[asterisk-bugs] [Asterisk 0014913]: limitonpeers=yes vs call-limit=1 didn't work on asterisk 1.4.24

Asterisk Bug Tracker noreply at bugs.digium.com
Thu Apr 16 16:43:50 CDT 2009


The following issue has been UPDATED. 
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http://bugs.digium.com/view.php?id=14913 
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Reported By:                sybasesql
Assigned To:                
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Project:                    Asterisk
Issue ID:                   14913
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     closed
Asterisk Version:           1.4.24 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
Resolution:                 no change required
Fixed in Version:           
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Date Submitted:             2009-04-16 12:57 CDT
Last Modified:              2009-04-16 16:43 CDT
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Summary:                    limitonpeers=yes vs call-limit=1 didn't work on
asterisk 1.4.24
Description: 
Hello,

In asterisk 1.4.24 limitonpeers=yes vs call-limit=1 didn't work.

My setup:
[xxxxxxxxxxx]
type=friend
accountcode=xxxxxxxxxxx
username=xxxxxxxxxxx
secret=xxxxxxxxxxx
host=dynamic
qualify=yes
dtmfmode=rfc2833
nat=yes
canreinvite=no
context=xxxxxxxxxxx
callerid="xxxxxxxxxxx" <xxxxxxxxxxx>
usereqphone=yes
call-limit=1
limitonpeers=yes
disallow=all
allow=alaw

Errors:
[Apr 16 21:33:21] ERROR[23739] chan_sip.c: Call from peer 'xxxxxxxxxxx'
rejected due to usage limit of 1
[Apr 16 21:33:21] NOTICE[23739] chan_sip.c: Failed to place call for user
xxxxxxxxxxx, too many calls

Dscription:
User xxxxxxxxxxx accept inbound call and immediate make outbound call
(redirect inbound) through his peer xxxxxxxxxxx, but got error "rejected
due to usage limit of 1".

====================================================================== 

---------------------------------------------------------------------- 
 (0103343) lmadsen (administrator) - 2009-04-16 16:43
 http://bugs.digium.com/view.php?id=14913#c103343 
---------------------------------------------------------------------- 
Based on a conversation I had with file on IRC, this appears to be working
exactly as it should per your configuration. The incoming call is most
likely matching on the IP address ('peer' structure), and not the 'user'
structure (which would be user based matching), and thus when you try to
dial out, it already has a channel usage of '1'. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-04-16 16:43 lmadsen        Note Added: 0103343                          
2009-04-16 16:43 lmadsen        Status                   new => closed       
2009-04-16 16:43 lmadsen        Resolution               open => no change
required
======================================================================




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