similar to: Play a file

Displaying 20 results from an estimated 1000 matches similar to: "Play a file"

2003 Oct 15
4
indications.conf
Hi, I?m trying to make * work with Brazilian analog signalling.. I?m using the following in indications.conf file... [br] description = Brasil ringcadence = 1000,4000 dial = 425 busy = 425/250,0/250 ring = 425/1000,0/4000 callwaiting = 425/60,0/250,425/60,0/5000 I changed zaptel.conf to loadzone=br #loadzone=fr #loadzone=de #loadzone=uk #loadzone=fi #loadzone=jp #loadzone=sp #loadzone=no
2004 May 13
0
MGCP channel problem
Hello I have a problem with my MGCP voice gateway. I use D-Link DG104S Boot PROM Version 3.0B38-D Firmware Version 3.0T86-D I tried asterisk v 0.7.2 and I am using latest CVS version now. When I dial a number very fast, or when I use a redial function, my asterisk receives coupled digits. My co-worker called number 245005111, these are a few lines of my debug. The identifier of first digit
2005 Jun 26
3
cdr and billing
Hello ; how can i enable billing only while using specific trunk (ex:zap) but internal sip calls will not be counted specifically how to make all outbound is counted i am using asterisk mysql cdr enabled -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050626/0faf0974/attachment.htm
2007 Sep 21
1
Authenticate() application and CDR
Dear all, I'm trying to configure Asterisk to be able to ask the caller to enter a given password in order to continue dialplan execution. I've tested this feature using the Authenticate application like this: exten => _X./5219,1,Answer exten => _X./5219,2,Authenticate(1234,a) exten => _X./5219,3,Playback(pin-number-accepted) exten => _X./5219,4,Dial(SIP/${EXTEN},120)
2007 Feb 15
7
Call forwarding
Hi All, I'm using asterisk 1.2.15 and call forwarding doesnt work for me. from my extensions.conf: ; Unconditional Call Forward exten => _*21*X.,1,NoCDR exten => _*21*X.,2,Set(DB(CFIM/${CALLERID(NUM)})=${EXTEN:4}) exten => _*21*X.,3,Playback(vm-saved) exten => _*21*X.,4,Hangup exten => #21#,1,NoCDR exten => #21#,2,DBdel(CFIM/${CALLERID(NUM)}) exten =>
2005 Mar 01
1
NoCDR Warning
Hi, When I use NoCDR application I obtain this warning in console log: Mar 1 11:16:08 WARNING[3513]: cdr.c:114 ast_cdr_free: CDR on channel 'SIP/492-7371' not posted Mar 1 11:16:08 WARNING[3513]: cdr.c:116 ast_cdr_free: CDR on channel 'SIP/492-7371' lacks end Can someone explain to me what is due? Thanks.
2020 Jul 10
2
Way to start CDR when call is bridged ?
Hi, in dialplan -Asterisk 16.2 from Debian Buster- we have  same = n,Dial(PJSIP/101&PJSIP/102&PJSIP/103,15,tT) If thew call is not answered after 20 seconds, we launch a new dial with same and/or other extensions  same = n,Dial(PJSIP/101&PJSIP/104&PJSIP/110,20,tT) Looking in CDR we have at the end of the call (here we called 3 extensions which where ringing, let say 110
2004 Jan 08
4
2nd call leg status?
