[asterisk-bugs] [Asterisk 0010228]: Extension Status does not change to 8 durring ring

noreply at bugs.digium.com noreply at bugs.digium.com
Wed Aug 8 10:15:21 CDT 2007


A NOTE has been added to this issue. 
====================================================================== 
http://bugs.digium.com/view.php?id=10228 
====================================================================== 
Reported By:                swolfe
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   10228
Category:                   Core/*
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     feedback
Asterisk Version:           1.4.8  
SVN Branch (only for SVN checkouts, not tarball releases): N/A  
SVN Revision (number only!):  
Disclaimer on File?:        No 
Request Review:              
====================================================================== 
Date Submitted:             07-18-2007 10:55 CDT
Last Modified:              08-08-2007 10:15 CDT
====================================================================== 
Summary:                    Extension Status does not change to 8 durring ring
Description: 
When calling from an IAX extension to a SIP extension the API command:

Action: ExtensionState
Exten: 200

Gives a status of 0

******************************** 
Response: Success
Message: Extension Status
Exten: 200
Context: default
Hint: SIP/200
Status: 0 
******************************** 


Where calling from SIP to IAX gives the correct Status
******************************** 
Response: Success
Message: Extension Status
Exten: 702
Context: default
Hint: IAX2/702
Status: 8 
******************************** 

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

---------------------------------------------------------------------- 
 swolfe - 08-08-07 10:15  
---------------------------------------------------------------------- 
After adding the call-limit entry and the limitonpeer there is still no
change to the origianl issue of the sip client showing ringing through the
API. See below for extensionstate output.

I have reposted my sip.conf so you can see what I have. 

Sip.conf
[general]
context=default                 ; Default context for incoming calls
allowoverlap=no                 ; Disable overlap dialing support.
(Default is yes)
bindport=5060                   ; UDP Port to bind to (SIP standard port
is 5060)
bindaddr=0.0.0.0                ; IP address to bind to (0.0.0.0 binds to
all)
srvlookup=yes                   ; Enable DNS SRV lookups on outbound
calls
notifyhold = yes
limitonpeer = yes
notifyringing = yes ; Notify subscriptions on RINGING state (default: no)
notifyhold = yes
call-limit = 300


[351]
type=friend
;context = incomming_local
fullname = Spider Man
host = dynamic
mailbox = 351
secret = 1234
vmsecret = 9844
dtmfmode = rfc2833
canreinvite = yes
nat = no
qualify = yes



Action: ExtensionState shows the following  while 351 is in fact ringing.


***********************************
Response: Success
Message: Extension Status
Exten: 351
Context: default
Hint: SIP/351
Status: 0 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
08-08-07 10:15  swolfe         Note Added: 0068618                          
======================================================================




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