Siemens Gigaset S450IP and C450IP Phones

[couplands]

Hi,


I was wondering if anyone has got the new Siemens Gigaset S450 IP or C450 IP phones (these are the VOIP not the pure analogue versions) working on an Epygi. I cannot register the phone with the Epygi and get either "Registration failed" or "Server not accessible" errors on the phone.


The account on the Epygi is valid as I have tested successfully with other VOIP phones including the new Siemens Gigaset SL75 wifi (this has a similar but not identical SIP setup screen). The phone and Epygi firmware are the latest.


The Siemens Gigaset S450IP and C450IP phones are attractive as they are quality DECT handsets/bases with both SIPVOIP and analogue connections.


Many thanks, Lindsay

[couplands]

It's always the case... try for a couple of days to get something working, then post a message, then trip over the answer a few minutes later!.


The phone registers fine -


- If the Authentication Name is used in addition to the username and password... the other models didn't need the Authentication name filled in.


- Don't trust the Siemens Telephony/Connections/Status screen... it can say that there is a registrationfailure when the device has in fact registered Ok. This may be an issue with the screen finishing refreshing before the registration is completed.

[nlucas]

couplands hi!

lol! i was posting the same thing...and you give me the answer...

it work's fine now!!!



best regards!

[ppvote]

Indeed you get a refresh LONG before it has had a chance to connect. Press f5 in your browser to refresh.

I do have issues with mine sometimes if I do a power cycle - I am set up as a remote extension (as I am 100miles away from the main office) and after a power cycle it fails to register. Though if someone at a different location registers their phone as me, then I register again, it seems to come back - I think there is some VERY strange routing issue there.

One question for you though - which I have not been able to resolve.

Can you get the R key to work RELIABLY to stick a call on Hold and transfer it? Or is it possible to tell the Epygi to use the * key or something else instead?

Seems to be OK if you use it as analogue, but not as VoIP.smileys/smiley7.gif

Thanks

[davrays]

Well, I am not familiar with those phones, but would like to make one general statement: on SIP phones there is no need to user "R" key: normal SIP phone has special keys to make hold and transfer. Moreover, as far as I know Quadro will not recognise the OOB DTMF 16, if phone sends that. Resume: forget about "R" key on SIP phones.

[ppvote]

Well the bad news is that even on the latest version of firmware (020810000000/041.00) the Siemens phone does NOT HAVE a special key for Hold / Transfer

The soft keys go to Ext (which will allow you to make an external call if you have another VoIP port available) and options - none of which are helpful...

Holding down other keys does not appear to assist either.

Ideas on a postcard? These are great phones, but not getting too far with them...

FYI - the 'r' key works 'sometimes' when you use the phone as an analogue extension.

[davrays]

Well, theoretically the"R" should work 'always' when the phone is used as analogue phone. If there are problems, it could be because of too long or too short "flash" timeout configured on phone. Quadro accepts by default the flash timeouts from 100 to 800 msec. If phone flash timeout is bigger (or equal to 800 msec), it can be adjusted either from the phone, or from Quadro's "linesconfig.cgi" webpage.


Concerning using this phone as SIP phone... Any SIP phone should be able to send re-INVITE (for hold) requests and REFER (for transfer)request. If this phone doesn't have special keys for hold or transfer, anyway it has to have some way to force it to send the mentioned SIP requests. Probably there is some tricky way described in documentation of the phone...


Regards,


DVREdited by: davrays

[ppvote]

OK - spoke with Siemens and IT DOES NOT do re-INVITE or REFER.. the ONLY 'sort of' hold you can get is to hit the button that says INT (as if you were going to transfer to another handset on the same base) which will put them on hold, but not so that you can transfer them off anywhere.

They said they will request it, but also suggest anyone who wants it requests it also by going to http://gigaset.siemens.com/ clicking on contact in the top right and filling in the form - my suggestion would be to put it under the options soft key (right hand one) whilst in a VoIP call.

smileys/smiley7.gif not very happy with the functionality of the phone - it is NOT true VoIP - also just checked out the price of the door bell unit (HC400 - it is about £400 on Ebay Germany!) smileys/smiley7.gif

[AramK]

So davrays were quite right in his first comment, describing "normal IP Phones" and stating "forget about "R" key on SIP phones." :) Conclusion: Not everything that shines is gold.

[edgarlopes]

Hi everyone.


