Summary:	ASTERISK-09899: Extension Status does not change to 8 durring ring
Reporter:	Scott Wolfe (swolfe)	Labels:
Date Opened:	2007-07-18 10:55:39	Date Closed:	2011-06-07 14:07:51
Priority:	Minor	Regression?	No
Status:	Closed/Complete	Components:
Versions:		Frequency of Occurrence
Related Issues:
Environment:		Attachments:
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 ******************************** ****** ADDITIONAL INFORMATION ****** This is in 1.4.8 but that option is not available from the asterisk version drop down below.
Comments:	By: Joshua C. Colp (jcolp) 2007-08-06 13:54:22 Please attach your sip.conf minus passwords and core show hints. Thanks! By: Scott Wolfe (swolfe) 2007-08-06 14:14:42 localhost*CLI> core show hints localhost*CLI>    -= Registered Asterisk Dial Plan Hints =-                   7000@default             : IAX2/7000             State:Idle            Watchers  0                   6000@default             : SIP/6000              State:Idle            Watchers  0                    351@default             : SIP/351               State:Unavailable     Watchers  0 Here is SIP.CONF with all the comments taken out. [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 [351] type=friend ;context = incomming_local fullname = Spider Man host = dynamic mailbox = 351 secret = XXXX vmsecret = XXXX dtmfmode = rfc2833 canreinvite = yes nat = no qualify = yes By: Joshua C. Colp (jcolp) 2007-08-08 08:23:57 A call limit must be set to get proper status and limitonpeer = yes must be set. Try those settings and see if it makes it behave as you need. By: Scott Wolfe (swolfe) 2007-08-08 10:15:21 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 By: Gregory Hinton Nietsky (irroot) 2007-08-08 10:39:52 the call-limit is per peer setting and not a global setting ... insert 20c and try again it will probably werk ... By: Scott Wolfe (swolfe) 2007-08-08 11:09:48 It does. Thanks for help clearing this up for me. By: Joshua C. Colp (jcolp) 2007-08-16 13:28:30 Closed, configuration issue.