search for: callfile

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() i...

...ayedRetry: 20354 0 (1400072406) nope.... DelayedRetry: 20354 0 (1400072706) waiting for someone.... DelayedRetry: 20354 0 (1400073006) to do manual work Asterisk log: [May 14 14:35:06] DEBUG[20421] pbx_spool.c: Delaying retry since we're currently running '/var/spool/asterisk/outgoing/callfile' [May 14 14:40:06] DEBUG[20421] pbx_spool.c: Delaying retry since we're currently running '/var/spool/asterisk/outgoing/callfile' [May 14 14:45:06] DEBUG[20421] pbx_spool.c: Delaying retry since we're currently running '/var/spool/asterisk/outgoing/callfile' [May 14 14:5...

...e able to attach an > > additional piece of information along with the call record There is a > > userfield in the SQL database, which is a VARCHAR(255) and would be > > plenty for what we need. Is there a way to set the userfield of the CDR > > database from within such a callfile? > > Yes, adding a Set field in your call file (see > http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out), you'll be able > to pass everything you need to your dialplan, and then, from there, write > everything you need to your CDR. I've got it working now! Thanks Ol...

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/aste...

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: 3...

...ell. However, it would be nice to be able to attach an additional piece of information along with the call record There is a userfield in the SQL database, which is a VARCHAR(255) and would be plenty for what we need. Is there a way to set the userfield of the CDR database from within such a callfile? -- AJS

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...

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

...ogic? -Justin -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Joshua Colp Sent: Tuesday, February 17, 2015 4:07 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Callfile problem - Unable to find codec translation path from (nothing) 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...

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) > > [2...

...t; _X.,n,Dial(DAHDI/g51/w${EXTEN},15,r) exten => _X.,n,Hangup and here is a sample call file: Channel: Local/5551212 at mis-phone/n MaxRetries: 50 RetryTime: 5 WaitTime: 5 Archive: yes Extension: ernestine ip monitor failure for dozer ping Context: outbound-swift 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 16:56:02] WARNING[9...

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 =>

...tinations" and "custom applications" such that users can dial a code from their desks and enable or disable forwarding via the above contexts. That works fine. Now I whipped up a C program to create a call file to do the same thing from the command line: [snip] fprintf(callfile, "Channel: Local/*71 at custom-callfwd/n\n"); fprintf(callfile, "Application: Playback\n"); fprintf(callfile, "Data: hello-world\n"); [snip] When I run this it creates the call file and I see this in the console: -- Attempting call on Local/*71 at custom-cal...

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 hang up, the exit...

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 w...

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