similar to: Call file structure and syntax

Displaying 20 results from an estimated 700 matches similar to: "Call file structure and syntax"

2011 Mar 03
6
[1.4] Forcing Asterisk/Zaptel to wait until callee answers?
Hello I need to write a script that will dial a list of customers and play a message. I couldn't find a way to tell Asterisk/Zaptel to wait until the callee has actually picked up the phone before proceeding with Playback(): ============ ;call made through Dial(): Doesn't proceed after off-hook/hangup [internal] exten => 8888,1,Dial(Zap/1/${IPPI}) exten => 8888,n,NoOp(We never
2012 Aug 01
2
Problem with callfile and CDR
Good afternoon list. I am experiencing a problem with the CDR and callfiles. What is happening is this: When generating a call with a callfile, everything works perfectly, but the CDR is recorded in the table when they answer the call destination. The field disposition is being recorded correctly, but the duration field is marked with the ring time and billsec is marked with 0. This just happens
2014 Jan 31
2
callfiles.call
hello list, i have created a callfiles with my asterisk 1.4.43 like: Channel: SIP/watara/06xxxxxxxx MaxRetries: 10 RetryTime: 5 WaitTime: 20 Context: mycontext Extension: s Priority: 1 extensions.conf mycontext exten => s,1,Ringing() exten => s,n,Playback(hello-world) exten => s,n,Dial(SIP/105) exten => s,n,Hangup() it works with one number how can i do in order to create a
2006 Nov 03
1
Clearing Outgoing Call Queue
I have an app that generates callfiles in the outgoing queue, which connect a channel to an AGI (Perl script) at an extension. The AGI calls the Dial command over a SIP channel. Sometimes I need to stop the outgoing calls after the requests have been made. I delete the callfiles from the outgoing directory, but there are still some calls "in the pipeline". Especially if Dials failed at
2010 Jul 06
2
Can't dial out through AMI
SIP user => Asterisk 1.6 server => SIP Trunk => external destination: works AMI script => Asterisk 1.6 server => SIP Trunk => external destination: Failed to authenticate on INVITE to '"asterisk" <sip:asterisk@(ipaddr)>;tag=alphanumeric' I?ve tried doing things like ?include => contextwithtrunk" in various places, googling, re-reading relevant
2013 Oct 31
3
Realtime Call Files
Hi all, Is there any way of originating calls in future without using call files? We have 2 servers (1 active at a time). If we use call files with modification date in future, on the 1st server and it dies and, we activate the second server but we lose the call files. I could have a cronjob on both servers and create callfiles reading execution time from database, but this involves some other
2008 May 27
2
ForkCDR
Hello, CDR fans! I'm looking at some issues brought forward over time: 12726/10668: someone wants me to revert the changes I made via bug 10668, last Sept; (that's they are messing him up. And I didn't do the change suggested in ForkCDR, for fear of lousing up folks depending on current behavior. Which probably sparked: 11721 :
2009 Jan 16
2
CDR problems -- two call legs create only one CDR. Using ForkCDR() not even working.
