asterisk users - Apr 2009 - Siemens Gigaset Phones get mute function.

Hi, I know this is a little OT but there are many Asterisk users of the 
excellent Siemens DECT/VOIP phones like the S685IP and 475IP and this is 
probably newsworthy for them.

One of the biggest bug bears has been no mute function on the handset.

When I woke up this morning, the handset told me there was a firmware 
update. I updated and then visited the web site to find out what had 
been fixed (quite a lot of new features have been added):

http://gigaset.com/shc/0,1935,hq_en_0_152411_rArNrNrNrN_variation%253A-5_pageType%253Adownloads_imagePos%253A0,00.html#content

Gigaset C470 IP / C475 IP / S675 IP / S685 IP Firmware update 04/2009

Download version: 02184

New features:

     * E-mail viewer (with C47H, S45, S67H, S68H handsets)
     * Mute function. Turn off the handset's microphone during an 
external call with the left display key.
     * Send and receive SMS messages via VoIP*
     * VoIP: If the telephone cannot establish a VoIP connection, it 
automatically dials via the fixed line network (auto-fallback to PSTN).
     * VoIP: Call transfer via R key
     * VoIP: An incoming call indicated in parallel at different VoIP 
devices (parallel ringing) will not be stored in the "Missed calls"
list
if the call was accepted at one of the devices.*
     * Online directory: display the postal codes in search results.*
     * Online directory: when starting a new search the cities used in 
the last searches will be displayed.
     * Fixed line access codes can be stored in the phone.
     * Some languages will be loaded on to the base via the internet, 
depending on the language set on the handset.
     * Extended RTP port range (1024-55000)
     * DHCP Option 114 implemented.*
     * DHCP Option 120 implemented.*
     * Web Configurator: option to specify whether the "area code" is 
dialled as well.
     * Web Configurator: display RTP port range
     * Web Configurator: new languages - Arabic and Russian
     * Web Configurator: enhanced PIN protection - warning if the 
default pin (0000) has not been changed.

Improvements:

     * The automatic search function for firmware updates is enabled 
even if the internet connection is temporarily interrupted during the night.
     * The indication "Anonymous call activated" will no longer be 
displayed in the idle mode of the handset.
     * The country code is synchronised between base station and handset.

Hello,

Using odbc voicemail and mysql, i have a problem.
After 12 seconds recording, asterisk stop recording and hangup.

I have changed the settings in voicemail.conf to allow 180 seconds but,...

Any hint?

Olivier

On Wed, Apr 8, 2009 at 9:05 AM, Alan Lord (News) <alanslists at gmail.com>
wrote:
> Hi, I know this is a little OT but there are many Asterisk users of the
> excellent Siemens DECT/VOIP phones like the S685IP and 475IP and this is
> probably newsworthy for them.
Oh, yes! This is the greatest news since sliced bread!

>? * VoIP: An incoming call indicated in parallel at different VoIP
> devices (parallel ringing) will not be stored in the "Missed
calls" list
> if the call was accepted at one of the devices.*
This was a good idea, too.

Thanks for the post Alan, I saw the update notice on the phoine but
didn't go for it until I ready this!

r

2009/4/8 Alan Lord (News) <alanslists at gmail.com>
>
>     * DHCP Option 114 implemented.*
>     * DHCP Option 120 implemented.*
> <http://lists.digium.com/mailman/listinfo/asterisk-users>
>
What does it imply ?
Provisionning from DHCP server ?
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2009/4/8 Olivier <oza-4h07 at myamail.com>
>
> 2009/4/8 Alan Lord (News) <alanslists at gmail.com>
>
>>
>>     * DHCP Option 114 implemented.*
>>     * DHCP Option 120 implemented.*
>> <http://lists.digium.com/mailman/listinfo/asterisk-users>
>>
> What does it imply ?
> Provisionning from DHCP server ?
>
114 is for passing a URL to be displayed after boot
120 is for passing SIP servers (RFC3361)

So, I guess that would be a yes...

d
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