Hi, okay heres what I want to do .. simple ivr, we take a call, answer it, play a menu, dial out based on options. No problems so far. The CDR always shows the call as answered as I answer the 1st leg to play the prompts, I am actually more interested in if the 2nd leg - the outbound part - has been answered or not before the call is hungup. How can I get this and record the information in
2013 Jun 24
2
Asterisk-11 loop behaviour
Arch = x86_64 OS = CentOS-6.4 (freepbx) Asterisk = 11.4.0 FreePBX = 2.11.0.2 Snom870 Handsets We are in the process of moving to an Asterisk based PBX. At the moment most things work as we wish. However, I have just notices that when I force a reload using 'amportal a reload' I see this loop start in 'asterisk -rvvvvvvvvvv': > Channel Local/s at tc-maint-000002a4;1
2007 Dec 06
1
s, CDR and NoCDR in v1.4.10.1
I am running 1.4.10.1. I have a macro that is called from default for a certain extension (both below). I added NoCDR to s to try and stop extra CDR records, but I am still getting them. Any idea how to stop them? extensions.conf: [macro-STDEXT] exten =s,1,NoCDR() exten =s,2,Dial(${ARG1},30,Tt) exten =s,3,Goto(s-${DIALSTATUS},1) exten =s-NOANSWER,1,Voicemail(${ARG2}|u) exten
2008 Mar 11
3
Call tracing - Asterisk 1.4
Hi guys I've just read this about the upcoming release of * 1.6: ?Better reporting through a new call event logging capability in Asterisk 1.6 will allow complete tracking of events that take place during a call. The goal, according to Fleming, is to provide more detail than traditional CDR (Call Detail Recording) features offer and to allow for more granular tracking and auditing.? That
2005 Jul 25
1
sendDTMF at pickup
Hi everyone: The following code dials our prefix, sends a beep, and sends a DTMF "c" tone, then dials the phone number. I need to send the DTMF only if the phone is answered. [voip] exten=>i,1,NoCDR() exten=>i,2,Hangup() exten=>s,1,Wait(2) exten=>s,2,Background(beep||) exten=>s,3,DigitTimeout(6) exten=>s,4,ResponseTimeout(10) exten=>s,5,SendDTMF(c)
2008 Apr 03
1
Sending audio to a channel
I have a voicemail application that users can listen to messages and leave messages. I am looking for a way to play a beep tone to a user when a new message is received when they are on the phone. Here is what I have come up with: in extensions.conf: [beepvoicemail] exten => 1000,1,answer() exten => 1000,2,NoCDR() exten => 1000,3,wait(2) exten => 1000,4,Set(TIMEOUT(absolute)=5)
2007 Jan 24
3
setting up AMD
I'm trying get this working. I've looked through the list, and can't see how to get AMD to print out more. I have it call and say Hello like I normally would. I've tried to say more and less doesn't seem to matter. After I hangup it does recognize hangup. Here's logging during an attempt where I make outbound call and answer, but then hangup after 1-2 seconds: Jan 24
2009 Jul 15
1
ResetCDR after GotoIf doesn't set dst correctly, Is this a bug?
(Both on Asterisk 1.2 and 1.4) I was struggling to find out why my CDR was recording dst = h after a call hangup. It was working fine until I added a GotoIf statement before ResetCDR to calculate some value for userfield column. Today I tested and found out that if ResetCDR is put after GotoIf (or after if in AEL), it doesn't record correct value in dst column, and isntead puts 'h'
2009 Jan 07
2
How to use AMD "Answering Machine Detect" ?
Hi everybody, Happy New Year ! I'm trying to detect if a call was answered by a machine (linke voicemail systems) or a human. I would like to use AMD (Answering Machine Detect) command, but with my configuration it was not possible get there. Follow my dialplan: exten => _[789].,1,NoCDR exten => _[789].,n,Dial(SIP/${EXTEN}@111,60) exten => _[789].,n,AMD
2005 Jan 28
4
FW: FAQ missing info? Asterisk@home V 0.4
Just installed V 0.4 of asterisk@home Programmed up 3 sip budgetone extensions, they call call each other fine. Tried to dial '9' for an outside line through an X100P to a packet8 ATA but got 'all circuits are busy now'. Here is the console output. == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/30-8d25' -- Executing
2009 Apr 01
2
Extract a MOS value from Asterisk CDR
Hello all, I'm tring to retrieve a formula to calculate a MOS value from Asterisk RTCP stats... Have you got any idea how to do it? Thanks I'm reading all G.107 ITU docs to retrieve something... I'm saving the SIP RTCP stats with: [macro-hangupcall] exten => s,1,Set(CDR(userfield)=${CHANNEL(rtpqos|audio|all)}) exten => s,n,ResetCDR(vw) exten => s,n,NoCDR() So I retrieve
2009 Jul 14
3
Why CDR is recording dst value = h?
For a new project, I have written a dialplan and it is pretty straight forward: The [dialout] context dials out a number, and h extension in this context writes the CDR. But what is happening is that if the callee hangs up first, all values in the CDR are fine, but if the caller hangs up first, the 'dst' column is always 'h'. I put a NoOp right in the begining of this macro to
2012 Aug 05
3
Voice Mail beep / tone detection
Though asterisk support AMD which is based on silence detection but I did not found support of tone / beep detection in asterisk to record a voice message for answering machines after detecting tone Will appreciate any help in this regard Best Regards *Tahir Almas* Managing Partner ICT Innovations http://www.ictinnovations.com Leveraging open source in ICT Unified Communication Telemarketing