I'm studing the same problem for the last 4 months. I have 27 S450IP and 10 SL37h in some of my costumers. This means that I have a real problem to solve ASAP.smileys/smiley7.gif


Here some good news smileys/smiley4.gif(I haven't tested yet!!! But nextThursday I have the results)


From Siemens Website:
Gigaset S450 IP (version 02081)


Download version: 02081


Firmware Update 09/2007 Version 02081

New Features:

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. smileys/smiley4.gif(Depends on the provider).
Call rejection via Onhook-Key added.
IP address exchange via paging call.
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. [/list]


Improvements:


Message presentation improved.
Phone menu adjusted.
G729 Codec support improved.[/list]


Now it's only missing the information "How to transfer or holda SIP call"smileys/smiley18.gif.


Well, lets see what I can do and next week I post here the ("GOOD" I hope) results.


Best Regards

[AramK]

That's a good news, Edgar, because many our customers are using that phones and usually they think that the problem comes from Quadro, but not from phones itself. We had really hard time withsuch requests, explaining them the root of problem,so if you aware that some people are using that phones with Quadro, please informthem about the phones' specifications. For now let's wait and hope that the"VoIP Call Transfer function", as it is announced,will be implemented according to standards.

[ppvote]

Here is a quick screen shot of the new options, but I don't think this really helps - as said before - I have been told the best thing is to get as many people to put in a request through their website as possible...

http://www.cinimod.co.uk/epygi/1.gif

I do not know if any of you are aware of Gradwell as a trunk provider, but they also run a per extension service - they have a special setting for those poor souls who run Siemens devices at the moment!

The nice tech guy (Tom) also told me just now that he believes Siemens are working on proper SIP transfer and Hold commands - no idea how long though!

http://www.cinimod.co.uk/epygi/2.gif

[AramK]

You are quite right, this new option will not work, because as davrays wrote before, this is based on sending OoB DTMF 16, which is not recognised by Quadro. This is not a true IP (SIP) Hold/Transfer way.

[ppvote]

I was hoping the 'call transfer' bit at the bottom may help...

Here is another issue for you with these otherwise good phones.

Have you tried the S450with a headset?
When you plug in the headset it does NOT disable the phone mic, so the phone still picks up the background noise! (Which if you have the handset on your desk and are typing can be quite loud!)

I have logged this with Siemens and apparently it is a known issue - not sure if or when they will fix it though!

Wonder if it is worth getting a different make of DECT phone to use with the base... any ideas which others they support?

[jonathand131]

I've brought a Siemens 470IP which is almost the same phone (model number change depending on country) and I've the same problem : I can not make call transfer.

I've seen two options in phone configuration interface but I don't know which value I should write to make it work.

If someone has a suggestion ...

Options and current value :

Hook Flash (R-key)
Application Type: dtmf-relay
Application Signal: 16

[davrays]

Hmm... does this phone work with any PBX?.. I mean if its method of transfer is not based on REFER or any other SIP message, I doubt it could work. It is really strange why people are implementing phone such way, that is doesn't work correctly with any 3-rd party IPPBX...

See this this thread:
http://www.trixbox.org/forums/trixbox-forums/help/flash-hook-r-key-problem-dect-siemens-470-ip.
Looks the transfer doesn't work with Trixbox too.

The only way to make the transfer on that phone to work, is to see how it can send SIP REFER message. As I told above: forget about "R" button on SIP phone. That is ridiculous, I don't see how and why any SIP phone could use that...
May it be that this phone allows to configure some codes starting with "R" to send normal SIP REFER messages to PBX?

[jonathand131]

There was a firmware update today for my C470IP :
In the release note ( http://gigaset.com/repository/1693/169367/readme_S6x5_IP_C47x_IP_S685_IP_eng_V02184.txt ), it is mentioned "VoIP: Call transfer via R key".

This is good news but after updating my phone, the R key still doesn't work with IP call. But, I've found that I can do transfer call using "Ext.call" function which is displayed during a call and then using "Options/Call transfer". Don't know if this feature was added by this firmware update or if it was already available before.

[KSComs]

Hmm... does this phone work with any PBX?.. I mean if its method of transfer is not based on REFER or any other SIP message, I doubt it could work. It is really strange why people are implementing phone such way, that is doesn't work correctly with any 3-rd party IPPBX...

See this this thread:
http://www.trixbox.org/forums/trixbox-forums/help/flash-hook-r-key-problem-dect-siemens-470-ip.
Looks the transfer doesn't work with Trixbox too.

