by zecdav » Thu Jun 28, 2012 7:26 am
This is what appears in asterisk console:
== Using SIP RTP CoS mark 5
-- Executing [200@employees:1] Playback("SIP/204-00000141", "silence/1") in new stack
-- <SIP/204-00000141> Playing 'silence/1.alaw' (language 'en')
-- Executing [200@employees:2] Macro("SIP/104-00000141", "queue,200") in new stack
-- Executing [s@macro-queue:1] Queue("SIP/104-00000141", "150") in new stack
-- Started music on hold, class 'default', on SIP/104-00000141
== Using SIP RTP CoS mark 5
== Using SIP RTP CoS mark 5
-- SIP/101-00000143 connected line has changed. Saving it until answer for SIP/5204-00000141
-- SIP/101-00000142 connected line has changed. Saving it until answer for SIP/5204-00000141
-- SIP/102-00000142 is ringing
-- SIP/101-00000143 is ringing
-- SIP/101-00000143 is ringing
-- SIP/101-00000143 is ringing
-- SIP/102-00000142 connected line has changed. Saving it until answer for SIP/5204-00000141
-- SIP/102-00000142 answered SIP/104-00000141
-- Stopped music on hold on SIP/104-00000141
-- Remotely bridging SIP/104-00000141 and SIP/102-00000142
== Spawn extension (macro-queue, s, 1) exited non-zero on 'SIP/104-00000141' in macro 'queue'
== Spawn extension (employees, 101, 2) exited non-zero on 'SIP/104-00000141'
How you can see in the log above, caller 104 calls 200. The call is passed to the queue 150 (member 101 e 102).
Now both the phones correctly rings but when 102 pickup the call the other member 101 continues ringing.