[asterisk-commits] branch bweschke/polycom_acd_functions - r7735 /team/bweschke/polycom_acd_func...

asterisk-commits at lists.digium.com asterisk-commits at lists.digium.com
Tue Jan 3 08:33:26 CST 2006


Author: bweschke
Date: Tue Jan  3 08:33:24 2006
New Revision: 7735

URL: http://svn.digium.com/view/asterisk?rev=7735&view=rev
Log:
 Added README for this functionality in /doc


Added:
    team/bweschke/polycom_acd_functions/doc/README.polycom_agents

Added: team/bweschke/polycom_acd_functions/doc/README.polycom_agents
URL: http://svn.digium.com/view/asterisk/team/bweschke/polycom_acd_functions/doc/README.polycom_agents?rev=7735&view=auto
==============================================================================
--- team/bweschke/polycom_acd_functions/doc/README.polycom_agents (added)
+++ team/bweschke/polycom_acd_functions/doc/README.polycom_agents Tue Jan  3 08:33:24 2006
@@ -1,0 +1,34 @@
+ This branch contains code to allow for integration with the Polycom 
+SoundPoint IP phones and Asterisk's agent infrastructure. You can 
+login/logout an agent and pause/unpause them from queue(s) via 
+soft-buttons on the phone.
+
+ Pre-requisites / assumptions / caveats:
+ * Agent-IDs defined in agents.conf must NOT have an agentid that 
+conflicts with a SIP device ID/username. If you intermingle these, you 
+will more than likely get failed login attempts, but could get other 
+really undesirable results on your SIP channel. This happens because 
+Asterisk doesn't yet support multiple authentication realms in SIP and 
+won't until chan_sip3 is available. I will not be introducing any 
+"workarounds" to overcome this limitation between now and the time these 
+features make it into chan_sip3. To do so would compromise the security 
+of the SIP channel authentication in place now.
+ * We are assuming that the From: header / username on the SIP device is 
+the extension that we're registering the Agent to be called back at. If 
+the username doesn't match the extension in extensions.conf that the 
+device can be reached at, the Agent login will work, but when it comes 
+time to dial that agent, the call will be going to the wrong location.
+
+ New sip.conf parameters that allow this to work:
+ * The device must be "type=friend" in order for the device to be able 
+to use this functionality. This is generally a safe assumption for 
+Polycom phones that are directly connected to the Queue engine.
+ * agentlogin=yes in the device definition. If you do not have this, 
+your login attempts will always fail because the digestusername on 
+authentication will not match the username in the From SIP header, and 
+prior to this code, this was generally a no-no and a security violation 
+that resulted in an immediate negative SIP message. The default is "no", 
+so if you want this device to be able to login an agent from the phone, 
+you must specify this parameter and set it to "yes".
+ * agentcbcontext=default in the device definition. This is the context 
+for which calls back to the agent will be sent. Default is "default".



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