Summary: | ASTERISK-03596: Call acknowledgement for a Directed Dial to a specific Agent. | ||
Reporter: | Kristopher Lalletti (kris2k) | Labels: | |
Date Opened: | 2005-02-27 12:37:02.000-0600 | Date Closed: | 2008-01-15 15:30:41.000-0600 |
Priority: | Minor | Regression? | No |
Status: | Closed/Complete | Components: | Applications/app_queue |
Versions: | Frequency of Occurrence | ||
Related Issues: | |||
Environment: | Attachments: | ||
Description: | A logged-in agent from AgentCallBackLogin receives a call directly to the agent (without going through the Queue but by dialing Agent/ID), does not have to acknowledge to find himself with a live conversation without having to press pound(#) key. I have included verbose output of the series of events, where once the call answered, sound is being exchanged, but we notice that the call states indicate otherwise. I then acknowledge the call with the pound key, sound still gets exchanged, but now when we look at the channels, their states are back to normal. ****** ADDITIONAL INFORMATION ****** == CDR updated on SIP/kris-cisco-602-1fdc -- Executing Goto("SIP/kris-cisco-602-1fdc", "prxAgentExtensions|100101|1") in new stack -- Goto (prxAgentExtensions,100101,1) -- Executing Dial("SIP/kris-cisco-602-1fdc", "Agent/100101|120") in new stack -- outgoing agentcall, to agent '100101', on 'Local/101@prxAgentDialPlan-0c82,1' -- Called 100101 -- Executing Dial("Local/101@prxAgentDialPlan-0c82,2", "SIP/kris-hit-602") in new stack -- Called kris-hit-602 -- SIP/kris-hit-602-a3c0 is ringing -- Agent/100101 is ringing -- SIP/kris-hit-602-a3c0 answered Local/101@prxAgentDialPlan-0c82,2 -- Agent/100101 stopped sounds -- Local/101@prxAgentDialPlan-0c82,1 answered, waiting for '#' to acknowledge Feb 27 13:35:46 NOTICE[5643]: channel.c:1352 ast_read: Dropping incompatible voice frame on SIP/kris-hit-602-a3c0 of format g729 since our native format has changed to ulaw == Spawn extension (prxAgentDialPlan, 101, 1) exited non-zero on 'Local/101@prxAgentDialPlan-0c82,2' sip*CLI> show channels Channel (Context Extension Pri ) State Appl. Data Agent/100101 (prxAgentDialPlan s 1 ) Down AppDial (Outgoing Line) SIP/kris-hit-602-a3c0 (prxAgentDialPlan s 1 ) Up (None) (None) SIP/kris-cisco-602-1fdc (prxAgentExtensions 100101 1 ) Up Dial Agent/100101|120 3 active channel(s) sip*CLI> sip show channels Peer User/ANR Call ID Seq (Tx/Rx) Format Last Msg 216.58.12.241 kris-hit-6 48fc7501731 00102/00000 ulaw Tx: ACK 216.58.12.241 kris-cisco 000d6527-b8 00101/00102 g729 Rx: ACK 2 active SIP channel(s) -- SIP/kris-hit-602-a3c0 acknowledged -- Agent/100101 answered SIP/kris-cisco-602-1fdc sip*CLI> show channels Channel (Context Extension Pri ) State Appl. Data Agent/100101 (prxAgentDialPlan s 1 ) Up Bridged Call SIP/kris-cisco-602-1fdc SIP/kris-hit-602-a3c0 (prxAgentDialPlan s 1 ) Up (None) (None) SIP/kris-cisco-602-1fdc (prxAgentExtensions 100101 1 ) Up Dial Agent/100101|120 3 active channel(s) sip*CLI> sip show channels Peer User/ANR Call ID Seq (Tx/Rx) Format Last Msg 216.58.12.241 kris-hit-6 48fc7501731 00102/00000 ulaw Tx: ACK 216.58.12.241 kris-cisco 000d6527-b8 00101/00102 g729 Rx: ACK 2 active SIP channel(s) sip*CLI> | ||
Comments: | By: Kevin P. Fleming (kpfleming) 2005-02-27 15:36:02.000-0600 Yuck... does this mean that chan_agent should _not_ be issuing an 'answer' on the channel until after the agent has acked the call? By: Mark Spencer (markster) 2005-02-27 18:45:02.000-0600 Fixed in CVS head. We don't create the answer until after the ack. The issue is that we setup full duplex audio (which we should) before the call is considered answered. When calling an agent, it's probably reasnoable to not pass the agent's audio to the customer until they've answered, but this is really just changing the behavior of the way agents work more so than really fixing a bug. I don't *think* anyone would count on the previous behavior, but still, the real question is whether Russell wants to backport this to stable. By: Russell Bryant (russell) 2005-04-05 03:39:33 fixed in 1.0 By: Digium Subversion (svnbot) 2008-01-15 15:26:27.000-0600 Repository: asterisk Revision: 5103 U trunk/channels/chan_agent.c ------------------------------------------------------------------------ r5103 | markster | 2008-01-15 15:26:26 -0600 (Tue, 15 Jan 2008) | 2 lines Don't pass agent audio until call is confirmed (bug ASTERISK-3596) ------------------------------------------------------------------------ http://svn.digium.com/view/asterisk?view=rev&revision=5103 By: Digium Subversion (svnbot) 2008-01-15 15:30:41.000-0600 Repository: asterisk Revision: 5400 U branches/v1-0/CHANGES U branches/v1-0/channels/chan_agent.c ------------------------------------------------------------------------ r5400 | russell | 2008-01-15 15:30:41 -0600 (Tue, 15 Jan 2008) | 2 lines don't pass audio until the call has been acked if configured to do so (bug ASTERISK-3596) ------------------------------------------------------------------------ http://svn.digium.com/view/asterisk?view=rev&revision=5400 |