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Siemens Gigaset S675IP

Created by: mali,Last modification on Thu 25 of Jun, 2009 [03:30 UTC] by e42006

Siemens Gigaset S675 & S685IP

Image

Summary

A VoIP DECT Phone based on the Siemens Chagall Platform. The Siemens Gigaset S675IP/S685IP is a successor to the Siemens Gigaset S450IP. It adds the following features:

  • Integrated answering machine
  • Improved handset with a larger display (S67H for the S675IP, S68H with bluetooth support for the S685IP)
  • Screen saver displaying RSS feeds and other information sources

Official NEW Firmware version V184 (NEWS : Available April 08, 2009)


Firmware Update 04/2009 Version V02184:

Full version number from web interface:
Firmware version: 021840000000 / 043.00
EEPROM version: 125


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.

* provider-dependent 



Official NEW Firmware version V140 (was available January 17, 2009)


Firmware Update 01/2009 Version V02140:

(After FW update country settings for automatic time settings are back to defaults. Telephony/Advanced settings are affected as well. )

Full version number from web interface:

Firmware version: 021400000000 / 043.00
EEPROM version: 125

Improvements:
===============
- Name replacement from online directory caused an error, for very long names.





Official NEW Firmware version V139 (was available from November 25th 2008)


An new official firmware version was announced for the end of october, and is released per 25 oktober 2008.

Gigaset C470 IP / C475 IP / S675 IP / S685 IP Firmware update 11/2008

Download version: 02139

New Features:

   * Online directory for Switzerland provided by tel.search.ch
   * HTTP Proxy support. For example online directory can be used in company environments via an HTTP proxy server.

Improvements:

   * Fast media switching improved (due to partial loss of provider cost rate announcements, e.g. at Sipgate)
   * Indication of VoIP waiting calls improved
   * Firmware update procedure starts before VoIP connection assistant.
   * Store the complete caller information in the "Missed Calls" list, if it is retrieved from online directory
   * Optimised presentation of search results in Yellow Pages (on C47H, S67H, S68H handsets)
   * Gigaset.net assistant starts if the user opens Gigaset.net directory for the first time and there is no corresponding Gigaset.net nickname. 
      It does no longer start after the VoIP connection assistant.
   * The phone deregisters from registrar server, once the SIP account is disabled.
   * Wideband support for A58H/C38H handsets improved
   * SMS wizard disabled (subscription to SMS service is no longer necessary)
   * Handling of IP dialling improved
   * Automatic user login by accessing the Gigaset.net configuration page via Web configurator
   * Handling of multiple network mailboxes improved
   * Translation errors fixed
   * Support of new daylight saving time rule for New Zealand
   * Sipgate echo test supported again (handling of RTP sequence numbers)



Obsolete (not official) pre-release

2008-10-22 Version V02133: 021330000000 / 041.00 EEPROM version: 124

To acces this pre-release goto: gigaset.siemens.com/chagall/T_HJL/chagall133_02.bin
(URL doesn't work : 404) Siemens has been removed this beta firmware which apparently was the only way to fix their "missing calls" bug!
Download location is in below section, there should be no copyright problems as the firmware contains GPL open source software.

New Features:

  • Not yet known, no release notes enclosed

Improvements:
  • Not confirmed yet, no release notes enclosed BUT per reports this fixes the critical "missing calls" bug whereby the phone just drops registrations and ignores incoming calls.

UPDATE: From a source that would like to not be named: http://drop.io/chagall133_02
This is Siemens Gigaset S675IP - Pre-release Firmware 2008-10-22 Version V02133: 021330000000 / 041.00 EEPROM version: 124
Good luck guys! I have not been able to test it myself.



Unofficial pre-releases


2008-10-15 Version V02131: 021310000000 / 041.00 EEPROM version: 124
To acces this pre-release goto: gigaset.siemens.com/chagall/T_HJL/chagall131_02.bin

2008-10-09 Version V02129: 021290000000 / 041.00 EEPROM version: 124
To acces this pre-release goto: gigaset.siemens.com/chagall/T_HJL/chagall129_02.bin

New Features:

  • Not yet known, no release notes enclosed

Improvements:

  • Not confirmed yet, no release notes enclosed


Official Firmware versions


2008-11-25 Version V02139: 021390000000 / 043.00 EEPROM version: 125 (043 could be country specific)

Release notes : gigaset.siemens.com

New Features:
  • Online directory for Switzerland provided by tel.search.ch
  • HTTP Proxy support. For example online directory can be used in company environments via an HTTP proxy server.