Hello, When I bridge an incoming and outgoing call (attempting to simulate call-forwarding) I'm only getting one CDR -- that of the outgoing call. A (PSTN) calls B (residing on Asterisk) and the Asterisk calls C (cell phone on PSTN) and bridges the call. The only CDR created is from B to C. I have even tried using Answer() and ForkCDR() to get two CDRs, but to no avail. I am starting to
2009 Sep 02
1
Skype for Asterisk callfile question
Hi list, To make outgoing calls by skype i would like to have our crm app create callfiles like we do for normal calls. If i read the instructions it says this : ---quote--- The syntax for making an outgoing call using Skype for Asterisk is as follows: Dial(Skype/[<originator>@]<destination>) ---unquote--- So i create a callfile that looks like this: --- Channel: SIP/228
2006 Jan 28
3
(Un)PauseQeueMamber usage
Does anyone have an example of hoe to use these two commands? I have read he documentation, and I am still unclear on where this command goes, as part of extensions.conf or where? If someone could post a working example it would be most helpful. Regards to all, Joe
2008 Feb 18
1
ForkCdr in 1.4.*
Hello, I'm looking for a way to restore old behaviour (before Arkadia patch #0010668) of ForkCDR application in 1.4.18 I've done some research directly in the code (cdr.c & forkcdr.c), but can't find any flag. I am just f*c*ed or do you have something to suggest ? :) Thank you for help. Mathieu
2007 May 15
2
Originate and ForkCDR()
Hi, I'm tryng to place a call through Asterisk Manager Originate Action. Since I want separate CDR for each of the two legs of the call, I'm forking CDR with ForkCDR as the first Channel has picked up. The problem is that, while the first CDR is fine, in the second one the "answer" field is always empty, "billsec" field is 0 and "disposition" field is
2011 Jun 24
3
t.38 virtual fax software?
Can anyone recommend some kind of virtual t.38 fax software? I'd like to test/debug some of the t.38 stuff, but it'd be much easier if I had a software client that could just generate the faxes from a workstation, rather than having to sit with the fax machine + t.38 ata to source faxes from. There doesn't seem to be much out there, and the stuff that's out there is kind of
2009 Mar 12
4
log to cdr each dialpan action, not only one record for each call
Hi to all. What can i do if a customer needs to log in the CDR all the dialpan actions related to a call? I mean, not only the lastapp e the lastdata but all the dialpan actions! I know that the actual CDR system store one record for each call (and for billing purposes this can be correct) but in some cases the approach needed is something similar to the queue_log. I know that exists ResetCDR
2017 Jun 26
4
Autodialer - call simultaneously to both ends
Hello List, I'm working on an autodialer project. At the moment I use the Originate application then I "throw" it to an extension where I Dial() the other party and then both legs are bridged. The problem is that the Dial() will only run after the Originate finish its bit and I have lots of wasted time or even worse, the remote party hanging the call because instead of a human
2010 Apr 05
2
spool directories and filename
Hi, Is it possible to configure Asterisk to fetch for files from the spool directory in different directories? For example, fetch voicemail files in /abc/voicemail and call files in /cde/outgoing ?. And is it possible to configure the filename that Asterisk gives to files, like voicemail files? Thanks, Ricardo
2007 Jun 12
3
CDR changes in Trunk -- Transfers, CDRs, Life, and Everything
I have created an asterisk.org blog entry: http://www.asterisk.org/node/48358 to describe what I will shortly be committing to trunk to correct the weaknesses of CDRs, that asterisk users and developers have been complaining about for quite some time. Highlights: Restructuring the code and philosophy of CDRs. Plans to eliminate the ForkCDR() application Plans to create
2010 Sep 23
1
Forking a call
Hi, Using 1.6.2.13. I'd like to know how I can force Asterisk to fork a call. To simplify things, Let's say I have an out context (for outbound calls) and an in (for inbound). If person A wants to call person B, and both are on my servers, I don`t want to send the call out. I want all this to happen internally on my server. The problem is if I use some condition to send
2009 Apr 14
4
Ignoring time spent waiting in queue in CDR
Hello, I'm working on an Asterisk configuration for a call center, and they bill based on the time spent talking to an agent, but not for any time spent waiting in a queue. The CDR information contains the entire duration of the call as billable seconds, including time spent waiting in the queue. I would like the billable seconds to only include the time spent actually talking to an agent.
2011 Mar 28
2
Dialplan help: hang up incoming call and call the number back
Hi, I'm trying to setup Asterisk so that: 1. I call a specific number that goes to a defined extension from my phone (an external line). 2. Asterisk notes my phone number (the CLID) and hangs up without picking up the call. 3. Asterisk initiates a call to my phone and prompts me for a passkey. 4. Asterisk validates the passkey and lets me enter another number (say FOO). 5. Asterisk dials FOO