A VoIP DECT Phone based on the Siemens Chagall Platform.

The Siemens Gigaset S675IP and S685 IP are successors to this phone.

It is similar to the Siemens Gigaset C450IP. Some of the differences result from the different handset. The S450IP uses a S45 handset.

• Larger display than C450IP
• Multi-Base Capability (the handset can be registered to multiple base stations)
• Jabber-compatible Instant Messenger
• POP3 Inbox Status Display
• Support for up to 6 SIP accounts, which may be assigned to one or more of the registered handsets for incoming/ourgoing calls.
• Support for 2 simultaneous DECT channels.
• Support for 2 SIP calls in parallel (in addition to using the analoge line)
• Support for uploading and downloading the phonebooks of the individual handsets

See also: Siemens phones, Siemens Gigaset Blog on S450 IP

07/2008 - Firmware: 021230000000 / 038.00 EEPROM: 121

New Features:

• Compatibility with A58H and C38H Handsets (including Infoscreensaver as Liveticker)
• New languages: Brazilian Portuguese, Russian (country and handset dependent)
• BEL, NLD: Autoconfigurationcode-enquiry added in the Handset's connection assistant

Improvements:

• NAT Traversal improved
• STUN can be disabled for Gigaset.net
• Email-Notification: display of date and subject improved, lenght of Email address up to 74 characters
• Display IP-Address without leading "0"s during the paging call
• Entries out of the calls list will be fully copied into the telephone directory, if the name was displayed * from online phonebook.
• Web configurator: does not show "********" for empty password
• SMS Status Report function improved
• SIP Protocol implementation improved
• Incorrect recall after a successful call transfer fixed
• Display Scandinavian characters
• Answering machine functionality improved

13/12/2007 - Firmware: 020970000000 / 041.00 EPROM: 114

Note that MWI now works with asterisk mailboxes

New Features:

• ECO-DECT is supported
• Net AM calls will be displayed at the handset
• It is possible to select international online phonebook provider
• Advanced Online phonebook features usable (Depends on Provider)
• The number of incoming calls will be replaced by the caller’s name of the online phonebook (Depends on provider)
• The SIP account can be activated with a code (Depends on provider)
• Calling Line Identity Restriction (CLIR) for VoIP (Depends on Provider)
• Directory entries for Gigaset.net and Net Directories will be transferred during registration (Handset: C45, S45, SL55, SL56, C47H, SL37H, S67H)
• Gigaset.net call forwarding
• Default Line configuration via Web Configurator
• Handset name configuration via Web Configurator
• Display of called number (like COLP in ISDN)
• With an active online connection the Web Configurator can be reached with "www.Gigaset-config.com"

Improvements:

• SIP UDP registration improved
• It is possible to edit international Prefix and Local Area Code
• It is possible to use ", ', >, <, & in Gigaset.net Nicknames
• Unfounded Stun Requests avoided
• NTP requests minimized
• Wideband (G.722) function improved
• E-Mail and Messenger function improved
• Suffix dialing and dialing plan function improved
• Echo suppression for VoIP improved
• Online phonebook function improved
• Directory transfer to the Gigaset SL1 handset is possible
• IP dialing improved
• Notepad *.vcf files can be transmitted to the handset
• Info Screen problem with Gigaset C47H Handset fixed.
• It is possible to use quotations in the "Display Name" of the SIP profile

14/09/2007 - Firmware: 020810000000 / 041.00 EPROM: 103

• Fixed CLI length issue, now truncates instead of displaying 'Unknown'
• R-button now works in SIP mode (You need to enable this in advanced settings, Telephony / Advanced Settings page to includes Hook-Flash call transfer). See also this thread on call transfer. User report: When pressed Asterisk 1.4.13 reports this error and nothing happens: "WARNING[905]: chan_sip.c:2773 sip_indicate: Don't know how to indicate condition 9"

New Features/Improvements according to Siemens, see Release Notes, manuals

• VoIP Wideband function added (DTMF Outband signalization only). This feature can be used with S67H and SL37H handsets only.
• Line selection via account index added (#0 - #9).
• Dialing Plan added.
• VoIP Call Transfer function added. (Depends on the provider).
• Call rejection via Onhook-Key added.
• IP address exchange via paging call.
• NTP/SNTP protocol support added (automatic Time/Date adjustment).
• E-Mail username length is enlarged up to 50 characters (configurable via Web Configurator).
• VoIP Accounts can be stored without user name.
• Info service functionality added e.g. Weather forecast as Screensaver. This feature can be used with C47H, S67H and SL37H handsets only.
• Call Waiting Rejection function added.
• If you try an outgoing call for the first time with a wrongly configured VoIP or Gigaset.net account, you are being asked if you want to start the installation assistant.
• Message presentation improved.
• Phone menu adjusted.
• G729 Codec support improved.

