Thank you for the comments! On 170801 1119, Richard Mudgett wrote:
On Thu, Jul 27, 2017 at 10:31 PM, Kirill Katsnelson <k...@smartaction.com> wrote: The variable approach that you have implemented in the gerrit review ( https://gerrit.asterisk.org/#/c/6118/ ) is a good way and certainly the easier way. You are able to build on existing functionality rather than having to create a lot of support infrastructure if you tried to go the dialplan function route. Another thought arguing in favor of channel variables is that the channel being blind transferred by the REFER is not necessarily a SIP channel.
This is why I tried to keep variable and hash names non-specific to channel or message--I am rather thinking of "some additional data sent by the transferrer". So the pjsip channel can certainly share the same names. I do not know enough of other channels with similar concept applicable, but I believe it can represent most additional data, whatever it is.
As to documentation, you would need to add a section in sip.conf.sample to describe the functionality as it is currently chan_sip specific. If you add support for the functionality in chan_pjsip/res_pjsip then it would need documentation in the pjsip.conf.sample file.
We are not yet on pjsip; we have been developing on chan_sip specifics for a very long time. There are dependencies on AMI events and actions, specific documented behaviors, and probably even some undocumented ("it just worked"), you name it. So it's in the plans, but we are not there yet. I'll certainly add the same functionality to pjsip when we are migrating, we depend on it now.
I suggest using an asterisk '*' as the match any header value instead so an empty or non-existing variable can mean the feature is disabled.
Certainly, a very good point about unassigning the variable, thanks! -kkm -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
