[asterisk-dev] Overlapped dialling on SIP trunks for outgoing calls

[Klaus Darilion]

Tilghman Lesher wrote:
> On Monday 28 April 2008 15:08, Klaus Darilion wrote:
>> How do SIP phones with overlap dialing handle the following scenario:
>> 1. User presses "1"
>> 2. SIP phone sends INVITE sip:1 at ....
>> 3. user presses "2"
>> 4. what happens now? does the phone CANCEL sip:1 at ... and INVITE sip:12 at ..?
> 
> That depends on 1) what's in your dialplan, and 2) how the remote end
> responds.  If the remote end sends a 484, trunk will now cause Dial to exit
> with status AST_PBX_INCOMPLETE, which will allow the dialplan to accept
> additional digits.

Sorry - I have to make myself more clear. I am interested in the phone's 
behavior.

Consider the scenario where Asterisk itself does not know if the number 
is complete but forwards the number upstream (SIP or ZAP or whatelse).

This might look like:

phone              Asterisk              PSTN
---INVITE 1---------->
<--------100----------
                       ---INVITE 1---------->
                       <------484/Cause 28---
<--------484----------
---------ACK--------->

---INVITE 12--------->
<--------100----------
...

In the previous example it is rather clear. But what if the answer from 
the PSTN takes longer as the user dials? E.g.


---INVITE 1---------->
<--------100----------
                       ---INVITE 1---------->
                       (waiting for response)
(now user pressed digit 2)

How will a SIP phone which support overlap dialing behave now? Will it 
CANCEL 1 and INVITE 12 or will it wait for an 200/484 response bevor it 
decides if it will send a new INVITE?

Thanks
Klaus