search for: func_callerid

...gt; codec_lpc10.so load => codec_ulaw.so load => format_gsm.so load => format_pcm.so load => format_wav.so load => format_wav_gsm.so load => res_agi.so load => res_clioriginate.so load => res_fax.so load => res_musiconhold.so load => res_timing_timerfd.so load => func_callerid.so load => func_cdr.so load => func_channel.so load => func_cut.so load => func_math.so load => func_rand.so load => func_strings.so load => func_global.so load => cdr_csv.so Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: <http:...

I have asterisk 1.8.2.3 + A102D Sangoma card 2 port T1. when i am starting asterisk i am getting this error on console. func_callerid.so => (Party ID related dialplan functions (Caller-ID, Connected-line, Redirecting)) == Registered application 'PrivacyManager' app_privacy.so => (Require phone number to be entered, if no CallerID sent) == Registered custom function 'TIMEOUT' func_timeout.so => (Chan...

...ATE' func_extstate.so => (Gets an extension's state in the dialplan) == Registered application 'DAHDIBarge' app_dahdibarge.so => (Barge in on DAHDI channel application) == Registered custom function 'CALLERPRES' == Registered custom function 'CALLERID' func_callerid.so => (Caller ID related dialplan functions) [2014-02-12 22:40:07] NOTICE[13842]: codec_g729a.c:760 load_module: G.729A transcoding module version 1.6.0_3.1.4, Copyright (C) 1999-2009 Digium, Inc. [2014-02-12 22:40:07] NOTICE[13842]: codec_g729a.c:761 load_module: This module is supplied under a...

...related to res_rtp_asterisk.so. So the old config was: [modules] autoload=no load => res_rtp_asterisk.so load => res_http_websocket.so load => chan_local.so load => codec_ulaw.so load => codec_alaw.so load => pbx_config.so load => chan_sip.so load => app_dial.so load => func_callerid.so load => func_cut.so load => func_logic.so [global] Since Asterisk 16 (Debian Buster version) I have a dependency problem, where res_rtp_asterisk.so is dependent on res_pjproject.so: When I try to make a call: [Dec 22 22:00:55] ERROR[6093][C-00000001]: rtp_engine.c:474 ast_rtp_instance_...

...text for text frames (Reported by N A) * ASTERISK-30361 - json.h: Add missing ast_json_object_real_get (Reported by N A) * ASTERISK-30280 - Create capability to assign a Media Experience Score to RTP streams (Reported by George Joseph) * ASTERISK-30332 - func_callerid: Warn if invalid redirecting reason provided (Reported by N A) For a full list of changes in this release, please see the ChangeLog: https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.0 Thank you for your continued support of Asterisk! -------------- next part -----...

...text for text frames (Reported by N A) * ASTERISK-30361 - json.h: Add missing ast_json_object_real_get (Reported by N A) * ASTERISK-30280 - Create capability to assign a Media Experience Score to RTP streams (Reported by George Joseph) * ASTERISK-30332 - func_callerid: Warn if invalid redirecting reason provided (Reported by N A) For a full list of changes in this release, please see the ChangeLog: https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.0 Thank you for your continued support of Asterisk! -------------- next part -----...

...text for text frames (Reported by N A) * ASTERISK-30361 - json.h: Add missing ast_json_object_real_get (Reported by N A) * ASTERISK-30280 - Create capability to assign a Media Experience Score to RTP streams (Reported by George Joseph) * ASTERISK-30332 - func_callerid: Warn if invalid redirecting reason provided (Reported by N A) For a full list of changes in this release, please see the ChangeLog: https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0 Thank you for your continued support of Asterisk! -------------- next part ------...

Hi, I am using: Asterisk 11.2.0 libpri 1.4.12 Dahdi: 2.6.1 Sangoma E1-Card with Wanpipe-Drivers 3.5.28 I call my asterisk box via SIP and connect the call to an AGI-Script. Within the script I do EXEC SetCallerPres prohib or EXEC SetCallerPres prohib_not_screened But I get the following error: ast*CLI> == Using SIP RTP CoS mark 5 -- Executing [100 at sip:1]

...text for text frames (Reported by N A) * ASTERISK-30361 - json.h: Add missing ast_json_object_real_get (Reported by N A) * ASTERISK-30280 - Create capability to assign a Media Experience Score to RTP streams (Reported by George Joseph) * ASTERISK-30332 - func_callerid: Warn if invalid redirecting reason provided (Reported by N A) For a full list of changes in this release, please see the ChangeLog: https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0 Thank you for your continued support of Asterisk! -------------- next part ------...

