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[commit] r10995 - freeswitch/trunk/debian
by FreeSWITCH SVN 29 Dec '08

29 Dec '08

Author: mcollins Date: Mon Dec 29 18:32:48 2008 New Revision: 10995 Log: Update debian/changelog with 1.0.2 changes Modified: freeswitch/trunk/debian/changelog Modified: freeswitch/trunk/debian/changelog ============================================================================== --- freeswitch/trunk/debian/changelog (original) +++ freeswitch/trunk/debian/changelog Mon Dec 29 18:32:48 2008 @@ -1,3 +1,296 @@ +freeswitch (1.0.2-1) unstable; urgency=low + + * all: don't add module interfaces before returning from error conditions in module load functions (MDXMLINT-36) + * all: fixed multiple memory leaks + * all: improved module unloading/reloading support + * build: add support for --switchconfdir (FSBUILD-84) + * build: fixed netbsd build + * build: make freeswitch stop graceflly with /etc/init.d/freeswitch stop on debian add working dir to start-stop-dir so freeswitch dumps core in workdir + * build: multiple packaging fixes + * build: user freeswitch not added to audio group on deb install (FSBUILD-95) + * Configuration: many updates to default configuration + * core: Add ability to choose uuid from originate string, originate_uuid var (use at your own risk) + * core: add bridge_generate_comfort_noise option for bridge to generate comfort noise to the A leg when there is no audio on the B leg + * core: add chan vars to param event during hangup hook + * core: add exec directive to xml preprocessor (not available on windows) + * core: add force_transfer_dialplan and force_transfer_context variables + * core: add hashing to event header lookup + * core: add hits to tone_detect + * core: add last_dtmf_duration variable + * core: add msleep function to swigged languages + * core: add park_after_bridge variable + * core: add per leg timeouts and specific cause codes in reject_on_single_fail + * core: add runtime selection of the module dir (FSCORE-198) + * core: add scheduler support for heartbeat + * core: add session heartbeat feature + * core: add session.destroy psuedo method to sort of destroy a session at least for the sake of FS + * core: add session.unsetInputCallback + * core: add strftime format string validation for user supplied values + * core: add vars param to switch_ivr_originate for recursion (MODAPP-175) + * core: added a "group" concept to the user directory + * core: added ability to do dns lookup to find ip with host: like stun: (FSCORE-219) + * core: added better locking for codec changes during a call + * core: added current_application and current_application_data variables + * core: added error/ magic endpoint so modules can return error causes in situations like sofia_contact + * core: added read_result channel variable + * core: added support for "F" to indicate flash in dtmf (FSCORE-213) + * core: allow calls to be stolen from originate + * core: allow you to get the privacy bits in the caller_profile + * core: change dso code to load symbols local + * core: changes core flags to be array based so we have more + * core: eavesdrop causes the people being eavesdropped on to not hear ach other (MODAPP-140) + * core: expose time table to variable interface via caller field lookup + * core: fix 100% cpu when sending parked call to moh (FSCORE-234) + * core: fix bridge app to make sure both channels are ready for media when one is only in ringing state + * core: fix buffer overflow (FSCORE-188) + * core: fix conference dial by allowing multiple braces in originate, fix bad pointer op (FSCORE-208) + * core: fix double detection of DTMF in IVR (FSCORE-221) + * core: fix hangup_after_bridge is false on a bridge started with the intercept app + * core: fix issue where pid file is accidentally truncated + * core: fix ivr timeout (FSCORE-181) + * core: fix memory leak in alias tab completion code + * core: fix min digits in read app + * core: fix out-of-bounds pointer in variable expansion (FSCORE-171) + * core: fix segfault in media bugs when in bypass media (FSCORE-193) + * core: fix segfault on gtalk to sip calls (FSCORE-212) + * core: fix segfault on reloadxml (FSCORE-176) + * core: fix segfault on trasfering eavesdopping party (FSCORE-210) + * core: fix segfault using switch_system function (FSCORE-196) + * core: fix session.bridge + * core: fix setting effective_caller_id_name / effective_caller_id_number on bridge dialstring (MODAPP-122) + * core: fix stream_raw_write (MODAPP-145) + * core: fix using resampling on ringback file + * core: fixed performance bottleneck in sqlite db's + * core: fixed race in reloadxml + * core: increment app before execute in case it returns to execute it will go to the next item in the list and not the same + * core: ivr menu max_failures and max_timeouts now default to 3 if not specified or invalid (less than 1) values are specified (FSCORE-244) + * core: ivr_menu max-timeouts option, result in ivr_menu_status var (FSCORE-183) + * core: let b legs use park_after_bridge too + * core: make events less verbose unless verbose_events is set + * core: parse private events during originate + * core: pass pdd data to a leg on oubound calls using bridge + * core: prevent crash in crazy situation with xml interface lookup failures (FSCORE-169) + * core: reduce cpu requirement for generated comfort noise + * core: remove interface names from core db on unload + * core: reworked timing resulting in significant performance increase and better rtp timing + * core: rewrite switch_play_and_get_digits (MODAPP-166) + * core: session.recordFile never terminates (MODLANG-79) + * core: session.transfer make dialplan and context optional + * core: set_user app now sets domain vars as well as user vars + * core: tone_detect not triggering app after tone detection (MODAPP-182) + * core: unprivileged user setting bigger stack for switch_system thread failure (FSCORE-197) + * core: user_exists returns false when fetching a user from XML Curl or other xml interfaces + * libesl: added c event socket library and fs_cli + * libsndfile: fix autoconf 2.62 support (LBSNDF-5) + * mod commands: add "all" modifier to "break" command + * mod_celt: added new module + * mod_commands: Add support for more than 2 variables to uuid_setvar_multi (MODAPP-171) + * mod_commands: Add support for passing the cause of hangup to the uuid_kill command (FSCORE-217) + * mod_commands: add attr lookup to user_data + * mod_commands: add domain_exists fsapi command + * mod_commands: add eval fsapi command + * mod_commands: add flush_dtmf app and uuid_flush_dtmf api command + * mod_commands: add fsctl send_sighup, fsctl shutdown asap, unsched_api commands + * mod_commands: add fsctl shutdown [elegant|restart|cancel] + * mod_commands: add new syntax to uuid_setvar to allow you to unset a var. <uuid> <var> [value] (MODAPP-167) + * mod_commands: add reload fsapi command to reload a module + * mod_commands: add system fsapi and application (MODAPP-138) + * mod_commands: added hupall fsapi command + * mod_commands: added strftime_tz api command + * mod_commands: break all now stops broadcast too + * mod_commands: fix api command sent through sched_api was getting the last char lopped off + * mod_commands: fix race on transfer with -both + * mod_commands: fix system dialplan app problems (MODAPP-86) + * mod_commands: only send content-type on status when it really is http. + * mod_conference: add fsapi to stop async playback too + * mod_conference: add video caps to mod_conference with video follow audio + * mod_conference: better sound prefix handling when using say: and allow say: on kick sounds. + * mod_conference: fix race in record + * mod_conference: fix runaway thread when floor holder has no video and other people do have video + * mod_conference: fix seg when kicking many members quickly (MODAPP-129) + * mod_conference: fix segfault on invalid chat event + * mod_conference: perpetual sound does not auto-mute, you can do that yourself if you want it + * mod_dialplan_xml: add Hunt- vars in dialplan lookup after transfer + * mod_dialplan_xml: fail call on extensions with nested conditions + * mod_dingaling: (LBDING-7) fix segfault on os x + * mod_dingaling: end call on ice timeout + * mod_dingaling: fix presence on jabber to be less protocol ambiguous + * mod_dingaling: fix segfault (LBDING-10) + * mod_dingaling: update to support latest client from google + * mod_dptools: add a mechanism to tell if a file played from sendmsg over event socket + * mod_dptools: add playback_terminator support to phrase and say app + * mod_dptools: add playback_terminator_used variable (MODAPP-132) + * mod_dptools: add presence application + * mod_dptools: fix originate api not parsing users properly (FSCORE-246) + * mod_dptools: fix record and record_session to create directory if it does not exist (FSCORE-250) + * mod_dptools: fixed limit and + parsing bug in record_session app (MODAPP-148) + * mod_dptools: remove_bugs added to remove all media bugs on a session + * mod_erlang_event: add new module + * mod_event_socket: missing : after Content-Length in event socket (MODEVENT-33) + * mod_event_socket: add event socket listener filters + * mod_event_socket: add stateful listener fsapi commands for ajax-y type event interface over http + * mod_event_socket: fix arg parsing errors (MODEVENT-34) + * mod_event_socket: fix shutdown segfault race (MODEVENT-32) + * mod_event_socket: inbound connection to event_socket can now take over an existing channel with 'myevents <uuid>' to take on the behaviour of an outbound socket + * mod_event_socket: let any channel get messages + * mod_event_socket: make event socket wait for hangup on outbound mode and send disconnect message + * mod_expr: fix endless loop + * mod_fax: new module + * mod_fifo: add fifo_consumer_wrapup_time var (MODAPP-117) + * mod_fifo: added callback agents + * mod_fifo: honor keyword silence (MODAPP-118) + * mod_flite: added windows build + * mod_fsv: fix in a windows enviroment opening the record file in text mode. (MODAPP-169) + * mod_http: added new module + * mod_java: updated to new module api to support read/write locks on interface + * mod_limit: accept dialplan context for transfer (MODAPP-161) + * mod_limit: added hashtable based limit functions + * mod_limit: prevent empty error log message (MODAPP-134) + * mod_local_stream: add start_local_stream and stop_local_stream fsapi commands to start/stop dynamically (MODFORM-13) + * mod_local_stream: