[asterisk-bugs] [Asterisk 0011180]: call limits not work as expected (limitonpeer, busy-level)
noreply at bugs.digium.com
noreply at bugs.digium.com
Mon Nov 19 12:29:00 CST 2007
The following issue has been REOPENED.
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http://bugs.digium.com/view.php?id=11180
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Reported By: pj
Assigned To: oej
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Project: Asterisk
Issue ID: 11180
Category: Channels/chan_sip/Subscriptions
Reproducibility: always
Severity: minor
Priority: normal
Status: feedback
Asterisk Version: SVN
SVN Branch (only for SVN checkouts, not tarball releases): trunk
SVN Revision (number only!): 89081
Disclaimer on File?: N/A
Request Review:
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Date Submitted: 11-07-2007 05:05 CST
Last Modified: 11-19-2007 12:29 CST
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Summary: call limits not work as expected (limitonpeer,
busy-level)
Description:
probably most wanted scenario is to let users place unlimited number of
outgoing calls from their sip phones (to be able to transfer etc.), but
indicate busy condition for calls to their phones, when one call already
exist (no matter if incomming or outgoing).
to achieve this, I set 'limitonpeer=yes' in [general] to be able to limit
only outgoing calls from asterisk to sip device (peer from asterisk
perspective, incomming call from sip phone perspective) and set
call-limit=1 in type=friend phone definition.
but seems, that limitonpeer=yes is actualy not working, because from sip
phone I can place only single call, so limited is also 'user' part in
asterisk phone definition.
I tried also set call-limit=2 and busy-level=1, but this permit call to
sip phone, even if I have already placed call from this phone. I think that
'busy-level' should work or count calls fot both directions, to correctly
indicate 'busy' condition if user already has call.
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pj - 11-19-07 12:29
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Hi, thanks for patch, still some issues:
- to be consistent with sip.conf 'busy-level', shouldn't be also sippeer
item called busy-level, instead of busylevel (or maybe change to busylevel
in sip.conf)?
- why we can't use comma "," to separate arguments in SIPPEER, I think,
that pipe "|" is now deprecated
- I think, it should work without need to GROUP_COUNT & SIPPEER busylimit
checking, eg. when try to dial extension, where busylevel limit is reached,
simply do not call this extension
- if I make call FROM sip phone (ie. asterisks friend user part), 'sip
show inuse' still shows zero in 'in use' column of 'user' configuration
part
* User name In use Limit
324 0 2
instead calls are counted in 'peer' section,
when I make call FROM phone, it shows:
* Peer name In use Limit
324 1/0/0 2
- 'limit 2' is still displayed in 'user' part, even if I have
'limitonpeer=yes' in sip.conf
when I have one call FROM sip phone and make concurent call TO phone, 'sip
show inuse' displays:
324 2/1/0 2
but after hangup it displays some weird values, even if phone s idle,
like:
* Peer name In use Limit
324 -1/0/0 2
it can be repaired, only with restarting asterisk
Issue History
Date Modified Username Field Change
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11-19-07 12:29 pj Note Added: 0073982
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