[asterisk-dev] ARI - Add Support for custom SIP Headers with Originate

[Olle E. Johansson]

On 09 Mar 2015, at 08:54, Nir Simionovich <nir.simionovich at gmail.com> wrote:

> Cool, too bad it isn't documented. I'll add it into PHPARI as well.
> 


I believe that this was one of the reasons we created the support for _VARIABLE and __VARIABLE
in Asterisk. The ability to reach over to the new channel from the old channel was required
for a number of things we wanted to do.

The code is very old and at the time it was seem as a bad hack to use the channel
variables for this.  The decision was not to expose this part while trying to figure out a better
way to do it. It's still around, about ten years later. :-)

I have mentioned this a number of times on various mailing lists, so it should be known,
even though it is not documented.

Make sure that if you add variables not using the dialplan function,  you must use high numbers and 
decrement so the risk of a collision is lower.

/O


> On Mar 8, 2015 6:18 PM, "Matthew Jordan" <mjordan at digium.com> wrote:
> 
> On Sun, Mar 8, 2015 at 10:51 AM, Nir Simionovich <nir.simionovich at gmail.com> wrote:
> Ok, I'll have a look into that one.
> 
> On Sun, Mar 8, 2015 at 1:03 PM, Olle E. Johansson <oej at edvina.net> wrote:
> 
> On 08 Mar 2015, at 09:52, Nir Simionovich <nir.simionovich at gmail.com> wrote:
> 
> > Hi All,
> >
> >   So, I've been banging my head against an issue with ARI. While Channel Originate enables
> > you to originate channels, you can't really do a "SIPAddHeader" type functionality in there.
> >
> >   Originally, I was under impression that endpoints/message should be able to give me the functionality I wanted, but it didn't.
> >
> >   So, I realized that the functionality I'm looking for doesn't really exist.
> >
> >   Question, are we missing a feature here? or is there an alternative method of achieving the
> > same functionality?
> If you can add channel variables, you can add SIP headers.
> Look at a dump of the channel after you executed SIPaddheader to figure out how it works.
> Add two headers, and run dumpchan().
> 
> You should be able to do it with just the channel variable "SIPADDHEADER", that is:
> 
> SIPADDHEADER=X-CustomHeader-1: foo
> SIPADDHEADER=X-CustomHeader-2: bar
> 
> These can be specified in the /channels operation's JSON body.
> 
> WIth chan_pjsip, headers are manipulated using a dialplan function, so there shouldn't be any issue there.
> 
> -- 
> Matthew Jordan
> Digium, Inc. | Director of Technology
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> Check us out at: http://digium.com & http://asterisk.org
> 
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