fix leak and improve error checking + * mod_local_stream: fix seg when no timer name specified in config file. (MODFORM-16) + * mod_loopback: add new module + * mod_lua: add local scripts directory support (MODLANG-86) + * mod_lua: don't eval blank string + * mod_lua: fix originate + * mod_lua: fix segfault (MODLANG-77) + * mod_lua: update to lua 5.1.4 (MODLANG-87) + * mod_lumenvox: removed + * mod_managed: new module replaces mod_mono now supports native .net runtime on windows as well + * mod_opal: added to trunk (still very beta) + * mod_perl: fix segfault (MODLANG-77) + * mod_pocketsphinx: fix rpm build + * mod_portaudio: fix cpu race on inbound call to pa when no ring file is set + * mod_radius_cdr: dictionary update for cause code changes (MODEVENT-27) + * mod_radius_cdr: fix unload (MODEVENT-29) + * mod_shout: add stereo recording broadcast support + * mod_shout: added windows build + * mod_shout: fix segfault when recording mp3's (MODFORM-12) + * mod_shout: improved stability of mp3 decoding + * mod_siren: added new module + * mod_sndfile added support to record 16bit for the various rates including 48kHz + * mod_sofia: Add filter to "sofia status profile XXX" (MODENDP-138) + * mod_sofia: Add force-register-db-domain which works in conjunction with force-register-domain. + * mod_sofia: Add optional <variables> and <params> tag to <gateway> tag. + * mod_sofia: Challenge the right realm when to_host is outside the users domain. (MODENDP-136) + * mod_sofia: Improve notify messages through a proxy (MODENDP-147) + * mod_sofia: MWI for multiple domains (MODAPP-126) + * mod_sofia: Move "a=sendrecv" from session to media section of SDP (MODENDP-148) + * mod_sofia: add 200 OK re-invite without sdp + * mod_sofia: add custom sofia::gateway_state event (MODENDP-112) + * mod_sofia: add fire events for the refer SIP NOTIFY event package (MODENDP-152) + * mod_sofia: add more params for xml_curl directory lookup + * mod_sofia: add new auto vals for challenge-realm param <param name="challenge-realm" value="auto_from|auto_to|<hardcoded_val>"/> + * mod_sofia: add option to turn of auto_restart of sofia profiles on ip change + * mod_sofia: add params to use sip callid as uuid on inbound calls and uuid as sip callid on outbound calls + * mod_sofia: add parsing of Privacy header for privacy info (MODENDP-133) + * mod_sofia: add proto_specific_hangup_cause to both legs + * mod_sofia: add proxy 3pcc mode + * mod_sofia: add redirect variable to channel as well as partner channe (MODENDP-135) + * mod_sofia: add sip-forbid-register to user params to refuse to let a certian user register + * mod_sofia: add sip: into register-proxy when it's not specified + * mod_sofia: add sip_history_info var for inbound invites. + * mod_sofia: add sip_via_protocol variable + * mod_sofia: add sofia xmlstatus (MODENDP-156) + * mod_sofia: add support for params other than Replaces in Refer-To (MODENDP-143) + * mod_sofia: add support for profiles sharing databases so that you can have a domain that uses multiple profiles for split dns type setups + * mod_sofia: add support for refer transfer involving multiple machines + * mod_sofia: add support to send a notify in the invite dialog by specifying the uuid of the call. (SFSIP-92) + * mod_sofia: add suppress_from_cidname var to not have display name in from header (MODENDP-153) + * mod_sofia: added sip_hangup_disposition variable + * mod_sofia: allow send_message and notify events to send a message/notify without a body if needed. + * mod_sofia: append -1 .. -N postfix after any X-headers as vars that have the same name + * mod_sofia: cache auth_gateway_name in sofia for challenged bye + * mod_sofia: cancel proxy or no-media mode if you purposely answer or pre_answer + * mod_sofia: correct result code mapping for Unallocated Number (MODENDP-124) + * mod_sofia: disable 100rel by default + * mod_sofia: don't accept crypto in the RTP/AVP (MODENDP-126) + * mod_sofia: don't put CN in sdp answer if it was not in the offer. + * mod_sofia: fix Incorrect IP address shows up in SDP "o" field when multiple external IPs available and FS not bound to first (MODENDP-132) + * mod_sofia: fix Wrong RTP media port destination after reinvite/UNHOLD (SFSIP-82) + * mod_sofia: fix bug on linksys where they lie about the ptime and handle linksys transfer problem + * mod_sofia: fix chat (send an IM) assumes that the user's profile is the same as their domain, which isn't necessarily so (SFSIP-83) + * mod_sofia: fix dtmf handling of broken info dtmf endpoints + * mod_sofia: fix eyebeam presence to be RFC compliant (MODENDP-144) + * mod_sofia: fix ip change detection when in proxy mode + * mod_sofia: fix register_proxy ignoring the paramaters (MODENDP-121) + * mod_sofia: fix remote session refresh triggers request glare (MODENDP-131) + * mod_sofia: fix rtp auto adjust running when it should not + * mod_sofia: fix rtp sent to wrong port after some re-INVITE scenarios (MODENDP-141) + * mod_sofia: fix sending of cn packets across bridge when we shouldn't + * mod_sofia: fix sqlite issue with select of the sip contact + * mod_sofia: fixed segfault on invalid presence payload + * mod_sofia: gateway ping needs to look for 501 (SFSIP-78) + * mod_sofia: handle multi contact register responses and register timeout better + * mod_sofia: improve gateway resilience + * mod_sofia: log ip and port you get reply to invite from + * mod_sofia: make multiple-registations=true use the contact method and call-id option to do it the old way + * mod_sofia: make proxy mode pull the port from m=image as well + * mod_sofia: make register-proxy preserve the url composed from proxy but target the packets to desired address (MODENDP-121) + * mod_sofia: many fixes for sonus rtp issues silence_when_idle=400 chanvar to send generated silence duing sleeps etc + * mod_sofia: many fixes in presence handling + * mod_sofia: passthrough t.38 fixes + * mod_sofia: pick ipv4 or ipv6 based on sipip instead of having mixed in sdp + * mod_sofia: send NOTIFY on TCP/UDP depending on the SUBSCRIBE (SFSIP-104) + * mod_sofia: setting profile option multiple-registrations=contact key multi reg off the contact string + * mod_sofia: wait for a reply on refer + * mod_soundtouch: fixes and improvements, many options changed (MODAPP-149) + * mod_soundtouch: updated to new module api + * mod_spidermonkey: Segmentation fault in check_hangup_hook at mod_spidermonkey.c:1589 (MODLANG-74) + * mod_spidermonkey: fix bug in apiExecute + * mod_spidermonkey: fix memory pool handling and leaks + * mod_spidermonkey: limit recursion busting through the stack (FSCORE-202) + * mod_spidermonkey: make session.getVariable return blank string not the word false + * mod_spidermonkey_curl: add optional content-type arg + * mod_spidermonkey_odbc: fix numRows and add numCols + * mod_spidermonkey_odbc: fix segfault (MODLANG-75) + * mod_stress: new module for voice stress analysis + * mod_syslog: don't log blank lines (FSCORE-163) + * mod_tone_stream: let silence_stream://0 indicate perpetual silence + * mod_vmd: add new module to detect voicemail "beep" + * mod_voicemail: Add vm_alternate_greet_id param to directory entry (MODAPP-174) + * mod_voicemail: Patch to add voicemail preference controlling date announcement new param 'play-date-announcement' to values 'first' 'last' or 'never' defaults to first to retain previous behavior (MODAPP-121) + * mod_voicemail: Update mwi light after delete vm via web. (MODAPP-124) + * mod_voicemail: add ability to get to options without listening to every saved message (MODAPP-115) + * mod_voicemail: add ability to skip greeting when leaving a voicemail. (MODAPP-181) + * mod_voicemail: add change-pass-key config file option + * mod_voicemail: add forwarding support + * mod_voicemail: add local dtmf driven alternat vm pass + * mod_voicemail: add proper notification of a vm message being too short + * mod_voicemail: add support for auth via a1-hash + * mod_voicemail: add the "storage-dir" parameter to be set on a per-user basis (MODAPP-133) + * mod_voicemail: add voicemail_greeting_path variable + * mod_voicemail: added voicemail_alternate_greet_id variable + * mod_voicemail: allow changing of password from voicemail to update user directory if using non-static config (MODAPP-156) + * mod_voicemail: created email date (int overflow) (MODAPP-125) + * mod_voicemail: don't try to deliver vm when no file was recorded. (MODAPP-133) + * mod_voicemail: fix MWI with xml_curl used for directory (MODAPP-176) + * mod_voicemail: fix Voicemail messages occasionally lost / stranded (MODAPP-178) + * mod_voicemail: fix invalid event after message deleted (MODAPP-170) + * mod_voicemail: fix mwi for phones with multiple registrations problem (MODAPP-153) + * mod_voicemail: fix voicemail segfault on incorrect password (FSCORE-187) + * mod_voicemail: fix voicemail_inject error handling (MODAPP-133) + * mod_voicemail: fix voicemail_inject usage api call + * mod_voicemail: improve error checking (MODAPP-142) + * mod_voicemail: localize notification emails (MODAPP-139) + * mod_voicemail: make more multi-domain friendly (MODAPP-162) + * mod_voicemail: make playback created file macros optional (MODAPP-150) + * mod_voicemail: recognize operator key in more places (MODAPP-159) + * mod_voicemail: web interface displays incorrect created / last heard dates (MODAPP-123) + * mod_wanpipe: removed + * mod_xml_cdr: add https support + * mod_xml_cdr: add optional a-leg prefix to xml cdr filenames (MDXMLINT-39) + * mod_xml_cdr: add support for fallback webserver for cdr posting (FSCORE-238) + * mod_xml_curl: Allow specification of HTTP method, and dynamic expansion of variables in URI. (MDXMLINT-41) + * mod_xml_curl: added redirect following (max 10) + * mod_xml_ldap: almost a complete rewrite of this module + * mod_xml_rpc: allow setting of global realm without a global user (MDXMLINT-45) + * mod_xml_rpc: fix multiple segfaults + * mod_xml_rpc: fix segfault on originate via http + * sofia-sip: updated to 1.12.10 (plus a few patches) + + -- Mike Jerris <mike(a)jerris.com> Mon, 29 Dec 2008 14:46:00 -0500 + freeswitch (1.0.1-1) unstable; urgency=low * FIX: prevent intercept race condition that can also be solved with continue_on_fail=originator_cancel @@ -295,7 +588,7 @@ Fixed mod_iax race conditions Fixed ptime negotiation issues when re-packetizing Added ip based acl lists - + * -- Michael Jerris <mike(a)jerris.com> Wed, 9 Apr 2008 12:58:22 -0400 freeswitch (1.0~rc1-1) unstable; urgency=low

