[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