...eading symbols from /opt/asterisk/lib/modules/func_base64.so...done. Loaded symbols for /opt/asterisk/lib/modules/func_base64.so Reading symbols from /opt/asterisk/lib/modules/func_cdr.so...done. Loaded symbols for /opt/asterisk/lib/modules/func_cdr.so Reading symbols from /opt/asterisk/lib/modules/func_callerid.so...done. Loaded symbols for /opt/asterisk/lib/modules/func_callerid.so Reading symbols from /opt/asterisk/lib/modules/func_channel.so...done. Loaded symbols for /opt/asterisk/lib/modules/func_channel.so Reading symbols from /opt/asterisk/lib/modules/func_cut.so...done. Loaded symbols for /opt/ast...

.../opt/asterisk/lib/modules/func_base64.so...done. > Loaded symbols for /opt/asterisk/lib/modules/func_base64.so > Reading symbols from /opt/asterisk/lib/modules/func_cdr.so...done. > Loaded symbols for /opt/asterisk/lib/modules/func_cdr.so > Reading symbols from /opt/asterisk/lib/modules/func_callerid.so...done. > Loaded symbols for /opt/asterisk/lib/modules/func_callerid.so > Reading symbols from /opt/asterisk/lib/modules/func_channel.so...done. > Loaded symbols for /opt/asterisk/lib/modules/func_channel.so > Reading symbols from /opt/asterisk/lib/modules/func_cut.so...done. > Lo...

...- load => pbx_config.so ; Requires N/A ;load => pbx_dundi.so ; Requires res_crypto.so ;load => pbx_functions.so ; Requires N/A ;load => pbx_loopback.so ; Requires N/A ;load => pbx_realtime.so ; Requires N/A ;load => pbx_spool.so ; Requires N/A ; Functions -- ;load => func_callerid.so ; Database Call Detail Records -- load => cdr_csv.so ; Requires N/A ;load => cdr_custom.so ; Requires N/A ;load => cdr_manager.so ; Requires N/A ;load => cdr_odbc.so ; Requires N/A ;load => cdr_pgsql.so ; Requires N/A ; Channels -- ;load => chan_agent.so ; Require...

...nf': Found [cdr_custom.so] => (Customizable Comma Separated Values CDR Backend) == Parsing '/etc/asterisk/cdr_custom.conf': Not found (No such file or directory) Jun 14 13:48:05 WARNING[15142]: cdr_custom.c:85 load_config: Failed to load configuration file. Module not activated. [func_callerid.so] => (Caller ID related dialplan function) == Registered custom function CALLERID == Manager registered action DBGet == Manager registered action DBPut == Parsing '/etc/asterisk/enum.conf': Found == Parsing '/etc/asterisk/dnsmgr.conf': Not found (No such file or direc...

I have been doing a lot of reading forums and elsewhere but am somehow unable to connect the dots. Here is what I am trying to accomplish initially and then wish for it to grow bigger from here on. I have two POTS (Analog) line that would connect to the Asterisk Box. I have, to begin with 5 IP phones (PoE), all connected to a switch. Asterisk Box with a LAN card also connects to the same switch.

...c/asterisk-1.4.20.1/cdr' for x in cdr_csv.so cdr_custom.so cdr_manager.so; do /usr/bin/install -c -m 755 $x /usr/lib/asterisk/modules ; done make[1]: Leaving directory `/usr/src/asterisk-1.4.20.1/cdr' make[1]: Entering directory `/usr/src/asterisk-1.4.20.1/funcs' for x in func_base64.so func_callerid.so func_cdr.so func_channel.so func_cut.so func_db.so func_enum.so func_env.so func_global.so func_groupcount.so func_language.so func_logic.so func_math.so func_md5.so func_moh.so func_rand.so func_realtime.so func_sha1.so func_strings.so func_timeout.so func_uri.so; do /usr/bin/install -c -m 755...