1 0

[commit] r10994 - freeswitch/trunk/src
by FreeSWITCH SVN 29 Dec '08

29 Dec '08

Author: anthm Date: Mon Dec 29 17:41:03 2008 New Revision: 10994 Log: fix FSCORE-261 Modified: freeswitch/trunk/src/switch_core.c Modified: freeswitch/trunk/src/switch_core.c ============================================================================== --- freeswitch/trunk/src/switch_core.c (original) +++ freeswitch/trunk/src/switch_core.c Mon Dec 29 17:41:03 2008 @@ -168,7 +168,7 @@ SWITCH_DECLARE(void) switch_core_remove_state_handler(const switch_state_handler_table_t *state_handler) { - int index, total = 0; + int index, tmp_index = 0; const switch_state_handler_table_t *tmp[SWITCH_MAX_STATE_HANDLERS+1] = { 0 }; switch_mutex_lock(runtime.global_mutex); @@ -179,13 +179,12 @@ if (cur == state_handler) { continue; } - tmp[index] = runtime.state_handlers[index]; - total++; + tmp[tmp_index++] = cur; } runtime.state_handler_index = 0; - for (index = 0; index < total; index++) { + for (index = 0; index < tmp_index; index++) { runtime.state_handlers[runtime.state_handler_index++] = tmp[index]; } switch_mutex_unlock(runtime.global_mutex);

1 0

[commit] r10993 - in freeswitch/trunk/src: . include mod/applications/mod_voicemail
by FreeSWITCH SVN 29 Dec '08

29 Dec '08

Author: anthm Date: Mon Dec 29 17:26:10 2008 New Revision: 10993 Log: fix voicemail to be in line with groups Modified: freeswitch/trunk/src/include/switch_xml.h freeswitch/trunk/src/mod/applications/mod_voicemail/mod_voicemail.c freeswitch/trunk/src/switch_xml.c Modified: freeswitch/trunk/src/include/switch_xml.h ============================================================================== --- freeswitch/trunk/src/include/switch_xml.h (original) +++ freeswitch/trunk/src/include/switch_xml.h Mon Dec 29 17:26:10 2008 @@ -350,6 +350,8 @@ switch_xml_t *root, switch_xml_t *domain, switch_xml_t *user, switch_xml_t *ingroup, switch_event_t *params); +SWITCH_DECLARE(switch_status_t) switch_xml_locate_user_in_domain(const char *user_name, switch_xml_t domain, switch_xml_t *user, switch_xml_t *ingroup); + ///\brief open a config in the core registry ///\param file_path the name of the config section e.g. modules.conf ///\param node a pointer to point to the node if it is found Modified: freeswitch/trunk/src/mod/applications/mod_voicemail/mod_voicemail.c ============================================================================== --- freeswitch/trunk/src/mod/applications/mod_voicemail/mod_voicemail.c (original) +++ freeswitch/trunk/src/mod/applications/mod_voicemail/mod_voicemail.c Mon Dec 29 17:26:10 2008 @@ -2261,7 +2261,9 @@ } /*switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "Send mail is %d, var is %s\n", send_mail, var);*/ } - switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "Send mail is %d and that's my final answer\n", send_mail); + + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "Deliver VM to %s@%s\n", myid, domain_name); + if (!switch_strlen_zero(vm_storage_dir)) { dir_path = switch_mprintf("%s%s%s", vm_storage_dir, SWITCH_PATH_SEPARATOR, myid); } else if (!switch_strlen_zero(profile->storage_dir)) { @@ -2507,7 +2509,7 @@ vm_profile_t *profile; char *dup = NULL, *user = NULL, *domain = NULL; switch_status_t status = SWITCH_STATUS_SUCCESS; - int istag = 0, isall = 0; + int isgroup = 0, isall = 0; int argc = 0; char *argv[4]; char *box, *path, *cid_num, *cid_name; @@ -2540,9 +2542,9 @@ domain = user; } - if (switch_stristr("tag=", user)) { - user += 4; - istag++; + if (switch_stristr("group=", user)) { + user += 6; + isgroup++; } else if (user == domain) { isall++; } @@ -2567,6 +2569,17 @@ switch_event_create(&my_params, SWITCH_EVENT_REQUEST_PARAMS); switch_assert(my_params); + + if (isgroup) { + switch_event_add_header_string(my_params, SWITCH_STACK_BOTTOM, "group", user); + } else { + if (isall) { + switch_event_add_header_string(my_params, SWITCH_STACK_BOTTOM, "user", "_all_"); + } else { + switch_event_add_header_string(my_params, SWITCH_STACK_BOTTOM, "user", user); + } + } + switch_event_add_header_string(my_params, SWITCH_STACK_BOTTOM, "domain", domain); switch_event_add_header_string(my_params, SWITCH_STACK_BOTTOM, "purpose", "publish-vm"); @@ -2581,24 +2594,66 @@ switch_core_new_memory_pool(&pool); - if (!isall && !istag) { - if ((ut = switch_xml_find_child(x_domain, "user", "id", user))) { + + if (isgroup) { + switch_xml_t group = NULL, groups = NULL, users = NULL; + if ((groups = switch_xml_child(x_domain, "groups"))) { + if ((group = switch_xml_find_child_multi(groups, "group", "name", user, NULL))) { + if ((users = switch_xml_child(group, "users"))) { + for (ut = switch_xml_child(users, "user"); ut; ut = ut->next) { + const char *type = switch_xml_attr_soft(ut, "type"); + + if (!strcasecmp(type, "pointer")) { + const char *uname = switch_xml_attr_soft(ut, "id"); + switch_xml_t ux; + + if (switch_xml_locate_user_in_domain(uname, x_domain, &ux, NULL) == SWITCH_STATUS_SUCCESS) { + switch_event_create(&my_params, SWITCH_EVENT_REQUEST_PARAMS); + status = deliver_vm(profile, ux, domain, path, 0, "B", my_params, pool, cid_name, cid_num, SWITCH_TRUE); + switch_event_destroy(&my_params); + } + continue; + } + + switch_event_create(&my_params, SWITCH_EVENT_REQUEST_PARAMS); + status = deliver_vm(profile, ut, domain, path, 0, "B", my_params, pool, cid_name, cid_num, SWITCH_TRUE); + switch_event_destroy(&my_params); + } + } + } else { + switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_WARNING, "Cannot locate group %s\n", user); + } + } + } else if (isall) { + switch_xml_t group = NULL, groups = NULL, users = NULL; + if ((groups = switch_xml_child(x_domain, "groups"))) { + for (group = switch_xml_child(groups, "group"); group; group = group->next) { + if ((users = switch_xml_child(group, "users"))) { + for (ut = switch_xml_child(users, "user"); ut; ut = ut->next) { + const char *type = switch_xml_attr_soft(ut, "type"); + + if (!strcasecmp(type, "pointer")) { + continue; + } + + switch_event_create(&my_params, SWITCH_EVENT_REQUEST_PARAMS); + status = deliver_vm(profile, ut, domain, path, 0, "B", my_params, pool, cid_name, cid_num, SWITCH_TRUE); + switch_event_destroy(&my_params); + } + } + } + } + + } else { + switch_xml_t x_group = NULL; + + if ((status = switch_xml_locate_user_in_domain(user, x_domain, &ut, &x_group)) == SWITCH_STATUS_SUCCESS) { switch_event_create(&my_params, SWITCH_EVENT_REQUEST_PARAMS); status = deliver_vm(profile, ut, domain, path, 0, "B", my_params, pool, cid_name, cid_num, SWITCH_TRUE); switch_event_destroy(&my_params); } else { status = SWITCH_STATUS_FALSE; } - } else { - for (ut = switch_xml_child(x_domain, "user"); ut; ut = ut->next) { - const char *tag; - - if (isall || (istag && (tag=switch_xml_attr(ut, "vm-tag")) && !strcasecmp(tag, user))) { - switch_event_create(&my_params, SWITCH_EVENT_REQUEST_PARAMS); - status = deliver_vm(profile, ut, domain, path, 0, "B", my_params, pool, cid_name, cid_num, SWITCH_TRUE); - switch_event_destroy(&my_params); - } - } } switch_core_destroy_memory_pool(&pool); @@ -3562,7 +3617,7 @@ } } -#define VM_INJECT_USAGE "[tag=]<box> <sound_file> [<cid_num>] [<cid_name>]" +#define VM_INJECT_USAGE "[group=]<box> <sound_file> [<cid_num>] [<cid_name>]" SWITCH_STANDARD_API(voicemail_inject_api_function) { if (voicemail_inject(cmd) == SWITCH_STATUS_SUCCESS) { Modified: freeswitch/trunk/src/switch_xml.c ============================================================================== --- freeswitch/trunk/src/switch_xml.c (original) +++ freeswitch/trunk/src/switch_xml.c Mon Dec 29 17:26:10 2008 @@ -1634,7 +1634,6 @@ return status; } - static switch_status_t find_user_in_tag(switch_xml_t tag, const char *ip, const char *user_name, const char *key, switch_event_t *params, switch_xml_t *user) { const char *type = "!pointer"; @@ -1670,6 +1669,26 @@ } +SWITCH_DECLARE(switch_status_t) switch_xml_locate_user_in_domain(const char *user_name, switch_xml_t domain, switch_xml_t *user, switch_xml_t *ingroup) +{ + switch_xml_t group = NULL, groups = NULL, users = NULL; + switch_status_t status = SWITCH_STATUS_FALSE; + + if ((groups = switch_xml_child(domain, "groups"))) { + for (group = switch_xml_child(groups, "group"); group; group = group->next) { + if ((users = switch_xml_child(group, "users"))) { + if ((status = find_user_in_tag(users, NULL, user_name, "id", NULL, user)) == SWITCH_STATUS_SUCCESS) { + if (ingroup) { + *ingroup = group; + } + break; + } + } + } + } + + return status; +} SWITCH_DECLARE(switch_status_t) switch_xml_locate_user(const char *key, const char *user_name,

