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