The only way to make the transfer on that phone to work, is to see how it can send SIP REFER message. As I told above: forget about "R" button on SIP phone. That is ridiculous, I don't see how and why any SIP phone could use that...
May it be that this phone allows to configure some codes starting with "R" to send normal SIP REFER messages to PBX?


Actually david maybe it has a standard PSTN connection and if you had some form of call waiting activated on your PSTN line or if you had the phone connected to a tdm pbx extension then the (R) key would be used as a recall facility or a key for sending a flash ( timed loop break - usually around 100milliseconds or 600 depending on which side of the sea you are from )... sometimes it is used as transfer....

Regards

Kev

[KSComs]

By the way ... under Asterisk, it is still possible to use this phone to transfer an incoming call ... without the use of a transfer key ... it is by use of a feature code enabled under feature.conf called blind transfer by use of the # key.

Sending this will inadvertantly set up a transfer from the handset ... now I dont know if we can accomodate a hack like this in perhaps a feature page to be provided by Epygi in the future or not...

Personally, I would like to see other more important things than getting budget cordless telephones to work ( my opinion - but heck im use to Kirk Dect ) such as Park groups for groups of extensions.... ie 3 extensions able to use park 1 to 2, 5 extns being able to use park 3 - 8 etc....

Mr Hrant please make it so !!! I been good.... :D

Kev

[davrays]

Well, as soon as this "Ext.call"->"Options/Call transfer" function exists, I think it doesn't really worth thinking about special support for intruducing a different special code for such phone...
What about the rest, I'll remind Hrant about your "more important thing", Kevin :)

Best regards,
David

[jonathand131]

There was a firmware update today for my C470IP :
In the release note ( http://gigaset.com/repository/1693/169367/readme_S6x5_IP_C47x_IP_S685_IP_eng_V02184.txt ), it is mentioned "VoIP: Call transfer via R key".

This is good news but after updating my phone, the R key still doesn't work with IP call. But, I've found that I can do transfer call using "Ext.call" function which is displayed during a call and then using "Options/Call transfer". Don't know if this feature was added by this firmware update or if it was already available before.

In fact, my phone wasn't at the latest version.

Now, there is a new option in the phone's web interface which is disabled by default but that you can activate to use R key to do call transfer.

But it still doesn't work for me.

"Use the R key to initiate call transfer with the SIP Refer method.: Yes No"

[davrays]

So at least theoretically the problem should be resolved now. Did you check if the phone is really sending REFER when you press the R key with the new option activated?

[jonathand131]

It appears that it only send a "Flash" RTP Event.

Here is RTP part of sent IP frame:

80e5038900d4bc00f3bef1fe100a00a0


In phone's web interface, for "DTMF over VoIP connections Send settings", both "RFC 2833" and "SIP Info" are activated and "Use the R key to initiate call transfer with the SIP Refer method" is set to "Yes".

[davrays]

You need to force the phone to not send "SIP Info", nor the "flash" in RTP (RFC 2833). Itr should send the SIP REFER only. Can you try to disable both "RFC 2833" and "SIP Info", to see what does it send in that case, when you press R key and dial a number to tranfser to?

[jonathand131]

When the three options "audio", "RFC 2833" and "SIP Info" are disabled, nothing appends when I press the "R" key during a call.

Moreover, dial keys doesn't work when these options are disabled.

[davrays]

Well, we could do a lot of assumptions, but easier to call to their Customer Care hotline and ask if they have bug there and how/when they are going to fix that... :) For France I guess it should be here: http://gigaset.com/shc/0,1935,fr_fr_0_37374_rArNrNrNrN,00.html.

If the option tells "Use the R key to initiate call transfer with the SIP Refer method", and that option is set to "Yes", the phone just should send REFER. If it doesn't, they have to explain...

Best regards,
David

[nkoanyane]

I also have the same problem.
I have a quadro 2x and four gigaset A58H ip phones.
I cant figure how to register the phones in the quadro 2x.


Please help.

[davrays]

I don't think you have the same problem. The only problem with the Gigaset phones, is related to transfer only. The rest is working ok for anyone who posted here, including registration.

What specific problem do you have with registration? What values did you enter on Quadro and on Gigaset to register?

[EricM]

Hi,

I read all the pots because i have the same problem... impossible to forward. But i saw that e firmware update (c470 IP) add two new item on the web configurator "Direct transfert" and "supervisor transfert".