1 0

[commit] r10992 - freeswitch/trunk/docs
by FreeSWITCH SVN 29 Dec '08

29 Dec '08

Author: mikej Date: Mon Dec 29 17:19:09 2008 New Revision: 10992 Log: Change-log Modified: freeswitch/trunk/docs/ChangeLog Modified: freeswitch/trunk/docs/ChangeLog ============================================================================== --- freeswitch/trunk/docs/ChangeLog (original) +++ freeswitch/trunk/docs/ChangeLog Mon Dec 29 17:19:09 2008 @@ -1,3 +1,294 @@ +freeswitch (1.0.2) + + all: don't add module interfaces before returning from error conditions in module load functions (MDXMLINT-36) + all: fixed multiple memory leaks + all: improved module unloading/reloading support + build: add support for --switchconfdir (FSBUILD-84) + build: fixed netbsd build + build: make freeswitch stop graceflly with /etc/init.d/freeswitch stop on debian add working dir to start-stop-dir so freeswitch dumps core in workdir + build: multiple packaging fixes + build: user freeswitch not added to audio group on deb install (FSBUILD-95) + Configuration: many updates to default configuration + core: Add ability to choose uuid from originate string, originate_uuid var (use at your own risk) + core: add bridge_generate_comfort_noise option for bridge to generate comfort noise to the A leg when there is no audio on the B leg + core: add chan vars to param event during hangup hook + core: add exec directive to xml preprocessor (not available on windows) + core: add force_transfer_dialplan and force_transfer_context variables + core: add hashing to event header lookup + core: add hits to tone_detect + core: add last_dtmf_duration variable + core: add msleep function to swigged languages + core: add park_after_bridge variable + core: add per leg timeouts and specific cause codes in reject_on_single_fail + core: add runtime selection of the module dir (FSCORE-198) + core: add scheduler support for heartbeat + core: add session heartbeat feature + core: add session.destroy psuedo method to sort of destroy a session at least for the sake of FS + core: add session.unsetInputCallback + core: add strftime format string validation for user supplied values + core: add vars param to switch_ivr_originate for recursion (MODAPP-175) + core: added a "group" concept to the user directory + core: added ability to do dns lookup to find ip with host: like stun: (FSCORE-219) + core: added better locking for codec changes during a call + core: added current_application and current_application_data variables + core: added error/ magic endpoint so modules can return error causes in situations like sofia_contact + core: added read_result channel variable + core: added support for "F" to indicate flash in dtmf (FSCORE-213) + core: allow calls to be stolen from originate + core: allow you to get the privacy bits in the caller_profile + core: change dso code to load symbols local + core: changes core flags to be array based so we have more + core: eavesdrop causes the people being eavesdropped on to not hear ach other (MODAPP-140) + core: expose time table to variable interface via caller field lookup + core: fix 100% cpu when sending parked call to moh (FSCORE-234) + core: fix bridge app to make sure both channels are ready for media when one is only in ringing state + core: fix buffer overflow (FSCORE-188) + core: fix conference dial by allowing multiple braces in originate, fix bad pointer op (FSCORE-208) + core: fix double detection of DTMF in IVR (FSCORE-221) + core: fix hangup_after_bridge is false on a bridge started with the intercept app + core: fix issue where pid file is accidentally truncated + core: fix ivr timeout (FSCORE-181) + core: fix memory leak in alias tab completion code + core: fix min digits in read app + core: fix out-of-bounds pointer in variable expansion (FSCORE-171) + core: fix segfault in media bugs when in bypass media (FSCORE-193) + core: fix segfault on gtalk to sip calls (FSCORE-212) + core: fix segfault on reloadxml (FSCORE-176) + core: fix segfault on trasfering eavesdopping party (FSCORE-210) + core: fix segfault using switch_system function (FSCORE-196) + core: fix session.bridge + core: fix setting effective_caller_id_name / effective_caller_id_number on bridge dialstring (MODAPP-122) + core: fix stream_raw_write (MODAPP-145) + core: fix using resampling on ringback file + core: fixed performance bottleneck in sqlite db's + core: fixed race in reloadxml + core: increment app before execute in case it returns to execute it will go to the next item in the list and not the same + core: ivr menu max_failures and max_timeouts now default to 3 if not specified or invalid (less than 1) values are specified (FSCORE-244) + core: ivr_menu max-timeouts option, result in ivr_menu_status var (FSCORE-183) + core: let b legs use park_after_bridge too + core: make events less verbose unless verbose_events is set + core: parse private events during originate + core: pass pdd data to a leg on oubound calls using bridge + core: prevent crash in crazy situation with xml interface lookup failures (FSCORE-169) + core: reduce cpu requirement for generated comfort noise + core: remove interface names from core db on unload + core: reworked timing resulting in significant performance increase and better rtp timing + core: rewrite switch_play_and_get_digits (MODAPP-166) + core: session.recordFile never terminates (MODLANG-79) + core: session.transfer make dialplan and context optional + core: set_user app now sets domain vars as well as user vars + core: tone_detect not triggering app after tone detection (MODAPP-182) + core: unprivileged user setting bigger stack for switch_system thread failure (FSCORE-197) + core: user_exists returns false when fetching a user from XML Curl or other xml interfaces + libesl: added c event socket library and fs_cli + libsndfile: fix autoconf 2.62 support (LBSNDF-5) + mod commands: add "all" modifier to "break" command + mod_celt: added new module + mod_commands: Add support for more than 2 variables to uuid_setvar_multi (MODAPP-171) + mod_commands: Add support for passing the cause of hangup to the uuid_kill command (FSCORE-217) + mod_commands: add attr lookup to user_data + mod_commands: add domain_exists fsapi command + mod_commands: add eval fsapi command + mod_commands: add flush_dtmf app and uuid_flush_dtmf api command + mod_commands: add fsctl send_sighup, fsctl shutdown asap, unsched_api commands + mod_commands: add fsctl shutdown [elegant|restart|cancel] + mod_commands: add new syntax to uuid_setvar to allow you to unset a var. <uuid> <var> [value] (MODAPP-167) + mod_commands: add reload fsapi command to reload a module + mod_commands: add system fsapi and application (MODAPP-138) + mod_commands: added hupall fsapi command + mod_commands: added strftime_tz api command + mod_commands: break all now stops broadcast too + mod_commands: fix api command sent through sched_api was getting the last char lopped off + mod_commands: fix race on transfer with -both + mod_commands: fix system dialplan app problems (MODAPP-86) + mod_commands: only send content-type on status when it really is http. + mod_conference: add fsapi to stop async playback too + mod_conference: add video caps to mod_conference with video follow audio + mod_conference: better sound prefix handling when using say: and allow say: on kick sounds. + mod_conference: fix race in record + mod_conference: fix runaway thread when floor holder has no video and other people do have video + mod_conference: fix seg when kicking many members quickly (MODAPP-129) + mod_conference: fix segfault on invalid chat event + mod_conference: perpetual sound does not auto-mute, you can do that yourself if you want it + mod_dialplan_xml: add Hunt- vars in dialplan lookup after transfer + mod_dialplan_xml: fail call on extensions with nested conditions + mod_dingaling: (LBDING-7) fix segfault on os x + mod_dingaling: end call on ice timeout + mod_dingaling: fix presence on jabber to be less protocol ambiguous + mod_dingaling: fix segfault (LBDING-10) + mod_dingaling: update to support latest client from google + mod_dptools: add a mechanism to tell if a file played from sendmsg over event socket + mod_dptools: add playback_terminator support to phrase and say app + mod_dptools: add playback_terminator_used variable (MODAPP-132) + mod_dptools: add presence application + mod_dptools: fix originate api not parsing users properly (FSCORE-246) + mod_dptools: fix record and record_session to create directory if it does not exist (FSCORE-250) + mod_dptools: fixed limit and + parsing bug in record_session app (MODAPP-148) + mod_dptools: remove_bugs added to remove all media bugs on a session + mod_erlang_event: add new module + mod_event_socket: missing : after Content-Length in event socket (MODEVENT-33) + mod_event_socket: add event socket listener filters + mod_event_socket: add stateful listener fsapi commands for ajax-y type event interface over http + mod_event_socket: fix arg parsing errors (MODEVENT-34) + mod_event_socket: fix shutdown segfault race (MODEVENT-32) + mod_event_socket: inbound connection to event_socket can now take over an existing channel with 'myevents <uuid>' to take on the behaviour of an outbound socket + mod_event_socket: let any channel get messages + mod_event_socket: make event socket wait for hangup on outbound mode and send disconnect message + mod_expr: fix endless loop + mod_fax: new module + mod_fifo: add fifo_consumer_wrapup_time var (MODAPP-117) + mod_fifo: added callback agents + mod_fifo: honor keyword silence (MODAPP-118) + mod_flite: added windows build + mod_fsv: fix in a windows enviroment opening the record file in text mode. (MODAPP-169) + mod_http: added new module + mod_java: updated to new module api to support read/write locks on interface + mod_limit: accept dialplan context for transfer (MODAPP-161) + mod_limit: added hashtable based limit functions + mod_limit: prevent empty error log message (MODAPP-134) + mod_local_stream: