[asterisk-commits] oej: branch 1.4 r48113 - /branches/1.4/configs/sip.conf.sample

[asterisk-commits at lists.digium.com]

Author: oej
Date: Wed Nov 29 13:56:56 2006
New Revision: 48113

URL: http://svn.digium.com/view/asterisk?view=rev&rev=48113
Log:
Explain the use device status system implemented in SIP for subscriptions,
queues and manager a bit better.

Like in 1.2, you will get more detailed information if you set a call 
limit for a device. When the call limit is reached, the status system will
report a device as busy.

For queues, setting a call limit per SIP device is propably a requirement.

In most cases, it will work much better if you only use type=peer and not
type=friend. We might decide to backport the new setting from trunk to
apply all call limits to the peer part of a friend only.


Modified:
    branches/1.4/configs/sip.conf.sample

Modified: branches/1.4/configs/sip.conf.sample
URL: http://svn.digium.com/view/asterisk/branches/1.4/configs/sip.conf.sample?view=diff&rev=48113&r1=48112&r2=48113
==============================================================================
--- branches/1.4/configs/sip.conf.sample (original)
+++ branches/1.4/configs/sip.conf.sample Wed Nov 29 13:56:56 2006
@@ -164,6 +164,15 @@
 ; You can subscribe to the status of extensions with a "hint" priority
 ; (See extensions.conf.sample for examples)
 ; chan_sip support two major formats for notifications: dialog-info and SIMPLE 
+;
+; You will get more detailed reports (busy etc) if you have a call limit set
+; for a device. When the call limit is filled, we will indicate busy. Note that
+; you need at least 2 in order to be able to do attended transfers.
+;
+; For queues, you will need this level of detail in status reporting, regardless
+; if you use SIP subscriptions. Queues and manager use the same internal interface
+; for reading status information.
+;
 ; Note: Subscriptions does not work if you have a realtime dialplan and use the
 ; realtime switch.
 ;