[asterisk-bugs] [Asterisk 0011723]: caller side of sip hints not updated
noreply at bugs.digium.com
noreply at bugs.digium.com
Fri Jan 11 02:01:58 CST 2008
A NOTE has been added to this issue.
======================================================================
http://bugs.digium.com/view.php?id=11723
======================================================================
Reported By: mostyn
Assigned To:
======================================================================
Project: Asterisk
Issue ID: 11723
Category: Core-General
Reproducibility: always
Severity: minor
Priority: normal
Status: feedback
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:
======================================================================
Date Submitted: 01-09-2008 21:30 CST
Last Modified: 01-11-2008 02:01 CST
======================================================================
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
======================================================================
----------------------------------------------------------------------
mostyn - 01-11-08 02:01
----------------------------------------------------------------------
I have attached three files,
full: a full log with debug and verbose enabled, both set to 9
sip_show_peer_and_user: the output from sip show user 10{2,3,4} and sip
show peer 10{2,3,4}
sip_show_inuse: the output from sip show inuse
Issue History
Date Modified Username Field Change
======================================================================
01-11-08 02:01 mostyn Note Added: 0076698
======================================================================
More information about the asterisk-bugs
mailing list