add start_local_stream and stop_local_stream fsapi commands to start/stop dynamically (MODFORM-13) + mod_local_stream: fix leak and improve error checking + mod_local_stream: fix seg when no timer name specified in config file. (MODFORM-16) + mod_loopback: add new module + mod_lua: add local scripts directory support (MODLANG-86) + mod_lua: don't eval blank string + mod_lua: fix originate + mod_lua: fix segfault (MODLANG-77) + mod_lua: update to lua 5.1.4 (MODLANG-87) + mod_lumenvox: removed + mod_managed: new module replaces mod_mono now supports native .net runtime on windows as well + mod_opal: added to trunk (still very beta) + mod_perl: fix segfault (MODLANG-77) + mod_pocketsphinx: fix rpm build + mod_portaudio: fix cpu race on inbound call to pa when no ring file is set + mod_radius_cdr: dictionary update for cause code changes (MODEVENT-27) + mod_radius_cdr: fix unload (MODEVENT-29) + mod_shout: add stereo recording broadcast support + mod_shout: added windows build + mod_shout: fix segfault when recording mp3's (MODFORM-12) + mod_shout: improved stability of mp3 decoding + mod_siren: added new module + mod_sndfile added support to record 16bit for the various rates including 48kHz + mod_sofia: Add filter to "sofia status profile XXX" (MODENDP-138) + mod_sofia: Add force-register-db-domain which works in conjunction with force-register-domain. + mod_sofia: Add optional <variables> and <params> tag to <gateway> tag. + mod_sofia: Challenge the right realm when to_host is outside the users domain. (MODENDP-136) + mod_sofia: Improve notify messages through a proxy (MODENDP-147) + mod_sofia: MWI for multiple domains (MODAPP-126) + mod_sofia: Move "a=sendrecv" from session to media section of SDP (MODENDP-148) + mod_sofia: add 200 OK re-invite without sdp + mod_sofia: add custom sofia::gateway_state event (MODENDP-112) + mod_sofia: add fire events for the refer SIP NOTIFY event package (MODENDP-152) + mod_sofia: add more params for xml_curl directory lookup + mod_sofia: add new auto vals for challenge-realm param <param name="challenge-realm" value="auto_from|auto_to|<hardcoded_val>"/> + mod_sofia: add option to turn of auto_restart of sofia profiles on ip change + mod_sofia: add params to use sip callid as uuid on inbound calls and uuid as sip callid on outbound calls + mod_sofia: add parsing of Privacy header for privacy info (MODENDP-133) + mod_sofia: add proto_specific_hangup_cause to both legs + mod_sofia: add proxy 3pcc mode + mod_sofia: add redirect variable to channel as well as partner channe (MODENDP-135) + mod_sofia: add sip-forbid-register to user params to refuse to let a certian user register + mod_sofia: add sip: into register-proxy when it's not specified + mod_sofia: add sip_history_info var for inbound invites. + mod_sofia: add sip_via_protocol variable + mod_sofia: add sofia xmlstatus (MODENDP-156) + mod_sofia: add support for params other than Replaces in Refer-To (MODENDP-143) + mod_sofia: add support for profiles sharing databases so that you can have a domain that uses multiple profiles for split dns type setups + mod_sofia: add support for refer transfer involving multiple machines + mod_sofia: add support to send a notify in the invite dialog by specifying the uuid of the call. (SFSIP-92) + mod_sofia: add suppress_from_cidname var to not have display name in from header (MODENDP-153) + mod_sofia: added sip_hangup_disposition variable + mod_sofia: allow send_message and notify events to send a message/notify without a body if needed. + mod_sofia: append -1 .. -N postfix after any X-headers as vars that have the same name + mod_sofia: cache auth_gateway_name in sofia for challenged bye + mod_sofia: cancel proxy or no-media mode if you purposely answer or pre_answer + mod_sofia: correct result code mapping for Unallocated Number (MODENDP-124) + mod_sofia: disable 100rel by default + mod_sofia: don't accept crypto in the RTP/AVP (MODENDP-126) + mod_sofia: don't put CN in sdp answer if it was not in the offer. + mod_sofia: fix Incorrect IP address shows up in SDP "o" field when multiple external IPs available and FS not bound to first (MODENDP-132) + mod_sofia: fix Wrong RTP media port destination after reinvite/UNHOLD (SFSIP-82) + mod_sofia: fix bug on linksys where they lie about the ptime and handle linksys transfer problem + mod_sofia: fix chat (send an IM) assumes that the user's profile is the same as their domain, which isn't necessarily so (SFSIP-83) + mod_sofia: fix dtmf handling of broken info dtmf endpoints + mod_sofia: fix eyebeam presence to be RFC compliant (MODENDP-144) + mod_sofia: fix ip change detection when in proxy mode + mod_sofia: fix register_proxy ignoring the paramaters (MODENDP-121) + mod_sofia: fix remote session refresh triggers request glare (MODENDP-131) + mod_sofia: fix rtp auto adjust running when it should not + mod_sofia: fix rtp sent to wrong port after some re-INVITE scenarios (MODENDP-141) + mod_sofia: fix sending of cn packets across bridge when we shouldn't + mod_sofia: fix sqlite issue with select of the sip contact + mod_sofia: fixed segfault on invalid presence payload + mod_sofia: gateway ping needs to look for 501 (SFSIP-78) + mod_sofia: handle multi contact register responses and register timeout better + mod_sofia: improve gateway resilience + mod_sofia: log ip and port you get reply to invite from + mod_sofia: make multiple-registations=true use the contact method and call-id option to do it the old way + mod_sofia: make proxy mode pull the port from m=image as well + mod_sofia: make register-proxy preserve the url composed from proxy but target the packets to desired address (MODENDP-121) + mod_sofia: many fixes for sonus rtp issues silence_when_idle=400 chanvar to send generated silence duing sleeps etc + mod_sofia: many fixes in presence handling + mod_sofia: passthrough t.38 fixes + mod_sofia: pick ipv4 or ipv6 based on sipip instead of having mixed in sdp + mod_sofia: send NOTIFY on TCP/UDP depending on the SUBSCRIBE (SFSIP-104) + mod_sofia: setting profile option multiple-registrations=contact key multi reg off the contact string + mod_sofia: wait for a reply on refer + mod_soundtouch: fixes and improvements, many options changed (MODAPP-149) + mod_soundtouch: updated to new module api + mod_spidermonkey: Segmentation fault in check_hangup_hook at mod_spidermonkey.c:1589 (MODLANG-74) + mod_spidermonkey: fix bug in apiExecute + mod_spidermonkey: fix memory pool handling and leaks + mod_spidermonkey: limit recursion busting through the stack (FSCORE-202) + mod_spidermonkey: make session.getVariable return blank string not the word false + mod_spidermonkey_curl: add optional content-type arg + mod_spidermonkey_odbc: fix numRows and add numCols + mod_spidermonkey_odbc: fix segfault (MODLANG-75) + mod_stress: new module for voice stress analysis + mod_syslog: don't log blank lines (FSCORE-163) + mod_tone_stream: let silence_stream://0 indicate perpetual silence + mod_vmd: add new module to detect voicemail "beep" + mod_voicemail: Add vm_alternate_greet_id param to directory entry (MODAPP-174) + mod_voicemail: Patch to add voicemail preference controlling date announcement new param 'play-date-announcement' to values 'first' 'last' or 'never' defaults to first to retain previous behavior (MODAPP-121) + mod_voicemail: Update mwi light after delete vm via web. (MODAPP-124) + mod_voicemail: add ability to get to options without listening to every saved message (MODAPP-115) + mod_voicemail: add ability to skip greeting when leaving a voicemail. (MODAPP-181) + mod_voicemail: add change-pass-key config file option + mod_voicemail: add forwarding support + mod_voicemail: add local dtmf driven alternat vm pass + mod_voicemail: add proper notification of a vm message being too short + mod_voicemail: add support for auth via a1-hash + mod_voicemail: add the "storage-dir" parameter to be set on a per-user basis (MODAPP-133) + mod_voicemail: add voicemail_greeting_path variable + mod_voicemail: added voicemail_alternate_greet_id variable + mod_voicemail: allow changing of password from voicemail to update user directory if using non-static config (MODAPP-156) + mod_voicemail: created email date (int overflow) (MODAPP-125) + mod_voicemail: don't try to deliver vm when no file was recorded. (MODAPP-133) + mod_voicemail: fix MWI with xml_curl used for directory (MODAPP-176) + mod_voicemail: fix Voicemail messages occasionally lost / stranded (MODAPP-178) + mod_voicemail: fix invalid event after message deleted (MODAPP-170) + mod_voicemail: fix mwi for phones with multiple registrations problem (MODAPP-153) + mod_voicemail: fix voicemail segfault on incorrect password (FSCORE-187) + mod_voicemail: fix voicemail_inject error handling (MODAPP-133) + mod_voicemail: fix voicemail_inject usage api call + mod_voicemail: improve error checking (MODAPP-142) + mod_voicemail: localize notification emails (MODAPP-139) + mod_voicemail: make more multi-domain friendly (MODAPP-162) + mod_voicemail: make playback created file macros optional (MODAPP-150) + mod_voicemail: recognize operator key in more places (MODAPP-159) + mod_voicemail: web interface displays incorrect created / last heard dates (MODAPP-123) + mod_wanpipe: removed + mod_xml_cdr: add https support + mod_xml_cdr: add optional a-leg prefix to xml cdr filenames (MDXMLINT-39) + mod_xml_cdr: add support for fallback webserver for cdr posting (FSCORE-238) + mod_xml_curl: Allow specification of HTTP method, and dynamic expansion of variables in URI. (MDXMLINT-41) + mod_xml_curl: added redirect following (max 10) + mod_xml_ldap: almost a complete rewrite of this module + mod_xml_rpc: allow setting of global realm without a global user (MDXMLINT-45) + mod_xml_rpc: fix multiple segfaults + mod_xml_rpc: fix segfault on originate via http + sofia-sip: updated to 1.12.10 (plus a few patches) + freeswitch (1.0.1) FIX: prevent intercept race condition that can also be solved with continue_on_fail=originator_cancel

