[Asterisk-Dev] Interface names (in AddQueueMember)
Tjardick van der Kraan
tjardick at vanderkraan.net
Mon Aug 18 01:05:05 MST 2003
The AddQueueMember is allready fixed in CVS by mark.
follow: http://bugs.digium.com/bug_view_page.php?bug_id=0000034
to keep informed :)
Greetings,
Tj
----- Original Message -----
From: "Brian West" <brian at bkw.org>
To: <asterisk-dev at lists.digium.com>
Sent: Thursday, August 14, 2003 5:28 AM
Subject: RE: [Asterisk-Dev] Interface names (in AddQueueMember)
> We have ${CHANNEL} but that includes the -XXXX stuff. If we could have
> another var without that..it would be great... like ${DEVICE} or
> something.
>
> bkw
>
> On Wed, 13 Aug 2003, Benjamin Miller wrote:
>
> > This seems to be an important request.
> > Adding a member that is the generic version and not the specific version
> > to the structure would be _very_ valuable. There are a number of other
> > places that the "-XXXX" has to be stripped to do comparisons, such as in
> > the manager interface, etc.
> > There are times when the specific instance of the channel are needed
> > (such as call monitoring), and other where the generic reference are
> > needed, this being an excellent example.
> > Mark, would this be too difficult to do or are we thinking about this
> > the wrong way?
> > Ben
> >
> > -----Original Message-----
> > From: Jordyn Buchanan [mailto:jbuchanan at registrypro.pro]
> > Sent: Wednesday, August 13, 2003 12:16 PM
> > To: asterisk-dev at lists.digium.com
> > Subject: [Asterisk-Dev] Interface names (in AddQueueMember)
> >
> >
> > [My first attempts to send this to the list seems not to have worked; I
> > apologize for any duplicates.]
> >
> > Hello:
> >
> > I've been playing a bit with the AddQueueMember application and have
> > realized that its handling of SIP interfaces (and probably other types
> > that use a similar channel naming convention) is a bit wrong.
> >
> > Basically, if you don't specify an interface to be added to the queue,
> > it will take the name of the calling channel and add that to the queue.
> > In the case of a SIP channel, this is something like "SIP/blah-XXXX"
> > where XXXX is some random string to differentiate between various calls
> > to/from the same entity. Adding this whole thing
> >
> > I've put together a fix to this that looks for SIP interfaces and
> > truncates the name beginning with the final "-". This works fine, but
> > won't work for other types of channels that use similar naming
> > conventions unless I manually add similar logic for each of them.
> >
> > So, finally, to a question: is there a generalized way to derive the
> > callable name of a given channel other than stripping off some of the
> > text, as I've been doing? I was hoping that the ast_channel structure
> > would contain something useful, but I don' t see anything that will
> > obviously help me out.
> >
> > Jordyn
> > _______________________________________________
> > Asterisk-Dev mailing list
> > Asterisk-Dev at lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-dev
> > _______________________________________________
> > Asterisk-Dev mailing list
> > Asterisk-Dev at lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-dev
> >
> _______________________________________________
> Asterisk-Dev mailing list
> Asterisk-Dev at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-dev
>
>
More information about the asterisk-dev
mailing list