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[asterisk-users] Anyone used Siemens SIP/Dect phones?

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alanslists at gmail.com
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Posted: Sat Mar 22, 2008 4:08 am Post subject: [asterisk-users] Anyone used Siemens SIP/Dect phones?

Hi all, I am close to purchasing some new DECT phones for our home office here in the UK. We use Asterisk and I am sorely tempted by the Siemens C475IP or the "soon-to-become-available-in-the-uk" S685IP. Both systems have great feature sets and, on-paper at least, look to be the bee's knees. Anyone got any skeletons on them? Thanks Alan -- The way out is open! http://www.theopensourcerer.com

gordon+asterisk at dro...
Guest

Posted: Sat Mar 22, 2008 11:50 am Post subject: [asterisk-users] Anyone used Siemens SIP/Dect phones?

On Sat, 22 Mar 2008, Alan Lord wrote:

Quote:

Hi all,

I am close to purchasing some new DECT phones for our home office here
in the UK.

We use Asterisk and I am sorely tempted by the Siemens C475IP or the
"soon-to-become-available-in-the-uk" S685IP.

Both systems have great feature sets and, on-paper at least, look to be
the bee's knees.

Anyone got any skeletons on them?

I've deployed a number of Siemens C460IP's.

They're really good and coverage is excellent, but for one thing: They
base stations lose registration with the asterisk box after some time and
need rebooting. I've read that this affects the S450's too, but I've no
1st hand experience of them. I've emailled Siemens and posted on one of
their forums where others have experienced the same issues, but have
nothing back at all. I could live with the, if they had a remote reboot
facility, but they don't seem to.

I'd look at the Snom M3's but they're more expensive that my customers
will pay )-: And had a report of them being unstable too )-:

So for the time being I'm sticking to ATA's and cheap analogue DECT phones
for those customers who want them.

Gordon

mgraves at mstvp.com
Guest

Posted: Sat Mar 22, 2008 12:05 pm Post subject: [asterisk-users] Anyone used Siemens SIP/Dect phones?

On Sat, 22 Mar 2008 16:50:39 +0000 (GMT), Gordon Henderson wrote:

Quote:

I'd look at the Snom M3's but they're more expensive that my customers
will pay )-: And had a report of them being unstable too )-:

So for the time being I'm sticking to ATA's and cheap analogue DECT phones
for those customers who want them.

I can speak from experience that the snom m3 works well with Asterisk.
I've had no trouble with staying registered.

My Asterisk installation is still based on v1.2 since it's Astlinux
0.4.8. I'm hoping to get into much newer code real soon now.

Michael
--
Michael Graves
mgraves<at>mstvp.com
http://blog.mgraves.org
o713-861-4005
c713-201-1262
sip:mjgraves at pixelpower.onsip.com
skype mjgraves
54245 at fwd.pulver.com

alanslists at gmail.com
Guest

Posted: Sat Mar 22, 2008 1:02 pm Post subject: [asterisk-users] Anyone used Siemens SIP/Dect phones?

Gordon Henderson wrote:
<snip />

Quote:

Anyone got any skeletons on them?

As it is for home use that doesn't sound like a show stopper - I'll only
have one base station and three handsets.

Quote:

I'd look at the Snom M3's but they're more expensive that my customers
will pay )-: And had a report of them being unstable too )-:

The Snom does look good but it has one severe limitation for my needs -
no PSTN (Analogue) port on the base station...

Quote:

So for the time being I'm sticking to ATA's and cheap analogue DECT phones
for those customers who want them.

Gordon

Thanks

Al
--
The way out is open!
http://www.theopensourcerer.com

gordon+asterisk at dro...
Guest

Posted: Sat Mar 22, 2008 1:19 pm Post subject: [asterisk-users] Anyone used Siemens SIP/Dect phones?

On Sat, 22 Mar 2008, Alan Lord wrote:

Quote:

Gordon Henderson wrote:
<snip />

Quote:

Anyone got any skeletons on them?

As it is for home use that doesn't sound like a show stopper - I'll only
have one base station and three handsets.

Ah, in that case, you'll likely be fine. The ones I've deployed are all
rather busy (with the exception on the one I use at home!) One place has
to power cycle their bases 2-3 times a week, mine is OK for a month or 2.

Gordon

robl at linx.net
Guest

Posted: Sat Mar 22, 2008 3:19 pm Post subject: [asterisk-users] Anyone used Siemens SIP/Dect phones?

On Sat, Mar 22, 2008 at 09:08:43AM +0000, Alan Lord wrote:

Quote:

Have been using the C460IP phones and they seem to work okay, the range on
them is excellent. I haven't had any problems with the base de-registering
from asterisk though. (maybe a NAT timeout issue?) They are very simple to
configure.

Limitations:

- No SIP call transfer feature (that I can find)
- Doesn't have any remote provisioning features (yet)
- Doesn't have any ability to forward calls from the analog side
<-> SIP side, which is a shame as that would be handy.
- Base station supports multiple phones, but you can only register each
handset with one base station. So if you have multiple base stations, you
can't take advantage of that feature (i.e, allow the same handset to be
used in multiple locations. Other DECT handsets that support multiple
bases are available though.)
- I find it a bit quiet on the sound quality, sometimes a problem with
background noise.
- No VoIP message waiting indicator

Maybe some or all of these are addressed in the C475IP?