Unofficial Improvement:
  • "Missing calls" bug solved

Improvements:
  • Fast media switching improved (due to partial loss of provider cost rate announcements, e.g. at Sipgate)
  • Indication of VoIP waiting calls improved
  • Firmware update procedure starts before VoIP connection assistant.
  • Store the complete caller information in the "Missed Calls" list, if it is retrieved from online directory
  • Optimised presentation of search results in Yellow Pages (on C47H, S67H, S68H handsets)
  • Gigaset.net assistant starts if the user opens Gigaset.net directory for the first time and there is no corresponding Gigaset.net nickname. It does no longer start after the VoIP connection assistant.
  • The phone deregisters from registrar server, once the SIP account is disabled.
  • Wideband support for A58H/C38H handsets improved
  • SMS wizard disabled (subscription to SMS service is no longer necessary)
  • Handling of IP dialling improved
  • Automatic user login by accessing the Gigaset.net configuration page via Web configurator
  • Handling of multiple network mailboxes improved
  • Translation errors fixed
  • Support of new daylight saving time rule for New Zealand
  • Sipgate echo test supported again (handling of RTP sequence numbers)


2008-07-19 Version V02123: 021230000000 / 041.00 EEPROM version: 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

2007-12-13: Version V02097: 020970000000 / 041.00; EEPROM version: 114
Release notes from gigaset.siemens.com, international site:

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




RSS Feeds and Info display

The RSS feed to be displayed is configured through the http://gigaset.net/myaccount web site. The data will be fetched by the gigaset.net server, and delivered to your phone. The server will check for updates every 30 minutes. The phone will only display the contents of the 'title' tag of the first 'entry'.

It is possible to intercept the info display pages and feed custom data to it if you can redirect "info.gigaset.net" somehow, i.e. by running your own name server. Instructions for this are available here.


Known problems

  • First 1.0 - 1.5s of conversation is cut off (you are not hearing your counterparty when calling out)
  • "Call answered elsewhere" reason code in aborted calls is ignored
  • Crash on startup (thus causing reboot loop) when using multiple network voicemail with Asterisk (at least with 1.6.0).
  • Only two SIP invites are processed in parallel, making it impossible to use individual SIP accounts per handset and calling more than 2 with a call distribution. Use an additional "call multiple handsets" SIP account as a workaround if you have less than 6 handsets.
  • No G.722 (HDSP) when using a DECT repeater
  • No support for MWI on VoIP channels, see comments. fixed with latest firmware
  • If the sip.conf section for the S675 IP contains a 'voicemail=<mailbox>' statement, asterisk will complain about '415 Unsupported Media Type' each time asterisk tries to tell the phone about the voicemail status. As PieterB pointed out, asterisk actually ignores the list of accepted applications sent by the phone. fixed with latest firmware
  • The RSS feed only displays the title of the first entry in the feed, and it does not seem to accept all common feeds.
  • Menu 8-5-1 (Call List Type: Missed calls/All calls) is missing. (Does not apply to FW 2123)
    • Workaround: To activate "All calls", press Menu (right arrow), enter 8 5 then 910, display shows: 910 SET (0), enter 1, press OK to activate.
  • CRITICAL BUG with lastest firmware, you will loose calls ! (Does not apply to FW 2123)
    • After a short period of time (1 to 5 days), handsets are no more ringing on all calls, you need to reboot DECT base.(Does not apply to FW 2123)
    • A ticket is open with Siemens Support for this problem.
    • look at discussion tab
  • Possible software crash with low MTU values. TCP-Stack does not react cleanly to PMTU-Discovery.
    • In a VPN environment, the MTU of a link may be set to a very low value to allow safe coexistence of VoIP and data applications (i.e. MTU < 576).
    • All HTTP traffic is using IP packets with DF bit set.
    • When accessing the gigaset.net HTTP services the phone may receive ICMP "fragmentation needed" packets with the proper MTU, but doesn't adjust the packet size.
    • Instead of just failing to access the requested web resource, the phone will retry several times (8-10) and finally reboot or crash (no more IP responses, DECT service down).
    • This misbehavior applies to all firmware including v139.

EDITORS: Do not not use terms like "latest" or "today". They get outdated too quickly.


Remote control

A small bash script to up/download phonebooks and reboot the base is available here.


Available from

Information mainly entered by companies that want to sell stuff.



Notice: You will hardly find a power adapter for this phone, not because the 6.5V power switch is difficult to be found but the adapter that switches to the phone is not a standard one. You have to cut the cord and you loose the guarantee.

Comments

Comments Filter
222

333S685IP Stop Inbound PSTN Calls?

by andymc, Friday 01 of May, 2009 [07:13:51 UTC]
 Can the S685 stop inbound calls from the PSTN from ringing the handsets. I want Asterisk to handle the PSTN calls, but still have PSTN fallback during a network outage. I have seen the options to deselect each handset from the PSTN.

