[asterisk-users] Issue when reloading
module 'cdr_odbc' (ODBC CDR Backend) -- Reloading module 'cdr_manager' (Asterisk Manager Interface CDR Backend) -- Reloading module 'cdr_custom' (Customizable Comma Separated Values CDR Backend) -- Reloading module 'cdr_csv' (Comma Separated Values CDR Backend) -- Reloading module 'cdr_adaptive_odbc' (Adaptive ODBC CDR backend)* == Parsing '/etc/asterisk/cdr_adaptive_odbc.conf': == Found -- Reloading module 'app_voicemail' (Comedian Mail (Voicemail System) with ODBC Storage) -- Reloading module 'app_rpt' (Radio Repeater/Remote Base Application) -- Reloading module 'app_queue' (True Call Queueing) [Feb 2 08:19:41] NOTICE[1625]: app_queue.c:5660 reload_queue_rules: queuerules.conf has not changed since it was last loaded. Not taking any action. * == Parsing '/etc/asterisk/queues.conf': == Found == Parsing '/etc/asterisk/queues_general_additional.conf': == Found == Parsing '/etc/asterisk/queues_custom_general.conf': == Found == Parsing '/etc/asterisk/queues_custom.conf': == Found == Parsing '/etc/asterisk/queues_additional.conf': == Found == Parsing '/etc/asterisk/queues_post_custom.conf': == Found* -- Reloading module 'app_playback' (Sound File Playback Application) -- Reloading module 'app_minivm' (Mini VoiceMail (A minimal Voicemail e-mail System)) -- Reloading module 'app_meetme' (MeetMe conference bridge) == Parsing '/etc/asterisk/meetme.conf': == Found [Feb 2 08:19:41] NOTICE[1625]: app_meetme.c:6427 load_config: A reload of the SLA configuration has been requested and will be completed when the system is idle. -- Reloading module 'app_followme' (Find-Me/Follow-Me Application) -- Reloading module 'app_amd' (Answering Machine Detection Application) -- Reloading module 'res_config_mysql.so' (MySQL RealTime Configuration Driver) == MySQL RealTime reloaded. -- Reloading module 'cdr_addon_mysql.so' (MySQL CDR Backend) -- Remote UNIX connection -- Remote UNIX connection disconnected -- Remote UNIX connection -- Remote UNIX connection disconnected I´m with 1.6.2.1 with the original .conf, I´ve made only few changes. What could be happening? If I send a reload without editing any .conf file it doesnt reload chan_agent for example, so my call center gets crazy. Thank you very much!! and sorry about my poor english, Pablo Bernasconi -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Best way to detect fax in Asterisk 1.6??
Hello, I´ve found information about NVFax, app_fax, NVBackgroundDetect, rxfax, etc But which is the best way for *detecting fax in Asterisk 1.6*??? I will use it in an automatic dialer. Thank you very much, Pablo Bernasconi ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to send a digit to a channel??
Hello! I need to send a digit to a channel of an established call, from outside of Asterisk, I suppose it must be from the AMI. I want to send a * for example, but in addition to reproducing the sound of that digit (I dont care thatl), I need that the digit sent actually performs an action. For example if I have configured that the attended transfer in Asterisk is #2, I need somehow to be able to send the # and the 2 for the transfer menu begins. Any help with this??? I have proved with PlayDTMF, but all it does is play the sound of the digit, but nothing happens... Please need help! Thank you very much. Pablo Bernasconi ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem sending a DTMF remotely. Please need help!!
Ivan, first of all thank you for your answer. The manager function PlayDTMF only generates sound, and the dialplan function SendDTMF only generates sound too, I´d prove it and the same result... So, how can I really send a DTMF to a channel?? and not just the audio.. Thank you very much, Pablo 2009/10/8 Ivan Stepaniuk i...@albafotonica.com Pablo Bernasconi wrote: My Asterisk version is 1.6.0.15, but I`ve tried it in 1.6.0.6 and 1.6.1.6 version and the same happens. I dont know what I am missing... Please help me. Pablo, I did not answer in the first place because I am not completely sure, but just guessing, PlayDTMF just generates DTMF sounds, inband, and not info/rfc2833 messages, that's probably why you hear the tones but nothing happens. Just my two cents, perhaps it throws some light. -- Iván Stepaniuk Alba Fotónica S.L. www.albafotonica.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem sending a DTMF remotely. Please need help!!!
