[asterisk-dev] PJSIP Dialog-Info+XML enhancement

[Hans-Peter Jansen]

On Mittwoch, 14. November 2018 12:26:06 Joshua C. Colp wrote:
> On Wed, Nov 14, 2018, at 12:01 PM, Hans-Peter Jansen wrote:
> > 
> > Remote is still wrong, it's a local extension, and I also have no idea
> > ATM,
> > where to fetch call-id, local-tag and remote-tag attributes. It also makes
> > asterisk not to exit gracefully anymore after hitting ^C.
> > 
> > May I kindly ask you to take a look at it?
> 
> The chan_sip module appears to use the information of the underlying
> subscription for the call-id and tags. This can be retrieved using the
> ast_sip_subscription_get_dialog function to get the dialog, and then
> accessed on it[1]. Specifically call_id, local, and remote.

Great, thanks, will look into this in the next few days...

> Can you clarify
> what you mean by "Remote is still wrong".

The remote entry:  
  <remote>
   <identity display>sip:000413414123 at 192.168.23.2</identity>
   <target uri="sip:000413414123 at 192.168.23.2" />
  </remote>
should look like:
  <remote>
   <identity display="">sip:012345678900 at 192.168.23.2</identity>
   <target uri="sip:012345678900 at 192.168.23.2"/>
  </remote>

IOW, it should display the caller, not the sip address of the callee (which 
isn't immediately obvious indeed, sorry for being too sparse here..). 000413 
is the Snom manufacturer ethercode...

What the code needs is fetching the caller from the callee. The sip code does 
something similar to what I'm doing here:

	char *remote = ast_strdupa(state_data->remote);

but this seems to point to the callee (ringing device) for some reason.

> As well on a cursory glance I
> don't see anything that would explain the ^C problem, but it's possible
> there is a memory issue within it that I don't see immediately.

Let's get the dialog working, this can come thereafter.

Thanks, Joshua, for your continuous support. Much appreciated.

Cheers,
Pete