[Asterisk-Dev] Interface names (in AddQueueMember)
Santosh Prasad
sprasad at hubris.net
Wed Aug 13 10:42:03 MST 2003
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
More information about the asterisk-dev
mailing list