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Fri 29 of Aug, 2008 [07:52 UTC]

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Siemens Gigaset C450IP

Created by: dominig,Last modification on Mon 21 of Apr, 2008 [11:59 UTC] by Mol99

SIEMENS Gigaset C450 IP


This is the first mass market DECT / SIP phone. It started shipment in mainland Europe during summer 2006. It sells over the internet for less than €85 (£58, $109) including shipment.
Note that simliar HW is available without the IP connection (model C450). Be careful when you order your unit.

The UK variant of this phone is the C460IP. It is the same thing, and even says C450IP in the web interface.

The phone is based on the Siemens Chagall Platform.

Siemens provides detailled documentations (including user manuals in various languages) over the Web.
http://gigaset.siemens.com/shc/0,1935,hq_en_0_122378_rArNrNrNrN,00.html

Highlights

  • Dual mode: Easy switch from VoIP calls to fixed line calls by single keypress
  • Easy configuration of internet telephony (VoIP) without a PC (from hanset) or with a PC via the integrated Web server
  • 1 VoIP call and 1 fixed line call in parallel with multiple handsets
  • Convenient handsfree talking
  • STUN/NAT support
  • DHCP or fixed IP address allocation
  • Port allocation fully configurable
  • Dialling by directory numbers and IP addresses
  • Support of DECT repeater for extended Wireless range.
  • Multiple language support
  • GAP compliant (support up to 6 DECT phone connected to a single base)

The Gigaset C450 is an entry phone in the Siemens Gigaset product line, the addition of IP connectivity via a standard Ethernet 10/100 BaseT interface and SIP protocol within a reasonable price bracket, opens large applications where wireless operation and SIP are required.
The Ethernet connectivity allows the unit to work without a PC.
The dual mode (SIP/PSDN) enables incoming call on legacy interface with existing directory number and safe emmergency outgoing calls.
The presentation of the unit with an independant base and a dedicated phone charger/support will simplify cabling.

Integration with well known VoIP operators is preset. Alternative configurations can be entered directly via the handset or the integrated Web server. Multiple languages support is also provided for the integrated Web server.
Simple debug (SIP message status display) can be activated on the handset for a debug without PC. This function is very usful in case of professional application where remote user support is required.

Configuration in Asterisk is straitforward and interworking with other asterisk phone is transparent.
Under Asterisk the follwoing codecs can be activated on the phone G726, G729 , G711 aLaw and uLaw

The Gigaset C450IP is based on an Open Software which can be requested for free to Siemens. A future version which will support PPPoE is announced for later.
Source code can be downloaded at http://www.siemens.com/developer/c450ip

Note : The support for Direct Dial In DDI to multiple handset registered on a single base, is not described in the manual and likely not available neither for SIP or PSDN incoming calls.

VoIP User Review of the Gigaset C450

There is another Siemens VoIP DECT Phone named Siemens Gigaset S450IP, which seems to share a common base station with the C450IP but uses a different Handset.

QoS

RTP data: Class Selector 5, Differentiated Service Code Point 46
Resulting DiffServ value: 0xB8
SIP data: Sent with standard 0x00
Older firmware versions used Class Selector 6 for RTP, DiffServ 0xC0

Bugs - Missing key features in firmware version 00 / 034.00 (EEPROM version 70)


  • The "R" key, which is the hook flash button when used in PSTN mode, does nothing in VOIP mode, so I cannot put a call on hold and transfer it. I am using Asterisk and DTMF signalling via SIP info. Siemens are working on this feature currently for S450IP hopefully also in C450IP too

  • UPDATE Dec/06: This bug is still present even in the latets releases.
  • Alert_Info is not supported for distinctive ring. This would not be a big problem if there was not a side effect. Using Asterisk and ring groups for example, it is desirable to use Alert_info for external call distinctive ring. Then, when the C450IP receive the Alert_Info message it simply does not ring ! There is no simple way inside Asterisk to solve this problem, except manually editing configuration files to allow for selective Alert_Info. But it is a nightmare with FreePBX as the main files are updated at each GUI refresh...
  • EDIT: Actually, there is a way to have alert_info work with other phones and still have the C450IP work with the same configuration. Even though the C450IP will not have a discinctive ring. Just make sure that you set your alert_info variable in the following way: exten => s,1,SetVar(_ALERT_INFO=<something>). Note the < and > in the command. The C450IP will still ring, and at least the other phones I have tried with will accept that syntax.

