FW: [asterisk-users] After upgrade to 1.4 transfers don't workproperly

[Savoy, Kevin - Williston, ND]

No one knows what the Notify answer on an owned channel is?

Anyone?

-----Original Message-----
From: Savoy, Kevin - Williston, ND 
Sent: Monday, February 12, 2007 11:14 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: FW: [asterisk-users] After upgrade to 1.4 transfers don't workproperly

Sorry if this is a repeat but I didn't receive a copy of it so I'm not sure it actually posted.


The below 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 at lists.digium.com [mailto:asterisk-users-bounces at 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.

> 
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