[asterisk-bugs] [Asterisk 0011723]: caller side of sip hints not updated

[noreply at bugs.digium.com]

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:              02-07-2008 13:57 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
====================================================================== 

---------------------------------------------------------------------- 
 jb - 02-07-08 13:57  
---------------------------------------------------------------------- 
[general]
limitonpeers = yes

To the sip.conf, I got it to work.

I recon that there is some confusion regarding this field, as it sometimes
is referred to as limitonpeers and other times as limitonpeer (without the
s). Having limitonpeers under the definition of friend gave a cryptic error
message at SIP reload, stating that the type was not defined. 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
02-07-08 13:57  jb             Note Added: 0081871                          
======================================================================