[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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