 BUT, I saw a post suggesting that having no handsets selected for the PSTN line would cause ALL handsets to ring.

 Could a current S685 owner do me a (BIG) favor, and test this? ie. Call inbound on PSTN without any handsets selected to PSTN. Hopefully no ringing handsets.
222

333Another S685IP SIP registration problems

by rhcp, Monday 09 of February, 2009 [19:40:20 UTC]
you're not alone.
since latest firmware update i do not have sip registration anymore.
that sucks. siemens sucks.
how could they let suck problems in firmwares...
222

333S685IP SIP registration problems

by simple_sipman, Friday 06 of February, 2009 [11:46:13 UTC]
We suffered a lot of problems with the S685IP and registration - well not specifically registration actually though that is what showed on the handsets.

Every so often for no apparent reason the handsets would show "Provider Registration Failed". On tracing the fault actually it was the fact that the network card on the S685IP base had stopped. There is nothing you can do to revive this situation except for powering the base off and on again. This can even happen whilst in the middle of a call and happened once or twice a day but the pain is that you don't know that is has happened unless you are constantly monitoring the handset. We missed many many calls this way.

We tried a second base, that did the same thing but more frequently. This one would fail 5 or 6 times a day. Many attempts to resolve the issue drew blanks so we have now ditched the S685IP and are using Snom M3's.

We quite liked the Siemens unit, it did everything we wanted and felt better quality than the M3's do but we just couldn't live with the unreliability.

Simple


222

333S685IP SIP registration problems

by simonqro, Monday 19 of January, 2009 [16:02:25 UTC]
I’m having lots of problems with the S658s and Asterisk (1.4.22). I assume the problem is with the base though, not Asterisk.

I have 2x bases; one with 5 phones and another with 3 - each has their own SIP account on the Asterisk box.
The phones de-register themselves from Asterisk randomly (and regularly!) making the SIP channel unreachable. I'm not using NAT.
Some of you here have reported the same issue and it's frustrating after all the time with Siemens they still can't fix it.
I contacted their Support number earlier today to no avail; the usual have you checked your router type messages (after explaining it's all local LAN traffic)

So, I have unfortunately drawn a blank as I can only attribute the issue to firmware on the base (have updated to latest available as of Jan 09 - 021400000000 / 043.00).

Are others still experiencing the similar problems? Or successes?
Can anyone recommend a stable working DECT SIP phone for enterprise use?

Many thanks.
222

333After 4 days update worked

by nbsberlin, Sunday 14 of December, 2008 [16:12:38 UTC]
It took apr. 4 days for the base station to attempt loading the new firmware, but then it worked.
Here is a first quick impression, and I have to say, my two main issues have have been resolved.
- Load process was a pain
- Net AM works with Asterisk
- It does not seem to loose connection any more if multiple SIP accounts are configured

Only Caveat I have seen so far: It was crashing Asterisk if a Net-AM was configured on an deactivated line (I had two lines set up but 3 Net-AM) and Asterisk would crash completly after the first call had been received. However this could also be an Asterisk thing. Since it was a config issue, not much of a problem.

222

333Re: Firmware missing?

by emmanuel38, Thursday 11 of December, 2008 [17:31:18 UTC]
It should work without special setting (gigaset.siemens.com/chagall).

If it doesn't work, it's something like that (no warranty) :
http://gigaset.siemens.com/chagall/2/chagall139_02.bin

222

333Re: Critical Bug on last firmware

by nbsberlin, Monday 08 of December, 2008 [05:23:19 UTC]
Me too... I figured out that it is depending on the number of SIP accounts I have configured. The more I use the worse it get's. With 3 accounts it happend almost daily, with 2 accounts maybe once a week, with one it happend every other week...

And the update does not work either, nor does the MWI.
222

333Firmware missing?

by nbsberlin, Monday 08 of December, 2008 [05:19:53 UTC]
Tried updating today and got into an endless loop of the base - it tries downloading the firmware but no success. Attempted to download the firmware manually following the logic of previous links:
http://gigaset.siemens.com/chagall/T_HJL/chagall139_02.bin
However: I also get the usual 404 message. Did the firmware disapear from the siemens server?


222

333New firmware Dec 2008 - bugs fixed?

by kb, Tuesday 02 of December, 2008 [21:34:53 UTC]
Any reports on whether the new firmware actually resolves the issues mentioned below?
I'm hoping the silence is a good sign that people are happy, but would still be nice to hear from someone.

222

333Re: Critical bug / downgrade firmware ?

by owagner, Sunday 16 of November, 2008 [21:34:40 UTC]
For the SIP registration error, try to only set ONE voicemail account in the SIP settings (for one of the SIP accounts).