[Asterisk-Users] Re: Dial group continues to ring after answer - SNOM phones and solution

[Mike Meyer]

Asterisk Users;

	Just wanted to let you know I fixed my problem. 

To follow up on my own testing of the situation, I find that the
continued ringing after pickup only occured on the SNOM phones in the
group. The Grandstream phones stop ringing when another phone picks up.

Having turned on SIP debugging I have verified that the cancel message
is sent to the SNOM phone (and others in the group) when one of the
group phones is picked up, as expected. 

It appears that the SNOMs don't handle the cancel message the same as
the Grandstream. I was using SIP 2.03o firmware on the SNOM which is the
latest official release.

It seems that these phones even though they are set to do automatic
update, they do not. Or perhaps that capability was broken in the
firmware version I had last updated to.

THE SOLUTION:
To remedy the problem I upgraded to version 3.52 beta version. Also
2.04g fixes this problem as well. 

I had to create my own internal TFTP server and flash update to 3.52.
The standard update process did not work to go beyond 2.03y or 2.04g. I
tried 2.05e & f and these would never come out of boot.

MORAL TO THE STORY: Keep your SIP phone firmware up to date.

SNOM support is telling me to upgrade to 3.54. I don't see this one
listed on the standard update URL. I am a little leery about moving to
that one. 

Now to upgrade my GrandStream's. They seem to be stuck at an old version
as well.

Thanks,
Mike Meyer

On Tue, 2004-10-05 at 16:47, Mike Meyer wrote:
> Asterisk Users:
> 
> 	We have our * dial plan set up to ring 5 phones in the office and it
> delivers the call to the first that picks up their receiver.
> 	The problem is that after the pickup, everyone else's SIP phone keeps
> ringing at least once and sometimes twice. This interferes with the
> conversation, while others pick up the phone and get nothing.
> 
> 	Does anyone else have similar problems or have a solution to stop the
> ring once answered? My dial statement looks like the following and has a
> timeout of 15 seconds.
> 
> exten => MainTeam,1,Dial(${MainTeamChannels},15,tr)
> exten => MainTeam,2,Voicemail(u${MainTeam_EXT})
> ...
> note the variables MainTeamChannels define the SIP phone channels
> defined in sip.conf and MainTeam_EXT is the voicemail box for this group
> extension.
> 
> 	As an alternative, I am considering doing a round robin on a call group
> or pickup group and implementing call pickup.
> 
> Any ideas welcome.
> 
> Mike Meyer
> 
> 
>