[asterisk-users] [HELP]: Auto-answering calls placed from call files

[Tech Support]

    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-bounces at lists.digium.com
[mailto:asterisk-users-bounces at 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 <srmccann at 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

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