[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
======================================================================
More information about the asterisk-bugs
mailing list