Siemens costumer care forum: click here


Update Firmware from local Webserver

You will most likely have wondered, whether you will require "siemens.gigaset.com" to be available and resolveable by your DNS for updating your C450IP's firmware.

Obviuosly, you will need to get the files from "siemens.gigaset.com" before you can supply them on your local webserver (e.g. if you have more devices in a small business or if you're just a freak like me). You will need to setup a directory on your webserver which is accessible from within your network. It doesn't matter how you call it and how the url will look to the phone BUT you need to observe the following:

Where /~ is the root directory for the webserver-directory you want your C450IP to look for the new Firmware, you need to have a directory /~/0.
If you copy & paste the following PHP-script to /~ and execute it, it will get all the files that are needed for a successfull firmware update:

<?php

/* This script is designed to get firmware files for Siemens Gigaset
   C450IP from Siemens' server and copy them to a local directory
   for possible re-supplying within a small home/business network.

   This script is freely useable but please remember to inform
   author Kai Michael Poppe at voip@poppe-online.de about any
   of your changes if you find it appropriate to do so.

   GNU Public License applies.

   Version 0.1 as of December, 28th 2006 00:30 UTC +0000
*/

$baseurl = "https://gigaset.siemens.com/chagall/1/0/";
$basedir = "0/";

function returnFilenames($file) {
  $baseurl = "https://gigaset.siemens.com/chagall/1/0/";
  $fcont = file_get_contents($baseurl.$file);
  preg_match_all("/(\.\.\/)?([a-z0-9_]*\.bin)/",$fcont,$fhits);
  return($thits[0]);
}

$getFilenames = Array("master.bin","../baselines.bin");
$x = returnFilenames("../baselines.bin");
$getFilenames[] = $x[0];

foreach($getFilenames as $fil) {
  $fp = fopen($basedir.$fil,"w+");
  fputs($fp,file_get_contents($baseurl.$fil));
  fclose($fp);
}

?>

You are welcome to test this script and report your findings and possible suggestions for extensions to voip@poppe-online.de

Configuring Asterisk and C450IP to work together


Although it seems very strange, you need to set the following options in the C450IP configuration to the IP (or DNS name if you have a DNS server) of your asterisk server:
  • Domain
  • Proxy-Server
  • Registrar-Server
Furthermore, the Realm-option needs to be _empty_!

In sip.conf, the user you choos for your phone, needs to have (besides all the neat stuff) at least this parameters:
  • host=dynamic (even though you might have given the phone a static IP address. If you set host=IP Address asterisk and phone will work together, but both will show registration errors)
  • dtmfmode=inband (which is, if you want to have a voice/dtmf controlled menu working when the phone calls a specific extenstion)

These information apply for: Asterisk 1.0.7-BRIstuffed-0.2.0-RC7k and Phone Firmware 00 / 034.00

For firmware version 038.00
  • host=dynamic
  • dtmfmode=rfc2833 ;more stable solution


Enabling call transfer feature

Although C450IP has a "R", it seems Siemens is not willing to make it work over IP.
However, it is possible to take advantage of asterisk's builtin transfer capability. Here is how:

  • make sure chan_local.so is loaded (e.g. checkout modules.conf)
  • edit features.conf and have at least something like this (of course you can change key mappings as you like):
[general]
featuredigittimeout = 999 ; default of 500 is short
transferdigittimeout => 5 ; default of 3 is short

[featuremap]
blindxfer => #1
atxfer => #2

With firmware version 038.00 no changes to extensions.conf where needed!

  • change your dialplan (extensions.conf) to make sure that each Dial() entry has the 't' or 'T' option active (the lowercase 't' permits the called party to initiate the transfer by pressing '#', the uppercase 'T' permits the calling party to initate the transfer; for instance, you will set a 'T' for outgoing calls and a 't' for incoming calls, while a'tT' is recommended for calls between internal extensions)
[extensions]
exten => 210,1,Dial(SIP/210,,oTt)

[outgoingtrunk]
exten => _XXX.,1,Dial(Zap/g1/${EXTEN},,T)

  • make sure your C450IP configuration has dtmf tones sent via SIP INFO (optionally also via inband audio)

That's it, it works on Asterisk 1.2.x (never tested on 1.4.x but there should be no difference at all)


Available From





See Also


Comments

Comments Filter
222

333Calling IP address?

by javame, Tuesday 12 of February, 2008 [09:13:07 UTC]
Hello, I have some problem with this nice phone. I need to use it in the plant, where is possible only calling of IP address, not number extensions. From C450 I can call IP of all phones, but when I want to call IP of C450, others phones said it is busy. Has anybody the same problem? Would anybody know the solution?

