[asterisk-bugs] [Asterisk 0011723]: caller side of sip hints not updated
noreply at bugs.digium.com
noreply at bugs.digium.com
Fri Feb 8 09:44:10 CST 2008
The following issue has been RESOLVED.
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http://bugs.digium.com/view.php?id=11723
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Reported By: mostyn
Assigned To: file
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Project: Asterisk
Issue ID: 11723
Category: Core-General
Reproducibility: always
Severity: minor
Priority: normal
Status: resolved
Asterisk Version: 1.4.17
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Disclaimer on File?: N/A
Request Review:
Resolution: no change required
Fixed in Version:
======================================================================
Date Submitted: 01-09-2008 21:30 CST
Last Modified: 02-08-2008 09:44 CST
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Summary: caller side of sip hints not updated
Description:
With asterisk 1.4.17, only the called end of a sip->sip call has its status
set to InUse, the caller is marked as Idle. This occurs when using
type=friend but also when I create SIP user/peer pairs explicitly.
sip.conf:
[general]
bindport=5060
bindaddr=0.0.0.0
autocreatepeer=no
allowguest=no
language=en
disallow=all
allow=alaw ; Fat codec
allow=ulaw ; Fat codec
allow=gsm ; 13 Kbps, free
qualify=yes
[102]
type=friend
username=102
secret=102
accountcode=102
canreinvite=no
limitonpeers=yes
call-limit=2
context=default
subscribecontext=default
host=dynamic
nat=yes
qualify=yes
disallow=all
allow=alaw
allow=ulaw
allow=gsm
[103]
type=friend
username=103
secret=103
accountcode=103
canreinvite=no
limitonpeers=yes
call-limit=2
context=default
subscribecontext=default
host=dynamic
nat=yes
qualify=yes
disallow=all
allow=alaw
allow=ulaw
allow=gsm
[104]
type=friend
username=104
secret=104
accountcode=104
canreinvite=no
limitonpeers=yes
call-limit=2
context=default
subscribecontext=default
host=dynamic
nat=yes
qualify=yes
disallow=all
allow=alaw
allow=ulaw
allow=gsm
extensions.conf:
[general]
static=yes
writeprotect=yes
[globals]
[default]
exten => 102,1,Dial(SIP/102)
exten => 103,1,Dial(SIP/103)
exten => 104,1,Dial(SIP/104)
exten => 2,1,Pickup(102)
exten => 3,1,Pickup(103)
exten => 4,1,Pickup(104)
exten => 102,hint,SIP/102
exten => 103,hint,SIP/103
exten => 104,hint,SIP/104
when 103 calls 104, "show hints":
-= Registered Asterisk Dial Plan Hints =-
104 at default : SIP/104
State:InUse Watchers 1
103 at default : SIP/103
State:Idle Watchers 1
102 at default : SIP/102
State:Idle Watchers 0
show channels:
Channel Location State Application(Data)
SIP/104-006e0c70 (None) Up Bridged
Call(SIP/103-006e6850)
SIP/103-006e6850 104 at default:1 Up Dial(SIP/104)
2 active channels
1 active call
and when 104 calls 103:
-= Registered Asterisk Dial Plan Hints =-
104 at default : SIP/104
State:Idle Watchers 1
103 at default : SIP/103
State:InUse Watchers 1
102 at default : SIP/102
State:Idle Watchers 0
Channel Location State Application(Data)
SIP/103-006e0c70 (None) Up Bridged
Call(SIP/104-006e6850)
SIP/104-006e6850 103 at default:1 Up Dial(SIP/103)
2 active channels
1 active call
======================================================================
----------------------------------------------------------------------
file - 02-08-08 09:44
----------------------------------------------------------------------
As described this is a configuration issue. You must have limitonpeers set
to yes in order to have devicestate correctly report status. This option
can only be set globally.
Issue History
Date Modified Username Field Change
======================================================================
02-08-08 09:44 file Status feedback => resolved
02-08-08 09:44 file Resolution open => no change
required
02-08-08 09:44 file Assigned To => file
02-08-08 09:44 file Note Added: 0081917
======================================================================
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