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  #1 (permalink)  
Old 09-14-2007, 08:40 AM
Alister
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Default Siemens S450IP call transfer

Hi,

I know a few of you have been waiting for a firmware update for the
Siemens S450IP which
addresses the issue of call transfer using hook-flash. Untill now,
this has been disabled on
these phones for VoIP calls.

Well, today I noticed that the latest firmware has changed the
available settings in the
Telephony / Advanced Settings page to include Hook-Flash call
Transfer. All we have to do
now is figure out how to make it work with Asterisk.

The firmware, downloaded this morning, reports the following version:

Firmware version: 020810000000 / 041.00
EEPROM version: 103

Just thought I'd let you know.


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  #2 (permalink)  
Old 09-14-2007, 09:24 AM
Tim
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Default Re: Siemens S450IP call transfer

Alister wrote:
> Well, today I noticed that the latest firmware has changed the
> available settings in the
> Telephony / Advanced Settings page to include Hook-Flash call
> Transfer. All we have to do
> now is figure out how to make it work with Asterisk.


It does work with Asterisk. We've been using it for the last couple of
months.

The other new feature is that the clock gets set from an NTP server.


Release notes:
http://gigaset.siemens.com/shc/0,193...123868,00.html

Additional user manual for the new features:
http://shc-download.siemens.com/repo...en_GBR_USA.pdf


Tim

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  #3 (permalink)  
Old 09-14-2007, 01:17 PM
Alister
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Default Re: Siemens S450IP call transfer

On Sep 14, 9:24 am, Tim <nutn...@kooky.org> wrote:
> Alister wrote:
>
> > Well, today I noticed that the latest firmware has changed the

>
> > available settings in the
> > Telephony / Advanced Settings page to include Hook-Flash call
> > Transfer. All we have to do
> > now is figure out how to make it work with Asterisk.

>
> It does work with Asterisk. We've been using it for the last couple of
> months.
>
> The other new feature is that the clock gets set from an NTP server.
>
> Release notes:http://gigaset.siemens.com/shc/0,193...NrNrNrN_prodId...
>
> Additional user manual for the new features:http://shc-download.siemens.com/repo...C455IP_S450IP_...
>
> Tim


I would be grateful if you could post details of the asterisk / phone
settings that work.
We currently use features.conf for transfers on these phones, but it
would be great
to get them using hook-flash like all our other desktop phones do.
Had a quick experiment this AM but failed miserably. :-(

Thanks very much

Alister.

btw you second link didn't work for me.


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  #4 (permalink)  
Old 09-14-2007, 04:23 PM
Tim
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Default Re: Siemens S450IP call transfer

Alister wrote:
> I would be grateful if you could post details of the asterisk / phone
> settings that work.


I don't think we had to set anything.

I know you use the menu to do the transfer.

I think, you press ext-call to make another call. Then press menu and
transfer.

> btw you second link didn't work for me.


Works here.

Tim

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  #5 (permalink)  
Old 09-18-2007, 11:23 AM
Paul Hayes
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Default Re: Siemens S450IP call transfer

Alister wrote:

>
> I would be grateful if you could post details of the asterisk / phone
> settings that work.
> We currently use features.conf for transfers on these phones, but it
> would be great
> to get them using hook-flash like all our other desktop phones do.
> Had a quick experiment this AM but failed miserably. :-(
>
> Thanks very much
>
> Alister.
>
> btw you second link didn't work for me.
>


The additional user manual is also here:

http://www.provu.co.uk/pdf/siemens/S...res_sept07.pdf

Your other desk phones don't use hook flash, Asterisk doesn't listen for
hook flash messages on SIP channels, in fact the Siemens phone will send
a hook flash signal to Asterisk if you press the R key but Asterisk
ignores it.

Your desk phones use normal SIP methods to perform transfers (putting a
call on hold with a special Invite message, doing a transfer with a
Refer message...), this functionality is what has been added to the
Siemens phone. Whilst in a call you need to press the "ext call"
button, then you can announce the call to the person on the other end
and select the "transfer" option in the menu on the phone. If you have
the "transfer on hangup" option set to yes then you can just hangup and
the call will transfer, this method can also be used for a blind
transfer, just hangup while the 2nd call is ringing.

As for Asterisk config, if transfers are already working on other SIP
phones you have (using proper SIP transfers, not the ## code) then there
should be no extra config required. That was certainly the case for me
in our office and I've been using this firmware for a couple of months
in BETA before it was released.

cheers,
Paul.

--
Working Email:

paul-at-polog40-dot-co-dot-uk

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  #6 (permalink)  
Old 09-19-2007, 09:43 AM
Paul Hayes
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Posts: n/a
Default Re: Siemens S450IP call transfer

Jono wrote:
> Paul Hayes formulated the question :
>> Alister wrote:
>>
>>>
>>> I would be grateful if you could post details of the asterisk / phone
>>> settings that work.
>>> We currently use features.conf for transfers on these phones, but it
>>> would be great
>>> to get them using hook-flash like all our other desktop phones do.
>>> Had a quick experiment this AM but failed miserably. :-(
>>>
>>> Thanks very much
>>>
>>> Alister.
>>>
>>> btw you second link didn't work for me.
>>>

>>
>> The additional user manual is also here:
>>
>> http://www.provu.co.uk/pdf/siemens/S...res_sept07.pdf
>>

>
> Hi Paul,
>
> Do the "new features" apply to both Siemens models?
>
>


Only the S450IP.

cheers,
Paul.

