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

This worked. Great and thanks

-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Carlos Chavez
Sent: Wednesday, February 07, 2007 5:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] After upgrade to 1.4 transfers don't
workproperly

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

Ok that worked for normal transfers. Now here is another situation. When we try
to transfer a call directly to voicemail it plays the voicemail message but we
can't transfer the call. The only way I could get it to work was to do a
conference and then drop out of that conference.

My dial plan for direct dialing is:

exten=>_*40XX,n,Voicemail(${EXTEN:1},u)

When this is attempted the following message shows up on the CLI of Asterisk:

[Feb  9 11:52:03] WARNING[5054]: chan_sip.c:12310 handle_response: Notify answer
on an owned channel?

Can anyone tell me what this means and what I can do to fix this?

Thanks

-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Carlos Chavez
Sent: Wednesday, February 07, 2007 5:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] After upgrade to 1.4 transfers don't
workproperly

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