We're using Asterisk 14.7.6, and we're able to route particular
extensions
to a corresponding automated member via SIP. Each extension has its own
specific content, and all the automated members are configured to handle
any of these extensions. We would like to achieve some scaling by putting
all the members in a single queue.
Here are some problems we've run into:
1) How should these members be declared in queues.conf? We couldn't find
much about automated members in the Asterisk Book (
http://the-asterisk-book.com/1.6/queues.conf.html ) nor the wiki (
https://wiki.asterisk.org/wiki/display/AST/Building+Queues).
For example, all of our extensions start with 773, and we're starting with
two automated members (declared in sip.conf as CB1 and CB2).
member => SIP/773X at CB1
member => SIP/773X at CB2
Is this a valid way to declare these members and the extensions they can
handle?
2) When I configure the queue that way, it looks like calls don't make it
to the automated agents. Is there a way to inspect Asterisk to see what
extension is being provided to the agents by the queue?
== Using SIP RTP CoS mark 5
> 0x7f32f8013380 -- Strict RTP learning after remote address set to:
(IP address of caller):24804
-- Executing [77314a31e9c0-6173-4b41-9938-be3325139088 at default:1]
Answer("SIP/1000-00000012", "") in new stack
> 0x7f32f8013380 -- Strict RTP switching to RTP target address (IP
address of caller):24804 as source
-- Auto fallthrough, channel 'SIP/1000-00000012' status is
'UNKNOWN'
Cheers,
David
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