It's a bit late now, but we missed updating this validation logic for the new CLI arguments. I've committed that to branches/1.4, FWIW.
On 2009-03-30 09:44:16, Mark Michelson wrote:
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(Updated 2009-03-30 09:44:16)
Review request for Asterisk Developers.
This is a backport of functionality that is already in all releases of Asterisk 1.6.
The functionality being added here is the ability to associate an interface with a queue member which is used to determine that member's device state.
A common usage for such functionality is to accurately determine the device state of a queue member who is called via the Agent or Local channel
technologies. By specifying a more "concrete" interface to watch for device state, the queue application can more accurately decide the availability
of the queue member.
The policy for Asterisk 1.4 is to not introduce new functionality, and only make bug fixes. This case is one where the line has been blurred significantly
and so community feedback would be helpful. From a development standpoint, this violates the policy since most of what is being added is a new option for
queue members. From many users' perspectives, however, this is a means of solving a deficiency in the basic operations of queues. Please take a look at the
diff attached and comment on whether this would be problematic. It is implemented in such a way that any old dialplans, CLI commands, manager commands, etc.
will not need to be updated unless you are making use of this new functionality.
In internal discussions, the only point that was brought up was a possible problem that could aris if using persistent members. If you upgraded to a version
that had the state interface functionality in it and then downgraded to a version which did not have state interface functionality, then the queue members'
names loaded from the database would be wrong (they would have a semicolon and the state interface appended). Since the members' names are only used for display
purposes, since the error may be easily corrected by reloading the queue configuration,and since the number of people likely to be affected by this is very
small, I feel it's a risk worth taking.
Another potential problem I have thought of is that Asterisk 1.4 does not contain func_devstate. This means that this solution will only work if
you know for sure what endpoint will be dialed when a call is placed from a queue. To give an example, if you know that calling Local/2000 will always
end up dialing SIP/2000, then this functionality will work great for you. If calling Local/2000 may end up dialing SIP/2000 or potentially SIP/3000 based
on some dialplan logic, then this new functionality will not help you since you cannot use custom device states in the dialplan.
In the context "internal," extension 1000 is as follows:
exten => 1000,1,Dial(SIP/1000)
If it is not clear, All calls to Agent/1000 will eventually go to SIP/1000. I added
a line to my queues.conf file for my queue:
member => Agent/1000,,,SIP/1000
When I place an outbound call from SIP/1000, issuing the "queue show" command shows that
Agent/1000 is "in use." Attempting to place a call to the queue will not cause SIP/1000 to
ring since I have ringinuse=no for my queue. When SIP/1000 hangs up, the status returns to
"not in use."
As another test, I tried implementing a common problem. In my queues.conf file, I added this
to my queue:
member => Local/1000,,,SIP/1000
As you may be able to guess, Local/1000 always dials SIP/1000. I placed a call to my queue
and answered with SIP/1000. At this point, issuing a "queue show" command showed that the
member Local/1000 was "in use." Then, from SIP/1000 I transferred the call to another endpoint.
Issuing a "queue show" command then showed Local/1000 as being "not in use."
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