwitch should always use 16000S/s and also use "G722:9/8000" in SDP! Read RFC3551, 4.5.2: ...Even though the actual sampling rate for G.722 audio is 16,000 Hz, the RTP clock rate for the G722 payload format is 8,000 Hz because that value was erroneously assigned in RFC 1890 and must remain unchanged for backward compatibility. ... I tried the 16kHz-implementation with a SIP phone (siemens optipoint 410 with "G722:9/8000") and it seems to work. In the g722.h file is mentioned that the 8k option is used for fast and flexible interworking with narrow band telephony. I think this is a special case (we also support G711 for this). Our default should be 16000S/s and "G722:9/8000". What do u think? Thomas _______________________________________________ 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

3 2

Bridging dingaling to sofia
by Kieran O'Loughlin 21 Mar '07

21 Mar '07

Hey all, I've been playing around with trying to bridge dingaling to sofia for some time now. I've had some success. 1) I can call to googletalk using a local sip registration 2) I can call to googletalk using a remote sip registration 3) I can call from googletalk to a local sip registration 4) I can call from a local sip registration to a remote sip registration In case my terminology is bad when I say local registration I mean that the soft-phone is registered with freeswitch. When I say remote sip registration I mean that the soft-phone is registered with my standard SIP provider. The problem is if I call from googletalk and attempt to bridge the call to my remote sip provider the call rings, but there is no audio. I've tracked through the console and here is the piece that's different. This piece never shows up if I attempt to bridge the call to a remote sip registration. I copied this from the console when bridging to a local sip registration. 2007-03-21 12:56:13 [DEBUG] mod_sofia.c:4982 event_callback() event [nua_r_invite] status [180][Ringing] session: sofia/XX.XX.XX.XX/kieran 2007-03-21 12:56:13 [DEBUG] mod_sofia.c:4982 event_callback() event [nua_i_state] status [180][Ringing] session: sofia/XX.XX.XX.XX/kieran 2007-03-21 12:56:13 [DEBUG] mod_sofia.c:2991 sip_i_state() Channel sofia/XX.XX.XX.XX/kieran entering state [proceeding] 2007-03-21 12:56:13 [NOTICE] mod_sofia.c:3018 sip_i_state() Ring-Ready sofia/XX.XX.XX.XX/kieran! 2007-03-21 12:56:13 [INFO] switch_core.c:1872 2007-03-21 12:56:13 [INFO] switch_core.c:1872 switch_core_session_receive_message() Kill DingaLing/1004j [BREAK] 2007-03-21 12:56:13 [DEBUG] mod_dingaling.c:1182 channel_kill_channel() DingaLing/1004j CHANNEL KILL 2007-03-21 12:56:15 [DEBUG] mod_sofia.c:2991 sip_i_state() Channel sofia/XX.XX.XX.XX/kieran entering state [ready] 2007-03-21 12:56:15 [DEBUG] mod_sofia.c:2996 sip_i_state() Remote SDP: v=0 o=- 7 2 IN IP4 192.168.2.53 s=CounterPath X-Lite 3.0 c=IN IP4 192.168.2.53 t=0 0 m=audio 37468 RTP/AVP 0 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 2007-03-21 12:56:15 [DEBUG] mod_sofia.c:2595 negotiate_sdp() Codec Compare [PCMU:0]/[PCMU:0] 2007-03-21 12:56:15 [INFO] mod_sofia.c:1578 tech_set_codec() Set Codec sofia/XX.XX.XX.XX/kieran PCMU/8000 20 ms 2007-03-21 12:56:15 [DEBUG] mod_sofia.c:2571 negotiate_sdp() Set 2833 dtmf payload to 101 2007-03-21 12:56:15 [INFO] mod_sofia.c:1635 activate_rtp() RTP [sofia/XX.XX.XX.XX/kieran] XX.XX.XX.XX:16386->192.168.2.53:37468 codec: 0 ms: 20 2007-03-21 12:56:15 [DEBUG] switch_rtp.c:487 switch_rtp_create() Starting timer [soft] 160 bytes per 20000ms 2007-03-21 12:56:15 [NOTICE] mod_sofia.c:3247 sip_i_state() Channel [sofia/XX.XX.XX.XX/kieran] has been answered 2007-03-21 12:56:15 [DEBUG] switch_ivr.c:3074 switch_ivr_originate() Originate Resulted in Success: [sofia/XX.XX.XX.XX/kieran] The weird thing is that if I call the same extension in default_context.xml using a sip phone registered locally it bridges without any problem to the remote sip registration. Can anyone please help with this? I've been battling it for a long time now. I've learned a lot which is good though :-). By the way I just downloaded the last svn version today, so I couldn't be on a more recent version :-) Also if this isn't the best place to address this type of question if anyone could point me in the right direction that would be greatly appreciated. Thanks for any help. Kieran.

1 0

two problems with mod_conference
by Mark D. Anderson 21 Mar '07

21 Mar '07

I'm currently running head (4687) but these problems have been unchanged since I've started playing with mod_conference about 2 months ago. I have tried a cross-product of two ITSPs (vitelity and teliax) either landline caller or cell caller mod_conference "native", or a javascript based on confroom.js I have found some differences in behavior in these 8 cases. 1. Missing 1 second of initial audio (callers don't hear the beginning of the "please enter your pin" message). It doesn't matter whether I move the session.answer() earlier. This happens except for {teliax, landline, *} 2. Doubling (and tripling) of incoming dtmf digits. It is quite hard to get it to accept a pin, because it keeps thinking I've entered a digit twice, when I haven't. As far as I can see, there is no user configuration to tune dtmf configuration (like min/max length of a tone, or echo cancellation). I also don't know if there is a way to force freeswitch/sofia to use only a particular dtmf system. Note that the problem seems worse when running javascript confroom.js, but it happens both ways. This happens with both ITSPs. I see it except for {*, landline, native mod_conference.c}. Any suggestion for causes, fixes, ways to diagnose further, etc., are welcome. -mda

2 1