,reporting,originate,all My Asterisk version is 1.6.0.15, but I`ve tried it in 1.6.0.6 and 1.6.1.6 version and the same happens. I dont know what I am missing... Please help me. Thank you very much. Pablo Bernasconi ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem sending a DTMF remotely. Please need help!!
,originate,all My Asterisk version is 1.6.0.15, but I`ve tried it in 1.6.0.6 and 1.6.1.6 version and the same happens. I dont know what I am missing... Please help me. Thank you very much. Pablo Bernasconi ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sending a DTMF remotely with PlayDTMF problem.
Hello, I need to be able to send a DTMF to an existing channel remotely. So I made a php script to do such with the Manager command PlayDTMF. I need it for example to start a transfer. isb177*CLI features show Builtin Feature Default Current --- --- --- Pickup*8 *8 Blind Transfer# #8 Attended Transfer *2 One Touch Monitor *1 Disconnect Call * ** Park Call One Touch MixMonitor Dynamic Feature Default Current --- --- --- (none) Call parking Parking extension : 70 Parking context : parkedcalls Parked call extensions: 71-78 My script is: #!/usr/bin/php -q ?php error_reporting (E_ALL); set_time_limit(60); ob_implicit_flush(false); $ip_asterisk = 127.0.0.1; $user_asterisk = admin; $pass_asterisk = forward; $canal = SIP/1000-0a292360; //hardcodeado $oSocket = fsockopen($ip_asterisk, 5038, $errnum, $errdesc) or die(Connection to host failed); fputs($oSocket, Action: login\r\n); fputs($oSocket, Username: $user_asterisk\r\n); fputs($oSocket, Secret: $pass_asterisk\r\n\r\n); fputs($oSocket, Action: PlayDTMF\r\n); fputs($oSocket, Channel: $canal\r\n); fputs($oSocket, Digit: #\r\n\r\n); usleep(50); fputs($oSocket, Action: PlayDTMF\r\n); fputs($oSocket, Channel: $canal\r\n); fputs($oSocket, Digit: 8\r\n\r\n); usleep(50); fputs($oSocket, Action: Logoff\r\n\r\n); // Carga toda la respuesta recibida en un string $loaded = ; while (!feof($oSocket)){ $buffer = fgets($oSocket, 4096); $loaded .= $buffer; } $vec = explode(\n, $loaded); $len = count($vec); print_r($vec); ? The script output is: Array ( [0] = Asterisk Call Manager/1.1 [1] = Response: Success [2] = Message: Authentication accepted [3] = [4] = Response: Success [5] = Message: DTMF successfully queued [6] = [7] = Response: Success [8] = Message: DTMF successfully queued [9] = [10] = Response: Goodbye [11] = Message: Thanks for all the fish. [12] = [13] = ) When I run the script I can hear the two digit (only the audio) but nothing happens, the Transfer menu doesnt start. The Cli shows: [Oct 2 11:14:46] DEBUG[30054]: manager.c:2776 process_message: Manager received command 'login' == Manager 'admin' logged on from 127.0.0.1 [Oct 2 11:14:46] DEBUG[30054]: manager.c:2776 process_message: Manager received command 'PlayDTMF' [Oct 2 11:14:46] DEBUG[30054]: channel.c:2055 ast_waitfor_nandfds: Thread -1216881776 Blocking 'SIP/1000-0a292360', already blocked by thread -1217414256 in procedure ast_waitfor_nandfds [Oct 2 11:14:47] DEBUG[29533]: channel.c:3341 ast_write: Deadlock avoided for write to channel 'SIP/1000-0a292360' [Oct 2 11:14:47] DEBUG[30054]: