On Tue, Oct 30, 2018 at 2:02 PM Stephen Davies <stephen.l.dav...@gmail.com>
wrote:
> Hi,
>
> I'm looking to see if I've missed something - if not I think there is an
> ARI limitation worth thinking about.
>
> app_dial offers a flag "I" which is supposed to block propagation of
> connected line updates through the dial bridge. Is there anything like
> this that I can do when setting up and managing a bridge with ARI?
>
> My problem is like so: I have an incoming call from an "external" source
> to a DDI. I want to send that source a Remote-Party-Id containing the DDI
> dialled an a name like "ABC Corp". And I don't want that to change even
> with subsequent transfers etc.
>
> I've got this working for the initial incoming call. My problem comes
> when the dialled phone does a transfer - specifically I've need testing
> with attended transfer.
>
> (An attended transfer starts with the transferer initiating an "enquiry
> call". So two bridges. On the REFER Asterisk does an attended transfer
> with type "link" and inserts a Local channel pair between the two bridges,
> replacing the two channels to the transferring phone.
>
> At this point Asterisk sends the original caller a connected line update
> saying that "you are now talking to 201 Abu". And tells the target phone
> "you are now talking to the external caller".
>
> Which is true, no doubt, but in this case I want to block the connected
> line update being sent to the original caller.
>
> Is there any way in the ARI API? I can't find it.
>
> In related news, I tried to "isolate" my external caller by passing the
> incoming call through a Local channel with the I option on the dial. To
> create a point where the connected line update with be blocked. My
> reasoning is then that the Attended transfer can do what it wants to the
> channels in the bridge, but the flag in the dial will isolate the original
> calling channel My ARI app can directly tinker with the channel if I do
> want to adjust the connected line info - but its 100% under my control.
>
> So I made my dialplan do this:
>
> [Oct 30 17:51:21] VERBOSE[20289][C-00000015] pbx.c: Executing
> [2710900XXXX@from-drachtio:1] Goto("PJSIP/drachtio-sip1-00000045",
> "to-stasis-ariapp,2710900XXXX,1") in new stack
> [Oct 30 17:51:21] VERBOSE[20289][C-00000015] pbx_builtins.c: Goto
> (to-stasis-ariapp,2710900XXXX,1)
> [Oct 30 17:51:21] VERBOSE[20289][C-00000015] pbx.c: Executing
> [2710900XXXX@to-stasis-ariapp:1] Set("PJSIP/drachtio-sip1-00000045",
> "app=ariapp") in new stack
> [Oct 30 17:51:21] VERBOSE[20289][C-00000015] pbx.c: Executing
> [2710900XXXX@to-stasis-ariapp:2] ExecIf("PJSIP/drachtio-sip1-00000045",
> "0?Set(app=ariapp)") in new stack
> [Oct 30 17:51:21] VERBOSE[20289][C-00000015] pbx.c: Executing
> [2710900XXXX@to-stasis-ariapp:3] Set("PJSIP/drachtio-sip1-00000045",
> "direction=inbound") in new stack
> [Oct 30 17:51:21] VERBOSE[20289][C-00000015] pbx.c: Executing
> [2710900XXXX@to-stasis-ariapp:4] GotoIf("PJSIP/drachtio-sip1-00000045",
> "1?to-stasis-ariapp-via-local,2710900XXXX,1") in new stack
> [Oct 30 17:51:21] VERBOSE[20289][C-00000015] pbx_builtins.c: Goto
> (to-stasis-ariapp-via-local,2710900XXXX,1)
> [Oct 30 17:51:21] VERBOSE[20289][C-00000015] pbx.c: Executing
> [2710900XXXX@to-stasis-ariapp-via-local:1]
> NoOp("PJSIP/drachtio-sip1-00000045", "Routing inbound call "0878XXXXXX"
> <0878XXXXXX> -> 2710900XXXX via Local channel to block connected line
> updates") in new stack
> [Oct 30 17:51:21] VERBOSE[20289][C-00000015] pbx.c: Executing
> [2710900XXXX@to-stasis-ariapp-via-local:2]
> Dial("PJSIP/drachtio-sip1-00000045",
> "Local/2710900XXXX@to-stasis-ariapp-inbound/n,,I") in new stack
> [Oct 30 17:51:21] VERBOSE[20289][C-00000015] app_dial.c: Called
> Local/2710900XXXX@to-stasis-ariapp-inbound/n
> [Oct 30 17:51:21] VERBOSE[20291][C-00000015] pbx.c: Executing
> [2710900XXXX@to-stasis-ariapp-inbound:1]
> NoOp("Local/2710900XXXX@to-stasis-ariapp-inbound-00000006;2", "Routing
> inbound call "0878XXXXXX" <0878XXXXXX> -> 2710900XXXX other side of local
> channel") in new stack
> [Oct 30 17:51:21] VERBOSE[20291][C-00000015] pbx.c: Executing
> [2710900XXXX@to-stasis-ariapp-inbound:2]
> Stasis("Local/2710900XXXX@to-stasis-ariapp-inbound-00000006;2",
> "ariapp,"0878XXXXXX <0878XXXXXX>,2710900XXXX,inbound") in new stack
>
> So what's interesting about this is that is does prevent connected line
> updates during the initial setup of the call:
>
> [Oct 30 17:51:24] VERBOSE[20289][C-00000015] app_dial.c: Connected line
> update to PJSIP/drachtio-sip1-00000045 prevented.
>
> However the attended transfer still results in a connected-line update
> being sent put to the calling channel through the dial into the Local
> channel despite the I flag.
>
> My Asterisk is "Asterisk GIT-master-9189c26M" - which is a master branch
> from shortly before 16 was released.
>
> Thanks for any pointers.
>
Look at
https://gerrit.asterisk.org/#/c/asterisk/+/10286/
which has been recently committed to the current branches and
https://gerrit.asterisk.org/#/c/asterisk/+/10480/
which is nearly ready to get merged.
The other thing is to always use the connected line interception routine to
manipulate the connected line information from what you are currently
connected with to how you need it.
https://wiki.asterisk.org/wiki/display/AST/Manipulating+Party+ID+Information
https://wiki.asterisk.org/wiki/display/AST/Party+ID+Interception+Macros+and+Routines
For the interception routine you can use the CONNECTEDLINE(tag) value to
identify endpoint membership such as tenants.
Note that there was a recent fix to the tag value in chan_pjsip:
https://gerrit.asterisk.org/#/c/asterisk/+/10476/
Richard
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