1 0

[commit] r10991 - in freeswitch/trunk/src: . include mod/endpoints/mod_sofia
by FreeSWITCH SVN 29 Dec '08

29 Dec '08

Author: anthm Date: Mon Dec 29 16:23:17 2008 New Revision: 10991 Log: add rtp_bug flags so we can keep track of all the bugs we have to emulate Modified: freeswitch/trunk/src/include/switch_rtp.h freeswitch/trunk/src/include/switch_types.h freeswitch/trunk/src/mod/endpoints/mod_sofia/mod_sofia.h freeswitch/trunk/src/mod/endpoints/mod_sofia/sofia_glue.c freeswitch/trunk/src/switch_rtp.c Modified: freeswitch/trunk/src/include/switch_rtp.h ============================================================================== --- freeswitch/trunk/src/include/switch_rtp.h (original) +++ freeswitch/trunk/src/include/switch_rtp.h Mon Dec 29 16:23:17 2008 @@ -431,6 +431,8 @@ SWITCH_DECLARE(switch_status_t) switch_rtp_activate_stun_ping(switch_rtp_t *rtp_session, const char *stun_ip, switch_port_t stun_port, uint32_t packet_count, switch_bool_t funny); +SWITCH_DECLARE(void) switch_rtp_intentional_bugs(switch_rtp_t *rtp_session, switch_rtp_bug_flag_t bugs); + /*! \} */ Modified: freeswitch/trunk/src/include/switch_types.h ============================================================================== --- freeswitch/trunk/src/include/switch_types.h (original) +++ freeswitch/trunk/src/include/switch_types.h Mon Dec 29 16:23:17 2008 @@ -460,6 +460,38 @@ } switch_rtp_flag_enum_t; typedef uint32_t switch_rtp_flag_t; +typedef enum { + RTP_BUG_NONE = 0, /* won't be using this one much ;) */ + + RTP_BUG_CISCO_SKIP_MARK_BIT_2833 = (1 << 0), + /* Some Cisco devices get mad when you send the mark bit on new 2833 because it makes + them flush their jitterbuffer and the dtmf along with it. + + This flag will disable the sending of the mark bit on the first DTMF packet. + */ + + + RTP_BUG_SONUS_SEND_INVALID_TIMESTAMP_2833 = (1 << 1) + /* + Sonus wrongly expects that, when sending a multi-packet 2833 DTMF event, The sender + should increment the RTP timestamp in each packet when, in reality, the sender should + send the same exact timestamp and increment the duration field in the 2833 payload. + This allows a reconstruction of the duration if any of the packets are lost. + + final_duration - initial_timestamp = total_samples + + However, if the duration value exceeds the space allocated (16 bits), The sender should increment + the timestamp one unit and reset the duration to 0. + + Always sending a duration of 0 with a new timestamp should be tolerated but is rarely intentional + and is mistakenly done by many devices. + The issue is that the Sonus expects everyone to do it this way instead of tolerating eiher way. + Sonus will actually ignore every packet with the same timestamp before concluding if it's DTMF. + + This flag will cause each packet to have a new timestamp. + */ + +} switch_rtp_bug_flag_t; #ifdef _MSC_VER #pragma pack(push, r1, 1) Modified: freeswitch/trunk/src/mod/endpoints/mod_sofia/mod_sofia.h ============================================================================== --- freeswitch/trunk/src/mod/endpoints/mod_sofia/mod_sofia.h (original) +++ freeswitch/trunk/src/mod/endpoints/mod_sofia/mod_sofia.h Mon Dec 29 16:23:17 2008 @@ -208,7 +208,7 @@ TFLAG_NOHUP = (1 << 18), TFLAG_NOSDP_REINVITE = (1 << 19), TFLAG_NAT = (1 << 20), - TFLAG_BUGGY_2833 = (1 << 21), + TFLAG_USEME = (1 << 21), TFLAG_SIP_HOLD = (1 << 22), TFLAG_INB_NOMEDIA = (1 << 23), TFLAG_LATE_NEGOTIATION = (1 << 24), @@ -552,6 +552,7 @@ uint32_t mismatch_count; uint32_t last_codec_ms; nua_event_t want_event; + switch_rtp_bug_flag_t rtp_bugs; }; struct callback_t { Modified: freeswitch/trunk/src/mod/endpoints/mod_sofia/sofia_glue.c ============================================================================== --- freeswitch/trunk/src/mod/endpoints/mod_sofia/sofia_glue.c (original) +++ freeswitch/trunk/src/mod/endpoints/mod_sofia/sofia_glue.c Mon Dec 29 16:23:17 2008 @@ -1775,10 +1775,6 @@ flags = (switch_rtp_flag_t) (SWITCH_RTP_FLAG_DATAWAIT); } - if (switch_test_flag(tech_pvt, TFLAG_BUGGY_2833)) { - flags |= SWITCH_RTP_FLAG_BUGGY_2833; - } - if ((val = switch_channel_get_variable(tech_pvt->channel, "dtmf_type"))) { if (!strcasecmp(val, "rfc2833")) { tech_pvt->dtmf_type = DTMF_2833; @@ -1923,7 +1919,9 @@ tech_pvt->ssrc = switch_rtp_get_ssrc(tech_pvt->rtp_session); switch_set_flag(tech_pvt, TFLAG_RTP); switch_set_flag(tech_pvt, TFLAG_IO); - + + switch_rtp_intentional_bugs(tech_pvt->rtp_session, tech_pvt->rtp_bugs); + if ((vad_in && inb) || (vad_out && !inb)) { switch_rtp_enable_vad(tech_pvt->rtp_session, tech_pvt->session, &tech_pvt->read_codec, SWITCH_VAD_FLAG_TALKING); switch_set_flag(tech_pvt, TFLAG_VAD); @@ -2205,12 +2203,12 @@ if ((tech_pvt->origin = switch_core_session_strdup(session, (char *) sdp->sdp_origin->o_username))) { if (strstr(tech_pvt->origin, "CiscoSystemsSIP-GW-UserAgent")) { - switch_set_flag_locked(tech_pvt, TFLAG_BUGGY_2833); + tech_pvt->rtp_bugs |= RTP_BUG_CISCO_SKIP_MARK_BIT_2833; switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "Activate Buggy RFC2833 Mode!\n"); } if (strstr(tech_pvt->origin, "Sonus_UAC")) { - switch_set_flag_locked(tech_pvt, TFLAG_BUGGY_2833); + tech_pvt->rtp_bugs |= RTP_BUG_SONUS_SEND_INVALID_TIMESTAMP_2833; switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_WARNING, "Hello,\nI see you have a Sonus!\n" "FYI, Sonus cannot follow the RFC on the proper way to send DTMF.\n" Modified: freeswitch/trunk/src/switch_rtp.c ============================================================================== --- freeswitch/trunk/src/switch_rtp.c (original) +++ freeswitch/trunk/src/switch_rtp.c Mon Dec 29 16:23:17 2008 @@ -199,6 +199,7 @@ switch_port_t stun_port; int from_auto; uint32_t cng_count; + switch_rtp_bug_flag_t rtp_bugs; }; static int global_init = 0; @@ -486,6 +487,11 @@ return port; } +SWITCH_DECLARE(void) switch_rtp_intentional_bugs(switch_rtp_t *rtp_session, switch_rtp_bug_flag_t bugs) +{ + rtp_session->rtp_bugs = bugs; +} + SWITCH_DECLARE(switch_status_t) switch_rtp_set_local_address(switch_rtp_t *rtp_session, const char *host, switch_port_t port, const char **err) { switch_socket_t *new_sock = NULL, *old_sock = NULL; @@ -1204,7 +1210,7 @@ rtp_session->dtmf_data.timestamp_dtmf, rtp_session->dtmf_data.out_digit_sofar, rtp_session->dtmf_data.out_digit_sub_sofar, rtp_session->dtmf_data.out_digit_dur, rtp_session->seq); - if (switch_test_flag(rtp_session, SWITCH_RTP_FLAG_BUGGY_2833)) { + if (rtp_session->rtp_bugs & RTP_BUG_SONUS_SEND_INVALID_TIMESTAMP_2833) { rtp_session->dtmf_data.timestamp_dtmf = rtp_session->last_write_ts + samples; } } @@ -1257,7 +1263,7 @@ switch_rtp_write_manual(rtp_session, rtp_session->dtmf_data.out_digit_packet, 4, - switch_test_flag(rtp_session, SWITCH_RTP_FLAG_BUGGY_2833) ? 0 : 1, + rtp_session->rtp_bugs & RTP_BUG_CISCO_SKIP_MARK_BIT_2833 ? 0 : 1, rtp_session->te, rtp_session->dtmf_data.timestamp_dtmf, &flags); switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "Send start packet for [%c] ts=%u dur=%d/%d/%d seq=%d\n",

