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Fri 08 of Aug, 2008 [21:20 UTC]

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Siemens Gigaset S675IP

Created by: mali,Last modification on Sat 19 of Jul, 2008 [09:10 UTC] by ASort

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Siemens Gigaset S675 & S685IP

Image

Summary

A VoIP DECT Phone based on the Siemens Chagall Platform. The Siemens Gigaset S675 IP is a successor to the Siemens Gigaset S450IP. It adds the following features:

  • Integrated answering machine
  • Improved handset with a larger display
  • Screen saver displaying RSS feeds

The S685 IP has an identical base station, but uses a more advanced handset which offers bluetooth hands-free operation.



Firmware versions

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

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'.

Known problems

  • 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.
    • 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.


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

333Re: Jabber Account per handset?

by ASort, Thursday 03 of July, 2008 [11:45:21 UTC]
Yes, it is only possible to set up one Jabber account pr base station (S675IP).
The same goes for email check. (it should be one pr handset !)
222

333MWI Problem

by strikegun, Friday 25 of April, 2008 [07:34:22 UTC]
I just got a mail from Siemens:

Das Gigaset S675IP akzeptiert (RFC-konform) nur SIP-Notify Messages die auch vorher via Subscribe-Message angefordert wurden.
Die meisten übrigen SIP-Telefone am Markt akzeptieren jede (auch nicht-subscribte) Messages. Asterisk unterstützt erst ab Version 1.4 SIP MWI-Subscriptions.
Prüfen Sie bitte, ob Sie eine Asterisk Version > 1.4 nutzen.


In english it means that the S675IP only take SIP-Notify Messages wich are requested before. Asterisk > 1.4 uses the SIP MWI-Subscriptions.
I hope this information helps.
Richy
222

333Problem MWI with Firmware version: 020970000000 / 041.00; EEPROM version: 114 not resolved

by llevet, Tuesday 12 of February, 2008 [11:24:54 UTC]
Good product. But miss some feature ...


Message waiting indicator still doesn't work on voip with firmware version 02097 (12/2007).
S675IP doesn't fully compatible with rfc3265 MWI (http://www.ietf.org/rfc/rfc3265.txt),
The error is:
-- Got SIP response 481 "Call Leg/Transaction Does Not Exist" back from <IP>

Works on asterisk PBX (version 1.2.25 and 1.4.17), but without MWI. It is a big problem for me.

Waiting for a new firmware, and hope it include MWI bugfix ...

P.S : have same problem whit a c470ip (same firmware)

Ludo.
222

333call disconnects every 30-32 minutes from SIP server

by dkounal, Monday 14 of January, 2008 [17:20:53 UTC]
For some strange reason, it does something every 30-32 minutes. If a call is on the way at this time it disconnects and the screen says that connection with SIP provider was lost. That happens with asterisk v1.4.10 and the same with a voip provider (i-call). This does not happen with other voip devices in the same configuration that work in connection with the same servers. The asterisk logs show a disconnection from the the phone in the debug.
The same happens with a C430 IP

Any idea?

Problem fixed: I disabled all communication services, Gigaset.net, e-mail checking, etc and now everything is fixed. My e-mail was checked every 30 minutes and probably this caused the problem, my full mailbox probably. You can add it as bug probably!
222

333MWI still not working for me

by jadler, Monday 31 of December, 2007 [01:27:09 UTC]
Sure, with latest fw I do not get the 415 error, I get another error instead. Really nice bugfix, Siemens... :-(

Yes, they added the possibility to check voicemail on other servers, not just on the one in the base station (that I have never and will never use), but at least my Asterisk will give me other errors when I try to use it. Have you had better luck, please let me know, and tell me how.

Add: The error is:
-- Got SIP response 481 "Call Leg/Transaction Does Not Exist" back from <IP>

I just left a message for myself, my Linksys SPA941 lights up right away, the Siemens S675IP still has no clue about any messages waiting.

I am just waiting for the day when I can buy a Snom M3 (or a few of them) and ditch this Siemens crap.
222

333Re: 415 Unsupported Media Type

by neilplatform1, Sunday 30 of December, 2007 [21:37:03 UTC]
I believe this is now fixed with the latest firmware.
222

333Re: 415 Unsupported Media Type

by olivier1010, Saturday 29 of December, 2007 [15:31:53 UTC]

As this phone is the only one to exhibit this MWI problem, i think that Siemens should correct this asap, Asterisk is a defacto standard now. Siemens is making users unhappy trying to use special methods in their firmwares.


222

333MWI does not work with Asterisk

by olivier1010, Saturday 29 of December, 2007 [15:28:04 UTC]

This phone seems to have the same problem as the C470IP do have.

The message waiting indicator does not work. In rare occasions it is possible to get the MWI lamp flashing. Most of the time it is after the base has been rebooted. After a first message indication it will not work again for subsequent messages.

I would have expected a better tested hardware from Siemens. I ask myself if they are producing IP phones to earn money or only to write their Name on the phone body.

We had some big missing functionalities with the C450IP (no IP call transfer for example), this has never been corrected. I hope that the C470IP / S675IP firmware will receive more attention from Siemens.


Hopefully we have now other serious DECT SIP manufacturers.


222

333Jabber Account per handset?

by mmcnamee, Monday 17 of December, 2007 [12:43:34 UTC]
I have just spoken to Siemens technical support, with a pre-sales technical enquiry. They told me that the S450IP and indeed the S675IP only supports one Jabber account per Base-station, not one account per handset. Can anyone who has either system confirm this either way? Ideally, each handset would be assigned to a particular user, and their Jabber account registered on their own handset, this would also allow inter-hanset messaging, I'm at a loss to believe Siemens would have implemented this as one account per system!

Can someone with either set of phones confirm this either way, I'll even setup some accounts for you to test it if you need that!! :)

Cheers,
Mark
222

333Re: 415 Unsupported Media Type

by jadler, Tuesday 06 of November, 2007 [19:46:40 UTC]
OK, I found one solution, not too elegant though.

In sip.conf, comment out or remove the mailbox= statement:

;mailbox=42

Then asterisk will not try to tell the phone any voicemail status. I wish there was a more elegant and functional solution, but it will have to do for now.