Although I haven't tried this for this particular example, instead of using a .call file, you could probably originate a call using Ryan Bullock's Asterisk::AMI PERL module http://search.cpan.org/~greenbean/Asterisk-AMI-v0.2.8. It's one of the most valuable tools that I have and I've written literally hundreds of PERL scripts using it. You should check it out. It's got good documentation and examples to go along with it. I also use the AGISpeedy FastAGI package written in PERL and there's also an AGISpeedy package written in php that is also a valuable tool. Regards; John
-----Original Message----- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matthew Jordan Sent: Tuesday, January 28, 2014 12:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] [HELP]: Auto-answering calls placed from call files On Tue, Jan 28, 2014 at 10:56 AM, Steve McCann <srmcc...@gmail.com> wrote: > Hello All, > > I've asked this on the asterisk-dev list, so sorry for cross-posting. > So far I'm not sure how to accomplish this without looking at the > source code or looking at some other way to get around this issue. > > > I'm trying to have an automated call to an Aastra SIP phone and have > the call auto-answeredby the phone. I know that a SIP call placed to > the phone can be auto-answered if a certain SIP header is added to the > call. I am able to apply the SIP headers manually and get that working > (using > SIPAddHeader(Alert-Info: info=alert-autoanswer) in the dialplan, but > for call files, I don't seem to be able to edit any of the sip headers > - there is only basic customizations allowed to setup the calls. > > Does anyone know how I could place automated outgoing calls that would > have the proper sip headers added to it that would allow the call to > be auto-answered? > > I've also posted this question to the forums here: > http://forums.asterisk.org/viewtopic.php?f=1&t=89190 > > Many thanks, > Steve > This isn't a development question, as it doesn't relate to the actual Asterisk source code itself. Cross-posting across the -dev and -users lists isn't helpful either, as pretty much everyone who is subscribed to the asterisk-dev list is also subscribed to the asterisk-users list. As SIPAddHeader is a dialplan application and not a dialplan function, it cannot be used from a call file. One approach to performing an outbound call that requires SIPAddHeader - and that doesn't rely on undocumented behaviour - is to use the call file to create a Local channel in the dialplan that dials the SIP channel, and use SIPAddHeader from there. A quick Google indicates others have used a similar approach in the past as well [1]. [1] http://lists.digium.com/pipermail/asterisk-users/2008-January/204375.html Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com & http://asterisk.org -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