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[commit] r10990 - freeswitch/trunk/libs/esl
by FreeSWITCH SVN 29 Dec '08

29 Dec '08

Author: anthm Date: Mon Dec 29 13:40:40 2008 New Revision: 10990 Log: don't display console_clean logs Modified: freeswitch/trunk/libs/esl/fs_cli.c Modified: freeswitch/trunk/libs/esl/fs_cli.c ============================================================================== --- freeswitch/trunk/libs/esl/fs_cli.c (original) +++ freeswitch/trunk/libs/esl/fs_cli.c Mon Dec 29 13:40:40 2008 @@ -86,17 +86,17 @@ int level = 0, tchannel = 0; const char *lname = esl_event_get_header(handle->last_event, "log-level"); const char *channel = esl_event_get_header(handle->last_event, "text-channel"); - const char *file = esl_event_get_header(handle->last_event, "log-file"); + //const char *file = esl_event_get_header(handle->last_event, "log-file"); if (channel) { tchannel = atoi(channel); } - + if (lname) { level = atoi(lname); } - - if (tchannel == 0 || (file && !strcmp(file, "switch_console.c"))) { + + if (tchannel == 0) { #ifdef WIN32 DWORD len = (DWORD) strlen(handle->last_event->body); DWORD outbytes = 0;

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[commit] r10989 - in freeswitch/branches/gmaruzz/src/mod/endpoints/mod_skypiax: . configs
by FreeSWITCH SVN 29 Dec '08