I'm not a specialist, i try some différent configuration but without succes... too much settings.
Anyone succeed to forward a voip call ?
Please, help me to choose the rights settings below :
DTMF over VoIP connections

Send settings: [Check Box] Auto [Check Box]Audio [Check Box] RFC 2833 [Check Box] SIP Info

Note : When using G.722-Codecs (wide-band connection) DTMF Signals cannot be transmitted over audio.
Call Transfer

Use the R key to initiate call transfer with the SIP Refer method.: [Check Box]Yes [Check Box] No

Transfer Call by On-Hook: [Check Box]Yes [Check Box] No
Derive target address: [Check Box]from SIP URL [Check Box]from SIP contact header
Find target addr. automatically: [Check Box]Yes [Check Box] No
Hold on transfer target:
[Check Box]For attended transfer
[Check Box]For unattended transfer

Note : Hook Flash (R-key) R key settings are disabled because the R key is being used for call transfer.




Anyone succeed to forward a voip call ?

[davrays]

Eric, I don't have that Siemens Gigaset phone to play with, so I cannot tell you how to make it work. Hopefully someone has managed to configure it correctly, and will reply to you.

But I can tell you the right values (the ones which should be the best to use with the Quadro) for some of the options you asked:

DTMF over VoIP connections
Send settings: RFC 2833

Call Transfer
Use the R key to initiate call transfer with the SIP Refer method.: Yes
Transfer Call by On-Hook: this is up to you. This defines whether you want the call to be dropped or transferred, when you put down the handset, while having one holded and one active call.
Derive target address: from SIP URL
Find target addr. automatically: ? (see help for this device)
Hold on transfer target: ?

Best regards,
David

[EricM]

Thank a lot. I'm going testing and back to u.

Thanks a lot

Regards

[EricM]

It doesn't work. The key 'R' do nothing... Thanks.
Best regards

[Narval]

It seems it won't exist any firmware upgrade for C450 IP.

Firmware upgrade is only available for S450 IP.

C450 IP is (sorrowly) trashcan ready !

[kahalary-harry]

Thanks all for the tips on how to register these units, I have a customer who wants to try and use his left over stock of C455IP sets.

So here is the jist: They register on the Epygi (Q6L on 5.1.18), I can call the assigned extension and answer any call to it, internal and external, can transfer calls to it, but I can't make calls from it, not even internal.

There are very few settings one can change on the C455IP with regards to call and line preference, and even less on the hand set it self.

Is there any thing I can look at on the Q6L? I might be missing a simple setting some where.

The SNOM phones I installed on the same Q6L work just fine, AA works fine.... I told the customer that there is no known solution for the problem of transferring calls from one of those C455IP's, but they just want basic functionality (make and receive calls).

Thanks for any suggestions :)

Regards

[KSComs]

With the SNOM M3 Units being Superceded, and work very well the Epygi Quadro's, why not tell the customer to use EBAY and sell the over stock of C455IP units, then buy all of your new proposed SNOM M3's that you are going to sell him at the right price ?

Hows that for a solution and one that will work : )

It is still a suggestion none the least but would privide a WORKING solution that is tried and tested and you do not have to loose too much sleep over it nor does your customer!!!.

Just a 2.5 cents worth...

Kev

[kahalary-harry]

Hi Kev - thanks, my original quote to customer was for SNOM300's all round. They do not really need portable phones, however the customer has had the wool pulled over their eyes a few times in the past few years by resellers of other brand PBX's, hence their plea to try and make it work with those C455IP sets. I did tell them that it is not a tested phone set / solution from an Epygi point of view, and no guarantee that we will get them to work, so this is on a best effort basis.
This is one of those customers who will recommend us to their colleagues and clients if we can deliver good service.
Not with standing, they do appreciate that they might have to settle for new hardware.

I am trying to get some support from the local Siemens crowd on this, will let you guys know how that works out (!).

Thanks and regards

[KSComs]

Out of interest, and I must admit I havent scanned the rest of the thread, but does the handsets have any DTMF settings that you could play with ?

Any chance of taking a snap shot of the GUI and posting here of the base station Sip area?

It might be that someone might recognise an area or menu that will hold the key ... just a thought..

Kev

[kahalary-harry]

yes, it does have some settings about DTMF / SIP, the link below provides a reference guide to the GUI, the images are not that clear, and there is not much elaboration on the actual settings other than a few short notes.

http://www.asteriskguru.com/tutorials/c455ip.html

The online help menu accessible through the GUI is not all that helpful, but I will see what else I can find.