manager.c:2776 process_message: Manager received command 'PlayDTMF' [Oct 2 11:14:47] DEBUG[30054]: channel.c:2055 ast_waitfor_nandfds: Thread -1216881776 Blocking 'SIP/1000-0a292360', already blocked by thread -1217414256 in procedure ast_waitfor_nandfds [Oct 2 11:14:47] DEBUG[29533]: channel.c:3341 ast_write: Deadlock avoided for write to channel 'SIP/1000-0a292360' [Oct 2 11:14:47] DEBUG[30054]: manager.c:2776 process_message: Manager received command 'Logoff' == Manager 'admin' logged off from 127.0.0.1 BUT, if I press #8 in the softphone, I can hear the two digit and inmediately the Transfer menu begins playing 'pbx-transfer.gsm'. And the Cli output in this case is: [Oct 2 11:09:17] DEBUG[29533]: rtp.c:1148 ast_rtcp_read: Got RTCP report of 60 bytes [Oct 2 11:09:20] DEBUG[29533]: rtp.c:958 process_rfc2833: - RTP 2833 Event: 000b (len = 4) [Oct 2 11:09:20] DEBUG[29533]: rtp.c:806 send_dtmf: Sending dtmf: 35 (#), at 192.168.0.148 [Oct 2 11:09:20] DTMF[29533]: channel.c:2840 __ast_read: DTMF begin '#' received on SIP/1000-0a292360 [Oct 2 11:09:20] DTMF[29533]: channel.c:2850 __ast_read: DTMF begin passthrough '#' on SIP/1000-0a292360 [Oct 2 11:09:20] DEBUG[29533]: channel.c:4806 ast_generic_bridge: Got DTMF begin on channel (SIP/1000-0a292360) [Oct 2 11:09:20] DEBUG[29533]: channel.c:5150 ast_channel_bridge: Bridge stops bridging channels SIP/1000-0a292360 and SIP/1001-0a026408 [Oct 2 11:09:20] DEBUG[29533]: rtp.c:958 process_rfc2833: - RTP 2833 Event: 000b (len = 4) [Oct 2 11:09:20] DEBUG[29533]: rtp.c:958 process_rfc2833: - RTP 2833 Event: 000b (len = 4) [Oct 2 11:09:20] DEBUG[29533]: rtp.c:958 process_rfc2833: - RTP 2833 Event: 000b (len = 4) [Oct 2 11:09:20] DEBUG[29533]: rtp.c:958 process_rfc2833: - RTP 2833 Event: 000b (len = 4) [Oct 2 11:09:20] DEBUG[29533]: rtp.c:806 send_dtmf: Sending dtmf: 35 (#), at 192.168.0.148 [Oct 2 11:09:20] DTMF[29533]: channel.c:2768 __ast_read: DTMF end '#' received on SIP/1000-0a292360, duration 80 ms [Oct 2 11:09:20] DTMF[29533]: channel.c:2808 __ast_read: DTMF end
[asterisk-users] Problem sending a DTMF remotely. Please need help...
,originate,all My Asterisk version is 1.6.0.15, but I`ve tried it in 1.6.0.6 and 1.6.1.6 version and the same happens. I dont know what I am missing... Please help me. Thank you very much. Pablo Bernasconi ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Flash Operator Panel with Asterisk 1.6
Hi, I know this is not a 100% Asterisk question, but is there anyone who has the Flash Operator Panel working with Asterisk 1.6?? In asternic.org there is a version that show call status but you cant make transfers or originate a call. Has anyone fixed the op_server.pl file to fully work with Asterisk 1.6??? Thank you very much, Pablo Bernasconi ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk - Nortel integration via SIP protocol
Hi, I need to integrate my Asterisk with a Nortel Meridian 11, but I can´t use PRI, Analog lines, etc. It has to be via SIP protocol, and there is few information about this type of integration. Could someone please help me?? Thanks, Pablo ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users