--
Working Email:

paul-at-polog40-dot-co-dot-uk

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  #7 (permalink)  
Old 11-04-2007, 06:53 AM
Junior Member
 
Join Date: Nov 2007
Posts: 2
Default New bugs introduced?

I have Firmware version: 020810000000 / 041.00 and EEPROM version: 103. Now my phone is almost useless.

Seems like the phone will fail to register after some time, resulting no calls can be put out nor calls can be received. Checking the asterisk logs seems to show that the phone does try to make the call, but when asterisk asks for authentication information, the phone just ignores, causing asterisk to continually retransmit the authentication request.

Secondly, after some time, the phone will refuse any connection requests. It remains visible when pinged, but access to the web configuration page gets a connection refused. The phone will still be able to try to put out calls to asterisk, but the previous mentioned problem would have occurred before then.

The only way to temporarily fix this is to unplug the Ethernet cable and replug it back in and the problem would be back after a few minutes.

Finally, when the phone does work, audio is choppy. I can't determine the cause of this problem other than that different codecs were used to bridge the phone and the incoming call even though both of them have the same set of codecs. I tried using a softphone in place of the S450 and the audio has no problems.

Anyone knows how I can fix this? Simens is not responding to any support emails and there is no way to downgrade the phone nor reflash the current version (in case its just a bad flash). I paid big bucks for this phone and now its nothing but paperweight.
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  #8 (permalink)  
Old 11-04-2007, 03:46 PM
Tim
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Posts: n/a
Default Re: Siemens S450IP call transfer

booya wrote:
> Anyone knows how I can fix this? Simens is not responding to any
> support emails and there is no way to downgrade the phone nor reflash
> the current version (in case its just a bad flash). I paid big bucks for
> this phone and now its nothing but paperweight.
>


Have you contacted the place you bought it from to see if they can help?

Tim

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  #9 (permalink)  
Old 11-04-2007, 06:12 PM
Cjay
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Posts: n/a
Default Re: Siemens S450IP call transfer

booya wrote:
> I have Firmware version: 020810000000 / 041.00 and EEPROM version: 103.
> Now my phone is almost useless.
>
> Seems like the phone will fail to register after some time, resulting
> no calls can be put out nor calls can be received. Checking the asterisk
> logs seems to show that the phone does try to make the call, but when
> asterisk asks for authentication information, the phone just ignores,
> causing asterisk to continually retransmit the authentication request.
>
> Secondly, after some time, the phone will refuse any connection
> requests. It remains visible when pinged, but access to the web
> configuration page gets a connection refused. The phone will still be
> able to try to put out calls to asterisk, but the previous mentioned
> problem would have occurred before then.
>
> The only way to temporarily fix this is to unplug the Ethernet cable
> and replug it back in and the problem would be back after a few
> minutes.
>
> Finally, when the phone does work, audio is choppy. I can't determine
> the cause of this problem other than that different codecs were used to
> bridge the phone and the incoming call even though both of them have the
> same set of codecs. I tried using a softphone in place of the S450 and
> the audio has no problems.
>
> Anyone knows how I can fix this? Simens is not responding to any
> support emails and there is no way to downgrade the phone nor reflash
> the current version (in case its just a bad flash). I paid big bucks for
> this phone and now its nothing but paperweight.
>
>
> ------------------------------------------------------------------------
> View this thread: http://www.wirelessforums.org/uk-telecom-voip/siemens-s450ip-call-transfer-28930.html
> http://www.wirelessforums.org
>


My S450IP is running exactly the same firmware/EPROM versions with no
apparent problem so problem may not be due the phone? The only
difference perhaps is that I run mine under 3cx (Windows PBX) rather
than asterisk?

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  #10 (permalink)  
Old 11-04-2007, 07:52 PM
TheFug
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Posts: n/a
Default Re: Siemens S450IP call transfer

