asterisk users - Feb 2007 - After upgrade to 1.4 transfers don't work properly

I have discovered an issue on my system after upgrading from 1.2.13 to
1.4. A call comes in on a T1 line and goes to a Polycom 501 SIP phone. I
have confirmed this on multiple phones. When the called person answers
and tries to transfer the call to another extension, the call
successfully transfers, however the person answering the transfer cannot
hear the person that called in, the caller. My dial command simply is 

 

exten=>4000,1,Dial(SIP/4000,40,t)

 

This DID work before when we were on 1.2.13. 

 

The CLI displays the following message:

 

handle_response: Notify answer on an owned channel?

 

I searched the web and found similar issues but not the same.

 

http://bugs.digium.com/view.php?id=8696

 

This one has the error, however I don't get a segment fault and
supposedly this was fixed in revision 50032. How do I get this revision?
I'm guessing it's in a non-tested svn release which I don't think I
want
to install in a production system.

 

Anyone else have this issue? Any ideas on how to fix this or get around
it?

 

_____________________

Kevin Savoy

Business Unit Telecom Analyst

2218 4th Ave W

Williston, ND 58801

Ph: 701-774-4023

Fax: 701-774-2901

http://www.novo1.com <http://www.novo1.com/> 

Novo 1 is a service mark of Novo 1, Inc

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20070207/424100f4/attachment.htm

On Wed, 2007-02-07 at 14:12 -0600, Savoy, Kevin - Williston, ND
wrote:
> I have discovered an issue on my system after upgrading from 1.2.13 to
> 1.4. A call comes in on a T1 line and goes to a Polycom 501 SIP phone.
> I have confirmed this on multiple phones. When the called person
> answers and tries to transfer the call to another extension, the call
> successfully transfers, however the person answering the transfer
> cannot hear the person that called in, the caller. My dial command
> simply is 
> 
>  
> 
	I had exactly the same problem when upgrading to 1.4 and I solved by
making sure canreinvite=no is in sip.conf for every phone.
> 
-- 
Telecomunicaciones Abiertas de Mexico S.A. de C.V.
Carlos Ch?vez Prats
Director de Tecnolog?a
+52-55-91169161 ext 2001
-------------- next part --------------
A non-text attachment was scrubbed...
Name: not available
Type: application/pgp-signature
Size: 189 bytes
Desc: This is a digitally signed message part
Url :
http://lists.digium.com/pipermail/asterisk-users/attachments/20070207/f94428a1/attachment.pgp