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 Posted: Tue Jan 28, 2014 12:54 pm Post subject: [asterisk-users] [HELP]: Auto-answering calls placed from ca |
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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@lists.digium.com
[mailto:asterisk-users-bounces@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@gmail.com> wrote:
| Quote: | 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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