[Asterisk-Dev] Interface names (in AddQueueMember)

[Brian West]

I do this like so now also.

bkw

On Wed, 13 Aug 2003, Santosh Prasad wrote:

> Hello Jordyn,
>
> I am using AddQueueMember as shown below and it works fine
> for SIP and H.323. If you are using chan_h323 for H.323 then specify the
> full ip address for "interface" in AddQueueMember.
>
> My extensions.conf is as follows:
>
> [queue]
> exten =>9000,1,Answer
> exten =>9000,2,Queue(techsupport|TtH)
> exten =>9000,3,WaitMusicOnHold(10)
> exten =>9000,4,Voicemail(u4050)
> exten =>9000,5,Playback(vm-goodbye)
>
> exten =>8000,1,AddQueueMember(techsupport|SIP/${CALLERIDNUM})
> exten =>8000,2,Playback(auth-thankyou)
> exten =>8000,3,Hangup
> exten =>8001,1,RemoveQueueMember(techsupport|SIP/${CALLERIDNUM})
> exten =>8001,2,Playback(auth-thankyou)
> exten =>8001,3,Hangup
>
> exten =>8050,1,AddQueueMember(techsupport|H323/4050 at x.x.x.x)
> exten =>8050,2,Hangup
> exten =>8051,1,RemoveQueueMember(techsupport|H323/4050 at x.x.x.x)
> exten =>8051,2,Hangup
>
> Hope this helps
>
> Best
> Santosh
>
>
> On Wed, Aug 13, 2003 at 10:15:49AM -0700, Jordyn Buchanan wrote:
> > [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.
> >
> > _______________________________________________
> > 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
>