search for: app_dumpchan

.../lib/asterisk/modules/app_queue.so: undefined symbol: global_app_buf [Aug 11 22:00:01] NOTICE[20173]: res_smdi.c:1330 load_module: No SMDI interfaces are available to listen on, not starting SMDI listener. [Aug 11 22:00:01] WARNING[20173]: loader.c:376 load_dynamic_module: Error loading module 'app_dumpchan.so': /usr/lib/asterisk/modules/app_dumpchan.so: undefined symbol: global_app_buf [Aug 11 22:00:01] WARNING[20173]: loader.c:376 load_dynamic_module: Error loading module 'app_voicemail.so': /usr/lib/asterisk/modules/app_voicemail.so: undefined symbol: global_app_buf [Aug 11 22:00:01] WA...

...k/modules/pbx_realtime.so: undefined symbol: ast_load_realtime_multientry Aug 14 15:03:33 WARNING[4824]: Loading module pbx_realtime.so failed! Aug 14 15:03:46 WARNING[4836]: Unknown directive 'parkingtime' at line 9 of features.conf Aug 14 15:03:46 WARNING[4836]: /usr/lib/asterisk/modules/app_dumpchan.so: undefined symbol: ast_register_file_version Aug 14 15:03:46 WARNING[4836]: Loading module app_dumpchan.so failed! Aug 14 15:04:13 WARNING[4848]: Unknown directive 'parkingtime' at line 9 of features.conf Aug 14 15:04:13 WARNING[4848]: /usr/lib/asterisk/modules/app_while.so: undefined s...

...I don't have the /usr/lib/asterisk/modules/codec_g729a.so file in place it works. I use or used to use the licensed G729 Codec from Digium. This is the error message from asterisk -vvg: [app_playback.so] => (Sound File Playback Application) == Registered application 'Playback' [app_dumpchan.so] => (Dump Info About The Calling Channel) == Registered application 'DumpChan' [app_zapateller.so] => (Block Telemarketers with Special Information Tone) == Registered application 'Zapateller' [codec_ilbc.so] => (iLBC/PCM16 (signed linear) Codec Translator) == Re...

...the number of modules to a minimum ? Here is my modules.conf : [modules] autoload=yes load => pbx_config.so load => pbx_spool.so load => chan_local.so load => chan_iax2.so load => chan_sip.so load => app_authenticate.so load => app_cdr.so load => app_dial.so load => app_dumpchan.so load => app_echo.so load => app_exec.so load => app_hangup.so load => app_macro.so load => app_originate.so load => app_playback.so load => app_playtones.so load => app_record.so load => app_userevent.so load => codec_adpcm.so load => codec_alaw.so load => co...

...; which they both do. Not sure why I even do that. Perhaps I was planning some other feature that I never finished. Or does VoiceMail() use it? Someone (in private email for some reason so I won't give his name) suggested DumpChan() but I can't seem to make that work, even if I load app_dumpchan. -- D'Arcy J.M. Cain Vybe Networks Inc. http://www.VybeNetworks.com/ IM:darcy at Vex.Net VoIP: sip:darcy at VybeNetworks.com

On 2017-04-19 11:57 AM, J Montoya or A J Stiles wrote: > I fished this out of an old extensions.conf from a defunct project. It might > be relevant to your use case: > > exten => 1571,1,NoOp(Call to 1571: voicemail retrieval) > exten => 1571,n,AGI(lookup_caller_id.agi,${CALLERID(num)}) > exten => 1571,n,NoOp(CLID is ${clid}) > exten =>

...ng symbols from /opt/asterisk/lib/modules/app_directory.so...done. Loaded symbols for /opt/asterisk/lib/modules/app_directory.so Reading symbols from /opt/asterisk/lib/modules/app_disa.so...done. Loaded symbols for /opt/asterisk/lib/modules/app_disa.so Reading symbols from /opt/asterisk/lib/modules/app_dumpchan.so...done. Loaded symbols for /opt/asterisk/lib/modules/app_dumpchan.so Reading symbols from /opt/asterisk/lib/modules/app_echo.so...done. Loaded symbols for /opt/asterisk/lib/modules/app_echo.so Reading symbols from /opt/asterisk/lib/modules/app_exec.so...done. Loaded symbols for /opt/asterisk/lib...