--
Robert Lister - London Internet Exchange - http://www.linx.net/
sip:robl at linx.net - inoc-dba:5459*710 - tel: +44 (0)20 7645 3510

lists at minotaur.cc
Guest

Posted: Sat Mar 22, 2008 4:36 pm Post subject: [asterisk-users] Anyone used Siemens SIP/Dect phones?

Quote:

They're really good and coverage is excellent, but for one thing: They
base stations lose registration with the asterisk box after some time and
need rebooting.

I've not played with the C460s, but have deployed plenty of S450s, and they did indeed exhibit this behaviour in earlier firmware versions, and where the asterisk server is over a less-than-ideal connection.

The S450s are pretty good and largely problem free given the following conditions:
1) The base station and the asterisk server are on the same LAN
2) You only register one handset per S450 base - registering multiple handsets seems to lead to all sorts of stability issues

Regards,

Chris
--
C.M. Bagnall, Director, Minotaur I.T. Limited
For full contact details visit http://www.minotaur.it
This email is made from 100% recycled electrons

lists at minotaur.cc
Guest

Posted: Sat Mar 22, 2008 4:39 pm Post subject: [asterisk-users] Anyone used Siemens SIP/Dect phones?

Can?t comment on the C460, but the S450 definitely doesn't have these issues:

Quote:

- No SIP call transfer feature (that I can find)

Hit "ext call" during a call, create a new call, then you can SIP transfer between them.

Quote:

- Base station supports multiple phones, but you can only register each
handset with one base station. So if you have multiple base stations, you
can't take advantage of that feature (i.e, allow the same handset to be
used in multiple locations. Other DECT handsets that support multiple
bases are available though.)

S450 handsets will register to 4 bases.

Quote:

- I find it a bit quiet on the sound quality, sometimes a problem with
background noise.

Should be an option in the web interface to adjust volume - I set all the ones I deploy to "high"

Regards,

Chris
--
C.M. Bagnall, Director, Minotaur I.T. Limited
For full contact details visit http://www.minotaur.it
This email is made from 100% recycled electrons

robl at linx.net
Guest

Posted: Sun Mar 23, 2008 6:15 am Post subject: [asterisk-users] Anyone used Siemens SIP/Dect phones?

On Sat, Mar 22, 2008 at 09:39:47PM -0000, Chris Bagnall wrote:

Quote:

Can?t comment on the C460, but the S450 definitely doesn't have these issues:

Quote:

- No SIP call transfer feature (that I can find)

Hit "ext call" during a call, create a new call, then you can SIP
transfer between them.

Where is that? I don't seem to get that option. What I want is an announced
call transfer to another SIP device.

Quote:

S450 handsets will register to 4 bases.

I've got S460IP, that that only seems to allow one base station.

Quote:

- I find it a bit quiet on the sound quality, sometimes a problem with
background noise.

Should be an option in the web interface to adjust volume - I set all the
ones I deploy to "high"

Yeah, tried that. Still a bit quiet when compared to other handsets.
--
Robert Lister - London Internet Exchange - http://www.linx.net/
sip:robl at linx.net - inoc-dba:5459*710 - tel: +44 (0)20 7645 3510

dcotton at linuxautrem...
Guest

Posted: Sun Mar 23, 2008 6:33 am Post subject: [asterisk-users] Anyone used Siemens SIP/Dect phones?

Robert Lister wrote:

Quote:

On Sat, Mar 22, 2008 at 09:39:47PM -0000, Chris Bagnall wrote:

Quote:

Can?t comment on the C460, but the S450 definitely doesn't have these issues:

Quote:

- No SIP call transfer feature (that I can find)

Hit "ext call" during a call, create a new call, then you can SIP
transfer between them.

Where is that? I don't seem to get that option. What I want is an announced
call transfer to another SIP device.

I've always used # which, in my features, conf is configured for
attended transfer and ## which is configured as blind.

Dave Cotton

gordon+asterisk at dro...
Guest

Posted: Sun Mar 23, 2008 7:02 am Post subject: [asterisk-users] Anyone used Siemens SIP/Dect phones?

On Sun, 23 Mar 2008, Robert Lister wrote:

Quote:

On Sat, Mar 22, 2008 at 09:39:47PM -0000, Chris Bagnall wrote:

Quote:

Can??t comment on the C460, but the S450 definitely doesn't have these issues:

Quote:

- No SIP call transfer feature (that I can find)

Hit "ext call" during a call, create a new call, then you can SIP
transfer between them.

Where is that? I don't seem to get that option. What I want is an announced
call transfer to another SIP device.

The only way I've found with the Siemens that I have is to use the
asterisk native features - So edit features.conf appropriately, I have:

[general]
xfersound = beep ; to indicate an attended transfer is complete
xferfailsound = beeperr ; to indicate a failed transfer
transferdigittimeout = 8 ; Number of seconds to wait between digits when transfering a call
featuredigittimeout = 999 ; Max time (ms) between digits for activation. Default is 500
[featuremap]
blindxfer => #1 ; Blind transfer
atxfer => ## ; Attended transfer
disconnect => #0 ; Disconnect

Then you need to make sure you use the 't' option in Dial() statements,
then you dial ##, asterisk says 'Transfer', then you dial the extension,
speak to them & hangup to make the transfer, or #0 to get back to the
original caller.

It's a PITA, but it works on all phones regardless of their own features,
buttons, etc. so I usually teach people that before the phone specific
stuff.

Gordon

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