(sorry for my english, I've never been good student)
222

333Registration problem

by silencer, Tuesday 23 of October, 2007 [14:06:40 UTC]
I have C450IP with firmware: 01070 and eeprom: 87

I can register on asterisk but the first time, the register always fails.

Then even if i'm registred, i can not answer calls and if i place a call from the c450ip, i have a one way audio !

The phone is not behind a nat...

I have made some tests in july and it was working perfectly.... so do not upgrade your phones ! As you can not downgrade after.

if somebody have an advise, I'm interested in !
222

333Put text on the screen of the C450IP

by johanvloet, Sunday 05 of August, 2007 [17:24:29 UTC]
HI,

Is it posible to send some information to the screen of de C450IP, It has 3 lines which I could use to giver some information back from a computer within the same network. Or does any body know if it's possible or if there is special software for it. Can it been done by changing the opensource software from siemens?

Greetings and thanks

Johan
222

333Put text on the screen of the C450IP

by johanvloet, Sunday 05 of August, 2007 [17:17:14 UTC]
HI,

Is it posible to send some information to the screen of de C450IP, It has 3 lines which I could use to giver some information back from a computer within the same network. Or does any body know if it's possible or if there is special software for it. Can it been done by changing the opensource software from siemens?

Greetings and thanks

Johan
222

333Problems with C460IP

by barns, Friday 20 of April, 2007 [16:53:06 UTC]
Hi,

I have just bought a C450IP, which seems to work ok apart from 3 bugs and 1 lack of a feature.

1) No MWI
2) 15 Character CLI Name Limit, that replaces the CLI Name with "unknown" if it is greater than 15 characters.
3) Disconnects and re-registers every 3 minutes, very annoying if you're on a call.
4) A Nice feature would be to be able to turn of missed-call notification, as when used in a call group it sits there flashing saying i've missed 30 calls.

Current Software

Firmware version: 010590000000 / 038.00
EEPROM version: 86
222

333problems with eeprom version 84

by babyloon00, Wednesday 07 of March, 2007 [09:23:19 UTC]
I already use 3 Gigaset C450 IP (eeprom 71) in my company.
They work very well.

Recently, the last buyed was with eeprom 84.
Every 3 minute, it disconnect and show a message saying "server ont accessible"

I try some changes in the phone config, but it did'n work.

Does anybody get this kind of issues?

Thx
222

333Dialplan issue ...

by drogon, Sunday 25 of February, 2007 [22:15:13 UTC]
Another thing to note with these phones is that they will treat a number that ends in a star as a number to go out over the POTS line, rather than the VoIP "line", so if you have any "star codes" in your dialplan that end with a star, then they won't work!
222

333Re: C450IP call transfer

by drogon, Saturday 24 of February, 2007 [22:48:32 UTC]
Just bought a pair of these. Very nice phones - lacking in MWI function though which is a real shame - my 4 year old pure analogue DECT system lights up it's little mailbox icon when theres voicemail waiting, why can't these...

As for call transfer - I had to make a few more changes in features.conf:

featuredigittimeout = 999 ; The default of 500 was too short
transferdigittimeout => 8 ; again, default too short

I also found that with the latest firmware, 00 / 038.00, eeprom version 84
then the DTMF setting of rfc8233 worked just as well.

I'm also using:

featuremap
blindxfer => #1 ; Blind transfer
atxfer => ## ; Attended transfer
disconnect => #0 ; Disconnect

Which is going well. (The people who'll be using these will be mostly doing attended transfers, so ## to kick it off, and #0 to recover the original caller makes life easier for them).

G
222

333C450IP call transfer

by agpastore, Thursday 15 of February, 2007 [09:38:33 UTC]
I've added the required steps to enable C450IP call transfer with the '#' key in the main page. Check it out.
222

333Re: C450IP in a professional installation

by mmto60, Wednesday 14 of February, 2007 [12:24:58 UTC]
Can you post sip.conf and extensions.conf to try to configure like you?.

I can't transfer a call between extensions. I try to do it using "R" or "#" and I can't.