asterisk users - Mar 2005 - Is it an AGI bug in 1.06? IAX Calls going to wrong extension with AGI.

I am using PBXware for configuring users and extensions.
Pbxware uses Internal script called init.sh to process the calls 
based on its own version of extensions.conf defined in the GUI.
 
I have IAX2 Extensions 56 and 101 and SIP extensions 50 and 51. 
 
I have used IAX2 extension 101 and dialed SIP Extension 51
 
But the PBXWare's Init.sh  AGI command identifies the DNIS 
as another IAX Extension - extension 56, instead of SIP Extension 51 
and sends the call there.
 
I tried the same with Extension 50 and the result is the same? 
is this an AGI Bug or a bug in the GUI Software. 
 
Has anyone tried this before and had such problem?
 
 
VAR:  agi_request: init.sh                                        ;(
Init.sh is sent from PBXware)
VAR:  agi_channel: IAX2/101@101/2 <mailto:IAX2/101@101/2>     
VAR:  agi_language: en    
VAR:  agi_type: IAX2    
VAR:  agi_uniqueid: asterisk-28947-1110463619.0    
VAR:  agi_callerid: Seshu Kanuri <101>    
VAR:  agi_dnid: 56                                                ;
Actual number dialed was 51
VAR:  agi_rdnis: unknown    
VAR:  agi_context: default    
VAR:  agi_extension: 56    
VAR:  agi_priority: 1    
VAR:  agi_enhanced: 0.0    
VAR:  agi_accountcode:    
   Detected protocol 'iax2' ...  200 result=1  
   Detected caller '101' ...  200 result=1  
   Set limit - 24  200 result=1  
   Limit not exceeded (1 < 24) for localextensions  200 result=1  
   Set limit - 2  200 result=1  
   Limit not exceeded (1 < 2) for 101_out  200 result=1  
   Detecting destination for '56' ...  200 result=1  
   Found Destination localextensions (range 56 - 56)  200 result=1  
   Setting destination 'localextensions' ...  200 result=1  
   This is local extension, skipping Time Based Dialing/miniLCR ...  200
result=1  
   Set limit - 24  200 result=1  
   Limit not exceeded (2 < 24) for localextensions  200 result=1  
   Detecting Vertical Services ...  200 result=1  
   Set limit - 2  200 result=1  
   Limit not exceeded (1 < 2) for 56_in  200 result=1  
   Checking for channel IAX2/56/56 ...  200 result=1  
APP:  exec ChanIsAvail IAX2/56/56  200 result=-1  
   Channel is not available ...  200 result=1  
   Dialing Voicemail 56 ...  200 result=1  
APP:  exec Voicemail u56  200 result=-1  
APP:  answer  200 result=0  
   Playing macro 'vm-goodbye' ...  200 result=1  
APP:  stream file vm-goodbye  200 result=-1 endpos=6880  
 
 
Any clues or pointers?
 
Seshu
 
  _____  

From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Dennis
Webb
Sent: Thursday, March 10, 2005 4:32 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Polycom phones do not talk to each other
andcannot answer when we pickup


Never used pbxware, but the context the sip phones dial out using
specified in sip.conf needs to include the dialplan context of the
phones in extensions.conf.

On Thu, 2005-03-10 at 15:08, Kanuri, Seshu (Company IT) wrote: 

	We have bought PBXware GUI from Bicom systems and configured
extensions
	with Polycom Phones as UAs.
	
	The Polycom Phones can dial out and make calls but I cannot make
	extension to extension calling.
	
	Googling did not help much.
	
	As you may be aware PBXware is a closed source software GUI from
Bicom
	Systems for configuring extensions. It is a good tool to
configure and
	manage users and phones but it does not allow to do any of the
	customization tasks that are possible by directly editing the
.conf
	files, which may be required in for Polycom.
	
	However if this is an issue of configuration on the Phone
itself, we
	want to be able to make changes and fix this problem.
	
	Any tips?
	
	Seshu 
	--------------------------------------------------------
	 
	NOTICE: If received in error, please destroy and notify sender.
Sender does not waive confidentiality or privilege, and use is
prohibited. 
	 
	_______________________________________________
	Asterisk-Users mailing list
	Asterisk-Users@lists.digium.com
	http://lists.digium.com/mailman/listinfo/asterisk-users
<http://lists.digium.com/mailman/listinfo/asterisk-users> 
	To UNSUBSCRIBE or update options visit:
	   http://lists.digium.com/mailman/listinfo/asterisk-users
<http://lists.digium.com/mailman/listinfo/asterisk-users> 
--------------------------------------------------------
 
NOTICE: If received in error, please destroy and notify sender.  Sender does not
waive confidentiality or privilege, and use is prohibited.
 
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20050311/df203642/attachment.htm

On Fri, 2005-03-11 at 12:26 -0500, Kanuri, Seshu (Company IT)
wrote:
> I am using PBXware for configuring users and extensions.
> Pbxware uses Internal script called init.sh to process the calls 
> based on its own version of extensions.conf defined in the GUI.
>  
> I have IAX2 Extensions 56 and 101 and SIP extensions 50 and 51. 
>  
> I have used IAX2 extension 101 and dialed SIP Extension 51
>  
> But the PBXWare's Init.sh  AGI command identifies the DNIS 
> as another IAX Extension - extension 56, instead of SIP Extension 51 
> and sends the call there.
Just a quick thought here, as the vast majority doesn't have access or
at the minimal don't use the software you are using to do config and as
it is an agi script outside of asterisk, you should go to the vendor of
PBXWare and see what they say.
-- 
Steven Critchfield <critch@basesys.com>