I am doing a distributed asterisk environment with the following setup
asterisk1 = queue server with ‘toasterisk2’ as a SIP trunk to asterisk2
asterisk2 = extension register server with ‘toasterisk1’ as a SIP trunk to asterisk1
Here is my queue in asterisk1:
[2000]
announce-frequency=0
announce-holdtime=no
announce-position=no
autofill=no
eventmemberstatus=no
eventwhencalled=no
joinempty=yes
leavewhenempty=no
maxlen=0
periodic-announce-frequency=0
queue-callswaiting=silence/1
queue-thereare=silence/1
queue-youarenext=silence/1
reportholdtime=no
retry=5
ringinuse=yes
servicelevel=60
strategy=ringall
timeout=15
weight=0
wrapuptime=0
Here is my extension in asterisk2:
[100]
deny=0.0.0.0/0.0.0.0
secret=specialt007
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
type=friend
nat=yes
port=5060
qualify=yes
callgroup=
pickupgroup=
dial=SIP/100
mailbox=100@device
permit=0.0.0.0/0.0.0.0
callerid=device <100>
callcounter=yes
faxdetect=no
I am using FreePBX on asterisk1 and asterisk2.I am using tigase XMPP for my distributed device state. My problem is that I cannot figure out how to correctly log extensions from asterisk2 in and out of dynamic queues on asterisk1. If I just log the agents in with: "queue add member SIP/toasterisk2/100 to 2000 the queue works fine sends the call OK and the device state is OK, but the ‘queue show’ command always shows the extension as Unavailable. I really want to be able to add extensions with the ‘queue add member SIP/100 to 2000’. When I try to change the dial string in the extension to dial=SIP/toasterisk2/100 and add the queue member with ‘queue add member SIP/100 to 2000’ my queues will not ring the extensions at all and i’m stuck with endless music on hold. Is there any way I can set up my extensions to be added with ‘queue add member SIP/100 to 2000’ which will correctly carry the device states and queue status?