booya schreef:
> I have Firmware version: 020810000000 / 041.00 and EEPROM version: 103.
> Now my phone is almost useless.
>
> Seems like the phone will fail to register after some time, resulting
> no calls can be put out nor calls can be received. Checking the asterisk
> logs seems to show that the phone does try to make the call, but when
> asterisk asks for authentication information, the phone just ignores,
> causing asterisk to continually retransmit the authentication request.
>
> Secondly, after some time, the phone will refuse any connection
> requests. It remains visible when pinged, but access to the web
> configuration page gets a connection refused. The phone will still be
> able to try to put out calls to asterisk, but the previous mentioned
> problem would have occurred before then.
>
> The only way to temporarily fix this is to unplug the Ethernet cable
> and replug it back in and the problem would be back after a few
> minutes.
>
> Finally, when the phone does work, audio is choppy. I can't determine
> the cause of this problem other than that different codecs were used to
> bridge the phone and the incoming call even though both of them have the
> same set of codecs. I tried using a softphone in place of the S450 and
> the audio has no problems.
>
> Anyone knows how I can fix this? Simens is not responding to any
> support emails and there is no way to downgrade the phone nor reflash
> the current version (in case its just a bad flash). I paid big bucks for
> this phone and now its nothing but paperweight.
>
>
> ------------------------------------------------------------------------
> View this thread: http://www.wirelessforums.org/showthread.php?t=28930
> http://www.wirelessforums.org
>


you haven't changed anything in your router settings since ?

--
The Fug.

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  #11 (permalink)  
Old 11-04-2007, 07:54 PM
TheFug
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Posts: n/a
Default Re: Siemens S450IP call transfer

booya schreef:
> I have Firmware version: 020810000000 / 041.00 and EEPROM version: 103.
> Now my phone is almost useless.
>
> Seems like the phone will fail to register after some time, resulting
> no calls can be put out nor calls can be received. Checking the asterisk
> logs seems to show that the phone does try to make the call, but when
> asterisk asks for authentication information, the phone just ignores,
> causing asterisk to continually retransmit the authentication request.
>
> Secondly, after some time, the phone will refuse any connection
> requests. It remains visible when pinged, but access to the web
> configuration page gets a connection refused. The phone will still be
> able to try to put out calls to asterisk, but the previous mentioned
> problem would have occurred before then.
>
> The only way to temporarily fix this is to unplug the Ethernet cable
> and replug it back in and the problem would be back after a few
> minutes.
>
> Finally, when the phone does work, audio is choppy. I can't determine
> the cause of this problem other than that different codecs were used to
> bridge the phone and the incoming call even though both of them have the
> same set of codecs. I tried using a softphone in place of the S450 and
> the audio has no problems.
>
> Anyone knows how I can fix this? Simens is not responding to any
> support emails and there is no way to downgrade the phone nor reflash
> the current version (in case its just a bad flash). I paid big bucks for
> this phone and now its nothing but paperweight.
>
>
> ------------------------------------------------------------------------
> View this thread: http://www.wirelessforums.org/showthread.php?t=28930
> http://www.wirelessforums.org
>

Did you change the id name of your phone, or has it been changed after
the fw upgrade ?

--
The Fug.

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  #12 (permalink)  
Old 11-04-2007, 10:59 PM
Christian
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Posts: n/a
Default Re: Siemens S450IP call transfer

> I have Firmware version: 020810000000 / 041.00 and EEPROM version:
> 103. Now my phone is almost useless.
> Seems like the phone will fail to register after some time, resulting
> no calls can be put out nor calls can be received.


So this happened after a firmware-upgrade? I have read about incidents with
S450IP-firmware-upgrades on a German forum, although their problems don't
seem entirely the same...

If you haven't tried it yet: a cold start seemed to help in the German
cases: unplug the power supply & plug it back in while pushing & holding the
registration-button on the base station for 5 seconds or so... You'll lose
the settings of the phone of course, but hey... if it helps...

Christian



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  #13 (permalink)  
Old 11-06-2007, 12:03 PM
Junior Member
 
Join Date: Nov 2007
Posts: 2
Default

Thank you all for your replies.

I have solved the problem, although I do not know which is the real solving step.

In hopes that it will help someone, here is what I did:

I did the cold reset once before, but I remember that some devices do not reset well with just a power cycle and a reset button press, and memory corruption would occur. Thus I:
1) Reset the base settings via the phone menu
2) Do a cold reset
3) Do another cold reset as soon as the phone boots up
4) Reset the base settings via the phone menu
5) Do another cold reset
6) Do another cold reset as soon as the phone boots up
7) Reset the base settings via the phone menu
Hoping the repeated resets would fully reset the phone.

Next, I thought since the problem is with authentication problems, it could be my ultra-long extension password (20 characters long). Thus I shortened it to 8 characters.

And since my asterisk logs seem to tell that the phone is not responding to authentication requests it sends, I thought maybe its a NAT issue. My asterisk server is set to only respond to requests from my network domain (phone.***.com). However, since the phone is on the same network as the server, I decided to just enter the IP address into all the fields. I thought maybe that's the cause of the authentication issues. Thus I changed the registry address to my domain name.

That solved the authentication problem (I hope, its been almost two days) but audio does not work. Changing all the other IP addresses into my domain name does not help. Finally, I decided to use a public STUN server, and things worked.

The funny thing though is that now I can use either my domain name or IP address and it'll still work. I've actually tried the above changes (not together, separately) before but it didn't work, funny that it worked when I did all of them together.

Thus, I guess the fixing steps could either be:
1) The cold reset process
2) Changing the password length
3) Using a STUN server

The stuttering problem is also solved.

I know this is confusing, but I hope it helps someone.
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