search for: callfiles

Good afternoon, I'm trying to write a simple callback context, but i need to hangup an incoming call and then call the origin number back, the problem is that asterisk stops processing the call after Hangup() application then it is not able to dial the origin number back. Sorry for the grammatical erros. -------------- next part -------------- An HTML attachment was scrubbed... URL:

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...

I am using Realtime extensions as well, in case that would matter. Following problem arises from time to time, a call will successfully terminate: [May 14 14:31:41] VERBOSE[3274] pbx_realtime.c: -- Executing [t at project_init:1] Hangup("SIP/peer-2-00002f7e", "") [May 14 14:31:41] VERBOSE[3274] pbx.c: == Spawn extension (project_init, t, 1) exited non-zero on

On Monday 20 Dec 2010, Olivier wrote: > 2010/12/20 A J Stiles <asterisk_list at earthshod.co.uk> > > > Our current Asterisk 1.6.2.9 setup includes a CGI auto-dial application > > (written by someone else before me) which sets up calls by creating > > files of > > the general form > > > > Channel: SIP/$INSIDE_NUMBER > > Context: $CONTEXT >

Hi, how may I define in a callfile the CallerID presentation to be used for the requested call, eg. set it to prohibited? TIA, Bruno -------------- next part -------------- A non-text attachment was scrubbed... Name: Bruno.Voigt.vcf Type: text/x-vcard Size: 270 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20050913/fcb5c595/Bruno.Voigt.vcf

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...

Hi! I am trying to do a callfiel for autodialing but when I move the callfile to outdialing folder asterisk seems like if did the call but it doesnt. I put here my callfile and that I get when asterisk begins to do the call If anybody has idea, pls. Tell me TIA ;;----CallFile----- Channel: Zap/g1/8093908270 Callerid: 8093908270 MaxRetries: 2 RetryTime: 300 WaitTime: 45

Our current Asterisk 1.6.2.9 setup includes a CGI auto-dial application (written by someone else before me) which sets up calls by creating files of the general form Channel: SIP/$INSIDE_NUMBER Context: $CONTEXT Extension: $OUTSIDE_NUMBER Priority: 1 CallerId: $INSIDE_NUMBER in /var/spool/asterisk/outgoing/ . It works very well. However, it would be nice to be able to attach an additional

On Tuesday 22 September 2020 at 13:27:27, Joshua C. Colp wrote: > On Tue, Sep 22, 2020 at 7:37 AM Antony Stone wrote: > > Hi. > > > > (Asterisk 16.2.1) > > > > I'm using AMI Originate to initiate calls, and I'm passing some > > additional data in to the dialplan context using the Variable: > > parameter. Works fine. > > > >

Hello, I'm using Asterisk 1.6.2.0 and I like to call extension #8 from callfile. Unfortunately the #-char ist interpreted just as comment. I got a "Invalid file contents in /var/spool/asterisk/outgoing/callfile, deleting" from asterisk. When I try to escape with \ oder use quotes, I got: \#8,1 failed so falling back to exten 's' or "#8",1 failed so falling back

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

Joshua, If I'm understanding this correctly, you're saying that the Playback is failing because it isn't connected to anything on the other end, because the Dial() failed. When the channel is created on the "OutgoingSpoolFailed" extension, what context is it created in, one of the origin legs? Is there a way detect this condition in the target context ([outbound-swift]),

Justin Killen wrote: <snip> > > Whenever I try to copy this callfile into /var/spool/asterisk/outgoing/ > I get these 3 lines repeating over and over (I?m not 100% sure which > entry is first): > > [2015-02-16 16:56:02] WARNING[9737][C-0000f8a7]: channel.c:5353 > set_format: Unable to find a codec translation path from (nothing) to (slin) > > [2015-02-16

Hi, I copied a setup from an older 1.8.5 installation to an 11.15 installation, and I'm having problems getting call files to work. Here is the extension setup I'm using: [outbound-swift] exten => _[a-zA-Z].,1,Answer exten => _[a-zA-Z].,n,Playback(AAA/check_ip_failure) ;exten => _[a-zA-Z].,1,Swift("${EXTEN}") exten => _[a-zA-Z].,n,Goto(1) [mis-phone] exten =>

Hello For some reason, when dialing out through a call file and the remote line is busy, Asterisk doesn't jump to the "failed" extension in the context used by the call file: ====== call file Channel: Zap/1/5551234 Context: callbacktest Extension: start Priority: 1 MaxRetries: 1 ====== extension.conf [callbacktest] exten => start,1,NoOp(Status is ${DIALSTATUS}) exten =>

I am sure this is simple, but have been struggling. I want to create a call file that dials out a particular Dahdi channel to enable call forwarding on a POTS line. I have this in extensions.conf: [custom-callfwd] exten => s,1,Answer exten => s,n,Dial(DAHDI/4-1/*717157750) exten => s,n,Verbose(${DIALSTATUS}) exten => s,n,Hangup [custom-callfwdcanc] exten => s,1,Answer exten

I have Meetme conferencing compiled for Debian as per http://powerontech.com/freepbx-on-debian.htm . I drop a callfile in the outgoing dir, and it intitiates a call to a local extension as a channel, but the call seems to block on the Meetme() command. That extension completes the outgoing Dial(SIP) command to my phone, announcing that leg is the only member of the conference, and just waits. If I

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 wi...

Hi there, I?ve been looking to do an outbound dialer for systems alerting, etc. and have in large part followed the recipe here: http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out That and the associated pages at voip-info give a basic set of recipes for callfiles, but nowhere there or in my copy of the O?Reilly book by Meggelen, Madsen, & Smith can I find a detailed discussion of what goes into a callfile, how to get it to do things like interact with the shell (I?d like ?Press 2? in my outbound call to do something of value), etc. I?ve googled around...

Hi all, I had a chance to use some call conferences that had some very neat functionalities: - When you call you are first asked for your name - When someone joins the conference a message "<name> is now joining the conference." is played. - When someone leaves the room a message "<name> has left to conference." is played. How can I set MeetMe/Asterisk to have