29 Dec '08

Author: gmaruzz Date: Mon Dec 29 13:36:45 2008 New Revision: 10989 Log: skypiax: config and startup files, and make it compile on 64bit machines, let's hope it still compile on 32bit. I'll check it tomorrow Modified: freeswitch/branches/gmaruzz/src/mod/endpoints/mod_skypiax/configs/startskype.sh freeswitch/branches/gmaruzz/src/mod/endpoints/mod_skypiax/skypiax_protocol.c Modified: freeswitch/branches/gmaruzz/src/mod/endpoints/mod_skypiax/configs/startskype.sh ============================================================================== --- freeswitch/branches/gmaruzz/src/mod/endpoints/mod_skypiax/configs/startskype.sh (original) +++ freeswitch/branches/gmaruzz/src/mod/endpoints/mod_skypiax/configs/startskype.sh Mon Dec 29 13:36:45 2008 @@ -1,3 +1,6 @@ +# remember to add here the removing of all the installed snd-* modules, so you're sure only the snd-dummy driver will be around +rmmod snd_hda_intel + # you need three dummy soundcard for 20 Skype client instances, because each dummy soundcard can handle a max of 8 Skype instances # the enable= module parameter tells how many cards to start. For each additional card, add a comma and a 1 # configure the first 8 Skype client instances to use the hw:Dummy_0, the next 8 instances to use hw:Dummy_1, etc for all three devices (Play, Capture, Ring) @@ -5,125 +8,125 @@ sleep 5 #start the fake X server on a given port -Xvfb :101 & +/usr/bin/Xvfb :101 & sleep 5 # start a Skype client instance that will connect to the X server above, and will login to the Skype network using the "username password" you send to it on stdin. Here xxx would be the password and skypiax1 the username -echo "skypiax1 xxx"| DISPLAY=:101 /usr/bin/skype.static --pipelogin & +echo "skypiax1 xxx"| DISPLAY=:101 /usr/bin/skype --pipelogin & sleep 10 -Xvfb :102 & +/usr/bin/Xvfb :102 & sleep 5 -echo "skypiax2 xxx"| DISPLAY=:102 /usr/bin/skype.static --pipelogin & +echo "skypiax2 xxx"| DISPLAY=:102 /usr/bin/skype --pipelogin & sleep 10 -Xvfb :103 & +/usr/bin/Xvfb :103 & sleep 5 -echo "skypiax3 xxx"| DISPLAY=:103 /usr/bin/skype.static --pipelogin & +echo "skypiax3 xxx"| DISPLAY=:103 /usr/bin/skype --pipelogin & sleep 10 -Xvfb :104 & +/usr/bin/Xvfb :104 & sleep 5 -echo "skypiax4 xxx"| DISPLAY=:104 /usr/bin/skype.static --pipelogin & +echo "skypiax4 xxx"| DISPLAY=:104 /usr/bin/skype --pipelogin & sleep 10 -Xvfb :105 & +/usr/bin/Xvfb :105 & sleep 5 -echo "skypiax5 xxx"| DISPLAY=:105 /usr/bin/skype.static --pipelogin & +echo "skypiax5 xxx"| DISPLAY=:105 /usr/bin/skype --pipelogin & sleep 10 -Xvfb :106 & +/usr/bin/Xvfb :106 & sleep 5 -echo "skypiax6 xxx"| DISPLAY=:106 /usr/bin/skype.static --pipelogin & +echo "skypiax6 xxx"| DISPLAY=:106 /usr/bin/skype --pipelogin & sleep 10 -Xvfb :107 & +/usr/bin/Xvfb :107 & sleep 5 -echo "skypiax7 xxx"| DISPLAY=:107 /usr/bin/skype.static --pipelogin & +echo "skypiax7 xxx"| DISPLAY=:107 /usr/bin/skype --pipelogin & sleep 10 -Xvfb :108 & +/usr/bin/Xvfb :108 & sleep 5 -echo "skypiax8 xxx"| DISPLAY=:108 /usr/bin/skype.static --pipelogin & +echo "skypiax8 xxx"| DISPLAY=:108 /usr/bin/skype --pipelogin & sleep 10 -Xvfb :109 & +/usr/bin/Xvfb :109 & sleep 5 -echo "skypiax9 xxx"| DISPLAY=:109 /usr/bin/skype.static --pipelogin & +echo "skypiax9 xxx"| DISPLAY=:109 /usr/bin/skype --pipelogin & sleep 10 -Xvfb :110 & +/usr/bin/Xvfb :110 & sleep 5 -echo "skypiax10 xxx"| DISPLAY=:110 /usr/bin/skype.static --pipelogin & +echo "skypiax10 xxx"| DISPLAY=:110 /usr/bin/skype --pipelogin & sleep 10 -Xvfb :111 & +/usr/bin/Xvfb :111 & sleep 5 -echo "skypiax11 xxx"| DISPLAY=:111 /usr/bin/skype.static --pipelogin & +echo "skypiax11 xxx"| DISPLAY=:111 /usr/bin/skype --pipelogin & sleep 10 -Xvfb :112 & +/usr/bin/Xvfb :112 & sleep 5 -echo "skypiax12 xxx"| DISPLAY=:112 /usr/bin/skype.static --pipelogin & +echo "skypiax12 xxx"| DISPLAY=:112 /usr/bin/skype --pipelogin & sleep 10 -Xvfb :113 & +/usr/bin/Xvfb :113 & sleep 5 -echo "skypiax13 xxx"| DISPLAY=:113 /usr/bin/skype.static --pipelogin & +echo "skypiax13 xxx"| DISPLAY=:113 /usr/bin/skype --pipelogin & sleep 10 -Xvfb :114 & +/usr/bin/Xvfb :114 & sleep 5 -echo "skypiax14 xxx"| DISPLAY=:114 /usr/bin/skype.static --pipelogin & +echo "skypiax14 xxx"| DISPLAY=:114 /usr/bin/skype --pipelogin & sleep 10 -Xvfb :115 & +/usr/bin/Xvfb :115 & sleep 5 -echo "skypiax15 xxx"| DISPLAY=:115 /usr/bin/skype.static --pipelogin & +echo "skypiax15 xxx"| DISPLAY=:115 /usr/bin/skype --pipelogin & sleep 10 -Xvfb :116 & +/usr/bin/Xvfb :116 & sleep 5 -echo "skypiax16 xxx"| DISPLAY=:116 /usr/bin/skype.static --pipelogin & +echo "skypiax16 xxx"| DISPLAY=:116 /usr/bin/skype --pipelogin & sleep 10 -Xvfb :117 & +/usr/bin/Xvfb :117 & sleep 5 -echo "skypiax17 xxx"| DISPLAY=:117 /usr/bin/skype.static --pipelogin & +echo "skypiax17 xxx"| DISPLAY=:117 /usr/bin/skype --pipelogin & sleep 10 -Xvfb :118 & +/usr/bin/Xvfb :118 & sleep 5 -echo "skypiax18 xxx"| DISPLAY=:118 /usr/bin/skype.static --pipelogin & +echo "skypiax18 xxx"| DISPLAY=:118 /usr/bin/skype --pipelogin & sleep 10 -Xvfb :119 & +/usr/bin/Xvfb :119 & sleep 5 -echo "skypiax19 xxx"| DISPLAY=:119 /usr/bin/skype.static --pipelogin & +echo "skypiax19 xxx"| DISPLAY=:119 /usr/bin/skype --pipelogin & sleep 10 -Xvfb :120 & +/usr/bin/Xvfb :120 & sleep 5 -echo "skypiax20 xxx"| DISPLAY=:120 /usr/bin/skype.static --pipelogin & +echo "skypiax20 xxx"| DISPLAY=:120 /usr/bin/skype --pipelogin & sleep 10 Modified: freeswitch/branches/gmaruzz/src/mod/endpoints/mod_skypiax/skypiax_protocol.c ============================================================================== --- freeswitch/branches/gmaruzz/src/mod/endpoints/mod_skypiax/skypiax_protocol.c (original) +++ freeswitch/branches/gmaruzz/src/mod/endpoints/mod_skypiax/skypiax_protocol.c Mon Dec 29 13:36:45 2008 @@ -190,7 +190,7 @@ howmany = write(p->audiopipe[1], srv_in, howmany); #endif /* WIN32 */ p->audiobuf_is_loaded = 0; - WARNINGA("SRV PIPE WRITE=====> req=%d recv=%d to sent=%d sent=%d\n", SKYPIAX_P_LOG, sizeof(short)*SAMPLES_PER_FRAME, len, len, howmany); + WARNINGA("SRV PIPE WRITE=====> req=%lu recv=%d to sent=%d sent=%u\n", SKYPIAX_P_LOG, sizeof(short)*SAMPLES_PER_FRAME, len, len, howmany); }else if (len == SAMPLES_PER_FRAME * sizeof(short) / 2) { #if 1 @@ -570,7 +570,7 @@ if (samples != SAMPLES_PER_FRAME * sizeof(short)) { if (samples) - ERRORA("SRV PIPE READ=====> NOT GOOD samples=%d expected=%d\n", SKYPIAX_P_LOG, + ERRORA("SRV PIPE READ=====> NOT GOOD samples=%u expected=%lu\n", SKYPIAX_P_LOG, samples, SAMPLES_PER_FRAME * sizeof(short)); //do nothing } else {

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[commit] r10988 - freeswitch/branches/gmaruzz/src/mod/endpoints/mod_skypiax/configs
by FreeSWITCH SVN 29 Dec '08

29 Dec '08

Author: gmaruzz Date: Mon Dec 29 07:47:42 2008 New Revision: 10988 Log: skypiax: updated config and startup files Modified: freeswitch/branches/gmaruzz/src/mod/endpoints/mod_skypiax/configs/startskype.sh Modified: freeswitch/branches/gmaruzz/src/mod/endpoints/mod_skypiax/configs/startskype.sh ============================================================================== --- freeswitch/branches/gmaruzz/src/mod/endpoints/mod_skypiax/configs/startskype.sh (original) +++ freeswitch/branches/gmaruzz/src/mod/endpoints/mod_skypiax/configs/startskype.sh Mon Dec 29 07:47:42 2008 @@ -1,9 +1,14 @@ +# you need three dummy soundcard for 20 Skype client instances, because each dummy soundcard can handle a max of 8 Skype instances +# the enable= module parameter tells how many cards to start. For each additional card, add a comma and a 1 +# configure the first 8 Skype client instances to use the hw:Dummy_0, the next 8 instances to use hw:Dummy_1, etc for all three devices (Play, Capture, Ring) modprobe snd-dummy enable=1,1,1 +sleep 5 - +#start the fake X server on a given port Xvfb :101 & sleep 5 +# start a Skype client instance that will connect to the X server above, and will login to the Skype network using the "username password" you send to it on stdin. Here xxx would be the password and skypiax1 the username echo "skypiax1 xxx"| DISPLAY=:101 /usr/bin/skype.static --pipelogin & sleep 10

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[commit] r10987 - freeswitch/branches/gmaruzz/src/mod/endpoints/mod_skypiax/configs
by FreeSWITCH SVN 29 Dec '08

29 Dec '08

Author: gmaruzz Date: Mon Dec 29 07:36:06 2008 New Revision: 10987 Log: skypiax: updated config and startup files Added: freeswitch/branches/gmaruzz/src/mod/endpoints/mod_skypiax/configs/startskype.sh - copied unchanged from r10986, /freeswitch/branches/gmaruzz/src/mod/endpoints/mod_skypiax/configs/command1 Removed: freeswitch/branches/gmaruzz/src/mod/endpoints/mod_skypiax/configs/command1

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[commit] r10986 - freeswitch/branches/gmaruzz/src/mod/endpoints/mod_skypiax/configs
by FreeSWITCH SVN 29 Dec '08

29 Dec '08

Author: gmaruzz Date: Mon Dec 29 07:35:55 2008 New Revision: 10986 Log: skypiax: updated config and startup files Removed: freeswitch/branches/gmaruzz/src/mod/endpoints/mod_skypiax/configs/command freeswitch/branches/gmaruzz/src/mod/endpoints/mod_skypiax/configs/default.xml freeswitch/branches/gmaruzz/src/mod/endpoints/mod_skypiax/configs/modules.conf.xml freeswitch/branches/gmaruzz/src/mod/endpoints/mod_skypiax/configs/openzap.conf.xml freeswitch/branches/gmaruzz/src/mod/endpoints/mod_skypiax/configs/portaudio.conf.xml freeswitch/branches/gmaruzz/src/mod/endpoints/mod_skypiax/configs/zaptel.conf Modified: freeswitch/branches/gmaruzz/src/mod/endpoints/mod_skypiax/configs/command1 Modified: freeswitch/branches/gmaruzz/src/mod/endpoints/mod_skypiax/configs/command1 ============================================================================== --- freeswitch/branches/gmaruzz/src/mod/endpoints/mod_skypiax/configs/command1 (original) +++ freeswitch/branches/gmaruzz/src/mod/endpoints/mod_skypiax/configs/command1 Mon Dec 29 07:35:55 2008 @@ -1,3 +1,6 @@ +modprobe snd-dummy enable=1,1,1 + + Xvfb :101 & sleep 5

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