Hi,
we’ve got the following configuration on our Asterisk:
Several extensions that put the call into a queue with a different queue per extension. The queues have one static member each, which is the SIP address of another Asterisk (running 1.6).
When calls come in on an extension, only the first call causes the queue to call the queue member. All following calls are queued without a call going out to the queue member. When SIP debugging is turned on, you can see that only for the first call an INVITE is sent to the queue member, no INVITEs are sent on the subsequent calls. When doing a “module reload”, the next call (or the currently queued call) comes through, all subsequent calls don’t. The behavior can be observed for each queue: you can call all queues in a row several times and for the first call on the queue the member will be called for all following calls the member won’t be called. Doing a “SIP show channels” shows no active channels after the first call was hung up.
I had this behavior consistently on two different computers after updating my Asterisk from 1.4.24 to 1.4.35. After updating both to 1.6.2, i only got the behavior on one of them.
Can anybody give me an explanation or further troubleshooting steps? Is this a bug in Asterisk?
This is the queue configuration:
[Star_Test_20]
maxlen=0
reportholdtime=no
periodic-announce-frequency=60
strategy=ringall
joinempty=yes
announce-round-seconds=10
retry=5
announce-holdtime=no
announce-frequency=60
timeout=10000
music=default
autofill=yes
ringinuse=yes
member => SIP/test_delti_20@StarThis is the SIP configuration for the static member:
[Star]
qualify=yes
nat=no
host=xx.yy.zz
callerid=myId
dtmfmode=rfc2833
context=from-outside
type=friend
insecure=very
canreinvite=no
disallow=all
allow=g729