PLEASE LIST ANY OTHER BUG FIXES HERE

The current firmware version 41 seems to work fine with Asterisk and OpenSer for some users. The "realm must be empty"-effect seems to be gone. There might be issues with outbound calling with a S450IP and Asterisk 1.4.x (statuscode 415).

Feature requests

• Provide a list of registered handsets through a web-interface
• Enable the R-button in SIP mode fixed 14/09/2007
• Enable MWI for VoIP
• Add a means to put a caller on hold
• Add a 'mute' option for the microphone
• Option to disable missed calls alert (see note below)
• Option to add an extra leading 0 and a pause for use in PABX
• Allow SIP transfer/consult between terminals of same base (not possible now since they are needed 3 SIP channels in parallel, therefore you will need to use the 'internal' transfer method)
• Allow the use of the SIP Header "Call-Info:\;answer-after=0" to make the phone automatically answer in loudspeaker mode without ringing or user interaction

Known limitations

Firmware Version 41

• The Asterisk logfiles says "Got SIP response 415 "Unsupported Media Type" back from <ip_of_s450ip>. This is only an annoyance, not an error, and it can be easily fixed by not having a 'mailbox=' entry in sip.conf since there is no support for MWI.

Firmware Version 38

• R-button does not work in SIP mode (fixed in firmware 41)
• Caller name length limited to 15 characters

Firmware Version 34

• R-button does not work in SIP mode

Firmware Version xx

• Now with support for up to six VoIP providers (not just 1 with max. 4 accounts)

Details

Cannot disable missed calls alert

If this phone is part of a ringing group it would be nice to be able to disable the missed call alert. So that it does not sit there flashing.

R-button does not work in SIP mode (fixed)

The R-button, which is used to request for transfer etc. in POTS-mode, has no function in SIP-mode. The behaviour is mentioned in the manual. Siemens does not seem to consider this a bug.

Using SIP is not possible to transfer/consult between S450IP terminals of same base

Each base allows just two simultaneous DECT channels. Imagine a SIP enviroment in which each S450IP terminal has an associated SIP account.
If a S450IP is in a call with Asterisk (for example) and the user press "Consult" to initiate other call while the first one is on hold, then he cannot call to other terminal of same base, because that would be 2 DECT channels more:

• First DECT channel: call between S450IP-1 and Asterisk (on hold but alive).
• Second DECT channel: outgoing call from S450IP-1 to SIP account of S450IP-2.
• Third DECT channel: incoming call from SIP proxy to S450IP-2 (NOT POSSIBLE).

So, using SIP you cannot transfer or consult to other terminal in same base, you will instead need to use the 'internal' transfer method between handsets that are registered to the same base.

Comments:

• According to a Siemens dealer this should be changed with a recent firmware upgrade (September 2007) . Can anybody with a S450IP please confirm or refute this claim?
• I confirm that it doesn't work (November 2007, version 41.00)

Caller Name length limitation

The length of the caller name (as set by CALLERID(name) oder CALLERID(all)) must be limited to 15 characters or less. If a longer string is transmitted by the SIP server, the phone will ignore that name and display "Unknown" (or the corresponding localized string). It does not show the first 15 characters!

E-mail

Only one POP address can be checked, and only subject and sender will be shown
POP client does not support SSL - so does not work with Google or Yahoo

HTTP proxy

There is currently no support for a HTTP proxy.

Jabber

Jabber works via port 80 (?). Doesn't work with Google Talk becaseu gtalk requires TLS which the S450 doesn't support.

Line assignment

Each handset has either pstn or a specified voip provider as the default outgoing line. It was not possible to select the outgoing line/identity on a case-by-case basis until the release of firware 41 which allows particular outgoing numbers to be associated with a specific provider or the pstn line via the basic Dialling Plan feature acessible from the web interface, and also allows a number to be dialled with a postifxed identifier, eg ***********#0 will dial via the pstn while ******#2 will dial via the second voip provider and ***********#9 will dial Siemens' own gigaset.net provider.

Firmware update

In order for the firmware update to work behind a LAN router you will need to forward the ftp data port 20 to your phone (with update URL
gigaset.siemens.com/chagall). However, other workarounds using an update URL within your local LAN should also be possible.

photo from base station board :

---

Available from:

• Belgium: VoIPsolutions At 94,99 Euro; Reseller prices available (Other models also available; S675 IP; SL75; C450 IP; C470 IP; C475 IP; Repeater; Doorphone; ...)
• UK and Worldwide: VoIPon - Shipping UK and Worldwide