Thanks Kev

(Edited: most of this post was about how to solve the problem of registering the base station on the Epygi and then later on about the fact that the "R" key does not allow call transfer, even though there is a setting in the GUI that changes the protocol to IP PBX call transfer. FYI)

[KSComs]

Ok.... I have the answer for you.... they can use the Handset via the PSTN connection to the telephone system, the only thing is you'd need to provide an analogue ata to interface to the Handset, a SPA3102 will cost around $35 us which will give you 2 x Analogue ports on the 6L per ATA .. So if it is a matter of using those devices, they can send and receive a call and transfer the call to other extensions etc.. just not use the SIP component.

The reason i say this ... on the Website it mentions the following words - "8. R-key
(not for VoIP connections)"

~~~~~~~~~~~~~~~~~~~~~~~~~

Other thing.. on the following PAGE what do you have as the DTMF .. which is checked ?

http://www.asteriskguru.com/tutorials/c455ip_image275600.png (http://www.asteriskguru.com/tutorials/c455ip_image275600.png)

If all checked .. uncheck everything except rfc2833 - does that help ?

Kev

[kahalary-harry]

Hi Kev - good news: I got the C455IP working in terms of receiving and making internal and external calls (I suggested to use ATA's, but for a few bucks extra they might as well buy SNOM300's and not have to worry about all that hardware).

There were basically 2 settings messing me around - could be because I'm a newby of course :).

I downloaded a use guide for a newer model from the Siemens website to get a reference of their terminology:

For example in the "Other Provider" configuration, I had the DNS server as their actual DNS server, after reading that use guide I noticed I was supposed to enter the Epygi's IP in there. As soon as I made that change, I was able to dial internal numbers.
The second item was under the "Number Assignment" section - with all the changes, I overlooked that those settings refer to an active "Other Provider" - once I set outgoing calls to use the "Outgoing Provider" that is set up to register to the Epygi, I was able to dial out normally. They have 2 sets registered to the base station, initially I didn't know which set was what. Any way, that was solved now :).

I could not go into detail into the call transfer, all though I did try briefly with RFC 2833. Being a Friday afternoon, the customer wanted to close up shop, but he is over the moon that he can use his C455IP's now for basic functionality.

I am going back on Monday to configure the remaining 5 units, then I will look at the transfer and see if we can get it to work.

Thanks for your input and help, I appreciate it.

Have a good week end

[kahalary-harry]

Ok, this is very interesting:

Two things I figured out today;

1. Call Transfer:
I again looked at the manual of a newer IP gigaset, to try and see how to initiate a call transfer with the "R" key, ie first press "R" then number etc, when I read that one can access the menu function to "conference" parties together. So I got a call going on one of he C455IP's, then accessed the menu. At first I used the "Internal Call", but that failed, then I navigated to "External Call", dialed one of the other extensions on the Epygi, once the other party answered, I disconnected the call on the C455IP by pressing the "red" key, and voila, call was transferred. Then I realized that the Gigaset sees any thing that is NOT on the base station as "external calls", thus allowing the call to go to what ever digits are dialed. How many of the features on the base station influence this, I don't know, but I set it to that RFC2833, enabled the features in the attached screen shot.

(I can't shrink this image to 19.5KB, sorry)

2. Firmware:
Happy that I got the call transfer working, I moved along and started adding the remaining C455IP's to the system, then realized that they did not all have the same level of firmware, about 3 in particular seem to be on a much older version. The problem is that on those, there is no option to set "VoIP" as a default for out going calls. They all register to the Epygi, and can receive calls, but can't dial out because I can not enable that feature. When I clicked on the upgrade firmware button, it told me it's busy with downloading, all though 2 hours later it showed the same message. Will go back tomorrow and see if the unit has a later firmware version :).

But the bottom line is on those old units, use the Gigaset's "External Call" to transfer calls to ohter extensions (via the menu on the phone), and perhaps on any other Gigaset that does not provide proper IP Protocol for the "R" key.

I know, first prize sell supported phones to customer, but hey, some times you just have to do what you have to do, hey?

[kahalary-harry]

Here is a little add on: I found about 3 of those C455IP base stations with older firmware, where you do not have the option to select VoIP for outgoing calls. At first I tried to upgrade the firmware directly from the base station, but it kept telling me it's already in progress - however nothing seemed to happen.
The way to upgrade those base stations is to select the firmware upgrade option from a paired Gigaset. Once it has 02223 (currently the latest), you can proceed to select VoIP in the number assignment and make calls through the Epigy. Before then, the base station registers, and can receive calls only.

Ok, I think that is all the value I can add to this thread :)