.../asterisk/lib/modules/app_directory.so...done. > Loaded symbols for /opt/asterisk/lib/modules/app_directory.so > Reading symbols from /opt/asterisk/lib/modules/app_disa.so...done. > Loaded symbols for /opt/asterisk/lib/modules/app_disa.so > Reading symbols from /opt/asterisk/lib/modules/app_dumpchan.so...done. > Loaded symbols for /opt/asterisk/lib/modules/app_dumpchan.so > Reading symbols from /opt/asterisk/lib/modules/app_echo.so...done. > Loaded symbols for /opt/asterisk/lib/modules/app_echo.so > Reading symbols from /opt/asterisk/lib/modules/app_exec.so...done. > Loaded symb...

...equires N/A ;load => app_cut.so ; Requires N/A ;load => app_db.so ; Requires N/A load => app_dial.so ; Requires res_features.so, res_musiconhold.so ;load => app_dictate.so ; Requires N/A load => app_directory.so ; Requires N/A ;load => app_disa.so ; Requires N/A ;load => app_dumpchan.so ; Requires N/A load => app_echo.so ; Requires N/A ;load => app_enumlookup.so ; Requires N/A load => app_eval.so ; Requires N/A ;load => app_exec.so ; Requires N/A load => app_festival.so ; Requires N/A load => app_forkcdr.so ; Requires N/A ;load => app_getcpeid.so ; R...

...tities.) == Registered application 'ForkCDR' [app_math.so] => (Basic Math Functions) == Registered application 'Math' [app_realtime.so] => (Realtime Data Lookup/Rewrite) == Registered application 'RealTimeUpdate' == Registered application 'RealTime' [app_dumpchan.so] => (Dump Info About The Calling Channel) == Registered application 'DumpChan' [app_waitforsilence.so] => (Wait For Silence) == Registered application 'WaitForSilence' [app_while.so] => (While Loops and Conditional Execution) == Registered application 'While&...

Anyone knows where can I find spandsp? Official site seems permanently down... TIA, Simone.

Anyone know how to fix this? gcc -shared -Xlinker -x -o app_rxfax.so app_rxfax.c -lspandsp -ltiff In file included from app_rxfax.c:14: /usr/include/asterisk/lock.h: In function `ast_mutex_init': /usr/include/asterisk/lock.h:302: error: `PTHREAD_MUTEX_RECURSIVE' undeclared (first use in this function) /usr/include/asterisk/lock.h:302: error: (Each undeclared identifier is reported

I am attempting to update my Asterisk installation from 1.0 to the latest stable version. When I use CVS checkout, I am receiving the following messages on chan_sip.c: RCS file: /usr/cvsroot/asterisk/channels/chan_sip.c,v retrieving revision 1.510.2.25 retrieving revision 1.510.2.27 Merging differences between 1.510.2.25 and 1.510.2.27 into chan_sip.c M asterisk/channels/chan_sip.c Then, when

...asterisk-1.4.20.1/apps' for x in app_adsiprog.so app_alarmreceiver.so app_amd.so app_authenticate.so app_cbmysql.so app_cdr.so app_chanisavail.so app_channelredirect.so app_chanspy.so app_controlplayback.so app_db.so app_dial.so app_dictate.so app_directed_pickup.so app_directory.so app_disa.so app_dumpchan.so app_echo.so app_exec.so app_externalivr.so app_festival.so app_flash.so app_followme.so app_forkcdr.so app_getcpeid.so app_hasnewvoicemail.so app_ices.so app_image.so app_lookupblacklist.so app_lookupcidname.so app_macro.so app_meetme.so app_milliwatt.so app_mixmonitor.so app_morsecode.so app_mp...

Has anyone had any luck with PostgreSQL and Realtime? The realtime instructions on voip-info seem pretty straight forward... just not woking for me. I've included all of my config files below, and my console output. Entire console bootup output: [root@abox asterisk]# /usr/sbin/asterisk -vvvvvvc == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing